From bd8a71a7b0f50da9350d202d325c3926ffd6b189 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 3 Jan 2009 16:56:56 +0000 Subject: ALSA: Reduce boilerplate for new jack types Use a lookup table rather than explicit code to map input subsystem jack types into ASoC ones, implemented as suggested by Takashi Iwai. Signed-off-by: Mark Brown --- include/sound/jack.h | 3 +++ sound/core/jack.c | 44 ++++++++++++++++++++------------------------ 2 files changed, 23 insertions(+), 24 deletions(-) diff --git a/include/sound/jack.h b/include/sound/jack.h index 2e0315cdd0d..85266a2f5c6 100644 --- a/include/sound/jack.h +++ b/include/sound/jack.h @@ -30,6 +30,9 @@ struct input_dev; /** * Jack types which can be reported. These values are used as a * bitmask. + * + * Note that this must be kept in sync with the lookup table in + * sound/core/jack.c. */ enum snd_jack_types { SND_JACK_HEADPHONE = 0x0001, diff --git a/sound/core/jack.c b/sound/core/jack.c index dd4a12dc09a..b2da10c9916 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -23,6 +23,13 @@ #include #include +static int jack_types[] = { + SW_HEADPHONE_INSERT, + SW_MICROPHONE_INSERT, + SW_LINEOUT_INSERT, + SW_JACK_PHYSICAL_INSERT, +}; + static int snd_jack_dev_free(struct snd_device *device) { struct snd_jack *jack = device->device_data; @@ -79,6 +86,7 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, { struct snd_jack *jack; int err; + int i; static struct snd_device_ops ops = { .dev_free = snd_jack_dev_free, .dev_register = snd_jack_dev_register, @@ -100,18 +108,10 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, jack->type = type; - if (type & SND_JACK_HEADPHONE) - input_set_capability(jack->input_dev, EV_SW, - SW_HEADPHONE_INSERT); - if (type & SND_JACK_LINEOUT) - input_set_capability(jack->input_dev, EV_SW, - SW_LINEOUT_INSERT); - if (type & SND_JACK_MICROPHONE) - input_set_capability(jack->input_dev, EV_SW, - SW_MICROPHONE_INSERT); - if (type & SND_JACK_MECHANICAL) - input_set_capability(jack->input_dev, EV_SW, - SW_JACK_PHYSICAL_INSERT); + for (i = 0; i < ARRAY_SIZE(jack_types); i++) + if (type & (1 << i)) + input_set_capability(jack->input_dev, EV_SW, + jack_types[i]); err = snd_device_new(card, SNDRV_DEV_JACK, jack, &ops); if (err < 0) @@ -154,21 +154,17 @@ EXPORT_SYMBOL(snd_jack_set_parent); */ void snd_jack_report(struct snd_jack *jack, int status) { + int i; + if (!jack) return; - if (jack->type & SND_JACK_HEADPHONE) - input_report_switch(jack->input_dev, SW_HEADPHONE_INSERT, - status & SND_JACK_HEADPHONE); - if (jack->type & SND_JACK_LINEOUT) - input_report_switch(jack->input_dev, SW_LINEOUT_INSERT, - status & SND_JACK_LINEOUT); - if (jack->type & SND_JACK_MICROPHONE) - input_report_switch(jack->input_dev, SW_MICROPHONE_INSERT, - status & SND_JACK_MICROPHONE); - if (jack->type & SND_JACK_MECHANICAL) - input_report_switch(jack->input_dev, SW_JACK_PHYSICAL_INSERT, - status & SND_JACK_MECHANICAL); + for (i = 0; i < ARRAY_SIZE(jack_types); i++) { + int testbit = 1 << i; + if (jack->type & testbit) + input_report_switch(jack->input_dev, jack_types[i], + status & testbit); + } input_sync(jack->input_dev); } -- cgit v1.2.3-70-g09d2 From d506fc322ec2af04fc47be83d796a1c9e1a16022 Mon Sep 17 00:00:00 2001 From: Jani Nikula Date: Wed, 7 Jan 2009 11:54:25 +0200 Subject: ALSA: Add support for video out to the jack reporting API Add support for reporting new jack types SND_JACK_VIDEOOUT and SND_JACK_AVOUT (a combination of LINEOUT and VIDEOOUT) to the jack reporting API. Also add the corresponding SW_VIDEOOUT_INSERT switch to the input system header. Signed-off-by: Jani Nikula Signed-off-by: Mark Brown --- include/linux/input.h | 1 + include/sound/jack.h | 2 ++ sound/core/jack.c | 1 + 3 files changed, 4 insertions(+) diff --git a/include/linux/input.h b/include/linux/input.h index 9a6355f74db..adc13322d1d 100644 --- a/include/linux/input.h +++ b/include/linux/input.h @@ -661,6 +661,7 @@ struct input_absinfo { #define SW_DOCK 0x05 /* set = plugged into dock */ #define SW_LINEOUT_INSERT 0x06 /* set = inserted */ #define SW_JACK_PHYSICAL_INSERT 0x07 /* set = mechanical switch set */ +#define SW_VIDEOOUT_INSERT 0x08 /* set = inserted */ #define SW_MAX 0x0f #define SW_CNT (SW_MAX+1) diff --git a/include/sound/jack.h b/include/sound/jack.h index 85266a2f5c6..6b013c6f6a0 100644 --- a/include/sound/jack.h +++ b/include/sound/jack.h @@ -40,6 +40,8 @@ enum snd_jack_types { SND_JACK_HEADSET = SND_JACK_HEADPHONE | SND_JACK_MICROPHONE, SND_JACK_LINEOUT = 0x0004, SND_JACK_MECHANICAL = 0x0008, /* If detected separately */ + SND_JACK_VIDEOOUT = 0x0010, + SND_JACK_AVOUT = SND_JACK_LINEOUT | SND_JACK_VIDEOOUT, }; struct snd_jack { diff --git a/sound/core/jack.c b/sound/core/jack.c index b2da10c9916..43b10d6e522 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -28,6 +28,7 @@ static int jack_types[] = { SW_MICROPHONE_INSERT, SW_LINEOUT_INSERT, SW_JACK_PHYSICAL_INSERT, + SW_VIDEOOUT_INSERT, }; static int snd_jack_dev_free(struct snd_device *device) -- cgit v1.2.3-70-g09d2 From ca9c1aaec4187fc9922cfb6b283fffef89286943 Mon Sep 17 00:00:00 2001 From: Ian Molton Date: Tue, 6 Jan 2009 20:11:51 +0000 Subject: ASoC: dapm: Allow explictly named mixer controls This patch allows you to define the mixer paths as having the same name as the paths they represent. This is required to support codecs such as the wm9705 neatly without extra controls in the alsa mixer. Signed-off-by: Ian Molton --- Documentation/sound/alsa/soc/dapm.txt | 3 +++ include/sound/soc-dapm.h | 11 ++++++++ sound/soc/soc-dapm.c | 47 +++++++++++++++++++++++++++-------- 3 files changed, 50 insertions(+), 11 deletions(-) diff --git a/Documentation/sound/alsa/soc/dapm.txt b/Documentation/sound/alsa/soc/dapm.txt index 46f9684d0b2..9e6763264a2 100644 --- a/Documentation/sound/alsa/soc/dapm.txt +++ b/Documentation/sound/alsa/soc/dapm.txt @@ -116,6 +116,9 @@ SOC_DAPM_SINGLE("HiFi Playback Switch", WM8731_APANA, 4, 1, 0), SND_SOC_DAPM_MIXER("Output Mixer", WM8731_PWR, 4, 1, wm8731_output_mixer_controls, ARRAY_SIZE(wm8731_output_mixer_controls)), +If you dont want the mixer elements prefixed with the name of the mixer widget, +you can use SND_SOC_DAPM_MIXER_NAMED_CTL instead. the parameters are the same +as for SND_SOC_DAPM_MIXER. 2.3 Platform/Machine domain Widgets ----------------------------------- diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 4af1083e328..cc99dd40493 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -76,6 +76,11 @@ wcontrols, wncontrols)\ { .id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \ .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = wncontrols} +#define SND_SOC_DAPM_MIXER_NAMED_CTL(wname, wreg, wshift, winvert, \ + wcontrols, wncontrols)\ +{ .id = snd_soc_dapm_mixer_named_ctl, .name = wname, .reg = wreg, \ + .shift = wshift, .invert = winvert, .kcontrols = wcontrols, \ + .num_kcontrols = wncontrols} #define SND_SOC_DAPM_MICBIAS(wname, wreg, wshift, winvert) \ { .id = snd_soc_dapm_micbias, .name = wname, .reg = wreg, .shift = wshift, \ .invert = winvert, .kcontrols = NULL, .num_kcontrols = 0} @@ -101,6 +106,11 @@ { .id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \ .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = wncontrols, \ .event = wevent, .event_flags = wflags} +#define SND_SOC_DAPM_MIXER_NAMED_CTL_E(wname, wreg, wshift, winvert, \ + wcontrols, wncontrols, wevent, wflags) \ +{ .id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \ + .invert = winvert, .kcontrols = wcontrols, \ + .num_kcontrols = wncontrols, .event = wevent, .event_flags = wflags} #define SND_SOC_DAPM_MICBIAS_E(wname, wreg, wshift, winvert, wevent, wflags) \ { .id = snd_soc_dapm_micbias, .name = wname, .reg = wreg, .shift = wshift, \ .invert = winvert, .kcontrols = NULL, .num_kcontrols = 0, \ @@ -263,6 +273,7 @@ enum snd_soc_dapm_type { snd_soc_dapm_mux, /* selects 1 analog signal from many inputs */ snd_soc_dapm_value_mux, /* selects 1 analog signal from many inputs */ snd_soc_dapm_mixer, /* mixes several analog signals together */ + snd_soc_dapm_mixer_named_ctl, /* mixer with named controls */ snd_soc_dapm_pga, /* programmable gain/attenuation (volume) */ snd_soc_dapm_adc, /* analog to digital converter */ snd_soc_dapm_dac, /* digital to analog converter */ diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index ad0d801677c..6362c7641ce 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -54,14 +54,15 @@ static int dapm_up_seq[] = { snd_soc_dapm_pre, snd_soc_dapm_micbias, snd_soc_dapm_mic, snd_soc_dapm_mux, snd_soc_dapm_value_mux, snd_soc_dapm_dac, - snd_soc_dapm_mixer, snd_soc_dapm_pga, snd_soc_dapm_adc, snd_soc_dapm_hp, - snd_soc_dapm_spk, snd_soc_dapm_post + snd_soc_dapm_mixer, snd_soc_dapm_mixer_named_ctl, snd_soc_dapm_pga, + snd_soc_dapm_adc, snd_soc_dapm_hp, snd_soc_dapm_spk, snd_soc_dapm_post }; + static int dapm_down_seq[] = { snd_soc_dapm_pre, snd_soc_dapm_adc, snd_soc_dapm_hp, snd_soc_dapm_spk, - snd_soc_dapm_pga, snd_soc_dapm_mixer, snd_soc_dapm_dac, snd_soc_dapm_mic, - snd_soc_dapm_micbias, snd_soc_dapm_mux, snd_soc_dapm_value_mux, - snd_soc_dapm_post + snd_soc_dapm_pga, snd_soc_dapm_mixer_named_ctl, snd_soc_dapm_mixer, + snd_soc_dapm_dac, snd_soc_dapm_mic, snd_soc_dapm_micbias, + snd_soc_dapm_mux, snd_soc_dapm_value_mux, snd_soc_dapm_post }; static int dapm_status = 1; @@ -101,7 +102,8 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, { switch (w->id) { case snd_soc_dapm_switch: - case snd_soc_dapm_mixer: { + case snd_soc_dapm_mixer: + case snd_soc_dapm_mixer_named_ctl: { int val; struct soc_mixer_control *mc = (struct soc_mixer_control *) w->kcontrols[i].private_value; @@ -347,15 +349,33 @@ static int dapm_new_mixer(struct snd_soc_codec *codec, if (path->name != (char*)w->kcontrols[i].name) continue; - /* add dapm control with long name */ - name_len = 2 + strlen(w->name) - + strlen(w->kcontrols[i].name); + /* add dapm control with long name. + * for dapm_mixer this is the concatenation of the + * mixer and kcontrol name. + * for dapm_mixer_named_ctl this is simply the + * kcontrol name. + */ + name_len = strlen(w->kcontrols[i].name) + 1; + if (w->id == snd_soc_dapm_mixer) + name_len += 1 + strlen(w->name); + path->long_name = kmalloc(name_len, GFP_KERNEL); + if (path->long_name == NULL) return -ENOMEM; - snprintf(path->long_name, name_len, "%s %s", - w->name, w->kcontrols[i].name); + switch (w->id) { + case snd_soc_dapm_mixer: + default: + snprintf(path->long_name, name_len, "%s %s", + w->name, w->kcontrols[i].name); + break; + case snd_soc_dapm_mixer_named_ctl: + snprintf(path->long_name, name_len, "%s", + w->kcontrols[i].name); + break; + } + path->long_name[name_len - 1] = '\0'; path->kcontrol = snd_soc_cnew(&w->kcontrols[i], w, @@ -711,6 +731,7 @@ static void dbg_dump_dapm(struct snd_soc_codec* codec, const char *action) case snd_soc_dapm_adc: case snd_soc_dapm_pga: case snd_soc_dapm_mixer: + case snd_soc_dapm_mixer_named_ctl: if (w->name) { in = is_connected_input_ep(w); dapm_clear_walk(w->codec); @@ -822,6 +843,7 @@ static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, int found = 0; if (widget->id != snd_soc_dapm_mixer && + widget->id != snd_soc_dapm_mixer_named_ctl && widget->id != snd_soc_dapm_switch) return -ENODEV; @@ -875,6 +897,7 @@ static ssize_t dapm_widget_show(struct device *dev, case snd_soc_dapm_adc: case snd_soc_dapm_pga: case snd_soc_dapm_mixer: + case snd_soc_dapm_mixer_named_ctl: if (w->name) count += sprintf(buf + count, "%s: %s\n", w->name, w->power ? "On":"Off"); @@ -1058,6 +1081,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_codec *codec, break; case snd_soc_dapm_switch: case snd_soc_dapm_mixer: + case snd_soc_dapm_mixer_named_ctl: ret = dapm_connect_mixer(codec, wsource, wsink, path, control); if (ret != 0) goto err; @@ -1135,6 +1159,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec) switch(w->id) { case snd_soc_dapm_switch: case snd_soc_dapm_mixer: + case snd_soc_dapm_mixer_named_ctl: dapm_new_mixer(codec, w); break; case snd_soc_dapm_mux: -- cgit v1.2.3-70-g09d2 From 3195954da9cdb1cadb2059921c62e69d376c624f Mon Sep 17 00:00:00 2001 From: Andrea Borgia Date: Wed, 7 Jan 2009 22:58:50 +0100 Subject: ALSA: preliminary support for Toshiba SB-0500 The Toshiba Multimedia Center SB-0500 is a rebranded version of the Creative Technology SB Live! 24-bit External: it shares the same chipset and only has minor cosmetic differences. Remote controller works with alsa_usb module, basic audio is there and mixer controls are mostly untested. Signed-off-by: Andrea Borgia Signed-off-by: Takashi Iwai --- sound/usb/usbmixer.c | 15 ++++++++++----- sound/usb/usbmixer_maps.c | 5 +++++ 2 files changed, 15 insertions(+), 5 deletions(-) diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index 00397c8a765..bc8bd00047a 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -66,6 +66,7 @@ static const struct rc_config { { USB_ID(0x041e, 0x3000), 0, 1, 2, 1, 18, 0x0013 }, /* Extigy */ { USB_ID(0x041e, 0x3020), 2, 1, 6, 6, 18, 0x0013 }, /* Audigy 2 NX */ { USB_ID(0x041e, 0x3040), 2, 2, 6, 6, 2, 0x6e91 }, /* Live! 24-bit */ + { USB_ID(0x041e, 0x3048), 2, 2, 6, 6, 2, 0x6e91 }, /* Toshiba SB0500 */ }; struct usb_mixer_interface { @@ -1706,7 +1707,8 @@ static void snd_usb_mixer_memory_change(struct usb_mixer_interface *mixer, break; /* live24ext: 4 = line-in jack */ case 3: /* hp-out jack (may actuate Mute) */ - if (mixer->chip->usb_id == USB_ID(0x041e, 0x3040)) + if (mixer->chip->usb_id == USB_ID(0x041e, 0x3040) || + mixer->chip->usb_id == USB_ID(0x041e, 0x3048)) snd_usb_mixer_notify_id(mixer, mixer->rc_cfg->mute_mixer_id); break; default: @@ -1956,8 +1958,9 @@ static int snd_audigy2nx_controls_create(struct usb_mixer_interface *mixer) int i, err; for (i = 0; i < ARRAY_SIZE(snd_audigy2nx_controls); ++i) { - if (i > 1 && /* Live24ext has 2 LEDs only */ - mixer->chip->usb_id == USB_ID(0x041e, 0x3040)) + if (i > 1 && /* Live24ext has 2 LEDs only */ + (mixer->chip->usb_id == USB_ID(0x041e, 0x3040) || + mixer->chip->usb_id == USB_ID(0x041e, 0x3048))) break; err = snd_ctl_add(mixer->chip->card, snd_ctl_new1(&snd_audigy2nx_controls[i], mixer)); @@ -1994,7 +1997,8 @@ static void snd_audigy2nx_proc_read(struct snd_info_entry *entry, snd_iprintf(buffer, "%s jacks\n\n", mixer->chip->card->shortname); if (mixer->chip->usb_id == USB_ID(0x041e, 0x3020)) jacks = jacks_audigy2nx; - else if (mixer->chip->usb_id == USB_ID(0x041e, 0x3040)) + else if (mixer->chip->usb_id == USB_ID(0x041e, 0x3040) || + mixer->chip->usb_id == USB_ID(0x041e, 0x3048)) jacks = jacks_live24ext; else return; @@ -2044,7 +2048,8 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, goto _error; if (mixer->chip->usb_id == USB_ID(0x041e, 0x3020) || - mixer->chip->usb_id == USB_ID(0x041e, 0x3040)) { + mixer->chip->usb_id == USB_ID(0x041e, 0x3040) || + mixer->chip->usb_id == USB_ID(0x041e, 0x3048)) { struct snd_info_entry *entry; if ((err = snd_audigy2nx_controls_create(mixer)) < 0) diff --git a/sound/usb/usbmixer_maps.c b/sound/usb/usbmixer_maps.c index d755be0ad81..f41214f3ad6 100644 --- a/sound/usb/usbmixer_maps.c +++ b/sound/usb/usbmixer_maps.c @@ -284,6 +284,11 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = { .id = USB_ID(0x041e, 0x3040), .map = live24ext_map, }, + { + .id = USB_ID(0x041e, 0x3048), + .map = audigy2nx_map, + .selector_map = audigy2nx_selectors, + }, { /* Hercules DJ Console (Windows Edition) */ .id = USB_ID(0x06f8, 0xb000), -- cgit v1.2.3-70-g09d2 From 1649923dd52ce914be98bff0ae352344ef04f305 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 7 Jan 2009 18:25:13 +0000 Subject: ASoC: Constify pin names for DAPM pin status APIs Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 8 ++++---- sound/soc/soc-dapm.c | 10 +++++----- 2 files changed, 9 insertions(+), 9 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index cc99dd40493..075244ef41e 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -260,10 +260,10 @@ int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev, int snd_soc_dapm_sys_add(struct device *dev); /* dapm audio pin control and status */ -int snd_soc_dapm_enable_pin(struct snd_soc_codec *codec, char *pin); -int snd_soc_dapm_disable_pin(struct snd_soc_codec *codec, char *pin); -int snd_soc_dapm_nc_pin(struct snd_soc_codec *codec, char *pin); -int snd_soc_dapm_get_pin_status(struct snd_soc_codec *codec, char *pin); +int snd_soc_dapm_enable_pin(struct snd_soc_codec *codec, const char *pin); +int snd_soc_dapm_disable_pin(struct snd_soc_codec *codec, const char *pin); +int snd_soc_dapm_nc_pin(struct snd_soc_codec *codec, const char *pin); +int snd_soc_dapm_get_pin_status(struct snd_soc_codec *codec, const char *pin); int snd_soc_dapm_sync(struct snd_soc_codec *codec); /* dapm widget types */ diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 6362c7641ce..a35ce69d9d7 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -961,7 +961,7 @@ static void dapm_free_widgets(struct snd_soc_codec *codec) } static int snd_soc_dapm_set_pin(struct snd_soc_codec *codec, - char *pin, int status) + const char *pin, int status) { struct snd_soc_dapm_widget *w; @@ -1643,7 +1643,7 @@ int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev, * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to * do any widget power switching. */ -int snd_soc_dapm_enable_pin(struct snd_soc_codec *codec, char *pin) +int snd_soc_dapm_enable_pin(struct snd_soc_codec *codec, const char *pin) { return snd_soc_dapm_set_pin(codec, pin, 1); } @@ -1658,7 +1658,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_enable_pin); * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to * do any widget power switching. */ -int snd_soc_dapm_disable_pin(struct snd_soc_codec *codec, char *pin) +int snd_soc_dapm_disable_pin(struct snd_soc_codec *codec, const char *pin) { return snd_soc_dapm_set_pin(codec, pin, 0); } @@ -1678,7 +1678,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_disable_pin); * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to * do any widget power switching. */ -int snd_soc_dapm_nc_pin(struct snd_soc_codec *codec, char *pin) +int snd_soc_dapm_nc_pin(struct snd_soc_codec *codec, const char *pin) { return snd_soc_dapm_set_pin(codec, pin, 0); } @@ -1693,7 +1693,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_nc_pin); * * Returns 1 for connected otherwise 0. */ -int snd_soc_dapm_get_pin_status(struct snd_soc_codec *codec, char *pin) +int snd_soc_dapm_get_pin_status(struct snd_soc_codec *codec, const char *pin) { struct snd_soc_dapm_widget *w; -- cgit v1.2.3-70-g09d2 From 8a2cd6180f8fa00111843c2f4a4f4361995358e0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 7 Jan 2009 17:31:10 +0000 Subject: ASoC: Add jack reporting interface This patch adds a jack reporting interface to ASoC. This wraps the ALSA core jack detection functionality and provides integration with DAPM to automatically update the power state of pins based on the jack state. Since embedded platforms can have multiple detecton methods used for a single jack (eg, separate microphone and headphone detection) the report function allows specification of which bits are being updated on a given report. The expected usage is that machine drivers will create jack objects and then configure jack detection methods to update that jack. Signed-off-by: Mark Brown --- include/sound/soc.h | 32 ++++++++++++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 2 +- sound/soc/soc-jack.c | 138 +++++++++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 172 insertions(+), 1 deletion(-) create mode 100644 sound/soc/soc-jack.c diff --git a/include/sound/soc.h b/include/sound/soc.h index 9b930d34211..9c3ef6a3e9f 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -154,6 +154,8 @@ enum snd_soc_bias_level { SND_SOC_BIAS_OFF, }; +struct snd_jack; +struct snd_soc_card; struct snd_soc_device; struct snd_soc_pcm_stream; struct snd_soc_ops; @@ -164,6 +166,8 @@ struct snd_soc_platform; struct snd_soc_codec; struct soc_enum; struct snd_soc_ac97_ops; +struct snd_soc_jack; +struct snd_soc_jack_pin; typedef int (*hw_write_t)(void *,const char* ,int); typedef int (*hw_read_t)(void *,char* ,int); @@ -184,6 +188,13 @@ int snd_soc_init_card(struct snd_soc_device *socdev); int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream, const struct snd_pcm_hardware *hw); +/* Jack reporting */ +int snd_soc_jack_new(struct snd_soc_card *card, const char *id, int type, + struct snd_soc_jack *jack); +void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask); +int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count, + struct snd_soc_jack_pin *pins); + /* codec IO */ #define snd_soc_read(codec, reg) codec->read(codec, reg) #define snd_soc_write(codec, reg, value) codec->write(codec, reg, value) @@ -239,6 +250,27 @@ int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol, int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +/** + * struct snd_soc_jack_pin - Describes a pin to update based on jack detection + * + * @pin: name of the pin to update + * @mask: bits to check for in reported jack status + * @invert: if non-zero then pin is enabled when status is not reported + */ +struct snd_soc_jack_pin { + struct list_head list; + const char *pin; + int mask; + bool invert; +}; + +struct snd_soc_jack { + struct snd_jack *jack; + struct snd_soc_card *card; + struct list_head pins; + int status; +}; + /* SoC PCM stream information */ struct snd_soc_pcm_stream { char *stream_name; diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index ef025c66cc6..3d2bb6fc6dc 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -6,6 +6,7 @@ menuconfig SND_SOC tristate "ALSA for SoC audio support" select SND_PCM select AC97_BUS if SND_SOC_AC97_BUS + select SND_JACK if INPUT=y || INPUT=SND ---help--- If you want ASoC support, you should say Y here and also to the diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 86a9b1f5b0f..0237879fd41 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,4 +1,4 @@ -snd-soc-core-objs := soc-core.o soc-dapm.o +snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o obj-$(CONFIG_SND_SOC) += codecs/ diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c new file mode 100644 index 00000000000..8cc00c3cdf3 --- /dev/null +++ b/sound/soc/soc-jack.c @@ -0,0 +1,138 @@ +/* + * soc-jack.c -- ALSA SoC jack handling + * + * Copyright 2008 Wolfson Microelectronics PLC. + * + * Author: Mark Brown + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include +#include +#include + +/** + * snd_soc_jack_new - Create a new jack + * @card: ASoC card + * @id: an identifying string for this jack + * @type: a bitmask of enum snd_jack_type values that can be detected by + * this jack + * @jack: structure to use for the jack + * + * Creates a new jack object. + * + * Returns zero if successful, or a negative error code on failure. + * On success jack will be initialised. + */ +int snd_soc_jack_new(struct snd_soc_card *card, const char *id, int type, + struct snd_soc_jack *jack) +{ + jack->card = card; + INIT_LIST_HEAD(&jack->pins); + + return snd_jack_new(card->socdev->codec->card, id, type, &jack->jack); +} +EXPORT_SYMBOL_GPL(snd_soc_jack_new); + +/** + * snd_soc_jack_report - Report the current status for a jack + * + * @jack: the jack + * @status: a bitmask of enum snd_jack_type values that are currently detected. + * @mask: a bitmask of enum snd_jack_type values that being reported. + * + * If configured using snd_soc_jack_add_pins() then the associated + * DAPM pins will be enabled or disabled as appropriate and DAPM + * synchronised. + * + * Note: This function uses mutexes and should be called from a + * context which can sleep (such as a workqueue). + */ +void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) +{ + struct snd_soc_codec *codec = jack->card->socdev->codec; + struct snd_soc_jack_pin *pin; + int enable; + int oldstatus; + + if (!jack) { + WARN_ON_ONCE(!jack); + return; + } + + mutex_lock(&codec->mutex); + + oldstatus = jack->status; + + jack->status &= ~mask; + jack->status |= status; + + /* The DAPM sync is expensive enough to be worth skipping */ + if (jack->status == oldstatus) + goto out; + + list_for_each_entry(pin, &jack->pins, list) { + enable = pin->mask & status; + + if (pin->invert) + enable = !enable; + + if (enable) + snd_soc_dapm_enable_pin(codec, pin->pin); + else + snd_soc_dapm_disable_pin(codec, pin->pin); + } + + snd_soc_dapm_sync(codec); + + snd_jack_report(jack->jack, status); + +out: + mutex_unlock(&codec->mutex); +} +EXPORT_SYMBOL_GPL(snd_soc_jack_report); + +/** + * snd_soc_jack_add_pins - Associate DAPM pins with an ASoC jack + * + * @jack: ASoC jack + * @count: Number of pins + * @pins: Array of pins + * + * After this function has been called the DAPM pins specified in the + * pins array will have their status updated to reflect the current + * state of the jack whenever the jack status is updated. + */ +int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count, + struct snd_soc_jack_pin *pins) +{ + int i; + + for (i = 0; i < count; i++) { + if (!pins[i].pin) { + printk(KERN_ERR "No name for pin %d\n", i); + return -EINVAL; + } + if (!pins[i].mask) { + printk(KERN_ERR "No mask for pin %d (%s)\n", i, + pins[i].pin); + return -EINVAL; + } + + INIT_LIST_HEAD(&pins[i].list); + list_add(&(pins[i].list), &jack->pins); + } + + /* Update to reflect the last reported status; canned jack + * implementations are likely to set their state before the + * card has an opportunity to associate pins. + */ + snd_soc_jack_report(jack, 0, 0); + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_jack_add_pins); -- cgit v1.2.3-70-g09d2 From a6ba2b2dabb583e7820e567fb309d771b50cb9ff Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 8 Jan 2009 15:16:16 +0000 Subject: ASoC: Implement WM8350 headphone jack detection Signed-off-by: Mark Brown --- include/linux/mfd/wm8350/audio.h | 1 + sound/soc/codecs/wm8350.c | 116 +++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8350.h | 8 +++ 3 files changed, 125 insertions(+) diff --git a/include/linux/mfd/wm8350/audio.h b/include/linux/mfd/wm8350/audio.h index af95a1d2f3a..d899dc0223b 100644 --- a/include/linux/mfd/wm8350/audio.h +++ b/include/linux/mfd/wm8350/audio.h @@ -490,6 +490,7 @@ /* * R231 (0xE7) - Jack Status */ +#define WM8350_JACK_L_LVL 0x0800 #define WM8350_JACK_R_LVL 0x0400 /* diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index e3989d406f5..47a9dabb523 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -51,10 +51,17 @@ struct wm8350_output { u16 mute; }; +struct wm8350_jack_data { + struct snd_soc_jack *jack; + int report; +}; + struct wm8350_data { struct snd_soc_codec codec; struct wm8350_output out1; struct wm8350_output out2; + struct wm8350_jack_data hpl; + struct wm8350_jack_data hpr; struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)]; }; @@ -1328,6 +1335,95 @@ static int wm8350_resume(struct platform_device *pdev) return 0; } +static void wm8350_hp_jack_handler(struct wm8350 *wm8350, int irq, void *data) +{ + struct wm8350_data *priv = data; + u16 reg; + int report; + int mask; + struct wm8350_jack_data *jack = NULL; + + switch (irq) { + case WM8350_IRQ_CODEC_JCK_DET_L: + jack = &priv->hpl; + mask = WM8350_JACK_L_LVL; + break; + + case WM8350_IRQ_CODEC_JCK_DET_R: + jack = &priv->hpr; + mask = WM8350_JACK_R_LVL; + break; + + default: + BUG(); + } + + if (!jack->jack) { + dev_warn(wm8350->dev, "Jack interrupt called with no jack\n"); + return; + } + + /* Debounce */ + msleep(200); + + reg = wm8350_reg_read(wm8350, WM8350_JACK_PIN_STATUS); + if (reg & mask) + report = jack->report; + else + report = 0; + + snd_soc_jack_report(jack->jack, report, jack->report); +} + +/** + * wm8350_hp_jack_detect - Enable headphone jack detection. + * + * @codec: WM8350 codec + * @which: left or right jack detect signal + * @jack: jack to report detection events on + * @report: value to report + * + * Enables the headphone jack detection of the WM8350. + */ +int wm8350_hp_jack_detect(struct snd_soc_codec *codec, enum wm8350_jack which, + struct snd_soc_jack *jack, int report) +{ + struct wm8350_data *priv = codec->private_data; + struct wm8350 *wm8350 = codec->control_data; + int irq; + int ena; + + switch (which) { + case WM8350_JDL: + priv->hpl.jack = jack; + priv->hpl.report = report; + irq = WM8350_IRQ_CODEC_JCK_DET_L; + ena = WM8350_JDL_ENA; + break; + + case WM8350_JDR: + priv->hpr.jack = jack; + priv->hpr.report = report; + irq = WM8350_IRQ_CODEC_JCK_DET_R; + ena = WM8350_JDR_ENA; + break; + + default: + return -EINVAL; + } + + wm8350_set_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_TOCLK_ENA); + wm8350_set_bits(wm8350, WM8350_JACK_DETECT, ena); + + /* Sync status */ + wm8350_hp_jack_handler(wm8350, irq, priv); + + wm8350_unmask_irq(wm8350, irq); + + return 0; +} +EXPORT_SYMBOL_GPL(wm8350_hp_jack_detect); + static struct snd_soc_codec *wm8350_codec; static int wm8350_probe(struct platform_device *pdev) @@ -1381,6 +1477,13 @@ static int wm8350_probe(struct platform_device *pdev) wm8350_set_bits(wm8350, WM8350_ROUT2_VOLUME, WM8350_OUT2_VU | WM8350_OUT2R_MUTE); + wm8350_mask_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L); + wm8350_mask_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R); + wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L, + wm8350_hp_jack_handler, priv); + wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R, + wm8350_hp_jack_handler, priv); + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) { dev_err(&pdev->dev, "failed to create pcms\n"); @@ -1411,8 +1514,21 @@ static int wm8350_remove(struct platform_device *pdev) struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->codec; struct wm8350 *wm8350 = codec->control_data; + struct wm8350_data *priv = codec->private_data; int ret; + wm8350_clear_bits(wm8350, WM8350_JACK_DETECT, + WM8350_JDL_ENA | WM8350_JDR_ENA); + wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_TOCLK_ENA); + + wm8350_mask_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L); + wm8350_mask_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R); + wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L); + wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R); + + priv->hpl.jack = NULL; + priv->hpr.jack = NULL; + /* cancel any work waiting to be queued. */ ret = cancel_delayed_work(&codec->delayed_work); diff --git a/sound/soc/codecs/wm8350.h b/sound/soc/codecs/wm8350.h index cc2887aa6c3..d11bd9288cf 100644 --- a/sound/soc/codecs/wm8350.h +++ b/sound/soc/codecs/wm8350.h @@ -17,4 +17,12 @@ extern struct snd_soc_dai wm8350_dai; extern struct snd_soc_codec_device soc_codec_dev_wm8350; +enum wm8350_jack { + WM8350_JDL = 1, + WM8350_JDR = 2, +}; + +int wm8350_hp_jack_detect(struct snd_soc_codec *codec, enum wm8350_jack which, + struct snd_soc_jack *jack, int report); + #endif -- cgit v1.2.3-70-g09d2 From 3e8e1952e3a3dd59b11233a532ca68e6471742d9 Mon Sep 17 00:00:00 2001 From: Ian Molton Date: Fri, 9 Jan 2009 00:23:21 +0000 Subject: ASoC: cleanup duplicated code. Many codec drivers were implementing cookie-cutter copies of the function that adds kcontrols to the codec. This patch moves this code to a common function snd_soc_add_controls() in soc-core.c and updates all drivers using copies of this function to use the new common version. [Edited to raise priority of error log message and document parameters. -- broonie] Signed-off-by: Ian Molton Signed-off-by: Mark Brown --- include/sound/soc.h | 2 ++ sound/soc/codecs/ad1980.c | 17 ++------------ sound/soc/codecs/ak4535.c | 18 ++------------- sound/soc/codecs/ssm2602.c | 18 ++------------- sound/soc/codecs/tlv320aic23.c | 21 ++---------------- sound/soc/codecs/tlv320aic3x.c | 19 ++-------------- sound/soc/codecs/twl4030.c | 19 ++-------------- sound/soc/codecs/uda134x.c | 50 ++++++++++++++---------------------------- sound/soc/codecs/uda1380.c | 18 ++------------- sound/soc/codecs/wm8350.c | 18 ++------------- sound/soc/codecs/wm8510.c | 19 ++-------------- sound/soc/codecs/wm8580.c | 17 ++------------ sound/soc/codecs/wm8728.c | 18 ++------------- sound/soc/codecs/wm8731.c | 19 ++-------------- sound/soc/codecs/wm8750.c | 18 ++------------- sound/soc/codecs/wm8753.c | 18 ++------------- sound/soc/codecs/wm8900.c | 19 ++-------------- sound/soc/codecs/wm8903.c | 18 ++------------- sound/soc/codecs/wm8971.c | 18 ++------------- sound/soc/codecs/wm8990.c | 18 ++------------- sound/soc/codecs/wm9712.c | 18 ++------------- sound/soc/codecs/wm9713.c | 18 ++------------- sound/soc/soc-core.c | 31 ++++++++++++++++++++++++++ 23 files changed, 89 insertions(+), 360 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 9c3ef6a3e9f..9d5a0362a05 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -214,6 +214,8 @@ void snd_soc_free_ac97_codec(struct snd_soc_codec *codec); */ struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template, void *data, char *long_name); +int snd_soc_add_controls(struct snd_soc_codec *codec, + const struct snd_kcontrol_new *controls, int num_controls); int snd_soc_info_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); int snd_soc_info_enum_ext(struct snd_kcontrol *kcontrol, diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 73fdbb4d4a3..c3c5d0eee37 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -93,20 +93,6 @@ SOC_ENUM("Capture Source", ad1980_cap_src), SOC_SINGLE("Mic Boost Switch", AC97_MIC, 6, 1, 0), }; -/* add non dapm controls */ -static int ad1980_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(ad1980_snd_ac97_controls); i++) { - err = snd_ctl_add(codec->card, snd_soc_cnew( - &ad1980_snd_ac97_controls[i], codec, NULL)); - if (err < 0) - return err; - } - return 0; -} - static unsigned int ac97_read(struct snd_soc_codec *codec, unsigned int reg) { @@ -269,7 +255,8 @@ static int ad1980_soc_probe(struct platform_device *pdev) ext_status = ac97_read(codec, AC97_EXTENDED_STATUS); ac97_write(codec, AC97_EXTENDED_STATUS, ext_status&~0x3800); - ad1980_add_controls(codec); + snd_soc_add_controls(codec, ad1980_snd_ac97_controls, + ARRAY_SIZE(ad1980_snd_ac97_controls)); ret = snd_soc_init_card(socdev); if (ret < 0) { printk(KERN_ERR "ad1980: failed to register card\n"); diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 81300d8d42c..f17c363cb1d 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -155,21 +155,6 @@ static const struct snd_kcontrol_new ak4535_snd_controls[] = { SOC_SINGLE("Mic Sidetone Volume", AK4535_VOL, 4, 7, 0), }; -/* add non dapm controls */ -static int ak4535_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(ak4535_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&ak4535_snd_controls[i], codec, NULL)); - if (err < 0) - return err; - } - - return 0; -} - /* Mono 1 Mixer */ static const struct snd_kcontrol_new ak4535_mono1_mixer_controls[] = { SOC_DAPM_SINGLE("Mic Sidetone Switch", AK4535_SIG1, 4, 1, 0), @@ -510,7 +495,8 @@ static int ak4535_init(struct snd_soc_device *socdev) /* power on device */ ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - ak4535_add_controls(codec); + snd_soc_add_controls(codec, ak4535_snd_controls, + ARRAY_SIZE(ak4535_snd_controls)); ak4535_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index cac37361676..ec7fe3b7b0c 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -151,21 +151,6 @@ SOC_ENUM("Capture Source", ssm2602_enum[0]), SOC_ENUM("Playback De-emphasis", ssm2602_enum[1]), }; -/* add non dapm controls */ -static int ssm2602_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(ssm2602_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&ssm2602_snd_controls[i], codec, NULL)); - if (err < 0) - return err; - } - - return 0; -} - /* Output Mixer */ static const struct snd_kcontrol_new ssm2602_output_mixer_controls[] = { SOC_DAPM_SINGLE("Line Bypass Switch", SSM2602_APANA, 3, 1, 0), @@ -622,7 +607,8 @@ static int ssm2602_init(struct snd_soc_device *socdev) APANA_ENABLE_MIC_BOOST); ssm2602_write(codec, SSM2602_PWR, 0); - ssm2602_add_controls(codec); + snd_soc_add_controls(codec, ssm2602_snd_controls, + ARRAY_SIZE(ssm2602_snd_controls)); ssm2602_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index cfdea007c4c..a0e47c1dcd6 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -183,24 +183,6 @@ static const struct snd_kcontrol_new tlv320aic23_snd_controls[] = { SOC_ENUM("Playback De-emphasis", tlv320aic23_deemph), }; -/* add non dapm controls */ -static int tlv320aic23_add_controls(struct snd_soc_codec *codec) -{ - - int err, i; - - for (i = 0; i < ARRAY_SIZE(tlv320aic23_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&tlv320aic23_snd_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - - return 0; - -} - /* PGA Mixer controls for Line and Mic switch */ static const struct snd_kcontrol_new tlv320aic23_output_mixer_controls[] = { SOC_DAPM_SINGLE("Line Bypass Switch", TLV320AIC23_ANLG, 3, 1, 0), @@ -718,7 +700,8 @@ static int tlv320aic23_init(struct snd_soc_device *socdev) tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x1); - tlv320aic23_add_controls(codec); + snd_soc_add_controls(codec, tlv320aic23_snd_controls, + ARRAY_SIZE(tlv320aic23_snd_controls)); tlv320aic23_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index b47a749c5ea..36ab0198ca3 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -311,22 +311,6 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { SOC_ENUM("ADC HPF Cut-off", aic3x_enum[ADC_HPF_ENUM]), }; -/* add non dapm controls */ -static int aic3x_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(aic3x_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&aic3x_snd_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - - return 0; -} - /* Left DAC Mux */ static const struct snd_kcontrol_new aic3x_left_dac_mux_controls = SOC_DAPM_ENUM("Route", aic3x_enum[LDAC_ENUM]); @@ -1224,7 +1208,8 @@ static int aic3x_init(struct snd_soc_device *socdev) aic3x_write(codec, AIC3X_GPIO1_REG, (setup->gpio_func[0] & 0xf) << 4); aic3x_write(codec, AIC3X_GPIO2_REG, (setup->gpio_func[1] & 0xf) << 4); - aic3x_add_controls(codec); + snd_soc_add_controls(codec, aic3x_snd_controls, + ARRAY_SIZE(aic3x_snd_controls)); aic3x_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index fd0f338374a..253063fd319 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -670,22 +670,6 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = { 0, 3, 5, 0, input_gain_tlv), }; -/* add non dapm controls */ -static int twl4030_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(twl4030_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&twl4030_snd_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - - return 0; -} - static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { /* Left channel inputs */ SND_SOC_DAPM_INPUT("MAINMIC"), @@ -1233,7 +1217,8 @@ static int twl4030_init(struct snd_soc_device *socdev) /* power on device */ twl4030_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - twl4030_add_controls(codec); + snd_soc_add_controls(codec, twl4030_snd_controls, + ARRAY_SIZE(twl4030_snd_controls)); twl4030_add_widgets(codec); ret = snd_soc_init_card(socdev); diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index a2c5064a774..277825d155a 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -431,39 +431,6 @@ SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]), SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0), }; -static int uda134x_add_controls(struct snd_soc_codec *codec) -{ - int err, i, n; - const struct snd_kcontrol_new *ctrls; - struct uda134x_platform_data *pd = codec->control_data; - - switch (pd->model) { - case UDA134X_UDA1340: - case UDA134X_UDA1344: - n = ARRAY_SIZE(uda1340_snd_controls); - ctrls = uda1340_snd_controls; - break; - case UDA134X_UDA1341: - n = ARRAY_SIZE(uda1341_snd_controls); - ctrls = uda1341_snd_controls; - break; - default: - printk(KERN_ERR "%s unkown codec type: %d", - __func__, pd->model); - return -EINVAL; - } - - for (i = 0; i < n; i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&ctrls[i], - codec, NULL)); - if (err < 0) - return err; - } - - return 0; -} - struct snd_soc_dai uda134x_dai = { .name = "UDA134X", /* playback capabilities */ @@ -572,7 +539,22 @@ static int uda134x_soc_probe(struct platform_device *pdev) goto pcm_err; } - ret = uda134x_add_controls(codec); + switch (pd->model) { + case UDA134X_UDA1340: + case UDA134X_UDA1344: + ret = snd_soc_add_controls(codec, uda1340_snd_controls, + ARRAY_SIZE(uda1340_snd_controls)); + break; + case UDA134X_UDA1341: + ret = snd_soc_add_controls(codec, uda1341_snd_controls, + ARRAY_SIZE(uda1341_snd_controls)); + break; + default: + printk(KERN_ERR "%s unkown codec type: %d", + __func__, pd->model); + return -EINVAL; + } + if (ret < 0) { printk(KERN_ERR "UDA134X: failed to register controls\n"); goto pcm_err; diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index e6bf0844fbf..a957b4365b9 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -271,21 +271,6 @@ static const struct snd_kcontrol_new uda1380_snd_controls[] = { SOC_SINGLE("AGC Switch", UDA1380_AGC, 0, 1, 0), }; -/* add non dapm controls */ -static int uda1380_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(uda1380_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&uda1380_snd_controls[i], codec, NULL)); - if (err < 0) - return err; - } - - return 0; -} - /* Input mux */ static const struct snd_kcontrol_new uda1380_input_mux_control = SOC_DAPM_ENUM("Route", uda1380_input_sel_enum); @@ -675,7 +660,8 @@ static int uda1380_init(struct snd_soc_device *socdev, int dac_clk) } /* uda1380 init */ - uda1380_add_controls(codec); + snd_soc_add_controls(codec, uda1380_snd_controls, + ARRAY_SIZE(uda1380_snd_controls)); uda1380_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 47a9dabb523..2e0db29b499 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -782,21 +782,6 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Beep", NULL, "IN3R PGA"}, }; -static int wm8350_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm8350_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8350_snd_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - - return 0; -} - static int wm8350_add_widgets(struct snd_soc_codec *codec) { int ret; @@ -1490,7 +1475,8 @@ static int wm8350_probe(struct platform_device *pdev) return ret; } - wm8350_add_controls(codec); + snd_soc_add_controls(codec, wm8350_snd_controls, + ARRAY_SIZE(wm8350_snd_controls)); wm8350_add_widgets(codec); wm8350_set_bias_level(codec, SND_SOC_BIAS_STANDBY); diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 40f8238df71..abe7cce8771 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -171,22 +171,6 @@ SOC_SINGLE("Capture Boost(+20dB)", WM8510_ADCBOOST, 8, 1, 0), SOC_SINGLE("Mono Playback Switch", WM8510_MONOMIX, 6, 1, 1), }; -/* add non dapm controls */ -static int wm8510_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm8510_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8510_snd_controls[i], codec, - NULL)); - if (err < 0) - return err; - } - - return 0; -} - /* Speaker Output Mixer */ static const struct snd_kcontrol_new wm8510_speaker_mixer_controls[] = { SOC_DAPM_SINGLE("Line Bypass Switch", WM8510_SPKMIX, 1, 1, 0), @@ -656,7 +640,8 @@ static int wm8510_init(struct snd_soc_device *socdev) /* power on device */ codec->bias_level = SND_SOC_BIAS_OFF; wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - wm8510_add_controls(codec); + snd_soc_add_controls(codec, wm8510_snd_controls, + ARRAY_SIZE(wm8510_snd_controls)); wm8510_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index d004e584529..9b75a377453 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -330,20 +330,6 @@ SOC_DOUBLE("ADC Mute Switch", WM8580_ADC_CONTROL1, 0, 1, 1, 0), SOC_SINGLE("ADC High-Pass Filter Switch", WM8580_ADC_CONTROL1, 4, 1, 0), }; -/* Add non-DAPM controls */ -static int wm8580_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm8580_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8580_snd_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - return 0; -} static const struct snd_soc_dapm_widget wm8580_dapm_widgets[] = { SND_SOC_DAPM_DAC("DAC1", "Playback", WM8580_PWRDN1, 2, 1), SND_SOC_DAPM_DAC("DAC2", "Playback", WM8580_PWRDN1, 3, 1), @@ -866,7 +852,8 @@ static int wm8580_init(struct snd_soc_device *socdev) goto pcm_err; } - wm8580_add_controls(codec); + snd_soc_add_controls(codec, wm8580_snd_controls, + ARRAY_SIZE(wm8580_snd_controls)); wm8580_add_widgets(codec); ret = snd_soc_init_card(socdev); diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index 80b11983e13..defa310bc7d 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -92,21 +92,6 @@ SOC_DOUBLE_R_TLV("Digital Playback Volume", WM8728_DACLVOL, WM8728_DACRVOL, SOC_SINGLE("Deemphasis", WM8728_DACCTL, 1, 1, 0), }; -static int wm8728_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm8728_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8728_snd_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - - return 0; -} - /* * DAPM controls. */ @@ -330,7 +315,8 @@ static int wm8728_init(struct snd_soc_device *socdev) /* power on device */ wm8728_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - wm8728_add_controls(codec); + snd_soc_add_controls(codec, wm8728_snd_controls, + ARRAY_SIZE(wm8728_snd_controls)); wm8728_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index c444b9f2701..96d6e1aeaf4 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -129,22 +129,6 @@ SOC_SINGLE("Store DC Offset Switch", WM8731_APDIGI, 4, 1, 0), SOC_ENUM("Playback De-emphasis", wm8731_enum[1]), }; -/* add non dapm controls */ -static int wm8731_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm8731_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8731_snd_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - - return 0; -} - /* Output Mixer */ static const struct snd_kcontrol_new wm8731_output_mixer_controls[] = { SOC_DAPM_SINGLE("Line Bypass Switch", WM8731_APANA, 3, 1, 0), @@ -543,7 +527,8 @@ static int wm8731_init(struct snd_soc_device *socdev) reg = wm8731_read_reg_cache(codec, WM8731_RINVOL); wm8731_write(codec, WM8731_RINVOL, reg & ~0x0100); - wm8731_add_controls(codec); + snd_soc_add_controls(codec, wm8731_snd_controls, + ARRAY_SIZE(wm8731_snd_controls)); wm8731_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 5997fa68e0d..1578569793a 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -231,21 +231,6 @@ SOC_SINGLE("Mono Playback Volume", WM8750_MOUTV, 0, 127, 0), }; -/* add non dapm controls */ -static int wm8750_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm8750_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8750_snd_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - return 0; -} - /* * DAPM Controls */ @@ -816,7 +801,8 @@ static int wm8750_init(struct snd_soc_device *socdev) reg = wm8750_read_reg_cache(codec, WM8750_RINVOL); wm8750_write(codec, WM8750_RINVOL, reg | 0x0100); - wm8750_add_controls(codec); + snd_soc_add_controls(codec, wm8750_snd_controls, + ARRAY_SIZE(wm8750_snd_controls)); wm8750_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 6c21b50c937..7283178e0eb 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -339,21 +339,6 @@ SOC_ENUM("ADC Data Select", wm8753_enum[27]), SOC_ENUM("ROUT2 Phase", wm8753_enum[28]), }; -/* add non dapm controls */ -static int wm8753_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm8753_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8753_snd_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - return 0; -} - /* * _DAPM_ Controls */ @@ -1603,7 +1588,8 @@ static int wm8753_init(struct snd_soc_device *socdev) reg = wm8753_read_reg_cache(codec, WM8753_RINVOL); wm8753_write(codec, WM8753_RINVOL, reg | 0x0100); - wm8753_add_controls(codec); + snd_soc_add_controls(codec, wm8753_snd_controls, + ARRAY_SIZE(wm8753_snd_controls)); wm8753_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 6767de10ded..1e08d4f065f 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -517,22 +517,6 @@ SOC_SINGLE("LINEOUT2 LP -12dB", WM8900_REG_LOUTMIXCTL1, }; -/* add non dapm controls */ -static int wm8900_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm8900_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8900_snd_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - - return 0; -} - static const struct snd_kcontrol_new wm8900_dapm_loutput2_control = SOC_DAPM_SINGLE("LINEOUT2L Switch", WM8900_REG_POWER3, 6, 1, 0); @@ -1439,7 +1423,8 @@ static int wm8900_probe(struct platform_device *pdev) goto pcm_err; } - wm8900_add_controls(codec); + snd_soc_add_controls(codec, wm8900_snd_controls, + ARRAY_SIZE(wm8900_snd_controls)); wm8900_add_widgets(codec); ret = snd_soc_init_card(socdev); diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index bde74546db4..6ff34b957dc 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -744,21 +744,6 @@ SOC_DOUBLE_R_TLV("Speaker Volume", 0, 63, 0, out_tlv), }; -static int wm8903_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm8903_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8903_snd_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - - return 0; -} - static const struct snd_kcontrol_new linput_mode_mux = SOC_DAPM_ENUM("Left Input Mode Mux", linput_mode_enum); @@ -1737,7 +1722,8 @@ static int wm8903_probe(struct platform_device *pdev) goto err; } - wm8903_add_controls(socdev->codec); + snd_soc_add_controls(socdev->codec, wm8903_snd_controls, + ARRAY_SIZE(wm8903_snd_controls)); wm8903_add_widgets(socdev->codec); ret = snd_soc_init_card(socdev); diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 88ead7f8dd9..c8bd9b06f33 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -195,21 +195,6 @@ static const struct snd_kcontrol_new wm8971_snd_controls[] = { SOC_DOUBLE_R("Mic Boost", WM8971_LADCIN, WM8971_RADCIN, 4, 3, 0), }; -/* add non-DAPM controls */ -static int wm8971_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm8971_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8971_snd_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - return 0; -} - /* * DAPM Controls */ @@ -745,7 +730,8 @@ static int wm8971_init(struct snd_soc_device *socdev) reg = wm8971_read_reg_cache(codec, WM8971_RINVOL); wm8971_write(codec, WM8971_RINVOL, reg | 0x0100); - wm8971_add_controls(codec); + snd_soc_add_controls(codec, wm8971_snd_controls, + ARRAY_SIZE(wm8971_snd_controls)); wm8971_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 5b5afc14447..6b2778632d5 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -417,21 +417,6 @@ SOC_SINGLE("RIN34 Mute Switch", WM8990_RIGHT_LINE_INPUT_3_4_VOLUME, }; -/* add non dapm controls */ -static int wm8990_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm8990_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8990_snd_controls[i], codec, - NULL)); - if (err < 0) - return err; - } - return 0; -} - /* * _DAPM_ Controls */ @@ -1460,7 +1445,8 @@ static int wm8990_init(struct snd_soc_device *socdev) wm8990_write(codec, WM8990_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8)); wm8990_write(codec, WM8990_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8)); - wm8990_add_controls(codec); + snd_soc_add_controls(codec, wm8990_snd_controls, + ARRAY_SIZE(wm8990_snd_controls)); wm8990_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index af83d629078..1b0ace0f4dc 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -154,21 +154,6 @@ SOC_SINGLE("Mic 2 Volume", AC97_MIC, 0, 31, 1), SOC_SINGLE("Mic 20dB Boost Switch", AC97_MIC, 7, 1, 0), }; -/* add non dapm controls */ -static int wm9712_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm9712_snd_ac97_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm9712_snd_ac97_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - return 0; -} - /* We have to create a fake left and right HP mixers because * the codec only has a single control that is shared by both channels. * This makes it impossible to determine the audio path. @@ -698,7 +683,8 @@ static int wm9712_soc_probe(struct platform_device *pdev) ac97_write(codec, AC97_VIDEO, ac97_read(codec, AC97_VIDEO) | 0x3000); wm9712_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - wm9712_add_controls(codec); + snd_soc_add_controls(codec, wm9712_snd_ac97_controls, + ARRAY_SIZE(wm9712_snd_ac97_controls)); wm9712_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index f3ca8aaf013..a45622620db 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -190,21 +190,6 @@ SOC_SINGLE("3D Lower Cut-off Switch", AC97_REC_GAIN_MIC, 4, 1, 0), SOC_SINGLE("3D Depth", AC97_REC_GAIN_MIC, 0, 15, 1), }; -/* add non dapm controls */ -static int wm9713_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm9713_snd_ac97_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm9713_snd_ac97_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - return 0; -} - /* We have to create a fake left and right HP mixers because * the codec only has a single control that is shared by both channels. * This makes it impossible to determine the audio path using the current @@ -1245,7 +1230,8 @@ static int wm9713_soc_probe(struct platform_device *pdev) reg = ac97_read(codec, AC97_CD) & 0x7fff; ac97_write(codec, AC97_CD, reg); - wm9713_add_controls(codec); + snd_soc_add_controls(codec, wm9713_snd_ac97_controls, + ARRAY_SIZE(wm9713_snd_ac97_controls)); wm9713_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 6cbe7e82f23..d3b97a7542e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1494,6 +1494,37 @@ struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template, } EXPORT_SYMBOL_GPL(snd_soc_cnew); +/** + * snd_soc_add_controls - add an array of controls to a codec. + * Convienience function to add a list of controls. Many codecs were + * duplicating this code. + * + * @codec: codec to add controls to + * @controls: array of controls to add + * @num_controls: number of elements in the array + * + * Return 0 for success, else error. + */ +int snd_soc_add_controls(struct snd_soc_codec *codec, + const struct snd_kcontrol_new *controls, int num_controls) +{ + struct snd_card *card = codec->card; + int err, i; + + for (i = 0; i < num_controls; i++) { + const struct snd_kcontrol_new *control = &controls[i]; + err = snd_ctl_add(card, snd_soc_cnew(control, codec, NULL)); + if (err < 0) { + dev_err(codec->dev, "%s: Failed to add %s\n", + codec->name, control->name); + return err; + } + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_add_controls); + /** * snd_soc_info_enum_double - enumerated double mixer info callback * @kcontrol: mixer control -- cgit v1.2.3-70-g09d2 From 199f7978730a4bbd88038fd84212b30759579f1a Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Fri, 9 Jan 2009 23:10:52 +0100 Subject: ALSA: wss-lib: move AD1845 frequency setting into wss-lib This is required to allow the sscape driver to autodetect installed codec. Also, do not create a timer if detected codec has no hardware timer (e.g. AD1848). Signed-off-by: Krzysztof Helt Cc: Rene Herman Signed-off-by: Takashi Iwai --- sound/isa/sscape.c | 113 ++++-------------------------------------------- sound/isa/wss/wss_lib.c | 40 +++++++++++++++++ 2 files changed, 48 insertions(+), 105 deletions(-) diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index 48a16d86583..bc449166d18 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -129,9 +129,6 @@ enum GA_REG { #define DMA_8BIT 0x80 -#define AD1845_FREQ_SEL_MSB 0x16 -#define AD1845_FREQ_SEL_LSB 0x17 - enum card_type { SSCAPE, SSCAPE_PNP, @@ -954,82 +951,6 @@ static int __devinit create_mpu401(struct snd_card *card, int devnum, unsigned l } -/* - * Override for the CS4231 playback format function. - * The AD1845 has much simpler format and rate selection. - */ -static void ad1845_playback_format(struct snd_wss *chip, - struct snd_pcm_hw_params *params, - unsigned char format) -{ - unsigned long flags; - unsigned rate = params_rate(params); - - /* - * The AD1845 can't handle sample frequencies - * outside of 4 kHZ to 50 kHZ - */ - if (rate > 50000) - rate = 50000; - else if (rate < 4000) - rate = 4000; - - spin_lock_irqsave(&chip->reg_lock, flags); - - /* - * Program the AD1845 correctly for the playback stream. - * Note that we do NOT need to toggle the MCE bit because - * the PLAYBACK_ENABLE bit of the Interface Configuration - * register is set. - * - * NOTE: We seem to need to write to the MSB before the LSB - * to get the correct sample frequency. - */ - snd_wss_out(chip, CS4231_PLAYBK_FORMAT, (format & 0xf0)); - snd_wss_out(chip, AD1845_FREQ_SEL_MSB, (unsigned char) (rate >> 8)); - snd_wss_out(chip, AD1845_FREQ_SEL_LSB, (unsigned char) rate); - - spin_unlock_irqrestore(&chip->reg_lock, flags); -} - -/* - * Override for the CS4231 capture format function. - * The AD1845 has much simpler format and rate selection. - */ -static void ad1845_capture_format(struct snd_wss *chip, - struct snd_pcm_hw_params *params, - unsigned char format) -{ - unsigned long flags; - unsigned rate = params_rate(params); - - /* - * The AD1845 can't handle sample frequencies - * outside of 4 kHZ to 50 kHZ - */ - if (rate > 50000) - rate = 50000; - else if (rate < 4000) - rate = 4000; - - spin_lock_irqsave(&chip->reg_lock, flags); - - /* - * Program the AD1845 correctly for the playback stream. - * Note that we do NOT need to toggle the MCE bit because - * the CAPTURE_ENABLE bit of the Interface Configuration - * register is set. - * - * NOTE: We seem to need to write to the MSB before the LSB - * to get the correct sample frequency. - */ - snd_wss_out(chip, CS4231_REC_FORMAT, (format & 0xf0)); - snd_wss_out(chip, AD1845_FREQ_SEL_MSB, (unsigned char) (rate >> 8)); - snd_wss_out(chip, AD1845_FREQ_SEL_LSB, (unsigned char) rate); - - spin_unlock_irqrestore(&chip->reg_lock, flags); -} - /* * Create an AD1845 PCM subdevice on the SoundScape. The AD1845 * is very much like a CS4231, with a few extra bits. We will @@ -1055,11 +976,6 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, unsigned long flags; struct snd_pcm *pcm; -#define AD1845_FREQ_SEL_ENABLE 0x08 - -#define AD1845_PWR_DOWN_CTRL 0x1b -#define AD1845_CRYS_CLOCK_SEL 0x1d - /* * It turns out that the PLAYBACK_ENABLE bit is set * by the lowlevel driver ... @@ -1074,7 +990,6 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, */ if (sscape->type != SSCAPE_VIVO) { - int val; /* * The input clock frequency on the SoundScape must * be 14.31818 MHz, because we must set this register @@ -1082,22 +997,10 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, */ snd_wss_mce_up(chip); spin_lock_irqsave(&chip->reg_lock, flags); - snd_wss_out(chip, AD1845_CRYS_CLOCK_SEL, 0x20); + snd_wss_out(chip, AD1845_CLOCK, 0x20); spin_unlock_irqrestore(&chip->reg_lock, flags); snd_wss_mce_down(chip); - /* - * More custom configuration: - * a) select "mode 2" and provide a current drive of 8mA - * b) enable frequency selection (for capture/playback) - */ - spin_lock_irqsave(&chip->reg_lock, flags); - snd_wss_out(chip, CS4231_MISC_INFO, - CS4231_MODE2 | 0x10); - val = snd_wss_in(chip, AD1845_PWR_DOWN_CTRL); - snd_wss_out(chip, AD1845_PWR_DOWN_CTRL, - val | AD1845_FREQ_SEL_ENABLE); - spin_unlock_irqrestore(&chip->reg_lock, flags); } err = snd_wss_pcm(chip, 0, &pcm); @@ -1113,11 +1016,13 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, "for AD1845 chip\n"); goto _error; } - err = snd_wss_timer(chip, 0, NULL); - if (err < 0) { - snd_printk(KERN_ERR "sscape: No timer device " - "for AD1845 chip\n"); - goto _error; + if (chip->hardware != WSS_HW_AD1848) { + err = snd_wss_timer(chip, 0, NULL); + if (err < 0) { + snd_printk(KERN_ERR "sscape: No timer device " + "for AD1845 chip\n"); + goto _error; + } } if (sscape->type != SSCAPE_VIVO) { @@ -1128,8 +1033,6 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, "MIDI mixer control\n"); goto _error; } - chip->set_playback_format = ad1845_playback_format; - chip->set_capture_format = ad1845_capture_format; } strcpy(card->driver, "SoundScape"); diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c index 3d6c5f2838a..13299aebd07 100644 --- a/sound/isa/wss/wss_lib.c +++ b/sound/isa/wss/wss_lib.c @@ -646,6 +646,24 @@ static void snd_wss_playback_format(struct snd_wss *chip, full_calib = 0; } spin_unlock_irqrestore(&chip->reg_lock, flags); + } else if (chip->hardware == WSS_HW_AD1845) { + unsigned rate = params_rate(params); + + /* + * Program the AD1845 correctly for the playback stream. + * Note that we do NOT need to toggle the MCE bit because + * the PLAYBACK_ENABLE bit of the Interface Configuration + * register is set. + * + * NOTE: We seem to need to write to the MSB before the LSB + * to get the correct sample frequency. + */ + spin_lock_irqsave(&chip->reg_lock, flags); + snd_wss_out(chip, CS4231_PLAYBK_FORMAT, (pdfr & 0xf0)); + snd_wss_out(chip, AD1845_UPR_FREQ_SEL, (rate >> 8) & 0xff); + snd_wss_out(chip, AD1845_LWR_FREQ_SEL, rate & 0xff); + full_calib = 0; + spin_unlock_irqrestore(&chip->reg_lock, flags); } if (full_calib) { snd_wss_mce_up(chip); @@ -690,6 +708,24 @@ static void snd_wss_capture_format(struct snd_wss *chip, full_calib = 0; } spin_unlock_irqrestore(&chip->reg_lock, flags); + } else if (chip->hardware == WSS_HW_AD1845) { + unsigned rate = params_rate(params); + + /* + * Program the AD1845 correctly for the capture stream. + * Note that we do NOT need to toggle the MCE bit because + * the PLAYBACK_ENABLE bit of the Interface Configuration + * register is set. + * + * NOTE: We seem to need to write to the MSB before the LSB + * to get the correct sample frequency. + */ + spin_lock_irqsave(&chip->reg_lock, flags); + snd_wss_out(chip, CS4231_REC_FORMAT, (cdfr & 0xf0)); + snd_wss_out(chip, AD1845_UPR_FREQ_SEL, (rate >> 8) & 0xff); + snd_wss_out(chip, AD1845_LWR_FREQ_SEL, rate & 0xff); + full_calib = 0; + spin_unlock_irqrestore(&chip->reg_lock, flags); } if (full_calib) { snd_wss_mce_up(chip); @@ -1314,6 +1350,10 @@ static int snd_wss_probe(struct snd_wss *chip) chip->image[CS4231_ALT_FEATURE_2] = chip->hardware == WSS_HW_INTERWAVE ? 0xc2 : 0x01; } + /* enable fine grained frequency selection */ + if (chip->hardware == WSS_HW_AD1845) + chip->image[AD1845_PWR_DOWN] = 8; + ptr = (unsigned char *) &chip->image; regnum = (chip->hardware & WSS_HW_AD1848_MASK) ? 16 : 32; snd_wss_mce_down(chip); -- cgit v1.2.3-70-g09d2 From 53fb1e63599438bd5f6fbb852023d80916d83983 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 28 Dec 2008 16:32:08 +0100 Subject: ALSA: Introduce snd_card_create() Introduced snd_card_create() function as a replacement of snd_card_new(). The new function returns a negative error code so that the probe callback can return the proper error code, while snd_card_new() can give only NULL check. The old snd_card_new() is still provided as an inline function but with __deprecated attribute. It'll be removed soon later. Signed-off-by: Takashi Iwai --- include/sound/core.h | 14 +++++++++++++- sound/core/init.c | 47 +++++++++++++++++++++++++++++++---------------- 2 files changed, 44 insertions(+), 17 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index f632484bc74..25420c3b551 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -296,8 +296,20 @@ int snd_card_locked(int card); extern int (*snd_mixer_oss_notify_callback)(struct snd_card *card, int cmd); #endif +int snd_card_create(int idx, const char *id, + struct module *module, int extra_size, + struct snd_card **card_ret); + +static inline __deprecated struct snd_card *snd_card_new(int idx, const char *id, - struct module *module, int extra_size); + struct module *module, int extra_size) +{ + struct snd_card *card; + if (snd_card_create(idx, id, module, extra_size, &card) < 0) + return NULL; + return card; +} + int snd_card_disconnect(struct snd_card *card); int snd_card_free(struct snd_card *card); int snd_card_free_when_closed(struct snd_card *card); diff --git a/sound/core/init.c b/sound/core/init.c index 0d5520c415d..dc4b80c7f31 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -121,31 +121,44 @@ static inline int init_info_for_card(struct snd_card *card) #endif /** - * snd_card_new - create and initialize a soundcard structure + * snd_card_create - create and initialize a soundcard structure * @idx: card index (address) [0 ... (SNDRV_CARDS-1)] * @xid: card identification (ASCII string) * @module: top level module for locking * @extra_size: allocate this extra size after the main soundcard structure + * @card_ret: the pointer to store the created card instance * * Creates and initializes a soundcard structure. * - * Returns kmallocated snd_card structure. Creates the ALSA control interface - * (which is blocked until snd_card_register function is called). + * The function allocates snd_card instance via kzalloc with the given + * space for the driver to use freely. The allocated struct is stored + * in the given card_ret pointer. + * + * Returns zero if successful or a negative error code. */ -struct snd_card *snd_card_new(int idx, const char *xid, - struct module *module, int extra_size) +int snd_card_create(int idx, const char *xid, + struct module *module, int extra_size, + struct snd_card **card_ret) { struct snd_card *card; int err, idx2; + if (snd_BUG_ON(!card_ret)) + return -EINVAL; + *card_ret = NULL; + if (extra_size < 0) extra_size = 0; card = kzalloc(sizeof(*card) + extra_size, GFP_KERNEL); - if (card == NULL) - return NULL; + if (!card) + return -ENOMEM; if (xid) { - if (!snd_info_check_reserved_words(xid)) + if (!snd_info_check_reserved_words(xid)) { + snd_printk(KERN_ERR + "given id string '%s' is reserved.\n", xid); + err = -EBUSY; goto __error; + } strlcpy(card->id, xid, sizeof(card->id)); } err = 0; @@ -202,26 +215,28 @@ struct snd_card *snd_card_new(int idx, const char *xid, #endif /* the control interface cannot be accessed from the user space until */ /* snd_cards_bitmask and snd_cards are set with snd_card_register */ - if ((err = snd_ctl_create(card)) < 0) { - snd_printd("unable to register control minors\n"); + err = snd_ctl_create(card); + if (err < 0) { + snd_printk(KERN_ERR "unable to register control minors\n"); goto __error; } - if ((err = snd_info_card_create(card)) < 0) { - snd_printd("unable to create card info\n"); + err = snd_info_card_create(card); + if (err < 0) { + snd_printk(KERN_ERR "unable to create card info\n"); goto __error_ctl; } if (extra_size > 0) card->private_data = (char *)card + sizeof(struct snd_card); - return card; + *card_ret = card; + return 0; __error_ctl: snd_device_free_all(card, SNDRV_DEV_CMD_PRE); __error: kfree(card); - return NULL; + return err; } - -EXPORT_SYMBOL(snd_card_new); +EXPORT_SYMBOL(snd_card_create); /* return non-zero if a card is already locked */ int snd_card_locked(int card) -- cgit v1.2.3-70-g09d2 From c95eadd2f1afd2ba643e85a8dfc9079a3f03ae47 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 28 Dec 2008 16:43:35 +0100 Subject: ALSA: Convert to snd_card_create() in sound/isa/* Convert from snd_card_new() to the new snd_card_create() function. Signed-off-by: Takashi Iwai --- sound/isa/ad1816a/ad1816a.c | 7 ++++--- sound/isa/ad1848/ad1848.c | 6 +++--- sound/isa/adlib.c | 6 +++--- sound/isa/als100.c | 7 ++++--- sound/isa/azt2320.c | 7 ++++--- sound/isa/cmi8330.c | 26 ++++++++++++++------------ sound/isa/cs423x/cs4231.c | 6 +++--- sound/isa/cs423x/cs4236.c | 7 ++++--- sound/isa/dt019x.c | 7 ++++--- sound/isa/es1688/es1688.c | 6 +++--- sound/isa/es18xx.c | 7 +++++-- sound/isa/gus/gusclassic.c | 6 +++--- sound/isa/gus/gusextreme.c | 6 +++--- sound/isa/gus/gusmax.c | 8 ++++---- sound/isa/gus/interwave.c | 7 ++++--- sound/isa/opl3sa2.c | 31 +++++++++++++++++-------------- sound/isa/opti9xx/miro.c | 7 ++++--- sound/isa/opti9xx/opti92x-ad1848.c | 6 ++++-- sound/isa/sb/es968.c | 7 ++++--- sound/isa/sb/sb16.c | 9 ++++++--- sound/isa/sb/sb8.c | 8 ++++---- sound/isa/sc6000.c | 6 +++--- sound/isa/sgalaxy.c | 6 +++--- sound/isa/sscape.c | 16 ++++++++-------- sound/isa/wavefront/wavefront.c | 7 ++++--- 25 files changed, 122 insertions(+), 100 deletions(-) diff --git a/sound/isa/ad1816a/ad1816a.c b/sound/isa/ad1816a/ad1816a.c index 77524244a84..9660e598232 100644 --- a/sound/isa/ad1816a/ad1816a.c +++ b/sound/isa/ad1816a/ad1816a.c @@ -157,9 +157,10 @@ static int __devinit snd_card_ad1816a_probe(int dev, struct pnp_card_link *pcard struct snd_ad1816a *chip; struct snd_opl3 *opl3; - if ((card = snd_card_new(index[dev], id[dev], THIS_MODULE, - sizeof(struct snd_card_ad1816a))) == NULL) - return -ENOMEM; + error = snd_card_create(index[dev], id[dev], THIS_MODULE, + sizeof(struct snd_card_ad1816a), &card); + if (error < 0) + return error; acard = (struct snd_card_ad1816a *)card->private_data; if ((error = snd_card_ad1816a_pnp(dev, acard, pcard, pid))) { diff --git a/sound/isa/ad1848/ad1848.c b/sound/isa/ad1848/ad1848.c index 223a6c03881..4beeb6f98e0 100644 --- a/sound/isa/ad1848/ad1848.c +++ b/sound/isa/ad1848/ad1848.c @@ -91,9 +91,9 @@ static int __devinit snd_ad1848_probe(struct device *dev, unsigned int n) struct snd_pcm *pcm; int error; - card = snd_card_new(index[n], id[n], THIS_MODULE, 0); - if (!card) - return -EINVAL; + error = snd_card_create(index[n], id[n], THIS_MODULE, 0, &card); + if (error < 0) + return error; error = snd_wss_create(card, port[n], -1, irq[n], dma1[n], -1, thinkpad[n] ? WSS_HW_THINKPAD : WSS_HW_DETECT, diff --git a/sound/isa/adlib.c b/sound/isa/adlib.c index 374b7177e11..7465ae036e0 100644 --- a/sound/isa/adlib.c +++ b/sound/isa/adlib.c @@ -53,10 +53,10 @@ static int __devinit snd_adlib_probe(struct device *dev, unsigned int n) struct snd_opl3 *opl3; int error; - card = snd_card_new(index[n], id[n], THIS_MODULE, 0); - if (!card) { + error = snd_card_create(index[n], id[n], THIS_MODULE, 0, &card); + if (error < 0) { dev_err(dev, "could not create card\n"); - return -EINVAL; + return error; } card->private_data = request_region(port[n], 4, CRD_NAME); diff --git a/sound/isa/als100.c b/sound/isa/als100.c index f1ce30f379c..5fd52e4d707 100644 --- a/sound/isa/als100.c +++ b/sound/isa/als100.c @@ -163,9 +163,10 @@ static int __devinit snd_card_als100_probe(int dev, struct snd_card_als100 *acard; struct snd_opl3 *opl3; - if ((card = snd_card_new(index[dev], id[dev], THIS_MODULE, - sizeof(struct snd_card_als100))) == NULL) - return -ENOMEM; + error = snd_card_create(index[dev], id[dev], THIS_MODULE, + sizeof(struct snd_card_als100), &card); + if (error < 0) + return error; acard = card->private_data; if ((error = snd_card_als100_pnp(dev, acard, pcard, pid))) { diff --git a/sound/isa/azt2320.c b/sound/isa/azt2320.c index 3e74d1a3928..f7aa637b0d1 100644 --- a/sound/isa/azt2320.c +++ b/sound/isa/azt2320.c @@ -184,9 +184,10 @@ static int __devinit snd_card_azt2320_probe(int dev, struct snd_wss *chip; struct snd_opl3 *opl3; - if ((card = snd_card_new(index[dev], id[dev], THIS_MODULE, - sizeof(struct snd_card_azt2320))) == NULL) - return -ENOMEM; + error = snd_card_create(index[dev], id[dev], THIS_MODULE, + sizeof(struct snd_card_azt2320), &card); + if (error < 0) + return error; acard = (struct snd_card_azt2320 *)card->private_data; if ((error = snd_card_azt2320_pnp(dev, acard, pcard, pid))) { diff --git a/sound/isa/cmi8330.c b/sound/isa/cmi8330.c index e49aec700a5..24e60902f8c 100644 --- a/sound/isa/cmi8330.c +++ b/sound/isa/cmi8330.c @@ -467,20 +467,22 @@ static int snd_cmi8330_resume(struct snd_card *card) #define PFX "cmi8330: " -static struct snd_card *snd_cmi8330_card_new(int dev) +static int snd_cmi8330_card_new(int dev, struct snd_card **cardp) { struct snd_card *card; struct snd_cmi8330 *acard; + int err; - card = snd_card_new(index[dev], id[dev], THIS_MODULE, - sizeof(struct snd_cmi8330)); - if (card == NULL) { + err = snd_card_create(index[dev], id[dev], THIS_MODULE, + sizeof(struct snd_cmi8330), &card); + if (err < 0) { snd_printk(KERN_ERR PFX "could not get a new card\n"); - return NULL; + return err; } acard = card->private_data; acard->card = card; - return card; + *cardp = card; + return 0; } static int __devinit snd_cmi8330_probe(struct snd_card *card, int dev) @@ -564,9 +566,9 @@ static int __devinit snd_cmi8330_isa_probe(struct device *pdev, struct snd_card *card; int err; - card = snd_cmi8330_card_new(dev); - if (! card) - return -ENOMEM; + err = snd_cmi8330_card_new(dev, &card); + if (err < 0) + return err; snd_card_set_dev(card, pdev); if ((err = snd_cmi8330_probe(card, dev)) < 0) { snd_card_free(card); @@ -628,9 +630,9 @@ static int __devinit snd_cmi8330_pnp_detect(struct pnp_card_link *pcard, if (dev >= SNDRV_CARDS) return -ENODEV; - card = snd_cmi8330_card_new(dev); - if (! card) - return -ENOMEM; + res = snd_cmi8330_card_new(dev, &card); + if (res < 0) + return res; if ((res = snd_cmi8330_pnp(dev, card->private_data, pcard, pid)) < 0) { snd_printk(KERN_ERR PFX "PnP detection failed\n"); snd_card_free(card); diff --git a/sound/isa/cs423x/cs4231.c b/sound/isa/cs423x/cs4231.c index f019d449e2d..cb9153e75b8 100644 --- a/sound/isa/cs423x/cs4231.c +++ b/sound/isa/cs423x/cs4231.c @@ -95,9 +95,9 @@ static int __devinit snd_cs4231_probe(struct device *dev, unsigned int n) struct snd_pcm *pcm; int error; - card = snd_card_new(index[n], id[n], THIS_MODULE, 0); - if (!card) - return -EINVAL; + error = snd_card_create(index[n], id[n], THIS_MODULE, 0, &card); + if (error < 0) + return error; error = snd_wss_create(card, port[n], -1, irq[n], dma1[n], dma2[n], WSS_HW_DETECT, 0, &chip); diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c index 019c9401663..db830682804 100644 --- a/sound/isa/cs423x/cs4236.c +++ b/sound/isa/cs423x/cs4236.c @@ -385,10 +385,11 @@ static void snd_card_cs4236_free(struct snd_card *card) static struct snd_card *snd_cs423x_card_new(int dev) { struct snd_card *card; + int err; - card = snd_card_new(index[dev], id[dev], THIS_MODULE, - sizeof(struct snd_card_cs4236)); - if (card == NULL) + err = snd_card_create(index[dev], id[dev], THIS_MODULE, + sizeof(struct snd_card_cs4236), &card); + if (err < 0) return NULL; card->private_free = snd_card_cs4236_free; return card; diff --git a/sound/isa/dt019x.c b/sound/isa/dt019x.c index a0242c3b613..80f5b1af9be 100644 --- a/sound/isa/dt019x.c +++ b/sound/isa/dt019x.c @@ -150,9 +150,10 @@ static int __devinit snd_card_dt019x_probe(int dev, struct pnp_card_link *pcard, struct snd_card_dt019x *acard; struct snd_opl3 *opl3; - if ((card = snd_card_new(index[dev], id[dev], THIS_MODULE, - sizeof(struct snd_card_dt019x))) == NULL) - return -ENOMEM; + error = snd_card_create(index[dev], id[dev], THIS_MODULE, + sizeof(struct snd_card_dt019x), &card); + if (error < 0) + return error; acard = card->private_data; snd_card_set_dev(card, &pcard->card->dev); diff --git a/sound/isa/es1688/es1688.c b/sound/isa/es1688/es1688.c index b46377139cf..d746750410e 100644 --- a/sound/isa/es1688/es1688.c +++ b/sound/isa/es1688/es1688.c @@ -122,9 +122,9 @@ static int __devinit snd_es1688_probe(struct device *dev, unsigned int n) struct snd_pcm *pcm; int error; - card = snd_card_new(index[n], id[n], THIS_MODULE, 0); - if (!card) - return -EINVAL; + error = snd_card_create(index[n], id[n], THIS_MODULE, 0, &card); + if (error < 0) + return error; error = snd_es1688_legacy_create(card, dev, n, &chip); if (error < 0) diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index 90498e4ca26..c24c6322fcc 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -2127,8 +2127,11 @@ static int __devinit snd_audiodrive_pnpc(int dev, struct snd_audiodrive *acard, static struct snd_card *snd_es18xx_card_new(int dev) { - return snd_card_new(index[dev], id[dev], THIS_MODULE, - sizeof(struct snd_audiodrive)); + struct snd_card *card; + if (snd_card_create(index[dev], id[dev], THIS_MODULE, + sizeof(struct snd_audiodrive), &card) < 0) + return NULL; + return card; } static int __devinit snd_audiodrive_probe(struct snd_card *card, int dev) diff --git a/sound/isa/gus/gusclassic.c b/sound/isa/gus/gusclassic.c index 426532a4d73..086b8f0e0f9 100644 --- a/sound/isa/gus/gusclassic.c +++ b/sound/isa/gus/gusclassic.c @@ -148,9 +148,9 @@ static int __devinit snd_gusclassic_probe(struct device *dev, unsigned int n) struct snd_gus_card *gus; int error; - card = snd_card_new(index[n], id[n], THIS_MODULE, 0); - if (!card) - return -EINVAL; + error = snd_card_create(index[n], id[n], THIS_MODULE, 0, &card); + if (error < 0) + return error; if (pcm_channels[n] < 2) pcm_channels[n] = 2; diff --git a/sound/isa/gus/gusextreme.c b/sound/isa/gus/gusextreme.c index 7ad4c3b41a8..180a8dea6bd 100644 --- a/sound/isa/gus/gusextreme.c +++ b/sound/isa/gus/gusextreme.c @@ -241,9 +241,9 @@ static int __devinit snd_gusextreme_probe(struct device *dev, unsigned int n) struct snd_opl3 *opl3; int error; - card = snd_card_new(index[n], id[n], THIS_MODULE, 0); - if (!card) - return -EINVAL; + error = snd_card_create(index[n], id[n], THIS_MODULE, 0, &card); + if (error < 0) + return error; if (mpu_port[n] == SNDRV_AUTO_PORT) mpu_port[n] = 0; diff --git a/sound/isa/gus/gusmax.c b/sound/isa/gus/gusmax.c index f94c1976e63..f26eac8d811 100644 --- a/sound/isa/gus/gusmax.c +++ b/sound/isa/gus/gusmax.c @@ -214,10 +214,10 @@ static int __devinit snd_gusmax_probe(struct device *pdev, unsigned int dev) struct snd_wss *wss; struct snd_gusmax *maxcard; - card = snd_card_new(index[dev], id[dev], THIS_MODULE, - sizeof(struct snd_gusmax)); - if (card == NULL) - return -ENOMEM; + err = snd_card_create(index[dev], id[dev], THIS_MODULE, + sizeof(struct snd_gusmax), &card); + if (err < 0) + return err; card->private_free = snd_gusmax_free; maxcard = (struct snd_gusmax *)card->private_data; maxcard->card = card; diff --git a/sound/isa/gus/interwave.c b/sound/isa/gus/interwave.c index 5faecfb602d..e040c763891 100644 --- a/sound/isa/gus/interwave.c +++ b/sound/isa/gus/interwave.c @@ -630,10 +630,11 @@ static struct snd_card *snd_interwave_card_new(int dev) { struct snd_card *card; struct snd_interwave *iwcard; + int err; - card = snd_card_new(index[dev], id[dev], THIS_MODULE, - sizeof(struct snd_interwave)); - if (card == NULL) + err = snd_card_create(index[dev], id[dev], THIS_MODULE, + sizeof(struct snd_interwave), &card); + if (err < 0) return NULL; iwcard = card->private_data; iwcard->card = card; diff --git a/sound/isa/opl3sa2.c b/sound/isa/opl3sa2.c index 58c972b2af0..645491a5302 100644 --- a/sound/isa/opl3sa2.c +++ b/sound/isa/opl3sa2.c @@ -617,21 +617,24 @@ static void snd_opl3sa2_free(struct snd_card *card) release_and_free_resource(chip->res_port); } -static struct snd_card *snd_opl3sa2_card_new(int dev) +static int snd_opl3sa2_card_new(int dev, struct snd_card **cardp) { struct snd_card *card; struct snd_opl3sa2 *chip; + int err; - card = snd_card_new(index[dev], id[dev], THIS_MODULE, sizeof(struct snd_opl3sa2)); - if (card == NULL) - return NULL; + err = snd_card_create(index[dev], id[dev], THIS_MODULE, + sizeof(struct snd_opl3sa2), &card); + if (err < 0) + return err; strcpy(card->driver, "OPL3SA2"); strcpy(card->shortname, "Yamaha OPL3-SA2"); chip = card->private_data; spin_lock_init(&chip->reg_lock); chip->irq = -1; card->private_free = snd_opl3sa2_free; - return card; + *cardp = card; + return 0; } static int __devinit snd_opl3sa2_probe(struct snd_card *card, int dev) @@ -723,9 +726,9 @@ static int __devinit snd_opl3sa2_pnp_detect(struct pnp_dev *pdev, if (dev >= SNDRV_CARDS) return -ENODEV; - card = snd_opl3sa2_card_new(dev); - if (! card) - return -ENOMEM; + err = snd_opl3sa2_card_new(dev, &card); + if (err < 0) + return err; if ((err = snd_opl3sa2_pnp(dev, card->private_data, pdev)) < 0) { snd_card_free(card); return err; @@ -789,9 +792,9 @@ static int __devinit snd_opl3sa2_pnp_cdetect(struct pnp_card_link *pcard, if (dev >= SNDRV_CARDS) return -ENODEV; - card = snd_opl3sa2_card_new(dev); - if (! card) - return -ENOMEM; + err = snd_opl3sa2_card_new(dev, &card); + if (err < 0) + return err; if ((err = snd_opl3sa2_pnp(dev, card->private_data, pdev)) < 0) { snd_card_free(card); return err; @@ -870,9 +873,9 @@ static int __devinit snd_opl3sa2_isa_probe(struct device *pdev, struct snd_card *card; int err; - card = snd_opl3sa2_card_new(dev); - if (! card) - return -ENOMEM; + err = snd_opl3sa2_card_new(dev, &card); + if (err < 0) + return err; snd_card_set_dev(card, pdev); if ((err = snd_opl3sa2_probe(card, dev)) < 0) { snd_card_free(card); diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index 440755cc001..02e30d7c6a9 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -1228,9 +1228,10 @@ static int __devinit snd_miro_probe(struct device *devptr, unsigned int n) struct snd_pcm *pcm; struct snd_rawmidi *rmidi; - if (!(card = snd_card_new(index, id, THIS_MODULE, - sizeof(struct snd_miro)))) - return -ENOMEM; + error = snd_card_create(index, id, THIS_MODULE, + sizeof(struct snd_miro), &card); + if (error < 0) + return error; card->private_free = snd_card_miro_free; miro = card->private_data; diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index 19706b0d849..5750f38bb79 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -833,9 +833,11 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) static struct snd_card *snd_opti9xx_card_new(void) { struct snd_card *card; + int err; - card = snd_card_new(index, id, THIS_MODULE, sizeof(struct snd_opti9xx)); - if (! card) + err = snd_card_create(index, id, THIS_MODULE, + sizeof(struct snd_opti9xx), &card); + if (err < 0) return NULL; card->private_free = snd_card_opti9xx_free; return card; diff --git a/sound/isa/sb/es968.c b/sound/isa/sb/es968.c index c8c8e214c84..cafc3a7316a 100644 --- a/sound/isa/sb/es968.c +++ b/sound/isa/sb/es968.c @@ -108,9 +108,10 @@ static int __devinit snd_card_es968_probe(int dev, struct snd_card *card; struct snd_card_es968 *acard; - if ((card = snd_card_new(index[dev], id[dev], THIS_MODULE, - sizeof(struct snd_card_es968))) == NULL) - return -ENOMEM; + error = snd_card_create(index[dev], id[dev], THIS_MODULE, + sizeof(struct snd_card_es968), &card); + if (error < 0) + return error; acard = card->private_data; if ((error = snd_card_es968_pnp(dev, acard, pcard, pid))) { snd_card_free(card); diff --git a/sound/isa/sb/sb16.c b/sound/isa/sb/sb16.c index 2c201f78ce5..adf4fdd2c4a 100644 --- a/sound/isa/sb/sb16.c +++ b/sound/isa/sb/sb16.c @@ -326,9 +326,12 @@ static void snd_sb16_free(struct snd_card *card) static struct snd_card *snd_sb16_card_new(int dev) { - struct snd_card *card = snd_card_new(index[dev], id[dev], THIS_MODULE, - sizeof(struct snd_card_sb16)); - if (card == NULL) + struct snd_card *card; + int err; + + err = snd_card_create(index[dev], id[dev], THIS_MODULE, + sizeof(struct snd_card_sb16), &card); + if (err < 0) return NULL; card->private_free = snd_sb16_free; return card; diff --git a/sound/isa/sb/sb8.c b/sound/isa/sb/sb8.c index ea06877be4b..3cd57ee5466 100644 --- a/sound/isa/sb/sb8.c +++ b/sound/isa/sb/sb8.c @@ -103,10 +103,10 @@ static int __devinit snd_sb8_probe(struct device *pdev, unsigned int dev) struct snd_opl3 *opl3; int err; - card = snd_card_new(index[dev], id[dev], THIS_MODULE, - sizeof(struct snd_sb8)); - if (card == NULL) - return -ENOMEM; + err = snd_card_create(index[dev], id[dev], THIS_MODULE, + sizeof(struct snd_sb8), &card); + if (err < 0) + return err; acard = card->private_data; card->private_free = snd_sb8_free; diff --git a/sound/isa/sc6000.c b/sound/isa/sc6000.c index ca35924dc3b..7a1470376c6 100644 --- a/sound/isa/sc6000.c +++ b/sound/isa/sc6000.c @@ -489,9 +489,9 @@ static int __devinit snd_sc6000_probe(struct device *devptr, unsigned int dev) char __iomem *vmss_port; - card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); - if (!card) - return -ENOMEM; + err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); + if (err < 0) + return err; if (xirq == SNDRV_AUTO_IRQ) { xirq = snd_legacy_find_free_irq(possible_irqs); diff --git a/sound/isa/sgalaxy.c b/sound/isa/sgalaxy.c index 2c7503bf127..6fe27b9d944 100644 --- a/sound/isa/sgalaxy.c +++ b/sound/isa/sgalaxy.c @@ -243,9 +243,9 @@ static int __devinit snd_sgalaxy_probe(struct device *devptr, unsigned int dev) struct snd_card *card; struct snd_wss *chip; - card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); - if (card == NULL) - return -ENOMEM; + err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); + if (err < 0) + return err; xirq = irq[dev]; if (xirq == SNDRV_AUTO_IRQ) { diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index 48a16d86583..4025fb558c5 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -1357,10 +1357,10 @@ static int __devinit snd_sscape_probe(struct device *pdev, unsigned int dev) struct soundscape *sscape; int ret; - card = snd_card_new(index[dev], id[dev], THIS_MODULE, - sizeof(struct soundscape)); - if (!card) - return -ENOMEM; + ret = snd_card_create(index[dev], id[dev], THIS_MODULE, + sizeof(struct soundscape), &card); + if (ret < 0) + return ret; sscape = get_card_soundscape(card); sscape->type = SSCAPE; @@ -1462,10 +1462,10 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard, * Create a new ALSA sound card entry, in anticipation * of detecting our hardware ... */ - card = snd_card_new(index[idx], id[idx], THIS_MODULE, - sizeof(struct soundscape)); - if (!card) - return -ENOMEM; + ret = snd_card_create(index[idx], id[idx], THIS_MODULE, + sizeof(struct soundscape), &card); + if (ret < 0) + return ret; sscape = get_card_soundscape(card); diff --git a/sound/isa/wavefront/wavefront.c b/sound/isa/wavefront/wavefront.c index 4c095bc7c72..82b8fb74690 100644 --- a/sound/isa/wavefront/wavefront.c +++ b/sound/isa/wavefront/wavefront.c @@ -342,10 +342,11 @@ static struct snd_card *snd_wavefront_card_new(int dev) { struct snd_card *card; snd_wavefront_card_t *acard; + int err; - card = snd_card_new (index[dev], id[dev], THIS_MODULE, - sizeof(snd_wavefront_card_t)); - if (card == NULL) + err = snd_card_create(index[dev], id[dev], THIS_MODULE, + sizeof(snd_wavefront_card_t), &card); + if (err < 0) return NULL; acard = card->private_data; -- cgit v1.2.3-70-g09d2 From e58de7baf7de11f01a675cbbf6ecc8a2758b9ca5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 28 Dec 2008 16:44:30 +0100 Subject: ALSA: Convert to snd_card_create() in sound/pci/* Convert from snd_card_new() to the new snd_card_create() function in sound/pci/*. Signed-off-by: Takashi Iwai --- sound/pci/ad1889.c | 6 +++--- sound/pci/ali5451/ali5451.c | 6 +++--- sound/pci/als300.c | 6 +++--- sound/pci/als4000.c | 9 +++++---- sound/pci/atiixp.c | 6 +++--- sound/pci/atiixp_modem.c | 6 +++--- sound/pci/au88x0/au88x0.c | 6 +++--- sound/pci/aw2/aw2-alsa.c | 6 +++--- sound/pci/azt3328.c | 6 +++--- sound/pci/bt87x.c | 6 +++--- sound/pci/ca0106/ca0106_main.c | 6 +++--- sound/pci/cmipci.c | 6 +++--- sound/pci/cs4281.c | 6 +++--- sound/pci/cs46xx/cs46xx.c | 6 +++--- sound/pci/cs5530.c | 6 +++--- sound/pci/cs5535audio/cs5535audio.c | 6 +++--- sound/pci/echoaudio/echoaudio.c | 6 +++--- sound/pci/emu10k1/emu10k1.c | 6 +++--- sound/pci/emu10k1/emu10k1x.c | 6 +++--- sound/pci/ens1370.c | 6 +++--- sound/pci/es1938.c | 6 +++--- sound/pci/es1968.c | 6 +++--- sound/pci/fm801.c | 6 +++--- sound/pci/hda/hda_intel.c | 6 +++--- sound/pci/ice1712/ice1712.c | 6 +++--- sound/pci/ice1712/ice1724.c | 6 +++--- sound/pci/intel8x0.c | 6 +++--- sound/pci/intel8x0m.c | 6 +++--- sound/pci/korg1212/korg1212.c | 6 +++--- sound/pci/maestro3.c | 6 +++--- sound/pci/mixart/mixart.c | 6 +++--- sound/pci/nm256/nm256.c | 6 +++--- sound/pci/oxygen/oxygen_lib.c | 8 ++++---- sound/pci/pcxhr/pcxhr.c | 6 +++--- sound/pci/riptide/riptide.c | 6 +++--- sound/pci/rme32.c | 7 ++++--- sound/pci/rme96.c | 7 ++++--- sound/pci/rme9652/hdsp.c | 6 ++++-- sound/pci/rme9652/hdspm.c | 8 ++++---- sound/pci/rme9652/rme9652.c | 8 ++++---- sound/pci/sis7019.c | 5 ++--- sound/pci/sonicvibes.c | 6 +++--- sound/pci/trident/trident.c | 6 +++--- sound/pci/via82xx.c | 6 +++--- sound/pci/via82xx_modem.c | 6 +++--- sound/pci/vx222/vx222.c | 6 +++--- sound/pci/ymfpci/ymfpci.c | 6 +++--- 47 files changed, 148 insertions(+), 144 deletions(-) diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c index a7f38e63303..d1f242bd0ac 100644 --- a/sound/pci/ad1889.c +++ b/sound/pci/ad1889.c @@ -995,10 +995,10 @@ snd_ad1889_probe(struct pci_dev *pci, } /* (2) */ - card = snd_card_new(index[devno], id[devno], THIS_MODULE, 0); + err = snd_card_create(index[devno], id[devno], THIS_MODULE, 0, &card); /* XXX REVISIT: we can probably allocate chip in this call */ - if (card == NULL) - return -ENOMEM; + if (err < 0) + return err; strcpy(card->driver, "AD1889"); strcpy(card->shortname, "Analog Devices AD1889"); diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index 1a0fd65ec28..b36c551da56 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -2307,9 +2307,9 @@ static int __devinit snd_ali_probe(struct pci_dev *pci, snd_ali_printk("probe ...\n"); - card = snd_card_new(index, id, THIS_MODULE, 0); - if (!card) - return -ENOMEM; + err = snd_card_create(index, id, THIS_MODULE, 0, &card); + if (err < 0) + return err; err = snd_ali_create(card, pci, pcm_channels, spdif, &codec); if (err < 0) diff --git a/sound/pci/als300.c b/sound/pci/als300.c index 8df6824b51c..f557c155db4 100644 --- a/sound/pci/als300.c +++ b/sound/pci/als300.c @@ -812,10 +812,10 @@ static int __devinit snd_als300_probe(struct pci_dev *pci, return -ENOENT; } - card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); + err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); - if (card == NULL) - return -ENOMEM; + if (err < 0) + return err; chip_type = pci_id->driver_data; diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c index ba570053d4d..542a0c65a92 100644 --- a/sound/pci/als4000.c +++ b/sound/pci/als4000.c @@ -889,12 +889,13 @@ static int __devinit snd_card_als4000_probe(struct pci_dev *pci, pci_write_config_word(pci, PCI_COMMAND, word | PCI_COMMAND_IO); pci_set_master(pci); - card = snd_card_new(index[dev], id[dev], THIS_MODULE, - sizeof(*acard) /* private_data: acard */); - if (card == NULL) { + err = snd_card_create(index[dev], id[dev], THIS_MODULE, + sizeof(*acard) /* private_data: acard */, + &card); + if (err < 0) { pci_release_regions(pci); pci_disable_device(pci); - return -ENOMEM; + return err; } acard = card->private_data; diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index 226fe8237d3..9ce8548c03e 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -1645,9 +1645,9 @@ static int __devinit snd_atiixp_probe(struct pci_dev *pci, struct atiixp *chip; int err; - card = snd_card_new(index, id, THIS_MODULE, 0); - if (card == NULL) - return -ENOMEM; + err = snd_card_create(index, id, THIS_MODULE, 0, &card); + if (err < 0) + return err; strcpy(card->driver, spdif_aclink ? "ATIIXP" : "ATIIXP-SPDMA"); strcpy(card->shortname, "ATI IXP"); diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c index 0e6e5cc1c50..c3136cccc55 100644 --- a/sound/pci/atiixp_modem.c +++ b/sound/pci/atiixp_modem.c @@ -1288,9 +1288,9 @@ static int __devinit snd_atiixp_probe(struct pci_dev *pci, struct atiixp_modem *chip; int err; - card = snd_card_new(index, id, THIS_MODULE, 0); - if (card == NULL) - return -ENOMEM; + err = snd_card_create(index, id, THIS_MODULE, 0, &card); + if (err < 0) + return err; strcpy(card->driver, "ATIIXP-MODEM"); strcpy(card->shortname, "ATI IXP Modem"); diff --git a/sound/pci/au88x0/au88x0.c b/sound/pci/au88x0/au88x0.c index a36d4d1fd41..9ec122383ee 100644 --- a/sound/pci/au88x0/au88x0.c +++ b/sound/pci/au88x0/au88x0.c @@ -250,9 +250,9 @@ snd_vortex_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) return -ENOENT; } // (2) - card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); - if (card == NULL) - return -ENOMEM; + err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); + if (err < 0) + return err; // (3) if ((err = snd_vortex_create(card, pci, &chip)) < 0) { diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c index 3f00ddf450f..eefcbf648ee 100644 --- a/sound/pci/aw2/aw2-alsa.c +++ b/sound/pci/aw2/aw2-alsa.c @@ -368,9 +368,9 @@ static int __devinit snd_aw2_probe(struct pci_dev *pci, } /* (2) Create card instance */ - card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); - if (card == NULL) - return -ENOMEM; + err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); + if (err < 0) + return err; /* (3) Create main component */ err = snd_aw2_create(card, pci, &chip); diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 333007c523a..1df96e76c48 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -2216,9 +2216,9 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) return -ENOENT; } - card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); - if (card == NULL) - return -ENOMEM; + err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); + if (err < 0) + return err; strcpy(card->driver, "AZF3328"); strcpy(card->shortname, "Aztech AZF3328 (PCI168)"); diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c index 1aa1c040254..a299340519d 100644 --- a/sound/pci/bt87x.c +++ b/sound/pci/bt87x.c @@ -888,9 +888,9 @@ static int __devinit snd_bt87x_probe(struct pci_dev *pci, return -ENOENT; } - card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); - if (!card) - return -ENOMEM; + err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); + if (err < 0) + return err; err = snd_bt87x_create(card, pci, &chip); if (err < 0) diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index 0e62205d408..b116456e770 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -1707,9 +1707,9 @@ static int __devinit snd_ca0106_probe(struct pci_dev *pci, return -ENOENT; } - card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); - if (card == NULL) - return -ENOMEM; + err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); + if (err < 0) + return err; err = snd_ca0106_create(dev, card, pci, &chip); if (err < 0) diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index 1a74ca62c31..c7899c32aba 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -3272,9 +3272,9 @@ static int __devinit snd_cmipci_probe(struct pci_dev *pci, return -ENOENT; } - card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); - if (card == NULL) - return -ENOMEM; + err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); + if (err < 0) + return err; switch (pci->device) { case PCI_DEVICE_ID_CMEDIA_CM8738: diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c index 192e7842e18..b9b07f46463 100644 --- a/sound/pci/cs4281.c +++ b/sound/pci/cs4281.c @@ -1925,9 +1925,9 @@ static int __devinit snd_cs4281_probe(struct pci_dev *pci, return -ENOENT; } - card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); - if (card == NULL) - return -ENOMEM; + err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); + if (err < 0) + return err; if ((err = snd_cs4281_create(card, pci, &chip, dual_codec[dev])) < 0) { snd_card_free(card); diff --git a/sound/pci/cs46xx/cs46xx.c b/sound/pci/cs46xx/cs46xx.c index e876b3263e4..c9b3e3d48cb 100644 --- a/sound/pci/cs46xx/cs46xx.c +++ b/sound/pci/cs46xx/cs46xx.c @@ -88,9 +88,9 @@ static int __devinit snd_card_cs46xx_probe(struct pci_dev *pci, return -ENOENT; } - card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); - if (card == NULL) - return -ENOMEM; + err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); + if (err < 0) + return err; if ((err = snd_cs46xx_create(card, pci, external_amp[dev], thinkpad[dev], &chip)) < 0) { diff --git a/sound/pci/cs5530.c b/sound/pci/cs5530.c index 6dea5b5cc77..dc464321d0f 100644 --- a/sound/pci/cs5530.c +++ b/sound/pci/cs5530.c @@ -258,10 +258,10 @@ static int __devinit snd_cs5530_probe(struct pci_dev *pci, return -ENOENT; } - card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); + err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); - if (card == NULL) - return -ENOMEM; + if (err < 0) + return err; err = snd_cs5530_create(card, pci, &chip); if (err < 0) { diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c index 826e6dec2e9..ac1d72e0a1e 100644 --- a/sound/pci/cs5535audio/cs5535audio.c +++ b/sound/pci/cs5535audio/cs5535audio.c @@ -353,9 +353,9 @@ static int __devinit snd_cs5535audio_probe(struct pci_dev *pci, return -ENOENT; } - card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); - if (card == NULL) - return -ENOMEM; + err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); + if (err < 0) + return err; if ((err = snd_cs5535audio_create(card, pci, &cs5535au)) < 0) goto probefail_out; diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 8dbc5c4ba42..9d015a76eb6 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -1997,9 +1997,9 @@ static int __devinit snd_echo_probe(struct pci_dev *pci, DE_INIT(("Echoaudio driver starting...\n")); i = 0; - card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); - if (card == NULL) - return -ENOMEM; + err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); + if (err < 0) + return err; snd_card_set_dev(card, &pci->dev); diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c index 8354c1a8331..c7f3b994101 100644 --- a/sound/pci/emu10k1/emu10k1.c +++ b/sound/pci/emu10k1/emu10k1.c @@ -114,9 +114,9 @@ static int __devinit snd_card_emu10k1_probe(struct pci_dev *pci, return -ENOENT; } - card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); - if (card == NULL) - return -ENOMEM; + err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); + if (err < 0) + return err; if (max_buffer_size[dev] < 32) max_buffer_size[dev] = 32; else if (max_buffer_size[dev] > 1024) diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index 5ff4dbb62da..31542adc6b7 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -1544,9 +1544,9 @@ static int __devinit snd_emu10k1x_probe(struct pci_dev *pci, return -ENOENT; } - card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); - if (card == NULL) - return -ENOMEM; + err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); + if (err < 0) + return err; if ((err = snd_emu10k1x_create(card, pci, &chip)) < 0) { snd_card_free(card); diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index 9bf95367c88..e00614cbcef 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -2409,9 +2409,9 @@ static int __devinit snd_audiopci_probe(struct pci_dev *pci, return -ENOENT; } - card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); - if (card == NULL) - return -ENOMEM; + err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); + if (err < 0) + return err; if ((err = snd_ensoniq_create(card, pci, &ensoniq)) < 0) { snd_card_free(card); diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c index 4cd9a1faaec..34a78afc26d 100644 --- a/sound/pci/es1938.c +++ b/sound/pci/es1938.c @@ -1799,9 +1799,9 @@ static int __devinit snd_es1938_probe(struct pci_dev *pci, return -ENOENT; } - card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); - if (card == NULL) - return -ENOMEM; + err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); + if (err < 0) + return err; for (idx = 0; idx < 5; idx++) { if (pci_resource_start(pci, idx) == 0 || !(pci_resource_flags(pci, idx) & IORESOURCE_IO)) { diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index e9c3794bbcb..dc97e811614 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -2645,9 +2645,9 @@ static int __devinit snd_es1968_probe(struct pci_dev *pci, return -ENOENT; } - card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); - if (!card) - return -ENOMEM; + err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); + if (err < 0) + return err; if (total_bufsize[dev] < 128) total_bufsize[dev] = 128; diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index c129f9e2072..60cdb9e0b68 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -1468,9 +1468,9 @@ static int __devinit snd_card_fm801_probe(struct pci_dev *pci, return -ENOENT; } - card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); - if (card == NULL) - return -ENOMEM; + err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); + if (err < 0) + return err; if ((err = snd_fm801_create(card, pci, tea575x_tuner[dev], &chip)) < 0) { snd_card_free(card); return err; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index f04de115ee1..ad5df2ae6f7 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2335,10 +2335,10 @@ static int __devinit azx_probe(struct pci_dev *pci, return -ENOENT; } - card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); - if (!card) { + err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); + if (err < 0) { snd_printk(KERN_ERR SFX "Error creating card!\n"); - return -ENOMEM; + return err; } err = azx_create(card, pci, dev, pci_id->driver_data, &chip); diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 58d7cda03de..bab1c700f49 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -2648,9 +2648,9 @@ static int __devinit snd_ice1712_probe(struct pci_dev *pci, return -ENOENT; } - card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); - if (card == NULL) - return -ENOMEM; + err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); + if (err < 0) + return err; strcpy(card->driver, "ICE1712"); strcpy(card->shortname, "ICEnsemble ICE1712"); diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index bb8d8c766b9..7ff36d3f0f4 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -2456,9 +2456,9 @@ static int __devinit snd_vt1724_probe(struct pci_dev *pci, return -ENOENT; } - card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); - if (card == NULL) - return -ENOMEM; + err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); + if (err < 0) + return err; strcpy(card->driver, "ICE1724"); strcpy(card->shortname, "ICEnsemble ICE1724"); diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 19d3391e229..671ff65db02 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -3058,9 +3058,9 @@ static int __devinit snd_intel8x0_probe(struct pci_dev *pci, int err; struct shortname_table *name; - card = snd_card_new(index, id, THIS_MODULE, 0); - if (card == NULL) - return -ENOMEM; + err = snd_card_create(index, id, THIS_MODULE, 0, &card); + if (err < 0) + return err; if (spdif_aclink < 0) spdif_aclink = check_default_spdif_aclink(pci); diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index 93449e46456..33a843c1931 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -1269,9 +1269,9 @@ static int __devinit snd_intel8x0m_probe(struct pci_dev *pci, int err; struct shortname_table *name; - card = snd_card_new(index, id, THIS_MODULE, 0); - if (card == NULL) - return -ENOMEM; + err = snd_card_create(index, id, THIS_MODULE, 0, &card); + if (err < 0) + return err; strcpy(card->driver, "ICH-MODEM"); strcpy(card->shortname, "Intel ICH"); diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c index 5f8006b4275..8b79969034b 100644 --- a/sound/pci/korg1212/korg1212.c +++ b/sound/pci/korg1212/korg1212.c @@ -2443,9 +2443,9 @@ snd_korg1212_probe(struct pci_dev *pci, dev++; return -ENOENT; } - card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); - if (card == NULL) - return -ENOMEM; + err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); + if (err < 0) + return err; if ((err = snd_korg1212_create(card, pci, &korg1212)) < 0) { snd_card_free(card); diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index 59bbaf8f3e5..70141548f25 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -2691,9 +2691,9 @@ snd_m3_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) return -ENOENT; } - card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); - if (card == NULL) - return -ENOMEM; + err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); + if (err < 0) + return err; switch (pci->device) { case PCI_DEVICE_ID_ESS_ALLEGRO: diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index f23a73577c2..bfc19e36c4b 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -1365,12 +1365,12 @@ static int __devinit snd_mixart_probe(struct pci_dev *pci, else idx = index[dev] + i; snprintf(tmpid, sizeof(tmpid), "%s-%d", id[dev] ? id[dev] : "MIXART", i); - card = snd_card_new(idx, tmpid, THIS_MODULE, 0); + err = snd_card_create(idx, tmpid, THIS_MODULE, 0, &card); - if (! card) { + if (err < 0) { snd_printk(KERN_ERR "cannot allocate the card %d\n", i); snd_mixart_free(mgr); - return -ENOMEM; + return err; } strcpy(card->driver, CARD_NAME); diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c index 50c9f8a0508..522a040855d 100644 --- a/sound/pci/nm256/nm256.c +++ b/sound/pci/nm256/nm256.c @@ -1668,9 +1668,9 @@ static int __devinit snd_nm256_probe(struct pci_dev *pci, } } - card = snd_card_new(index, id, THIS_MODULE, 0); - if (card == NULL) - return -ENOMEM; + err = snd_card_create(index, id, THIS_MODULE, 0, &card); + if (err < 0) + return err; switch (pci->device) { case PCI_DEVICE_ID_NEOMAGIC_NM256AV_AUDIO: diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index 84f481d41ef..9c81e0b0511 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -459,10 +459,10 @@ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id, struct oxygen *chip; int err; - card = snd_card_new(index, id, model->owner, - sizeof *chip + model->model_data_size); - if (!card) - return -ENOMEM; + err = snd_card_create(index, id, model->owner, + sizeof(*chip) + model->model_data_size, &card); + if (err < 0) + return err; chip = card->private_data; chip->card = card; diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index 27cf2c28d11..7f95459c8b1 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -1510,12 +1510,12 @@ static int __devinit pcxhr_probe(struct pci_dev *pci, snprintf(tmpid, sizeof(tmpid), "%s-%d", id[dev] ? id[dev] : card_name, i); - card = snd_card_new(idx, tmpid, THIS_MODULE, 0); + err = snd_card_create(idx, tmpid, THIS_MODULE, 0, &card); - if (! card) { + if (err < 0) { snd_printk(KERN_ERR "cannot allocate the card %d\n", i); pcxhr_free(mgr); - return -ENOMEM; + return err; } strcpy(card->driver, DRIVER_NAME); diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index 3caacfb9d8e..6f1034417a0 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -2102,9 +2102,9 @@ snd_card_riptide_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) return -ENOENT; } - card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); - if (card == NULL) - return -ENOMEM; + err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); + if (err < 0) + return err; if ((err = snd_riptide_create(card, pci, &chip)) < 0) { snd_card_free(card); return err; diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c index e7ef3a1a25a..d7b966e7c4c 100644 --- a/sound/pci/rme32.c +++ b/sound/pci/rme32.c @@ -1941,9 +1941,10 @@ snd_rme32_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) return -ENOENT; } - if ((card = snd_card_new(index[dev], id[dev], THIS_MODULE, - sizeof(struct rme32))) == NULL) - return -ENOMEM; + err = snd_card_create(index[dev], id[dev], THIS_MODULE, + sizeof(struct rme32), &card); + if (err < 0) + return err; card->private_free = snd_rme32_card_free; rme32 = (struct rme32 *) card->private_data; rme32->card = card; diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c index 3fdd488d097..55fb1c131f5 100644 --- a/sound/pci/rme96.c +++ b/sound/pci/rme96.c @@ -2348,9 +2348,10 @@ snd_rme96_probe(struct pci_dev *pci, dev++; return -ENOENT; } - if ((card = snd_card_new(index[dev], id[dev], THIS_MODULE, - sizeof(struct rme96))) == NULL) - return -ENOMEM; + err = snd_card_create(index[dev], id[dev], THIS_MODULE, + sizeof(struct rme96), &card); + if (err < 0) + return err; card->private_free = snd_rme96_card_free; rme96 = (struct rme96 *)card->private_data; rme96->card = card; diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 44d0c15e2b7..05b3f795a16 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -5158,8 +5158,10 @@ static int __devinit snd_hdsp_probe(struct pci_dev *pci, return -ENOENT; } - if (!(card = snd_card_new(index[dev], id[dev], THIS_MODULE, sizeof(struct hdsp)))) - return -ENOMEM; + err = snd_card_create(index[dev], id[dev], THIS_MODULE, + sizeof(struct hdsp), &card); + if (err < 0) + return err; hdsp = (struct hdsp *) card->private_data; card->private_free = snd_hdsp_card_free; diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 71231cf1b2b..d4b4e0d0fee 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -4503,10 +4503,10 @@ static int __devinit snd_hdspm_probe(struct pci_dev *pci, return -ENOENT; } - card = snd_card_new(index[dev], id[dev], - THIS_MODULE, sizeof(struct hdspm)); - if (!card) - return -ENOMEM; + err = snd_card_create(index[dev], id[dev], + THIS_MODULE, sizeof(struct hdspm), &card); + if (err < 0) + return err; hdspm = card->private_data; card->private_free = snd_hdspm_card_free; diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c index 2570907134d..bc539abb210 100644 --- a/sound/pci/rme9652/rme9652.c +++ b/sound/pci/rme9652/rme9652.c @@ -2594,11 +2594,11 @@ static int __devinit snd_rme9652_probe(struct pci_dev *pci, return -ENOENT; } - card = snd_card_new(index[dev], id[dev], THIS_MODULE, - sizeof(struct snd_rme9652)); + err = snd_card_create(index[dev], id[dev], THIS_MODULE, + sizeof(struct snd_rme9652), &card); - if (!card) - return -ENOMEM; + if (err < 0) + return err; rme9652 = (struct snd_rme9652 *) card->private_data; card->private_free = snd_rme9652_card_free; diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c index df2007e3be7..baf6d8e3dab 100644 --- a/sound/pci/sis7019.c +++ b/sound/pci/sis7019.c @@ -1387,9 +1387,8 @@ static int __devinit snd_sis7019_probe(struct pci_dev *pci, if (!enable) goto error_out; - rc = -ENOMEM; - card = snd_card_new(index, id, THIS_MODULE, sizeof(*sis)); - if (!card) + rc = snd_card_create(index, id, THIS_MODULE, sizeof(*sis), &card); + if (rc < 0) goto error_out; strcpy(card->driver, "SiS7019"); diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c index cd408b86c83..c5601b0ad7c 100644 --- a/sound/pci/sonicvibes.c +++ b/sound/pci/sonicvibes.c @@ -1423,9 +1423,9 @@ static int __devinit snd_sonic_probe(struct pci_dev *pci, return -ENOENT; } - card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); - if (card == NULL) - return -ENOMEM; + err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); + if (err < 0) + return err; for (idx = 0; idx < 5; idx++) { if (pci_resource_start(pci, idx) == 0 || !(pci_resource_flags(pci, idx) & IORESOURCE_IO)) { diff --git a/sound/pci/trident/trident.c b/sound/pci/trident/trident.c index d94b16ffb38..21cef97d478 100644 --- a/sound/pci/trident/trident.c +++ b/sound/pci/trident/trident.c @@ -89,9 +89,9 @@ static int __devinit snd_trident_probe(struct pci_dev *pci, return -ENOENT; } - card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); - if (card == NULL) - return -ENOMEM; + err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); + if (err < 0) + return err; if ((err = snd_trident_create(card, pci, pcm_channels[dev], diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 1aafe956ee2..d8705547dae 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -2433,9 +2433,9 @@ static int __devinit snd_via82xx_probe(struct pci_dev *pci, unsigned int i; int err; - card = snd_card_new(index, id, THIS_MODULE, 0); - if (card == NULL) - return -ENOMEM; + err = snd_card_create(index, id, THIS_MODULE, 0, &card); + if (err < 0) + return err; card_type = pci_id->driver_data; switch (card_type) { diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c index 5bd79d2a5a1..c086b762c15 100644 --- a/sound/pci/via82xx_modem.c +++ b/sound/pci/via82xx_modem.c @@ -1167,9 +1167,9 @@ static int __devinit snd_via82xx_probe(struct pci_dev *pci, unsigned int i; int err; - card = snd_card_new(index, id, THIS_MODULE, 0); - if (card == NULL) - return -ENOMEM; + err = snd_card_create(index, id, THIS_MODULE, 0, &card); + if (err < 0) + return err; card_type = pci_id->driver_data; switch (card_type) { diff --git a/sound/pci/vx222/vx222.c b/sound/pci/vx222/vx222.c index acc352f4a44..fc9136c3e0d 100644 --- a/sound/pci/vx222/vx222.c +++ b/sound/pci/vx222/vx222.c @@ -204,9 +204,9 @@ static int __devinit snd_vx222_probe(struct pci_dev *pci, return -ENOENT; } - card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); - if (card == NULL) - return -ENOMEM; + err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); + if (err < 0) + return err; switch ((int)pci_id->driver_data) { case VX_PCI_VX222_OLD: diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c index 2631a554845..4af66661f9b 100644 --- a/sound/pci/ymfpci/ymfpci.c +++ b/sound/pci/ymfpci/ymfpci.c @@ -187,9 +187,9 @@ static int __devinit snd_card_ymfpci_probe(struct pci_dev *pci, return -ENOENT; } - card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); - if (card == NULL) - return -ENOMEM; + err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); + if (err < 0) + return err; switch (pci_id->device) { case 0x0004: str = "YMF724"; model = "DS-1"; break; -- cgit v1.2.3-70-g09d2 From bd7dd77c2a05c530684eea2e3af16449ae9c5d52 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 28 Dec 2008 16:45:02 +0100 Subject: ALSA: Convert to snd_card_create() in other sound/* Convert from snd_card_new() to the new snd_card_create() function in other sound subdirectories. Signed-off-by: Takashi Iwai --- sound/aoa/core/alsa.c | 7 ++++--- sound/arm/aaci.c | 7 ++++--- sound/arm/pxa2xx-ac97.c | 7 +++---- sound/arm/sa11xx-uda1341.c | 7 ++++--- sound/drivers/dummy.c | 8 ++++---- sound/drivers/ml403-ac97cr.c | 6 +++--- sound/drivers/mpu401/mpu401.c | 6 +++--- sound/drivers/mtpav.c | 6 +++--- sound/drivers/mts64.c | 6 +++--- sound/drivers/pcsp/pcsp.c | 6 +++--- sound/drivers/portman2x4.c | 6 +++--- sound/drivers/serial-u16550.c | 6 +++--- sound/drivers/virmidi.c | 8 ++++---- sound/mips/au1x00.c | 7 ++++--- sound/mips/hal2.c | 6 +++--- sound/mips/sgio2audio.c | 6 +++--- sound/parisc/harmony.c | 6 +++--- sound/pcmcia/pdaudiocf/pdaudiocf.c | 8 ++++---- sound/pcmcia/vx/vxpocket.c | 8 ++++---- sound/ppc/powermac.c | 6 +++--- sound/ppc/snd_ps3.c | 6 ++---- sound/sh/aica.c | 8 ++++---- sound/soc/soc-core.c | 8 ++++---- sound/sparc/amd7930.c | 7 ++++--- sound/sparc/cs4231.c | 9 +++++---- sound/sparc/dbri.c | 8 ++++---- sound/spi/at73c213.c | 7 +++---- sound/usb/caiaq/caiaq-device.c | 7 ++++--- sound/usb/usbaudio.c | 6 +++--- sound/usb/usx2y/us122l.c | 8 +++++--- sound/usb/usx2y/usbusx2y.c | 7 +++++-- 31 files changed, 111 insertions(+), 103 deletions(-) diff --git a/sound/aoa/core/alsa.c b/sound/aoa/core/alsa.c index 61785046358..0fa3855b479 100644 --- a/sound/aoa/core/alsa.c +++ b/sound/aoa/core/alsa.c @@ -23,9 +23,10 @@ int aoa_alsa_init(char *name, struct module *mod, struct device *dev) /* cannot be EEXIST due to usage in aoa_fabric_register */ return -EBUSY; - alsa_card = snd_card_new(index, name, mod, sizeof(struct aoa_card)); - if (!alsa_card) - return -ENOMEM; + err = snd_card_create(index, name, mod, sizeof(struct aoa_card), + &alsa_card); + if (err < 0) + return err; aoa_card = alsa_card->private_data; aoa_card->alsa_card = alsa_card; alsa_card->dev = dev; diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index 89096e811a4..7d39aac9ec1 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -995,10 +995,11 @@ static struct aaci * __devinit aaci_init_card(struct amba_device *dev) { struct aaci *aaci; struct snd_card *card; + int err; - card = snd_card_new(SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1, - THIS_MODULE, sizeof(struct aaci)); - if (card == NULL) + err = snd_card_create(SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1, + THIS_MODULE, sizeof(struct aaci), &card); + if (err < 0) return NULL; card->private_free = aaci_free_card; diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index 85cf591d4e1..7ed100c80a5 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -173,10 +173,9 @@ static int __devinit pxa2xx_ac97_probe(struct platform_device *dev) struct snd_ac97_template ac97_template; int ret; - ret = -ENOMEM; - card = snd_card_new(SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1, - THIS_MODULE, 0); - if (!card) + ret = snd_card_create(SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1, + THIS_MODULE, 0, &card); + if (ret < 0) goto err; card->dev = &dev->dev; diff --git a/sound/arm/sa11xx-uda1341.c b/sound/arm/sa11xx-uda1341.c index 1dcd51d81d1..51d708c31e6 100644 --- a/sound/arm/sa11xx-uda1341.c +++ b/sound/arm/sa11xx-uda1341.c @@ -887,9 +887,10 @@ static int __devinit sa11xx_uda1341_probe(struct platform_device *devptr) struct sa11xx_uda1341 *chip; /* register the soundcard */ - card = snd_card_new(-1, id, THIS_MODULE, sizeof(struct sa11xx_uda1341)); - if (card == NULL) - return -ENOMEM; + err = snd_card_create(-1, id, THIS_MODULE, + sizeof(struct sa11xx_uda1341), &card); + if (err < 0) + return err; chip = card->private_data; spin_lock_init(&chip->s[0].dma_lock); diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index 73be7e14a60..54239d2e099 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -588,10 +588,10 @@ static int __devinit snd_dummy_probe(struct platform_device *devptr) int idx, err; int dev = devptr->id; - card = snd_card_new(index[dev], id[dev], THIS_MODULE, - sizeof(struct snd_dummy)); - if (card == NULL) - return -ENOMEM; + err = snd_card_create(index[dev], id[dev], THIS_MODULE, + sizeof(struct snd_dummy), &card); + if (err < 0) + return err; dummy = card->private_data; dummy->card = card; for (idx = 0; idx < MAX_PCM_DEVICES && idx < pcm_devs[dev]; idx++) { diff --git a/sound/drivers/ml403-ac97cr.c b/sound/drivers/ml403-ac97cr.c index 7783843ca9a..1950ffce2b5 100644 --- a/sound/drivers/ml403-ac97cr.c +++ b/sound/drivers/ml403-ac97cr.c @@ -1279,9 +1279,9 @@ static int __devinit snd_ml403_ac97cr_probe(struct platform_device *pfdev) if (!enable[dev]) return -ENOENT; - card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); - if (card == NULL) - return -ENOMEM; + err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); + if (err < 0) + return err; err = snd_ml403_ac97cr_create(card, pfdev, &ml403_ac97cr); if (err < 0) { PDEBUG(INIT_FAILURE, "probe(): create failed!\n"); diff --git a/sound/drivers/mpu401/mpu401.c b/sound/drivers/mpu401/mpu401.c index 5b996f3faba..149d05a8202 100644 --- a/sound/drivers/mpu401/mpu401.c +++ b/sound/drivers/mpu401/mpu401.c @@ -73,9 +73,9 @@ static int snd_mpu401_create(int dev, struct snd_card **rcard) snd_printk(KERN_ERR "the uart_enter option is obsolete; remove it\n"); *rcard = NULL; - card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); - if (card == NULL) - return -ENOMEM; + err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); + if (err < 0) + return err; strcpy(card->driver, "MPU-401 UART"); strcpy(card->shortname, card->driver); sprintf(card->longname, "%s at %#lx, ", card->shortname, port[dev]); diff --git a/sound/drivers/mtpav.c b/sound/drivers/mtpav.c index 5b89c0883d6..c3e9833dcfd 100644 --- a/sound/drivers/mtpav.c +++ b/sound/drivers/mtpav.c @@ -696,9 +696,9 @@ static int __devinit snd_mtpav_probe(struct platform_device *dev) int err; struct mtpav *mtp_card; - card = snd_card_new(index, id, THIS_MODULE, sizeof(*mtp_card)); - if (! card) - return -ENOMEM; + err = snd_card_create(index, id, THIS_MODULE, sizeof(*mtp_card), &card); + if (err < 0) + return err; mtp_card = card->private_data; spin_lock_init(&mtp_card->spinlock); diff --git a/sound/drivers/mts64.c b/sound/drivers/mts64.c index 87ba1ddc011..33d9db782e0 100644 --- a/sound/drivers/mts64.c +++ b/sound/drivers/mts64.c @@ -957,10 +957,10 @@ static int __devinit snd_mts64_probe(struct platform_device *pdev) if ((err = snd_mts64_probe_port(p)) < 0) return err; - card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); - if (card == NULL) { + err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); + if (err < 0) { snd_printd("Cannot create card\n"); - return -ENOMEM; + return err; } strcpy(card->driver, DRIVER_NAME); strcpy(card->shortname, "ESI " CARD_NAME); diff --git a/sound/drivers/pcsp/pcsp.c b/sound/drivers/pcsp/pcsp.c index a4049eb94d3..aa2ae07a76d 100644 --- a/sound/drivers/pcsp/pcsp.c +++ b/sound/drivers/pcsp/pcsp.c @@ -98,9 +98,9 @@ static int __devinit snd_card_pcsp_probe(int devnum, struct device *dev) hrtimer_init(&pcsp_chip.timer, CLOCK_MONOTONIC, HRTIMER_MODE_REL); pcsp_chip.timer.function = pcsp_do_timer; - card = snd_card_new(index, id, THIS_MODULE, 0); - if (!card) - return -ENOMEM; + err = snd_card_create(index, id, THIS_MODULE, 0, &card); + if (err < 0) + return err; err = snd_pcsp_create(card); if (err < 0) { diff --git a/sound/drivers/portman2x4.c b/sound/drivers/portman2x4.c index b1c047ec19a..60158e2e0ea 100644 --- a/sound/drivers/portman2x4.c +++ b/sound/drivers/portman2x4.c @@ -746,10 +746,10 @@ static int __devinit snd_portman_probe(struct platform_device *pdev) if ((err = snd_portman_probe_port(p)) < 0) return err; - card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); - if (card == NULL) { + err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); + if (err < 0) { snd_printd("Cannot create card\n"); - return -ENOMEM; + return err; } strcpy(card->driver, DRIVER_NAME); strcpy(card->shortname, CARD_NAME); diff --git a/sound/drivers/serial-u16550.c b/sound/drivers/serial-u16550.c index d8aab9da97c..891d081e482 100644 --- a/sound/drivers/serial-u16550.c +++ b/sound/drivers/serial-u16550.c @@ -936,9 +936,9 @@ static int __devinit snd_serial_probe(struct platform_device *devptr) return -ENODEV; } - card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); - if (card == NULL) - return -ENOMEM; + err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); + if (err < 0) + return err; strcpy(card->driver, "Serial"); strcpy(card->shortname, "Serial MIDI (UART16550A)"); diff --git a/sound/drivers/virmidi.c b/sound/drivers/virmidi.c index f79e3614079..6f48711818f 100644 --- a/sound/drivers/virmidi.c +++ b/sound/drivers/virmidi.c @@ -90,10 +90,10 @@ static int __devinit snd_virmidi_probe(struct platform_device *devptr) int idx, err; int dev = devptr->id; - card = snd_card_new(index[dev], id[dev], THIS_MODULE, - sizeof(struct snd_card_virmidi)); - if (card == NULL) - return -ENOMEM; + err = snd_card_create(index[dev], id[dev], THIS_MODULE, + sizeof(struct snd_card_virmidi), &card); + if (err < 0) + return err; vmidi = (struct snd_card_virmidi *)card->private_data; vmidi->card = card; diff --git a/sound/mips/au1x00.c b/sound/mips/au1x00.c index 1881cec11e7..99e1391b2eb 100644 --- a/sound/mips/au1x00.c +++ b/sound/mips/au1x00.c @@ -636,9 +636,10 @@ au1000_init(void) struct snd_card *card; struct snd_au1000 *au1000; - card = snd_card_new(-1, "AC97", THIS_MODULE, sizeof(struct snd_au1000)); - if (card == NULL) - return -ENOMEM; + err = snd_card_create(-1, "AC97", THIS_MODULE, + sizeof(struct snd_au1000), &card); + if (err < 0) + return err; card->private_free = snd_au1000_free; au1000 = card->private_data; diff --git a/sound/mips/hal2.c b/sound/mips/hal2.c index db495be0186..c52691c2fc4 100644 --- a/sound/mips/hal2.c +++ b/sound/mips/hal2.c @@ -878,9 +878,9 @@ static int __devinit hal2_probe(struct platform_device *pdev) struct snd_hal2 *chip; int err; - card = snd_card_new(index, id, THIS_MODULE, 0); - if (card == NULL) - return -ENOMEM; + err = snd_card_create(index, id, THIS_MODULE, 0, &card); + if (err < 0) + return err; err = hal2_create(card, &chip); if (err < 0) { diff --git a/sound/mips/sgio2audio.c b/sound/mips/sgio2audio.c index 4c63504348d..66f3b48ceaf 100644 --- a/sound/mips/sgio2audio.c +++ b/sound/mips/sgio2audio.c @@ -936,9 +936,9 @@ static int __devinit snd_sgio2audio_probe(struct platform_device *pdev) struct snd_sgio2audio *chip; int err; - card = snd_card_new(index, id, THIS_MODULE, 0); - if (card == NULL) - return -ENOMEM; + err = snd_card_create(index, id, THIS_MODULE, 0, &card); + if (err < 0) + return err; err = snd_sgio2audio_create(card, &chip); if (err < 0) { diff --git a/sound/parisc/harmony.c b/sound/parisc/harmony.c index 41f870f8a11..6055fd6d3b3 100644 --- a/sound/parisc/harmony.c +++ b/sound/parisc/harmony.c @@ -975,9 +975,9 @@ snd_harmony_probe(struct parisc_device *padev) struct snd_card *card; struct snd_harmony *h; - card = snd_card_new(index, id, THIS_MODULE, 0); - if (card == NULL) - return -ENOMEM; + err = snd_card_create(index, id, THIS_MODULE, 0, &card); + if (err < 0) + return err; err = snd_harmony_create(card, padev, &h); if (err < 0) diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.c b/sound/pcmcia/pdaudiocf/pdaudiocf.c index 819aaaac432..183f6615c68 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf.c @@ -91,7 +91,7 @@ static int snd_pdacf_dev_free(struct snd_device *device) */ static int snd_pdacf_probe(struct pcmcia_device *link) { - int i; + int i, err; struct snd_pdacf *pdacf; struct snd_card *card; static struct snd_device_ops ops = { @@ -112,10 +112,10 @@ static int snd_pdacf_probe(struct pcmcia_device *link) return -ENODEV; /* disabled explicitly */ /* ok, create a card instance */ - card = snd_card_new(index[i], id[i], THIS_MODULE, 0); - if (card == NULL) { + err = snd_card_create(index[i], id[i], THIS_MODULE, 0, &card); + if (err < 0) { snd_printk(KERN_ERR "pdacf: cannot create a card instance\n"); - return -ENOMEM; + return err; } pdacf = snd_pdacf_create(card); diff --git a/sound/pcmcia/vx/vxpocket.c b/sound/pcmcia/vx/vxpocket.c index 706602a4060..087ded8a8d7 100644 --- a/sound/pcmcia/vx/vxpocket.c +++ b/sound/pcmcia/vx/vxpocket.c @@ -292,7 +292,7 @@ static int vxpocket_probe(struct pcmcia_device *p_dev) { struct snd_card *card; struct snd_vxpocket *vxp; - int i; + int i, err; /* find an empty slot from the card list */ for (i = 0; i < SNDRV_CARDS; i++) { @@ -307,10 +307,10 @@ static int vxpocket_probe(struct pcmcia_device *p_dev) return -ENODEV; /* disabled explicitly */ /* ok, create a card instance */ - card = snd_card_new(index[i], id[i], THIS_MODULE, 0); - if (card == NULL) { + err = snd_card_create(index[i], id[i], THIS_MODULE, 0, &card); + if (err < 0) { snd_printk(KERN_ERR "vxpocket: cannot create a card instance\n"); - return -ENOMEM; + return err; } vxp = snd_vxpocket_new(card, ibl[i], p_dev); diff --git a/sound/ppc/powermac.c b/sound/ppc/powermac.c index c936225771b..2e18ed0ea89 100644 --- a/sound/ppc/powermac.c +++ b/sound/ppc/powermac.c @@ -58,9 +58,9 @@ static int __init snd_pmac_probe(struct platform_device *devptr) char *name_ext; int err; - card = snd_card_new(index, id, THIS_MODULE, 0); - if (card == NULL) - return -ENOMEM; + err = snd_card_create(index, id, THIS_MODULE, 0, &card); + if (err < 0) + return err; if ((err = snd_pmac_new(card, &chip)) < 0) goto __error; diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c index 8f9e3859c37..ef2c3f41717 100644 --- a/sound/ppc/snd_ps3.c +++ b/sound/ppc/snd_ps3.c @@ -969,11 +969,9 @@ static int __init snd_ps3_driver_probe(struct ps3_system_bus_device *dev) } /* create card instance */ - the_card.card = snd_card_new(index, id, THIS_MODULE, 0); - if (!the_card.card) { - ret = -ENXIO; + ret = snd_card_create(index, id, THIS_MODULE, 0, &the_card.card); + if (ret < 0) goto clean_irq; - } strcpy(the_card.card->driver, "PS3"); strcpy(the_card.card->shortname, "PS3"); diff --git a/sound/sh/aica.c b/sound/sh/aica.c index 7c920f3e7fe..f551233c5a0 100644 --- a/sound/sh/aica.c +++ b/sound/sh/aica.c @@ -609,11 +609,11 @@ static int __devinit snd_aica_probe(struct platform_device *devptr) dreamcastcard = kmalloc(sizeof(struct snd_card_aica), GFP_KERNEL); if (unlikely(!dreamcastcard)) return -ENOMEM; - dreamcastcard->card = - snd_card_new(index, SND_AICA_DRIVER, THIS_MODULE, 0); - if (unlikely(!dreamcastcard->card)) { + err = snd_card_create(index, SND_AICA_DRIVER, THIS_MODULE, 0, + &dreamcastcard->card); + if (unlikely(err < 0)) { kfree(dreamcastcard); - return -ENODEV; + return err; } strcpy(dreamcastcard->card->driver, "snd_aica"); strcpy(dreamcastcard->card->shortname, SND_AICA_DRIVER); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 6cbe7e82f23..318dfdd54d7 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1311,17 +1311,17 @@ int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid) { struct snd_soc_codec *codec = socdev->codec; struct snd_soc_card *card = socdev->card; - int ret = 0, i; + int ret, i; mutex_lock(&codec->mutex); /* register a sound card */ - codec->card = snd_card_new(idx, xid, codec->owner, 0); - if (!codec->card) { + ret = snd_card_create(idx, xid, codec->owner, 0, &codec->card); + if (ret < 0) { printk(KERN_ERR "asoc: can't create sound card for codec %s\n", codec->name); mutex_unlock(&codec->mutex); - return -ENODEV; + return ret; } codec->card->dev = socdev->dev; diff --git a/sound/sparc/amd7930.c b/sound/sparc/amd7930.c index f87933e4881..ba38912614b 100644 --- a/sound/sparc/amd7930.c +++ b/sound/sparc/amd7930.c @@ -1018,9 +1018,10 @@ static int __devinit amd7930_sbus_probe(struct of_device *op, const struct of_de return -ENOENT; } - card = snd_card_new(index[dev_num], id[dev_num], THIS_MODULE, 0); - if (card == NULL) - return -ENOMEM; + err = snd_card_create(index[dev_num], id[dev_num], THIS_MODULE, 0, + &card); + if (err < 0) + return err; strcpy(card->driver, "AMD7930"); strcpy(card->shortname, "Sun AMD7930"); diff --git a/sound/sparc/cs4231.c b/sound/sparc/cs4231.c index 41c38758747..7d93fa705cc 100644 --- a/sound/sparc/cs4231.c +++ b/sound/sparc/cs4231.c @@ -1563,6 +1563,7 @@ static int __init cs4231_attach_begin(struct snd_card **rcard) { struct snd_card *card; struct snd_cs4231 *chip; + int err; *rcard = NULL; @@ -1574,10 +1575,10 @@ static int __init cs4231_attach_begin(struct snd_card **rcard) return -ENOENT; } - card = snd_card_new(index[dev], id[dev], THIS_MODULE, - sizeof(struct snd_cs4231)); - if (card == NULL) - return -ENOMEM; + err = snd_card_create(index[dev], id[dev], THIS_MODULE, + sizeof(struct snd_cs4231), &card); + if (err < 0) + return err; strcpy(card->driver, "CS4231"); strcpy(card->shortname, "Sun CS4231"); diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index 23ed6f04a71..af95ff1e126 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -2612,10 +2612,10 @@ static int __devinit dbri_probe(struct of_device *op, const struct of_device_id return -ENODEV; } - card = snd_card_new(index[dev], id[dev], THIS_MODULE, - sizeof(struct snd_dbri)); - if (card == NULL) - return -ENOMEM; + err = snd_card_create(index[dev], id[dev], THIS_MODULE, + sizeof(struct snd_dbri), &card); + if (err < 0) + return err; strcpy(card->driver, "DBRI"); strcpy(card->shortname, "Sun DBRI"); diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c index 09802e8a6fb..4c7b051f9d1 100644 --- a/sound/spi/at73c213.c +++ b/sound/spi/at73c213.c @@ -965,12 +965,11 @@ static int __devinit snd_at73c213_probe(struct spi_device *spi) return PTR_ERR(board->dac_clk); } - retval = -ENOMEM; - /* Allocate "card" using some unused identifiers. */ snprintf(id, sizeof id, "at73c213_%d", board->ssc_id); - card = snd_card_new(-1, id, THIS_MODULE, sizeof(struct snd_at73c213)); - if (!card) + retval = snd_card_create(-1, id, THIS_MODULE, + sizeof(struct snd_at73c213), &card); + if (retval < 0) goto out; chip = card->private_data; diff --git a/sound/usb/caiaq/caiaq-device.c b/sound/usb/caiaq/caiaq-device.c index a62500e387a..63a2c1d5779 100644 --- a/sound/usb/caiaq/caiaq-device.c +++ b/sound/usb/caiaq/caiaq-device.c @@ -339,6 +339,7 @@ static void __devinit setup_card(struct snd_usb_caiaqdev *dev) static struct snd_card* create_card(struct usb_device* usb_dev) { int devnum; + int err; struct snd_card *card; struct snd_usb_caiaqdev *dev; @@ -349,9 +350,9 @@ static struct snd_card* create_card(struct usb_device* usb_dev) if (devnum >= SNDRV_CARDS) return NULL; - card = snd_card_new(index[devnum], id[devnum], THIS_MODULE, - sizeof(struct snd_usb_caiaqdev)); - if (!card) + err = snd_card_create(index[devnum], id[devnum], THIS_MODULE, + sizeof(struct snd_usb_caiaqdev), &card); + if (err < 0) return NULL; dev = caiaqdev(card); diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index c709b956322..eec32e1a302 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -3463,10 +3463,10 @@ static int snd_usb_audio_create(struct usb_device *dev, int idx, return -ENXIO; } - card = snd_card_new(index[idx], id[idx], THIS_MODULE, 0); - if (card == NULL) { + err = snd_card_create(index[idx], id[idx], THIS_MODULE, 0, &card); + if (err < 0) { snd_printk(KERN_ERR "cannot create card instance %d\n", idx); - return -ENOMEM; + return err; } chip = kzalloc(sizeof(*chip), GFP_KERNEL); diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c index 73e59f4403a..b21bb475c0f 100644 --- a/sound/usb/usx2y/us122l.c +++ b/sound/usb/usx2y/us122l.c @@ -482,14 +482,16 @@ static struct snd_card *usx2y_create_card(struct usb_device *device) { int dev; struct snd_card *card; + int err; + for (dev = 0; dev < SNDRV_CARDS; ++dev) if (enable[dev] && !snd_us122l_card_used[dev]) break; if (dev >= SNDRV_CARDS) return NULL; - card = snd_card_new(index[dev], id[dev], THIS_MODULE, - sizeof(struct us122l)); - if (!card) + err = snd_card_create(index[dev], id[dev], THIS_MODULE, + sizeof(struct us122l), &card); + if (err < 0) return NULL; snd_us122l_card_used[US122L(card)->chip.index = dev] = 1; diff --git a/sound/usb/usx2y/usbusx2y.c b/sound/usb/usx2y/usbusx2y.c index 11639bd72a5..b848a180638 100644 --- a/sound/usb/usx2y/usbusx2y.c +++ b/sound/usb/usx2y/usbusx2y.c @@ -337,13 +337,16 @@ static struct snd_card *usX2Y_create_card(struct usb_device *device) { int dev; struct snd_card * card; + int err; + for (dev = 0; dev < SNDRV_CARDS; ++dev) if (enable[dev] && !snd_usX2Y_card_used[dev]) break; if (dev >= SNDRV_CARDS) return NULL; - card = snd_card_new(index[dev], id[dev], THIS_MODULE, sizeof(struct usX2Ydev)); - if (!card) + err = snd_card_create(index[dev], id[dev], THIS_MODULE, + sizeof(struct usX2Ydev), &card); + if (err < 0) return NULL; snd_usX2Y_card_used[usX2Y(card)->chip.index = dev] = 1; card->private_free = snd_usX2Y_card_private_free; -- cgit v1.2.3-70-g09d2 From d453379bc5d34d7f55b55931245de5ac1896fd8d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 28 Dec 2008 16:45:34 +0100 Subject: ALSA: Update description of snd_card_create() in documents Signed-off-by: Takashi Iwai --- .../sound/alsa/DocBook/writing-an-alsa-driver.tmpl | 44 ++++++++++++---------- 1 file changed, 25 insertions(+), 19 deletions(-) diff --git a/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl b/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl index 87a7c07ab65..320384c1791 100644 --- a/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl +++ b/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl @@ -492,9 +492,9 @@ } /* (2) */ - card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); - if (card == NULL) - return -ENOMEM; + err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); + if (err < 0) + return err; /* (3) */ err = snd_mychip_create(card, pci, &chip); @@ -590,8 +590,9 @@ @@ -809,26 +810,28 @@ As mentioned above, to create a card instance, call - snd_card_new(). + snd_card_create(). - The function takes four arguments, the card-index number, the + The function takes five arguments, the card-index number, the id string, the module pointer (usually THIS_MODULE), - and the size of extra-data space. The last argument is used to + the size of extra-data space, and the pointer to return the + card instance. The extra_size argument is used to allocate card->private_data for the chip-specific data. Note that these data - are allocated by snd_card_new(). + are allocated by snd_card_create(). @@ -915,15 +918,16 @@
- 1. Allocating via <function>snd_card_new()</function>. + 1. Allocating via <function>snd_card_create()</function>. As mentioned above, you can pass the extra-data-length - to the 4th argument of snd_card_new(), i.e. + to the 4th argument of snd_card_create(), i.e. @@ -952,8 +956,8 @@ After allocating a card instance via - snd_card_new() (with - NULL on the 4th arg), call + snd_card_create() (with + 0 on the 4th arg), call kzalloc(). @@ -961,7 +965,7 @@ @@ -5750,8 +5754,9 @@ struct _snd_pcm_runtime { .... struct snd_card *card; struct mychip *chip; + int err; .... - card = snd_card_new(index[dev], id[dev], THIS_MODULE, NULL); + err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); .... chip = kzalloc(sizeof(*chip), GFP_KERNEL); .... @@ -5763,7 +5768,7 @@ struct _snd_pcm_runtime { When you created the chip data with - snd_card_new(), it's anyway accessible + snd_card_create(), it's anyway accessible via private_data field. @@ -5775,9 +5780,10 @@ struct _snd_pcm_runtime { .... struct snd_card *card; struct mychip *chip; + int err; .... - card = snd_card_new(index[dev], id[dev], THIS_MODULE, - sizeof(struct mychip)); + err = snd_card_create(index[dev], id[dev], THIS_MODULE, + sizeof(struct mychip), &card); .... chip = card->private_data; .... -- cgit v1.2.3-70-g09d2 From 3e7fb9f7ec00fd7cefd0d8e83df0cff86ce12515 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 28 Dec 2008 16:47:30 +0100 Subject: ALSA: Return proper error code at probe in sound/isa/* Some drivers in sound/isa/* don't handle the error code properly from snd_card_create(). This patch fixes these places. Signed-off-by: Takashi Iwai --- sound/isa/cs423x/cs4236.c | 25 +++++++++++++------------ sound/isa/es18xx.c | 27 ++++++++++++--------------- sound/isa/gus/interwave.c | 19 ++++++++++--------- sound/isa/opti9xx/opti92x-ad1848.c | 20 ++++++++++---------- sound/isa/sb/sb16.c | 19 ++++++++++--------- sound/isa/wavefront/wavefront.c | 19 ++++++++++--------- 6 files changed, 65 insertions(+), 64 deletions(-) diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c index db830682804..f7845986f46 100644 --- a/sound/isa/cs423x/cs4236.c +++ b/sound/isa/cs423x/cs4236.c @@ -382,7 +382,7 @@ static void snd_card_cs4236_free(struct snd_card *card) release_and_free_resource(acard->res_sb_port); } -static struct snd_card *snd_cs423x_card_new(int dev) +static int snd_cs423x_card_new(int dev, struct snd_card **cardp) { struct snd_card *card; int err; @@ -390,9 +390,10 @@ static struct snd_card *snd_cs423x_card_new(int dev) err = snd_card_create(index[dev], id[dev], THIS_MODULE, sizeof(struct snd_card_cs4236), &card); if (err < 0) - return NULL; + return err; card->private_free = snd_card_cs4236_free; - return card; + *cardp = card; + return 0; } static int __devinit snd_cs423x_probe(struct snd_card *card, int dev) @@ -513,9 +514,9 @@ static int __devinit snd_cs423x_isa_probe(struct device *pdev, struct snd_card *card; int err; - card = snd_cs423x_card_new(dev); - if (! card) - return -ENOMEM; + err = snd_cs423x_card_new(dev, &card); + if (err < 0) + return err; snd_card_set_dev(card, pdev); if ((err = snd_cs423x_probe(card, dev)) < 0) { snd_card_free(card); @@ -595,9 +596,9 @@ static int __devinit snd_cs4232_pnpbios_detect(struct pnp_dev *pdev, if (dev >= SNDRV_CARDS) return -ENODEV; - card = snd_cs423x_card_new(dev); - if (! card) - return -ENOMEM; + err = snd_cs423x_card_new(dev, &card); + if (err < 0) + return err; if ((err = snd_card_cs4232_pnp(dev, card->private_data, pdev)) < 0) { printk(KERN_ERR "PnP BIOS detection failed for " IDENT "\n"); snd_card_free(card); @@ -657,9 +658,9 @@ static int __devinit snd_cs423x_pnpc_detect(struct pnp_card_link *pcard, if (dev >= SNDRV_CARDS) return -ENODEV; - card = snd_cs423x_card_new(dev); - if (! card) - return -ENOMEM; + res = snd_cs423x_card_new(dev, &card); + if (res < 0) + return res; if ((res = snd_card_cs423x_pnpc(dev, card->private_data, pcard, pid)) < 0) { printk(KERN_ERR "isapnp detection failed and probing for " IDENT " is not supported\n"); diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index c24c6322fcc..8cfbff73a83 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -2125,13 +2125,10 @@ static int __devinit snd_audiodrive_pnpc(int dev, struct snd_audiodrive *acard, #define is_isapnp_selected(dev) 0 #endif -static struct snd_card *snd_es18xx_card_new(int dev) +static int snd_es18xx_card_new(int dev, struct snd_card **cardp) { - struct snd_card *card; - if (snd_card_create(index[dev], id[dev], THIS_MODULE, - sizeof(struct snd_audiodrive), &card) < 0) - return NULL; - return card; + return snd_card_create(index[dev], id[dev], THIS_MODULE, + sizeof(struct snd_audiodrive), cardp); } static int __devinit snd_audiodrive_probe(struct snd_card *card, int dev) @@ -2200,9 +2197,9 @@ static int __devinit snd_es18xx_isa_probe1(int dev, struct device *devptr) struct snd_card *card; int err; - card = snd_es18xx_card_new(dev); - if (! card) - return -ENOMEM; + err = snd_es18xx_card_new(dev, &card); + if (err < 0) + return err; snd_card_set_dev(card, devptr); if ((err = snd_audiodrive_probe(card, dev)) < 0) { snd_card_free(card); @@ -2306,9 +2303,9 @@ static int __devinit snd_audiodrive_pnp_detect(struct pnp_dev *pdev, if (dev >= SNDRV_CARDS) return -ENODEV; - card = snd_es18xx_card_new(dev); - if (! card) - return -ENOMEM; + err = snd_es18xx_card_new(dev, &card); + if (err < 0) + return err; if ((err = snd_audiodrive_pnp(dev, card->private_data, pdev)) < 0) { snd_card_free(card); return err; @@ -2365,9 +2362,9 @@ static int __devinit snd_audiodrive_pnpc_detect(struct pnp_card_link *pcard, if (dev >= SNDRV_CARDS) return -ENODEV; - card = snd_es18xx_card_new(dev); - if (! card) - return -ENOMEM; + res = snd_es18xx_card_new(dev, &card); + if (res < 0) + return res; if ((res = snd_audiodrive_pnpc(dev, card->private_data, pcard, pid)) < 0) { snd_card_free(card); diff --git a/sound/isa/gus/interwave.c b/sound/isa/gus/interwave.c index e040c763891..50e429a120d 100644 --- a/sound/isa/gus/interwave.c +++ b/sound/isa/gus/interwave.c @@ -626,7 +626,7 @@ static void snd_interwave_free(struct snd_card *card) free_irq(iwcard->irq, (void *)iwcard); } -static struct snd_card *snd_interwave_card_new(int dev) +static int snd_interwave_card_new(int dev, struct snd_card **cardp) { struct snd_card *card; struct snd_interwave *iwcard; @@ -635,12 +635,13 @@ static struct snd_card *snd_interwave_card_new(int dev) err = snd_card_create(index[dev], id[dev], THIS_MODULE, sizeof(struct snd_interwave), &card); if (err < 0) - return NULL; + return err; iwcard = card->private_data; iwcard->card = card; iwcard->irq = -1; card->private_free = snd_interwave_free; - return card; + *cardp = card; + return 0; } static int __devinit snd_interwave_probe(struct snd_card *card, int dev) @@ -779,9 +780,9 @@ static int __devinit snd_interwave_isa_probe1(int dev, struct device *devptr) struct snd_card *card; int err; - card = snd_interwave_card_new(dev); - if (! card) - return -ENOMEM; + err = snd_interwave_card_new(dev, &card); + if (err < 0) + return err; snd_card_set_dev(card, devptr); if ((err = snd_interwave_probe(card, dev)) < 0) { @@ -877,9 +878,9 @@ static int __devinit snd_interwave_pnp_detect(struct pnp_card_link *pcard, if (dev >= SNDRV_CARDS) return -ENODEV; - card = snd_interwave_card_new(dev); - if (! card) - return -ENOMEM; + res = snd_interwave_card_new(dev, &card); + if (res < 0) + return res; if ((res = snd_interwave_pnp(dev, card->private_data, pcard, pid)) < 0) { snd_card_free(card); diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index 5750f38bb79..87a4feb5010 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -830,17 +830,17 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) return snd_card_register(card); } -static struct snd_card *snd_opti9xx_card_new(void) +static int snd_opti9xx_card_new(struct snd_card **cardp) { struct snd_card *card; - int err; err = snd_card_create(index, id, THIS_MODULE, sizeof(struct snd_opti9xx), &card); if (err < 0) - return NULL; + return err; card->private_free = snd_card_opti9xx_free; - return card; + *cardp = card; + return 0; } static int __devinit snd_opti9xx_isa_match(struct device *devptr, @@ -905,9 +905,9 @@ static int __devinit snd_opti9xx_isa_probe(struct device *devptr, } #endif - card = snd_opti9xx_card_new(); - if (! card) - return -ENOMEM; + error = snd_opti9xx_card_new(&card); + if (error < 0) + return error; if ((error = snd_card_opti9xx_detect(card, card->private_data)) < 0) { snd_card_free(card); @@ -952,9 +952,9 @@ static int __devinit snd_opti9xx_pnp_probe(struct pnp_card_link *pcard, return -EBUSY; if (! isapnp) return -ENODEV; - card = snd_opti9xx_card_new(); - if (! card) - return -ENOMEM; + error = snd_opti9xx_card_new(&card); + if (error < 0) + return error; chip = card->private_data; hw = snd_card_opti9xx_pnp(chip, pcard, pid); diff --git a/sound/isa/sb/sb16.c b/sound/isa/sb/sb16.c index adf4fdd2c4a..519c36346de 100644 --- a/sound/isa/sb/sb16.c +++ b/sound/isa/sb/sb16.c @@ -324,7 +324,7 @@ static void snd_sb16_free(struct snd_card *card) #define is_isapnp_selected(dev) 0 #endif -static struct snd_card *snd_sb16_card_new(int dev) +static int snd_sb16_card_new(int dev, struct snd_card **cardp) { struct snd_card *card; int err; @@ -332,9 +332,10 @@ static struct snd_card *snd_sb16_card_new(int dev) err = snd_card_create(index[dev], id[dev], THIS_MODULE, sizeof(struct snd_card_sb16), &card); if (err < 0) - return NULL; + return err; card->private_free = snd_sb16_free; - return card; + *cardp = card; + return 0; } static int __devinit snd_sb16_probe(struct snd_card *card, int dev) @@ -492,9 +493,9 @@ static int __devinit snd_sb16_isa_probe1(int dev, struct device *pdev) struct snd_card *card; int err; - card = snd_sb16_card_new(dev); - if (! card) - return -ENOMEM; + err = snd_sb16_card_new(dev, &card); + if (err < 0) + return err; acard = card->private_data; /* non-PnP FM port address is hardwired with base port address */ @@ -613,9 +614,9 @@ static int __devinit snd_sb16_pnp_detect(struct pnp_card_link *pcard, for ( ; dev < SNDRV_CARDS; dev++) { if (!enable[dev] || !isapnp[dev]) continue; - card = snd_sb16_card_new(dev); - if (! card) - return -ENOMEM; + res = snd_sb16_card_new(dev, &card); + if (res < 0) + return res; snd_card_set_dev(card, &pcard->card->dev); if ((res = snd_card_sb16_pnp(dev, card->private_data, pcard, pid)) < 0 || (res = snd_sb16_probe(card, dev)) < 0) { diff --git a/sound/isa/wavefront/wavefront.c b/sound/isa/wavefront/wavefront.c index 82b8fb74690..95898b2b7b5 100644 --- a/sound/isa/wavefront/wavefront.c +++ b/sound/isa/wavefront/wavefront.c @@ -338,7 +338,7 @@ snd_wavefront_free(struct snd_card *card) } } -static struct snd_card *snd_wavefront_card_new(int dev) +static int snd_wavefront_card_new(int dev, struct snd_card **cardp) { struct snd_card *card; snd_wavefront_card_t *acard; @@ -347,7 +347,7 @@ static struct snd_card *snd_wavefront_card_new(int dev) err = snd_card_create(index[dev], id[dev], THIS_MODULE, sizeof(snd_wavefront_card_t), &card); if (err < 0) - return NULL; + return err; acard = card->private_data; acard->wavefront.irq = -1; @@ -358,7 +358,8 @@ static struct snd_card *snd_wavefront_card_new(int dev) acard->wavefront.card = card; card->private_free = snd_wavefront_free; - return card; + *cardp = card; + return 0; } static int __devinit @@ -568,9 +569,9 @@ static int __devinit snd_wavefront_isa_probe(struct device *pdev, struct snd_card *card; int err; - card = snd_wavefront_card_new(dev); - if (! card) - return -ENOMEM; + err = snd_wavefront_card_new(dev, &card); + if (err < 0) + return err; snd_card_set_dev(card, pdev); if ((err = snd_wavefront_probe(card, dev)) < 0) { snd_card_free(card); @@ -617,9 +618,9 @@ static int __devinit snd_wavefront_pnp_detect(struct pnp_card_link *pcard, if (dev >= SNDRV_CARDS) return -ENODEV; - card = snd_wavefront_card_new(dev); - if (! card) - return -ENOMEM; + res = snd_wavefront_card_new(dev, &card); + if (res < 0) + return res; if (snd_wavefront_pnp (dev, card->private_data, pcard, pid) < 0) { if (cs4232_pcm_port[dev] == SNDRV_AUTO_PORT) { -- cgit v1.2.3-70-g09d2 From 51721f70acaca5aa056b07c5cbe58e62662c068c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 28 Dec 2008 16:55:08 +0100 Subject: ALSA: Return proper error code at probe in sound/usb/* Some drivers in soudn/usb/* don't handle the error code properly from snd_card_create(). This patch fixes these places. Signed-off-by: Takashi Iwai --- sound/usb/caiaq/caiaq-device.c | 15 +++++++------ sound/usb/usx2y/us122l.c | 51 +++++++++++++++++++++++++----------------- sound/usb/usx2y/usbusx2y.c | 45 +++++++++++++++++++++++-------------- 3 files changed, 67 insertions(+), 44 deletions(-) diff --git a/sound/usb/caiaq/caiaq-device.c b/sound/usb/caiaq/caiaq-device.c index 63a2c1d5779..55a9075cb09 100644 --- a/sound/usb/caiaq/caiaq-device.c +++ b/sound/usb/caiaq/caiaq-device.c @@ -336,7 +336,7 @@ static void __devinit setup_card(struct snd_usb_caiaqdev *dev) log("Unable to set up control system (ret=%d)\n", ret); } -static struct snd_card* create_card(struct usb_device* usb_dev) +static int create_card(struct usb_device* usb_dev, struct snd_card **cardp) { int devnum; int err; @@ -348,12 +348,12 @@ static struct snd_card* create_card(struct usb_device* usb_dev) break; if (devnum >= SNDRV_CARDS) - return NULL; + return -ENODEV; err = snd_card_create(index[devnum], id[devnum], THIS_MODULE, sizeof(struct snd_usb_caiaqdev), &card); if (err < 0) - return NULL; + return err; dev = caiaqdev(card); dev->chip.dev = usb_dev; @@ -363,7 +363,8 @@ static struct snd_card* create_card(struct usb_device* usb_dev) spin_lock_init(&dev->spinlock); snd_card_set_dev(card, &usb_dev->dev); - return card; + *cardp = card; + return 0; } static int __devinit init_card(struct snd_usb_caiaqdev *dev) @@ -442,10 +443,10 @@ static int __devinit snd_probe(struct usb_interface *intf, struct snd_card *card; struct usb_device *device = interface_to_usbdev(intf); - card = create_card(device); + ret = create_card(device, &card); - if (!card) - return -ENOMEM; + if (ret < 0) + return ret; usb_set_intfdata(intf, card); ret = init_card(caiaqdev(card)); diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c index b21bb475c0f..98276aafefe 100644 --- a/sound/usb/usx2y/us122l.c +++ b/sound/usb/usx2y/us122l.c @@ -478,7 +478,7 @@ static bool us122l_create_card(struct snd_card *card) return true; } -static struct snd_card *usx2y_create_card(struct usb_device *device) +static int usx2y_create_card(struct usb_device *device, struct snd_card **cardp) { int dev; struct snd_card *card; @@ -488,11 +488,11 @@ static struct snd_card *usx2y_create_card(struct usb_device *device) if (enable[dev] && !snd_us122l_card_used[dev]) break; if (dev >= SNDRV_CARDS) - return NULL; + return -ENODEV; err = snd_card_create(index[dev], id[dev], THIS_MODULE, sizeof(struct us122l), &card); if (err < 0) - return NULL; + return err; snd_us122l_card_used[US122L(card)->chip.index = dev] = 1; US122L(card)->chip.dev = device; @@ -511,46 +511,57 @@ static struct snd_card *usx2y_create_card(struct usb_device *device) US122L(card)->chip.dev->devnum ); snd_card_set_dev(card, &device->dev); - return card; + *cardp = card; + return 0; } -static void *us122l_usb_probe(struct usb_interface *intf, - const struct usb_device_id *device_id) +static int us122l_usb_probe(struct usb_interface *intf, + const struct usb_device_id *device_id, + struct snd_card **cardp) { struct usb_device *device = interface_to_usbdev(intf); - struct snd_card *card = usx2y_create_card(device); + struct snd_card *card; + int err; - if (!card) - return NULL; + err = usx2y_create_card(device, &card); + if (err < 0) + return err; - if (!us122l_create_card(card) || - snd_card_register(card) < 0) { + if (!us122l_create_card(card)) { snd_card_free(card); - return NULL; + return -EINVAL; + } + + err = snd_card_register(card); + if (err < 0) { + snd_card_free(card); + return err; } usb_get_dev(device); - return card; + *cardp = card; + return 0; } static int snd_us122l_probe(struct usb_interface *intf, const struct usb_device_id *id) { struct snd_card *card; + int err; + snd_printdd(KERN_DEBUG"%p:%i\n", intf, intf->cur_altsetting->desc.bInterfaceNumber); if (intf->cur_altsetting->desc.bInterfaceNumber != 1) return 0; - card = us122l_usb_probe(usb_get_intf(intf), id); - - if (card) { - usb_set_intfdata(intf, card); - return 0; + err = us122l_usb_probe(usb_get_intf(intf), id, &card); + if (err < 0) { + usb_put_intf(intf); + return err; } - usb_put_intf(intf); - return -EIO; + usb_set_intfdata(intf, card); + return 0; } static void snd_us122l_disconnect(struct usb_interface *intf) diff --git a/sound/usb/usx2y/usbusx2y.c b/sound/usb/usx2y/usbusx2y.c index b848a180638..af8b8495405 100644 --- a/sound/usb/usx2y/usbusx2y.c +++ b/sound/usb/usx2y/usbusx2y.c @@ -333,7 +333,7 @@ static struct usb_device_id snd_usX2Y_usb_id_table[] = { { /* terminator */ } }; -static struct snd_card *usX2Y_create_card(struct usb_device *device) +static int usX2Y_create_card(struct usb_device *device, struct snd_card **cardp) { int dev; struct snd_card * card; @@ -343,11 +343,11 @@ static struct snd_card *usX2Y_create_card(struct usb_device *device) if (enable[dev] && !snd_usX2Y_card_used[dev]) break; if (dev >= SNDRV_CARDS) - return NULL; + return -ENODEV; err = snd_card_create(index[dev], id[dev], THIS_MODULE, sizeof(struct usX2Ydev), &card); if (err < 0) - return NULL; + return err; snd_usX2Y_card_used[usX2Y(card)->chip.index = dev] = 1; card->private_free = snd_usX2Y_card_private_free; usX2Y(card)->chip.dev = device; @@ -365,26 +365,36 @@ static struct snd_card *usX2Y_create_card(struct usb_device *device) usX2Y(card)->chip.dev->bus->busnum, usX2Y(card)->chip.dev->devnum ); snd_card_set_dev(card, &device->dev); - return card; + *cardp = card; + return 0; } -static void *usX2Y_usb_probe(struct usb_device *device, struct usb_interface *intf, const struct usb_device_id *device_id) +static int usX2Y_usb_probe(struct usb_device *device, + struct usb_interface *intf, + const struct usb_device_id *device_id, + struct snd_card **cardp) { int err; struct snd_card * card; + + *cardp = NULL; if (le16_to_cpu(device->descriptor.idVendor) != 0x1604 || (le16_to_cpu(device->descriptor.idProduct) != USB_ID_US122 && le16_to_cpu(device->descriptor.idProduct) != USB_ID_US224 && - le16_to_cpu(device->descriptor.idProduct) != USB_ID_US428) || - !(card = usX2Y_create_card(device))) - return NULL; + le16_to_cpu(device->descriptor.idProduct) != USB_ID_US428)) + return -EINVAL; + + err = usX2Y_create_card(device, &card); + if (err < 0) + return err; if ((err = usX2Y_hwdep_new(card, device)) < 0 || (err = snd_card_register(card)) < 0) { snd_card_free(card); - return NULL; + return err; } - return card; + *cardp = card; + return 0; } /* @@ -392,13 +402,14 @@ static void *usX2Y_usb_probe(struct usb_device *device, struct usb_interface *in */ static int snd_usX2Y_probe(struct usb_interface *intf, const struct usb_device_id *id) { - void *chip; - chip = usX2Y_usb_probe(interface_to_usbdev(intf), intf, id); - if (chip) { - usb_set_intfdata(intf, chip); - return 0; - } else - return -EIO; + struct snd_card *card; + int err; + + err = usX2Y_usb_probe(interface_to_usbdev(intf), intf, id, &card); + if (err < 0) + return err; + dev_set_drvdata(&intf->dev, card); + return 0; } static void snd_usX2Y_disconnect(struct usb_interface *intf) -- cgit v1.2.3-70-g09d2 From aa3d75d80de464cf23af1d068a5e22f1527b6957 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 28 Dec 2008 16:59:41 +0100 Subject: ALSA: pdaudiocf - Fix missing free in the error path Added the missing snd_card_free() in the error path of probe callback. Signed-off-by: Takashi Iwai --- sound/pcmcia/pdaudiocf/pdaudiocf.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.c b/sound/pcmcia/pdaudiocf/pdaudiocf.c index 183f6615c68..ec51569fd50 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf.c @@ -119,8 +119,10 @@ static int snd_pdacf_probe(struct pcmcia_device *link) } pdacf = snd_pdacf_create(card); - if (! pdacf) + if (!pdacf) { + snd_card_free(card); return -EIO; + } if (snd_device_new(card, SNDRV_DEV_LOWLEVEL, pdacf, &ops) < 0) { kfree(pdacf); -- cgit v1.2.3-70-g09d2 From 2fa51107c9aa80ae95b4524198442cdea82d08a3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 28 Dec 2008 17:03:56 +0100 Subject: ALSA: Return proper error code at probe in sound/pcmcia/* Signed-off-by: Takashi Iwai --- sound/pcmcia/pdaudiocf/pdaudiocf.c | 7 ++++--- sound/pcmcia/vx/vxpocket.c | 24 ++++++++++++++---------- 2 files changed, 18 insertions(+), 13 deletions(-) diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.c b/sound/pcmcia/pdaudiocf/pdaudiocf.c index ec51569fd50..7dea74b71cf 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf.c @@ -121,13 +121,14 @@ static int snd_pdacf_probe(struct pcmcia_device *link) pdacf = snd_pdacf_create(card); if (!pdacf) { snd_card_free(card); - return -EIO; + return -ENOMEM; } - if (snd_device_new(card, SNDRV_DEV_LOWLEVEL, pdacf, &ops) < 0) { + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, pdacf, &ops); + if (err < 0) { kfree(pdacf); snd_card_free(card); - return -ENODEV; + return err; } snd_card_set_dev(card, &handle_to_dev(link)); diff --git a/sound/pcmcia/vx/vxpocket.c b/sound/pcmcia/vx/vxpocket.c index 087ded8a8d7..7445cc8a47d 100644 --- a/sound/pcmcia/vx/vxpocket.c +++ b/sound/pcmcia/vx/vxpocket.c @@ -130,23 +130,26 @@ static struct snd_vx_hardware vxp440_hw = { /* * create vxpocket instance */ -static struct snd_vxpocket *snd_vxpocket_new(struct snd_card *card, int ibl, - struct pcmcia_device *link) +static int snd_vxpocket_new(struct snd_card *card, int ibl, + struct pcmcia_device *link, + struct snd_vxpocket **chip_ret) { struct vx_core *chip; struct snd_vxpocket *vxp; static struct snd_device_ops ops = { .dev_free = snd_vxpocket_dev_free, }; + int err; chip = snd_vx_create(card, &vxpocket_hw, &snd_vxpocket_ops, sizeof(struct snd_vxpocket) - sizeof(struct vx_core)); - if (! chip) - return NULL; + if (!chip) + return -ENOMEM; - if (snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops) < 0) { + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) { kfree(chip); - return NULL; + return err; } chip->ibl.size = ibl; @@ -169,7 +172,8 @@ static struct snd_vxpocket *snd_vxpocket_new(struct snd_card *card, int ibl, link->conf.ConfigIndex = 1; link->conf.Present = PRESENT_OPTION; - return vxp; + *chip_ret = vxp; + return 0; } @@ -313,10 +317,10 @@ static int vxpocket_probe(struct pcmcia_device *p_dev) return err; } - vxp = snd_vxpocket_new(card, ibl[i], p_dev); - if (! vxp) { + err = snd_vxpocket_new(card, ibl[i], p_dev, &vxp); + if (err < 0) { snd_card_free(card); - return -ENODEV; + return err; } card->private_data = vxp; -- cgit v1.2.3-70-g09d2 From 758021bfa9ea25c58e62d2f68512628b19502ce7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 12 Jan 2009 15:17:09 +0100 Subject: drivers/media: Convert to snd_card_create() Convert from snd_card_new() to the new snd_card_create() function. Signed-off-by: Takashi Iwai --- drivers/media/video/cx88/cx88-alsa.c | 7 ++++--- drivers/media/video/em28xx/em28xx-audio.c | 7 ++++--- drivers/media/video/saa7134/saa7134-alsa.c | 8 ++++---- 3 files changed, 12 insertions(+), 10 deletions(-) diff --git a/drivers/media/video/cx88/cx88-alsa.c b/drivers/media/video/cx88/cx88-alsa.c index 66c755c116d..ce98d955231 100644 --- a/drivers/media/video/cx88/cx88-alsa.c +++ b/drivers/media/video/cx88/cx88-alsa.c @@ -803,9 +803,10 @@ static int __devinit cx88_audio_initdev(struct pci_dev *pci, return (-ENOENT); } - card = snd_card_new(index[devno], id[devno], THIS_MODULE, sizeof(snd_cx88_card_t)); - if (!card) - return (-ENOMEM); + err = snd_card_create(index[devno], id[devno], THIS_MODULE, + sizeof(snd_cx88_card_t), &card); + if (err < 0) + return err; card->private_free = snd_cx88_dev_free; diff --git a/drivers/media/video/em28xx/em28xx-audio.c b/drivers/media/video/em28xx/em28xx-audio.c index 94378ccb750..66579508e17 100644 --- a/drivers/media/video/em28xx/em28xx-audio.c +++ b/drivers/media/video/em28xx/em28xx-audio.c @@ -438,9 +438,10 @@ static int em28xx_audio_init(struct em28xx *dev) printk(KERN_INFO "em28xx-audio.c: Copyright (C) 2006 Markus " "Rechberger\n"); - card = snd_card_new(index[devnr], "Em28xx Audio", THIS_MODULE, 0); - if (card == NULL) - return -ENOMEM; + err = snd_card_create(index[devnr], "Em28xx Audio", THIS_MODULE, 0, + &card); + if (err < 0) + return err; spin_lock_init(&adev->slock); err = snd_pcm_new(card, "Em28xx Audio", 0, 0, 1, &pcm); diff --git a/drivers/media/video/saa7134/saa7134-alsa.c b/drivers/media/video/saa7134/saa7134-alsa.c index 26194a0ce92..482be1436e9 100644 --- a/drivers/media/video/saa7134/saa7134-alsa.c +++ b/drivers/media/video/saa7134/saa7134-alsa.c @@ -990,10 +990,10 @@ static int alsa_card_saa7134_create(struct saa7134_dev *dev, int devnum) if (!enable[devnum]) return -ENODEV; - card = snd_card_new(index[devnum], id[devnum], THIS_MODULE, sizeof(snd_card_saa7134_t)); - - if (card == NULL) - return -ENOMEM; + err = snd_card_create(index[devnum], id[devnum], THIS_MODULE, + sizeof(snd_card_saa7134_t), &card); + if (err < 0) + return err; strcpy(card->driver, "SAA7134"); -- cgit v1.2.3-70-g09d2 From 6ff1871617a3ea1eeaf88b42f652f9a311826bad Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 12 Jan 2009 15:18:28 +0100 Subject: drivers/staging: Convert to snd_card_create() for go7007 Convert from snd_card_new to the new snd_card_create() for go7007. Signed-off-by: Takashi Iwai --- drivers/staging/go7007/snd-go7007.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/drivers/staging/go7007/snd-go7007.c b/drivers/staging/go7007/snd-go7007.c index a7de401f61a..cd19be6c00e 100644 --- a/drivers/staging/go7007/snd-go7007.c +++ b/drivers/staging/go7007/snd-go7007.c @@ -248,10 +248,11 @@ int go7007_snd_init(struct go7007 *go) spin_lock_init(&gosnd->lock); gosnd->hw_ptr = gosnd->w_idx = gosnd->avail = 0; gosnd->capturing = 0; - gosnd->card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); - if (gosnd->card == NULL) { + ret = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, + &gosnd->card); + if (ret < 0) { kfree(gosnd); - return -ENOMEM; + return ret; } ret = snd_device_new(gosnd->card, SNDRV_DEV_LOWLEVEL, go, &go7007_snd_device_ops); -- cgit v1.2.3-70-g09d2 From 183c6e0fb4e39c860960de4abd7541bd260491bb Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 12 Jan 2009 15:19:08 +0100 Subject: drivers/usb/gadget: Convert to snd_card_create() Convert from snd_card_new() to the new snd_card_create() function for gmidi. Signed-off-by: Takashi Iwai --- drivers/usb/gadget/gmidi.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/drivers/usb/gadget/gmidi.c b/drivers/usb/gadget/gmidi.c index 60d3f9e9b51..14e09abbddf 100644 --- a/drivers/usb/gadget/gmidi.c +++ b/drivers/usb/gadget/gmidi.c @@ -1099,10 +1099,9 @@ static int gmidi_register_card(struct gmidi_device *dev) .dev_free = gmidi_snd_free, }; - card = snd_card_new(index, id, THIS_MODULE, 0); - if (!card) { - ERROR(dev, "snd_card_new failed\n"); - err = -ENOMEM; + err = snd_card_create(index, id, THIS_MODULE, 0, &card); + if (err < 0) { + ERROR(dev, "snd_card_create failed\n"); goto fail; } dev->card = card; -- cgit v1.2.3-70-g09d2 From 554b91edec1c588b889a7357ff201c0a450e31ff Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Mon, 12 Jan 2009 21:25:04 +0100 Subject: ALSA: sscape: fix incorrect timeout after microcode upload A comment states that one should wait up to 5 secs while a waiting loop waits only 5 system ticks. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/sscape.c | 24 ++++++++++++------------ 1 file changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index bc449166d18..6a7f842b962 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -393,20 +393,20 @@ static int sscape_wait_dma_unsafe(unsigned io_base, enum GA_REG reg, unsigned ti */ static int obp_startup_ack(struct soundscape *s, unsigned timeout) { - while (timeout != 0) { + unsigned long end_time = jiffies + msecs_to_jiffies(timeout); + + do { unsigned long flags; unsigned char x; - schedule_timeout_uninterruptible(1); - spin_lock_irqsave(&s->lock, flags); x = inb(HOST_DATA_IO(s->io_base)); spin_unlock_irqrestore(&s->lock, flags); if ((x & 0xfe) == 0xfe) return 1; - --timeout; - } /* while */ + msleep(10); + } while (time_before(jiffies, end_time)); return 0; } @@ -420,20 +420,20 @@ static int obp_startup_ack(struct soundscape *s, unsigned timeout) */ static int host_startup_ack(struct soundscape *s, unsigned timeout) { - while (timeout != 0) { + unsigned long end_time = jiffies + msecs_to_jiffies(timeout); + + do { unsigned long flags; unsigned char x; - schedule_timeout_uninterruptible(1); - spin_lock_irqsave(&s->lock, flags); x = inb(HOST_DATA_IO(s->io_base)); spin_unlock_irqrestore(&s->lock, flags); if (x == 0xfe) return 1; - --timeout; - } /* while */ + msleep(10); + } while (time_before(jiffies, end_time)); return 0; } @@ -529,10 +529,10 @@ static int upload_dma_data(struct soundscape *s, * give it 5 seconds (max) ... */ ret = 0; - if (!obp_startup_ack(s, 5)) { + if (!obp_startup_ack(s, 5000)) { snd_printk(KERN_ERR "sscape: No response from on-board processor after upload\n"); ret = -EAGAIN; - } else if (!host_startup_ack(s, 5)) { + } else if (!host_startup_ack(s, 5000)) { snd_printk(KERN_ERR "sscape: SoundScape failed to initialise\n"); ret = -EAGAIN; } -- cgit v1.2.3-70-g09d2 From dc61b66fc724f89d357c43e2319d2cb7bec1e517 Mon Sep 17 00:00:00 2001 From: Andrea Borgia Date: Mon, 12 Jan 2009 23:17:47 +0100 Subject: ALSA: rename "Device" to "Toshiba SB-0500" via quirks Signed-off-by: Andrea Borgia Signed-off-by: Takashi Iwai --- sound/usb/usbquirks.h | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index 92115755d98..d59323ecd57 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -39,6 +39,16 @@ .idProduct = prod, \ .bInterfaceClass = USB_CLASS_VENDOR_SPEC +/* Creative/Toshiba Multimedia Center SB-0500 */ +{ + USB_DEVICE(0x041e, 0x3048), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Toshiba", + .product_name = "SB-0500", + .ifnum = QUIRK_NO_INTERFACE + } +}, + /* Creative/E-Mu devices */ { USB_DEVICE(0x041e, 0x3010), -- cgit v1.2.3-70-g09d2 From b1a0aac05f044e78a589bfd7a9e2334aa640eb45 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 14 Jan 2009 09:34:06 +0100 Subject: ALSA: opti9xx - Fix build breakage by snd_card_create() conversion Add a missing variable declaration. Signed-off-by: Takashi Iwai --- sound/isa/opti9xx/opti92x-ad1848.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index 87a4feb5010..cd6e60a6a4e 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -833,6 +833,7 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) static int snd_opti9xx_card_new(struct snd_card **cardp) { struct snd_card *card; + int err; err = snd_card_create(index, id, THIS_MODULE, sizeof(struct snd_opti9xx), &card); -- cgit v1.2.3-70-g09d2 From 641b4879444c0edb276fedca5c2fcbd2e5c70044 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 15 Jan 2009 17:05:24 +0100 Subject: ALSA: usb-audio - Cache mixer values Cache mixer values in usb-audio driver to reduce too excessive accesses to the hardware. Signed-off-by: Takashi Iwai --- sound/usb/usbmixer.c | 122 +++++++++++++++++++++++++++++---------------------- 1 file changed, 70 insertions(+), 52 deletions(-) diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index 00397c8a765..c07b3f8485e 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -110,6 +110,8 @@ struct mixer_build { const struct usbmix_selector_map *selector_map; }; +#define MAX_CHANNELS 10 /* max logical channels */ + struct usb_mixer_elem_info { struct usb_mixer_interface *mixer; struct usb_mixer_elem_info *next_id_elem; /* list of controls with same id */ @@ -120,6 +122,8 @@ struct usb_mixer_elem_info { int channels; int val_type; int min, max, res; + int cached; + int cache_val[MAX_CHANNELS]; u8 initialized; }; @@ -181,8 +185,6 @@ enum { USB_PROC_DCR_RELEASE = 6, }; -#define MAX_CHANNELS 10 /* max logical channels */ - /* * manual mapping of mixer names @@ -376,11 +378,35 @@ static int get_cur_ctl_value(struct usb_mixer_elem_info *cval, int validx, int * } /* channel = 0: master, 1 = first channel */ -static inline int get_cur_mix_value(struct usb_mixer_elem_info *cval, int channel, int *value) +static inline int get_cur_mix_raw(struct usb_mixer_elem_info *cval, + int channel, int *value) { return get_ctl_value(cval, GET_CUR, (cval->control << 8) | channel, value); } +static int get_cur_mix_value(struct usb_mixer_elem_info *cval, + int channel, int index, int *value) +{ + int err; + + if (cval->cached & (1 << channel)) { + *value = cval->cache_val[index]; + return 0; + } + err = get_cur_mix_raw(cval, channel, value); + if (err < 0) { + if (!cval->mixer->ignore_ctl_error) + snd_printd(KERN_ERR "cannot get current value for " + "control %d ch %d: err = %d\n", + cval->control, channel, err); + return err; + } + cval->cached |= 1 << channel; + cval->cache_val[index] = *value; + return 0; +} + + /* * set a mixer value */ @@ -412,9 +438,17 @@ static int set_cur_ctl_value(struct usb_mixer_elem_info *cval, int validx, int v return set_ctl_value(cval, SET_CUR, validx, value); } -static inline int set_cur_mix_value(struct usb_mixer_elem_info *cval, int channel, int value) +static int set_cur_mix_value(struct usb_mixer_elem_info *cval, int channel, + int index, int value) { - return set_ctl_value(cval, SET_CUR, (cval->control << 8) | channel, value); + int err; + err = set_ctl_value(cval, SET_CUR, (cval->control << 8) | channel, + value); + if (err < 0) + return err; + cval->cached |= 1 << channel; + cval->cache_val[index] = value; + return 0; } /* @@ -718,7 +752,7 @@ static int get_min_max(struct usb_mixer_elem_info *cval, int default_min) if (cval->min + cval->res < cval->max) { int last_valid_res = cval->res; int saved, test, check; - get_cur_mix_value(cval, minchn, &saved); + get_cur_mix_raw(cval, minchn, &saved); for (;;) { test = saved; if (test < cval->max) @@ -726,8 +760,8 @@ static int get_min_max(struct usb_mixer_elem_info *cval, int default_min) else test -= cval->res; if (test < cval->min || test > cval->max || - set_cur_mix_value(cval, minchn, test) || - get_cur_mix_value(cval, minchn, &check)) { + set_cur_mix_value(cval, minchn, 0, test) || + get_cur_mix_raw(cval, minchn, &check)) { cval->res = last_valid_res; break; } @@ -735,7 +769,7 @@ static int get_min_max(struct usb_mixer_elem_info *cval, int default_min) break; cval->res *= 2; } - set_cur_mix_value(cval, minchn, saved); + set_cur_mix_value(cval, minchn, 0, saved); } cval->initialized = 1; @@ -775,35 +809,25 @@ static int mixer_ctl_feature_get(struct snd_kcontrol *kcontrol, struct snd_ctl_e struct usb_mixer_elem_info *cval = kcontrol->private_data; int c, cnt, val, err; + ucontrol->value.integer.value[0] = cval->min; if (cval->cmask) { cnt = 0; for (c = 0; c < MAX_CHANNELS; c++) { - if (cval->cmask & (1 << c)) { - err = get_cur_mix_value(cval, c + 1, &val); - if (err < 0) { - if (cval->mixer->ignore_ctl_error) { - ucontrol->value.integer.value[0] = cval->min; - return 0; - } - snd_printd(KERN_ERR "cannot get current value for control %d ch %d: err = %d\n", cval->control, c + 1, err); - return err; - } - val = get_relative_value(cval, val); - ucontrol->value.integer.value[cnt] = val; - cnt++; - } + if (!(cval->cmask & (1 << c))) + continue; + err = get_cur_mix_value(cval, c + 1, cnt, &val); + if (err < 0) + return cval->mixer->ignore_ctl_error ? 0 : err; + val = get_relative_value(cval, val); + ucontrol->value.integer.value[cnt] = val; + cnt++; } + return 0; } else { /* master channel */ - err = get_cur_mix_value(cval, 0, &val); - if (err < 0) { - if (cval->mixer->ignore_ctl_error) { - ucontrol->value.integer.value[0] = cval->min; - return 0; - } - snd_printd(KERN_ERR "cannot get current value for control %d master ch: err = %d\n", cval->control, err); - return err; - } + err = get_cur_mix_value(cval, 0, 0, &val); + if (err < 0) + return cval->mixer->ignore_ctl_error ? 0 : err; val = get_relative_value(cval, val); ucontrol->value.integer.value[0] = val; } @@ -820,34 +844,28 @@ static int mixer_ctl_feature_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e if (cval->cmask) { cnt = 0; for (c = 0; c < MAX_CHANNELS; c++) { - if (cval->cmask & (1 << c)) { - err = get_cur_mix_value(cval, c + 1, &oval); - if (err < 0) { - if (cval->mixer->ignore_ctl_error) - return 0; - return err; - } - val = ucontrol->value.integer.value[cnt]; - val = get_abs_value(cval, val); - if (oval != val) { - set_cur_mix_value(cval, c + 1, val); - changed = 1; - } - get_cur_mix_value(cval, c + 1, &val); - cnt++; + if (!(cval->cmask & (1 << c))) + continue; + err = get_cur_mix_value(cval, c + 1, cnt, &oval); + if (err < 0) + return cval->mixer->ignore_ctl_error ? 0 : err; + val = ucontrol->value.integer.value[cnt]; + val = get_abs_value(cval, val); + if (oval != val) { + set_cur_mix_value(cval, c + 1, cnt, val); + changed = 1; } + cnt++; } } else { /* master channel */ - err = get_cur_mix_value(cval, 0, &oval); - if (err < 0 && cval->mixer->ignore_ctl_error) - return 0; + err = get_cur_mix_value(cval, 0, 0, &oval); if (err < 0) - return err; + return cval->mixer->ignore_ctl_error ? 0 : err; val = ucontrol->value.integer.value[0]; val = get_abs_value(cval, val); if (val != oval) { - set_cur_mix_value(cval, 0, val); + set_cur_mix_value(cval, 0, 0, val); changed = 1; } } -- cgit v1.2.3-70-g09d2 From 45e513b689b8b0a01ec2b01cc21816e4780d7ea6 Mon Sep 17 00:00:00 2001 From: Johannes Berg Date: Thu, 15 Jan 2009 18:21:48 +0100 Subject: ALSA: snd-aoa: handle older machines This patch changes snd-aoa to handle some older machines that are currently handled by snd-powermac. snd-aoa has a number of advantages though, notably it can autoload better and is generally a more modern driver. By hardcoding the accepted device-ids (last hunk of the patch) I'm trying to avoid regressions because this driver will otherwise load automatically and not let snd-powermac load. People who are unhappy with snd-powermac and have a device-id property in the device tree are encouraged to read this patch and make a patch to amend this as appropriate. Signed-off-by: Johannes Berg Signed-off-by: Takashi Iwai --- sound/aoa/fabrics/layout.c | 74 +++++++++++++++++++++++++++++++--------- sound/aoa/soundbus/i2sbus/core.c | 22 +++++++++--- 2 files changed, 74 insertions(+), 22 deletions(-) diff --git a/sound/aoa/fabrics/layout.c b/sound/aoa/fabrics/layout.c index ad60f5d10e8..d9b1d22a62c 100644 --- a/sound/aoa/fabrics/layout.c +++ b/sound/aoa/fabrics/layout.c @@ -1,16 +1,14 @@ /* - * Apple Onboard Audio driver -- layout fabric + * Apple Onboard Audio driver -- layout/machine id fabric * - * Copyright 2006 Johannes Berg + * Copyright 2006-2008 Johannes Berg * * GPL v2, can be found in COPYING. * * - * This fabric module looks for sound codecs - * based on the layout-id property in the device tree. - * + * This fabric module looks for sound codecs based on the + * layout-id or device-id property in the device tree. */ - #include #include #include @@ -63,7 +61,7 @@ struct codec_connect_info { #define LAYOUT_FLAG_COMBO_LINEOUT_SPDIF (1<<0) struct layout { - unsigned int layout_id; + unsigned int layout_id, device_id; struct codec_connect_info codecs[MAX_CODECS_PER_BUS]; int flags; @@ -111,6 +109,10 @@ MODULE_ALIAS("sound-layout-96"); MODULE_ALIAS("sound-layout-98"); MODULE_ALIAS("sound-layout-100"); +MODULE_ALIAS("aoa-device-id-14"); +MODULE_ALIAS("aoa-device-id-22"); +MODULE_ALIAS("aoa-device-id-35"); + /* onyx with all but microphone connected */ static struct codec_connection onyx_connections_nomic[] = { { @@ -518,6 +520,27 @@ static struct layout layouts[] = { .connections = onyx_connections_noheadphones, }, }, + /* PowerMac3,4 */ + { .device_id = 14, + .codecs[0] = { + .name = "tas", + .connections = tas_connections_noline, + }, + }, + /* PowerMac3,6 */ + { .device_id = 22, + .codecs[0] = { + .name = "tas", + .connections = tas_connections_all, + }, + }, + /* PowerBook5,2 */ + { .device_id = 35, + .codecs[0] = { + .name = "tas", + .connections = tas_connections_all, + }, + }, {} }; @@ -526,7 +549,7 @@ static struct layout *find_layout_by_id(unsigned int id) struct layout *l; l = layouts; - while (l->layout_id) { + while (l->codecs[0].name) { if (l->layout_id == id) return l; l++; @@ -534,6 +557,19 @@ static struct layout *find_layout_by_id(unsigned int id) return NULL; } +static struct layout *find_layout_by_device(unsigned int id) +{ + struct layout *l; + + l = layouts; + while (l->codecs[0].name) { + if (l->device_id == id) + return l; + l++; + } + return NULL; +} + static void use_layout(struct layout *l) { int i; @@ -938,8 +974,8 @@ static struct aoa_fabric layout_fabric = { static int aoa_fabric_layout_probe(struct soundbus_dev *sdev) { struct device_node *sound = NULL; - const unsigned int *layout_id; - struct layout *layout; + const unsigned int *id; + struct layout *layout = NULL; struct layout_dev *ldev = NULL; int err; @@ -952,15 +988,18 @@ static int aoa_fabric_layout_probe(struct soundbus_dev *sdev) if (sound->type && strcasecmp(sound->type, "soundchip") == 0) break; } - if (!sound) return -ENODEV; + if (!sound) + return -ENODEV; - layout_id = of_get_property(sound, "layout-id", NULL); - if (!layout_id) - goto outnodev; - printk(KERN_INFO "snd-aoa-fabric-layout: found bus with layout %d\n", - *layout_id); + id = of_get_property(sound, "layout-id", NULL); + if (id) { + layout = find_layout_by_id(*id); + } else { + id = of_get_property(sound, "device-id", NULL); + if (id) + layout = find_layout_by_device(*id); + } - layout = find_layout_by_id(*layout_id); if (!layout) { printk(KERN_ERR "snd-aoa-fabric-layout: unknown layout\n"); goto outnodev; @@ -976,6 +1015,7 @@ static int aoa_fabric_layout_probe(struct soundbus_dev *sdev) ldev->layout = layout; ldev->gpio.node = sound->parent; switch (layout->layout_id) { + case 0: /* anything with device_id, not layout_id */ case 41: /* that unknown machine no one seems to have */ case 51: /* PowerBook5,4 */ case 58: /* Mac Mini */ diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c index be468edf3ec..418c84c99d6 100644 --- a/sound/aoa/soundbus/i2sbus/core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -1,7 +1,7 @@ /* * i2sbus driver * - * Copyright 2006 Johannes Berg + * Copyright 2006-2008 Johannes Berg * * GPL v2, can be found in COPYING. */ @@ -186,13 +186,25 @@ static int i2sbus_add_dev(struct macio_dev *macio, } } if (i == 1) { - const u32 *layout_id = - of_get_property(sound, "layout-id", NULL); - if (layout_id) { - layout = *layout_id; + const u32 *id = of_get_property(sound, "layout-id", NULL); + + if (id) { + layout = *id; snprintf(dev->sound.modalias, 32, "sound-layout-%d", layout); ok = 1; + } else { + id = of_get_property(sound, "device-id", NULL); + /* + * We probably cannot handle all device-id machines, + * so restrict to those we do handle for now. + */ + if (id && (*id == 22 || *id == 14 || *id == 35)) { + snprintf(dev->sound.modalias, 32, + "aoa-device-id-%d", *id); + ok = 1; + layout = -1; + } } } /* for the time being, until we can handle non-layout-id -- cgit v1.2.3-70-g09d2 From 5f17e79cdf530b1a6090c65730e5656ac9c19eaa Mon Sep 17 00:00:00 2001 From: Johannes Berg Date: Thu, 15 Jan 2009 18:22:31 +0100 Subject: ALSA: snd-aoa: handle master-amp if present Some machines have a master amp GPIO that needs to be toggled to get sound output, in addition to speaker/headphone/line-out amps. This makes snd-aoa handle it, if present in the device tree, thus making snd-aoa be able to output sound on PowerMac3,6, which was previously handled by snd-powermac which also doesn't use the master amp GPIO. Signed-off-by: Johannes Berg Signed-off-by: Takashi Iwai --- sound/aoa/aoa-gpio.h | 2 ++ sound/aoa/core/gpio-feature.c | 17 ++++++++++++++++- sound/aoa/fabrics/layout.c | 7 +++++++ 3 files changed, 25 insertions(+), 1 deletion(-) diff --git a/sound/aoa/aoa-gpio.h b/sound/aoa/aoa-gpio.h index ee64f5de896..6065b0344e2 100644 --- a/sound/aoa/aoa-gpio.h +++ b/sound/aoa/aoa-gpio.h @@ -34,10 +34,12 @@ struct gpio_methods { void (*set_headphone)(struct gpio_runtime *rt, int on); void (*set_speakers)(struct gpio_runtime *rt, int on); void (*set_lineout)(struct gpio_runtime *rt, int on); + void (*set_master)(struct gpio_runtime *rt, int on); int (*get_headphone)(struct gpio_runtime *rt); int (*get_speakers)(struct gpio_runtime *rt); int (*get_lineout)(struct gpio_runtime *rt); + int (*get_master)(struct gpio_runtime *rt); void (*set_hw_reset)(struct gpio_runtime *rt, int on); diff --git a/sound/aoa/core/gpio-feature.c b/sound/aoa/core/gpio-feature.c index c93ad5dec66..de8e03afa97 100644 --- a/sound/aoa/core/gpio-feature.c +++ b/sound/aoa/core/gpio-feature.c @@ -14,7 +14,7 @@ #include #include "../aoa.h" -/* TODO: these are 20 global variables +/* TODO: these are lots of global variables * that aren't used on most machines... * Move them into a dynamically allocated * structure and use that. @@ -23,6 +23,7 @@ /* these are the GPIO numbers (register addresses as offsets into * the GPIO space) */ static int headphone_mute_gpio; +static int master_mute_gpio; static int amp_mute_gpio; static int lineout_mute_gpio; static int hw_reset_gpio; @@ -32,6 +33,7 @@ static int linein_detect_gpio; /* see the SWITCH_GPIO macro */ static int headphone_mute_gpio_activestate; +static int master_mute_gpio_activestate; static int amp_mute_gpio_activestate; static int lineout_mute_gpio_activestate; static int hw_reset_gpio_activestate; @@ -156,6 +158,7 @@ static int ftr_gpio_get_##name(struct gpio_runtime *rt) \ FTR_GPIO(headphone, 0); FTR_GPIO(amp, 1); FTR_GPIO(lineout, 2); +FTR_GPIO(master, 3); static void ftr_gpio_set_hw_reset(struct gpio_runtime *rt, int on) { @@ -172,6 +175,8 @@ static void ftr_gpio_set_hw_reset(struct gpio_runtime *rt, int on) hw_reset_gpio, v); } +static struct gpio_methods methods; + static void ftr_gpio_all_amps_off(struct gpio_runtime *rt) { int saved; @@ -181,6 +186,8 @@ static void ftr_gpio_all_amps_off(struct gpio_runtime *rt) ftr_gpio_set_headphone(rt, 0); ftr_gpio_set_amp(rt, 0); ftr_gpio_set_lineout(rt, 0); + if (methods.set_master) + ftr_gpio_set_master(rt, 0); rt->implementation_private = saved; } @@ -193,6 +200,8 @@ static void ftr_gpio_all_amps_restore(struct gpio_runtime *rt) ftr_gpio_set_headphone(rt, (s>>0)&1); ftr_gpio_set_amp(rt, (s>>1)&1); ftr_gpio_set_lineout(rt, (s>>2)&1); + if (methods.set_master) + ftr_gpio_set_master(rt, (s>>3)&1); } static void ftr_handle_notify(struct work_struct *work) @@ -231,6 +240,12 @@ static void ftr_gpio_init(struct gpio_runtime *rt) get_gpio("hw-reset", "audio-hw-reset", &hw_reset_gpio, &hw_reset_gpio_activestate); + if (get_gpio("master-mute", NULL, + &master_mute_gpio, + &master_mute_gpio_activestate)) { + methods.set_master = ftr_gpio_set_master; + methods.get_master = ftr_gpio_get_master; + } headphone_detect_node = get_gpio("headphone-detect", NULL, &headphone_detect_gpio, diff --git a/sound/aoa/fabrics/layout.c b/sound/aoa/fabrics/layout.c index d9b1d22a62c..fbf5c933baa 100644 --- a/sound/aoa/fabrics/layout.c +++ b/sound/aoa/fabrics/layout.c @@ -600,6 +600,7 @@ struct layout_dev { struct snd_kcontrol *headphone_ctrl; struct snd_kcontrol *lineout_ctrl; struct snd_kcontrol *speaker_ctrl; + struct snd_kcontrol *master_ctrl; struct snd_kcontrol *headphone_detected_ctrl; struct snd_kcontrol *lineout_detected_ctrl; @@ -651,6 +652,7 @@ static struct snd_kcontrol_new n##_ctl = { \ AMP_CONTROL(headphone, "Headphone Switch"); AMP_CONTROL(speakers, "Speakers Switch"); AMP_CONTROL(lineout, "Line-Out Switch"); +AMP_CONTROL(master, "Master Switch"); static int detect_choice_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -891,6 +893,11 @@ static void layout_attached_codec(struct aoa_codec *codec) lineout = codec->gpio->methods->get_detect(codec->gpio, AOA_NOTIFY_LINE_OUT); + if (codec->gpio->methods->set_master) { + ctl = snd_ctl_new1(&master_ctl, codec->gpio); + ldev->master_ctrl = ctl; + aoa_snd_ctl_add(ctl); + } while (cc->connected) { if (cc->connected & CC_SPEAKERS) { if (headphones <= 0 && lineout <= 0) -- cgit v1.2.3-70-g09d2 From ac37373b6463d32955c6ac6b753d5e5b0946a791 Mon Sep 17 00:00:00 2001 From: Hugo Villeneuve Date: Thu, 15 Jan 2009 15:40:35 -0500 Subject: ASoC: DaVinci: Fix SFFSDR compilation error. Remove dependency on sffsdr_fpga_set_codec_fs() when the SFFSDR FPGA module is not selected. Signed-off-by: Hugo Villeneuve Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-sffsdr.c | 20 +++++++++++++++++--- 1 file changed, 17 insertions(+), 3 deletions(-) diff --git a/sound/soc/davinci/davinci-sffsdr.c b/sound/soc/davinci/davinci-sffsdr.c index 4935d1bcbd8..50baef1fe5b 100644 --- a/sound/soc/davinci/davinci-sffsdr.c +++ b/sound/soc/davinci/davinci-sffsdr.c @@ -25,7 +25,9 @@ #include #include +#ifdef CONFIG_SFFSDR_FPGA #include +#endif #include #include @@ -43,6 +45,17 @@ static int sffsdr_hw_params(struct snd_pcm_substream *substream, int fs; int ret = 0; + /* Fsref can be 32000, 44100 or 48000. */ + fs = params_rate(params); + +#ifndef CONFIG_SFFSDR_FPGA + /* Without the FPGA module, the Fs is fixed at 44100 Hz */ + if (fs != 44100) { + pr_debug("warning: only 44.1 kHz is supported without SFFSDR FPGA module\n"); + return -EINVAL; + } +#endif + /* Set cpu DAI configuration: * CLKX and CLKR are the inputs for the Sample Rate Generator. * FSX and FSR are outputs, driven by the sample Rate Generator. */ @@ -53,12 +66,13 @@ static int sffsdr_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - /* Fsref can be 32000, 44100 or 48000. */ - fs = params_rate(params); - pr_debug("sffsdr_hw_params: rate = %d Hz\n", fs); +#ifndef CONFIG_SFFSDR_FPGA + return 0; +#else return sffsdr_fpga_set_codec_fs(fs); +#endif } static struct snd_soc_ops sffsdr_ops = { -- cgit v1.2.3-70-g09d2 From 2165592b837e086f2b94835a2d81e6f3199c1319 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Fri, 16 Jan 2009 11:03:19 +0100 Subject: ALSA: snd-usb-caiaq: support for two more audio devices - Added support for two new audio devices from Native Instuments, 'Audio4DJ' and 'GuitarRig mobile' - Add missing statement about 'Session IO' in Kconfig help text - Version number bumped to 1.3.11 Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/Kconfig | 3 +++ sound/usb/caiaq/caiaq-audio.c | 5 +++-- sound/usb/caiaq/caiaq-control.c | 15 ++++++++++++--- sound/usb/caiaq/caiaq-device.c | 16 ++++++++++++++-- sound/usb/caiaq/caiaq-device.h | 2 ++ 5 files changed, 34 insertions(+), 7 deletions(-) diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index 4f0eac9bff1..523aec188cc 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -48,7 +48,10 @@ config SND_USB_CAIAQ * Native Instruments Kore Controller * Native Instruments Kore Controller 2 * Native Instruments Audio Kontrol 1 + * Native Instruments Audio 4 DJ * Native Instruments Audio 8 DJ + * Native Instruments Guitar Rig Session I/O + * Native Instruments Guitar Rig mobile To compile this driver as a module, choose M here: the module will be called snd-usb-caiaq. diff --git a/sound/usb/caiaq/caiaq-audio.c b/sound/usb/caiaq/caiaq-audio.c index b3a60332583..fc6d571eeac 100644 --- a/sound/usb/caiaq/caiaq-audio.c +++ b/sound/usb/caiaq/caiaq-audio.c @@ -638,9 +638,10 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AK1): case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL3): case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_SESSIONIO): - dev->samplerates |= SNDRV_PCM_RATE_88200; + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_GUITARRIGMOBILE): dev->samplerates |= SNDRV_PCM_RATE_192000; - break; + /* fall thru */ + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ): case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ): dev->samplerates |= SNDRV_PCM_RATE_88200; break; diff --git a/sound/usb/caiaq/caiaq-control.c b/sound/usb/caiaq/caiaq-control.c index ccd763dd716..6ac5489a0f2 100644 --- a/sound/usb/caiaq/caiaq-control.c +++ b/sound/usb/caiaq/caiaq-control.c @@ -39,14 +39,15 @@ static int control_info(struct snd_kcontrol *kcontrol, struct snd_usb_caiaqdev *dev = caiaqdev(chip->card); int pos = kcontrol->private_value; int is_intval = pos & CNT_INTVAL; + unsigned int id = dev->chip.usb_id; uinfo->count = 1; pos &= ~CNT_INTVAL; - if (dev->chip.usb_id == - USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ) + if (((id == USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ)) || + (id == USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ))) && (pos == 0)) { - /* current input mode of A8DJ */ + /* current input mode of A8DJ and A4DJ */ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->value.integer.min = 0; uinfo->value.integer.max = 2; @@ -247,6 +248,10 @@ static struct caiaq_controller a8dj_controller[] = { { "Software lock", 40 } }; +static struct caiaq_controller a4dj_controller[] = { + { "Current input mode", 0 | CNT_INTVAL } +}; + static int __devinit add_controls(struct caiaq_controller *c, int num, struct snd_usb_caiaqdev *dev) { @@ -295,6 +300,10 @@ int __devinit snd_usb_caiaq_control_init(struct snd_usb_caiaqdev *dev) ret = add_controls(a8dj_controller, ARRAY_SIZE(a8dj_controller), dev); break; + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ): + ret = add_controls(a4dj_controller, + ARRAY_SIZE(a4dj_controller), dev); + break; } return ret; diff --git a/sound/usb/caiaq/caiaq-device.c b/sound/usb/caiaq/caiaq-device.c index 41c36b055f6..d09fc2a88cf 100644 --- a/sound/usb/caiaq/caiaq-device.c +++ b/sound/usb/caiaq/caiaq-device.c @@ -42,15 +42,17 @@ #endif MODULE_AUTHOR("Daniel Mack "); -MODULE_DESCRIPTION("caiaq USB audio, version 1.3.10"); +MODULE_DESCRIPTION("caiaq USB audio, version 1.3.11"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," "{Native Instruments, RigKontrol3}," "{Native Instruments, Kore Controller}," "{Native Instruments, Kore Controller 2}," "{Native Instruments, Audio Kontrol 1}," + "{Native Instruments, Audio 4 DJ}," "{Native Instruments, Audio 8 DJ}," - "{Native Instruments, Session I/O}}"); + "{Native Instruments, Session I/O}," + "{Native Instruments, GuitarRig mobile}"); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-max */ static char* id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* Id for this card */ @@ -116,6 +118,16 @@ static struct usb_device_id snd_usb_id_table[] = { .idVendor = USB_VID_NATIVEINSTRUMENTS, .idProduct = USB_PID_SESSIONIO }, + { + .match_flags = USB_DEVICE_ID_MATCH_DEVICE, + .idVendor = USB_VID_NATIVEINSTRUMENTS, + .idProduct = USB_PID_GUITARRIGMOBILE + }, + { + .match_flags = USB_DEVICE_ID_MATCH_DEVICE, + .idVendor = USB_VID_NATIVEINSTRUMENTS, + .idProduct = USB_PID_AUDIO4DJ + }, { /* terminator */ } }; diff --git a/sound/usb/caiaq/caiaq-device.h b/sound/usb/caiaq/caiaq-device.h index ab56e738c5f..0560c327d99 100644 --- a/sound/usb/caiaq/caiaq-device.h +++ b/sound/usb/caiaq/caiaq-device.h @@ -10,8 +10,10 @@ #define USB_PID_KORECONTROLLER 0x4711 #define USB_PID_KORECONTROLLER2 0x4712 #define USB_PID_AK1 0x0815 +#define USB_PID_AUDIO4DJ 0x0839 #define USB_PID_AUDIO8DJ 0x1978 #define USB_PID_SESSIONIO 0x1915 +#define USB_PID_GUITARRIGMOBILE 0x0d8d #define EP1_BUFSIZE 64 #define CAIAQ_USB_STR_LEN 0xff -- cgit v1.2.3-70-g09d2 From 2aceefefc891e85d336c1d95d9d89fd785f5d44c Mon Sep 17 00:00:00 2001 From: Ian Molton Date: Fri, 16 Jan 2009 11:04:18 +0000 Subject: ASoC: Driver for the WM9705 AC97 codec. This driver adds support for the wm9705 ac97 codec. The driver supports audio input and output. Signed-off-by: Ian Molton Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 3 + sound/soc/codecs/wm9705.c | 410 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm9705.h | 14 ++ 4 files changed, 431 insertions(+) create mode 100644 sound/soc/codecs/wm9705.c create mode 100644 sound/soc/codecs/wm9705.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index d0e0d691ae5..cb5fcd605ac 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -34,6 +34,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8903 if I2C select SND_SOC_WM8971 if I2C select SND_SOC_WM8990 if I2C + select SND_SOC_WM9705 if SND_SOC_AC97_BUS select SND_SOC_WM9712 if SND_SOC_AC97_BUS select SND_SOC_WM9713 if SND_SOC_AC97_BUS help @@ -144,6 +145,9 @@ config SND_SOC_WM8971 config SND_SOC_WM8990 tristate +config SND_SOC_WM9705 + tristate + config SND_SOC_WM9712 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index c4ddc9aa2bb..3664cdc300b 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -23,6 +23,7 @@ snd-soc-wm8900-objs := wm8900.o snd-soc-wm8903-objs := wm8903.o snd-soc-wm8971-objs := wm8971.o snd-soc-wm8990-objs := wm8990.o +snd-soc-wm9705-objs := wm9705.o snd-soc-wm9712-objs := wm9712.o snd-soc-wm9713-objs := wm9713.o @@ -51,5 +52,7 @@ obj-$(CONFIG_SND_SOC_WM8900) += snd-soc-wm8900.o obj-$(CONFIG_SND_SOC_WM8903) += snd-soc-wm8903.o obj-$(CONFIG_SND_SOC_WM8971) += snd-soc-wm8971.o obj-$(CONFIG_SND_SOC_WM8990) += snd-soc-wm8990.o +obj-$(CONFIG_SND_SOC_WM8991) += snd-soc-wm8991.o +obj-$(CONFIG_SND_SOC_WM9705) += snd-soc-wm9705.o obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c new file mode 100644 index 00000000000..cb26b6a77ff --- /dev/null +++ b/sound/soc/codecs/wm9705.c @@ -0,0 +1,410 @@ +/* + * wm9705.c -- ALSA Soc WM9705 codec support + * + * Copyright 2008 Ian Molton + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; Version 2 of the License only. + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +/* + * WM9705 register cache + */ +static const u16 wm9705_reg[] = { + 0x6150, 0x8000, 0x8000, 0x8000, /* 0x0 */ + 0x0000, 0x8000, 0x8008, 0x8008, /* 0x8 */ + 0x8808, 0x8808, 0x8808, 0x8808, /* 0x10 */ + 0x8808, 0x0000, 0x8000, 0x0000, /* 0x18 */ + 0x0000, 0x0000, 0x0000, 0x000f, /* 0x20 */ + 0x0605, 0x0000, 0xbb80, 0x0000, /* 0x28 */ + 0x0000, 0xbb80, 0x0000, 0x0000, /* 0x30 */ + 0x0000, 0x2000, 0x0000, 0x0000, /* 0x38 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 0x40 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 0x48 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 0x50 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 0x58 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 0x60 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 0x68 */ + 0x0000, 0x0808, 0x0000, 0x0006, /* 0x70 */ + 0x0000, 0x0000, 0x574d, 0x4c05, /* 0x78 */ +}; + +static const struct snd_kcontrol_new wm9705_snd_ac97_controls[] = { + SOC_DOUBLE("Master Playback Volume", AC97_MASTER, 8, 0, 31, 1), + SOC_SINGLE("Master Playback Switch", AC97_MASTER, 15, 1, 1), + SOC_DOUBLE("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1), + SOC_SINGLE("Headphone Playback Switch", AC97_HEADPHONE, 15, 1, 1), + SOC_DOUBLE("PCM Playback Volume", AC97_PCM, 8, 0, 31, 1), + SOC_SINGLE("PCM Playback Switch", AC97_PCM, 15, 1, 1), + SOC_SINGLE("Mono Playback Volume", AC97_MASTER_MONO, 0, 31, 1), + SOC_SINGLE("Mono Playback Switch", AC97_MASTER_MONO, 15, 1, 1), + SOC_SINGLE("PCBeep Playback Volume", AC97_PC_BEEP, 1, 15, 1), + SOC_SINGLE("Phone Playback Volume", AC97_PHONE, 0, 31, 1), + SOC_DOUBLE("Line Playback Volume", AC97_LINE, 8, 0, 31, 1), + SOC_DOUBLE("CD Playback Volume", AC97_CD, 8, 0, 31, 1), + SOC_SINGLE("Mic Playback Volume", AC97_MIC, 0, 31, 1), + SOC_SINGLE("Mic 20dB Boost Switch", AC97_MIC, 6, 1, 0), + SOC_DOUBLE("PCM Capture Volume", AC97_REC_GAIN, 8, 0, 15, 0), + SOC_SINGLE("PCM Capture Switch", AC97_REC_GAIN, 15, 1, 1), +}; + +static const char *wm9705_mic[] = {"Mic 1", "Mic 2"}; +static const char *wm9705_rec_sel[] = {"Mic", "CD", "NC", "NC", + "Line", "Stereo Mix", "Mono Mix", "Phone"}; + +static const struct soc_enum wm9705_enum_mic = + SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 8, 2, wm9705_mic); +static const struct soc_enum wm9705_enum_rec_l = + SOC_ENUM_SINGLE(AC97_REC_SEL, 8, 8, wm9705_rec_sel); +static const struct soc_enum wm9705_enum_rec_r = + SOC_ENUM_SINGLE(AC97_REC_SEL, 0, 8, wm9705_rec_sel); + +/* Headphone Mixer */ +static const struct snd_kcontrol_new wm9705_hp_mixer_controls[] = { + SOC_DAPM_SINGLE("PCBeep Playback Switch", AC97_PC_BEEP, 15, 1, 1), + SOC_DAPM_SINGLE("CD Playback Switch", AC97_CD, 15, 1, 1), + SOC_DAPM_SINGLE("Mic Playback Switch", AC97_MIC, 15, 1, 1), + SOC_DAPM_SINGLE("Phone Playback Switch", AC97_PHONE, 15, 1, 1), + SOC_DAPM_SINGLE("Line Playback Switch", AC97_LINE, 15, 1, 1), +}; + +/* Mic source */ +static const struct snd_kcontrol_new wm9705_mic_src_controls = + SOC_DAPM_ENUM("Route", wm9705_enum_mic); + +/* Capture source */ +static const struct snd_kcontrol_new wm9705_capture_selectl_controls = + SOC_DAPM_ENUM("Route", wm9705_enum_rec_l); +static const struct snd_kcontrol_new wm9705_capture_selectr_controls = + SOC_DAPM_ENUM("Route", wm9705_enum_rec_r); + +/* DAPM widgets */ +static const struct snd_soc_dapm_widget wm9705_dapm_widgets[] = { + SND_SOC_DAPM_MUX("Mic Source", SND_SOC_NOPM, 0, 0, + &wm9705_mic_src_controls), + SND_SOC_DAPM_MUX("Left Capture Source", SND_SOC_NOPM, 0, 0, + &wm9705_capture_selectl_controls), + SND_SOC_DAPM_MUX("Right Capture Source", SND_SOC_NOPM, 0, 0, + &wm9705_capture_selectr_controls), + SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback", + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback", + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_MIXER_NAMED_CTL("HP Mixer", SND_SOC_NOPM, 0, 0, + &wm9705_hp_mixer_controls[0], + ARRAY_SIZE(wm9705_hp_mixer_controls)), + SND_SOC_DAPM_MIXER("Mono Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_PGA("Headphone PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Speaker PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Line PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Line out PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Mono PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Phone PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Mic PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PCBEEP PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("CD PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("ADC PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("HPOUTL"), + SND_SOC_DAPM_OUTPUT("HPOUTR"), + SND_SOC_DAPM_OUTPUT("LOUT"), + SND_SOC_DAPM_OUTPUT("ROUT"), + SND_SOC_DAPM_OUTPUT("MONOOUT"), + SND_SOC_DAPM_INPUT("PHONE"), + SND_SOC_DAPM_INPUT("LINEINL"), + SND_SOC_DAPM_INPUT("LINEINR"), + SND_SOC_DAPM_INPUT("CDINL"), + SND_SOC_DAPM_INPUT("CDINR"), + SND_SOC_DAPM_INPUT("PCBEEP"), + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), +}; + +/* Audio map + * WM9705 has no switches to disable the route from the inputs to the HP mixer + * so in order to prevent active inputs from forcing the audio outputs to be + * constantly enabled, we use the mutes on those inputs to simulate such + * controls. + */ +static const struct snd_soc_dapm_route audio_map[] = { + /* HP mixer */ + {"HP Mixer", "PCBeep Playback Switch", "PCBEEP PGA"}, + {"HP Mixer", "CD Playback Switch", "CD PGA"}, + {"HP Mixer", "Mic Playback Switch", "Mic PGA"}, + {"HP Mixer", "Phone Playback Switch", "Phone PGA"}, + {"HP Mixer", "Line Playback Switch", "Line PGA"}, + {"HP Mixer", NULL, "Left DAC"}, + {"HP Mixer", NULL, "Right DAC"}, + + /* mono mixer */ + {"Mono Mixer", NULL, "HP Mixer"}, + + /* outputs */ + {"Headphone PGA", NULL, "HP Mixer"}, + {"HPOUTL", NULL, "Headphone PGA"}, + {"HPOUTR", NULL, "Headphone PGA"}, + {"Line out PGA", NULL, "HP Mixer"}, + {"LOUT", NULL, "Line out PGA"}, + {"ROUT", NULL, "Line out PGA"}, + {"Mono PGA", NULL, "Mono Mixer"}, + {"MONOOUT", NULL, "Mono PGA"}, + + /* inputs */ + {"CD PGA", NULL, "CDINL"}, + {"CD PGA", NULL, "CDINR"}, + {"Line PGA", NULL, "LINEINL"}, + {"Line PGA", NULL, "LINEINR"}, + {"Phone PGA", NULL, "PHONE"}, + {"Mic Source", "Mic 1", "MIC1"}, + {"Mic Source", "Mic 2", "MIC2"}, + {"Mic PGA", NULL, "Mic Source"}, + {"PCBEEP PGA", NULL, "PCBEEP"}, + + /* Left capture selector */ + {"Left Capture Source", "Mic", "Mic Source"}, + {"Left Capture Source", "CD", "CDINL"}, + {"Left Capture Source", "Line", "LINEINL"}, + {"Left Capture Source", "Stereo Mix", "HP Mixer"}, + {"Left Capture Source", "Mono Mix", "HP Mixer"}, + {"Left Capture Source", "Phone", "PHONE"}, + + /* Right capture source */ + {"Right Capture Source", "Mic", "Mic Source"}, + {"Right Capture Source", "CD", "CDINR"}, + {"Right Capture Source", "Line", "LINEINR"}, + {"Right Capture Source", "Stereo Mix", "HP Mixer"}, + {"Right Capture Source", "Mono Mix", "HP Mixer"}, + {"Right Capture Source", "Phone", "PHONE"}, + + {"ADC PGA", NULL, "Left Capture Source"}, + {"ADC PGA", NULL, "Right Capture Source"}, + + /* ADC's */ + {"Left ADC", NULL, "ADC PGA"}, + {"Right ADC", NULL, "ADC PGA"}, +}; + +static int wm9705_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, wm9705_dapm_widgets, + ARRAY_SIZE(wm9705_dapm_widgets)); + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_new_widgets(codec); + + return 0; +} + +/* We use a register cache to enhance read performance. */ +static unsigned int ac97_read(struct snd_soc_codec *codec, unsigned int reg) +{ + u16 *cache = codec->reg_cache; + + switch (reg) { + case AC97_RESET: + case AC97_VENDOR_ID1: + case AC97_VENDOR_ID2: + return soc_ac97_ops.read(codec->ac97, reg); + default: + reg = reg >> 1; + + if (reg >= (ARRAY_SIZE(wm9705_reg))) + return -EIO; + + return cache[reg]; + } +} + +static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int val) +{ + u16 *cache = codec->reg_cache; + + soc_ac97_ops.write(codec->ac97, reg, val); + reg = reg >> 1; + if (reg < (ARRAY_SIZE(wm9705_reg))) + cache[reg] = val; + + return 0; +} + +static int ac97_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + int reg; + u16 vra; + + vra = ac97_read(codec, AC97_EXTENDED_STATUS); + ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + reg = AC97_PCM_FRONT_DAC_RATE; + else + reg = AC97_PCM_LR_ADC_RATE; + + return ac97_write(codec, reg, runtime->rate); +} + +#define WM9705_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | \ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000) + +struct snd_soc_dai wm9705_dai[] = { + { + .name = "AC97 HiFi", + .ac97_control = 1, + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM9705_AC97_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "HiFi Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM9705_AC97_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = { + .prepare = ac97_prepare, + }, + }, + { + .name = "AC97 Aux", + .playback = { + .stream_name = "Aux Playback", + .channels_min = 1, + .channels_max = 1, + .rates = WM9705_AC97_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + } +}; +EXPORT_SYMBOL_GPL(wm9705_dai); + +static int wm9705_reset(struct snd_soc_codec *codec) +{ + if (soc_ac97_ops.reset) { + soc_ac97_ops.reset(codec->ac97); + if (ac97_read(codec, 0) == wm9705_reg[0]) + return 0; /* Success */ + } + + return -EIO; +} + +static int wm9705_soc_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + printk(KERN_INFO "WM9705 SoC Audio Codec\n"); + + socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (socdev->codec == NULL) + return -ENOMEM; + codec = socdev->codec; + mutex_init(&codec->mutex); + + codec->reg_cache = kmemdup(wm9705_reg, sizeof(wm9705_reg), GFP_KERNEL); + if (codec->reg_cache == NULL) { + ret = -ENOMEM; + goto cache_err; + } + codec->reg_cache_size = sizeof(wm9705_reg); + codec->reg_cache_step = 2; + + codec->name = "WM9705"; + codec->owner = THIS_MODULE; + codec->dai = wm9705_dai; + codec->num_dai = ARRAY_SIZE(wm9705_dai); + codec->write = ac97_write; + codec->read = ac97_read; + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); + if (ret < 0) { + printk(KERN_ERR "wm9705: failed to register AC97 codec\n"); + goto codec_err; + } + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) + goto pcm_err; + + ret = wm9705_reset(codec); + if (ret) + goto reset_err; + + snd_soc_add_controls(codec, wm9705_snd_ac97_controls, + ARRAY_SIZE(wm9705_snd_ac97_controls)); + wm9705_add_widgets(codec); + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + printk(KERN_ERR "wm9705: failed to register card\n"); + goto pcm_err; + } + + return 0; + +reset_err: + snd_soc_free_pcms(socdev); +pcm_err: + snd_soc_free_ac97_codec(codec); +codec_err: + kfree(codec->reg_cache); +cache_err: + kfree(socdev->codec); + socdev->codec = NULL; + return ret; +} + +static int wm9705_soc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + if (codec == NULL) + return 0; + + snd_soc_dapm_free(socdev); + snd_soc_free_pcms(socdev); + snd_soc_free_ac97_codec(codec); + kfree(codec->reg_cache); + kfree(codec); + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm9705 = { + .probe = wm9705_soc_probe, + .remove = wm9705_soc_remove, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm9705); + +MODULE_DESCRIPTION("ASoC WM9705 driver"); +MODULE_AUTHOR("Ian Molton"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/wm9705.h b/sound/soc/codecs/wm9705.h new file mode 100644 index 00000000000..d380f110f9e --- /dev/null +++ b/sound/soc/codecs/wm9705.h @@ -0,0 +1,14 @@ +/* + * wm9705.h -- WM9705 Soc Audio driver + */ + +#ifndef _WM9705_H +#define _WM9705_H + +#define WM9705_DAI_AC97_HIFI 0 +#define WM9705_DAI_AC97_AUX 1 + +extern struct snd_soc_dai wm9705_dai[2]; +extern struct snd_soc_codec_device soc_codec_dev_wm9705; + +#endif -- cgit v1.2.3-70-g09d2 From a7e2e735dcf98717150d3c8eaa731de8038af05a Mon Sep 17 00:00:00 2001 From: Ian Molton Date: Thu, 8 Jan 2009 21:03:55 +0000 Subject: ASoC: machine driver for Toshiba e750 This patch adds support for the wm9705 ac97 codec as used in the Toshiba e750 PDA. It includes support for powering up / down the external headphone and speaker amplifiers on this machine. Signed-off-by: Ian Molton Signed-off-by: Mark Brown --- arch/arm/mach-pxa/e750.c | 5 + arch/arm/mach-pxa/include/mach/eseries-gpio.h | 5 + sound/soc/pxa/Kconfig | 9 ++ sound/soc/pxa/Makefile | 2 + sound/soc/pxa/e750_wm9705.c | 189 ++++++++++++++++++++++++++ 5 files changed, 210 insertions(+) create mode 100644 sound/soc/pxa/e750_wm9705.c diff --git a/arch/arm/mach-pxa/e750.c b/arch/arm/mach-pxa/e750.c index be1ab8edb97..665066fd280 100644 --- a/arch/arm/mach-pxa/e750.c +++ b/arch/arm/mach-pxa/e750.c @@ -133,6 +133,11 @@ static unsigned long e750_pin_config[] __initdata = { /* IrDA */ GPIO38_GPIO | MFP_LPM_DRIVE_HIGH, + /* Audio power control */ + GPIO4_GPIO, /* Headphone amp power */ + GPIO7_GPIO, /* Speaker amp power */ + GPIO37_GPIO, /* Headphone detect */ + /* PC Card */ GPIO8_GPIO, /* CD0 */ GPIO44_GPIO, /* CD1 */ diff --git a/arch/arm/mach-pxa/include/mach/eseries-gpio.h b/arch/arm/mach-pxa/include/mach/eseries-gpio.h index efbd2aa9ece..02b28e0ed73 100644 --- a/arch/arm/mach-pxa/include/mach/eseries-gpio.h +++ b/arch/arm/mach-pxa/include/mach/eseries-gpio.h @@ -45,6 +45,11 @@ /* e7xx IrDA power control */ #define GPIO_E7XX_IR_OFF 38 +/* e750 audio control GPIOs */ +#define GPIO_E750_HP_AMP_OFF 4 +#define GPIO_E750_SPK_AMP_OFF 7 +#define GPIO_E750_HP_DETECT 37 + /* ASIC related GPIOs */ #define GPIO_ESERIES_TMIO_IRQ 5 #define GPIO_ESERIES_TMIO_PCLR 19 diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index f82e1069947..b9b1a3f5d67 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -61,6 +61,15 @@ config SND_PXA2XX_SOC_TOSA Say Y if you want to add support for SoC audio on Sharp Zaurus SL-C6000x models (Tosa). +config SND_PXA2XX_SOC_E750 + tristate "SoC AC97 Audio support for e750" + depends on SND_PXA2XX_SOC && MACH_E750 + select SND_SOC_WM9705 + select SND_PXA2XX_SOC_AC97 + help + Say Y if you want to add support for SoC audio on the + toshiba e750 PDA + config SND_PXA2XX_SOC_E800 tristate "SoC AC97 Audio support for e800" depends on SND_PXA2XX_SOC && MACH_E800 diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 08a9f279772..c7d4cceeed9 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -13,6 +13,7 @@ obj-$(CONFIG_SND_PXA_SOC_SSP) += snd-soc-pxa-ssp.o snd-soc-corgi-objs := corgi.o snd-soc-poodle-objs := poodle.o snd-soc-tosa-objs := tosa.o +snd-soc-e750-objs := e750_wm9705.o snd-soc-e800-objs := e800_wm9712.o snd-soc-spitz-objs := spitz.o snd-soc-em-x270-objs := em-x270.o @@ -22,6 +23,7 @@ snd-soc-zylonite-objs := zylonite.o obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o obj-$(CONFIG_SND_PXA2XX_SOC_TOSA) += snd-soc-tosa.o +obj-$(CONFIG_SND_PXA2XX_SOC_E750) += snd-soc-e750.o obj-$(CONFIG_SND_PXA2XX_SOC_E800) += snd-soc-e800.o obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o obj-$(CONFIG_SND_PXA2XX_SOC_EM_X270) += snd-soc-em-x270.o diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c new file mode 100644 index 00000000000..20fbdcfa9f7 --- /dev/null +++ b/sound/soc/pxa/e750_wm9705.c @@ -0,0 +1,189 @@ +/* + * e750-wm9705.c -- SoC audio for e750 + * + * Copyright 2007 (c) Ian Molton + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; version 2 ONLY. + * + */ + +#include +#include +#include + +#include +#include +#include +#include + +#include +#include +#include +#include + +#include + +#include "../codecs/wm9705.h" +#include "pxa2xx-pcm.h" +#include "pxa2xx-ac97.h" + +static int e750_spk_amp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + if (event & SND_SOC_DAPM_PRE_PMU) + gpio_set_value(GPIO_E750_SPK_AMP_OFF, 0); + else if (event & SND_SOC_DAPM_POST_PMD) + gpio_set_value(GPIO_E750_SPK_AMP_OFF, 1); + + return 0; +} + +static int e750_hp_amp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + if (event & SND_SOC_DAPM_PRE_PMU) + gpio_set_value(GPIO_E750_HP_AMP_OFF, 0); + else if (event & SND_SOC_DAPM_POST_PMD) + gpio_set_value(GPIO_E750_HP_AMP_OFF, 1); + + return 0; +} + +static const struct snd_soc_dapm_widget e750_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_MIC("Mic (Internal)", NULL), + SND_SOC_DAPM_PGA_E("Headphone Amp", SND_SOC_NOPM, 0, 0, NULL, 0, + e750_hp_amp_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PGA_E("Speaker Amp", SND_SOC_NOPM, 0, 0, NULL, 0, + e750_spk_amp_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + {"Headphone Amp", NULL, "HPOUTL"}, + {"Headphone Amp", NULL, "HPOUTR"}, + {"Headphone Jack", NULL, "Headphone Amp"}, + + {"Speaker Amp", NULL, "MONOOUT"}, + {"Speaker", NULL, "Speaker Amp"}, + + {"MIC1", NULL, "Mic (Internal)"}, +}; + +static int e750_ac97_init(struct snd_soc_codec *codec) +{ + snd_soc_dapm_nc_pin(codec, "LOUT"); + snd_soc_dapm_nc_pin(codec, "ROUT"); + snd_soc_dapm_nc_pin(codec, "PHONE"); + snd_soc_dapm_nc_pin(codec, "LINEINL"); + snd_soc_dapm_nc_pin(codec, "LINEINR"); + snd_soc_dapm_nc_pin(codec, "CDINL"); + snd_soc_dapm_nc_pin(codec, "CDINR"); + snd_soc_dapm_nc_pin(codec, "PCBEEP"); + snd_soc_dapm_nc_pin(codec, "MIC2"); + + snd_soc_dapm_new_controls(codec, e750_dapm_widgets, + ARRAY_SIZE(e750_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link e750_dai[] = { + { + .name = "AC97", + .stream_name = "AC97 HiFi", + .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI], + .codec_dai = &wm9705_dai[WM9705_DAI_AC97_HIFI], + .init = e750_ac97_init, + /* use ops to check startup state */ + }, + { + .name = "AC97 Aux", + .stream_name = "AC97 Aux", + .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX], + .codec_dai = &wm9705_dai[WM9705_DAI_AC97_AUX], + }, +}; + +static struct snd_soc_card e750 = { + .name = "Toshiba e750", + .platform = &pxa2xx_soc_platform, + .dai_link = e750_dai, + .num_links = ARRAY_SIZE(e750_dai), +}; + +static struct snd_soc_device e750_snd_devdata = { + .card = &e750, + .codec_dev = &soc_codec_dev_wm9705, +}; + +static struct platform_device *e750_snd_device; + +static int __init e750_init(void) +{ + int ret; + + if (!machine_is_e750()) + return -ENODEV; + + ret = gpio_request(GPIO_E750_HP_AMP_OFF, "Headphone amp"); + if (ret) + return ret; + + ret = gpio_request(GPIO_E750_SPK_AMP_OFF, "Speaker amp"); + if (ret) + goto free_hp_amp_gpio; + + ret = gpio_direction_output(GPIO_E750_HP_AMP_OFF, 1); + if (ret) + goto free_spk_amp_gpio; + + ret = gpio_direction_output(GPIO_E750_SPK_AMP_OFF, 1); + if (ret) + goto free_spk_amp_gpio; + + e750_snd_device = platform_device_alloc("soc-audio", -1); + if (!e750_snd_device) { + ret = -ENOMEM; + goto free_spk_amp_gpio; + } + + platform_set_drvdata(e750_snd_device, &e750_snd_devdata); + e750_snd_devdata.dev = &e750_snd_device->dev; + ret = platform_device_add(e750_snd_device); + + if (!ret) + return 0; + +/* Fail gracefully */ + platform_device_put(e750_snd_device); +free_spk_amp_gpio: + gpio_free(GPIO_E750_SPK_AMP_OFF); +free_hp_amp_gpio: + gpio_free(GPIO_E750_HP_AMP_OFF); + + return ret; +} + +static void __exit e750_exit(void) +{ + platform_device_unregister(e750_snd_device); + gpio_free(GPIO_E750_SPK_AMP_OFF); + gpio_free(GPIO_E750_HP_AMP_OFF); +} + +module_init(e750_init); +module_exit(e750_exit); + +/* Module information */ +MODULE_AUTHOR("Ian Molton "); +MODULE_DESCRIPTION("ALSA SoC driver for e750"); +MODULE_LICENSE("GPL v2"); -- cgit v1.2.3-70-g09d2 From 0465c7aa6fbab89de820442aed449ceb8d9145a6 Mon Sep 17 00:00:00 2001 From: Ian Molton Date: Thu, 8 Jan 2009 21:16:05 +0000 Subject: ASoC: machine driver for Toshiba e800 This patch adds support for the wm9712 ac97 codec as used in the Toshiba e800 PDA. It includes support for powering up / down the external headphone and speaker amplifiers on this machine. Signed-off-by: Ian Molton Signed-off-by: Mark Brown --- arch/arm/mach-pxa/include/mach/eseries-gpio.h | 5 ++ sound/soc/pxa/e800_wm9712.c | 116 ++++++++++++++++++++++---- 2 files changed, 107 insertions(+), 14 deletions(-) diff --git a/arch/arm/mach-pxa/include/mach/eseries-gpio.h b/arch/arm/mach-pxa/include/mach/eseries-gpio.h index 02b28e0ed73..6d6e4d8fa4c 100644 --- a/arch/arm/mach-pxa/include/mach/eseries-gpio.h +++ b/arch/arm/mach-pxa/include/mach/eseries-gpio.h @@ -50,6 +50,11 @@ #define GPIO_E750_SPK_AMP_OFF 7 #define GPIO_E750_HP_DETECT 37 +/* e800 audio control GPIOs */ +#define GPIO_E800_HP_DETECT 81 +#define GPIO_E800_HP_AMP_OFF 82 +#define GPIO_E800_SPK_AMP_ON 83 + /* ASIC related GPIOs */ #define GPIO_ESERIES_TMIO_IRQ 5 #define GPIO_ESERIES_TMIO_PCLR 19 diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c index 2e3386dfa0f..78a1770b986 100644 --- a/sound/soc/pxa/e800_wm9712.c +++ b/sound/soc/pxa/e800_wm9712.c @@ -1,8 +1,6 @@ /* * e800-wm9712.c -- SoC audio for e800 * - * Based on tosa.c - * * Copyright 2007 (c) Ian Molton * * This program is free software; you can redistribute it and/or modify it @@ -13,31 +11,96 @@ #include #include -#include +#include #include #include #include #include -#include #include #include #include +#include + +#include #include "../codecs/wm9712.h" #include "pxa2xx-pcm.h" #include "pxa2xx-ac97.h" -static struct snd_soc_card e800; +static int e800_spk_amp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + if (event & SND_SOC_DAPM_PRE_PMU) + gpio_set_value(GPIO_E800_SPK_AMP_ON, 1); + else if (event & SND_SOC_DAPM_POST_PMD) + gpio_set_value(GPIO_E800_SPK_AMP_ON, 0); -static struct snd_soc_dai_link e800_dai[] = { + return 0; +} + +static int e800_hp_amp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) { - .name = "AC97 Aux", - .stream_name = "AC97 Aux", - .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX], - .codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX], -}, + if (event & SND_SOC_DAPM_PRE_PMU) + gpio_set_value(GPIO_E800_HP_AMP_OFF, 0); + else if (event & SND_SOC_DAPM_POST_PMD) + gpio_set_value(GPIO_E800_HP_AMP_OFF, 1); + + return 0; +} + +static const struct snd_soc_dapm_widget e800_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Mic (Internal1)", NULL), + SND_SOC_DAPM_MIC("Mic (Internal2)", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_PGA_E("Headphone Amp", SND_SOC_NOPM, 0, 0, NULL, 0, + e800_hp_amp_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PGA_E("Speaker Amp", SND_SOC_NOPM, 0, 0, NULL, 0, + e800_spk_amp_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + {"Headphone Jack", NULL, "HPOUTL"}, + {"Headphone Jack", NULL, "HPOUTR"}, + {"Headphone Jack", NULL, "Headphone Amp"}, + + {"Speaker Amp", NULL, "MONOOUT"}, + {"Speaker", NULL, "Speaker Amp"}, + + {"MIC1", NULL, "Mic (Internal1)"}, + {"MIC2", NULL, "Mic (Internal2)"}, +}; + +static int e800_ac97_init(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, e800_dapm_widgets, + ARRAY_SIZE(e800_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link e800_dai[] = { + { + .name = "AC97", + .stream_name = "AC97 HiFi", + .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI], + .codec_dai = &wm9712_dai[WM9712_DAI_AC97_HIFI], + .init = e800_ac97_init, + }, + { + .name = "AC97 Aux", + .stream_name = "AC97 Aux", + .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX], + .codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX], + }, }; static struct snd_soc_card e800 = { @@ -61,6 +124,22 @@ static int __init e800_init(void) if (!machine_is_e800()) return -ENODEV; + ret = gpio_request(GPIO_E800_HP_AMP_OFF, "Headphone amp"); + if (ret) + return ret; + + ret = gpio_request(GPIO_E800_SPK_AMP_ON, "Speaker amp"); + if (ret) + goto free_hp_amp_gpio; + + ret = gpio_direction_output(GPIO_E800_HP_AMP_OFF, 1); + if (ret) + goto free_spk_amp_gpio; + + ret = gpio_direction_output(GPIO_E800_SPK_AMP_ON, 1); + if (ret) + goto free_spk_amp_gpio; + e800_snd_device = platform_device_alloc("soc-audio", -1); if (!e800_snd_device) return -ENOMEM; @@ -69,8 +148,15 @@ static int __init e800_init(void) e800_snd_devdata.dev = &e800_snd_device->dev; ret = platform_device_add(e800_snd_device); - if (ret) - platform_device_put(e800_snd_device); + if (!ret) + return 0; + +/* Fail gracefully */ + platform_device_put(e800_snd_device); +free_spk_amp_gpio: + gpio_free(GPIO_E800_SPK_AMP_ON); +free_hp_amp_gpio: + gpio_free(GPIO_E800_HP_AMP_OFF); return ret; } @@ -78,6 +164,8 @@ static int __init e800_init(void) static void __exit e800_exit(void) { platform_device_unregister(e800_snd_device); + gpio_free(GPIO_E800_SPK_AMP_ON); + gpio_free(GPIO_E800_HP_AMP_OFF); } module_init(e800_init); @@ -86,4 +174,4 @@ module_exit(e800_exit); /* Module information */ MODULE_AUTHOR("Ian Molton "); MODULE_DESCRIPTION("ALSA SoC driver for e800"); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); -- cgit v1.2.3-70-g09d2 From c42f69bb064333624dcc1452ed109441c3c9e7b4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 16 Jan 2009 16:31:03 +0000 Subject: ASoC: Ignore output frequency for WM9713 PLL The WM9713 driver does not support configuring the PLL output frequency so the output frequency parameter is irrelevant. Allow users to set it to zero by ignoring it. Signed-off-by: Mark Brown --- sound/soc/codecs/wm9713.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index a45622620db..e636d8a18ed 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -32,7 +32,6 @@ struct wm9713_priv { u32 pll_in; /* PLL input frequency */ - u32 pll_out; /* PLL output frequency */ }; static unsigned int ac97_read(struct snd_soc_codec *codec, @@ -723,13 +722,13 @@ static int wm9713_set_pll(struct snd_soc_codec *codec, struct _pll_div pll_div; /* turn PLL off ? */ - if (freq_in == 0 || freq_out == 0) { + if (freq_in == 0) { /* disable PLL power and select ext source */ reg = ac97_read(codec, AC97_HANDSET_RATE); ac97_write(codec, AC97_HANDSET_RATE, reg | 0x0080); reg = ac97_read(codec, AC97_EXTENDED_MID); ac97_write(codec, AC97_EXTENDED_MID, reg | 0x0200); - wm9713->pll_out = 0; + wm9713->pll_in = 0; return 0; } @@ -773,7 +772,6 @@ static int wm9713_set_pll(struct snd_soc_codec *codec, ac97_write(codec, AC97_EXTENDED_MID, reg & 0xfdff); reg = ac97_read(codec, AC97_HANDSET_RATE); ac97_write(codec, AC97_HANDSET_RATE, reg & 0xff7f); - wm9713->pll_out = freq_out; wm9713->pll_in = freq_in; /* wait 10ms AC97 link frames for the link to stabilise */ @@ -1149,8 +1147,8 @@ static int wm9713_soc_resume(struct platform_device *pdev) wm9713_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* do we need to re-start the PLL ? */ - if (wm9713->pll_out) - wm9713_set_pll(codec, 0, wm9713->pll_in, wm9713->pll_out); + if (wm9713->pll_in) + wm9713_set_pll(codec, 0, wm9713->pll_in, 0); /* only synchronise the codec if warm reset failed */ if (ret == 0) { -- cgit v1.2.3-70-g09d2 From 9ef344f89ac41116d4ab138b0941c784a3ab8cf4 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Fri, 16 Jan 2009 22:47:30 +0100 Subject: ALSA: wss-lib: remove "pops" before each played sound A WSS codec is autocalibrated each time before playing sound. Do only one calibration during codec initialization. Complete snd_wss_calibrate_mute to mute loopback volume as well. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/wss/wss_lib.c | 41 +++++++++++++---------------------------- 1 file changed, 13 insertions(+), 28 deletions(-) diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c index 13299aebd07..f0c0be5bb68 100644 --- a/sound/isa/wss/wss_lib.c +++ b/sound/isa/wss/wss_lib.c @@ -181,25 +181,6 @@ static void snd_wss_wait(struct snd_wss *chip) udelay(100); } -static void snd_wss_outm(struct snd_wss *chip, unsigned char reg, - unsigned char mask, unsigned char value) -{ - unsigned char tmp = (chip->image[reg] & mask) | value; - - snd_wss_wait(chip); -#ifdef CONFIG_SND_DEBUG - if (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) - snd_printk("outm: auto calibration time out - reg = 0x%x, value = 0x%x\n", reg, value); -#endif - chip->image[reg] = tmp; - if (!chip->calibrate_mute) { - wss_outb(chip, CS4231P(REGSEL), chip->mce_bit | reg); - wmb(); - wss_outb(chip, CS4231P(REG), tmp); - mb(); - } -} - static void snd_wss_dout(struct snd_wss *chip, unsigned char reg, unsigned char value) { @@ -587,7 +568,15 @@ static void snd_wss_calibrate_mute(struct snd_wss *chip, int mute) chip->image[CS4231_RIGHT_INPUT]); snd_wss_dout(chip, CS4231_LOOPBACK, chip->image[CS4231_LOOPBACK]); + } else { + snd_wss_dout(chip, CS4231_LEFT_INPUT, + 0); + snd_wss_dout(chip, CS4231_RIGHT_INPUT, + 0); + snd_wss_dout(chip, CS4231_LOOPBACK, + 0xfd); } + snd_wss_dout(chip, CS4231_AUX1_LEFT_INPUT, mute | chip->image[CS4231_AUX1_LEFT_INPUT]); snd_wss_dout(chip, CS4231_AUX1_RIGHT_INPUT, @@ -630,7 +619,6 @@ static void snd_wss_playback_format(struct snd_wss *chip, int full_calib = 1; mutex_lock(&chip->mce_mutex); - snd_wss_calibrate_mute(chip, 1); if (chip->hardware == WSS_HW_CS4231A || (chip->hardware & WSS_HW_CS4232_MASK)) { spin_lock_irqsave(&chip->reg_lock, flags); @@ -681,7 +669,6 @@ static void snd_wss_playback_format(struct snd_wss *chip, udelay(100); /* this seems to help */ snd_wss_mce_down(chip); } - snd_wss_calibrate_mute(chip, 0); mutex_unlock(&chip->mce_mutex); } @@ -693,7 +680,6 @@ static void snd_wss_capture_format(struct snd_wss *chip, int full_calib = 1; mutex_lock(&chip->mce_mutex); - snd_wss_calibrate_mute(chip, 1); if (chip->hardware == WSS_HW_CS4231A || (chip->hardware & WSS_HW_CS4232_MASK)) { spin_lock_irqsave(&chip->reg_lock, flags); @@ -750,7 +736,6 @@ static void snd_wss_capture_format(struct snd_wss *chip, spin_unlock_irqrestore(&chip->reg_lock, flags); snd_wss_mce_down(chip); } - snd_wss_calibrate_mute(chip, 0); mutex_unlock(&chip->mce_mutex); } @@ -807,6 +792,7 @@ static void snd_wss_init(struct snd_wss *chip) { unsigned long flags; + snd_wss_calibrate_mute(chip, 1); snd_wss_mce_down(chip); #ifdef SNDRV_DEBUG_MCE @@ -830,6 +816,8 @@ static void snd_wss_init(struct snd_wss *chip) snd_wss_mce_up(chip); spin_lock_irqsave(&chip->reg_lock, flags); + chip->image[CS4231_IFACE_CTRL] &= ~CS4231_AUTOCALIB; + snd_wss_out(chip, CS4231_IFACE_CTRL, chip->image[CS4231_IFACE_CTRL]); snd_wss_out(chip, CS4231_ALT_FEATURE_1, chip->image[CS4231_ALT_FEATURE_1]); spin_unlock_irqrestore(&chip->reg_lock, flags); @@ -863,6 +851,7 @@ static void snd_wss_init(struct snd_wss *chip) chip->image[CS4231_REC_FORMAT]); spin_unlock_irqrestore(&chip->reg_lock, flags); snd_wss_mce_down(chip); + snd_wss_calibrate_mute(chip, 0); #ifdef SNDRV_DEBUG_MCE snd_printk("init: (5)\n"); @@ -921,8 +910,6 @@ static void snd_wss_close(struct snd_wss *chip, unsigned int mode) mutex_unlock(&chip->open_mutex); return; } - snd_wss_calibrate_mute(chip, 1); - /* disable IRQ */ spin_lock_irqsave(&chip->reg_lock, flags); if (!(chip->hardware & WSS_HW_AD1848_MASK)) @@ -955,8 +942,6 @@ static void snd_wss_close(struct snd_wss *chip, unsigned int mode) wss_outb(chip, CS4231P(STATUS), 0); /* clear IRQ */ spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_wss_calibrate_mute(chip, 0); - chip->mode = 0; mutex_unlock(&chip->open_mutex); } @@ -1149,7 +1134,7 @@ irqreturn_t snd_wss_interrupt(int irq, void *dev_id) if (chip->hardware & WSS_HW_AD1848_MASK) wss_outb(chip, CS4231P(STATUS), 0); else - snd_wss_outm(chip, CS4231_IRQ_STATUS, status, 0); + snd_wss_out(chip, CS4231_IRQ_STATUS, status); spin_unlock(&chip->reg_lock); return IRQ_HANDLED; } -- cgit v1.2.3-70-g09d2 From 8693290b9038f32b6b9bafd97b7e18465d62655b Mon Sep 17 00:00:00 2001 From: Andreas Bergmeier Date: Sun, 18 Jan 2009 18:48:03 +0100 Subject: ALSA: usb-audio - Quirk for Serato phono Ignore errors (wrong usb interface data) found when using the serato scratch live box with alsa Thus the alsa controls can be accessed (beware: they don't work though - but at least it's one ugly error message less) Signed-off-by: Andreas Bergmeier Signed-off-by: Takashi Iwai --- sound/usb/usbmixer_maps.c | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+) diff --git a/sound/usb/usbmixer_maps.c b/sound/usb/usbmixer_maps.c index f41214f3ad6..3e5d66cf1f5 100644 --- a/sound/usb/usbmixer_maps.c +++ b/sound/usb/usbmixer_maps.c @@ -261,6 +261,22 @@ static struct usbmix_name_map aureon_51_2_map[] = { {} /* terminator */ }; +static struct usbmix_name_map scratch_live_map[] = { + /* 1: IT Line 1 (USB streaming) */ + /* 2: OT Line 1 (Speaker) */ + /* 3: IT Line 1 (Line connector) */ + { 4, "Line 1 In" }, /* FU */ + /* 5: OT Line 1 (USB streaming) */ + /* 6: IT Line 2 (USB streaming) */ + /* 7: OT Line 2 (Speaker) */ + /* 8: IT Line 2 (Line connector) */ + { 9, "Line 2 In" }, /* FU */ + /* 10: OT Line 2 (USB streaming) */ + /* 11: IT Mic (Line connector) */ + /* 12: OT Mic (USB streaming) */ + { 0 } /* terminator */ +}; + /* * Control map entries */ @@ -316,6 +332,11 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = { .id = USB_ID(0x0ccd, 0x0028), .map = aureon_51_2_map, }, + { + .id = USB_ID(0x13e5, 0x0001), + .map = scratch_live_map, + .ignore_ctl_error = 1, + }, { 0 } /* terminator */ }; -- cgit v1.2.3-70-g09d2 From f3a374e55a60f7ca57335c24ef875731b6683147 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 19 Jan 2009 14:30:48 +0100 Subject: ALSA: ca0106 - Add quirk for GA-G1975X mobo Giga-byte GA-G1975X mobo has a CA0106 on-board chip. Reference: bnc#395807 https://bugzilla.novell.com/show_bug.cgi?id=395807 Signed-off-by: Takashi Iwai --- sound/pci/ca0106/ca0106_main.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index 0e62205d408..3aac7e6489c 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -255,6 +255,14 @@ static struct snd_ca0106_details ca0106_chip_details[] = { .gpio_type = 2, .i2c_adc = 1, .spi_dac = 1 } , + /* Giga-byte GA-G1975X mobo + * Novell bnc#395807 + */ + /* FIXME: the GPIO and I2C setting aren't tested well */ + { .serial = 0x1458a006, + .name = "Giga-byte GA-G1975X", + .gpio_type = 1, + .i2c_adc = 1 }, /* Shuttle XPC SD31P which has an onboard Creative Labs * Sound Blaster Live! 24-bit EAX * high-definition 7.1 audio processor". -- cgit v1.2.3-70-g09d2 From cade9f8a9cf1cd41f6f9e8850c6a0465a21248c3 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Mon, 19 Jan 2009 12:08:58 +0100 Subject: ALSA: Release v1.0.19 Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- include/sound/version.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/version.h b/include/sound/version.h index 2b48237e23b..a7e74e23ad2 100644 --- a/include/sound/version.h +++ b/include/sound/version.h @@ -1,3 +1,3 @@ /* include/version.h */ -#define CONFIG_SND_VERSION "1.0.18a" +#define CONFIG_SND_VERSION "1.0.19" #define CONFIG_SND_DATE "" -- cgit v1.2.3-70-g09d2 From 2c782f5981a022f7a238d550af5daa75c8acf382 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 16 Jan 2009 16:35:52 +0000 Subject: ASoC: Implement support for CLK_POUT as MCLK on Zylonite The Zylonite supports switching the MCLK for the WM9713 between the AC97CLK and CLK_POUT outputs of the PXA processor via switch SW15 on the board. This patch adds support for configuring the system to use CLK_POUT. Unfortunately it is not possible to read the state of SW15 from software so this feature is controlled by a module option 'clk_pout' which should be set to a non-zero value to enable the use of CLK_POUT. Signed-off-by: Mark Brown --- sound/soc/pxa/zylonite.c | 101 +++++++++++++++++++++++++++++++++++++++-------- 1 file changed, 84 insertions(+), 17 deletions(-) diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index f8e9ecd589d..8541b679f6e 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -14,6 +14,7 @@ #include #include #include +#include #include #include #include @@ -26,6 +27,17 @@ #include "pxa2xx-ac97.h" #include "pxa-ssp.h" +/* + * There is a physical switch SW15 on the board which changes the MCLK + * for the WM9713 between the standard AC97 master clock and the + * output of the CLK_POUT signal from the PXA. + */ +static int clk_pout; +module_param(clk_pout, int, 0); +MODULE_PARM_DESC(clk_pout, "Use CLK_POUT as WM9713 MCLK (SW15 on board)."); + +static struct clk *pout; + static struct snd_soc_card zylonite; static const struct snd_soc_dapm_widget zylonite_dapm_widgets[] = { @@ -61,10 +73,8 @@ static const struct snd_soc_dapm_route audio_map[] = { static int zylonite_wm9713_init(struct snd_soc_codec *codec) { - /* Currently we only support use of the AC97 clock here. If - * CLK_POUT is selected by SW15 then the clock API will need - * to be used to request and enable it here. - */ + if (clk_pout) + snd_soc_dai_set_pll(&codec->dai[0], 0, clk_get_rate(pout), 0); snd_soc_dapm_new_controls(codec, zylonite_dapm_widgets, ARRAY_SIZE(zylonite_dapm_widgets)); @@ -85,7 +95,6 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - unsigned int pll_out = 0; unsigned int acds = 0; unsigned int wm9713_div = 0; int ret = 0; @@ -93,16 +102,13 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream, switch (params_rate(params)) { case 8000: wm9713_div = 12; - pll_out = 2048000; break; case 16000: wm9713_div = 6; - pll_out = 4096000; break; case 48000: default: wm9713_div = 2; - pll_out = 12288000; acds = 1; break; } @@ -123,10 +129,6 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, pll_out); - if (ret < 0) - return ret; - ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_AUDIO_DIV_ACDS, acds); if (ret < 0) return ret; @@ -135,11 +137,12 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - /* Note that if the PLL is in use the WM9713_PCMCLK_PLL_DIV needs - * to be set instead. - */ - ret = snd_soc_dai_set_clkdiv(codec_dai, WM9713_PCMCLK_DIV, - WM9713_PCMDIV(wm9713_div)); + if (clk_pout) + ret = snd_soc_dai_set_clkdiv(codec_dai, WM9713_PCMCLK_PLL_DIV, + WM9713_PCMDIV(wm9713_div)); + else + ret = snd_soc_dai_set_clkdiv(codec_dai, WM9713_PCMCLK_DIV, + WM9713_PCMDIV(wm9713_div)); if (ret < 0) return ret; @@ -173,8 +176,72 @@ static struct snd_soc_dai_link zylonite_dai[] = { }, }; +static int zylonite_probe(struct platform_device *pdev) +{ + int ret; + + if (clk_pout) { + pout = clk_get(NULL, "CLK_POUT"); + if (IS_ERR(pout)) { + dev_err(&pdev->dev, "Unable to obtain CLK_POUT: %ld\n", + PTR_ERR(pout)); + return PTR_ERR(pout); + } + + ret = clk_enable(pout); + if (ret != 0) { + dev_err(&pdev->dev, "Unable to enable CLK_POUT: %d\n", + ret); + clk_put(pout); + return ret; + } + + dev_dbg(&pdev->dev, "MCLK enabled at %luHz\n", + clk_get_rate(pout)); + } + + return 0; +} + +static int zylonite_remove(struct platform_device *pdev) +{ + if (clk_pout) { + clk_disable(pout); + clk_put(pout); + } + + return 0; +} + +static int zylonite_suspend_post(struct platform_device *pdev, + pm_message_t state) +{ + if (clk_pout) + clk_disable(pout); + + return 0; +} + +static int zylonite_resume_pre(struct platform_device *pdev) +{ + int ret = 0; + + if (clk_pout) { + ret = clk_enable(pout); + if (ret != 0) + dev_err(&pdev->dev, "Unable to enable CLK_POUT: %d\n", + ret); + } + + return ret; +} + static struct snd_soc_card zylonite = { .name = "Zylonite", + .probe = &zylonite_probe, + .remove = &zylonite_remove, + .suspend_post = &zylonite_suspend_post, + .resume_pre = &zylonite_resume_pre, .platform = &pxa2xx_soc_platform, .dai_link = zylonite_dai, .num_links = ARRAY_SIZE(zylonite_dai), -- cgit v1.2.3-70-g09d2 From 28796eaf806502b9bd86cbacf8edbc14c80c14b0 Mon Sep 17 00:00:00 2001 From: Ian Molton Date: Sat, 17 Jan 2009 15:11:06 +0000 Subject: ASoC: machine support for Toshiba e740 PDA This patch provides suupport for the wm9705 AC97 codec on the Toshiba e740. Note: The e740 has a hard headphone switch that turns the speaker off and is not software detectable or controlable. Also both headphone and speaker amps share a common output enable. Signed-off-by: Ian Molton Signed-off-by: Mark Brown --- arch/arm/mach-pxa/e740.c | 5 + arch/arm/mach-pxa/include/mach/eseries-gpio.h | 5 + sound/soc/pxa/Kconfig | 9 ++ sound/soc/pxa/Makefile | 2 + sound/soc/pxa/e740_wm9705.c | 213 ++++++++++++++++++++++++++ 5 files changed, 234 insertions(+) create mode 100644 sound/soc/pxa/e740_wm9705.c diff --git a/arch/arm/mach-pxa/e740.c b/arch/arm/mach-pxa/e740.c index 6d48e00f4f0..a6fff782e7a 100644 --- a/arch/arm/mach-pxa/e740.c +++ b/arch/arm/mach-pxa/e740.c @@ -135,6 +135,11 @@ static unsigned long e740_pin_config[] __initdata = { /* IrDA */ GPIO38_GPIO | MFP_LPM_DRIVE_HIGH, + /* Audio power control */ + GPIO16_GPIO, /* AC97 codec AVDD2 supply (analogue power) */ + GPIO40_GPIO, /* Mic amp power */ + GPIO41_GPIO, /* Headphone amp power */ + /* PC Card */ GPIO8_GPIO, /* CD0 */ GPIO44_GPIO, /* CD1 */ diff --git a/arch/arm/mach-pxa/include/mach/eseries-gpio.h b/arch/arm/mach-pxa/include/mach/eseries-gpio.h index 6d6e4d8fa4c..f3e5509820d 100644 --- a/arch/arm/mach-pxa/include/mach/eseries-gpio.h +++ b/arch/arm/mach-pxa/include/mach/eseries-gpio.h @@ -45,6 +45,11 @@ /* e7xx IrDA power control */ #define GPIO_E7XX_IR_OFF 38 +/* e740 audio control GPIOs */ +#define GPIO_E740_WM9705_nAVDD2 16 +#define GPIO_E740_MIC_ON 40 +#define GPIO_E740_AMP_ON 41 + /* e750 audio control GPIOs */ #define GPIO_E750_HP_AMP_OFF 4 #define GPIO_E750_SPK_AMP_OFF 7 diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index b9b1a3f5d67..958ac3fe15d 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -61,6 +61,15 @@ config SND_PXA2XX_SOC_TOSA Say Y if you want to add support for SoC audio on Sharp Zaurus SL-C6000x models (Tosa). +config SND_PXA2XX_SOC_E740 + tristate "SoC AC97 Audio support for e740" + depends on SND_PXA2XX_SOC && MACH_E740 + select SND_SOC_WM9705 + select SND_PXA2XX_SOC_AC97 + help + Say Y if you want to add support for SoC audio on the + toshiba e740 PDA + config SND_PXA2XX_SOC_E750 tristate "SoC AC97 Audio support for e750" depends on SND_PXA2XX_SOC && MACH_E750 diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index c7d4cceeed9..97a51a8c936 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -13,6 +13,7 @@ obj-$(CONFIG_SND_PXA_SOC_SSP) += snd-soc-pxa-ssp.o snd-soc-corgi-objs := corgi.o snd-soc-poodle-objs := poodle.o snd-soc-tosa-objs := tosa.o +snd-soc-e740-objs := e740_wm9705.o snd-soc-e750-objs := e750_wm9705.o snd-soc-e800-objs := e800_wm9712.o snd-soc-spitz-objs := spitz.o @@ -23,6 +24,7 @@ snd-soc-zylonite-objs := zylonite.o obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o obj-$(CONFIG_SND_PXA2XX_SOC_TOSA) += snd-soc-tosa.o +obj-$(CONFIG_SND_PXA2XX_SOC_E740) += snd-soc-e740.o obj-$(CONFIG_SND_PXA2XX_SOC_E750) += snd-soc-e750.o obj-$(CONFIG_SND_PXA2XX_SOC_E800) += snd-soc-e800.o obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c new file mode 100644 index 00000000000..ac361765173 --- /dev/null +++ b/sound/soc/pxa/e740_wm9705.c @@ -0,0 +1,213 @@ +/* + * e740-wm9705.c -- SoC audio for e740 + * + * Copyright 2007 (c) Ian Molton + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; version 2 ONLY. + * + */ + +#include +#include +#include + +#include +#include +#include +#include + +#include +#include +#include +#include + +#include + +#include "../codecs/wm9705.h" +#include "pxa2xx-pcm.h" +#include "pxa2xx-ac97.h" + + +#define E740_AUDIO_OUT 1 +#define E740_AUDIO_IN 2 + +static int e740_audio_power; + +static void e740_sync_audio_power(int status) +{ + gpio_set_value(GPIO_E740_WM9705_nAVDD2, !status); + gpio_set_value(GPIO_E740_AMP_ON, (status & E740_AUDIO_OUT) ? 1 : 0); + gpio_set_value(GPIO_E740_MIC_ON, (status & E740_AUDIO_IN) ? 1 : 0); +} + +static int e740_mic_amp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + if (event & SND_SOC_DAPM_PRE_PMU) + e740_audio_power |= E740_AUDIO_IN; + else if (event & SND_SOC_DAPM_POST_PMD) + e740_audio_power &= ~E740_AUDIO_IN; + + e740_sync_audio_power(e740_audio_power); + + return 0; +} + +static int e740_output_amp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + if (event & SND_SOC_DAPM_PRE_PMU) + e740_audio_power |= E740_AUDIO_OUT; + else if (event & SND_SOC_DAPM_POST_PMD) + e740_audio_power &= ~E740_AUDIO_OUT; + + e740_sync_audio_power(e740_audio_power); + + return 0; +} + +static const struct snd_soc_dapm_widget e740_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_MIC("Mic (Internal)", NULL), + SND_SOC_DAPM_PGA_E("Output Amp", SND_SOC_NOPM, 0, 0, NULL, 0, + e740_output_amp_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PGA_E("Mic Amp", SND_SOC_NOPM, 0, 0, NULL, 0, + e740_mic_amp_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + {"Output Amp", NULL, "LOUT"}, + {"Output Amp", NULL, "ROUT"}, + {"Output Amp", NULL, "MONOOUT"}, + + {"Speaker", NULL, "Output Amp"}, + {"Headphone Jack", NULL, "Output Amp"}, + + {"MIC1", NULL, "Mic Amp"}, + {"Mic Amp", NULL, "Mic (Internal)"}, +}; + +static int e740_ac97_init(struct snd_soc_codec *codec) +{ + snd_soc_dapm_nc_pin(codec, "HPOUTL"); + snd_soc_dapm_nc_pin(codec, "HPOUTR"); + snd_soc_dapm_nc_pin(codec, "PHONE"); + snd_soc_dapm_nc_pin(codec, "LINEINL"); + snd_soc_dapm_nc_pin(codec, "LINEINR"); + snd_soc_dapm_nc_pin(codec, "CDINL"); + snd_soc_dapm_nc_pin(codec, "CDINR"); + snd_soc_dapm_nc_pin(codec, "PCBEEP"); + snd_soc_dapm_nc_pin(codec, "MIC2"); + + snd_soc_dapm_new_controls(codec, e740_dapm_widgets, + ARRAY_SIZE(e740_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link e740_dai[] = { + { + .name = "AC97", + .stream_name = "AC97 HiFi", + .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI], + .codec_dai = &wm9705_dai[WM9705_DAI_AC97_HIFI], + .init = e740_ac97_init, + }, + { + .name = "AC97 Aux", + .stream_name = "AC97 Aux", + .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX], + .codec_dai = &wm9705_dai[WM9705_DAI_AC97_AUX], + }, +}; + +static struct snd_soc_card e740 = { + .name = "Toshiba e740", + .platform = &pxa2xx_soc_platform, + .dai_link = e740_dai, + .num_links = ARRAY_SIZE(e740_dai), +}; + +static struct snd_soc_device e740_snd_devdata = { + .card = &e740, + .codec_dev = &soc_codec_dev_wm9705, +}; + +static struct platform_device *e740_snd_device; + +static int __init e740_init(void) +{ + int ret; + + if (!machine_is_e740()) + return -ENODEV; + + ret = gpio_request(GPIO_E740_MIC_ON, "Mic amp"); + if (ret) + return ret; + + ret = gpio_request(GPIO_E740_AMP_ON, "Output amp"); + if (ret) + goto free_mic_amp_gpio; + + ret = gpio_request(GPIO_E740_WM9705_nAVDD2, "Audio power"); + if (ret) + goto free_op_amp_gpio; + + /* Disable audio */ + ret = gpio_direction_output(GPIO_E740_MIC_ON, 0); + if (ret) + goto free_apwr_gpio; + ret = gpio_direction_output(GPIO_E740_AMP_ON, 0); + if (ret) + goto free_apwr_gpio; + ret = gpio_direction_output(GPIO_E740_WM9705_nAVDD2, 1); + if (ret) + goto free_apwr_gpio; + + e740_snd_device = platform_device_alloc("soc-audio", -1); + if (!e740_snd_device) { + ret = -ENOMEM; + goto free_apwr_gpio; + } + + platform_set_drvdata(e740_snd_device, &e740_snd_devdata); + e740_snd_devdata.dev = &e740_snd_device->dev; + ret = platform_device_add(e740_snd_device); + + if (!ret) + return 0; + +/* Fail gracefully */ + platform_device_put(e740_snd_device); +free_apwr_gpio: + gpio_free(GPIO_E740_WM9705_nAVDD2); +free_op_amp_gpio: + gpio_free(GPIO_E740_AMP_ON); +free_mic_amp_gpio: + gpio_free(GPIO_E740_MIC_ON); + + return ret; +} + +static void __exit e740_exit(void) +{ + platform_device_unregister(e740_snd_device); +} + +module_init(e740_init); +module_exit(e740_exit); + +/* Module information */ +MODULE_AUTHOR("Ian Molton "); +MODULE_DESCRIPTION("ALSA SoC driver for e740"); +MODULE_LICENSE("GPL v2"); -- cgit v1.2.3-70-g09d2 From 91432e976ff1323e5dd6f52498969602953c6ee9 Mon Sep 17 00:00:00 2001 From: Ian Molton Date: Sat, 17 Jan 2009 17:44:23 +0000 Subject: ASoC: fixes to caching implementations This patch takes fixes a number of bugs in the caching code used by several ASoC codec drivers. Mostly off-by-one fixes. Signed-off-by: Ian Molton Signed-off-by: Mark Brown --- sound/soc/codecs/ac97.c | 2 -- sound/soc/codecs/ad1980.c | 4 ++-- sound/soc/codecs/twl4030.c | 3 +++ sound/soc/codecs/wm8580.c | 4 ++-- sound/soc/codecs/wm8728.c | 4 ++-- sound/soc/codecs/wm8753.c | 4 ++-- sound/soc/codecs/wm8990.c | 4 ++-- sound/soc/codecs/wm9712.c | 4 ++-- sound/soc/codecs/wm9713.c | 4 ++-- 9 files changed, 17 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index fb53e6511af..89d41277616 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -123,7 +123,6 @@ bus_err: snd_soc_free_pcms(socdev); err: - kfree(socdev->codec->reg_cache); kfree(socdev->codec); socdev->codec = NULL; return ret; @@ -138,7 +137,6 @@ static int ac97_soc_remove(struct platform_device *pdev) return 0; snd_soc_free_pcms(socdev); - kfree(socdev->codec->reg_cache); kfree(socdev->codec); return 0; diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index c3c5d0eee37..faf358758e1 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -109,7 +109,7 @@ static unsigned int ac97_read(struct snd_soc_codec *codec, default: reg = reg >> 1; - if (reg >= (ARRAY_SIZE(ad1980_reg))) + if (reg >= ARRAY_SIZE(ad1980_reg)) return -EINVAL; return cache[reg]; @@ -123,7 +123,7 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, soc_ac97_ops.write(codec->ac97, reg, val); reg = reg >> 1; - if (reg < (ARRAY_SIZE(ad1980_reg))) + if (reg < ARRAY_SIZE(ad1980_reg)) cache[reg] = val; return 0; diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index ddc9f37d863..f530c1e6d9e 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -125,6 +125,9 @@ static inline unsigned int twl4030_read_reg_cache(struct snd_soc_codec *codec, { u8 *cache = codec->reg_cache; + if (reg >= TWL4030_CACHEREGNUM) + return -EIO; + return cache[reg]; } diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 9b75a377453..3faf0e70ce1 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -200,7 +200,7 @@ static inline unsigned int wm8580_read_reg_cache(struct snd_soc_codec *codec, unsigned int reg) { u16 *cache = codec->reg_cache; - BUG_ON(reg > ARRAY_SIZE(wm8580_reg)); + BUG_ON(reg >= ARRAY_SIZE(wm8580_reg)); return cache[reg]; } @@ -223,7 +223,7 @@ static int wm8580_write(struct snd_soc_codec *codec, unsigned int reg, { u8 data[2]; - BUG_ON(reg > ARRAY_SIZE(wm8580_reg)); + BUG_ON(reg >= ARRAY_SIZE(wm8580_reg)); /* Registers are 9 bits wide */ value &= 0x1ff; diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index defa310bc7d..f90dc52e975 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -47,7 +47,7 @@ static inline unsigned int wm8728_read_reg_cache(struct snd_soc_codec *codec, unsigned int reg) { u16 *cache = codec->reg_cache; - BUG_ON(reg > ARRAY_SIZE(wm8728_reg_defaults)); + BUG_ON(reg >= ARRAY_SIZE(wm8728_reg_defaults)); return cache[reg]; } @@ -55,7 +55,7 @@ static inline void wm8728_write_reg_cache(struct snd_soc_codec *codec, u16 reg, unsigned int value) { u16 *cache = codec->reg_cache; - BUG_ON(reg > ARRAY_SIZE(wm8728_reg_defaults)); + BUG_ON(reg >= ARRAY_SIZE(wm8728_reg_defaults)); cache[reg] = value; } diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 7283178e0eb..5a1c1fca120 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -97,7 +97,7 @@ static inline unsigned int wm8753_read_reg_cache(struct snd_soc_codec *codec, unsigned int reg) { u16 *cache = codec->reg_cache; - if (reg < 1 || reg > (ARRAY_SIZE(wm8753_reg) + 1)) + if (reg < 1 || reg >= (ARRAY_SIZE(wm8753_reg) + 1)) return -1; return cache[reg - 1]; } @@ -109,7 +109,7 @@ static inline void wm8753_write_reg_cache(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { u16 *cache = codec->reg_cache; - if (reg < 1 || reg > 0x3f) + if (reg < 1 || reg >= (ARRAY_SIZE(wm8753_reg) + 1)) return; cache[reg - 1] = value; } diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 6b2778632d5..f93c0955ed9 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -116,7 +116,7 @@ static inline unsigned int wm8990_read_reg_cache(struct snd_soc_codec *codec, unsigned int reg) { u16 *cache = codec->reg_cache; - BUG_ON(reg > (ARRAY_SIZE(wm8990_reg)) - 1); + BUG_ON(reg >= ARRAY_SIZE(wm8990_reg)); return cache[reg]; } @@ -129,7 +129,7 @@ static inline void wm8990_write_reg_cache(struct snd_soc_codec *codec, u16 *cache = codec->reg_cache; /* Reset register and reserved registers are uncached */ - if (reg == 0 || reg > ARRAY_SIZE(wm8990_reg) - 1) + if (reg == 0 || reg >= ARRAY_SIZE(wm8990_reg)) return; cache[reg] = value; diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 1b0ace0f4dc..4dc90d67530 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -452,7 +452,7 @@ static unsigned int ac97_read(struct snd_soc_codec *codec, else { reg = reg >> 1; - if (reg > (ARRAY_SIZE(wm9712_reg))) + if (reg >= (ARRAY_SIZE(wm9712_reg))) return -EIO; return cache[reg]; @@ -466,7 +466,7 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, soc_ac97_ops.write(codec->ac97, reg, val); reg = reg >> 1; - if (reg <= (ARRAY_SIZE(wm9712_reg))) + if (reg < (ARRAY_SIZE(wm9712_reg))) cache[reg] = val; return 0; diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index e636d8a18ed..0e60e16973d 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -620,7 +620,7 @@ static unsigned int ac97_read(struct snd_soc_codec *codec, else { reg = reg >> 1; - if (reg > (ARRAY_SIZE(wm9713_reg))) + if (reg >= (ARRAY_SIZE(wm9713_reg))) return -EIO; return cache[reg]; @@ -634,7 +634,7 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, if (reg < 0x7c) soc_ac97_ops.write(codec->ac97, reg, val); reg = reg >> 1; - if (reg <= (ARRAY_SIZE(wm9713_reg))) + if (reg < (ARRAY_SIZE(wm9713_reg))) cache[reg] = val; return 0; -- cgit v1.2.3-70-g09d2 From b2a19d02396c92294abcddee5bd9bd49cc4e4d1c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 17 Jan 2009 19:14:26 +0000 Subject: ASoC: Staticise PCM operations tables The PCM operations tables are not exported directly but are instead included in the platform structure so should be declared static. Signed-off-by: Mark Brown --- sound/soc/atmel/atmel-pcm.c | 2 +- sound/soc/au1x/dbdma2.c | 2 +- sound/soc/blackfin/bf5xx-ac97-pcm.c | 2 +- sound/soc/blackfin/bf5xx-i2s-pcm.c | 2 +- sound/soc/davinci/davinci-pcm.c | 2 +- sound/soc/omap/omap-pcm.c | 2 +- 6 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c index 3dcdc4e3cfa..9ef6b96373f 100644 --- a/sound/soc/atmel/atmel-pcm.c +++ b/sound/soc/atmel/atmel-pcm.c @@ -347,7 +347,7 @@ static int atmel_pcm_mmap(struct snd_pcm_substream *substream, vma->vm_end - vma->vm_start, vma->vm_page_prot); } -struct snd_pcm_ops atmel_pcm_ops = { +static struct snd_pcm_ops atmel_pcm_ops = { .open = atmel_pcm_open, .close = atmel_pcm_close, .ioctl = snd_pcm_lib_ioctl, diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index bc8d654576c..30490a25914 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -305,7 +305,7 @@ static int au1xpsc_pcm_close(struct snd_pcm_substream *substream) return 0; } -struct snd_pcm_ops au1xpsc_pcm_ops = { +static struct snd_pcm_ops au1xpsc_pcm_ops = { .open = au1xpsc_pcm_open, .close = au1xpsc_pcm_close, .ioctl = snd_pcm_lib_ioctl, diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c index 8067cfafa3a..8cfed1a5dcb 100644 --- a/sound/soc/blackfin/bf5xx-ac97-pcm.c +++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c @@ -297,7 +297,7 @@ static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel, } #endif -struct snd_pcm_ops bf5xx_pcm_ac97_ops = { +static struct snd_pcm_ops bf5xx_pcm_ac97_ops = { .open = bf5xx_pcm_open, .ioctl = snd_pcm_lib_ioctl, .hw_params = bf5xx_pcm_hw_params, diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c index 53d290b3ea4..1318c4f627b 100644 --- a/sound/soc/blackfin/bf5xx-i2s-pcm.c +++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c @@ -184,7 +184,7 @@ static int bf5xx_pcm_mmap(struct snd_pcm_substream *substream, return 0 ; } -struct snd_pcm_ops bf5xx_pcm_i2s_ops = { +static struct snd_pcm_ops bf5xx_pcm_i2s_ops = { .open = bf5xx_pcm_open, .ioctl = snd_pcm_lib_ioctl, .hw_params = bf5xx_pcm_hw_params, diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 366049d8578..7af3b5b3a53 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -286,7 +286,7 @@ static int davinci_pcm_mmap(struct snd_pcm_substream *substream, runtime->dma_bytes); } -struct snd_pcm_ops davinci_pcm_ops = { +static struct snd_pcm_ops davinci_pcm_ops = { .open = davinci_pcm_open, .close = davinci_pcm_close, .ioctl = snd_pcm_lib_ioctl, diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index b0362dfd5b7..607a38c7ae4 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -264,7 +264,7 @@ static int omap_pcm_mmap(struct snd_pcm_substream *substream, runtime->dma_bytes); } -struct snd_pcm_ops omap_pcm_ops = { +static struct snd_pcm_ops omap_pcm_ops = { .open = omap_pcm_open, .close = omap_pcm_close, .ioctl = snd_pcm_lib_ioctl, -- cgit v1.2.3-70-g09d2 From 29fdbec2dcb1ce364812778271056aa9516ff3ed Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 20 Jan 2009 13:07:55 +0100 Subject: ALSA: hda - Add extra volume offset to standard volume amp macros Added the volume offset to base for the standard volume controls to handle elements with too big volume scales like -96dB..0dB. For such elements, you can set the base volume to reduce the range. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 45 +++++++++++++++++++++++++++++++++++++-------- sound/pci/hda/hda_local.h | 5 ++++- 2 files changed, 41 insertions(+), 9 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index b7bba7dc7cf..0cf2424ada6 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1119,6 +1119,7 @@ int snd_hda_mixer_amp_volume_info(struct snd_kcontrol *kcontrol, u16 nid = get_amp_nid(kcontrol); u8 chs = get_amp_channels(kcontrol); int dir = get_amp_direction(kcontrol); + unsigned int ofs = get_amp_offset(kcontrol); u32 caps; caps = query_amp_caps(codec, nid, dir); @@ -1130,6 +1131,8 @@ int snd_hda_mixer_amp_volume_info(struct snd_kcontrol *kcontrol, kcontrol->id.name); return -EINVAL; } + if (ofs < caps) + caps -= ofs; uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = chs == 3 ? 2 : 1; uinfo->value.integer.min = 0; @@ -1138,6 +1141,32 @@ int snd_hda_mixer_amp_volume_info(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_volume_info); + +static inline unsigned int +read_amp_value(struct hda_codec *codec, hda_nid_t nid, + int ch, int dir, int idx, unsigned int ofs) +{ + unsigned int val; + val = snd_hda_codec_amp_read(codec, nid, ch, dir, idx); + val &= HDA_AMP_VOLMASK; + if (val >= ofs) + val -= ofs; + else + val = 0; + return val; +} + +static inline int +update_amp_value(struct hda_codec *codec, hda_nid_t nid, + int ch, int dir, int idx, unsigned int ofs, + unsigned int val) +{ + if (val > 0) + val += ofs; + return snd_hda_codec_amp_update(codec, nid, ch, dir, idx, + HDA_AMP_VOLMASK, val); +} + int snd_hda_mixer_amp_volume_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -1146,14 +1175,13 @@ int snd_hda_mixer_amp_volume_get(struct snd_kcontrol *kcontrol, int chs = get_amp_channels(kcontrol); int dir = get_amp_direction(kcontrol); int idx = get_amp_index(kcontrol); + unsigned int ofs = get_amp_offset(kcontrol); long *valp = ucontrol->value.integer.value; if (chs & 1) - *valp++ = snd_hda_codec_amp_read(codec, nid, 0, dir, idx) - & HDA_AMP_VOLMASK; + *valp++ = read_amp_value(codec, nid, 0, dir, idx, ofs); if (chs & 2) - *valp = snd_hda_codec_amp_read(codec, nid, 1, dir, idx) - & HDA_AMP_VOLMASK; + *valp = read_amp_value(codec, nid, 1, dir, idx, ofs); return 0; } EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_volume_get); @@ -1166,18 +1194,17 @@ int snd_hda_mixer_amp_volume_put(struct snd_kcontrol *kcontrol, int chs = get_amp_channels(kcontrol); int dir = get_amp_direction(kcontrol); int idx = get_amp_index(kcontrol); + unsigned int ofs = get_amp_offset(kcontrol); long *valp = ucontrol->value.integer.value; int change = 0; snd_hda_power_up(codec); if (chs & 1) { - change = snd_hda_codec_amp_update(codec, nid, 0, dir, idx, - 0x7f, *valp); + change = update_amp_value(codec, nid, 0, dir, idx, ofs, *valp); valp++; } if (chs & 2) - change |= snd_hda_codec_amp_update(codec, nid, 1, dir, idx, - 0x7f, *valp); + change |= update_amp_value(codec, nid, 1, dir, idx, ofs, *valp); snd_hda_power_down(codec); return change; } @@ -1189,6 +1216,7 @@ int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag, struct hda_codec *codec = snd_kcontrol_chip(kcontrol); hda_nid_t nid = get_amp_nid(kcontrol); int dir = get_amp_direction(kcontrol); + unsigned int ofs = get_amp_offset(kcontrol); u32 caps, val1, val2; if (size < 4 * sizeof(unsigned int)) @@ -1197,6 +1225,7 @@ int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag, val2 = (caps & AC_AMPCAP_STEP_SIZE) >> AC_AMPCAP_STEP_SIZE_SHIFT; val2 = (val2 + 1) * 25; val1 = -((caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT); + val1 += ofs; val1 = ((int)val1) * ((int)val2); if (put_user(SNDRV_CTL_TLVT_DB_SCALE, _tlv)) return -EFAULT; diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 1dd8716c387..d53ce1f8541 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -26,8 +26,10 @@ /* * for mixer controls */ +#define HDA_COMPOSE_AMP_VAL_OFS(nid,chs,idx,dir,ofs) \ + ((nid) | ((chs)<<16) | ((dir)<<18) | ((idx)<<19) | ((ofs)<<23)) #define HDA_COMPOSE_AMP_VAL(nid,chs,idx,dir) \ - ((nid) | ((chs)<<16) | ((dir)<<18) | ((idx)<<19)) + HDA_COMPOSE_AMP_VAL_OFS(nid, chs, idx, dir, 0) /* mono volume with index (index=0,1,...) (channel=1,2) */ #define HDA_CODEC_VOLUME_MONO_IDX(xname, xcidx, nid, channel, xindex, direction) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx, \ @@ -456,6 +458,7 @@ int snd_hda_check_amp_list_power(struct hda_codec *codec, #define get_amp_channels(kc) (((kc)->private_value >> 16) & 0x3) #define get_amp_direction(kc) (((kc)->private_value >> 18) & 0x1) #define get_amp_index(kc) (((kc)->private_value >> 19) & 0xf) +#define get_amp_offset(kc) (((kc)->private_value >> 23) & 0x3f) /* * CEA Short Audio Descriptor data -- cgit v1.2.3-70-g09d2 From 7c7767ebe2fa847c91a0dd5551ca422aba359473 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 20 Jan 2009 15:28:38 +0100 Subject: ALSA: hda - Halve too large volume scales for STAC/IDT codecs STAC/IDT codecs have often too large volume scales such as -96dB, and exposing this as is results in too large scale in percentage representation. This patch adds the check of the volume scale and halves the volume range if it's too large automatically. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 41 +++++++++++++++++++++++++++++++++-------- 1 file changed, 33 insertions(+), 8 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index a4d4afe6b4f..c2d4abee3b0 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -166,6 +166,7 @@ struct sigmatel_spec { unsigned int alt_switch: 1; unsigned int hp_detect: 1; unsigned int spdif_mute: 1; + unsigned int check_volume_offset:1; /* gpio lines */ unsigned int eapd_mask; @@ -202,6 +203,8 @@ struct sigmatel_spec { hda_nid_t hp_dacs[5]; hda_nid_t speaker_dacs[5]; + int volume_offset; + /* capture */ hda_nid_t *adc_nids; unsigned int num_adcs; @@ -1297,6 +1300,8 @@ static int stac92xx_build_controls(struct hda_codec *codec) unsigned int vmaster_tlv[4]; snd_hda_set_vmaster_tlv(codec, spec->multiout.dac_nids[0], HDA_OUTPUT, vmaster_tlv); + /* correct volume offset */ + vmaster_tlv[2] += vmaster_tlv[3] * spec->volume_offset; err = snd_hda_add_vmaster(codec, "Master Playback Volume", vmaster_tlv, slave_vols); if (err < 0) @@ -2980,14 +2985,34 @@ static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec) } /* create volume control/switch for the given prefx type */ -static int create_controls(struct sigmatel_spec *spec, const char *pfx, hda_nid_t nid, int chs) +static int create_controls(struct hda_codec *codec, const char *pfx, + hda_nid_t nid, int chs) { + struct sigmatel_spec *spec = codec->spec; char name[32]; int err; + if (!spec->check_volume_offset) { + unsigned int caps, step, nums, db_scale; + caps = query_amp_caps(codec, nid, HDA_OUTPUT); + step = (caps & AC_AMPCAP_STEP_SIZE) >> + AC_AMPCAP_STEP_SIZE_SHIFT; + step = (step + 1) * 25; /* in .01dB unit */ + nums = (caps & AC_AMPCAP_NUM_STEPS) >> + AC_AMPCAP_NUM_STEPS_SHIFT; + db_scale = nums * step; + /* if dB scale is over -64dB, and finer enough, + * let's reduce it to half + */ + if (db_scale > 6400 && nums >= 0x1f) + spec->volume_offset = nums / 2; + spec->check_volume_offset = 1; + } + sprintf(name, "%s Playback Volume", pfx); err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, name, - HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT)); + HDA_COMPOSE_AMP_VAL_OFS(nid, chs, 0, HDA_OUTPUT, + spec->volume_offset)); if (err < 0) return err; sprintf(name, "%s Playback Switch", pfx); @@ -3053,10 +3078,10 @@ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec, nid = spec->multiout.dac_nids[i]; if (i == 2) { /* Center/LFE */ - err = create_controls(spec, "Center", nid, 1); + err = create_controls(codec, "Center", nid, 1); if (err < 0) return err; - err = create_controls(spec, "LFE", nid, 2); + err = create_controls(codec, "LFE", nid, 2); if (err < 0) return err; @@ -3084,7 +3109,7 @@ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec, break; } } - err = create_controls(spec, name, nid, 3); + err = create_controls(codec, name, nid, 3); if (err < 0) return err; } @@ -3139,7 +3164,7 @@ static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec, nid = spec->hp_dacs[i]; if (!nid) continue; - err = create_controls(spec, pfxs[nums++], nid, 3); + err = create_controls(codec, pfxs[nums++], nid, 3); if (err < 0) return err; } @@ -3153,7 +3178,7 @@ static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec, nid = spec->speaker_dacs[i]; if (!nid) continue; - err = create_controls(spec, pfxs[nums++], nid, 3); + err = create_controls(codec, pfxs[nums++], nid, 3); if (err < 0) return err; } @@ -3729,7 +3754,7 @@ static int stac9200_auto_create_lfe_ctls(struct hda_codec *codec, } if (lfe_pin) { - err = create_controls(spec, "LFE", lfe_pin, 1); + err = create_controls(codec, "LFE", lfe_pin, 1); if (err < 0) return err; } -- cgit v1.2.3-70-g09d2 From 89ce9e87083216389d2ff5740cc60f835537d8d0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 20 Jan 2009 17:15:57 +0100 Subject: ALSA: hda - Add debug prints for digital I/O pin detections Add the debug prints for digital I/O pin detections in snd_hda_parse_pin_def_config() function. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index b7bba7dc7cf..c03de0bc399 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3499,6 +3499,8 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, cfg->hp_pins[1], cfg->hp_pins[2], cfg->hp_pins[3], cfg->hp_pins[4]); snd_printd(" mono: mono_out=0x%x\n", cfg->mono_out_pin); + if (cfg->dig_out_pin) + snd_printd(" dig-out=0x%x\n", cfg->dig_out_pin); snd_printd(" inputs: mic=0x%x, fmic=0x%x, line=0x%x, fline=0x%x," " cd=0x%x, aux=0x%x\n", cfg->input_pins[AUTO_PIN_MIC], @@ -3507,6 +3509,8 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, cfg->input_pins[AUTO_PIN_FRONT_LINE], cfg->input_pins[AUTO_PIN_CD], cfg->input_pins[AUTO_PIN_AUX]); + if (cfg->dig_out_pin) + snd_printd(" dig-in=0x%x\n", cfg->dig_in_pin); return 0; } -- cgit v1.2.3-70-g09d2 From 1b52ae701fedf97f9984e73b6a1fe2444230871b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 20 Jan 2009 17:17:29 +0100 Subject: ALSA: hda - Detect non-SPDIF digital I/O Accept non-SPDIF digital I/O pins as the digital pins. These are usually corresponding to HDMI I/O. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index c03de0bc399..2d6f72ca014 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3390,9 +3390,11 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, cfg->input_pins[AUTO_PIN_AUX] = nid; break; case AC_JACK_SPDIF_OUT: + case AC_JACK_DIG_OTHER_OUT: cfg->dig_out_pin = nid; break; case AC_JACK_SPDIF_IN: + case AC_JACK_DIG_OTHER_IN: cfg->dig_in_pin = nid; break; } -- cgit v1.2.3-70-g09d2 From caa10b6e808a4d65eb0306f0006308244f2b8d79 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 20 Jan 2009 17:19:01 +0100 Subject: ALSA: hda - Improve auto-probing of STAC9872 codec Use the standard STAC/IDT auto-probing routine for non-static STAC9872 codec probing. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 58 ++++++++++++++++++++++++++++++++++-------- 1 file changed, 48 insertions(+), 10 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index a4d4afe6b4f..b6e797d1c21 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -5511,24 +5511,62 @@ static struct snd_pci_quirk stac9872_cfg_tbl[] = { {} }; +static struct snd_kcontrol_new stac9872_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x09, 0, HDA_INPUT), + STAC_INPUT_SOURCE(1), + { } /* end */ +}; + +static hda_nid_t stac9872_pin_nids[] = { + 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f, + 0x11, 0x13, 0x14, +}; + +static hda_nid_t stac9872_adc_nids[] = { + 0x8 /*,0x6*/ +}; + +static hda_nid_t stac9872_mux_nids[] = { + 0x15 +}; + static int patch_stac9872(struct hda_codec *codec) { struct sigmatel_spec *spec; - int board_config; - board_config = snd_hda_check_board_config(codec, STAC_9872_MODELS, - stac9872_models, - stac9872_cfg_tbl); - if (board_config < 0) - /* unknown config, let generic-parser do its job... */ - return snd_hda_parse_generic_codec(codec); - spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) return -ENOMEM; - codec->spec = spec; - switch (board_config) { + + spec->board_config = snd_hda_check_board_config(codec, STAC_9872_MODELS, + stac9872_models, + stac9872_cfg_tbl); + if (spec->board_config < 0) { + int err; + + spec->num_pins = ARRAY_SIZE(stac9872_pin_nids); + spec->pin_nids = stac9872_pin_nids; + spec->multiout.dac_nids = spec->dac_nids; + spec->num_adcs = ARRAY_SIZE(stac9872_adc_nids); + spec->adc_nids = stac9872_adc_nids; + spec->num_muxes = ARRAY_SIZE(stac9872_mux_nids); + spec->mux_nids = stac9872_mux_nids; + spec->mixer = stac9872_mixer; + spec->init = vaio_init; + + err = stac92xx_parse_auto_config(codec, 0x10, 0x12); + if (err < 0) { + stac92xx_free(codec); + return -EINVAL; + } + spec->input_mux = &spec->private_imux; + codec->patch_ops = stac92xx_patch_ops; + return 0; + } + + switch (spec->board_config) { case CXD9872RD_VAIO: case STAC9872AK_VAIO: case STAC9872K_VAIO: -- cgit v1.2.3-70-g09d2 From 41b5b01afb71226653282951965d5efa9d7b843d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 20 Jan 2009 18:21:23 +0100 Subject: ALSA: hda - Don't break the PCM creation loop Don't break the loop in snd_hda_codec_build_pcms() even if the item has no substreams. It's possible that it's an empty item and the next item containing the valid substreams (e.g. realtek codecs may create the analog and alt-analog but no digitl streams). Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 2d6f72ca014..0129e95672a 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2613,7 +2613,7 @@ int snd_hda_codec_build_pcms(struct hda_codec *codec) int dev; if (!cpcm->stream[0].substreams && !cpcm->stream[1].substreams) - return 0; /* no substreams assigned */ + continue; /* no substreams assigned */ if (!cpcm->pcm) { dev = get_empty_pcm_device(codec->bus, cpcm->pcm_type); -- cgit v1.2.3-70-g09d2 From 2297bd6e526ce1469279284ffda9140f8d60ea84 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 20 Jan 2009 18:24:13 +0100 Subject: ALSA: hda - Check HDMI jack types in the auto configuration Add dig_out_type and dig_in_type fields to autocfg struct. A proper HDA_PCM_TYPE_* value is assigned to these fields according to the pin-jack location type value. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 8 ++++++++ sound/pci/hda/hda_local.h | 2 ++ 2 files changed, 10 insertions(+) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 0129e95672a..dd419ce43d9 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3392,10 +3392,18 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, case AC_JACK_SPDIF_OUT: case AC_JACK_DIG_OTHER_OUT: cfg->dig_out_pin = nid; + if (loc == AC_JACK_LOC_HDMI) + cfg->dig_out_type = HDA_PCM_TYPE_HDMI; + else + cfg->dig_out_type = HDA_PCM_TYPE_SPDIF; break; case AC_JACK_SPDIF_IN: case AC_JACK_DIG_OTHER_IN: cfg->dig_in_pin = nid; + if (loc == AC_JACK_LOC_HDMI) + cfg->dig_in_type = HDA_PCM_TYPE_HDMI; + else + cfg->dig_in_type = HDA_PCM_TYPE_SPDIF; break; } } diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 1dd8716c387..a4ecd77a451 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -355,6 +355,8 @@ struct auto_pin_cfg { hda_nid_t dig_out_pin; hda_nid_t dig_in_pin; hda_nid_t mono_out_pin; + int dig_out_type; /* HDA_PCM_TYPE_XXX */ + int dig_in_type; /* HDA_PCM_TYPE_XXX */ }; #define get_defcfg_connect(cfg) \ -- cgit v1.2.3-70-g09d2 From 8c441982fdc00f77b7aa609061c6411f47bcceda Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 20 Jan 2009 18:30:20 +0100 Subject: ALSA: hda - Assign proper digital I/O type for STAC/IDT Assign the proper PCM digital I/O type (HDA_PCM_TYPE_*) for the digital I/O on STAC/IDT codecs. HDA_PCM_TYPE_HDMI is assigned for the HDMI I/O. A similar framework is implemented to patch_realtek.c, but it's not set up and still using only HDA_PCM_TYPE_SPDIF yet. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 6 +++++- sound/pci/hda/patch_sigmatel.c | 2 +- 2 files changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 5d249a547fb..4fdae06162e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -269,6 +269,7 @@ struct alc_spec { * dig_out_nid and hp_nid are optional */ hda_nid_t alt_dac_nid; + int dig_out_type; /* capture */ unsigned int num_adc_nids; @@ -3087,7 +3088,10 @@ static int alc_build_pcms(struct hda_codec *codec) codec->num_pcms = 2; info = spec->pcm_rec + 1; info->name = spec->stream_name_digital; - info->pcm_type = HDA_PCM_TYPE_SPDIF; + if (spec->dig_out_type) + info->pcm_type = spec->dig_out_type; + else + info->pcm_type = HDA_PCM_TYPE_SPDIF; if (spec->multiout.dig_out_nid && spec->stream_digital_playback) { info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *(spec->stream_digital_playback); diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index b6e797d1c21..1dd448e85bc 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2553,7 +2553,7 @@ static int stac92xx_build_pcms(struct hda_codec *codec) codec->num_pcms++; info++; info->name = "STAC92xx Digital"; - info->pcm_type = HDA_PCM_TYPE_SPDIF; + info->pcm_type = spec->autocfg.dig_out_type; if (spec->multiout.dig_out_nid) { info->stream[SNDRV_PCM_STREAM_PLAYBACK] = stac92xx_pcm_digital_playback; info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dig_out_nid; -- cgit v1.2.3-70-g09d2 From e64f14f4e570d6ec5bc88abac92a3a27150756d7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 20 Jan 2009 18:32:55 +0100 Subject: ALSA: hda - Allow digital-only I/O on ALC262 codec Some laptops like VAIO have multiple codecs and uses ALC262 only for the SPIDF output without analog I/O. So far, the codec-parser assumes the presence of analog I/O and returned an error for such a case. This patch adds some hacks to allow the digital-only configuration for ALC262. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 43 +++++++++++++++++++++++++++++++++---------- 1 file changed, 33 insertions(+), 10 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4fdae06162e..4cfa78c5439 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -306,6 +306,9 @@ struct alc_spec { unsigned int jack_present: 1; unsigned int master_sw: 1; + /* other flags */ + unsigned int no_analog :1; /* digital I/O only */ + /* for virtual master */ hda_nid_t vmaster_nid; #ifdef CONFIG_SND_HDA_POWER_SAVE @@ -2019,11 +2022,13 @@ static int alc_build_controls(struct hda_codec *codec) spec->multiout.dig_out_nid); if (err < 0) return err; - err = snd_hda_create_spdif_share_sw(codec, - &spec->multiout); - if (err < 0) - return err; - spec->multiout.share_spdif = 1; + if (!spec->no_analog) { + err = snd_hda_create_spdif_share_sw(codec, + &spec->multiout); + if (err < 0) + return err; + spec->multiout.share_spdif = 1; + } } if (spec->dig_in_nid) { err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid); @@ -2032,7 +2037,8 @@ static int alc_build_controls(struct hda_codec *codec) } /* if we have no master control, let's create it */ - if (!snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) { + if (!spec->no_analog && + !snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) { unsigned int vmaster_tlv[4]; snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid, HDA_OUTPUT, vmaster_tlv); @@ -2041,7 +2047,8 @@ static int alc_build_controls(struct hda_codec *codec) if (err < 0) return err; } - if (!snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) { + if (!spec->no_analog && + !snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) { err = snd_hda_add_vmaster(codec, "Master Playback Switch", NULL, alc_slave_sws); if (err < 0) @@ -3060,6 +3067,9 @@ static int alc_build_pcms(struct hda_codec *codec) codec->num_pcms = 1; codec->pcm_info = info; + if (spec->no_analog) + goto skip_analog; + info->name = spec->stream_name_analog; if (spec->stream_analog_playback) { if (snd_BUG_ON(!spec->multiout.dac_nids)) @@ -3083,6 +3093,7 @@ static int alc_build_pcms(struct hda_codec *codec) } } + skip_analog: /* SPDIF for stream index #1 */ if (spec->multiout.dig_out_nid || spec->dig_in_nid) { codec->num_pcms = 2; @@ -3106,6 +3117,9 @@ static int alc_build_pcms(struct hda_codec *codec) codec->spdif_status_reset = 1; } + if (spec->no_analog) + return 0; + /* If the use of more than one ADC is requested for the current * model, configure a second analog capture-only PCM. */ @@ -10468,8 +10482,14 @@ static int alc262_parse_auto_config(struct hda_codec *codec) alc262_ignore); if (err < 0) return err; - if (!spec->autocfg.line_outs) + if (!spec->autocfg.line_outs) { + if (spec->autocfg.dig_out_pin || spec->autocfg.dig_in_pin) { + spec->multiout.max_channels = 2; + spec->no_analog = 1; + goto dig_only; + } return 0; /* can't find valid BIOS pin config */ + } err = alc262_auto_create_multi_out_ctls(spec, &spec->autocfg); if (err < 0) return err; @@ -10479,8 +10499,11 @@ static int alc262_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_out_pin) + dig_only: + if (spec->autocfg.dig_out_pin) { spec->multiout.dig_out_nid = ALC262_DIGOUT_NID; + spec->dig_out_type = spec->autocfg.dig_out_type; + } if (spec->autocfg.dig_in_pin) spec->dig_in_nid = ALC262_DIGIN_NID; @@ -10875,7 +10898,7 @@ static int patch_alc262(struct hda_codec *codec) spec->capsrc_nids = alc262_capsrc_nids; } } - if (!spec->cap_mixer) + if (!spec->cap_mixer && !spec->no_analog) set_capture_mixer(spec); spec->vmaster_nid = 0x0c; -- cgit v1.2.3-70-g09d2 From 75d91f9bc6d36b8d0ceef1cb75a4ac2b5c8a51d0 Mon Sep 17 00:00:00 2001 From: Timur Tabi Date: Mon, 19 Jan 2009 11:57:46 -0600 Subject: ASoC: Allow Freescale MPC8610 audio drivers to be compiled as modules Change the Kconfig and Makefile options for Freescale MPC8610 audio drivers so that they can be compiled as modules, and simplify the Kconfig choices so that only the platform is selected. Also fix the naming of the driver files to conform to ALSA standards. [Removed extraneous SND_SOC dependency -- broonie] Signed-off-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/fsl/Kconfig | 16 ++++++++-------- sound/soc/fsl/Makefile | 7 +++++-- 2 files changed, 13 insertions(+), 10 deletions(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 95c12b26fe3..c7c78c39cfe 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -1,17 +1,17 @@ config SND_SOC_OF_SIMPLE tristate +# ASoC platform support for the Freescale MPC8610 SOC. This compiles drivers +# for the SSI and the Elo DMA controller. You will still need to select +# a platform driver and a codec driver. config SND_SOC_MPC8610 - bool "ALSA SoC support for the MPC8610 SOC" - depends on MPC8610_HPCD - default y if MPC8610 - help - Say Y if you want to add support for codecs attached to the SSI - device on an MPC8610. + tristate + depends on MPC8610 config SND_SOC_MPC8610_HPCD - bool "ALSA SoC support for the Freescale MPC8610 HPCD board" - depends on SND_SOC_MPC8610 + tristate "ALSA SoC support for the Freescale MPC8610 HPCD board" + depends on MPC8610_HPCD + select SND_SOC_MPC8610 select SND_SOC_CS4270 select SND_SOC_CS4270_VD33_ERRATA default y if MPC8610_HPCD diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 035da4afec3..f85134c8638 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -2,10 +2,13 @@ obj-$(CONFIG_SND_SOC_OF_SIMPLE) += soc-of-simple.o # MPC8610 HPCD Machine Support -obj-$(CONFIG_SND_SOC_MPC8610_HPCD) += mpc8610_hpcd.o +snd-soc-mpc8610-hpcd-objs := mpc8610_hpcd.o +obj-$(CONFIG_SND_SOC_MPC8610_HPCD) += snd-soc-mpc8610-hpcd.o # MPC8610 Platform Support -obj-$(CONFIG_SND_SOC_MPC8610) += fsl_ssi.o fsl_dma.o +snd-soc-fsl-ssi-objs := fsl_ssi.o +snd-soc-fsl-dma-objs := fsl_dma.o +obj-$(CONFIG_SND_SOC_MPC8610) += snd-soc-fsl-ssi.o snd-soc-fsl-dma.o obj-$(CONFIG_SND_SOC_MPC5200_I2S) += mpc5200_psc_i2s.o -- cgit v1.2.3-70-g09d2 From 927b0aea93bb324d743e575659e10d6d76818e4b Mon Sep 17 00:00:00 2001 From: Ian Molton Date: Mon, 19 Jan 2009 17:23:11 +0000 Subject: ASoC: Fix WM9705 capture switch name This patch fixes the acpture switch name so that it better reflects its purpose. Signed-off-by: Ian Molton Signed-off-by: Mark Brown --- sound/soc/codecs/wm9705.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index cb26b6a77ff..5e1937ac0b5 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -57,8 +57,8 @@ static const struct snd_kcontrol_new wm9705_snd_ac97_controls[] = { SOC_DOUBLE("CD Playback Volume", AC97_CD, 8, 0, 31, 1), SOC_SINGLE("Mic Playback Volume", AC97_MIC, 0, 31, 1), SOC_SINGLE("Mic 20dB Boost Switch", AC97_MIC, 6, 1, 0), - SOC_DOUBLE("PCM Capture Volume", AC97_REC_GAIN, 8, 0, 15, 0), - SOC_SINGLE("PCM Capture Switch", AC97_REC_GAIN, 15, 1, 1), + SOC_DOUBLE("Capture Volume", AC97_REC_GAIN, 8, 0, 15, 0), + SOC_SINGLE("Capture Switch", AC97_REC_GAIN, 15, 1, 1), }; static const char *wm9705_mic[] = {"Mic 1", "Mic 2"}; -- cgit v1.2.3-70-g09d2 From 1e137f929bb490ff615ea475ac3904d58b0cdd5e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 21 Jan 2009 07:41:22 +0100 Subject: ALSA: hda - Clean up old VAIO hack codes for STAC9872 Get rid of old VAIO static hack codes for STAC9872 and use the BIOS auto-parser for all models. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 238 ++++------------------------------------- 1 file changed, 21 insertions(+), 217 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 775f8581906..dbe8b1201ef 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -5351,172 +5351,12 @@ static int patch_stac9205(struct hda_codec *codec) * STAC9872 hack */ -/* static config for Sony VAIO FE550G and Sony VAIO AR */ -static hda_nid_t vaio_dacs[] = { 0x2 }; -#define VAIO_HP_DAC 0x5 -static hda_nid_t vaio_adcs[] = { 0x8 /*,0x6*/ }; -static hda_nid_t vaio_mux_nids[] = { 0x15 }; - -static struct hda_input_mux vaio_mux = { - .num_items = 3, - .items = { - /* { "HP", 0x0 }, */ - { "Mic Jack", 0x1 }, - { "Internal Mic", 0x2 }, - { "PCM", 0x3 }, - } -}; - -static struct hda_verb vaio_init[] = { - {0x0a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, /* HP <- 0x2 */ - {0x0a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | STAC_HP_EVENT}, - {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, /* Speaker <- 0x5 */ - {0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? (<- 0x2) */ - {0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, /* CD */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? */ - {0x15, AC_VERB_SET_CONNECT_SEL, 0x1}, /* mic-sel: 0a,0d,14,02 */ - {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* HP */ - {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Speaker */ - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* capture sw/vol -> 0x8 */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* CD-in -> 0x6 */ - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Mic-in -> 0x9 */ - {} -}; - -static struct hda_verb vaio_ar_init[] = { - {0x0a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, /* HP <- 0x2 */ - {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, /* Speaker <- 0x5 */ - {0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? (<- 0x2) */ - {0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, /* CD */ -/* {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },*/ /* Optical Out */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? */ +static struct hda_verb stac9872_core_init[] = { {0x15, AC_VERB_SET_CONNECT_SEL, 0x1}, /* mic-sel: 0a,0d,14,02 */ - {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* HP */ - {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Speaker */ -/* {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},*/ /* Optical Out */ - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* capture sw/vol -> 0x8 */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* CD-in -> 0x6 */ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Mic-in -> 0x9 */ {} }; -static struct snd_kcontrol_new vaio_mixer[] = { - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x02, 0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x02, 0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x05, 0, HDA_OUTPUT), - HDA_CODEC_MUTE("Speaker Playback Switch", 0x05, 0, HDA_OUTPUT), - /* HDA_CODEC_VOLUME("CD Capture Volume", 0x07, 0, HDA_INPUT), */ - HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x09, 0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .count = 1, - .info = stac92xx_mux_enum_info, - .get = stac92xx_mux_enum_get, - .put = stac92xx_mux_enum_put, - }, - {} -}; - -static struct snd_kcontrol_new vaio_ar_mixer[] = { - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x02, 0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x02, 0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x05, 0, HDA_OUTPUT), - HDA_CODEC_MUTE("Speaker Playback Switch", 0x05, 0, HDA_OUTPUT), - /* HDA_CODEC_VOLUME("CD Capture Volume", 0x07, 0, HDA_INPUT), */ - HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x09, 0, HDA_INPUT), - /*HDA_CODEC_MUTE("Optical Out Switch", 0x10, 0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Optical Out Volume", 0x10, 0, HDA_OUTPUT),*/ - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .count = 1, - .info = stac92xx_mux_enum_info, - .get = stac92xx_mux_enum_get, - .put = stac92xx_mux_enum_put, - }, - {} -}; - -static struct hda_codec_ops stac9872_patch_ops = { - .build_controls = stac92xx_build_controls, - .build_pcms = stac92xx_build_pcms, - .init = stac92xx_init, - .free = stac92xx_free, -#ifdef SND_HDA_NEEDS_RESUME - .resume = stac92xx_resume, -#endif -}; - -static int stac9872_vaio_init(struct hda_codec *codec) -{ - int err; - - err = stac92xx_init(codec); - if (err < 0) - return err; - if (codec->patch_ops.unsol_event) - codec->patch_ops.unsol_event(codec, STAC_HP_EVENT << 26); - return 0; -} - -static void stac9872_vaio_hp_detect(struct hda_codec *codec, unsigned int res) -{ - if (get_pin_presence(codec, 0x0a)) { - stac92xx_reset_pinctl(codec, 0x0f, AC_PINCTL_OUT_EN); - stac92xx_set_pinctl(codec, 0x0a, AC_PINCTL_OUT_EN); - } else { - stac92xx_reset_pinctl(codec, 0x0a, AC_PINCTL_OUT_EN); - stac92xx_set_pinctl(codec, 0x0f, AC_PINCTL_OUT_EN); - } -} - -static void stac9872_vaio_unsol_event(struct hda_codec *codec, unsigned int res) -{ - switch (res >> 26) { - case STAC_HP_EVENT: - stac9872_vaio_hp_detect(codec, res); - break; - } -} - -static struct hda_codec_ops stac9872_vaio_patch_ops = { - .build_controls = stac92xx_build_controls, - .build_pcms = stac92xx_build_pcms, - .init = stac9872_vaio_init, - .free = stac92xx_free, - .unsol_event = stac9872_vaio_unsol_event, -#ifdef CONFIG_PM - .resume = stac92xx_resume, -#endif -}; - -enum { /* FE and SZ series. id=0x83847661 and subsys=0x104D0700 or 104D1000. */ - CXD9872RD_VAIO, - /* Unknown. id=0x83847662 and subsys=0x104D1200 or 104D1000. */ - STAC9872AK_VAIO, - /* Unknown. id=0x83847661 and subsys=0x104D1200. */ - STAC9872K_VAIO, - /* AR Series. id=0x83847664 and subsys=104D1300 */ - CXD9872AKD_VAIO, - STAC_9872_MODELS, -}; - -static const char *stac9872_models[STAC_9872_MODELS] = { - [CXD9872RD_VAIO] = "vaio", - [CXD9872AKD_VAIO] = "vaio-ar", -}; - -static struct snd_pci_quirk stac9872_cfg_tbl[] = { - SND_PCI_QUIRK(0x104d, 0x81e6, "Sony VAIO F/S", CXD9872RD_VAIO), - SND_PCI_QUIRK(0x104d, 0x81ef, "Sony VAIO F/S", CXD9872RD_VAIO), - SND_PCI_QUIRK(0x104d, 0x81fd, "Sony VAIO AR", CXD9872AKD_VAIO), - SND_PCI_QUIRK(0x104d, 0x8205, "Sony VAIO AR", CXD9872AKD_VAIO), - {} -}; - static struct snd_kcontrol_new stac9872_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x09, 0, HDA_INPUT), @@ -5540,72 +5380,36 @@ static hda_nid_t stac9872_mux_nids[] = { static int patch_stac9872(struct hda_codec *codec) { struct sigmatel_spec *spec; + int err; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) return -ENOMEM; codec->spec = spec; +#if 0 /* no model right now */ spec->board_config = snd_hda_check_board_config(codec, STAC_9872_MODELS, stac9872_models, stac9872_cfg_tbl); - if (spec->board_config < 0) { - int err; - - spec->num_pins = ARRAY_SIZE(stac9872_pin_nids); - spec->pin_nids = stac9872_pin_nids; - spec->multiout.dac_nids = spec->dac_nids; - spec->num_adcs = ARRAY_SIZE(stac9872_adc_nids); - spec->adc_nids = stac9872_adc_nids; - spec->num_muxes = ARRAY_SIZE(stac9872_mux_nids); - spec->mux_nids = stac9872_mux_nids; - spec->mixer = stac9872_mixer; - spec->init = vaio_init; - - err = stac92xx_parse_auto_config(codec, 0x10, 0x12); - if (err < 0) { - stac92xx_free(codec); - return -EINVAL; - } - spec->input_mux = &spec->private_imux; - codec->patch_ops = stac92xx_patch_ops; - return 0; - } - - switch (spec->board_config) { - case CXD9872RD_VAIO: - case STAC9872AK_VAIO: - case STAC9872K_VAIO: - spec->mixer = vaio_mixer; - spec->init = vaio_init; - spec->multiout.max_channels = 2; - spec->multiout.num_dacs = ARRAY_SIZE(vaio_dacs); - spec->multiout.dac_nids = vaio_dacs; - spec->multiout.hp_nid = VAIO_HP_DAC; - spec->num_adcs = ARRAY_SIZE(vaio_adcs); - spec->adc_nids = vaio_adcs; - spec->num_pwrs = 0; - spec->input_mux = &vaio_mux; - spec->mux_nids = vaio_mux_nids; - codec->patch_ops = stac9872_vaio_patch_ops; - break; - - case CXD9872AKD_VAIO: - spec->mixer = vaio_ar_mixer; - spec->init = vaio_ar_init; - spec->multiout.max_channels = 2; - spec->multiout.num_dacs = ARRAY_SIZE(vaio_dacs); - spec->multiout.dac_nids = vaio_dacs; - spec->multiout.hp_nid = VAIO_HP_DAC; - spec->num_adcs = ARRAY_SIZE(vaio_adcs); - spec->num_pwrs = 0; - spec->adc_nids = vaio_adcs; - spec->input_mux = &vaio_mux; - spec->mux_nids = vaio_mux_nids; - codec->patch_ops = stac9872_patch_ops; - break; - } +#endif + spec->num_pins = ARRAY_SIZE(stac9872_pin_nids); + spec->pin_nids = stac9872_pin_nids; + spec->multiout.dac_nids = spec->dac_nids; + spec->num_adcs = ARRAY_SIZE(stac9872_adc_nids); + spec->adc_nids = stac9872_adc_nids; + spec->num_muxes = ARRAY_SIZE(stac9872_mux_nids); + spec->mux_nids = stac9872_mux_nids; + spec->mixer = stac9872_mixer; + spec->init = stac9872_core_init; + + err = stac92xx_parse_auto_config(codec, 0x10, 0x12); + if (err < 0) { + stac92xx_free(codec); + return -EINVAL; + } + spec->input_mux = &spec->private_imux; + codec->patch_ops = stac92xx_patch_ops; return 0; } -- cgit v1.2.3-70-g09d2 From 08989930f91e4802b94e03eb54e5385bac112811 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 21 Jan 2009 07:43:23 +0100 Subject: ALSA: hda - Remove old models for STAC9872 from the document Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 64eb1100eec..75914bcdce7 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -352,5 +352,4 @@ STAC92HD83* STAC9872 ======== - vaio Setup for VAIO FE550G/SZ110 - vaio-ar Setup for VAIO AR + N/A -- cgit v1.2.3-70-g09d2 From 48972cc5101dee24243c1b53d409cc27880e7a29 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Wed, 21 Jan 2009 08:18:16 +0100 Subject: ALSA: cmi8330: add OPL3 support Add OPL3 handling to the driver and volume control for FM synthesis. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/Kconfig | 1 + sound/isa/cmi8330.c | 30 ++++++++++++++++++++++++++++-- 2 files changed, 29 insertions(+), 2 deletions(-) diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index ce0aa044e27..be2d377ff90 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -94,6 +94,7 @@ config SND_CMI8330 tristate "C-Media CMI8330" select SND_WSS_LIB select SND_SB16_DSP + select SND_OPL3_LIB help Say Y here to include support for soundcards based on the C-Media CMI8330 chip. diff --git a/sound/isa/cmi8330.c b/sound/isa/cmi8330.c index e49aec700a5..dec6ea52cc4 100644 --- a/sound/isa/cmi8330.c +++ b/sound/isa/cmi8330.c @@ -51,6 +51,7 @@ #include #include #include +#include #include #include @@ -79,6 +80,7 @@ static int sbdma16[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; static long wssport[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; static int wssirq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; static int wssdma[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; +static long fmport[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for CMI8330 soundcard."); @@ -107,6 +109,8 @@ MODULE_PARM_DESC(wssirq, "IRQ # for CMI8330 WSS driver."); module_param_array(wssdma, int, NULL, 0444); MODULE_PARM_DESC(wssdma, "DMA for CMI8330 WSS driver."); +module_param_array(fmport, long, NULL, 0444); +MODULE_PARM_DESC(fmport, "FM port # for CMI8330 driver."); #ifdef CONFIG_PNP static int isa_registered; static int pnp_registered; @@ -219,8 +223,10 @@ WSS_SINGLE("3D Control - Switch", 0, CMI8330_RMUX3D, 5, 1, 1), WSS_SINGLE("PC Speaker Playback Volume", 0, CMI8330_OUTPUTVOL, 3, 3, 0), -WSS_SINGLE("FM Playback Switch", 0, - CMI8330_RECMUX, 3, 1, 1), +WSS_DOUBLE("FM Playback Switch", 0, + CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), +WSS_DOUBLE("FM Playback Volume", 0, + CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1), WSS_SINGLE(SNDRV_CTL_NAME_IEC958("Input ", CAPTURE, SWITCH), 0, CMI8330_RMUX3D, 7, 1, 1), WSS_SINGLE(SNDRV_CTL_NAME_IEC958("Input ", PLAYBACK, SWITCH), 0, @@ -333,6 +339,7 @@ static int __devinit snd_cmi8330_pnp(int dev, struct snd_cmi8330 *acard, wssport[dev] = pnp_port_start(pdev, 0); wssdma[dev] = pnp_dma(pdev, 0); wssirq[dev] = pnp_irq(pdev, 0); + fmport[dev] = pnp_port_start(pdev, 1); /* allocate SB16 resources */ pdev = acard->play; @@ -487,6 +494,7 @@ static int __devinit snd_cmi8330_probe(struct snd_card *card, int dev) { struct snd_cmi8330 *acard; int i, err; + struct snd_opl3 *opl3; acard = card->private_data; err = snd_wss_create(card, wssport[dev] + 4, -1, @@ -530,6 +538,24 @@ static int __devinit snd_cmi8330_probe(struct snd_card *card, int dev) snd_printk(KERN_ERR PFX "failed to create pcms\n"); return err; } + if (fmport[dev] != SNDRV_AUTO_PORT) { + if (snd_opl3_create(card, + fmport[dev], fmport[dev] + 2, + OPL3_HW_AUTO, 0, &opl3) < 0) { + snd_printk(KERN_ERR PFX + "no OPL device at 0x%lx-0x%lx ?\n", + fmport[dev], fmport[dev] + 2); + } else { + err = snd_opl3_timer_new(opl3, 0, 1); + if (err < 0) + return err; + + err = snd_opl3_hwdep_new(opl3, 0, 1, NULL); + if (err < 0) + return err; + } + } + strcpy(card->driver, "CMI8330/C3D"); strcpy(card->shortname, "C-Media CMI8330/C3D"); -- cgit v1.2.3-70-g09d2 From c9864fd30a28aceef5293f28559c4a2f5a20d7d5 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Wed, 21 Jan 2009 08:19:27 +0100 Subject: ALSA: sscape: use common MPU401 macros Remove local macros which redefines the common ones. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/sscape.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index 6a7f842b962..681e2237acb 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -89,9 +89,6 @@ MODULE_DEVICE_TABLE(pnp_card, sscape_pnpids); #endif -#define MPU401_IO(i) ((i) + 0) -#define MIDI_DATA_IO(i) ((i) + 0) -#define MIDI_CTRL_IO(i) ((i) + 1) #define HOST_CTRL_IO(i) ((i) + 2) #define HOST_DATA_IO(i) ((i) + 3) #define ODIE_ADDR_IO(i) ((i) + 4) @@ -327,7 +324,7 @@ static int host_write_ctrl_unsafe(unsigned io_base, unsigned char data, */ static inline int verify_mpu401(const struct snd_mpu401 * mpu) { - return ((inb(MIDI_CTRL_IO(mpu->port)) & 0xc0) == 0x80); + return ((inb(MPU401C(mpu)) & 0xc0) == 0x80); } /* @@ -335,7 +332,7 @@ static inline int verify_mpu401(const struct snd_mpu401 * mpu) */ static inline void initialise_mpu401(const struct snd_mpu401 * mpu) { - outb(0, MIDI_DATA_IO(mpu->port)); + outb(0, MPU401D(mpu)); } /* @@ -1191,12 +1188,11 @@ static int __devinit create_sscape(int dev, struct snd_card *card) } #define MIDI_DEVNUM 0 if (sscape->type != SSCAPE_VIVO) { - err = create_mpu401(card, MIDI_DEVNUM, - MPU401_IO(xport), mpu_irq[dev]); + err = create_mpu401(card, MIDI_DEVNUM, xport, mpu_irq[dev]); if (err < 0) { printk(KERN_ERR "sscape: Failed to create " "MPU-401 device at 0x%x\n", - MPU401_IO(xport)); + xport); goto _release_dma; } -- cgit v1.2.3-70-g09d2 From 80c509fdd74f3b158267374cc55156965c8bf930 Mon Sep 17 00:00:00 2001 From: Steve Sakoman Date: Tue, 20 Jan 2009 23:05:27 -0800 Subject: ASoC: Complete Beagleboard support Commit dc06102a0c8b5aa0dd7f9a40ce241e793c252a87 in the asoc tree did not include the necessary Kconfig and Makefile changes. This patch completes the support for Beagleboard Signed-off-by: Steve Sakoman Acked-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/omap/Kconfig | 10 ++++++++++ sound/soc/omap/Makefile | 2 ++ 2 files changed, 12 insertions(+) diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 4f7f0401458..ccd8973683d 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -55,3 +55,13 @@ config SND_OMAP_SOC_OMAP3_PANDORA select SND_SOC_TWL4030 help Say Y if you want to add support for SoC audio on the OMAP3 Pandora. + +config SND_OMAP_SOC_OMAP3_BEAGLE + tristate "SoC Audio support for OMAP3 Beagle" + depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3_BEAGLE + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for SoC audio on the Beagleboard. + + diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 76fedd96e36..0c9e4ac3766 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -12,6 +12,7 @@ snd-soc-overo-objs := overo.o snd-soc-omap2evm-objs := omap2evm.o snd-soc-sdp3430-objs := sdp3430.o snd-soc-omap3pandora-objs := omap3pandora.o +snd-soc-omap3beagle-objs := omap3beagle.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o @@ -19,3 +20,4 @@ obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o +obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o -- cgit v1.2.3-70-g09d2 From aa9c293ae46d71f5add0761bce8db67b162e3f29 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Wed, 21 Jan 2009 15:08:03 +0100 Subject: ALSA: do not create OPL3 timers if there is no OPL3 irq wired Most cards have OPL3 FM synthetiser but they do not have OPL3 interrupt wired to a sound chip or CPU. Do not create OPL3 timers for such cards as the timers are useless witthout interrupt. This patch removes OPL3 timers for following alsa drivers: snd-ad1816a, snd-opti93x, snd-opti92x, snd-sc6000, snd-cmi8330. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/ad1816a/ad1816a.c | 7 ++----- sound/isa/cmi8330.c | 4 ---- sound/isa/opti9xx/opti92x-ad1848.c | 10 ++-------- sound/isa/sc6000.c | 4 ---- 4 files changed, 4 insertions(+), 21 deletions(-) diff --git a/sound/isa/ad1816a/ad1816a.c b/sound/isa/ad1816a/ad1816a.c index 77524244a84..3810833d3a8 100644 --- a/sound/isa/ad1816a/ad1816a.c +++ b/sound/isa/ad1816a/ad1816a.c @@ -207,11 +207,8 @@ static int __devinit snd_card_ad1816a_probe(int dev, struct pnp_card_link *pcard OPL3_HW_AUTO, 0, &opl3) < 0) { printk(KERN_ERR PFX "no OPL device at 0x%lx-0x%lx.\n", fm_port[dev], fm_port[dev] + 2); } else { - if ((error = snd_opl3_timer_new(opl3, 1, 2)) < 0) { - snd_card_free(card); - return error; - } - if ((error = snd_opl3_hwdep_new(opl3, 0, 1, NULL)) < 0) { + error = snd_opl3_hwdep_new(opl3, 0, 1, NULL); + if (error < 0) { snd_card_free(card); return error; } diff --git a/sound/isa/cmi8330.c b/sound/isa/cmi8330.c index dec6ea52cc4..11543795741 100644 --- a/sound/isa/cmi8330.c +++ b/sound/isa/cmi8330.c @@ -546,10 +546,6 @@ static int __devinit snd_cmi8330_probe(struct snd_card *card, int dev) "no OPL device at 0x%lx-0x%lx ?\n", fmport[dev], fmport[dev] + 2); } else { - err = snd_opl3_timer_new(opl3, 0, 1); - if (err < 0) - return err; - err = snd_opl3_hwdep_new(opl3, 0, 1, NULL); if (err < 0) return err; diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index 19706b0d849..5deb7e69a02 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -815,14 +815,8 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) chip->fm_port, chip->fm_port + 4 - 1); } if (opl3) { -#ifdef CS4231 - const int t1dev = 1; -#else - const int t1dev = 0; -#endif - if ((error = snd_opl3_timer_new(opl3, t1dev, t1dev+1)) < 0) - return error; - if ((error = snd_opl3_hwdep_new(opl3, 0, 1, &synth)) < 0) + error = snd_opl3_hwdep_new(opl3, 0, 1, &synth); + if (error < 0) return error; } } diff --git a/sound/isa/sc6000.c b/sound/isa/sc6000.c index ca35924dc3b..bbc53692e68 100644 --- a/sound/isa/sc6000.c +++ b/sound/isa/sc6000.c @@ -576,10 +576,6 @@ static int __devinit snd_sc6000_probe(struct device *devptr, unsigned int dev) snd_printk(KERN_ERR PFX "no OPL device at 0x%x-0x%x ?\n", 0x388, 0x388 + 2); } else { - err = snd_opl3_timer_new(opl3, 0, 1); - if (err < 0) - goto err_unmap2; - err = snd_opl3_hwdep_new(opl3, 0, 1, NULL); if (err < 0) goto err_unmap2; -- cgit v1.2.3-70-g09d2 From a17ac45a5da76f851faf0b6502f66c1205159469 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Wed, 21 Jan 2009 15:14:09 +0100 Subject: ALSA: ad1816a: enable hardware timer Enable hardware timer with 10 usec resolution. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- include/sound/ad1816a.h | 2 ++ sound/isa/ad1816a/ad1816a.c | 7 +++++++ sound/isa/ad1816a/ad1816a_lib.c | 5 ----- 3 files changed, 9 insertions(+), 5 deletions(-) diff --git a/include/sound/ad1816a.h b/include/sound/ad1816a.h index b3aa62ee3c8..d010858c33c 100644 --- a/include/sound/ad1816a.h +++ b/include/sound/ad1816a.h @@ -169,5 +169,7 @@ extern int snd_ad1816a_create(struct snd_card *card, unsigned long port, extern int snd_ad1816a_pcm(struct snd_ad1816a *chip, int device, struct snd_pcm **rpcm); extern int snd_ad1816a_mixer(struct snd_ad1816a *chip); +extern int snd_ad1816a_timer(struct snd_ad1816a *chip, int device, + struct snd_timer **rtimer); #endif /* __SOUND_AD1816A_H */ diff --git a/sound/isa/ad1816a/ad1816a.c b/sound/isa/ad1816a/ad1816a.c index 3810833d3a8..15f60107a11 100644 --- a/sound/isa/ad1816a/ad1816a.c +++ b/sound/isa/ad1816a/ad1816a.c @@ -156,6 +156,7 @@ static int __devinit snd_card_ad1816a_probe(int dev, struct pnp_card_link *pcard struct snd_card_ad1816a *acard; struct snd_ad1816a *chip; struct snd_opl3 *opl3; + struct snd_timer *timer; if ((card = snd_card_new(index[dev], id[dev], THIS_MODULE, sizeof(struct snd_card_ad1816a))) == NULL) @@ -194,6 +195,12 @@ static int __devinit snd_card_ad1816a_probe(int dev, struct pnp_card_link *pcard return error; } + error = snd_ad1816a_timer(chip, 0, &timer); + if (error < 0) { + snd_card_free(card); + return error; + } + if (mpu_port[dev] > 0) { if (snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, mpu_port[dev], 0, mpu_irq[dev], IRQF_DISABLED, diff --git a/sound/isa/ad1816a/ad1816a_lib.c b/sound/isa/ad1816a/ad1816a_lib.c index 3bfca7c59ba..1c9e01ecac0 100644 --- a/sound/isa/ad1816a/ad1816a_lib.c +++ b/sound/isa/ad1816a/ad1816a_lib.c @@ -377,7 +377,6 @@ static struct snd_pcm_hardware snd_ad1816a_capture = { .fifo_size = 0, }; -#if 0 /* not used now */ static int snd_ad1816a_timer_close(struct snd_timer *timer) { struct snd_ad1816a *chip = snd_timer_chip(timer); @@ -442,8 +441,6 @@ static struct snd_timer_hardware snd_ad1816a_timer_table = { .start = snd_ad1816a_timer_start, .stop = snd_ad1816a_timer_stop, }; -#endif /* not used now */ - static int snd_ad1816a_playback_open(struct snd_pcm_substream *substream) { @@ -687,7 +684,6 @@ int __devinit snd_ad1816a_pcm(struct snd_ad1816a *chip, int device, struct snd_p return 0; } -#if 0 /* not used now */ int __devinit snd_ad1816a_timer(struct snd_ad1816a *chip, int device, struct snd_timer **rtimer) { struct snd_timer *timer; @@ -709,7 +705,6 @@ int __devinit snd_ad1816a_timer(struct snd_ad1816a *chip, int device, struct snd *rtimer = timer; return 0; } -#endif /* not used now */ /* * -- cgit v1.2.3-70-g09d2 From 8ce8419829998c91b33200894a0db5e1441d6952 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 22 Jan 2009 16:59:20 +0100 Subject: ALSA: hda - Avoid to set the pin control again if already set Check the present pin control bit and avoid the write if it's already set in patch_sigmatel.c. This will reduce the number of verb execs at jack plugging. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 20 +++++++++++++------- 1 file changed, 13 insertions(+), 7 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 0fa6c593d1d..11634a4478e 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4126,7 +4126,9 @@ static void stac92xx_free(struct hda_codec *codec) static void stac92xx_set_pinctl(struct hda_codec *codec, hda_nid_t nid, unsigned int flag) { - unsigned int pin_ctl = snd_hda_codec_read(codec, nid, + unsigned int old_ctl, pin_ctl; + + pin_ctl = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0x00); if (pin_ctl & AC_PINCTL_IN_EN) { @@ -4140,14 +4142,17 @@ static void stac92xx_set_pinctl(struct hda_codec *codec, hda_nid_t nid, return; } + old_ctl = pin_ctl; /* if setting pin direction bits, clear the current direction bits first */ if (flag & (AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN)) pin_ctl &= ~(AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN); - snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - pin_ctl | flag); + pin_ctl |= flag; + if (old_ctl != pin_ctl) + snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + pin_ctl); } static void stac92xx_reset_pinctl(struct hda_codec *codec, hda_nid_t nid, @@ -4155,9 +4160,10 @@ static void stac92xx_reset_pinctl(struct hda_codec *codec, hda_nid_t nid, { unsigned int pin_ctl = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0x00); - snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - pin_ctl & ~flag); + if (pin_ctl & flag) + snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + pin_ctl & ~flag); } static int get_pin_presence(struct hda_codec *codec, hda_nid_t nid) -- cgit v1.2.3-70-g09d2 From d9a4268ee92ba1a2355c892a3add1fa66856b510 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 22 Jan 2009 17:40:18 +0100 Subject: ALSA: hda - Add quirk for Gateway %1616 laptop Gateway T1616 laptop needs EAPD always on while the current STAC9205 code turns off per HP plug. Added a new model "eapd" to keep it on. Reference: Novell bnc#467597 https://bugzilla.novell.com/show_bug.cgi?id=467597 Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 + sound/pci/hda/patch_sigmatel.c | 10 +++++++++- 2 files changed, 10 insertions(+), 1 deletion(-) diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 75914bcdce7..ef6b22e2541 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -285,6 +285,7 @@ STAC9205/9254 dell-m42 Dell (unknown) dell-m43 Dell Precision dell-m44 Dell Inspiron + eapd Keep EAPD on (e.g. Gateway T1616) STAC9220/9221 ============= diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 3f85731055c..ed2fa431b03 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -66,6 +66,7 @@ enum { STAC_9205_DELL_M42, STAC_9205_DELL_M43, STAC_9205_DELL_M44, + STAC_9205_EAPD, STAC_9205_MODELS }; @@ -2240,6 +2241,7 @@ static unsigned int *stac9205_brd_tbl[STAC_9205_MODELS] = { [STAC_9205_DELL_M42] = dell_9205_m42_pin_configs, [STAC_9205_DELL_M43] = dell_9205_m43_pin_configs, [STAC_9205_DELL_M44] = dell_9205_m44_pin_configs, + [STAC_9205_EAPD] = NULL, }; static const char *stac9205_models[STAC_9205_MODELS] = { @@ -2247,12 +2249,14 @@ static const char *stac9205_models[STAC_9205_MODELS] = { [STAC_9205_DELL_M42] = "dell-m42", [STAC_9205_DELL_M43] = "dell-m43", [STAC_9205_DELL_M44] = "dell-m44", + [STAC_9205_EAPD] = "eapd", }; static struct snd_pci_quirk stac9205_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_9205_REF), + /* Dell */ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f1, "unknown Dell", STAC_9205_DELL_M42), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f2, @@ -2283,6 +2287,8 @@ static struct snd_pci_quirk stac9205_cfg_tbl[] = { "Dell Inspiron", STAC_9205_DELL_M44), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0228, "Dell Vostro 1500", STAC_9205_DELL_M42), + /* Gateway */ + SND_PCI_QUIRK(0x107b, 0x0565, "Gateway T1616", STAC_9205_EAPD), {} /* terminator */ }; @@ -5320,7 +5326,9 @@ static int patch_stac9205(struct hda_codec *codec) spec->aloopback_mask = 0x40; spec->aloopback_shift = 0; - spec->eapd_switch = 1; + /* Turn on/off EAPD per HP plugging */ + if (spec->board_config != STAC_9205_EAPD) + spec->eapd_switch = 1; spec->multiout.dac_nids = spec->dac_nids; switch (spec->board_config){ -- cgit v1.2.3-70-g09d2 From fd8757aed16470e088ecdad96ffd30f86c34424d Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Wed, 21 Jan 2009 17:45:12 -0500 Subject: Add PCI DFI vendor ID Add a define for DFI PCI vendor id. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai --- include/linux/pci_ids.h | 2 ++ 1 file changed, 2 insertions(+) diff --git a/include/linux/pci_ids.h b/include/linux/pci_ids.h index d543365518a..6b339766b7a 100644 --- a/include/linux/pci_ids.h +++ b/include/linux/pci_ids.h @@ -2106,6 +2106,8 @@ #define PCI_DEVICE_ID_MELLANOX_SINAI_OLD 0x5e8c #define PCI_DEVICE_ID_MELLANOX_SINAI 0x6274 +#define PCI_VENDOR_ID_DFI 0x15bd + #define PCI_VENDOR_ID_QUICKNET 0x15e2 #define PCI_DEVICE_ID_QUICKNET_XJ 0x0500 -- cgit v1.2.3-70-g09d2 From 577aa2c195045599275b54356969ae19f34e7a66 Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Thu, 22 Jan 2009 22:55:44 -0500 Subject: ALSA: hda: add reference board SND_PCI_QUIRK Add another LanParty reference board SND_PCI_QUIRK to quirk lists of all codec families. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 212d8c09a67..3fbe22053b3 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1517,6 +1517,8 @@ static struct snd_pci_quirk stac9200_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, + "DFI LanParty", STAC_REF), /* Dell laptops have BIOS problem */ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01a8, "unknown Dell", STAC_9200_DELL_D21), @@ -1666,6 +1668,7 @@ static struct snd_pci_quirk stac925x_codec_id_cfg_tbl[] = { static struct snd_pci_quirk stac925x_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, "DFI LanParty", STAC_REF), SND_PCI_QUIRK(0x8384, 0x7632, "Stac9202 Reference Board", STAC_REF), /* Default table for unknown ID */ @@ -1709,6 +1712,8 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_92HD73XX_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, + "DFI LanParty", STAC_92HD73XX_REF), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0254, "Dell Studio 1535", STAC_DELL_M6_DMIC), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0255, @@ -1753,6 +1758,8 @@ static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_92HD83XXX_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, + "DFI LanParty", STAC_92HD83XXX_REF), {} /* terminator */ }; @@ -1802,6 +1809,8 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_92HD71BXX_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, + "DFI LanParty", STAC_92HD71BXX_REF), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30f2, "HP dv5", STAC_HP_M4), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30f4, @@ -1992,6 +2001,8 @@ static struct snd_pci_quirk stac922x_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_D945_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, + "DFI LanParty", STAC_D945_REF), /* Intel 945G based systems */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x0101, "Intel D945G", STAC_D945GTP3), @@ -2148,6 +2159,8 @@ static struct snd_pci_quirk stac927x_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_D965_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, + "DFI LanParty", STAC_D965_REF), /* Intel 946 based systems */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x3d01, "Intel D946", STAC_D965_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0xa301, "Intel D946", STAC_D965_3ST), @@ -2259,6 +2272,8 @@ static struct snd_pci_quirk stac9205_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_9205_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, + "DFI LanParty", STAC_9205_REF), /* Dell */ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f1, "unknown Dell", STAC_9205_DELL_M42), -- cgit v1.2.3-70-g09d2 From 8056d47e77a0f7e3c99c114deab4859d31496075 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 23 Jan 2009 09:09:43 +0100 Subject: ALSA: hda - Add model=ref for Intel board with STAC9221 An intel board (8086:0204) works only with model=ref. Reference: Novell bug #406529 https://bugzilla.novell.com/show_bug.cgi?id=406529 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 3fbe22053b3..4ee9f7fc772 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2056,6 +2056,9 @@ static struct snd_pci_quirk stac922x_cfg_tbl[] = { "Intel D945P", STAC_D945GTP3), SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x0707, "Intel D945P", STAC_D945GTP5), + /* other intel */ + SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x0204, + "Intel D945", STAC_D945_REF), /* other systems */ /* Apple Intel Mac (Mac Mini, MacBook, MacBook Pro...) */ SND_PCI_QUIRK(0x8384, 0x7680, -- cgit v1.2.3-70-g09d2 From e3c75964666a27cec46d2cccf2d9806336becd48 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 23 Jan 2009 11:57:22 +0100 Subject: ALSA: hda - Create "Input Source" control dynamically for STAC/IDT Instead of fixed kcontrol_new element, build "Input Source" controls dynamically. If the number of input-source items is 0 or 1, we don't need to create such a control. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 78 ++++++++++++++++++++++++++++-------------- 1 file changed, 53 insertions(+), 25 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index b3c3a02a422..80a4c288b31 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -958,16 +958,6 @@ static struct hda_verb stac9205_core_init[] = { .private_value = HDA_COMPOSE_AMP_VAL(nid, chs, idx, dir) \ } -#define STAC_INPUT_SOURCE(cnt) \ - { \ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ - .name = "Input Source", \ - .count = cnt, \ - .info = stac92xx_mux_enum_info, \ - .get = stac92xx_mux_enum_get, \ - .put = stac92xx_mux_enum_put, \ - } - #define STAC_ANALOG_LOOPBACK(verb_read, verb_write, cnt) \ { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ @@ -982,7 +972,6 @@ static struct hda_verb stac9205_core_init[] = { static struct snd_kcontrol_new stac9200_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0xb, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Master Playback Switch", 0xb, 0, HDA_OUTPUT), - STAC_INPUT_SOURCE(1), HDA_CODEC_VOLUME("Capture Volume", 0x0a, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x0a, 0, HDA_OUTPUT), { } /* end */ @@ -1098,7 +1087,6 @@ static struct snd_kcontrol_new stac92hd83xxx_mixer[] = { }; static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { - STAC_INPUT_SOURCE(2), STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), @@ -1127,7 +1115,6 @@ static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { }; static struct snd_kcontrol_new stac92hd71bxx_mixer[] = { - STAC_INPUT_SOURCE(2), STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), @@ -1141,14 +1128,12 @@ static struct snd_kcontrol_new stac92hd71bxx_mixer[] = { static struct snd_kcontrol_new stac925x_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x0e, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Master Playback Switch", 0x0e, 0, HDA_OUTPUT), - STAC_INPUT_SOURCE(1), HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x14, 0, HDA_OUTPUT), { } /* end */ }; static struct snd_kcontrol_new stac9205_mixer[] = { - STAC_INPUT_SOURCE(2), STAC_ANALOG_LOOPBACK(0xFE0, 0x7E0, 1), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1b, 0x0, HDA_INPUT), @@ -1161,7 +1146,6 @@ static struct snd_kcontrol_new stac9205_mixer[] = { /* This needs to be generated dynamically based on sequence */ static struct snd_kcontrol_new stac922x_mixer[] = { - STAC_INPUT_SOURCE(2), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x17, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x17, 0x0, HDA_INPUT), @@ -1172,7 +1156,6 @@ static struct snd_kcontrol_new stac922x_mixer[] = { static struct snd_kcontrol_new stac927x_mixer[] = { - STAC_INPUT_SOURCE(3), STAC_ANALOG_LOOPBACK(0xFEB, 0x7EB, 1), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x18, 0x0, HDA_INPUT), @@ -2777,22 +2760,37 @@ static struct snd_kcontrol_new stac92xx_control_templates[] = { }; /* add dynamic controls */ -static int stac92xx_add_control_temp(struct sigmatel_spec *spec, - struct snd_kcontrol_new *ktemp, - int idx, const char *name, - unsigned long val) +static struct snd_kcontrol_new * +stac_control_new(struct sigmatel_spec *spec, + struct snd_kcontrol_new *ktemp, + const char *name) { struct snd_kcontrol_new *knew; snd_array_init(&spec->kctls, sizeof(*knew), 32); knew = snd_array_new(&spec->kctls); if (!knew) - return -ENOMEM; + return NULL; *knew = *ktemp; - knew->index = idx; knew->name = kstrdup(name, GFP_KERNEL); - if (!knew->name) + if (!knew->name) { + /* roolback */ + memset(knew, 0, sizeof(*knew)); + spec->kctls.alloced--; + return NULL; + } + return knew; +} + +static int stac92xx_add_control_temp(struct sigmatel_spec *spec, + struct snd_kcontrol_new *ktemp, + int idx, const char *name, + unsigned long val) +{ + struct snd_kcontrol_new *knew = stac_control_new(spec, ktemp, name); + if (!knew) return -ENOMEM; + knew->index = idx; knew->private_value = val; return 0; } @@ -2814,6 +2812,29 @@ static inline int stac92xx_add_control(struct sigmatel_spec *spec, int type, return stac92xx_add_control_idx(spec, type, 0, name, val); } +static struct snd_kcontrol_new stac_input_src_temp = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Input Source", + .info = stac92xx_mux_enum_info, + .get = stac92xx_mux_enum_get, + .put = stac92xx_mux_enum_put, +}; + +static int stac92xx_add_input_source(struct sigmatel_spec *spec) +{ + struct snd_kcontrol_new *knew; + struct hda_input_mux *imux = &spec->private_imux; + + if (!spec->num_adcs || imux->num_items <= 1) + return 0; /* no need for input source control */ + knew = stac_control_new(spec, &stac_input_src_temp, + stac_input_src_temp.name); + if (!knew) + return -ENOMEM; + knew->count = spec->num_adcs; + return 0; +} + /* check whether the line-input can be used as line-out */ static hda_nid_t check_line_out_switch(struct hda_codec *codec) { @@ -3699,6 +3720,10 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out return err; } + err = stac92xx_add_input_source(spec); + if (err < 0) + return err; + spec->multiout.max_channels = spec->multiout.num_dacs * 2; if (spec->multiout.max_channels > 2) spec->surr_switch = 1; @@ -3812,6 +3837,10 @@ static int stac9200_parse_auto_config(struct hda_codec *codec) return err; } + err = stac92xx_add_input_source(spec); + if (err < 0) + return err; + if (spec->autocfg.dig_out_pin) spec->multiout.dig_out_nid = 0x05; if (spec->autocfg.dig_in_pin) @@ -5426,7 +5455,6 @@ static struct hda_verb stac9872_core_init[] = { static struct snd_kcontrol_new stac9872_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x09, 0, HDA_INPUT), - STAC_INPUT_SOURCE(1), { } /* end */ }; -- cgit v1.2.3-70-g09d2 From ff637d38ea6b9c54f708a2b9edabc1b0c73c6d0a Mon Sep 17 00:00:00 2001 From: Timur Tabi Date: Thu, 22 Jan 2009 18:23:39 -0600 Subject: ASoC: remove stand-alone mode support from CS4270 codec driver The CS4270 supports stand-alone mode, where the codec is not connect to the I2C or SPI buses. Instead, input voltages configure the codec at power-on. The CS4270 ASoC device driver has partial support for this mode, but the code was never tested, and partial support doesn't help anyone. It also made the rest of the code more complicated than necessary. [Removed redundant CS4270 dependency on I2C -- broonie] Signed-off-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/codecs/cs4270.c | 92 ++++++++++++++--------------------------------- sound/soc/fsl/Kconfig | 3 +- 2 files changed, 29 insertions(+), 66 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index f1aa0c34421..2e4ce04925e 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -12,11 +12,7 @@ * * Current features/limitations: * - * 1) Software mode is supported. Stand-alone mode is automatically - * selected if I2C is disabled or if a CS4270 is not found on the I2C - * bus. However, stand-alone mode is only partially implemented because - * there is no mechanism yet for this driver and the machine driver to - * communicate the values of the M0, M1, MCLK1, and MCLK2 pins. + * 1) Software mode is supported. Stand-alone mode is not supported. * 2) Only I2C is supported, not SPI * 3) Only Master mode is supported, not Slave. * 4) The machine driver's 'startup' function must call @@ -33,14 +29,6 @@ #include #include -#include "cs4270.h" - -/* If I2C is defined, then we support software mode. However, if we're - not compiled as module but I2C is, then we can't use I2C calls. */ -#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) -#define USE_I2C -#endif - /* Private data for the CS4270 */ struct cs4270_private { unsigned int mclk; /* Input frequency of the MCLK pin */ @@ -60,8 +48,6 @@ struct cs4270_private { SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE) -#ifdef USE_I2C - /* CS4270 registers addresses */ #define CS4270_CHIPID 0x01 /* Chip ID */ #define CS4270_PWRCTL 0x02 /* Power Control */ @@ -271,17 +257,6 @@ static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai, return ret; } -/* - * A list of addresses on which this CS4270 could use. I2C addresses are - * 7 bits. For the CS4270, the upper four bits are always 1001, and the - * lower three bits are determined via the AD2, AD1, and AD0 pins - * (respectively). - */ -static const unsigned short normal_i2c[] = { - 0x48, 0x49, 0x4A, 0x4B, 0x4C, 0x4D, 0x4E, 0x4F, I2C_CLIENT_END -}; -I2C_CLIENT_INSMOD; - /* * Pre-fill the CS4270 register cache. * @@ -476,7 +451,6 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, } #ifdef CONFIG_SND_SOC_CS4270_HWMUTE - /* * Set the CS4270 external mute * @@ -501,32 +475,16 @@ static int cs4270_mute(struct snd_soc_dai *dai, int mute) return snd_soc_write(codec, CS4270_MUTE, reg6); } - +#else +#define cs4270_mute NULL #endif -static int cs4270_i2c_probe(struct i2c_client *, const struct i2c_device_id *); - /* A list of non-DAPM controls that the CS4270 supports */ static const struct snd_kcontrol_new cs4270_snd_controls[] = { SOC_DOUBLE_R("Master Playback Volume", CS4270_VOLA, CS4270_VOLB, 0, 0xFF, 1) }; -static const struct i2c_device_id cs4270_id[] = { - {"cs4270", 0}, - {} -}; -MODULE_DEVICE_TABLE(i2c, cs4270_id); - -static struct i2c_driver cs4270_i2c_driver = { - .driver = { - .name = "CS4270 I2C", - .owner = THIS_MODULE, - }, - .id_table = cs4270_id, - .probe = cs4270_i2c_probe, -}; - /* * Global variable to store socdev for i2c probe function. * @@ -633,7 +591,20 @@ error: return ret; } -#endif /* USE_I2C*/ +static const struct i2c_device_id cs4270_id[] = { + {"cs4270", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, cs4270_id); + +static struct i2c_driver cs4270_i2c_driver = { + .driver = { + .name = "cs4270", + .owner = THIS_MODULE, + }, + .id_table = cs4270_id, + .probe = cs4270_i2c_probe, +}; struct snd_soc_dai cs4270_dai = { .name = "CS4270", @@ -698,7 +669,6 @@ static int cs4270_probe(struct platform_device *pdev) goto error_free_codec; } -#ifdef USE_I2C cs4270_socdev = socdev; ret = i2c_add_driver(&cs4270_i2c_driver); @@ -708,20 +678,16 @@ static int cs4270_probe(struct platform_device *pdev) } /* Did we find a CS4270 on the I2C bus? */ - if (codec->control_data) { - /* Initialize codec ops */ - cs4270_dai.ops.hw_params = cs4270_hw_params; - cs4270_dai.ops.set_sysclk = cs4270_set_dai_sysclk; - cs4270_dai.ops.set_fmt = cs4270_set_dai_fmt; -#ifdef CONFIG_SND_SOC_CS4270_HWMUTE - cs4270_dai.ops.digital_mute = cs4270_mute; -#endif - } else - printk(KERN_INFO "cs4270: no I2C device found, " - "using stand-alone mode\n"); -#else - printk(KERN_INFO "cs4270: I2C disabled, using stand-alone mode\n"); -#endif + if (!codec->control_data) { + printk(KERN_ERR "cs4270: failed to attach driver"); + goto error_del_driver; + } + + /* Initialize codec ops */ + cs4270_dai.ops.hw_params = cs4270_hw_params; + cs4270_dai.ops.set_sysclk = cs4270_set_dai_sysclk; + cs4270_dai.ops.set_fmt = cs4270_set_dai_fmt; + cs4270_dai.ops.digital_mute = cs4270_mute; ret = snd_soc_init_card(socdev); if (ret < 0) { @@ -732,11 +698,9 @@ static int cs4270_probe(struct platform_device *pdev) return 0; error_del_driver: -#ifdef USE_I2C i2c_del_driver(&cs4270_i2c_driver); error_free_pcms: -#endif snd_soc_free_pcms(socdev); error_free_codec: @@ -752,9 +716,7 @@ static int cs4270_remove(struct platform_device *pdev) snd_soc_free_pcms(socdev); -#ifdef USE_I2C i2c_del_driver(&cs4270_i2c_driver); -#endif kfree(socdev->codec); socdev->codec = NULL; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index c7c78c39cfe..9fc90828337 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -10,7 +10,8 @@ config SND_SOC_MPC8610 config SND_SOC_MPC8610_HPCD tristate "ALSA SoC support for the Freescale MPC8610 HPCD board" - depends on MPC8610_HPCD + # I2C is necessary for the CS4270 driver + depends on MPC8610_HPCD && I2C select SND_SOC_MPC8610 select SND_SOC_CS4270 select SND_SOC_CS4270_VD33_ERRATA -- cgit v1.2.3-70-g09d2 From c91cf25ebfbf3a5b336cbaa46646d37dd3d33127 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 23 Jan 2009 11:23:32 +0000 Subject: ASoC: Fix merge with PXA tree Fix a merge issue caused by context overlap. Reported-by: Stephen Rothwell Signed-off-by: Mark Brown --- sound/soc/pxa/e800_wm9712.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c index 78a1770b986..bc019cdce42 100644 --- a/sound/soc/pxa/e800_wm9712.c +++ b/sound/soc/pxa/e800_wm9712.c @@ -18,13 +18,10 @@ #include #include -#include -#include +#include #include #include -#include - #include "../codecs/wm9712.h" #include "pxa2xx-pcm.h" #include "pxa2xx-ac97.h" -- cgit v1.2.3-70-g09d2 From 6d6e17de4f64131e9c976fd524d73aaec268178f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 23 Jan 2009 12:33:54 +0100 Subject: ALSA: hda - Fix initial verbs for mic-boosts on AD1981HD The mic boosts (NID 0x08 and 0x18) are input-amps, not output-amps. Fix the initial verbs for them. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 2e7371ec2e2..9a902c2f05a 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1407,8 +1407,8 @@ static struct hda_verb ad1981_init_verbs[] = { {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, {0x1f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, /* Mic boost: 0dB */ - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* Record selector: Front mic */ {0x15, AC_VERB_SET_CONNECT_SEL, 0x0}, {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, -- cgit v1.2.3-70-g09d2 From 19a2d3e9b99ffa264adf1138bd8d8aef8909dca9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 23 Jan 2009 12:35:25 +0100 Subject: ALSA: hda - Remove invalid amp initializations for AD1988* codecs The ADC widgets on AD1988* codecs have no amp controls. Remove invalid initialization verbs. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 12 ------------ 1 file changed, 12 deletions(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 9a902c2f05a..52bc85dd6f5 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -2288,10 +2288,6 @@ static struct hda_verb ad1988_capture_init_verbs[] = { {0x0c, AC_VERB_SET_CONNECT_SEL, 0x1}, {0x0d, AC_VERB_SET_CONNECT_SEL, 0x1}, {0x0e, AC_VERB_SET_CONNECT_SEL, 0x1}, - /* ADCs; muted */ - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, { } }; @@ -2399,10 +2395,6 @@ static struct hda_verb ad1988_3stack_init_verbs[] = { {0x0c, AC_VERB_SET_CONNECT_SEL, 0x1}, {0x0d, AC_VERB_SET_CONNECT_SEL, 0x1}, {0x0e, AC_VERB_SET_CONNECT_SEL, 0x1}, - /* ADCs; muted */ - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Analog Mix output amp */ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */ { } @@ -2474,10 +2466,6 @@ static struct hda_verb ad1988_laptop_init_verbs[] = { {0x0c, AC_VERB_SET_CONNECT_SEL, 0x1}, {0x0d, AC_VERB_SET_CONNECT_SEL, 0x1}, {0x0e, AC_VERB_SET_CONNECT_SEL, 0x1}, - /* ADCs; muted */ - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Analog Mix output amp */ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */ { } -- cgit v1.2.3-70-g09d2 From 60e388e89c9e258a51a0995ddd9e18fdebcdbe12 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 23 Jan 2009 12:37:09 +0100 Subject: ALSA: hda - Fix invalid verbs for mic-boosts on AD1884* The mic-boosts (0x14 and 0x15) on AD1884* codecs are input-amps, not output-amps. Fix the invalid initialization verbs. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 52bc85dd6f5..a7298d28a0d 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -3183,10 +3183,10 @@ static struct hda_verb ad1884_init_verbs[] = { {0x0e, AC_VERB_SET_CONNECT_SEL, 0x1}, /* Port-B (front mic) pin */ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* Port-C (rear mic) pin */ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* Analog mixer; mute as default */ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, @@ -3601,10 +3601,10 @@ static struct hda_verb ad1884a_init_verbs[] = { {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Port-B (front mic) pin */ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* Port-C (rear line-in) pin */ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* Port-E (rear mic) pin */ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, -- cgit v1.2.3-70-g09d2 From f6fca2e93c9ad3c704f02aaabe4359a8af16fbbb Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 23 Jan 2009 11:40:26 +0000 Subject: ASoC: Remove unneeded e7x0 inclusion of pxa-regs.h and hardware.h pxa-regs.h and hardware.h are not intended for use directly in driver code and references to them have been removed in other code - remove them from the newly added e740 and e750 machine drivers. Signed-off-by: Mark Brown --- sound/soc/pxa/e740_wm9705.c | 2 -- sound/soc/pxa/e750_wm9705.c | 2 -- 2 files changed, 4 deletions(-) diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c index ac361765173..7cd2f89d7b1 100644 --- a/sound/soc/pxa/e740_wm9705.c +++ b/sound/soc/pxa/e740_wm9705.c @@ -18,8 +18,6 @@ #include #include -#include -#include #include #include diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c index 20fbdcfa9f7..8dceccc5e05 100644 --- a/sound/soc/pxa/e750_wm9705.c +++ b/sound/soc/pxa/e750_wm9705.c @@ -18,8 +18,6 @@ #include #include -#include -#include #include #include -- cgit v1.2.3-70-g09d2 From a435869cacbb581920df23411416bed533748bf1 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 23 Jan 2009 11:49:45 +0000 Subject: ASoC: Configure SSP port PLL for Zylonite Signed-off-by: Mark Brown --- sound/soc/pxa/zylonite.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index 8541b679f6e..ec2fb764b24 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -95,6 +95,7 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + unsigned int pll_out = 0; unsigned int acds = 0; unsigned int wm9713_div = 0; int ret = 0; @@ -102,13 +103,16 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream, switch (params_rate(params)) { case 8000: wm9713_div = 12; + pll_out = 2048000; break; case 16000: wm9713_div = 6; + pll_out = 4096000; break; case 48000: default: wm9713_div = 2; + pll_out = 12288000; acds = 1; break; } @@ -129,6 +133,10 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; + ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, pll_out); + if (ret < 0) + return ret; + ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_AUDIO_DIV_ACDS, acds); if (ret < 0) return ret; -- cgit v1.2.3-70-g09d2 From 4cfb91c6d764b18e81bfb6e6779e07bcecbb197c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 23 Jan 2009 12:53:09 +0100 Subject: ALSA: hda - Fix invalid amp init for ALC268 codec Fix some invalid AMP initializations for ALC268 codecs. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 18 ++---------------- 1 file changed, 2 insertions(+), 16 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4cfa78c5439..863ab957204 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -11279,19 +11279,13 @@ static void alc267_quanta_il1_unsol_event(struct hda_codec *codec, static struct hda_verb alc268_base_init_verbs[] = { /* Unmute DAC0-1 and set vol = 0 */ {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, /* * Set up output mixers (0x0c - 0x0e) */ /* set vol=0 to output mixers */ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x0e, AC_VERB_SET_CONNECT_SEL, 0x00}, {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, @@ -11310,9 +11304,7 @@ static struct hda_verb alc268_base_init_verbs[] = { {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* set PCBEEP vol = 0, mute connections */ {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, @@ -11334,10 +11326,8 @@ static struct hda_verb alc268_base_init_verbs[] = { */ static struct hda_verb alc268_volume_init_verbs[] = { /* set output DAC */ - {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, @@ -11345,16 +11335,12 @@ static struct hda_verb alc268_volume_init_verbs[] = { {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* set PCBEEP vol = 0, mute connections */ {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, -- cgit v1.2.3-70-g09d2 From 70040c07402ef5a3fad2133daffb7ee61b0d4641 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 23 Jan 2009 14:18:11 +0100 Subject: ALSA: hda - Fix wrong initial verb for AD1984 thinkpad model The docking mic-boost (0x25) has no mute bit. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index a7298d28a0d..e934e2c187d 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -3337,7 +3337,7 @@ static struct hda_verb ad1984_thinkpad_init_verbs[] = { {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* docking mic boost */ - {0x25, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x25, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* Analog mixer - docking mic; mute as default */ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* enable EAPD bit */ -- cgit v1.2.3-70-g09d2 From 55aef4508598d59c2baea7e2a3e6dfed415bbfc0 Mon Sep 17 00:00:00 2001 From: Markus Bollinger Date: Fri, 23 Jan 2009 14:45:41 +0100 Subject: ALSA: pcxhr - add support for gpio ports and minor bug fix - add support for gpio ports (2 GPI, 2 GPO) of pcxhr stereo cards - minor bugfixes : allow setting clock to internal by the mixer even if there is no stream (but monitoring) Signed-off-by: Markus Bollinger Signed-off-by: Takashi Iwai --- sound/pci/pcxhr/pcxhr.c | 41 +++++++++++++++++++++++++++++++++++++++++ sound/pci/pcxhr/pcxhr.h | 5 +++-- sound/pci/pcxhr/pcxhr_mix22.c | 40 ++++++++++++++++++++++++++++++++++++---- sound/pci/pcxhr/pcxhr_mix22.h | 3 +++ sound/pci/pcxhr/pcxhr_mixer.c | 8 ++++++-- 5 files changed, 89 insertions(+), 8 deletions(-) diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index 27cf2c28d11..ca89106f8c5 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -1334,6 +1334,40 @@ static void pcxhr_proc_sync(struct snd_info_entry *entry, snd_iprintf(buffer, "\n"); } +static void pcxhr_proc_gpio_read(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct snd_pcxhr *chip = entry->private_data; + struct pcxhr_mgr *mgr = chip->mgr; + /* commands available when embedded DSP is running */ + if (mgr->dsp_loaded & (1 << PCXHR_FIRMWARE_DSP_MAIN_INDEX)) { + /* gpio ports on stereo boards only available */ + int value = 0; + hr222_read_gpio(mgr, 1, &value); /* GPI */ + snd_iprintf(buffer, "GPI: 0x%x\n", value); + hr222_read_gpio(mgr, 0, &value); /* GP0 */ + snd_iprintf(buffer, "GPO: 0x%x\n", value); + } else + snd_iprintf(buffer, "no firmware loaded\n"); + snd_iprintf(buffer, "\n"); +} +static void pcxhr_proc_gpo_write(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct snd_pcxhr *chip = entry->private_data; + struct pcxhr_mgr *mgr = chip->mgr; + char line[64]; + int value; + /* commands available when embedded DSP is running */ + if (!(mgr->dsp_loaded & (1 << PCXHR_FIRMWARE_DSP_MAIN_INDEX))) + return; + while (!snd_info_get_line(buffer, line, sizeof(line))) { + if (sscanf(line, "GPO: 0x%x", &value) != 1) + continue; + hr222_write_gpo(mgr, value); /* GP0 */ + } +} + static void __devinit pcxhr_proc_init(struct snd_pcxhr *chip) { struct snd_info_entry *entry; @@ -1342,6 +1376,13 @@ static void __devinit pcxhr_proc_init(struct snd_pcxhr *chip) snd_info_set_text_ops(entry, chip, pcxhr_proc_info); if (! snd_card_proc_new(chip->card, "sync", &entry)) snd_info_set_text_ops(entry, chip, pcxhr_proc_sync); + /* gpio available on stereo sound cards only */ + if (chip->mgr->is_hr_stereo && + !snd_card_proc_new(chip->card, "gpio", &entry)) { + snd_info_set_text_ops(entry, chip, pcxhr_proc_gpio_read); + entry->c.text.write = pcxhr_proc_gpo_write; + entry->mode |= S_IWUSR; + } } /* end of proc interface */ diff --git a/sound/pci/pcxhr/pcxhr.h b/sound/pci/pcxhr/pcxhr.h index 84131a916c9..ac9c3b3bb4e 100644 --- a/sound/pci/pcxhr/pcxhr.h +++ b/sound/pci/pcxhr/pcxhr.h @@ -27,8 +27,8 @@ #include #include -#define PCXHR_DRIVER_VERSION 0x000905 /* 0.9.5 */ -#define PCXHR_DRIVER_VERSION_STRING "0.9.5" /* 0.9.5 */ +#define PCXHR_DRIVER_VERSION 0x000906 /* 0.9.6 */ +#define PCXHR_DRIVER_VERSION_STRING "0.9.6" /* 0.9.6 */ #define PCXHR_MAX_CARDS 6 @@ -124,6 +124,7 @@ struct pcxhr_mgr { unsigned char xlx_cfg; /* copy of PCXHR_XLX_CFG register */ unsigned char xlx_selmic; /* copy of PCXHR_XLX_SELMIC register */ + unsigned char dsp_reset; /* copy of PCXHR_DSP_RESET register */ }; diff --git a/sound/pci/pcxhr/pcxhr_mix22.c b/sound/pci/pcxhr/pcxhr_mix22.c index ff019126b67..1cb82c0a9cb 100644 --- a/sound/pci/pcxhr/pcxhr_mix22.c +++ b/sound/pci/pcxhr/pcxhr_mix22.c @@ -53,6 +53,8 @@ #define PCXHR_DSP_RESET_DSP 0x01 #define PCXHR_DSP_RESET_MUTE 0x02 #define PCXHR_DSP_RESET_CODEC 0x08 +#define PCXHR_DSP_RESET_GPO_OFFSET 5 +#define PCXHR_DSP_RESET_GPO_MASK 0x60 /* values for PCHR_XLX_CFG register */ #define PCXHR_CFG_SYNCDSP_MASK 0x80 @@ -81,6 +83,8 @@ /* values for PCHR_XLX_STATUS register - READ */ #define PCXHR_STAT_SRC_LOCK 0x01 #define PCXHR_STAT_LEVEL_IN 0x02 +#define PCXHR_STAT_GPI_OFFSET 2 +#define PCXHR_STAT_GPI_MASK 0x0C #define PCXHR_STAT_MIC_CAPS 0x10 /* values for PCHR_XLX_STATUS register - WRITE */ #define PCXHR_STAT_FREQ_SYNC_MASK 0x01 @@ -291,10 +295,11 @@ int hr222_sub_init(struct pcxhr_mgr *mgr) PCXHR_OUTPB(mgr, PCXHR_DSP_RESET, PCXHR_DSP_RESET_DSP); msleep(5); - PCXHR_OUTPB(mgr, PCXHR_DSP_RESET, - PCXHR_DSP_RESET_DSP | - PCXHR_DSP_RESET_MUTE | - PCXHR_DSP_RESET_CODEC); + mgr->dsp_reset = PCXHR_DSP_RESET_DSP | + PCXHR_DSP_RESET_MUTE | + PCXHR_DSP_RESET_CODEC; + PCXHR_OUTPB(mgr, PCXHR_DSP_RESET, mgr->dsp_reset); + /* hr222_write_gpo(mgr, 0); does the same */ msleep(5); /* config AKM */ @@ -496,6 +501,33 @@ int hr222_get_external_clock(struct pcxhr_mgr *mgr, } +int hr222_read_gpio(struct pcxhr_mgr *mgr, int is_gpi, int *value) +{ + if (is_gpi) { + unsigned char reg = PCXHR_INPB(mgr, PCXHR_XLX_STATUS); + *value = (int)(reg & PCXHR_STAT_GPI_MASK) >> + PCXHR_STAT_GPI_OFFSET; + } else { + *value = (int)(mgr->dsp_reset & PCXHR_DSP_RESET_GPO_MASK) >> + PCXHR_DSP_RESET_GPO_OFFSET; + } + return 0; +} + + +int hr222_write_gpo(struct pcxhr_mgr *mgr, int value) +{ + unsigned char reg = mgr->dsp_reset & ~PCXHR_DSP_RESET_GPO_MASK; + + reg |= (unsigned char)(value << PCXHR_DSP_RESET_GPO_OFFSET) & + PCXHR_DSP_RESET_GPO_MASK; + + PCXHR_OUTPB(mgr, PCXHR_DSP_RESET, reg); + mgr->dsp_reset = reg; + return 0; +} + + int hr222_update_analog_audio_level(struct snd_pcxhr *chip, int is_capture, int channel) { diff --git a/sound/pci/pcxhr/pcxhr_mix22.h b/sound/pci/pcxhr/pcxhr_mix22.h index 6b318b2f010..5a37a0007e8 100644 --- a/sound/pci/pcxhr/pcxhr_mix22.h +++ b/sound/pci/pcxhr/pcxhr_mix22.h @@ -32,6 +32,9 @@ int hr222_get_external_clock(struct pcxhr_mgr *mgr, enum pcxhr_clock_type clock_type, int *sample_rate); +int hr222_read_gpio(struct pcxhr_mgr *mgr, int is_gpi, int *value); +int hr222_write_gpo(struct pcxhr_mgr *mgr, int value); + #define HR222_LINE_PLAYBACK_LEVEL_MIN 0 /* -25.5 dB */ #define HR222_LINE_PLAYBACK_ZERO_LEVEL 51 /* 0.0 dB */ #define HR222_LINE_PLAYBACK_LEVEL_MAX 99 /* +24.0 dB */ diff --git a/sound/pci/pcxhr/pcxhr_mixer.c b/sound/pci/pcxhr/pcxhr_mixer.c index 2436e374586..fec04934462 100644 --- a/sound/pci/pcxhr/pcxhr_mixer.c +++ b/sound/pci/pcxhr/pcxhr_mixer.c @@ -789,11 +789,15 @@ static int pcxhr_clock_type_put(struct snd_kcontrol *kcontrol, if (mgr->use_clock_type != ucontrol->value.enumerated.item[0]) { mutex_lock(&mgr->setup_mutex); mgr->use_clock_type = ucontrol->value.enumerated.item[0]; - if (mgr->use_clock_type) + rate = 0; + if (mgr->use_clock_type != PCXHR_CLOCK_TYPE_INTERNAL) { pcxhr_get_external_clock(mgr, mgr->use_clock_type, &rate); - else + } else { rate = mgr->sample_rate; + if (!rate) + rate = 48000; + } if (rate) { pcxhr_set_clock(mgr, rate); if (mgr->sample_rate) -- cgit v1.2.3-70-g09d2 From ef963dcf6879e500e6559b4327f6cbdc4439198e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 23 Jan 2009 14:53:58 +0000 Subject: ASoC: Fix spurious codec driver dependencies Kbuild ignores dependency from things that are themselves selected so ASoC machine drivers need to ensure that the control bus is being built. This also avoids issues where multiple buses are supported by a given codec. Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 ---- sound/soc/omap/Kconfig | 4 ++-- 2 files changed, 2 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index cb5fcd605ac..656f180b2c1 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -91,7 +91,6 @@ config SND_SOC_SSM2602 config SND_SOC_TLV320AIC23 tristate - depends on I2C config SND_SOC_TLV320AIC26 tristate "TI TLV320AIC26 Codec support" if SND_SOC_OF_SIMPLE @@ -99,15 +98,12 @@ config SND_SOC_TLV320AIC26 config SND_SOC_TLV320AIC3X tristate - depends on I2C config SND_SOC_TWL4030 tristate - depends on TWL4030_CORE config SND_SOC_UDA134X tristate - select SND_SOC_L3 config SND_SOC_UDA1380 tristate diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index ccd8973683d..675732e724d 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -8,7 +8,7 @@ config SND_OMAP_SOC_MCBSP config SND_OMAP_SOC_N810 tristate "SoC Audio support for Nokia N810" - depends on SND_OMAP_SOC && MACH_NOKIA_N810 + depends on SND_OMAP_SOC && MACH_NOKIA_N810 && I2C select SND_OMAP_SOC_MCBSP select OMAP_MUX select SND_SOC_TLV320AIC3X @@ -17,7 +17,7 @@ config SND_OMAP_SOC_N810 config SND_OMAP_SOC_OSK5912 tristate "SoC Audio support for omap osk5912" - depends on SND_OMAP_SOC && MACH_OMAP_OSK + depends on SND_OMAP_SOC && MACH_OMAP_OSK && I2C select SND_OMAP_SOC_MCBSP select SND_SOC_TLV320AIC23 help -- cgit v1.2.3-70-g09d2 From 01e097d6c409a6eb64758dce9fcde0c70073fe36 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 23 Jan 2009 15:07:45 +0000 Subject: ASoC: Include header file in cs4270 and wm9705 Ensures that the DAI and socdev exported by the codec match up with their exported prototype. Signed-off-by: Mark Brown --- sound/soc/codecs/cs4270.c | 2 ++ sound/soc/codecs/wm9705.c | 2 ++ 2 files changed, 4 insertions(+) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 2e4ce04925e..e2130d7b1e4 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -29,6 +29,8 @@ #include #include +#include "cs4270.h" + /* Private data for the CS4270 */ struct cs4270_private { unsigned int mclk; /* Input frequency of the MCLK pin */ diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index 5e1937ac0b5..d5c81bb3dec 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -20,6 +20,8 @@ #include #include +#include "wm9705.h" + /* * WM9705 register cache */ -- cgit v1.2.3-70-g09d2 From 070504ade7a95a0f4395673717f3bb7d41793ca8 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 23 Jan 2009 15:34:54 +0000 Subject: ASoC: Fix L3 bus handling in Kconfig It has no external dependencies but needs to be selected for L3 based codecs to work. Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 1 + sound/soc/s3c24xx/Kconfig | 1 + 2 files changed, 2 insertions(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 656f180b2c1..a195303603e 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -10,6 +10,7 @@ config SND_SOC_I2C_AND_SPI config SND_SOC_ALL_CODECS tristate "Build all ASoC CODEC drivers" + select SND_SOC_L3 select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS select SND_SOC_AD1980 if SND_SOC_AC97_BUS select SND_SOC_AD73311 if I2C diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index fcd03acf10f..e05a71084c3 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -48,4 +48,5 @@ config SND_S3C24XX_SOC_S3C24XX_UDA134X tristate "SoC I2S Audio support UDA134X wired to a S3C24XX" depends on SND_S3C24XX_SOC select SND_S3C24XX_SOC_I2S + select SND_SOC_L3 select SND_SOC_UDA134X -- cgit v1.2.3-70-g09d2 From 0db4d0705260dd4bddf1e5a5441c58bdf08bdc9f Mon Sep 17 00:00:00 2001 From: Timur Tabi Date: Fri, 23 Jan 2009 16:31:19 -0600 Subject: ASoC: improve I2C initialization code in CS4270 driver Further improvements in the I2C initialization sequence of the CS4270 driver. All ASoC initialization is now done in the I2C probe function. Signed-off-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/codecs/cs4270.c | 252 +++++++++++++++++++++------------------------- 1 file changed, 113 insertions(+), 139 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index e2130d7b1e4..2aa12fdbd2c 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -31,12 +31,6 @@ #include "cs4270.h" -/* Private data for the CS4270 */ -struct cs4270_private { - unsigned int mclk; /* Input frequency of the MCLK pin */ - unsigned int mode; /* The mode (I2S or left-justified) */ -}; - /* * The codec isn't really big-endian or little-endian, since the I2S * interface requires data to be sent serially with the MSbit first. @@ -109,6 +103,14 @@ struct cs4270_private { #define CS4270_MUTE_DAC_A 0x01 #define CS4270_MUTE_DAC_B 0x02 +/* Private data for the CS4270 */ +struct cs4270_private { + struct snd_soc_codec codec; + u8 reg_cache[CS4270_NUMREGS]; + unsigned int mclk; /* Input frequency of the MCLK pin */ + unsigned int mode; /* The mode (I2S or left-justified) */ +}; + /* * Clock Ratio Selection for Master Mode with I2C enabled * @@ -504,6 +506,31 @@ static const struct snd_kcontrol_new cs4270_snd_controls[] = { */ static struct snd_soc_device *cs4270_socdev; +struct snd_soc_dai cs4270_dai = { + .name = "cs4270", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = 0, + .formats = CS4270_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = 0, + .formats = CS4270_FORMATS, + }, + .ops = { + .hw_params = cs4270_hw_params, + .set_sysclk = cs4270_set_dai_sysclk, + .set_fmt = cs4270_set_dai_fmt, + .digital_mute = cs4270_mute, + }, +}; +EXPORT_SYMBOL_GPL(cs4270_dai); + /* * Initialize the I2C interface of the CS4270 * @@ -517,47 +544,52 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client, const struct i2c_device_id *id) { struct snd_soc_device *socdev = cs4270_socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec; + struct cs4270_private *cs4270; int i; int ret = 0; - /* Probing all possible addresses has one drawback: if there are - multiple CS4270s on the bus, then you cannot specify which - socdev is matched with which CS4270. For now, we just reject - this I2C device if the socdev already has one attached. */ - if (codec->control_data) - return -ENODEV; - - /* Note: codec_dai->codec is NULL here */ - - codec->reg_cache = kzalloc(CS4270_NUMREGS, GFP_KERNEL); - if (!codec->reg_cache) { - printk(KERN_ERR "cs4270: could not allocate register cache\n"); - ret = -ENOMEM; - goto error; - } - /* Verify that we have a CS4270 */ ret = i2c_smbus_read_byte_data(i2c_client, CS4270_CHIPID); if (ret < 0) { printk(KERN_ERR "cs4270: failed to read I2C\n"); - goto error; + return ret; } /* The top four bits of the chip ID should be 1100. */ if ((ret & 0xF0) != 0xC0) { - /* The device at this address is not a CS4270 codec */ - ret = -ENODEV; - goto error; + printk(KERN_ERR "cs4270: device at addr %X is not a CS4270\n", + i2c_client->addr); + return -ENODEV; } printk(KERN_INFO "cs4270: found device at I2C address %X\n", i2c_client->addr); printk(KERN_INFO "cs4270: hardware revision %X\n", ret & 0xF); + /* Allocate enough space for the snd_soc_codec structure + and our private data together. */ + cs4270 = kzalloc(sizeof(struct cs4270_private), GFP_KERNEL); + if (!cs4270) { + printk(KERN_ERR "cs4270: Could not allocate codec structure\n"); + return -ENOMEM; + } + codec = &cs4270->codec; + socdev->codec = codec; + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->name = "CS4270"; + codec->owner = THIS_MODULE; + codec->dai = &cs4270_dai; + codec->num_dai = 1; + codec->private_data = cs4270; codec->control_data = i2c_client; codec->read = cs4270_read_reg_cache; codec->write = cs4270_i2c_write; + codec->reg_cache = cs4270->reg_cache; codec->reg_cache_size = CS4270_NUMREGS; /* The I2C interface is set up, so pre-fill our register cache */ @@ -565,35 +597,72 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client, ret = cs4270_fill_cache(codec); if (ret < 0) { printk(KERN_ERR "cs4270: failed to fill register cache\n"); - goto error; + goto error_free_codec; + } + + /* Register PCMs */ + + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "cs4270: failed to create PCMs\n"); + goto error_free_codec; } /* Add the non-DAPM controls */ for (i = 0; i < ARRAY_SIZE(cs4270_snd_controls); i++) { - struct snd_kcontrol *kctrl = - snd_soc_cnew(&cs4270_snd_controls[i], codec, NULL); + struct snd_kcontrol *kctrl; + + kctrl = snd_soc_cnew(&cs4270_snd_controls[i], codec, NULL); + if (!kctrl) { + printk(KERN_ERR "cs4270: error creating control '%s'\n", + cs4270_snd_controls[i].name); + ret = -ENOMEM; + goto error_free_pcms; + } ret = snd_ctl_add(codec->card, kctrl); - if (ret < 0) - goto error; + if (ret < 0) { + printk(KERN_ERR "cs4270: error adding control '%s'\n", + cs4270_snd_controls[i].name); + goto error_free_pcms; + } } - i2c_set_clientdata(i2c_client, codec); + /* Initialize the SOC device */ + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + printk(KERN_ERR "cs4270: failed to register card\n"); + goto error_free_pcms;; + } + + i2c_set_clientdata(i2c_client, socdev); return 0; -error: - codec->control_data = NULL; +error_free_pcms: + snd_soc_free_pcms(socdev); - kfree(codec->reg_cache); - codec->reg_cache = NULL; - codec->reg_cache_size = 0; +error_free_codec: + kfree(cs4270); return ret; } -static const struct i2c_device_id cs4270_id[] = { +static int cs4270_i2c_remove(struct i2c_client *i2c_client) +{ + struct snd_soc_device *socdev = i2c_get_clientdata(i2c_client); + struct snd_soc_codec *codec = socdev->codec; + struct cs4270_private *cs4270 = codec->private_data; + + snd_soc_free_pcms(socdev); + kfree(cs4270); + + return 0; +} + +static struct i2c_device_id cs4270_id[] = { {"cs4270", 0}, {} }; @@ -606,27 +675,9 @@ static struct i2c_driver cs4270_i2c_driver = { }, .id_table = cs4270_id, .probe = cs4270_i2c_probe, + .remove = cs4270_i2c_remove, }; -struct snd_soc_dai cs4270_dai = { - .name = "CS4270", - .playback = { - .stream_name = "Playback", - .channels_min = 1, - .channels_max = 2, - .rates = 0, - .formats = CS4270_FORMATS, - }, - .capture = { - .stream_name = "Capture", - .channels_min = 1, - .channels_max = 2, - .rates = 0, - .formats = CS4270_FORMATS, - }, -}; -EXPORT_SYMBOL_GPL(cs4270_dai); - /* * ASoC probe function * @@ -635,94 +686,15 @@ EXPORT_SYMBOL_GPL(cs4270_dai); */ static int cs4270_probe(struct platform_device *pdev) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec; - int ret = 0; - - printk(KERN_INFO "CS4270 ALSA SoC Codec\n"); - - /* Allocate enough space for the snd_soc_codec structure - and our private data together. */ - codec = kzalloc(ALIGN(sizeof(struct snd_soc_codec), 4) + - sizeof(struct cs4270_private), GFP_KERNEL); - if (!codec) { - printk(KERN_ERR "cs4270: Could not allocate codec structure\n"); - return -ENOMEM; - } - - mutex_init(&codec->mutex); - INIT_LIST_HEAD(&codec->dapm_widgets); - INIT_LIST_HEAD(&codec->dapm_paths); - - codec->name = "CS4270"; - codec->owner = THIS_MODULE; - codec->dai = &cs4270_dai; - codec->num_dai = 1; - codec->private_data = (void *) codec + - ALIGN(sizeof(struct snd_soc_codec), 4); - - socdev->codec = codec; - - /* Register PCMs */ - - ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); - if (ret < 0) { - printk(KERN_ERR "cs4270: failed to create PCMs\n"); - goto error_free_codec; - } - - cs4270_socdev = socdev; - - ret = i2c_add_driver(&cs4270_i2c_driver); - if (ret) { - printk(KERN_ERR "cs4270: failed to attach driver"); - goto error_free_pcms; - } - - /* Did we find a CS4270 on the I2C bus? */ - if (!codec->control_data) { - printk(KERN_ERR "cs4270: failed to attach driver"); - goto error_del_driver; - } + cs4270_socdev = platform_get_drvdata(pdev);; - /* Initialize codec ops */ - cs4270_dai.ops.hw_params = cs4270_hw_params; - cs4270_dai.ops.set_sysclk = cs4270_set_dai_sysclk; - cs4270_dai.ops.set_fmt = cs4270_set_dai_fmt; - cs4270_dai.ops.digital_mute = cs4270_mute; - - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "cs4270: failed to register card\n"); - goto error_del_driver; - } - - return 0; - -error_del_driver: - i2c_del_driver(&cs4270_i2c_driver); - -error_free_pcms: - snd_soc_free_pcms(socdev); - -error_free_codec: - kfree(socdev->codec); - socdev->codec = NULL; - - return ret; + return i2c_add_driver(&cs4270_i2c_driver); } static int cs4270_remove(struct platform_device *pdev) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - - snd_soc_free_pcms(socdev); - i2c_del_driver(&cs4270_i2c_driver); - kfree(socdev->codec); - socdev->codec = NULL; - return 0; } @@ -740,6 +712,8 @@ EXPORT_SYMBOL_GPL(soc_codec_device_cs4270); static int __init cs4270_init(void) { + printk(KERN_INFO "Cirrus Logic CS4270 ALSA SoC Codec Driver\n"); + return snd_soc_register_dai(&cs4270_dai); } module_init(cs4270_init); -- cgit v1.2.3-70-g09d2 From ca8d33fc9fafe373362d35107f01fba1e73fb966 Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Mon, 26 Jan 2009 09:33:52 -0500 Subject: ALSA: hda: 92hd71xxx disable unmute support for codecs that don't have input amps Some revisions of the 92hd71xxx codec families don't have input amps on ports 0xa, 0xd and 0xf, so probe the widget caps on port 0xa and check for support, if found run snd_hda_sequence_write_cache() on the stac92hd71xxx_unmute_core_init verb list. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 20 ++++++++++++-------- 1 file changed, 12 insertions(+), 8 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 80a4c288b31..03b26426611 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -858,26 +858,25 @@ static struct hda_verb stac92hd83xxx_core_init[] = { static struct hda_verb stac92hd71bxx_core_init[] = { /* set master volume and direct control */ { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, - /* unmute right and left channels for nodes 0x0a, 0xd, 0x0f */ - { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {} }; -#define HD_DISABLE_PORTF 2 +#define HD_DISABLE_PORTF 1 static struct hda_verb stac92hd71bxx_analog_core_init[] = { /* start of config #1 */ /* connect port 0f to audio mixer */ { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x2}, - /* unmute right and left channels for node 0x0f */ - { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* start of config #2 */ /* set master volume and direct control */ { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, - /* unmute right and left channels for nodes 0x0a, 0xd */ + {} +}; + +static struct hda_verb stac92hd71bxx_unmute_core_init[] = { + /* unmute right and left channels for nodes 0x0f, 0xa, 0x0d */ + { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {} @@ -4942,6 +4941,7 @@ static struct hda_input_mux stac92hd71bxx_dmux = { static int patch_stac92hd71bxx(struct hda_codec *codec) { struct sigmatel_spec *spec; + struct hda_verb *unmute_init = stac92hd71bxx_unmute_core_init; int err = 0; spec = kzalloc(sizeof(*spec), GFP_KERNEL); @@ -5015,6 +5015,7 @@ again: /* disable VSW */ spec->init = &stac92hd71bxx_analog_core_init[HD_DISABLE_PORTF]; + unmute_init++; stac_change_pin_config(codec, 0xf, 0x40f000f0); break; case 0x111d7603: /* 6 Port with Analog Mixer */ @@ -5031,6 +5032,9 @@ again: codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; } + if (get_wcaps(codec, 0xa) & AC_WCAP_IN_AMP) + snd_hda_sequence_write_cache(codec, unmute_init); + spec->aloopback_mask = 0x50; spec->aloopback_shift = 0; -- cgit v1.2.3-70-g09d2 From b7eb4a06e9980973755b7e95a6d97fb8decbf8fd Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 26 Jan 2009 08:08:34 +0100 Subject: sound: usb-audio: use normal number of frames for no-data URBs When sending a silence URB (before playback has started, or when it is paused), use the number of frames that would be normally sent instead of a single frame so that the rate at which completion interrupts arrive is consistent. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/usb/usbaudio.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index c709b956322..417d557ed64 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -525,7 +525,7 @@ static int snd_usb_audio_next_packet_size(struct snd_usb_substream *subs) /* * Prepare urb for streaming before playback starts or when paused. * - * We don't have any data, so we send a frame of silence. + * We don't have any data, so we send silence. */ static int prepare_nodata_playback_urb(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime, @@ -537,13 +537,13 @@ static int prepare_nodata_playback_urb(struct snd_usb_substream *subs, offs = 0; urb->dev = ctx->subs->dev; - urb->number_of_packets = subs->packs_per_ms; - for (i = 0; i < subs->packs_per_ms; ++i) { + for (i = 0; i < ctx->packets; ++i) { counts = snd_usb_audio_next_packet_size(subs); urb->iso_frame_desc[i].offset = offs * stride; urb->iso_frame_desc[i].length = counts * stride; offs += counts; } + urb->number_of_packets = ctx->packets; urb->transfer_buffer_length = offs * stride; memset(urb->transfer_buffer, subs->cur_audiofmt->format == SNDRV_PCM_FORMAT_U8 ? 0x80 : 0, -- cgit v1.2.3-70-g09d2 From 4d788e040b72d2a46ea3ba726b7fa0b65de06c88 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 26 Jan 2009 08:09:28 +0100 Subject: sound: usb-audio: limit playback queue length Once our URBs contain enough packets, queueing more URBs does not give us any additional underrun protection (as we use double-buffering) but just increases latency unnecessarily. Therefore, we try to limit the queue length to some reasonable value. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/usb/usbaudio.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 417d557ed64..f3d4de23fed 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -108,6 +108,7 @@ MODULE_PARM_DESC(ignore_ctl_error, #define MAX_URBS 8 #define SYNC_URBS 4 /* always four urbs for sync */ #define MIN_PACKS_URB 1 /* minimum 1 packet per urb */ +#define MAX_QUEUE 24 /* try not to exceed this queue length, in ms */ struct audioformat { struct list_head list; @@ -1079,7 +1080,7 @@ static int init_substream_urbs(struct snd_usb_substream *subs, unsigned int peri /* decide how many packets to be used */ if (is_playback) { - unsigned int minsize; + unsigned int minsize, maxpacks; /* determine how small a packet can be */ minsize = (subs->freqn >> (16 - subs->datainterval)) * (frame_bits >> 3); @@ -1094,6 +1095,12 @@ static int init_substream_urbs(struct snd_usb_substream *subs, unsigned int peri /* we need at least two URBs for queueing */ if (total_packs < 2 * MIN_PACKS_URB * packs_per_ms) total_packs = 2 * MIN_PACKS_URB * packs_per_ms; + else { + /* and we don't want too long a queue either */ + maxpacks = max((unsigned int)MAX_QUEUE, urb_packs * 2); + if (total_packs > maxpacks * packs_per_ms) + total_packs = maxpacks * packs_per_ms; + } } else { total_packs = MAX_URBS * urb_packs; } -- cgit v1.2.3-70-g09d2 From 160389c8d21c8139a93191c2e5ca2273f311ed4e Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 26 Jan 2009 08:10:19 +0100 Subject: sound: usb-audio: make URB sizes more equal Distribute the packets evenly among the URBs, instead of making all URBs except the last one to have the maximum size. This makes the timing of pointer updates more regular and removes some special cases from the code. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/usb/usbaudio.c | 29 +++++------------------------ 1 file changed, 5 insertions(+), 24 deletions(-) diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index f3d4de23fed..44485b29f67 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -1035,9 +1035,9 @@ static void release_substream_urbs(struct snd_usb_substream *subs, int force) static int init_substream_urbs(struct snd_usb_substream *subs, unsigned int period_bytes, unsigned int rate, unsigned int frame_bits) { - unsigned int maxsize, n, i; + unsigned int maxsize, i; int is_playback = subs->direction == SNDRV_PCM_STREAM_PLAYBACK; - unsigned int npacks[MAX_URBS], urb_packs, total_packs, packs_per_ms; + unsigned int urb_packs, total_packs, packs_per_ms; /* calculate the frequency in 16.16 format */ if (snd_usb_get_speed(subs->dev) == USB_SPEED_FULL) @@ -1109,31 +1109,11 @@ static int init_substream_urbs(struct snd_usb_substream *subs, unsigned int peri /* too much... */ subs->nurbs = MAX_URBS; total_packs = MAX_URBS * urb_packs; - } - n = total_packs; - for (i = 0; i < subs->nurbs; i++) { - npacks[i] = n > urb_packs ? urb_packs : n; - n -= urb_packs; - } - if (subs->nurbs <= 1) { + } else if (subs->nurbs < 2) { /* too little - we need at least two packets * to ensure contiguous playback/capture */ subs->nurbs = 2; - npacks[0] = (total_packs + 1) / 2; - npacks[1] = total_packs - npacks[0]; - } else if (npacks[subs->nurbs-1] < MIN_PACKS_URB * packs_per_ms) { - /* the last packet is too small.. */ - if (subs->nurbs > 2) { - /* merge to the first one */ - npacks[0] += npacks[subs->nurbs - 1]; - subs->nurbs--; - } else { - /* divide to two */ - subs->nurbs = 2; - npacks[0] = (total_packs + 1) / 2; - npacks[1] = total_packs - npacks[0]; - } } /* allocate and initialize data urbs */ @@ -1141,7 +1121,8 @@ static int init_substream_urbs(struct snd_usb_substream *subs, unsigned int peri struct snd_urb_ctx *u = &subs->dataurb[i]; u->index = i; u->subs = subs; - u->packets = npacks[i]; + u->packets = (i + 1) * total_packs / subs->nurbs + - i * total_packs / subs->nurbs; u->buffer_size = maxsize * u->packets; if (subs->fmt_type == USB_FORMAT_TYPE_II) u->packets++; /* for transfer delimiter */ -- cgit v1.2.3-70-g09d2 From 6627a653bceb3a54e55e5cdc478ec5b8d5c9cc44 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 23 Jan 2009 22:55:23 +0000 Subject: ASoC: Push the codec runtime storage into the card structure This is a further stage on the road to refactoring away the ASoC platform device. Signed-off-by: Mark Brown --- include/sound/soc.h | 3 ++- sound/soc/codecs/ac97.c | 20 ++++++++++---------- sound/soc/codecs/ad1980.c | 12 ++++++------ sound/soc/codecs/ad73311.c | 8 ++++---- sound/soc/codecs/ak4535.c | 14 +++++++------- sound/soc/codecs/cs4270.c | 6 +++--- sound/soc/codecs/pcm3008.c | 12 ++++++------ sound/soc/codecs/ssm2602.c | 20 ++++++++++---------- sound/soc/codecs/tlv320aic23.c | 18 +++++++++--------- sound/soc/codecs/tlv320aic26.c | 4 ++-- sound/soc/codecs/tlv320aic3x.c | 14 +++++++------- sound/soc/codecs/twl4030.c | 12 ++++++------ sound/soc/codecs/uda134x.c | 18 +++++++++--------- sound/soc/codecs/uda1380.c | 18 +++++++++--------- sound/soc/codecs/wm8350.c | 10 +++++----- sound/soc/codecs/wm8510.c | 16 ++++++++-------- sound/soc/codecs/wm8580.c | 10 +++++----- sound/soc/codecs/wm8728.c | 16 ++++++++-------- sound/soc/codecs/wm8731.c | 20 ++++++++++---------- sound/soc/codecs/wm8750.c | 16 ++++++++-------- sound/soc/codecs/wm8753.c | 18 +++++++++--------- sound/soc/codecs/wm8900.c | 8 ++++---- sound/soc/codecs/wm8903.c | 18 +++++++++--------- sound/soc/codecs/wm8971.c | 14 +++++++------- sound/soc/codecs/wm8990.c | 14 +++++++------- sound/soc/codecs/wm9705.c | 15 ++++++++------- sound/soc/codecs/wm9712.c | 21 +++++++++++---------- sound/soc/codecs/wm9713.c | 17 +++++++++-------- sound/soc/soc-core.c | 37 +++++++++++++++++-------------------- sound/soc/soc-dapm.c | 6 +++--- sound/soc/soc-jack.c | 4 ++-- 31 files changed, 220 insertions(+), 219 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 7039343e8a7..0e773526416 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -418,6 +418,8 @@ struct snd_soc_card { struct snd_soc_device *socdev; + struct snd_soc_codec *codec; + struct snd_soc_platform *platform; struct delayed_work delayed_work; struct work_struct deferred_resume_work; @@ -427,7 +429,6 @@ struct snd_soc_card { struct snd_soc_device { struct device *dev; struct snd_soc_card *card; - struct snd_soc_codec *codec; struct snd_soc_codec_device *codec_dev; void *codec_data; }; diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index 89d41277616..11f84b6e5cb 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -30,7 +30,7 @@ static int ac97_prepare(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int reg = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? AC97_PCM_FRONT_DAC_RATE : AC97_PCM_LR_ADC_RATE; @@ -84,10 +84,10 @@ static int ac97_soc_probe(struct platform_device *pdev) printk(KERN_INFO "AC97 SoC Audio Codec %s\n", AC97_VERSION); - socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); - if (!socdev->codec) + socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (!socdev->card->codec) return -ENOMEM; - codec = socdev->codec; + codec = socdev->card->codec; mutex_init(&codec->mutex); codec->name = "AC97"; @@ -123,21 +123,21 @@ bus_err: snd_soc_free_pcms(socdev); err: - kfree(socdev->codec); - socdev->codec = NULL; + kfree(socdev->card->codec); + socdev->card->codec = NULL; return ret; } static int ac97_soc_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; if (!codec) return 0; snd_soc_free_pcms(socdev); - kfree(socdev->codec); + kfree(socdev->card->codec); return 0; } @@ -147,7 +147,7 @@ static int ac97_soc_suspend(struct platform_device *pdev, pm_message_t msg) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - snd_ac97_suspend(socdev->codec->ac97); + snd_ac97_suspend(socdev->card->codec->ac97); return 0; } @@ -156,7 +156,7 @@ static int ac97_soc_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - snd_ac97_resume(socdev->codec->ac97); + snd_ac97_resume(socdev->card->codec->ac97); return 0; } diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index faf358758e1..ddb3b08ac23 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -186,10 +186,10 @@ static int ad1980_soc_probe(struct platform_device *pdev) printk(KERN_INFO "AD1980 SoC Audio Codec\n"); - socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); - if (socdev->codec == NULL) + socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (socdev->card->codec == NULL) return -ENOMEM; - codec = socdev->codec; + codec = socdev->card->codec; mutex_init(&codec->mutex); codec->reg_cache = @@ -275,15 +275,15 @@ codec_err: kfree(codec->reg_cache); cache_err: - kfree(socdev->codec); - socdev->codec = NULL; + kfree(socdev->card->codec); + socdev->card->codec = NULL; return ret; } static int ad1980_soc_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; if (codec == NULL) return 0; diff --git a/sound/soc/codecs/ad73311.c b/sound/soc/codecs/ad73311.c index b09289a1e55..e61dac5e7b8 100644 --- a/sound/soc/codecs/ad73311.c +++ b/sound/soc/codecs/ad73311.c @@ -53,7 +53,7 @@ static int ad73311_soc_probe(struct platform_device *pdev) codec->owner = THIS_MODULE; codec->dai = &ad73311_dai; codec->num_dai = 1; - socdev->codec = codec; + socdev->card->codec = codec; INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); @@ -75,15 +75,15 @@ static int ad73311_soc_probe(struct platform_device *pdev) register_err: snd_soc_free_pcms(socdev); pcm_err: - kfree(socdev->codec); - socdev->codec = NULL; + kfree(socdev->card->codec); + socdev->card->codec = NULL; return ret; } static int ad73311_soc_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; if (codec == NULL) return 0; diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index f17c363cb1d..d56e6bb1fed 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -329,7 +329,7 @@ static int ak4535_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; struct ak4535_priv *ak4535 = codec->private_data; u8 mode2 = ak4535_read_reg_cache(codec, AK4535_MODE2) & ~(0x3 << 5); int rate = params_rate(params), fs = 256; @@ -447,7 +447,7 @@ EXPORT_SYMBOL_GPL(ak4535_dai); static int ak4535_suspend(struct platform_device *pdev, pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; ak4535_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; @@ -456,7 +456,7 @@ static int ak4535_suspend(struct platform_device *pdev, pm_message_t state) static int ak4535_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; ak4535_sync(codec); ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY); ak4535_set_bias_level(codec, codec->suspend_bias_level); @@ -469,7 +469,7 @@ static int ak4535_resume(struct platform_device *pdev) */ static int ak4535_init(struct snd_soc_device *socdev) { - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int ret = 0; codec->name = "AK4535"; @@ -523,7 +523,7 @@ static int ak4535_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct snd_soc_device *socdev = ak4535_socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int ret; i2c_set_clientdata(i2c, codec); @@ -622,7 +622,7 @@ static int ak4535_probe(struct platform_device *pdev) } codec->private_data = ak4535; - socdev->codec = codec; + socdev->card->codec = codec; mutex_init(&codec->mutex); INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); @@ -649,7 +649,7 @@ static int ak4535_probe(struct platform_device *pdev) static int ak4535_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; if (codec->control_data) ak4535_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 2aa12fdbd2c..21253b48289 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -350,7 +350,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; struct cs4270_private *cs4270 = codec->private_data; int ret; unsigned int i; @@ -575,7 +575,7 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client, return -ENOMEM; } codec = &cs4270->codec; - socdev->codec = codec; + socdev->card->codec = codec; mutex_init(&codec->mutex); INIT_LIST_HEAD(&codec->dapm_widgets); @@ -653,7 +653,7 @@ error_free_codec: static int cs4270_i2c_remove(struct i2c_client *i2c_client) { struct snd_soc_device *socdev = i2c_get_clientdata(i2c_client); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; struct cs4270_private *cs4270 = codec->private_data; snd_soc_free_pcms(socdev); diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c index 9a3e67e5319..5cda9e6b5a7 100644 --- a/sound/soc/codecs/pcm3008.c +++ b/sound/soc/codecs/pcm3008.c @@ -67,11 +67,11 @@ static int pcm3008_soc_probe(struct platform_device *pdev) printk(KERN_INFO "PCM3008 SoC Audio Codec %s\n", PCM3008_VERSION); - socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); - if (!socdev->codec) + socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (!socdev->card->codec) return -ENOMEM; - codec = socdev->codec; + codec = socdev->card->codec; mutex_init(&codec->mutex); codec->name = "PCM3008"; @@ -139,7 +139,7 @@ gpio_err: card_err: snd_soc_free_pcms(socdev); pcm_err: - kfree(socdev->codec); + kfree(socdev->card->codec); return ret; } @@ -147,7 +147,7 @@ pcm_err: static int pcm3008_soc_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; struct pcm3008_setup_data *setup = socdev->codec_data; if (!codec) @@ -155,7 +155,7 @@ static int pcm3008_soc_remove(struct platform_device *pdev) pcm3008_gpio_free(setup); snd_soc_free_pcms(socdev); - kfree(socdev->codec); + kfree(socdev->card->codec); return 0; } diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index ec7fe3b7b0c..58e225dadc7 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -276,7 +276,7 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream, u16 srate; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; struct ssm2602_priv *ssm2602 = codec->private_data; struct i2c_client *i2c = codec->control_data; u16 iface = ssm2602_read_reg_cache(codec, SSM2602_IFACE) & 0xfff3; @@ -321,7 +321,7 @@ static int ssm2602_startup(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; struct ssm2602_priv *ssm2602 = codec->private_data; struct i2c_client *i2c = codec->control_data; struct snd_pcm_runtime *master_runtime; @@ -358,7 +358,7 @@ static int ssm2602_pcm_prepare(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; /* set active */ ssm2602_write(codec, SSM2602_ACTIVE, ACTIVE_ACTIVATE_CODEC); @@ -370,7 +370,7 @@ static void ssm2602_shutdown(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; struct ssm2602_priv *ssm2602 = codec->private_data; /* deactivate */ if (!codec->active) @@ -535,7 +535,7 @@ EXPORT_SYMBOL_GPL(ssm2602_dai); static int ssm2602_suspend(struct platform_device *pdev, pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; ssm2602_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; @@ -544,7 +544,7 @@ static int ssm2602_suspend(struct platform_device *pdev, pm_message_t state) static int ssm2602_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int i; u8 data[2]; u16 *cache = codec->reg_cache; @@ -566,7 +566,7 @@ static int ssm2602_resume(struct platform_device *pdev) */ static int ssm2602_init(struct snd_soc_device *socdev) { - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int reg, ret = 0; codec->name = "SSM2602"; @@ -639,7 +639,7 @@ static int ssm2602_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct snd_soc_device *socdev = ssm2602_socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int ret; i2c_set_clientdata(i2c, codec); @@ -733,7 +733,7 @@ static int ssm2602_probe(struct platform_device *pdev) } codec->private_data = ssm2602; - socdev->codec = codec; + socdev->card->codec = codec; mutex_init(&codec->mutex); INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); @@ -754,7 +754,7 @@ static int ssm2602_probe(struct platform_device *pdev) static int ssm2602_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; if (codec->control_data) ssm2602_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index a0e47c1dcd6..8b20c360adf 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -405,7 +405,7 @@ static int tlv320aic23_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; u16 iface_reg; int ret; struct aic23 *aic23 = container_of(codec, struct aic23, codec); @@ -453,7 +453,7 @@ static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; /* set active */ tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0001); @@ -466,7 +466,7 @@ static void tlv320aic23_shutdown(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; struct aic23 *aic23 = container_of(codec, struct aic23, codec); /* deactivate */ @@ -609,7 +609,7 @@ static int tlv320aic23_suspend(struct platform_device *pdev, pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0); tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF); @@ -620,7 +620,7 @@ static int tlv320aic23_suspend(struct platform_device *pdev, static int tlv320aic23_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int i; u16 reg; @@ -642,7 +642,7 @@ static int tlv320aic23_resume(struct platform_device *pdev) */ static int tlv320aic23_init(struct snd_soc_device *socdev) { - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int ret = 0; u16 reg; @@ -729,7 +729,7 @@ static int tlv320aic23_codec_probe(struct i2c_client *i2c, const struct i2c_device_id *i2c_id) { struct snd_soc_device *socdev = tlv320aic23_socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int ret; if (!i2c_check_functionality(i2c->adapter, I2C_FUNC_SMBUS_BYTE_DATA)) @@ -787,7 +787,7 @@ static int tlv320aic23_probe(struct platform_device *pdev) if (aic23 == NULL) return -ENOMEM; codec = &aic23->codec; - socdev->codec = codec; + socdev->card->codec = codec; mutex_init(&codec->mutex); INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); @@ -806,7 +806,7 @@ static int tlv320aic23_probe(struct platform_device *pdev) static int tlv320aic23_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; struct aic23 *aic23 = container_of(codec, struct aic23, codec); if (codec->control_data) diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index 29f2f1a017f..229e464cf71 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -130,7 +130,7 @@ static int aic26_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; struct aic26 *aic26 = codec->private_data; int fsref, divisor, wlen, pval, jval, dval, qval; u16 reg; @@ -338,7 +338,7 @@ static int aic26_probe(struct platform_device *pdev) return -ENODEV; } codec = &aic26->codec; - socdev->codec = codec; + socdev->card->codec = codec; dev_dbg(&pdev->dev, "Registering PCMs, dev=%p, socdev->dev=%p\n", &pdev->dev, socdev->dev); diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 36ab0198ca3..ba64b0c617e 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -727,7 +727,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; struct aic3x_priv *aic3x = codec->private_data; int codec_clk = 0, bypass_pll = 0, fsref, last_clk = 0; u8 data, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1; @@ -1079,7 +1079,7 @@ EXPORT_SYMBOL_GPL(aic3x_dai); static int aic3x_suspend(struct platform_device *pdev, pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; aic3x_set_bias_level(codec, SND_SOC_BIAS_OFF); @@ -1089,7 +1089,7 @@ static int aic3x_suspend(struct platform_device *pdev, pm_message_t state) static int aic3x_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int i; u8 data[2]; u8 *cache = codec->reg_cache; @@ -1112,7 +1112,7 @@ static int aic3x_resume(struct platform_device *pdev) */ static int aic3x_init(struct snd_soc_device *socdev) { - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; struct aic3x_setup_data *setup = socdev->codec_data; int reg, ret = 0; @@ -1243,7 +1243,7 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct snd_soc_device *socdev = aic3x_socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int ret; i2c_set_clientdata(i2c, codec); @@ -1348,7 +1348,7 @@ static int aic3x_probe(struct platform_device *pdev) } codec->private_data = aic3x; - socdev->codec = codec; + socdev->card->codec = codec; mutex_init(&codec->mutex); INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); @@ -1374,7 +1374,7 @@ static int aic3x_probe(struct platform_device *pdev) static int aic3x_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; /* power down chip */ if (codec->control_data) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index f530c1e6d9e..796f34cac85 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -981,7 +981,7 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; u8 mode, old_mode, format, old_format; @@ -1166,7 +1166,7 @@ EXPORT_SYMBOL_GPL(twl4030_dai); static int twl4030_suspend(struct platform_device *pdev, pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; twl4030_set_bias_level(codec, SND_SOC_BIAS_OFF); @@ -1176,7 +1176,7 @@ static int twl4030_suspend(struct platform_device *pdev, pm_message_t state) static int twl4030_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; twl4030_set_bias_level(codec, SND_SOC_BIAS_STANDBY); twl4030_set_bias_level(codec, codec->suspend_bias_level); @@ -1190,7 +1190,7 @@ static int twl4030_resume(struct platform_device *pdev) static int twl4030_init(struct snd_soc_device *socdev) { - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int ret = 0; printk(KERN_INFO "TWL4030 Audio Codec init \n"); @@ -1251,7 +1251,7 @@ static int twl4030_probe(struct platform_device *pdev) if (codec == NULL) return -ENOMEM; - socdev->codec = codec; + socdev->card->codec = codec; mutex_init(&codec->mutex); INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); @@ -1265,7 +1265,7 @@ static int twl4030_probe(struct platform_device *pdev) static int twl4030_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; printk(KERN_INFO "TWL4030 Audio Codec remove\n"); snd_soc_free_pcms(socdev); diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index 277825d155a..661599295ca 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -173,7 +173,7 @@ static int uda134x_startup(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; struct uda134x_priv *uda134x = codec->private_data; struct snd_pcm_runtime *master_runtime; @@ -206,7 +206,7 @@ static void uda134x_shutdown(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; struct uda134x_priv *uda134x = codec->private_data; if (uda134x->master_substream == substream) @@ -221,7 +221,7 @@ static int uda134x_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; struct uda134x_priv *uda134x = codec->private_data; u8 hw_params; @@ -492,11 +492,11 @@ static int uda134x_soc_probe(struct platform_device *pdev) return -EINVAL; } - socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); - if (socdev->codec == NULL) + socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (socdev->card->codec == NULL) return ret; - codec = socdev->codec; + codec = socdev->card->codec; uda134x = kzalloc(sizeof(struct uda134x_priv), GFP_KERNEL); if (uda134x == NULL) @@ -584,7 +584,7 @@ priv_err: static int uda134x_soc_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); uda134x_set_bias_level(codec, SND_SOC_BIAS_OFF); @@ -604,7 +604,7 @@ static int uda134x_soc_suspend(struct platform_device *pdev, pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); uda134x_set_bias_level(codec, SND_SOC_BIAS_OFF); @@ -614,7 +614,7 @@ static int uda134x_soc_suspend(struct platform_device *pdev, static int uda134x_soc_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; uda134x_set_bias_level(codec, SND_SOC_BIAS_PREPARE); uda134x_set_bias_level(codec, SND_SOC_BIAS_ON); diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index a957b4365b9..98e4a6560f0 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -397,7 +397,7 @@ static int uda1380_pcm_prepare(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int reg, reg_start, reg_end, clk; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { @@ -430,7 +430,7 @@ static int uda1380_pcm_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK); /* set WSPLL power and divider if running from this clock */ @@ -469,7 +469,7 @@ static void uda1380_pcm_shutdown(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK); /* shut down WSPLL power if running from this clock */ @@ -591,7 +591,7 @@ EXPORT_SYMBOL_GPL(uda1380_dai); static int uda1380_suspend(struct platform_device *pdev, pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; uda1380_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; @@ -600,7 +600,7 @@ static int uda1380_suspend(struct platform_device *pdev, pm_message_t state) static int uda1380_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int i; u8 data[2]; u16 *cache = codec->reg_cache; @@ -622,7 +622,7 @@ static int uda1380_resume(struct platform_device *pdev) */ static int uda1380_init(struct snd_soc_device *socdev, int dac_clk) { - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int ret = 0; codec->name = "UDA1380"; @@ -688,7 +688,7 @@ static int uda1380_i2c_probe(struct i2c_client *i2c, { struct snd_soc_device *socdev = uda1380_socdev; struct uda1380_setup_data *setup = socdev->codec_data; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int ret; i2c_set_clientdata(i2c, codec); @@ -779,7 +779,7 @@ static int uda1380_probe(struct platform_device *pdev) if (codec == NULL) return -ENOMEM; - socdev->codec = codec; + socdev->card->codec = codec; mutex_init(&codec->mutex); INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); @@ -803,7 +803,7 @@ static int uda1380_probe(struct platform_device *pdev) static int uda1380_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; if (codec->control_data) uda1380_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 2e0db29b499..75d3438ccb8 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1301,7 +1301,7 @@ static int wm8350_set_bias_level(struct snd_soc_codec *codec, static int wm8350_suspend(struct platform_device *pdev, pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; wm8350_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; @@ -1310,7 +1310,7 @@ static int wm8350_suspend(struct platform_device *pdev, pm_message_t state) static int wm8350_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; wm8350_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -1423,8 +1423,8 @@ static int wm8350_probe(struct platform_device *pdev) BUG_ON(!wm8350_codec); - socdev->codec = wm8350_codec; - codec = socdev->codec; + socdev->card->codec = wm8350_codec; + codec = socdev->card->codec; wm8350 = codec->control_data; priv = codec->private_data; @@ -1498,7 +1498,7 @@ card_err: static int wm8350_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; struct wm8350 *wm8350 = codec->control_data; struct wm8350_data *priv = codec->private_data; int ret; diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index abe7cce8771..f01078cfbd7 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -452,7 +452,7 @@ static int wm8510_pcm_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; u16 iface = wm8510_read_reg_cache(codec, WM8510_IFACE) & 0x19f; u16 adn = wm8510_read_reg_cache(codec, WM8510_ADD) & 0x1f1; @@ -581,7 +581,7 @@ EXPORT_SYMBOL_GPL(wm8510_dai); static int wm8510_suspend(struct platform_device *pdev, pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; wm8510_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; @@ -590,7 +590,7 @@ static int wm8510_suspend(struct platform_device *pdev, pm_message_t state) static int wm8510_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int i; u8 data[2]; u16 *cache = codec->reg_cache; @@ -612,7 +612,7 @@ static int wm8510_resume(struct platform_device *pdev) */ static int wm8510_init(struct snd_soc_device *socdev) { - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int ret = 0; codec->name = "WM8510"; @@ -670,7 +670,7 @@ static int wm8510_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct snd_soc_device *socdev = wm8510_socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int ret; i2c_set_clientdata(i2c, codec); @@ -751,7 +751,7 @@ err_driver: static int __devinit wm8510_spi_probe(struct spi_device *spi) { struct snd_soc_device *socdev = wm8510_socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int ret; codec->control_data = spi; @@ -817,7 +817,7 @@ static int wm8510_probe(struct platform_device *pdev) if (codec == NULL) return -ENOMEM; - socdev->codec = codec; + socdev->card->codec = codec; mutex_init(&codec->mutex); INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); @@ -847,7 +847,7 @@ static int wm8510_probe(struct platform_device *pdev) static int wm8510_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; if (codec->control_data) wm8510_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 3faf0e70ce1..d3c51ba5e6f 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -539,7 +539,7 @@ static int wm8580_paif_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; u16 paifb = wm8580_read(codec, WM8580_PAIF3 + dai->id); paifb &= ~WM8580_AIF_LENGTH_MASK; @@ -816,7 +816,7 @@ EXPORT_SYMBOL_GPL(wm8580_dai); */ static int wm8580_init(struct snd_soc_device *socdev) { - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int ret = 0; codec->name = "WM8580"; @@ -888,7 +888,7 @@ static int wm8580_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct snd_soc_device *socdev = wm8580_socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int ret; i2c_set_clientdata(i2c, codec); @@ -986,7 +986,7 @@ static int wm8580_probe(struct platform_device *pdev) } codec->private_data = wm8580; - socdev->codec = codec; + socdev->card->codec = codec; mutex_init(&codec->mutex); INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); @@ -1007,7 +1007,7 @@ static int wm8580_probe(struct platform_device *pdev) static int wm8580_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; if (codec->control_data) wm8580_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index f90dc52e975..f8363b30889 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -137,7 +137,7 @@ static int wm8728_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; u16 dac = wm8728_read_reg_cache(codec, WM8728_DACCTL); dac &= ~0x18; @@ -264,7 +264,7 @@ EXPORT_SYMBOL_GPL(wm8728_dai); static int wm8728_suspend(struct platform_device *pdev, pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; wm8728_set_bias_level(codec, SND_SOC_BIAS_OFF); @@ -274,7 +274,7 @@ static int wm8728_suspend(struct platform_device *pdev, pm_message_t state) static int wm8728_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; wm8728_set_bias_level(codec, codec->suspend_bias_level); @@ -287,7 +287,7 @@ static int wm8728_resume(struct platform_device *pdev) */ static int wm8728_init(struct snd_soc_device *socdev) { - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int ret = 0; codec->name = "WM8728"; @@ -349,7 +349,7 @@ static int wm8728_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct snd_soc_device *socdev = wm8728_socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int ret; i2c_set_clientdata(i2c, codec); @@ -430,7 +430,7 @@ err_driver: static int __devinit wm8728_spi_probe(struct spi_device *spi) { struct snd_soc_device *socdev = wm8728_socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int ret; codec->control_data = spi; @@ -494,7 +494,7 @@ static int wm8728_probe(struct platform_device *pdev) if (codec == NULL) return -ENOMEM; - socdev->codec = codec; + socdev->card->codec = codec; mutex_init(&codec->mutex); INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); @@ -527,7 +527,7 @@ static int wm8728_probe(struct platform_device *pdev) static int wm8728_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; if (codec->control_data) wm8728_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 96d6e1aeaf4..0150fe53a65 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -253,7 +253,7 @@ static int wm8731_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; struct wm8731_priv *wm8731 = codec->private_data; u16 iface = wm8731_read_reg_cache(codec, WM8731_IFACE) & 0xfff3; int i = get_coeff(wm8731->sysclk, params_rate(params)); @@ -283,7 +283,7 @@ static int wm8731_pcm_prepare(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; /* set active */ wm8731_write(codec, WM8731_ACTIVE, 0x0001); @@ -296,7 +296,7 @@ static void wm8731_shutdown(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; /* deactivate */ if (!codec->active) { @@ -458,7 +458,7 @@ EXPORT_SYMBOL_GPL(wm8731_dai); static int wm8731_suspend(struct platform_device *pdev, pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; wm8731_write(codec, WM8731_ACTIVE, 0x0); wm8731_set_bias_level(codec, SND_SOC_BIAS_OFF); @@ -468,7 +468,7 @@ static int wm8731_suspend(struct platform_device *pdev, pm_message_t state) static int wm8731_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int i; u8 data[2]; u16 *cache = codec->reg_cache; @@ -490,7 +490,7 @@ static int wm8731_resume(struct platform_device *pdev) */ static int wm8731_init(struct snd_soc_device *socdev) { - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int reg, ret = 0; codec->name = "WM8731"; @@ -561,7 +561,7 @@ static int wm8731_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct snd_soc_device *socdev = wm8731_socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int ret; i2c_set_clientdata(i2c, codec); @@ -642,7 +642,7 @@ err_driver: static int __devinit wm8731_spi_probe(struct spi_device *spi) { struct snd_soc_device *socdev = wm8731_socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int ret; codec->control_data = spi; @@ -716,7 +716,7 @@ static int wm8731_probe(struct platform_device *pdev) } codec->private_data = wm8731; - socdev->codec = codec; + socdev->card->codec = codec; mutex_init(&codec->mutex); INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); @@ -750,7 +750,7 @@ static int wm8731_probe(struct platform_device *pdev) static int wm8731_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; if (codec->control_data) wm8731_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 1578569793a..96afb86addc 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -604,7 +604,7 @@ static int wm8750_pcm_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; struct wm8750_priv *wm8750 = codec->private_data; u16 iface = wm8750_read_reg_cache(codec, WM8750_IFACE) & 0x1f3; u16 srate = wm8750_read_reg_cache(codec, WM8750_SRATE) & 0x1c0; @@ -712,7 +712,7 @@ static void wm8750_work(struct work_struct *work) static int wm8750_suspend(struct platform_device *pdev, pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; wm8750_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; @@ -721,7 +721,7 @@ static int wm8750_suspend(struct platform_device *pdev, pm_message_t state) static int wm8750_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int i; u8 data[2]; u16 *cache = codec->reg_cache; @@ -754,7 +754,7 @@ static int wm8750_resume(struct platform_device *pdev) */ static int wm8750_init(struct snd_soc_device *socdev) { - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int reg, ret = 0; codec->name = "WM8750"; @@ -836,7 +836,7 @@ static int wm8750_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct snd_soc_device *socdev = wm8750_socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int ret; i2c_set_clientdata(i2c, codec); @@ -917,7 +917,7 @@ err_driver: static int __devinit wm8750_spi_probe(struct spi_device *spi) { struct snd_soc_device *socdev = wm8750_socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int ret; codec->control_data = spi; @@ -989,7 +989,7 @@ static int wm8750_probe(struct platform_device *pdev) } codec->private_data = wm8750; - socdev->codec = codec; + socdev->card->codec = codec; mutex_init(&codec->mutex); INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); @@ -1043,7 +1043,7 @@ static int run_delayed_work(struct delayed_work *dwork) static int wm8750_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; if (codec->control_data) wm8750_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 5a1c1fca120..502766dce86 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -912,7 +912,7 @@ static int wm8753_pcm_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; struct wm8753_priv *wm8753 = codec->private_data; u16 voice = wm8753_read_reg_cache(codec, WM8753_PCM) & 0x01f3; u16 srate = wm8753_read_reg_cache(codec, WM8753_SRATE1) & 0x017f; @@ -1146,7 +1146,7 @@ static int wm8753_i2s_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; struct wm8753_priv *wm8753 = codec->private_data; u16 srate = wm8753_read_reg_cache(codec, WM8753_SRATE1) & 0x01c0; u16 hifi = wm8753_read_reg_cache(codec, WM8753_HIFI) & 0x01f3; @@ -1483,7 +1483,7 @@ static void wm8753_work(struct work_struct *work) static int wm8753_suspend(struct platform_device *pdev, pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; /* we only need to suspend if we are a valid card */ if (!codec->card) @@ -1496,7 +1496,7 @@ static int wm8753_suspend(struct platform_device *pdev, pm_message_t state) static int wm8753_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int i; u8 data[2]; u16 *cache = codec->reg_cache; @@ -1533,7 +1533,7 @@ static int wm8753_resume(struct platform_device *pdev) */ static int wm8753_init(struct snd_soc_device *socdev) { - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int reg, ret = 0; codec->name = "WM8753"; @@ -1624,7 +1624,7 @@ static int wm8753_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct snd_soc_device *socdev = wm8753_socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int ret; i2c_set_clientdata(i2c, codec); @@ -1705,7 +1705,7 @@ err_driver: static int __devinit wm8753_spi_probe(struct spi_device *spi) { struct snd_soc_device *socdev = wm8753_socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int ret; codec->control_data = spi; @@ -1780,7 +1780,7 @@ static int wm8753_probe(struct platform_device *pdev) } codec->private_data = wm8753; - socdev->codec = codec; + socdev->card->codec = codec; mutex_init(&codec->mutex); INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); @@ -1832,7 +1832,7 @@ static int run_delayed_work(struct delayed_work *dwork) static int wm8753_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; if (codec->control_data) wm8753_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 1e08d4f065f..85c0f1bc676 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -720,7 +720,7 @@ static int wm8900_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; u16 reg; reg = wm8900_read(codec, WM8900_REG_AUDIO1) & ~0x60; @@ -1210,7 +1210,7 @@ static int wm8900_set_bias_level(struct snd_soc_codec *codec, static int wm8900_suspend(struct platform_device *pdev, pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; struct wm8900_priv *wm8900 = codec->private_data; int fll_out = wm8900->fll_out; int fll_in = wm8900->fll_in; @@ -1234,7 +1234,7 @@ static int wm8900_suspend(struct platform_device *pdev, pm_message_t state) static int wm8900_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; struct wm8900_priv *wm8900 = codec->private_data; u16 *cache; int i, ret; @@ -1414,7 +1414,7 @@ static int wm8900_probe(struct platform_device *pdev) } codec = wm8900_codec; - socdev->codec = codec; + socdev->card->codec = codec; /* Register pcms */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 6ff34b957dc..d36b2b1edf1 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1261,7 +1261,7 @@ static int wm8903_startup(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; struct wm8903_priv *wm8903 = codec->private_data; struct i2c_client *i2c = codec->control_data; struct snd_pcm_runtime *master_runtime; @@ -1303,7 +1303,7 @@ static void wm8903_shutdown(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; struct wm8903_priv *wm8903 = codec->private_data; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) @@ -1323,7 +1323,7 @@ static int wm8903_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; struct wm8903_priv *wm8903 = codec->private_data; struct i2c_client *i2c = codec->control_data; int fs = params_rate(params); @@ -1527,7 +1527,7 @@ EXPORT_SYMBOL_GPL(wm8903_dai); static int wm8903_suspend(struct platform_device *pdev, pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; wm8903_set_bias_level(codec, SND_SOC_BIAS_OFF); @@ -1537,7 +1537,7 @@ static int wm8903_suspend(struct platform_device *pdev, pm_message_t state) static int wm8903_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; struct i2c_client *i2c = codec->control_data; int i; u16 *reg_cache = codec->reg_cache; @@ -1713,7 +1713,7 @@ static int wm8903_probe(struct platform_device *pdev) goto err; } - socdev->codec = wm8903_codec; + socdev->card->codec = wm8903_codec; /* register pcms */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); @@ -1722,9 +1722,9 @@ static int wm8903_probe(struct platform_device *pdev) goto err; } - snd_soc_add_controls(socdev->codec, wm8903_snd_controls, + snd_soc_add_controls(socdev->card->codec, wm8903_snd_controls, ARRAY_SIZE(wm8903_snd_controls)); - wm8903_add_widgets(socdev->codec); + wm8903_add_widgets(socdev->card->codec); ret = snd_soc_init_card(socdev); if (ret < 0) { @@ -1745,7 +1745,7 @@ err: static int wm8903_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; if (codec->control_data) wm8903_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index c8bd9b06f33..24d4c905a01 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -531,7 +531,7 @@ static int wm8971_pcm_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; struct wm8971_priv *wm8971 = codec->private_data; u16 iface = wm8971_read_reg_cache(codec, WM8971_IFACE) & 0x1f3; u16 srate = wm8971_read_reg_cache(codec, WM8971_SRATE) & 0x1c0; @@ -637,7 +637,7 @@ static void wm8971_work(struct work_struct *work) static int wm8971_suspend(struct platform_device *pdev, pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; wm8971_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; @@ -646,7 +646,7 @@ static int wm8971_suspend(struct platform_device *pdev, pm_message_t state) static int wm8971_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int i; u8 data[2]; u16 *cache = codec->reg_cache; @@ -677,7 +677,7 @@ static int wm8971_resume(struct platform_device *pdev) static int wm8971_init(struct snd_soc_device *socdev) { - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int reg, ret = 0; codec->name = "WM8971"; @@ -758,7 +758,7 @@ static int wm8971_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct snd_soc_device *socdev = wm8971_socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int ret; i2c_set_clientdata(i2c, codec); @@ -859,7 +859,7 @@ static int wm8971_probe(struct platform_device *pdev) } codec->private_data = wm8971; - socdev->codec = codec; + socdev->card->codec = codec; mutex_init(&codec->mutex); INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); @@ -894,7 +894,7 @@ static int wm8971_probe(struct platform_device *pdev) static int wm8971_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; if (codec->control_data) wm8971_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index f93c0955ed9..6af1d399b31 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1162,7 +1162,7 @@ static int wm8990_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; u16 audio1 = wm8990_read_reg_cache(codec, WM8990_AUDIO_INTERFACE_1); audio1 &= ~WM8990_AIF_WL_MASK; @@ -1361,7 +1361,7 @@ EXPORT_SYMBOL_GPL(wm8990_dai); static int wm8990_suspend(struct platform_device *pdev, pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; /* we only need to suspend if we are a valid card */ if (!codec->card) @@ -1374,7 +1374,7 @@ static int wm8990_suspend(struct platform_device *pdev, pm_message_t state) static int wm8990_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int i; u8 data[2]; u16 *cache = codec->reg_cache; @@ -1402,7 +1402,7 @@ static int wm8990_resume(struct platform_device *pdev) */ static int wm8990_init(struct snd_soc_device *socdev) { - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; u16 reg; int ret = 0; @@ -1480,7 +1480,7 @@ static int wm8990_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct snd_soc_device *socdev = wm8990_socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int ret; i2c_set_clientdata(i2c, codec); @@ -1579,7 +1579,7 @@ static int wm8990_probe(struct platform_device *pdev) } codec->private_data = wm8990; - socdev->codec = codec; + socdev->card->codec = codec; mutex_init(&codec->mutex); INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); @@ -1605,7 +1605,7 @@ static int wm8990_probe(struct platform_device *pdev) static int wm8990_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; if (codec->control_data) wm8990_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index d5c81bb3dec..2e9e06b2daa 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -249,7 +249,7 @@ static int ac97_prepare(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int reg; u16 vra; @@ -323,10 +323,11 @@ static int wm9705_soc_probe(struct platform_device *pdev) printk(KERN_INFO "WM9705 SoC Audio Codec\n"); - socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); - if (socdev->codec == NULL) + socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec), + GFP_KERNEL); + if (socdev->card->codec == NULL) return -ENOMEM; - codec = socdev->codec; + codec = socdev->card->codec; mutex_init(&codec->mutex); codec->reg_cache = kmemdup(wm9705_reg, sizeof(wm9705_reg), GFP_KERNEL); @@ -380,15 +381,15 @@ pcm_err: codec_err: kfree(codec->reg_cache); cache_err: - kfree(socdev->codec); - socdev->codec = NULL; + kfree(socdev->card->codec); + socdev->card->codec = NULL; return ret; } static int wm9705_soc_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; if (codec == NULL) return 0; diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 4dc90d67530..b3a8be77676 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -478,7 +478,7 @@ static int ac97_prepare(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int reg; u16 vra; @@ -499,7 +499,7 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; u16 vra, xsle; vra = ac97_read(codec, AC97_EXTENDED_STATUS); @@ -592,7 +592,7 @@ static int wm9712_soc_suspend(struct platform_device *pdev, pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; wm9712_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; @@ -601,7 +601,7 @@ static int wm9712_soc_suspend(struct platform_device *pdev, static int wm9712_soc_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int i, ret; u16 *cache = codec->reg_cache; @@ -637,10 +637,11 @@ static int wm9712_soc_probe(struct platform_device *pdev) printk(KERN_INFO "WM9711/WM9712 SoC Audio Codec %s\n", WM9712_VERSION); - socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); - if (socdev->codec == NULL) + socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec), + GFP_KERNEL); + if (socdev->card->codec == NULL) return -ENOMEM; - codec = socdev->codec; + codec = socdev->card->codec; mutex_init(&codec->mutex); codec->reg_cache = kmemdup(wm9712_reg, sizeof(wm9712_reg), GFP_KERNEL); @@ -704,15 +705,15 @@ codec_err: kfree(codec->reg_cache); cache_err: - kfree(socdev->codec); - socdev->codec = NULL; + kfree(socdev->card->codec); + socdev->card->codec = NULL; return ret; } static int wm9712_soc_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; if (codec == NULL) return 0; diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 0e60e16973d..54db9c52498 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1115,7 +1115,7 @@ static int wm9713_soc_suspend(struct platform_device *pdev, pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; u16 reg; /* Disable everything except touchpanel - that will be handled @@ -1133,7 +1133,7 @@ static int wm9713_soc_suspend(struct platform_device *pdev, static int wm9713_soc_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; struct wm9713_priv *wm9713 = codec->private_data; int i, ret; u16 *cache = codec->reg_cache; @@ -1174,10 +1174,11 @@ static int wm9713_soc_probe(struct platform_device *pdev) printk(KERN_INFO "WM9713/WM9714 SoC Audio Codec %s\n", WM9713_VERSION); - socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); - if (socdev->codec == NULL) + socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec), + GFP_KERNEL); + if (socdev->card->codec == NULL) return -ENOMEM; - codec = socdev->codec; + codec = socdev->card->codec; mutex_init(&codec->mutex); codec->reg_cache = kmemdup(wm9713_reg, sizeof(wm9713_reg), GFP_KERNEL); @@ -1249,15 +1250,15 @@ priv_err: kfree(codec->reg_cache); cache_err: - kfree(socdev->codec); - socdev->codec = NULL; + kfree(socdev->card->codec); + socdev->card->codec = NULL; return ret; } static int wm9713_soc_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; if (codec == NULL) return 0; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 8313d52a6e8..f18c7a3e36d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -234,7 +234,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) cpu_dai->capture.active = codec_dai->capture.active = 1; cpu_dai->active = codec_dai->active = 1; cpu_dai->runtime = runtime; - socdev->codec->active++; + card->codec->active++; mutex_unlock(&pcm_mutex); return 0; @@ -264,7 +264,7 @@ static void close_delayed_work(struct work_struct *work) struct snd_soc_card *card = container_of(work, struct snd_soc_card, delayed_work.work); struct snd_soc_device *socdev = card->socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = card->codec; struct snd_soc_dai *codec_dai; int i; @@ -319,7 +319,7 @@ static int soc_codec_close(struct snd_pcm_substream *substream) struct snd_soc_platform *platform = card->platform; struct snd_soc_dai *cpu_dai = machine->cpu_dai; struct snd_soc_dai *codec_dai = machine->codec_dai; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = card->codec; mutex_lock(&pcm_mutex); @@ -387,7 +387,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) struct snd_soc_platform *platform = card->platform; struct snd_soc_dai *cpu_dai = machine->cpu_dai; struct snd_soc_dai *codec_dai = machine->codec_dai; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = card->codec; int ret = 0; mutex_lock(&pcm_mutex); @@ -553,7 +553,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) struct snd_soc_platform *platform = card->platform; struct snd_soc_dai *cpu_dai = machine->cpu_dai; struct snd_soc_dai *codec_dai = machine->codec_dai; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = card->codec; mutex_lock(&pcm_mutex); @@ -629,7 +629,7 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) struct snd_soc_card *card = socdev->card; struct snd_soc_platform *platform = card->platform; struct snd_soc_codec_device *codec_dev = socdev->codec_dev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = card->codec; int i; /* Due to the resume being scheduled into a workqueue we could @@ -705,7 +705,7 @@ static void soc_resume_deferred(struct work_struct *work) struct snd_soc_device *socdev = card->socdev; struct snd_soc_platform *platform = card->platform; struct snd_soc_codec_device *codec_dev = socdev->codec_dev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = card->codec; struct platform_device *pdev = to_platform_device(socdev->dev); int i; @@ -982,8 +982,8 @@ static struct platform_driver soc_driver = { static int soc_new_pcm(struct snd_soc_device *socdev, struct snd_soc_dai_link *dai_link, int num) { - struct snd_soc_codec *codec = socdev->codec; struct snd_soc_card *card = socdev->card; + struct snd_soc_codec *codec = card->codec; struct snd_soc_platform *platform = card->platform; struct snd_soc_dai *codec_dai = dai_link->codec_dai; struct snd_soc_dai *cpu_dai = dai_link->cpu_dai; @@ -998,7 +998,7 @@ static int soc_new_pcm(struct snd_soc_device *socdev, rtd->dai = dai_link; rtd->socdev = socdev; - codec_dai->codec = socdev->codec; + codec_dai->codec = card->codec; /* check client and interface hw capabilities */ sprintf(new_name, "%s %s-%d", dai_link->stream_name, codec_dai->name, @@ -1048,9 +1048,8 @@ static int soc_new_pcm(struct snd_soc_device *socdev, } /* codec register dump */ -static ssize_t soc_codec_reg_show(struct snd_soc_device *devdata, char *buf) +static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf) { - struct snd_soc_codec *codec = devdata->codec; int i, step = 1, count = 0; if (!codec->reg_cache_size) @@ -1090,7 +1089,7 @@ static ssize_t codec_reg_show(struct device *dev, struct device_attribute *attr, char *buf) { struct snd_soc_device *devdata = dev_get_drvdata(dev); - return soc_codec_reg_show(devdata, buf); + return soc_codec_reg_show(devdata->card->codec, buf); } static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL); @@ -1107,12 +1106,10 @@ static ssize_t codec_reg_read_file(struct file *file, char __user *user_buf, { ssize_t ret; struct snd_soc_codec *codec = file->private_data; - struct device *card_dev = codec->card->dev; - struct snd_soc_device *devdata = card_dev->driver_data; char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL); if (!buf) return -ENOMEM; - ret = soc_codec_reg_show(devdata, buf); + ret = soc_codec_reg_show(codec, buf); if (ret >= 0) ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret); kfree(buf); @@ -1309,8 +1306,8 @@ EXPORT_SYMBOL_GPL(snd_soc_test_bits); */ int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid) { - struct snd_soc_codec *codec = socdev->codec; struct snd_soc_card *card = socdev->card; + struct snd_soc_codec *codec = card->codec; int ret = 0, i; mutex_lock(&codec->mutex); @@ -1355,8 +1352,8 @@ EXPORT_SYMBOL_GPL(snd_soc_new_pcms); */ int snd_soc_init_card(struct snd_soc_device *socdev) { - struct snd_soc_codec *codec = socdev->codec; struct snd_soc_card *card = socdev->card; + struct snd_soc_codec *codec = card->codec; int ret = 0, i, ac97 = 0, err = 0; for (i = 0; i < card->num_links; i++) { @@ -1404,7 +1401,7 @@ int snd_soc_init_card(struct snd_soc_device *socdev) if (err < 0) printk(KERN_WARNING "asoc: failed to add codec sysfs files\n"); - soc_init_codec_debugfs(socdev->codec); + soc_init_codec_debugfs(codec); mutex_unlock(&codec->mutex); out: @@ -1421,14 +1418,14 @@ EXPORT_SYMBOL_GPL(snd_soc_init_card); */ void snd_soc_free_pcms(struct snd_soc_device *socdev) { - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; #ifdef CONFIG_SND_SOC_AC97_BUS struct snd_soc_dai *codec_dai; int i; #endif mutex_lock(&codec->mutex); - soc_cleanup_codec_debugfs(socdev->codec); + soc_cleanup_codec_debugfs(codec); #ifdef CONFIG_SND_SOC_AC97_BUS for (i = 0; i < codec->num_dai; i++) { codec_dai = &codec->dai[i]; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 54b4564b82b..f4a8753c84c 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -817,7 +817,7 @@ static ssize_t dapm_widget_show(struct device *dev, struct device_attribute *attr, char *buf) { struct snd_soc_device *devdata = dev_get_drvdata(dev); - struct snd_soc_codec *codec = devdata->codec; + struct snd_soc_codec *codec = devdata->card->codec; struct snd_soc_dapm_widget *w; int count = 0; char *state = "not set"; @@ -1552,8 +1552,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_stream_event); int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev, enum snd_soc_bias_level level) { - struct snd_soc_codec *codec = socdev->codec; struct snd_soc_card *card = socdev->card; + struct snd_soc_codec *codec = socdev->card->codec; int ret = 0; if (card->set_bias_level) @@ -1645,7 +1645,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_get_pin_status); */ void snd_soc_dapm_free(struct snd_soc_device *socdev) { - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; snd_soc_dapm_sys_remove(socdev->dev); dapm_free_widgets(codec); diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 8cc00c3cdf3..ab64a30bedd 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -34,7 +34,7 @@ int snd_soc_jack_new(struct snd_soc_card *card, const char *id, int type, jack->card = card; INIT_LIST_HEAD(&jack->pins); - return snd_jack_new(card->socdev->codec->card, id, type, &jack->jack); + return snd_jack_new(card->codec->card, id, type, &jack->jack); } EXPORT_SYMBOL_GPL(snd_soc_jack_new); @@ -54,7 +54,7 @@ EXPORT_SYMBOL_GPL(snd_soc_jack_new); */ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) { - struct snd_soc_codec *codec = jack->card->socdev->codec; + struct snd_soc_codec *codec = jack->card->codec; struct snd_soc_jack_pin *pin; int enable; int oldstatus; -- cgit v1.2.3-70-g09d2 From b9d710b3c530ed91e8683933fe94c7605d175bf5 Mon Sep 17 00:00:00 2001 From: Andreas Bergmeier Date: Sat, 24 Jan 2009 12:15:14 +0100 Subject: ALSA: usbaudio - use printf format instead of hardcoding it Rather use printf format instead of hardcoding prefix like 0x. A next step would be to predefine certain formats. Signed-off-by: Andreas Bergmeier Signed-off-by: Takashi Iwai --- sound/usb/usbaudio.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 44485b29f67..4636926d12d 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -1280,14 +1280,14 @@ static int init_usb_sample_rate(struct usb_device *dev, int iface, if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), SET_CUR, USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_OUT, SAMPLING_FREQ_CONTROL << 8, ep, data, 3, 1000)) < 0) { - snd_printk(KERN_ERR "%d:%d:%d: cannot set freq %d to ep 0x%x\n", + snd_printk(KERN_ERR "%d:%d:%d: cannot set freq %d to ep %#x\n", dev->devnum, iface, fmt->altsetting, rate, ep); return err; } if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), GET_CUR, USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_IN, SAMPLING_FREQ_CONTROL << 8, ep, data, 3, 1000)) < 0) { - snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq at ep 0x%x\n", + snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq at ep %#x\n", dev->devnum, iface, fmt->altsetting, ep); return 0; /* some devices don't support reading */ } @@ -1456,7 +1456,7 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, channels = params_channels(hw_params); fmt = find_format(subs, format, rate, channels); if (!fmt) { - snd_printd(KERN_DEBUG "cannot set format: format = 0x%x, rate = %d, channels = %d\n", + snd_printd(KERN_DEBUG "cannot set format: format = %#x, rate = %d, channels = %d\n", format, rate, channels); return -EINVAL; } @@ -2148,7 +2148,7 @@ static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct s fp = list_entry(p, struct audioformat, list); snd_iprintf(buffer, " Interface %d\n", fp->iface); snd_iprintf(buffer, " Altset %d\n", fp->altsetting); - snd_iprintf(buffer, " Format: 0x%x\n", fp->format); + snd_iprintf(buffer, " Format: %#x\n", fp->format); snd_iprintf(buffer, " Channels: %d\n", fp->channels); snd_iprintf(buffer, " Endpoint: %d %s (%s)\n", fp->endpoint & USB_ENDPOINT_NUMBER_MASK, @@ -2168,7 +2168,7 @@ static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct s snd_iprintf(buffer, "\n"); } // snd_iprintf(buffer, " Max Packet Size = %d\n", fp->maxpacksize); - // snd_iprintf(buffer, " EP Attribute = 0x%x\n", fp->attributes); + // snd_iprintf(buffer, " EP Attribute = %#x\n", fp->attributes); } } @@ -2607,7 +2607,7 @@ static int parse_audio_format_ii(struct snd_usb_audio *chip, struct audioformat fp->format = SNDRV_PCM_FORMAT_MPEG; break; default: - snd_printd(KERN_INFO "%d:%u:%d : unknown format tag 0x%x is detected. processed as MPEG.\n", + snd_printd(KERN_INFO "%d:%u:%d : unknown format tag %#x is detected. processed as MPEG.\n", chip->dev->devnum, fp->iface, fp->altsetting, format); fp->format = SNDRV_PCM_FORMAT_MPEG; break; @@ -2805,7 +2805,7 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) continue; } - snd_printdd(KERN_INFO "%d:%u:%d: add audio endpoint 0x%x\n", dev->devnum, iface_no, altno, fp->endpoint); + snd_printdd(KERN_INFO "%d:%u:%d: add audio endpoint %#x\n", dev->devnum, iface_no, altno, fp->endpoint); err = add_audio_endpoint(chip, stream, fp); if (err < 0) { kfree(fp->rate_table); -- cgit v1.2.3-70-g09d2 From 3fc93030e5a792fdd0da3321487f5cbfd1143c2b Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 27 Jan 2009 11:29:39 +0200 Subject: ASoC: TWL4030: Syncronize the reg_cache for ANAMICL after the offset cancelation The offset cancelation bit in ANAMICL register is self cleanig. Make sure that the reg_cache holds the same value as the HW register. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 796f34cac85..24419afd319 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -913,6 +913,9 @@ static void twl4030_power_up(struct snd_soc_codec *codec) ((byte & TWL4030_CNCL_OFFSET_START) == TWL4030_CNCL_OFFSET_START)); + /* Make sure that the reg_cache has the same value as the HW */ + twl4030_write_reg_cache(codec, TWL4030_REG_ANAMICL, byte); + /* anti-pop when changing analog gain */ regmisc1 = twl4030_read_reg_cache(codec, TWL4030_REG_MISC_SET_1); twl4030_write(codec, TWL4030_REG_MISC_SET_1, -- cgit v1.2.3-70-g09d2 From db04e2c58a65364218b89f1372b4b3b78d206423 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 27 Jan 2009 11:29:40 +0200 Subject: ASoC: TWL4030: Code clean up for codec power up and down Merge the codec up and down functions to a simple one. Codec is powered down by default (reg_cache change). Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 43 +++++++++++++++++-------------------------- 1 file changed, 17 insertions(+), 26 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 24419afd319..af7b433d4f5 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -42,7 +42,7 @@ */ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0x00, /* this register not used */ - 0x93, /* REG_CODEC_MODE (0x1) */ + 0x91, /* REG_CODEC_MODE (0x1) */ 0xc3, /* REG_OPTION (0x2) */ 0x00, /* REG_UNKNOWN (0x3) */ 0x00, /* REG_MICBIAS_CTL (0x4) */ @@ -154,26 +154,17 @@ static int twl4030_write(struct snd_soc_codec *codec, return twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value, reg); } -static void twl4030_clear_codecpdz(struct snd_soc_codec *codec) +static void twl4030_codec_enable(struct snd_soc_codec *codec, int enable) { u8 mode; mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE); - twl4030_write(codec, TWL4030_REG_CODEC_MODE, - mode & ~TWL4030_CODECPDZ); + if (enable) + mode |= TWL4030_CODECPDZ; + else + mode &= ~TWL4030_CODECPDZ; - /* REVISIT: this delay is present in TI sample drivers */ - /* but there seems to be no TRM requirement for it */ - udelay(10); -} - -static void twl4030_set_codecpdz(struct snd_soc_codec *codec) -{ - u8 mode; - - mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE); - twl4030_write(codec, TWL4030_REG_CODEC_MODE, - mode | TWL4030_CODECPDZ); + twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode); /* REVISIT: this delay is present in TI sample drivers */ /* but there seems to be no TRM requirement for it */ @@ -185,7 +176,7 @@ static void twl4030_init_chip(struct snd_soc_codec *codec) int i; /* clear CODECPDZ prior to setting register defaults */ - twl4030_clear_codecpdz(codec); + twl4030_codec_enable(codec, 0); /* set all audio section registers to reasonable defaults */ for (i = TWL4030_REG_OPTION; i <= TWL4030_REG_MISC_SET_2; i++) @@ -895,7 +886,7 @@ static void twl4030_power_up(struct snd_soc_codec *codec) int i = 0; /* set CODECPDZ to turn on codec */ - twl4030_set_codecpdz(codec); + twl4030_codec_enable(codec, 1); /* initiate offset cancellation */ anamicl = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL); @@ -922,8 +913,8 @@ static void twl4030_power_up(struct snd_soc_codec *codec) regmisc1 | TWL4030_SMOOTH_ANAVOL_EN); /* toggle CODECPDZ as per TRM */ - twl4030_clear_codecpdz(codec); - twl4030_set_codecpdz(codec); + twl4030_codec_enable(codec, 0); + twl4030_codec_enable(codec, 1); /* program anti-pop with bias ramp delay */ popn = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET); @@ -952,7 +943,7 @@ static void twl4030_power_down(struct snd_soc_codec *codec) twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn); /* power down */ - twl4030_clear_codecpdz(codec); + twl4030_codec_enable(codec, 0); } static int twl4030_set_bias_level(struct snd_soc_codec *codec, @@ -1030,7 +1021,7 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, if (mode != old_mode) { /* change rate and set CODECPDZ */ twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode); - twl4030_set_codecpdz(codec); + twl4030_codec_enable(codec, 1); } /* sample size */ @@ -1053,13 +1044,13 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, if (format != old_format) { /* clear CODECPDZ before changing format (codec requirement) */ - twl4030_clear_codecpdz(codec); + twl4030_codec_enable(codec, 0); /* change format */ twl4030_write(codec, TWL4030_REG_AUDIO_IF, format); /* set CODECPDZ afterwards */ - twl4030_set_codecpdz(codec); + twl4030_codec_enable(codec, 1); } return 0; } @@ -1129,13 +1120,13 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, if (format != old_format) { /* clear CODECPDZ before changing format (codec requirement) */ - twl4030_clear_codecpdz(codec); + twl4030_codec_enable(codec, 0); /* change format */ twl4030_write(codec, TWL4030_REG_AUDIO_IF, format); /* set CODECPDZ afterwards */ - twl4030_set_codecpdz(codec); + twl4030_codec_enable(codec, 1); } return 0; -- cgit v1.2.3-70-g09d2 From aad749e51a66d473f5cef4a050e3e36795261be3 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 27 Jan 2009 11:29:41 +0200 Subject: ASoC: TWL4030: Enable Headset Left anti-pop/bias ramp only if the Headset Left is in use The Headset Left anti-pop and bias ramp does not need to be enabled, if the headset is not in use. Move the code to DAPM event handler for HeadsetL. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 71 +++++++++++++++++++++++++++++----------------- 1 file changed, 45 insertions(+), 26 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index af7b433d4f5..900486ef633 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -414,6 +414,47 @@ static int handsfree_event(struct snd_soc_dapm_widget *w, return 0; } +static int headsetl_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + unsigned char hs_gain, hs_pop; + + /* Save the current volume */ + hs_gain = twl4030_read_reg_cache(w->codec, TWL4030_REG_HS_GAIN_SET); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* Do the anti-pop/bias ramp enable according to the TRM */ + hs_pop = TWL4030_RAMP_DELAY_645MS; + twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop); + hs_pop |= TWL4030_VMID_EN; + twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop); + /* Is this needed? Can we just use whatever gain here? */ + twl4030_write(w->codec, TWL4030_REG_HS_GAIN_SET, + (hs_gain & (~0x0f)) | 0x0a); + hs_pop |= TWL4030_RAMP_EN; + twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop); + + /* Restore the original volume */ + twl4030_write(w->codec, TWL4030_REG_HS_GAIN_SET, hs_gain); + break; + case SND_SOC_DAPM_POST_PMD: + /* Do the anti-pop/bias ramp disable according to the TRM */ + hs_pop = twl4030_read_reg_cache(w->codec, + TWL4030_REG_HS_POPN_SET); + hs_pop &= ~TWL4030_RAMP_EN; + twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop); + /* Bypass the reg_cache to mute the headset */ + twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, + hs_gain & (~0x0f), + TWL4030_REG_HS_GAIN_SET); + hs_pop &= ~TWL4030_VMID_EN; + twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop); + break; + } + return 0; +} + /* * Some of the gain controls in TWL (mostly those which are associated with * the outputs) are implemented in an interesting way: @@ -720,8 +761,9 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_VALUE_MUX("PredriveR Mux", SND_SOC_NOPM, 0, 0, &twl4030_dapm_predriver_control), /* HeadsetL/R */ - SND_SOC_DAPM_MUX("HeadsetL Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_hsol_control), + SND_SOC_DAPM_MUX_E("HeadsetL Mux", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_hsol_control, headsetl_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_MUX("HeadsetR Mux", SND_SOC_NOPM, 0, 0, &twl4030_dapm_hsor_control), /* CarkitL/R */ @@ -882,7 +924,7 @@ static int twl4030_add_widgets(struct snd_soc_codec *codec) static void twl4030_power_up(struct snd_soc_codec *codec) { - u8 anamicl, regmisc1, byte, popn; + u8 anamicl, regmisc1, byte; int i = 0; /* set CODECPDZ to turn on codec */ @@ -915,33 +957,10 @@ static void twl4030_power_up(struct snd_soc_codec *codec) /* toggle CODECPDZ as per TRM */ twl4030_codec_enable(codec, 0); twl4030_codec_enable(codec, 1); - - /* program anti-pop with bias ramp delay */ - popn = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET); - popn &= TWL4030_RAMP_DELAY; - popn |= TWL4030_RAMP_DELAY_645MS; - twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn); - popn |= TWL4030_VMID_EN; - twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn); - - /* enable anti-pop ramp */ - popn |= TWL4030_RAMP_EN; - twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn); } static void twl4030_power_down(struct snd_soc_codec *codec) { - u8 popn; - - /* disable anti-pop ramp */ - popn = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET); - popn &= ~TWL4030_RAMP_EN; - twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn); - - /* disable bias out */ - popn &= ~TWL4030_VMID_EN; - twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn); - /* power down */ twl4030_codec_enable(codec, 0); } -- cgit v1.2.3-70-g09d2 From fb2a2f84908460c18c3894963990518b224dd9ff Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 27 Jan 2009 11:29:42 +0200 Subject: ASoC: TWL4030: Physical ADC and amplifier power switch change Change the power switches for the physical ADC and for the amplifiers for the analog capture path to map more closely the actual path inside of the codec. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 26 +++++++++++++------------- 1 file changed, 13 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 900486ef633..8fe059e3e9a 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -802,16 +802,16 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD| SND_SOC_DAPM_POST_REG), - /* Analog input muxes with power switch for the physical ADCL/R */ + /* Analog input muxes with switch for the capture amplifiers */ SND_SOC_DAPM_VALUE_MUX("Analog Left Capture Route", - TWL4030_REG_AVADC_CTL, 3, 0, &twl4030_dapm_analoglmic_control), + TWL4030_REG_ANAMICL, 4, 0, &twl4030_dapm_analoglmic_control), SND_SOC_DAPM_VALUE_MUX("Analog Right Capture Route", - TWL4030_REG_AVADC_CTL, 1, 0, &twl4030_dapm_analogrmic_control), + TWL4030_REG_ANAMICR, 4, 0, &twl4030_dapm_analogrmic_control), - SND_SOC_DAPM_PGA("Analog Left Amplifier", - TWL4030_REG_ANAMICL, 4, 0, NULL, 0), - SND_SOC_DAPM_PGA("Analog Right Amplifier", - TWL4030_REG_ANAMICR, 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("ADC Physical Left", + TWL4030_REG_AVADC_CTL, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA("ADC Physical Right", + TWL4030_REG_AVADC_CTL, 1, 0, NULL, 0), SND_SOC_DAPM_PGA("Digimic0 Enable", TWL4030_REG_ADCMICSEL, 1, 0, NULL, 0), @@ -885,23 +885,23 @@ static const struct snd_soc_dapm_route intercon[] = { {"Analog Right Capture Route", "Sub mic", "SUBMIC"}, {"Analog Right Capture Route", "AUXR", "AUXR"}, - {"Analog Left Amplifier", NULL, "Analog Left Capture Route"}, - {"Analog Right Amplifier", NULL, "Analog Right Capture Route"}, + {"ADC Physical Left", NULL, "Analog Left Capture Route"}, + {"ADC Physical Right", NULL, "Analog Right Capture Route"}, {"Digimic0 Enable", NULL, "DIGIMIC0"}, {"Digimic1 Enable", NULL, "DIGIMIC1"}, /* TX1 Left capture path */ - {"TX1 Capture Route", "Analog", "Analog Left Amplifier"}, + {"TX1 Capture Route", "Analog", "ADC Physical Left"}, {"TX1 Capture Route", "Digimic0", "Digimic0 Enable"}, /* TX1 Right capture path */ - {"TX1 Capture Route", "Analog", "Analog Right Amplifier"}, + {"TX1 Capture Route", "Analog", "ADC Physical Right"}, {"TX1 Capture Route", "Digimic0", "Digimic0 Enable"}, /* TX2 Left capture path */ - {"TX2 Capture Route", "Analog", "Analog Left Amplifier"}, + {"TX2 Capture Route", "Analog", "ADC Physical Left"}, {"TX2 Capture Route", "Digimic1", "Digimic1 Enable"}, /* TX2 Right capture path */ - {"TX2 Capture Route", "Analog", "Analog Right Amplifier"}, + {"TX2 Capture Route", "Analog", "ADC Physical Right"}, {"TX2 Capture Route", "Digimic1", "Digimic1 Enable"}, {"ADC Virtual Left1", NULL, "TX1 Capture Route"}, -- cgit v1.2.3-70-g09d2 From 006f367e38fb45e2f161c0f500c74449ae63e866 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 27 Jan 2009 11:29:43 +0200 Subject: ASoC: TWL4030: Move the twl4030_power_up and _power_down function Move the twl4030_power_up and twl4030_power_down function earlier to facilitate the analog bypass implementation. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 85 +++++++++++++++++++++++----------------------- 1 file changed, 42 insertions(+), 43 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 8fe059e3e9a..f985bef40a3 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -184,6 +184,48 @@ static void twl4030_init_chip(struct snd_soc_codec *codec) } +static void twl4030_power_up(struct snd_soc_codec *codec) +{ + u8 anamicl, regmisc1, byte; + int i = 0; + + /* set CODECPDZ to turn on codec */ + twl4030_codec_enable(codec, 1); + + /* initiate offset cancellation */ + anamicl = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL); + twl4030_write(codec, TWL4030_REG_ANAMICL, + anamicl | TWL4030_CNCL_OFFSET_START); + + /* wait for offset cancellation to complete */ + do { + /* this takes a little while, so don't slam i2c */ + udelay(2000); + twl4030_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &byte, + TWL4030_REG_ANAMICL); + } while ((i++ < 100) && + ((byte & TWL4030_CNCL_OFFSET_START) == + TWL4030_CNCL_OFFSET_START)); + + /* Make sure that the reg_cache has the same value as the HW */ + twl4030_write_reg_cache(codec, TWL4030_REG_ANAMICL, byte); + + /* anti-pop when changing analog gain */ + regmisc1 = twl4030_read_reg_cache(codec, TWL4030_REG_MISC_SET_1); + twl4030_write(codec, TWL4030_REG_MISC_SET_1, + regmisc1 | TWL4030_SMOOTH_ANAVOL_EN); + + /* toggle CODECPDZ as per TRM */ + twl4030_codec_enable(codec, 0); + twl4030_codec_enable(codec, 1); +} + +static void twl4030_power_down(struct snd_soc_codec *codec) +{ + /* power down */ + twl4030_codec_enable(codec, 0); +} + /* Earpiece */ static const char *twl4030_earpiece_texts[] = {"Off", "DACL1", "DACL2", "DACR1"}; @@ -922,49 +964,6 @@ static int twl4030_add_widgets(struct snd_soc_codec *codec) return 0; } -static void twl4030_power_up(struct snd_soc_codec *codec) -{ - u8 anamicl, regmisc1, byte; - int i = 0; - - /* set CODECPDZ to turn on codec */ - twl4030_codec_enable(codec, 1); - - /* initiate offset cancellation */ - anamicl = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL); - twl4030_write(codec, TWL4030_REG_ANAMICL, - anamicl | TWL4030_CNCL_OFFSET_START); - - - /* wait for offset cancellation to complete */ - do { - /* this takes a little while, so don't slam i2c */ - udelay(2000); - twl4030_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &byte, - TWL4030_REG_ANAMICL); - } while ((i++ < 100) && - ((byte & TWL4030_CNCL_OFFSET_START) == - TWL4030_CNCL_OFFSET_START)); - - /* Make sure that the reg_cache has the same value as the HW */ - twl4030_write_reg_cache(codec, TWL4030_REG_ANAMICL, byte); - - /* anti-pop when changing analog gain */ - regmisc1 = twl4030_read_reg_cache(codec, TWL4030_REG_MISC_SET_1); - twl4030_write(codec, TWL4030_REG_MISC_SET_1, - regmisc1 | TWL4030_SMOOTH_ANAVOL_EN); - - /* toggle CODECPDZ as per TRM */ - twl4030_codec_enable(codec, 0); - twl4030_codec_enable(codec, 1); -} - -static void twl4030_power_down(struct snd_soc_codec *codec) -{ - /* power down */ - twl4030_codec_enable(codec, 0); -} - static int twl4030_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { -- cgit v1.2.3-70-g09d2 From f6c6383502751ceb6f2f3579ad22578ca44f91f5 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Sat, 24 Jan 2009 13:35:28 +0100 Subject: ALSA: Turtle Beach Multisound Classic/Pinnacle driver This is driver for Turtle Beach Multisound cards: Classic, Fiji and Pinnacle. Tested pcm playback and recording and MIDI playback on Multisound Pinnacle. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/Kconfig | 31 + sound/isa/msnd/Makefile | 9 + sound/isa/msnd/msnd.c | 702 +++++++++++++++++++ sound/isa/msnd/msnd.h | 308 +++++++++ sound/isa/msnd/msnd_classic.c | 3 + sound/isa/msnd/msnd_classic.h | 129 ++++ sound/isa/msnd/msnd_midi.c | 180 +++++ sound/isa/msnd/msnd_pinnacle.c | 1235 ++++++++++++++++++++++++++++++++++ sound/isa/msnd/msnd_pinnacle.h | 181 +++++ sound/isa/msnd/msnd_pinnacle_mixer.c | 343 ++++++++++ 10 files changed, 3121 insertions(+) create mode 100644 sound/isa/msnd/Makefile create mode 100644 sound/isa/msnd/msnd.c create mode 100644 sound/isa/msnd/msnd.h create mode 100644 sound/isa/msnd/msnd_classic.c create mode 100644 sound/isa/msnd/msnd_classic.h create mode 100644 sound/isa/msnd/msnd_midi.c create mode 100644 sound/isa/msnd/msnd_pinnacle.c create mode 100644 sound/isa/msnd/msnd_pinnacle.h create mode 100644 sound/isa/msnd/msnd_pinnacle_mixer.c diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index ce0aa044e27..a74725950b0 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -411,5 +411,36 @@ config SND_WAVEFRONT_FIRMWARE_IN_KERNEL you need to install the firmware files from the alsa-firmware package. +config SND_MSND_PINNACLE + tristate "Turtle Beach MultiSound Pinnacle/Fiji driver" + depends on X86 && EXPERIMENTAL + select FW_LOADER + select SND_MPU401_UART + select SND_PCM + help + Say Y to include support for Turtle Beach MultiSound Pinnacle/ + Fiji soundcards. + + To compile this driver as a module, choose M here: the module + will be called snd-msnd-pinnacle. + +config SND_MSND_CLASSIC + tristate "Support for Turtle Beach MultiSound Classic, Tahiti, Monterey" + depends on X86 && EXPERIMENTAL + select FW_LOADER + select SND_MPU401_UART + select SND_PCM + help + Say M here if you have a Turtle Beach MultiSound Classic, Tahiti or + Monterey (not for the Pinnacle or Fiji). + + See for important information + about this driver. Note that it has been discontinued, but the + Voyetra Turtle Beach knowledge base entry for it is still available + at . + + To compile this driver as a module, choose M here: the module + will be called snd-msnd-classic. + endif # SND_ISA diff --git a/sound/isa/msnd/Makefile b/sound/isa/msnd/Makefile new file mode 100644 index 00000000000..2171c0aa2f6 --- /dev/null +++ b/sound/isa/msnd/Makefile @@ -0,0 +1,9 @@ + +snd-msnd-lib-objs := msnd.o msnd_midi.o msnd_pinnacle_mixer.o +snd-msnd-pinnacle-objs := msnd_pinnacle.o +snd-msnd-classic-objs := msnd_classic.o + +# Toplevel Module Dependency +obj-$(CONFIG_SND_MSND_PINNACLE) += snd-msnd-pinnacle.o snd-msnd-lib.o +obj-$(CONFIG_SND_MSND_CLASSIC) += snd-msnd-classic.o snd-msnd-lib.o + diff --git a/sound/isa/msnd/msnd.c b/sound/isa/msnd/msnd.c new file mode 100644 index 00000000000..264e08212c6 --- /dev/null +++ b/sound/isa/msnd/msnd.c @@ -0,0 +1,702 @@ +/********************************************************************* + * + * 2002/06/30 Karsten Wiese: + * removed kernel-version dependencies. + * ripped from linux kernel 2.4.18 (OSS Implementation) by me. + * In the OSS Version, this file is compiled to a separate MODULE, + * that is used by the pinnacle and the classic driver. + * since there is no classic driver for alsa yet (i dont have a classic + * & writing one blindfold is difficult) this file's object is statically + * linked into the pinnacle-driver-module for now. look for the string + * "uncomment this to make this a module again" + * to do guess what. + * + * the following is a copy of the 2.4.18 OSS FREE file-heading comment: + * + * msnd.c - Driver Base + * + * Turtle Beach MultiSound Sound Card Driver for Linux + * + * Copyright (C) 1998 Andrew Veliath + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + * + ********************************************************************/ + +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include + +#include "msnd.h" + +#define LOGNAME "msnd" + + +void snd_msnd_init_queue(void *base, int start, int size) +{ + writew(PCTODSP_BASED(start), base + JQS_wStart); + writew(PCTODSP_OFFSET(size) - 1, base + JQS_wSize); + writew(0, base + JQS_wHead); + writew(0, base + JQS_wTail); +} +EXPORT_SYMBOL(snd_msnd_init_queue); + +static int snd_msnd_wait_TXDE(struct snd_msnd *dev) +{ + unsigned int io = dev->io; + int timeout = 1000; + + while (timeout-- > 0) + if (inb(io + HP_ISR) & HPISR_TXDE) + return 0; + + return -EIO; +} + +static int snd_msnd_wait_HC0(struct snd_msnd *dev) +{ + unsigned int io = dev->io; + int timeout = 1000; + + while (timeout-- > 0) + if (!(inb(io + HP_CVR) & HPCVR_HC)) + return 0; + + return -EIO; +} + +int snd_msnd_send_dsp_cmd(struct snd_msnd *dev, u8 cmd) +{ + unsigned long flags; + + spin_lock_irqsave(&dev->lock, flags); + if (snd_msnd_wait_HC0(dev) == 0) { + outb(cmd, dev->io + HP_CVR); + spin_unlock_irqrestore(&dev->lock, flags); + return 0; + } + spin_unlock_irqrestore(&dev->lock, flags); + + snd_printd(KERN_ERR LOGNAME ": Send DSP command timeout\n"); + + return -EIO; +} +EXPORT_SYMBOL(snd_msnd_send_dsp_cmd); + +int snd_msnd_send_word(struct snd_msnd *dev, unsigned char high, + unsigned char mid, unsigned char low) +{ + unsigned int io = dev->io; + + if (snd_msnd_wait_TXDE(dev) == 0) { + outb(high, io + HP_TXH); + outb(mid, io + HP_TXM); + outb(low, io + HP_TXL); + return 0; + } + + snd_printd(KERN_ERR LOGNAME ": Send host word timeout\n"); + + return -EIO; +} +EXPORT_SYMBOL(snd_msnd_send_word); + +int snd_msnd_upload_host(struct snd_msnd *dev, const u8 *bin, int len) +{ + int i; + + if (len % 3 != 0) { + snd_printk(KERN_ERR LOGNAME + ": Upload host data not multiple of 3!\n"); + return -EINVAL; + } + + for (i = 0; i < len; i += 3) + if (snd_msnd_send_word(dev, bin[i], bin[i + 1], bin[i + 2])) + return -EIO; + + inb(dev->io + HP_RXL); + inb(dev->io + HP_CVR); + + return 0; +} +EXPORT_SYMBOL(snd_msnd_upload_host); + +int snd_msnd_enable_irq(struct snd_msnd *dev) +{ + unsigned long flags; + + if (dev->irq_ref++) + return 0; + + snd_printdd(LOGNAME ": Enabling IRQ\n"); + + spin_lock_irqsave(&dev->lock, flags); + if (snd_msnd_wait_TXDE(dev) == 0) { + outb(inb(dev->io + HP_ICR) | HPICR_TREQ, dev->io + HP_ICR); + if (dev->type == msndClassic) + outb(dev->irqid, dev->io + HP_IRQM); + + outb(inb(dev->io + HP_ICR) & ~HPICR_TREQ, dev->io + HP_ICR); + outb(inb(dev->io + HP_ICR) | HPICR_RREQ, dev->io + HP_ICR); + enable_irq(dev->irq); + snd_msnd_init_queue(dev->DSPQ, dev->dspq_data_buff, + dev->dspq_buff_size); + spin_unlock_irqrestore(&dev->lock, flags); + return 0; + } + spin_unlock_irqrestore(&dev->lock, flags); + + snd_printd(KERN_ERR LOGNAME ": Enable IRQ failed\n"); + + return -EIO; +} +EXPORT_SYMBOL(snd_msnd_enable_irq); + +int snd_msnd_disable_irq(struct snd_msnd *dev) +{ + unsigned long flags; + + if (--dev->irq_ref > 0) + return 0; + + if (dev->irq_ref < 0) + snd_printd(KERN_WARNING LOGNAME ": IRQ ref count is %d\n", + dev->irq_ref); + + snd_printdd(LOGNAME ": Disabling IRQ\n"); + + spin_lock_irqsave(&dev->lock, flags); + if (snd_msnd_wait_TXDE(dev) == 0) { + outb(inb(dev->io + HP_ICR) & ~HPICR_RREQ, dev->io + HP_ICR); + if (dev->type == msndClassic) + outb(HPIRQ_NONE, dev->io + HP_IRQM); + disable_irq(dev->irq); + spin_unlock_irqrestore(&dev->lock, flags); + return 0; + } + spin_unlock_irqrestore(&dev->lock, flags); + + snd_printd(KERN_ERR LOGNAME ": Disable IRQ failed\n"); + + return -EIO; +} +EXPORT_SYMBOL(snd_msnd_disable_irq); + +static inline long get_play_delay_jiffies(struct snd_msnd *chip, long size) +{ + long tmp = (size * HZ * chip->play_sample_size) / 8; + return tmp / (chip->play_sample_rate * chip->play_channels); +} + +static void snd_msnd_dsp_write_flush(struct snd_msnd *chip) +{ + if (!(chip->mode & FMODE_WRITE) || !test_bit(F_WRITING, &chip->flags)) + return; + set_bit(F_WRITEFLUSH, &chip->flags); +/* interruptible_sleep_on_timeout( + &chip->writeflush, + get_play_delay_jiffies(&chip, chip->DAPF.len));*/ + clear_bit(F_WRITEFLUSH, &chip->flags); + if (!signal_pending(current)) + schedule_timeout_interruptible( + get_play_delay_jiffies(chip, chip->play_period_bytes)); + clear_bit(F_WRITING, &chip->flags); +} + +void snd_msnd_dsp_halt(struct snd_msnd *chip, struct file *file) +{ + if ((file ? file->f_mode : chip->mode) & FMODE_READ) { + clear_bit(F_READING, &chip->flags); + snd_msnd_send_dsp_cmd(chip, HDEX_RECORD_STOP); + snd_msnd_disable_irq(chip); + if (file) { + snd_printd(KERN_INFO LOGNAME + ": Stopping read for %p\n", file); + chip->mode &= ~FMODE_READ; + } + clear_bit(F_AUDIO_READ_INUSE, &chip->flags); + } + if ((file ? file->f_mode : chip->mode) & FMODE_WRITE) { + if (test_bit(F_WRITING, &chip->flags)) { + snd_msnd_dsp_write_flush(chip); + snd_msnd_send_dsp_cmd(chip, HDEX_PLAY_STOP); + } + snd_msnd_disable_irq(chip); + if (file) { + snd_printd(KERN_INFO + LOGNAME ": Stopping write for %p\n", file); + chip->mode &= ~FMODE_WRITE; + } + clear_bit(F_AUDIO_WRITE_INUSE, &chip->flags); + } +} +EXPORT_SYMBOL(snd_msnd_dsp_halt); + + +int snd_msnd_DARQ(struct snd_msnd *chip, int bank) +{ + int /*size, n,*/ timeout = 3; + u16 wTmp; + /* void *DAQD; */ + + /* Increment the tail and check for queue wrap */ + wTmp = readw(chip->DARQ + JQS_wTail) + PCTODSP_OFFSET(DAQDS__size); + if (wTmp > readw(chip->DARQ + JQS_wSize)) + wTmp = 0; + while (wTmp == readw(chip->DARQ + JQS_wHead) && timeout--) + udelay(1); + + if (chip->capturePeriods == 2) { + void *pDAQ = chip->mappedbase + DARQ_DATA_BUFF + + bank * DAQDS__size + DAQDS_wStart; + unsigned short offset = 0x3000 + chip->capturePeriodBytes; + + if (readw(pDAQ) != PCTODSP_BASED(0x3000)) + offset = 0x3000; + writew(PCTODSP_BASED(offset), pDAQ); + } + + writew(wTmp, chip->DARQ + JQS_wTail); + +#if 0 + /* Get our digital audio queue struct */ + DAQD = bank * DAQDS__size + chip->mappedbase + DARQ_DATA_BUFF; + + /* Get length of data */ + size = readw(DAQD + DAQDS_wSize); + + /* Read data from the head (unprotected bank 1 access okay + since this is only called inside an interrupt) */ + outb(HPBLKSEL_1, chip->io + HP_BLKS); + n = msnd_fifo_write(&chip->DARF, + (char *)(chip->base + bank * DAR_BUFF_SIZE), + size, 0); + if (n <= 0) { + outb(HPBLKSEL_0, chip->io + HP_BLKS); + return n; + } + outb(HPBLKSEL_0, chip->io + HP_BLKS); +#endif + + return 1; +} +EXPORT_SYMBOL(snd_msnd_DARQ); + +int snd_msnd_DAPQ(struct snd_msnd *chip, int start) +{ + u16 DAPQ_tail; + int protect = start, nbanks = 0; + void *DAQD; + static int play_banks_submitted; + /* unsigned long flags; + spin_lock_irqsave(&chip->lock, flags); not necessary */ + + DAPQ_tail = readw(chip->DAPQ + JQS_wTail); + while (DAPQ_tail != readw(chip->DAPQ + JQS_wHead) || start) { + int bank_num = DAPQ_tail / PCTODSP_OFFSET(DAQDS__size); + + if (start) { + start = 0; + play_banks_submitted = 0; + } + + /* Get our digital audio queue struct */ + DAQD = bank_num * DAQDS__size + chip->mappedbase + + DAPQ_DATA_BUFF; + + /* Write size of this bank */ + writew(chip->play_period_bytes, DAQD + DAQDS_wSize); + if (play_banks_submitted < 3) + ++play_banks_submitted; + else if (chip->playPeriods == 2) { + unsigned short offset = chip->play_period_bytes; + + if (readw(DAQD + DAQDS_wStart) != PCTODSP_BASED(0x0)) + offset = 0; + + writew(PCTODSP_BASED(offset), DAQD + DAQDS_wStart); + } + ++nbanks; + + /* Then advance the tail */ + /* + if (protect) + snd_printd(KERN_INFO "B %X %lX\n", + bank_num, xtime.tv_usec); + */ + + DAPQ_tail = (++bank_num % 3) * PCTODSP_OFFSET(DAQDS__size); + writew(DAPQ_tail, chip->DAPQ + JQS_wTail); + /* Tell the DSP to play the bank */ + snd_msnd_send_dsp_cmd(chip, HDEX_PLAY_START); + if (protect) + if (2 == bank_num) + break; + } + /* + if (protect) + snd_printd(KERN_INFO "%lX\n", xtime.tv_usec); + */ + /* spin_unlock_irqrestore(&chip->lock, flags); not necessary */ + return nbanks; +} +EXPORT_SYMBOL(snd_msnd_DAPQ); + +static void snd_msnd_play_reset_queue(struct snd_msnd *chip, + unsigned int pcm_periods, + unsigned int pcm_count) +{ + int n; + void *pDAQ = chip->mappedbase + DAPQ_DATA_BUFF; + + chip->last_playbank = -1; + chip->playLimit = pcm_count * (pcm_periods - 1); + chip->playPeriods = pcm_periods; + writew(PCTODSP_OFFSET(0 * DAQDS__size), chip->DAPQ + JQS_wHead); + writew(PCTODSP_OFFSET(0 * DAQDS__size), chip->DAPQ + JQS_wTail); + + chip->play_period_bytes = pcm_count; + + for (n = 0; n < pcm_periods; ++n, pDAQ += DAQDS__size) { + writew(PCTODSP_BASED((u32)(pcm_count * n)), + pDAQ + DAQDS_wStart); + writew(0, pDAQ + DAQDS_wSize); + writew(1, pDAQ + DAQDS_wFormat); + writew(chip->play_sample_size, pDAQ + DAQDS_wSampleSize); + writew(chip->play_channels, pDAQ + DAQDS_wChannels); + writew(chip->play_sample_rate, pDAQ + DAQDS_wSampleRate); + writew(HIMT_PLAY_DONE * 0x100 + n, pDAQ + DAQDS_wIntMsg); + writew(n, pDAQ + DAQDS_wFlags); + } +} + +static void snd_msnd_capture_reset_queue(struct snd_msnd *chip, + unsigned int pcm_periods, + unsigned int pcm_count) +{ + int n; + void *pDAQ; + /* unsigned long flags; */ + + /* snd_msnd_init_queue(chip->DARQ, DARQ_DATA_BUFF, DARQ_BUFF_SIZE); */ + + chip->last_recbank = 2; + chip->captureLimit = pcm_count * (pcm_periods - 1); + chip->capturePeriods = pcm_periods; + writew(PCTODSP_OFFSET(0 * DAQDS__size), chip->DARQ + JQS_wHead); + writew(PCTODSP_OFFSET(chip->last_recbank * DAQDS__size), + chip->DARQ + JQS_wTail); + +#if 0 /* Critical section: bank 1 access. this is how the OSS driver does it:*/ + spin_lock_irqsave(&chip->lock, flags); + outb(HPBLKSEL_1, chip->io + HP_BLKS); + memset_io(chip->mappedbase, 0, DAR_BUFF_SIZE * 3); + outb(HPBLKSEL_0, chip->io + HP_BLKS); + spin_unlock_irqrestore(&chip->lock, flags); +#endif + + chip->capturePeriodBytes = pcm_count; + snd_printdd("snd_msnd_capture_reset_queue() %i\n", pcm_count); + + pDAQ = chip->mappedbase + DARQ_DATA_BUFF; + + for (n = 0; n < pcm_periods; ++n, pDAQ += DAQDS__size) { + u32 tmp = pcm_count * n; + + writew(PCTODSP_BASED(tmp + 0x3000), pDAQ + DAQDS_wStart); + writew(pcm_count, pDAQ + DAQDS_wSize); + writew(1, pDAQ + DAQDS_wFormat); + writew(chip->capture_sample_size, pDAQ + DAQDS_wSampleSize); + writew(chip->capture_channels, pDAQ + DAQDS_wChannels); + writew(chip->capture_sample_rate, pDAQ + DAQDS_wSampleRate); + writew(HIMT_RECORD_DONE * 0x100 + n, pDAQ + DAQDS_wIntMsg); + writew(n, pDAQ + DAQDS_wFlags); + } +} + +static struct snd_pcm_hardware snd_msnd_playback = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP_VALID, + .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE, + .rates = SNDRV_PCM_RATE_8000_48000, + .rate_min = 8000, + .rate_max = 48000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = 0x3000, + .period_bytes_min = 0x40, + .period_bytes_max = 0x1800, + .periods_min = 2, + .periods_max = 3, + .fifo_size = 0, +}; + +static struct snd_pcm_hardware snd_msnd_capture = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP_VALID, + .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE, + .rates = SNDRV_PCM_RATE_8000_48000, + .rate_min = 8000, + .rate_max = 48000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = 0x3000, + .period_bytes_min = 0x40, + .period_bytes_max = 0x1800, + .periods_min = 2, + .periods_max = 3, + .fifo_size = 0, +}; + + +static int snd_msnd_playback_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + + set_bit(F_AUDIO_WRITE_INUSE, &chip->flags); + clear_bit(F_WRITING, &chip->flags); + snd_msnd_enable_irq(chip); + + runtime->dma_area = chip->mappedbase; + runtime->dma_bytes = 0x3000; + + chip->playback_substream = substream; + runtime->hw = snd_msnd_playback; + return 0; +} + +static int snd_msnd_playback_close(struct snd_pcm_substream *substream) +{ + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + + snd_msnd_disable_irq(chip); + clear_bit(F_AUDIO_WRITE_INUSE, &chip->flags); + return 0; +} + + +static int snd_msnd_playback_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + int i; + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + void *pDAQ = chip->mappedbase + DAPQ_DATA_BUFF; + + chip->play_sample_size = snd_pcm_format_width(params_format(params)); + chip->play_channels = params_channels(params); + chip->play_sample_rate = params_rate(params); + + for (i = 0; i < 3; ++i, pDAQ += DAQDS__size) { + writew(chip->play_sample_size, pDAQ + DAQDS_wSampleSize); + writew(chip->play_channels, pDAQ + DAQDS_wChannels); + writew(chip->play_sample_rate, pDAQ + DAQDS_wSampleRate); + } + /* dont do this here: + * snd_msnd_calibrate_adc(chip->play_sample_rate); + */ + + return 0; +} + +static int snd_msnd_playback_prepare(struct snd_pcm_substream *substream) +{ + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + unsigned int pcm_size = snd_pcm_lib_buffer_bytes(substream); + unsigned int pcm_count = snd_pcm_lib_period_bytes(substream); + unsigned int pcm_periods = pcm_size / pcm_count; + + snd_msnd_play_reset_queue(chip, pcm_periods, pcm_count); + chip->playDMAPos = 0; + return 0; +} + +static int snd_msnd_playback_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + int result = 0; + + if (cmd == SNDRV_PCM_TRIGGER_START) { + snd_printdd("snd_msnd_playback_trigger(START)\n"); + chip->banksPlayed = 0; + set_bit(F_WRITING, &chip->flags); + snd_msnd_DAPQ(chip, 1); + } else if (cmd == SNDRV_PCM_TRIGGER_STOP) { + snd_printdd("snd_msnd_playback_trigger(STop)\n"); + /* interrupt diagnostic, comment this out later */ + clear_bit(F_WRITING, &chip->flags); + snd_msnd_send_dsp_cmd(chip, HDEX_PLAY_STOP); + } else { + snd_printd(KERN_ERR "snd_msnd_playback_trigger(?????)\n"); + result = -EINVAL; + } + + snd_printdd("snd_msnd_playback_trigger() ENDE\n"); + return result; +} + +static snd_pcm_uframes_t +snd_msnd_playback_pointer(struct snd_pcm_substream *substream) +{ + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + + return bytes_to_frames(substream->runtime, chip->playDMAPos); +} + + +static struct snd_pcm_ops snd_msnd_playback_ops = { + .open = snd_msnd_playback_open, + .close = snd_msnd_playback_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_msnd_playback_hw_params, + .prepare = snd_msnd_playback_prepare, + .trigger = snd_msnd_playback_trigger, + .pointer = snd_msnd_playback_pointer, +}; + +static int snd_msnd_capture_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + + set_bit(F_AUDIO_READ_INUSE, &chip->flags); + snd_msnd_enable_irq(chip); + runtime->dma_area = chip->mappedbase + 0x3000; + runtime->dma_bytes = 0x3000; + memset(runtime->dma_area, 0, runtime->dma_bytes); + chip->capture_substream = substream; + runtime->hw = snd_msnd_capture; + return 0; +} + +static int snd_msnd_capture_close(struct snd_pcm_substream *substream) +{ + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + + snd_msnd_disable_irq(chip); + clear_bit(F_AUDIO_READ_INUSE, &chip->flags); + return 0; +} + +static int snd_msnd_capture_prepare(struct snd_pcm_substream *substream) +{ + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + unsigned int pcm_size = snd_pcm_lib_buffer_bytes(substream); + unsigned int pcm_count = snd_pcm_lib_period_bytes(substream); + unsigned int pcm_periods = pcm_size / pcm_count; + + snd_msnd_capture_reset_queue(chip, pcm_periods, pcm_count); + chip->captureDMAPos = 0; + return 0; +} + +static int snd_msnd_capture_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + + if (cmd == SNDRV_PCM_TRIGGER_START) { + chip->last_recbank = -1; + set_bit(F_READING, &chip->flags); + if (snd_msnd_send_dsp_cmd(chip, HDEX_RECORD_START) == 0) + return 0; + + clear_bit(F_READING, &chip->flags); + } else if (cmd == SNDRV_PCM_TRIGGER_STOP) { + clear_bit(F_READING, &chip->flags); + snd_msnd_send_dsp_cmd(chip, HDEX_RECORD_STOP); + return 0; + } + return -EINVAL; +} + + +static snd_pcm_uframes_t +snd_msnd_capture_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + + return bytes_to_frames(runtime, chip->captureDMAPos); +} + + +static int snd_msnd_capture_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + int i; + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + void *pDAQ = chip->mappedbase + DARQ_DATA_BUFF; + + chip->capture_sample_size = snd_pcm_format_width(params_format(params)); + chip->capture_channels = params_channels(params); + chip->capture_sample_rate = params_rate(params); + + for (i = 0; i < 3; ++i, pDAQ += DAQDS__size) { + writew(chip->capture_sample_size, pDAQ + DAQDS_wSampleSize); + writew(chip->capture_channels, pDAQ + DAQDS_wChannels); + writew(chip->capture_sample_rate, pDAQ + DAQDS_wSampleRate); + } + return 0; +} + + +static struct snd_pcm_ops snd_msnd_capture_ops = { + .open = snd_msnd_capture_open, + .close = snd_msnd_capture_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_msnd_capture_hw_params, + .prepare = snd_msnd_capture_prepare, + .trigger = snd_msnd_capture_trigger, + .pointer = snd_msnd_capture_pointer, +}; + + +int snd_msnd_pcm(struct snd_card *card, int device, + struct snd_pcm **rpcm) +{ + struct snd_msnd *chip = card->private_data; + struct snd_pcm *pcm; + int err; + + err = snd_pcm_new(card, "MSNDPINNACLE", device, 1, 1, &pcm); + if (err < 0) + return err; + + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_msnd_playback_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_msnd_capture_ops); + + pcm->private_data = chip; + strcpy(pcm->name, "Hurricane"); + + + if (rpcm) + *rpcm = pcm; + return 0; +} +EXPORT_SYMBOL(snd_msnd_pcm); + diff --git a/sound/isa/msnd/msnd.h b/sound/isa/msnd/msnd.h new file mode 100644 index 00000000000..3773e242b58 --- /dev/null +++ b/sound/isa/msnd/msnd.h @@ -0,0 +1,308 @@ +/********************************************************************* + * + * msnd.h + * + * Turtle Beach MultiSound Sound Card Driver for Linux + * + * Some parts of this header file were derived from the Turtle Beach + * MultiSound Driver Development Kit. + * + * Copyright (C) 1998 Andrew Veliath + * Copyright (C) 1993 Turtle Beach Systems, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + * + ********************************************************************/ +#ifndef __MSND_H +#define __MSND_H + +#define DEFSAMPLERATE 44100 +#define DEFSAMPLESIZE SNDRV_PCM_FORMAT_S16 +#define DEFCHANNELS 1 + +#define SRAM_BANK_SIZE 0x8000 +#define SRAM_CNTL_START 0x7F00 +#define SMA_STRUCT_START 0x7F40 + +#define DSP_BASE_ADDR 0x4000 +#define DSP_BANK_BASE 0x4000 + +#define AGND 0x01 +#define SIGNAL 0x02 + +#define EXT_DSP_BIT_DCAL 0x0001 +#define EXT_DSP_BIT_MIDI_CON 0x0002 + +#define BUFFSIZE 0x8000 +#define HOSTQ_SIZE 0x40 + +#define DAP_BUFF_SIZE 0x2400 + +#define DAPQ_STRUCT_SIZE 0x10 +#define DARQ_STRUCT_SIZE 0x10 +#define DAPQ_BUFF_SIZE (3 * 0x10) +#define DARQ_BUFF_SIZE (3 * 0x10) +#define MODQ_BUFF_SIZE 0x400 + +#define DAPQ_DATA_BUFF 0x6C00 +#define DARQ_DATA_BUFF 0x6C30 +#define MODQ_DATA_BUFF 0x6C60 +#define MIDQ_DATA_BUFF 0x7060 + +#define DAPQ_OFFSET SRAM_CNTL_START +#define DARQ_OFFSET (SRAM_CNTL_START + 0x08) +#define MODQ_OFFSET (SRAM_CNTL_START + 0x10) +#define MIDQ_OFFSET (SRAM_CNTL_START + 0x18) +#define DSPQ_OFFSET (SRAM_CNTL_START + 0x20) + +#define HP_ICR 0x00 +#define HP_CVR 0x01 +#define HP_ISR 0x02 +#define HP_IVR 0x03 +#define HP_NU 0x04 +#define HP_INFO 0x04 +#define HP_TXH 0x05 +#define HP_RXH 0x05 +#define HP_TXM 0x06 +#define HP_RXM 0x06 +#define HP_TXL 0x07 +#define HP_RXL 0x07 + +#define HP_ICR_DEF 0x00 +#define HP_CVR_DEF 0x12 +#define HP_ISR_DEF 0x06 +#define HP_IVR_DEF 0x0f +#define HP_NU_DEF 0x00 + +#define HP_IRQM 0x09 + +#define HPR_BLRC 0x08 +#define HPR_SPR1 0x09 +#define HPR_SPR2 0x0A +#define HPR_TCL0 0x0B +#define HPR_TCL1 0x0C +#define HPR_TCL2 0x0D +#define HPR_TCL3 0x0E +#define HPR_TCL4 0x0F + +#define HPICR_INIT 0x80 +#define HPICR_HM1 0x40 +#define HPICR_HM0 0x20 +#define HPICR_HF1 0x10 +#define HPICR_HF0 0x08 +#define HPICR_TREQ 0x02 +#define HPICR_RREQ 0x01 + +#define HPCVR_HC 0x80 + +#define HPISR_HREQ 0x80 +#define HPISR_DMA 0x40 +#define HPISR_HF3 0x10 +#define HPISR_HF2 0x08 +#define HPISR_TRDY 0x04 +#define HPISR_TXDE 0x02 +#define HPISR_RXDF 0x01 + +#define HPIO_290 0 +#define HPIO_260 1 +#define HPIO_250 2 +#define HPIO_240 3 +#define HPIO_230 4 +#define HPIO_220 5 +#define HPIO_210 6 +#define HPIO_3E0 7 + +#define HPMEM_NONE 0 +#define HPMEM_B000 1 +#define HPMEM_C800 2 +#define HPMEM_D000 3 +#define HPMEM_D400 4 +#define HPMEM_D800 5 +#define HPMEM_E000 6 +#define HPMEM_E800 7 + +#define HPIRQ_NONE 0 +#define HPIRQ_5 1 +#define HPIRQ_7 2 +#define HPIRQ_9 3 +#define HPIRQ_10 4 +#define HPIRQ_11 5 +#define HPIRQ_12 6 +#define HPIRQ_15 7 + +#define HIMT_PLAY_DONE 0x00 +#define HIMT_RECORD_DONE 0x01 +#define HIMT_MIDI_EOS 0x02 +#define HIMT_MIDI_OUT 0x03 + +#define HIMT_MIDI_IN_UCHAR 0x0E +#define HIMT_DSP 0x0F + +#define HDEX_BASE 0x92 +#define HDEX_PLAY_START (0 + HDEX_BASE) +#define HDEX_PLAY_STOP (1 + HDEX_BASE) +#define HDEX_PLAY_PAUSE (2 + HDEX_BASE) +#define HDEX_PLAY_RESUME (3 + HDEX_BASE) +#define HDEX_RECORD_START (4 + HDEX_BASE) +#define HDEX_RECORD_STOP (5 + HDEX_BASE) +#define HDEX_MIDI_IN_START (6 + HDEX_BASE) +#define HDEX_MIDI_IN_STOP (7 + HDEX_BASE) +#define HDEX_MIDI_OUT_START (8 + HDEX_BASE) +#define HDEX_MIDI_OUT_STOP (9 + HDEX_BASE) +#define HDEX_AUX_REQ (10 + HDEX_BASE) + +#define HDEXAR_CLEAR_PEAKS 1 +#define HDEXAR_IN_SET_POTS 2 +#define HDEXAR_AUX_SET_POTS 3 +#define HDEXAR_CAL_A_TO_D 4 +#define HDEXAR_RD_EXT_DSP_BITS 5 + +/* Pinnacle only HDEXAR defs */ +#define HDEXAR_SET_ANA_IN 0 +#define HDEXAR_SET_SYNTH_IN 4 +#define HDEXAR_READ_DAT_IN 5 +#define HDEXAR_MIC_SET_POTS 6 +#define HDEXAR_SET_DAT_IN 7 + +#define HDEXAR_SET_SYNTH_48 8 +#define HDEXAR_SET_SYNTH_44 9 + +#define HIWORD(l) ((u16)((((u32)(l)) >> 16) & 0xFFFF)) +#define LOWORD(l) ((u16)(u32)(l)) +#define HIBYTE(w) ((u8)(((u16)(w) >> 8) & 0xFF)) +#define LOBYTE(w) ((u8)(w)) +#define MAKELONG(low, hi) ((long)(((u16)(low))|(((u32)((u16)(hi)))<<16))) +#define MAKEWORD(low, hi) ((u16)(((u8)(low))|(((u16)((u8)(hi)))<<8))) + +#define PCTODSP_OFFSET(w) (u16)((w)/2) +#define PCTODSP_BASED(w) (u16)(((w)/2) + DSP_BASE_ADDR) +#define DSPTOPC_BASED(w) (((w) - DSP_BASE_ADDR) * 2) + +#ifdef SLOWIO +# undef outb +# undef inb +# define outb outb_p +# define inb inb_p +#endif + +/* JobQueueStruct */ +#define JQS_wStart 0x00 +#define JQS_wSize 0x02 +#define JQS_wHead 0x04 +#define JQS_wTail 0x06 +#define JQS__size 0x08 + +/* DAQueueDataStruct */ +#define DAQDS_wStart 0x00 +#define DAQDS_wSize 0x02 +#define DAQDS_wFormat 0x04 +#define DAQDS_wSampleSize 0x06 +#define DAQDS_wChannels 0x08 +#define DAQDS_wSampleRate 0x0A +#define DAQDS_wIntMsg 0x0C +#define DAQDS_wFlags 0x0E +#define DAQDS__size 0x10 + +#include + +struct snd_msnd { + void __iomem *mappedbase; + int play_period_bytes; + int playLimit; + int playPeriods; + int playDMAPos; + int banksPlayed; + int captureDMAPos; + int capturePeriodBytes; + int captureLimit; + int capturePeriods; + struct snd_card *card; + void *msndmidi_mpu; + struct snd_rawmidi *rmidi; + + /* Hardware resources */ + long io; + int memid, irqid; + int irq, irq_ref; + unsigned long base; + + /* Motorola 56k DSP SMA */ + void __iomem *SMA; + void __iomem *DAPQ; + void __iomem *DARQ; + void __iomem *MODQ; + void __iomem *MIDQ; + void __iomem *DSPQ; + int dspq_data_buff, dspq_buff_size; + + /* State variables */ + enum { msndClassic, msndPinnacle } type; + mode_t mode; + unsigned long flags; +#define F_RESETTING 0 +#define F_HAVEDIGITAL 1 +#define F_AUDIO_WRITE_INUSE 2 +#define F_WRITING 3 +#define F_WRITEBLOCK 4 +#define F_WRITEFLUSH 5 +#define F_AUDIO_READ_INUSE 6 +#define F_READING 7 +#define F_READBLOCK 8 +#define F_EXT_MIDI_INUSE 9 +#define F_HDR_MIDI_INUSE 10 +#define F_DISABLE_WRITE_NDELAY 11 + spinlock_t lock; + spinlock_t mixer_lock; + int nresets; + unsigned recsrc; +#define LEVEL_ENTRIES 32 + int left_levels[LEVEL_ENTRIES]; + int right_levels[LEVEL_ENTRIES]; + int calibrate_signal; + int play_sample_size, play_sample_rate, play_channels; + int play_ndelay; + int capture_sample_size, capture_sample_rate, capture_channels; + int capture_ndelay; + u8 bCurrentMidiPatch; + + int last_playbank, last_recbank; + struct snd_pcm_substream *playback_substream; + struct snd_pcm_substream *capture_substream; + +}; + +void snd_msnd_init_queue(void *base, int start, int size); + +int snd_msnd_send_dsp_cmd(struct snd_msnd *chip, u8 cmd); +int snd_msnd_send_word(struct snd_msnd *chip, + unsigned char high, + unsigned char mid, + unsigned char low); +int snd_msnd_upload_host(struct snd_msnd *chip, + const u8 *bin, int len); +int snd_msnd_enable_irq(struct snd_msnd *chip); +int snd_msnd_disable_irq(struct snd_msnd *chip); +void snd_msnd_dsp_halt(struct snd_msnd *chip, struct file *file); +int snd_msnd_DAPQ(struct snd_msnd *chip, int start); +int snd_msnd_DARQ(struct snd_msnd *chip, int start); +int snd_msnd_pcm(struct snd_card *card, int device, struct snd_pcm **rpcm); + +int snd_msndmidi_new(struct snd_card *card, int device); +void snd_msndmidi_input_read(void *mpu); + +void snd_msndmix_setup(struct snd_msnd *chip); +int __devinit snd_msndmix_new(struct snd_card *card); +int snd_msndmix_force_recsrc(struct snd_msnd *chip, int recsrc); +#endif /* __MSND_H */ diff --git a/sound/isa/msnd/msnd_classic.c b/sound/isa/msnd/msnd_classic.c new file mode 100644 index 00000000000..3b23a096fa4 --- /dev/null +++ b/sound/isa/msnd/msnd_classic.c @@ -0,0 +1,3 @@ +/* The work is in msnd_pinnacle.c, just define MSND_CLASSIC before it. */ +#define MSND_CLASSIC +#include "msnd_pinnacle.c" diff --git a/sound/isa/msnd/msnd_classic.h b/sound/isa/msnd/msnd_classic.h new file mode 100644 index 00000000000..f18d5fa5baf --- /dev/null +++ b/sound/isa/msnd/msnd_classic.h @@ -0,0 +1,129 @@ +/********************************************************************* + * + * msnd_classic.h + * + * Turtle Beach MultiSound Sound Card Driver for Linux + * + * Some parts of this header file were derived from the Turtle Beach + * MultiSound Driver Development Kit. + * + * Copyright (C) 1998 Andrew Veliath + * Copyright (C) 1993 Turtle Beach Systems, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + * + ********************************************************************/ +#ifndef __MSND_CLASSIC_H +#define __MSND_CLASSIC_H + +#define DSP_NUMIO 0x10 + +#define HP_MEMM 0x08 + +#define HP_BITM 0x0E +#define HP_WAIT 0x0D +#define HP_DSPR 0x0A +#define HP_PROR 0x0B +#define HP_BLKS 0x0C + +#define HPPRORESET_OFF 0 +#define HPPRORESET_ON 1 + +#define HPDSPRESET_OFF 0 +#define HPDSPRESET_ON 1 + +#define HPBLKSEL_0 0 +#define HPBLKSEL_1 1 + +#define HPWAITSTATE_0 0 +#define HPWAITSTATE_1 1 + +#define HPBITMODE_16 0 +#define HPBITMODE_8 1 + +#define HIDSP_INT_PLAY_UNDER 0x00 +#define HIDSP_INT_RECORD_OVER 0x01 +#define HIDSP_INPUT_CLIPPING 0x02 +#define HIDSP_MIDI_IN_OVER 0x10 +#define HIDSP_MIDI_OVERRUN_ERR 0x13 + +#define TIME_PRO_RESET_DONE 0x028A +#define TIME_PRO_SYSEX 0x0040 +#define TIME_PRO_RESET 0x0032 + +#define DAR_BUFF_SIZE 0x2000 + +#define MIDQ_BUFF_SIZE 0x200 +#define DSPQ_BUFF_SIZE 0x40 + +#define DSPQ_DATA_BUFF 0x7260 + +#define MOP_SYNTH 0x10 +#define MOP_EXTOUT 0x32 +#define MOP_EXTTHRU 0x02 +#define MOP_OUTMASK 0x01 + +#define MIP_EXTIN 0x01 +#define MIP_SYNTH 0x00 +#define MIP_INMASK 0x32 + +/* Classic SMA Common Data */ +#define SMA_wCurrPlayBytes 0x0000 +#define SMA_wCurrRecordBytes 0x0002 +#define SMA_wCurrPlayVolLeft 0x0004 +#define SMA_wCurrPlayVolRight 0x0006 +#define SMA_wCurrInVolLeft 0x0008 +#define SMA_wCurrInVolRight 0x000a +#define SMA_wUser_3 0x000c +#define SMA_wUser_4 0x000e +#define SMA_dwUser_5 0x0010 +#define SMA_dwUser_6 0x0014 +#define SMA_wUser_7 0x0018 +#define SMA_wReserved_A 0x001a +#define SMA_wReserved_B 0x001c +#define SMA_wReserved_C 0x001e +#define SMA_wReserved_D 0x0020 +#define SMA_wReserved_E 0x0022 +#define SMA_wReserved_F 0x0024 +#define SMA_wReserved_G 0x0026 +#define SMA_wReserved_H 0x0028 +#define SMA_wCurrDSPStatusFlags 0x002a +#define SMA_wCurrHostStatusFlags 0x002c +#define SMA_wCurrInputTagBits 0x002e +#define SMA_wCurrLeftPeak 0x0030 +#define SMA_wCurrRightPeak 0x0032 +#define SMA_wExtDSPbits 0x0034 +#define SMA_bExtHostbits 0x0036 +#define SMA_bBoardLevel 0x0037 +#define SMA_bInPotPosRight 0x0038 +#define SMA_bInPotPosLeft 0x0039 +#define SMA_bAuxPotPosRight 0x003a +#define SMA_bAuxPotPosLeft 0x003b +#define SMA_wCurrMastVolLeft 0x003c +#define SMA_wCurrMastVolRight 0x003e +#define SMA_bUser_12 0x0040 +#define SMA_bUser_13 0x0041 +#define SMA_wUser_14 0x0042 +#define SMA_wUser_15 0x0044 +#define SMA_wCalFreqAtoD 0x0046 +#define SMA_wUser_16 0x0048 +#define SMA_wUser_17 0x004a +#define SMA__size 0x004c + +#define INITCODEFILE "turtlebeach/msndinit.bin" +#define PERMCODEFILE "turtlebeach/msndperm.bin" +#define LONGNAME "MultiSound (Classic/Monterey/Tahiti)" + +#endif /* __MSND_CLASSIC_H */ diff --git a/sound/isa/msnd/msnd_midi.c b/sound/isa/msnd/msnd_midi.c new file mode 100644 index 00000000000..cb9aa4c4edd --- /dev/null +++ b/sound/isa/msnd/msnd_midi.c @@ -0,0 +1,180 @@ +/* + * Copyright (c) by Jaroslav Kysela + * Copyright (c) 2009 by Krzysztof Helt + * Routines for control of MPU-401 in UART mode + * + * MPU-401 supports UART mode which is not capable generate transmit + * interrupts thus output is done via polling. Also, if irq < 0, then + * input is done also via polling. Do not expect good performance. + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#include +#include +#include +#include +#include +#include + +#include "msnd.h" + +#define MSNDMIDI_MODE_BIT_INPUT 0 +#define MSNDMIDI_MODE_BIT_OUTPUT 1 +#define MSNDMIDI_MODE_BIT_INPUT_TRIGGER 2 +#define MSNDMIDI_MODE_BIT_OUTPUT_TRIGGER 3 + +struct snd_msndmidi { + struct snd_msnd *dev; + + unsigned long mode; /* MSNDMIDI_MODE_XXXX */ + + struct snd_rawmidi_substream *substream_input; + + spinlock_t input_lock; +}; + +/* + * input/output open/close - protected by open_mutex in rawmidi.c + */ +static int snd_msndmidi_input_open(struct snd_rawmidi_substream *substream) +{ + struct snd_msndmidi *mpu; + + snd_printdd("snd_msndmidi_input_open()\n"); + + mpu = substream->rmidi->private_data; + + mpu->substream_input = substream; + + snd_msnd_enable_irq(mpu->dev); + + snd_msnd_send_dsp_cmd(mpu->dev, HDEX_MIDI_IN_START); + set_bit(MSNDMIDI_MODE_BIT_INPUT, &mpu->mode); + return 0; +} + +static int snd_msndmidi_input_close(struct snd_rawmidi_substream *substream) +{ + struct snd_msndmidi *mpu; + + mpu = substream->rmidi->private_data; + snd_msnd_send_dsp_cmd(mpu->dev, HDEX_MIDI_IN_STOP); + clear_bit(MSNDMIDI_MODE_BIT_INPUT, &mpu->mode); + mpu->substream_input = NULL; + snd_msnd_disable_irq(mpu->dev); + return 0; +} + +static void snd_msndmidi_input_drop(struct snd_msndmidi *mpu) +{ + u16 tail; + + tail = readw(mpu->dev->MIDQ + JQS_wTail); + writew(tail, mpu->dev->MIDQ + JQS_wHead); +} + +/* + * trigger input + */ +static void snd_msndmidi_input_trigger(struct snd_rawmidi_substream *substream, + int up) +{ + unsigned long flags; + struct snd_msndmidi *mpu; + + snd_printdd("snd_msndmidi_input_trigger(, %i)\n", up); + + mpu = substream->rmidi->private_data; + spin_lock_irqsave(&mpu->input_lock, flags); + if (up) { + if (!test_and_set_bit(MSNDMIDI_MODE_BIT_INPUT_TRIGGER, + &mpu->mode)) + snd_msndmidi_input_drop(mpu); + } else { + clear_bit(MSNDMIDI_MODE_BIT_INPUT_TRIGGER, &mpu->mode); + } + spin_unlock_irqrestore(&mpu->input_lock, flags); + if (up) + snd_msndmidi_input_read(mpu); +} + +void snd_msndmidi_input_read(void *mpuv) +{ + unsigned long flags; + struct snd_msndmidi *mpu = mpuv; + void *pwMIDQData = mpu->dev->mappedbase + MIDQ_DATA_BUFF; + + spin_lock_irqsave(&mpu->input_lock, flags); + while (readw(mpu->dev->MIDQ + JQS_wTail) != + readw(mpu->dev->MIDQ + JQS_wHead)) { + u16 wTmp, val; + val = readw(pwMIDQData + 2 * readw(mpu->dev->MIDQ + JQS_wHead)); + + if (test_bit(MSNDMIDI_MODE_BIT_INPUT_TRIGGER, + &mpu->mode)) + snd_rawmidi_receive(mpu->substream_input, + (unsigned char *)&val, 1); + + wTmp = readw(mpu->dev->MIDQ + JQS_wHead) + 1; + if (wTmp > readw(mpu->dev->MIDQ + JQS_wSize)) + writew(0, mpu->dev->MIDQ + JQS_wHead); + else + writew(wTmp, mpu->dev->MIDQ + JQS_wHead); + } + spin_unlock_irqrestore(&mpu->input_lock, flags); +} +EXPORT_SYMBOL(snd_msndmidi_input_read); + +static struct snd_rawmidi_ops snd_msndmidi_input = { + .open = snd_msndmidi_input_open, + .close = snd_msndmidi_input_close, + .trigger = snd_msndmidi_input_trigger, +}; + +static void snd_msndmidi_free(struct snd_rawmidi *rmidi) +{ + struct snd_msndmidi *mpu = rmidi->private_data; + kfree(mpu); +} + +int snd_msndmidi_new(struct snd_card *card, int device) +{ + struct snd_msnd *chip = card->private_data; + struct snd_msndmidi *mpu; + struct snd_rawmidi *rmidi; + int err; + + err = snd_rawmidi_new(card, "MSND-MIDI", device, 1, 1, &rmidi); + if (err < 0) + return err; + mpu = kcalloc(1, sizeof(*mpu), GFP_KERNEL); + if (mpu == NULL) { + snd_device_free(card, rmidi); + return -ENOMEM; + } + mpu->dev = chip; + chip->msndmidi_mpu = mpu; + rmidi->private_data = mpu; + rmidi->private_free = snd_msndmidi_free; + spin_lock_init(&mpu->input_lock); + strcpy(rmidi->name, "MSND MIDI"); + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, + &snd_msndmidi_input); + rmidi->info_flags |= SNDRV_RAWMIDI_INFO_INPUT; + return 0; +} diff --git a/sound/isa/msnd/msnd_pinnacle.c b/sound/isa/msnd/msnd_pinnacle.c new file mode 100644 index 00000000000..70559223e8f --- /dev/null +++ b/sound/isa/msnd/msnd_pinnacle.c @@ -0,0 +1,1235 @@ +/********************************************************************* + * + * Linux multisound pinnacle/fiji driver for ALSA. + * + * 2002/06/30 Karsten Wiese: + * for now this is only used to build a pinnacle / fiji driver. + * the OSS parent of this code is designed to also support + * the multisound classic via the file msnd_classic.c. + * to make it easier for some brave heart to implemt classic + * support in alsa, i left all the MSND_CLASSIC tokens in this file. + * but for now this untested & undone. + * + * + * ripped from linux kernel 2.4.18 by Karsten Wiese. + * + * the following is a copy of the 2.4.18 OSS FREE file-heading comment: + * + * Turtle Beach MultiSound Sound Card Driver for Linux + * msnd_pinnacle.c / msnd_classic.c + * + * -- If MSND_CLASSIC is defined: + * + * -> driver for Turtle Beach Classic/Monterey/Tahiti + * + * -- Else + * + * -> driver for Turtle Beach Pinnacle/Fiji + * + * 12-3-2000 Modified IO port validation Steve Sycamore + * + * Copyright (C) 1998 Andrew Veliath + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + * + ********************************************************************/ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include + +#ifdef MSND_CLASSIC +# ifndef __alpha__ +# define SLOWIO +# endif +#endif +#include "msnd.h" +#ifdef MSND_CLASSIC +# include "msnd_classic.h" +# define LOGNAME "msnd_classic" +#else +# include "msnd_pinnacle.h" +# define LOGNAME "snd_msnd_pinnacle" +#endif + +static void __devinit set_default_audio_parameters(struct snd_msnd *chip) +{ + chip->play_sample_size = DEFSAMPLESIZE; + chip->play_sample_rate = DEFSAMPLERATE; + chip->play_channels = DEFCHANNELS; + chip->capture_sample_size = DEFSAMPLESIZE; + chip->capture_sample_rate = DEFSAMPLERATE; + chip->capture_channels = DEFCHANNELS; +} + +static void snd_msnd_eval_dsp_msg(struct snd_msnd *chip, u16 wMessage) +{ + switch (HIBYTE(wMessage)) { + case HIMT_PLAY_DONE: { + if (chip->banksPlayed < 3) + snd_printdd("%08X: HIMT_PLAY_DONE: %i\n", + (unsigned)jiffies, LOBYTE(wMessage)); + + if (chip->last_playbank == LOBYTE(wMessage)) { + snd_printdd("chip.last_playbank == LOBYTE(wMessage)\n"); + break; + } + chip->banksPlayed++; + + if (test_bit(F_WRITING, &chip->flags)) + snd_msnd_DAPQ(chip, 0); + + chip->last_playbank = LOBYTE(wMessage); + chip->playDMAPos += chip->play_period_bytes; + if (chip->playDMAPos > chip->playLimit) + chip->playDMAPos = 0; + snd_pcm_period_elapsed(chip->playback_substream); + + break; + } + case HIMT_RECORD_DONE: + if (chip->last_recbank == LOBYTE(wMessage)) + break; + chip->last_recbank = LOBYTE(wMessage); + chip->captureDMAPos += chip->capturePeriodBytes; + if (chip->captureDMAPos > (chip->captureLimit)) + chip->captureDMAPos = 0; + + if (test_bit(F_READING, &chip->flags)) + snd_msnd_DARQ(chip, chip->last_recbank); + + snd_pcm_period_elapsed(chip->capture_substream); + break; + + case HIMT_DSP: + switch (LOBYTE(wMessage)) { +#ifndef MSND_CLASSIC + case HIDSP_PLAY_UNDER: +#endif + case HIDSP_INT_PLAY_UNDER: + snd_printd(KERN_WARNING LOGNAME ": Play underflow %i\n", + chip->banksPlayed); + if (chip->banksPlayed > 2) + clear_bit(F_WRITING, &chip->flags); + break; + + case HIDSP_INT_RECORD_OVER: + snd_printd(KERN_WARNING LOGNAME ": Record overflow\n"); + clear_bit(F_READING, &chip->flags); + break; + + default: + snd_printd(KERN_WARNING LOGNAME + ": DSP message %d 0x%02x\n", + LOBYTE(wMessage), LOBYTE(wMessage)); + break; + } + break; + + case HIMT_MIDI_IN_UCHAR: + if (chip->msndmidi_mpu) + snd_msndmidi_input_read(chip->msndmidi_mpu); + break; + + default: + snd_printd(KERN_WARNING LOGNAME ": HIMT message %d 0x%02x\n", + HIBYTE(wMessage), HIBYTE(wMessage)); + break; + } +} + +static irqreturn_t snd_msnd_interrupt(int irq, void *dev_id) +{ + struct snd_msnd *chip = dev_id; + void *pwDSPQData = chip->mappedbase + DSPQ_DATA_BUFF; + + /* Send ack to DSP */ + /* inb(chip->io + HP_RXL); */ + + /* Evaluate queued DSP messages */ + while (readw(chip->DSPQ + JQS_wTail) != readw(chip->DSPQ + JQS_wHead)) { + u16 wTmp; + + snd_msnd_eval_dsp_msg(chip, + readw(pwDSPQData + 2 * readw(chip->DSPQ + JQS_wHead))); + + wTmp = readw(chip->DSPQ + JQS_wHead) + 1; + if (wTmp > readw(chip->DSPQ + JQS_wSize)) + writew(0, chip->DSPQ + JQS_wHead); + else + writew(wTmp, chip->DSPQ + JQS_wHead); + } + /* Send ack to DSP */ + inb(chip->io + HP_RXL); + return IRQ_HANDLED; +} + + +static int snd_msnd_reset_dsp(long io, unsigned char *info) +{ + int timeout = 100; + + outb(HPDSPRESET_ON, io + HP_DSPR); + msleep(1); +#ifndef MSND_CLASSIC + if (info) + *info = inb(io + HP_INFO); +#endif + outb(HPDSPRESET_OFF, io + HP_DSPR); + msleep(1); + while (timeout-- > 0) { + if (inb(io + HP_CVR) == HP_CVR_DEF) + return 0; + msleep(1); + } + snd_printk(KERN_ERR LOGNAME ": Cannot reset DSP\n"); + + return -EIO; +} + +static int __devinit snd_msnd_probe(struct snd_card *card) +{ + struct snd_msnd *chip = card->private_data; + unsigned char info; +#ifndef MSND_CLASSIC + char *xv, *rev = NULL; + char *pin = "TB Pinnacle", *fiji = "TB Fiji"; + char *pinfiji = "TB Pinnacle/Fiji"; +#endif + + if (!request_region(chip->io, DSP_NUMIO, "probing")) { + snd_printk(KERN_ERR LOGNAME ": I/O port conflict\n"); + return -ENODEV; + } + + if (snd_msnd_reset_dsp(chip->io, &info) < 0) { + release_region(chip->io, DSP_NUMIO); + return -ENODEV; + } + +#ifdef MSND_CLASSIC + strcpy(card->shortname, "Classic/Tahiti/Monterey"); + strcpy(card->longname, "Turtle Beach Multisound"); + printk(KERN_INFO LOGNAME ": %s, " + "I/O 0x%lx-0x%lx, IRQ %d, memory mapped to 0x%lX-0x%lX\n", + card->shortname, + chip->io, chip->io + DSP_NUMIO - 1, + chip->irq, + chip->base, chip->base + 0x7fff); +#else + switch (info >> 4) { + case 0xf: + xv = "<= 1.15"; + break; + case 0x1: + xv = "1.18/1.2"; + break; + case 0x2: + xv = "1.3"; + break; + case 0x3: + xv = "1.4"; + break; + default: + xv = "unknown"; + break; + } + + switch (info & 0x7) { + case 0x0: + rev = "I"; + strcpy(card->shortname, pin); + break; + case 0x1: + rev = "F"; + strcpy(card->shortname, pin); + break; + case 0x2: + rev = "G"; + strcpy(card->shortname, pin); + break; + case 0x3: + rev = "H"; + strcpy(card->shortname, pin); + break; + case 0x4: + rev = "E"; + strcpy(card->shortname, fiji); + break; + case 0x5: + rev = "C"; + strcpy(card->shortname, fiji); + break; + case 0x6: + rev = "D"; + strcpy(card->shortname, fiji); + break; + case 0x7: + rev = "A-B (Fiji) or A-E (Pinnacle)"; + strcpy(card->shortname, pinfiji); + break; + } + strcpy(card->longname, "Turtle Beach Multisound Pinnacle"); + printk(KERN_INFO LOGNAME ": %s revision %s, Xilinx version %s, " + "I/O 0x%lx-0x%lx, IRQ %d, memory mapped to 0x%lX-0x%lX\n", + card->shortname, + rev, xv, + chip->io, chip->io + DSP_NUMIO - 1, + chip->irq, + chip->base, chip->base + 0x7fff); +#endif + + release_region(chip->io, DSP_NUMIO); + return 0; +} + +static int snd_msnd_init_sma(struct snd_msnd *chip) +{ + static int initted; + u16 mastVolLeft, mastVolRight; + unsigned long flags; + +#ifdef MSND_CLASSIC + outb(chip->memid, chip->io + HP_MEMM); +#endif + outb(HPBLKSEL_0, chip->io + HP_BLKS); + /* Motorola 56k shared memory base */ + chip->SMA = chip->mappedbase + SMA_STRUCT_START; + + if (initted) { + mastVolLeft = readw(chip->SMA + SMA_wCurrMastVolLeft); + mastVolRight = readw(chip->SMA + SMA_wCurrMastVolRight); + } else + mastVolLeft = mastVolRight = 0; + memset_io(chip->mappedbase, 0, 0x8000); + + /* Critical section: bank 1 access */ + spin_lock_irqsave(&chip->lock, flags); + outb(HPBLKSEL_1, chip->io + HP_BLKS); + memset_io(chip->mappedbase, 0, 0x8000); + outb(HPBLKSEL_0, chip->io + HP_BLKS); + spin_unlock_irqrestore(&chip->lock, flags); + + /* Digital audio play queue */ + chip->DAPQ = chip->mappedbase + DAPQ_OFFSET; + snd_msnd_init_queue(chip->DAPQ, DAPQ_DATA_BUFF, DAPQ_BUFF_SIZE); + + /* Digital audio record queue */ + chip->DARQ = chip->mappedbase + DARQ_OFFSET; + snd_msnd_init_queue(chip->DARQ, DARQ_DATA_BUFF, DARQ_BUFF_SIZE); + + /* MIDI out queue */ + chip->MODQ = chip->mappedbase + MODQ_OFFSET; + snd_msnd_init_queue(chip->MODQ, MODQ_DATA_BUFF, MODQ_BUFF_SIZE); + + /* MIDI in queue */ + chip->MIDQ = chip->mappedbase + MIDQ_OFFSET; + snd_msnd_init_queue(chip->MIDQ, MIDQ_DATA_BUFF, MIDQ_BUFF_SIZE); + + /* DSP -> host message queue */ + chip->DSPQ = chip->mappedbase + DSPQ_OFFSET; + snd_msnd_init_queue(chip->DSPQ, DSPQ_DATA_BUFF, DSPQ_BUFF_SIZE); + + /* Setup some DSP values */ +#ifndef MSND_CLASSIC + writew(1, chip->SMA + SMA_wCurrPlayFormat); + writew(chip->play_sample_size, chip->SMA + SMA_wCurrPlaySampleSize); + writew(chip->play_channels, chip->SMA + SMA_wCurrPlayChannels); + writew(chip->play_sample_rate, chip->SMA + SMA_wCurrPlaySampleRate); +#endif + writew(chip->play_sample_rate, chip->SMA + SMA_wCalFreqAtoD); + writew(mastVolLeft, chip->SMA + SMA_wCurrMastVolLeft); + writew(mastVolRight, chip->SMA + SMA_wCurrMastVolRight); +#ifndef MSND_CLASSIC + writel(0x00010000, chip->SMA + SMA_dwCurrPlayPitch); + writel(0x00000001, chip->SMA + SMA_dwCurrPlayRate); +#endif + writew(0x303, chip->SMA + SMA_wCurrInputTagBits); + + initted = 1; + + return 0; +} + + +static int upload_dsp_code(struct snd_card *card) +{ + struct snd_msnd *chip = card->private_data; + const struct firmware *init_fw = NULL, *perm_fw = NULL; + int err; + + outb(HPBLKSEL_0, chip->io + HP_BLKS); + + err = request_firmware(&init_fw, INITCODEFILE, card->dev); + if (err < 0) { + printk(KERN_ERR LOGNAME ": Error loading " INITCODEFILE); + goto cleanup1; + } + err = request_firmware(&perm_fw, PERMCODEFILE, card->dev); + if (err < 0) { + printk(KERN_ERR LOGNAME ": Error loading " PERMCODEFILE); + goto cleanup; + } + + memcpy_toio(chip->mappedbase, perm_fw->data, perm_fw->size); + if (snd_msnd_upload_host(chip, init_fw->data, init_fw->size) < 0) { + printk(KERN_WARNING LOGNAME ": Error uploading to DSP\n"); + err = -ENODEV; + goto cleanup; + } + printk(KERN_INFO LOGNAME ": DSP firmware uploaded\n"); + err = 0; + +cleanup: + release_firmware(perm_fw); +cleanup1: + release_firmware(init_fw); + return err; +} + +#ifdef MSND_CLASSIC +static void reset_proteus(struct snd_msnd *chip) +{ + outb(HPPRORESET_ON, chip->io + HP_PROR); + msleep(TIME_PRO_RESET); + outb(HPPRORESET_OFF, chip->io + HP_PROR); + msleep(TIME_PRO_RESET_DONE); +} +#endif + +static int snd_msnd_initialize(struct snd_card *card) +{ + struct snd_msnd *chip = card->private_data; + int err, timeout; + +#ifdef MSND_CLASSIC + outb(HPWAITSTATE_0, chip->io + HP_WAIT); + outb(HPBITMODE_16, chip->io + HP_BITM); + + reset_proteus(chip); +#endif + err = snd_msnd_init_sma(chip); + if (err < 0) { + printk(KERN_WARNING LOGNAME ": Cannot initialize SMA\n"); + return err; + } + + err = snd_msnd_reset_dsp(chip->io, NULL); + if (err < 0) + return err; + + err = upload_dsp_code(card); + if (err < 0) { + printk(KERN_WARNING LOGNAME ": Cannot upload DSP code\n"); + return err; + } + + timeout = 200; + + while (readw(chip->mappedbase)) { + msleep(1); + if (!timeout--) { + snd_printd(KERN_ERR LOGNAME ": DSP reset timeout\n"); + return -EIO; + } + } + + snd_msndmix_setup(chip); + return 0; +} + +static int snd_msnd_dsp_full_reset(struct snd_card *card) +{ + struct snd_msnd *chip = card->private_data; + int rv; + + if (test_bit(F_RESETTING, &chip->flags) || ++chip->nresets > 10) + return 0; + + set_bit(F_RESETTING, &chip->flags); + snd_msnd_dsp_halt(chip, NULL); /* Unconditionally halt */ + + rv = snd_msnd_initialize(card); + if (rv) + printk(KERN_WARNING LOGNAME ": DSP reset failed\n"); + snd_msndmix_force_recsrc(chip, 0); + clear_bit(F_RESETTING, &chip->flags); + return rv; +} + +static int snd_msnd_dev_free(struct snd_device *device) +{ + snd_printdd("snd_msnd_chip_free()\n"); + return 0; +} + +static int snd_msnd_send_dsp_cmd_chk(struct snd_msnd *chip, u8 cmd) +{ + if (snd_msnd_send_dsp_cmd(chip, cmd) == 0) + return 0; + snd_msnd_dsp_full_reset(chip->card); + return snd_msnd_send_dsp_cmd(chip, cmd); +} + +static int __devinit snd_msnd_calibrate_adc(struct snd_msnd *chip, u16 srate) +{ + snd_printdd("snd_msnd_calibrate_adc(%i)\n", srate); + writew(srate, chip->SMA + SMA_wCalFreqAtoD); + if (chip->calibrate_signal == 0) + writew(readw(chip->SMA + SMA_wCurrHostStatusFlags) + | 0x0001, chip->SMA + SMA_wCurrHostStatusFlags); + else + writew(readw(chip->SMA + SMA_wCurrHostStatusFlags) + & ~0x0001, chip->SMA + SMA_wCurrHostStatusFlags); + if (snd_msnd_send_word(chip, 0, 0, HDEXAR_CAL_A_TO_D) == 0 && + snd_msnd_send_dsp_cmd_chk(chip, HDEX_AUX_REQ) == 0) { + schedule_timeout_interruptible(msecs_to_jiffies(333)); + return 0; + } + printk(KERN_WARNING LOGNAME ": ADC calibration failed\n"); + return -EIO; +} + +/* + * ALSA callback function, called when attempting to open the MIDI device. + */ +static int snd_msnd_mpu401_open(struct snd_mpu401 *mpu) +{ + snd_msnd_enable_irq(mpu->private_data); + snd_msnd_send_dsp_cmd(mpu->private_data, HDEX_MIDI_IN_START); + return 0; +} + +static void snd_msnd_mpu401_close(struct snd_mpu401 *mpu) +{ + snd_msnd_send_dsp_cmd(mpu->private_data, HDEX_MIDI_IN_STOP); + snd_msnd_disable_irq(mpu->private_data); +} + +static long mpu_io[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; +static int mpu_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; + +static int __devinit snd_msnd_attach(struct snd_card *card) +{ + struct snd_msnd *chip = card->private_data; + int err; + static struct snd_device_ops ops = { + .dev_free = snd_msnd_dev_free, + }; + + err = request_irq(chip->irq, snd_msnd_interrupt, 0, card->shortname, + chip); + if (err < 0) { + printk(KERN_ERR LOGNAME ": Couldn't grab IRQ %d\n", chip->irq); + return err; + } + request_region(chip->io, DSP_NUMIO, card->shortname); + + if (!request_mem_region(chip->base, BUFFSIZE, card->shortname)) { + printk(KERN_ERR LOGNAME + ": unable to grab memory region 0x%lx-0x%lx\n", + chip->base, chip->base + BUFFSIZE - 1); + release_region(chip->io, DSP_NUMIO); + free_irq(chip->irq, chip); + return -EBUSY; + } + chip->mappedbase = ioremap_nocache(chip->base, 0x8000); + if (!chip->mappedbase) { + printk(KERN_ERR LOGNAME + ": unable to map memory region 0x%lx-0x%lx\n", + chip->base, chip->base + BUFFSIZE - 1); + err = -EIO; + goto err_release_region; + } + + err = snd_msnd_dsp_full_reset(card); + if (err < 0) + goto err_release_region; + + /* Register device */ + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) + goto err_release_region; + + err = snd_msnd_pcm(card, 0, NULL); + if (err < 0) { + printk(KERN_ERR LOGNAME ": error creating new PCM device\n"); + goto err_release_region; + } + + err = snd_msndmix_new(card); + if (err < 0) { + printk(KERN_ERR LOGNAME ": error creating new Mixer device\n"); + goto err_release_region; + } + + + if (mpu_io[0] != SNDRV_AUTO_PORT) { + struct snd_mpu401 *mpu; + + err = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, + mpu_io[0], + MPU401_MODE_INPUT | + MPU401_MODE_OUTPUT, + mpu_irq[0], IRQF_DISABLED, + &chip->rmidi); + if (err < 0) { + printk(KERN_ERR LOGNAME + ": error creating new Midi device\n"); + goto err_release_region; + } + mpu = chip->rmidi->private_data; + + mpu->open_input = snd_msnd_mpu401_open; + mpu->close_input = snd_msnd_mpu401_close; + mpu->private_data = chip; + } + + disable_irq(chip->irq); + snd_msnd_calibrate_adc(chip, chip->play_sample_rate); + snd_msndmix_force_recsrc(chip, 0); + + err = snd_card_register(card); + if (err < 0) + goto err_release_region; + + return 0; + +err_release_region: + if (chip->mappedbase) + iounmap(chip->mappedbase); + release_mem_region(chip->base, BUFFSIZE); + release_region(chip->io, DSP_NUMIO); + free_irq(chip->irq, chip); + return err; +} + + +static void __devexit snd_msnd_unload(struct snd_card *card) +{ + struct snd_msnd *chip = card->private_data; + + iounmap(chip->mappedbase); + release_mem_region(chip->base, BUFFSIZE); + release_region(chip->io, DSP_NUMIO); + free_irq(chip->irq, chip); + snd_card_free(card); +} + +#ifndef MSND_CLASSIC + +/* Pinnacle/Fiji Logical Device Configuration */ + +static int __devinit snd_msnd_write_cfg(int cfg, int reg, int value) +{ + outb(reg, cfg); + outb(value, cfg + 1); + if (value != inb(cfg + 1)) { + printk(KERN_ERR LOGNAME ": snd_msnd_write_cfg: I/O error\n"); + return -EIO; + } + return 0; +} + +static int __devinit snd_msnd_write_cfg_io0(int cfg, int num, u16 io) +{ + if (snd_msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) + return -EIO; + if (snd_msnd_write_cfg(cfg, IREG_IO0_BASEHI, HIBYTE(io))) + return -EIO; + if (snd_msnd_write_cfg(cfg, IREG_IO0_BASELO, LOBYTE(io))) + return -EIO; + return 0; +} + +static int __devinit snd_msnd_write_cfg_io1(int cfg, int num, u16 io) +{ + if (snd_msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) + return -EIO; + if (snd_msnd_write_cfg(cfg, IREG_IO1_BASEHI, HIBYTE(io))) + return -EIO; + if (snd_msnd_write_cfg(cfg, IREG_IO1_BASELO, LOBYTE(io))) + return -EIO; + return 0; +} + +static int __devinit snd_msnd_write_cfg_irq(int cfg, int num, u16 irq) +{ + if (snd_msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) + return -EIO; + if (snd_msnd_write_cfg(cfg, IREG_IRQ_NUMBER, LOBYTE(irq))) + return -EIO; + if (snd_msnd_write_cfg(cfg, IREG_IRQ_TYPE, IRQTYPE_EDGE)) + return -EIO; + return 0; +} + +static int __devinit snd_msnd_write_cfg_mem(int cfg, int num, int mem) +{ + u16 wmem; + + mem >>= 8; + wmem = (u16)(mem & 0xfff); + if (snd_msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) + return -EIO; + if (snd_msnd_write_cfg(cfg, IREG_MEMBASEHI, HIBYTE(wmem))) + return -EIO; + if (snd_msnd_write_cfg(cfg, IREG_MEMBASELO, LOBYTE(wmem))) + return -EIO; + if (wmem && snd_msnd_write_cfg(cfg, IREG_MEMCONTROL, + MEMTYPE_HIADDR | MEMTYPE_16BIT)) + return -EIO; + return 0; +} + +static int __devinit snd_msnd_activate_logical(int cfg, int num) +{ + if (snd_msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) + return -EIO; + if (snd_msnd_write_cfg(cfg, IREG_ACTIVATE, LD_ACTIVATE)) + return -EIO; + return 0; +} + +static int __devinit snd_msnd_write_cfg_logical(int cfg, int num, u16 io0, + u16 io1, u16 irq, int mem) +{ + if (snd_msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) + return -EIO; + if (snd_msnd_write_cfg_io0(cfg, num, io0)) + return -EIO; + if (snd_msnd_write_cfg_io1(cfg, num, io1)) + return -EIO; + if (snd_msnd_write_cfg_irq(cfg, num, irq)) + return -EIO; + if (snd_msnd_write_cfg_mem(cfg, num, mem)) + return -EIO; + if (snd_msnd_activate_logical(cfg, num)) + return -EIO; + return 0; +} + +static int __devinit snd_msnd_pinnacle_cfg_reset(int cfg) +{ + int i; + + /* Reset devices if told to */ + printk(KERN_INFO LOGNAME ": Resetting all devices\n"); + for (i = 0; i < 4; ++i) + if (snd_msnd_write_cfg_logical(cfg, i, 0, 0, 0, 0)) + return -EIO; + + return 0; +} +#endif + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ + +module_param_array(index, int, NULL, S_IRUGO); +MODULE_PARM_DESC(index, "Index value for msnd_pinnacle soundcard."); +module_param_array(id, charp, NULL, S_IRUGO); +MODULE_PARM_DESC(id, "ID string for msnd_pinnacle soundcard."); + +static long io[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; +static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; +static long mem[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; + +static long cfg[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; + +#ifndef MSND_CLASSIC +/* Extra Peripheral Configuration (Default: Disable) */ +static long ide_io0[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; +static long ide_io1[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; +static int ide_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; + +static long joystick_io[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; +/* If we have the digital daugherboard... */ +static int digital[SNDRV_CARDS]; + +/* Extra Peripheral Configuration */ +static int reset[SNDRV_CARDS]; +#endif + +static int write_ndelay[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)] = 1 }; + +static int calibrate_signal; + +#ifdef CONFIG_PNP +static int isapnp[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; +module_param_array(isapnp, bool, NULL, 0444); +MODULE_PARM_DESC(isapnp, "ISA PnP detection for specified soundcard."); +#endif + +MODULE_AUTHOR("Karsten Wiese "); +MODULE_DESCRIPTION("Turtle Beach " LONGNAME " Linux Driver"); +MODULE_LICENSE("GPL"); +MODULE_FIRMWARE(INITCODEFILE); +MODULE_FIRMWARE(PERMCODEFILE); + +module_param_array(io, long, NULL, S_IRUGO); +MODULE_PARM_DESC(io, "IO port #"); +module_param_array(irq, int, NULL, S_IRUGO); +module_param_array(mem, long, NULL, S_IRUGO); +module_param_array(write_ndelay, int, NULL, S_IRUGO); +module_param(calibrate_signal, int, S_IRUGO); +#ifndef MSND_CLASSIC +module_param_array(digital, int, NULL, S_IRUGO); +module_param_array(cfg, long, NULL, S_IRUGO); +module_param_array(reset, int, 0, S_IRUGO); +module_param_array(mpu_io, long, NULL, S_IRUGO); +module_param_array(mpu_irq, int, NULL, S_IRUGO); +module_param_array(ide_io0, long, NULL, S_IRUGO); +module_param_array(ide_io1, long, NULL, S_IRUGO); +module_param_array(ide_irq, int, NULL, S_IRUGO); +module_param_array(joystick_io, long, NULL, S_IRUGO); +#endif + + +static int __devinit snd_msnd_isa_match(struct device *pdev, unsigned int i) +{ + if (io[i] == SNDRV_AUTO_PORT) + return 0; + + if (irq[i] == SNDRV_AUTO_PORT || mem[i] == SNDRV_AUTO_PORT) { + printk(KERN_WARNING LOGNAME ": io, irq and mem must be set\n"); + return 0; + } + +#ifdef MSND_CLASSIC + if (!(io[i] == 0x290 || + io[i] == 0x260 || + io[i] == 0x250 || + io[i] == 0x240 || + io[i] == 0x230 || + io[i] == 0x220 || + io[i] == 0x210 || + io[i] == 0x3e0)) { + printk(KERN_ERR LOGNAME ": \"io\" - DSP I/O base must be set " + " to 0x210, 0x220, 0x230, 0x240, 0x250, 0x260, 0x290, " + "or 0x3E0\n"); + return 0; + } +#else + if (io[i] < 0x100 || io[i] > 0x3e0 || (io[i] % 0x10) != 0) { + printk(KERN_ERR LOGNAME + ": \"io\" - DSP I/O base must within the range 0x100 " + "to 0x3E0 and must be evenly divisible by 0x10\n"); + return 0; + } +#endif /* MSND_CLASSIC */ + + if (!(irq[i] == 5 || + irq[i] == 7 || + irq[i] == 9 || + irq[i] == 10 || + irq[i] == 11 || + irq[i] == 12)) { + printk(KERN_ERR LOGNAME + ": \"irq\" - must be set to 5, 7, 9, 10, 11 or 12\n"); + return 0; + } + + if (!(mem[i] == 0xb0000 || + mem[i] == 0xc8000 || + mem[i] == 0xd0000 || + mem[i] == 0xd8000 || + mem[i] == 0xe0000 || + mem[i] == 0xe8000)) { + printk(KERN_ERR LOGNAME ": \"mem\" - must be set to " + "0xb0000, 0xc8000, 0xd0000, 0xd8000, 0xe0000 or " + "0xe8000\n"); + return 0; + } + +#ifndef MSND_CLASSIC + if (cfg[i] == SNDRV_AUTO_PORT) { + printk(KERN_INFO LOGNAME ": Assuming PnP mode\n"); + } else if (cfg[i] != 0x250 && cfg[i] != 0x260 && cfg[i] != 0x270) { + printk(KERN_INFO LOGNAME + ": Config port must be 0x250, 0x260 or 0x270 " + "(or unspecified for PnP mode)\n"); + return 0; + } +#endif /* MSND_CLASSIC */ + + return 1; +} + +static int __devinit snd_msnd_isa_probe(struct device *pdev, unsigned int idx) +{ + int err; + struct snd_card *card; + struct snd_msnd *chip; + + if (isapnp[idx] || cfg[idx] == SNDRV_AUTO_PORT) { + printk(KERN_INFO LOGNAME ": Assuming PnP mode\n"); + return -ENODEV; + } + + err = snd_card_create(index[idx], id[idx], THIS_MODULE, + sizeof(struct snd_msnd), &card); + if (err < 0) + return err; + + snd_card_set_dev(card, pdev); + chip = card->private_data; + chip->card = card; + +#ifdef MSND_CLASSIC + switch (irq[idx]) { + case 5: + chip->irqid = HPIRQ_5; break; + case 7: + chip->irqid = HPIRQ_7; break; + case 9: + chip->irqid = HPIRQ_9; break; + case 10: + chip->irqid = HPIRQ_10; break; + case 11: + chip->irqid = HPIRQ_11; break; + case 12: + chip->irqid = HPIRQ_12; break; + } + + switch (mem[idx]) { + case 0xb0000: + chip->memid = HPMEM_B000; break; + case 0xc8000: + chip->memid = HPMEM_C800; break; + case 0xd0000: + chip->memid = HPMEM_D000; break; + case 0xd8000: + chip->memid = HPMEM_D800; break; + case 0xe0000: + chip->memid = HPMEM_E000; break; + case 0xe8000: + chip->memid = HPMEM_E800; break; + } +#else + printk(KERN_INFO LOGNAME ": Non-PnP mode: configuring at port 0x%lx\n", + cfg[idx]); + + if (!request_region(cfg[idx], 2, "Pinnacle/Fiji Config")) { + printk(KERN_ERR LOGNAME ": Config port 0x%lx conflict\n", + cfg[idx]); + snd_card_free(card); + return -EIO; + } + if (reset[idx]) + if (snd_msnd_pinnacle_cfg_reset(cfg[idx])) { + err = -EIO; + goto cfg_error; + } + + /* DSP */ + err = snd_msnd_write_cfg_logical(cfg[idx], 0, + io[idx], 0, + irq[idx], mem[idx]); + + if (err) + goto cfg_error; + + /* The following are Pinnacle specific */ + + /* MPU */ + if (mpu_io[idx] != SNDRV_AUTO_PORT + && mpu_irq[idx] != SNDRV_AUTO_IRQ) { + printk(KERN_INFO LOGNAME + ": Configuring MPU to I/O 0x%lx IRQ %d\n", + mpu_io[idx], mpu_irq[idx]); + err = snd_msnd_write_cfg_logical(cfg[idx], 1, + mpu_io[idx], 0, + mpu_irq[idx], 0); + + if (err) + goto cfg_error; + } + + /* IDE */ + if (ide_io0[idx] != SNDRV_AUTO_PORT + && ide_io1[idx] != SNDRV_AUTO_PORT + && ide_irq[idx] != SNDRV_AUTO_IRQ) { + printk(KERN_INFO LOGNAME + ": Configuring IDE to I/O 0x%lx, 0x%lx IRQ %d\n", + ide_io0[idx], ide_io1[idx], ide_irq[idx]); + err = snd_msnd_write_cfg_logical(cfg[idx], 2, + ide_io0[idx], ide_io1[idx], + ide_irq[idx], 0); + + if (err) + goto cfg_error; + } + + /* Joystick */ + if (joystick_io[idx] != SNDRV_AUTO_PORT) { + printk(KERN_INFO LOGNAME + ": Configuring joystick to I/O 0x%lx\n", + joystick_io[idx]); + err = snd_msnd_write_cfg_logical(cfg[idx], 3, + joystick_io[idx], 0, + 0, 0); + + if (err) + goto cfg_error; + } + release_region(cfg[idx], 2); + +#endif /* MSND_CLASSIC */ + + set_default_audio_parameters(chip); +#ifdef MSND_CLASSIC + chip->type = msndClassic; +#else + chip->type = msndPinnacle; +#endif + chip->io = io[idx]; + chip->irq = irq[idx]; + chip->base = mem[idx]; + + chip->calibrate_signal = calibrate_signal ? 1 : 0; + chip->recsrc = 0; + chip->dspq_data_buff = DSPQ_DATA_BUFF; + chip->dspq_buff_size = DSPQ_BUFF_SIZE; + if (write_ndelay[idx]) + clear_bit(F_DISABLE_WRITE_NDELAY, &chip->flags); + else + set_bit(F_DISABLE_WRITE_NDELAY, &chip->flags); +#ifndef MSND_CLASSIC + if (digital[idx]) + set_bit(F_HAVEDIGITAL, &chip->flags); +#endif + spin_lock_init(&chip->lock); + err = snd_msnd_probe(card); + if (err < 0) { + printk(KERN_ERR LOGNAME ": Probe failed\n"); + snd_card_free(card); + return err; + } + + err = snd_msnd_attach(card); + if (err < 0) { + printk(KERN_ERR LOGNAME ": Attach failed\n"); + snd_card_free(card); + return err; + } + dev_set_drvdata(pdev, card); + + return 0; + +#ifndef MSND_CLASSIC +cfg_error: + release_region(cfg[idx], 2); + snd_card_free(card); + return err; +#endif +} + +static int __devexit snd_msnd_isa_remove(struct device *pdev, unsigned int dev) +{ + snd_msnd_unload(dev_get_drvdata(pdev)); + dev_set_drvdata(pdev, NULL); + return 0; +} + +#define DEV_NAME "msnd-pinnacle" + +static struct isa_driver snd_msnd_driver = { + .match = snd_msnd_isa_match, + .probe = snd_msnd_isa_probe, + .remove = __devexit_p(snd_msnd_isa_remove), + /* FIXME: suspend, resume */ + .driver = { + .name = DEV_NAME + }, +}; + +#ifdef CONFIG_PNP +static int __devinit snd_msnd_pnp_detect(struct pnp_card_link *pcard, + const struct pnp_card_device_id *pid) +{ + static int idx; + struct pnp_dev *pnp_dev; + struct pnp_dev *mpu_dev; + struct snd_card *card; + struct snd_msnd *chip; + int ret; + + for ( ; idx < SNDRV_CARDS; idx++) { + if (isapnp[idx]) + break; + } + if (idx >= SNDRV_CARDS) + return -ENODEV; + + /* + * Check that we still have room for another sound card ... + */ + pnp_dev = pnp_request_card_device(pcard, pid->devs[0].id, NULL); + if (!pnp_dev) + return -ENODEV; + + mpu_dev = pnp_request_card_device(pcard, pid->devs[1].id, NULL); + if (!mpu_dev) + return -ENODEV; + + if (!pnp_is_active(pnp_dev) && pnp_activate_dev(pnp_dev) < 0) { + printk(KERN_INFO "msnd_pinnacle: device is inactive\n"); + return -EBUSY; + } + + if (!pnp_is_active(mpu_dev) && pnp_activate_dev(mpu_dev) < 0) { + printk(KERN_INFO "msnd_pinnacle: MPU device is inactive\n"); + return -EBUSY; + } + + /* + * Create a new ALSA sound card entry, in anticipation + * of detecting our hardware ... + */ + ret = snd_card_create(index[idx], id[idx], THIS_MODULE, + sizeof(struct snd_msnd), &card); + if (ret < 0) + return ret; + + chip = card->private_data; + chip->card = card; + snd_card_set_dev(card, &pcard->card->dev); + + /* + * Read the correct parameters off the ISA PnP bus ... + */ + io[idx] = pnp_port_start(pnp_dev, 0); + irq[idx] = pnp_irq(pnp_dev, 0); + mem[idx] = pnp_mem_start(pnp_dev, 0); + mpu_io[idx] = pnp_port_start(mpu_dev, 0); + mpu_irq[idx] = pnp_irq(mpu_dev, 0); + + set_default_audio_parameters(chip); +#ifdef MSND_CLASSIC + chip->type = msndClassic; +#else + chip->type = msndPinnacle; +#endif + chip->io = io[idx]; + chip->irq = irq[idx]; + chip->base = mem[idx]; + + chip->calibrate_signal = calibrate_signal ? 1 : 0; + chip->recsrc = 0; + chip->dspq_data_buff = DSPQ_DATA_BUFF; + chip->dspq_buff_size = DSPQ_BUFF_SIZE; + if (write_ndelay[idx]) + clear_bit(F_DISABLE_WRITE_NDELAY, &chip->flags); + else + set_bit(F_DISABLE_WRITE_NDELAY, &chip->flags); +#ifndef MSND_CLASSIC + if (digital[idx]) + set_bit(F_HAVEDIGITAL, &chip->flags); +#endif + spin_lock_init(&chip->lock); + ret = snd_msnd_probe(card); + if (ret < 0) { + printk(KERN_ERR LOGNAME ": Probe failed\n"); + goto _release_card; + } + + ret = snd_msnd_attach(card); + if (ret < 0) { + printk(KERN_ERR LOGNAME ": Attach failed\n"); + goto _release_card; + } + + pnp_set_card_drvdata(pcard, card); + ++idx; + return 0; + +_release_card: + snd_card_free(card); + return ret; +} + +static void __devexit snd_msnd_pnp_remove(struct pnp_card_link *pcard) +{ + snd_msnd_unload(pnp_get_card_drvdata(pcard)); + pnp_set_card_drvdata(pcard, NULL); +} + +static int isa_registered; +static int pnp_registered; + +static struct pnp_card_device_id msnd_pnpids[] = { + /* Pinnacle PnP */ + { .id = "BVJ0440", .devs = { { "TBS0000" }, { "TBS0001" } } }, + { .id = "" } /* end */ +}; + +MODULE_DEVICE_TABLE(pnp_card, msnd_pnpids); + +static struct pnp_card_driver msnd_pnpc_driver = { + .flags = PNP_DRIVER_RES_DO_NOT_CHANGE, + .name = "msnd_pinnacle", + .id_table = msnd_pnpids, + .probe = snd_msnd_pnp_detect, + .remove = __devexit_p(snd_msnd_pnp_remove), +}; +#endif /* CONFIG_PNP */ + +static int __init snd_msnd_init(void) +{ + int err; + + err = isa_register_driver(&snd_msnd_driver, SNDRV_CARDS); +#ifdef CONFIG_PNP + if (!err) + isa_registered = 1; + + err = pnp_register_card_driver(&msnd_pnpc_driver); + if (!err) + pnp_registered = 1; + + if (isa_registered) + err = 0; +#endif + return err; +} + +static void __exit snd_msnd_exit(void) +{ +#ifdef CONFIG_PNP + if (pnp_registered) + pnp_unregister_card_driver(&msnd_pnpc_driver); + if (isa_registered) +#endif + isa_unregister_driver(&snd_msnd_driver); +} + +module_init(snd_msnd_init); +module_exit(snd_msnd_exit); + diff --git a/sound/isa/msnd/msnd_pinnacle.h b/sound/isa/msnd/msnd_pinnacle.h new file mode 100644 index 00000000000..48318d1ee34 --- /dev/null +++ b/sound/isa/msnd/msnd_pinnacle.h @@ -0,0 +1,181 @@ +/********************************************************************* + * + * msnd_pinnacle.h + * + * Turtle Beach MultiSound Sound Card Driver for Linux + * + * Some parts of this header file were derived from the Turtle Beach + * MultiSound Driver Development Kit. + * + * Copyright (C) 1998 Andrew Veliath + * Copyright (C) 1993 Turtle Beach Systems, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + * + ********************************************************************/ +#ifndef __MSND_PINNACLE_H +#define __MSND_PINNACLE_H + +#define DSP_NUMIO 0x08 + +#define IREG_LOGDEVICE 0x07 +#define IREG_ACTIVATE 0x30 +#define LD_ACTIVATE 0x01 +#define LD_DISACTIVATE 0x00 +#define IREG_EECONTROL 0x3F +#define IREG_MEMBASEHI 0x40 +#define IREG_MEMBASELO 0x41 +#define IREG_MEMCONTROL 0x42 +#define IREG_MEMRANGEHI 0x43 +#define IREG_MEMRANGELO 0x44 +#define MEMTYPE_8BIT 0x00 +#define MEMTYPE_16BIT 0x02 +#define MEMTYPE_RANGE 0x00 +#define MEMTYPE_HIADDR 0x01 +#define IREG_IO0_BASEHI 0x60 +#define IREG_IO0_BASELO 0x61 +#define IREG_IO1_BASEHI 0x62 +#define IREG_IO1_BASELO 0x63 +#define IREG_IRQ_NUMBER 0x70 +#define IREG_IRQ_TYPE 0x71 +#define IRQTYPE_HIGH 0x02 +#define IRQTYPE_LOW 0x00 +#define IRQTYPE_LEVEL 0x01 +#define IRQTYPE_EDGE 0x00 + +#define HP_DSPR 0x04 +#define HP_BLKS 0x04 + +#define HPDSPRESET_OFF 2 +#define HPDSPRESET_ON 0 + +#define HPBLKSEL_0 2 +#define HPBLKSEL_1 3 + +#define HIMT_DAT_OFF 0x03 + +#define HIDSP_PLAY_UNDER 0x00 +#define HIDSP_INT_PLAY_UNDER 0x01 +#define HIDSP_SSI_TX_UNDER 0x02 +#define HIDSP_RECQ_OVERFLOW 0x08 +#define HIDSP_INT_RECORD_OVER 0x09 +#define HIDSP_SSI_RX_OVERFLOW 0x0a + +#define HIDSP_MIDI_IN_OVER 0x10 + +#define HIDSP_MIDI_FRAME_ERR 0x11 +#define HIDSP_MIDI_PARITY_ERR 0x12 +#define HIDSP_MIDI_OVERRUN_ERR 0x13 + +#define HIDSP_INPUT_CLIPPING 0x20 +#define HIDSP_MIX_CLIPPING 0x30 +#define HIDSP_DAT_IN_OFF 0x21 + +#define TIME_PRO_RESET_DONE 0x028A +#define TIME_PRO_SYSEX 0x001E +#define TIME_PRO_RESET 0x0032 + +#define DAR_BUFF_SIZE 0x1000 + +#define MIDQ_BUFF_SIZE 0x800 +#define DSPQ_BUFF_SIZE 0x5A0 + +#define DSPQ_DATA_BUFF 0x7860 + +#define MOP_WAVEHDR 0 +#define MOP_EXTOUT 1 +#define MOP_HWINIT 0xfe +#define MOP_NONE 0xff +#define MOP_MAX 1 + +#define MIP_EXTIN 0 +#define MIP_WAVEHDR 1 +#define MIP_HWINIT 0xfe +#define MIP_MAX 1 + +/* Pinnacle/Fiji SMA Common Data */ +#define SMA_wCurrPlayBytes 0x0000 +#define SMA_wCurrRecordBytes 0x0002 +#define SMA_wCurrPlayVolLeft 0x0004 +#define SMA_wCurrPlayVolRight 0x0006 +#define SMA_wCurrInVolLeft 0x0008 +#define SMA_wCurrInVolRight 0x000a +#define SMA_wCurrMHdrVolLeft 0x000c +#define SMA_wCurrMHdrVolRight 0x000e +#define SMA_dwCurrPlayPitch 0x0010 +#define SMA_dwCurrPlayRate 0x0014 +#define SMA_wCurrMIDIIOPatch 0x0018 +#define SMA_wCurrPlayFormat 0x001a +#define SMA_wCurrPlaySampleSize 0x001c +#define SMA_wCurrPlayChannels 0x001e +#define SMA_wCurrPlaySampleRate 0x0020 +#define SMA_wCurrRecordFormat 0x0022 +#define SMA_wCurrRecordSampleSize 0x0024 +#define SMA_wCurrRecordChannels 0x0026 +#define SMA_wCurrRecordSampleRate 0x0028 +#define SMA_wCurrDSPStatusFlags 0x002a +#define SMA_wCurrHostStatusFlags 0x002c +#define SMA_wCurrInputTagBits 0x002e +#define SMA_wCurrLeftPeak 0x0030 +#define SMA_wCurrRightPeak 0x0032 +#define SMA_bMicPotPosLeft 0x0034 +#define SMA_bMicPotPosRight 0x0035 +#define SMA_bMicPotMaxLeft 0x0036 +#define SMA_bMicPotMaxRight 0x0037 +#define SMA_bInPotPosLeft 0x0038 +#define SMA_bInPotPosRight 0x0039 +#define SMA_bAuxPotPosLeft 0x003a +#define SMA_bAuxPotPosRight 0x003b +#define SMA_bInPotMaxLeft 0x003c +#define SMA_bInPotMaxRight 0x003d +#define SMA_bAuxPotMaxLeft 0x003e +#define SMA_bAuxPotMaxRight 0x003f +#define SMA_bInPotMaxMethod 0x0040 +#define SMA_bAuxPotMaxMethod 0x0041 +#define SMA_wCurrMastVolLeft 0x0042 +#define SMA_wCurrMastVolRight 0x0044 +#define SMA_wCalFreqAtoD 0x0046 +#define SMA_wCurrAuxVolLeft 0x0048 +#define SMA_wCurrAuxVolRight 0x004a +#define SMA_wCurrPlay1VolLeft 0x004c +#define SMA_wCurrPlay1VolRight 0x004e +#define SMA_wCurrPlay2VolLeft 0x0050 +#define SMA_wCurrPlay2VolRight 0x0052 +#define SMA_wCurrPlay3VolLeft 0x0054 +#define SMA_wCurrPlay3VolRight 0x0056 +#define SMA_wCurrPlay4VolLeft 0x0058 +#define SMA_wCurrPlay4VolRight 0x005a +#define SMA_wCurrPlay1PeakLeft 0x005c +#define SMA_wCurrPlay1PeakRight 0x005e +#define SMA_wCurrPlay2PeakLeft 0x0060 +#define SMA_wCurrPlay2PeakRight 0x0062 +#define SMA_wCurrPlay3PeakLeft 0x0064 +#define SMA_wCurrPlay3PeakRight 0x0066 +#define SMA_wCurrPlay4PeakLeft 0x0068 +#define SMA_wCurrPlay4PeakRight 0x006a +#define SMA_wCurrPlayPeakLeft 0x006c +#define SMA_wCurrPlayPeakRight 0x006e +#define SMA_wCurrDATSR 0x0070 +#define SMA_wCurrDATRXCHNL 0x0072 +#define SMA_wCurrDATTXCHNL 0x0074 +#define SMA_wCurrDATRXRate 0x0076 +#define SMA_dwDSPPlayCount 0x0078 +#define SMA__size 0x007c + +#define INITCODEFILE "turtlebeach/pndspini.bin" +#define PERMCODEFILE "turtlebeach/pndsperm.bin" +#define LONGNAME "MultiSound (Pinnacle/Fiji)" + +#endif /* __MSND_PINNACLE_H */ diff --git a/sound/isa/msnd/msnd_pinnacle_mixer.c b/sound/isa/msnd/msnd_pinnacle_mixer.c new file mode 100644 index 00000000000..494058a1a50 --- /dev/null +++ b/sound/isa/msnd/msnd_pinnacle_mixer.c @@ -0,0 +1,343 @@ +/*************************************************************************** + msnd_pinnacle_mixer.c - description + ------------------- + begin : Fre Jun 7 2002 + copyright : (C) 2002 by karsten wiese + email : annabellesgarden@yahoo.de + ***************************************************************************/ + +/*************************************************************************** + * * + * This program is free software; you can redistribute it and/or modify * + * it under the terms of the GNU General Public License as published by * + * the Free Software Foundation; either version 2 of the License, or * + * (at your option) any later version. * + * * + ***************************************************************************/ + +#include + +#include +#include +#include "msnd.h" +#include "msnd_pinnacle.h" + + +#define MSND_MIXER_VOLUME 0 +#define MSND_MIXER_PCM 1 +#define MSND_MIXER_AUX 2 /* Input source 1 (aux1) */ +#define MSND_MIXER_IMIX 3 /* Recording monitor */ +#define MSND_MIXER_SYNTH 4 +#define MSND_MIXER_SPEAKER 5 +#define MSND_MIXER_LINE 6 +#define MSND_MIXER_MIC 7 +#define MSND_MIXER_RECLEV 11 /* Recording level */ +#define MSND_MIXER_IGAIN 12 /* Input gain */ +#define MSND_MIXER_OGAIN 13 /* Output gain */ +#define MSND_MIXER_DIGITAL 17 /* Digital (input) 1 */ + +/* Device mask bits */ + +#define MSND_MASK_VOLUME (1 << MSND_MIXER_VOLUME) +#define MSND_MASK_SYNTH (1 << MSND_MIXER_SYNTH) +#define MSND_MASK_PCM (1 << MSND_MIXER_PCM) +#define MSND_MASK_SPEAKER (1 << MSND_MIXER_SPEAKER) +#define MSND_MASK_LINE (1 << MSND_MIXER_LINE) +#define MSND_MASK_MIC (1 << MSND_MIXER_MIC) +#define MSND_MASK_IMIX (1 << MSND_MIXER_IMIX) +#define MSND_MASK_RECLEV (1 << MSND_MIXER_RECLEV) +#define MSND_MASK_IGAIN (1 << MSND_MIXER_IGAIN) +#define MSND_MASK_OGAIN (1 << MSND_MIXER_OGAIN) +#define MSND_MASK_AUX (1 << MSND_MIXER_AUX) +#define MSND_MASK_DIGITAL (1 << MSND_MIXER_DIGITAL) + +static int snd_msndmix_info_mux(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static char *texts[3] = { + "Analog", "MASS", "SPDIF", + }; + struct snd_msnd *chip = snd_kcontrol_chip(kcontrol); + unsigned items = test_bit(F_HAVEDIGITAL, &chip->flags) ? 3 : 2; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = items; + if (uinfo->value.enumerated.item >= items) + uinfo->value.enumerated.item = items - 1; + strcpy(uinfo->value.enumerated.name, + texts[uinfo->value.enumerated.item]); + return 0; +} + +static int snd_msndmix_get_mux(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_msnd *chip = snd_kcontrol_chip(kcontrol); + /* MSND_MASK_IMIX is the default */ + ucontrol->value.enumerated.item[0] = 0; + + if (chip->recsrc & MSND_MASK_SYNTH) { + ucontrol->value.enumerated.item[0] = 1; + } else if ((chip->recsrc & MSND_MASK_DIGITAL) && + test_bit(F_HAVEDIGITAL, &chip->flags)) { + ucontrol->value.enumerated.item[0] = 2; + } + + + return 0; +} + +static int snd_msndmix_set_mux(struct snd_msnd *chip, int val) +{ + unsigned newrecsrc; + int change; + unsigned char msndbyte; + + switch (val) { + case 0: + newrecsrc = MSND_MASK_IMIX; + msndbyte = HDEXAR_SET_ANA_IN; + break; + case 1: + newrecsrc = MSND_MASK_SYNTH; + msndbyte = HDEXAR_SET_SYNTH_IN; + break; + case 2: + newrecsrc = MSND_MASK_DIGITAL; + msndbyte = HDEXAR_SET_DAT_IN; + break; + default: + return -EINVAL; + } + change = newrecsrc != chip->recsrc; + if (change) { + change = 0; + if (!snd_msnd_send_word(chip, 0, 0, msndbyte)) + if (!snd_msnd_send_dsp_cmd(chip, HDEX_AUX_REQ)) { + chip->recsrc = newrecsrc; + change = 1; + } + } + return change; +} + +static int snd_msndmix_put_mux(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_msnd *msnd = snd_kcontrol_chip(kcontrol); + return snd_msndmix_set_mux(msnd, ucontrol->value.enumerated.item[0]); +} + + +static int snd_msndmix_volume_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 100; + return 0; +} + +static int snd_msndmix_volume_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_msnd *msnd = snd_kcontrol_chip(kcontrol); + int addr = kcontrol->private_value; + unsigned long flags; + + spin_lock_irqsave(&msnd->mixer_lock, flags); + ucontrol->value.integer.value[0] = msnd->left_levels[addr] * 100; + ucontrol->value.integer.value[0] /= 0xFFFF; + ucontrol->value.integer.value[1] = msnd->right_levels[addr] * 100; + ucontrol->value.integer.value[1] /= 0xFFFF; + spin_unlock_irqrestore(&msnd->mixer_lock, flags); + return 0; +} + +#define update_volm(a, b) \ + do { \ + writew((dev->left_levels[a] >> 1) * \ + readw(dev->SMA + SMA_wCurrMastVolLeft) / 0xffff, \ + dev->SMA + SMA_##b##Left); \ + writew((dev->right_levels[a] >> 1) * \ + readw(dev->SMA + SMA_wCurrMastVolRight) / 0xffff, \ + dev->SMA + SMA_##b##Right); \ + } while (0); + +#define update_potm(d, s, ar) \ + do { \ + writeb((dev->left_levels[d] >> 8) * \ + readw(dev->SMA + SMA_wCurrMastVolLeft) / 0xffff, \ + dev->SMA + SMA_##s##Left); \ + writeb((dev->right_levels[d] >> 8) * \ + readw(dev->SMA + SMA_wCurrMastVolRight) / 0xffff, \ + dev->SMA + SMA_##s##Right); \ + if (snd_msnd_send_word(dev, 0, 0, ar) == 0) \ + snd_msnd_send_dsp_cmd(dev, HDEX_AUX_REQ); \ + } while (0); + +#define update_pot(d, s, ar) \ + do { \ + writeb(dev->left_levels[d] >> 8, \ + dev->SMA + SMA_##s##Left); \ + writeb(dev->right_levels[d] >> 8, \ + dev->SMA + SMA_##s##Right); \ + if (snd_msnd_send_word(dev, 0, 0, ar) == 0) \ + snd_msnd_send_dsp_cmd(dev, HDEX_AUX_REQ); \ + } while (0); + + +static int snd_msndmix_set(struct snd_msnd *dev, int d, int left, int right) +{ + int bLeft, bRight; + int wLeft, wRight; + int updatemaster = 0; + + if (d >= LEVEL_ENTRIES) + return -EINVAL; + + bLeft = left * 0xff / 100; + wLeft = left * 0xffff / 100; + + bRight = right * 0xff / 100; + wRight = right * 0xffff / 100; + + dev->left_levels[d] = wLeft; + dev->right_levels[d] = wRight; + + switch (d) { + /* master volume unscaled controls */ + case MSND_MIXER_LINE: /* line pot control */ + /* scaled by IMIX in digital mix */ + writeb(bLeft, dev->SMA + SMA_bInPotPosLeft); + writeb(bRight, dev->SMA + SMA_bInPotPosRight); + if (snd_msnd_send_word(dev, 0, 0, HDEXAR_IN_SET_POTS) == 0) + snd_msnd_send_dsp_cmd(dev, HDEX_AUX_REQ); + break; + case MSND_MIXER_MIC: /* mic pot control */ + if (dev->type == msndClassic) + return -EINVAL; + /* scaled by IMIX in digital mix */ + writeb(bLeft, dev->SMA + SMA_bMicPotPosLeft); + writeb(bRight, dev->SMA + SMA_bMicPotPosRight); + if (snd_msnd_send_word(dev, 0, 0, HDEXAR_MIC_SET_POTS) == 0) + snd_msnd_send_dsp_cmd(dev, HDEX_AUX_REQ); + break; + case MSND_MIXER_VOLUME: /* master volume */ + writew(wLeft, dev->SMA + SMA_wCurrMastVolLeft); + writew(wRight, dev->SMA + SMA_wCurrMastVolRight); + /* fall through */ + + case MSND_MIXER_AUX: /* aux pot control */ + /* scaled by master volume */ + /* fall through */ + + /* digital controls */ + case MSND_MIXER_SYNTH: /* synth vol (dsp mix) */ + case MSND_MIXER_PCM: /* pcm vol (dsp mix) */ + case MSND_MIXER_IMIX: /* input monitor (dsp mix) */ + /* scaled by master volume */ + updatemaster = 1; + break; + + default: + return -EINVAL; + } + + if (updatemaster) { + /* update master volume scaled controls */ + update_volm(MSND_MIXER_PCM, wCurrPlayVol); + update_volm(MSND_MIXER_IMIX, wCurrInVol); + if (dev->type == msndPinnacle) + update_volm(MSND_MIXER_SYNTH, wCurrMHdrVol); + update_potm(MSND_MIXER_AUX, bAuxPotPos, HDEXAR_AUX_SET_POTS); + } + + return 0; +} + +static int snd_msndmix_volume_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_msnd *msnd = snd_kcontrol_chip(kcontrol); + int change, addr = kcontrol->private_value; + int left, right; + unsigned long flags; + + left = ucontrol->value.integer.value[0] % 101; + right = ucontrol->value.integer.value[1] % 101; + spin_lock_irqsave(&msnd->mixer_lock, flags); + change = msnd->left_levels[addr] != left + || msnd->right_levels[addr] != right; + snd_msndmix_set(msnd, addr, left, right); + spin_unlock_irqrestore(&msnd->mixer_lock, flags); + return change; +} + + +#define DUMMY_VOLUME(xname, xindex, addr) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ + .info = snd_msndmix_volume_info, \ + .get = snd_msndmix_volume_get, .put = snd_msndmix_volume_put, \ + .private_value = addr } + + +static struct snd_kcontrol_new snd_msnd_controls[] = { +DUMMY_VOLUME("Master Volume", 0, MSND_MIXER_VOLUME), +DUMMY_VOLUME("PCM Volume", 0, MSND_MIXER_PCM), +DUMMY_VOLUME("Aux Volume", 0, MSND_MIXER_AUX), +DUMMY_VOLUME("Line Volume", 0, MSND_MIXER_LINE), +DUMMY_VOLUME("Mic Volume", 0, MSND_MIXER_MIC), +DUMMY_VOLUME("Monitor", 0, MSND_MIXER_IMIX), +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .info = snd_msndmix_info_mux, + .get = snd_msndmix_get_mux, + .put = snd_msndmix_put_mux, +} +}; + + +int __devinit snd_msndmix_new(struct snd_card *card) +{ + struct snd_msnd *chip = card->private_data; + unsigned int idx; + int err; + + if (snd_BUG_ON(!chip)) + return -EINVAL; + spin_lock_init(&chip->mixer_lock); + strcpy(card->mixername, "MSND Pinnacle Mixer"); + + for (idx = 0; idx < ARRAY_SIZE(snd_msnd_controls); idx++) + err = snd_ctl_add(card, + snd_ctl_new1(snd_msnd_controls + idx, chip)); + if (err < 0) + return err; + + return 0; +} +EXPORT_SYMBOL(snd_msndmix_new); + +void snd_msndmix_setup(struct snd_msnd *dev) +{ + update_pot(MSND_MIXER_LINE, bInPotPos, HDEXAR_IN_SET_POTS); + update_potm(MSND_MIXER_AUX, bAuxPotPos, HDEXAR_AUX_SET_POTS); + update_volm(MSND_MIXER_PCM, wCurrPlayVol); + update_volm(MSND_MIXER_IMIX, wCurrInVol); + if (dev->type == msndPinnacle) { + update_pot(MSND_MIXER_MIC, bMicPotPos, HDEXAR_MIC_SET_POTS); + update_volm(MSND_MIXER_SYNTH, wCurrMHdrVol); + } +} +EXPORT_SYMBOL(snd_msndmix_setup); + +int snd_msndmix_force_recsrc(struct snd_msnd *dev, int recsrc) +{ + dev->recsrc = -1; + return snd_msndmix_set_mux(dev, recsrc); +} +EXPORT_SYMBOL(snd_msndmix_force_recsrc); -- cgit v1.2.3-70-g09d2 From c96330b083ce88b9fea428df99b4631f1b6410ef Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 28 Jan 2009 08:23:03 +0100 Subject: ALSA: Add description of new snd-msnd-* drivers Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/ALSA-Configuration.txt | 48 +++++++++++++++++++++++++ 1 file changed, 48 insertions(+) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 841a9365d5f..ba7b14a13ab 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -1185,6 +1185,54 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. This module supports multiple devices and PnP. + Module snd-msnd-classic + ----------------------- + + Module for Turtle Beach MultiSound Classic, Tahiti or Monterey + soundcards. + + io - Port # for msnd-classic card + irq - IRQ # for msnd-classic card + mem - Memory address (0xb0000, 0xc8000, 0xd0000, 0xd8000, + 0xe0000 or 0xe8000) + write_ndelay - enable write ndelay (default = 1) + calibrate_signal - calibrate signal (default = 0) + isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) + digital - Digital daughterboard present (default = 0) + cfg - Config port (0x250, 0x260 or 0x270) default = PnP + reset - Reset all devices + mpu_io - MPU401 I/O port + mpu_irq - MPU401 irq# + ide_io0 - IDE port #0 + ide_io1 - IDE port #1 + ide_irq - IDE irq# + joystick_io - Joystick I/O port + + The driver requires firmware files "turtlebeach/msndinit.bin" and + "turtlebeach/msndperm.bin" in the proper firmware directory. + + See Documentation/sound/oss/MultiSound for important information + about this driver. Note that it has been discontinued, but the + Voyetra Turtle Beach knowledge base entry for it is still available + at + http://www.turtlebeach.com/site/kb_ftp/790.asp + + Module snd-msnd-pinnacle + ------------------------ + + Module for Turtle Beach MultiSound Pinnacle/Fiji soundcards. + + io - Port # for pinnacle/fiji card + irq - IRQ # for pinnalce/fiji card + mem - Memory address (0xb0000, 0xc8000, 0xd0000, 0xd8000, + 0xe0000 or 0xe8000) + write_ndelay - enable write ndelay (default = 1) + calibrate_signal - calibrate signal (default = 0) + isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) + + The driver requires firmware files "turtlebeach/pndspini.bin" and + "turtlebeach/pndsperm.bin" in the proper firmware directory. + Module snd-mtpav ---------------- -- cgit v1.2.3-70-g09d2 From a5f7c47391ca15c3e2f8e2aa46fb089408541bcd Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 28 Jan 2009 09:02:52 +0100 Subject: ALSA: enable build of snd-msnd-* drivers Added the missing msnd directory to Makefile. Signed-off-by: Takashi Iwai --- sound/isa/Makefile | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/isa/Makefile b/sound/isa/Makefile index 63af13d901a..b906b9a1a81 100644 --- a/sound/isa/Makefile +++ b/sound/isa/Makefile @@ -26,5 +26,5 @@ obj-$(CONFIG_SND_SC6000) += snd-sc6000.o obj-$(CONFIG_SND_SGALAXY) += snd-sgalaxy.o obj-$(CONFIG_SND_SSCAPE) += snd-sscape.o -obj-$(CONFIG_SND) += ad1816a/ ad1848/ cs423x/ es1688/ gus/ opti9xx/ \ +obj-$(CONFIG_SND) += ad1816a/ ad1848/ cs423x/ es1688/ gus/ msnd/ opti9xx/ \ sb/ wavefront/ wss/ -- cgit v1.2.3-70-g09d2 From e3e9c5e7096f6379ca8fa78413b2055fa29f4530 Mon Sep 17 00:00:00 2001 From: Thadeu Lima de Souza Cascardo Date: Wed, 28 Jan 2009 12:40:42 -0200 Subject: ALSA: Don't cold reset AC97 codecs in some ICH chipsets Check in a quirk list if it should do cold reset when AC97 power saving is enabled. Some devices do not resume properly when cold reset, although power saving works OK. Signed-off-by: Thadeu Lima de Souza Cascardo Signed-off-by: Takashi Iwai --- sound/pci/intel8x0.c | 68 +++++++++++++++++++++++++++++++++++++++------------- 1 file changed, 52 insertions(+), 16 deletions(-) diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 19d3391e229..b37bd268301 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -2287,23 +2287,23 @@ static void do_ali_reset(struct intel8x0 *chip) iputdword(chip, ICHREG(ALI_INTERRUPTSR), 0x00000000); } -static int snd_intel8x0_ich_chip_init(struct intel8x0 *chip, int probing) -{ - unsigned long end_time; - unsigned int cnt, status, nstatus; - - /* put logic to right state */ - /* first clear status bits */ - status = ICH_RCS | ICH_MCINT | ICH_POINT | ICH_PIINT; - if (chip->device_type == DEVICE_NFORCE) - status |= ICH_NVSPINT; - cnt = igetdword(chip, ICHREG(GLOB_STA)); - iputdword(chip, ICHREG(GLOB_STA), cnt & status); +#ifdef CONFIG_SND_AC97_POWER_SAVE +static struct snd_pci_quirk ich_chip_reset_mode[] = { + SND_PCI_QUIRK(0x1014, 0x051f, "Thinkpad R32", 1), + { } /* end */ +}; +static int snd_intel8x0_ich_chip_cold_reset(struct intel8x0 *chip) +{ + unsigned int cnt; /* ACLink on, 2 channels */ + + if (snd_pci_quirk_lookup(chip->pci, ich_chip_reset_mode)) + return -EIO; + cnt = igetdword(chip, ICHREG(GLOB_CNT)); cnt &= ~(ICH_ACLINK | ICH_PCM_246_MASK); -#ifdef CONFIG_SND_AC97_POWER_SAVE + /* do cold reset - the full ac97 powerdown may leave the controller * in a warm state but actually it cannot communicate with the codec. */ @@ -2312,22 +2312,58 @@ static int snd_intel8x0_ich_chip_init(struct intel8x0 *chip, int probing) udelay(10); iputdword(chip, ICHREG(GLOB_CNT), cnt | ICH_AC97COLD); msleep(1); + return 0; +} +#define snd_intel8x0_ich_chip_can_cold_reset(chip) \ + (!snd_pci_quirk_lookup(chip->pci, ich_chip_reset_mode)) #else +#define snd_intel8x0_ich_chip_cold_reset(x) do { } while (0) +#define snd_intel8x0_ich_chip_can_cold_reset(chip) (0) +#endif + +static int snd_intel8x0_ich_chip_reset(struct intel8x0 *chip) +{ + unsigned long end_time; + unsigned int cnt; + /* ACLink on, 2 channels */ + cnt = igetdword(chip, ICHREG(GLOB_CNT)); + cnt &= ~(ICH_ACLINK | ICH_PCM_246_MASK); /* finish cold or do warm reset */ cnt |= (cnt & ICH_AC97COLD) == 0 ? ICH_AC97COLD : ICH_AC97WARM; iputdword(chip, ICHREG(GLOB_CNT), cnt); end_time = (jiffies + (HZ / 4)) + 1; do { if ((igetdword(chip, ICHREG(GLOB_CNT)) & ICH_AC97WARM) == 0) - goto __ok; + return 0; schedule_timeout_uninterruptible(1); } while (time_after_eq(end_time, jiffies)); snd_printk(KERN_ERR "AC'97 warm reset still in progress? [0x%x]\n", igetdword(chip, ICHREG(GLOB_CNT))); return -EIO; +} + +static int snd_intel8x0_ich_chip_init(struct intel8x0 *chip, int probing) +{ + unsigned long end_time; + unsigned int status, nstatus; + unsigned int cnt; + int err; + + /* put logic to right state */ + /* first clear status bits */ + status = ICH_RCS | ICH_MCINT | ICH_POINT | ICH_PIINT; + if (chip->device_type == DEVICE_NFORCE) + status |= ICH_NVSPINT; + cnt = igetdword(chip, ICHREG(GLOB_STA)); + iputdword(chip, ICHREG(GLOB_STA), cnt & status); + + if (snd_intel8x0_ich_chip_can_cold_reset(chip)) + err = snd_intel8x0_ich_chip_cold_reset(chip); + else + err = snd_intel8x0_ich_chip_reset(chip); + if (err < 0) + return err; - __ok: -#endif if (probing) { /* wait for any codec ready status. * Once it becomes ready it should remain ready -- cgit v1.2.3-70-g09d2 From e167280070cccd2e0cde94f73ded0a4b08bf6412 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 28 Jan 2009 16:05:16 +0100 Subject: ALSA: intel8x0 - Fix build with CONFIG_SND_AC97_POWERSAVE=n Signed-off-by: Takashi Iwai --- sound/pci/intel8x0.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index b37bd268301..b13ef1e2a4a 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -2317,7 +2317,7 @@ static int snd_intel8x0_ich_chip_cold_reset(struct intel8x0 *chip) #define snd_intel8x0_ich_chip_can_cold_reset(chip) \ (!snd_pci_quirk_lookup(chip->pci, ich_chip_reset_mode)) #else -#define snd_intel8x0_ich_chip_cold_reset(x) do { } while (0) +#define snd_intel8x0_ich_chip_cold_reset(chip) 0 #define snd_intel8x0_ich_chip_can_cold_reset(chip) (0) #endif -- cgit v1.2.3-70-g09d2 From 61b9b9b109217b2bfb128c3ca24d8f8c839a425f Mon Sep 17 00:00:00 2001 From: Herton Ronaldo Krzesinski Date: Wed, 28 Jan 2009 09:16:33 -0200 Subject: ALSA: hda - Consider additional capture source/selector in ALC889 Currently code for capture source support in ALC889 only considers capture mixers. This change adds additional support for ADC+selector present in ALC889, taking into account also the presence of an additional DMIC connection item in the selector. Signed-off-by: Herton Ronaldo Krzesinski Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 105 +++++++++++++++++++++++++++++++----------- 1 file changed, 77 insertions(+), 28 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 863ab957204..d81cb5eb8c5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -238,6 +238,13 @@ enum { ALC883_MODEL_LAST, }; +/* styles of capture selection */ +enum { + CAPT_MUX = 0, /* only mux based */ + CAPT_MIX, /* only mixer based */ + CAPT_1MUX_MIX, /* first mux and other mixers */ +}; + /* for GPIO Poll */ #define GPIO_MASK 0x03 @@ -276,7 +283,7 @@ struct alc_spec { hda_nid_t *adc_nids; hda_nid_t *capsrc_nids; hda_nid_t dig_in_nid; /* digital-in NID; optional */ - unsigned char is_mix_capture; /* matrix-style capture (non-mux) */ + int capture_style; /* capture style (CAPT_*) */ /* capture source */ unsigned int num_mux_defs; @@ -294,7 +301,7 @@ struct alc_spec { /* dynamic controls, init_verbs and input_mux */ struct auto_pin_cfg autocfg; struct snd_array kctls; - struct hda_input_mux private_imux; + struct hda_input_mux private_imux[3]; hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS]; /* hooks */ @@ -396,7 +403,8 @@ static int alc_mux_enum_put(struct snd_kcontrol *kcontrol, mux_idx = adc_idx >= spec->num_mux_defs ? 0 : adc_idx; imux = &spec->input_mux[mux_idx]; - if (spec->is_mix_capture) { + if (spec->capture_style && + !(spec->capture_style == CAPT_1MUX_MIX && !adc_idx)) { /* Matrix-mixer style (e.g. ALC882) */ unsigned int *cur_val = &spec->cur_mux[adc_idx]; unsigned int i, idx; @@ -4130,7 +4138,7 @@ static int new_analog_input(struct alc_spec *spec, hda_nid_t pin, static int alc880_auto_create_analog_input_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) { - struct hda_input_mux *imux = &spec->private_imux; + struct hda_input_mux *imux = &spec->private_imux[0]; int i, err, idx; for (i = 0; i < AUTO_PIN_LAST; i++) { @@ -4279,7 +4287,7 @@ static int alc880_parse_auto_config(struct hda_codec *codec) add_verb(spec, alc880_volume_init_verbs); spec->num_mux_defs = 1; - spec->input_mux = &spec->private_imux; + spec->input_mux = &spec->private_imux[0]; store_pin_configs(codec); return 1; @@ -5487,7 +5495,7 @@ static int alc260_auto_create_multi_out_ctls(struct alc_spec *spec, static int alc260_auto_create_analog_input_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) { - struct hda_input_mux *imux = &spec->private_imux; + struct hda_input_mux *imux = &spec->private_imux[0]; int i, err, idx; for (i = 0; i < AUTO_PIN_LAST; i++) { @@ -5647,7 +5655,7 @@ static int alc260_parse_auto_config(struct hda_codec *codec) add_verb(spec, alc260_volume_init_verbs); spec->num_mux_defs = 1; - spec->input_mux = &spec->private_imux; + spec->input_mux = &spec->private_imux[0]; store_pin_configs(codec); return 1; @@ -7087,7 +7095,7 @@ static int patch_alc882(struct hda_codec *codec) spec->stream_digital_playback = &alc882_pcm_digital_playback; spec->stream_digital_capture = &alc882_pcm_digital_capture; - spec->is_mix_capture = 1; /* matrix-style capture */ + spec->capture_style = CAPT_MIX; /* matrix-style capture */ if (!spec->adc_nids && spec->input_mux) { /* check whether NID 0x07 is valid */ unsigned int wcap = get_wcaps(codec, 0x07); @@ -7155,10 +7163,14 @@ static hda_nid_t alc883_adc_nids_rev[2] = { 0x09, 0x08 }; +#define alc889_adc_nids alc880_adc_nids + static hda_nid_t alc883_capsrc_nids[2] = { 0x23, 0x22 }; static hda_nid_t alc883_capsrc_nids_rev[2] = { 0x22, 0x23 }; +#define alc889_capsrc_nids alc882_capsrc_nids + /* input MUX */ /* FIXME: should be a matrix-type input source selection */ @@ -8977,6 +8989,8 @@ static int alc883_parse_auto_config(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; int err = alc880_parse_auto_config(codec); + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; if (err < 0) return err; @@ -8990,6 +9004,26 @@ static int alc883_parse_auto_config(struct hda_codec *codec) /* hack - override the init verbs */ spec->init_verbs[0] = alc883_auto_init_verbs; + /* setup input_mux for ALC889 */ + if (codec->vendor_id == 0x10ec0889) { + /* digital-mic input pin is excluded in alc880_auto_create..() + * because it's under 0x18 + */ + if (cfg->input_pins[AUTO_PIN_MIC] == 0x12 || + cfg->input_pins[AUTO_PIN_FRONT_MIC] == 0x12) { + struct hda_input_mux *imux = &spec->private_imux[0]; + for (i = 1; i < 3; i++) + memcpy(&spec->private_imux[i], + &spec->private_imux[0], + sizeof(spec->private_imux[0])); + imux->items[imux->num_items].label = "Int DMic"; + imux->items[imux->num_items].index = 0x0b; + imux->num_items++; + spec->num_mux_defs = 3; + spec->input_mux = spec->private_imux; + } + } + return 1; /* config found */ } @@ -9053,14 +9087,36 @@ static int patch_alc883(struct hda_codec *codec) spec->stream_name_analog = "ALC888 Analog"; spec->stream_name_digital = "ALC888 Digital"; } + if (!spec->num_adc_nids) { + spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids); + spec->adc_nids = alc883_adc_nids; + } + if (!spec->capsrc_nids) + spec->capsrc_nids = alc883_capsrc_nids; + spec->capture_style = CAPT_MIX; /* matrix-style capture */ break; case 0x10ec0889: spec->stream_name_analog = "ALC889 Analog"; spec->stream_name_digital = "ALC889 Digital"; + if (!spec->num_adc_nids) { + spec->num_adc_nids = ARRAY_SIZE(alc889_adc_nids); + spec->adc_nids = alc889_adc_nids; + } + if (!spec->capsrc_nids) + spec->capsrc_nids = alc889_capsrc_nids; + spec->capture_style = CAPT_1MUX_MIX; /* 1mux/Nmix-style + capture */ break; default: spec->stream_name_analog = "ALC883 Analog"; spec->stream_name_digital = "ALC883 Digital"; + if (!spec->num_adc_nids) { + spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids); + spec->adc_nids = alc883_adc_nids; + } + if (!spec->capsrc_nids) + spec->capsrc_nids = alc883_capsrc_nids; + spec->capture_style = CAPT_MIX; /* matrix-style capture */ break; } @@ -9071,13 +9127,6 @@ static int patch_alc883(struct hda_codec *codec) spec->stream_digital_playback = &alc883_pcm_digital_playback; spec->stream_digital_capture = &alc883_pcm_digital_capture; - if (!spec->num_adc_nids) { - spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids); - spec->adc_nids = alc883_adc_nids; - } - if (!spec->capsrc_nids) - spec->capsrc_nids = alc883_capsrc_nids; - spec->is_mix_capture = 1; /* matrix-style capture */ if (!spec->cap_mixer) set_capture_mixer(spec); @@ -10512,7 +10561,7 @@ static int alc262_parse_auto_config(struct hda_codec *codec) add_verb(spec, alc262_volume_init_verbs); spec->num_mux_defs = 1; - spec->input_mux = &spec->private_imux; + spec->input_mux = &spec->private_imux[0]; err = alc_auto_add_mic_boost(codec); if (err < 0) @@ -10881,7 +10930,7 @@ static int patch_alc262(struct hda_codec *codec) spec->stream_digital_playback = &alc262_pcm_digital_playback; spec->stream_digital_capture = &alc262_pcm_digital_capture; - spec->is_mix_capture = 1; + spec->capture_style = CAPT_MIX; if (!spec->adc_nids && spec->input_mux) { /* check whether NID 0x07 is valid */ unsigned int wcap = get_wcaps(codec, 0x07); @@ -11539,7 +11588,7 @@ static int alc268_auto_create_multi_out_ctls(struct alc_spec *spec, static int alc268_auto_create_analog_input_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) { - struct hda_input_mux *imux = &spec->private_imux; + struct hda_input_mux *imux = &spec->private_imux[0]; int i, idx1; for (i = 0; i < AUTO_PIN_LAST; i++) { @@ -11657,7 +11706,7 @@ static int alc268_parse_auto_config(struct hda_codec *codec) add_verb(spec, alc268_volume_init_verbs); spec->num_mux_defs = 1; - spec->input_mux = &spec->private_imux; + spec->input_mux = &spec->private_imux[0]; err = alc_auto_add_mic_boost(codec); if (err < 0) @@ -12511,7 +12560,7 @@ static int alc269_auto_create_analog_input_ctls(struct alc_spec *spec, */ if (cfg->input_pins[AUTO_PIN_MIC] == 0x12 || cfg->input_pins[AUTO_PIN_FRONT_MIC] == 0x12) { - struct hda_input_mux *imux = &spec->private_imux; + struct hda_input_mux *imux = &spec->private_imux[0]; imux->items[imux->num_items].label = "Int Mic"; imux->items[imux->num_items].index = 0x05; imux->num_items++; @@ -12567,7 +12616,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec) add_verb(spec, alc269_init_verbs); spec->num_mux_defs = 1; - spec->input_mux = &spec->private_imux; + spec->input_mux = &spec->private_imux[0]; /* set default input source */ snd_hda_codec_write_cache(codec, alc269_capsrc_nids[0], 0, AC_VERB_SET_CONNECT_SEL, @@ -13483,7 +13532,7 @@ static int alc861_auto_create_hp_ctls(struct alc_spec *spec, hda_nid_t pin) static int alc861_auto_create_analog_input_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) { - struct hda_input_mux *imux = &spec->private_imux; + struct hda_input_mux *imux = &spec->private_imux[0]; int i, err, idx, idx1; for (i = 0; i < AUTO_PIN_LAST; i++) { @@ -13620,7 +13669,7 @@ static int alc861_parse_auto_config(struct hda_codec *codec) add_verb(spec, alc861_auto_init_verbs); spec->num_mux_defs = 1; - spec->input_mux = &spec->private_imux; + spec->input_mux = &spec->private_imux[0]; spec->adc_nids = alc861_adc_nids; spec->num_adc_nids = ARRAY_SIZE(alc861_adc_nids); @@ -14724,7 +14773,7 @@ static int alc861vd_parse_auto_config(struct hda_codec *codec) add_verb(spec, alc861vd_volume_init_verbs); spec->num_mux_defs = 1; - spec->input_mux = &spec->private_imux; + spec->input_mux = &spec->private_imux[0]; err = alc_auto_add_mic_boost(codec); if (err < 0) @@ -14803,7 +14852,7 @@ static int patch_alc861vd(struct hda_codec *codec) spec->adc_nids = alc861vd_adc_nids; spec->num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids); spec->capsrc_nids = alc861vd_capsrc_nids; - spec->is_mix_capture = 1; + spec->capture_style = CAPT_MIX; set_capture_mixer(spec); @@ -16397,7 +16446,7 @@ static int alc662_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin, static int alc662_auto_create_analog_input_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) { - struct hda_input_mux *imux = &spec->private_imux; + struct hda_input_mux *imux = &spec->private_imux[0]; int i, err, idx; for (i = 0; i < AUTO_PIN_LAST; i++) { @@ -16528,7 +16577,7 @@ static int alc662_parse_auto_config(struct hda_codec *codec) add_mixer(spec, spec->kctls.list); spec->num_mux_defs = 1; - spec->input_mux = &spec->private_imux; + spec->input_mux = &spec->private_imux[0]; add_verb(spec, alc662_auto_init_verbs); if (codec->vendor_id == 0x10ec0663) @@ -16613,7 +16662,7 @@ static int patch_alc662(struct hda_codec *codec) spec->adc_nids = alc662_adc_nids; spec->num_adc_nids = ARRAY_SIZE(alc662_adc_nids); spec->capsrc_nids = alc662_capsrc_nids; - spec->is_mix_capture = 1; + spec->capture_style = CAPT_MIX; if (!spec->cap_mixer) set_capture_mixer(spec); -- cgit v1.2.3-70-g09d2 From 328cc6dfaadad614449eca1c75559e64c5054fea Mon Sep 17 00:00:00 2001 From: Thadeu Lima de Souza Cascardo Date: Wed, 28 Jan 2009 15:39:22 -0200 Subject: ALSA: AC97: Print AC97 flags in proc file to make debug it easier While debugging some code paths in AC97 codec patches and its suspend and resume functions, getting to know the flags has proved useful to follow those code paths. Signed-off-by: Thadeu Lima de Souza Cascardo Signed-off-by: Takashi Iwai --- sound/pci/ac97/ac97_proc.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/ac97/ac97_proc.c b/sound/pci/ac97/ac97_proc.c index 060ea59d5f0..73b17d526c8 100644 --- a/sound/pci/ac97/ac97_proc.c +++ b/sound/pci/ac97/ac97_proc.c @@ -125,6 +125,8 @@ static void snd_ac97_proc_read_main(struct snd_ac97 *ac97, struct snd_info_buffe snd_iprintf(buffer, "PCI Subsys Device: 0x%04x\n\n", ac97->subsystem_device); + snd_iprintf(buffer, "Flags: %x\n", ac97->flags); + if ((ac97->ext_id & AC97_EI_REV_MASK) >= AC97_EI_REV_23) { val = snd_ac97_read(ac97, AC97_INT_PAGING); snd_ac97_update_bits(ac97, AC97_INT_PAGING, -- cgit v1.2.3-70-g09d2 From b833b5ec0411adc2255053a0e0ec536d97e5784e Mon Sep 17 00:00:00 2001 From: Thadeu Lima de Souza Cascardo Date: Wed, 28 Jan 2009 18:20:06 -0200 Subject: ALSA: AC97: Fix function name type in comment s/updat/update/ Signed-off-by: Thadeu Lima de Souza Cascardo Signed-off-by: Takashi Iwai --- sound/pci/ac97/ac97_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index e2b843b4f9d..27551e963e5 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -383,7 +383,7 @@ int snd_ac97_update_bits(struct snd_ac97 *ac97, unsigned short reg, unsigned sho EXPORT_SYMBOL(snd_ac97_update_bits); -/* no lock version - see snd_ac97_updat_bits() */ +/* no lock version - see snd_ac97_update_bits() */ int snd_ac97_update_bits_nolock(struct snd_ac97 *ac97, unsigned short reg, unsigned short mask, unsigned short value) { -- cgit v1.2.3-70-g09d2 From 56305757f0b64b7d5dd02fd54c6dfc0989868f31 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Thu, 29 Jan 2009 11:44:24 +0100 Subject: ALSA: sscape: update Kconfig description about SoundScape cards The SoundScape driver handles more cards then described. Signed-off-by: Takashi Iwai --- sound/isa/Kconfig | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index ce0aa044e27..542c1ead14b 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -377,14 +377,17 @@ config SND_SGALAXY will be called snd-sgalaxy. config SND_SSCAPE - tristate "Ensoniq SoundScape PnP driver" + tristate "Ensoniq SoundScape driver" select SND_HWDEP select SND_MPU401_UART select SND_WSS_LIB help - Say Y here to include support for Ensoniq SoundScape PnP + Say Y here to include support for Ensoniq SoundScape soundcards. + The PCM audio is supported on SoundScape Classic, Elite, PnP + and VIVO cards. The MIDI support is very experimental. + To compile this driver as a module, choose M here: the module will be called snd-sscape. -- cgit v1.2.3-70-g09d2 From 0a898e6e500ec8ab98000896fe243c4c0e91c72a Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Thu, 29 Jan 2009 11:46:45 +0100 Subject: ALSA: gus: update debug messages Convert some of them to snd_printdd() and update arguments to make them compilable. Signed-off-by: Takashi Iwai --- sound/isa/gus/gus_dma.c | 24 +++++++++++++++--------- 1 file changed, 15 insertions(+), 9 deletions(-) diff --git a/sound/isa/gus/gus_dma.c b/sound/isa/gus/gus_dma.c index f45f6116c77..cf8cd3c26a5 100644 --- a/sound/isa/gus/gus_dma.c +++ b/sound/isa/gus/gus_dma.c @@ -45,7 +45,8 @@ static void snd_gf1_dma_program(struct snd_gus_card * gus, unsigned char dma_cmd; unsigned int address_high; - // snd_printk("dma_transfer: addr=0x%x, buf=0x%lx, count=0x%x\n", addr, (long) buf, count); + snd_printdd("dma_transfer: addr=0x%x, buf=0x%lx, count=0x%x\n", + addr, buf_addr, count); if (gus->gf1.dma1 > 3) { if (gus->gf1.enh_mode) { @@ -142,7 +143,9 @@ static void snd_gf1_dma_interrupt(struct snd_gus_card * gus) snd_gf1_dma_program(gus, block->addr, block->buf_addr, block->count, (unsigned short) block->cmd); kfree(block); #if 0 - printk("program dma (IRQ) - addr = 0x%x, buffer = 0x%lx, count = 0x%x, cmd = 0x%x\n", addr, (long) buffer, count, cmd); + snd_printd(KERN_DEBUG "program dma (IRQ) - " + "addr = 0x%x, buffer = 0x%lx, count = 0x%x, cmd = 0x%x\n", + block->addr, block->buf_addr, block->count, block->cmd); #endif } @@ -203,13 +206,16 @@ int snd_gf1_dma_transfer_block(struct snd_gus_card * gus, } *block = *__block; block->next = NULL; -#if 0 - printk("addr = 0x%x, buffer = 0x%lx, count = 0x%x, cmd = 0x%x\n", block->addr, (long) block->buffer, block->count, block->cmd); -#endif -#if 0 - printk("gus->gf1.dma_data_pcm_last = 0x%lx\n", (long)gus->gf1.dma_data_pcm_last); - printk("gus->gf1.dma_data_pcm = 0x%lx\n", (long)gus->gf1.dma_data_pcm); -#endif + + snd_printdd("addr = 0x%x, buffer = 0x%lx, count = 0x%x, cmd = 0x%x\n", + block->addr, (long) block->buffer, block->count, + block->cmd); + + snd_printdd("gus->gf1.dma_data_pcm_last = 0x%lx\n", + (long)gus->gf1.dma_data_pcm_last); + snd_printdd("gus->gf1.dma_data_pcm = 0x%lx\n", + (long)gus->gf1.dma_data_pcm); + spin_lock_irqsave(&gus->dma_lock, flags); if (synth) { if (gus->gf1.dma_data_synth_last) { -- cgit v1.2.3-70-g09d2 From c97dff84e0d9a4e0b7048e033d33511e3897c859 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Thu, 29 Jan 2009 11:48:14 +0100 Subject: ALSA: cmi8330: add MPU-401 support Add MPU-401 port support for the chip. Also, update some error messages and description. Signed-off-by: Takashi Iwai --- sound/isa/Kconfig | 1 + sound/isa/cmi8330.c | 42 ++++++++++++++++++++++++++++++++++++------ 2 files changed, 37 insertions(+), 6 deletions(-) diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index be2d377ff90..5915dc41c0e 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -95,6 +95,7 @@ config SND_CMI8330 select SND_WSS_LIB select SND_SB16_DSP select SND_OPL3_LIB + select SND_MPU401_UART help Say Y here to include support for soundcards based on the C-Media CMI8330 chip. diff --git a/sound/isa/cmi8330.c b/sound/isa/cmi8330.c index 11543795741..9ca8122f7ba 100644 --- a/sound/isa/cmi8330.c +++ b/sound/isa/cmi8330.c @@ -31,11 +31,11 @@ * To quickly load the module, * * modprobe -a snd-cmi8330 sbport=0x220 sbirq=5 sbdma8=1 - * sbdma16=5 wssport=0x530 wssirq=11 wssdma=0 + * sbdma16=5 wssport=0x530 wssirq=11 wssdma=0 fmport=0x388 * * This card has two mixers and two PCM devices. I've cheesed it such * that recording and playback can be done through the same device. - * The driver "magically" routes the capturing to the AD1848 codec, + * The driver "magically" routes the capturing to the CMI8330 codec, * and playback to the SB16 codec. This allows for full-duplex mode * to some extent. * The utilities in alsa-utils are aware of both devices, so passing @@ -52,6 +52,7 @@ #include #include #include +#include #include #include @@ -81,6 +82,8 @@ static long wssport[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; static int wssirq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; static int wssdma[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; static long fmport[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; +static long mpuport[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; +static int mpuirq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for CMI8330 soundcard."); @@ -111,6 +114,10 @@ MODULE_PARM_DESC(wssdma, "DMA for CMI8330 WSS driver."); module_param_array(fmport, long, NULL, 0444); MODULE_PARM_DESC(fmport, "FM port # for CMI8330 driver."); +module_param_array(mpuport, long, NULL, 0444); +MODULE_PARM_DESC(mpuport, "MPU-401 port # for CMI8330 driver."); +module_param_array(mpuirq, int, NULL, 0444); +MODULE_PARM_DESC(mpuirq, "IRQ # for CMI8330 MPU-401 port."); #ifdef CONFIG_PNP static int isa_registered; static int pnp_registered; @@ -153,6 +160,7 @@ struct snd_cmi8330 { #ifdef CONFIG_PNP struct pnp_dev *cap; struct pnp_dev *play; + struct pnp_dev *mpu; #endif struct snd_card *card; struct snd_wss *wss; @@ -169,7 +177,7 @@ struct snd_cmi8330 { #ifdef CONFIG_PNP static struct pnp_card_device_id snd_cmi8330_pnpids[] = { - { .id = "CMI0001", .devs = { { "@@@0001" }, { "@X@0001" } } }, + { .id = "CMI0001", .devs = { { "@@@0001" }, { "@X@0001" }, { "@H@0001" } } }, { .id = "" } }; @@ -329,11 +337,15 @@ static int __devinit snd_cmi8330_pnp(int dev, struct snd_cmi8330 *acard, if (acard->play == NULL) return -EBUSY; + acard->mpu = pnp_request_card_device(card, id->devs[2].id, NULL); + if (acard->play == NULL) + return -EBUSY; + pdev = acard->cap; err = pnp_activate_dev(pdev); if (err < 0) { - snd_printk(KERN_ERR "CMI8330/C3D (AD1848) PnP configure failure\n"); + snd_printk(KERN_ERR "CMI8330/C3D PnP configure failure\n"); return -EBUSY; } wssport[dev] = pnp_port_start(pdev, 0); @@ -354,6 +366,17 @@ static int __devinit snd_cmi8330_pnp(int dev, struct snd_cmi8330 *acard, sbdma16[dev] = pnp_dma(pdev, 1); sbirq[dev] = pnp_irq(pdev, 0); + /* allocate MPU-401 resources */ + pdev = acard->mpu; + + err = pnp_activate_dev(pdev); + if (err < 0) { + snd_printk(KERN_ERR + "CMI8330/C3D (MPU-401) PnP configure failure\n"); + return -EBUSY; + } + mpuport[dev] = pnp_port_start(pdev, 0); + mpuirq[dev] = pnp_irq(pdev, 0); return 0; } #endif @@ -502,11 +525,11 @@ static int __devinit snd_cmi8330_probe(struct snd_card *card, int dev) wssdma[dev], -1, WSS_HW_DETECT, 0, &acard->wss); if (err < 0) { - snd_printk(KERN_ERR PFX "(AD1848) device busy??\n"); + snd_printk(KERN_ERR PFX "(CMI8330) device busy??\n"); return err; } if (acard->wss->hardware != WSS_HW_CMI8330) { - snd_printk(KERN_ERR PFX "(AD1848) not found during probe\n"); + snd_printk(KERN_ERR PFX "(CMI8330) not found during probe\n"); return -ENODEV; } @@ -552,6 +575,13 @@ static int __devinit snd_cmi8330_probe(struct snd_card *card, int dev) } } + if (mpuport[dev] != SNDRV_AUTO_PORT) { + if (snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, + mpuport[dev], 0, mpuirq[dev], + IRQF_DISABLED, NULL) < 0) + printk(KERN_ERR PFX "no MPU-401 device at 0x%lx.\n", + mpuport[dev]); + } strcpy(card->driver, "CMI8330/C3D"); strcpy(card->shortname, "C-Media CMI8330/C3D"); -- cgit v1.2.3-70-g09d2 From 9e128fddcc589db4e7d9e8328f656ae4a21a2808 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 29 Jan 2009 11:49:10 +0100 Subject: ALSA: Add missing description of snd-cmi8330 module parameters Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/ALSA-Configuration.txt | 3 +++ 1 file changed, 3 insertions(+) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 841a9365d5f..7134a8f7044 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -346,6 +346,9 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. sbirq - IRQ # for CMI8330 chip (SB16) sbdma8 - 8bit DMA # for CMI8330 chip (SB16) sbdma16 - 16bit DMA # for CMI8330 chip (SB16) + fmport - (optional) OPL3 I/O port + mpuport - (optional) MPU401 I/O port + mpuirq - (optional) MPU401 irq # This module supports multiple cards and autoprobe. -- cgit v1.2.3-70-g09d2 From 7393958f630ac91e591e62058f2bdb61523ec60c Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 29 Jan 2009 14:57:50 +0200 Subject: ASoC: TWL4030: Add analog loopback support This patch adds the analog loopback/bypass support for twl4030 codec. Details for the implementation: It seams that the analog loopback needs the DAC powered on on the channel, where the loopback is selected. The switch for the DACs has been moved from the DAPM_DAC to the "Analog XX Playback Mixer". In this way the DAC will be powered while the audio playback is used or/and the loopback is enabled for the channel. The twl4030 codec powering has been reworked. Now the codec will be powered as long as it does not receives the SND_SOC_BIAS_OFF event. The exceptions are when the given change in the registers needs the codec power down/up cycle in order to take effect. Otherwise the codec is on. When the codec enters to STANDBY state and none of the loopback paths are enabled, than the amplifiers, which are no in the DAPM path are forced to turn off and the PLL is disabled. When playback/capture starts the disabled gains are restored and the PLL is enabled. When one of the loopback enabled in STANDBY mode, the disabled gains are restored and the PLL is enabled also. In short: the codec always goes to the lowest power state based on the bias_level and the bypass_state. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 212 ++++++++++++++++++++++++++++++++++++++++++--- sound/soc/codecs/twl4030.h | 15 ++++ 2 files changed, 214 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index f985bef40a3..c26854b398d 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -117,6 +117,13 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0x00, /* REG_MISC_SET_2 (0x49) */ }; +/* codec private data */ +struct twl4030_priv { + unsigned int bypass_state; + unsigned int codec_powered; + unsigned int codec_muted; +}; + /* * read twl4030 register cache */ @@ -156,8 +163,12 @@ static int twl4030_write(struct snd_soc_codec *codec, static void twl4030_codec_enable(struct snd_soc_codec *codec, int enable) { + struct twl4030_priv *twl4030 = codec->private_data; u8 mode; + if (enable == twl4030->codec_powered) + return; + mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE); if (enable) mode |= TWL4030_CODECPDZ; @@ -165,6 +176,7 @@ static void twl4030_codec_enable(struct snd_soc_codec *codec, int enable) mode &= ~TWL4030_CODECPDZ; twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode); + twl4030->codec_powered = enable; /* REVISIT: this delay is present in TI sample drivers */ /* but there seems to be no TRM requirement for it */ @@ -184,11 +196,82 @@ static void twl4030_init_chip(struct snd_soc_codec *codec) } +static void twl4030_codec_mute(struct snd_soc_codec *codec, int mute) +{ + struct twl4030_priv *twl4030 = codec->private_data; + u8 reg_val; + + if (mute == twl4030->codec_muted) + return; + + if (mute) { + /* Bypass the reg_cache and mute the volumes + * Headset mute is done in it's own event handler + * Things to mute: Earpiece, PreDrivL/R, CarkitL/R + */ + reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_EAR_CTL); + twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, + reg_val & (~TWL4030_EAR_GAIN), + TWL4030_REG_EAR_CTL); + + reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_PREDL_CTL); + twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, + reg_val & (~TWL4030_PREDL_GAIN), + TWL4030_REG_PREDL_CTL); + reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_PREDR_CTL); + twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, + reg_val & (~TWL4030_PREDR_GAIN), + TWL4030_REG_PREDL_CTL); + + reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_PRECKL_CTL); + twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, + reg_val & (~TWL4030_PRECKL_GAIN), + TWL4030_REG_PRECKL_CTL); + reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_PRECKR_CTL); + twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, + reg_val & (~TWL4030_PRECKL_GAIN), + TWL4030_REG_PRECKR_CTL); + + /* Disable PLL */ + reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_APLL_CTL); + reg_val &= ~TWL4030_APLL_EN; + twl4030_write(codec, TWL4030_REG_APLL_CTL, reg_val); + } else { + /* Restore the volumes + * Headset mute is done in it's own event handler + * Things to restore: Earpiece, PreDrivL/R, CarkitL/R + */ + twl4030_write(codec, TWL4030_REG_EAR_CTL, + twl4030_read_reg_cache(codec, TWL4030_REG_EAR_CTL)); + + twl4030_write(codec, TWL4030_REG_PREDL_CTL, + twl4030_read_reg_cache(codec, TWL4030_REG_PREDL_CTL)); + twl4030_write(codec, TWL4030_REG_PREDR_CTL, + twl4030_read_reg_cache(codec, TWL4030_REG_PREDR_CTL)); + + twl4030_write(codec, TWL4030_REG_PRECKL_CTL, + twl4030_read_reg_cache(codec, TWL4030_REG_PRECKL_CTL)); + twl4030_write(codec, TWL4030_REG_PRECKR_CTL, + twl4030_read_reg_cache(codec, TWL4030_REG_PRECKR_CTL)); + + /* Enable PLL */ + reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_APLL_CTL); + reg_val |= TWL4030_APLL_EN; + twl4030_write(codec, TWL4030_REG_APLL_CTL, reg_val); + } + + twl4030->codec_muted = mute; +} + static void twl4030_power_up(struct snd_soc_codec *codec) { + struct twl4030_priv *twl4030 = codec->private_data; u8 anamicl, regmisc1, byte; int i = 0; + if (twl4030->codec_powered) + return; + /* set CODECPDZ to turn on codec */ twl4030_codec_enable(codec, 1); @@ -220,6 +303,9 @@ static void twl4030_power_up(struct snd_soc_codec *codec) twl4030_codec_enable(codec, 1); } +/* + * Unconditional power down + */ static void twl4030_power_down(struct snd_soc_codec *codec) { /* power down */ @@ -402,6 +488,22 @@ static const struct soc_enum twl4030_micpathtx2_enum = static const struct snd_kcontrol_new twl4030_dapm_micpathtx2_control = SOC_DAPM_ENUM("Route", twl4030_micpathtx2_enum); +/* Analog bypass for AudioR1 */ +static const struct snd_kcontrol_new twl4030_dapm_abypassr1_control = + SOC_DAPM_SINGLE("Switch", TWL4030_REG_ARXR1_APGA_CTL, 2, 1, 0); + +/* Analog bypass for AudioL1 */ +static const struct snd_kcontrol_new twl4030_dapm_abypassl1_control = + SOC_DAPM_SINGLE("Switch", TWL4030_REG_ARXL1_APGA_CTL, 2, 1, 0); + +/* Analog bypass for AudioR2 */ +static const struct snd_kcontrol_new twl4030_dapm_abypassr2_control = + SOC_DAPM_SINGLE("Switch", TWL4030_REG_ARXR2_APGA_CTL, 2, 1, 0); + +/* Analog bypass for AudioL2 */ +static const struct snd_kcontrol_new twl4030_dapm_abypassl2_control = + SOC_DAPM_SINGLE("Switch", TWL4030_REG_ARXL2_APGA_CTL, 2, 1, 0); + static int micpath_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -497,6 +599,31 @@ static int headsetl_event(struct snd_soc_dapm_widget *w, return 0; } +static int bypass_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct soc_mixer_control *m = + (struct soc_mixer_control *)w->kcontrols->private_value; + struct twl4030_priv *twl4030 = w->codec->private_data; + unsigned char reg; + + reg = twl4030_read_reg_cache(w->codec, m->reg); + if (reg & (1 << m->shift)) + twl4030->bypass_state |= + (1 << (m->reg - TWL4030_REG_ARXL1_APGA_CTL)); + else + twl4030->bypass_state &= + ~(1 << (m->reg - TWL4030_REG_ARXL1_APGA_CTL)); + + if (w->codec->bias_level == SND_SOC_BIAS_STANDBY) { + if (twl4030->bypass_state) + twl4030_codec_mute(w->codec, 0); + else + twl4030_codec_mute(w->codec, 1); + } + return 0; +} + /* * Some of the gain controls in TWL (mostly those which are associated with * the outputs) are implemented in an interesting way: @@ -775,13 +902,13 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { /* DACs */ SND_SOC_DAPM_DAC("DAC Right1", "Right Front Playback", - TWL4030_REG_AVDAC_CTL, 0, 0), + SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC("DAC Left1", "Left Front Playback", - TWL4030_REG_AVDAC_CTL, 1, 0), + SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC("DAC Right2", "Right Rear Playback", - TWL4030_REG_AVDAC_CTL, 2, 0), + SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC("DAC Left2", "Left Rear Playback", - TWL4030_REG_AVDAC_CTL, 3, 0), + SND_SOC_NOPM, 0, 0), /* Analog PGAs */ SND_SOC_DAPM_PGA("ARXR1_APGA", TWL4030_REG_ARXR1_APGA_CTL, @@ -793,6 +920,29 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_PGA("ARXL2_APGA", TWL4030_REG_ARXL2_APGA_CTL, 0, 0, NULL, 0), + /* Analog bypasses */ + SND_SOC_DAPM_SWITCH_E("Right1 Analog Loopback", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_abypassr1_control, bypass_event, + SND_SOC_DAPM_POST_REG), + SND_SOC_DAPM_SWITCH_E("Left1 Analog Loopback", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_abypassl1_control, + bypass_event, SND_SOC_DAPM_POST_REG), + SND_SOC_DAPM_SWITCH_E("Right2 Analog Loopback", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_abypassr2_control, + bypass_event, SND_SOC_DAPM_POST_REG), + SND_SOC_DAPM_SWITCH_E("Left2 Analog Loopback", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_abypassl2_control, + bypass_event, SND_SOC_DAPM_POST_REG), + + SND_SOC_DAPM_MIXER("Analog R1 Playback Mixer", TWL4030_REG_AVDAC_CTL, + 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Analog L1 Playback Mixer", TWL4030_REG_AVDAC_CTL, + 1, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Analog R2 Playback Mixer", TWL4030_REG_AVDAC_CTL, + 2, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Analog L2 Playback Mixer", TWL4030_REG_AVDAC_CTL, + 3, 0, NULL, 0), + /* Output MUX controls */ /* Earpiece */ SND_SOC_DAPM_VALUE_MUX("Earpiece Mux", SND_SOC_NOPM, 0, 0, @@ -863,13 +1013,19 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_MICBIAS("Mic Bias 1", TWL4030_REG_MICBIAS_CTL, 0, 0), SND_SOC_DAPM_MICBIAS("Mic Bias 2", TWL4030_REG_MICBIAS_CTL, 1, 0), SND_SOC_DAPM_MICBIAS("Headset Mic Bias", TWL4030_REG_MICBIAS_CTL, 2, 0), + }; static const struct snd_soc_dapm_route intercon[] = { - {"ARXL1_APGA", NULL, "DAC Left1"}, - {"ARXR1_APGA", NULL, "DAC Right1"}, - {"ARXL2_APGA", NULL, "DAC Left2"}, - {"ARXR2_APGA", NULL, "DAC Right2"}, + {"Analog L1 Playback Mixer", NULL, "DAC Left1"}, + {"Analog R1 Playback Mixer", NULL, "DAC Right1"}, + {"Analog L2 Playback Mixer", NULL, "DAC Left2"}, + {"Analog R2 Playback Mixer", NULL, "DAC Right2"}, + + {"ARXL1_APGA", NULL, "Analog L1 Playback Mixer"}, + {"ARXR1_APGA", NULL, "Analog R1 Playback Mixer"}, + {"ARXL2_APGA", NULL, "Analog L2 Playback Mixer"}, + {"ARXR2_APGA", NULL, "Analog R2 Playback Mixer"}, /* Internal playback routings */ /* Earpiece */ @@ -951,6 +1107,17 @@ static const struct snd_soc_dapm_route intercon[] = { {"ADC Virtual Left2", NULL, "TX2 Capture Route"}, {"ADC Virtual Right2", NULL, "TX2 Capture Route"}, + /* Analog bypass routes */ + {"Right1 Analog Loopback", "Switch", "Analog Right Capture Route"}, + {"Left1 Analog Loopback", "Switch", "Analog Left Capture Route"}, + {"Right2 Analog Loopback", "Switch", "Analog Right Capture Route"}, + {"Left2 Analog Loopback", "Switch", "Analog Left Capture Route"}, + + {"Analog R1 Playback Mixer", NULL, "Right1 Analog Loopback"}, + {"Analog L1 Playback Mixer", NULL, "Left1 Analog Loopback"}, + {"Analog R2 Playback Mixer", NULL, "Right2 Analog Loopback"}, + {"Analog L2 Playback Mixer", NULL, "Left2 Analog Loopback"}, + }; static int twl4030_add_widgets(struct snd_soc_codec *codec) @@ -967,16 +1134,25 @@ static int twl4030_add_widgets(struct snd_soc_codec *codec) static int twl4030_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct twl4030_priv *twl4030 = codec->private_data; + switch (level) { case SND_SOC_BIAS_ON: - twl4030_power_up(codec); + twl4030_codec_mute(codec, 0); break; case SND_SOC_BIAS_PREPARE: - /* TODO: develop a twl4030_prepare function */ + twl4030_power_up(codec); + if (twl4030->bypass_state) + twl4030_codec_mute(codec, 0); + else + twl4030_codec_mute(codec, 1); break; case SND_SOC_BIAS_STANDBY: - /* TODO: develop a twl4030_standby function */ - twl4030_power_down(codec); + twl4030_power_up(codec); + if (twl4030->bypass_state) + twl4030_codec_mute(codec, 0); + else + twl4030_codec_mute(codec, 1); break; case SND_SOC_BIAS_OFF: twl4030_power_down(codec); @@ -996,7 +1172,6 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = socdev->card->codec; u8 mode, old_mode, format, old_format; - /* bit rate */ old_mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE) & ~TWL4030_CODECPDZ; @@ -1038,6 +1213,7 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, if (mode != old_mode) { /* change rate and set CODECPDZ */ + twl4030_codec_enable(codec, 0); twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode); twl4030_codec_enable(codec, 1); } @@ -1258,11 +1434,19 @@ static int twl4030_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec; + struct twl4030_priv *twl4030; codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); if (codec == NULL) return -ENOMEM; + twl4030 = kzalloc(sizeof(struct twl4030_priv), GFP_KERNEL); + if (twl4030 == NULL) { + kfree(codec); + return -ENOMEM; + } + + codec->private_data = twl4030; socdev->card->codec = codec; mutex_init(&codec->mutex); INIT_LIST_HEAD(&codec->dapm_widgets); @@ -1280,8 +1464,10 @@ static int twl4030_remove(struct platform_device *pdev) struct snd_soc_codec *codec = socdev->card->codec; printk(KERN_INFO "TWL4030 Audio Codec remove\n"); + twl4030_set_bias_level(codec, SND_SOC_BIAS_OFF); snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); + kfree(codec->private_data); kfree(codec); return 0; diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index 442e5a82861..33dbb144dad 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -170,6 +170,9 @@ #define TWL4030_CLK256FS_EN 0x02 #define TWL4030_AIF_EN 0x01 +/* EAR_CTL (0x21) */ +#define TWL4030_EAR_GAIN 0x30 + /* HS_GAIN_SET (0x23) Fields */ #define TWL4030_HSR_GAIN 0x0C @@ -198,6 +201,18 @@ #define TWL4030_RAMP_DELAY_2581MS 0x1C #define TWL4030_RAMP_EN 0x02 +/* PREDL_CTL (0x25) */ +#define TWL4030_PREDL_GAIN 0x30 + +/* PREDR_CTL (0x26) */ +#define TWL4030_PREDR_GAIN 0x30 + +/* PRECKL_CTL (0x27) */ +#define TWL4030_PRECKL_GAIN 0x30 + +/* PRECKR_CTL (0x28) */ +#define TWL4030_PRECKR_GAIN 0x30 + /* HFL_CTL (0x29, 0x2A) Fields */ #define TWL4030_HF_CTL_HB_EN 0x04 #define TWL4030_HF_CTL_LOOP_EN 0x08 -- cgit v1.2.3-70-g09d2 From b98b7b347eed333d6fa2f74770beb8106e576cc6 Mon Sep 17 00:00:00 2001 From: Herton Ronaldo Krzesinski Date: Thu, 29 Jan 2009 13:18:31 -0200 Subject: ALSA: hda - make alc882_auto_init_input_src aware of selectors In the case of having a selector instead of mixer while initing input sources, the case that happens with ALC889, we must select instead of muting input. Problem was found while testing with hda-emu. Signed-off-by: Herton Ronaldo Krzesinski Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 21 ++++++++++++++++++--- 1 file changed, 18 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d81cb5eb8c5..3666cc5dc3b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6924,18 +6924,21 @@ static void alc882_auto_init_analog_input(struct hda_codec *codec) static void alc882_auto_init_input_src(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - const struct hda_input_mux *imux = spec->input_mux; int c; for (c = 0; c < spec->num_adc_nids; c++) { hda_nid_t conn_list[HDA_MAX_NUM_INPUTS]; hda_nid_t nid = spec->capsrc_nids[c]; + unsigned int mux_idx; + const struct hda_input_mux *imux; int conns, mute, idx, item; conns = snd_hda_get_connections(codec, nid, conn_list, ARRAY_SIZE(conn_list)); if (conns < 0) continue; + mux_idx = c >= spec->num_mux_defs ? 0 : c; + imux = &spec->input_mux[mux_idx]; for (idx = 0; idx < conns; idx++) { /* if the current connection is the selected one, * unmute it as default - otherwise mute it @@ -6948,8 +6951,20 @@ static void alc882_auto_init_input_src(struct hda_codec *codec) break; } } - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_AMP_GAIN_MUTE, mute); + /* check if we have a selector or mixer + * we could check for the widget type instead, but + * just check for Amp-In presence (in case of mixer + * without amp-in there is something wrong, this + * function shouldn't be used or capsrc nid is wrong) + */ + if (get_wcaps(codec, nid) & AC_WCAP_IN_AMP) + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + mute); + else if (mute != AMP_IN_MUTE(idx)) + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_CONNECT_SEL, + idx); } } } -- cgit v1.2.3-70-g09d2 From 04eb093c7c81d118efeb96228f69bc0179f71897 Mon Sep 17 00:00:00 2001 From: Timur Tabi Date: Thu, 29 Jan 2009 14:28:37 -0600 Subject: ASoC: fix initialization order of the CS4270 codec driver ASoC codec drivers typically serve two masters: the I2C bus and ASoC itself. When a codec driver registers with ASoC, a probe function is called. Most codec drivers call ASoC first, and then register with the I2C bus in the ASoC probe function. However, in order to support multiple codecs on one board, it's easier if the codec driver is probed via the I2C bus first. This is because the call to i2c_add_driver() can result in the I2C probe function being called multiple times - once for each codec. In the current design, the driver registers once with ASoC, and in the ASoC probe function, it calls i2c_add_driver(). The results in the I2C probe function being called multiple times before the driver can register with ASoC again. The new design has the driver call i2c_add_driver() first. In the I2C probe function, the driver registers with ASoC. This allows the ASoC probe function to be called once per I2C device. Also add code to check if the I2C probe function is called more than once, since that is not supported with the current ASoC design. Signed-off-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/codecs/cs4270.c | 177 +++++++++++++++++++++++++--------------------- 1 file changed, 97 insertions(+), 80 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 21253b48289..adc1150ddb0 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -490,21 +490,17 @@ static const struct snd_kcontrol_new cs4270_snd_controls[] = { }; /* - * Global variable to store socdev for i2c probe function. + * Global variable to store codec for the ASoC probe function. * * If struct i2c_driver had a private_data field, we wouldn't need to use - * cs4270_socdec. This is the only way to pass the socdev structure to - * cs4270_i2c_probe(). - * - * The real solution to cs4270_socdev is to create a mechanism - * that maps I2C addresses to snd_soc_device structures. Perhaps the - * creation of the snd_soc_device object should be moved out of - * cs4270_probe() and into cs4270_i2c_probe(), but that would make this - * driver dependent on I2C. The CS4270 supports "stand-alone" mode, whereby - * the chip is *not* connected to the I2C bus, but is instead configured via - * input pins. + * cs4270_codec. This is the only way to pass the codec structure from + * cs4270_i2c_probe() to cs4270_probe(). Unfortunately, there is no good + * way to synchronize these two functions. cs4270_i2c_probe() can be called + * multiple times before cs4270_probe() is called even once. So for now, we + * also only allow cs4270_i2c_probe() to be run once. That means that we do + * not support more than one cs4270 device in the system, at least for now. */ -static struct snd_soc_device *cs4270_socdev; +static struct snd_soc_codec *cs4270_codec; struct snd_soc_dai cs4270_dai = { .name = "cs4270", @@ -531,6 +527,70 @@ struct snd_soc_dai cs4270_dai = { }; EXPORT_SYMBOL_GPL(cs4270_dai); +/* + * ASoC probe function + */ +static int cs4270_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = cs4270_codec; + unsigned int i; + int ret; + + /* Connect the codec to the socdev. snd_soc_new_pcms() needs this. */ + socdev->card->codec = codec; + + /* Register PCMs */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "cs4270: failed to create PCMs\n"); + return ret; + } + + /* Add the non-DAPM controls */ + for (i = 0; i < ARRAY_SIZE(cs4270_snd_controls); i++) { + struct snd_kcontrol *kctrl; + + kctrl = snd_soc_cnew(&cs4270_snd_controls[i], codec, NULL); + if (!kctrl) { + printk(KERN_ERR "cs4270: error creating control '%s'\n", + cs4270_snd_controls[i].name); + ret = -ENOMEM; + goto error_free_pcms; + } + + ret = snd_ctl_add(codec->card, kctrl); + if (ret < 0) { + printk(KERN_ERR "cs4270: error adding control '%s'\n", + cs4270_snd_controls[i].name); + goto error_free_pcms; + } + } + + /* And finally, register the socdev */ + ret = snd_soc_init_card(socdev); + if (ret < 0) { + printk(KERN_ERR "cs4270: failed to register card\n"); + goto error_free_pcms; + } + + return 0; + +error_free_pcms: + snd_soc_free_pcms(socdev); + + return ret; +} + +static int cs4270_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + + return 0; +}; + /* * Initialize the I2C interface of the CS4270 * @@ -543,17 +603,27 @@ EXPORT_SYMBOL_GPL(cs4270_dai); static int cs4270_i2c_probe(struct i2c_client *i2c_client, const struct i2c_device_id *id) { - struct snd_soc_device *socdev = cs4270_socdev; struct snd_soc_codec *codec; struct cs4270_private *cs4270; - int i; - int ret = 0; + int ret; + + /* For now, we only support one cs4270 device in the system. See the + * comment for cs4270_codec. + */ + if (cs4270_codec) { + printk(KERN_ERR "cs4270: ignoring CS4270 at addr %X\n", + i2c_client->addr); + printk(KERN_ERR "cs4270: only one CS4270 per board allowed\n"); + /* Should we return something other than ENODEV here? */ + return -ENODEV; + } /* Verify that we have a CS4270 */ ret = i2c_smbus_read_byte_data(i2c_client, CS4270_CHIPID); if (ret < 0) { - printk(KERN_ERR "cs4270: failed to read I2C\n"); + printk(KERN_ERR "cs4270: failed to read I2C at addr %X\n", + i2c_client->addr); return ret; } /* The top four bits of the chip ID should be 1100. */ @@ -575,7 +645,7 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client, return -ENOMEM; } codec = &cs4270->codec; - socdev->card->codec = codec; + cs4270_codec = codec; mutex_init(&codec->mutex); INIT_LIST_HEAD(&codec->dapm_widgets); @@ -600,50 +670,20 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client, goto error_free_codec; } - /* Register PCMs */ - - ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + /* Register the DAI. If all the other ASoC driver have already + * registered, then this will call our probe function, so + * cs4270_codec needs to be ready. + */ + ret = snd_soc_register_dai(&cs4270_dai); if (ret < 0) { - printk(KERN_ERR "cs4270: failed to create PCMs\n"); + printk(KERN_ERR "cs4270: failed to register DAIe\n"); goto error_free_codec; } - /* Add the non-DAPM controls */ - - for (i = 0; i < ARRAY_SIZE(cs4270_snd_controls); i++) { - struct snd_kcontrol *kctrl; - - kctrl = snd_soc_cnew(&cs4270_snd_controls[i], codec, NULL); - if (!kctrl) { - printk(KERN_ERR "cs4270: error creating control '%s'\n", - cs4270_snd_controls[i].name); - ret = -ENOMEM; - goto error_free_pcms; - } - - ret = snd_ctl_add(codec->card, kctrl); - if (ret < 0) { - printk(KERN_ERR "cs4270: error adding control '%s'\n", - cs4270_snd_controls[i].name); - goto error_free_pcms; - } - } - - /* Initialize the SOC device */ - - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "cs4270: failed to register card\n"); - goto error_free_pcms;; - } - - i2c_set_clientdata(i2c_client, socdev); + i2c_set_clientdata(i2c_client, cs4270); return 0; -error_free_pcms: - snd_soc_free_pcms(socdev); - error_free_codec: kfree(cs4270); @@ -652,11 +692,8 @@ error_free_codec: static int cs4270_i2c_remove(struct i2c_client *i2c_client) { - struct snd_soc_device *socdev = i2c_get_clientdata(i2c_client); - struct snd_soc_codec *codec = socdev->card->codec; - struct cs4270_private *cs4270 = codec->private_data; + struct cs4270_private *cs4270 = i2c_get_clientdata(i2c_client); - snd_soc_free_pcms(socdev); kfree(cs4270); return 0; @@ -678,26 +715,6 @@ static struct i2c_driver cs4270_i2c_driver = { .remove = cs4270_i2c_remove, }; -/* - * ASoC probe function - * - * This function is called when the machine driver calls - * platform_device_add(). - */ -static int cs4270_probe(struct platform_device *pdev) -{ - cs4270_socdev = platform_get_drvdata(pdev);; - - return i2c_add_driver(&cs4270_i2c_driver); -} - -static int cs4270_remove(struct platform_device *pdev) -{ - i2c_del_driver(&cs4270_i2c_driver); - - return 0; -} - /* * ASoC codec device structure * @@ -714,13 +731,13 @@ static int __init cs4270_init(void) { printk(KERN_INFO "Cirrus Logic CS4270 ALSA SoC Codec Driver\n"); - return snd_soc_register_dai(&cs4270_dai); + return i2c_add_driver(&cs4270_i2c_driver); } module_init(cs4270_init); static void __exit cs4270_exit(void) { - snd_soc_unregister_dai(&cs4270_dai); + i2c_del_driver(&cs4270_i2c_driver); } module_exit(cs4270_exit); -- cgit v1.2.3-70-g09d2 From 880abd42d0891635e988b0a2cfb0942cf79fa2c3 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Fri, 30 Jan 2009 19:20:29 +0100 Subject: ALSA: ess1688: fix OPL3 port setting The ess1688 driver uses the same port for PCM audio (SB compatible) and OPL3 synthesis. It is not always right so allow to choose a different port for OPL3 synthesis. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/es1688/es1688.c | 23 ++++++++++++++++------- 1 file changed, 16 insertions(+), 7 deletions(-) diff --git a/sound/isa/es1688/es1688.c b/sound/isa/es1688/es1688.c index b46377139cf..b0eb0cf6050 100644 --- a/sound/isa/es1688/es1688.c +++ b/sound/isa/es1688/es1688.c @@ -49,6 +49,7 @@ static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */ static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x220,0x240,0x260 */ +static long fm_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* Usually 0x388 */ static long mpu_port[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = -1}; static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 5,7,9,10 */ static int mpu_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 5,7,9,10 */ @@ -65,6 +66,8 @@ MODULE_PARM_DESC(port, "Port # for " CRD_NAME " driver."); module_param_array(mpu_port, long, NULL, 0444); MODULE_PARM_DESC(mpu_port, "MPU-401 port # for " CRD_NAME " driver."); module_param_array(irq, int, NULL, 0444); +module_param_array(fm_port, long, NULL, 0444); +MODULE_PARM_DESC(fm_port, "FM port # for ES1688 driver."); MODULE_PARM_DESC(irq, "IRQ # for " CRD_NAME " driver."); module_param_array(mpu_irq, int, NULL, 0444); MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for " CRD_NAME " driver."); @@ -143,13 +146,19 @@ static int __devinit snd_es1688_probe(struct device *dev, unsigned int n) sprintf(card->longname, "%s at 0x%lx, irq %i, dma %i", pcm->name, chip->port, chip->irq, chip->dma8); - if (snd_opl3_create(card, chip->port, chip->port + 2, - OPL3_HW_OPL3, 0, &opl3) < 0) - dev_warn(dev, "opl3 not detected at 0x%lx\n", chip->port); - else { - error = snd_opl3_hwdep_new(opl3, 0, 1, NULL); - if (error < 0) - goto out; + if (fm_port[n] == SNDRV_AUTO_PORT) + fm_port[n] = port[n]; /* share the same port */ + + if (fm_port[n] > 0) { + if (snd_opl3_create(card, fm_port[n], fm_port[n] + 2, + OPL3_HW_OPL3, 0, &opl3) < 0) + dev_warn(dev, + "opl3 not detected at 0x%lx\n", fm_port[n]); + else { + error = snd_opl3_hwdep_new(opl3, 0, 1, NULL); + if (error < 0) + goto out; + } } if (mpu_irq[n] >= 0 && mpu_irq[n] != SNDRV_AUTO_IRQ && -- cgit v1.2.3-70-g09d2 From 504a06d8b05cb5b214c9b97752d8451e88d9ef81 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 30 Jan 2009 19:59:10 +0100 Subject: ALSA: Add description of new fm_port option for snd-es1688 driver Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/ALSA-Configuration.txt | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 7134a8f7044..a763b76afe5 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -609,6 +609,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. Module for ESS AudioDrive ES-1688 and ES-688 sound cards. port - port # for ES-1688 chip (0x220,0x240,0x260) + fm_port - port # for OPL3 (option; share the same port as default) mpu_port - port # for MPU-401 port (0x300,0x310,0x320,0x330), -1 = disable (default) irq - IRQ # for ES-1688 chip (5,7,9,10) mpu_irq - IRQ # for MPU-401 port (5,7,9,10) -- cgit v1.2.3-70-g09d2 From ff7bf02f630ae93cad4feda0f6a5a19b25a5019a Mon Sep 17 00:00:00 2001 From: Timur Tabi Date: Fri, 30 Jan 2009 11:14:49 -0600 Subject: ASoC: fix documentation in CS4270 codec driver Spruce up the documentation in the CS4270 codec. Use kerneldoc where appropriate. Fix incorrect comments. Signed-off-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/codecs/cs4270.c | 163 +++++++++++++++++++++++++++++----------------- 1 file changed, 105 insertions(+), 58 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index adc1150ddb0..e5f5afdd342 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -3,10 +3,10 @@ * * Author: Timur Tabi * - * Copyright 2007 Freescale Semiconductor, Inc. This file is licensed under - * the terms of the GNU General Public License version 2. This program - * is licensed "as is" without any warranty of any kind, whether express - * or implied. + * Copyright 2007-2009 Freescale Semiconductor, Inc. This file is licensed + * under the terms of the GNU General Public License version 2. This + * program is licensed "as is" without any warranty of any kind, whether + * express or implied. * * This is an ASoC device driver for the Cirrus Logic CS4270 codec. * @@ -111,8 +111,13 @@ struct cs4270_private { unsigned int mode; /* The mode (I2S or left-justified) */ }; -/* - * Clock Ratio Selection for Master Mode with I2C enabled +/** + * struct cs4270_mode_ratios - clock ratio tables + * @ratio: the ratio of MCLK to the sample rate + * @speed_mode: the Speed Mode bits to set in the Mode Control register for + * this ratio + * @mclk: the Ratio Select bits to set in the Mode Control register for this + * ratio * * The data for this chart is taken from Table 5 of the CS4270 reference * manual. @@ -121,31 +126,30 @@ struct cs4270_private { * It is also used by cs4270_set_dai_sysclk() to tell ALSA which sampling * rates the CS4270 currently supports. * - * Each element in this array corresponds to the ratios in mclk_ratios[]. - * These two arrays need to be in sync. - * - * 'speed_mode' is the corresponding bit pattern to be written to the + * @speed_mode is the corresponding bit pattern to be written to the * MODE bits of the Mode Control Register * - * 'mclk' is the corresponding bit pattern to be wirten to the MCLK bits of + * @mclk is the corresponding bit pattern to be wirten to the MCLK bits of * the Mode Control Register. * * In situations where a single ratio is represented by multiple speed * modes, we favor the slowest speed. E.g, for a ratio of 128, we pick * double-speed instead of quad-speed. However, the CS4270 errata states - * that Divide-By-1.5 can cause failures, so we avoid that mode where + * that divide-By-1.5 can cause failures, so we avoid that mode where * possible. * - * ERRATA: There is an errata for the CS4270 where divide-by-1.5 does not - * work if VD = 3.3V. If this effects you, select the + * Errata: There is an errata for the CS4270 where divide-by-1.5 does not + * work if Vd is 3.3V. If this effects you, select the * CONFIG_SND_SOC_CS4270_VD33_ERRATA Kconfig option, and the driver will * never select any sample rates that require divide-by-1.5. */ -static struct { +struct cs4270_mode_ratios { unsigned int ratio; u8 speed_mode; u8 mclk; -} cs4270_mode_ratios[] = { +}; + +static struct cs4270_mode_ratios[] = { {64, CS4270_MODE_4X, CS4270_MODE_DIV1}, #ifndef CONFIG_SND_SOC_CS4270_VD33_ERRATA {96, CS4270_MODE_4X, CS4270_MODE_DIV15}, @@ -162,34 +166,27 @@ static struct { /* The number of MCLK/LRCK ratios supported by the CS4270 */ #define NUM_MCLK_RATIOS ARRAY_SIZE(cs4270_mode_ratios) -/* - * Determine the CS4270 samples rates. +/** + * cs4270_set_dai_sysclk - determine the CS4270 samples rates. + * @codec_dai: the codec DAI + * @clk_id: the clock ID (ignored) + * @freq: the MCLK input frequency + * @dir: the clock direction (ignored) * - * 'freq' is the input frequency to MCLK. The other parameters are ignored. + * This function is used to tell the codec driver what the input MCLK + * frequency is. * * The value of MCLK is used to determine which sample rates are supported * by the CS4270. The ratio of MCLK / Fs must be equal to one of nine - * support values: 64, 96, 128, 192, 256, 384, 512, 768, and 1024. + * supported values - 64, 96, 128, 192, 256, 384, 512, 768, and 1024. * * This function calculates the nine ratios and determines which ones match * a standard sample rate. If there's a match, then it is added to the list - * of support sample rates. + * of supported sample rates. * * This function must be called by the machine driver's 'startup' function, * otherwise the list of supported sample rates will not be available in * time for ALSA. - * - * Note that in stand-alone mode, the sample rate is determined by input - * pins M0, M1, MDIV1, and MDIV2. Also in stand-alone mode, divide-by-3 - * is not a programmable option. However, divide-by-3 is not an available - * option in stand-alone mode. This cases two problems: a ratio of 768 is - * not available (it requires divide-by-3) and B) ratios 192 and 384 can - * only be selected with divide-by-1.5, but there is an errate that make - * this selection difficult. - * - * In addition, there is no mechanism for communicating with the machine - * driver what the input settings can be. This would need to be implemented - * for stand-alone mode to work. */ static int cs4270_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) @@ -230,8 +227,10 @@ static int cs4270_set_dai_sysclk(struct snd_soc_dai *codec_dai, return 0; } -/* - * Configure the codec for the selected audio format +/** + * cs4270_set_dai_fmt - configure the codec for the selected audio format + * @codec_dai: the codec DAI + * @format: a SND_SOC_DAIFMT_x value indicating the data format * * This function takes a bitmask of SND_SOC_DAIFMT_x bits and programs the * codec accordingly. @@ -261,8 +260,16 @@ static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai, return ret; } -/* - * Pre-fill the CS4270 register cache. +/** + * cs4270_fill_cache - pre-fill the CS4270 register cache. + * @codec: the codec for this CS4270 + * + * This function fills in the CS4270 register cache by reading the register + * values from the hardware. + * + * This CS4270 registers are cached to avoid excessive I2C I/O operations. + * After the initial read to pre-fill the cache, the CS4270 never updates + * the register values, so we won't have a cache coherency problem. * * We use the auto-increment feature of the CS4270 to read all registers in * one shot. @@ -285,12 +292,17 @@ static int cs4270_fill_cache(struct snd_soc_codec *codec) return 0; } -/* - * Read from the CS4270 register cache. +/** + * cs4270_read_reg_cache - read from the CS4270 register cache. + * @codec: the codec for this CS4270 + * @reg: the register to read + * + * This function returns the value for a given register. It reads only from + * the register cache, not the hardware itself. * * This CS4270 registers are cached to avoid excessive I2C I/O operations. * After the initial read to pre-fill the cache, the CS4270 never updates - * the register values, so we won't have a cache coherncy problem. + * the register values, so we won't have a cache coherency problem. */ static unsigned int cs4270_read_reg_cache(struct snd_soc_codec *codec, unsigned int reg) @@ -303,8 +315,11 @@ static unsigned int cs4270_read_reg_cache(struct snd_soc_codec *codec, return cache[reg - CS4270_FIRSTREG]; } -/* - * Write to a CS4270 register via the I2C bus. +/** + * cs4270_i2c_write - write to a CS4270 register via the I2C bus. + * @codec: the codec for this CS4270 + * @reg: the register to write + * @value: the value to write to the register * * This function writes the given value to the given CS4270 register, and * also updates the register cache. @@ -336,11 +351,17 @@ static int cs4270_i2c_write(struct snd_soc_codec *codec, unsigned int reg, return 0; } -/* - * Program the CS4270 with the given hardware parameters. +/** + * cs4270_hw_params - program the CS4270 with the given hardware parameters. + * @substream: the audio stream + * @params: the hardware parameters to set + * @dai: the SOC DAI (ignored) * - * The .ops functions are used to provide board-specific data, like - * input frequencies, to this driver. This function takes that information, + * This function programs the hardware with the values provided. + * Specifically, the sample rate and the data format. + * + * The .ops functions are used to provide board-specific data, like input + * frequencies, to this driver. This function takes that information, * combines it with the hardware parameters provided, and programs the * hardware accordingly. */ @@ -455,8 +476,10 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, } #ifdef CONFIG_SND_SOC_CS4270_HWMUTE -/* - * Set the CS4270 external mute +/** + * cs4270_mute - enable/disable the CS4270 external mute + * @dai: the SOC DAI + * @mute: 0 = disable mute, 1 = enable mute * * This function toggles the mute bits in the MUTE register. The CS4270's * mute capability is intended for external muting circuitry, so if the @@ -490,7 +513,7 @@ static const struct snd_kcontrol_new cs4270_snd_controls[] = { }; /* - * Global variable to store codec for the ASoC probe function. + * cs4270_codec - global variable to store codec for the ASoC probe function * * If struct i2c_driver had a private_data field, we wouldn't need to use * cs4270_codec. This is the only way to pass the codec structure from @@ -527,8 +550,12 @@ struct snd_soc_dai cs4270_dai = { }; EXPORT_SYMBOL_GPL(cs4270_dai); -/* - * ASoC probe function +/** + * cs4270_probe - ASoC probe function + * @pdev: platform device + * + * This function is called when ASoC has all the pieces it needs to + * instantiate a sound driver. */ static int cs4270_probe(struct platform_device *pdev) { @@ -582,6 +609,12 @@ error_free_pcms: return ret; } +/** + * cs4270_remove - ASoC remove function + * @pdev: platform device + * + * This function is the counterpart to cs4270_probe(). + */ static int cs4270_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); @@ -591,14 +624,13 @@ static int cs4270_remove(struct platform_device *pdev) return 0; }; -/* - * Initialize the I2C interface of the CS4270 - * - * This function is called for whenever the I2C subsystem finds a device - * at a particular address. +/** + * cs4270_i2c_probe - initialize the I2C interface of the CS4270 + * @i2c_client: the I2C client object + * @id: the I2C device ID (ignored) * - * Note: snd_soc_new_pcms() must be called before this function can be called, - * because of snd_ctl_add(). + * This function is called whenever the I2C subsystem finds a device that + * matches the device ID given via a prior call to i2c_add_driver(). */ static int cs4270_i2c_probe(struct i2c_client *i2c_client, const struct i2c_device_id *id) @@ -690,6 +722,12 @@ error_free_codec: return ret; } +/** + * cs4270_i2c_remove - remove an I2C device + * @i2c_client: the I2C client object + * + * This function is the counterpart to cs4270_i2c_probe(). + */ static int cs4270_i2c_remove(struct i2c_client *i2c_client) { struct cs4270_private *cs4270 = i2c_get_clientdata(i2c_client); @@ -699,12 +737,21 @@ static int cs4270_i2c_remove(struct i2c_client *i2c_client) return 0; } +/* + * cs4270_id - I2C device IDs supported by this driver + */ static struct i2c_device_id cs4270_id[] = { {"cs4270", 0}, {} }; MODULE_DEVICE_TABLE(i2c, cs4270_id); +/* + * cs4270_i2c_driver - I2C device identification + * + * This structure tells the I2C subsystem how to identify and support a + * given I2C device type. + */ static struct i2c_driver cs4270_i2c_driver = { .driver = { .name = "cs4270", -- cgit v1.2.3-70-g09d2 From 8f008062943c8565e855dda8a6681f641d7e71f9 Mon Sep 17 00:00:00 2001 From: Grazvydas Ignotas Date: Sat, 31 Jan 2009 16:29:24 +0200 Subject: ASoC: Update OMAP3 pandora board file Update pandora board file for recent TWL4030 codec changes. Also move output related snd_soc_dapm_nc_pin() calls to omap3pandora_out_init(), where they belong. Signed-off-by: Grazvydas Ignotas Signed-off-by: Mark Brown --- sound/soc/omap/omap3pandora.c | 49 +++++++++++++++++++++++++------------------ 1 file changed, 29 insertions(+), 20 deletions(-) diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index fcc2f5d9a87..fe282d4ef42 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -143,7 +143,7 @@ static const struct snd_soc_dapm_widget omap3pandora_out_dapm_widgets[] = { }; static const struct snd_soc_dapm_widget omap3pandora_in_dapm_widgets[] = { - SND_SOC_DAPM_MIC("Mic (Internal)", NULL), + SND_SOC_DAPM_MIC("Mic (internal)", NULL), SND_SOC_DAPM_MIC("Mic (external)", NULL), SND_SOC_DAPM_LINE("Line In", NULL), }; @@ -155,16 +155,33 @@ static const struct snd_soc_dapm_route omap3pandora_out_map[] = { }; static const struct snd_soc_dapm_route omap3pandora_in_map[] = { - {"INL", NULL, "Line In"}, - {"INR", NULL, "Line In"}, - {"INL", NULL, "Mic (Internal)"}, - {"INR", NULL, "Mic (external)"}, + {"AUXL", NULL, "Line In"}, + {"AUXR", NULL, "Line In"}, + + {"MAINMIC", NULL, "Mic Bias 1"}, + {"Mic Bias 1", NULL, "Mic (internal)"}, + + {"SUBMIC", NULL, "Mic Bias 2"}, + {"Mic Bias 2", NULL, "Mic (external)"}, }; static int omap3pandora_out_init(struct snd_soc_codec *codec) { int ret; + /* All TWL4030 output pins are floating */ + snd_soc_dapm_nc_pin(codec, "OUTL"); + snd_soc_dapm_nc_pin(codec, "OUTR"); + snd_soc_dapm_nc_pin(codec, "EARPIECE"); + snd_soc_dapm_nc_pin(codec, "PREDRIVEL"); + snd_soc_dapm_nc_pin(codec, "PREDRIVER"); + snd_soc_dapm_nc_pin(codec, "HSOL"); + snd_soc_dapm_nc_pin(codec, "HSOR"); + snd_soc_dapm_nc_pin(codec, "CARKITL"); + snd_soc_dapm_nc_pin(codec, "CARKITR"); + snd_soc_dapm_nc_pin(codec, "HFL"); + snd_soc_dapm_nc_pin(codec, "HFR"); + ret = snd_soc_dapm_new_controls(codec, omap3pandora_out_dapm_widgets, ARRAY_SIZE(omap3pandora_out_dapm_widgets)); if (ret < 0) @@ -180,18 +197,11 @@ static int omap3pandora_in_init(struct snd_soc_codec *codec) { int ret; - /* All TWL4030 output pins are floating */ - snd_soc_dapm_nc_pin(codec, "OUTL"), - snd_soc_dapm_nc_pin(codec, "OUTR"), - snd_soc_dapm_nc_pin(codec, "EARPIECE"), - snd_soc_dapm_nc_pin(codec, "PREDRIVEL"), - snd_soc_dapm_nc_pin(codec, "PREDRIVER"), - snd_soc_dapm_nc_pin(codec, "HSOL"), - snd_soc_dapm_nc_pin(codec, "HSOR"), - snd_soc_dapm_nc_pin(codec, "CARKITL"), - snd_soc_dapm_nc_pin(codec, "CARKITR"), - snd_soc_dapm_nc_pin(codec, "HFL"), - snd_soc_dapm_nc_pin(codec, "HFR"), + /* Not comnnected */ + snd_soc_dapm_nc_pin(codec, "HSMIC"); + snd_soc_dapm_nc_pin(codec, "CARKITMIC"); + snd_soc_dapm_nc_pin(codec, "DIGIMIC0"); + snd_soc_dapm_nc_pin(codec, "DIGIMIC1"); ret = snd_soc_dapm_new_controls(codec, omap3pandora_in_dapm_widgets, ARRAY_SIZE(omap3pandora_in_dapm_widgets)); @@ -251,10 +261,9 @@ static int __init omap3pandora_soc_init(void) { int ret; - if (!machine_is_omap3_pandora()) { - pr_debug(PREFIX "Not OMAP3 Pandora\n"); + if (!machine_is_omap3_pandora()) return -ENODEV; - } + pr_info("OMAP3 Pandora SoC init\n"); ret = gpio_request(OMAP3_PANDORA_DAC_POWER_GPIO, "dac_power"); -- cgit v1.2.3-70-g09d2 From d563ffa6b319a4e401d096db9014a947590ca081 Mon Sep 17 00:00:00 2001 From: Tim Blechmann Date: Sat, 31 Jan 2009 18:01:13 +0100 Subject: ALSA: pcxhr: fix trivial typo Signed-off-by: Tim Blechmann Signed-off-by: Takashi Iwai --- sound/pci/pcxhr/pcxhr_core.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/pcxhr/pcxhr_core.h b/sound/pci/pcxhr/pcxhr_core.h index bbbd66d13a6..be0173796cd 100644 --- a/sound/pci/pcxhr/pcxhr_core.h +++ b/sound/pci/pcxhr/pcxhr_core.h @@ -1,7 +1,7 @@ /* * Driver for Digigram pcxhr compatible soundcards * - * low level interface with interrupt ans message handling + * low level interface with interrupt and message handling * * Copyright (c) 2004 by Digigram * -- cgit v1.2.3-70-g09d2 From 5aa13a94098ef5fc1bb0a7f531fdda8864ae67ff Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Sun, 1 Feb 2009 21:13:15 +0100 Subject: ALSA: msnd: add module description and license for the snd-msnd-lib The missing module license generates warning during module loading. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/msnd/msnd.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/isa/msnd/msnd.c b/sound/isa/msnd/msnd.c index 264e08212c6..906454413ed 100644 --- a/sound/isa/msnd/msnd.c +++ b/sound/isa/msnd/msnd.c @@ -700,3 +700,6 @@ int snd_msnd_pcm(struct snd_card *card, int device, } EXPORT_SYMBOL(snd_msnd_pcm); +MODULE_DESCRIPTION("Common routines for Turtle Beach Multisound drivers"); +MODULE_LICENSE("GPL"); + -- cgit v1.2.3-70-g09d2 From e683ec4697c74c7d04ff8e90ec625ac34e25a7d8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 12 Nov 2008 16:42:44 +0100 Subject: ALSA: ice1724 - Dynamic MIDI TX irq control MIDI_TX IRQ seems always pending when any bytes on FIFO is available. Thus, it's better to enable MPU_TX only when any bytres are really stored in the substream, and disables immediately when the queue becomes empty. Signed-off-by: Takashi Iwai --- sound/pci/ice1712/ice1724.c | 43 +++++++++++++++++++++++++++---------------- 1 file changed, 27 insertions(+), 16 deletions(-) diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index bb8d8c766b9..eb7872dec5a 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -241,6 +241,8 @@ get_rawmidi_substream(struct snd_ice1712 *ice, unsigned int stream) struct snd_rawmidi_substream, list); } +static void enable_midi_irq(struct snd_ice1712 *ice, u8 flag, int enable); + static void vt1724_midi_write(struct snd_ice1712 *ice) { struct snd_rawmidi_substream *s; @@ -254,6 +256,11 @@ static void vt1724_midi_write(struct snd_ice1712 *ice) for (i = 0; i < count; ++i) outb(buffer[i], ICEREG1724(ice, MPU_DATA)); } + /* mask irq when all bytes have been transmitted. + * enabled again in output_trigger when the new data comes in. + */ + enable_midi_irq(ice, VT1724_IRQ_MPU_TX, + !snd_rawmidi_transmit_empty(s)); } static void vt1724_midi_read(struct snd_ice1712 *ice) @@ -272,31 +279,34 @@ static void vt1724_midi_read(struct snd_ice1712 *ice) } } -static void vt1724_enable_midi_irq(struct snd_rawmidi_substream *substream, - u8 flag, int enable) +/* call with ice->reg_lock */ +static void enable_midi_irq(struct snd_ice1712 *ice, u8 flag, int enable) { - struct snd_ice1712 *ice = substream->rmidi->private_data; - u8 mask; - - spin_lock_irq(&ice->reg_lock); - mask = inb(ICEREG1724(ice, IRQMASK)); + u8 mask = inb(ICEREG1724(ice, IRQMASK)); if (enable) mask &= ~flag; else mask |= flag; outb(mask, ICEREG1724(ice, IRQMASK)); +} + +static void vt1724_enable_midi_irq(struct snd_rawmidi_substream *substream, + u8 flag, int enable) +{ + struct snd_ice1712 *ice = substream->rmidi->private_data; + + spin_lock_irq(&ice->reg_lock); + enable_midi_irq(ice, flag, enable); spin_unlock_irq(&ice->reg_lock); } static int vt1724_midi_output_open(struct snd_rawmidi_substream *s) { - vt1724_enable_midi_irq(s, VT1724_IRQ_MPU_TX, 1); return 0; } static int vt1724_midi_output_close(struct snd_rawmidi_substream *s) { - vt1724_enable_midi_irq(s, VT1724_IRQ_MPU_TX, 0); return 0; } @@ -311,6 +321,7 @@ static void vt1724_midi_output_trigger(struct snd_rawmidi_substream *s, int up) vt1724_midi_write(ice); } else { ice->midi_output = 0; + enable_midi_irq(ice, VT1724_IRQ_MPU_TX, 0); } spin_unlock_irqrestore(&ice->reg_lock, flags); } @@ -320,6 +331,7 @@ static void vt1724_midi_output_drain(struct snd_rawmidi_substream *s) struct snd_ice1712 *ice = s->rmidi->private_data; unsigned long timeout; + vt1724_enable_midi_irq(s, VT1724_IRQ_MPU_TX, 0); /* 32 bytes should be transmitted in less than about 12 ms */ timeout = jiffies + msecs_to_jiffies(15); do { @@ -389,24 +401,24 @@ static irqreturn_t snd_vt1724_interrupt(int irq, void *dev_id) status &= status_mask; if (status == 0) break; + spin_lock(&ice->reg_lock); if (++timeout > 10) { status = inb(ICEREG1724(ice, IRQSTAT)); printk(KERN_ERR "ice1724: Too long irq loop, " "status = 0x%x\n", status); if (status & VT1724_IRQ_MPU_TX) { printk(KERN_ERR "ice1724: Disabling MPU_TX\n"); - outb(inb(ICEREG1724(ice, IRQMASK)) | - VT1724_IRQ_MPU_TX, - ICEREG1724(ice, IRQMASK)); + enable_midi_irq(ice, VT1724_IRQ_MPU_TX, 0); } + spin_unlock(&ice->reg_lock); break; } handled = 1; if (status & VT1724_IRQ_MPU_TX) { - spin_lock(&ice->reg_lock); if (ice->midi_output) vt1724_midi_write(ice); - spin_unlock(&ice->reg_lock); + else + enable_midi_irq(ice, VT1724_IRQ_MPU_TX, 0); /* Due to mysterical reasons, MPU_TX is always * generated (and can't be cleared) when a PCM * playback is going. So let's ignore at the @@ -415,15 +427,14 @@ static irqreturn_t snd_vt1724_interrupt(int irq, void *dev_id) status_mask &= ~VT1724_IRQ_MPU_TX; } if (status & VT1724_IRQ_MPU_RX) { - spin_lock(&ice->reg_lock); if (ice->midi_input) vt1724_midi_read(ice); else vt1724_midi_clear_rx(ice); - spin_unlock(&ice->reg_lock); } /* ack MPU irq */ outb(status, ICEREG1724(ice, IRQSTAT)); + spin_unlock(&ice->reg_lock); if (status & VT1724_IRQ_MTPCM) { /* * Multi-track PCM -- cgit v1.2.3-70-g09d2 From d9fb7fbddc9a14569aad517984c1a5b0b07002ea Mon Sep 17 00:00:00 2001 From: Timur Tabi Date: Mon, 2 Feb 2009 14:50:45 -0600 Subject: ASoC: fix build break in CS4270 codec driver Fix a oversight in the CS4270 codec driver that caused a build break. Signed-off-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/codecs/cs4270.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index e5f5afdd342..7962874258f 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -149,7 +149,7 @@ struct cs4270_mode_ratios { u8 mclk; }; -static struct cs4270_mode_ratios[] = { +static struct cs4270_mode_ratios cs4270_mode_ratios[] = { {64, CS4270_MODE_4X, CS4270_MODE_DIV1}, #ifndef CONFIG_SND_SOC_CS4270_VD33_ERRATA {96, CS4270_MODE_4X, CS4270_MODE_DIV15}, -- cgit v1.2.3-70-g09d2 From a6c255e0945160b76eabe983f59e2129e0c66246 Mon Sep 17 00:00:00 2001 From: Timur Tabi Date: Mon, 2 Feb 2009 15:08:29 -0600 Subject: ASoC: fix message display in CS4270 codec driver Replace printk calls with dev_xxx calls. Set the 'dev' field of the codec and codec_dai structures so that these calls work. Signed-off-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/codecs/cs4270.c | 66 ++++++++++++++++++++++++++++------------------- 1 file changed, 39 insertions(+), 27 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 7962874258f..2c79a24186f 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -212,7 +212,7 @@ static int cs4270_set_dai_sysclk(struct snd_soc_dai *codec_dai, rates &= ~SNDRV_PCM_RATE_KNOT; if (!rates) { - printk(KERN_ERR "cs4270: could not find a valid sample rate\n"); + dev_err(codec->dev, "could not find a valid sample rate\n"); return -EINVAL; } @@ -253,7 +253,7 @@ static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai, cs4270->mode = format & SND_SOC_DAIFMT_FORMAT_MASK; break; default: - printk(KERN_ERR "cs4270: invalid DAI format\n"); + dev_err(codec->dev, "invalid dai format\n"); ret = -EINVAL; } @@ -284,7 +284,7 @@ static int cs4270_fill_cache(struct snd_soc_codec *codec) CS4270_FIRSTREG | 0x80, CS4270_NUMREGS, cache); if (length != CS4270_NUMREGS) { - printk(KERN_ERR "cs4270: I2C read failure, addr=0x%x\n", + dev_err(codec->dev, "i2c read failure, addr=0x%x\n", i2c_client->addr); return -EIO; } @@ -340,7 +340,7 @@ static int cs4270_i2c_write(struct snd_soc_codec *codec, unsigned int reg, if (cache[reg - CS4270_FIRSTREG] != value) { struct i2c_client *client = codec->control_data; if (i2c_smbus_write_byte_data(client, reg, value)) { - printk(KERN_ERR "cs4270: I2C write failed\n"); + dev_err(codec->dev, "i2c write failed\n"); return -EIO; } @@ -391,7 +391,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, if (i == NUM_MCLK_RATIOS) { /* We did not find a matching ratio */ - printk(KERN_ERR "cs4270: could not find matching ratio\n"); + dev_err(codec->dev, "could not find matching ratio\n"); return -EINVAL; } @@ -401,7 +401,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, CS4270_PWRCTL_PDN_ADC | CS4270_PWRCTL_PDN_DAC | CS4270_PWRCTL_PDN); if (ret < 0) { - printk(KERN_ERR "cs4270: I2C write failed\n"); + dev_err(codec->dev, "i2c write failed\n"); return ret; } @@ -413,7 +413,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, ret = snd_soc_write(codec, CS4270_MODE, reg); if (ret < 0) { - printk(KERN_ERR "cs4270: I2C write failed\n"); + dev_err(codec->dev, "i2c write failed\n"); return ret; } @@ -430,13 +430,13 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, reg |= CS4270_FORMAT_DAC_LJ | CS4270_FORMAT_ADC_LJ; break; default: - printk(KERN_ERR "cs4270: unknown format\n"); + dev_err(codec->dev, "unknown dai format\n"); return -EINVAL; } ret = snd_soc_write(codec, CS4270_FORMAT, reg); if (ret < 0) { - printk(KERN_ERR "cs4270: I2C write failed\n"); + dev_err(codec->dev, "i2c write failed\n"); return ret; } @@ -447,7 +447,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, reg &= ~CS4270_MUTE_AUTO; ret = snd_soc_write(codec, CS4270_MUTE, reg); if (ret < 0) { - printk(KERN_ERR "cs4270: I2C write failed\n"); + dev_err(codec->dev, "i2c write failed\n"); return ret; } @@ -460,7 +460,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, reg &= ~(CS4270_TRANS_SOFT | CS4270_TRANS_ZERO); ret = cs4270_i2c_write(codec, CS4270_TRANS, reg); if (ret < 0) { - printk(KERN_ERR "I2C write failed\n"); + dev_err(codec->dev, "i2c write failed\n"); return ret; } @@ -468,7 +468,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, ret = snd_soc_write(codec, CS4270_PWRCTL, 0); if (ret < 0) { - printk(KERN_ERR "cs4270: I2C write failed\n"); + dev_err(codec->dev, "i2c write failed\n"); return ret; } @@ -570,7 +570,7 @@ static int cs4270_probe(struct platform_device *pdev) /* Register PCMs */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) { - printk(KERN_ERR "cs4270: failed to create PCMs\n"); + dev_err(codec->dev, "failed to create pcms\n"); return ret; } @@ -580,7 +580,7 @@ static int cs4270_probe(struct platform_device *pdev) kctrl = snd_soc_cnew(&cs4270_snd_controls[i], codec, NULL); if (!kctrl) { - printk(KERN_ERR "cs4270: error creating control '%s'\n", + dev_err(codec->dev, "error creating control '%s'\n", cs4270_snd_controls[i].name); ret = -ENOMEM; goto error_free_pcms; @@ -588,7 +588,7 @@ static int cs4270_probe(struct platform_device *pdev) ret = snd_ctl_add(codec->card, kctrl); if (ret < 0) { - printk(KERN_ERR "cs4270: error adding control '%s'\n", + dev_err(codec->dev, "error adding control '%s'\n", cs4270_snd_controls[i].name); goto error_free_pcms; } @@ -597,7 +597,7 @@ static int cs4270_probe(struct platform_device *pdev) /* And finally, register the socdev */ ret = snd_soc_init_card(socdev); if (ret < 0) { - printk(KERN_ERR "cs4270: failed to register card\n"); + dev_err(codec->dev, "failed to register card\n"); goto error_free_pcms; } @@ -643,9 +643,9 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client, * comment for cs4270_codec. */ if (cs4270_codec) { - printk(KERN_ERR "cs4270: ignoring CS4270 at addr %X\n", + dev_err(&i2c_client->dev, "ignoring CS4270 at addr %X\n", i2c_client->addr); - printk(KERN_ERR "cs4270: only one CS4270 per board allowed\n"); + dev_err(&i2c_client->dev, "only one per board allowed\n"); /* Should we return something other than ENODEV here? */ return -ENODEV; } @@ -654,35 +654,35 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client, ret = i2c_smbus_read_byte_data(i2c_client, CS4270_CHIPID); if (ret < 0) { - printk(KERN_ERR "cs4270: failed to read I2C at addr %X\n", + dev_err(&i2c_client->dev, "failed to read i2c at addr %X\n", i2c_client->addr); return ret; } /* The top four bits of the chip ID should be 1100. */ if ((ret & 0xF0) != 0xC0) { - printk(KERN_ERR "cs4270: device at addr %X is not a CS4270\n", + dev_err(&i2c_client->dev, "device at addr %X is not a CS4270\n", i2c_client->addr); return -ENODEV; } - printk(KERN_INFO "cs4270: found device at I2C address %X\n", + dev_info(&i2c_client->dev, "found device at i2c address %X\n", i2c_client->addr); - printk(KERN_INFO "cs4270: hardware revision %X\n", ret & 0xF); + dev_info(&i2c_client->dev, "hardware revision %X\n", ret & 0xF); /* Allocate enough space for the snd_soc_codec structure and our private data together. */ cs4270 = kzalloc(sizeof(struct cs4270_private), GFP_KERNEL); if (!cs4270) { - printk(KERN_ERR "cs4270: Could not allocate codec structure\n"); + dev_err(&i2c_client->dev, "could not allocate codec\n"); return -ENOMEM; } codec = &cs4270->codec; - cs4270_codec = codec; mutex_init(&codec->mutex); INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); + codec->dev = &i2c_client->dev; codec->name = "CS4270"; codec->owner = THIS_MODULE; codec->dai = &cs4270_dai; @@ -698,17 +698,25 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client, ret = cs4270_fill_cache(codec); if (ret < 0) { - printk(KERN_ERR "cs4270: failed to fill register cache\n"); + dev_err(&i2c_client->dev, "failed to fill register cache\n"); goto error_free_codec; } + /* Initialize the DAI. Normally, we'd prefer to have a kmalloc'd DAI + * structure for each CS4270 device, but the machine driver needs to + * have a pointer to the DAI structure, so for now it must be a global + * variable. + */ + cs4270_dai.dev = &i2c_client->dev; + /* Register the DAI. If all the other ASoC driver have already * registered, then this will call our probe function, so * cs4270_codec needs to be ready. */ + cs4270_codec = codec; ret = snd_soc_register_dai(&cs4270_dai); if (ret < 0) { - printk(KERN_ERR "cs4270: failed to register DAIe\n"); + dev_err(&i2c_client->dev, "failed to register DAIe\n"); goto error_free_codec; } @@ -718,6 +726,8 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client, error_free_codec: kfree(cs4270); + cs4270_codec = NULL; + cs4270_dai.dev = NULL; return ret; } @@ -733,6 +743,8 @@ static int cs4270_i2c_remove(struct i2c_client *i2c_client) struct cs4270_private *cs4270 = i2c_get_clientdata(i2c_client); kfree(cs4270); + cs4270_codec = NULL; + cs4270_dai.dev = NULL; return 0; } @@ -776,7 +788,7 @@ EXPORT_SYMBOL_GPL(soc_codec_device_cs4270); static int __init cs4270_init(void) { - printk(KERN_INFO "Cirrus Logic CS4270 ALSA SoC Codec Driver\n"); + pr_info("Cirrus Logic CS4270 ALSA SoC Codec Driver\n"); return i2c_add_driver(&cs4270_i2c_driver); } -- cgit v1.2.3-70-g09d2 From ba340e825f4b892782779abd0f93bcff7e763567 Mon Sep 17 00:00:00 2001 From: Tony Vroon Date: Mon, 2 Feb 2009 19:01:30 +0000 Subject: ALSA: hda - Add tyan model for Realtek ALC262 The Realtek ALC262 on the Tyan Thunder n6650W (S2915-E) mainboard has a rather odd configuration template. As a result, the white AUX connector can not be used. This rewrites the default config to more accurately reflect the connector layout, colour and function. Unfortunately the black CD_IN connector, which is suspected to be widget 0x1c refuses to switch into input (0x20), instead opting to remain on 0x0. As such, no mixer controls are exposed for it. Autoswitching is implemented between the front headphone output and back line output. Signed-off-by: Tony Vroon Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 77 +++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 77 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0c81d92c3d7..bd9ef336389 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -103,6 +103,7 @@ enum { ALC262_NEC, ALC262_TOSHIBA_S06, ALC262_TOSHIBA_RX1, + ALC262_TYAN, ALC262_AUTO, ALC262_MODEL_LAST /* last tag */ }; @@ -9509,6 +9510,67 @@ static struct snd_kcontrol_new alc262_benq_t31_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc262_tyan_mixer[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Master Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Aux Playback Volume", 0x0b, 0x06, HDA_INPUT), + HDA_CODEC_MUTE("Aux Playback Switch", 0x0b, 0x06, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), + { } /* end */ +}; + +static struct hda_verb alc262_tyan_verbs[] = { + /* Headphone automute */ + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, + + /* P11 AUX_IN, white 4-pin connector */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x14, AC_VERB_SET_CONFIG_DEFAULT_BYTES_1, 0xe1}, + {0x14, AC_VERB_SET_CONFIG_DEFAULT_BYTES_2, 0x93}, + {0x14, AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, 0x19}, + + {} +}; + +/* unsolicited event for HP jack sensing */ +static void alc262_tyan_automute(struct hda_codec *codec) +{ + unsigned int mute; + unsigned int present; + + snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_SET_PIN_SENSE, 0); + present = snd_hda_codec_read(codec, 0x1b, 0, + AC_VERB_GET_PIN_SENSE, 0); + present = (present & 0x80000000) != 0; + if (present) { + /* mute line output on ATX panel */ + snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, + HDA_AMP_MUTE, HDA_AMP_MUTE); + } else { + /* unmute line output if necessary */ + mute = snd_hda_codec_amp_read(codec, 0x1b, 0, HDA_OUTPUT, 0); + snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, + HDA_AMP_MUTE, mute); + } +} + +static void alc262_tyan_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + if ((res >> 26) != ALC880_HP_EVENT) + return; + alc262_tyan_automute(codec); +} + #define alc262_capture_mixer alc882_capture_mixer #define alc262_capture_alt_mixer alc882_capture_alt_mixer @@ -10626,6 +10688,7 @@ static const char *alc262_models[ALC262_MODEL_LAST] = { [ALC262_ULTRA] = "ultra", [ALC262_LENOVO_3000] = "lenovo-3000", [ALC262_NEC] = "nec", + [ALC262_TYAN] = "tyan", [ALC262_AUTO] = "auto", }; @@ -10666,6 +10729,7 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x1179, 0xff7b, "Toshiba S06", ALC262_TOSHIBA_S06), SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FUJITSU), SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FUJITSU), + SND_PCI_QUIRK(0x10f1, 0x2915, "Tyan Thunder n6650W", ALC262_TYAN), SND_PCI_QUIRK(0x144d, 0xc032, "Samsung Q1 Ultra", ALC262_ULTRA), SND_PCI_QUIRK(0x144d, 0xc039, "Samsung Q1U EL", ALC262_ULTRA), SND_PCI_QUIRK(0x144d, 0xc510, "Samsung Q45", ALC262_HIPPO), @@ -10884,6 +10948,19 @@ static struct alc_config_preset alc262_presets[] = { .unsol_event = alc262_hippo_unsol_event, .init_hook = alc262_hippo_automute, }, + [ALC262_TYAN] = { + .mixers = { alc262_tyan_mixer }, + .init_verbs = { alc262_init_verbs, alc262_tyan_verbs}, + .num_dacs = ARRAY_SIZE(alc262_dac_nids), + .dac_nids = alc262_dac_nids, + .hp_nid = 0x02, + .dig_out_nid = ALC262_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc262_modes), + .channel_mode = alc262_modes, + .input_mux = &alc262_capture_source, + .unsol_event = alc262_tyan_unsol_event, + .init_hook = alc262_tyan_automute, + }, }; static int patch_alc262(struct hda_codec *codec) -- cgit v1.2.3-70-g09d2 From 123848e77623b9996288e85433985439c157fcd0 Mon Sep 17 00:00:00 2001 From: Tony Vroon Date: Tue, 3 Feb 2009 11:13:34 +0000 Subject: ALSA: Document tyan model for Realtek ALC262 As just pointed out to me, the new tyan model for ALC262 was implemented but not documented. This adds the board to the list, using both its marketing name (Thunder n6650W) and its model number (S2915-E). Signed-off-by: Tony Vroon Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index c9df9db5835..8f40999a456 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -56,6 +56,7 @@ ALC262 sony-assamd Sony ASSAMD toshiba-s06 Toshiba S06 toshiba-rx1 Toshiba RX1 + tyan Tyan Thunder n6650W (S2915-E) ultra Samsung Q1 Ultra Vista model lenovo-3000 Lenovo 3000 y410 nec NEC Versa S9100 -- cgit v1.2.3-70-g09d2 From 111f6fbeb73fc350fe3a08c4ecd0ccdf3e13bef0 Mon Sep 17 00:00:00 2001 From: Vasily Khoruzhick Date: Tue, 3 Feb 2009 15:52:56 +0200 Subject: ASoC: Don't unconditionally use the PLL in UDA1380 Without this fix driver switches to WSPLL in uda1380_pcm_prepare even if SYSCLK was chosen (uda1380_pcm_prepare modifies UDA1380_CLK register to disable R00_DAC_CLK before flushing reg cache) Signed-off-by: Vasily Khoruzhick Signed-off-by: Mark Brown --- sound/soc/codecs/uda1380.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 98e4a6560f0..6e4a1770ce8 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -418,8 +418,8 @@ static int uda1380_pcm_prepare(struct snd_pcm_substream *substream, uda1380_write(codec, reg, uda1380_read_reg_cache(codec, reg)); } - /* FIXME enable DAC_CLK */ - uda1380_write(codec, UDA1380_CLK, clk | R00_DAC_CLK); + /* FIXME restore DAC_CLK */ + uda1380_write(codec, UDA1380_CLK, clk); return 0; } -- cgit v1.2.3-70-g09d2 From 5b2474425ed2a625b75dcd8d648701e473b7d764 Mon Sep 17 00:00:00 2001 From: Philipp Zabel Date: Tue, 3 Feb 2009 17:19:40 +0100 Subject: ASoC: uda1380: split set_dai_fmt into _both, _playback and _capture variants This patch splits set_dai_fmt into three variants (single interface, dual interface playback only, dual interface capture only) so that data input and output formats can be configured separately for dual interface setups. Signed-off-by: Philipp Zabel Tested-by: Vasily Khoruzhick Signed-off-by: Mark Brown --- sound/soc/codecs/uda1380.c | 68 +++++++++++++++++++++++++++++++++++++++++----- 1 file changed, 61 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 6e4a1770ce8..5242b8156b3 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -356,7 +356,7 @@ static int uda1380_add_widgets(struct snd_soc_codec *codec) return 0; } -static int uda1380_set_dai_fmt(struct snd_soc_dai *codec_dai, +static int uda1380_set_dai_fmt_both(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -366,16 +366,70 @@ static int uda1380_set_dai_fmt(struct snd_soc_dai *codec_dai, iface = uda1380_read_reg_cache(codec, UDA1380_IFACE); iface &= ~(R01_SFORI_MASK | R01_SIM | R01_SFORO_MASK); - /* FIXME: how to select I2S for DATAO and MSB for DATAI correctly? */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: iface |= R01_SFORI_I2S | R01_SFORO_I2S; break; case SND_SOC_DAIFMT_LSB: - iface |= R01_SFORI_LSB16 | R01_SFORO_I2S; + iface |= R01_SFORI_LSB16 | R01_SFORO_LSB16; break; case SND_SOC_DAIFMT_MSB: - iface |= R01_SFORI_MSB | R01_SFORO_I2S; + iface |= R01_SFORI_MSB | R01_SFORO_MSB; + } + + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) == SND_SOC_DAIFMT_CBM_CFM) + iface |= R01_SIM; + + uda1380_write(codec, UDA1380_IFACE, iface); + + return 0; +} + +static int uda1380_set_dai_fmt_playback(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + int iface; + + /* set up DAI based upon fmt */ + iface = uda1380_read_reg_cache(codec, UDA1380_IFACE); + iface &= ~R01_SFORI_MASK; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= R01_SFORI_I2S; + break; + case SND_SOC_DAIFMT_LSB: + iface |= R01_SFORI_LSB16; + break; + case SND_SOC_DAIFMT_MSB: + iface |= R01_SFORI_MSB; + } + + uda1380_write(codec, UDA1380_IFACE, iface); + + return 0; +} + +static int uda1380_set_dai_fmt_capture(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + int iface; + + /* set up DAI based upon fmt */ + iface = uda1380_read_reg_cache(codec, UDA1380_IFACE); + iface &= ~(R01_SIM | R01_SFORO_MASK); + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= R01_SFORO_I2S; + break; + case SND_SOC_DAIFMT_LSB: + iface |= R01_SFORO_LSB16; + break; + case SND_SOC_DAIFMT_MSB: + iface |= R01_SFORO_MSB; } if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) == SND_SOC_DAIFMT_CBM_CFM) @@ -549,7 +603,7 @@ struct snd_soc_dai uda1380_dai[] = { .shutdown = uda1380_pcm_shutdown, .prepare = uda1380_pcm_prepare, .digital_mute = uda1380_mute, - .set_fmt = uda1380_set_dai_fmt, + .set_fmt = uda1380_set_dai_fmt_both, }, }, { /* playback only - dual interface */ @@ -566,7 +620,7 @@ struct snd_soc_dai uda1380_dai[] = { .shutdown = uda1380_pcm_shutdown, .prepare = uda1380_pcm_prepare, .digital_mute = uda1380_mute, - .set_fmt = uda1380_set_dai_fmt, + .set_fmt = uda1380_set_dai_fmt_playback, }, }, { /* capture only - dual interface*/ @@ -582,7 +636,7 @@ struct snd_soc_dai uda1380_dai[] = { .hw_params = uda1380_pcm_hw_params, .shutdown = uda1380_pcm_shutdown, .prepare = uda1380_pcm_prepare, - .set_fmt = uda1380_set_dai_fmt, + .set_fmt = uda1380_set_dai_fmt_capture, }, }, }; -- cgit v1.2.3-70-g09d2 From 0664678a84c653bde844c7d91646259a25c6188b Mon Sep 17 00:00:00 2001 From: Philipp Zabel Date: Tue, 3 Feb 2009 21:18:26 +0100 Subject: ASoC: pxa-ssp: fix SSP port request PXA2xx/3xx SSP ports start from 1, not 0. Thus, the probe function requested the wrong SSP port. Correcting this unveiled another bug where ssp_init tries to request the already-requested SSP port again. So this patch replaces the ssp_init/exit calls with their internals from mach-pxa/ssp.c, leaving out the redundant ssp_request and the unneeded IRQ request. Effectively, that leaves us with not much more than enabling/disabling the SSP clock. Signed-off-by: Philipp Zabel Signed-off-by: Mark Brown --- sound/soc/pxa/pxa-ssp.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 73cb6b4c2f2..4a973ab710b 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -21,6 +21,8 @@ #include #include +#include + #include #include #include @@ -221,9 +223,9 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream, int ret = 0; if (!cpu_dai->active) { - ret = ssp_init(&priv->dev, cpu_dai->id + 1, SSP_NO_IRQ); - if (ret < 0) - return ret; + priv->dev.port = cpu_dai->id + 1; + priv->dev.irq = NO_IRQ; + clk_enable(priv->dev.ssp->clk); ssp_disable(&priv->dev); } return ret; @@ -238,7 +240,7 @@ static void pxa_ssp_shutdown(struct snd_pcm_substream *substream, if (!cpu_dai->active) { ssp_disable(&priv->dev); - ssp_exit(&priv->dev); + clk_disable(priv->dev.ssp->clk); } } @@ -751,7 +753,7 @@ static int pxa_ssp_probe(struct platform_device *pdev, if (!priv) return -ENOMEM; - priv->dev.ssp = ssp_request(dai->id, "SoC audio"); + priv->dev.ssp = ssp_request(dai->id + 1, "SoC audio"); if (priv->dev.ssp == NULL) { ret = -ENODEV; goto err_priv; -- cgit v1.2.3-70-g09d2 From 680cd53652d8bfb2b97d8c0248d1afb82de6b61d Mon Sep 17 00:00:00 2001 From: Kusanagi Kouichi Date: Thu, 5 Feb 2009 00:00:58 +0900 Subject: ALSA: hda: Add digital beep generator support for Realtek codecs. A digital beep generator can be used via input layer. Signed-off-by: Kusanagi Kouichi Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_beep.h | 2 +- sound/pci/hda/patch_realtek.c | 62 +++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 63 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h index b9679f081ca..51bf6a5daf3 100644 --- a/sound/pci/hda/hda_beep.h +++ b/sound/pci/hda/hda_beep.h @@ -39,7 +39,7 @@ struct hda_beep { int snd_hda_attach_beep_device(struct hda_codec *codec, int nid); void snd_hda_detach_beep_device(struct hda_codec *codec); #else -#define snd_hda_attach_beep_device(...) +#define snd_hda_attach_beep_device(...) 0 #define snd_hda_detach_beep_device(...) #endif #endif diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index bd9ef336389..0faa41bfc8b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -30,6 +30,7 @@ #include #include "hda_codec.h" #include "hda_local.h" +#include "hda_beep.h" #define ALC880_FRONT_EVENT 0x01 #define ALC880_DCVOL_EVENT 0x02 @@ -3187,6 +3188,7 @@ static void alc_free(struct hda_codec *codec) alc_free_kctls(codec); kfree(spec); + snd_hda_detach_beep_device(codec); codec->spec = NULL; /* to be sure */ } @@ -4355,6 +4357,12 @@ static int patch_alc880(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC880_AUTO) setup_preset(spec, &alc880_presets[board_config]); @@ -5882,6 +5890,12 @@ static int patch_alc260(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC260_AUTO) setup_preset(spec, &alc260_presets[board_config]); @@ -7093,6 +7107,12 @@ static int patch_alc882(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC882_AUTO) setup_preset(spec, &alc882_presets[board_config]); @@ -9093,6 +9113,12 @@ static int patch_alc883(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC883_AUTO) setup_preset(spec, &alc883_presets[board_config]); @@ -11013,6 +11039,12 @@ static int patch_alc262(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC262_AUTO) setup_preset(spec, &alc262_presets[board_config]); @@ -12051,6 +12083,12 @@ static int patch_alc268(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC268_AUTO) setup_preset(spec, &alc268_presets[board_config]); @@ -12885,6 +12923,12 @@ static int patch_alc269(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC269_AUTO) setup_preset(spec, &alc269_presets[board_config]); @@ -13978,6 +14022,12 @@ static int patch_alc861(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x23); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC861_AUTO) setup_preset(spec, &alc861_presets[board_config]); @@ -14924,6 +14974,12 @@ static int patch_alc861vd(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x23); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC861VD_AUTO) setup_preset(spec, &alc861vd_presets[board_config]); @@ -16733,6 +16789,12 @@ static int patch_alc662(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC662_AUTO) setup_preset(spec, &alc662_presets[board_config]); -- cgit v1.2.3-70-g09d2 From 453e37b37521b613f0927fcf53ccd93ad3a8b3ae Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Wed, 4 Feb 2009 17:41:32 +0100 Subject: ALSA: sscape: drop redundant fields from soundscape struct The wss_base is disuised parameter for one function. It is converted to function parameter. The code_type is only set but never read. It is removed. The midi_vol is set only to 0 so it does not work as detection of change in midi volume. It is fixed. The xport variable is alias to the port[dev]. Use the port[dev] directly to increase readability. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/sscape.c | 44 ++++++++++++++++---------------------------- 1 file changed, 16 insertions(+), 28 deletions(-) diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index 681e2237acb..33c1258029f 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -135,8 +135,6 @@ enum card_type { struct soundscape { spinlock_t lock; unsigned io_base; - unsigned wss_base; - int codec_type; int ic_type; enum card_type type; struct resource *io_res; @@ -726,13 +724,7 @@ static int sscape_midi_get(struct snd_kcontrol *kctl, unsigned long flags; spin_lock_irqsave(&s->lock, flags); - set_host_mode_unsafe(s->io_base); - - if (host_write_ctrl_unsafe(s->io_base, CMD_GET_MIDI_VOL, 100)) { - uctl->value.integer.value[0] = host_read_ctrl_unsafe(s->io_base, 100); - } - - set_midi_mode_unsafe(s->io_base); + uctl->value.integer.value[0] = s->midi_vol; spin_unlock_irqrestore(&s->lock, flags); return 0; } @@ -767,6 +759,7 @@ static int sscape_midi_put(struct snd_kcontrol *kctl, change = (host_write_ctrl_unsafe(s->io_base, CMD_SET_MIDI_VOL, 100) && host_write_ctrl_unsafe(s->io_base, ((unsigned char) uctl->value.integer. value[0]) & 127, 100) && host_write_ctrl_unsafe(s->io_base, CMD_XXX_MIDI_VOL, 100)); + s->midi_vol = (unsigned char) uctl->value.integer.value[0] & 127; __skip_change: /* @@ -809,12 +802,11 @@ static unsigned __devinit get_irq_config(int irq) * Perform certain arcane port-checks to see whether there * is a SoundScape board lurking behind the given ports. */ -static int __devinit detect_sscape(struct soundscape *s) +static int __devinit detect_sscape(struct soundscape *s, long wss_io) { unsigned long flags; unsigned d; int retval = 0; - int codec = s->wss_base; spin_lock_irqsave(&s->lock, flags); @@ -830,13 +822,11 @@ static int __devinit detect_sscape(struct soundscape *s) if ((d & 0x80) != 0) goto _done; - if (d == 0) { - s->codec_type = 1; + if (d == 0) s->ic_type = IC_ODIE; - } else if ((d & 0x60) != 0) { - s->codec_type = 2; + else if ((d & 0x60) != 0) s->ic_type = IC_OPUS; - } else + else goto _done; outb(0xfa, ODIE_ADDR_IO(s->io_base)); @@ -856,10 +846,10 @@ static int __devinit detect_sscape(struct soundscape *s) sscape_write_unsafe(s->io_base, GA_HMCTL_REG, d | 0xc0); if (s->type == SSCAPE_VIVO) - codec += 4; + wss_io += 4; /* wait for WSS codec */ for (d = 0; d < 500; d++) { - if ((inb(codec) & 0x80) == 0) + if ((inb(wss_io) & 0x80) == 0) break; spin_unlock_irqrestore(&s->lock, flags); msleep(1); @@ -1057,7 +1047,6 @@ static int __devinit create_sscape(int dev, struct snd_card *card) unsigned dma_cfg; unsigned irq_cfg; unsigned mpu_irq_cfg; - unsigned xport; struct resource *io_res; struct resource *wss_res; unsigned long flags; @@ -1077,15 +1066,15 @@ static int __devinit create_sscape(int dev, struct snd_card *card) printk(KERN_ERR "sscape: Invalid IRQ %d\n", mpu_irq[dev]); return -ENXIO; } - xport = port[dev]; /* * Grab IO ports that we will need to probe so that we * can detect and control this hardware ... */ - io_res = request_region(xport, 8, "SoundScape"); + io_res = request_region(port[dev], 8, "SoundScape"); if (!io_res) { - snd_printk(KERN_ERR "sscape: can't grab port 0x%x\n", xport); + snd_printk(KERN_ERR + "sscape: can't grab port 0x%lx\n", port[dev]); return -EBUSY; } wss_res = NULL; @@ -1112,10 +1101,9 @@ static int __devinit create_sscape(int dev, struct snd_card *card) spin_lock_init(&sscape->fwlock); sscape->io_res = io_res; sscape->wss_res = wss_res; - sscape->io_base = xport; - sscape->wss_base = wss_port[dev]; + sscape->io_base = port[dev]; - if (!detect_sscape(sscape)) { + if (!detect_sscape(sscape, wss_port[dev])) { printk(KERN_ERR "sscape: hardware not detected at 0x%x\n", sscape->io_base); err = -ENODEV; goto _release_dma; @@ -1188,11 +1176,11 @@ static int __devinit create_sscape(int dev, struct snd_card *card) } #define MIDI_DEVNUM 0 if (sscape->type != SSCAPE_VIVO) { - err = create_mpu401(card, MIDI_DEVNUM, xport, mpu_irq[dev]); + err = create_mpu401(card, MIDI_DEVNUM, port[dev], mpu_irq[dev]); if (err < 0) { printk(KERN_ERR "sscape: Failed to create " - "MPU-401 device at 0x%x\n", - xport); + "MPU-401 device at 0x%lx\n", + port[dev]); goto _release_dma; } -- cgit v1.2.3-70-g09d2 From 5e7476243ad755fa1d8be2b1774d0aeb16bb48df Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 4 Feb 2009 18:28:42 +0100 Subject: ALSA: msnd - Fix build error with CONFIG_PNP=n sound/isa/msnd/msnd_pinnacle.c:891: error: 'isapnp' undeclared (first use in this function) Signed-off-by: Takashi Iwai --- sound/isa/msnd/msnd_pinnacle.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/isa/msnd/msnd_pinnacle.c b/sound/isa/msnd/msnd_pinnacle.c index 70559223e8f..60b6abd7161 100644 --- a/sound/isa/msnd/msnd_pinnacle.c +++ b/sound/isa/msnd/msnd_pinnacle.c @@ -785,6 +785,9 @@ static int calibrate_signal; static int isapnp[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; module_param_array(isapnp, bool, NULL, 0444); MODULE_PARM_DESC(isapnp, "ISA PnP detection for specified soundcard."); +#define has_isapnp(x) isapnp[x] +#else +#define has_isapnp(x) 0 #endif MODULE_AUTHOR("Karsten Wiese "); @@ -888,7 +891,7 @@ static int __devinit snd_msnd_isa_probe(struct device *pdev, unsigned int idx) struct snd_card *card; struct snd_msnd *chip; - if (isapnp[idx] || cfg[idx] == SNDRV_AUTO_PORT) { + if (has_isapnp(idx) || cfg[idx] == SNDRV_AUTO_PORT) { printk(KERN_INFO LOGNAME ": Assuming PnP mode\n"); return -ENODEV; } @@ -1082,7 +1085,7 @@ static int __devinit snd_msnd_pnp_detect(struct pnp_card_link *pcard, int ret; for ( ; idx < SNDRV_CARDS; idx++) { - if (isapnp[idx]) + if (has_isapnp(idx)) break; } if (idx >= SNDRV_CARDS) -- cgit v1.2.3-70-g09d2 From 616f89e74cd954e04ae4f8bad6a3dc8730a4a47a Mon Sep 17 00:00:00 2001 From: Herton Ronaldo Krzesinski Date: Wed, 4 Feb 2009 11:23:19 -0500 Subject: ALSA: hda - Additional pin nids for STAC92HD71Bx and STAC92HD75Bx codecs Current code for STAC92HD71Bx and STAC92HD75Bx doesn't consider pin complexes 0x20 and 0x27. Also for 4 port models, nids 0x0e and 0x0f are vendor reserved. This commit changes code so it'll consider the additional pin complexes for models that have it, and avoid reserved nids to be touched on 4 port models. Signed-off-by: Herton Ronaldo Krzesinski Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 59 ++++++++++++++++++++++++++++++------------ 1 file changed, 43 insertions(+), 16 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index a7df81efed2..58c9ff9d27f 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -481,10 +481,17 @@ static hda_nid_t stac92hd83xxx_pin_nids[14] = { 0x0f, 0x10, 0x11, 0x12, 0x13, 0x1d, 0x1e, 0x1f, 0x20 }; -static hda_nid_t stac92hd71bxx_pin_nids[11] = { + +#define STAC92HD71BXX_NUM_PINS 13 +static hda_nid_t stac92hd71bxx_pin_nids_4port[STAC92HD71BXX_NUM_PINS] = { + 0x0a, 0x0b, 0x0c, 0x0d, 0x00, + 0x00, 0x14, 0x18, 0x19, 0x1e, + 0x1f, 0x20, 0x27 +}; +static hda_nid_t stac92hd71bxx_pin_nids_6port[STAC92HD71BXX_NUM_PINS] = { 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f, 0x14, 0x18, 0x19, 0x1e, - 0x1f, + 0x1f, 0x20, 0x27 }; static hda_nid_t stac927x_pin_nids[14] = { @@ -1745,28 +1752,32 @@ static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { {} /* terminator */ }; -static unsigned int ref92hd71bxx_pin_configs[11] = { +static unsigned int ref92hd71bxx_pin_configs[STAC92HD71BXX_NUM_PINS] = { 0x02214030, 0x02a19040, 0x01a19020, 0x01014010, 0x0181302e, 0x01014010, 0x01019020, 0x90a000f0, - 0x90a000f0, 0x01452050, 0x01452050, + 0x90a000f0, 0x01452050, 0x01452050, 0x00000000, + 0x00000000 }; -static unsigned int dell_m4_1_pin_configs[11] = { +static unsigned int dell_m4_1_pin_configs[STAC92HD71BXX_NUM_PINS] = { 0x0421101f, 0x04a11221, 0x40f000f0, 0x90170110, 0x23a1902e, 0x23014250, 0x40f000f0, 0x90a000f0, - 0x40f000f0, 0x4f0000f0, 0x4f0000f0, + 0x40f000f0, 0x4f0000f0, 0x4f0000f0, 0x00000000, + 0x00000000 }; -static unsigned int dell_m4_2_pin_configs[11] = { +static unsigned int dell_m4_2_pin_configs[STAC92HD71BXX_NUM_PINS] = { 0x0421101f, 0x04a11221, 0x90a70330, 0x90170110, 0x23a1902e, 0x23014250, 0x40f000f0, 0x40f000f0, - 0x40f000f0, 0x044413b0, 0x044413b0, + 0x40f000f0, 0x044413b0, 0x044413b0, 0x00000000, + 0x00000000 }; -static unsigned int dell_m4_3_pin_configs[11] = { +static unsigned int dell_m4_3_pin_configs[STAC92HD71BXX_NUM_PINS] = { 0x0421101f, 0x04a11221, 0x90a70330, 0x90170110, 0x40f000f0, 0x40f000f0, 0x40f000f0, 0x90a000f0, - 0x40f000f0, 0x044413b0, 0x044413b0, + 0x40f000f0, 0x044413b0, 0x044413b0, 0x00000000, + 0x00000000 }; static unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = { @@ -2311,7 +2322,9 @@ static int stac92xx_save_bios_config_regs(struct hda_codec *codec) for (i = 0; i < spec->num_pins; i++) { hda_nid_t nid = spec->pin_nids[i]; unsigned int pin_cfg; - + + if (!nid) + continue; pin_cfg = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONFIG_DEFAULT, 0x00); snd_printdd(KERN_INFO "hda_codec: pin nid %2.2x bios pin config %8.8x\n", @@ -2354,8 +2367,9 @@ static void stac92xx_set_config_regs(struct hda_codec *codec) return; for (i = 0; i < spec->num_pins; i++) - stac92xx_set_config_reg(codec, spec->pin_nids[i], - spec->pin_configs[i]); + if (spec->pin_nids[i] && spec->pin_configs[i]) + stac92xx_set_config_reg(codec, spec->pin_nids[i], + spec->pin_configs[i]); } static int stac_save_pin_cfgs(struct hda_codec *codec, unsigned int *pins) @@ -4952,9 +4966,21 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) codec->spec = spec; codec->patch_ops = stac92xx_patch_ops; - spec->num_pins = ARRAY_SIZE(stac92hd71bxx_pin_nids); + spec->num_pins = STAC92HD71BXX_NUM_PINS; + switch (codec->vendor_id) { + case 0x111d76b6: + case 0x111d76b7: + spec->pin_nids = stac92hd71bxx_pin_nids_4port; + break; + case 0x111d7603: + case 0x111d7608: + /* On 92HD75Bx 0x27 isn't a pin nid */ + spec->num_pins--; + /* fallthrough */ + default: + spec->pin_nids = stac92hd71bxx_pin_nids_6port; + } spec->num_pwrs = ARRAY_SIZE(stac92hd71bxx_pwr_nids); - spec->pin_nids = stac92hd71bxx_pin_nids; memcpy(&spec->private_dimux, &stac92hd71bxx_dmux, sizeof(stac92hd71bxx_dmux)); spec->board_config = snd_hda_check_board_config(codec, @@ -5018,7 +5044,8 @@ again: /* disable VSW */ spec->init = &stac92hd71bxx_analog_core_init[HD_DISABLE_PORTF]; unmute_init++; - stac_change_pin_config(codec, 0xf, 0x40f000f0); + stac_change_pin_config(codec, 0x0f, 0x40f000f0); + stac_change_pin_config(codec, 0x19, 0x40f000f3); break; case 0x111d7603: /* 6 Port with Analog Mixer */ if ((codec->revision_id & 0xf) == 1) -- cgit v1.2.3-70-g09d2 From 6df703aefc81252447c69d24d2863007de2338e9 Mon Sep 17 00:00:00 2001 From: Herton Ronaldo Krzesinski Date: Wed, 4 Feb 2009 11:34:22 -0500 Subject: ALSA: hda - Dynamic detection of dmics/dmuxes/smuxes in stac92hd71bxx Detect the number of connected ports and number of smuxes dynamically, looking at pin configs, using new introduced functions stac92hd71bxx_connected_ports and stac92hd71bxx_connected_smuxes. Also use proper input mux configuration for 4port and 5port models. Signed-off-by: Herton Ronaldo Krzesinski Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 99 +++++++++++++++++++++++++++++++++++++----- 1 file changed, 87 insertions(+), 12 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 58c9ff9d27f..c36c1c0f957 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4944,7 +4944,16 @@ again: return 0; } -static struct hda_input_mux stac92hd71bxx_dmux = { +static struct hda_input_mux stac92hd71bxx_dmux_nomixer = { + .num_items = 3, + .items = { + { "Analog Inputs", 0x00 }, + { "Digital Mic 1", 0x02 }, + { "Digital Mic 2", 0x03 }, + } +}; + +static struct hda_input_mux stac92hd71bxx_dmux_amixer = { .num_items = 4, .items = { { "Analog Inputs", 0x00 }, @@ -4954,11 +4963,57 @@ static struct hda_input_mux stac92hd71bxx_dmux = { } }; +static int stac92hd71bxx_connected_ports(struct hda_codec *codec, + hda_nid_t *nids, int num_nids) +{ + struct sigmatel_spec *spec = codec->spec; + int idx, num; + unsigned int def_conf; + + for (num = 0; num < num_nids; num++) { + for (idx = 0; idx < spec->num_pins; idx++) + if (spec->pin_nids[idx] == nids[num]) + break; + if (idx >= spec->num_pins) + break; + def_conf = get_defcfg_connect(spec->pin_configs[idx]); + if (def_conf == AC_JACK_PORT_NONE) + break; + } + return num; +} + +static int stac92hd71bxx_connected_smuxes(struct hda_codec *codec, + hda_nid_t dig0pin) +{ + struct sigmatel_spec *spec = codec->spec; + int idx; + + for (idx = 0; idx < spec->num_pins; idx++) + if (spec->pin_nids[idx] == dig0pin) + break; + if ((idx + 2) >= spec->num_pins) + return 0; + + /* dig1pin case */ + if (get_defcfg_connect(spec->pin_configs[idx+1]) != AC_JACK_PORT_NONE) + return 2; + + /* dig0pin + dig2pin case */ + if (get_defcfg_connect(spec->pin_configs[idx+2]) != AC_JACK_PORT_NONE) + return 2; + if (get_defcfg_connect(spec->pin_configs[idx]) != AC_JACK_PORT_NONE) + return 1; + else + return 0; +} + static int patch_stac92hd71bxx(struct hda_codec *codec) { struct sigmatel_spec *spec; struct hda_verb *unmute_init = stac92hd71bxx_unmute_core_init; int err = 0; + unsigned int ndmic_nids = 0; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -4981,8 +5036,6 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) spec->pin_nids = stac92hd71bxx_pin_nids_6port; } spec->num_pwrs = ARRAY_SIZE(stac92hd71bxx_pwr_nids); - memcpy(&spec->private_dimux, &stac92hd71bxx_dmux, - sizeof(stac92hd71bxx_dmux)); spec->board_config = snd_hda_check_board_config(codec, STAC_92HD71BXX_MODELS, stac92hd71bxx_models, @@ -5007,16 +5060,32 @@ again: spec->gpio_data = 0x01; } + spec->dmic_nids = stac92hd71bxx_dmic_nids; + spec->dmux_nids = stac92hd71bxx_dmux_nids; + switch (codec->vendor_id) { case 0x111d76b6: /* 4 Port without Analog Mixer */ case 0x111d76b7: case 0x111d76b4: /* 6 Port without Analog Mixer */ case 0x111d76b5: + memcpy(&spec->private_dimux, &stac92hd71bxx_dmux_nomixer, + sizeof(stac92hd71bxx_dmux_nomixer)); spec->mixer = stac92hd71bxx_mixer; spec->init = stac92hd71bxx_core_init; codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; + spec->num_dmics = stac92hd71bxx_connected_ports(codec, + stac92hd71bxx_dmic_nids, + STAC92HD71BXX_NUM_DMICS); + if (spec->num_dmics) { + spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids); + spec->dinput_mux = &spec->private_dimux; + ndmic_nids = ARRAY_SIZE(stac92hd71bxx_dmic_nids) - 1; + } break; case 0x111d7608: /* 5 Port with Analog Mixer */ + memcpy(&spec->private_dimux, &stac92hd71bxx_dmux_amixer, + sizeof(stac92hd71bxx_dmux_amixer)); + spec->private_dimux.num_items--; switch (spec->board_config) { case STAC_HP_M4: /* Enable VREF power saving on GPIO1 detect */ @@ -5046,6 +5115,12 @@ again: unmute_init++; stac_change_pin_config(codec, 0x0f, 0x40f000f0); stac_change_pin_config(codec, 0x19, 0x40f000f3); + stac92hd71bxx_dmic_nids[STAC92HD71BXX_NUM_DMICS - 1] = 0; + spec->num_dmics = stac92hd71bxx_connected_ports(codec, + stac92hd71bxx_dmic_nids, + STAC92HD71BXX_NUM_DMICS - 1); + spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids); + ndmic_nids = ARRAY_SIZE(stac92hd71bxx_dmic_nids) - 2; break; case 0x111d7603: /* 6 Port with Analog Mixer */ if ((codec->revision_id & 0xf) == 1) @@ -5055,10 +5130,17 @@ again: spec->num_pwrs = 0; /* fallthru */ default: + memcpy(&spec->private_dimux, &stac92hd71bxx_dmux_amixer, + sizeof(stac92hd71bxx_dmux_amixer)); spec->dinput_mux = &spec->private_dimux; spec->mixer = stac92hd71bxx_analog_mixer; spec->init = stac92hd71bxx_analog_core_init; codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; + spec->num_dmics = stac92hd71bxx_connected_ports(codec, + stac92hd71bxx_dmic_nids, + STAC92HD71BXX_NUM_DMICS); + spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids); + ndmic_nids = ARRAY_SIZE(stac92hd71bxx_dmic_nids) - 1; } if (get_wcaps(codec, 0xa) & AC_WCAP_IN_AMP) @@ -5071,13 +5153,12 @@ again: spec->digbeep_nid = 0x26; spec->mux_nids = stac92hd71bxx_mux_nids; spec->adc_nids = stac92hd71bxx_adc_nids; - spec->dmic_nids = stac92hd71bxx_dmic_nids; - spec->dmux_nids = stac92hd71bxx_dmux_nids; spec->smux_nids = stac92hd71bxx_smux_nids; spec->pwr_nids = stac92hd71bxx_pwr_nids; spec->num_muxes = ARRAY_SIZE(stac92hd71bxx_mux_nids); spec->num_adcs = ARRAY_SIZE(stac92hd71bxx_adc_nids); + spec->num_smuxes = stac92hd71bxx_connected_smuxes(codec, 0x1e); switch (spec->board_config) { case STAC_HP_M4: @@ -5097,17 +5178,11 @@ again: spec->num_smuxes = 0; spec->num_dmuxes = 0; break; - default: - spec->num_dmics = STAC92HD71BXX_NUM_DMICS; - spec->num_smuxes = ARRAY_SIZE(stac92hd71bxx_smux_nids); - spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids); }; spec->multiout.dac_nids = spec->dac_nids; if (spec->dinput_mux) - spec->private_dimux.num_items += - spec->num_dmics - - (ARRAY_SIZE(stac92hd71bxx_dmic_nids) - 1); + spec->private_dimux.num_items += spec->num_dmics - ndmic_nids; err = stac92xx_parse_auto_config(codec, 0x21, 0x23); if (!err) { -- cgit v1.2.3-70-g09d2 From 29d4ab4d6e996ef4c71910c915611151c34f1c75 Mon Sep 17 00:00:00 2001 From: Herton Ronaldo Krzesinski Date: Wed, 4 Feb 2009 11:37:27 -0500 Subject: ALSA: hda - Don't call stac92xx_parse_auto_config with wrong dig_in Don't use uneeded/wrong third parameter for stac92xx_parse_auto_config in patch_stac92hd71bxx (no SPDIF in). Signed-off-by: Herton Ronaldo Krzesinski Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index c36c1c0f957..0b00110a5a0 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -5184,7 +5184,7 @@ again: if (spec->dinput_mux) spec->private_dimux.num_items += spec->num_dmics - ndmic_nids; - err = stac92xx_parse_auto_config(codec, 0x21, 0x23); + err = stac92xx_parse_auto_config(codec, 0x21, 0); if (!err) { if (spec->board_config < 0) { printk(KERN_WARNING "hda_codec: No auto-config is " -- cgit v1.2.3-70-g09d2 From 45c1d85bcc6438454d104966c30fd2497ae1cdd7 Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Wed, 4 Feb 2009 17:49:41 -0500 Subject: ALSA: hda: Added stac378x digital slave out struct Added the ADATOut nid to a slave digital outs struct to allow output via the DigOut pin. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 0b00110a5a0..85dc642d113 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -404,6 +404,10 @@ static hda_nid_t stac922x_mux_nids[2] = { 0x12, 0x13, }; +static hda_nid_t stac927x_slave_dig_outs[2] = { + 0x1f, 0, +}; + static hda_nid_t stac927x_adc_nids[3] = { 0x07, 0x08, 0x09 }; @@ -5320,6 +5324,7 @@ static int patch_stac927x(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; + codec->slave_dig_outs = stac927x_slave_dig_outs; spec->num_pins = ARRAY_SIZE(stac927x_pin_nids); spec->pin_nids = stac927x_pin_nids; spec->board_config = snd_hda_check_board_config(codec, STAC_927X_MODELS, -- cgit v1.2.3-70-g09d2 From 345d0b1964df83a6c3fff815fabd34e37265581f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 5 Feb 2009 09:10:20 +0100 Subject: ALSA: hwdep - Make open callback optional Don't require the open callback as mandatory. Now all hwdeps ops can be optional. Signed-off-by: Takashi Iwai --- sound/core/hwdep.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/core/hwdep.c b/sound/core/hwdep.c index 195cafc5a55..a70ee7f1ed9 100644 --- a/sound/core/hwdep.c +++ b/sound/core/hwdep.c @@ -99,9 +99,6 @@ static int snd_hwdep_open(struct inode *inode, struct file * file) if (hw == NULL) return -ENODEV; - if (!hw->ops.open) - return -ENXIO; - if (!try_module_get(hw->card->module)) return -EFAULT; @@ -113,6 +110,10 @@ static int snd_hwdep_open(struct inode *inode, struct file * file) err = -EBUSY; break; } + if (!hw->ops.open) { + err = 0; + break; + } err = hw->ops.open(hw, file); if (err >= 0) break; @@ -151,7 +152,7 @@ static int snd_hwdep_open(struct inode *inode, struct file * file) static int snd_hwdep_release(struct inode *inode, struct file * file) { - int err = -ENXIO; + int err = 0; struct snd_hwdep *hw = file->private_data; struct module *mod = hw->card->module; -- cgit v1.2.3-70-g09d2 From e0d80648c0037b8b815317a52b782d4ea0c287f0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 5 Feb 2009 09:17:50 +0100 Subject: ALSA: hwdep - Fix coding style Fix misc coding style issues in hwdep.h and add some comments. Signed-off-by: Takashi Iwai --- include/sound/hwdep.h | 38 ++++++++++++++++++++++++-------------- 1 file changed, 24 insertions(+), 14 deletions(-) diff --git a/include/sound/hwdep.h b/include/sound/hwdep.h index d9eea013c75..8c05e47a409 100644 --- a/include/sound/hwdep.h +++ b/include/sound/hwdep.h @@ -27,18 +27,28 @@ struct snd_hwdep; +/* hwdep file ops; all ops can be NULL */ struct snd_hwdep_ops { - long long (*llseek) (struct snd_hwdep *hw, struct file * file, long long offset, int orig); - long (*read) (struct snd_hwdep *hw, char __user *buf, long count, loff_t *offset); - long (*write) (struct snd_hwdep *hw, const char __user *buf, long count, loff_t *offset); - int (*open) (struct snd_hwdep * hw, struct file * file); - int (*release) (struct snd_hwdep *hw, struct file * file); - unsigned int (*poll) (struct snd_hwdep *hw, struct file * file, poll_table * wait); - int (*ioctl) (struct snd_hwdep *hw, struct file * file, unsigned int cmd, unsigned long arg); - int (*ioctl_compat) (struct snd_hwdep *hw, struct file * file, unsigned int cmd, unsigned long arg); - int (*mmap) (struct snd_hwdep *hw, struct file * file, struct vm_area_struct * vma); - int (*dsp_status) (struct snd_hwdep *hw, struct snd_hwdep_dsp_status *status); - int (*dsp_load) (struct snd_hwdep *hw, struct snd_hwdep_dsp_image *image); + long long (*llseek)(struct snd_hwdep *hw, struct file *file, + long long offset, int orig); + long (*read)(struct snd_hwdep *hw, char __user *buf, + long count, loff_t *offset); + long (*write)(struct snd_hwdep *hw, const char __user *buf, + long count, loff_t *offset); + int (*open)(struct snd_hwdep *hw, struct file * file); + int (*release)(struct snd_hwdep *hw, struct file * file); + unsigned int (*poll)(struct snd_hwdep *hw, struct file *file, + poll_table *wait); + int (*ioctl)(struct snd_hwdep *hw, struct file *file, + unsigned int cmd, unsigned long arg); + int (*ioctl_compat)(struct snd_hwdep *hw, struct file *file, + unsigned int cmd, unsigned long arg); + int (*mmap)(struct snd_hwdep *hw, struct file *file, + struct vm_area_struct *vma); + int (*dsp_status)(struct snd_hwdep *hw, + struct snd_hwdep_dsp_status *status); + int (*dsp_load)(struct snd_hwdep *hw, + struct snd_hwdep_dsp_image *image); }; struct snd_hwdep { @@ -61,9 +71,9 @@ struct snd_hwdep { void (*private_free) (struct snd_hwdep *hwdep); struct mutex open_mutex; - int used; - unsigned int dsp_loaded; - unsigned int exclusive: 1; + int used; /* reference counter */ + unsigned int dsp_loaded; /* bit fields of loaded dsp indices */ + unsigned int exclusive:1; /* exclusive access mode */ }; extern int snd_hwdep_new(struct snd_card *card, char *id, int device, -- cgit v1.2.3-70-g09d2 From 28b7e343ee63454d563a71d2d5f769fc297fd5ad Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 5 Feb 2009 09:28:08 +0100 Subject: ALSA: Remove superfluous hwdep ops Remove NOP hwdep ops in sound drivers. Signed-off-by: Takashi Iwai --- sound/drivers/vx/vx_hwdep.c | 12 ------------ sound/pci/mixart/mixart_hwdep.c | 12 ------------ sound/pci/pcxhr/pcxhr_hwdep.c | 12 ------------ sound/pci/rme9652/hdsp.c | 9 --------- sound/pci/rme9652/hdspm.c | 9 --------- sound/synth/emux/emux_hwdep.c | 21 --------------------- sound/usb/usbmixer.c | 22 +--------------------- sound/usb/usx2y/usX2Yhwdep.c | 12 ------------ 8 files changed, 1 insertion(+), 108 deletions(-) diff --git a/sound/drivers/vx/vx_hwdep.c b/sound/drivers/vx/vx_hwdep.c index 8d6362e2d4c..46df8817c18 100644 --- a/sound/drivers/vx/vx_hwdep.c +++ b/sound/drivers/vx/vx_hwdep.c @@ -119,16 +119,6 @@ void snd_vx_free_firmware(struct vx_core *chip) #else /* old style firmware loading */ -static int vx_hwdep_open(struct snd_hwdep *hw, struct file *file) -{ - return 0; -} - -static int vx_hwdep_release(struct snd_hwdep *hw, struct file *file) -{ - return 0; -} - static int vx_hwdep_dsp_status(struct snd_hwdep *hw, struct snd_hwdep_dsp_status *info) { @@ -243,8 +233,6 @@ int snd_vx_setup_firmware(struct vx_core *chip) hw->iface = SNDRV_HWDEP_IFACE_VX; hw->private_data = chip; - hw->ops.open = vx_hwdep_open; - hw->ops.release = vx_hwdep_release; hw->ops.dsp_status = vx_hwdep_dsp_status; hw->ops.dsp_load = vx_hwdep_dsp_load; hw->exclusive = 1; diff --git a/sound/pci/mixart/mixart_hwdep.c b/sound/pci/mixart/mixart_hwdep.c index 3782b52bc0e..fa4de985fc4 100644 --- a/sound/pci/mixart/mixart_hwdep.c +++ b/sound/pci/mixart/mixart_hwdep.c @@ -581,16 +581,6 @@ MODULE_FIRMWARE("mixart/miXart8AES.xlx"); /* miXart hwdep interface id string */ #define SND_MIXART_HWDEP_ID "miXart Loader" -static int mixart_hwdep_open(struct snd_hwdep *hw, struct file *file) -{ - return 0; -} - -static int mixart_hwdep_release(struct snd_hwdep *hw, struct file *file) -{ - return 0; -} - static int mixart_hwdep_dsp_status(struct snd_hwdep *hw, struct snd_hwdep_dsp_status *info) { @@ -643,8 +633,6 @@ int snd_mixart_setup_firmware(struct mixart_mgr *mgr) hw->iface = SNDRV_HWDEP_IFACE_MIXART; hw->private_data = mgr; - hw->ops.open = mixart_hwdep_open; - hw->ops.release = mixart_hwdep_release; hw->ops.dsp_status = mixart_hwdep_dsp_status; hw->ops.dsp_load = mixart_hwdep_dsp_load; hw->exclusive = 1; diff --git a/sound/pci/pcxhr/pcxhr_hwdep.c b/sound/pci/pcxhr/pcxhr_hwdep.c index 592743a298b..17cb1233a90 100644 --- a/sound/pci/pcxhr/pcxhr_hwdep.c +++ b/sound/pci/pcxhr/pcxhr_hwdep.c @@ -471,16 +471,6 @@ static int pcxhr_hwdep_dsp_load(struct snd_hwdep *hw, return 0; } -static int pcxhr_hwdep_open(struct snd_hwdep *hw, struct file *file) -{ - return 0; -} - -static int pcxhr_hwdep_release(struct snd_hwdep *hw, struct file *file) -{ - return 0; -} - int pcxhr_setup_firmware(struct pcxhr_mgr *mgr) { int err; @@ -495,8 +485,6 @@ int pcxhr_setup_firmware(struct pcxhr_mgr *mgr) hw->iface = SNDRV_HWDEP_IFACE_PCXHR; hw->private_data = mgr; - hw->ops.open = pcxhr_hwdep_open; - hw->ops.release = pcxhr_hwdep_release; hw->ops.dsp_status = pcxhr_hwdep_dsp_status; hw->ops.dsp_load = pcxhr_hwdep_dsp_load; hw->exclusive = 1; diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 44d0c15e2b7..2434609b2d3 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -4413,13 +4413,6 @@ static int snd_hdsp_capture_release(struct snd_pcm_substream *substream) return 0; } -static int snd_hdsp_hwdep_dummy_op(struct snd_hwdep *hw, struct file *file) -{ - /* we have nothing to initialize but the call is required */ - return 0; -} - - /* helper functions for copying meter values */ static inline int copy_u32_le(void __user *dest, void __iomem *src) { @@ -4738,9 +4731,7 @@ static int snd_hdsp_create_hwdep(struct snd_card *card, struct hdsp *hdsp) hw->private_data = hdsp; strcpy(hw->name, "HDSP hwdep interface"); - hw->ops.open = snd_hdsp_hwdep_dummy_op; hw->ops.ioctl = snd_hdsp_hwdep_ioctl; - hw->ops.release = snd_hdsp_hwdep_dummy_op; return 0; } diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 71231cf1b2b..df2034eb235 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -4100,13 +4100,6 @@ static int snd_hdspm_capture_release(struct snd_pcm_substream *substream) return 0; } -static int snd_hdspm_hwdep_dummy_op(struct snd_hwdep * hw, struct file *file) -{ - /* we have nothing to initialize but the call is required */ - return 0; -} - - static int snd_hdspm_hwdep_ioctl(struct snd_hwdep * hw, struct file *file, unsigned int cmd, unsigned long arg) { @@ -4213,9 +4206,7 @@ static int __devinit snd_hdspm_create_hwdep(struct snd_card *card, hw->private_data = hdspm; strcpy(hw->name, "HDSPM hwdep interface"); - hw->ops.open = snd_hdspm_hwdep_dummy_op; hw->ops.ioctl = snd_hdspm_hwdep_ioctl; - hw->ops.release = snd_hdspm_hwdep_dummy_op; return 0; } diff --git a/sound/synth/emux/emux_hwdep.c b/sound/synth/emux/emux_hwdep.c index 0a5391436ad..ff0b2a8fd25 100644 --- a/sound/synth/emux/emux_hwdep.c +++ b/sound/synth/emux/emux_hwdep.c @@ -24,25 +24,6 @@ #include #include "emux_voice.h" -/* - * open the hwdep device - */ -static int -snd_emux_hwdep_open(struct snd_hwdep *hw, struct file *file) -{ - return 0; -} - - -/* - * close the device - */ -static int -snd_emux_hwdep_release(struct snd_hwdep *hw, struct file *file) -{ - return 0; -} - #define TMP_CLIENT_ID 0x1001 @@ -146,8 +127,6 @@ snd_emux_init_hwdep(struct snd_emux *emu) emu->hwdep = hw; strcpy(hw->name, SNDRV_EMUX_HWDEP_NAME); hw->iface = SNDRV_HWDEP_IFACE_EMUX_WAVETABLE; - hw->ops.open = snd_emux_hwdep_open; - hw->ops.release = snd_emux_hwdep_release; hw->ops.ioctl = snd_emux_hwdep_ioctl; hw->exclusive = 1; hw->private_data = emu; diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index 00397c8a765..2bde79216fa 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -78,7 +78,6 @@ struct usb_mixer_interface { /* Sound Blaster remote control stuff */ const struct rc_config *rc_cfg; - unsigned long rc_hwdep_open; u32 rc_code; wait_queue_head_t rc_waitq; struct urb *rc_urb; @@ -1797,24 +1796,6 @@ static void snd_usb_soundblaster_remote_complete(struct urb *urb) wake_up(&mixer->rc_waitq); } -static int snd_usb_sbrc_hwdep_open(struct snd_hwdep *hw, struct file *file) -{ - struct usb_mixer_interface *mixer = hw->private_data; - - if (test_and_set_bit(0, &mixer->rc_hwdep_open)) - return -EBUSY; - return 0; -} - -static int snd_usb_sbrc_hwdep_release(struct snd_hwdep *hw, struct file *file) -{ - struct usb_mixer_interface *mixer = hw->private_data; - - clear_bit(0, &mixer->rc_hwdep_open); - smp_mb__after_clear_bit(); - return 0; -} - static long snd_usb_sbrc_hwdep_read(struct snd_hwdep *hw, char __user *buf, long count, loff_t *offset) { @@ -1867,9 +1848,8 @@ static int snd_usb_soundblaster_remote_init(struct usb_mixer_interface *mixer) hwdep->iface = SNDRV_HWDEP_IFACE_SB_RC; hwdep->private_data = mixer; hwdep->ops.read = snd_usb_sbrc_hwdep_read; - hwdep->ops.open = snd_usb_sbrc_hwdep_open; - hwdep->ops.release = snd_usb_sbrc_hwdep_release; hwdep->ops.poll = snd_usb_sbrc_hwdep_poll; + hwdep->exclusive = 1; mixer->rc_urb = usb_alloc_urb(0, GFP_KERNEL); if (!mixer->rc_urb) diff --git a/sound/usb/usx2y/usX2Yhwdep.c b/sound/usb/usx2y/usX2Yhwdep.c index 1558a5c4094..a26d8d83d3e 100644 --- a/sound/usb/usx2y/usX2Yhwdep.c +++ b/sound/usb/usx2y/usX2Yhwdep.c @@ -106,16 +106,6 @@ static unsigned int snd_us428ctls_poll(struct snd_hwdep *hw, struct file *file, } -static int snd_usX2Y_hwdep_open(struct snd_hwdep *hw, struct file *file) -{ - return 0; -} - -static int snd_usX2Y_hwdep_release(struct snd_hwdep *hw, struct file *file) -{ - return 0; -} - static int snd_usX2Y_hwdep_dsp_status(struct snd_hwdep *hw, struct snd_hwdep_dsp_status *info) { @@ -267,8 +257,6 @@ int usX2Y_hwdep_new(struct snd_card *card, struct usb_device* device) hw->iface = SNDRV_HWDEP_IFACE_USX2Y; hw->private_data = usX2Y(card); - hw->ops.open = snd_usX2Y_hwdep_open; - hw->ops.release = snd_usX2Y_hwdep_release; hw->ops.dsp_status = snd_usX2Y_hwdep_dsp_status; hw->ops.dsp_load = snd_usX2Y_hwdep_dsp_load; hw->ops.mmap = snd_us428ctls_mmap; -- cgit v1.2.3-70-g09d2 From 705350f8bd6b44fda3f0dcc3e6f4b453da4378dd Mon Sep 17 00:00:00 2001 From: Mark Hills Date: Wed, 4 Feb 2009 22:34:30 +0000 Subject: ALSA: snd-usb-caiaq: Send the correct command when setting controls Fixes a bug where an incorrect command was sent which had no effect on the device. Signed-off-by: Mark Hills Acked-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/caiaq/caiaq-control.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/caiaq/caiaq-control.c b/sound/usb/caiaq/caiaq-control.c index 6ac5489a0f2..1f9531d0fce 100644 --- a/sound/usb/caiaq/caiaq-control.c +++ b/sound/usb/caiaq/caiaq-control.c @@ -94,7 +94,7 @@ static int control_put(struct snd_kcontrol *kcontrol, if (pos & CNT_INTVAL) { dev->control_state[pos & ~CNT_INTVAL] = ucontrol->value.integer.value[0]; - snd_usb_caiaq_send_command(dev, EP1_CMD_DIMM_LEDS, + snd_usb_caiaq_send_command(dev, EP1_CMD_WRITE_IO, dev->control_state, sizeof(dev->control_state)); } else { if (ucontrol->value.integer.value[0]) -- cgit v1.2.3-70-g09d2 From e3ca4c9982e3b84da859ca20a3ca0a9d5bda8c30 Mon Sep 17 00:00:00 2001 From: Mark Hills Date: Wed, 4 Feb 2009 22:34:31 +0000 Subject: ALSA: snd-usb-caiaq: Set default input mode of A4DJ Do not start the device with input mode undefined. Mimic the behaviour of the Audio 8 DJ and start in phono input mode. Signed-off-by: Mark Hills Acked-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/caiaq/caiaq-device.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/usb/caiaq/caiaq-device.c b/sound/usb/caiaq/caiaq-device.c index d09fc2a88cf..94610dda8ab 100644 --- a/sound/usb/caiaq/caiaq-device.c +++ b/sound/usb/caiaq/caiaq-device.c @@ -312,6 +312,12 @@ static void __devinit setup_card(struct snd_usb_caiaqdev *dev) } break; + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ): + /* Audio 4 DJ - default input mode to phono */ + dev->control_state[0] = 2; + snd_usb_caiaq_send_command(dev, EP1_CMD_WRITE_IO, + dev->control_state, 1); + break; } if (dev->spec.num_analog_audio_out + -- cgit v1.2.3-70-g09d2 From 9a9527ed49f45e75a5b005592a261ab2bd7c1b1d Mon Sep 17 00:00:00 2001 From: Mark Hills Date: Wed, 4 Feb 2009 22:34:32 +0000 Subject: ALSA: snd-usb-caiaq: Do not expose hardware input mode 0 of A4DJ In the context of the Audio 4 DJ (when compared to Audio 8 DJ), hardware input mode 0 is not used. Expose modes 1 (line) and 2 (phono) to the user as modes 0 and 1 respectively. Signed-off-by: Mark Hills Acked-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/caiaq/caiaq-control.c | 32 +++++++++++++++++++++++++++++--- 1 file changed, 29 insertions(+), 3 deletions(-) diff --git a/sound/usb/caiaq/caiaq-control.c b/sound/usb/caiaq/caiaq-control.c index 1f9531d0fce..136ef34300d 100644 --- a/sound/usb/caiaq/caiaq-control.c +++ b/sound/usb/caiaq/caiaq-control.c @@ -44,16 +44,24 @@ static int control_info(struct snd_kcontrol *kcontrol, uinfo->count = 1; pos &= ~CNT_INTVAL; - if (((id == USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ)) || - (id == USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ))) + if (id == USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ) && (pos == 0)) { - /* current input mode of A8DJ and A4DJ */ + /* current input mode of A8DJ */ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->value.integer.min = 0; uinfo->value.integer.max = 2; return 0; } + if (id == USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ) + && (pos == 0)) { + /* current input mode of A4DJ */ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; + } + if (is_intval) { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->value.integer.min = 0; @@ -74,6 +82,14 @@ static int control_get(struct snd_kcontrol *kcontrol, struct snd_usb_caiaqdev *dev = caiaqdev(chip->card); int pos = kcontrol->private_value; + if (dev->chip.usb_id == + USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ)) { + /* A4DJ has only one control */ + /* do not expose hardware input mode 0 */ + ucontrol->value.integer.value[0] = dev->control_state[0] - 1; + return 0; + } + if (pos & CNT_INTVAL) ucontrol->value.integer.value[0] = dev->control_state[pos & ~CNT_INTVAL]; @@ -91,6 +107,16 @@ static int control_put(struct snd_kcontrol *kcontrol, struct snd_usb_caiaqdev *dev = caiaqdev(chip->card); int pos = kcontrol->private_value; + if (dev->chip.usb_id == + USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ)) { + /* A4DJ has only one control */ + /* do not expose hardware input mode 0 */ + dev->control_state[0] = ucontrol->value.integer.value[0] + 1; + snd_usb_caiaq_send_command(dev, EP1_CMD_WRITE_IO, + dev->control_state, sizeof(dev->control_state)); + return 1; + } + if (pos & CNT_INTVAL) { dev->control_state[pos & ~CNT_INTVAL] = ucontrol->value.integer.value[0]; -- cgit v1.2.3-70-g09d2 From a8564155a9cb3b5c4a18afc451679a1f02c647b5 Mon Sep 17 00:00:00 2001 From: Mark Hills Date: Wed, 4 Feb 2009 22:34:33 +0000 Subject: ALSA: snd-usb-caiaq: Remove duplicate A8DJ control Remove a duplicate control which causes an error when it is registered, and causes later controls to not be registered. The device does not have a fourth ground lift control. Signed-off-by: Mark Hills Acked-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/caiaq/caiaq-control.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/usb/caiaq/caiaq-control.c b/sound/usb/caiaq/caiaq-control.c index 136ef34300d..e92c2bbf4fe 100644 --- a/sound/usb/caiaq/caiaq-control.c +++ b/sound/usb/caiaq/caiaq-control.c @@ -270,7 +270,6 @@ static struct caiaq_controller a8dj_controller[] = { { "GND lift for TC Vinyl mode", 24 + 0 }, { "GND lift for TC CD/Line mode", 24 + 1 }, { "GND lift for phono mode", 24 + 2 }, - { "GND lift for TC Vinyl mode", 24 + 3 }, { "Software lock", 40 } }; -- cgit v1.2.3-70-g09d2 From 238c0270baade3a542c1497712dd8e66cc9cc476 Mon Sep 17 00:00:00 2001 From: Mark Hills Date: Wed, 4 Feb 2009 22:34:34 +0000 Subject: ALSA: snd-usb-caiaq: Increase version number to 1.3.12 Indicates fixes affecting control messages and switching of input mode on Audio 8 DJ and Audio 4 DJ. Signed-off-by: Mark Hills Acked-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/caiaq/caiaq-device.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/caiaq/caiaq-device.c b/sound/usb/caiaq/caiaq-device.c index 94610dda8ab..5736669df2d 100644 --- a/sound/usb/caiaq/caiaq-device.c +++ b/sound/usb/caiaq/caiaq-device.c @@ -42,7 +42,7 @@ #endif MODULE_AUTHOR("Daniel Mack "); -MODULE_DESCRIPTION("caiaq USB audio, version 1.3.11"); +MODULE_DESCRIPTION("caiaq USB audio, version 1.3.12"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," "{Native Instruments, RigKontrol3}," -- cgit v1.2.3-70-g09d2 From 67f7857ab12e9f8005ef988f0b667396e07622c2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 5 Feb 2009 12:14:52 +0100 Subject: ALSA: hda - Add quirk for HP zenith laptop Added model=laptop for another HP laptop (103c:3072) with AD1984A codec. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index e934e2c187d..6e348d03b71 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -3890,6 +3890,7 @@ static struct snd_pci_quirk ad1884a_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x3030, "HP", AD1884A_MOBILE), SND_PCI_QUIRK(0x103c, 0x3037, "HP 2230s", AD1884A_LAPTOP), SND_PCI_QUIRK(0x103c, 0x3056, "HP", AD1884A_MOBILE), + SND_PCI_QUIRK(0x103c, 0x3072, "HP", AD1884A_LAPTOP), SND_PCI_QUIRK(0x103c, 0x30e6, "HP 6730b", AD1884A_LAPTOP), SND_PCI_QUIRK(0x103c, 0x30e7, "HP EliteBook 8530p", AD1884A_LAPTOP), SND_PCI_QUIRK(0x103c, 0x3614, "HP 6730s", AD1884A_LAPTOP), -- cgit v1.2.3-70-g09d2 From 632da7321b7e9fa5375956280f8a0f380836c22d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 5 Feb 2009 15:02:06 +0100 Subject: ALSA: hda - Add quirk for another HP laptop Add model=laptop entry for another HP laptop (103c:3077) with AD1984A. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 6e348d03b71..30399cbf819 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -3891,6 +3891,7 @@ static struct snd_pci_quirk ad1884a_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x3037, "HP 2230s", AD1884A_LAPTOP), SND_PCI_QUIRK(0x103c, 0x3056, "HP", AD1884A_MOBILE), SND_PCI_QUIRK(0x103c, 0x3072, "HP", AD1884A_LAPTOP), + SND_PCI_QUIRK(0x103c, 0x3077, "HP", AD1884A_LAPTOP), SND_PCI_QUIRK(0x103c, 0x30e6, "HP 6730b", AD1884A_LAPTOP), SND_PCI_QUIRK(0x103c, 0x30e7, "HP EliteBook 8530p", AD1884A_LAPTOP), SND_PCI_QUIRK(0x103c, 0x3614, "HP 6730s", AD1884A_LAPTOP), -- cgit v1.2.3-70-g09d2 From e6161653094f14b1add10efe3493a2e526fe9538 Mon Sep 17 00:00:00 2001 From: Tim Blechmann Date: Thu, 5 Feb 2009 13:01:54 +0100 Subject: ALSA: snd_pcm_new api cleanup Impact: cleanup snd_pcm_new takes a char *id argument, although it is not modifying the string. it can therefore be declared as const char *id. Signed-off-by: Tim Blechmann Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 2 +- sound/core/pcm.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 40c5a6fa6bc..ee0e887e49d 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -451,7 +451,7 @@ struct snd_pcm_notify { extern const struct file_operations snd_pcm_f_ops[2]; -int snd_pcm_new(struct snd_card *card, char *id, int device, +int snd_pcm_new(struct snd_card *card, const char *id, int device, int playback_count, int capture_count, struct snd_pcm **rpcm); int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count); diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 192a433a240..583453e2355 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -692,7 +692,7 @@ EXPORT_SYMBOL(snd_pcm_new_stream); * * Returns zero if successful, or a negative error code on failure. */ -int snd_pcm_new(struct snd_card *card, char *id, int device, +int snd_pcm_new(struct snd_card *card, const char *id, int device, int playback_count, int capture_count, struct snd_pcm ** rpcm) { -- cgit v1.2.3-70-g09d2 From e4967d6016b7785edafdb871e6d3e4426cb4bd1f Mon Sep 17 00:00:00 2001 From: Hans-Christian Egtvedt Date: Thu, 5 Feb 2009 13:10:59 +0100 Subject: ALSA: Add ALSA driver for Atmel Audio Bitstream DAC This patch adds ALSA support for the Audio Bistream DAC found on Atmel AVR32 devices. The ABDAC is an Atmel IP which might show up on AT91 devices in the future, hence making a generic driver which can be utilized by AT91 arch if needed. Datasheet describing the ABDAC peripheral is available in the AT32AP7000 datasheet, http://www.atmel.com/dyn/products/datasheets.asp?family_id=682 Tested on ATSTK1006 + ATSTK1000 with a class D amplifier stage. Signed-off-by: Hans-Christian Egtvedt Signed-off-by: Takashi Iwai --- include/sound/atmel-abdac.h | 23 ++ sound/atmel/Kconfig | 11 + sound/atmel/Makefile | 3 + sound/atmel/abdac.c | 602 ++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 639 insertions(+) create mode 100644 include/sound/atmel-abdac.h create mode 100644 sound/atmel/Kconfig create mode 100644 sound/atmel/Makefile create mode 100644 sound/atmel/abdac.c diff --git a/include/sound/atmel-abdac.h b/include/sound/atmel-abdac.h new file mode 100644 index 00000000000..edff6a8ba1b --- /dev/null +++ b/include/sound/atmel-abdac.h @@ -0,0 +1,23 @@ +/* + * Driver for the Atmel Audio Bitstream DAC (ABDAC) + * + * Copyright (C) 2009 Atmel Corporation + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License version 2 as published + * by the Free Software Foundation. + */ +#ifndef __INCLUDE_SOUND_ATMEL_ABDAC_H +#define __INCLUDE_SOUND_ATMEL_ABDAC_H + +#include + +/** + * struct atmel_abdac_pdata - board specific ABDAC configuration + * @dws: DMA slave interface to use for sound playback. + */ +struct atmel_abdac_pdata { + struct dw_dma_slave dws; +}; + +#endif /* __INCLUDE_SOUND_ATMEL_ABDAC_H */ diff --git a/sound/atmel/Kconfig b/sound/atmel/Kconfig new file mode 100644 index 00000000000..715318e3670 --- /dev/null +++ b/sound/atmel/Kconfig @@ -0,0 +1,11 @@ +menu "Atmel devices (AVR32 and AT91)" + depends on AVR32 || ARCH_AT91 + +config SND_ATMEL_ABDAC + tristate "Atmel Audio Bitstream DAC (ABDAC) driver" + select SND_PCM + depends on DW_DMAC && AVR32 + help + ALSA sound driver for the Atmel Audio Bitstream DAC (ABDAC). + +endmenu diff --git a/sound/atmel/Makefile b/sound/atmel/Makefile new file mode 100644 index 00000000000..c5a8213f9cb --- /dev/null +++ b/sound/atmel/Makefile @@ -0,0 +1,3 @@ +snd-atmel-abdac-objs := abdac.o + +obj-$(CONFIG_SND_ATMEL_ABDAC) += snd-atmel-abdac.o diff --git a/sound/atmel/abdac.c b/sound/atmel/abdac.c new file mode 100644 index 00000000000..28b3c7f7cfe --- /dev/null +++ b/sound/atmel/abdac.c @@ -0,0 +1,602 @@ +/* + * Driver for the Atmel on-chip Audio Bitstream DAC (ABDAC) + * + * Copyright (C) 2006-2009 Atmel Corporation + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License version 2 as published by + * the Free Software Foundation. + */ +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include + +/* DAC register offsets */ +#define DAC_DATA 0x0000 +#define DAC_CTRL 0x0008 +#define DAC_INT_MASK 0x000c +#define DAC_INT_EN 0x0010 +#define DAC_INT_DIS 0x0014 +#define DAC_INT_CLR 0x0018 +#define DAC_INT_STATUS 0x001c + +/* Bitfields in CTRL */ +#define DAC_SWAP_OFFSET 30 +#define DAC_SWAP_SIZE 1 +#define DAC_EN_OFFSET 31 +#define DAC_EN_SIZE 1 + +/* Bitfields in INT_MASK/INT_EN/INT_DIS/INT_STATUS/INT_CLR */ +#define DAC_UNDERRUN_OFFSET 28 +#define DAC_UNDERRUN_SIZE 1 +#define DAC_TX_READY_OFFSET 29 +#define DAC_TX_READY_SIZE 1 + +/* Bit manipulation macros */ +#define DAC_BIT(name) \ + (1 << DAC_##name##_OFFSET) +#define DAC_BF(name, value) \ + (((value) & ((1 << DAC_##name##_SIZE) - 1)) \ + << DAC_##name##_OFFSET) +#define DAC_BFEXT(name, value) \ + (((value) >> DAC_##name##_OFFSET) \ + & ((1 << DAC_##name##_SIZE) - 1)) +#define DAC_BFINS(name, value, old) \ + (((old) & ~(((1 << DAC_##name##_SIZE) - 1) \ + << DAC_##name##_OFFSET)) \ + | DAC_BF(name, value)) + +/* Register access macros */ +#define dac_readl(port, reg) \ + __raw_readl((port)->regs + DAC_##reg) +#define dac_writel(port, reg, value) \ + __raw_writel((value), (port)->regs + DAC_##reg) + +/* + * ABDAC supports a maximum of 6 different rates from a generic clock. The + * generic clock has a power of two divider, which gives 6 steps from 192 kHz + * to 5112 Hz. + */ +#define MAX_NUM_RATES 6 +/* ALSA seems to use rates between 192000 Hz and 5112 Hz. */ +#define RATE_MAX 192000 +#define RATE_MIN 5112 + +enum { + DMA_READY = 0, +}; + +struct atmel_abdac_dma { + struct dma_chan *chan; + struct dw_cyclic_desc *cdesc; +}; + +struct atmel_abdac { + struct clk *pclk; + struct clk *sample_clk; + struct platform_device *pdev; + struct atmel_abdac_dma dma; + + struct snd_pcm_hw_constraint_list constraints_rates; + struct snd_pcm_substream *substream; + struct snd_card *card; + struct snd_pcm *pcm; + + void __iomem *regs; + unsigned long flags; + unsigned int rates[MAX_NUM_RATES]; + unsigned int rates_num; + int irq; +}; + +#define get_dac(card) ((struct atmel_abdac *)(card)->private_data) + +/* This function is called by the DMA driver. */ +static void atmel_abdac_dma_period_done(void *arg) +{ + struct atmel_abdac *dac = arg; + snd_pcm_period_elapsed(dac->substream); +} + +static int atmel_abdac_prepare_dma(struct atmel_abdac *dac, + struct snd_pcm_substream *substream, + enum dma_data_direction direction) +{ + struct dma_chan *chan = dac->dma.chan; + struct dw_cyclic_desc *cdesc; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned long buffer_len, period_len; + + /* + * We don't do DMA on "complex" transfers, i.e. with + * non-halfword-aligned buffers or lengths. + */ + if (runtime->dma_addr & 1 || runtime->buffer_size & 1) { + dev_dbg(&dac->pdev->dev, "too complex transfer\n"); + return -EINVAL; + } + + buffer_len = frames_to_bytes(runtime, runtime->buffer_size); + period_len = frames_to_bytes(runtime, runtime->period_size); + + cdesc = dw_dma_cyclic_prep(chan, runtime->dma_addr, buffer_len, + period_len, DMA_TO_DEVICE); + if (IS_ERR(cdesc)) { + dev_dbg(&dac->pdev->dev, "could not prepare cyclic DMA\n"); + return PTR_ERR(cdesc); + } + + cdesc->period_callback = atmel_abdac_dma_period_done; + cdesc->period_callback_param = dac; + + dac->dma.cdesc = cdesc; + + set_bit(DMA_READY, &dac->flags); + + return 0; +} + +static struct snd_pcm_hardware atmel_abdac_hw = { + .info = (SNDRV_PCM_INFO_MMAP + | SNDRV_PCM_INFO_MMAP_VALID + | SNDRV_PCM_INFO_INTERLEAVED + | SNDRV_PCM_INFO_BLOCK_TRANSFER + | SNDRV_PCM_INFO_RESUME + | SNDRV_PCM_INFO_PAUSE), + .formats = (SNDRV_PCM_FMTBIT_S16_BE), + .rates = (SNDRV_PCM_RATE_KNOT), + .rate_min = RATE_MIN, + .rate_max = RATE_MAX, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 64 * 4096, + .period_bytes_min = 4096, + .period_bytes_max = 4096, + .periods_min = 4, + .periods_max = 64, +}; + +static int atmel_abdac_open(struct snd_pcm_substream *substream) +{ + struct atmel_abdac *dac = snd_pcm_substream_chip(substream); + + dac->substream = substream; + atmel_abdac_hw.rate_max = dac->rates[dac->rates_num - 1]; + atmel_abdac_hw.rate_min = dac->rates[0]; + substream->runtime->hw = atmel_abdac_hw; + + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &dac->constraints_rates); +} + +static int atmel_abdac_close(struct snd_pcm_substream *substream) +{ + struct atmel_abdac *dac = snd_pcm_substream_chip(substream); + dac->substream = NULL; + return 0; +} + +static int atmel_abdac_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct atmel_abdac *dac = snd_pcm_substream_chip(substream); + int retval; + + retval = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + if (retval < 0) + return retval; + /* snd_pcm_lib_malloc_pages returns 1 if buffer is changed. */ + if (retval == 1) + if (test_and_clear_bit(DMA_READY, &dac->flags)) + dw_dma_cyclic_free(dac->dma.chan); + + return retval; +} + +static int atmel_abdac_hw_free(struct snd_pcm_substream *substream) +{ + struct atmel_abdac *dac = snd_pcm_substream_chip(substream); + if (test_and_clear_bit(DMA_READY, &dac->flags)) + dw_dma_cyclic_free(dac->dma.chan); + return snd_pcm_lib_free_pages(substream); +} + +static int atmel_abdac_prepare(struct snd_pcm_substream *substream) +{ + struct atmel_abdac *dac = snd_pcm_substream_chip(substream); + int retval; + + retval = clk_set_rate(dac->sample_clk, 256 * substream->runtime->rate); + if (retval) + return retval; + + if (!test_bit(DMA_READY, &dac->flags)) + retval = atmel_abdac_prepare_dma(dac, substream, DMA_TO_DEVICE); + + return retval; +} + +static int atmel_abdac_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct atmel_abdac *dac = snd_pcm_substream_chip(substream); + int retval = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: /* fall through */ + case SNDRV_PCM_TRIGGER_RESUME: /* fall through */ + case SNDRV_PCM_TRIGGER_START: + clk_enable(dac->sample_clk); + retval = dw_dma_cyclic_start(dac->dma.chan); + if (retval) + goto out; + dac_writel(dac, CTRL, DAC_BIT(EN)); + break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: /* fall through */ + case SNDRV_PCM_TRIGGER_SUSPEND: /* fall through */ + case SNDRV_PCM_TRIGGER_STOP: + dw_dma_cyclic_stop(dac->dma.chan); + dac_writel(dac, DATA, 0); + dac_writel(dac, CTRL, 0); + clk_disable(dac->sample_clk); + break; + default: + retval = -EINVAL; + break; + } +out: + return retval; +} + +static snd_pcm_uframes_t +atmel_abdac_pointer(struct snd_pcm_substream *substream) +{ + struct atmel_abdac *dac = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + snd_pcm_uframes_t frames; + unsigned long bytes; + + bytes = dw_dma_get_src_addr(dac->dma.chan); + bytes -= runtime->dma_addr; + + frames = bytes_to_frames(runtime, bytes); + if (frames >= runtime->buffer_size) + frames -= runtime->buffer_size; + + return frames; +} + +static irqreturn_t abdac_interrupt(int irq, void *dev_id) +{ + struct atmel_abdac *dac = dev_id; + u32 status; + + status = dac_readl(dac, INT_STATUS); + if (status & DAC_BIT(UNDERRUN)) { + dev_err(&dac->pdev->dev, "underrun detected\n"); + dac_writel(dac, INT_CLR, DAC_BIT(UNDERRUN)); + } else { + dev_err(&dac->pdev->dev, "spurious interrupt (status=0x%x)\n", + status); + dac_writel(dac, INT_CLR, status); + } + + return IRQ_HANDLED; +} + +static struct snd_pcm_ops atmel_abdac_ops = { + .open = atmel_abdac_open, + .close = atmel_abdac_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = atmel_abdac_hw_params, + .hw_free = atmel_abdac_hw_free, + .prepare = atmel_abdac_prepare, + .trigger = atmel_abdac_trigger, + .pointer = atmel_abdac_pointer, +}; + +static int __devinit atmel_abdac_pcm_new(struct atmel_abdac *dac) +{ + struct snd_pcm_hardware hw = atmel_abdac_hw; + struct snd_pcm *pcm; + int retval; + + retval = snd_pcm_new(dac->card, dac->card->shortname, + dac->pdev->id, 1, 0, &pcm); + if (retval) + return retval; + + strcpy(pcm->name, dac->card->shortname); + pcm->private_data = dac; + pcm->info_flags = 0; + dac->pcm = pcm; + + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &atmel_abdac_ops); + + retval = snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + &dac->pdev->dev, hw.periods_min * hw.period_bytes_min, + hw.buffer_bytes_max); + + return retval; +} + +static bool filter(struct dma_chan *chan, void *slave) +{ + struct dw_dma_slave *dws = slave; + + if (dws->dma_dev == chan->device->dev) { + chan->private = dws; + return true; + } else + return false; +} + +static int set_sample_rates(struct atmel_abdac *dac) +{ + long new_rate = RATE_MAX; + int retval = -EINVAL; + int index = 0; + + /* we start at 192 kHz and work our way down to 5112 Hz */ + while (new_rate >= RATE_MIN && index < (MAX_NUM_RATES + 1)) { + new_rate = clk_round_rate(dac->sample_clk, 256 * new_rate); + if (new_rate < 0) + break; + /* make sure we are below the ABDAC clock */ + if (new_rate <= clk_get_rate(dac->pclk)) { + dac->rates[index] = new_rate / 256; + index++; + } + /* divide by 256 and then by two to get next rate */ + new_rate /= 256 * 2; + } + + if (index) { + int i; + + /* reverse array, smallest go first */ + for (i = 0; i < (index / 2); i++) { + unsigned int tmp = dac->rates[index - 1 - i]; + dac->rates[index - 1 - i] = dac->rates[i]; + dac->rates[i] = tmp; + } + + dac->constraints_rates.count = index; + dac->constraints_rates.list = dac->rates; + dac->constraints_rates.mask = 0; + dac->rates_num = index; + + retval = 0; + } + + return retval; +} + +static int __devinit atmel_abdac_probe(struct platform_device *pdev) +{ + struct snd_card *card; + struct atmel_abdac *dac; + struct resource *regs; + struct atmel_abdac_pdata *pdata; + struct clk *pclk; + struct clk *sample_clk; + int retval; + int irq; + + regs = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!regs) { + dev_dbg(&pdev->dev, "no memory resource\n"); + return -ENXIO; + } + + irq = platform_get_irq(pdev, 0); + if (irq < 0) { + dev_dbg(&pdev->dev, "could not get IRQ number\n"); + return irq; + } + + pdata = pdev->dev.platform_data; + if (!pdata) { + dev_dbg(&pdev->dev, "no platform data\n"); + return -ENXIO; + } + + pclk = clk_get(&pdev->dev, "pclk"); + if (IS_ERR(pclk)) { + dev_dbg(&pdev->dev, "no peripheral clock\n"); + return PTR_ERR(pclk); + } + sample_clk = clk_get(&pdev->dev, "sample_clk"); + if (IS_ERR(pclk)) { + dev_dbg(&pdev->dev, "no sample clock\n"); + retval = PTR_ERR(pclk); + goto out_put_pclk; + } + clk_enable(pclk); + + retval = snd_card_create(SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1, + THIS_MODULE, sizeof(struct atmel_abdac), &card); + if (retval) { + dev_dbg(&pdev->dev, "could not create sound card device\n"); + goto out_put_sample_clk; + } + + dac = get_dac(card); + + dac->irq = irq; + dac->card = card; + dac->pclk = pclk; + dac->sample_clk = sample_clk; + dac->pdev = pdev; + + retval = set_sample_rates(dac); + if (retval < 0) { + dev_dbg(&pdev->dev, "could not set supported rates\n"); + goto out_free_card; + } + + dac->regs = ioremap(regs->start, regs->end - regs->start + 1); + if (!dac->regs) { + dev_dbg(&pdev->dev, "could not remap register memory\n"); + goto out_free_card; + } + + /* make sure the DAC is silent and disabled */ + dac_writel(dac, DATA, 0); + dac_writel(dac, CTRL, 0); + + retval = request_irq(irq, abdac_interrupt, 0, "abdac", dac); + if (retval) { + dev_dbg(&pdev->dev, "could not request irq\n"); + goto out_unmap_regs; + } + + snd_card_set_dev(card, &pdev->dev); + + if (pdata->dws.dma_dev) { + struct dw_dma_slave *dws = &pdata->dws; + dma_cap_mask_t mask; + + dws->tx_reg = regs->start + DAC_DATA; + + dma_cap_zero(mask); + dma_cap_set(DMA_SLAVE, mask); + + dac->dma.chan = dma_request_channel(mask, filter, dws); + } + if (!pdata->dws.dma_dev || !dac->dma.chan) { + dev_dbg(&pdev->dev, "DMA not available\n"); + retval = -ENODEV; + goto out_unset_card_dev; + } + + strcpy(card->driver, "Atmel ABDAC"); + strcpy(card->shortname, "Atmel ABDAC"); + sprintf(card->longname, "Atmel Audio Bitstream DAC"); + + retval = atmel_abdac_pcm_new(dac); + if (retval) { + dev_dbg(&pdev->dev, "could not register ABDAC pcm device\n"); + goto out_release_dma; + } + + retval = snd_card_register(card); + if (retval) { + dev_dbg(&pdev->dev, "could not register sound card\n"); + goto out_release_dma; + } + + platform_set_drvdata(pdev, card); + + dev_info(&pdev->dev, "Atmel ABDAC at 0x%p using %s\n", + dac->regs, dac->dma.chan->dev->device.bus_id); + + return retval; + +out_release_dma: + dma_release_channel(dac->dma.chan); + dac->dma.chan = NULL; +out_unset_card_dev: + snd_card_set_dev(card, NULL); + free_irq(irq, dac); +out_unmap_regs: + iounmap(dac->regs); +out_free_card: + snd_card_free(card); +out_put_sample_clk: + clk_put(sample_clk); + clk_disable(pclk); +out_put_pclk: + clk_put(pclk); + return retval; +} + +#ifdef CONFIG_PM +static int atmel_abdac_suspend(struct platform_device *pdev, pm_message_t msg) +{ + struct snd_card *card = platform_get_drvdata(pdev); + struct atmel_abdac *dac = card->private_data; + + dw_dma_cyclic_stop(dac->dma.chan); + clk_disable(dac->sample_clk); + clk_disable(dac->pclk); + + return 0; +} + +static int atmel_abdac_resume(struct platform_device *pdev) +{ + struct snd_card *card = platform_get_drvdata(pdev); + struct atmel_abdac *dac = card->private_data; + + clk_enable(dac->pclk); + clk_enable(dac->sample_clk); + if (test_bit(DMA_READY, &dac->flags)) + dw_dma_cyclic_start(dac->dma.chan); + + return 0; +} +#else +#define atmel_abdac_suspend NULL +#define atmel_abdac_resume NULL +#endif + +static int __devexit atmel_abdac_remove(struct platform_device *pdev) +{ + struct snd_card *card = platform_get_drvdata(pdev); + struct atmel_abdac *dac = get_dac(card); + + clk_put(dac->sample_clk); + clk_disable(dac->pclk); + clk_put(dac->pclk); + + dma_release_channel(dac->dma.chan); + dac->dma.chan = NULL; + snd_card_set_dev(card, NULL); + iounmap(dac->regs); + free_irq(dac->irq, dac); + snd_card_free(card); + + platform_set_drvdata(pdev, NULL); + + return 0; +} + +static struct platform_driver atmel_abdac_driver = { + .remove = __devexit_p(atmel_abdac_remove), + .driver = { + .name = "atmel_abdac", + }, + .suspend = atmel_abdac_suspend, + .resume = atmel_abdac_resume, +}; + +static int __init atmel_abdac_init(void) +{ + return platform_driver_probe(&atmel_abdac_driver, + atmel_abdac_probe); +} +module_init(atmel_abdac_init); + +static void __exit atmel_abdac_exit(void) +{ + platform_driver_unregister(&atmel_abdac_driver); +} +module_exit(atmel_abdac_exit); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Driver for Atmel Audio Bitstream DAC (ABDAC)"); +MODULE_AUTHOR("Hans-Christian Egtvedt "); -- cgit v1.2.3-70-g09d2 From 4ede028f8716523fc31e0d3d01b81405613dfb8f Mon Sep 17 00:00:00 2001 From: Hans-Christian Egtvedt Date: Thu, 5 Feb 2009 13:11:00 +0100 Subject: ALSA: Add ALSA driver for Atmel AC97 controller This patch adds ALSA support for the AC97 controller found on Atmel AVR32 devices. Tested on ATSTK1006 + ATSTK1000 with a development board with a AC97 codec. Signed-off-by: Hans-Christian Egtvedt Signed-off-by: Takashi Iwai --- include/sound/atmel-ac97c.h | 40 ++ sound/atmel/Kconfig | 8 + sound/atmel/Makefile | 2 + sound/atmel/ac97c.c | 932 ++++++++++++++++++++++++++++++++++++++++++++ sound/atmel/ac97c.h | 71 ++++ 5 files changed, 1053 insertions(+) create mode 100644 include/sound/atmel-ac97c.h create mode 100644 sound/atmel/ac97c.c create mode 100644 sound/atmel/ac97c.h diff --git a/include/sound/atmel-ac97c.h b/include/sound/atmel-ac97c.h new file mode 100644 index 00000000000..e6aabdb4586 --- /dev/null +++ b/include/sound/atmel-ac97c.h @@ -0,0 +1,40 @@ +/* + * Driver for the Atmel AC97C controller + * + * Copyright (C) 2005-2009 Atmel Corporation + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License version 2 as published + * by the Free Software Foundation. + */ +#ifndef __INCLUDE_SOUND_ATMEL_AC97C_H +#define __INCLUDE_SOUND_ATMEL_AC97C_H + +#include + +#define AC97C_CAPTURE 0x01 +#define AC97C_PLAYBACK 0x02 +#define AC97C_BOTH (AC97C_CAPTURE | AC97C_PLAYBACK) + +/** + * struct atmel_ac97c_pdata - board specific AC97C configuration + * @rx_dws: DMA slave interface to use for sound capture. + * @tx_dws: DMA slave interface to use for sound playback. + * @reset_pin: GPIO pin wired to the reset input on the external AC97 codec, + * optional to use, set to -ENODEV if not in use. AC97 layer will + * try to do a software reset of the external codec anyway. + * @flags: Flags for which directions should be enabled. + * + * If the user do not want to use a DMA channel for playback or capture, i.e. + * only one feature is required on the board. The slave for playback or capture + * can be set to NULL. The AC97C driver will take use of this when setting up + * the sound streams. + */ +struct ac97c_platform_data { + struct dw_dma_slave rx_dws; + struct dw_dma_slave tx_dws; + unsigned int flags; + int reset_pin; +}; + +#endif /* __INCLUDE_SOUND_ATMEL_AC97C_H */ diff --git a/sound/atmel/Kconfig b/sound/atmel/Kconfig index 715318e3670..6c228a91940 100644 --- a/sound/atmel/Kconfig +++ b/sound/atmel/Kconfig @@ -8,4 +8,12 @@ config SND_ATMEL_ABDAC help ALSA sound driver for the Atmel Audio Bitstream DAC (ABDAC). +config SND_ATMEL_AC97C + tristate "Atmel AC97 Controller (AC97C) driver" + select SND_PCM + select SND_AC97_CODEC + depends on DW_DMAC && AVR32 + help + ALSA sound driver for the Atmel AC97 controller. + endmenu diff --git a/sound/atmel/Makefile b/sound/atmel/Makefile index c5a8213f9cb..219dcfac608 100644 --- a/sound/atmel/Makefile +++ b/sound/atmel/Makefile @@ -1,3 +1,5 @@ snd-atmel-abdac-objs := abdac.o +snd-atmel-ac97c-objs := ac97c.o obj-$(CONFIG_SND_ATMEL_ABDAC) += snd-atmel-abdac.o +obj-$(CONFIG_SND_ATMEL_AC97C) += snd-atmel-ac97c.o diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c new file mode 100644 index 00000000000..dd72e00e5ae --- /dev/null +++ b/sound/atmel/ac97c.c @@ -0,0 +1,932 @@ +/* + * Driver for the Atmel AC97C controller + * + * Copyright (C) 2005-2009 Atmel Corporation + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License version 2 as published by + * the Free Software Foundation. + */ +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include +#include +#include + +#include + +#include "ac97c.h" + +enum { + DMA_TX_READY = 0, + DMA_RX_READY, + DMA_TX_CHAN_PRESENT, + DMA_RX_CHAN_PRESENT, +}; + +/* Serialize access to opened variable */ +static DEFINE_MUTEX(opened_mutex); + +struct atmel_ac97c_dma { + struct dma_chan *rx_chan; + struct dma_chan *tx_chan; +}; + +struct atmel_ac97c { + struct clk *pclk; + struct platform_device *pdev; + struct atmel_ac97c_dma dma; + + struct snd_pcm_substream *playback_substream; + struct snd_pcm_substream *capture_substream; + struct snd_card *card; + struct snd_pcm *pcm; + struct snd_ac97 *ac97; + struct snd_ac97_bus *ac97_bus; + + u64 cur_format; + unsigned int cur_rate; + unsigned long flags; + /* Serialize access to opened variable */ + spinlock_t lock; + void __iomem *regs; + int opened; + int reset_pin; +}; + +#define get_chip(card) ((struct atmel_ac97c *)(card)->private_data) + +#define ac97c_writel(chip, reg, val) \ + __raw_writel((val), (chip)->regs + AC97C_##reg) +#define ac97c_readl(chip, reg) \ + __raw_readl((chip)->regs + AC97C_##reg) + +/* This function is called by the DMA driver. */ +static void atmel_ac97c_dma_playback_period_done(void *arg) +{ + struct atmel_ac97c *chip = arg; + snd_pcm_period_elapsed(chip->playback_substream); +} + +static void atmel_ac97c_dma_capture_period_done(void *arg) +{ + struct atmel_ac97c *chip = arg; + snd_pcm_period_elapsed(chip->capture_substream); +} + +static int atmel_ac97c_prepare_dma(struct atmel_ac97c *chip, + struct snd_pcm_substream *substream, + enum dma_data_direction direction) +{ + struct dma_chan *chan; + struct dw_cyclic_desc *cdesc; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned long buffer_len, period_len; + + /* + * We don't do DMA on "complex" transfers, i.e. with + * non-halfword-aligned buffers or lengths. + */ + if (runtime->dma_addr & 1 || runtime->buffer_size & 1) { + dev_dbg(&chip->pdev->dev, "too complex transfer\n"); + return -EINVAL; + } + + if (direction == DMA_TO_DEVICE) + chan = chip->dma.tx_chan; + else + chan = chip->dma.rx_chan; + + buffer_len = frames_to_bytes(runtime, runtime->buffer_size); + period_len = frames_to_bytes(runtime, runtime->period_size); + + cdesc = dw_dma_cyclic_prep(chan, runtime->dma_addr, buffer_len, + period_len, direction); + if (IS_ERR(cdesc)) { + dev_dbg(&chip->pdev->dev, "could not prepare cyclic DMA\n"); + return PTR_ERR(cdesc); + } + + if (direction == DMA_TO_DEVICE) { + cdesc->period_callback = atmel_ac97c_dma_playback_period_done; + set_bit(DMA_TX_READY, &chip->flags); + } else { + cdesc->period_callback = atmel_ac97c_dma_capture_period_done; + set_bit(DMA_RX_READY, &chip->flags); + } + + cdesc->period_callback_param = chip; + + return 0; +} + +static struct snd_pcm_hardware atmel_ac97c_hw = { + .info = (SNDRV_PCM_INFO_MMAP + | SNDRV_PCM_INFO_MMAP_VALID + | SNDRV_PCM_INFO_INTERLEAVED + | SNDRV_PCM_INFO_BLOCK_TRANSFER + | SNDRV_PCM_INFO_JOINT_DUPLEX + | SNDRV_PCM_INFO_RESUME + | SNDRV_PCM_INFO_PAUSE), + .formats = (SNDRV_PCM_FMTBIT_S16_BE + | SNDRV_PCM_FMTBIT_S16_LE), + .rates = (SNDRV_PCM_RATE_CONTINUOUS), + .rate_min = 4000, + .rate_max = 48000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = 64 * 4096, + .period_bytes_min = 4096, + .period_bytes_max = 4096, + .periods_min = 4, + .periods_max = 64, +}; + +static int atmel_ac97c_playback_open(struct snd_pcm_substream *substream) +{ + struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + mutex_lock(&opened_mutex); + chip->opened++; + runtime->hw = atmel_ac97c_hw; + if (chip->cur_rate) { + runtime->hw.rate_min = chip->cur_rate; + runtime->hw.rate_max = chip->cur_rate; + } + if (chip->cur_format) + runtime->hw.formats = (1ULL << chip->cur_format); + mutex_unlock(&opened_mutex); + chip->playback_substream = substream; + return 0; +} + +static int atmel_ac97c_capture_open(struct snd_pcm_substream *substream) +{ + struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + mutex_lock(&opened_mutex); + chip->opened++; + runtime->hw = atmel_ac97c_hw; + if (chip->cur_rate) { + runtime->hw.rate_min = chip->cur_rate; + runtime->hw.rate_max = chip->cur_rate; + } + if (chip->cur_format) + runtime->hw.formats = (1ULL << chip->cur_format); + mutex_unlock(&opened_mutex); + chip->capture_substream = substream; + return 0; +} + +static int atmel_ac97c_playback_close(struct snd_pcm_substream *substream) +{ + struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); + + mutex_lock(&opened_mutex); + chip->opened--; + if (!chip->opened) { + chip->cur_rate = 0; + chip->cur_format = 0; + } + mutex_unlock(&opened_mutex); + + chip->playback_substream = NULL; + + return 0; +} + +static int atmel_ac97c_capture_close(struct snd_pcm_substream *substream) +{ + struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); + + mutex_lock(&opened_mutex); + chip->opened--; + if (!chip->opened) { + chip->cur_rate = 0; + chip->cur_format = 0; + } + mutex_unlock(&opened_mutex); + + chip->capture_substream = NULL; + + return 0; +} + +static int atmel_ac97c_playback_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); + int retval; + + retval = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + if (retval < 0) + return retval; + /* snd_pcm_lib_malloc_pages returns 1 if buffer is changed. */ + if (retval == 1) + if (test_and_clear_bit(DMA_TX_READY, &chip->flags)) + dw_dma_cyclic_free(chip->dma.tx_chan); + + /* Set restrictions to params. */ + mutex_lock(&opened_mutex); + chip->cur_rate = params_rate(hw_params); + chip->cur_format = params_format(hw_params); + mutex_unlock(&opened_mutex); + + return retval; +} + +static int atmel_ac97c_capture_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); + int retval; + + retval = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + if (retval < 0) + return retval; + /* snd_pcm_lib_malloc_pages returns 1 if buffer is changed. */ + if (retval == 1) + if (test_and_clear_bit(DMA_RX_READY, &chip->flags)) + dw_dma_cyclic_free(chip->dma.rx_chan); + + /* Set restrictions to params. */ + mutex_lock(&opened_mutex); + chip->cur_rate = params_rate(hw_params); + chip->cur_format = params_format(hw_params); + mutex_unlock(&opened_mutex); + + return retval; +} + +static int atmel_ac97c_playback_hw_free(struct snd_pcm_substream *substream) +{ + struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); + if (test_and_clear_bit(DMA_TX_READY, &chip->flags)) + dw_dma_cyclic_free(chip->dma.tx_chan); + return snd_pcm_lib_free_pages(substream); +} + +static int atmel_ac97c_capture_hw_free(struct snd_pcm_substream *substream) +{ + struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); + if (test_and_clear_bit(DMA_RX_READY, &chip->flags)) + dw_dma_cyclic_free(chip->dma.rx_chan); + return snd_pcm_lib_free_pages(substream); +} + +static int atmel_ac97c_playback_prepare(struct snd_pcm_substream *substream) +{ + struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned long word = 0; + int retval; + + /* assign channels to AC97C channel A */ + switch (runtime->channels) { + case 1: + word |= AC97C_CH_ASSIGN(PCM_LEFT, A); + break; + case 2: + word |= AC97C_CH_ASSIGN(PCM_LEFT, A) + | AC97C_CH_ASSIGN(PCM_RIGHT, A); + break; + default: + /* TODO: support more than two channels */ + return -EINVAL; + break; + } + ac97c_writel(chip, OCA, word); + + /* configure sample format and size */ + word = AC97C_CMR_DMAEN | AC97C_CMR_SIZE_16; + + switch (runtime->format) { + case SNDRV_PCM_FORMAT_S16_LE: + word |= AC97C_CMR_CEM_LITTLE; + break; + case SNDRV_PCM_FORMAT_S16_BE: /* fall through */ + default: + word &= ~(AC97C_CMR_CEM_LITTLE); + break; + } + + ac97c_writel(chip, CAMR, word); + + /* set variable rate if needed */ + if (runtime->rate != 48000) { + word = ac97c_readl(chip, MR); + word |= AC97C_MR_VRA; + ac97c_writel(chip, MR, word); + } else { + word = ac97c_readl(chip, MR); + word &= ~(AC97C_MR_VRA); + ac97c_writel(chip, MR, word); + } + + retval = snd_ac97_set_rate(chip->ac97, AC97_PCM_FRONT_DAC_RATE, + runtime->rate); + if (retval) + dev_dbg(&chip->pdev->dev, "could not set rate %d Hz\n", + runtime->rate); + + if (!test_bit(DMA_TX_READY, &chip->flags)) + retval = atmel_ac97c_prepare_dma(chip, substream, + DMA_TO_DEVICE); + + return retval; +} + +static int atmel_ac97c_capture_prepare(struct snd_pcm_substream *substream) +{ + struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned long word = 0; + int retval; + + /* assign channels to AC97C channel A */ + switch (runtime->channels) { + case 1: + word |= AC97C_CH_ASSIGN(PCM_LEFT, A); + break; + case 2: + word |= AC97C_CH_ASSIGN(PCM_LEFT, A) + | AC97C_CH_ASSIGN(PCM_RIGHT, A); + break; + default: + /* TODO: support more than two channels */ + return -EINVAL; + break; + } + ac97c_writel(chip, ICA, word); + + /* configure sample format and size */ + word = AC97C_CMR_DMAEN | AC97C_CMR_SIZE_16; + + switch (runtime->format) { + case SNDRV_PCM_FORMAT_S16_LE: + word |= AC97C_CMR_CEM_LITTLE; + break; + case SNDRV_PCM_FORMAT_S16_BE: /* fall through */ + default: + word &= ~(AC97C_CMR_CEM_LITTLE); + break; + } + + ac97c_writel(chip, CAMR, word); + + /* set variable rate if needed */ + if (runtime->rate != 48000) { + word = ac97c_readl(chip, MR); + word |= AC97C_MR_VRA; + ac97c_writel(chip, MR, word); + } else { + word = ac97c_readl(chip, MR); + word &= ~(AC97C_MR_VRA); + ac97c_writel(chip, MR, word); + } + + retval = snd_ac97_set_rate(chip->ac97, AC97_PCM_LR_ADC_RATE, + runtime->rate); + if (retval) + dev_dbg(&chip->pdev->dev, "could not set rate %d Hz\n", + runtime->rate); + + if (!test_bit(DMA_RX_READY, &chip->flags)) + retval = atmel_ac97c_prepare_dma(chip, substream, + DMA_FROM_DEVICE); + + return retval; +} + +static int +atmel_ac97c_playback_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); + unsigned long camr; + int retval = 0; + + camr = ac97c_readl(chip, CAMR); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: /* fall through */ + case SNDRV_PCM_TRIGGER_RESUME: /* fall through */ + case SNDRV_PCM_TRIGGER_START: + retval = dw_dma_cyclic_start(chip->dma.tx_chan); + if (retval) + goto out; + camr |= AC97C_CMR_CENA; + break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: /* fall through */ + case SNDRV_PCM_TRIGGER_SUSPEND: /* fall through */ + case SNDRV_PCM_TRIGGER_STOP: + dw_dma_cyclic_stop(chip->dma.tx_chan); + if (chip->opened <= 1) + camr &= ~AC97C_CMR_CENA; + break; + default: + retval = -EINVAL; + goto out; + } + + ac97c_writel(chip, CAMR, camr); +out: + return retval; +} + +static int +atmel_ac97c_capture_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); + unsigned long camr; + int retval = 0; + + camr = ac97c_readl(chip, CAMR); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: /* fall through */ + case SNDRV_PCM_TRIGGER_RESUME: /* fall through */ + case SNDRV_PCM_TRIGGER_START: + retval = dw_dma_cyclic_start(chip->dma.rx_chan); + if (retval) + goto out; + camr |= AC97C_CMR_CENA; + break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: /* fall through */ + case SNDRV_PCM_TRIGGER_SUSPEND: /* fall through */ + case SNDRV_PCM_TRIGGER_STOP: + dw_dma_cyclic_stop(chip->dma.rx_chan); + if (chip->opened <= 1) + camr &= ~AC97C_CMR_CENA; + break; + default: + retval = -EINVAL; + break; + } + + ac97c_writel(chip, CAMR, camr); +out: + return retval; +} + +static snd_pcm_uframes_t +atmel_ac97c_playback_pointer(struct snd_pcm_substream *substream) +{ + struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + snd_pcm_uframes_t frames; + unsigned long bytes; + + bytes = dw_dma_get_src_addr(chip->dma.tx_chan); + bytes -= runtime->dma_addr; + + frames = bytes_to_frames(runtime, bytes); + if (frames >= runtime->buffer_size) + frames -= runtime->buffer_size; + return frames; +} + +static snd_pcm_uframes_t +atmel_ac97c_capture_pointer(struct snd_pcm_substream *substream) +{ + struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + snd_pcm_uframes_t frames; + unsigned long bytes; + + bytes = dw_dma_get_dst_addr(chip->dma.rx_chan); + bytes -= runtime->dma_addr; + + frames = bytes_to_frames(runtime, bytes); + if (frames >= runtime->buffer_size) + frames -= runtime->buffer_size; + return frames; +} + +static struct snd_pcm_ops atmel_ac97_playback_ops = { + .open = atmel_ac97c_playback_open, + .close = atmel_ac97c_playback_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = atmel_ac97c_playback_hw_params, + .hw_free = atmel_ac97c_playback_hw_free, + .prepare = atmel_ac97c_playback_prepare, + .trigger = atmel_ac97c_playback_trigger, + .pointer = atmel_ac97c_playback_pointer, +}; + +static struct snd_pcm_ops atmel_ac97_capture_ops = { + .open = atmel_ac97c_capture_open, + .close = atmel_ac97c_capture_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = atmel_ac97c_capture_hw_params, + .hw_free = atmel_ac97c_capture_hw_free, + .prepare = atmel_ac97c_capture_prepare, + .trigger = atmel_ac97c_capture_trigger, + .pointer = atmel_ac97c_capture_pointer, +}; + +static int __devinit atmel_ac97c_pcm_new(struct atmel_ac97c *chip) +{ + struct snd_pcm *pcm; + struct snd_pcm_hardware hw = atmel_ac97c_hw; + int capture, playback, retval; + + capture = test_bit(DMA_RX_CHAN_PRESENT, &chip->flags); + playback = test_bit(DMA_TX_CHAN_PRESENT, &chip->flags); + + retval = snd_pcm_new(chip->card, chip->card->shortname, + chip->pdev->id, playback, capture, &pcm); + if (retval) + return retval; + + if (capture) + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, + &atmel_ac97_capture_ops); + if (playback) + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &atmel_ac97_playback_ops); + + retval = snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + &chip->pdev->dev, hw.periods_min * hw.period_bytes_min, + hw.buffer_bytes_max); + if (retval) + return retval; + + pcm->private_data = chip; + pcm->info_flags = 0; + strcpy(pcm->name, chip->card->shortname); + chip->pcm = pcm; + + return 0; +} + +static int atmel_ac97c_mixer_new(struct atmel_ac97c *chip) +{ + struct snd_ac97_template template; + memset(&template, 0, sizeof(template)); + template.private_data = chip; + return snd_ac97_mixer(chip->ac97_bus, &template, &chip->ac97); +} + +static void atmel_ac97c_write(struct snd_ac97 *ac97, unsigned short reg, + unsigned short val) +{ + struct atmel_ac97c *chip = get_chip(ac97); + unsigned long word; + int timeout = 40; + + word = (reg & 0x7f) << 16 | val; + + do { + if (ac97c_readl(chip, COSR) & AC97C_CSR_TXRDY) { + ac97c_writel(chip, COTHR, word); + return; + } + udelay(1); + } while (--timeout); + + dev_dbg(&chip->pdev->dev, "codec write timeout\n"); +} + +static unsigned short atmel_ac97c_read(struct snd_ac97 *ac97, + unsigned short reg) +{ + struct atmel_ac97c *chip = get_chip(ac97); + unsigned long word; + int timeout = 40; + int write = 10; + + word = (0x80 | (reg & 0x7f)) << 16; + + if ((ac97c_readl(chip, COSR) & AC97C_CSR_RXRDY) != 0) + ac97c_readl(chip, CORHR); + +retry_write: + timeout = 40; + + do { + if ((ac97c_readl(chip, COSR) & AC97C_CSR_TXRDY) != 0) { + ac97c_writel(chip, COTHR, word); + goto read_reg; + } + udelay(10); + } while (--timeout); + + if (!--write) + goto timed_out; + goto retry_write; + +read_reg: + do { + if ((ac97c_readl(chip, COSR) & AC97C_CSR_RXRDY) != 0) { + unsigned short val = ac97c_readl(chip, CORHR); + return val; + } + udelay(10); + } while (--timeout); + + if (!--write) + goto timed_out; + goto retry_write; + +timed_out: + dev_dbg(&chip->pdev->dev, "codec read timeout\n"); + return 0xffff; +} + +static bool filter(struct dma_chan *chan, void *slave) +{ + struct dw_dma_slave *dws = slave; + + if (dws->dma_dev == chan->device->dev) { + chan->private = dws; + return true; + } else + return false; +} + +static void atmel_ac97c_reset(struct atmel_ac97c *chip) +{ + ac97c_writel(chip, MR, AC97C_MR_WRST); + + if (gpio_is_valid(chip->reset_pin)) { + gpio_set_value(chip->reset_pin, 0); + /* AC97 v2.2 specifications says minimum 1 us. */ + udelay(10); + gpio_set_value(chip->reset_pin, 1); + } + + udelay(1); + ac97c_writel(chip, MR, AC97C_MR_ENA); +} + +static int __devinit atmel_ac97c_probe(struct platform_device *pdev) +{ + struct snd_card *card; + struct atmel_ac97c *chip; + struct resource *regs; + struct ac97c_platform_data *pdata; + struct clk *pclk; + static struct snd_ac97_bus_ops ops = { + .write = atmel_ac97c_write, + .read = atmel_ac97c_read, + }; + int retval; + + regs = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!regs) { + dev_dbg(&pdev->dev, "no memory resource\n"); + return -ENXIO; + } + + pdata = pdev->dev.platform_data; + if (!pdata) { + dev_dbg(&pdev->dev, "no platform data\n"); + return -ENXIO; + } + + pclk = clk_get(&pdev->dev, "pclk"); + if (IS_ERR(pclk)) { + dev_dbg(&pdev->dev, "no peripheral clock\n"); + return PTR_ERR(pclk); + } + clk_enable(pclk); + + retval = snd_card_create(SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1, + THIS_MODULE, sizeof(struct atmel_ac97c), &card); + if (retval) { + dev_dbg(&pdev->dev, "could not create sound card device\n"); + goto err_snd_card_new; + } + + chip = get_chip(card); + + spin_lock_init(&chip->lock); + + strcpy(card->driver, "Atmel AC97C"); + strcpy(card->shortname, "Atmel AC97C"); + sprintf(card->longname, "Atmel AC97 controller"); + + chip->card = card; + chip->pclk = pclk; + chip->pdev = pdev; + chip->regs = ioremap(regs->start, regs->end - regs->start + 1); + + if (!chip->regs) { + dev_dbg(&pdev->dev, "could not remap register memory\n"); + goto err_ioremap; + } + + if (gpio_is_valid(pdata->reset_pin)) { + if (gpio_request(pdata->reset_pin, "reset_pin")) { + dev_dbg(&pdev->dev, "reset pin not available\n"); + chip->reset_pin = -ENODEV; + } else { + gpio_direction_output(pdata->reset_pin, 1); + chip->reset_pin = pdata->reset_pin; + } + } + + snd_card_set_dev(card, &pdev->dev); + + retval = snd_ac97_bus(card, 0, &ops, chip, &chip->ac97_bus); + if (retval) { + dev_dbg(&pdev->dev, "could not register on ac97 bus\n"); + goto err_ac97_bus; + } + + atmel_ac97c_reset(chip); + + retval = atmel_ac97c_mixer_new(chip); + if (retval) { + dev_dbg(&pdev->dev, "could not register ac97 mixer\n"); + goto err_ac97_bus; + } + + if (pdata->rx_dws.dma_dev) { + struct dw_dma_slave *dws = &pdata->rx_dws; + dma_cap_mask_t mask; + + dws->rx_reg = regs->start + AC97C_CARHR + 2; + + dma_cap_zero(mask); + dma_cap_set(DMA_SLAVE, mask); + + chip->dma.rx_chan = dma_request_channel(mask, filter, dws); + + dev_info(&chip->pdev->dev, "using %s for DMA RX\n", + chip->dma.rx_chan->dev->device.bus_id); + set_bit(DMA_RX_CHAN_PRESENT, &chip->flags); + } + + if (pdata->tx_dws.dma_dev) { + struct dw_dma_slave *dws = &pdata->tx_dws; + dma_cap_mask_t mask; + + dws->tx_reg = regs->start + AC97C_CATHR + 2; + + dma_cap_zero(mask); + dma_cap_set(DMA_SLAVE, mask); + + chip->dma.tx_chan = dma_request_channel(mask, filter, dws); + + dev_info(&chip->pdev->dev, "using %s for DMA TX\n", + chip->dma.tx_chan->dev->device.bus_id); + set_bit(DMA_TX_CHAN_PRESENT, &chip->flags); + } + + if (!test_bit(DMA_RX_CHAN_PRESENT, &chip->flags) && + !test_bit(DMA_TX_CHAN_PRESENT, &chip->flags)) { + dev_dbg(&pdev->dev, "DMA not available\n"); + retval = -ENODEV; + goto err_dma; + } + + retval = atmel_ac97c_pcm_new(chip); + if (retval) { + dev_dbg(&pdev->dev, "could not register ac97 pcm device\n"); + goto err_dma; + } + + retval = snd_card_register(card); + if (retval) { + dev_dbg(&pdev->dev, "could not register sound card\n"); + goto err_ac97_bus; + } + + platform_set_drvdata(pdev, card); + + dev_info(&pdev->dev, "Atmel AC97 controller at 0x%p\n", + chip->regs); + + return 0; + +err_dma: + if (test_bit(DMA_RX_CHAN_PRESENT, &chip->flags)) + dma_release_channel(chip->dma.rx_chan); + if (test_bit(DMA_TX_CHAN_PRESENT, &chip->flags)) + dma_release_channel(chip->dma.tx_chan); + clear_bit(DMA_RX_CHAN_PRESENT, &chip->flags); + clear_bit(DMA_TX_CHAN_PRESENT, &chip->flags); + chip->dma.rx_chan = NULL; + chip->dma.tx_chan = NULL; +err_ac97_bus: + snd_card_set_dev(card, NULL); + + if (gpio_is_valid(chip->reset_pin)) + gpio_free(chip->reset_pin); + + iounmap(chip->regs); +err_ioremap: + snd_card_free(card); +err_snd_card_new: + clk_disable(pclk); + clk_put(pclk); + return retval; +} + +#ifdef CONFIG_PM +static int atmel_ac97c_suspend(struct platform_device *pdev, pm_message_t msg) +{ + struct snd_card *card = platform_get_drvdata(pdev); + struct atmel_ac97c *chip = card->private_data; + + if (test_bit(DMA_RX_READY, &chip->flags)) + dw_dma_cyclic_stop(chip->dma.rx_chan); + if (test_bit(DMA_TX_READY, &chip->flags)) + dw_dma_cyclic_stop(chip->dma.tx_chan); + clk_disable(chip->pclk); + + return 0; +} + +static int atmel_ac97c_resume(struct platform_device *pdev) +{ + struct snd_card *card = platform_get_drvdata(pdev); + struct atmel_ac97c *chip = card->private_data; + + clk_enable(chip->pclk); + if (test_bit(DMA_RX_READY, &chip->flags)) + dw_dma_cyclic_start(chip->dma.rx_chan); + if (test_bit(DMA_TX_READY, &chip->flags)) + dw_dma_cyclic_start(chip->dma.tx_chan); + + return 0; +} +#else +#define atmel_ac97c_suspend NULL +#define atmel_ac97c_resume NULL +#endif + +static int __devexit atmel_ac97c_remove(struct platform_device *pdev) +{ + struct snd_card *card = platform_get_drvdata(pdev); + struct atmel_ac97c *chip = get_chip(card); + + if (gpio_is_valid(chip->reset_pin)) + gpio_free(chip->reset_pin); + + clk_disable(chip->pclk); + clk_put(chip->pclk); + iounmap(chip->regs); + + if (test_bit(DMA_RX_CHAN_PRESENT, &chip->flags)) + dma_release_channel(chip->dma.rx_chan); + if (test_bit(DMA_TX_CHAN_PRESENT, &chip->flags)) + dma_release_channel(chip->dma.tx_chan); + clear_bit(DMA_RX_CHAN_PRESENT, &chip->flags); + clear_bit(DMA_TX_CHAN_PRESENT, &chip->flags); + chip->dma.rx_chan = NULL; + chip->dma.tx_chan = NULL; + + snd_card_set_dev(card, NULL); + snd_card_free(card); + + platform_set_drvdata(pdev, NULL); + + return 0; +} + +static struct platform_driver atmel_ac97c_driver = { + .remove = __devexit_p(atmel_ac97c_remove), + .driver = { + .name = "atmel_ac97c", + }, + .suspend = atmel_ac97c_suspend, + .resume = atmel_ac97c_resume, +}; + +static int __init atmel_ac97c_init(void) +{ + return platform_driver_probe(&atmel_ac97c_driver, + atmel_ac97c_probe); +} +module_init(atmel_ac97c_init); + +static void __exit atmel_ac97c_exit(void) +{ + platform_driver_unregister(&atmel_ac97c_driver); +} +module_exit(atmel_ac97c_exit); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Driver for Atmel AC97 controller"); +MODULE_AUTHOR("Hans-Christian Egtvedt "); diff --git a/sound/atmel/ac97c.h b/sound/atmel/ac97c.h new file mode 100644 index 00000000000..c17bd582598 --- /dev/null +++ b/sound/atmel/ac97c.h @@ -0,0 +1,71 @@ +/* + * Register definitions for the Atmel AC97C controller + * + * Copyright (C) 2005-2009 Atmel Corporation + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License version 2 as published + * by the Free Software Foundation. + */ +#ifndef __SOUND_ATMEL_AC97C_H +#define __SOUND_ATMEL_AC97C_H + +#define AC97C_MR 0x08 +#define AC97C_ICA 0x10 +#define AC97C_OCA 0x14 +#define AC97C_CARHR 0x20 +#define AC97C_CATHR 0x24 +#define AC97C_CASR 0x28 +#define AC97C_CAMR 0x2c +#define AC97C_CBRHR 0x30 +#define AC97C_CBTHR 0x34 +#define AC97C_CBSR 0x38 +#define AC97C_CBMR 0x3c +#define AC97C_CORHR 0x40 +#define AC97C_COTHR 0x44 +#define AC97C_COSR 0x48 +#define AC97C_COMR 0x4c +#define AC97C_SR 0x50 +#define AC97C_IER 0x54 +#define AC97C_IDR 0x58 +#define AC97C_IMR 0x5c +#define AC97C_VERSION 0xfc + +#define AC97C_CATPR PDC_TPR +#define AC97C_CATCR PDC_TCR +#define AC97C_CATNPR PDC_TNPR +#define AC97C_CATNCR PDC_TNCR +#define AC97C_CARPR PDC_RPR +#define AC97C_CARCR PDC_RCR +#define AC97C_CARNPR PDC_RNPR +#define AC97C_CARNCR PDC_RNCR +#define AC97C_PTCR PDC_PTCR + +#define AC97C_MR_ENA (1 << 0) +#define AC97C_MR_WRST (1 << 1) +#define AC97C_MR_VRA (1 << 2) + +#define AC97C_CSR_TXRDY (1 << 0) +#define AC97C_CSR_UNRUN (1 << 2) +#define AC97C_CSR_RXRDY (1 << 4) +#define AC97C_CSR_ENDTX (1 << 10) +#define AC97C_CSR_ENDRX (1 << 14) + +#define AC97C_CMR_SIZE_20 (0 << 16) +#define AC97C_CMR_SIZE_18 (1 << 16) +#define AC97C_CMR_SIZE_16 (2 << 16) +#define AC97C_CMR_SIZE_10 (3 << 16) +#define AC97C_CMR_CEM_LITTLE (1 << 18) +#define AC97C_CMR_CEM_BIG (0 << 18) +#define AC97C_CMR_CENA (1 << 21) +#define AC97C_CMR_DMAEN (1 << 22) + +#define AC97C_SR_CAEVT (1 << 3) + +#define AC97C_CH_ASSIGN(slot, channel) \ + (AC97C_CHANNEL_##channel << (3 * (AC97_SLOT_##slot - 3))) +#define AC97C_CHANNEL_NONE 0x0 +#define AC97C_CHANNEL_A 0x1 +#define AC97C_CHANNEL_B 0x2 + +#endif /* __SOUND_ATMEL_AC97C_H */ -- cgit v1.2.3-70-g09d2 From 6c7578bb0a631d018a68e5f90554f29fbd928055 Mon Sep 17 00:00:00 2001 From: Hans-Christian Egtvedt Date: Thu, 5 Feb 2009 13:11:01 +0100 Subject: ALSA: Add Atmel ALSA drivers directory Signed-off-by: Hans-Christian Egtvedt Signed-off-by: Haavard Skinnemoen Signed-off-by: Takashi Iwai --- sound/Kconfig | 2 ++ sound/Makefile | 2 +- 2 files changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/Kconfig b/sound/Kconfig index 200aca1faa7..1eceb85287c 100644 --- a/sound/Kconfig +++ b/sound/Kconfig @@ -60,6 +60,8 @@ source "sound/aoa/Kconfig" source "sound/arm/Kconfig" +source "sound/atmel/Kconfig" + source "sound/spi/Kconfig" source "sound/mips/Kconfig" diff --git a/sound/Makefile b/sound/Makefile index c76d70716fa..ec467decfa7 100644 --- a/sound/Makefile +++ b/sound/Makefile @@ -6,7 +6,7 @@ obj-$(CONFIG_SOUND_PRIME) += sound_firmware.o obj-$(CONFIG_SOUND_PRIME) += oss/ obj-$(CONFIG_DMASOUND) += oss/ obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ sh/ synth/ usb/ \ - sparc/ spi/ parisc/ pcmcia/ mips/ soc/ + sparc/ spi/ parisc/ pcmcia/ mips/ soc/ atmel/ obj-$(CONFIG_SND_AOA) += aoa/ # This one must be compilable even if sound is configured out -- cgit v1.2.3-70-g09d2 From 76d498e43fa5f9f0a148dca8915cc7e9d9b9a643 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 5 Feb 2009 15:45:05 +0100 Subject: ALSA: wss - Add missing KERN_* prefix to printk Signed-off-by: Takashi Iwai --- sound/isa/wss/wss_lib.c | 76 ++++++++++++++++++++++++++++++++++--------------- 1 file changed, 53 insertions(+), 23 deletions(-) diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c index 3d6c5f2838a..8de5deda7ad 100644 --- a/sound/isa/wss/wss_lib.c +++ b/sound/isa/wss/wss_lib.c @@ -219,7 +219,8 @@ void snd_wss_out(struct snd_wss *chip, unsigned char reg, unsigned char value) snd_wss_wait(chip); #ifdef CONFIG_SND_DEBUG if (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) - snd_printk("out: auto calibration time out - reg = 0x%x, value = 0x%x\n", reg, value); + snd_printk(KERN_DEBUG "out: auto calibration time out " + "- reg = 0x%x, value = 0x%x\n", reg, value); #endif wss_outb(chip, CS4231P(REGSEL), chip->mce_bit | reg); wss_outb(chip, CS4231P(REG), value); @@ -235,7 +236,8 @@ unsigned char snd_wss_in(struct snd_wss *chip, unsigned char reg) snd_wss_wait(chip); #ifdef CONFIG_SND_DEBUG if (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) - snd_printk("in: auto calibration time out - reg = 0x%x\n", reg); + snd_printk(KERN_DEBUG "in: auto calibration time out " + "- reg = 0x%x\n", reg); #endif wss_outb(chip, CS4231P(REGSEL), chip->mce_bit | reg); mb(); @@ -252,7 +254,7 @@ void snd_cs4236_ext_out(struct snd_wss *chip, unsigned char reg, wss_outb(chip, CS4231P(REG), val); chip->eimage[CS4236_REG(reg)] = val; #if 0 - printk("ext out : reg = 0x%x, val = 0x%x\n", reg, val); + printk(KERN_DEBUG "ext out : reg = 0x%x, val = 0x%x\n", reg, val); #endif } EXPORT_SYMBOL(snd_cs4236_ext_out); @@ -268,7 +270,8 @@ unsigned char snd_cs4236_ext_in(struct snd_wss *chip, unsigned char reg) { unsigned char res; res = wss_inb(chip, CS4231P(REG)); - printk("ext in : reg = 0x%x, val = 0x%x\n", reg, res); + printk(KERN_DEBUG "ext in : reg = 0x%x, val = 0x%x\n", + reg, res); return res; } #endif @@ -394,13 +397,16 @@ void snd_wss_mce_up(struct snd_wss *chip) snd_wss_wait(chip); #ifdef CONFIG_SND_DEBUG if (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) - snd_printk("mce_up - auto calibration time out (0)\n"); + snd_printk(KERN_DEBUG + "mce_up - auto calibration time out (0)\n"); #endif spin_lock_irqsave(&chip->reg_lock, flags); chip->mce_bit |= CS4231_MCE; timeout = wss_inb(chip, CS4231P(REGSEL)); if (timeout == 0x80) - snd_printk("mce_up [0x%lx]: serious init problem - codec still busy\n", chip->port); + snd_printk(KERN_DEBUG "mce_up [0x%lx]: " + "serious init problem - codec still busy\n", + chip->port); if (!(timeout & CS4231_MCE)) wss_outb(chip, CS4231P(REGSEL), chip->mce_bit | (timeout & 0x1f)); @@ -419,7 +425,9 @@ void snd_wss_mce_down(struct snd_wss *chip) #ifdef CONFIG_SND_DEBUG if (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) - snd_printk("mce_down [0x%lx] - auto calibration time out (0)\n", (long)CS4231P(REGSEL)); + snd_printk(KERN_DEBUG "mce_down [0x%lx] - " + "auto calibration time out (0)\n", + (long)CS4231P(REGSEL)); #endif spin_lock_irqsave(&chip->reg_lock, flags); chip->mce_bit &= ~CS4231_MCE; @@ -427,7 +435,9 @@ void snd_wss_mce_down(struct snd_wss *chip) wss_outb(chip, CS4231P(REGSEL), chip->mce_bit | (timeout & 0x1f)); spin_unlock_irqrestore(&chip->reg_lock, flags); if (timeout == 0x80) - snd_printk("mce_down [0x%lx]: serious init problem - codec still busy\n", chip->port); + snd_printk(KERN_DEBUG "mce_down [0x%lx]: " + "serious init problem - codec still busy\n", + chip->port); if ((timeout & CS4231_MCE) == 0 || !(chip->hardware & hw_mask)) return; @@ -565,7 +575,7 @@ static unsigned char snd_wss_get_format(struct snd_wss *chip, if (channels > 1) rformat |= CS4231_STEREO; #if 0 - snd_printk("get_format: 0x%x (mode=0x%x)\n", format, mode); + snd_printk(KERN_DEBUG "get_format: 0x%x (mode=0x%x)\n", format, mode); #endif return rformat; } @@ -774,7 +784,7 @@ static void snd_wss_init(struct snd_wss *chip) snd_wss_mce_down(chip); #ifdef SNDRV_DEBUG_MCE - snd_printk("init: (1)\n"); + snd_printk(KERN_DEBUG "init: (1)\n"); #endif snd_wss_mce_up(chip); spin_lock_irqsave(&chip->reg_lock, flags); @@ -789,7 +799,7 @@ static void snd_wss_init(struct snd_wss *chip) snd_wss_mce_down(chip); #ifdef SNDRV_DEBUG_MCE - snd_printk("init: (2)\n"); + snd_printk(KERN_DEBUG "init: (2)\n"); #endif snd_wss_mce_up(chip); @@ -800,7 +810,7 @@ static void snd_wss_init(struct snd_wss *chip) snd_wss_mce_down(chip); #ifdef SNDRV_DEBUG_MCE - snd_printk("init: (3) - afei = 0x%x\n", + snd_printk(KERN_DEBUG "init: (3) - afei = 0x%x\n", chip->image[CS4231_ALT_FEATURE_1]); #endif @@ -817,7 +827,7 @@ static void snd_wss_init(struct snd_wss *chip) snd_wss_mce_down(chip); #ifdef SNDRV_DEBUG_MCE - snd_printk("init: (4)\n"); + snd_printk(KERN_DEBUG "init: (4)\n"); #endif snd_wss_mce_up(chip); @@ -829,7 +839,7 @@ static void snd_wss_init(struct snd_wss *chip) snd_wss_mce_down(chip); #ifdef SNDRV_DEBUG_MCE - snd_printk("init: (5)\n"); + snd_printk(KERN_DEBUG "init: (5)\n"); #endif } @@ -1278,7 +1288,8 @@ static int snd_wss_probe(struct snd_wss *chip) } else if (rev == 0x03) { chip->hardware = WSS_HW_CS4236B; } else { - snd_printk("unknown CS chip with version 0x%x\n", rev); + snd_printk(KERN_ERR + "unknown CS chip with version 0x%x\n", rev); return -ENODEV; /* unknown CS4231 chip? */ } } @@ -1342,7 +1353,10 @@ static int snd_wss_probe(struct snd_wss *chip) case 6: break; default: - snd_printk("unknown CS4235 chip (enhanced version = 0x%x)\n", id); + snd_printk(KERN_WARNING + "unknown CS4235 chip " + "(enhanced version = 0x%x)\n", + id); } } else if ((id & 0x1f) == 0x0b) { /* CS4236/B */ switch (id >> 5) { @@ -1353,7 +1367,10 @@ static int snd_wss_probe(struct snd_wss *chip) chip->hardware = WSS_HW_CS4236B; break; default: - snd_printk("unknown CS4236 chip (enhanced version = 0x%x)\n", id); + snd_printk(KERN_WARNING + "unknown CS4236 chip " + "(enhanced version = 0x%x)\n", + id); } } else if ((id & 0x1f) == 0x08) { /* CS4237B */ chip->hardware = WSS_HW_CS4237B; @@ -1364,7 +1381,10 @@ static int snd_wss_probe(struct snd_wss *chip) case 7: break; default: - snd_printk("unknown CS4237B chip (enhanced version = 0x%x)\n", id); + snd_printk(KERN_WARNING + "unknown CS4237B chip " + "(enhanced version = 0x%x)\n", + id); } } else if ((id & 0x1f) == 0x09) { /* CS4238B */ chip->hardware = WSS_HW_CS4238B; @@ -1374,7 +1394,10 @@ static int snd_wss_probe(struct snd_wss *chip) case 7: break; default: - snd_printk("unknown CS4238B chip (enhanced version = 0x%x)\n", id); + snd_printk(KERN_WARNING + "unknown CS4238B chip " + "(enhanced version = 0x%x)\n", + id); } } else if ((id & 0x1f) == 0x1e) { /* CS4239 */ chip->hardware = WSS_HW_CS4239; @@ -1384,10 +1407,15 @@ static int snd_wss_probe(struct snd_wss *chip) case 6: break; default: - snd_printk("unknown CS4239 chip (enhanced version = 0x%x)\n", id); + snd_printk(KERN_WARNING + "unknown CS4239 chip " + "(enhanced version = 0x%x)\n", + id); } } else { - snd_printk("unknown CS4236/CS423xB chip (enhanced version = 0x%x)\n", id); + snd_printk(KERN_WARNING + "unknown CS4236/CS423xB chip " + "(enhanced version = 0x%x)\n", id); } } } @@ -1618,7 +1646,8 @@ static void snd_wss_resume(struct snd_wss *chip) wss_outb(chip, CS4231P(REGSEL), chip->mce_bit | (timeout & 0x1f)); spin_unlock_irqrestore(&chip->reg_lock, flags); if (timeout == 0x80) - snd_printk("down [0x%lx]: serious init problem - codec still busy\n", chip->port); + snd_printk(KERN_ERR "down [0x%lx]: serious init problem " + "- codec still busy\n", chip->port); if ((timeout & CS4231_MCE) == 0 || !(chip->hardware & (WSS_HW_CS4231_MASK | WSS_HW_CS4232_MASK))) { return; @@ -1820,7 +1849,8 @@ int snd_wss_create(struct snd_card *card, #if 0 if (chip->hardware & WSS_HW_CS4232_MASK) { if (chip->res_cport == NULL) - snd_printk("CS4232 control port features are not accessible\n"); + snd_printk(KERN_ERR "CS4232 control port features are " + "not accessible\n"); } #endif -- cgit v1.2.3-70-g09d2 From 91f050604cc045a0b7aa0460d36eb6e0f0cb301a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 5 Feb 2009 15:46:48 +0100 Subject: ALSA: gus - Add missing KERN_* prefix to printk Signed-off-by: Takashi Iwai --- sound/isa/gus/gus_dma.c | 3 ++- sound/isa/gus/gus_irq.c | 6 ++++-- sound/isa/gus/gus_pcm.c | 26 ++++++++++++++++++++------ sound/isa/gus/gus_uart.c | 10 ++++++++-- sound/isa/gus/interwave.c | 16 +++++++++------- 5 files changed, 43 insertions(+), 18 deletions(-) diff --git a/sound/isa/gus/gus_dma.c b/sound/isa/gus/gus_dma.c index cf8cd3c26a5..36c27c83236 100644 --- a/sound/isa/gus/gus_dma.c +++ b/sound/isa/gus/gus_dma.c @@ -78,7 +78,8 @@ static void snd_gf1_dma_program(struct snd_gus_card * gus, snd_gf1_dma_ack(gus); snd_dma_program(gus->gf1.dma1, buf_addr, count, dma_cmd & SNDRV_GF1_DMA_READ ? DMA_MODE_READ : DMA_MODE_WRITE); #if 0 - snd_printk("address = 0x%x, count = 0x%x, dma_cmd = 0x%x\n", address << 1, count, dma_cmd); + snd_printk(KERN_DEBUG "address = 0x%x, count = 0x%x, dma_cmd = 0x%x\n", + address << 1, count, dma_cmd); #endif spin_lock_irqsave(&gus->reg_lock, flags); if (gus->gf1.enh_mode) { diff --git a/sound/isa/gus/gus_irq.c b/sound/isa/gus/gus_irq.c index 041894ddd01..2055aff71b5 100644 --- a/sound/isa/gus/gus_irq.c +++ b/sound/isa/gus/gus_irq.c @@ -41,7 +41,7 @@ __again: if (status == 0) return IRQ_RETVAL(handled); handled = 1; - // snd_printk("IRQ: status = 0x%x\n", status); + /* snd_printk(KERN_DEBUG "IRQ: status = 0x%x\n", status); */ if (status & 0x02) { STAT_ADD(gus->gf1.interrupt_stat_midi_in); if (gus->gf1.interrupt_handler_midi_in) @@ -65,7 +65,9 @@ __again: continue; /* multi request */ already |= _current_; /* mark request */ #if 0 - printk("voice = %i, voice_status = 0x%x, voice_verify = %i\n", voice, voice_status, inb(GUSP(gus, GF1PAGE))); + printk(KERN_DEBUG "voice = %i, voice_status = 0x%x, " + "voice_verify = %i\n", + voice, voice_status, inb(GUSP(gus, GF1PAGE))); #endif pvoice = &gus->gf1.voices[voice]; if (pvoice->use) { diff --git a/sound/isa/gus/gus_pcm.c b/sound/isa/gus/gus_pcm.c index 38510aeb21c..edb11eefdfe 100644 --- a/sound/isa/gus/gus_pcm.c +++ b/sound/isa/gus/gus_pcm.c @@ -82,7 +82,10 @@ static int snd_gf1_pcm_block_change(struct snd_pcm_substream *substream, count += offset & 31; offset &= ~31; - // snd_printk("block change - offset = 0x%x, count = 0x%x\n", offset, count); + /* + snd_printk(KERN_DEBUG "block change - offset = 0x%x, count = 0x%x\n", + offset, count); + */ memset(&block, 0, sizeof(block)); block.cmd = SNDRV_GF1_DMA_IRQ; if (snd_pcm_format_unsigned(runtime->format)) @@ -135,7 +138,11 @@ static void snd_gf1_pcm_trigger_up(struct snd_pcm_substream *substream) curr = begin + (pcmp->bpos * pcmp->block_size) / runtime->channels; end = curr + (pcmp->block_size / runtime->channels); end -= snd_pcm_format_width(runtime->format) == 16 ? 2 : 1; - // snd_printk("init: curr=0x%x, begin=0x%x, end=0x%x, ctrl=0x%x, ramp=0x%x, rate=0x%x\n", curr, begin, end, voice_ctrl, ramp_ctrl, rate); + /* + snd_printk(KERN_DEBUG "init: curr=0x%x, begin=0x%x, end=0x%x, " + "ctrl=0x%x, ramp=0x%x, rate=0x%x\n", + curr, begin, end, voice_ctrl, ramp_ctrl, rate); + */ pan = runtime->channels == 2 ? (!voice ? 1 : 14) : 8; vol = !voice ? gus->gf1.pcm_volume_level_left : gus->gf1.pcm_volume_level_right; spin_lock_irqsave(&gus->reg_lock, flags); @@ -205,9 +212,11 @@ static void snd_gf1_pcm_interrupt_wave(struct snd_gus_card * gus, ramp_ctrl = (snd_gf1_read8(gus, SNDRV_GF1_VB_VOLUME_CONTROL) & ~0xa4) | 0x03; #if 0 snd_gf1_select_voice(gus, pvoice->number); - printk("position = 0x%x\n", (snd_gf1_read_addr(gus, SNDRV_GF1_VA_CURRENT, voice_ctrl & 4) >> 4)); + printk(KERN_DEBUG "position = 0x%x\n", + (snd_gf1_read_addr(gus, SNDRV_GF1_VA_CURRENT, voice_ctrl & 4) >> 4)); snd_gf1_select_voice(gus, pcmp->pvoices[1]->number); - printk("position = 0x%x\n", (snd_gf1_read_addr(gus, SNDRV_GF1_VA_CURRENT, voice_ctrl & 4) >> 4)); + printk(KERN_DEBUG "position = 0x%x\n", + (snd_gf1_read_addr(gus, SNDRV_GF1_VA_CURRENT, voice_ctrl & 4) >> 4)); snd_gf1_select_voice(gus, pvoice->number); #endif pcmp->bpos++; @@ -299,7 +308,11 @@ static int snd_gf1_pcm_poke_block(struct snd_gus_card *gus, unsigned char *buf, unsigned int len; unsigned long flags; - // printk("poke block; buf = 0x%x, pos = %i, count = %i, port = 0x%x\n", (int)buf, pos, count, gus->gf1.port); + /* + printk(KERN_DEBUG + "poke block; buf = 0x%x, pos = %i, count = %i, port = 0x%x\n", + (int)buf, pos, count, gus->gf1.port); + */ while (count > 0) { len = count; if (len > 512) /* limit, to allow IRQ */ @@ -680,7 +693,8 @@ static int snd_gf1_pcm_playback_open(struct snd_pcm_substream *substream) runtime->private_free = snd_gf1_pcm_playback_free; #if 0 - printk("playback.buffer = 0x%lx, gf1.pcm_buffer = 0x%lx\n", (long) pcm->playback.buffer, (long) gus->gf1.pcm_buffer); + printk(KERN_DEBUG "playback.buffer = 0x%lx, gf1.pcm_buffer = 0x%lx\n", + (long) pcm->playback.buffer, (long) gus->gf1.pcm_buffer); #endif if ((err = snd_gf1_dma_init(gus)) < 0) return err; diff --git a/sound/isa/gus/gus_uart.c b/sound/isa/gus/gus_uart.c index f0af3f79b08..21cc42e4c4b 100644 --- a/sound/isa/gus/gus_uart.c +++ b/sound/isa/gus/gus_uart.c @@ -129,8 +129,14 @@ static int snd_gf1_uart_input_open(struct snd_rawmidi_substream *substream) } spin_unlock_irqrestore(&gus->uart_cmd_lock, flags); #if 0 - snd_printk("read init - enable = %i, cmd = 0x%x, stat = 0x%x\n", gus->uart_enable, gus->gf1.uart_cmd, snd_gf1_uart_stat(gus)); - snd_printk("[0x%x] reg (ctrl/status) = 0x%x, reg (data) = 0x%x (page = 0x%x)\n", gus->gf1.port + 0x100, inb(gus->gf1.port + 0x100), inb(gus->gf1.port + 0x101), inb(gus->gf1.port + 0x102)); + snd_printk(KERN_DEBUG + "read init - enable = %i, cmd = 0x%x, stat = 0x%x\n", + gus->uart_enable, gus->gf1.uart_cmd, snd_gf1_uart_stat(gus)); + snd_printk(KERN_DEBUG + "[0x%x] reg (ctrl/status) = 0x%x, reg (data) = 0x%x " + "(page = 0x%x)\n", + gus->gf1.port + 0x100, inb(gus->gf1.port + 0x100), + inb(gus->gf1.port + 0x101), inb(gus->gf1.port + 0x102)); #endif return 0; } diff --git a/sound/isa/gus/interwave.c b/sound/isa/gus/interwave.c index 5faecfb602d..418d49eef92 100644 --- a/sound/isa/gus/interwave.c +++ b/sound/isa/gus/interwave.c @@ -170,7 +170,7 @@ static void snd_interwave_i2c_setlines(struct snd_i2c_bus *bus, int ctrl, int da unsigned long port = bus->private_value; #if 0 - printk("i2c_setlines - 0x%lx <- %i,%i\n", port, ctrl, data); + printk(KERN_DEBUG "i2c_setlines - 0x%lx <- %i,%i\n", port, ctrl, data); #endif outb((data << 1) | ctrl, port); udelay(10); @@ -183,7 +183,7 @@ static int snd_interwave_i2c_getclockline(struct snd_i2c_bus *bus) res = inb(port) & 1; #if 0 - printk("i2c_getclockline - 0x%lx -> %i\n", port, res); + printk(KERN_DEBUG "i2c_getclockline - 0x%lx -> %i\n", port, res); #endif return res; } @@ -197,7 +197,7 @@ static int snd_interwave_i2c_getdataline(struct snd_i2c_bus *bus, int ack) udelay(10); res = (inb(port) & 2) >> 1; #if 0 - printk("i2c_getdataline - 0x%lx -> %i\n", port, res); + printk(KERN_DEBUG "i2c_getdataline - 0x%lx -> %i\n", port, res); #endif return res; } @@ -342,7 +342,8 @@ static void __devinit snd_interwave_bank_sizes(struct snd_gus_card * gus, int *s snd_gf1_poke(gus, local, d); snd_gf1_poke(gus, local + 1, d + 1); #if 0 - printk("d = 0x%x, local = 0x%x, local + 1 = 0x%x, idx << 22 = 0x%x\n", + printk(KERN_DEBUG "d = 0x%x, local = 0x%x, " + "local + 1 = 0x%x, idx << 22 = 0x%x\n", d, snd_gf1_peek(gus, local), snd_gf1_peek(gus, local + 1), @@ -356,7 +357,8 @@ static void __devinit snd_interwave_bank_sizes(struct snd_gus_card * gus, int *s } } #if 0 - printk("sizes: %i %i %i %i\n", sizes[0], sizes[1], sizes[2], sizes[3]); + printk(KERN_DEBUG "sizes: %i %i %i %i\n", + sizes[0], sizes[1], sizes[2], sizes[3]); #endif } @@ -410,12 +412,12 @@ static void __devinit snd_interwave_detect_memory(struct snd_gus_card * gus) lmct = (psizes[3] << 24) | (psizes[2] << 16) | (psizes[1] << 8) | psizes[0]; #if 0 - printk("lmct = 0x%08x\n", lmct); + printk(KERN_DEBUG "lmct = 0x%08x\n", lmct); #endif for (i = 0; i < ARRAY_SIZE(lmc); i++) if (lmct == lmc[i]) { #if 0 - printk("found !!! %i\n", i); + printk(KERN_DEBUG "found !!! %i\n", i); #endif snd_gf1_write16(gus, SNDRV_GF1_GW_MEMORY_CONFIG, (snd_gf1_look16(gus, SNDRV_GF1_GW_MEMORY_CONFIG) & 0xfff0) | i); snd_interwave_bank_sizes(gus, psizes); -- cgit v1.2.3-70-g09d2 From 4c9f1d3ed7e5f910b66dc4d1456cfac17e58cf0e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 5 Feb 2009 15:47:51 +0100 Subject: ALSA: isa/*: Add missing KERN_* prefix to printk Signed-off-by: Takashi Iwai --- sound/isa/ad1816a/ad1816a_lib.c | 6 +++--- sound/isa/cs423x/cs4236_lib.c | 21 ++++++++++++++------- sound/isa/es1688/es1688_lib.c | 23 +++++++++++++++-------- sound/isa/opl3sa2.c | 10 +++++++--- sound/isa/opti9xx/opti92x-ad1848.c | 30 +++++++++++++++++------------- sound/isa/wavefront/wavefront.c | 4 ++-- sound/isa/wavefront/wavefront_synth.c | 2 +- 7 files changed, 59 insertions(+), 37 deletions(-) diff --git a/sound/isa/ad1816a/ad1816a_lib.c b/sound/isa/ad1816a/ad1816a_lib.c index 1c9e01ecac0..05aef8b97e9 100644 --- a/sound/isa/ad1816a/ad1816a_lib.c +++ b/sound/isa/ad1816a/ad1816a_lib.c @@ -37,7 +37,7 @@ static inline int snd_ad1816a_busy_wait(struct snd_ad1816a *chip) if (inb(AD1816A_REG(AD1816A_CHIP_STATUS)) & AD1816A_READY) return 0; - snd_printk("chip busy.\n"); + snd_printk(KERN_WARNING "chip busy.\n"); return -EBUSY; } @@ -196,7 +196,7 @@ static int snd_ad1816a_trigger(struct snd_ad1816a *chip, unsigned char what, spin_unlock(&chip->lock); break; default: - snd_printk("invalid trigger mode 0x%x.\n", what); + snd_printk(KERN_WARNING "invalid trigger mode 0x%x.\n", what); error = -EINVAL; } @@ -565,7 +565,7 @@ static const char __devinit *snd_ad1816a_chip_id(struct snd_ad1816a *chip) case AD1816A_HW_AD1815: return "AD1815"; case AD1816A_HW_AD18MAX10: return "AD18max10"; default: - snd_printk("Unknown chip version %d:%d.\n", + snd_printk(KERN_WARNING "Unknown chip version %d:%d.\n", chip->version, chip->hardware); return "AD1816A - unknown"; } diff --git a/sound/isa/cs423x/cs4236_lib.c b/sound/isa/cs423x/cs4236_lib.c index 6a85fdc53b6..2406efdfd8d 100644 --- a/sound/isa/cs423x/cs4236_lib.c +++ b/sound/isa/cs423x/cs4236_lib.c @@ -286,7 +286,8 @@ int snd_cs4236_create(struct snd_card *card, if (hardware == WSS_HW_DETECT) hardware = WSS_HW_DETECT3; if (cport < 0x100) { - snd_printk("please, specify control port for CS4236+ chips\n"); + snd_printk(KERN_ERR "please, specify control port " + "for CS4236+ chips\n"); return -ENODEV; } err = snd_wss_create(card, port, cport, @@ -295,7 +296,8 @@ int snd_cs4236_create(struct snd_card *card, return err; if (!(chip->hardware & WSS_HW_CS4236B_MASK)) { - snd_printk("CS4236+: MODE3 and extended registers not available, hardware=0x%x\n",chip->hardware); + snd_printk(KERN_ERR "CS4236+: MODE3 and extended registers " + "not available, hardware=0x%x\n", chip->hardware); snd_device_free(card, chip); return -ENODEV; } @@ -303,16 +305,19 @@ int snd_cs4236_create(struct snd_card *card, { int idx; for (idx = 0; idx < 8; idx++) - snd_printk("CD%i = 0x%x\n", idx, inb(chip->cport + idx)); + snd_printk(KERN_DEBUG "CD%i = 0x%x\n", + idx, inb(chip->cport + idx)); for (idx = 0; idx < 9; idx++) - snd_printk("C%i = 0x%x\n", idx, snd_cs4236_ctrl_in(chip, idx)); + snd_printk(KERN_DEBUG "C%i = 0x%x\n", + idx, snd_cs4236_ctrl_in(chip, idx)); } #endif ver1 = snd_cs4236_ctrl_in(chip, 1); ver2 = snd_cs4236_ext_in(chip, CS4236_VERSION); snd_printdd("CS4236: [0x%lx] C1 (version) = 0x%x, ext = 0x%x\n", cport, ver1, ver2); if (ver1 != ver2) { - snd_printk("CS4236+ chip detected, but control port 0x%lx is not valid\n", cport); + snd_printk(KERN_ERR "CS4236+ chip detected, but " + "control port 0x%lx is not valid\n", cport); snd_device_free(card, chip); return -ENODEV; } @@ -883,7 +888,8 @@ static int snd_cs4236_get_iec958_switch(struct snd_kcontrol *kcontrol, struct sn spin_lock_irqsave(&chip->reg_lock, flags); ucontrol->value.integer.value[0] = chip->image[CS4231_ALT_FEATURE_1] & 0x02 ? 1 : 0; #if 0 - printk("get valid: ALT = 0x%x, C3 = 0x%x, C4 = 0x%x, C5 = 0x%x, C6 = 0x%x, C8 = 0x%x\n", + printk(KERN_DEBUG "get valid: ALT = 0x%x, C3 = 0x%x, C4 = 0x%x, " + "C5 = 0x%x, C6 = 0x%x, C8 = 0x%x\n", snd_wss_in(chip, CS4231_ALT_FEATURE_1), snd_cs4236_ctrl_in(chip, 3), snd_cs4236_ctrl_in(chip, 4), @@ -920,7 +926,8 @@ static int snd_cs4236_put_iec958_switch(struct snd_kcontrol *kcontrol, struct sn mutex_unlock(&chip->mce_mutex); #if 0 - printk("set valid: ALT = 0x%x, C3 = 0x%x, C4 = 0x%x, C5 = 0x%x, C6 = 0x%x, C8 = 0x%x\n", + printk(KERN_DEBUG "set valid: ALT = 0x%x, C3 = 0x%x, C4 = 0x%x, " + "C5 = 0x%x, C6 = 0x%x, C8 = 0x%x\n", snd_wss_in(chip, CS4231_ALT_FEATURE_1), snd_cs4236_ctrl_in(chip, 3), snd_cs4236_ctrl_in(chip, 4), diff --git a/sound/isa/es1688/es1688_lib.c b/sound/isa/es1688/es1688_lib.c index 4fbb508a817..4c6e14f87f2 100644 --- a/sound/isa/es1688/es1688_lib.c +++ b/sound/isa/es1688/es1688_lib.c @@ -45,7 +45,7 @@ static int snd_es1688_dsp_command(struct snd_es1688 *chip, unsigned char val) return 1; } #ifdef CONFIG_SND_DEBUG - printk("snd_es1688_dsp_command: timeout (0x%x)\n", val); + printk(KERN_DEBUG "snd_es1688_dsp_command: timeout (0x%x)\n", val); #endif return 0; } @@ -167,13 +167,16 @@ static int snd_es1688_probe(struct snd_es1688 *chip) hw = ES1688_HW_AUTO; switch (chip->version & 0xfff0) { case 0x4880: - snd_printk("[0x%lx] ESS: AudioDrive ES488 detected, but driver is in another place\n", chip->port); + snd_printk(KERN_ERR "[0x%lx] ESS: AudioDrive ES488 detected, " + "but driver is in another place\n", chip->port); return -ENODEV; case 0x6880: hw = (chip->version & 0x0f) >= 8 ? ES1688_HW_1688 : ES1688_HW_688; break; default: - snd_printk("[0x%lx] ESS: unknown AudioDrive chip with version 0x%x (Jazz16 soundcard?)\n", chip->port, chip->version); + snd_printk(KERN_ERR "[0x%lx] ESS: unknown AudioDrive chip " + "with version 0x%x (Jazz16 soundcard?)\n", + chip->port, chip->version); return -ENODEV; } @@ -223,7 +226,7 @@ static int snd_es1688_init(struct snd_es1688 * chip, int enable) } } #if 0 - snd_printk("mpu cfg = 0x%x\n", cfg); + snd_printk(KERN_DEBUG "mpu cfg = 0x%x\n", cfg); #endif spin_lock_irqsave(&chip->reg_lock, flags); snd_es1688_mixer_write(chip, 0x40, cfg); @@ -237,7 +240,9 @@ static int snd_es1688_init(struct snd_es1688 * chip, int enable) cfg = 0xf0; /* enable only DMA counter interrupt */ irq_bits = irqs[chip->irq & 0x0f]; if (irq_bits < 0) { - snd_printk("[0x%lx] ESS: bad IRQ %d for ES1688 chip!!\n", chip->port, chip->irq); + snd_printk(KERN_ERR "[0x%lx] ESS: bad IRQ %d " + "for ES1688 chip!!\n", + chip->port, chip->irq); #if 0 irq_bits = 0; cfg = 0x10; @@ -250,7 +255,8 @@ static int snd_es1688_init(struct snd_es1688 * chip, int enable) cfg = 0xf0; /* extended mode DMA enable */ dma = chip->dma8; if (dma > 3 || dma == 2) { - snd_printk("[0x%lx] ESS: bad DMA channel %d for ES1688 chip!!\n", chip->port, dma); + snd_printk(KERN_ERR "[0x%lx] ESS: bad DMA channel %d " + "for ES1688 chip!!\n", chip->port, dma); #if 0 dma_bits = 0; cfg = 0x00; /* disable all DMA */ @@ -341,8 +347,9 @@ static int snd_es1688_trigger(struct snd_es1688 *chip, int cmd, unsigned char va return -EINVAL; /* something is wrong */ } #if 0 - printk("trigger: val = 0x%x, value = 0x%x\n", val, value); - printk("trigger: pointer = 0x%x\n", snd_dma_pointer(chip->dma8, chip->dma_size)); + printk(KERN_DEBUG "trigger: val = 0x%x, value = 0x%x\n", val, value); + printk(KERN_DEBUG "trigger: pointer = 0x%x\n", + snd_dma_pointer(chip->dma8, chip->dma_size)); #endif snd_es1688_write(chip, 0xb8, (val & 0xf0) | value); spin_unlock(&chip->reg_lock); diff --git a/sound/isa/opl3sa2.c b/sound/isa/opl3sa2.c index 58c972b2af0..06810dfb9d9 100644 --- a/sound/isa/opl3sa2.c +++ b/sound/isa/opl3sa2.c @@ -179,12 +179,13 @@ static unsigned char __snd_opl3sa2_read(struct snd_opl3sa2 *chip, unsigned char unsigned char result; #if 0 outb(0x1d, port); /* password */ - printk("read [0x%lx] = 0x%x\n", port, inb(port)); + printk(KERN_DEBUG "read [0x%lx] = 0x%x\n", port, inb(port)); #endif outb(reg, chip->port); /* register */ result = inb(chip->port + 1); #if 0 - printk("read [0x%lx] = 0x%x [0x%x]\n", port, result, inb(port)); + printk(KERN_DEBUG "read [0x%lx] = 0x%x [0x%x]\n", + port, result, inb(port)); #endif return result; } @@ -233,7 +234,10 @@ static int __devinit snd_opl3sa2_detect(struct snd_card *card) snd_printk(KERN_ERR PFX "can't grab port 0x%lx\n", port); return -EBUSY; } - // snd_printk("REG 0A = 0x%x\n", snd_opl3sa2_read(chip, 0x0a)); + /* + snd_printk(KERN_DEBUG "REG 0A = 0x%x\n", + snd_opl3sa2_read(chip, 0x0a)); + */ chip->version = 0; tmp = snd_opl3sa2_read(chip, OPL3SA2_MISC); if (tmp == 0xff) { diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index 5deb7e69a02..d5bc0e03132 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -252,7 +252,7 @@ static int __devinit snd_opti9xx_init(struct snd_opti9xx *chip, #endif /* OPTi93X */ default: - snd_printk("chip %d not supported\n", hardware); + snd_printk(KERN_ERR "chip %d not supported\n", hardware); return -ENODEV; } return 0; @@ -294,7 +294,7 @@ static unsigned char snd_opti9xx_read(struct snd_opti9xx *chip, #endif /* OPTi93X */ default: - snd_printk("chip %d not supported\n", chip->hardware); + snd_printk(KERN_ERR "chip %d not supported\n", chip->hardware); } spin_unlock_irqrestore(&chip->lock, flags); @@ -336,7 +336,7 @@ static void snd_opti9xx_write(struct snd_opti9xx *chip, unsigned char reg, #endif /* OPTi93X */ default: - snd_printk("chip %d not supported\n", chip->hardware); + snd_printk(KERN_ERR "chip %d not supported\n", chip->hardware); } spin_unlock_irqrestore(&chip->lock, flags); @@ -412,7 +412,7 @@ static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip) #endif /* OPTi93X */ default: - snd_printk("chip %d not supported\n", chip->hardware); + snd_printk(KERN_ERR "chip %d not supported\n", chip->hardware); return -EINVAL; } @@ -430,7 +430,8 @@ static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip) wss_base_bits = 0x02; break; default: - snd_printk("WSS port 0x%lx not valid\n", chip->wss_base); + snd_printk(KERN_WARNING "WSS port 0x%lx not valid\n", + chip->wss_base); goto __skip_base; } snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(1), wss_base_bits << 4, 0x30); @@ -455,7 +456,7 @@ __skip_base: irq_bits = 0x04; break; default: - snd_printk("WSS irq # %d not valid\n", chip->irq); + snd_printk(KERN_WARNING "WSS irq # %d not valid\n", chip->irq); goto __skip_resources; } @@ -470,13 +471,14 @@ __skip_base: dma_bits = 0x03; break; default: - snd_printk("WSS dma1 # %d not valid\n", chip->dma1); + snd_printk(KERN_WARNING "WSS dma1 # %d not valid\n", + chip->dma1); goto __skip_resources; } #if defined(CS4231) || defined(OPTi93X) if (chip->dma1 == chip->dma2) { - snd_printk("don't want to share dmas\n"); + snd_printk(KERN_ERR "don't want to share dmas\n"); return -EBUSY; } @@ -485,7 +487,8 @@ __skip_base: case 1: break; default: - snd_printk("WSS dma2 # %d not valid\n", chip->dma2); + snd_printk(KERN_WARNING "WSS dma2 # %d not valid\n", + chip->dma2); goto __skip_resources; } dma_bits |= 0x04; @@ -516,7 +519,8 @@ __skip_resources: mpu_port_bits = 0x00; break; default: - snd_printk("MPU-401 port 0x%lx not valid\n", + snd_printk(KERN_WARNING + "MPU-401 port 0x%lx not valid\n", chip->mpu_port); goto __skip_mpu; } @@ -535,7 +539,7 @@ __skip_resources: mpu_irq_bits = 0x01; break; default: - snd_printk("MPU-401 irq # %d not valid\n", + snd_printk(KERN_WARNING "MPU-401 irq # %d not valid\n", chip->mpu_irq); goto __skip_mpu; } @@ -726,7 +730,7 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) if (chip->wss_base == SNDRV_AUTO_PORT) { chip->wss_base = snd_legacy_find_free_ioport(possible_ports, 4); if (chip->wss_base < 0) { - snd_printk("unable to find a free WSS port\n"); + snd_printk(KERN_ERR "unable to find a free WSS port\n"); return -EBUSY; } } @@ -891,7 +895,7 @@ static int __devinit snd_opti9xx_isa_probe(struct device *devptr, #if defined(CS4231) || defined(OPTi93X) if (dma2 == SNDRV_AUTO_DMA) { if ((dma2 = snd_legacy_find_free_dma(possible_dma2s[dma1 % 4])) < 0) { - snd_printk("unable to find a free DMA2\n"); + snd_printk(KERN_ERR "unable to find a free DMA2\n"); return -EBUSY; } } diff --git a/sound/isa/wavefront/wavefront.c b/sound/isa/wavefront/wavefront.c index 4c095bc7c72..c280e6220ae 100644 --- a/sound/isa/wavefront/wavefront.c +++ b/sound/isa/wavefront/wavefront.c @@ -551,11 +551,11 @@ static int __devinit snd_wavefront_isa_match(struct device *pdev, return 0; #endif if (cs4232_pcm_port[dev] == SNDRV_AUTO_PORT) { - snd_printk("specify CS4232 port\n"); + snd_printk(KERN_ERR "specify CS4232 port\n"); return 0; } if (ics2115_port[dev] == SNDRV_AUTO_PORT) { - snd_printk("specify ICS2115 port\n"); + snd_printk(KERN_ERR "specify ICS2115 port\n"); return 0; } return 1; diff --git a/sound/isa/wavefront/wavefront_synth.c b/sound/isa/wavefront/wavefront_synth.c index 4c410820a99..beb312cca75 100644 --- a/sound/isa/wavefront/wavefront_synth.c +++ b/sound/isa/wavefront/wavefront_synth.c @@ -633,7 +633,7 @@ wavefront_get_sample_status (snd_wavefront_t *dev, int assume_rom) wbuf[1] = i >> 7; if (snd_wavefront_cmd (dev, WFC_IDENTIFY_SAMPLE_TYPE, rbuf, wbuf)) { - snd_printk("cannot identify sample " + snd_printk(KERN_WARNING "cannot identify sample " "type of slot %d\n", i); dev->sample_status[i] = WF_ST_EMPTY; continue; -- cgit v1.2.3-70-g09d2 From 54530bded6ecf22d683423b66fc3cd6dddb249aa Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 5 Feb 2009 15:55:18 +0100 Subject: ALSA: usb - Add missing KERN_* prefix to printk Signed-off-by: Takashi Iwai --- sound/usb/usbaudio.c | 6 ++++-- sound/usb/usbmixer.c | 5 ++++- sound/usb/usx2y/usb_stream.c | 2 +- 3 files changed, 9 insertions(+), 4 deletions(-) diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 4636926d12d..c69cc6e4f54 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -1419,9 +1419,11 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) subs->cur_audiofmt = fmt; #if 0 - printk("setting done: format = %d, rate = %d..%d, channels = %d\n", + printk(KERN_DEBUG + "setting done: format = %d, rate = %d..%d, channels = %d\n", fmt->format, fmt->rate_min, fmt->rate_max, fmt->channels); - printk(" datapipe = 0x%0x, syncpipe = 0x%0x\n", + printk(KERN_DEBUG + " datapipe = 0x%0x, syncpipe = 0x%0x\n", subs->datapipe, subs->syncpipe); #endif diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index 330f2fbff2d..6615cd3b407 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -222,7 +222,10 @@ static int check_ignored_ctl(struct mixer_build *state, int unitid, int control) for (p = state->map; p->id; p++) { if (p->id == unitid && ! p->name && (! control || ! p->control || control == p->control)) { - // printk("ignored control %d:%d\n", unitid, control); + /* + printk(KERN_DEBUG "ignored control %d:%d\n", + unitid, control); + */ return 1; } } diff --git a/sound/usb/usx2y/usb_stream.c b/sound/usb/usx2y/usb_stream.c index 70b96355ca4..24393dafcb6 100644 --- a/sound/usb/usx2y/usb_stream.c +++ b/sound/usb/usx2y/usb_stream.c @@ -557,7 +557,7 @@ static void stream_start(struct usb_stream_kernel *sk, s->idle_insize -= max_diff - max_diff_0; s->idle_insize += urb_size - s->period_size; if (s->idle_insize < 0) { - snd_printk("%i %i %i\n", + snd_printk(KERN_WARNING "%i %i %i\n", s->idle_insize, urb_size, s->period_size); return; } else if (s->idle_insize == 0) { -- cgit v1.2.3-70-g09d2 From 939778aedd9386e13051a9e1d57c14cba2b6ae13 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 5 Feb 2009 15:57:55 +0100 Subject: ALSA: hda - Add missing KERN_* prefix to printk ... and disable the annoying debug message. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 5218118f01b..d2812ab729c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8265,7 +8265,7 @@ static void alc888_6st_dell_unsol_event(struct hda_codec *codec, { switch (res >> 26) { case ALC880_HP_EVENT: - printk("hp_event\n"); + /* printk(KERN_DEBUG "hp_event\n"); */ alc888_6st_dell_front_automute(codec); break; } @@ -16564,7 +16564,7 @@ static int alc662_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin, if (alc880_is_fixed_pin(pin)) { nid = alc880_idx_to_dac(alc880_fixed_pin_idx(pin)); - /* printk("DAC nid=%x\n",nid); */ + /* printk(KERN_DEBUG "DAC nid=%x\n",nid); */ /* specify the DAC as the extra output */ if (!spec->multiout.hp_nid) spec->multiout.hp_nid = nid; -- cgit v1.2.3-70-g09d2 From 006de267351aa3d836f3307370eae7ec16eac09d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 5 Feb 2009 15:51:04 +0100 Subject: ALSA: Add missing KERN_* prefix to printk in sound/core Signed-off-by: Takashi Iwai --- sound/core/oss/pcm_oss.c | 49 +++++++++++++++++++++++-------------- sound/core/oss/pcm_plugin.h | 4 +-- sound/core/pcm_native.c | 6 ++--- sound/core/seq/oss/seq_oss_device.h | 2 +- sound/core/seq/seq_prioq.c | 3 ++- 5 files changed, 39 insertions(+), 25 deletions(-) diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index e17836680f4..4b883595a85 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -1160,9 +1160,11 @@ snd_pcm_sframes_t snd_pcm_oss_write3(struct snd_pcm_substream *substream, const runtime->status->state == SNDRV_PCM_STATE_SUSPENDED) { #ifdef OSS_DEBUG if (runtime->status->state == SNDRV_PCM_STATE_XRUN) - printk("pcm_oss: write: recovering from XRUN\n"); + printk(KERN_DEBUG "pcm_oss: write: " + "recovering from XRUN\n"); else - printk("pcm_oss: write: recovering from SUSPEND\n"); + printk(KERN_DEBUG "pcm_oss: write: " + "recovering from SUSPEND\n"); #endif ret = snd_pcm_oss_prepare(substream); if (ret < 0) @@ -1196,9 +1198,11 @@ snd_pcm_sframes_t snd_pcm_oss_read3(struct snd_pcm_substream *substream, char *p runtime->status->state == SNDRV_PCM_STATE_SUSPENDED) { #ifdef OSS_DEBUG if (runtime->status->state == SNDRV_PCM_STATE_XRUN) - printk("pcm_oss: read: recovering from XRUN\n"); + printk(KERN_DEBUG "pcm_oss: read: " + "recovering from XRUN\n"); else - printk("pcm_oss: read: recovering from SUSPEND\n"); + printk(KERN_DEBUG "pcm_oss: read: " + "recovering from SUSPEND\n"); #endif ret = snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DRAIN, NULL); if (ret < 0) @@ -1242,9 +1246,11 @@ snd_pcm_sframes_t snd_pcm_oss_writev3(struct snd_pcm_substream *substream, void runtime->status->state == SNDRV_PCM_STATE_SUSPENDED) { #ifdef OSS_DEBUG if (runtime->status->state == SNDRV_PCM_STATE_XRUN) - printk("pcm_oss: writev: recovering from XRUN\n"); + printk(KERN_DEBUG "pcm_oss: writev: " + "recovering from XRUN\n"); else - printk("pcm_oss: writev: recovering from SUSPEND\n"); + printk(KERN_DEBUG "pcm_oss: writev: " + "recovering from SUSPEND\n"); #endif ret = snd_pcm_oss_prepare(substream); if (ret < 0) @@ -1278,9 +1284,11 @@ snd_pcm_sframes_t snd_pcm_oss_readv3(struct snd_pcm_substream *substream, void * runtime->status->state == SNDRV_PCM_STATE_SUSPENDED) { #ifdef OSS_DEBUG if (runtime->status->state == SNDRV_PCM_STATE_XRUN) - printk("pcm_oss: readv: recovering from XRUN\n"); + printk(KERN_DEBUG "pcm_oss: readv: " + "recovering from XRUN\n"); else - printk("pcm_oss: readv: recovering from SUSPEND\n"); + printk(KERN_DEBUG "pcm_oss: readv: " + "recovering from SUSPEND\n"); #endif ret = snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DRAIN, NULL); if (ret < 0) @@ -1533,7 +1541,7 @@ static int snd_pcm_oss_sync1(struct snd_pcm_substream *substream, size_t size) init_waitqueue_entry(&wait, current); add_wait_queue(&runtime->sleep, &wait); #ifdef OSS_DEBUG - printk("sync1: size = %li\n", size); + printk(KERN_DEBUG "sync1: size = %li\n", size); #endif while (1) { result = snd_pcm_oss_write2(substream, runtime->oss.buffer, size, 1); @@ -1590,7 +1598,7 @@ static int snd_pcm_oss_sync(struct snd_pcm_oss_file *pcm_oss_file) mutex_lock(&runtime->oss.params_lock); if (runtime->oss.buffer_used > 0) { #ifdef OSS_DEBUG - printk("sync: buffer_used\n"); + printk(KERN_DEBUG "sync: buffer_used\n"); #endif size = (8 * (runtime->oss.period_bytes - runtime->oss.buffer_used) + 7) / width; snd_pcm_format_set_silence(format, @@ -1603,7 +1611,7 @@ static int snd_pcm_oss_sync(struct snd_pcm_oss_file *pcm_oss_file) } } else if (runtime->oss.period_ptr > 0) { #ifdef OSS_DEBUG - printk("sync: period_ptr\n"); + printk(KERN_DEBUG "sync: period_ptr\n"); #endif size = runtime->oss.period_bytes - runtime->oss.period_ptr; snd_pcm_format_set_silence(format, @@ -1952,7 +1960,7 @@ static int snd_pcm_oss_set_trigger(struct snd_pcm_oss_file *pcm_oss_file, int tr int err, cmd; #ifdef OSS_DEBUG - printk("pcm_oss: trigger = 0x%x\n", trigger); + printk(KERN_DEBUG "pcm_oss: trigger = 0x%x\n", trigger); #endif psubstream = pcm_oss_file->streams[SNDRV_PCM_STREAM_PLAYBACK]; @@ -2170,7 +2178,9 @@ static int snd_pcm_oss_get_space(struct snd_pcm_oss_file *pcm_oss_file, int stre } #ifdef OSS_DEBUG - printk("pcm_oss: space: bytes = %i, fragments = %i, fragstotal = %i, fragsize = %i\n", info.bytes, info.fragments, info.fragstotal, info.fragsize); + printk(KERN_DEBUG "pcm_oss: space: bytes = %i, fragments = %i, " + "fragstotal = %i, fragsize = %i\n", + info.bytes, info.fragments, info.fragstotal, info.fragsize); #endif if (copy_to_user(_info, &info, sizeof(info))) return -EFAULT; @@ -2473,7 +2483,7 @@ static long snd_pcm_oss_ioctl(struct file *file, unsigned int cmd, unsigned long if (((cmd >> 8) & 0xff) != 'P') return -EINVAL; #ifdef OSS_DEBUG - printk("pcm_oss: ioctl = 0x%x\n", cmd); + printk(KERN_DEBUG "pcm_oss: ioctl = 0x%x\n", cmd); #endif switch (cmd) { case SNDCTL_DSP_RESET: @@ -2627,7 +2637,8 @@ static ssize_t snd_pcm_oss_read(struct file *file, char __user *buf, size_t coun #else { ssize_t res = snd_pcm_oss_read1(substream, buf, count); - printk("pcm_oss: read %li bytes (returned %li bytes)\n", (long)count, (long)res); + printk(KERN_DEBUG "pcm_oss: read %li bytes " + "(returned %li bytes)\n", (long)count, (long)res); return res; } #endif @@ -2646,7 +2657,8 @@ static ssize_t snd_pcm_oss_write(struct file *file, const char __user *buf, size substream->f_flags = file->f_flags & O_NONBLOCK; result = snd_pcm_oss_write1(substream, buf, count); #ifdef OSS_DEBUG - printk("pcm_oss: write %li bytes (wrote %li bytes)\n", (long)count, (long)result); + printk(KERN_DEBUG "pcm_oss: write %li bytes (wrote %li bytes)\n", + (long)count, (long)result); #endif return result; } @@ -2720,7 +2732,7 @@ static int snd_pcm_oss_mmap(struct file *file, struct vm_area_struct *area) int err; #ifdef OSS_DEBUG - printk("pcm_oss: mmap begin\n"); + printk(KERN_DEBUG "pcm_oss: mmap begin\n"); #endif pcm_oss_file = file->private_data; switch ((area->vm_flags & (VM_READ | VM_WRITE))) { @@ -2770,7 +2782,8 @@ static int snd_pcm_oss_mmap(struct file *file, struct vm_area_struct *area) runtime->silence_threshold = 0; runtime->silence_size = 0; #ifdef OSS_DEBUG - printk("pcm_oss: mmap ok, bytes = 0x%x\n", runtime->oss.mmap_bytes); + printk(KERN_DEBUG "pcm_oss: mmap ok, bytes = 0x%x\n", + runtime->oss.mmap_bytes); #endif /* In mmap mode we never stop */ runtime->stop_threshold = runtime->boundary; diff --git a/sound/core/oss/pcm_plugin.h b/sound/core/oss/pcm_plugin.h index ca2f4c39be4..b9afab60371 100644 --- a/sound/core/oss/pcm_plugin.h +++ b/sound/core/oss/pcm_plugin.h @@ -176,9 +176,9 @@ static inline int snd_pcm_plug_slave_format(int format, struct snd_mask *format_ #endif #ifdef PLUGIN_DEBUG -#define pdprintf( fmt, args... ) printk( "plugin: " fmt, ##args) +#define pdprintf(fmt, args...) printk(KERN_DEBUG "plugin: " fmt, ##args) #else -#define pdprintf( fmt, args... ) +#define pdprintf(fmt, args...) #endif #endif /* __PCM_PLUGIN_H */ diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index a789efc9df3..d9b8f537942 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -186,7 +186,7 @@ int snd_pcm_hw_refine(struct snd_pcm_substream *substream, if (!(params->rmask & (1 << k))) continue; #ifdef RULES_DEBUG - printk("%s = ", snd_pcm_hw_param_names[k]); + printk(KERN_DEBUG "%s = ", snd_pcm_hw_param_names[k]); printk("%04x%04x%04x%04x -> ", m->bits[3], m->bits[2], m->bits[1], m->bits[0]); #endif changed = snd_mask_refine(m, constrs_mask(constrs, k)); @@ -206,7 +206,7 @@ int snd_pcm_hw_refine(struct snd_pcm_substream *substream, if (!(params->rmask & (1 << k))) continue; #ifdef RULES_DEBUG - printk("%s = ", snd_pcm_hw_param_names[k]); + printk(KERN_DEBUG "%s = ", snd_pcm_hw_param_names[k]); if (i->empty) printk("empty"); else @@ -251,7 +251,7 @@ int snd_pcm_hw_refine(struct snd_pcm_substream *substream, if (!doit) continue; #ifdef RULES_DEBUG - printk("Rule %d [%p]: ", k, r->func); + printk(KERN_DEBUG "Rule %d [%p]: ", k, r->func); if (r->var >= 0) { printk("%s = ", snd_pcm_hw_param_names[r->var]); if (hw_is_mask(r->var)) { diff --git a/sound/core/seq/oss/seq_oss_device.h b/sound/core/seq/oss/seq_oss_device.h index bf8d2b4cb15..c0154a959d5 100644 --- a/sound/core/seq/oss/seq_oss_device.h +++ b/sound/core/seq/oss/seq_oss_device.h @@ -181,7 +181,7 @@ char *enabled_str(int bool); /* for debug */ #ifdef SNDRV_SEQ_OSS_DEBUG extern int seq_oss_debug; -#define debug_printk(x) do { if (seq_oss_debug > 0) snd_printk x; } while (0) +#define debug_printk(x) do { if (seq_oss_debug > 0) snd_printd x; } while (0) #else #define debug_printk(x) /**/ #endif diff --git a/sound/core/seq/seq_prioq.c b/sound/core/seq/seq_prioq.c index 0101a8b99b7..29896ab2340 100644 --- a/sound/core/seq/seq_prioq.c +++ b/sound/core/seq/seq_prioq.c @@ -321,7 +321,8 @@ void snd_seq_prioq_leave(struct snd_seq_prioq * f, int client, int timestamp) freeprev = cell; } else { #if 0 - printk("type = %i, source = %i, dest = %i, client = %i\n", + printk(KERN_DEBUG "type = %i, source = %i, dest = %i, " + "client = %i\n", cell->event.type, cell->event.source.client, cell->event.dest.client, -- cgit v1.2.3-70-g09d2 From 45203832df2fa9e94ca0a249ddb20d2b077e58cc Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 5 Feb 2009 15:51:50 +0100 Subject: ALSA: Add missing KERN_* prefix to printk in sound/drivers Signed-off-by: Takashi Iwai --- sound/drivers/mtpav.c | 12 +++++++----- sound/drivers/mts64.c | 2 +- sound/drivers/opl3/opl3_lib.c | 2 +- sound/drivers/opl3/opl3_midi.c | 30 +++++++++++++++--------------- sound/drivers/opl3/opl3_oss.c | 8 +++++--- sound/drivers/opl3/opl3_synth.c | 2 +- sound/drivers/pcsp/pcsp.c | 2 +- sound/drivers/serial-u16550.c | 18 ++++++++++++------ sound/drivers/virmidi.c | 4 +++- sound/drivers/vx/vx_core.c | 3 ++- 10 files changed, 48 insertions(+), 35 deletions(-) diff --git a/sound/drivers/mtpav.c b/sound/drivers/mtpav.c index 5b89c0883d6..6b26305ff0e 100644 --- a/sound/drivers/mtpav.c +++ b/sound/drivers/mtpav.c @@ -303,8 +303,10 @@ static void snd_mtpav_output_port_write(struct mtpav *mtp_card, snd_mtpav_send_byte(mtp_card, 0xf5); snd_mtpav_send_byte(mtp_card, portp->hwport); - //snd_printk("new outport: 0x%x\n", (unsigned int) portp->hwport); - + /* + snd_printk(KERN_DEBUG "new outport: 0x%x\n", + (unsigned int) portp->hwport); + */ if (!(outbyte & 0x80) && portp->running_status) snd_mtpav_send_byte(mtp_card, portp->running_status); } @@ -540,7 +542,7 @@ static void snd_mtpav_read_bytes(struct mtpav *mcrd) u8 sbyt = snd_mtpav_getreg(mcrd, SREG); - //printk("snd_mtpav_read_bytes() sbyt: 0x%x\n", sbyt); + /* printk(KERN_DEBUG "snd_mtpav_read_bytes() sbyt: 0x%x\n", sbyt); */ if (!(sbyt & SIGS_BYTE)) return; @@ -585,12 +587,12 @@ static irqreturn_t snd_mtpav_irqh(int irq, void *dev_id) static int __devinit snd_mtpav_get_ISA(struct mtpav * mcard) { if ((mcard->res_port = request_region(port, 3, "MotuMTPAV MIDI")) == NULL) { - snd_printk("MTVAP port 0x%lx is busy\n", port); + snd_printk(KERN_ERR "MTVAP port 0x%lx is busy\n", port); return -EBUSY; } mcard->port = port; if (request_irq(irq, snd_mtpav_irqh, IRQF_DISABLED, "MOTU MTPAV", mcard)) { - snd_printk("MTVAP IRQ %d busy\n", irq); + snd_printk(KERN_ERR "MTVAP IRQ %d busy\n", irq); return -EBUSY; } mcard->irq = irq; diff --git a/sound/drivers/mts64.c b/sound/drivers/mts64.c index 87ba1ddc011..1a05b2d64c9 100644 --- a/sound/drivers/mts64.c +++ b/sound/drivers/mts64.c @@ -1015,7 +1015,7 @@ static int __devinit snd_mts64_probe(struct platform_device *pdev) goto __err; } - snd_printk("ESI Miditerminal 4140 on 0x%lx\n", p->base); + snd_printk(KERN_INFO "ESI Miditerminal 4140 on 0x%lx\n", p->base); return 0; __err: diff --git a/sound/drivers/opl3/opl3_lib.c b/sound/drivers/opl3/opl3_lib.c index 780582340fe..6e31e46ca39 100644 --- a/sound/drivers/opl3/opl3_lib.c +++ b/sound/drivers/opl3/opl3_lib.c @@ -302,7 +302,7 @@ void snd_opl3_interrupt(struct snd_hwdep * hw) opl3 = hw->private_data; status = inb(opl3->l_port); #if 0 - snd_printk("AdLib IRQ status = 0x%x\n", status); + snd_printk(KERN_DEBUG "AdLib IRQ status = 0x%x\n", status); #endif if (!(status & 0x80)) return; diff --git a/sound/drivers/opl3/opl3_midi.c b/sound/drivers/opl3/opl3_midi.c index 16feafa2c51..6e7d09ae0e8 100644 --- a/sound/drivers/opl3/opl3_midi.c +++ b/sound/drivers/opl3/opl3_midi.c @@ -125,7 +125,7 @@ static void debug_alloc(struct snd_opl3 *opl3, char *s, int voice) { int i; char *str = "x.24"; - printk("time %.5i: %s [%.2i]: ", opl3->use_time, s, voice); + printk(KERN_DEBUG "time %.5i: %s [%.2i]: ", opl3->use_time, s, voice); for (i = 0; i < opl3->max_voices; i++) printk("%c", *(str + opl3->voices[i].state + 1)); printk("\n"); @@ -218,7 +218,7 @@ static int opl3_get_voice(struct snd_opl3 *opl3, int instr_4op, for (i = 0; i < END; i++) { if (best[i].voice >= 0) { #ifdef DEBUG_ALLOC - printk("%s %iop allocation on voice %i\n", + printk(KERN_DEBUG "%s %iop allocation on voice %i\n", alloc_type[i], instr_4op ? 4 : 2, best[i].voice); #endif @@ -317,7 +317,7 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan) opl3 = p; #ifdef DEBUG_MIDI - snd_printk("Note on, ch %i, inst %i, note %i, vel %i\n", + snd_printk(KERN_DEBUG "Note on, ch %i, inst %i, note %i, vel %i\n", chan->number, chan->midi_program, note, vel); #endif @@ -372,7 +372,7 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan) return; } #ifdef DEBUG_MIDI - snd_printk(" --> OPL%i instrument: %s\n", + snd_printk(KERN_DEBUG " --> OPL%i instrument: %s\n", instr_4op ? 3 : 2, patch->name); #endif /* in SYNTH mode, application takes care of voices */ @@ -431,7 +431,7 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan) } #ifdef DEBUG_MIDI - snd_printk(" --> setting OPL3 connection: 0x%x\n", + snd_printk(KERN_DEBUG " --> setting OPL3 connection: 0x%x\n", opl3->connection_reg); #endif /* @@ -466,7 +466,7 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan) /* Program the FM voice characteristics */ for (i = 0; i < (instr_4op ? 4 : 2); i++) { #ifdef DEBUG_MIDI - snd_printk(" --> programming operator %i\n", i); + snd_printk(KERN_DEBUG " --> programming operator %i\n", i); #endif op_offset = snd_opl3_regmap[voice_offset][i]; @@ -546,7 +546,7 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan) blocknum |= OPL3_KEYON_BIT; #ifdef DEBUG_MIDI - snd_printk(" --> trigger voice %i\n", voice); + snd_printk(KERN_DEBUG " --> trigger voice %i\n", voice); #endif /* Set OPL3 KEYON_BLOCK register of requested voice */ opl3_reg = reg_side | (OPL3_REG_KEYON_BLOCK + voice_offset); @@ -602,7 +602,7 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan) prg = extra_prg - 1; } #ifdef DEBUG_MIDI - snd_printk(" *** allocating extra program\n"); + snd_printk(KERN_DEBUG " *** allocating extra program\n"); #endif goto __extra_prg; } @@ -633,7 +633,7 @@ static void snd_opl3_kill_voice(struct snd_opl3 *opl3, int voice) /* kill voice */ #ifdef DEBUG_MIDI - snd_printk(" --> kill voice %i\n", voice); + snd_printk(KERN_DEBUG " --> kill voice %i\n", voice); #endif opl3_reg = reg_side | (OPL3_REG_KEYON_BLOCK + voice_offset); /* clear Key ON bit */ @@ -670,7 +670,7 @@ void snd_opl3_note_off(void *p, int note, int vel, struct snd_midi_channel *chan opl3 = p; #ifdef DEBUG_MIDI - snd_printk("Note off, ch %i, inst %i, note %i\n", + snd_printk(KERN_DEBUG "Note off, ch %i, inst %i, note %i\n", chan->number, chan->midi_program, note); #endif @@ -709,7 +709,7 @@ void snd_opl3_key_press(void *p, int note, int vel, struct snd_midi_channel *cha opl3 = p; #ifdef DEBUG_MIDI - snd_printk("Key pressure, ch#: %i, inst#: %i\n", + snd_printk(KERN_DEBUG "Key pressure, ch#: %i, inst#: %i\n", chan->number, chan->midi_program); #endif } @@ -723,7 +723,7 @@ void snd_opl3_terminate_note(void *p, int note, struct snd_midi_channel *chan) opl3 = p; #ifdef DEBUG_MIDI - snd_printk("Terminate note, ch#: %i, inst#: %i\n", + snd_printk(KERN_DEBUG "Terminate note, ch#: %i, inst#: %i\n", chan->number, chan->midi_program); #endif } @@ -812,7 +812,7 @@ void snd_opl3_control(void *p, int type, struct snd_midi_channel *chan) opl3 = p; #ifdef DEBUG_MIDI - snd_printk("Controller, TYPE = %i, ch#: %i, inst#: %i\n", + snd_printk(KERN_DEBUG "Controller, TYPE = %i, ch#: %i, inst#: %i\n", type, chan->number, chan->midi_program); #endif @@ -849,7 +849,7 @@ void snd_opl3_nrpn(void *p, struct snd_midi_channel *chan, opl3 = p; #ifdef DEBUG_MIDI - snd_printk("NRPN, ch#: %i, inst#: %i\n", + snd_printk(KERN_DEBUG "NRPN, ch#: %i, inst#: %i\n", chan->number, chan->midi_program); #endif } @@ -864,6 +864,6 @@ void snd_opl3_sysex(void *p, unsigned char *buf, int len, opl3 = p; #ifdef DEBUG_MIDI - snd_printk("SYSEX\n"); + snd_printk(KERN_DEBUG "SYSEX\n"); #endif } diff --git a/sound/drivers/opl3/opl3_oss.c b/sound/drivers/opl3/opl3_oss.c index 9a2271dc046..a54b1dc5cc7 100644 --- a/sound/drivers/opl3/opl3_oss.c +++ b/sound/drivers/opl3/opl3_oss.c @@ -220,14 +220,14 @@ static int snd_opl3_load_patch_seq_oss(struct snd_seq_oss_arg *arg, int format, return -EINVAL; if (count < (int)sizeof(sbi)) { - snd_printk("FM Error: Patch record too short\n"); + snd_printk(KERN_ERR "FM Error: Patch record too short\n"); return -EINVAL; } if (copy_from_user(&sbi, buf, sizeof(sbi))) return -EFAULT; if (sbi.channel < 0 || sbi.channel >= SBFM_MAXINSTR) { - snd_printk("FM Error: Invalid instrument number %d\n", + snd_printk(KERN_ERR "FM Error: Invalid instrument number %d\n", sbi.channel); return -EINVAL; } @@ -254,7 +254,9 @@ static int snd_opl3_ioctl_seq_oss(struct snd_seq_oss_arg *arg, unsigned int cmd, opl3 = arg->private_data; switch (cmd) { case SNDCTL_FM_LOAD_INSTR: - snd_printk("OPL3: Obsolete ioctl(SNDCTL_FM_LOAD_INSTR) used. Fix the program.\n"); + snd_printk(KERN_ERR "OPL3: " + "Obsolete ioctl(SNDCTL_FM_LOAD_INSTR) used. " + "Fix the program.\n"); return -EINVAL; case SNDCTL_SYNTH_MEMAVL: diff --git a/sound/drivers/opl3/opl3_synth.c b/sound/drivers/opl3/opl3_synth.c index 962bb9c8b9c..6d57b6441de 100644 --- a/sound/drivers/opl3/opl3_synth.c +++ b/sound/drivers/opl3/opl3_synth.c @@ -168,7 +168,7 @@ int snd_opl3_ioctl(struct snd_hwdep * hw, struct file *file, #ifdef CONFIG_SND_DEBUG default: - snd_printk("unknown IOCTL: 0x%x\n", cmd); + snd_printk(KERN_WARNING "unknown IOCTL: 0x%x\n", cmd); #endif } return -ENOTTY; diff --git a/sound/drivers/pcsp/pcsp.c b/sound/drivers/pcsp/pcsp.c index a4049eb94d3..c7c744c6fc0 100644 --- a/sound/drivers/pcsp/pcsp.c +++ b/sound/drivers/pcsp/pcsp.c @@ -57,7 +57,7 @@ static int __devinit snd_pcsp_create(struct snd_card *card) else min_div = MAX_DIV; #if PCSP_DEBUG - printk("PCSP: lpj=%li, min_div=%i, res=%li\n", + printk(KERN_DEBUG "PCSP: lpj=%li, min_div=%i, res=%li\n", loops_per_jiffy, min_div, tp.tv_nsec); #endif diff --git a/sound/drivers/serial-u16550.c b/sound/drivers/serial-u16550.c index d8aab9da97c..ff0a4151094 100644 --- a/sound/drivers/serial-u16550.c +++ b/sound/drivers/serial-u16550.c @@ -241,7 +241,8 @@ static void snd_uart16550_io_loop(struct snd_uart16550 * uart) snd_rawmidi_receive(uart->midi_input[substream], &c, 1); if (status & UART_LSR_OE) - snd_printk("%s: Overrun on device at 0x%lx\n", + snd_printk(KERN_WARNING + "%s: Overrun on device at 0x%lx\n", uart->rmidi->name, uart->base); } @@ -636,7 +637,8 @@ static int snd_uart16550_output_byte(struct snd_uart16550 *uart, } } else { if (!snd_uart16550_write_buffer(uart, midi_byte)) { - snd_printk("%s: Buffer overrun on device at 0x%lx\n", + snd_printk(KERN_WARNING + "%s: Buffer overrun on device at 0x%lx\n", uart->rmidi->name, uart->base); return 0; } @@ -815,7 +817,8 @@ static int __devinit snd_uart16550_create(struct snd_card *card, if (irq >= 0 && irq != SNDRV_AUTO_IRQ) { if (request_irq(irq, snd_uart16550_interrupt, IRQF_DISABLED, "Serial MIDI", uart)) { - snd_printk("irq %d busy. Using Polling.\n", irq); + snd_printk(KERN_WARNING + "irq %d busy. Using Polling.\n", irq); } else { uart->irq = irq; } @@ -919,19 +922,22 @@ static int __devinit snd_serial_probe(struct platform_device *devptr) case SNDRV_SERIAL_GENERIC: break; default: - snd_printk("Adaptor type is out of range 0-%d (%d)\n", + snd_printk(KERN_ERR + "Adaptor type is out of range 0-%d (%d)\n", SNDRV_SERIAL_MAX_ADAPTOR, adaptor[dev]); return -ENODEV; } if (outs[dev] < 1 || outs[dev] > SNDRV_SERIAL_MAX_OUTS) { - snd_printk("Count of outputs is out of range 1-%d (%d)\n", + snd_printk(KERN_ERR + "Count of outputs is out of range 1-%d (%d)\n", SNDRV_SERIAL_MAX_OUTS, outs[dev]); return -ENODEV; } if (ins[dev] < 1 || ins[dev] > SNDRV_SERIAL_MAX_INS) { - snd_printk("Count of inputs is out of range 1-%d (%d)\n", + snd_printk(KERN_ERR + "Count of inputs is out of range 1-%d (%d)\n", SNDRV_SERIAL_MAX_INS, ins[dev]); return -ENODEV; } diff --git a/sound/drivers/virmidi.c b/sound/drivers/virmidi.c index f79e3614079..1022e365606 100644 --- a/sound/drivers/virmidi.c +++ b/sound/drivers/virmidi.c @@ -98,7 +98,9 @@ static int __devinit snd_virmidi_probe(struct platform_device *devptr) vmidi->card = card; if (midi_devs[dev] > MAX_MIDI_DEVICES) { - snd_printk("too much midi devices for virmidi %d: force to use %d\n", dev, MAX_MIDI_DEVICES); + snd_printk(KERN_WARNING + "too much midi devices for virmidi %d: " + "force to use %d\n", dev, MAX_MIDI_DEVICES); midi_devs[dev] = MAX_MIDI_DEVICES; } for (idx = 0; idx < midi_devs[dev]; idx++) { diff --git a/sound/drivers/vx/vx_core.c b/sound/drivers/vx/vx_core.c index 14e3354be43..19c6e376c7c 100644 --- a/sound/drivers/vx/vx_core.c +++ b/sound/drivers/vx/vx_core.c @@ -688,7 +688,8 @@ int snd_vx_dsp_load(struct vx_core *chip, const struct firmware *dsp) image = dsp->data + i; /* Wait DSP ready for a new read */ if ((err = vx_wait_isr_bit(chip, ISR_TX_EMPTY)) < 0) { - printk("dsp loading error at position %d\n", i); + printk(KERN_ERR + "dsp loading error at position %d\n", i); return err; } cptr = image; -- cgit v1.2.3-70-g09d2 From 42b0158bdb1344b05cc1e98c363fba9e97137565 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 5 Feb 2009 16:01:46 +0100 Subject: ALSA: emux - Add missing KERN_* prefix to printk Signed-off-by: Takashi Iwai --- sound/synth/emux/emux_oss.c | 2 +- sound/synth/emux/emux_seq.c | 16 ++++++++-------- sound/synth/emux/emux_synth.c | 6 ++++-- sound/synth/emux/soundfont.c | 28 +++++++++++++++++----------- 4 files changed, 30 insertions(+), 22 deletions(-) diff --git a/sound/synth/emux/emux_oss.c b/sound/synth/emux/emux_oss.c index 5c47b6c0926..87e42206c4e 100644 --- a/sound/synth/emux/emux_oss.c +++ b/sound/synth/emux/emux_oss.c @@ -132,7 +132,7 @@ snd_emux_open_seq_oss(struct snd_seq_oss_arg *arg, void *closure) p = snd_emux_create_port(emu, tmpname, 32, 1, &callback); if (p == NULL) { - snd_printk("can't create port\n"); + snd_printk(KERN_ERR "can't create port\n"); snd_emux_dec_count(emu); mutex_unlock(&emu->register_mutex); return -ENOMEM; diff --git a/sound/synth/emux/emux_seq.c b/sound/synth/emux/emux_seq.c index 335aa2ce257..ca5f7effb4d 100644 --- a/sound/synth/emux/emux_seq.c +++ b/sound/synth/emux/emux_seq.c @@ -74,15 +74,15 @@ snd_emux_init_seq(struct snd_emux *emu, struct snd_card *card, int index) emu->client = snd_seq_create_kernel_client(card, index, "%s WaveTable", emu->name); if (emu->client < 0) { - snd_printk("can't create client\n"); + snd_printk(KERN_ERR "can't create client\n"); return -ENODEV; } if (emu->num_ports < 0) { - snd_printk("seqports must be greater than zero\n"); + snd_printk(KERN_WARNING "seqports must be greater than zero\n"); emu->num_ports = 1; } else if (emu->num_ports >= SNDRV_EMUX_MAX_PORTS) { - snd_printk("too many ports." + snd_printk(KERN_WARNING "too many ports." "limited max. ports %d\n", SNDRV_EMUX_MAX_PORTS); emu->num_ports = SNDRV_EMUX_MAX_PORTS; } @@ -100,7 +100,7 @@ snd_emux_init_seq(struct snd_emux *emu, struct snd_card *card, int index) p = snd_emux_create_port(emu, tmpname, MIDI_CHANNELS, 0, &pinfo); if (p == NULL) { - snd_printk("can't create port\n"); + snd_printk(KERN_ERR "can't create port\n"); return -ENOMEM; } @@ -147,12 +147,12 @@ snd_emux_create_port(struct snd_emux *emu, char *name, /* Allocate structures for this channel */ if ((p = kzalloc(sizeof(*p), GFP_KERNEL)) == NULL) { - snd_printk("no memory\n"); + snd_printk(KERN_ERR "no memory\n"); return NULL; } p->chset.channels = kcalloc(max_channels, sizeof(struct snd_midi_channel), GFP_KERNEL); if (p->chset.channels == NULL) { - snd_printk("no memory\n"); + snd_printk(KERN_ERR "no memory\n"); kfree(p); return NULL; } @@ -376,12 +376,12 @@ int snd_emux_init_virmidi(struct snd_emux *emu, struct snd_card *card) goto __error; } emu->vmidi[i] = rmidi; - //snd_printk("virmidi %d ok\n", i); + /* snd_printk(KERN_DEBUG "virmidi %d ok\n", i); */ } return 0; __error: - //snd_printk("error init..\n"); + /* snd_printk(KERN_DEBUG "error init..\n"); */ snd_emux_delete_virmidi(emu); return -ENOMEM; } diff --git a/sound/synth/emux/emux_synth.c b/sound/synth/emux/emux_synth.c index 2cc6f6f7906..3e921b386fd 100644 --- a/sound/synth/emux/emux_synth.c +++ b/sound/synth/emux/emux_synth.c @@ -956,7 +956,8 @@ void snd_emux_lock_voice(struct snd_emux *emu, int voice) if (emu->voices[voice].state == SNDRV_EMUX_ST_OFF) emu->voices[voice].state = SNDRV_EMUX_ST_LOCKED; else - snd_printk("invalid voice for lock %d (state = %x)\n", + snd_printk(KERN_WARNING + "invalid voice for lock %d (state = %x)\n", voice, emu->voices[voice].state); spin_unlock_irqrestore(&emu->voice_lock, flags); } @@ -973,7 +974,8 @@ void snd_emux_unlock_voice(struct snd_emux *emu, int voice) if (emu->voices[voice].state == SNDRV_EMUX_ST_LOCKED) emu->voices[voice].state = SNDRV_EMUX_ST_OFF; else - snd_printk("invalid voice for unlock %d (state = %x)\n", + snd_printk(KERN_WARNING + "invalid voice for unlock %d (state = %x)\n", voice, emu->voices[voice].state); spin_unlock_irqrestore(&emu->voice_lock, flags); } diff --git a/sound/synth/emux/soundfont.c b/sound/synth/emux/soundfont.c index 36d53bd317e..63c8f45c0c2 100644 --- a/sound/synth/emux/soundfont.c +++ b/sound/synth/emux/soundfont.c @@ -133,7 +133,7 @@ snd_soundfont_load(struct snd_sf_list *sflist, const void __user *data, int rc; if (count < (long)sizeof(patch)) { - snd_printk("patch record too small %ld\n", count); + snd_printk(KERN_ERR "patch record too small %ld\n", count); return -EINVAL; } if (copy_from_user(&patch, data, sizeof(patch))) @@ -143,15 +143,16 @@ snd_soundfont_load(struct snd_sf_list *sflist, const void __user *data, data += sizeof(patch); if (patch.key != SNDRV_OSS_SOUNDFONT_PATCH) { - snd_printk("'The wrong kind of patch' %x\n", patch.key); + snd_printk(KERN_ERR "The wrong kind of patch %x\n", patch.key); return -EINVAL; } if (count < patch.len) { - snd_printk("Patch too short %ld, need %d\n", count, patch.len); + snd_printk(KERN_ERR "Patch too short %ld, need %d\n", + count, patch.len); return -EINVAL; } if (patch.len < 0) { - snd_printk("poor length %d\n", patch.len); + snd_printk(KERN_ERR "poor length %d\n", patch.len); return -EINVAL; } @@ -195,7 +196,8 @@ snd_soundfont_load(struct snd_sf_list *sflist, const void __user *data, case SNDRV_SFNT_REMOVE_INFO: /* patch must be opened */ if (!sflist->currsf) { - snd_printk("soundfont: remove_info: patch not opened\n"); + snd_printk(KERN_ERR "soundfont: remove_info: " + "patch not opened\n"); rc = -EINVAL; } else { int bank, instr; @@ -531,7 +533,7 @@ load_info(struct snd_sf_list *sflist, const void __user *data, long count) return -EINVAL; if (count < (long)sizeof(hdr)) { - printk("Soundfont error: invalid patch zone length\n"); + printk(KERN_ERR "Soundfont error: invalid patch zone length\n"); return -EINVAL; } if (copy_from_user((char*)&hdr, data, sizeof(hdr))) @@ -541,12 +543,14 @@ load_info(struct snd_sf_list *sflist, const void __user *data, long count) count -= sizeof(hdr); if (hdr.nvoices <= 0 || hdr.nvoices >= 100) { - printk("Soundfont error: Illegal voice number %d\n", hdr.nvoices); + printk(KERN_ERR "Soundfont error: Illegal voice number %d\n", + hdr.nvoices); return -EINVAL; } if (count < (long)sizeof(struct soundfont_voice_info) * hdr.nvoices) { - printk("Soundfont Error: patch length(%ld) is smaller than nvoices(%d)\n", + printk(KERN_ERR "Soundfont Error: " + "patch length(%ld) is smaller than nvoices(%d)\n", count, hdr.nvoices); return -EINVAL; } @@ -952,7 +956,7 @@ load_guspatch(struct snd_sf_list *sflist, const char __user *data, int rc; if (count < (long)sizeof(patch)) { - snd_printk("patch record too small %ld\n", count); + snd_printk(KERN_ERR "patch record too small %ld\n", count); return -EINVAL; } if (copy_from_user(&patch, data, sizeof(patch))) @@ -1034,7 +1038,8 @@ load_guspatch(struct snd_sf_list *sflist, const char __user *data, /* panning position; -128 - 127 => 0-127 */ zone->v.pan = (patch.panning + 128) / 2; #if 0 - snd_printk("gus: basefrq=%d (ofs=%d) root=%d,tune=%d, range:%d-%d\n", + snd_printk(KERN_DEBUG + "gus: basefrq=%d (ofs=%d) root=%d,tune=%d, range:%d-%d\n", (int)patch.base_freq, zone->v.rate_offset, zone->v.root, zone->v.tune, zone->v.low, zone->v.high); #endif @@ -1068,7 +1073,8 @@ load_guspatch(struct snd_sf_list *sflist, const char __user *data, zone->v.parm.volrelease = 0x8000 | snd_sf_calc_parm_decay(release); zone->v.attenuation = calc_gus_attenuation(patch.env_offset[0]); #if 0 - snd_printk("gus: atkhld=%x, dcysus=%x, volrel=%x, att=%d\n", + snd_printk(KERN_DEBUG + "gus: atkhld=%x, dcysus=%x, volrel=%x, att=%d\n", zone->v.parm.volatkhld, zone->v.parm.voldcysus, zone->v.parm.volrelease, -- cgit v1.2.3-70-g09d2 From e2ea7cfc703cba3299d22db728516a0fc1a9717c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 5 Feb 2009 16:07:02 +0100 Subject: ALSA: Add missing KERN_* prefix to printk in sound/pci/ice1712 Signed-off-by: Takashi Iwai --- sound/pci/ice1712/ice1712.c | 2 +- sound/pci/ice1712/ice1724.c | 17 ++++++++++++++--- sound/pci/ice1712/juli.c | 5 +++-- sound/pci/ice1712/prodigy192.c | 13 +++++++++---- 4 files changed, 27 insertions(+), 10 deletions(-) diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 58d7cda03de..dcd3f4f89b4 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -458,7 +458,7 @@ static irqreturn_t snd_ice1712_interrupt(int irq, void *dev_id) u16 pbkstatus; struct snd_pcm_substream *substream; pbkstatus = inw(ICEDS(ice, INTSTAT)); - /* printk("pbkstatus = 0x%x\n", pbkstatus); */ + /* printk(KERN_DEBUG "pbkstatus = 0x%x\n", pbkstatus); */ for (idx = 0; idx < 6; idx++) { if ((pbkstatus & (3 << (idx * 2))) == 0) continue; diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index eb7872dec5a..da8c111e9e3 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -756,7 +756,14 @@ static int snd_vt1724_playback_pro_prepare(struct snd_pcm_substream *substream) spin_unlock_irq(&ice->reg_lock); - /* printk("pro prepare: ch = %d, addr = 0x%x, buffer = 0x%x, period = 0x%x\n", substream->runtime->channels, (unsigned int)substream->runtime->dma_addr, snd_pcm_lib_buffer_bytes(substream), snd_pcm_lib_period_bytes(substream)); */ + /* + printk(KERN_DEBUG "pro prepare: ch = %d, addr = 0x%x, " + "buffer = 0x%x, period = 0x%x\n", + substream->runtime->channels, + (unsigned int)substream->runtime->dma_addr, + snd_pcm_lib_buffer_bytes(substream), + snd_pcm_lib_period_bytes(substream)); + */ return 0; } @@ -2133,7 +2140,9 @@ unsigned char snd_vt1724_read_i2c(struct snd_ice1712 *ice, wait_i2c_busy(ice); val = inb(ICEREG1724(ice, I2C_DATA)); mutex_unlock(&ice->i2c_mutex); - /* printk("i2c_read: [0x%x,0x%x] = 0x%x\n", dev, addr, val); */ + /* + printk(KERN_DEBUG "i2c_read: [0x%x,0x%x] = 0x%x\n", dev, addr, val); + */ return val; } @@ -2142,7 +2151,9 @@ void snd_vt1724_write_i2c(struct snd_ice1712 *ice, { mutex_lock(&ice->i2c_mutex); wait_i2c_busy(ice); - /* printk("i2c_write: [0x%x,0x%x] = 0x%x\n", dev, addr, data); */ + /* + printk(KERN_DEBUG "i2c_write: [0x%x,0x%x] = 0x%x\n", dev, addr, data); + */ outb(addr, ICEREG1724(ice, I2C_BYTE_ADDR)); outb(data, ICEREG1724(ice, I2C_DATA)); outb(dev | VT1724_I2C_WRITE, ICEREG1724(ice, I2C_DEV_ADDR)); diff --git a/sound/pci/ice1712/juli.c b/sound/pci/ice1712/juli.c index c51659b9caf..fd948bfd9ae 100644 --- a/sound/pci/ice1712/juli.c +++ b/sound/pci/ice1712/juli.c @@ -345,8 +345,9 @@ static int juli_mute_put(struct snd_kcontrol *kcontrol, new_gpio = old_gpio & ~((unsigned int) kcontrol->private_value); } - /* printk("JULI - mute/unmute: control_value: 0x%x, old_gpio: 0x%x, \ - new_gpio 0x%x\n", + /* printk(KERN_DEBUG + "JULI - mute/unmute: control_value: 0x%x, old_gpio: 0x%x, " + "new_gpio 0x%x\n", (unsigned int)ucontrol->value.integer.value[0], old_gpio, new_gpio); */ if (old_gpio != new_gpio) { diff --git a/sound/pci/ice1712/prodigy192.c b/sound/pci/ice1712/prodigy192.c index 48d3679292a..2a8e5cd8f2d 100644 --- a/sound/pci/ice1712/prodigy192.c +++ b/sound/pci/ice1712/prodigy192.c @@ -133,8 +133,10 @@ static int stac9460_dac_mute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id) + STAC946X_LF_VOLUME; /* due to possible conflicts with stac9460_set_rate_val, mutexing */ mutex_lock(&spec->mute_mutex); - /*printk("Mute put: reg 0x%02x, ctrl value: 0x%02x\n", idx, - ucontrol->value.integer.value[0]);*/ + /* + printk(KERN_DEBUG "Mute put: reg 0x%02x, ctrl value: 0x%02x\n", idx, + ucontrol->value.integer.value[0]); + */ change = stac9460_dac_mute(ice, idx, ucontrol->value.integer.value[0]); mutex_unlock(&spec->mute_mutex); return change; @@ -185,7 +187,10 @@ static int stac9460_dac_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_el change = (ovol != nvol); if (change) { ovol = (0x7f - nvol) | (tmp & 0x80); - /*printk("DAC Volume: reg 0x%02x: 0x%02x\n", idx, ovol);*/ + /* + printk(KERN_DEBUG "DAC Volume: reg 0x%02x: 0x%02x\n", + idx, ovol); + */ stac9460_put(ice, idx, (0x7f - nvol) | (tmp & 0x80)); } return change; @@ -344,7 +349,7 @@ static void stac9460_set_rate_val(struct snd_ice1712 *ice, unsigned int rate) for (idx = 0; idx < 7 ; ++idx) changed[idx] = stac9460_dac_mute(ice, STAC946X_MASTER_VOLUME + idx, 0); - /*printk("Rate change: %d, new MC: 0x%02x\n", rate, new);*/ + /*printk(KERN_DEBUG "Rate change: %d, new MC: 0x%02x\n", rate, new);*/ stac9460_put(ice, STAC946X_MASTER_CLOCKING, new); udelay(10); /* unmuting - only originally unmuted dacs - -- cgit v1.2.3-70-g09d2 From 28a97c194cec477073ae341f15b836437d8ef8e5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 5 Feb 2009 16:08:14 +0100 Subject: ALSA: emu10k1 - Add missing KERN_* prefix to printk Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emu10k1_callback.c | 7 ++- sound/pci/emu10k1/emu10k1_main.c | 5 +- sound/pci/emu10k1/emufx.c | 11 ++-- sound/pci/emu10k1/emupcm.c | 37 ++++++++++--- sound/pci/emu10k1/io.c | 4 +- sound/pci/emu10k1/p16v.c | 100 +++++++++++++++++++++++++---------- sound/pci/emu10k1/voice.c | 12 +++-- 7 files changed, 130 insertions(+), 46 deletions(-) diff --git a/sound/pci/emu10k1/emu10k1_callback.c b/sound/pci/emu10k1/emu10k1_callback.c index 0e649dcdbf6..7ef949d99a5 100644 --- a/sound/pci/emu10k1/emu10k1_callback.c +++ b/sound/pci/emu10k1/emu10k1_callback.c @@ -103,7 +103,10 @@ snd_emu10k1_synth_get_voice(struct snd_emu10k1 *hw) int ch; vp = &emu->voices[best[i].voice]; if ((ch = vp->ch) < 0) { - //printk("synth_get_voice: ch < 0 (%d) ??", i); + /* + printk(KERN_WARNING + "synth_get_voice: ch < 0 (%d) ??", i); + */ continue; } vp->emu->num_voices--; @@ -335,7 +338,7 @@ start_voice(struct snd_emux_voice *vp) return -EINVAL; emem->map_locked++; if (snd_emu10k1_memblk_map(hw, emem) < 0) { - // printk("emu: cannot map!\n"); + /* printk(KERN_ERR "emu: cannot map!\n"); */ return -ENOMEM; } mapped_offset = snd_emu10k1_memblk_offset(emem) >> 1; diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 7958006a1d6..8343aecbd25 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -758,7 +758,8 @@ static int emu1010_firmware_thread(void *data) snd_printk(KERN_INFO "emu1010: Audio Dock Firmware loaded\n"); snd_emu1010_fpga_read(emu, EMU_DOCK_MAJOR_REV, &tmp); snd_emu1010_fpga_read(emu, EMU_DOCK_MINOR_REV, &tmp2); - snd_printk("Audio Dock ver:%d.%d\n", tmp, tmp2); + snd_printk(KERN_INFO "Audio Dock ver:%d.%d\n", + tmp, tmp2); /* Sync clocking between 1010 and Dock */ /* Allow DLL to settle */ msleep(10); @@ -887,7 +888,7 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 *emu) snd_printk(KERN_INFO "emu1010: Hana Firmware loaded\n"); snd_emu1010_fpga_read(emu, EMU_HANA_MAJOR_REV, &tmp); snd_emu1010_fpga_read(emu, EMU_HANA_MINOR_REV, &tmp2); - snd_printk("emu1010: Hana version: %d.%d\n", tmp, tmp2); + snd_printk(KERN_INFO "emu1010: Hana version: %d.%d\n", tmp, tmp2); /* Enable 48Volt power to Audio Dock */ snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, EMU_HANA_DOCK_PWR_ON); diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index 7dba08f0ab8..191e1cd9997 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -1519,7 +1519,7 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) /* A_PUT_STEREO_OUTPUT(A_EXTOUT_FRONT_L, A_EXTOUT_FRONT_R, playback + SND_EMU10K1_PLAYBACK_CHANNELS); */ if (emu->card_capabilities->emu_model) { /* EMU1010 Outputs from PCM Front, Rear, Center, LFE, Side */ - snd_printk("EMU outputs on\n"); + snd_printk(KERN_INFO "EMU outputs on\n"); for (z = 0; z < 8; z++) { if (emu->card_capabilities->ca0108_chip) { A_OP(icode, &ptr, iACC3, A3_EMU32OUT(z), A_GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + z), A_C_00000000, A_C_00000000); @@ -1567,7 +1567,7 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) if (emu->card_capabilities->emu_model) { if (emu->card_capabilities->ca0108_chip) { - snd_printk("EMU2 inputs on\n"); + snd_printk(KERN_INFO "EMU2 inputs on\n"); for (z = 0; z < 0x10; z++) { snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, @@ -1575,10 +1575,13 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) A_FXBUS2(z*2) ); } } else { - snd_printk("EMU inputs on\n"); + snd_printk(KERN_INFO "EMU inputs on\n"); /* Capture 16 (originally 8) channels of S32_LE sound */ - /* printk("emufx.c: gpr=0x%x, tmp=0x%x\n",gpr, tmp); */ + /* + printk(KERN_DEBUG "emufx.c: gpr=0x%x, tmp=0x%x\n", + gpr, tmp); + */ /* For the EMU1010: How to get 32bit values from the DSP. High 16bits into L, low 16bits into R. */ /* A_P16VIN(0) is delayed by one sample, * so all other A_P16VIN channels will need to also be delayed diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index cf9276ddad4..78f62fd404c 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -44,7 +44,7 @@ static void snd_emu10k1_pcm_interrupt(struct snd_emu10k1 *emu, if (epcm->substream == NULL) return; #if 0 - printk("IRQ: position = 0x%x, period = 0x%x, size = 0x%x\n", + printk(KERN_DEBUG "IRQ: position = 0x%x, period = 0x%x, size = 0x%x\n", epcm->substream->runtime->hw->pointer(emu, epcm->substream), snd_pcm_lib_period_bytes(epcm->substream), snd_pcm_lib_buffer_bytes(epcm->substream)); @@ -146,7 +146,11 @@ static int snd_emu10k1_pcm_channel_alloc(struct snd_emu10k1_pcm * epcm, int voic 1, &epcm->extra); if (err < 0) { - /* printk("pcm_channel_alloc: failed extra: voices=%d, frame=%d\n", voices, frame); */ + /* + printk(KERN_DEBUG "pcm_channel_alloc: " + "failed extra: voices=%d, frame=%d\n", + voices, frame); + */ for (i = 0; i < voices; i++) { snd_emu10k1_voice_free(epcm->emu, epcm->voices[i]); epcm->voices[i] = NULL; @@ -737,7 +741,10 @@ static int snd_emu10k1_playback_trigger(struct snd_pcm_substream *substream, struct snd_emu10k1_pcm_mixer *mix; int result = 0; - /* printk("trigger - emu10k1 = 0x%x, cmd = %i, pointer = %i\n", (int)emu, cmd, substream->ops->pointer(substream)); */ + /* + printk(KERN_DEBUG "trigger - emu10k1 = 0x%x, cmd = %i, pointer = %i\n", + (int)emu, cmd, substream->ops->pointer(substream)) + */ spin_lock(&emu->reg_lock); switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -786,7 +793,10 @@ static int snd_emu10k1_capture_trigger(struct snd_pcm_substream *substream, /* hmm this should cause full and half full interrupt to be raised? */ outl(epcm->capture_ipr, emu->port + IPR); snd_emu10k1_intr_enable(emu, epcm->capture_inte); - /* printk("adccr = 0x%x, adcbs = 0x%x\n", epcm->adccr, epcm->adcbs); */ + /* + printk(KERN_DEBUG "adccr = 0x%x, adcbs = 0x%x\n", + epcm->adccr, epcm->adcbs); + */ switch (epcm->type) { case CAPTURE_AC97ADC: snd_emu10k1_ptr_write(emu, ADCCR, 0, epcm->capture_cr_val); @@ -857,7 +867,11 @@ static snd_pcm_uframes_t snd_emu10k1_playback_pointer(struct snd_pcm_substream * ptr -= runtime->buffer_size; } #endif - /* printk("ptr = 0x%x, buffer_size = 0x%x, period_size = 0x%x\n", ptr, runtime->buffer_size, runtime->period_size); */ + /* + printk(KERN_DEBUG + "ptr = 0x%x, buffer_size = 0x%x, period_size = 0x%x\n", + ptr, runtime->buffer_size, runtime->period_size); + */ return ptr; } @@ -1546,7 +1560,11 @@ static void snd_emu10k1_fx8010_playback_tram_poke1(unsigned short *dst_left, unsigned int count, unsigned int tram_shift) { - /* printk("tram_poke1: dst_left = 0x%p, dst_right = 0x%p, src = 0x%p, count = 0x%x\n", dst_left, dst_right, src, count); */ + /* + printk(KERN_DEBUG "tram_poke1: dst_left = 0x%p, dst_right = 0x%p, " + "src = 0x%p, count = 0x%x\n", + dst_left, dst_right, src, count); + */ if ((tram_shift & 1) == 0) { while (count--) { *dst_left-- = *src++; @@ -1623,7 +1641,12 @@ static int snd_emu10k1_fx8010_playback_prepare(struct snd_pcm_substream *substre struct snd_emu10k1_fx8010_pcm *pcm = &emu->fx8010.pcm[substream->number]; unsigned int i; - /* printk("prepare: etram_pages = 0x%p, dma_area = 0x%x, buffer_size = 0x%x (0x%x)\n", emu->fx8010.etram_pages, runtime->dma_area, runtime->buffer_size, runtime->buffer_size << 2); */ + /* + printk(KERN_DEBUG "prepare: etram_pages = 0x%p, dma_area = 0x%x, " + "buffer_size = 0x%x (0x%x)\n", + emu->fx8010.etram_pages, runtime->dma_area, + runtime->buffer_size, runtime->buffer_size << 2); + */ memset(&pcm->pcm_rec, 0, sizeof(pcm->pcm_rec)); pcm->pcm_rec.hw_buffer_size = pcm->buffer_size * 2; /* byte size */ pcm->pcm_rec.sw_buffer_size = snd_pcm_lib_buffer_bytes(substream); diff --git a/sound/pci/emu10k1/io.c b/sound/pci/emu10k1/io.c index b5a802bdeb7..4bfc31d1b28 100644 --- a/sound/pci/emu10k1/io.c +++ b/sound/pci/emu10k1/io.c @@ -226,7 +226,9 @@ int snd_emu10k1_i2c_write(struct snd_emu10k1 *emu, break; if (timeout > 1000) { - snd_printk("emu10k1:I2C:timeout status=0x%x\n", status); + snd_printk(KERN_WARNING + "emu10k1:I2C:timeout status=0x%x\n", + status); break; } } diff --git a/sound/pci/emu10k1/p16v.c b/sound/pci/emu10k1/p16v.c index 749a21b6bd0..e617acaf10e 100644 --- a/sound/pci/emu10k1/p16v.c +++ b/sound/pci/emu10k1/p16v.c @@ -168,7 +168,7 @@ static void snd_p16v_pcm_free_substream(struct snd_pcm_runtime *runtime) struct snd_emu10k1_pcm *epcm = runtime->private_data; if (epcm) { - //snd_printk("epcm free: %p\n", epcm); + /* snd_printk(KERN_DEBUG "epcm free: %p\n", epcm); */ kfree(epcm); } } @@ -183,14 +183,16 @@ static int snd_p16v_pcm_open_playback_channel(struct snd_pcm_substream *substrea int err; epcm = kzalloc(sizeof(*epcm), GFP_KERNEL); - //snd_printk("epcm kcalloc: %p\n", epcm); + /* snd_printk(KERN_DEBUG "epcm kcalloc: %p\n", epcm); */ if (epcm == NULL) return -ENOMEM; epcm->emu = emu; epcm->substream = substream; - //snd_printk("epcm device=%d, channel_id=%d\n", substream->pcm->device, channel_id); - + /* + snd_printk(KERN_DEBUG "epcm device=%d, channel_id=%d\n", + substream->pcm->device, channel_id); + */ runtime->private_data = epcm; runtime->private_free = snd_p16v_pcm_free_substream; @@ -200,10 +202,15 @@ static int snd_p16v_pcm_open_playback_channel(struct snd_pcm_substream *substrea channel->number = channel_id; channel->use=1; - //snd_printk("p16v: open channel_id=%d, channel=%p, use=0x%x\n", channel_id, channel, channel->use); - //printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel); - //channel->interrupt = snd_p16v_pcm_channel_interrupt; - channel->epcm=epcm; +#if 0 /* debug */ + snd_printk(KERN_DEBUG + "p16v: open channel_id=%d, channel=%p, use=0x%x\n", + channel_id, channel, channel->use); + printk(KERN_DEBUG "open:channel_id=%d, chip=%p, channel=%p\n", + channel_id, chip, channel); +#endif /* debug */ + /* channel->interrupt = snd_p16v_pcm_channel_interrupt; */ + channel->epcm = epcm; if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) return err; @@ -224,14 +231,16 @@ static int snd_p16v_pcm_open_capture_channel(struct snd_pcm_substream *substream int err; epcm = kzalloc(sizeof(*epcm), GFP_KERNEL); - //snd_printk("epcm kcalloc: %p\n", epcm); + /* snd_printk(KERN_DEBUG "epcm kcalloc: %p\n", epcm); */ if (epcm == NULL) return -ENOMEM; epcm->emu = emu; epcm->substream = substream; - //snd_printk("epcm device=%d, channel_id=%d\n", substream->pcm->device, channel_id); - + /* + snd_printk(KERN_DEBUG "epcm device=%d, channel_id=%d\n", + substream->pcm->device, channel_id); + */ runtime->private_data = epcm; runtime->private_free = snd_p16v_pcm_free_substream; @@ -241,10 +250,15 @@ static int snd_p16v_pcm_open_capture_channel(struct snd_pcm_substream *substream channel->number = channel_id; channel->use=1; - //snd_printk("p16v: open channel_id=%d, channel=%p, use=0x%x\n", channel_id, channel, channel->use); - //printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel); - //channel->interrupt = snd_p16v_pcm_channel_interrupt; - channel->epcm=epcm; +#if 0 /* debug */ + snd_printk(KERN_DEBUG + "p16v: open channel_id=%d, channel=%p, use=0x%x\n", + channel_id, channel, channel->use); + printk(KERN_DEBUG "open:channel_id=%d, chip=%p, channel=%p\n", + channel_id, chip, channel); +#endif /* debug */ + /* channel->interrupt = snd_p16v_pcm_channel_interrupt; */ + channel->epcm = epcm; if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) return err; @@ -334,9 +348,19 @@ static int snd_p16v_pcm_prepare_playback(struct snd_pcm_substream *substream) int i; u32 tmp; - //snd_printk("prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, periods=%u, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, runtime->periods, frames_to_bytes(runtime, 1)); - //snd_printk("dma_addr=%x, dma_area=%p, table_base=%p\n",runtime->dma_addr, runtime->dma_area, table_base); - //snd_printk("dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n",emu->p16v_buffer.addr, emu->p16v_buffer.area, emu->p16v_buffer.bytes); +#if 0 /* debug */ + snd_printk(KERN_DEBUG "prepare:channel_number=%d, rate=%d, " + "format=0x%x, channels=%d, buffer_size=%ld, " + "period_size=%ld, periods=%u, frames_to_bytes=%d\n", + channel, runtime->rate, runtime->format, runtime->channels, + runtime->buffer_size, runtime->period_size, + runtime->periods, frames_to_bytes(runtime, 1)); + snd_printk(KERN_DEBUG "dma_addr=%x, dma_area=%p, table_base=%p\n", + runtime->dma_addr, runtime->dma_area, table_base); + snd_printk(KERN_DEBUG "dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n", + emu->p16v_buffer.addr, emu->p16v_buffer.area, + emu->p16v_buffer.bytes); +#endif /* debug */ tmp = snd_emu10k1_ptr_read(emu, A_SPDIF_SAMPLERATE, channel); switch (runtime->rate) { case 44100: @@ -379,7 +403,15 @@ static int snd_p16v_pcm_prepare_capture(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; int channel = substream->pcm->device - emu->p16v_device_offset; u32 tmp; - //printk("prepare capture:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, frames_to_bytes(runtime, 1)); + + /* + printk(KERN_DEBUG "prepare capture:channel_number=%d, rate=%d, " + "format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, " + "frames_to_bytes=%d\n", + channel, runtime->rate, runtime->format, runtime->channels, + runtime->buffer_size, runtime->period_size, + frames_to_bytes(runtime, 1)); + */ tmp = snd_emu10k1_ptr_read(emu, A_SPDIF_SAMPLERATE, channel); switch (runtime->rate) { case 44100: @@ -459,13 +491,13 @@ static int snd_p16v_pcm_trigger_playback(struct snd_pcm_substream *substream, runtime = s->runtime; epcm = runtime->private_data; channel = substream->pcm->device-emu->p16v_device_offset; - //snd_printk("p16v channel=%d\n",channel); + /* snd_printk(KERN_DEBUG "p16v channel=%d\n", channel); */ epcm->running = running; basic |= (0x1<buffer_size; printk(KERN_WARNING "buffer capture limited!\n"); } - //printk("ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n", ptr1, ptr2, ptr, (int)runtime->buffer_size, (int)runtime->period_size, (int)runtime->frame_bits, (int)runtime->rate); - + /* + printk(KERN_DEBUG "ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, " + "buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n", + ptr1, ptr2, ptr, (int)runtime->buffer_size, + (int)runtime->period_size, (int)runtime->frame_bits, + (int)runtime->rate); + */ return ptr; } @@ -592,7 +629,10 @@ int snd_p16v_free(struct snd_emu10k1 *chip) // release the data if (chip->p16v_buffer.area) { snd_dma_free_pages(&chip->p16v_buffer); - //snd_printk("period lables free: %p\n", &chip->p16v_buffer); + /* + snd_printk(KERN_DEBUG "period lables free: %p\n", + &chip->p16v_buffer); + */ } return 0; } @@ -604,7 +644,7 @@ int __devinit snd_p16v_pcm(struct snd_emu10k1 *emu, int device, struct snd_pcm * int err; int capture=1; - //snd_printk("snd_p16v_pcm called. device=%d\n", device); + /* snd_printk("KERN_DEBUG snd_p16v_pcm called. device=%d\n", device); */ emu->p16v_device_offset = device; if (rpcm) *rpcm = NULL; @@ -631,7 +671,10 @@ int __devinit snd_p16v_pcm(struct snd_emu10k1 *emu, int device, struct snd_pcm * snd_dma_pci_data(emu->pci), ((65536 - 64) * 8), ((65536 - 64) * 8))) < 0) return err; - //snd_printk("preallocate playback substream: err=%d\n", err); + /* + snd_printk(KERN_DEBUG + "preallocate playback substream: err=%d\n", err); + */ } for (substream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream; @@ -642,7 +685,10 @@ int __devinit snd_p16v_pcm(struct snd_emu10k1 *emu, int device, struct snd_pcm * snd_dma_pci_data(emu->pci), 65536 - 64, 65536 - 64)) < 0) return err; - //snd_printk("preallocate capture substream: err=%d\n", err); + /* + snd_printk(KERN_DEBUG + "preallocate capture substream: err=%d\n", err); + */ } if (rpcm) diff --git a/sound/pci/emu10k1/voice.c b/sound/pci/emu10k1/voice.c index d7300a1aa26..20b8da250bd 100644 --- a/sound/pci/emu10k1/voice.c +++ b/sound/pci/emu10k1/voice.c @@ -53,7 +53,10 @@ static int voice_alloc(struct snd_emu10k1 *emu, int type, int number, *rvoice = NULL; first_voice = last_voice = 0; for (i = emu->next_free_voice, j = 0; j < NUM_G ; i += number, j += number) { - // printk("i %d j %d next free %d!\n", i, j, emu->next_free_voice); + /* + printk(KERN_DEBUG "i %d j %d next free %d!\n", + i, j, emu->next_free_voice); + */ i %= NUM_G; /* stereo voices must be even/odd */ @@ -71,7 +74,7 @@ static int voice_alloc(struct snd_emu10k1 *emu, int type, int number, } } if (!skip) { - // printk("allocated voice %d\n", i); + /* printk(KERN_DEBUG "allocated voice %d\n", i); */ first_voice = i; last_voice = (i + number) % NUM_G; emu->next_free_voice = last_voice; @@ -84,7 +87,10 @@ static int voice_alloc(struct snd_emu10k1 *emu, int type, int number, for (i = 0; i < number; i++) { voice = &emu->voices[(first_voice + i) % NUM_G]; - // printk("voice alloc - %i, %i of %i\n", voice->number, idx-first_voice+1, number); + /* + printk(kERN_DEBUG "voice alloc - %i, %i of %i\n", + voice->number, idx-first_voice+1, number); + */ voice->use = 1; switch (type) { case EMU10K1_PCM: -- cgit v1.2.3-70-g09d2 From 14ab08610971eb1db572ad8ca63acd13bc4d4caf Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 5 Feb 2009 16:09:57 +0100 Subject: ALSA: intel8x0 - Add missing KERN_* prefix to printk Signed-off-by: Takashi Iwai --- sound/pci/intel8x0.c | 11 +++++++---- sound/pci/intel8x0m.c | 14 ++++++++++---- 2 files changed, 17 insertions(+), 8 deletions(-) diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index b13ef1e2a4a..0f7d1291190 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -689,7 +689,7 @@ static void snd_intel8x0_setup_periods(struct intel8x0 *chip, struct ichdev *ich bdbar[idx + 1] = cpu_to_le32(0x80000000 | /* interrupt on completion */ ichdev->fragsize >> ichdev->pos_shift); #if 0 - printk("bdbar[%i] = 0x%x [0x%x]\n", + printk(KERN_DEBUG "bdbar[%i] = 0x%x [0x%x]\n", idx + 0, bdbar[idx + 0], bdbar[idx + 1]); #endif } @@ -701,8 +701,10 @@ static void snd_intel8x0_setup_periods(struct intel8x0 *chip, struct ichdev *ich ichdev->lvi_frag = ICH_REG_LVI_MASK % ichdev->frags; ichdev->position = 0; #if 0 - printk("lvi_frag = %i, frags = %i, period_size = 0x%x, period_size1 = 0x%x\n", - ichdev->lvi_frag, ichdev->frags, ichdev->fragsize, ichdev->fragsize1); + printk(KERN_DEBUG "lvi_frag = %i, frags = %i, period_size = 0x%x, " + "period_size1 = 0x%x\n", + ichdev->lvi_frag, ichdev->frags, ichdev->fragsize, + ichdev->fragsize1); #endif /* clear interrupts */ iputbyte(chip, port + ichdev->roff_sr, ICH_FIFOE | ICH_BCIS | ICH_LVBCI); @@ -768,7 +770,8 @@ static inline void snd_intel8x0_update(struct intel8x0 *chip, struct ichdev *ich ichdev->lvi_frag %= ichdev->frags; ichdev->bdbar[ichdev->lvi * 2] = cpu_to_le32(ichdev->physbuf + ichdev->lvi_frag * ichdev->fragsize1); #if 0 - printk("new: bdbar[%i] = 0x%x [0x%x], prefetch = %i, all = 0x%x, 0x%x\n", + printk(KERN_DEBUG "new: bdbar[%i] = 0x%x [0x%x], prefetch = %i, " + "all = 0x%x, 0x%x\n", ichdev->lvi * 2, ichdev->bdbar[ichdev->lvi * 2], ichdev->bdbar[ichdev->lvi * 2 + 1], inb(ICH_REG_OFF_PIV + port), inl(port + 4), inb(port + ICH_REG_OFF_CR)); diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index 93449e46456..7c819fd824a 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -411,7 +411,10 @@ static void snd_intel8x0_setup_periods(struct intel8x0m *chip, struct ichdev *ic bdbar[idx + 0] = cpu_to_le32(ichdev->physbuf + (((idx >> 1) * ichdev->fragsize) % ichdev->size)); bdbar[idx + 1] = cpu_to_le32(0x80000000 | /* interrupt on completion */ ichdev->fragsize >> chip->pcm_pos_shift); - // printk("bdbar[%i] = 0x%x [0x%x]\n", idx + 0, bdbar[idx + 0], bdbar[idx + 1]); + /* + printk(KERN_DEBUG "bdbar[%i] = 0x%x [0x%x]\n", + idx + 0, bdbar[idx + 0], bdbar[idx + 1]); + */ } ichdev->frags = ichdev->size / ichdev->fragsize; } @@ -421,8 +424,10 @@ static void snd_intel8x0_setup_periods(struct intel8x0m *chip, struct ichdev *ic ichdev->lvi_frag = ICH_REG_LVI_MASK % ichdev->frags; ichdev->position = 0; #if 0 - printk("lvi_frag = %i, frags = %i, period_size = 0x%x, period_size1 = 0x%x\n", - ichdev->lvi_frag, ichdev->frags, ichdev->fragsize, ichdev->fragsize1); + printk(KERN_DEBUG "lvi_frag = %i, frags = %i, period_size = 0x%x, " + "period_size1 = 0x%x\n", + ichdev->lvi_frag, ichdev->frags, ichdev->fragsize, + ichdev->fragsize1); #endif /* clear interrupts */ iputbyte(chip, port + ichdev->roff_sr, ICH_FIFOE | ICH_BCIS | ICH_LVBCI); @@ -465,7 +470,8 @@ static inline void snd_intel8x0_update(struct intel8x0m *chip, struct ichdev *ic ichdev->lvi_frag * ichdev->fragsize1); #if 0 - printk("new: bdbar[%i] = 0x%x [0x%x], prefetch = %i, all = 0x%x, 0x%x\n", + printk(KERN_DEBUG "new: bdbar[%i] = 0x%x [0x%x], " + "prefetch = %i, all = 0x%x, 0x%x\n", ichdev->lvi * 2, ichdev->bdbar[ichdev->lvi * 2], ichdev->bdbar[ichdev->lvi * 2 + 1], inb(ICH_REG_OFF_PIV + port), inl(port + 4), inb(port + ICH_REG_OFF_CR)); -- cgit v1.2.3-70-g09d2 From ee419653a38de93b75a577851d9e4003cf0bbe07 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 5 Feb 2009 16:11:31 +0100 Subject: ALSA: Fix missing KERN_* prefix to printk in sound/pci Signed-off-by: Takashi Iwai --- sound/pci/ac97/ac97_codec.c | 5 +- sound/pci/ak4531_codec.c | 3 +- sound/pci/als300.c | 2 +- sound/pci/au88x0/au88x0_a3d.c | 7 ++- sound/pci/au88x0/au88x0_core.c | 19 +++++-- sound/pci/au88x0/au88x0_synth.c | 39 ++++++++++--- sound/pci/azt3328.c | 8 +-- sound/pci/ca0106/ca0106_main.c | 91 +++++++++++++++++++++++------- sound/pci/cs4281.c | 6 +- sound/pci/cs46xx/cs46xx_lib.c | 6 +- sound/pci/cs46xx/cs46xx_lib.h | 6 +- sound/pci/cs5535audio/cs5535audio.c | 2 +- sound/pci/ens1370.c | 3 +- sound/pci/es1938.c | 23 +++++--- sound/pci/mixart/mixart_hwdep.c | 46 ++++++++------- sound/pci/sonicvibes.c | 109 ++++++++++++++++++++++++------------ sound/pci/trident/trident_main.c | 57 ++++++++++--------- sound/pci/via82xx.c | 5 +- sound/pci/via82xx_modem.c | 5 +- sound/pci/vx222/vx222_ops.c | 8 ++- sound/pci/ymfpci/ymfpci_main.c | 14 ++++- 21 files changed, 318 insertions(+), 146 deletions(-) diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index e2b843b4f9d..bc707b60385 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -1643,7 +1643,10 @@ static int snd_ac97_modem_build(struct snd_card *card, struct snd_ac97 * ac97) { int err, idx; - //printk("AC97_GPIO_CFG = %x\n",snd_ac97_read(ac97,AC97_GPIO_CFG)); + /* + printk(KERN_DEBUG "AC97_GPIO_CFG = %x\n", + snd_ac97_read(ac97,AC97_GPIO_CFG)); + */ snd_ac97_write(ac97, AC97_GPIO_CFG, 0xffff & ~(AC97_GPIO_LINE1_OH)); snd_ac97_write(ac97, AC97_GPIO_POLARITY, 0xffff & ~(AC97_GPIO_LINE1_OH)); snd_ac97_write(ac97, AC97_GPIO_STICKY, 0xffff); diff --git a/sound/pci/ak4531_codec.c b/sound/pci/ak4531_codec.c index 0f819ddb3eb..fd135e3d8a8 100644 --- a/sound/pci/ak4531_codec.c +++ b/sound/pci/ak4531_codec.c @@ -51,7 +51,8 @@ static void snd_ak4531_dump(struct snd_ak4531 *ak4531) int idx; for (idx = 0; idx < 0x19; idx++) - printk("ak4531 0x%x: 0x%x\n", idx, ak4531->regs[idx]); + printk(KERN_DEBUG "ak4531 0x%x: 0x%x\n", + idx, ak4531->regs[idx]); } #endif diff --git a/sound/pci/als300.c b/sound/pci/als300.c index 8df6824b51c..a2c35c1081c 100644 --- a/sound/pci/als300.c +++ b/sound/pci/als300.c @@ -91,7 +91,7 @@ #define DEBUG_PLAY_REC 0 #if DEBUG_CALLS -#define snd_als300_dbgcalls(format, args...) printk(format, ##args) +#define snd_als300_dbgcalls(format, args...) printk(KERN_DEBUG format, ##args) #define snd_als300_dbgcallenter() printk(KERN_ERR "--> %s\n", __func__) #define snd_als300_dbgcallleave() printk(KERN_ERR "<-- %s\n", __func__) #else diff --git a/sound/pci/au88x0/au88x0_a3d.c b/sound/pci/au88x0/au88x0_a3d.c index 649849e540d..f4aa8ff6f5f 100644 --- a/sound/pci/au88x0/au88x0_a3d.c +++ b/sound/pci/au88x0/au88x0_a3d.c @@ -462,9 +462,10 @@ static void a3dsrc_ZeroSliceIO(a3dsrc_t * a) /* Reset Single A3D source. */ static void a3dsrc_ZeroState(a3dsrc_t * a) { - - //printk("vortex: ZeroState slice: %d, source %d\n", a->slice, a->source); - + /* + printk(KERN_DEBUG "vortex: ZeroState slice: %d, source %d\n", + a->slice, a->source); + */ a3dsrc_SetAtmosState(a, 0, 0, 0, 0); a3dsrc_SetHrtfState(a, A3dHrirZeros, A3dHrirZeros); a3dsrc_SetItdDline(a, A3dItdDlineZeros); diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c index b070e571451..e6a04d037c1 100644 --- a/sound/pci/au88x0/au88x0_core.c +++ b/sound/pci/au88x0/au88x0_core.c @@ -1135,7 +1135,10 @@ vortex_adbdma_setbuffers(vortex_t * vortex, int adbdma, snd_pcm_sgbuf_get_addr(dma->substream, 0)); break; } - //printk("vortex: cfg0 = 0x%x\nvortex: cfg1=0x%x\n", dma->cfg0, dma->cfg1); + /* + printk(KERN_DEBUG "vortex: cfg0 = 0x%x\nvortex: cfg1=0x%x\n", + dma->cfg0, dma->cfg1); + */ hwwrite(vortex->mmio, VORTEX_ADBDMA_BUFCFG0 + (adbdma << 3), dma->cfg0); hwwrite(vortex->mmio, VORTEX_ADBDMA_BUFCFG1 + (adbdma << 3), dma->cfg1); @@ -1959,7 +1962,7 @@ vortex_connect_codecplay(vortex_t * vortex, int en, unsigned char mixers[]) ADB_CODECOUT(0 + 4)); vortex_connection_mix_adb(vortex, en, 0x11, mixers[3], ADB_CODECOUT(1 + 4)); - //printk("SDAC detected "); + /* printk(KERN_DEBUG "SDAC detected "); */ } #else // Use plain direct output to codec. @@ -2013,7 +2016,11 @@ vortex_adb_checkinout(vortex_t * vortex, int resmap[], int out, int restype) resmap[restype] |= (1 << i); else vortex->dma_adb[i].resources[restype] |= (1 << i); - //printk("vortex: ResManager: type %d out %d\n", restype, i); + /* + printk(KERN_DEBUG + "vortex: ResManager: type %d out %d\n", + restype, i); + */ return i; } } @@ -2024,7 +2031,11 @@ vortex_adb_checkinout(vortex_t * vortex, int resmap[], int out, int restype) for (i = 0; i < qty; i++) { if (resmap[restype] & (1 << i)) { resmap[restype] &= ~(1 << i); - //printk("vortex: ResManager: type %d in %d\n",restype, i); + /* + printk(KERN_DEBUG + "vortex: ResManager: type %d in %d\n", + restype, i); + */ return i; } } diff --git a/sound/pci/au88x0/au88x0_synth.c b/sound/pci/au88x0/au88x0_synth.c index 978b856f562..2805e34bd41 100644 --- a/sound/pci/au88x0/au88x0_synth.c +++ b/sound/pci/au88x0/au88x0_synth.c @@ -213,38 +213,59 @@ vortex_wt_SetReg(vortex_t * vortex, unsigned char reg, int wt, switch (reg) { /* Voice specific parameters */ case 0: /* running */ - //printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_RUN(wt), (int)val); + /* + printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", + WT_RUN(wt), (int)val); + */ hwwrite(vortex->mmio, WT_RUN(wt), val); return 0xc; break; case 1: /* param 0 */ - //printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_PARM(wt,0), (int)val); + /* + printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", + WT_PARM(wt,0), (int)val); + */ hwwrite(vortex->mmio, WT_PARM(wt, 0), val); return 0xc; break; case 2: /* param 1 */ - //printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_PARM(wt,1), (int)val); + /* + printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", + WT_PARM(wt,1), (int)val); + */ hwwrite(vortex->mmio, WT_PARM(wt, 1), val); return 0xc; break; case 3: /* param 2 */ - //printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_PARM(wt,2), (int)val); + /* + printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", + WT_PARM(wt,2), (int)val); + */ hwwrite(vortex->mmio, WT_PARM(wt, 2), val); return 0xc; break; case 4: /* param 3 */ - //printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_PARM(wt,3), (int)val); + /* + printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", + WT_PARM(wt,3), (int)val); + */ hwwrite(vortex->mmio, WT_PARM(wt, 3), val); return 0xc; break; case 6: /* mute */ - //printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_MUTE(wt), (int)val); + /* + printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", + WT_MUTE(wt), (int)val); + */ hwwrite(vortex->mmio, WT_MUTE(wt), val); return 0xc; break; case 0xb: { /* delay */ - //printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_DELAY(wt,0), (int)val); + /* + printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", + WT_DELAY(wt,0), (int)val); + */ hwwrite(vortex->mmio, WT_DELAY(wt, 3), val); hwwrite(vortex->mmio, WT_DELAY(wt, 2), val); hwwrite(vortex->mmio, WT_DELAY(wt, 1), val); @@ -272,7 +293,9 @@ vortex_wt_SetReg(vortex_t * vortex, unsigned char reg, int wt, return 0; break; } - //printk("vortex: WT SetReg(0x%x) = 0x%08x\n", ecx, (int)val); + /* + printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", ecx, (int)val); + */ hwwrite(vortex->mmio, ecx, val); return 1; } diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 333007c523a..8121763b0c1 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -211,25 +211,25 @@ MODULE_SUPPORTED_DEVICE("{{Aztech,AZF3328}}"); #endif #if DEBUG_MIXER -#define snd_azf3328_dbgmixer(format, args...) printk(format, ##args) +#define snd_azf3328_dbgmixer(format, args...) printk(KERN_DEBUG format, ##args) #else #define snd_azf3328_dbgmixer(format, args...) #endif #if DEBUG_PLAY_REC -#define snd_azf3328_dbgplay(format, args...) printk(KERN_ERR format, ##args) +#define snd_azf3328_dbgplay(format, args...) printk(KERN_DEBUG format, ##args) #else #define snd_azf3328_dbgplay(format, args...) #endif #if DEBUG_MISC -#define snd_azf3328_dbgtimer(format, args...) printk(KERN_ERR format, ##args) +#define snd_azf3328_dbgtimer(format, args...) printk(KERN_DEBUG format, ##args) #else #define snd_azf3328_dbgtimer(format, args...) #endif #if DEBUG_GAME -#define snd_azf3328_dbggame(format, args...) printk(KERN_ERR format, ##args) +#define snd_azf3328_dbggame(format, args...) printk(KERN_DEBUG format, ##args) #else #define snd_azf3328_dbggame(format, args...) #endif diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index 0e62205d408..f2f8fd17ea4 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -404,7 +404,9 @@ int snd_ca0106_i2c_write(struct snd_ca0106 *emu, } tmp = reg << 25 | value << 16; - // snd_printk("I2C-write:reg=0x%x, value=0x%x\n", reg, value); + /* + snd_printk(KERN_DEBUG "I2C-write:reg=0x%x, value=0x%x\n", reg, value); + */ /* Not sure what this I2C channel controls. */ /* snd_ca0106_ptr_write(emu, I2C_D0, 0, tmp); */ @@ -422,7 +424,7 @@ int snd_ca0106_i2c_write(struct snd_ca0106 *emu, /* Wait till the transaction ends */ while (1) { status = snd_ca0106_ptr_read(emu, I2C_A, 0); - //snd_printk("I2C:status=0x%x\n", status); + /*snd_printk(KERN_DEBUG "I2C:status=0x%x\n", status);*/ timeout++; if ((status & I2C_A_ADC_START) == 0) break; @@ -521,7 +523,10 @@ static int snd_ca0106_pcm_open_playback_channel(struct snd_pcm_substream *substr channel->number = channel_id; channel->use = 1; - //printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel); + /* + printk(KERN_DEBUG "open:channel_id=%d, chip=%p, channel=%p\n", + channel_id, chip, channel); + */ //channel->interrupt = snd_ca0106_pcm_channel_interrupt; channel->epcm = epcm; if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) @@ -614,7 +619,10 @@ static int snd_ca0106_pcm_open_capture_channel(struct snd_pcm_substream *substre channel->number = channel_id; channel->use = 1; - //printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel); + /* + printk(KERN_DEBUG "open:channel_id=%d, chip=%p, channel=%p\n", + channel_id, chip, channel); + */ //channel->interrupt = snd_ca0106_pcm_channel_interrupt; channel->epcm = epcm; if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) @@ -705,9 +713,20 @@ static int snd_ca0106_pcm_prepare_playback(struct snd_pcm_substream *substream) u32 reg71; int i; - //snd_printk("prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, periods=%u, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, runtime->periods, frames_to_bytes(runtime, 1)); - //snd_printk("dma_addr=%x, dma_area=%p, table_base=%p\n",runtime->dma_addr, runtime->dma_area, table_base); - //snd_printk("dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n",emu->buffer.addr, emu->buffer.area, emu->buffer.bytes); +#if 0 /* debug */ + snd_printk(KERN_DEBUG + "prepare:channel_number=%d, rate=%d, format=0x%x, " + "channels=%d, buffer_size=%ld, period_size=%ld, " + "periods=%u, frames_to_bytes=%d\n", + channel, runtime->rate, runtime->format, + runtime->channels, runtime->buffer_size, + runtime->period_size, runtime->periods, + frames_to_bytes(runtime, 1)); + snd_printk(KERN_DEBUG "dma_addr=%x, dma_area=%p, table_base=%p\n", + runtime->dma_addr, runtime->dma_area, table_base); + snd_printk(KERN_DEBUG "dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n", + emu->buffer.addr, emu->buffer.area, emu->buffer.bytes); +#endif /* debug */ /* Rate can be set per channel. */ /* reg40 control host to fifo */ /* reg71 controls DAC rate. */ @@ -799,9 +818,20 @@ static int snd_ca0106_pcm_prepare_capture(struct snd_pcm_substream *substream) u32 reg71_set = 0; u32 reg71; - //snd_printk("prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, periods=%u, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, runtime->periods, frames_to_bytes(runtime, 1)); - //snd_printk("dma_addr=%x, dma_area=%p, table_base=%p\n",runtime->dma_addr, runtime->dma_area, table_base); - //snd_printk("dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n",emu->buffer.addr, emu->buffer.area, emu->buffer.bytes); +#if 0 /* debug */ + snd_printk(KERN_DEBUG + "prepare:channel_number=%d, rate=%d, format=0x%x, " + "channels=%d, buffer_size=%ld, period_size=%ld, " + "periods=%u, frames_to_bytes=%d\n", + channel, runtime->rate, runtime->format, + runtime->channels, runtime->buffer_size, + runtime->period_size, runtime->periods, + frames_to_bytes(runtime, 1)); + snd_printk(KERN_DEBUG "dma_addr=%x, dma_area=%p, table_base=%p\n", + runtime->dma_addr, runtime->dma_area, table_base); + snd_printk(KERN_DEBUG "dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n", + emu->buffer.addr, emu->buffer.area, emu->buffer.bytes); +#endif /* debug */ /* reg71 controls ADC rate. */ switch (runtime->rate) { case 44100: @@ -846,7 +876,14 @@ static int snd_ca0106_pcm_prepare_capture(struct snd_pcm_substream *substream) } - //printk("prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, frames_to_bytes(runtime, 1)); + /* + printk(KERN_DEBUG + "prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, " + "buffer_size=%ld, period_size=%ld, frames_to_bytes=%d\n", + channel, runtime->rate, runtime->format, runtime->channels, + runtime->buffer_size, runtime->period_size, + frames_to_bytes(runtime, 1)); + */ snd_ca0106_ptr_write(emu, 0x13, channel, 0); snd_ca0106_ptr_write(emu, CAPTURE_DMA_ADDR, channel, runtime->dma_addr); snd_ca0106_ptr_write(emu, CAPTURE_BUFFER_SIZE, channel, frames_to_bytes(runtime, runtime->buffer_size)<<16); // buffer size in bytes @@ -888,13 +925,13 @@ static int snd_ca0106_pcm_trigger_playback(struct snd_pcm_substream *substream, runtime = s->runtime; epcm = runtime->private_data; channel = epcm->channel_id; - /* snd_printk("channel=%d\n",channel); */ + /* snd_printk(KERN_DEBUG "channel=%d\n", channel); */ epcm->running = running; basic |= (0x1 << channel); extended |= (0x10 << channel); snd_pcm_trigger_done(s, substream); } - /* snd_printk("basic=0x%x, extended=0x%x\n",basic, extended); */ + /* snd_printk(KERN_DEBUG "basic=0x%x, extended=0x%x\n",basic, extended); */ switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -972,8 +1009,13 @@ snd_ca0106_pcm_pointer_playback(struct snd_pcm_substream *substream) ptr=ptr2; if (ptr >= runtime->buffer_size) ptr -= runtime->buffer_size; - //printk("ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n", ptr1, ptr2, ptr, (int)runtime->buffer_size, (int)runtime->period_size, (int)runtime->frame_bits, (int)runtime->rate); - + /* + printk(KERN_DEBUG "ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, " + "buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n", + ptr1, ptr2, ptr, (int)runtime->buffer_size, + (int)runtime->period_size, (int)runtime->frame_bits, + (int)runtime->rate); + */ return ptr; } @@ -995,8 +1037,13 @@ snd_ca0106_pcm_pointer_capture(struct snd_pcm_substream *substream) ptr=ptr2; if (ptr >= runtime->buffer_size) ptr -= runtime->buffer_size; - //printk("ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n", ptr1, ptr2, ptr, (int)runtime->buffer_size, (int)runtime->period_size, (int)runtime->frame_bits, (int)runtime->rate); - + /* + printk(KERN_DEBUG "ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, " + "buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n", + ptr1, ptr2, ptr, (int)runtime->buffer_size, + (int)runtime->period_size, (int)runtime->frame_bits, + (int)runtime->rate); + */ return ptr; } @@ -1181,8 +1228,12 @@ static irqreturn_t snd_ca0106_interrupt(int irq, void *dev_id) return IRQ_NONE; stat76 = snd_ca0106_ptr_read(chip, EXTENDED_INT, 0); - //snd_printk("interrupt status = 0x%08x, stat76=0x%08x\n", status, stat76); - //snd_printk("ptr=0x%08x\n",snd_ca0106_ptr_read(chip, PLAYBACK_POINTER, 0)); + /* + snd_printk(KERN_DEBUG "interrupt status = 0x%08x, stat76=0x%08x\n", + status, stat76); + snd_printk(KERN_DEBUG "ptr=0x%08x\n", + snd_ca0106_ptr_read(chip, PLAYBACK_POINTER, 0)); + */ mask = 0x11; /* 0x1 for one half, 0x10 for the other half period. */ for(i = 0; i < 4; i++) { pchannel = &(chip->playback_channels[i]); @@ -1470,7 +1521,7 @@ static void ca0106_init_chip(struct snd_ca0106 *chip, int resume) int size, n; size = ARRAY_SIZE(i2c_adc_init); - /* snd_printk("I2C:array size=0x%x\n", size); */ + /* snd_printk(KERN_DEBUG "I2C:array size=0x%x\n", size); */ for (n = 0; n < size; n++) snd_ca0106_i2c_write(chip, i2c_adc_init[n][0], i2c_adc_init[n][1]); diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c index 192e7842e18..415e88f2c62 100644 --- a/sound/pci/cs4281.c +++ b/sound/pci/cs4281.c @@ -834,7 +834,11 @@ static snd_pcm_uframes_t snd_cs4281_pointer(struct snd_pcm_substream *substream) struct cs4281_dma *dma = runtime->private_data; struct cs4281 *chip = snd_pcm_substream_chip(substream); - // printk("DCC = 0x%x, buffer_size = 0x%x, jiffies = %li\n", snd_cs4281_peekBA0(chip, dma->regDCC), runtime->buffer_size, jiffies); + /* + printk(KERN_DEBUG "DCC = 0x%x, buffer_size = 0x%x, jiffies = %li\n", + snd_cs4281_peekBA0(chip, dma->regDCC), runtime->buffer_size, + jiffies); + */ return runtime->buffer_size - snd_cs4281_peekBA0(chip, dma->regDCC) - 1; } diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index 8ab07aa6365..1be96ead424 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -194,7 +194,7 @@ static unsigned short snd_cs46xx_codec_read(struct snd_cs46xx *chip, * ACSDA = Status Data Register = 474h */ #if 0 - printk("e) reg = 0x%x, val = 0x%x, BA0_ACCAD = 0x%x\n", reg, + printk(KERN_DEBUG "e) reg = 0x%x, val = 0x%x, BA0_ACCAD = 0x%x\n", reg, snd_cs46xx_peekBA0(chip, BA0_ACSDA), snd_cs46xx_peekBA0(chip, BA0_ACCAD)); #endif @@ -428,8 +428,8 @@ static int cs46xx_wait_for_fifo(struct snd_cs46xx * chip,int retry_timeout) } if(status & SERBST_WBSY) { - snd_printk( KERN_ERR "cs46xx: failure waiting for FIFO command to complete\n"); - + snd_printk(KERN_ERR "cs46xx: failure waiting for " + "FIFO command to complete\n"); return -EINVAL; } diff --git a/sound/pci/cs46xx/cs46xx_lib.h b/sound/pci/cs46xx/cs46xx_lib.h index 018a7de5601..4eb55aa3361 100644 --- a/sound/pci/cs46xx/cs46xx_lib.h +++ b/sound/pci/cs46xx/cs46xx_lib.h @@ -62,7 +62,11 @@ static inline void snd_cs46xx_poke(struct snd_cs46xx *chip, unsigned long reg, u unsigned int bank = reg >> 16; unsigned int offset = reg & 0xffff; - /*if (bank == 0) printk("snd_cs46xx_poke: %04X - %08X\n",reg >> 2,val); */ + /* + if (bank == 0) + printk(KERN_DEBUG "snd_cs46xx_poke: %04X - %08X\n", + reg >> 2,val); + */ writel(val, chip->region.idx[bank+1].remap_addr + offset); } diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c index 826e6dec2e9..6506201d56f 100644 --- a/sound/pci/cs5535audio/cs5535audio.c +++ b/sound/pci/cs5535audio/cs5535audio.c @@ -312,7 +312,7 @@ static int __devinit snd_cs5535audio_create(struct snd_card *card, if (request_irq(pci->irq, snd_cs5535audio_interrupt, IRQF_SHARED, "CS5535 Audio", cs5535au)) { - snd_printk("unable to grab IRQ %d\n", pci->irq); + snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq); err = -EBUSY; goto sndfail; } diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index 9bf95367c88..17674b3406b 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -584,7 +584,8 @@ static void snd_es1370_codec_write(struct snd_ak4531 *ak4531, unsigned long end_time = jiffies + HZ / 10; #if 0 - printk("CODEC WRITE: reg = 0x%x, val = 0x%x (0x%x), creg = 0x%x\n", + printk(KERN_DEBUG + "CODEC WRITE: reg = 0x%x, val = 0x%x (0x%x), creg = 0x%x\n", reg, val, ES_1370_CODEC_WRITE(reg, val), ES_REG(ensoniq, 1370_CODEC)); #endif do { diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c index 4cd9a1faaec..e4ba84bed4a 100644 --- a/sound/pci/es1938.c +++ b/sound/pci/es1938.c @@ -1673,18 +1673,22 @@ static irqreturn_t snd_es1938_interrupt(int irq, void *dev_id) status = inb(SLIO_REG(chip, IRQCONTROL)); #if 0 - printk("Es1938debug - interrupt status: =0x%x\n", status); + printk(KERN_DEBUG "Es1938debug - interrupt status: =0x%x\n", status); #endif /* AUDIO 1 */ if (status & 0x10) { #if 0 - printk("Es1938debug - AUDIO channel 1 interrupt\n"); - printk("Es1938debug - AUDIO channel 1 DMAC DMA count: %u\n", + printk(KERN_DEBUG + "Es1938debug - AUDIO channel 1 interrupt\n"); + printk(KERN_DEBUG + "Es1938debug - AUDIO channel 1 DMAC DMA count: %u\n", inw(SLDM_REG(chip, DMACOUNT))); - printk("Es1938debug - AUDIO channel 1 DMAC DMA base: %u\n", + printk(KERN_DEBUG + "Es1938debug - AUDIO channel 1 DMAC DMA base: %u\n", inl(SLDM_REG(chip, DMAADDR))); - printk("Es1938debug - AUDIO channel 1 DMAC DMA status: 0x%x\n", + printk(KERN_DEBUG + "Es1938debug - AUDIO channel 1 DMAC DMA status: 0x%x\n", inl(SLDM_REG(chip, DMASTATUS))); #endif /* clear irq */ @@ -1699,10 +1703,13 @@ static irqreturn_t snd_es1938_interrupt(int irq, void *dev_id) /* AUDIO 2 */ if (status & 0x20) { #if 0 - printk("Es1938debug - AUDIO channel 2 interrupt\n"); - printk("Es1938debug - AUDIO channel 2 DMAC DMA count: %u\n", + printk(KERN_DEBUG + "Es1938debug - AUDIO channel 2 interrupt\n"); + printk(KERN_DEBUG + "Es1938debug - AUDIO channel 2 DMAC DMA count: %u\n", inw(SLIO_REG(chip, AUDIO2DMACOUNT))); - printk("Es1938debug - AUDIO channel 2 DMAC DMA base: %u\n", + printk(KERN_DEBUG + "Es1938debug - AUDIO channel 2 DMAC DMA base: %u\n", inl(SLIO_REG(chip, AUDIO2DMAADDR))); #endif diff --git a/sound/pci/mixart/mixart_hwdep.c b/sound/pci/mixart/mixart_hwdep.c index 3782b52bc0e..dda562081d7 100644 --- a/sound/pci/mixart/mixart_hwdep.c +++ b/sound/pci/mixart/mixart_hwdep.c @@ -345,8 +345,8 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw status_daught = readl_be( MIXART_MEM( mgr,MIXART_PSEUDOREG_DXLX_STATUS_OFFSET )); /* motherboard xilinx status 5 will say that the board is performing a reset */ - if( status_xilinx == 5 ) { - snd_printk( KERN_ERR "miXart is resetting !\n"); + if (status_xilinx == 5) { + snd_printk(KERN_ERR "miXart is resetting !\n"); return -EAGAIN; /* try again later */ } @@ -354,13 +354,14 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw case MIXART_MOTHERBOARD_XLX_INDEX: /* xilinx already loaded ? */ - if( status_xilinx == 4 ) { - snd_printk( KERN_DEBUG "xilinx is already loaded !\n"); + if (status_xilinx == 4) { + snd_printk(KERN_DEBUG "xilinx is already loaded !\n"); return 0; } /* the status should be 0 == "idle" */ - if( status_xilinx != 0 ) { - snd_printk( KERN_ERR "xilinx load error ! status = %d\n", status_xilinx); + if (status_xilinx != 0) { + snd_printk(KERN_ERR "xilinx load error ! status = %d\n", + status_xilinx); return -EIO; /* modprob -r may help ? */ } @@ -389,21 +390,23 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw case MIXART_MOTHERBOARD_ELF_INDEX: - if( status_elf == 4 ) { - snd_printk( KERN_DEBUG "elf file already loaded !\n"); + if (status_elf == 4) { + snd_printk(KERN_DEBUG "elf file already loaded !\n"); return 0; } /* the status should be 0 == "idle" */ - if( status_elf != 0 ) { - snd_printk( KERN_ERR "elf load error ! status = %d\n", status_elf); + if (status_elf != 0) { + snd_printk(KERN_ERR "elf load error ! status = %d\n", + status_elf); return -EIO; /* modprob -r may help ? */ } /* wait for xilinx status == 4 */ err = mixart_wait_nice_for_register_value( mgr, MIXART_PSEUDOREG_MXLX_STATUS_OFFSET, 1, 4, 500); /* 5sec */ if (err < 0) { - snd_printk( KERN_ERR "xilinx was not loaded or could not be started\n"); + snd_printk(KERN_ERR "xilinx was not loaded or " + "could not be started\n"); return err; } @@ -424,7 +427,7 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw /* wait for elf status == 4 */ err = mixart_wait_nice_for_register_value( mgr, MIXART_PSEUDOREG_ELF_STATUS_OFFSET, 1, 4, 300); /* 3sec */ if (err < 0) { - snd_printk( KERN_ERR "elf could not be started\n"); + snd_printk(KERN_ERR "elf could not be started\n"); return err; } @@ -437,15 +440,16 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw default: /* elf and xilinx should be loaded */ - if( (status_elf != 4) || (status_xilinx != 4) ) { - printk( KERN_ERR "xilinx or elf not successfully loaded\n"); + if (status_elf != 4 || status_xilinx != 4) { + printk(KERN_ERR "xilinx or elf not " + "successfully loaded\n"); return -EIO; /* modprob -r may help ? */ } /* wait for daughter detection != 0 */ err = mixart_wait_nice_for_register_value( mgr, MIXART_PSEUDOREG_DBRD_PRESENCE_OFFSET, 0, 0, 30); /* 300msec */ if (err < 0) { - snd_printk( KERN_ERR "error starting elf file\n"); + snd_printk(KERN_ERR "error starting elf file\n"); return err; } @@ -460,8 +464,9 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw return -EINVAL; /* daughter should be idle */ - if( status_daught != 0 ) { - printk( KERN_ERR "daughter load error ! status = %d\n", status_daught); + if (status_daught != 0) { + printk(KERN_ERR "daughter load error ! status = %d\n", + status_daught); return -EIO; /* modprob -r may help ? */ } @@ -480,7 +485,7 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw /* wait for status == 2 */ err = mixart_wait_nice_for_register_value( mgr, MIXART_PSEUDOREG_DXLX_STATUS_OFFSET, 1, 2, 30); /* 300msec */ if (err < 0) { - snd_printk( KERN_ERR "daughter board load error\n"); + snd_printk(KERN_ERR "daughter board load error\n"); return err; } @@ -502,7 +507,8 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw /* wait for daughter status == 3 */ err = mixart_wait_nice_for_register_value( mgr, MIXART_PSEUDOREG_DXLX_STATUS_OFFSET, 1, 3, 300); /* 3sec */ if (err < 0) { - snd_printk( KERN_ERR "daughter board could not be initialised\n"); + snd_printk(KERN_ERR + "daughter board could not be initialised\n"); return err; } @@ -512,7 +518,7 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw /* first communication with embedded */ err = mixart_first_init(mgr); if (err < 0) { - snd_printk( KERN_ERR "miXart could not be set up\n"); + snd_printk(KERN_ERR "miXart could not be set up\n"); return err; } diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c index cd408b86c83..e922b1887b1 100644 --- a/sound/pci/sonicvibes.c +++ b/sound/pci/sonicvibes.c @@ -273,7 +273,8 @@ static inline void snd_sonicvibes_setdmaa(struct sonicvibes * sonic, outl(count, sonic->dmaa_port + SV_DMA_COUNT0); outb(0x18, sonic->dmaa_port + SV_DMA_MODE); #if 0 - printk("program dmaa: addr = 0x%x, paddr = 0x%x\n", addr, inl(sonic->dmaa_port + SV_DMA_ADDR0)); + printk(KERN_DEBUG "program dmaa: addr = 0x%x, paddr = 0x%x\n", + addr, inl(sonic->dmaa_port + SV_DMA_ADDR0)); #endif } @@ -288,7 +289,8 @@ static inline void snd_sonicvibes_setdmac(struct sonicvibes * sonic, outl(count, sonic->dmac_port + SV_DMA_COUNT0); outb(0x14, sonic->dmac_port + SV_DMA_MODE); #if 0 - printk("program dmac: addr = 0x%x, paddr = 0x%x\n", addr, inl(sonic->dmac_port + SV_DMA_ADDR0)); + printk(KERN_DEBUG "program dmac: addr = 0x%x, paddr = 0x%x\n", + addr, inl(sonic->dmac_port + SV_DMA_ADDR0)); #endif } @@ -355,71 +357,104 @@ static unsigned char snd_sonicvibes_in(struct sonicvibes * sonic, unsigned char #if 0 static void snd_sonicvibes_debug(struct sonicvibes * sonic) { - printk("SV REGS: INDEX = 0x%02x ", inb(SV_REG(sonic, INDEX))); + printk(KERN_DEBUG + "SV REGS: INDEX = 0x%02x ", inb(SV_REG(sonic, INDEX))); printk(" STATUS = 0x%02x\n", inb(SV_REG(sonic, STATUS))); - printk(" 0x00: left input = 0x%02x ", snd_sonicvibes_in(sonic, 0x00)); + printk(KERN_DEBUG + " 0x00: left input = 0x%02x ", snd_sonicvibes_in(sonic, 0x00)); printk(" 0x20: synth rate low = 0x%02x\n", snd_sonicvibes_in(sonic, 0x20)); - printk(" 0x01: right input = 0x%02x ", snd_sonicvibes_in(sonic, 0x01)); + printk(KERN_DEBUG + " 0x01: right input = 0x%02x ", snd_sonicvibes_in(sonic, 0x01)); printk(" 0x21: synth rate high = 0x%02x\n", snd_sonicvibes_in(sonic, 0x21)); - printk(" 0x02: left AUX1 = 0x%02x ", snd_sonicvibes_in(sonic, 0x02)); + printk(KERN_DEBUG + " 0x02: left AUX1 = 0x%02x ", snd_sonicvibes_in(sonic, 0x02)); printk(" 0x22: ADC clock = 0x%02x\n", snd_sonicvibes_in(sonic, 0x22)); - printk(" 0x03: right AUX1 = 0x%02x ", snd_sonicvibes_in(sonic, 0x03)); + printk(KERN_DEBUG + " 0x03: right AUX1 = 0x%02x ", snd_sonicvibes_in(sonic, 0x03)); printk(" 0x23: ADC alt rate = 0x%02x\n", snd_sonicvibes_in(sonic, 0x23)); - printk(" 0x04: left CD = 0x%02x ", snd_sonicvibes_in(sonic, 0x04)); + printk(KERN_DEBUG + " 0x04: left CD = 0x%02x ", snd_sonicvibes_in(sonic, 0x04)); printk(" 0x24: ADC pll M = 0x%02x\n", snd_sonicvibes_in(sonic, 0x24)); - printk(" 0x05: right CD = 0x%02x ", snd_sonicvibes_in(sonic, 0x05)); + printk(KERN_DEBUG + " 0x05: right CD = 0x%02x ", snd_sonicvibes_in(sonic, 0x05)); printk(" 0x25: ADC pll N = 0x%02x\n", snd_sonicvibes_in(sonic, 0x25)); - printk(" 0x06: left line = 0x%02x ", snd_sonicvibes_in(sonic, 0x06)); + printk(KERN_DEBUG + " 0x06: left line = 0x%02x ", snd_sonicvibes_in(sonic, 0x06)); printk(" 0x26: Synth pll M = 0x%02x\n", snd_sonicvibes_in(sonic, 0x26)); - printk(" 0x07: right line = 0x%02x ", snd_sonicvibes_in(sonic, 0x07)); + printk(KERN_DEBUG + " 0x07: right line = 0x%02x ", snd_sonicvibes_in(sonic, 0x07)); printk(" 0x27: Synth pll N = 0x%02x\n", snd_sonicvibes_in(sonic, 0x27)); - printk(" 0x08: MIC = 0x%02x ", snd_sonicvibes_in(sonic, 0x08)); + printk(KERN_DEBUG + " 0x08: MIC = 0x%02x ", snd_sonicvibes_in(sonic, 0x08)); printk(" 0x28: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x28)); - printk(" 0x09: Game port = 0x%02x ", snd_sonicvibes_in(sonic, 0x09)); + printk(KERN_DEBUG + " 0x09: Game port = 0x%02x ", snd_sonicvibes_in(sonic, 0x09)); printk(" 0x29: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x29)); - printk(" 0x0a: left synth = 0x%02x ", snd_sonicvibes_in(sonic, 0x0a)); + printk(KERN_DEBUG + " 0x0a: left synth = 0x%02x ", snd_sonicvibes_in(sonic, 0x0a)); printk(" 0x2a: MPU401 = 0x%02x\n", snd_sonicvibes_in(sonic, 0x2a)); - printk(" 0x0b: right synth = 0x%02x ", snd_sonicvibes_in(sonic, 0x0b)); + printk(KERN_DEBUG + " 0x0b: right synth = 0x%02x ", snd_sonicvibes_in(sonic, 0x0b)); printk(" 0x2b: drive ctrl = 0x%02x\n", snd_sonicvibes_in(sonic, 0x2b)); - printk(" 0x0c: left AUX2 = 0x%02x ", snd_sonicvibes_in(sonic, 0x0c)); + printk(KERN_DEBUG + " 0x0c: left AUX2 = 0x%02x ", snd_sonicvibes_in(sonic, 0x0c)); printk(" 0x2c: SRS space = 0x%02x\n", snd_sonicvibes_in(sonic, 0x2c)); - printk(" 0x0d: right AUX2 = 0x%02x ", snd_sonicvibes_in(sonic, 0x0d)); + printk(KERN_DEBUG + " 0x0d: right AUX2 = 0x%02x ", snd_sonicvibes_in(sonic, 0x0d)); printk(" 0x2d: SRS center = 0x%02x\n", snd_sonicvibes_in(sonic, 0x2d)); - printk(" 0x0e: left analog = 0x%02x ", snd_sonicvibes_in(sonic, 0x0e)); + printk(KERN_DEBUG + " 0x0e: left analog = 0x%02x ", snd_sonicvibes_in(sonic, 0x0e)); printk(" 0x2e: wave source = 0x%02x\n", snd_sonicvibes_in(sonic, 0x2e)); - printk(" 0x0f: right analog = 0x%02x ", snd_sonicvibes_in(sonic, 0x0f)); + printk(KERN_DEBUG + " 0x0f: right analog = 0x%02x ", snd_sonicvibes_in(sonic, 0x0f)); printk(" 0x2f: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x2f)); - printk(" 0x10: left PCM = 0x%02x ", snd_sonicvibes_in(sonic, 0x10)); + printk(KERN_DEBUG + " 0x10: left PCM = 0x%02x ", snd_sonicvibes_in(sonic, 0x10)); printk(" 0x30: analog power = 0x%02x\n", snd_sonicvibes_in(sonic, 0x30)); - printk(" 0x11: right PCM = 0x%02x ", snd_sonicvibes_in(sonic, 0x11)); + printk(KERN_DEBUG + " 0x11: right PCM = 0x%02x ", snd_sonicvibes_in(sonic, 0x11)); printk(" 0x31: analog power = 0x%02x\n", snd_sonicvibes_in(sonic, 0x31)); - printk(" 0x12: DMA data format = 0x%02x ", snd_sonicvibes_in(sonic, 0x12)); + printk(KERN_DEBUG + " 0x12: DMA data format = 0x%02x ", snd_sonicvibes_in(sonic, 0x12)); printk(" 0x32: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x32)); - printk(" 0x13: P/C enable = 0x%02x ", snd_sonicvibes_in(sonic, 0x13)); + printk(KERN_DEBUG + " 0x13: P/C enable = 0x%02x ", snd_sonicvibes_in(sonic, 0x13)); printk(" 0x33: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x33)); - printk(" 0x14: U/D button = 0x%02x ", snd_sonicvibes_in(sonic, 0x14)); + printk(KERN_DEBUG + " 0x14: U/D button = 0x%02x ", snd_sonicvibes_in(sonic, 0x14)); printk(" 0x34: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x34)); - printk(" 0x15: revision = 0x%02x ", snd_sonicvibes_in(sonic, 0x15)); + printk(KERN_DEBUG + " 0x15: revision = 0x%02x ", snd_sonicvibes_in(sonic, 0x15)); printk(" 0x35: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x35)); - printk(" 0x16: ADC output ctrl = 0x%02x ", snd_sonicvibes_in(sonic, 0x16)); + printk(KERN_DEBUG + " 0x16: ADC output ctrl = 0x%02x ", snd_sonicvibes_in(sonic, 0x16)); printk(" 0x36: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x36)); - printk(" 0x17: --- = 0x%02x ", snd_sonicvibes_in(sonic, 0x17)); + printk(KERN_DEBUG + " 0x17: --- = 0x%02x ", snd_sonicvibes_in(sonic, 0x17)); printk(" 0x37: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x37)); - printk(" 0x18: DMA A upper cnt = 0x%02x ", snd_sonicvibes_in(sonic, 0x18)); + printk(KERN_DEBUG + " 0x18: DMA A upper cnt = 0x%02x ", snd_sonicvibes_in(sonic, 0x18)); printk(" 0x38: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x38)); - printk(" 0x19: DMA A lower cnt = 0x%02x ", snd_sonicvibes_in(sonic, 0x19)); + printk(KERN_DEBUG + " 0x19: DMA A lower cnt = 0x%02x ", snd_sonicvibes_in(sonic, 0x19)); printk(" 0x39: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x39)); - printk(" 0x1a: --- = 0x%02x ", snd_sonicvibes_in(sonic, 0x1a)); + printk(KERN_DEBUG + " 0x1a: --- = 0x%02x ", snd_sonicvibes_in(sonic, 0x1a)); printk(" 0x3a: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x3a)); - printk(" 0x1b: --- = 0x%02x ", snd_sonicvibes_in(sonic, 0x1b)); + printk(KERN_DEBUG + " 0x1b: --- = 0x%02x ", snd_sonicvibes_in(sonic, 0x1b)); printk(" 0x3b: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x3b)); - printk(" 0x1c: DMA C upper cnt = 0x%02x ", snd_sonicvibes_in(sonic, 0x1c)); + printk(KERN_DEBUG + " 0x1c: DMA C upper cnt = 0x%02x ", snd_sonicvibes_in(sonic, 0x1c)); printk(" 0x3c: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x3c)); - printk(" 0x1d: DMA C upper cnt = 0x%02x ", snd_sonicvibes_in(sonic, 0x1d)); + printk(KERN_DEBUG + " 0x1d: DMA C upper cnt = 0x%02x ", snd_sonicvibes_in(sonic, 0x1d)); printk(" 0x3d: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x3d)); - printk(" 0x1e: PCM rate low = 0x%02x ", snd_sonicvibes_in(sonic, 0x1e)); + printk(KERN_DEBUG + " 0x1e: PCM rate low = 0x%02x ", snd_sonicvibes_in(sonic, 0x1e)); printk(" 0x3e: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x3e)); - printk(" 0x1f: PCM rate high = 0x%02x ", snd_sonicvibes_in(sonic, 0x1f)); + printk(KERN_DEBUG + " 0x1f: PCM rate high = 0x%02x ", snd_sonicvibes_in(sonic, 0x1f)); printk(" 0x3f: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x3f)); } @@ -476,8 +511,8 @@ static void snd_sonicvibes_pll(unsigned int rate, *res_m = m; *res_n = n; #if 0 - printk("metric = %i, xm = %i, xn = %i\n", metric, xm, xn); - printk("pll: m = 0x%x, r = 0x%x, n = 0x%x\n", reg, m, r, n); + printk(KERN_DEBUG "metric = %i, xm = %i, xn = %i\n", metric, xm, xn); + printk(KERN_DEBUG "pll: m = 0x%x, r = 0x%x, n = 0x%x\n", reg, m, r, n); #endif } diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c index c612b435ca2..a9da9c18466 100644 --- a/sound/pci/trident/trident_main.c +++ b/sound/pci/trident/trident_main.c @@ -68,40 +68,40 @@ static void snd_trident_print_voice_regs(struct snd_trident *trident, int voice) { unsigned int val, tmp; - printk("Trident voice %i:\n", voice); + printk(KERN_DEBUG "Trident voice %i:\n", voice); outb(voice, TRID_REG(trident, T4D_LFO_GC_CIR)); val = inl(TRID_REG(trident, CH_LBA)); - printk("LBA: 0x%x\n", val); + printk(KERN_DEBUG "LBA: 0x%x\n", val); val = inl(TRID_REG(trident, CH_GVSEL_PAN_VOL_CTRL_EC)); - printk("GVSel: %i\n", val >> 31); - printk("Pan: 0x%x\n", (val >> 24) & 0x7f); - printk("Vol: 0x%x\n", (val >> 16) & 0xff); - printk("CTRL: 0x%x\n", (val >> 12) & 0x0f); - printk("EC: 0x%x\n", val & 0x0fff); + printk(KERN_DEBUG "GVSel: %i\n", val >> 31); + printk(KERN_DEBUG "Pan: 0x%x\n", (val >> 24) & 0x7f); + printk(KERN_DEBUG "Vol: 0x%x\n", (val >> 16) & 0xff); + printk(KERN_DEBUG "CTRL: 0x%x\n", (val >> 12) & 0x0f); + printk(KERN_DEBUG "EC: 0x%x\n", val & 0x0fff); if (trident->device != TRIDENT_DEVICE_ID_NX) { val = inl(TRID_REG(trident, CH_DX_CSO_ALPHA_FMS)); - printk("CSO: 0x%x\n", val >> 16); + printk(KERN_DEBUG "CSO: 0x%x\n", val >> 16); printk("Alpha: 0x%x\n", (val >> 4) & 0x0fff); - printk("FMS: 0x%x\n", val & 0x0f); + printk(KERN_DEBUG "FMS: 0x%x\n", val & 0x0f); val = inl(TRID_REG(trident, CH_DX_ESO_DELTA)); - printk("ESO: 0x%x\n", val >> 16); - printk("Delta: 0x%x\n", val & 0xffff); + printk(KERN_DEBUG "ESO: 0x%x\n", val >> 16); + printk(KERN_DEBUG "Delta: 0x%x\n", val & 0xffff); val = inl(TRID_REG(trident, CH_DX_FMC_RVOL_CVOL)); } else { // TRIDENT_DEVICE_ID_NX val = inl(TRID_REG(trident, CH_NX_DELTA_CSO)); tmp = (val >> 24) & 0xff; - printk("CSO: 0x%x\n", val & 0x00ffffff); + printk(KERN_DEBUG "CSO: 0x%x\n", val & 0x00ffffff); val = inl(TRID_REG(trident, CH_NX_DELTA_ESO)); tmp |= (val >> 16) & 0xff00; - printk("Delta: 0x%x\n", tmp); - printk("ESO: 0x%x\n", val & 0x00ffffff); + printk(KERN_DEBUG "Delta: 0x%x\n", tmp); + printk(KERN_DEBUG "ESO: 0x%x\n", val & 0x00ffffff); val = inl(TRID_REG(trident, CH_NX_ALPHA_FMS_FMC_RVOL_CVOL)); - printk("Alpha: 0x%x\n", val >> 20); - printk("FMS: 0x%x\n", (val >> 16) & 0x0f); + printk(KERN_DEBUG "Alpha: 0x%x\n", val >> 20); + printk(KERN_DEBUG "FMS: 0x%x\n", (val >> 16) & 0x0f); } - printk("FMC: 0x%x\n", (val >> 14) & 3); - printk("RVol: 0x%x\n", (val >> 7) & 0x7f); - printk("CVol: 0x%x\n", val & 0x7f); + printk(KERN_DEBUG "FMC: 0x%x\n", (val >> 14) & 3); + printk(KERN_DEBUG "RVol: 0x%x\n", (val >> 7) & 0x7f); + printk(KERN_DEBUG "CVol: 0x%x\n", val & 0x7f); } #endif @@ -496,12 +496,17 @@ void snd_trident_write_voice_regs(struct snd_trident * trident, outl(regs[4], TRID_REG(trident, CH_START + 16)); #if 0 - printk("written %i channel:\n", voice->number); - printk(" regs[0] = 0x%x/0x%x\n", regs[0], inl(TRID_REG(trident, CH_START + 0))); - printk(" regs[1] = 0x%x/0x%x\n", regs[1], inl(TRID_REG(trident, CH_START + 4))); - printk(" regs[2] = 0x%x/0x%x\n", regs[2], inl(TRID_REG(trident, CH_START + 8))); - printk(" regs[3] = 0x%x/0x%x\n", regs[3], inl(TRID_REG(trident, CH_START + 12))); - printk(" regs[4] = 0x%x/0x%x\n", regs[4], inl(TRID_REG(trident, CH_START + 16))); + printk(KERN_DEBUG "written %i channel:\n", voice->number); + printk(KERN_DEBUG " regs[0] = 0x%x/0x%x\n", + regs[0], inl(TRID_REG(trident, CH_START + 0))); + printk(KERN_DEBUG " regs[1] = 0x%x/0x%x\n", + regs[1], inl(TRID_REG(trident, CH_START + 4))); + printk(KERN_DEBUG " regs[2] = 0x%x/0x%x\n", + regs[2], inl(TRID_REG(trident, CH_START + 8))); + printk(KERN_DEBUG " regs[3] = 0x%x/0x%x\n", + regs[3], inl(TRID_REG(trident, CH_START + 12))); + printk(KERN_DEBUG " regs[4] = 0x%x/0x%x\n", + regs[4], inl(TRID_REG(trident, CH_START + 16))); #endif } @@ -583,7 +588,7 @@ static void snd_trident_write_vol_reg(struct snd_trident * trident, outb(voice->Vol >> 2, TRID_REG(trident, CH_GVSEL_PAN_VOL_CTRL_EC + 2)); break; case TRIDENT_DEVICE_ID_SI7018: - // printk("voice->Vol = 0x%x\n", voice->Vol); + /* printk(KERN_DEBUG "voice->Vol = 0x%x\n", voice->Vol); */ outw((voice->CTRL << 12) | voice->Vol, TRID_REG(trident, CH_GVSEL_PAN_VOL_CTRL_EC)); break; diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 1aafe956ee2..fc62d6380f8 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -466,7 +466,10 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre flag = VIA_TBL_BIT_FLAG; /* period boundary */ } else flag = 0; /* period continues to the next */ - // printk("via: tbl %d: at %d size %d (rest %d)\n", idx, ofs, r, rest); + /* + printk(KERN_DEBUG "via: tbl %d: at %d size %d " + "(rest %d)\n", idx, ofs, r, rest); + */ ((u32 *)dev->table.area)[(idx<<1) + 1] = cpu_to_le32(r | flag); dev->idx_table[idx].offset = ofs; dev->idx_table[idx].size = r; diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c index 5bd79d2a5a1..c0d9cc9dad4 100644 --- a/sound/pci/via82xx_modem.c +++ b/sound/pci/via82xx_modem.c @@ -328,7 +328,10 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre flag = VIA_TBL_BIT_FLAG; /* period boundary */ } else flag = 0; /* period continues to the next */ - // printk("via: tbl %d: at %d size %d (rest %d)\n", idx, ofs, r, rest); + /* + printk(KERN_DEBUG "via: tbl %d: at %d size %d " + "(rest %d)\n", idx, ofs, r, rest); + */ ((u32 *)dev->table.area)[(idx<<1) + 1] = cpu_to_le32(r | flag); dev->idx_table[idx].offset = ofs; dev->idx_table[idx].size = r; diff --git a/sound/pci/vx222/vx222_ops.c b/sound/pci/vx222/vx222_ops.c index 7e87f398ff0..c0efe449111 100644 --- a/sound/pci/vx222/vx222_ops.c +++ b/sound/pci/vx222/vx222_ops.c @@ -107,7 +107,9 @@ static unsigned char vx2_inb(struct vx_core *chip, int offset) static void vx2_outb(struct vx_core *chip, int offset, unsigned char val) { outb(val, vx2_reg_addr(chip, offset)); - //printk("outb: %x -> %x\n", val, vx2_reg_addr(chip, offset)); + /* + printk(KERN_DEBUG "outb: %x -> %x\n", val, vx2_reg_addr(chip, offset)); + */ } /** @@ -126,7 +128,9 @@ static unsigned int vx2_inl(struct vx_core *chip, int offset) */ static void vx2_outl(struct vx_core *chip, int offset, unsigned int val) { - // printk("outl: %x -> %x\n", val, vx2_reg_addr(chip, offset)); + /* + printk(KERN_DEBUG "outl: %x -> %x\n", val, vx2_reg_addr(chip, offset)); + */ outl(val, vx2_reg_addr(chip, offset)); } diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c index 90d0d62bd0b..2f0925236a1 100644 --- a/sound/pci/ymfpci/ymfpci_main.c +++ b/sound/pci/ymfpci/ymfpci_main.c @@ -318,7 +318,12 @@ static void snd_ymfpci_pcm_interrupt(struct snd_ymfpci *chip, struct snd_ymfpci_ ypcm->period_pos += delta; ypcm->last_pos = pos; if (ypcm->period_pos >= ypcm->period_size) { - // printk("done - active_bank = 0x%x, start = 0x%x\n", chip->active_bank, voice->bank[chip->active_bank].start); + /* + printk(KERN_DEBUG + "done - active_bank = 0x%x, start = 0x%x\n", + chip->active_bank, + voice->bank[chip->active_bank].start); + */ ypcm->period_pos %= ypcm->period_size; spin_unlock(&chip->reg_lock); snd_pcm_period_elapsed(ypcm->substream); @@ -366,7 +371,12 @@ static void snd_ymfpci_pcm_capture_interrupt(struct snd_pcm_substream *substream ypcm->last_pos = pos; if (ypcm->period_pos >= ypcm->period_size) { ypcm->period_pos %= ypcm->period_size; - // printk("done - active_bank = 0x%x, start = 0x%x\n", chip->active_bank, voice->bank[chip->active_bank].start); + /* + printk(KERN_DEBUG + "done - active_bank = 0x%x, start = 0x%x\n", + chip->active_bank, + voice->bank[chip->active_bank].start); + */ spin_unlock(&chip->reg_lock); snd_pcm_period_elapsed(substream); spin_lock(&chip->reg_lock); -- cgit v1.2.3-70-g09d2 From 2ebfb8eeb8f244f9d25937d31a947895cf819e26 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 5 Feb 2009 16:11:58 +0100 Subject: ALSA: Add missing KERN_* prefix to printk in other sound/* Signed-off-by: Takashi Iwai --- sound/arm/sa11xx-uda1341.c | 2 +- sound/mips/au1x00.c | 2 +- sound/pcmcia/pdaudiocf/pdaudiocf_core.c | 23 +++++++++++++++-------- sound/pcmcia/pdaudiocf/pdaudiocf_irq.c | 4 ++-- sound/sparc/amd7930.c | 5 +++-- 5 files changed, 22 insertions(+), 14 deletions(-) diff --git a/sound/arm/sa11xx-uda1341.c b/sound/arm/sa11xx-uda1341.c index 1dcd51d81d1..ed481a866a3 100644 --- a/sound/arm/sa11xx-uda1341.c +++ b/sound/arm/sa11xx-uda1341.c @@ -914,7 +914,7 @@ static int __devinit sa11xx_uda1341_probe(struct platform_device *devptr) snd_card_set_dev(card, &devptr->dev); if ((err = snd_card_register(card)) == 0) { - printk( KERN_INFO "iPAQ audio support initialized\n" ); + printk(KERN_INFO "iPAQ audio support initialized\n"); platform_set_drvdata(devptr, card); return 0; } diff --git a/sound/mips/au1x00.c b/sound/mips/au1x00.c index 1881cec11e7..7c1afc96ab8 100644 --- a/sound/mips/au1x00.c +++ b/sound/mips/au1x00.c @@ -678,7 +678,7 @@ au1000_init(void) return err; } - printk( KERN_INFO "ALSA AC97: Driver Initialized\n" ); + printk(KERN_INFO "ALSA AC97: Driver Initialized\n"); au1000_card = card; return 0; } diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_core.c b/sound/pcmcia/pdaudiocf/pdaudiocf_core.c index dfa40b0ed86..5d2afa0b0ce 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf_core.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf_core.c @@ -82,14 +82,21 @@ static void pdacf_ak4117_write(void *private_data, unsigned char reg, unsigned c #if 0 void pdacf_dump(struct snd_pdacf *chip) { - printk("PDAUDIOCF DUMP (0x%lx):\n", chip->port); - printk("WPD : 0x%x\n", inw(chip->port + PDAUDIOCF_REG_WDP)); - printk("RDP : 0x%x\n", inw(chip->port + PDAUDIOCF_REG_RDP)); - printk("TCR : 0x%x\n", inw(chip->port + PDAUDIOCF_REG_TCR)); - printk("SCR : 0x%x\n", inw(chip->port + PDAUDIOCF_REG_SCR)); - printk("ISR : 0x%x\n", inw(chip->port + PDAUDIOCF_REG_ISR)); - printk("IER : 0x%x\n", inw(chip->port + PDAUDIOCF_REG_IER)); - printk("AK_IFR : 0x%x\n", inw(chip->port + PDAUDIOCF_REG_AK_IFR)); + printk(KERN_DEBUG "PDAUDIOCF DUMP (0x%lx):\n", chip->port); + printk(KERN_DEBUG "WPD : 0x%x\n", + inw(chip->port + PDAUDIOCF_REG_WDP)); + printk(KERN_DEBUG "RDP : 0x%x\n", + inw(chip->port + PDAUDIOCF_REG_RDP)); + printk(KERN_DEBUG "TCR : 0x%x\n", + inw(chip->port + PDAUDIOCF_REG_TCR)); + printk(KERN_DEBUG "SCR : 0x%x\n", + inw(chip->port + PDAUDIOCF_REG_SCR)); + printk(KERN_DEBUG "ISR : 0x%x\n", + inw(chip->port + PDAUDIOCF_REG_ISR)); + printk(KERN_DEBUG "IER : 0x%x\n", + inw(chip->port + PDAUDIOCF_REG_IER)); + printk(KERN_DEBUG "AK_IFR : 0x%x\n", + inw(chip->port + PDAUDIOCF_REG_AK_IFR)); } #endif diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_irq.c b/sound/pcmcia/pdaudiocf/pdaudiocf_irq.c index ea903c8e90d..dcd32201bc8 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf_irq.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf_irq.c @@ -269,7 +269,7 @@ void pdacf_tasklet(unsigned long private_data) rdp = inw(chip->port + PDAUDIOCF_REG_RDP); wdp = inw(chip->port + PDAUDIOCF_REG_WDP); - // printk("TASKLET: rdp = %x, wdp = %x\n", rdp, wdp); + /* printk(KERN_DEBUG "TASKLET: rdp = %x, wdp = %x\n", rdp, wdp); */ size = wdp - rdp; if (size < 0) size += 0x10000; @@ -321,5 +321,5 @@ void pdacf_tasklet(unsigned long private_data) spin_lock(&chip->reg_lock); } spin_unlock(&chip->reg_lock); - // printk("TASKLET: end\n"); + /* printk(KERN_DEBUG "TASKLET: end\n"); */ } diff --git a/sound/sparc/amd7930.c b/sound/sparc/amd7930.c index f87933e4881..7cbc725934e 100644 --- a/sound/sparc/amd7930.c +++ b/sound/sparc/amd7930.c @@ -954,7 +954,8 @@ static int __devinit snd_amd7930_create(struct snd_card *card, amd->regs = of_ioremap(&op->resource[0], 0, resource_size(&op->resource[0]), "amd7930"); if (!amd->regs) { - snd_printk("amd7930-%d: Unable to map chip registers.\n", dev); + snd_printk(KERN_ERR + "amd7930-%d: Unable to map chip registers.\n", dev); return -EIO; } @@ -962,7 +963,7 @@ static int __devinit snd_amd7930_create(struct snd_card *card, if (request_irq(irq, snd_amd7930_interrupt, IRQF_DISABLED | IRQF_SHARED, "amd7930", amd)) { - snd_printk("amd7930-%d: Unable to grab IRQ %d\n", + snd_printk(KERN_ERR "amd7930-%d: Unable to grab IRQ %d\n", dev, irq); snd_amd7930_free(amd); return -EBUSY; -- cgit v1.2.3-70-g09d2 From dd542f169aaa35f4ac0d063e04b41c648a93887c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 5 Feb 2009 16:15:39 +0100 Subject: ALSA: ca0106 - Add missing KERN_* prefix to printk Signed-off-by: Takashi Iwai --- sound/pci/ca0106/ca0106_main.c | 91 ++++++++++++++++++++++++++++++++---------- 1 file changed, 71 insertions(+), 20 deletions(-) diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index 3aac7e6489c..dac8a5f040e 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -412,7 +412,9 @@ int snd_ca0106_i2c_write(struct snd_ca0106 *emu, } tmp = reg << 25 | value << 16; - // snd_printk("I2C-write:reg=0x%x, value=0x%x\n", reg, value); + /* + snd_printk(KERN_DEBUG "I2C-write:reg=0x%x, value=0x%x\n", reg, value); + */ /* Not sure what this I2C channel controls. */ /* snd_ca0106_ptr_write(emu, I2C_D0, 0, tmp); */ @@ -430,7 +432,7 @@ int snd_ca0106_i2c_write(struct snd_ca0106 *emu, /* Wait till the transaction ends */ while (1) { status = snd_ca0106_ptr_read(emu, I2C_A, 0); - //snd_printk("I2C:status=0x%x\n", status); + /*snd_printk(KERN_DEBUG "I2C:status=0x%x\n", status);*/ timeout++; if ((status & I2C_A_ADC_START) == 0) break; @@ -529,7 +531,10 @@ static int snd_ca0106_pcm_open_playback_channel(struct snd_pcm_substream *substr channel->number = channel_id; channel->use = 1; - //printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel); + /* + printk(KERN_DEBUG "open:channel_id=%d, chip=%p, channel=%p\n", + channel_id, chip, channel); + */ //channel->interrupt = snd_ca0106_pcm_channel_interrupt; channel->epcm = epcm; if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) @@ -622,7 +627,10 @@ static int snd_ca0106_pcm_open_capture_channel(struct snd_pcm_substream *substre channel->number = channel_id; channel->use = 1; - //printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel); + /* + printk(KERN_DEBUG "open:channel_id=%d, chip=%p, channel=%p\n", + channel_id, chip, channel); + */ //channel->interrupt = snd_ca0106_pcm_channel_interrupt; channel->epcm = epcm; if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) @@ -713,9 +721,20 @@ static int snd_ca0106_pcm_prepare_playback(struct snd_pcm_substream *substream) u32 reg71; int i; - //snd_printk("prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, periods=%u, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, runtime->periods, frames_to_bytes(runtime, 1)); - //snd_printk("dma_addr=%x, dma_area=%p, table_base=%p\n",runtime->dma_addr, runtime->dma_area, table_base); - //snd_printk("dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n",emu->buffer.addr, emu->buffer.area, emu->buffer.bytes); +#if 0 /* debug */ + snd_printk(KERN_DEBUG + "prepare:channel_number=%d, rate=%d, format=0x%x, " + "channels=%d, buffer_size=%ld, period_size=%ld, " + "periods=%u, frames_to_bytes=%d\n", + channel, runtime->rate, runtime->format, + runtime->channels, runtime->buffer_size, + runtime->period_size, runtime->periods, + frames_to_bytes(runtime, 1)); + snd_printk(KERN_DEBUG "dma_addr=%x, dma_area=%p, table_base=%p\n", + runtime->dma_addr, runtime->dma_area, table_base); + snd_printk(KERN_DEBUG "dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n", + emu->buffer.addr, emu->buffer.area, emu->buffer.bytes); +#endif /* debug */ /* Rate can be set per channel. */ /* reg40 control host to fifo */ /* reg71 controls DAC rate. */ @@ -807,9 +826,20 @@ static int snd_ca0106_pcm_prepare_capture(struct snd_pcm_substream *substream) u32 reg71_set = 0; u32 reg71; - //snd_printk("prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, periods=%u, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, runtime->periods, frames_to_bytes(runtime, 1)); - //snd_printk("dma_addr=%x, dma_area=%p, table_base=%p\n",runtime->dma_addr, runtime->dma_area, table_base); - //snd_printk("dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n",emu->buffer.addr, emu->buffer.area, emu->buffer.bytes); +#if 0 /* debug */ + snd_printk(KERN_DEBUG + "prepare:channel_number=%d, rate=%d, format=0x%x, " + "channels=%d, buffer_size=%ld, period_size=%ld, " + "periods=%u, frames_to_bytes=%d\n", + channel, runtime->rate, runtime->format, + runtime->channels, runtime->buffer_size, + runtime->period_size, runtime->periods, + frames_to_bytes(runtime, 1)); + snd_printk(KERN_DEBUG "dma_addr=%x, dma_area=%p, table_base=%p\n", + runtime->dma_addr, runtime->dma_area, table_base); + snd_printk(KERN_DEBUG "dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n", + emu->buffer.addr, emu->buffer.area, emu->buffer.bytes); +#endif /* debug */ /* reg71 controls ADC rate. */ switch (runtime->rate) { case 44100: @@ -854,7 +884,14 @@ static int snd_ca0106_pcm_prepare_capture(struct snd_pcm_substream *substream) } - //printk("prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, frames_to_bytes(runtime, 1)); + /* + printk(KERN_DEBUG + "prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, " + "buffer_size=%ld, period_size=%ld, frames_to_bytes=%d\n", + channel, runtime->rate, runtime->format, runtime->channels, + runtime->buffer_size, runtime->period_size, + frames_to_bytes(runtime, 1)); + */ snd_ca0106_ptr_write(emu, 0x13, channel, 0); snd_ca0106_ptr_write(emu, CAPTURE_DMA_ADDR, channel, runtime->dma_addr); snd_ca0106_ptr_write(emu, CAPTURE_BUFFER_SIZE, channel, frames_to_bytes(runtime, runtime->buffer_size)<<16); // buffer size in bytes @@ -896,13 +933,13 @@ static int snd_ca0106_pcm_trigger_playback(struct snd_pcm_substream *substream, runtime = s->runtime; epcm = runtime->private_data; channel = epcm->channel_id; - /* snd_printk("channel=%d\n",channel); */ + /* snd_printk(KERN_DEBUG "channel=%d\n", channel); */ epcm->running = running; basic |= (0x1 << channel); extended |= (0x10 << channel); snd_pcm_trigger_done(s, substream); } - /* snd_printk("basic=0x%x, extended=0x%x\n",basic, extended); */ + /* snd_printk(KERN_DEBUG "basic=0x%x, extended=0x%x\n",basic, extended); */ switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -980,8 +1017,13 @@ snd_ca0106_pcm_pointer_playback(struct snd_pcm_substream *substream) ptr=ptr2; if (ptr >= runtime->buffer_size) ptr -= runtime->buffer_size; - //printk("ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n", ptr1, ptr2, ptr, (int)runtime->buffer_size, (int)runtime->period_size, (int)runtime->frame_bits, (int)runtime->rate); - + /* + printk(KERN_DEBUG "ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, " + "buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n", + ptr1, ptr2, ptr, (int)runtime->buffer_size, + (int)runtime->period_size, (int)runtime->frame_bits, + (int)runtime->rate); + */ return ptr; } @@ -1003,8 +1045,13 @@ snd_ca0106_pcm_pointer_capture(struct snd_pcm_substream *substream) ptr=ptr2; if (ptr >= runtime->buffer_size) ptr -= runtime->buffer_size; - //printk("ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n", ptr1, ptr2, ptr, (int)runtime->buffer_size, (int)runtime->period_size, (int)runtime->frame_bits, (int)runtime->rate); - + /* + printk(KERN_DEBUG "ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, " + "buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n", + ptr1, ptr2, ptr, (int)runtime->buffer_size, + (int)runtime->period_size, (int)runtime->frame_bits, + (int)runtime->rate); + */ return ptr; } @@ -1189,8 +1236,12 @@ static irqreturn_t snd_ca0106_interrupt(int irq, void *dev_id) return IRQ_NONE; stat76 = snd_ca0106_ptr_read(chip, EXTENDED_INT, 0); - //snd_printk("interrupt status = 0x%08x, stat76=0x%08x\n", status, stat76); - //snd_printk("ptr=0x%08x\n",snd_ca0106_ptr_read(chip, PLAYBACK_POINTER, 0)); + /* + snd_printk(KERN_DEBUG "interrupt status = 0x%08x, stat76=0x%08x\n", + status, stat76); + snd_printk(KERN_DEBUG "ptr=0x%08x\n", + snd_ca0106_ptr_read(chip, PLAYBACK_POINTER, 0)); + */ mask = 0x11; /* 0x1 for one half, 0x10 for the other half period. */ for(i = 0; i < 4; i++) { pchannel = &(chip->playback_channels[i]); @@ -1478,7 +1529,7 @@ static void ca0106_init_chip(struct snd_ca0106 *chip, int resume) int size, n; size = ARRAY_SIZE(i2c_adc_init); - /* snd_printk("I2C:array size=0x%x\n", size); */ + /* snd_printk(KERN_DEBUG "I2C:array size=0x%x\n", size); */ for (n = 0; n < size; n++) snd_ca0106_i2c_write(chip, i2c_adc_init[n][0], i2c_adc_init[n][1]); -- cgit v1.2.3-70-g09d2 From b25c9da19889e33bb4ee2dff369fc46caa4543b0 Mon Sep 17 00:00:00 2001 From: Wu Fengguang Date: Fri, 6 Feb 2009 15:02:27 +0800 Subject: ALSA: enable concurrent digital outputs for ALC1200 Add the SPDIF pin as slave digital out to enable concurrent HDMI/SPDIF outputs for ASUS M3A-H/HDMI with ALC1200 codec. Tested-by: Thomas Schneider Signed-off-by: Wu Fengguang Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_local.h | 1 + sound/pci/hda/patch_realtek.c | 8 ++++++++ 2 files changed, 9 insertions(+) diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index ec687b206c0..4086491ed33 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -229,6 +229,7 @@ struct hda_multi_out { hda_nid_t hp_nid; /* optional DAC for HP, 0 when not exists */ hda_nid_t extra_out_nid[3]; /* optional DACs, 0 when not exists */ hda_nid_t dig_out_nid; /* digital out audio widget */ + hda_nid_t *slave_dig_outs; int max_channels; /* currently supported analog channels */ int dig_out_used; /* current usage of digital out (HDA_DIG_XXX) */ int no_share_stream; /* don't share a stream with multiple pins */ diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d2812ab729c..5194a58fafa 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -349,6 +349,7 @@ struct alc_config_preset { hda_nid_t *dac_nids; hda_nid_t dig_out_nid; /* optional */ hda_nid_t hp_nid; /* optional */ + hda_nid_t *slave_dig_outs; unsigned int num_adc_nids; hda_nid_t *adc_nids; hda_nid_t *capsrc_nids; @@ -824,6 +825,7 @@ static void setup_preset(struct alc_spec *spec, spec->multiout.num_dacs = preset->num_dacs; spec->multiout.dac_nids = preset->dac_nids; spec->multiout.dig_out_nid = preset->dig_out_nid; + spec->multiout.slave_dig_outs = preset->slave_dig_outs; spec->multiout.hp_nid = preset->hp_nid; spec->num_mux_defs = preset->num_mux_defs; @@ -3107,6 +3109,7 @@ static int alc_build_pcms(struct hda_codec *codec) /* SPDIF for stream index #1 */ if (spec->multiout.dig_out_nid || spec->dig_in_nid) { codec->num_pcms = 2; + codec->slave_dig_outs = spec->multiout.slave_dig_outs; info = spec->pcm_rec + 1; info->name = spec->stream_name_digital; if (spec->dig_out_type) @@ -8603,6 +8606,10 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { {} }; +static hda_nid_t alc1200_slave_dig_outs[] = { + ALC883_DIGOUT_NID, 0, +}; + static struct alc_config_preset alc883_presets[] = { [ALC883_3ST_2ch_DIG] = { .mixers = { alc883_3ST_2ch_mixer }, @@ -8943,6 +8950,7 @@ static struct alc_config_preset alc883_presets[] = { .dac_nids = alc883_dac_nids, .dig_out_nid = ALC1200_DIGOUT_NID, .dig_in_nid = ALC883_DIGIN_NID, + .slave_dig_outs = alc1200_slave_dig_outs, .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes), .channel_mode = alc883_sixstack_modes, .input_mux = &alc883_capture_source, -- cgit v1.2.3-70-g09d2 From 92a950ff2b7091a735c32a6e57b8136650bc7812 Mon Sep 17 00:00:00 2001 From: Mike Frysinger Date: Fri, 6 Feb 2009 18:12:34 +0800 Subject: ASoC: Blackfin: cleanup sport handling in ASoC Blackfin AC97 code - make sport number handling more dynamic as not all Blackfins have a linear sport map starting at 0 - indexes can be macroed away too Signed-off-by: Mike Frysinger Signed-off-by: Cliff Cai Signed-off-by: Bryan Wu Signed-off-by: Mark Brown --- sound/soc/blackfin/bf5xx-ac97.c | 92 +++++++++++++++-------------------------- 1 file changed, 33 insertions(+), 59 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c index 3be2be60576..5885702c78f 100644 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ b/sound/soc/blackfin/bf5xx-ac97.c @@ -31,72 +31,46 @@ #include "bf5xx-sport.h" #include "bf5xx-ac97.h" -#if defined(CONFIG_BF54x) -#define PIN_REQ_SPORT_0 {P_SPORT0_TFS, P_SPORT0_DTPRI, P_SPORT0_TSCLK, \ - P_SPORT0_RFS, P_SPORT0_DRPRI, P_SPORT0_RSCLK, 0} - -#define PIN_REQ_SPORT_1 {P_SPORT1_TFS, P_SPORT1_DTPRI, P_SPORT1_TSCLK, \ - P_SPORT1_RFS, P_SPORT1_DRPRI, P_SPORT1_RSCLK, 0} - -#define PIN_REQ_SPORT_2 {P_SPORT2_TFS, P_SPORT2_DTPRI, P_SPORT2_TSCLK, \ - P_SPORT2_RFS, P_SPORT2_DRPRI, P_SPORT2_RSCLK, 0} - -#define PIN_REQ_SPORT_3 {P_SPORT3_TFS, P_SPORT3_DTPRI, P_SPORT3_TSCLK, \ - P_SPORT3_RFS, P_SPORT3_DRPRI, P_SPORT3_RSCLK, 0} -#else -#define PIN_REQ_SPORT_0 {P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS, \ - P_SPORT0_DRPRI, P_SPORT0_RSCLK, 0} - -#define PIN_REQ_SPORT_1 {P_SPORT1_DTPRI, P_SPORT1_TSCLK, P_SPORT1_RFS, \ - P_SPORT1_DRPRI, P_SPORT1_RSCLK, 0} -#endif - static int *cmd_count; static int sport_num = CONFIG_SND_BF5XX_SPORT_NUM; +#define SPORT_REQ(x) \ + [x] = {P_SPORT##x##_TFS, P_SPORT##x##_DTPRI, P_SPORT##x##_TSCLK, \ + P_SPORT##x##_RFS, P_SPORT##x##_DRPRI, P_SPORT##x##_RSCLK, 0} static u16 sport_req[][7] = { - PIN_REQ_SPORT_0, -#ifdef PIN_REQ_SPORT_1 - PIN_REQ_SPORT_1, +#ifdef SPORT0_TCR1 + SPORT_REQ(0), #endif -#ifdef PIN_REQ_SPORT_2 - PIN_REQ_SPORT_2, +#ifdef SPORT1_TCR1 + SPORT_REQ(1), #endif -#ifdef PIN_REQ_SPORT_3 - PIN_REQ_SPORT_3, +#ifdef SPORT2_TCR1 + SPORT_REQ(2), #endif - }; +#ifdef SPORT3_TCR1 + SPORT_REQ(3), +#endif +}; +#define SPORT_PARAMS(x) \ + [x] = { \ + .dma_rx_chan = CH_SPORT##x##_RX, \ + .dma_tx_chan = CH_SPORT##x##_TX, \ + .err_irq = IRQ_SPORT##x##_ERROR, \ + .regs = (struct sport_register *)SPORT##x##_TCR1, \ + } static struct sport_param sport_params[4] = { - { - .dma_rx_chan = CH_SPORT0_RX, - .dma_tx_chan = CH_SPORT0_TX, - .err_irq = IRQ_SPORT0_ERROR, - .regs = (struct sport_register *)SPORT0_TCR1, - }, -#ifdef PIN_REQ_SPORT_1 - { - .dma_rx_chan = CH_SPORT1_RX, - .dma_tx_chan = CH_SPORT1_TX, - .err_irq = IRQ_SPORT1_ERROR, - .regs = (struct sport_register *)SPORT1_TCR1, - }, +#ifdef SPORT0_TCR1 + SPORT_PARAMS(0), #endif -#ifdef PIN_REQ_SPORT_2 - { - .dma_rx_chan = CH_SPORT2_RX, - .dma_tx_chan = CH_SPORT2_TX, - .err_irq = IRQ_SPORT2_ERROR, - .regs = (struct sport_register *)SPORT2_TCR1, - }, +#ifdef SPORT1_TCR1 + SPORT_PARAMS(1), #endif -#ifdef PIN_REQ_SPORT_3 - { - .dma_rx_chan = CH_SPORT3_RX, - .dma_tx_chan = CH_SPORT3_TX, - .err_irq = IRQ_SPORT3_ERROR, - .regs = (struct sport_register *)SPORT3_TCR1, - } +#ifdef SPORT2_TCR1 + SPORT_PARAMS(2), +#endif +#ifdef SPORT3_TCR1 + SPORT_PARAMS(3), #endif }; @@ -332,11 +306,11 @@ static int bf5xx_ac97_probe(struct platform_device *pdev, if (cmd_count == NULL) return -ENOMEM; - if (peripheral_request_list(&sport_req[sport_num][0], "soc-audio")) { + if (peripheral_request_list(sport_req[sport_num], "soc-audio")) { pr_err("Requesting Peripherals failed\n"); ret = -EFAULT; goto peripheral_err; - } + } #ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET /* Request PB3 as reset pin */ @@ -385,7 +359,7 @@ sport_err: gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM); #endif gpio_err: - peripheral_free_list(&sport_req[sport_num][0]); + peripheral_free_list(sport_req[sport_num]); peripheral_err: free_page((unsigned long)cmd_count); cmd_count = NULL; @@ -398,7 +372,7 @@ static void bf5xx_ac97_remove(struct platform_device *pdev, { free_page((unsigned long)cmd_count); cmd_count = NULL; - peripheral_free_list(&sport_req[sport_num][0]); + peripheral_free_list(sport_req[sport_num]); #ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM); #endif -- cgit v1.2.3-70-g09d2 From 8836c273e4d44d088157b7ccbd2c108cefe70565 Mon Sep 17 00:00:00 2001 From: Mike Frysinger Date: Fri, 6 Feb 2009 18:12:35 +0800 Subject: ASoC: Blackfin: drop unnecessary dma casts Signed-off-by: Mike Frysinger Signed-off-by: Bryan Wu Signed-off-by: Mark Brown --- sound/soc/blackfin/bf5xx-sport.c | 104 ++++++++++++++++----------------------- 1 file changed, 43 insertions(+), 61 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-sport.c b/sound/soc/blackfin/bf5xx-sport.c index 3b99e484d55..b7953c8cf83 100644 --- a/sound/soc/blackfin/bf5xx-sport.c +++ b/sound/soc/blackfin/bf5xx-sport.c @@ -133,7 +133,7 @@ static void setup_desc(struct dmasg *desc, void *buf, int fragcount, int i; for (i = 0; i < fragcount; ++i) { - desc[i].next_desc_addr = (unsigned long)&(desc[i + 1]); + desc[i].next_desc_addr = &(desc[i + 1]); desc[i].start_addr = (unsigned long)buf + i*fragsize; desc[i].cfg = cfg; desc[i].x_count = x_count; @@ -143,12 +143,12 @@ static void setup_desc(struct dmasg *desc, void *buf, int fragcount, } /* make circular */ - desc[fragcount-1].next_desc_addr = (unsigned long)desc; + desc[fragcount-1].next_desc_addr = desc; - pr_debug("setup desc: desc0=%p, next0=%lx, desc1=%p," - "next1=%lx\nx_count=%x,y_count=%x,addr=0x%lx,cfs=0x%x\n", - &(desc[0]), desc[0].next_desc_addr, - &(desc[1]), desc[1].next_desc_addr, + pr_debug("setup desc: desc0=%p, next0=%p, desc1=%p," + "next1=%p\nx_count=%x,y_count=%x,addr=0x%lx,cfs=0x%x\n", + desc, desc[0].next_desc_addr, + desc+1, desc[1].next_desc_addr, desc[0].x_count, desc[0].y_count, desc[0].start_addr, desc[0].cfg); } @@ -184,22 +184,20 @@ static inline int sport_hook_rx_dummy(struct sport_device *sport) BUG_ON(sport->curr_rx_desc == sport->dummy_rx_desc); /* Maybe the dummy buffer descriptor ring is damaged */ - sport->dummy_rx_desc->next_desc_addr = \ - (unsigned long)(sport->dummy_rx_desc+1); + sport->dummy_rx_desc->next_desc_addr = sport->dummy_rx_desc + 1; local_irq_save(flags); - desc = (struct dmasg *)get_dma_next_desc_ptr(sport->dma_rx_chan); + desc = get_dma_next_desc_ptr(sport->dma_rx_chan); /* Copy the descriptor which will be damaged to backup */ temp_desc = *desc; desc->x_count = 0xa; desc->y_count = 0; - desc->next_desc_addr = (unsigned long)(sport->dummy_rx_desc); + desc->next_desc_addr = sport->dummy_rx_desc; local_irq_restore(flags); /* Waiting for dummy buffer descriptor is already hooked*/ while ((get_dma_curr_desc_ptr(sport->dma_rx_chan) - - sizeof(struct dmasg)) != - (unsigned long)sport->dummy_rx_desc) - ; + sizeof(struct dmasg)) != sport->dummy_rx_desc) + continue; sport->curr_rx_desc = sport->dummy_rx_desc; /* Restore the damaged descriptor */ *desc = temp_desc; @@ -210,14 +208,12 @@ static inline int sport_hook_rx_dummy(struct sport_device *sport) static inline int sport_rx_dma_start(struct sport_device *sport, int dummy) { if (dummy) { - sport->dummy_rx_desc->next_desc_addr = \ - (unsigned long) sport->dummy_rx_desc; + sport->dummy_rx_desc->next_desc_addr = sport->dummy_rx_desc; sport->curr_rx_desc = sport->dummy_rx_desc; } else sport->curr_rx_desc = sport->dma_rx_desc; - set_dma_next_desc_addr(sport->dma_rx_chan, \ - (unsigned long)(sport->curr_rx_desc)); + set_dma_next_desc_addr(sport->dma_rx_chan, sport->curr_rx_desc); set_dma_x_count(sport->dma_rx_chan, 0); set_dma_x_modify(sport->dma_rx_chan, 0); set_dma_config(sport->dma_rx_chan, (DMAFLOW_LARGE | NDSIZE_9 | \ @@ -231,14 +227,12 @@ static inline int sport_rx_dma_start(struct sport_device *sport, int dummy) static inline int sport_tx_dma_start(struct sport_device *sport, int dummy) { if (dummy) { - sport->dummy_tx_desc->next_desc_addr = \ - (unsigned long) sport->dummy_tx_desc; + sport->dummy_tx_desc->next_desc_addr = sport->dummy_tx_desc; sport->curr_tx_desc = sport->dummy_tx_desc; } else sport->curr_tx_desc = sport->dma_tx_desc; - set_dma_next_desc_addr(sport->dma_tx_chan, \ - (unsigned long)(sport->curr_tx_desc)); + set_dma_next_desc_addr(sport->dma_tx_chan, sport->curr_tx_desc); set_dma_x_count(sport->dma_tx_chan, 0); set_dma_x_modify(sport->dma_tx_chan, 0); set_dma_config(sport->dma_tx_chan, @@ -261,11 +255,9 @@ int sport_rx_start(struct sport_device *sport) BUG_ON(sport->curr_rx_desc != sport->dummy_rx_desc); local_irq_save(flags); while ((get_dma_curr_desc_ptr(sport->dma_rx_chan) - - sizeof(struct dmasg)) != - (unsigned long)sport->dummy_rx_desc) - ; - sport->dummy_rx_desc->next_desc_addr = - (unsigned long)(sport->dma_rx_desc); + sizeof(struct dmasg)) != sport->dummy_rx_desc) + continue; + sport->dummy_rx_desc->next_desc_addr = sport->dma_rx_desc; local_irq_restore(flags); sport->curr_rx_desc = sport->dma_rx_desc; } else { @@ -310,23 +302,21 @@ static inline int sport_hook_tx_dummy(struct sport_device *sport) BUG_ON(sport->dummy_tx_desc == NULL); BUG_ON(sport->curr_tx_desc == sport->dummy_tx_desc); - sport->dummy_tx_desc->next_desc_addr = \ - (unsigned long)(sport->dummy_tx_desc+1); + sport->dummy_tx_desc->next_desc_addr = sport->dummy_tx_desc + 1; /* Shorten the time on last normal descriptor */ local_irq_save(flags); - desc = (struct dmasg *)get_dma_next_desc_ptr(sport->dma_tx_chan); + desc = get_dma_next_desc_ptr(sport->dma_tx_chan); /* Store the descriptor which will be damaged */ temp_desc = *desc; desc->x_count = 0xa; desc->y_count = 0; - desc->next_desc_addr = (unsigned long)(sport->dummy_tx_desc); + desc->next_desc_addr = sport->dummy_tx_desc; local_irq_restore(flags); /* Waiting for dummy buffer descriptor is already hooked*/ while ((get_dma_curr_desc_ptr(sport->dma_tx_chan) - \ - sizeof(struct dmasg)) != \ - (unsigned long)sport->dummy_tx_desc) - ; + sizeof(struct dmasg)) != sport->dummy_tx_desc) + continue; sport->curr_tx_desc = sport->dummy_tx_desc; /* Restore the damaged descriptor */ *desc = temp_desc; @@ -347,11 +337,9 @@ int sport_tx_start(struct sport_device *sport) /* Hook the normal buffer descriptor */ local_irq_save(flags); while ((get_dma_curr_desc_ptr(sport->dma_tx_chan) - - sizeof(struct dmasg)) != - (unsigned long)sport->dummy_tx_desc) - ; - sport->dummy_tx_desc->next_desc_addr = - (unsigned long)(sport->dma_tx_desc); + sizeof(struct dmasg)) != sport->dummy_tx_desc) + continue; + sport->dummy_tx_desc->next_desc_addr = sport->dma_tx_desc; local_irq_restore(flags); sport->curr_tx_desc = sport->dma_tx_desc; } else { @@ -536,19 +524,17 @@ static int sport_config_rx_dummy(struct sport_device *sport) unsigned config; pr_debug("%s entered\n", __func__); -#if L1_DATA_A_LENGTH != 0 - desc = (struct dmasg *) l1_data_sram_alloc(2 * sizeof(*desc)); -#else - { + if (L1_DATA_A_LENGTH) + desc = l1_data_sram_zalloc(2 * sizeof(*desc)); + else { dma_addr_t addr; desc = dma_alloc_coherent(NULL, 2 * sizeof(*desc), &addr, 0); + memset(desc, 0, 2 * sizeof(*desc)); } -#endif if (desc == NULL) { pr_err("Failed to allocate memory for dummy rx desc\n"); return -ENOMEM; } - memset(desc, 0, 2 * sizeof(*desc)); sport->dummy_rx_desc = desc; desc->start_addr = (unsigned long)sport->dummy_buf; config = DMAFLOW_LARGE | NDSIZE_9 | compute_wdsize(sport->wdsize) @@ -559,8 +545,8 @@ static int sport_config_rx_dummy(struct sport_device *sport) desc->y_count = 0; desc->y_modify = 0; memcpy(desc+1, desc, sizeof(*desc)); - desc->next_desc_addr = (unsigned long)(desc+1); - desc[1].next_desc_addr = (unsigned long)desc; + desc->next_desc_addr = desc + 1; + desc[1].next_desc_addr = desc; return 0; } @@ -571,19 +557,17 @@ static int sport_config_tx_dummy(struct sport_device *sport) pr_debug("%s entered\n", __func__); -#if L1_DATA_A_LENGTH != 0 - desc = (struct dmasg *) l1_data_sram_alloc(2 * sizeof(*desc)); -#else - { + if (L1_DATA_A_LENGTH) + desc = l1_data_sram_zalloc(2 * sizeof(*desc)); + else { dma_addr_t addr; desc = dma_alloc_coherent(NULL, 2 * sizeof(*desc), &addr, 0); + memset(desc, 0, 2 * sizeof(*desc)); } -#endif if (!desc) { pr_err("Failed to allocate memory for dummy tx desc\n"); return -ENOMEM; } - memset(desc, 0, 2 * sizeof(*desc)); sport->dummy_tx_desc = desc; desc->start_addr = (unsigned long)sport->dummy_buf + \ sport->dummy_count; @@ -595,8 +579,8 @@ static int sport_config_tx_dummy(struct sport_device *sport) desc->y_count = 0; desc->y_modify = 0; memcpy(desc+1, desc, sizeof(*desc)); - desc->next_desc_addr = (unsigned long)(desc+1); - desc[1].next_desc_addr = (unsigned long)desc; + desc->next_desc_addr = desc + 1; + desc[1].next_desc_addr = desc; return 0; } @@ -872,17 +856,15 @@ struct sport_device *sport_init(struct sport_param *param, unsigned wdsize, sport->wdsize = wdsize; sport->dummy_count = dummy_count; -#if L1_DATA_A_LENGTH != 0 - sport->dummy_buf = l1_data_sram_alloc(dummy_count * 2); -#else - sport->dummy_buf = kmalloc(dummy_count * 2, GFP_KERNEL); -#endif + if (L1_DATA_A_LENGTH) + sport->dummy_buf = l1_data_sram_zalloc(dummy_count * 2); + else + sport->dummy_buf = kzalloc(dummy_count * 2, GFP_KERNEL); if (sport->dummy_buf == NULL) { pr_err("Failed to allocate dummy buffer\n"); goto __error; } - memset(sport->dummy_buf, 0, dummy_count * 2); ret = sport_config_rx_dummy(sport); if (ret) { pr_err("Failed to config rx dummy ring\n"); @@ -939,6 +921,7 @@ void sport_done(struct sport_device *sport) sport = NULL; } EXPORT_SYMBOL(sport_done); + /* * It is only used to send several bytes when dma is not enabled * sport controller is configured but not enabled. @@ -1029,4 +1012,3 @@ EXPORT_SYMBOL(sport_send_and_recv); MODULE_AUTHOR("Roy Huang"); MODULE_DESCRIPTION("SPORT driver for ADI Blackfin"); MODULE_LICENSE("GPL"); - -- cgit v1.2.3-70-g09d2 From 85ef2375ef2ebbb2bf660ad3a27c644d0ebf1b1a Mon Sep 17 00:00:00 2001 From: Timur Tabi Date: Thu, 5 Feb 2009 17:56:02 -0600 Subject: ASoC: optimize init sequence of Freescale MPC8610 sound drivers In the Freescale MPC8610 sound drivers, relocate all code from the _prepare functions into the corresponding _hw_params functions. These drivers assumed that the sample size is known in the _prepare function and not in the _hw_params function, but this is not true. Move the code in fsl_dma_prepare() into fsl_dma_hw_param(). Create fsl_ssi_hw_params() and move the code from fsl_ssi_prepare() into it. Turn off snooping for DMA operations to/from I/O registers, since that's not necessary. Signed-off-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_dma.c | 178 +++++++++++++++++++++++------------------------- sound/soc/fsl/fsl_ssi.c | 19 +++--- 2 files changed, 94 insertions(+), 103 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 64993eda567..58a3fa49750 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -464,11 +464,7 @@ static int fsl_dma_open(struct snd_pcm_substream *substream) sizeof(struct fsl_dma_link_descriptor); for (i = 0; i < NUM_DMA_LINKS; i++) { - struct fsl_dma_link_descriptor *link = &dma_private->link[i]; - - link->source_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP); - link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP); - link->next = cpu_to_be64(temp_link); + dma_private->link[i].next = cpu_to_be64(temp_link); temp_link += sizeof(struct fsl_dma_link_descriptor); } @@ -525,79 +521,9 @@ static int fsl_dma_open(struct snd_pcm_substream *substream) * This function obtains hardware parameters about the opened stream and * programs the DMA controller accordingly. * - * Note that due to a quirk of the SSI's STX register, the target address - * for the DMA operations depends on the sample size. So we don't program - * the dest_addr (for playback -- source_addr for capture) fields in the - * link descriptors here. We do that in fsl_dma_prepare() - */ -static int fsl_dma_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *hw_params) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct fsl_dma_private *dma_private = runtime->private_data; - - dma_addr_t temp_addr; /* Pointer to next period */ - - unsigned int i; - - /* Get all the parameters we need */ - size_t buffer_size = params_buffer_bytes(hw_params); - size_t period_size = params_period_bytes(hw_params); - - /* Initialize our DMA tracking variables */ - dma_private->period_size = period_size; - dma_private->num_periods = params_periods(hw_params); - dma_private->dma_buf_end = dma_private->dma_buf_phys + buffer_size; - dma_private->dma_buf_next = dma_private->dma_buf_phys + - (NUM_DMA_LINKS * period_size); - if (dma_private->dma_buf_next >= dma_private->dma_buf_end) - dma_private->dma_buf_next = dma_private->dma_buf_phys; - - /* - * The actual address in STX0 (destination for playback, source for - * capture) is based on the sample size, but we don't know the sample - * size in this function, so we'll have to adjust that later. See - * comments in fsl_dma_prepare(). - * - * The DMA controller does not have a cache, so the CPU does not - * need to tell it to flush its cache. However, the DMA - * controller does need to tell the CPU to flush its cache. - * That's what the SNOOP bit does. - * - * Also, even though the DMA controller supports 36-bit addressing, for - * simplicity we currently support only 32-bit addresses for the audio - * buffer itself. - */ - temp_addr = substream->dma_buffer.addr; - - for (i = 0; i < NUM_DMA_LINKS; i++) { - struct fsl_dma_link_descriptor *link = &dma_private->link[i]; - - link->count = cpu_to_be32(period_size); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - link->source_addr = cpu_to_be32(temp_addr); - else - link->dest_addr = cpu_to_be32(temp_addr); - - temp_addr += period_size; - } - - return 0; -} - -/** - * fsl_dma_prepare - prepare the DMA registers for playback. - * - * This function is called after the specifics of the audio data are known, - * i.e. snd_pcm_runtime is initialized. - * - * In this function, we finish programming the registers of the DMA - * controller that are dependent on the sample size. - * - * One of the drawbacks with big-endian is that when copying integers of - * different sizes to a fixed-sized register, the address to which the - * integer must be copied is dependent on the size of the integer. + * One drawback of big-endian is that when copying integers of different + * sizes to a fixed-sized register, the address to which the integer must be + * copied is dependent on the size of the integer. * * For example, if P is the address of a 32-bit register, and X is a 32-bit * integer, then X should be copied to address P. However, if X is a 16-bit @@ -613,22 +539,58 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream, * and 8 bytes at a time). So we do not support packed 24-bit samples. * 24-bit data must be padded to 32 bits. */ -static int fsl_dma_prepare(struct snd_pcm_substream *substream) +static int fsl_dma_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) { struct snd_pcm_runtime *runtime = substream->runtime; struct fsl_dma_private *dma_private = runtime->private_data; + + /* Number of bits per sample */ + unsigned int sample_size = + snd_pcm_format_physical_width(params_format(hw_params)); + + /* Number of bytes per frame */ + unsigned int frame_size = 2 * (sample_size / 8); + + /* Bus address of SSI STX register */ + dma_addr_t ssi_sxx_phys = dma_private->ssi_sxx_phys; + + /* Size of the DMA buffer, in bytes */ + size_t buffer_size = params_buffer_bytes(hw_params); + + /* Number of bytes per period */ + size_t period_size = params_period_bytes(hw_params); + + /* Pointer to next period */ + dma_addr_t temp_addr = substream->dma_buffer.addr; + + /* Pointer to DMA controller */ struct ccsr_dma_channel __iomem *dma_channel = dma_private->dma_channel; - u32 mr; + + u32 mr; /* DMA Mode Register */ + unsigned int i; - dma_addr_t ssi_sxx_phys; /* Bus address of SSI STX register */ - unsigned int frame_size; /* Number of bytes per frame */ - ssi_sxx_phys = dma_private->ssi_sxx_phys; + /* Initialize our DMA tracking variables */ + dma_private->period_size = period_size; + dma_private->num_periods = params_periods(hw_params); + dma_private->dma_buf_end = dma_private->dma_buf_phys + buffer_size; + dma_private->dma_buf_next = dma_private->dma_buf_phys + + (NUM_DMA_LINKS * period_size); + + if (dma_private->dma_buf_next >= dma_private->dma_buf_end) + /* This happens if the number of periods == NUM_DMA_LINKS */ + dma_private->dma_buf_next = dma_private->dma_buf_phys; mr = in_be32(&dma_channel->mr) & ~(CCSR_DMA_MR_BWC_MASK | CCSR_DMA_MR_SAHTS_MASK | CCSR_DMA_MR_DAHTS_MASK); - switch (runtime->sample_bits) { + /* Due to a quirk of the SSI's STX register, the target address + * for the DMA operations depends on the sample size. So we calculate + * that offset here. While we're at it, also tell the DMA controller + * how much data to transfer per sample. + */ + switch (sample_size) { case 8: mr |= CCSR_DMA_MR_DAHTS_1 | CCSR_DMA_MR_SAHTS_1; ssi_sxx_phys += 3; @@ -641,12 +603,12 @@ static int fsl_dma_prepare(struct snd_pcm_substream *substream) mr |= CCSR_DMA_MR_DAHTS_4 | CCSR_DMA_MR_SAHTS_4; break; default: + /* We should never get here */ dev_err(substream->pcm->card->dev, - "unsupported sample size %u\n", runtime->sample_bits); + "unsupported sample size %u\n", sample_size); return -EINVAL; } - frame_size = runtime->frame_bits / 8; /* * BWC should always be a multiple of the frame size. BWC determines * how many bytes are sent/received before the DMA controller checks the @@ -655,7 +617,6 @@ static int fsl_dma_prepare(struct snd_pcm_substream *substream) * capture, the receive FIFO is triggered when it contains one frame, so * we want to receive one frame at a time. */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) mr |= CCSR_DMA_MR_BWC(2 * frame_size); else @@ -663,16 +624,48 @@ static int fsl_dma_prepare(struct snd_pcm_substream *substream) out_be32(&dma_channel->mr, mr); - /* - * Program the address of the DMA transfer to/from the SSI. - */ for (i = 0; i < NUM_DMA_LINKS; i++) { struct fsl_dma_link_descriptor *link = &dma_private->link[i]; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + link->count = cpu_to_be32(period_size); + + /* Even though the DMA controller supports 36-bit addressing, + * for simplicity we allow only 32-bit addresses for the audio + * buffer itself. This was enforced in fsl_dma_new() with the + * DMA mask. + * + * The snoop bit tells the DMA controller whether it should tell + * the ECM to snoop during a read or write to an address. For + * audio, we use DMA to transfer data between memory and an I/O + * device (the SSI's STX0 or SRX0 register). Snooping is only + * needed if there is a cache, so we need to snoop memory + * addresses only. For playback, that means we snoop the source + * but not the destination. For capture, we snoop the + * destination but not the source. + * + * Note that failing to snoop properly is unlikely to cause + * cache incoherency if the period size is larger than the + * size of L1 cache. This is because filling in one period will + * flush out the data for the previous period. So if you + * increased period_bytes_min to a large enough size, you might + * get more performance by not snooping, and you'll still be + * okay. + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + link->source_addr = cpu_to_be32(temp_addr); + link->source_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP); + link->dest_addr = cpu_to_be32(ssi_sxx_phys); - else + link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_NOSNOOP); + } else { link->source_addr = cpu_to_be32(ssi_sxx_phys); + link->source_attr = cpu_to_be32(CCSR_DMA_ATR_NOSNOOP); + + link->dest_addr = cpu_to_be32(temp_addr); + link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP); + } + + temp_addr += period_size; } return 0; @@ -808,7 +801,6 @@ static struct snd_pcm_ops fsl_dma_ops = { .ioctl = snd_pcm_lib_ioctl, .hw_params = fsl_dma_hw_params, .hw_free = fsl_dma_hw_free, - .prepare = fsl_dma_prepare, .pointer = fsl_dma_pointer, }; diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index c6d6eb71dc1..6844009833d 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -400,7 +400,7 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, } /** - * fsl_ssi_prepare: prepare the SSI. + * fsl_ssi_hw_params - program the sample size * * Most of the SSI registers have been programmed in the startup function, * but the word length must be programmed here. Unfortunately, programming @@ -412,20 +412,19 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, * Note: The SxCCR.DC and SxCCR.PM bits are only used if the SSI is the * clock master. */ -static int fsl_ssi_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) +static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params, struct snd_soc_dai *cpu_dai) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data; - - struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + struct fsl_ssi_private *ssi_private = cpu_dai->private_data; if (substream == ssi_private->first_stream) { + struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + unsigned int sample_size = + snd_pcm_format_width(params_format(hw_params)); u32 wl; /* The SSI should always be disabled at this points (SSIEN=0) */ - wl = CCSR_SSI_SxCCR_WL(snd_pcm_format_width(runtime->format)); + wl = CCSR_SSI_SxCCR_WL(sample_size); /* In synchronous mode, the SSI uses STCCR for capture */ clrsetbits_be32(&ssi->stccr, CCSR_SSI_SxCCR_WL_MASK, wl); @@ -579,7 +578,7 @@ static struct snd_soc_dai fsl_ssi_dai_template = { }, .ops = { .startup = fsl_ssi_startup, - .prepare = fsl_ssi_prepare, + .hw_params = fsl_ssi_hw_params, .shutdown = fsl_ssi_shutdown, .trigger = fsl_ssi_trigger, .set_sysclk = fsl_ssi_set_sysclk, -- cgit v1.2.3-70-g09d2 From 45bdd1c1bbac56876cb9c71649300013281e4b22 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 6 Feb 2009 16:11:25 +0100 Subject: ALSA: hda - Create beep mixer controls dynamically for Realtek codecs Create beep mixer controls dynamically for Realtek codecs instead of static arrays. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 147 ++++++++++++++---------------------------- 1 file changed, 47 insertions(+), 100 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 5194a58fafa..3b3b483e2a9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -255,6 +255,7 @@ struct alc_spec { struct snd_kcontrol_new *mixers[5]; /* mixer arrays */ unsigned int num_mixers; struct snd_kcontrol_new *cap_mixer; /* capture mixer */ + unsigned int beep_amp; /* beep amp value, set via set_beep_amp() */ const struct hda_verb *init_verbs[5]; /* initialization verbs * don't forget NULL @@ -937,7 +938,7 @@ static void alc_mic_automute(struct hda_codec *codec) HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); } #else -#define alc_mic_automute(codec) /* NOP */ +#define alc_mic_automute(codec) do {} while(0) /* NOP */ #endif /* disabled */ /* unsolicited event for HP jack sensing */ @@ -1389,8 +1390,6 @@ static struct snd_kcontrol_new alc888_base_mixer[] = { HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -1497,8 +1496,6 @@ static struct snd_kcontrol_new alc880_three_stack_mixer[] = { HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x3, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x3, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x19, 0x0, HDA_OUTPUT), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -1720,8 +1717,6 @@ static struct snd_kcontrol_new alc880_six_stack_mixer[] = { HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Channel Mode", @@ -1898,13 +1893,6 @@ static struct snd_kcontrol_new alc880_asus_w1v_mixer[] = { { } /* end */ }; -/* additional mixers to alc880_asus_mixer */ -static struct snd_kcontrol_new alc880_pcbeep_mixer[] = { - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), - { } /* end */ -}; - /* TCL S700 */ static struct snd_kcontrol_new alc880_tcl_s700_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), @@ -1937,8 +1925,6 @@ static struct snd_kcontrol_new alc880_uniwill_mixer[] = { HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Channel Mode", @@ -2013,6 +1999,13 @@ static const char *alc_slave_sws[] = { static void alc_free_kctls(struct hda_codec *codec); +/* additional beep mixers; the actual parameters are overwritten at build */ +static struct snd_kcontrol_new alc_beep_mixer[] = { + HDA_CODEC_VOLUME("Beep Playback Volume", 0, 0, HDA_INPUT), + HDA_CODEC_MUTE("Beep Playback Switch", 0, 0, HDA_INPUT), + { } /* end */ +}; + static int alc_build_controls(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -2048,6 +2041,21 @@ static int alc_build_controls(struct hda_codec *codec) return err; } + /* create beep controls if needed */ + if (spec->beep_amp) { + struct snd_kcontrol_new *knew; + for (knew = alc_beep_mixer; knew->name; knew++) { + struct snd_kcontrol *kctl; + kctl = snd_ctl_new1(knew, codec); + if (!kctl) + return -ENOMEM; + kctl->private_value = spec->beep_amp; + err = snd_hda_ctl_add(codec, kctl); + if (err < 0) + return err; + } + } + /* if we have no master control, let's create it */ if (!spec->no_analog && !snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) { @@ -3812,7 +3820,7 @@ static struct alc_config_preset alc880_presets[] = { .input_mux = &alc880_capture_source, }, [ALC880_UNIWILL_DIG] = { - .mixers = { alc880_asus_mixer, alc880_pcbeep_mixer }, + .mixers = { alc880_asus_mixer }, .init_verbs = { alc880_volume_init_verbs, alc880_pin_asus_init_verbs }, .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids), @@ -3850,8 +3858,7 @@ static struct alc_config_preset alc880_presets[] = { .init_hook = alc880_uniwill_p53_hp_automute, }, [ALC880_FUJITSU] = { - .mixers = { alc880_fujitsu_mixer, - alc880_pcbeep_mixer, }, + .mixers = { alc880_fujitsu_mixer }, .init_verbs = { alc880_volume_init_verbs, alc880_uniwill_p53_init_verbs, alc880_beep_init_verbs }, @@ -4310,10 +4317,6 @@ static void alc880_auto_init(struct hda_codec *codec) alc_inithook(codec); } -/* - * OK, here we have finally the patch for ALC880 - */ - static void set_capture_mixer(struct alc_spec *spec) { static struct snd_kcontrol_new *caps[3] = { @@ -4325,6 +4328,13 @@ static void set_capture_mixer(struct alc_spec *spec) spec->cap_mixer = caps[spec->num_adc_nids - 1]; } +#define set_beep_amp(spec, nid, idx, dir) \ + ((spec)->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 3, idx, dir)) + +/* + * OK, here we have finally the patch for ALC880 + */ + static int patch_alc880(struct hda_codec *codec) { struct alc_spec *spec; @@ -4392,6 +4402,7 @@ static int patch_alc880(struct hda_codec *codec) } } set_capture_mixer(spec); + set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); spec->vmaster_nid = 0x0c; @@ -4541,12 +4552,6 @@ static struct snd_kcontrol_new alc260_input_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc260_pc_beep_mixer[] = { - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x07, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x07, 0x05, HDA_INPUT), - { } /* end */ -}; - /* update HP, line and mono out pins according to the master switch */ static void alc260_hp_master_update(struct hda_codec *codec, hda_nid_t hp, hda_nid_t line, @@ -4738,8 +4743,6 @@ static struct snd_kcontrol_new alc260_fujitsu_mixer[] = { HDA_CODEC_VOLUME("Mic/Line Playback Volume", 0x07, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Mic/Line Playback Switch", 0x07, 0x0, HDA_INPUT), ALC_PIN_MODE("Mic/Line Jack Mode", 0x12, ALC_PIN_DIR_IN), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x07, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x07, 0x05, HDA_INPUT), HDA_CODEC_VOLUME("Speaker Playback Volume", 0x09, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Speaker Playback Switch", 0x09, 2, HDA_INPUT), { } /* end */ @@ -4784,8 +4787,6 @@ static struct snd_kcontrol_new alc260_acer_mixer[] = { HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT), HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT), ALC_PIN_MODE("Line Jack Mode", 0x14, ALC_PIN_DIR_INOUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x07, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x07, 0x05, HDA_INPUT), { } /* end */ }; @@ -4803,8 +4804,6 @@ static struct snd_kcontrol_new alc260_will_mixer[] = { ALC_PIN_MODE("Line Jack Mode", 0x14, ALC_PIN_DIR_INOUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x07, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x07, 0x05, HDA_INPUT), { } /* end */ }; @@ -5308,8 +5307,6 @@ static struct snd_kcontrol_new alc260_test_mixer[] = { HDA_CODEC_MUTE("LINE2 Playback Switch", 0x07, 0x03, HDA_INPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x07, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x07, 0x05, HDA_INPUT), HDA_CODEC_VOLUME("LINE-OUT loopback Playback Volume", 0x07, 0x06, HDA_INPUT), HDA_CODEC_MUTE("LINE-OUT loopback Playback Switch", 0x07, 0x06, HDA_INPUT), HDA_CODEC_VOLUME("HP-OUT loopback Playback Volume", 0x07, 0x7, HDA_INPUT), @@ -5737,8 +5734,7 @@ static struct snd_pci_quirk alc260_cfg_tbl[] = { static struct alc_config_preset alc260_presets[] = { [ALC260_BASIC] = { .mixers = { alc260_base_output_mixer, - alc260_input_mixer, - alc260_pc_beep_mixer }, + alc260_input_mixer }, .init_verbs = { alc260_init_verbs }, .num_dacs = ARRAY_SIZE(alc260_dac_nids), .dac_nids = alc260_dac_nids, @@ -5924,6 +5920,7 @@ static int patch_alc260(struct hda_codec *codec) } } set_capture_mixer(spec); + set_beep_amp(spec, 0x07, 0x05, HDA_INPUT); spec->vmaster_nid = 0x08; @@ -6095,8 +6092,6 @@ static struct snd_kcontrol_new alc882_base_mixer[] = { HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -6123,8 +6118,6 @@ static struct snd_kcontrol_new alc882_w2jc_mixer[] = { HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -6176,8 +6169,6 @@ static struct snd_kcontrol_new alc882_asus_a7m_mixer[] = { HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -6286,8 +6277,10 @@ static struct snd_kcontrol_new alc882_macpro_mixer[] = { HDA_CODEC_MUTE("Headphone Playback Switch", 0x18, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x01, HDA_INPUT), + /* FIXME: this looks suspicious... HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x02, HDA_INPUT), + */ { } /* end */ }; @@ -7153,6 +7146,7 @@ static int patch_alc882(struct hda_codec *codec) } } set_capture_mixer(spec); + set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); spec->vmaster_nid = 0x0c; @@ -7429,8 +7423,6 @@ static struct snd_kcontrol_new alc883_base_mixer[] = { HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -7493,8 +7485,6 @@ static struct snd_kcontrol_new alc883_3ST_2ch_mixer[] = { HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -7518,8 +7508,6 @@ static struct snd_kcontrol_new alc883_3ST_6ch_mixer[] = { HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -7544,8 +7532,6 @@ static struct snd_kcontrol_new alc883_3ST_6ch_intel_mixer[] = { HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Boost", 0x18, 0, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -7569,8 +7555,6 @@ static struct snd_kcontrol_new alc883_fivestack_mixer[] = { HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -9183,6 +9167,7 @@ static int patch_alc883(struct hda_codec *codec) if (!spec->cap_mixer) set_capture_mixer(spec); + set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); spec->vmaster_nid = 0x0c; @@ -9235,8 +9220,6 @@ static struct snd_kcontrol_new alc262_base_mixer[] = { HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), - /* HDA_CODEC_VOLUME("PC Beep Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Beep Playback Switch", 0x0b, 0x05, HDA_INPUT), */ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0D, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), @@ -9257,8 +9240,6 @@ static struct snd_kcontrol_new alc262_hippo1_mixer[] = { HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), - /* HDA_CODEC_VOLUME("PC Beep Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Beep Playback Switch", 0x0b, 0x05, HDA_INPUT), */ /*HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0D, 0x0, HDA_OUTPUT),*/ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), { } /* end */ @@ -9367,8 +9348,6 @@ static struct snd_kcontrol_new alc262_HP_BPC_mixer[] = { HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("PC Beep Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Beep Playback Switch", 0x0b, 0x05, HDA_INPUT), HDA_CODEC_VOLUME("AUX IN Playback Volume", 0x0b, 0x06, HDA_INPUT), HDA_CODEC_MUTE("AUX IN Playback Switch", 0x0b, 0x06, HDA_INPUT), { } /* end */ @@ -9397,8 +9376,6 @@ static struct snd_kcontrol_new alc262_HP_BPC_WildWest_mixer[] = { HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("PC Beep Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Beep Playback Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -10073,8 +10050,6 @@ static struct snd_kcontrol_new alc262_fujitsu_mixer[] = { }, HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Switch", 0x0b, 0x05, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), @@ -11085,6 +11060,7 @@ static int patch_alc262(struct hda_codec *codec) } if (!spec->cap_mixer && !spec->no_analog) set_capture_mixer(spec); + set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); spec->vmaster_nid = 0x0c; @@ -12205,8 +12181,6 @@ static struct snd_kcontrol_new alc269_base_mixer[] = { HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x0b, 0x4, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x0b, 0x4, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), @@ -12233,8 +12207,6 @@ static struct snd_kcontrol_new alc269_quanta_fl1_mixer[] = { HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_VOLUME("Internal Mic Boost", 0x19, 0, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x04, HDA_INPUT), { } }; @@ -12258,8 +12230,6 @@ static struct snd_kcontrol_new alc269_lifebook_mixer[] = { HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x0b, 0x03, HDA_INPUT), HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x0b, 0x03, HDA_INPUT), HDA_CODEC_VOLUME("Dock Mic Boost", 0x1b, 0, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x04, HDA_INPUT), { } }; @@ -12296,13 +12266,6 @@ static struct snd_kcontrol_new alc269_fujitsu_mixer[] = { { } /* end */ }; -/* beep control */ -static struct snd_kcontrol_new alc269_beep_mixer[] = { - HDA_CODEC_VOLUME("Beep Playback Volume", 0x0b, 0x4, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x0b, 0x4, HDA_INPUT), - { } /* end */ -}; - static struct hda_verb alc269_quanta_fl1_verbs[] = { {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, @@ -12749,13 +12712,6 @@ static int alc269_parse_auto_config(struct hda_codec *codec) if (spec->kctls.list) add_mixer(spec, spec->kctls.list); - /* create a beep mixer control if the pin 0x1d isn't assigned */ - for (i = 0; i < ARRAY_SIZE(spec->autocfg.input_pins); i++) - if (spec->autocfg.input_pins[i] == 0x1d) - break; - if (i >= ARRAY_SIZE(spec->autocfg.input_pins)) - add_mixer(spec, alc269_beep_mixer); - add_verb(spec, alc269_init_verbs); spec->num_mux_defs = 1; spec->input_mux = &spec->private_imux[0]; @@ -12868,7 +12824,7 @@ static struct alc_config_preset alc269_presets[] = { .init_hook = alc269_eeepc_dmic_inithook, }, [ALC269_FUJITSU] = { - .mixers = { alc269_fujitsu_mixer, alc269_beep_mixer }, + .mixers = { alc269_fujitsu_mixer }, .cap_mixer = alc269_epc_capture_mixer, .init_verbs = { alc269_init_verbs, alc269_eeepc_dmic_init_verbs }, @@ -12955,6 +12911,7 @@ static int patch_alc269(struct hda_codec *codec) spec->capsrc_nids = alc269_capsrc_nids; if (!spec->cap_mixer) set_capture_mixer(spec); + set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT); codec->patch_ops = alc_patch_ops; if (board_config == ALC269_AUTO) @@ -13205,8 +13162,6 @@ static struct snd_kcontrol_new alc861_asus_mixer[] = { static struct snd_kcontrol_new alc861_asus_laptop_mixer[] = { HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("PC Beep Playback Volume", 0x23, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("PC Beep Playback Switch", 0x23, 0x0, HDA_OUTPUT), { } }; @@ -14049,6 +14004,8 @@ static int patch_alc861(struct hda_codec *codec) spec->stream_digital_playback = &alc861_pcm_digital_playback; spec->stream_digital_capture = &alc861_pcm_digital_capture; + set_beep_amp(spec, 0x23, 0, HDA_OUTPUT); + spec->vmaster_nid = 0x03; codec->patch_ops = alc_patch_ops; @@ -14205,9 +14162,6 @@ static struct snd_kcontrol_new alc861vd_6st_mixer[] = { HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), - { } /* end */ }; @@ -14231,9 +14185,6 @@ static struct snd_kcontrol_new alc861vd_3st_mixer[] = { HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), - { } /* end */ }; @@ -14272,8 +14223,6 @@ static struct snd_kcontrol_new alc861vd_dallas_mixer[] = { HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT), HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("PC Beep Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Beep Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -15015,6 +14964,7 @@ static int patch_alc861vd(struct hda_codec *codec) spec->capture_style = CAPT_MIX; set_capture_mixer(spec); + set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); spec->vmaster_nid = 0x02; @@ -15203,8 +15153,6 @@ static struct snd_kcontrol_new alc662_3ST_2ch_mixer[] = { HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -15226,8 +15174,6 @@ static struct snd_kcontrol_new alc662_3ST_6ch_mixer[] = { HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -16832,6 +16778,7 @@ static int patch_alc662(struct hda_codec *codec) if (!spec->cap_mixer) set_capture_mixer(spec); + set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); spec->vmaster_nid = 0x02; -- cgit v1.2.3-70-g09d2 From c8dcdf829ca1827a802eae841dd04de8c9d6653f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 6 Feb 2009 16:21:20 +0100 Subject: ALSA: hda - Add missing NULL check in snd_hda_create_spdif_in_ctls() Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index c9158799ccb..93412f335dc 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1984,6 +1984,8 @@ int snd_hda_create_spdif_in_ctls(struct hda_codec *codec, hda_nid_t nid) } for (dig_mix = dig_in_ctls; dig_mix->name; dig_mix++) { kctl = snd_ctl_new1(dig_mix, codec); + if (!kctl) + return -ENOMEM; kctl->private_value = nid; err = snd_hda_ctl_add(codec, kctl); if (err < 0) -- cgit v1.2.3-70-g09d2 From c44765b8c8bfc883c9868ab7aef37d27b5b14be8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 6 Feb 2009 16:48:10 +0100 Subject: ALSA: hda - Clear codec->beep at release Clear codec->beep field in snd_hda_detach_beep_device() to be sure. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_beep.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 960fd797038..4de5bacd392 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -138,6 +138,7 @@ void snd_hda_detach_beep_device(struct hda_codec *codec) input_unregister_device(beep->dev); kfree(beep); + codec->beep = NULL; } } EXPORT_SYMBOL_HDA(snd_hda_detach_beep_device); -- cgit v1.2.3-70-g09d2 From a4ddeba9c8896cba8c6ce7a98c0b5c755c15a746 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 6 Feb 2009 17:21:09 +0100 Subject: ALSA: hda - Remove superfluous code in patch_realtek.c codec->spec is reset in the caller side. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 76934bc8b48..3d933e307b1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3202,7 +3202,6 @@ static void alc_free(struct hda_codec *codec) alc_free_kctls(codec); kfree(spec); snd_hda_detach_beep_device(codec); - codec->spec = NULL; /* to be sure */ } #ifdef SND_HDA_NEEDS_RESUME -- cgit v1.2.3-70-g09d2 From c5a4bcd0cac546c5d776af881c5e913ba4a9922d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 6 Feb 2009 17:22:05 +0100 Subject: ALSA: hda - Use digital beep for AD codecs Use digital beep instead of analog pc-beep for AD codecs. Create the beep mixer controls dynamically on demand. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 140 +++++++++++++++++++++++++++---------------- 1 file changed, 88 insertions(+), 52 deletions(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 30399cbf819..cc02f2df251 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -27,11 +27,12 @@ #include #include "hda_codec.h" #include "hda_local.h" +#include "hda_beep.h" struct ad198x_spec { struct snd_kcontrol_new *mixers[5]; int num_mixers; - + unsigned int beep_amp; /* beep amp value, set via set_beep_amp() */ const struct hda_verb *init_verbs[5]; /* initialization verbs * don't forget NULL termination! */ @@ -154,6 +155,16 @@ static const char *ad_slave_sws[] = { static void ad198x_free_kctls(struct hda_codec *codec); +/* additional beep mixers; the actual parameters are overwritten at build */ +static struct snd_kcontrol_new ad_beep_mixer[] = { + HDA_CODEC_VOLUME("Beep Playback Volume", 0, 0, HDA_OUTPUT), + HDA_CODEC_MUTE("Beep Playback Switch", 0, 0, HDA_OUTPUT), + { } /* end */ +}; + +#define set_beep_amp(spec, nid, idx, dir) \ + ((spec)->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 1, idx, dir)) /* mono */ + static int ad198x_build_controls(struct hda_codec *codec) { struct ad198x_spec *spec = codec->spec; @@ -181,6 +192,21 @@ static int ad198x_build_controls(struct hda_codec *codec) return err; } + /* create beep controls if needed */ + if (spec->beep_amp) { + struct snd_kcontrol_new *knew; + for (knew = ad_beep_mixer; knew->name; knew++) { + struct snd_kcontrol *kctl; + kctl = snd_ctl_new1(knew, codec); + if (!kctl) + return -ENOMEM; + kctl->private_value = spec->beep_amp; + err = snd_hda_ctl_add(codec, kctl); + if (err < 0) + return err; + } + } + /* if we have no master control, let's create it */ if (!snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) { unsigned int vmaster_tlv[4]; @@ -397,7 +423,8 @@ static void ad198x_free(struct hda_codec *codec) return; ad198x_free_kctls(codec); - kfree(codec->spec); + kfree(spec); + snd_hda_detach_beep_device(codec); } static struct hda_codec_ops ad198x_patch_ops = { @@ -536,8 +563,6 @@ static struct snd_kcontrol_new ad1986a_mixers[] = { HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Mic Boost", 0x0f, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x18, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x18, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Mono Playback Volume", 0x1e, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Mono Playback Switch", 0x1e, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT), @@ -601,8 +626,7 @@ static struct snd_kcontrol_new ad1986a_laptop_mixers[] = { HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Mic Boost", 0x0f, 0x0, HDA_OUTPUT), - /* HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x18, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x18, 0x0, HDA_OUTPUT), + /* HDA_CODEC_VOLUME("Mono Playback Volume", 0x1e, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Mono Playback Switch", 0x1e, 0x0, HDA_OUTPUT), */ HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT), @@ -800,8 +824,6 @@ static struct snd_kcontrol_new ad1986a_laptop_automute_mixers[] = { HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Mic Boost", 0x0f, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x18, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x18, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x12, 0x0, HDA_OUTPUT), { @@ -1026,7 +1048,7 @@ static int is_jack_available(struct hda_codec *codec, hda_nid_t nid) static int patch_ad1986a(struct hda_codec *codec) { struct ad198x_spec *spec; - int board_config; + int err, board_config; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -1034,6 +1056,13 @@ static int patch_ad1986a(struct hda_codec *codec) codec->spec = spec; + err = snd_hda_attach_beep_device(codec, 0x19); + if (err < 0) { + ad198x_free(codec); + return err; + } + set_beep_amp(spec, 0x18, 0, HDA_OUTPUT); + spec->multiout.max_channels = 6; spec->multiout.num_dacs = ARRAY_SIZE(ad1986a_dac_nids); spec->multiout.dac_nids = ad1986a_dac_nids; @@ -1213,8 +1242,6 @@ static struct snd_kcontrol_new ad1983_mixers[] = { HDA_CODEC_MUTE("Mic Playback Switch", 0x12, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Line Playback Volume", 0x13, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Line Playback Switch", 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("PC Speaker Playback Volume", 0x10, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("PC Speaker Playback Switch", 0x10, 1, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Mic Boost", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x15, 0x0, HDA_OUTPUT), @@ -1285,6 +1312,7 @@ static struct hda_amp_list ad1983_loopbacks[] = { static int patch_ad1983(struct hda_codec *codec) { struct ad198x_spec *spec; + int err; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -1292,6 +1320,13 @@ static int patch_ad1983(struct hda_codec *codec) codec->spec = spec; + err = snd_hda_attach_beep_device(codec, 0x10); + if (err < 0) { + ad198x_free(codec); + return err; + } + set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); + spec->multiout.max_channels = 2; spec->multiout.num_dacs = ARRAY_SIZE(ad1983_dac_nids); spec->multiout.dac_nids = ad1983_dac_nids; @@ -1361,8 +1396,6 @@ static struct snd_kcontrol_new ad1981_mixers[] = { HDA_CODEC_MUTE("Mic Playback Switch", 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x1d, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("PC Speaker Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("PC Speaker Playback Switch", 0x0d, 1, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Front Mic Boost", 0x08, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x18, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT), @@ -1685,7 +1718,7 @@ static struct snd_pci_quirk ad1981_cfg_tbl[] = { static int patch_ad1981(struct hda_codec *codec) { struct ad198x_spec *spec; - int board_config; + int err, board_config; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -1693,6 +1726,13 @@ static int patch_ad1981(struct hda_codec *codec) codec->spec = spec; + err = snd_hda_attach_beep_device(codec, 0x10); + if (err < 0) { + ad198x_free(codec); + return err; + } + set_beep_amp(spec, 0x0d, 0, HDA_OUTPUT); + spec->multiout.max_channels = 2; spec->multiout.num_dacs = ARRAY_SIZE(ad1981_dac_nids); spec->multiout.dac_nids = ad1981_dac_nids; @@ -1979,9 +2019,6 @@ static struct snd_kcontrol_new ad1988_6stack_mixers2[] = { HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x4, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x4, HDA_INPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x10, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x10, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT), @@ -2025,9 +2062,6 @@ static struct snd_kcontrol_new ad1988_3stack_mixers2[] = { HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x4, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x4, HDA_INPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x10, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x10, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT), @@ -2057,9 +2091,6 @@ static struct snd_kcontrol_new ad1988_laptop_mixers[] = { HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x1, HDA_INPUT), HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x10, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x10, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT), @@ -2919,7 +2950,7 @@ static struct snd_pci_quirk ad1988_cfg_tbl[] = { static int patch_ad1988(struct hda_codec *codec) { struct ad198x_spec *spec; - int board_config; + int err, board_config; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -2939,7 +2970,7 @@ static int patch_ad1988(struct hda_codec *codec) if (board_config == AD1988_AUTO) { /* automatic parse from the BIOS config */ - int err = ad1988_parse_auto_config(codec); + err = ad1988_parse_auto_config(codec); if (err < 0) { ad198x_free(codec); return err; @@ -2949,6 +2980,13 @@ static int patch_ad1988(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x10); + if (err < 0) { + ad198x_free(codec); + return err; + } + set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); + switch (board_config) { case AD1988_6STACK: case AD1988_6STACK_DIG: @@ -3105,12 +3143,6 @@ static struct snd_kcontrol_new ad1884_base_mixers[] = { HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x01, HDA_INPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x02, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x02, HDA_INPUT), - /* - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x20, 0x03, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x20, 0x03, HDA_INPUT), - HDA_CODEC_VOLUME("Digital Beep Playback Volume", 0x10, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Digital Beep Playback Switch", 0x10, 0x0, HDA_OUTPUT), - */ HDA_CODEC_VOLUME("Mic Boost", 0x15, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Boost", 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), @@ -3219,7 +3251,7 @@ static const char *ad1884_slave_vols[] = { "CD Playback Volume", "Internal Mic Playback Volume", "Docking Mic Playback Volume" - "Beep Playback Volume", + /* "Beep Playback Volume", */ "IEC958 Playback Volume", NULL }; @@ -3227,6 +3259,7 @@ static const char *ad1884_slave_vols[] = { static int patch_ad1884(struct hda_codec *codec) { struct ad198x_spec *spec; + int err; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -3234,6 +3267,13 @@ static int patch_ad1884(struct hda_codec *codec) codec->spec = spec; + err = snd_hda_attach_beep_device(codec, 0x10); + if (err < 0) { + ad198x_free(codec); + return err; + } + set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); + spec->multiout.max_channels = 2; spec->multiout.num_dacs = ARRAY_SIZE(ad1884_dac_nids); spec->multiout.dac_nids = ad1884_dac_nids; @@ -3300,8 +3340,6 @@ static struct snd_kcontrol_new ad1984_thinkpad_mixers[] = { HDA_CODEC_VOLUME("Mic Boost", 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Internal Mic Boost", 0x15, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Docking Mic Boost", 0x25, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT), HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT), @@ -3358,10 +3396,6 @@ static struct snd_kcontrol_new ad1984_dell_desktop_mixers[] = { HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT), HDA_CODEC_VOLUME("Line-In Playback Volume", 0x20, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Line-In Playback Switch", 0x20, 0x01, HDA_INPUT), - /* - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x20, 0x03, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x20, 0x03, HDA_INPUT), - */ HDA_CODEC_VOLUME("Line-In Boost", 0x15, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Boost", 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), @@ -3540,8 +3574,6 @@ static struct snd_kcontrol_new ad1884a_base_mixers[] = { HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x04, HDA_INPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x02, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Boost", 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Line Boost", 0x15, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x25, 0x0, HDA_OUTPUT), @@ -3674,8 +3706,6 @@ static struct snd_kcontrol_new ad1884a_laptop_mixers[] = { HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x20, 0x01, HDA_INPUT), HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x20, 0x04, HDA_INPUT), HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x20, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Internal Mic Boost", 0x15, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Dock Mic Boost", 0x25, 0x0, HDA_OUTPUT), @@ -3703,8 +3733,6 @@ static struct snd_kcontrol_new ad1884a_mobile_mixers[] = { HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT), HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT), HDA_CODEC_VOLUME("Mic Capture Volume", 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Internal Mic Capture Volume", 0x15, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), @@ -3815,8 +3843,6 @@ static struct snd_kcontrol_new ad1984a_thinkpad_mixers[] = { HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x00, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Internal Mic Boost", 0x17, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), @@ -3902,7 +3928,7 @@ static struct snd_pci_quirk ad1884a_cfg_tbl[] = { static int patch_ad1884a(struct hda_codec *codec) { struct ad198x_spec *spec; - int board_config; + int err, board_config; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -3910,6 +3936,13 @@ static int patch_ad1884a(struct hda_codec *codec) codec->spec = spec; + err = snd_hda_attach_beep_device(codec, 0x10); + if (err < 0) { + ad198x_free(codec); + return err; + } + set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); + spec->multiout.max_channels = 2; spec->multiout.num_dacs = ARRAY_SIZE(ad1884a_dac_nids); spec->multiout.dac_nids = ad1884a_dac_nids; @@ -4064,8 +4097,6 @@ static struct snd_kcontrol_new ad1882_loopback_mixers[] = { HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x04, HDA_INPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x06, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x06, HDA_INPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x07, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x07, HDA_INPUT), { } /* end */ }; @@ -4078,8 +4109,6 @@ static struct snd_kcontrol_new ad1882a_loopback_mixers[] = { HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x01, HDA_INPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x06, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x06, HDA_INPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x07, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x07, HDA_INPUT), HDA_CODEC_VOLUME("Digital Mic Boost", 0x1f, 0x0, HDA_INPUT), { } /* end */ }; @@ -4238,7 +4267,7 @@ static const char *ad1882_models[AD1986A_MODELS] = { static int patch_ad1882(struct hda_codec *codec) { struct ad198x_spec *spec; - int board_config; + int err, board_config; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -4246,6 +4275,13 @@ static int patch_ad1882(struct hda_codec *codec) codec->spec = spec; + err = snd_hda_attach_beep_device(codec, 0x10); + if (err < 0) { + ad198x_free(codec); + return err; + } + set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); + spec->multiout.max_channels = 6; spec->multiout.num_dacs = 3; spec->multiout.dac_nids = ad1882_dac_nids; -- cgit v1.2.3-70-g09d2 From cfb9fb5517faa9e61c7e874fc89ef9c9253a0902 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 6 Feb 2009 17:34:03 +0100 Subject: ALSA: hda - Fix unused variable compile warning MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit sound/pci/hda/patch_realtek.c:12693: warning: unused variable ‘i’ Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3d933e307b1..f594a096029 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12690,7 +12690,7 @@ static int alc269_auto_create_analog_input_ctls(struct alc_spec *spec, static int alc269_parse_auto_config(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - int i, err; + int err; static hda_nid_t alc269_ignore[] = { 0x1d, 0 }; err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, -- cgit v1.2.3-70-g09d2 From 8f0dc655f9efa3fc81b8cdaf5aa1f2779f8db46d Mon Sep 17 00:00:00 2001 From: Robert Jarzmik Date: Sat, 7 Feb 2009 14:01:58 +0100 Subject: ASoC: Add initial support of Mitac mioa701 device SoC. This machine driver enables sound functions on Mitac mio a701 smartphone. Build upon ASoC v1, it handles : - rear speaker - front speaker - microphone - GSM A global "Mio Mode" switch is not yet provided to cope with audio path setup. As balance on audio chip line is no more assured, an incorrect setup can produce a lot of heat and even fry the battery behind the wm9713 and the speaker amplifier. It doesn't cope with : - headset jack - mio master mode - master volume control This driver is backported from ASoc v2, and amputated from scenario setups and master volume control. [Minor mods for terminology in comments -- broonie] Signed-off-by: Robert Jarzmik Signed-off-by: Mark Brown --- sound/soc/pxa/Kconfig | 9 ++ sound/soc/pxa/Makefile | 2 + sound/soc/pxa/mioa701_wm9713.c | 250 +++++++++++++++++++++++++++++++++++++++++ 3 files changed, 261 insertions(+) create mode 100644 sound/soc/pxa/mioa701_wm9713.c diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 958ac3fe15d..5998ab366e8 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -115,3 +115,12 @@ config SND_SOC_ZYLONITE help Say Y if you want to add support for SoC audio on the Marvell Zylonite reference platform. + +config SND_PXA2XX_SOC_MIOA701 + tristate "SoC Audio support for MIO A701" + depends on SND_PXA2XX_SOC && MACH_MIOA701 + select SND_PXA2XX_SOC_AC97 + select SND_SOC_WM9713 + help + Say Y if you want to add support for SoC audio on the + MIO A701. diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 97a51a8c936..8ed881c5e5c 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -20,6 +20,7 @@ snd-soc-spitz-objs := spitz.o snd-soc-em-x270-objs := em-x270.o snd-soc-palm27x-objs := palm27x.o snd-soc-zylonite-objs := zylonite.o +snd-soc-mioa701-objs := mioa701_wm9713.o obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o @@ -30,4 +31,5 @@ obj-$(CONFIG_SND_PXA2XX_SOC_E800) += snd-soc-e800.o obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o obj-$(CONFIG_SND_PXA2XX_SOC_EM_X270) += snd-soc-em-x270.o obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o +obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c new file mode 100644 index 00000000000..19eda8bbfda --- /dev/null +++ b/sound/soc/pxa/mioa701_wm9713.c @@ -0,0 +1,250 @@ +/* + * Handles the Mitac mioa701 SoC system + * + * Copyright (C) 2008 Robert Jarzmik + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation in version 2 of the License. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + * This is a little schema of the sound interconnections : + * + * Sagem X200 Wolfson WM9713 + * +--------+ +-------------------+ Rear Speaker + * | | | | /-+ + * | +--->----->---+MONOIN SPKL+--->----+-+ | + * | GSM | | | | | | + * | +--->----->---+PCBEEP SPKR+--->----+-+ | + * | CHIP | | | \-+ + * | +---<-----<---+MONO | + * | | | | Front Speaker + * +--------+ | | /-+ + * | HPL+--->----+-+ | + * | | | | | + * | OUT3+--->----+-+ | + * | | \-+ + * | | + * | | Front Micro + * | | + + * | MIC1+-----<--+o+ + * | | + + * +-------------------+ --- + */ + +#include +#include +#include + +#include +#include + +#include +#include +#include +#include +#include +#include + +#include "pxa2xx-pcm.h" +#include "pxa2xx-ac97.h" +#include "../codecs/wm9713.h" + +#define ARRAY_AND_SIZE(x) (x), ARRAY_SIZE(x) + +#define AC97_GPIO_PULL 0x58 + +/* Use GPIO8 for rear speaker amplifier */ +static int rear_amp_power(struct snd_soc_codec *codec, int power) +{ + unsigned short reg; + + if (power) { + reg = snd_soc_read(codec, AC97_GPIO_CFG); + snd_soc_write(codec, AC97_GPIO_CFG, reg | 0x0100); + reg = snd_soc_read(codec, AC97_GPIO_PULL); + snd_soc_write(codec, AC97_GPIO_PULL, reg | (1<<15)); + } else { + reg = snd_soc_read(codec, AC97_GPIO_CFG); + snd_soc_write(codec, AC97_GPIO_CFG, reg & ~0x0100); + reg = snd_soc_read(codec, AC97_GPIO_PULL); + snd_soc_write(codec, AC97_GPIO_PULL, reg & ~(1<<15)); + } + + return 0; +} + +static int rear_amp_event(struct snd_soc_dapm_widget *widget, + struct snd_kcontrol *kctl, int event) +{ + struct snd_soc_codec *codec = widget->codec; + + return rear_amp_power(codec, SND_SOC_DAPM_EVENT_ON(event)); +} + +/* mioa701 machine dapm widgets */ +static const struct snd_soc_dapm_widget mioa701_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Front Speaker", NULL), + SND_SOC_DAPM_SPK("Rear Speaker", rear_amp_event), + SND_SOC_DAPM_MIC("Headset", NULL), + SND_SOC_DAPM_LINE("GSM Line Out", NULL), + SND_SOC_DAPM_LINE("GSM Line In", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Front Mic", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* Call Mic */ + {"Mic Bias", NULL, "Front Mic"}, + {"MIC1", NULL, "Mic Bias"}, + + /* Headset Mic */ + {"LINEL", NULL, "Headset Mic"}, + {"LINER", NULL, "Headset Mic"}, + + /* GSM Module */ + {"MONOIN", NULL, "GSM Line Out"}, + {"PCBEEP", NULL, "GSM Line Out"}, + {"GSM Line In", NULL, "MONO"}, + + /* headphone connected to HPL, HPR */ + {"Headset", NULL, "HPL"}, + {"Headset", NULL, "HPR"}, + + /* front speaker connected to HPL, OUT3 */ + {"Front Speaker", NULL, "HPL"}, + {"Front Speaker", NULL, "OUT3"}, + + /* rear speaker connected to SPKL, SPKR */ + {"Rear Speaker", NULL, "SPKL"}, + {"Rear Speaker", NULL, "SPKR"}, +}; + +static int mioa701_wm9713_init(struct snd_soc_codec *codec) +{ + unsigned short reg; + + /* Add mioa701 specific widgets */ + snd_soc_dapm_new_controls(codec, ARRAY_AND_SIZE(mioa701_dapm_widgets)); + + /* Set up mioa701 specific audio path audio_mapnects */ + snd_soc_dapm_add_routes(codec, ARRAY_AND_SIZE(audio_map)); + + /* Prepare GPIO8 for rear speaker amplifier */ + reg = codec->read(codec, AC97_GPIO_CFG); + codec->write(codec, AC97_GPIO_CFG, reg | 0x0100); + + /* Prepare MIC input */ + reg = codec->read(codec, AC97_3D_CONTROL); + codec->write(codec, AC97_3D_CONTROL, reg | 0xc000); + + snd_soc_dapm_enable_pin(codec, "Front Speaker"); + snd_soc_dapm_enable_pin(codec, "Rear Speaker"); + snd_soc_dapm_enable_pin(codec, "Front Mic"); + snd_soc_dapm_enable_pin(codec, "GSM Line In"); + snd_soc_dapm_enable_pin(codec, "GSM Line Out"); + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_ops mioa701_ops; + +static struct snd_soc_dai_link mioa701_dai[] = { + { + .name = "AC97", + .stream_name = "AC97 HiFi", + .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI], + .codec_dai = &wm9713_dai[WM9713_DAI_AC97_HIFI], + .init = mioa701_wm9713_init, + .ops = &mioa701_ops, + }, + { + .name = "AC97 Aux", + .stream_name = "AC97 Aux", + .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX], + .codec_dai = &wm9713_dai[WM9713_DAI_AC97_AUX], + .ops = &mioa701_ops, + }, +}; + +static struct snd_soc_card mioa701 = { + .name = "MioA701", + .platform = &pxa2xx_soc_platform, + .dai_link = mioa701_dai, + .num_links = ARRAY_SIZE(mioa701_dai), +}; + +static struct snd_soc_device mioa701_snd_devdata = { + .card = &mioa701, + .codec_dev = &soc_codec_dev_wm9713, +}; + +static struct platform_device *mioa701_snd_device; + +static int mioa701_wm9713_probe(struct platform_device *pdev) +{ + int ret; + + if (!machine_is_mioa701()) + return -ENODEV; + + dev_warn(&pdev->dev, "Be warned that incorrect mixers/muxes setup will" + "lead to overheating and possible destruction of your device." + "Do not use without a good knowledge of mio's board design!\n"); + + mioa701_snd_device = platform_device_alloc("soc-audio", -1); + if (!mioa701_snd_device) + return -ENOMEM; + + platform_set_drvdata(mioa701_snd_device, &mioa701_snd_devdata); + mioa701_snd_devdata.dev = &mioa701_snd_device->dev; + + ret = platform_device_add(mioa701_snd_device); + if (!ret) + return 0; + + platform_device_put(mioa701_snd_device); + return ret; +} + +static int __devexit mioa701_wm9713_remove(struct platform_device *pdev) +{ + platform_device_unregister(mioa701_snd_device); + return 0; +} + +static struct platform_driver mioa701_wm9713_driver = { + .probe = mioa701_wm9713_probe, + .remove = __devexit_p(mioa701_wm9713_remove), + .driver = { + .name = "mioa701-wm9713", + .owner = THIS_MODULE, + }, +}; + +static int __init mioa701_asoc_init(void) +{ + return platform_driver_register(&mioa701_wm9713_driver); +} + +static void __exit mioa701_asoc_exit(void) +{ + platform_driver_unregister(&mioa701_wm9713_driver); +} + +module_init(mioa701_asoc_init); +module_exit(mioa701_asoc_exit); + +/* Module information */ +MODULE_AUTHOR("Robert Jarzmik (rjarzmik@free.fr)"); +MODULE_DESCRIPTION("ALSA SoC WM9713 MIO A701"); +MODULE_LICENSE("GPL"); -- cgit v1.2.3-70-g09d2 From 67137a5d46d5a7c4cbdc66f03d1e2f397fe14b2b Mon Sep 17 00:00:00 2001 From: Roel Kluin Date: Sun, 8 Feb 2009 18:17:37 +0100 Subject: ASoC: count reaches 10001, not 10000. With a postfix increment count reaches 10001, not 10000. Signed-off-by: Mark Brown --- sound/soc/blackfin/bf5xx-ad73311.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ad73311.c b/sound/soc/blackfin/bf5xx-ad73311.c index 7f2a5e19907..edfbdc024e6 100644 --- a/sound/soc/blackfin/bf5xx-ad73311.c +++ b/sound/soc/blackfin/bf5xx-ad73311.c @@ -114,7 +114,7 @@ static int snd_ad73311_configure(void) SSYNC(); /* When TUVF is set, the data is already send out */ - while (!(status & TUVF) && count++ < 10000) { + while (!(status & TUVF) && ++count < 10000) { udelay(1); status = bfin_read_SPORT_STAT(); SSYNC(); @@ -123,7 +123,7 @@ static int snd_ad73311_configure(void) SSYNC(); local_irq_enable(); - if (count == 10000) { + if (count >= 10000) { printk(KERN_ERR "ad73311: failed to configure codec\n"); return -1; } -- cgit v1.2.3-70-g09d2 From 4e7f78f815412fd25b207b8c63a698b637c9621d Mon Sep 17 00:00:00 2001 From: Philipp Zabel Date: Thu, 5 Feb 2009 17:48:19 +0100 Subject: pxa/h5000: Setup I2S pins for pxa2xx-i2s The iPAQ h5000 has an AK4535 codec connected as I2S slave, PXA I2S providing SYSCLK. Signed-off-by: Philipp Zabel Acked-by: Eric Miao Signed-off-by: Mark Brown --- arch/arm/mach-pxa/h5000.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/arch/arm/mach-pxa/h5000.c b/arch/arm/mach-pxa/h5000.c index da6e4422c0f..295ec413d80 100644 --- a/arch/arm/mach-pxa/h5000.c +++ b/arch/arm/mach-pxa/h5000.c @@ -153,6 +153,13 @@ static unsigned long h5000_pin_config[] __initdata = { GPIO23_SSP1_SCLK, GPIO25_SSP1_TXD, GPIO26_SSP1_RXD, + + /* I2S */ + GPIO28_I2S_BITCLK_OUT, + GPIO29_I2S_SDATA_IN, + GPIO30_I2S_SDATA_OUT, + GPIO31_I2S_SYNC, + GPIO32_I2S_SYSCLK, }; /* -- cgit v1.2.3-70-g09d2 From 772885c1dc42f1992ac4fc937f1ed4ae9d42a31e Mon Sep 17 00:00:00 2001 From: Philipp Zabel Date: Thu, 5 Feb 2009 17:48:20 +0100 Subject: pxa/spitz: Setup I2S pins for pxa2xx-i2s The spitz has a WM8750 codec connected as I2S slave but doesn't use the PXA I2S system clock. Signed-off-by: Philipp Zabel Acked-by: Eric Miao Signed-off-by: Mark Brown --- arch/arm/mach-pxa/spitz.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/arch/arm/mach-pxa/spitz.c b/arch/arm/mach-pxa/spitz.c index 6d447c9ce8a..0d62d311d41 100644 --- a/arch/arm/mach-pxa/spitz.c +++ b/arch/arm/mach-pxa/spitz.c @@ -105,6 +105,12 @@ static unsigned long spitz_pin_config[] __initdata = { GPIO57_nIOIS16, GPIO104_PSKTSEL, + /* I2S */ + GPIO28_I2S_BITCLK_OUT, + GPIO29_I2S_SDATA_IN, + GPIO30_I2S_SDATA_OUT, + GPIO31_I2S_SYNC, + /* MMC */ GPIO32_MMC_CLK, GPIO112_MMC_CMD, -- cgit v1.2.3-70-g09d2 From 44dd2b9168350b82a671ce71666b99208ab2d973 Mon Sep 17 00:00:00 2001 From: Philipp Zabel Date: Thu, 5 Feb 2009 17:48:21 +0100 Subject: ASoC: pxa2xx-i2s: remove I2S pin setup This removes the calls to pxa_gpio_mode from the pxa2xx-i2s driver. Pin setup should be done during board init via pxa2xx_mfp_config instead. Signed-off-by: Philipp Zabel Acked-by: Eric Miao Signed-off-by: Mark Brown --- sound/soc/pxa/pxa2xx-i2s.c | 36 ------------------------------------ 1 file changed, 36 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 517991fb109..83b59d7fe96 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -25,20 +25,11 @@ #include #include -#include #include #include "pxa2xx-pcm.h" #include "pxa2xx-i2s.h" -struct pxa2xx_gpio { - u32 sys; - u32 rx; - u32 tx; - u32 clk; - u32 frm; -}; - /* * I2S Controller Register and Bit Definitions */ @@ -106,21 +97,6 @@ static struct pxa2xx_pcm_dma_params pxa2xx_i2s_pcm_stereo_in = { DCMD_BURST32 | DCMD_WIDTH4, }; -static struct pxa2xx_gpio gpio_bus[] = { - { /* I2S SoC Slave */ - .rx = GPIO29_SDATA_IN_I2S_MD, - .tx = GPIO30_SDATA_OUT_I2S_MD, - .clk = GPIO28_BITCLK_IN_I2S_MD, - .frm = GPIO31_SYNC_I2S_MD, - }, - { /* I2S SoC Master */ - .rx = GPIO29_SDATA_IN_I2S_MD, - .tx = GPIO30_SDATA_OUT_I2S_MD, - .clk = GPIO28_BITCLK_OUT_I2S_MD, - .frm = GPIO31_SYNC_I2S_MD, - }, -}; - static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -181,9 +157,6 @@ static int pxa2xx_i2s_set_dai_sysclk(struct snd_soc_dai *cpu_dai, if (clk_id != PXA2XX_I2S_SYSCLK) return -ENODEV; - if (pxa_i2s.master && dir == SND_SOC_CLOCK_OUT) - pxa_gpio_mode(gpio_bus[pxa_i2s.master].sys); - return 0; } @@ -194,10 +167,6 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - pxa_gpio_mode(gpio_bus[pxa_i2s.master].rx); - pxa_gpio_mode(gpio_bus[pxa_i2s.master].tx); - pxa_gpio_mode(gpio_bus[pxa_i2s.master].frm); - pxa_gpio_mode(gpio_bus[pxa_i2s.master].clk); BUG_ON(IS_ERR(clk_i2s)); clk_enable(clk_i2s); pxa_i2s_wait(); @@ -398,11 +367,6 @@ static struct platform_driver pxa2xx_i2s_driver = { static int __init pxa2xx_i2s_init(void) { - if (cpu_is_pxa27x()) - gpio_bus[1].sys = GPIO113_I2S_SYSCLK_MD; - else - gpio_bus[1].sys = GPIO32_SYSCLK_I2S_MD; - clk_i2s = ERR_PTR(-ENOENT); return platform_driver_register(&pxa2xx_i2s_driver); } -- cgit v1.2.3-70-g09d2 From 8663ae55f39e99c25242adb6242a191258a4eca1 Mon Sep 17 00:00:00 2001 From: Herton Ronaldo Krzesinski Date: Sun, 8 Feb 2009 19:50:34 -0200 Subject: ALSA: hda - Bind new ecs mobo id (1019:2950) to model=ecs202 This adds a new sound quirk entry (model=ecs202) for an ecs motherboard with IDT STAC9221 codec (1019:2950). Signed-off-by: Herton Ronaldo Krzesinski Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 85dc642d113..d16d5c60eec 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2108,6 +2108,8 @@ static struct snd_pci_quirk stac922x_cfg_tbl[] = { "ECS/PC chips", STAC_ECS_202), SND_PCI_QUIRK(0x1019, 0x2820, "ECS/PC chips", STAC_ECS_202), + SND_PCI_QUIRK(0x1019, 0x2950, + "ECS/PC chips", STAC_ECS_202), {} /* terminator */ }; -- cgit v1.2.3-70-g09d2 From 23c7b521c250b261dd97a7a06d5a2e74b56233d5 Mon Sep 17 00:00:00 2001 From: Herton Ronaldo Krzesinski Date: Sun, 8 Feb 2009 19:51:28 -0200 Subject: ALSA: hda - Don't touch non-existent port f on 4-port 92hd71bxx codecs When checking for input amps on pins 0x0a, 0x0d and 0x0f, and initializing them for 92hd71xxx codec models, we must skip nid 0x0f for 4-port models too like with 5-port models, as it is unused (nid 0x0f is vendor reserved in 4-port models). Signed-off-by: Herton Ronaldo Krzesinski Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index d16d5c60eec..2f4e090b055 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -5072,6 +5072,8 @@ again: switch (codec->vendor_id) { case 0x111d76b6: /* 4 Port without Analog Mixer */ case 0x111d76b7: + unmute_init++; + /* fallthru */ case 0x111d76b4: /* 6 Port without Analog Mixer */ case 0x111d76b5: memcpy(&spec->private_dimux, &stac92hd71bxx_dmux_nomixer, -- cgit v1.2.3-70-g09d2 From 8bd4bb7a35e8ebb015a531218614c48e10a3c4ee Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 30 Jan 2009 17:27:45 +0100 Subject: ALSA: Add subdevice_mask field to quirk entries Introduced a new field, subdevice_mask, which specifies the bitmask to match with the given subdevice ID. Signed-off-by: Takashi Iwai --- include/sound/core.h | 16 ++++++++++++++-- sound/core/misc.c | 10 ++++++---- 2 files changed, 20 insertions(+), 6 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index f632484bc74..f67952a61a2 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -446,21 +446,33 @@ static inline int __snd_bug_on(int cond) struct snd_pci_quirk { unsigned short subvendor; /* PCI subvendor ID */ unsigned short subdevice; /* PCI subdevice ID */ + unsigned short subdevice_mask; /* bitmask to match */ int value; /* value */ #ifdef CONFIG_SND_DEBUG_VERBOSE const char *name; /* name of the device (optional) */ #endif }; -#define _SND_PCI_QUIRK_ID(vend,dev) \ - .subvendor = (vend), .subdevice = (dev) +#define _SND_PCI_QUIRK_ID_MASK(vend, mask, dev) \ + .subvendor = (vend), .subdevice = (dev), .subdevice_mask = (mask) +#define _SND_PCI_QUIRK_ID(vend, dev) \ + _SND_PCI_QUIRK_ID_MASK(vend, 0xffff, dev) #define SND_PCI_QUIRK_ID(vend,dev) {_SND_PCI_QUIRK_ID(vend, dev)} #ifdef CONFIG_SND_DEBUG_VERBOSE #define SND_PCI_QUIRK(vend,dev,xname,val) \ {_SND_PCI_QUIRK_ID(vend, dev), .value = (val), .name = (xname)} +#define SND_PCI_QUIRK_VENDOR(vend, xname, val) \ + {_SND_PCI_QUIRK_ID_MASK(vend, 0, 0), .value = (val), .name = (xname)} +#define SND_PCI_QUIRK_MASK(vend, mask, dev, xname, val) \ + {_SND_PCI_QUIRK_ID_MASK(vend, mask, dev), \ + .value = (val), .name = (xname)} #else #define SND_PCI_QUIRK(vend,dev,xname,val) \ {_SND_PCI_QUIRK_ID(vend, dev), .value = (val)} +#define SND_PCI_QUIRK_MASK(vend, mask, dev, xname, val) \ + {_SND_PCI_QUIRK_ID_MASK(vend, mask, dev), .value = (val)} +#define SND_PCI_QUIRK_VENDOR(vend, xname, val) \ + {_SND_PCI_QUIRK_ID_MASK(vend, 0, 0), .value = (val)} #endif const struct snd_pci_quirk * diff --git a/sound/core/misc.c b/sound/core/misc.c index 38524f615d9..a9710e0c97a 100644 --- a/sound/core/misc.c +++ b/sound/core/misc.c @@ -95,12 +95,14 @@ snd_pci_quirk_lookup(struct pci_dev *pci, const struct snd_pci_quirk *list) { const struct snd_pci_quirk *q; - for (q = list; q->subvendor; q++) - if (q->subvendor == pci->subsystem_vendor && - (!q->subdevice || q->subdevice == pci->subsystem_device)) + for (q = list; q->subvendor; q++) { + if (q->subvendor != pci->subsystem_vendor) + continue; + if (!q->subdevice || + (pci->subsystem_device & q->subdevice_mask) == q->subdevice) return q; + } return NULL; } - EXPORT_SYMBOL(snd_pci_quirk_lookup); #endif -- cgit v1.2.3-70-g09d2 From dea0a5095b5e21306a81c496567043798fac7815 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 9 Feb 2009 17:14:52 +0100 Subject: ALSA: hda - Clean up quirk lists Clean up quirk lists with bit masks. Also, sorted in numerical order for alc662_cfg_tbl[]. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 10 ++--- sound/pci/hda/patch_conexant.c | 20 +++------ sound/pci/hda/patch_realtek.c | 97 ++++++++++++++++++++---------------------- sound/pci/hda/patch_sigmatel.c | 61 ++++---------------------- 4 files changed, 65 insertions(+), 123 deletions(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index cc02f2df251..6106dfe8ec0 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1015,10 +1015,8 @@ static struct snd_pci_quirk ad1986a_cfg_tbl[] = { SND_PCI_QUIRK(0x1179, 0xff40, "Toshiba", AD1986A_LAPTOP_EAPD), SND_PCI_QUIRK(0x144d, 0xb03c, "Samsung R55", AD1986A_3STACK), SND_PCI_QUIRK(0x144d, 0xc01e, "FSC V2060", AD1986A_LAPTOP), - SND_PCI_QUIRK(0x144d, 0xc023, "Samsung X60", AD1986A_SAMSUNG), - SND_PCI_QUIRK(0x144d, 0xc024, "Samsung R65", AD1986A_SAMSUNG), - SND_PCI_QUIRK(0x144d, 0xc026, "Samsung X11", AD1986A_SAMSUNG), SND_PCI_QUIRK(0x144d, 0xc027, "Samsung Q1", AD1986A_ULTRA), + SND_PCI_QUIRK_MASK(0x144d, 0xff00, 0xc000, "Samsung", AD1986A_SAMSUNG), SND_PCI_QUIRK(0x144d, 0xc504, "Samsung Q35", AD1986A_3STACK), SND_PCI_QUIRK(0x17aa, 0x1011, "Lenovo M55", AD1986A_LAPTOP), SND_PCI_QUIRK(0x17aa, 0x1017, "Lenovo A60", AD1986A_3STACK), @@ -1706,10 +1704,10 @@ static struct snd_pci_quirk ad1981_cfg_tbl[] = { SND_PCI_QUIRK(0x1014, 0x0597, "Lenovo Z60", AD1981_THINKPAD), SND_PCI_QUIRK(0x1014, 0x05b7, "Lenovo Z60m", AD1981_THINKPAD), /* All HP models */ - SND_PCI_QUIRK(0x103c, 0, "HP nx", AD1981_HP), + SND_PCI_QUIRK_VENDOR(0x103c, "HP nx", AD1981_HP), SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba U205", AD1981_TOSHIBA), /* Lenovo Thinkpad T60/X60/Z6xx */ - SND_PCI_QUIRK(0x17aa, 0, "Lenovo Thinkpad", AD1981_THINKPAD), + SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo Thinkpad", AD1981_THINKPAD), /* HP nx6320 (reversed SSID, H/W bug) */ SND_PCI_QUIRK(0x30b0, 0x103c, "HP nx6320", AD1981_HP), {} @@ -3481,7 +3479,7 @@ static const char *ad1984_models[AD1984_MODELS] = { static struct snd_pci_quirk ad1984_cfg_tbl[] = { /* Lenovo Thinkpad T61/X61 */ - SND_PCI_QUIRK(0x17aa, 0, "Lenovo Thinkpad", AD1984_THINKPAD), + SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo Thinkpad", AD1984_THINKPAD), SND_PCI_QUIRK(0x1028, 0x0214, "Dell T3400", AD1984_DELL_DESKTOP), {} }; diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 0177ef8f4c9..fdf876be712 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1002,15 +1002,9 @@ static const char *cxt5045_models[CXT5045_MODELS] = { }; static struct snd_pci_quirk cxt5045_cfg_tbl[] = { - SND_PCI_QUIRK(0x103c, 0x30a5, "HP", CXT5045_LAPTOP_HPSENSE), - SND_PCI_QUIRK(0x103c, 0x30b2, "HP DV Series", CXT5045_LAPTOP_HPSENSE), - SND_PCI_QUIRK(0x103c, 0x30b5, "HP DV2120", CXT5045_LAPTOP_HPSENSE), - SND_PCI_QUIRK(0x103c, 0x30b7, "HP DV6000Z", CXT5045_LAPTOP_HPSENSE), - SND_PCI_QUIRK(0x103c, 0x30bb, "HP DV8000", CXT5045_LAPTOP_HPSENSE), - SND_PCI_QUIRK(0x103c, 0x30cd, "HP DV Series", CXT5045_LAPTOP_HPSENSE), - SND_PCI_QUIRK(0x103c, 0x30cf, "HP DV9533EG", CXT5045_LAPTOP_HPSENSE), SND_PCI_QUIRK(0x103c, 0x30d5, "HP 530", CXT5045_LAPTOP_HP530), - SND_PCI_QUIRK(0x103c, 0x30d9, "HP Spartan", CXT5045_LAPTOP_HPSENSE), + SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3000, "HP DV Series", + CXT5045_LAPTOP_HPSENSE), SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba P105", CXT5045_LAPTOP_MICSENSE), SND_PCI_QUIRK(0x152d, 0x0753, "Benq R55E", CXT5045_BENQ), SND_PCI_QUIRK(0x1734, 0x10ad, "Fujitsu Si1520", CXT5045_LAPTOP_MICSENSE), @@ -1020,8 +1014,8 @@ static struct snd_pci_quirk cxt5045_cfg_tbl[] = { SND_PCI_QUIRK(0x1509, 0x1e40, "FIC", CXT5045_LAPTOP_HPMICSENSE), SND_PCI_QUIRK(0x1509, 0x2f05, "FIC", CXT5045_LAPTOP_HPMICSENSE), SND_PCI_QUIRK(0x1509, 0x2f06, "FIC", CXT5045_LAPTOP_HPMICSENSE), - SND_PCI_QUIRK(0x1631, 0xc106, "Packard Bell", CXT5045_LAPTOP_HPMICSENSE), - SND_PCI_QUIRK(0x1631, 0xc107, "Packard Bell", CXT5045_LAPTOP_HPMICSENSE), + SND_PCI_QUIRK_MASK(0x1631, 0xff00, 0xc100, "Packard Bell", + CXT5045_LAPTOP_HPMICSENSE), SND_PCI_QUIRK(0x8086, 0x2111, "Conexant Reference board", CXT5045_LAPTOP_HPSENSE), {} }; @@ -1571,11 +1565,9 @@ static const char *cxt5047_models[CXT5047_MODELS] = { }; static struct snd_pci_quirk cxt5047_cfg_tbl[] = { - SND_PCI_QUIRK(0x103c, 0x30a0, "HP DV1000", CXT5047_LAPTOP), SND_PCI_QUIRK(0x103c, 0x30a5, "HP DV5200T/DV8000T", CXT5047_LAPTOP_HP), - SND_PCI_QUIRK(0x103c, 0x30b2, "HP DV2000T/DV3000T", CXT5047_LAPTOP), - SND_PCI_QUIRK(0x103c, 0x30b5, "HP DV2000Z", CXT5047_LAPTOP), - SND_PCI_QUIRK(0x103c, 0x30cf, "HP DV6700", CXT5047_LAPTOP), + SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3000, "HP DV Series", + CXT5047_LAPTOP), SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba P100", CXT5047_LAPTOP_EAPD), {} }; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f594a096029..7ae8fad0189 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3598,7 +3598,7 @@ static struct snd_pci_quirk alc880_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x8181, "ASUS P4GPL", ALC880_ASUS_DIG), SND_PCI_QUIRK(0x1043, 0x8196, "ASUS P5GD1", ALC880_6ST), SND_PCI_QUIRK(0x1043, 0x81b4, "ASUS", ALC880_6ST), - SND_PCI_QUIRK(0x1043, 0, "ASUS", ALC880_ASUS), /* default ASUS */ + SND_PCI_QUIRK_VENDOR(0x1043, "ASUS", ALC880_ASUS), /* default ASUS */ SND_PCI_QUIRK(0x104d, 0x81a0, "Sony", ALC880_3ST), SND_PCI_QUIRK(0x104d, 0x81d6, "Sony", ALC880_3ST), SND_PCI_QUIRK(0x107b, 0x3032, "Gateway", ALC880_5ST), @@ -3641,7 +3641,8 @@ static struct snd_pci_quirk alc880_cfg_tbl[] = { SND_PCI_QUIRK(0x8086, 0xe400, "Intel mobo", ALC880_5ST_DIG), SND_PCI_QUIRK(0x8086, 0xe401, "Intel mobo", ALC880_5ST_DIG), SND_PCI_QUIRK(0x8086, 0xe402, "Intel mobo", ALC880_5ST_DIG), - SND_PCI_QUIRK(0x8086, 0, "Intel mobo", ALC880_3ST), /* default Intel */ + /* default Intel */ + SND_PCI_QUIRK_VENDOR(0x8086, "Intel mobo", ALC880_3ST), SND_PCI_QUIRK(0xa0a0, 0x0560, "AOpen i915GMm-HFS", ALC880_5ST_DIG), SND_PCI_QUIRK(0xe803, 0x1019, NULL, ALC880_6ST_DIG), {} @@ -8521,7 +8522,8 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { ALC888_ACER_ASPIRE_4930G), SND_PCI_QUIRK(0x1025, 0x015e, "Acer Aspire 6930G", ALC888_ACER_ASPIRE_4930G), - SND_PCI_QUIRK(0x1025, 0, "Acer laptop", ALC883_ACER), /* default Acer */ + /* default Acer */ + SND_PCI_QUIRK_VENDOR(0x1025, "Acer laptop", ALC883_ACER), SND_PCI_QUIRK(0x1028, 0x020d, "Dell Inspiron 530", ALC888_6ST_DELL), SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavillion", ALC883_6ST_DIG), SND_PCI_QUIRK(0x103c, 0x2a4f, "HP Samba", ALC888_3ST_HP), @@ -8566,7 +8568,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x147b, 0x1083, "Abit IP35-PRO", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1558, 0x0721, "Clevo laptop M720R", ALC883_CLEVO_M720), SND_PCI_QUIRK(0x1558, 0x0722, "Clevo laptop M720SR", ALC883_CLEVO_M720), - SND_PCI_QUIRK(0x1558, 0, "Clevo laptop", ALC883_LAPTOP_EAPD), + SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC883_LAPTOP_EAPD), SND_PCI_QUIRK(0x15d9, 0x8780, "Supermicro PDSBA", ALC883_3ST_6ch), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_MEDION), SND_PCI_QUIRK(0x1734, 0x1107, "FSC AMILO Xi2550", @@ -10707,14 +10709,10 @@ static const char *alc262_models[ALC262_MODEL_LAST] = { static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x1002, 0x437b, "Hippo", ALC262_HIPPO), SND_PCI_QUIRK(0x1033, 0x8895, "NEC Versa S9100", ALC262_NEC), - SND_PCI_QUIRK(0x103c, 0x12fe, "HP xw9400", ALC262_HP_BPC), - SND_PCI_QUIRK(0x103c, 0x12ff, "HP xw4550", ALC262_HP_BPC), - SND_PCI_QUIRK(0x103c, 0x1306, "HP xw8600", ALC262_HP_BPC), - SND_PCI_QUIRK(0x103c, 0x1307, "HP xw6600", ALC262_HP_BPC), - SND_PCI_QUIRK(0x103c, 0x1308, "HP xw4600", ALC262_HP_BPC), - SND_PCI_QUIRK(0x103c, 0x1309, "HP xw4*00", ALC262_HP_BPC), - SND_PCI_QUIRK(0x103c, 0x130a, "HP xw6*00", ALC262_HP_BPC), - SND_PCI_QUIRK(0x103c, 0x130b, "HP xw8*00", ALC262_HP_BPC), + SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1200, "HP xw series", + ALC262_HP_BPC), + SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1300, "HP xw series", + ALC262_HP_BPC), SND_PCI_QUIRK(0x103c, 0x2800, "HP D7000", ALC262_HP_BPC_D7000_WL), SND_PCI_QUIRK(0x103c, 0x2801, "HP D7000", ALC262_HP_BPC_D7000_WF), SND_PCI_QUIRK(0x103c, 0x2802, "HP D7000", ALC262_HP_BPC_D7000_WL), @@ -10742,8 +10740,8 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FUJITSU), SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FUJITSU), SND_PCI_QUIRK(0x10f1, 0x2915, "Tyan Thunder n6650W", ALC262_TYAN), - SND_PCI_QUIRK(0x144d, 0xc032, "Samsung Q1 Ultra", ALC262_ULTRA), - SND_PCI_QUIRK(0x144d, 0xc039, "Samsung Q1U EL", ALC262_ULTRA), + SND_PCI_QUIRK_MASK(0x144d, 0xff00, 0xc032, "Samsung Q1", + ALC262_ULTRA), SND_PCI_QUIRK(0x144d, 0xc510, "Samsung Q45", ALC262_HIPPO), SND_PCI_QUIRK(0x17aa, 0x384e, "Lenovo 3000 y410", ALC262_LENOVO_3000), SND_PCI_QUIRK(0x17ff, 0x0560, "Benq ED8", ALC262_BENQ_ED8), @@ -14534,9 +14532,7 @@ static struct snd_pci_quirk alc861vd_cfg_tbl[] = { SND_PCI_QUIRK(0x1179, 0xff03, "Toshiba P205", ALC861VD_LENOVO), SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba L30-149", ALC861VD_DALLAS), SND_PCI_QUIRK(0x1565, 0x820d, "Biostar NF61S SE", ALC861VD_6ST_DIG), - SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo", ALC861VD_LENOVO), - SND_PCI_QUIRK(0x17aa, 0x3802, "Lenovo 3000 C200", ALC861VD_LENOVO), - SND_PCI_QUIRK(0x17aa, 0x384e, "Lenovo 3000 N200", ALC861VD_LENOVO), + SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", ALC861VD_LENOVO), SND_PCI_QUIRK(0x1849, 0x0862, "ASRock K8NF6G-VSTA", ALC861VD_6ST_DIG), {} }; @@ -16150,56 +16146,55 @@ static const char *alc662_models[ALC662_MODEL_LAST] = { }; static struct snd_pci_quirk alc662_cfg_tbl[] = { - SND_PCI_QUIRK(0x1043, 0x1878, "ASUS M51VA", ALC663_ASUS_M51VA), - SND_PCI_QUIRK(0x1043, 0x19a3, "ASUS G50V", ALC663_ASUS_G50V), - SND_PCI_QUIRK(0x1043, 0x8290, "ASUS P5GC-MX", ALC662_3ST_6ch_DIG), - SND_PCI_QUIRK(0x1043, 0x82a1, "ASUS Eeepc", ALC662_ASUS_EEEPC_P701), - SND_PCI_QUIRK(0x1043, 0x82d1, "ASUS Eeepc EP20", ALC662_ASUS_EEEPC_EP20), - SND_PCI_QUIRK(0x1043, 0x1903, "ASUS F5GL", ALC663_ASUS_MODE1), - SND_PCI_QUIRK(0x1043, 0x1878, "ASUS M50Vr", ALC663_ASUS_MODE1), + SND_PCI_QUIRK(0x1019, 0x9087, "ECS", ALC662_ECS), SND_PCI_QUIRK(0x1043, 0x1000, "ASUS N50Vm", ALC663_ASUS_MODE1), - SND_PCI_QUIRK(0x1043, 0x19b3, "ASUS F7Z", ALC663_ASUS_MODE1), - SND_PCI_QUIRK(0x1043, 0x1953, "ASUS NB", ALC663_ASUS_MODE1), - SND_PCI_QUIRK(0x1043, 0x19a3, "ASUS NB", ALC663_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x1092, "ASUS NB", ALC663_ASUS_MODE3), + SND_PCI_QUIRK(0x1043, 0x11c3, "ASUS M70V", ALC663_ASUS_MODE3), SND_PCI_QUIRK(0x1043, 0x11d3, "ASUS NB", ALC663_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x11f3, "ASUS NB", ALC662_ASUS_MODE2), SND_PCI_QUIRK(0x1043, 0x1203, "ASUS NB", ALC663_ASUS_MODE1), - SND_PCI_QUIRK(0x1043, 0x19e3, "ASUS NB", ALC663_ASUS_MODE1), - SND_PCI_QUIRK(0x1043, 0x1993, "ASUS N20", ALC663_ASUS_MODE1), - SND_PCI_QUIRK(0x1043, 0x19c3, "ASUS F5Z/F6x", ALC662_ASUS_MODE2), SND_PCI_QUIRK(0x1043, 0x1339, "ASUS NB", ALC662_ASUS_MODE2), - SND_PCI_QUIRK(0x1043, 0x1913, "ASUS NB", ALC662_ASUS_MODE2), - SND_PCI_QUIRK(0x1043, 0x1843, "ASUS NB", ALC662_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x16c3, "ASUS NB", ALC662_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x1753, "ASUS NB", ALC662_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x1763, "ASUS NB", ALC663_ASUS_MODE6), + SND_PCI_QUIRK(0x1043, 0x1765, "ASUS NB", ALC663_ASUS_MODE6), + SND_PCI_QUIRK(0x1043, 0x1783, "ASUS NB", ALC662_ASUS_MODE2), SND_PCI_QUIRK(0x1043, 0x1813, "ASUS NB", ALC662_ASUS_MODE2), - SND_PCI_QUIRK(0x1043, 0x11f3, "ASUS NB", ALC662_ASUS_MODE2), - SND_PCI_QUIRK(0x1043, 0x1876, "ASUS NB", ALC662_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x1823, "ASUS NB", ALC663_ASUS_MODE5), + SND_PCI_QUIRK(0x1043, 0x1833, "ASUS NB", ALC663_ASUS_MODE6), + SND_PCI_QUIRK(0x1043, 0x1843, "ASUS NB", ALC662_ASUS_MODE2), SND_PCI_QUIRK(0x1043, 0x1864, "ASUS NB", ALC662_ASUS_MODE2), - SND_PCI_QUIRK(0x1043, 0x1783, "ASUS NB", ALC662_ASUS_MODE2), - SND_PCI_QUIRK(0x1043, 0x1753, "ASUS NB", ALC662_ASUS_MODE2), - SND_PCI_QUIRK(0x1043, 0x16c3, "ASUS NB", ALC662_ASUS_MODE2), - SND_PCI_QUIRK(0x1043, 0x1933, "ASUS F80Q", ALC662_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x1876, "ASUS NB", ALC662_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x1878, "ASUS M51VA", ALC663_ASUS_M51VA), + /*SND_PCI_QUIRK(0x1043, 0x1878, "ASUS M50Vr", ALC663_ASUS_MODE1),*/ SND_PCI_QUIRK(0x1043, 0x1893, "ASUS M50Vm", ALC663_ASUS_MODE3), - SND_PCI_QUIRK(0x1043, 0x11c3, "ASUS M70V", ALC663_ASUS_MODE3), - SND_PCI_QUIRK(0x1043, 0x1963, "ASUS X71C", ALC663_ASUS_MODE3), SND_PCI_QUIRK(0x1043, 0x1894, "ASUS X55", ALC663_ASUS_MODE3), - SND_PCI_QUIRK(0x1043, 0x1092, "ASUS NB", ALC663_ASUS_MODE3), + SND_PCI_QUIRK(0x1043, 0x1903, "ASUS F5GL", ALC663_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x1913, "ASUS NB", ALC662_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x1933, "ASUS F80Q", ALC662_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x1953, "ASUS NB", ALC663_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x1963, "ASUS X71C", ALC663_ASUS_MODE3), + SND_PCI_QUIRK(0x1043, 0x1993, "ASUS N20", ALC663_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x19a3, "ASUS G50V", ALC663_ASUS_G50V), + /*SND_PCI_QUIRK(0x1043, 0x19a3, "ASUS NB", ALC663_ASUS_MODE1),*/ + SND_PCI_QUIRK(0x1043, 0x19b3, "ASUS F7Z", ALC663_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x19c3, "ASUS F5Z/F6x", ALC662_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x19e3, "ASUS NB", ALC663_ASUS_MODE1), SND_PCI_QUIRK(0x1043, 0x19f3, "ASUS NB", ALC663_ASUS_MODE4), - SND_PCI_QUIRK(0x1043, 0x1823, "ASUS NB", ALC663_ASUS_MODE5), - SND_PCI_QUIRK(0x1043, 0x1833, "ASUS NB", ALC663_ASUS_MODE6), - SND_PCI_QUIRK(0x1043, 0x1763, "ASUS NB", ALC663_ASUS_MODE6), - SND_PCI_QUIRK(0x1043, 0x1765, "ASUS NB", ALC663_ASUS_MODE6), + SND_PCI_QUIRK(0x1043, 0x8290, "ASUS P5GC-MX", ALC662_3ST_6ch_DIG), + SND_PCI_QUIRK(0x1043, 0x82a1, "ASUS Eeepc", ALC662_ASUS_EEEPC_P701), + SND_PCI_QUIRK(0x1043, 0x82d1, "ASUS Eeepc EP20", ALC662_ASUS_EEEPC_EP20), + SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_ECS), SND_PCI_QUIRK(0x105b, 0x0d47, "Foxconn 45CMX/45GMX/45CMX-K", ALC662_3ST_6ch_DIG), - SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo", ALC662_LENOVO_101E), - SND_PCI_QUIRK(0x1019, 0x9087, "ECS", ALC662_ECS), - SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_ECS), SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L", ALC662_3ST_6ch_DIG), SND_PCI_QUIRK(0x1565, 0x820f, "Biostar TA780G M2+", ALC662_3ST_6ch_DIG), + SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo", ALC662_LENOVO_101E), SND_PCI_QUIRK(0x1849, 0x3662, "ASROCK K10N78FullHD-hSLI R3.0", ALC662_3ST_6ch_DIG), - SND_PCI_QUIRK(0x1854, 0x2000, "ASUS H13-2000", ALC663_ASUS_H13), - SND_PCI_QUIRK(0x1854, 0x2001, "ASUS H13-2001", ALC663_ASUS_H13), - SND_PCI_QUIRK(0x1854, 0x2002, "ASUS H13-2002", ALC663_ASUS_H13), + SND_PCI_QUIRK_MASK(0x1854, 0xf000, 0x2000, "ASUS H13-200x", + ALC663_ASUS_H13), {} }; diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 2f4e090b055..12b30884843 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2082,33 +2082,7 @@ static struct snd_pci_quirk stac922x_cfg_tbl[] = { SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01d7, "Dell XPS M1210", STAC_922X_DELL_M82), /* ECS/PC Chips boards */ - SND_PCI_QUIRK(0x1019, 0x2144, - "ECS/PC chips", STAC_ECS_202), - SND_PCI_QUIRK(0x1019, 0x2608, - "ECS/PC chips", STAC_ECS_202), - SND_PCI_QUIRK(0x1019, 0x2633, - "ECS/PC chips P17G/1333", STAC_ECS_202), - SND_PCI_QUIRK(0x1019, 0x2811, - "ECS/PC chips", STAC_ECS_202), - SND_PCI_QUIRK(0x1019, 0x2812, - "ECS/PC chips", STAC_ECS_202), - SND_PCI_QUIRK(0x1019, 0x2813, - "ECS/PC chips", STAC_ECS_202), - SND_PCI_QUIRK(0x1019, 0x2814, - "ECS/PC chips", STAC_ECS_202), - SND_PCI_QUIRK(0x1019, 0x2815, - "ECS/PC chips", STAC_ECS_202), - SND_PCI_QUIRK(0x1019, 0x2816, - "ECS/PC chips", STAC_ECS_202), - SND_PCI_QUIRK(0x1019, 0x2817, - "ECS/PC chips", STAC_ECS_202), - SND_PCI_QUIRK(0x1019, 0x2818, - "ECS/PC chips", STAC_ECS_202), - SND_PCI_QUIRK(0x1019, 0x2819, - "ECS/PC chips", STAC_ECS_202), - SND_PCI_QUIRK(0x1019, 0x2820, - "ECS/PC chips", STAC_ECS_202), - SND_PCI_QUIRK(0x1019, 0x2950, + SND_PCI_QUIRK_MASK(0x1019, 0xf000, 0x2000, "ECS/PC chips", STAC_ECS_202), {} /* terminator */ }; @@ -2169,22 +2143,10 @@ static struct snd_pci_quirk stac927x_cfg_tbl[] = { SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x3d01, "Intel D946", STAC_D965_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0xa301, "Intel D946", STAC_D965_3ST), /* 965 based 3 stack systems */ - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2116, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2115, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2114, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2113, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2112, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2111, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2110, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2009, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2008, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2007, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2006, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2005, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2004, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2003, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2002, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2001, "Intel D965", STAC_D965_3ST), + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_INTEL, 0xff00, 0x2100, + "Intel D965", STAC_D965_3ST), + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_INTEL, 0xff00, 0x2000, + "Intel D965", STAC_D965_3ST), /* Dell 3 stack systems */ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f7, "Dell XPS M1730", STAC_DELL_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01dd, "Dell Dimension E520", STAC_DELL_3ST), @@ -2200,15 +2162,10 @@ static struct snd_pci_quirk stac927x_cfg_tbl[] = { SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02ff, "Dell ", STAC_DELL_BIOS), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0209, "Dell XPS 1330", STAC_DELL_BIOS), /* 965 based 5 stack systems */ - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2301, "Intel D965", STAC_D965_5ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2302, "Intel D965", STAC_D965_5ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2303, "Intel D965", STAC_D965_5ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2304, "Intel D965", STAC_D965_5ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2305, "Intel D965", STAC_D965_5ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2501, "Intel D965", STAC_D965_5ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2502, "Intel D965", STAC_D965_5ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2503, "Intel D965", STAC_D965_5ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2504, "Intel D965", STAC_D965_5ST), + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_INTEL, 0xff00, 0x2300, + "Intel D965", STAC_D965_5ST), + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_INTEL, 0xff00, 0x2500, + "Intel D965", STAC_D965_5ST), {} /* terminator */ }; -- cgit v1.2.3-70-g09d2 From a85165c66c5640c37b67a94aa4e00fe45273bca1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 9 Feb 2009 17:15:50 +0100 Subject: ALSA: via82xx - Clean up quirk list Use SND_PCI_QUIRK_VENDOR() macro. Signed-off-by: Takashi Iwai --- sound/pci/via82xx.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index fc62d6380f8..a027896a220 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -2363,14 +2363,14 @@ static struct snd_pci_quirk dxs_whitelist[] __devinitdata = { SND_PCI_QUIRK(0x1019, 0x0996, "ESC Mobo", VIA_DXS_48K), SND_PCI_QUIRK(0x1019, 0x0a81, "ECS K7VTA3 v8.0", VIA_DXS_NO_VRA), SND_PCI_QUIRK(0x1019, 0x0a85, "ECS L7VMM2", VIA_DXS_NO_VRA), - SND_PCI_QUIRK(0x1019, 0, "ESC K8", VIA_DXS_SRC), + SND_PCI_QUIRK_VENDOR(0x1019, "ESC K8", VIA_DXS_SRC), SND_PCI_QUIRK(0x1019, 0xaa01, "ESC K8T890-A", VIA_DXS_SRC), SND_PCI_QUIRK(0x1025, 0x0033, "Acer Inspire 1353LM", VIA_DXS_NO_VRA), SND_PCI_QUIRK(0x1025, 0x0046, "Acer Aspire 1524 WLMi", VIA_DXS_SRC), - SND_PCI_QUIRK(0x1043, 0, "ASUS A7/A8", VIA_DXS_NO_VRA), - SND_PCI_QUIRK(0x1071, 0, "Diverse Notebook", VIA_DXS_NO_VRA), + SND_PCI_QUIRK_VENDOR(0x1043, "ASUS A7/A8", VIA_DXS_NO_VRA), + SND_PCI_QUIRK_VENDOR(0x1071, "Diverse Notebook", VIA_DXS_NO_VRA), SND_PCI_QUIRK(0x10cf, 0x118e, "FSC Laptop", VIA_DXS_ENABLE), - SND_PCI_QUIRK(0x1106, 0, "ASRock", VIA_DXS_SRC), + SND_PCI_QUIRK_VENDOR(0x1106, "ASRock", VIA_DXS_SRC), SND_PCI_QUIRK(0x1297, 0xa231, "Shuttle AK31v2", VIA_DXS_SRC), SND_PCI_QUIRK(0x1297, 0xa232, "Shuttle", VIA_DXS_SRC), SND_PCI_QUIRK(0x1297, 0xc160, "Shuttle Sk41G", VIA_DXS_SRC), @@ -2378,7 +2378,7 @@ static struct snd_pci_quirk dxs_whitelist[] __devinitdata = { SND_PCI_QUIRK(0x1462, 0x3800, "MSI KT266", VIA_DXS_ENABLE), SND_PCI_QUIRK(0x1462, 0x7120, "MSI KT4V", VIA_DXS_ENABLE), SND_PCI_QUIRK(0x1462, 0x7142, "MSI K8MM-V", VIA_DXS_ENABLE), - SND_PCI_QUIRK(0x1462, 0, "MSI Mobo", VIA_DXS_SRC), + SND_PCI_QUIRK_VENDOR(0x1462, "MSI Mobo", VIA_DXS_SRC), SND_PCI_QUIRK(0x147b, 0x1401, "ABIT KD7(-RAID)", VIA_DXS_ENABLE), SND_PCI_QUIRK(0x147b, 0x1411, "ABIT VA-20", VIA_DXS_ENABLE), SND_PCI_QUIRK(0x147b, 0x1413, "ABIT KV8 Pro", VIA_DXS_ENABLE), @@ -2392,11 +2392,11 @@ static struct snd_pci_quirk dxs_whitelist[] __devinitdata = { SND_PCI_QUIRK(0x161f, 0x2032, "m680x machines", VIA_DXS_48K), SND_PCI_QUIRK(0x1631, 0xe004, "PB EasyNote 3174", VIA_DXS_ENABLE), SND_PCI_QUIRK(0x1695, 0x3005, "EPoX EP-8K9A", VIA_DXS_ENABLE), - SND_PCI_QUIRK(0x1695, 0, "EPoX mobo", VIA_DXS_SRC), - SND_PCI_QUIRK(0x16f3, 0, "Jetway K8", VIA_DXS_SRC), - SND_PCI_QUIRK(0x1734, 0, "FSC Laptop", VIA_DXS_SRC), + SND_PCI_QUIRK_VENDOR(0x1695, "EPoX mobo", VIA_DXS_SRC), + SND_PCI_QUIRK_VENDOR(0x16f3, "Jetway K8", VIA_DXS_SRC), + SND_PCI_QUIRK_VENDOR(0x1734, "FSC Laptop", VIA_DXS_SRC), SND_PCI_QUIRK(0x1849, 0x3059, "ASRock K7VM2", VIA_DXS_NO_VRA), - SND_PCI_QUIRK(0x1849, 0, "ASRock mobo", VIA_DXS_SRC), + SND_PCI_QUIRK_VENDOR(0x1849, "ASRock mobo", VIA_DXS_SRC), SND_PCI_QUIRK(0x1919, 0x200a, "Soltek SL-K8", VIA_DXS_NO_VRA), SND_PCI_QUIRK(0x4005, 0x4710, "MSI K7T266", VIA_DXS_SRC), { } /* terminator */ -- cgit v1.2.3-70-g09d2 From b93f74f604c53b546fced33d11aee722560de249 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Mon, 9 Feb 2009 14:27:06 +0200 Subject: ASoC: TLV320AIC3X: Fix volume ranges This is a minor fix but helps to define dB ranges for volume controls. Only DAC digital volume has full register value range from 0 to 127 but ADC PGA gain and output stage volume controls don't. For ADC PGA, maximum value is 119 and then it saturates to the same gain value of 59.5 dB. For output stages, value 117 corresponds to -78.3 dB and is muted for values 118 and above. Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 38 +++++++++++++++++++------------------- 1 file changed, 19 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index ac73e692a99..cdb6dec14e1 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -255,51 +255,51 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { SOC_DOUBLE_R("PCM Playback Volume", LDAC_VOL, RDAC_VOL, 0, 0x7f, 1), SOC_DOUBLE_R("Line DAC Playback Volume", DACL1_2_LLOPM_VOL, - DACR1_2_RLOPM_VOL, 0, 0x7f, 1), + DACR1_2_RLOPM_VOL, 0, 118, 1), SOC_SINGLE("LineL Playback Switch", LLOPM_CTRL, 3, 0x01, 0), SOC_SINGLE("LineR Playback Switch", RLOPM_CTRL, 3, 0x01, 0), SOC_DOUBLE_R("LineL DAC Playback Volume", DACL1_2_LLOPM_VOL, - DACR1_2_LLOPM_VOL, 0, 0x7f, 1), + DACR1_2_LLOPM_VOL, 0, 118, 1), SOC_SINGLE("LineL Left PGA Bypass Playback Volume", PGAL_2_LLOPM_VOL, - 0, 0x7f, 1), + 0, 118, 1), SOC_SINGLE("LineR Right PGA Bypass Playback Volume", PGAR_2_RLOPM_VOL, - 0, 0x7f, 1), + 0, 118, 1), SOC_DOUBLE_R("LineL Line2 Bypass Playback Volume", LINE2L_2_LLOPM_VOL, - LINE2R_2_LLOPM_VOL, 0, 0x7f, 1), + LINE2R_2_LLOPM_VOL, 0, 118, 1), SOC_DOUBLE_R("LineR Line2 Bypass Playback Volume", LINE2L_2_RLOPM_VOL, - LINE2R_2_RLOPM_VOL, 0, 0x7f, 1), + LINE2R_2_RLOPM_VOL, 0, 118, 1), SOC_DOUBLE_R("Mono DAC Playback Volume", DACL1_2_MONOLOPM_VOL, - DACR1_2_MONOLOPM_VOL, 0, 0x7f, 1), + DACR1_2_MONOLOPM_VOL, 0, 118, 1), SOC_SINGLE("Mono DAC Playback Switch", MONOLOPM_CTRL, 3, 0x01, 0), SOC_DOUBLE_R("Mono PGA Bypass Playback Volume", PGAL_2_MONOLOPM_VOL, - PGAR_2_MONOLOPM_VOL, 0, 0x7f, 1), + PGAR_2_MONOLOPM_VOL, 0, 118, 1), SOC_DOUBLE_R("Mono Line2 Bypass Playback Volume", LINE2L_2_MONOLOPM_VOL, - LINE2R_2_MONOLOPM_VOL, 0, 0x7f, 1), + LINE2R_2_MONOLOPM_VOL, 0, 118, 1), SOC_DOUBLE_R("HP DAC Playback Volume", DACL1_2_HPLOUT_VOL, - DACR1_2_HPROUT_VOL, 0, 0x7f, 1), + DACR1_2_HPROUT_VOL, 0, 118, 1), SOC_DOUBLE_R("HP DAC Playback Switch", HPLOUT_CTRL, HPROUT_CTRL, 3, 0x01, 0), SOC_DOUBLE_R("HP Right PGA Bypass Playback Volume", PGAR_2_HPLOUT_VOL, - PGAR_2_HPROUT_VOL, 0, 0x7f, 1), + PGAR_2_HPROUT_VOL, 0, 118, 1), SOC_SINGLE("HPL PGA Bypass Playback Volume", PGAL_2_HPLOUT_VOL, - 0, 0x7f, 1), + 0, 118, 1), SOC_SINGLE("HPR PGA Bypass Playback Volume", PGAL_2_HPROUT_VOL, - 0, 0x7f, 1), + 0, 118, 1), SOC_DOUBLE_R("HP Line2 Bypass Playback Volume", LINE2L_2_HPLOUT_VOL, - LINE2R_2_HPROUT_VOL, 0, 0x7f, 1), + LINE2R_2_HPROUT_VOL, 0, 118, 1), SOC_DOUBLE_R("HPCOM DAC Playback Volume", DACL1_2_HPLCOM_VOL, - DACR1_2_HPRCOM_VOL, 0, 0x7f, 1), + DACR1_2_HPRCOM_VOL, 0, 118, 1), SOC_DOUBLE_R("HPCOM DAC Playback Switch", HPLCOM_CTRL, HPRCOM_CTRL, 3, 0x01, 0), SOC_SINGLE("HPLCOM PGA Bypass Playback Volume", PGAL_2_HPLCOM_VOL, - 0, 0x7f, 1), + 0, 118, 1), SOC_SINGLE("HPRCOM PGA Bypass Playback Volume", PGAL_2_HPRCOM_VOL, - 0, 0x7f, 1), + 0, 118, 1), SOC_DOUBLE_R("HPCOM Line2 Bypass Playback Volume", LINE2L_2_HPLCOM_VOL, - LINE2R_2_HPRCOM_VOL, 0, 0x7f, 1), + LINE2R_2_HPRCOM_VOL, 0, 118, 1), /* * Note: enable Automatic input Gain Controller with care. It can @@ -308,7 +308,7 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { SOC_DOUBLE_R("AGC Switch", LAGC_CTRL_A, RAGC_CTRL_A, 7, 0x01, 0), /* Input */ - SOC_DOUBLE_R("PGA Capture Volume", LADC_VOL, RADC_VOL, 0, 0x7f, 0), + SOC_DOUBLE_R("PGA Capture Volume", LADC_VOL, RADC_VOL, 0, 119, 0), SOC_DOUBLE_R("PGA Capture Switch", LADC_VOL, RADC_VOL, 7, 0x01, 1), SOC_ENUM("ADC HPF Cut-off", aic3x_enum[ADC_HPF_ENUM]), -- cgit v1.2.3-70-g09d2 From 7565fc38cc8c3a2742d63e957199d70a82d2faf1 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Mon, 9 Feb 2009 14:27:07 +0200 Subject: ASoC: TLV320AIC3X: Add TLV information for volume controls TLV320AIC3X volume controls are logarithmic. Export their dB ranges. Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 114 ++++++++++++++++++++++++++--------------- 1 file changed, 73 insertions(+), 41 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index cdb6dec14e1..d638e3f0728 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -45,6 +45,7 @@ #include #include #include +#include #include "tlv320aic3x.h" @@ -250,56 +251,86 @@ static const struct soc_enum aic3x_enum[] = { SOC_ENUM_DOUBLE(AIC3X_CODEC_DFILT_CTRL, 6, 4, 4, aic3x_adc_hpf), }; +/* + * DAC digital volumes. From -63.5 to 0 dB in 0.5 dB steps + */ +static DECLARE_TLV_DB_SCALE(dac_tlv, -6350, 50, 0); +/* ADC PGA gain volumes. From 0 to 59.5 dB in 0.5 dB steps */ +static DECLARE_TLV_DB_SCALE(adc_tlv, 0, 50, 0); +/* + * Output stage volumes. From -78.3 to 0 dB. Muted below -78.3 dB. + * Step size is approximately 0.5 dB over most of the scale but increasing + * near the very low levels. + * Define dB scale so that it is mostly correct for range about -55 to 0 dB + * but having increasing dB difference below that (and where it doesn't count + * so much). This setting shows -50 dB (actual is -50.3 dB) for register + * value 100 and -58.5 dB (actual is -78.3 dB) for register value 117. + */ +static DECLARE_TLV_DB_SCALE(output_stage_tlv, -5900, 50, 1); + static const struct snd_kcontrol_new aic3x_snd_controls[] = { /* Output */ - SOC_DOUBLE_R("PCM Playback Volume", LDAC_VOL, RDAC_VOL, 0, 0x7f, 1), + SOC_DOUBLE_R_TLV("PCM Playback Volume", + LDAC_VOL, RDAC_VOL, 0, 0x7f, 1, dac_tlv), - SOC_DOUBLE_R("Line DAC Playback Volume", DACL1_2_LLOPM_VOL, - DACR1_2_RLOPM_VOL, 0, 118, 1), + SOC_DOUBLE_R_TLV("Line DAC Playback Volume", + DACL1_2_LLOPM_VOL, DACR1_2_RLOPM_VOL, + 0, 118, 1, output_stage_tlv), SOC_SINGLE("LineL Playback Switch", LLOPM_CTRL, 3, 0x01, 0), SOC_SINGLE("LineR Playback Switch", RLOPM_CTRL, 3, 0x01, 0), - SOC_DOUBLE_R("LineL DAC Playback Volume", DACL1_2_LLOPM_VOL, - DACR1_2_LLOPM_VOL, 0, 118, 1), - SOC_SINGLE("LineL Left PGA Bypass Playback Volume", PGAL_2_LLOPM_VOL, - 0, 118, 1), - SOC_SINGLE("LineR Right PGA Bypass Playback Volume", PGAR_2_RLOPM_VOL, - 0, 118, 1), - SOC_DOUBLE_R("LineL Line2 Bypass Playback Volume", LINE2L_2_LLOPM_VOL, - LINE2R_2_LLOPM_VOL, 0, 118, 1), - SOC_DOUBLE_R("LineR Line2 Bypass Playback Volume", LINE2L_2_RLOPM_VOL, - LINE2R_2_RLOPM_VOL, 0, 118, 1), - - SOC_DOUBLE_R("Mono DAC Playback Volume", DACL1_2_MONOLOPM_VOL, - DACR1_2_MONOLOPM_VOL, 0, 118, 1), + SOC_DOUBLE_R_TLV("LineL DAC Playback Volume", + DACL1_2_LLOPM_VOL, DACR1_2_LLOPM_VOL, + 0, 118, 1, output_stage_tlv), + SOC_SINGLE_TLV("LineL Left PGA Bypass Playback Volume", + PGAL_2_LLOPM_VOL, 0, 118, 1, output_stage_tlv), + SOC_SINGLE_TLV("LineR Right PGA Bypass Playback Volume", + PGAR_2_RLOPM_VOL, 0, 118, 1, output_stage_tlv), + SOC_DOUBLE_R_TLV("LineL Line2 Bypass Playback Volume", + LINE2L_2_LLOPM_VOL, LINE2R_2_LLOPM_VOL, + 0, 118, 1, output_stage_tlv), + SOC_DOUBLE_R_TLV("LineR Line2 Bypass Playback Volume", + LINE2L_2_RLOPM_VOL, LINE2R_2_RLOPM_VOL, + 0, 118, 1, output_stage_tlv), + + SOC_DOUBLE_R_TLV("Mono DAC Playback Volume", + DACL1_2_MONOLOPM_VOL, DACR1_2_MONOLOPM_VOL, + 0, 118, 1, output_stage_tlv), SOC_SINGLE("Mono DAC Playback Switch", MONOLOPM_CTRL, 3, 0x01, 0), - SOC_DOUBLE_R("Mono PGA Bypass Playback Volume", PGAL_2_MONOLOPM_VOL, - PGAR_2_MONOLOPM_VOL, 0, 118, 1), - SOC_DOUBLE_R("Mono Line2 Bypass Playback Volume", LINE2L_2_MONOLOPM_VOL, - LINE2R_2_MONOLOPM_VOL, 0, 118, 1), - - SOC_DOUBLE_R("HP DAC Playback Volume", DACL1_2_HPLOUT_VOL, - DACR1_2_HPROUT_VOL, 0, 118, 1), + SOC_DOUBLE_R_TLV("Mono PGA Bypass Playback Volume", + PGAL_2_MONOLOPM_VOL, PGAR_2_MONOLOPM_VOL, + 0, 118, 1, output_stage_tlv), + SOC_DOUBLE_R_TLV("Mono Line2 Bypass Playback Volume", + LINE2L_2_MONOLOPM_VOL, LINE2R_2_MONOLOPM_VOL, + 0, 118, 1, output_stage_tlv), + + SOC_DOUBLE_R_TLV("HP DAC Playback Volume", + DACL1_2_HPLOUT_VOL, DACR1_2_HPROUT_VOL, + 0, 118, 1, output_stage_tlv), SOC_DOUBLE_R("HP DAC Playback Switch", HPLOUT_CTRL, HPROUT_CTRL, 3, 0x01, 0), - SOC_DOUBLE_R("HP Right PGA Bypass Playback Volume", PGAR_2_HPLOUT_VOL, - PGAR_2_HPROUT_VOL, 0, 118, 1), - SOC_SINGLE("HPL PGA Bypass Playback Volume", PGAL_2_HPLOUT_VOL, - 0, 118, 1), - SOC_SINGLE("HPR PGA Bypass Playback Volume", PGAL_2_HPROUT_VOL, - 0, 118, 1), - SOC_DOUBLE_R("HP Line2 Bypass Playback Volume", LINE2L_2_HPLOUT_VOL, - LINE2R_2_HPROUT_VOL, 0, 118, 1), - - SOC_DOUBLE_R("HPCOM DAC Playback Volume", DACL1_2_HPLCOM_VOL, - DACR1_2_HPRCOM_VOL, 0, 118, 1), + SOC_DOUBLE_R_TLV("HP Right PGA Bypass Playback Volume", + PGAR_2_HPLOUT_VOL, PGAR_2_HPROUT_VOL, + 0, 118, 1, output_stage_tlv), + SOC_SINGLE_TLV("HPL PGA Bypass Playback Volume", + PGAL_2_HPLOUT_VOL, 0, 118, 1, output_stage_tlv), + SOC_SINGLE_TLV("HPR PGA Bypass Playback Volume", + PGAL_2_HPROUT_VOL, 0, 118, 1, output_stage_tlv), + SOC_DOUBLE_R_TLV("HP Line2 Bypass Playback Volume", + LINE2L_2_HPLOUT_VOL, LINE2R_2_HPROUT_VOL, + 0, 118, 1, output_stage_tlv), + + SOC_DOUBLE_R_TLV("HPCOM DAC Playback Volume", + DACL1_2_HPLCOM_VOL, DACR1_2_HPRCOM_VOL, + 0, 118, 1, output_stage_tlv), SOC_DOUBLE_R("HPCOM DAC Playback Switch", HPLCOM_CTRL, HPRCOM_CTRL, 3, 0x01, 0), - SOC_SINGLE("HPLCOM PGA Bypass Playback Volume", PGAL_2_HPLCOM_VOL, - 0, 118, 1), - SOC_SINGLE("HPRCOM PGA Bypass Playback Volume", PGAL_2_HPRCOM_VOL, - 0, 118, 1), - SOC_DOUBLE_R("HPCOM Line2 Bypass Playback Volume", LINE2L_2_HPLCOM_VOL, - LINE2R_2_HPRCOM_VOL, 0, 118, 1), + SOC_SINGLE_TLV("HPLCOM PGA Bypass Playback Volume", + PGAL_2_HPLCOM_VOL, 0, 118, 1, output_stage_tlv), + SOC_SINGLE_TLV("HPRCOM PGA Bypass Playback Volume", + PGAL_2_HPRCOM_VOL, 0, 118, 1, output_stage_tlv), + SOC_DOUBLE_R_TLV("HPCOM Line2 Bypass Playback Volume", + LINE2L_2_HPLCOM_VOL, LINE2R_2_HPRCOM_VOL, + 0, 118, 1, output_stage_tlv), /* * Note: enable Automatic input Gain Controller with care. It can @@ -308,7 +339,8 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { SOC_DOUBLE_R("AGC Switch", LAGC_CTRL_A, RAGC_CTRL_A, 7, 0x01, 0), /* Input */ - SOC_DOUBLE_R("PGA Capture Volume", LADC_VOL, RADC_VOL, 0, 119, 0), + SOC_DOUBLE_R_TLV("PGA Capture Volume", LADC_VOL, RADC_VOL, + 0, 119, 0, adc_tlv), SOC_DOUBLE_R("PGA Capture Switch", LADC_VOL, RADC_VOL, 7, 0x01, 1), SOC_ENUM("ADC HPF Cut-off", aic3x_enum[ADC_HPF_ENUM]), -- cgit v1.2.3-70-g09d2 From 22971e3a77f193579be525a12f3ab91dbf241517 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Feb 2009 11:56:44 +0100 Subject: ALSA: hda - add digital beep support for ALC268 Added the digital beep support for ALC268. It was missing in the last patches. However, ALC268 has a strange pin use for widget 0x1d, which could be used as another purpose. So, the patch adds a check of the beep control before creating the hook for input layer. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 28 ++++++++++++++++++++++------ 1 file changed, 22 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7ae8fad0189..97eaf3b1d97 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -11885,7 +11885,7 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = { static struct alc_config_preset alc268_presets[] = { [ALC267_QUANTA_IL1] = { - .mixers = { alc267_quanta_il1_mixer }, + .mixers = { alc267_quanta_il1_mixer, alc268_beep_mixer }, .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, alc267_quanta_il1_verbs }, .num_dacs = ARRAY_SIZE(alc268_dac_nids), @@ -11967,7 +11967,8 @@ static struct alc_config_preset alc268_presets[] = { }, [ALC268_ACER_ASPIRE_ONE] = { .mixers = { alc268_acer_aspire_one_mixer, - alc268_capture_alt_mixer }, + alc268_beep_mixer, + alc268_capture_alt_mixer }, .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, alc268_acer_aspire_one_verbs }, .num_dacs = ARRAY_SIZE(alc268_dac_nids), @@ -12036,7 +12037,7 @@ static int patch_alc268(struct hda_codec *codec) { struct alc_spec *spec; int board_config; - int err; + int i, has_beep, err; spec = kcalloc(1, sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -12091,13 +12092,28 @@ static int patch_alc268(struct hda_codec *codec) spec->stream_digital_playback = &alc268_pcm_digital_playback; - if (!query_amp_caps(codec, 0x1d, HDA_INPUT)) - /* override the amp caps for beep generator */ - snd_hda_override_amp_caps(codec, 0x1d, HDA_INPUT, + has_beep = 0; + for (i = 0; i < spec->num_mixers; i++) { + if (spec->mixers[i] == alc268_beep_mixer) { + has_beep = 1; + break; + } + } + + if (has_beep) { + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } + if (!query_amp_caps(codec, 0x1d, HDA_INPUT)) + /* override the amp caps for beep generator */ + snd_hda_override_amp_caps(codec, 0x1d, HDA_INPUT, (0x0c << AC_AMPCAP_OFFSET_SHIFT) | (0x0c << AC_AMPCAP_NUM_STEPS_SHIFT) | (0x07 << AC_AMPCAP_STEP_SIZE_SHIFT) | (0 << AC_AMPCAP_MUTE_SHIFT)); + } if (!spec->adc_nids && spec->input_mux) { /* check whether NID 0x07 is valid */ -- cgit v1.2.3-70-g09d2 From 9e30d7718bb7402c7bdee631ad2aae2658c324f0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 11 Feb 2009 08:28:04 +0100 Subject: ASoC: Fix forgotten replacements of socdev->codec The snd_soc_codec was moved into socdev->card, but this change wasn't applied in some places. Fixed now. Signed-off-by: Takashi Iwai --- sound/soc/omap/n810.c | 2 +- sound/soc/pxa/corgi.c | 2 +- sound/soc/pxa/palm27x.c | 2 +- sound/soc/pxa/poodle.c | 2 +- sound/soc/pxa/spitz.c | 2 +- sound/soc/pxa/tosa.c | 2 +- 6 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 25593fee912..9f037cd0191 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -72,7 +72,7 @@ static int n810_startup(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->socdev->codec; + struct snd_soc_codec *codec = rtd->socdev->card->codec; snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2); diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 1ba25a55952..0d41be33d57 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -100,7 +100,7 @@ static void corgi_ext_control(struct snd_soc_codec *codec) static int corgi_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->socdev->codec; + struct snd_soc_codec *codec = rtd->socdev->card->codec; /* check the jack status at stream startup */ corgi_ext_control(codec); diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c index 4a9cf3083af..29958cd9dae 100644 --- a/sound/soc/pxa/palm27x.c +++ b/sound/soc/pxa/palm27x.c @@ -55,7 +55,7 @@ static void palm27x_ext_control(struct snd_soc_codec *codec) static int palm27x_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->socdev->codec; + struct snd_soc_codec *codec = rtd->socdev->card->codec; /* check the jack status at stream startup */ palm27x_ext_control(codec); diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index 6e9827189ff..3a62d4354ef 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -77,7 +77,7 @@ static void poodle_ext_control(struct snd_soc_codec *codec) static int poodle_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->socdev->codec; + struct snd_soc_codec *codec = rtd->socdev->card->codec; /* check the jack status at stream startup */ poodle_ext_control(codec); diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index a3b9e6bdf97..1aafd8c645a 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -109,7 +109,7 @@ static void spitz_ext_control(struct snd_soc_codec *codec) static int spitz_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->socdev->codec; + struct snd_soc_codec *codec = rtd->socdev->card->codec; /* check the jack status at stream startup */ spitz_ext_control(codec); diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index c77194f74c9..09b5bada03b 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -82,7 +82,7 @@ static void tosa_ext_control(struct snd_soc_codec *codec) static int tosa_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->socdev->codec; + struct snd_soc_codec *codec = rtd->socdev->card->codec; /* check the jack status at stream startup */ tosa_ext_control(codec); -- cgit v1.2.3-70-g09d2 From 32d2c7fa1344ddf51886eddf31e228d139501dc6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 11 Feb 2009 11:33:13 +0100 Subject: ALSA: hda - Fix a wrong pin check in snd_hda_parse_pin_def_config() Fixed a wrong pin check (a typo) for debug print of digital input pin. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 93412f335dc..95f10aec7a0 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3551,7 +3551,7 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, cfg->input_pins[AUTO_PIN_FRONT_LINE], cfg->input_pins[AUTO_PIN_CD], cfg->input_pins[AUTO_PIN_AUX]); - if (cfg->dig_out_pin) + if (cfg->dig_in_pin) snd_printd(" dig-in=0x%x\n", cfg->dig_in_pin); return 0; -- cgit v1.2.3-70-g09d2 From 1afa6e2e1d26d0b9d96785ee1823bf11c4c5f202 Mon Sep 17 00:00:00 2001 From: Roel Kluin Date: Wed, 11 Feb 2009 13:53:26 +0100 Subject: sound: OSS: dmabuf: too many loops loop adev->dmap_out->nbufs times Signed-off-by: Roel Kluin Signed-off-by: Takashi Iwai --- sound/oss/dmabuf.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/oss/dmabuf.c b/sound/oss/dmabuf.c index 1e90d769b62..1bfcf7e8854 100644 --- a/sound/oss/dmabuf.c +++ b/sound/oss/dmabuf.c @@ -439,7 +439,7 @@ int DMAbuf_sync(int dev) DMAbuf_launch_output(dev, dmap); adev->dmap_out->flags |= DMA_SYNCING; adev->dmap_out->underrun_count = 0; - while (!signal_pending(current) && n++ <= adev->dmap_out->nbufs && + while (!signal_pending(current) && n++ < adev->dmap_out->nbufs && adev->dmap_out->qlen && adev->dmap_out->underrun_count == 0) { long t = dmabuf_timeout(dmap); spin_unlock_irqrestore(&dmap->lock,flags); -- cgit v1.2.3-70-g09d2 From 0852d7a654f75d22a3c09fd7da4a3551bbb37740 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 11 Feb 2009 11:35:15 +0100 Subject: ALSA: hda - Detect multiple digital-out pins Detect multiple digital-out pins in snd_hda_parse_pin_defconfig(). The dig_out_pin and dig_out_type fields become arrays. The codec parser still doesn't use this multiple pins detection, though. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 17 ++++++++++------- sound/pci/hda/hda_local.h | 5 +++-- sound/pci/hda/patch_analog.c | 2 +- sound/pci/hda/patch_realtek.c | 20 ++++++++++---------- sound/pci/hda/patch_sigmatel.c | 10 +++++----- sound/pci/hda/patch_via.c | 10 +++++----- 6 files changed, 34 insertions(+), 30 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 95f10aec7a0..29eeb748561 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3423,11 +3423,13 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, break; case AC_JACK_SPDIF_OUT: case AC_JACK_DIG_OTHER_OUT: - cfg->dig_out_pin = nid; - if (loc == AC_JACK_LOC_HDMI) - cfg->dig_out_type = HDA_PCM_TYPE_HDMI; - else - cfg->dig_out_type = HDA_PCM_TYPE_SPDIF; + if (cfg->dig_outs >= ARRAY_SIZE(cfg->dig_out_pins)) + continue; + cfg->dig_out_pins[cfg->dig_outs] = nid; + cfg->dig_out_type[cfg->dig_outs] = + (loc == AC_JACK_LOC_HDMI) ? + HDA_PCM_TYPE_HDMI : HDA_PCM_TYPE_SPDIF; + cfg->dig_outs++; break; case AC_JACK_SPDIF_IN: case AC_JACK_DIG_OTHER_IN: @@ -3541,8 +3543,9 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, cfg->hp_pins[1], cfg->hp_pins[2], cfg->hp_pins[3], cfg->hp_pins[4]); snd_printd(" mono: mono_out=0x%x\n", cfg->mono_out_pin); - if (cfg->dig_out_pin) - snd_printd(" dig-out=0x%x\n", cfg->dig_out_pin); + if (cfg->dig_outs) + snd_printd(" dig-out=0x%x/0x%x\n", + cfg->dig_out_pins[0], cfg->dig_out_pins[1]); snd_printd(" inputs: mic=0x%x, fmic=0x%x, line=0x%x, fline=0x%x," " cd=0x%x, aux=0x%x\n", cfg->input_pins[AUTO_PIN_MIC], diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 4086491ed33..2ae6b53a462 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -355,10 +355,11 @@ struct auto_pin_cfg { int line_out_type; /* AUTO_PIN_XXX_OUT */ hda_nid_t hp_pins[AUTO_CFG_MAX_OUTS]; hda_nid_t input_pins[AUTO_PIN_LAST]; - hda_nid_t dig_out_pin; + int dig_outs; + hda_nid_t dig_out_pins[2]; hda_nid_t dig_in_pin; hda_nid_t mono_out_pin; - int dig_out_type; /* HDA_PCM_TYPE_XXX */ + int dig_out_type[2]; /* HDA_PCM_TYPE_XXX */ int dig_in_type; /* HDA_PCM_TYPE_XXX */ }; diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 6106dfe8ec0..d58c32b5b43 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -2898,7 +2898,7 @@ static int ad1988_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = AD1988_SPDIF_OUT; if (spec->autocfg.dig_in_pin) spec->dig_in_nid = AD1988_SPDIF_IN; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1db99df7950..e46251bceb9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4291,7 +4291,7 @@ static int alc880_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = ALC880_DIGOUT_NID; if (spec->autocfg.dig_in_pin) spec->dig_in_nid = ALC880_DIGIN_NID; @@ -5658,7 +5658,7 @@ static int alc260_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = 2; - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = ALC260_DIGOUT_NID; if (spec->kctls.list) add_mixer(spec, spec->kctls.list); @@ -10626,7 +10626,7 @@ static int alc262_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; if (!spec->autocfg.line_outs) { - if (spec->autocfg.dig_out_pin || spec->autocfg.dig_in_pin) { + if (spec->autocfg.dig_outs || spec->autocfg.dig_in_pin) { spec->multiout.max_channels = 2; spec->no_analog = 1; goto dig_only; @@ -10643,9 +10643,9 @@ static int alc262_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; dig_only: - if (spec->autocfg.dig_out_pin) { + if (spec->autocfg.dig_outs) { spec->multiout.dig_out_nid = ALC262_DIGOUT_NID; - spec->dig_out_type = spec->autocfg.dig_out_type; + spec->dig_out_type = spec->autocfg.dig_out_type[0]; } if (spec->autocfg.dig_in_pin) spec->dig_in_nid = ALC262_DIGIN_NID; @@ -11807,7 +11807,7 @@ static int alc268_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = 2; /* digital only support output */ - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = ALC268_DIGOUT_NID; if (spec->kctls.list) @@ -12722,7 +12722,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = ALC269_DIGOUT_NID; if (spec->kctls.list) @@ -13779,7 +13779,7 @@ static int alc861_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = ALC861_DIGOUT_NID; if (spec->kctls.list) @@ -14881,7 +14881,7 @@ static int alc861vd_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = ALC861VD_DIGOUT_NID; if (spec->kctls.list) @@ -16689,7 +16689,7 @@ static int alc662_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = ALC880_DIGOUT_NID; if (spec->kctls.list) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 12b30884843..1882c573587 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2546,7 +2546,7 @@ static int stac92xx_build_pcms(struct hda_codec *codec) codec->num_pcms++; info++; info->name = "STAC92xx Digital"; - info->pcm_type = spec->autocfg.dig_out_type; + info->pcm_type = spec->autocfg.dig_out_type[0]; if (spec->multiout.dig_out_nid) { info->stream[SNDRV_PCM_STREAM_PLAYBACK] = stac92xx_pcm_digital_playback; info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dig_out_nid; @@ -3706,7 +3706,7 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out if (spec->multiout.max_channels > 2) spec->surr_switch = 1; - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = dig_out; if (dig_in && spec->autocfg.dig_in_pin) spec->dig_in_nid = dig_in; @@ -3819,7 +3819,7 @@ static int stac9200_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = 0x05; if (spec->autocfg.dig_in_pin) spec->dig_in_nid = 0x04; @@ -4069,8 +4069,8 @@ static int stac92xx_init(struct hda_codec *codec) for (i = 0; i < spec->num_dmics; i++) stac92xx_auto_set_pinctl(codec, spec->dmic_nids[i], AC_PINCTL_IN_EN); - if (cfg->dig_out_pin) - stac92xx_auto_set_pinctl(codec, cfg->dig_out_pin, + if (cfg->dig_out_pins[0]) + stac92xx_auto_set_pinctl(codec, cfg->dig_out_pins[0], AC_PINCTL_OUT_EN); if (cfg->dig_in_pin) stac92xx_auto_set_pinctl(codec, cfg->dig_in_pin, diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index c761394cbe8..639b2ff510a 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1354,7 +1354,7 @@ static int vt1708_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = VT1708_DIGOUT_NID; if (spec->autocfg.dig_in_pin) spec->dig_in_nid = VT1708_DIGIN_NID; @@ -1827,7 +1827,7 @@ static int vt1709_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = VT1709_DIGOUT_NID; if (spec->autocfg.dig_in_pin) spec->dig_in_nid = VT1709_DIGIN_NID; @@ -2371,7 +2371,7 @@ static int vt1708B_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = VT1708B_DIGOUT_NID; if (spec->autocfg.dig_in_pin) spec->dig_in_nid = VT1708B_DIGIN_NID; @@ -2836,7 +2836,7 @@ static int vt1708S_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = VT1708S_DIGOUT_NID; spec->extra_dig_out_nid = 0x15; @@ -3155,7 +3155,7 @@ static int vt1702_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = VT1702_DIGOUT_NID; spec->extra_dig_out_nid = 0x1B; -- cgit v1.2.3-70-g09d2 From c98041f7d71890ac6aa2257d78ef175db44d2cd3 Mon Sep 17 00:00:00 2001 From: Herton Ronaldo Krzesinski Date: Wed, 11 Feb 2009 20:33:15 -0200 Subject: ALSA: hda - Cleanup setting of pin_configs in patch_stac927x After commit "ALSA: hda - Fix restore of pin configs at resume for STAC/IDT codecs", the introduced stac_save_pin_cfgs function checks already for pins == NULL case, saving then default pin configs from machine with stac92xx_save_bios_config_regs. So we can remove the extra checks when stac927x_brd_tbl[spec->board_config] == NULL. Signed-off-by: Herton Ronaldo Krzesinski Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 1882c573587..3c84817ccd2 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -5292,10 +5292,9 @@ static int patch_stac927x(struct hda_codec *codec) stac927x_models, stac927x_cfg_tbl); again: - if (spec->board_config < 0 || !stac927x_brd_tbl[spec->board_config]) { - if (spec->board_config < 0) - snd_printdd(KERN_INFO "hda_codec: Unknown model for" - "STAC927x, using BIOS defaults\n"); + if (spec->board_config < 0) { + snd_printdd(KERN_INFO "hda_codec: Unknown model for" + "STAC927x, using BIOS defaults\n"); err = stac92xx_save_bios_config_regs(codec); } else err = stac_save_pin_cfgs(codec, -- cgit v1.2.3-70-g09d2 From e930e99500e5bd055270c668cca8bd2f33056895 Mon Sep 17 00:00:00 2001 From: Harvey Harrison Date: Wed, 11 Feb 2009 14:49:30 -0800 Subject: ALSA: echoaudio - replace uses of __constant_{endian} The base versions handle constant folding now. Signed-off-by: Harvey Harrison Signed-off-by: Takashi Iwai --- sound/pci/echoaudio/echo3g_dsp.c | 2 +- sound/pci/echoaudio/echoaudio_3g.c | 3 +-- sound/pci/echoaudio/echoaudio_dsp.c | 6 +++--- sound/pci/echoaudio/gina20_dsp.c | 4 ++-- sound/pci/echoaudio/layla20_dsp.c | 4 ++-- sound/pci/echoaudio/mia_dsp.c | 4 ++-- sound/pci/echoaudio/midi.c | 4 ++-- 7 files changed, 13 insertions(+), 14 deletions(-) diff --git a/sound/pci/echoaudio/echo3g_dsp.c b/sound/pci/echoaudio/echo3g_dsp.c index 417e25add82..57967e58057 100644 --- a/sound/pci/echoaudio/echo3g_dsp.c +++ b/sound/pci/echoaudio/echo3g_dsp.c @@ -56,7 +56,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) } chip->comm_page->e3g_frq_register = - __constant_cpu_to_le32((E3G_MAGIC_NUMBER / 48000) - 2); + cpu_to_le32((E3G_MAGIC_NUMBER / 48000) - 2); chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; diff --git a/sound/pci/echoaudio/echoaudio_3g.c b/sound/pci/echoaudio/echoaudio_3g.c index c3736bbd819..e32a7489792 100644 --- a/sound/pci/echoaudio/echoaudio_3g.c +++ b/sound/pci/echoaudio/echoaudio_3g.c @@ -40,8 +40,7 @@ static int check_asic_status(struct echoaudio *chip) if (wait_handshake(chip)) return -EIO; - chip->comm_page->ext_box_status = - __constant_cpu_to_le32(E3G_ASIC_NOT_LOADED); + chip->comm_page->ext_box_status = cpu_to_le32(E3G_ASIC_NOT_LOADED); chip->asic_loaded = FALSE; clear_handshake(chip); send_vector(chip, DSP_VC_TEST_ASIC); diff --git a/sound/pci/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c index be0e18192de..4df51ef5e09 100644 --- a/sound/pci/echoaudio/echoaudio_dsp.c +++ b/sound/pci/echoaudio/echoaudio_dsp.c @@ -926,11 +926,11 @@ static int init_dsp_comm_page(struct echoaudio *chip) /* Init the comm page */ chip->comm_page->comm_size = - __constant_cpu_to_le32(sizeof(struct comm_page)); + cpu_to_le32(sizeof(struct comm_page)); chip->comm_page->handshake = 0xffffffff; chip->comm_page->midi_out_free_count = - __constant_cpu_to_le32(DSP_MIDI_OUT_FIFO_SIZE); - chip->comm_page->sample_rate = __constant_cpu_to_le32(44100); + cpu_to_le32(DSP_MIDI_OUT_FIFO_SIZE); + chip->comm_page->sample_rate = cpu_to_le32(44100); chip->sample_rate = 44100; /* Set line levels so we don't blast any inputs on startup */ diff --git a/sound/pci/echoaudio/gina20_dsp.c b/sound/pci/echoaudio/gina20_dsp.c index db6c952e9d7..3f1e7475fae 100644 --- a/sound/pci/echoaudio/gina20_dsp.c +++ b/sound/pci/echoaudio/gina20_dsp.c @@ -208,10 +208,10 @@ static int set_professional_spdif(struct echoaudio *chip, char prof) DE_ACT(("set_professional_spdif %d\n", prof)); if (prof) chip->comm_page->flags |= - __constant_cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF); + cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF); else chip->comm_page->flags &= - ~__constant_cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF); + ~cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF); chip->professional_spdif = prof; return update_flags(chip); } diff --git a/sound/pci/echoaudio/layla20_dsp.c b/sound/pci/echoaudio/layla20_dsp.c index ede75c6ca0f..83750e9fd7b 100644 --- a/sound/pci/echoaudio/layla20_dsp.c +++ b/sound/pci/echoaudio/layla20_dsp.c @@ -284,10 +284,10 @@ static int set_professional_spdif(struct echoaudio *chip, char prof) DE_ACT(("set_professional_spdif %d\n", prof)); if (prof) chip->comm_page->flags |= - __constant_cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF); + cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF); else chip->comm_page->flags &= - ~__constant_cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF); + ~cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF); chip->professional_spdif = prof; return update_flags(chip); } diff --git a/sound/pci/echoaudio/mia_dsp.c b/sound/pci/echoaudio/mia_dsp.c index 227386602f9..3eca16cb7f7 100644 --- a/sound/pci/echoaudio/mia_dsp.c +++ b/sound/pci/echoaudio/mia_dsp.c @@ -222,10 +222,10 @@ static int set_professional_spdif(struct echoaudio *chip, char prof) DE_ACT(("set_professional_spdif %d\n", prof)); if (prof) chip->comm_page->flags |= - __constant_cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF); + cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF); else chip->comm_page->flags &= - ~__constant_cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF); + ~cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF); chip->professional_spdif = prof; return update_flags(chip); } diff --git a/sound/pci/echoaudio/midi.c b/sound/pci/echoaudio/midi.c index 77bf2a83d99..a953d142cb4 100644 --- a/sound/pci/echoaudio/midi.c +++ b/sound/pci/echoaudio/midi.c @@ -44,10 +44,10 @@ static int enable_midi_input(struct echoaudio *chip, char enable) if (enable) { chip->mtc_state = MIDI_IN_STATE_NORMAL; chip->comm_page->flags |= - __constant_cpu_to_le32(DSP_FLAG_MIDI_INPUT); + cpu_to_le32(DSP_FLAG_MIDI_INPUT); } else chip->comm_page->flags &= - ~__constant_cpu_to_le32(DSP_FLAG_MIDI_INPUT); + ~cpu_to_le32(DSP_FLAG_MIDI_INPUT); clear_handshake(chip); return send_vector(chip, DSP_VC_UPDATE_FLAGS); -- cgit v1.2.3-70-g09d2 From f1eaaeec11982c6b529d4255987fdf507a5fa69e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 13 Feb 2009 08:16:55 +0100 Subject: ALSA: hda - Allow fixed codec-probe mask Some devices have broken BIOS and they don't set the codec probe-bit properly after cleared by the driver. This makes the driver skipping the necessary codec slots. Since BIOS update isn't always easy, now the semantics of probe_mask option is changed a bit. When it contains the bit 8 (0x100), the lower bits are used to probe that slots regardless of codec-probe bits returned by the hardware. For example, probe_mask=0x103 will force to probe the codec slot #0 and #1. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 24 +++++++++++++++++------- 1 file changed, 17 insertions(+), 7 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 11e791b965f..19886e4bc82 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -381,6 +381,7 @@ struct azx { /* HD codec */ unsigned short codec_mask; + int codec_probe_mask; /* copied from probe_mask option */ struct hda_bus *bus; /* CORB/RIRB */ @@ -1228,7 +1229,6 @@ static unsigned int azx_max_codecs[AZX_NUM_DRIVERS] __devinitdata = { }; static int __devinit azx_codec_create(struct azx *chip, const char *model, - unsigned int codec_probe_mask, int no_init) { struct hda_bus_template bus_temp; @@ -1261,7 +1261,7 @@ static int __devinit azx_codec_create(struct azx *chip, const char *model, /* First try to probe all given codec slots */ for (c = 0; c < max_slots; c++) { - if ((chip->codec_mask & (1 << c)) & codec_probe_mask) { + if ((chip->codec_mask & (1 << c)) & chip->codec_probe_mask) { if (probe_codec(chip, c) < 0) { /* Some BIOSen give you wrong codec addresses * that don't exist @@ -1285,7 +1285,7 @@ static int __devinit azx_codec_create(struct azx *chip, const char *model, /* Then create codec instances */ for (c = 0; c < max_slots; c++) { - if ((chip->codec_mask & (1 << c)) & codec_probe_mask) { + if ((chip->codec_mask & (1 << c)) & chip->codec_probe_mask) { struct hda_codec *codec; err = snd_hda_codec_new(chip->bus, c, !no_init, &codec); if (err < 0) @@ -2101,20 +2101,31 @@ static struct snd_pci_quirk probe_mask_list[] __devinitdata = { {} }; +#define AZX_FORCE_CODEC_MASK 0x100 + static void __devinit check_probe_mask(struct azx *chip, int dev) { const struct snd_pci_quirk *q; - if (probe_mask[dev] == -1) { + chip->codec_probe_mask = probe_mask[dev]; + if (chip->codec_probe_mask == -1) { q = snd_pci_quirk_lookup(chip->pci, probe_mask_list); if (q) { printk(KERN_INFO "hda_intel: probe_mask set to 0x%x " "for device %04x:%04x\n", q->value, q->subvendor, q->subdevice); - probe_mask[dev] = q->value; + chip->codec_probe_mask = q->value; } } + + /* check forced option */ + if (chip->codec_probe_mask != -1 && + (chip->codec_probe_mask & AZX_FORCE_CODEC_MASK)) { + chip->codec_mask = chip->codec_probe_mask & 0xff; + printk(KERN_INFO "hda_intel: codec_mask forced to 0x%x\n", + chip->codec_mask); + } } @@ -2347,8 +2358,7 @@ static int __devinit azx_probe(struct pci_dev *pci, card->private_data = chip; /* create codec instances */ - err = azx_codec_create(chip, model[dev], probe_mask[dev], - probe_only[dev]); + err = azx_codec_create(chip, model[dev], probe_only[dev]); if (err < 0) goto out_free; -- cgit v1.2.3-70-g09d2 From 20db7cb0acd0ba5a3b12f686148d670294a69366 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 13 Feb 2009 08:18:48 +0100 Subject: ALSA: hda - Add forced codec-slots for ASUS W5F ASUS W5F needs the fixed codec-slots to probe to override the BIOS problem. Tested-by: Giovanni Moser Frainer Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 19886e4bc82..e853e4a8bde 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2098,6 +2098,8 @@ static struct snd_pci_quirk probe_mask_list[] __devinitdata = { SND_PCI_QUIRK(0x1028, 0x20ac, "Dell Studio Desktop", 0x01), /* including bogus ALC268 in slot#2 that conflicts with ALC888 */ SND_PCI_QUIRK(0x17c0, 0x4085, "Medion MD96630", 0x01), + /* forced codec slots */ + SND_PCI_QUIRK(0x1046, 0x1262, "ASUS W5F", 0x103), {} }; -- cgit v1.2.3-70-g09d2 From ae374d667a54fb5e2c9c0c4e87b206bd665f3ad6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 13 Feb 2009 08:33:55 +0100 Subject: ALSA: hda - Update documentation Update documentation regarding codec probing; the new probe_only option and the new probe_mask usage. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/ALSA-Configuration.txt | 3 +++ Documentation/sound/alsa/HD-Audio.txt | 17 +++++++++++++++++ 2 files changed, 20 insertions(+) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 841a9365d5f..012afd7afb1 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -757,6 +757,9 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. model - force the model name position_fix - Fix DMA pointer (0 = auto, 1 = use LPIB, 2 = POSBUF) probe_mask - Bitmask to probe codecs (default = -1, meaning all slots) + When the bit 8 (0x100) is set, the lower 8 bits are used + as the "fixed" codec slots; i.e. the driver probes the + slots regardless what hardware reports back probe_only - Only probing and no codec initialization (default=off); Useful to check the initial codec status for debugging bdl_pos_adj - Specifies the DMA IRQ timing delay in samples. diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt index 8d68fff7183..99f7fbbe3e6 100644 --- a/Documentation/sound/alsa/HD-Audio.txt +++ b/Documentation/sound/alsa/HD-Audio.txt @@ -109,6 +109,13 @@ slot, pass `probe_mask=1`. For the first and the third slots, pass Since 2.6.29 kernel, the driver has a more robust probing method, so this error might happen rarely, though. +On a machine with a broken BIOS, sometimes you need to force the +driver to probe the codec slots the hardware doesn't report for use. +In such a case, turn the bit 8 (0x100) of `probe_mask` option on. +Then the rest 8 bits are passed as the codec slots to probe +unconditionally. For example, `probe_mask=0x103` will force to probe +the codec slots 0 and 1 no matter what the hardware reports. + Interrupt Handling ~~~~~~~~~~~~~~~~~~ @@ -461,6 +468,16 @@ run with `--no-upload` option, and attach the generated file. There are some other useful options. See `--help` option output for details. +When a probe error occurs or when the driver obviously assigns a +mismatched model, it'd be helpful to load the driver with +`probe_only=1` option (at best after the cold reboot) and run +alsa-info at this state. With this option, the driver won't configure +the mixer and PCM but just tries to probe the codec slot. After +probing, the proc file is available, so you can get the raw codec +information before modified by the driver. Of course, the driver +isn't usable with `probe_only=1`. But you can continue the +configuration via hwdep sysfs file if hda-reconfig option is enabled. + hda-verb ~~~~~~~~ -- cgit v1.2.3-70-g09d2 From 8bb0ac5573ff0879fef511e1a80a4a4db0316daa Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Thu, 12 Feb 2009 16:50:01 -0500 Subject: ALSA: hda: Add STAC_DELL_S14 quirk Add STAC_DELL_S14 quirk for new laptop series. Removed un-needed pins in pin_nids for stac92hd83xxx. Also reorganized connection selection code for the respective ports per quirk define. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 52 ++++++++++++++++++++++++++++-------------- 1 file changed, 35 insertions(+), 17 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 3c84817ccd2..1ebb36ca2e0 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -83,6 +83,7 @@ enum { enum { STAC_92HD83XXX_REF, STAC_92HD83XXX_PWR_REF, + STAC_DELL_S14, STAC_92HD83XXX_MODELS }; @@ -480,10 +481,9 @@ static hda_nid_t stac92hd73xx_pin_nids[13] = { 0x14, 0x22, 0x23 }; -static hda_nid_t stac92hd83xxx_pin_nids[14] = { +static hda_nid_t stac92hd83xxx_pin_nids[10] = { 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, - 0x0f, 0x10, 0x11, 0x12, 0x13, - 0x1d, 0x1e, 0x1f, 0x20 + 0x0f, 0x10, 0x11, 0x1f, 0x20, }; #define STAC92HD71BXX_NUM_PINS 13 @@ -857,9 +857,9 @@ static struct hda_verb stac92hd73xx_10ch_core_init[] = { }; static struct hda_verb stac92hd83xxx_core_init[] = { - { 0xa, AC_VERB_SET_CONNECT_SEL, 0x0}, - { 0xb, AC_VERB_SET_CONNECT_SEL, 0x0}, - { 0xd, AC_VERB_SET_CONNECT_SEL, 0x1}, + { 0xa, AC_VERB_SET_CONNECT_SEL, 0x1}, + { 0xb, AC_VERB_SET_CONNECT_SEL, 0x1}, + { 0xd, AC_VERB_SET_CONNECT_SEL, 0x0}, /* power state controls amps */ { 0x01, AC_VERB_SET_EAPD, 1 << 2}, @@ -1730,21 +1730,28 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = { {} /* terminator */ }; -static unsigned int ref92hd83xxx_pin_configs[14] = { +static unsigned int ref92hd83xxx_pin_configs[10] = { 0x02214030, 0x02211010, 0x02a19020, 0x02170130, 0x01014050, 0x01819040, 0x01014020, 0x90a3014e, - 0x40f000f0, 0x40f000f0, 0x40f000f0, 0x40f000f0, 0x01451160, 0x98560170, }; +static unsigned int dell_s14_pin_configs[10] = { + 0x02214030, 0x02211010, 0x02a19020, 0x01014050, + 0x40f000f0, 0x01819040, 0x40f000f0, 0x90a60160, + 0x40f000f0, 0x40f000f0, +}; + static unsigned int *stac92hd83xxx_brd_tbl[STAC_92HD83XXX_MODELS] = { [STAC_92HD83XXX_REF] = ref92hd83xxx_pin_configs, [STAC_92HD83XXX_PWR_REF] = ref92hd83xxx_pin_configs, + [STAC_DELL_S14] = dell_s14_pin_configs, }; static const char *stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = { [STAC_92HD83XXX_REF] = "ref", [STAC_92HD83XXX_PWR_REF] = "mic-ref", + [STAC_DELL_S14] = "dell-s14", }; static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { @@ -1753,6 +1760,8 @@ static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { "DFI LanParty", STAC_92HD83XXX_REF), SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, "DFI LanParty", STAC_92HD83XXX_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02ba, + "unknown Dell", STAC_DELL_S14), {} /* terminator */ }; @@ -4822,6 +4831,7 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) hda_nid_t conn[STAC92HD83_DAC_COUNT + 1]; int err; int num_dacs; + hda_nid_t nid; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -4840,15 +4850,6 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) spec->num_pwrs = ARRAY_SIZE(stac92hd83xxx_pwr_nids); spec->multiout.dac_nids = spec->dac_nids; - - /* set port 0xe to select the last DAC - */ - num_dacs = snd_hda_get_connections(codec, 0x0e, - conn, STAC92HD83_DAC_COUNT + 1) - 1; - - snd_hda_codec_write_cache(codec, 0xe, 0, - AC_VERB_SET_CONNECT_SEL, num_dacs); - spec->init = stac92hd83xxx_core_init; spec->mixer = stac92hd83xxx_mixer; spec->num_pins = ARRAY_SIZE(stac92hd83xxx_pin_nids); @@ -4900,6 +4901,23 @@ again: return err; } + switch (spec->board_config) { + case STAC_DELL_S14: + nid = 0xf; + break; + default: + nid = 0xe; + break; + } + + num_dacs = snd_hda_get_connections(codec, nid, + conn, STAC92HD83_DAC_COUNT + 1) - 1; + + /* set port X to select the last DAC + */ + snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_CONNECT_SEL, num_dacs); + codec->patch_ops = stac92xx_patch_ops; codec->proc_widget_hook = stac92hd_proc_hook; -- cgit v1.2.3-70-g09d2 From 27e089888fb1a3d1d13892262f9d522b03985044 Mon Sep 17 00:00:00 2001 From: Aristeu Sergio Rozanski Filho Date: Thu, 12 Feb 2009 17:50:37 -0500 Subject: ALSA: hda: add quirk for Lenovo X200 laptop dock Currently the HP connector on X200 dock doesn't detect when a HP is connected nor allows sound to be played using it. This patch fixes the problem by adding a quirk for this specific model. It's possible that others have the same NID (0x19) to report when dock HP is connected, but I don't have access to any. Please Cc me in the reply since I'm not subscribed to alsa-devel@. Signed-off-by: Aristeu Rozanski Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 + sound/pci/hda/patch_conexant.c | 40 ++++++++++++++++++++++++++++ 2 files changed, 41 insertions(+) diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 8f40999a456..0e52d273ce9 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -262,6 +262,7 @@ Conexant 5051 ============= laptop Basic Laptop config (default) hp HP Spartan laptop + lenovo-x200 Lenovo X200 laptop STAC9200 ======== diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index fdf876be712..b8de73ecfde 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1798,6 +1798,40 @@ static struct hda_verb cxt5051_init_verbs[] = { { } /* end */ }; +static struct hda_verb cxt5051_lenovo_x200_init_verbs[] = { + /* Line in, Mic */ + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, + /* SPK */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* HP, Amp */ + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x16, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* Docking HP */ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* DAC1 */ + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Record selector: Int mic */ + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x44}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1) | 0x44}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x44}, + /* SPDIF route: PCM */ + {0x1c, AC_VERB_SET_CONNECT_SEL, 0x0}, + /* EAPD */ + {0x1a, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ + {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT}, + {0x17, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CXT5051_PORTB_EVENT}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CXT5051_PORTC_EVENT}, + {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT}, + { } /* end */ +}; + /* initialize jack-sensing, too */ static int cxt5051_init(struct hda_codec *codec) { @@ -1815,18 +1849,21 @@ static int cxt5051_init(struct hda_codec *codec) enum { CXT5051_LAPTOP, /* Laptops w/ EAPD support */ CXT5051_HP, /* no docking */ + CXT5051_LENOVO_X200, /* Lenovo X200 laptop */ CXT5051_MODELS }; static const char *cxt5051_models[CXT5051_MODELS] = { [CXT5051_LAPTOP] = "laptop", [CXT5051_HP] = "hp", + [CXT5051_LENOVO_X200] = "lenovo-x200", }; static struct snd_pci_quirk cxt5051_cfg_tbl[] = { SND_PCI_QUIRK(0x14f1, 0x0101, "Conexant Reference board", CXT5051_LAPTOP), SND_PCI_QUIRK(0x14f1, 0x5051, "HP Spartan 1.1", CXT5051_HP), + SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT5051_LENOVO_X200), {} }; @@ -1867,6 +1904,9 @@ static int patch_cxt5051(struct hda_codec *codec) codec->patch_ops.unsol_event = cxt5051_hp_unsol_event; spec->mixers[0] = cxt5051_hp_mixers; break; + case CXT5051_LENOVO_X200: + spec->init_verbs[0] = cxt5051_lenovo_x200_init_verbs; + /* fallthru */ default: case CXT5051_LAPTOP: codec->patch_ops.unsol_event = cxt5051_hp_unsol_event; -- cgit v1.2.3-70-g09d2 From 946835074e026f4bbe9f3c2b091dca6346bd1474 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 13 Feb 2009 09:31:20 +0100 Subject: ALSA: hda - Add quirk for Acer AX1700-U3700A Force model=auto for Acer AX1700-U3700A with ALC888 codec. Since Acer devices are handlded as model=acer as default, the auto parsing has to be specified explicitly. (Maybe it's better rather to remove this default model=acer handling, though.) Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e46251bceb9..2306cca1b69 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8520,6 +8520,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { ALC888_ACER_ASPIRE_4930G), SND_PCI_QUIRK(0x1025, 0x013f, "Acer Aspire 5930G", ALC888_ACER_ASPIRE_4930G), + SND_PCI_QUIRK(0x1025, 0x0158, "Acer AX1700-U3700A", ALC883_AUTO), SND_PCI_QUIRK(0x1025, 0x015e, "Acer Aspire 6930G", ALC888_ACER_ASPIRE_4930G), /* default Acer */ -- cgit v1.2.3-70-g09d2 From 9b5f12e5a4029c1cd03784754687faef6d9e54fa Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 13 Feb 2009 11:47:37 +0100 Subject: ALSA: hda - Add proper cleanup for multiout-dig for ALC codecs The recent patch_realtek.c contains the slave digital-out support as well. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2306cca1b69..ef9b7ee3410 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2979,6 +2979,14 @@ static int alc880_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo, stream_tag, format, substream); } +static int alc880_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct alc_spec *spec = codec->spec; + return snd_hda_multi_out_dig_cleanup(codec, &spec->multiout); +} + static int alc880_dig_playback_pcm_close(struct hda_pcm_stream *hinfo, struct hda_codec *codec, struct snd_pcm_substream *substream) @@ -3062,7 +3070,8 @@ static struct hda_pcm_stream alc880_pcm_digital_playback = { .ops = { .open = alc880_dig_playback_pcm_open, .close = alc880_dig_playback_pcm_close, - .prepare = alc880_dig_playback_pcm_prepare + .prepare = alc880_dig_playback_pcm_prepare, + .cleanup = alc880_dig_playback_pcm_cleanup }, }; -- cgit v1.2.3-70-g09d2 From 6a05ac4afa90ac9c38fedd3f6940fe8da5d1fcf6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 13 Feb 2009 11:19:09 +0100 Subject: ALSA: hda - Support multiple digital outs with auto-probing Added the support of multiple digital outputs via auto-probing for Realtek ALC88x codecs. The multiple outputs are handled as slave streams, so only one PCM stream (and the corresponding IEC958* elements) is provided to control both digital outputs. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 22 +++++++++++++++++++--- 1 file changed, 19 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ef9b7ee3410..244de597c5b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -279,6 +279,7 @@ struct alc_spec { * dig_out_nid and hp_nid are optional */ hda_nid_t alt_dac_nid; + hda_nid_t slave_dig_outs[3]; /* optional - for auto-parsing */ int dig_out_type; /* capture */ @@ -4269,7 +4270,7 @@ static void alc880_auto_init_analog_input(struct hda_codec *codec) static int alc880_parse_auto_config(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - int err; + int i, err; static hda_nid_t alc880_ignore[] = { 0x1d, 0 }; err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, @@ -4300,8 +4301,23 @@ static int alc880_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_outs) - spec->multiout.dig_out_nid = ALC880_DIGOUT_NID; + /* check multiple SPDIF-out (for recent codecs) */ + for (i = 0; i < spec->autocfg.dig_outs; i++) { + hda_nid_t dig_nid; + err = snd_hda_get_connections(codec, + spec->autocfg.dig_out_pins[i], + &dig_nid, 1); + if (err < 0) + continue; + if (!i) + spec->multiout.dig_out_nid = dig_nid; + else { + spec->multiout.slave_dig_outs = spec->slave_dig_outs; + spec->slave_dig_outs[i - 1] = dig_nid; + if (i == ARRAY_SIZE(spec->slave_dig_outs) - 1) + break; + } + } if (spec->autocfg.dig_in_pin) spec->dig_in_nid = ALC880_DIGIN_NID; -- cgit v1.2.3-70-g09d2 From d5e9ba1d58b6da1c58a91113fc350ece97ec5a0b Mon Sep 17 00:00:00 2001 From: Timur Tabi Date: Tue, 3 Feb 2009 11:09:32 -0600 Subject: ASoC: add additional controls to the CS4270 codec driver Update the CS4270 codec driver to allow applications to use the mixer to control Digital Loopback, Soft Ramp, Zero Cross, Popguard, and Auto-Mute. Soft Ramp, Zero Cross, and Auto-Mute are disabled by the driver when it first initializes the hardware, but these features either don't work or interfere with normal ALSA behavior. However, they can now be re-enabled by an application if desired. Remove CONFIG_SND_SOC_CS4270_HWMUTE and always allow ASoC to control the mute bits. The driver previously and erroneously assumed that these bits control only external muting circuitry, but they also control internal muting circuitry, so they should always be used. Signed-off-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 6 --- sound/soc/codecs/cs4270.c | 93 ++++++++++++++++++++--------------------------- 2 files changed, 40 insertions(+), 59 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index a195303603e..628a591c728 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -67,12 +67,6 @@ config SND_SOC_AK4535 config SND_SOC_CS4270 tristate -# Cirrus Logic CS4270 Codec Hardware Mute Support -# Select if you have external muting circuitry attached to your CS4270. -config SND_SOC_CS4270_HWMUTE - bool - depends on SND_SOC_CS4270 - # Cirrus Logic CS4270 Codec VD = 3.3V Errata # Select if you are affected by the errata where the part will not function # if MCLK divide-by-1.5 is selected and VD is set to 3.3V. The driver will diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 2c79a24186f..cd4a9ee38e4 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -395,17 +395,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - /* Freeze and power-down the codec */ - - ret = snd_soc_write(codec, CS4270_PWRCTL, CS4270_PWRCTL_FREEZE | - CS4270_PWRCTL_PDN_ADC | CS4270_PWRCTL_PDN_DAC | - CS4270_PWRCTL_PDN); - if (ret < 0) { - dev_err(codec->dev, "i2c write failed\n"); - return ret; - } - - /* Program the mode control register */ + /* Set the sample rate */ reg = snd_soc_read(codec, CS4270_MODE); reg &= ~(CS4270_MODE_SPEED_MASK | CS4270_MODE_DIV_MASK); @@ -417,7 +407,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, return ret; } - /* Program the format register */ + /* Set the DAI format */ reg = snd_soc_read(codec, CS4270_FORMAT); reg &= ~(CS4270_FORMAT_DAC_MASK | CS4270_FORMAT_ADC_MASK); @@ -440,42 +430,9 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, return ret; } - /* Disable auto-mute. This feature appears to be buggy, because in - some situations, auto-mute will not deactivate when it should. */ - - reg = snd_soc_read(codec, CS4270_MUTE); - reg &= ~CS4270_MUTE_AUTO; - ret = snd_soc_write(codec, CS4270_MUTE, reg); - if (ret < 0) { - dev_err(codec->dev, "i2c write failed\n"); - return ret; - } - - /* Disable automatic volume control. It's enabled by default, and - * it causes volume change commands to be delayed, sometimes until - * after playback has started. - */ - - reg = cs4270_read_reg_cache(codec, CS4270_TRANS); - reg &= ~(CS4270_TRANS_SOFT | CS4270_TRANS_ZERO); - ret = cs4270_i2c_write(codec, CS4270_TRANS, reg); - if (ret < 0) { - dev_err(codec->dev, "i2c write failed\n"); - return ret; - } - - /* Thaw and power-up the codec */ - - ret = snd_soc_write(codec, CS4270_PWRCTL, 0); - if (ret < 0) { - dev_err(codec->dev, "i2c write failed\n"); - return ret; - } - return ret; } -#ifdef CONFIG_SND_SOC_CS4270_HWMUTE /** * cs4270_mute - enable/disable the CS4270 external mute * @dai: the SOC DAI @@ -494,22 +451,23 @@ static int cs4270_mute(struct snd_soc_dai *dai, int mute) reg6 = snd_soc_read(codec, CS4270_MUTE); if (mute) - reg6 |= CS4270_MUTE_ADC_A | CS4270_MUTE_ADC_B | - CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B; + reg6 |= CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B; else - reg6 &= ~(CS4270_MUTE_ADC_A | CS4270_MUTE_ADC_B | - CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B); + reg6 &= ~(CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B); return snd_soc_write(codec, CS4270_MUTE, reg6); } -#else -#define cs4270_mute NULL -#endif /* A list of non-DAPM controls that the CS4270 supports */ static const struct snd_kcontrol_new cs4270_snd_controls[] = { SOC_DOUBLE_R("Master Playback Volume", - CS4270_VOLA, CS4270_VOLB, 0, 0xFF, 1) + CS4270_VOLA, CS4270_VOLB, 0, 0xFF, 1), + SOC_SINGLE("Digital Sidetone Switch", CS4270_FORMAT, 5, 1, 0), + SOC_SINGLE("Soft Ramp Switch", CS4270_TRANS, 6, 1, 0), + SOC_SINGLE("Zero Cross Switch", CS4270_TRANS, 5, 1, 0), + SOC_SINGLE("Popguard Switch", CS4270_MODE, 0, 1, 1), + SOC_SINGLE("Auto-Mute Switch", CS4270_MUTE, 5, 1, 0), + SOC_DOUBLE("Master Capture Switch", CS4270_MUTE, 3, 4, 1, 0) }; /* @@ -637,6 +595,7 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client, { struct snd_soc_codec *codec; struct cs4270_private *cs4270; + unsigned int reg; int ret; /* For now, we only support one cs4270 device in the system. See the @@ -702,6 +661,34 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client, goto error_free_codec; } + /* Disable auto-mute. This feature appears to be buggy. In some + * situations, auto-mute will not deactivate when it should, so we want + * this feature disabled by default. An application (e.g. alsactl) can + * re-enabled it by using the controls. + */ + + reg = cs4270_read_reg_cache(codec, CS4270_MUTE); + reg &= ~CS4270_MUTE_AUTO; + ret = cs4270_i2c_write(codec, CS4270_MUTE, reg); + if (ret < 0) { + dev_err(&i2c_client->dev, "i2c write failed\n"); + return ret; + } + + /* Disable automatic volume control. The hardware enables, and it + * causes volume change commands to be delayed, sometimes until after + * playback has started. An application (e.g. alsactl) can + * re-enabled it by using the controls. + */ + + reg = cs4270_read_reg_cache(codec, CS4270_TRANS); + reg &= ~(CS4270_TRANS_SOFT | CS4270_TRANS_ZERO); + ret = cs4270_i2c_write(codec, CS4270_TRANS, reg); + if (ret < 0) { + dev_err(&i2c_client->dev, "i2c write failed\n"); + return ret; + } + /* Initialize the DAI. Normally, we'd prefer to have a kmalloc'd DAI * structure for each CS4270 device, but the machine driver needs to * have a pointer to the DAI structure, so for now it must be a global -- cgit v1.2.3-70-g09d2 From bf3dbe5c8c4b85f98c36d35432efa6573b75e6d3 Mon Sep 17 00:00:00 2001 From: Kevin Hilman Date: Fri, 13 Feb 2009 11:36:37 -0800 Subject: ASoC: Fix DaVinci module unload error Fix for the error when the audio module is unloaded. On unregistering the platform_device, platform_device_release will free the platform data.If platform data is static the kernel panics when it is freed. Instead use the platform device helper function to add data. This change has been tested on DM644x EVM, DM644x SFFSDR and DM355 EVM. Signed-off-by: Chaithrika U S Signed-off-by: Kevin Hilman Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-evm.c | 3 ++- sound/soc/davinci/davinci-sffsdr.c | 3 ++- 2 files changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 54851f31856..9b90b347007 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -186,7 +186,8 @@ static int __init evm_init(void) platform_set_drvdata(evm_snd_device, &evm_snd_devdata); evm_snd_devdata.dev = &evm_snd_device->dev; - evm_snd_device->dev.platform_data = &evm_snd_data; + platform_device_add_data(evm_snd_device, &evm_snd_data, + sizeof(evm_snd_data)); ret = platform_device_add_resources(evm_snd_device, evm_snd_resources, ARRAY_SIZE(evm_snd_resources)); diff --git a/sound/soc/davinci/davinci-sffsdr.c b/sound/soc/davinci/davinci-sffsdr.c index 50baef1fe5b..0bf81abba8c 100644 --- a/sound/soc/davinci/davinci-sffsdr.c +++ b/sound/soc/davinci/davinci-sffsdr.c @@ -141,7 +141,8 @@ static int __init sffsdr_init(void) platform_set_drvdata(sffsdr_snd_device, &sffsdr_snd_devdata); sffsdr_snd_devdata.dev = &sffsdr_snd_device->dev; - sffsdr_snd_device->dev.platform_data = &sffsdr_snd_data; + platform_device_add_data(sffsdr_snd_device, &sffsdr_snd_data, + sizeof(sffsdr_snd_data)); ret = platform_device_add_resources(sffsdr_snd_device, sffsdr_snd_resources, -- cgit v1.2.3-70-g09d2 From e2ea57a8df6da45f5f63ab7b56528a552f36fb72 Mon Sep 17 00:00:00 2001 From: Herton Ronaldo Krzesinski Date: Mon, 16 Feb 2009 10:23:00 +0100 Subject: ALSA: hda - Fix speaker output on HP DV4 1155-SE Force speaker pin config with model=hp-dv5 model for cases when bios doesn't set it up properly. All reported hp laptops using model=hp-dv5 model have speaker at pin 0x0d with same config, so it's safe to add this within hp-dv5 model. Reference: alsa-devel mailing list thread on http://mailman.alsa-project.org/pipermail/alsa-devel/2009-February/014390.html Signed-off-by: Herton Ronaldo Krzesinski Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index aeb5d2126da..7320059b713 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1823,6 +1823,8 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { "HP dv7", STAC_HP_DV5), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30f7, "HP dv4", STAC_HP_DV5), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30fb, + "HP dv7", STAC_HP_DV5), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30fc, "HP dv7", STAC_HP_M4), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3600, @@ -5170,6 +5172,10 @@ again: spec->num_smuxes = 0; spec->num_dmuxes = 0; break; + case STAC_HP_DV5: + stac_change_pin_config(codec, 0x0d, 0x90170010); + stac92xx_auto_set_pinctl(codec, 0x0d, AC_PINCTL_OUT_EN); + break; }; spec->multiout.dac_nids = spec->dac_nids; -- cgit v1.2.3-70-g09d2 From a259cb8eb784352ee9ce46a4fc6cd94fc2fbc68d Mon Sep 17 00:00:00 2001 From: Roel Kluin Date: Sun, 15 Feb 2009 20:51:19 +0100 Subject: sound: OSS: &&/|| typo in ad1848.c &&/|| typo Signed-off-by: Roel Kluin Signed-off-by: Takashi Iwai --- sound/oss/ad1848.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/oss/ad1848.c b/sound/oss/ad1848.c index 7cf9913a47b..a5b83568bdc 100644 --- a/sound/oss/ad1848.c +++ b/sound/oss/ad1848.c @@ -2107,7 +2107,7 @@ int ad1848_control(int cmd, int arg) switch (cmd) { case AD1848_SET_XTAL: /* Change clock frequency of AD1845 (only ) */ - if (devc->model != MD_1845 || devc->model != MD_1845_SSCAPE) + if (devc->model != MD_1845 && devc->model != MD_1845_SSCAPE) return -EINVAL; spin_lock_irqsave(&devc->lock,flags); ad_enter_MCE(devc); -- cgit v1.2.3-70-g09d2 From 2ae466f8cc522843fa9a456e46007dd98b052b13 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 16 Feb 2009 14:16:36 +0100 Subject: ALSA: hda - Cleanup IDT92HD7x HP quirks Clean up IDT92HD7x quirks for HP laptops with SND_PCI_QUIRK_MASK(). Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 20 +++++--------------- 1 file changed, 5 insertions(+), 15 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 7320059b713..d00a211a813 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1817,22 +1817,12 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { "DFI LanParty", STAC_92HD71BXX_REF), SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, "DFI LanParty", STAC_92HD71BXX_REF), - SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30f2, - "HP dv5", STAC_HP_M4), - SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30f4, - "HP dv7", STAC_HP_DV5), - SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30f7, - "HP dv4", STAC_HP_DV5), - SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30fb, - "HP dv7", STAC_HP_DV5), - SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30fc, - "HP dv7", STAC_HP_M4), - SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3600, - "HP dv5", STAC_HP_DV5), - SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3603, - "HP dv5", STAC_HP_DV5), + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x30f0, + "HP dv4-7", STAC_HP_DV5), + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x3600, + "HP dv4-7", STAC_HP_DV5), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x361a, - "unknown HP", STAC_HP_M4), + "HP mini 1000", STAC_HP_M4), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0233, "unknown Dell", STAC_DELL_M4_1), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0234, -- cgit v1.2.3-70-g09d2 From c23127566c7a54c8413bf1b99becea76072f467e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 16 Feb 2009 15:20:41 +0100 Subject: ALSA: hda - Clean up quirks for HP laptops with AD1984A Clean up quirks for HP laptops with AD1984A using SND_PCI_QUIRK_MASK() Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index af6b0035e2e..2c58d7b05ab 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -3923,8 +3923,7 @@ static struct snd_pci_quirk ad1884a_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x3030, "HP", AD1884A_MOBILE), SND_PCI_QUIRK(0x103c, 0x3037, "HP 2230s", AD1884A_LAPTOP), SND_PCI_QUIRK(0x103c, 0x3056, "HP", AD1884A_MOBILE), - SND_PCI_QUIRK(0x103c, 0x3072, "HP", AD1884A_LAPTOP), - SND_PCI_QUIRK(0x103c, 0x3077, "HP", AD1884A_LAPTOP), + SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x3070, "HP", AD1884A_MOBILE), SND_PCI_QUIRK(0x103c, 0x30e6, "HP 6730b", AD1884A_LAPTOP), SND_PCI_QUIRK(0x103c, 0x30e7, "HP EliteBook 8530p", AD1884A_LAPTOP), SND_PCI_QUIRK(0x103c, 0x3614, "HP 6730s", AD1884A_LAPTOP), -- cgit v1.2.3-70-g09d2 From c2b73d1458014a9f461b75bc1756a699a6c0781f Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Mon, 16 Feb 2009 21:38:37 +0100 Subject: ALSA: cs4236: cs4232 and cs4236 driver merge to solve PnP BIOS detection cs4232 and cs4236 driver merge to solve PnP BIOS detection. Also, the patch adds recognition if the chip is cs4236b+ or earlier part. This unifies drivers for both cs4232 and cs4236+ chips. It allows to use the PnP BIOS detection for the cs4236+ chips. Previously, only the snd-cs4232 could be detected by the PnP BIOS. The cs4232+ cards reports two separate PnP BIOS ids. The patch adds search for the second id to find out resources assigned to a control port. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- include/sound/wss.h | 1 + sound/isa/Kconfig | 23 ++----- sound/isa/cs423x/Makefile | 2 - sound/isa/cs423x/cs4232.c | 2 - sound/isa/cs423x/cs4236.c | 153 +++++++++++++++++++++------------------------- sound/isa/wss/wss_lib.c | 3 +- 6 files changed, 78 insertions(+), 106 deletions(-) delete mode 100644 sound/isa/cs423x/cs4232.c diff --git a/include/sound/wss.h b/include/sound/wss.h index fd01f22825c..6d65f322f1d 100644 --- a/include/sound/wss.h +++ b/include/sound/wss.h @@ -154,6 +154,7 @@ int snd_wss_create(struct snd_card *card, unsigned short hardware, unsigned short hwshare, struct snd_wss **rchip); +int snd_wss_free(struct snd_wss *chip); int snd_wss_pcm(struct snd_wss *chip, int device, struct snd_pcm **rpcm); int snd_wss_timer(struct snd_wss *chip, int device, struct snd_timer **rtimer); int snd_wss_mixer(struct snd_wss *chip); diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index 5915dc41c0e..4e06bbd9298 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -56,8 +56,8 @@ config SND_AD1848 Say Y here to include support for AD1848 (Analog Devices) or CS4248 (Cirrus Logic - Crystal Semiconductors) chips. - For newer chips from Cirrus Logic, use the CS4231, CS4232 or - CS4236+ drivers. + For newer chips from Cirrus Logic, use the CS4231 or CS4232+ + drivers. To compile this driver as a module, choose M here: the module will be called snd-ad1848. @@ -114,26 +114,15 @@ config SND_CS4231 To compile this driver as a module, choose M here: the module will be called snd-cs4231. -config SND_CS4232 - tristate "Generic Cirrus Logic CS4232 driver" - select SND_OPL3_LIB - select SND_MPU401_UART - select SND_WSS_LIB - help - Say Y here to include support for CS4232 chips from Cirrus - Logic - Crystal Semiconductors. - - To compile this driver as a module, choose M here: the module - will be called snd-cs4232. - config SND_CS4236 - tristate "Generic Cirrus Logic CS4236+ driver" + tristate "Generic Cirrus Logic CS4232/CS4236+ driver" select SND_OPL3_LIB select SND_MPU401_UART select SND_WSS_LIB help - Say Y to include support for CS4235,CS4236,CS4237B,CS4238B, - CS4239 chips from Cirrus Logic - Crystal Semiconductors. + Say Y to include support for CS4232,CS4235,CS4236,CS4237B, + CS4238B,CS4239 chips from Cirrus Logic - Crystal + Semiconductors. To compile this driver as a module, choose M here: the module will be called snd-cs4236. diff --git a/sound/isa/cs423x/Makefile b/sound/isa/cs423x/Makefile index 5870ca21ab5..732f66cc036 100644 --- a/sound/isa/cs423x/Makefile +++ b/sound/isa/cs423x/Makefile @@ -5,11 +5,9 @@ snd-cs4236-lib-objs := cs4236_lib.o snd-cs4231-objs := cs4231.o -snd-cs4232-objs := cs4232.o snd-cs4236-objs := cs4236.o # Toplevel Module Dependency obj-$(CONFIG_SND_CS4231) += snd-cs4231.o -obj-$(CONFIG_SND_CS4232) += snd-cs4232.o obj-$(CONFIG_SND_CS4236) += snd-cs4236.o snd-cs4236-lib.o diff --git a/sound/isa/cs423x/cs4232.c b/sound/isa/cs423x/cs4232.c deleted file mode 100644 index 9fad2e6c0c2..00000000000 --- a/sound/isa/cs423x/cs4232.c +++ /dev/null @@ -1,2 +0,0 @@ -#define CS4232 -#include "cs4236.c" diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c index f7845986f46..a076a6ce807 100644 --- a/sound/isa/cs423x/cs4236.c +++ b/sound/isa/cs423x/cs4236.c @@ -33,17 +33,14 @@ MODULE_AUTHOR("Jaroslav Kysela "); MODULE_LICENSE("GPL"); -#ifdef CS4232 -MODULE_DESCRIPTION("Cirrus Logic CS4232"); +MODULE_DESCRIPTION("Cirrus Logic CS4232-9"); MODULE_SUPPORTED_DEVICE("{{Turtle Beach,TBS-2000}," "{Turtle Beach,Tropez Plus}," "{SIC CrystalWave 32}," "{Hewlett Packard,Omnibook 5500}," "{TerraTec,Maestro 32/96}," - "{Philips,PCA70PS}}"); -#else -MODULE_DESCRIPTION("Cirrus Logic CS4235-9"); -MODULE_SUPPORTED_DEVICE("{{Crystal Semiconductors,CS4235}," + "{Philips,PCA70PS}}," + "{{Crystal Semiconductors,CS4235}," "{Crystal Semiconductors,CS4236}," "{Crystal Semiconductors,CS4237}," "{Crystal Semiconductors,CS4238}," @@ -70,15 +67,11 @@ MODULE_SUPPORTED_DEVICE("{{Crystal Semiconductors,CS4235}," "{Typhoon Soundsystem,CS4236B}," "{Turtle Beach,Malibu}," "{Unknown,Digital PC 5000 Onboard}}"); -#endif -#ifdef CS4232 -#define IDENT "CS4232" -#define DEV_NAME "cs4232" -#else -#define IDENT "CS4236+" -#define DEV_NAME "cs4236" -#endif +MODULE_ALIAS("snd_cs4232"); + +#define IDENT "CS4232+" +#define DEV_NAME "cs4232+" static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ @@ -128,9 +121,7 @@ MODULE_PARM_DESC(dma2, "DMA2 # for " IDENT " driver."); #ifdef CONFIG_PNP static int isa_registered; static int pnpc_registered; -#ifdef CS4232 static int pnp_registered; -#endif #endif /* CONFIG_PNP */ struct snd_card_cs4236 { @@ -145,11 +136,10 @@ struct snd_card_cs4236 { #ifdef CONFIG_PNP -#ifdef CS4232 /* * PNP BIOS */ -static const struct pnp_device_id snd_cs4232_pnpbiosids[] = { +static const struct pnp_device_id snd_cs423x_pnpbiosids[] = { { .id = "CSC0100" }, { .id = "CSC0000" }, /* Guillemot Turtlebeach something appears to be cs4232 compatible @@ -157,10 +147,8 @@ static const struct pnp_device_id snd_cs4232_pnpbiosids[] = { { .id = "GIM0100" }, { .id = "" } }; -MODULE_DEVICE_TABLE(pnp, snd_cs4232_pnpbiosids); -#endif /* CS4232 */ +MODULE_DEVICE_TABLE(pnp, snd_cs423x_pnpbiosids); -#ifdef CS4232 #define CS423X_ISAPNP_DRIVER "cs4232_isapnp" static struct pnp_card_device_id snd_cs423x_pnpids[] = { /* Philips PCA70PS */ @@ -179,12 +167,6 @@ static struct pnp_card_device_id snd_cs423x_pnpids[] = { { .id = "CSCf032", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } }, /* Netfinity 3000 on-board soundcard */ { .id = "CSCe825", .devs = { { "CSC0100" }, { "CSC0110" }, { "CSC010f" } } }, - /* --- */ - { .id = "" } /* end */ -}; -#else /* CS4236 */ -#define CS423X_ISAPNP_DRIVER "cs4236_isapnp" -static struct pnp_card_device_id snd_cs423x_pnpids[] = { /* Intel Marlin Spike Motherboard - CS4235 */ { .id = "CSC0225", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } }, /* Intel Marlin Spike Motherboard (#2) - CS4235 */ @@ -266,7 +248,6 @@ static struct pnp_card_device_id snd_cs423x_pnpids[] = { /* --- */ { .id = "" } /* end */ }; -#endif MODULE_DEVICE_TABLE(pnp_card, snd_cs423x_pnpids); @@ -323,17 +304,19 @@ static int __devinit snd_cs423x_pnp_init_mpu(int dev, struct pnp_dev *pdev) return 0; } -#ifdef CS4232 -static int __devinit snd_card_cs4232_pnp(int dev, struct snd_card_cs4236 *acard, - struct pnp_dev *pdev) +static int __devinit snd_card_cs423x_pnp(int dev, struct snd_card_cs4236 *acard, + struct pnp_dev *pdev, + struct pnp_dev *cdev) { acard->wss = pdev; if (snd_cs423x_pnp_init_wss(dev, acard->wss) < 0) return -EBUSY; - cport[dev] = -1; + if (cdev) + cport[dev] = pnp_port_start(cdev, 0); + else + cport[dev] = -1; return 0; } -#endif static int __devinit snd_card_cs423x_pnpc(int dev, struct snd_card_cs4236 *acard, struct pnp_card_link *card, @@ -411,40 +394,39 @@ static int __devinit snd_cs423x_probe(struct snd_card *card, int dev) return -EBUSY; } -#ifdef CS4232 err = snd_wss_create(card, port[dev], cport[dev], irq[dev], dma1[dev], dma2[dev], - WSS_HW_DETECT, 0, &chip); - if (err < 0) - return err; - acard->chip = chip; - - err = snd_wss_pcm(chip, 0, &pcm); - if (err < 0) - return err; - - err = snd_wss_mixer(chip); + WSS_HW_DETECT3, 0, &chip); if (err < 0) return err; - -#else /* CS4236 */ - err = snd_cs4236_create(card, - port[dev], cport[dev], - irq[dev], dma1[dev], dma2[dev], - WSS_HW_DETECT, 0, &chip); - if (err < 0) - return err; - acard->chip = chip; - - err = snd_cs4236_pcm(chip, 0, &pcm); - if (err < 0) - return err; - - err = snd_cs4236_mixer(chip); - if (err < 0) - return err; -#endif + if (chip->hardware & WSS_HW_CS4236B_MASK) { + snd_wss_free(chip); + err = snd_cs4236_create(card, + port[dev], cport[dev], + irq[dev], dma1[dev], dma2[dev], + WSS_HW_DETECT, 0, &chip); + if (err < 0) + return err; + acard->chip = chip; + + err = snd_cs4236_pcm(chip, 0, &pcm); + if (err < 0) + return err; + + err = snd_cs4236_mixer(chip); + if (err < 0) + return err; + } else { + acard->chip = chip; + err = snd_wss_pcm(chip, 0, &pcm); + if (err < 0) + return err; + + err = snd_wss_mixer(chip); + if (err < 0) + return err; + } strcpy(card->driver, pcm->name); strcpy(card->shortname, pcm->name); sprintf(card->longname, "%s at 0x%lx, irq %i, dma %i", @@ -579,13 +561,14 @@ static struct isa_driver cs423x_isa_driver = { #ifdef CONFIG_PNP -#ifdef CS4232 -static int __devinit snd_cs4232_pnpbios_detect(struct pnp_dev *pdev, +static int __devinit snd_cs423x_pnpbios_detect(struct pnp_dev *pdev, const struct pnp_device_id *id) { static int dev; int err; struct snd_card *card; + struct pnp_dev *cdev; + char cid[PNP_ID_LEN]; if (pnp_device_is_isapnp(pdev)) return -ENOENT; /* we have another procedure - card */ @@ -596,10 +579,19 @@ static int __devinit snd_cs4232_pnpbios_detect(struct pnp_dev *pdev, if (dev >= SNDRV_CARDS) return -ENODEV; + /* prepare second id */ + strcpy(cid, pdev->id[0].id); + cid[5] = '1'; + cdev = NULL; + list_for_each_entry(cdev, &(pdev->protocol->devices), protocol_list) { + if (!strcmp(cdev->id[0].id, cid)) + break; + } err = snd_cs423x_card_new(dev, &card); if (err < 0) return err; - if ((err = snd_card_cs4232_pnp(dev, card->private_data, pdev)) < 0) { + err = snd_card_cs423x_pnp(dev, card->private_data, pdev, cdev); + if (err < 0) { printk(KERN_ERR "PnP BIOS detection failed for " IDENT "\n"); snd_card_free(card); return err; @@ -614,35 +606,34 @@ static int __devinit snd_cs4232_pnpbios_detect(struct pnp_dev *pdev, return 0; } -static void __devexit snd_cs4232_pnp_remove(struct pnp_dev * pdev) +static void __devexit snd_cs423x_pnp_remove(struct pnp_dev *pdev) { snd_card_free(pnp_get_drvdata(pdev)); pnp_set_drvdata(pdev, NULL); } #ifdef CONFIG_PM -static int snd_cs4232_pnp_suspend(struct pnp_dev *pdev, pm_message_t state) +static int snd_cs423x_pnp_suspend(struct pnp_dev *pdev, pm_message_t state) { return snd_cs423x_suspend(pnp_get_drvdata(pdev)); } -static int snd_cs4232_pnp_resume(struct pnp_dev *pdev) +static int snd_cs423x_pnp_resume(struct pnp_dev *pdev) { return snd_cs423x_resume(pnp_get_drvdata(pdev)); } #endif -static struct pnp_driver cs4232_pnp_driver = { - .name = "cs4232-pnpbios", - .id_table = snd_cs4232_pnpbiosids, - .probe = snd_cs4232_pnpbios_detect, - .remove = __devexit_p(snd_cs4232_pnp_remove), +static struct pnp_driver cs423x_pnp_driver = { + .name = "cs423x-pnpbios", + .id_table = snd_cs423x_pnpbiosids, + .probe = snd_cs423x_pnpbios_detect, + .remove = __devexit_p(snd_cs423x_pnp_remove), #ifdef CONFIG_PM - .suspend = snd_cs4232_pnp_suspend, - .resume = snd_cs4232_pnp_resume, + .suspend = snd_cs423x_pnp_suspend, + .resume = snd_cs423x_pnp_resume, #endif }; -#endif /* CS4232 */ static int __devinit snd_cs423x_pnpc_detect(struct pnp_card_link *pcard, const struct pnp_card_device_id *pid) @@ -716,18 +707,14 @@ static int __init alsa_card_cs423x_init(void) #ifdef CONFIG_PNP if (!err) isa_registered = 1; -#ifdef CS4232 - err = pnp_register_driver(&cs4232_pnp_driver); + err = pnp_register_driver(&cs423x_pnp_driver); if (!err) pnp_registered = 1; -#endif err = pnp_register_card_driver(&cs423x_pnpc_driver); if (!err) pnpc_registered = 1; -#ifdef CS4232 if (pnp_registered) err = 0; -#endif if (isa_registered) err = 0; #endif @@ -739,10 +726,8 @@ static void __exit alsa_card_cs423x_exit(void) #ifdef CONFIG_PNP if (pnpc_registered) pnp_unregister_card_driver(&cs423x_pnpc_driver); -#ifdef CS4232 if (pnp_registered) - pnp_unregister_driver(&cs4232_pnp_driver); -#endif + pnp_unregister_driver(&cs423x_pnp_driver); if (isa_registered) #endif isa_unregister_driver(&cs423x_isa_driver); diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c index 8de5deda7ad..ac27832b2c6 100644 --- a/sound/isa/wss/wss_lib.c +++ b/sound/isa/wss/wss_lib.c @@ -1657,7 +1657,7 @@ static void snd_wss_resume(struct snd_wss *chip) } #endif /* CONFIG_PM */ -static int snd_wss_free(struct snd_wss *chip) +int snd_wss_free(struct snd_wss *chip) { release_and_free_resource(chip->res_port); release_and_free_resource(chip->res_cport); @@ -1680,6 +1680,7 @@ static int snd_wss_free(struct snd_wss *chip) kfree(chip); return 0; } +EXPORT_SYMBOL(snd_wss_free); static int snd_wss_dev_free(struct snd_device *device) { -- cgit v1.2.3-70-g09d2 From c844a5d38e4247fc71e371221cf762a2a44d565b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 16 Feb 2009 23:17:33 +0100 Subject: ALSA: Fix documentation for snd-cs4236 driver Updated; removal of snd-cs4232 entry. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/ALSA-Configuration.txt | 30 +++++-------------------- 1 file changed, 5 insertions(+), 25 deletions(-) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index a763b76afe5..57fe4f3ca2c 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -391,34 +391,11 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. The power-management is supported. - Module snd-cs4232 - ----------------- - - Module for sound cards based on CS4232/CS4232A ISA chips. - - isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) - - with isapnp=0, the following options are available: - - port - port # for CS4232 chip (PnP setup - 0x534) - cport - control port # for CS4232 chip (PnP setup - 0x120,0x210,0xf00) - mpu_port - port # for MPU-401 UART (PnP setup - 0x300), -1 = disable - fm_port - FM port # for CS4232 chip (PnP setup - 0x388), -1 = disable - irq - IRQ # for CS4232 chip (5,7,9,11,12,15) - mpu_irq - IRQ # for MPU-401 UART (9,11,12,15) - dma1 - first DMA # for CS4232 chip (0,1,3) - dma2 - second DMA # for Yamaha CS4232 chip (0,1,3), -1 = disable - - This module supports multiple cards. This module does not support autoprobe - (if ISA PnP is not used) thus main port must be specified!!! Other ports are - optional. - - The power-management is supported. - Module snd-cs4236 ----------------- - Module for sound cards based on CS4235/CS4236/CS4236B/CS4237B/ + Module for sound cards based on CS4232/CS4232A, + CS4235/CS4236/CS4236B/CS4237B/ CS4238B/CS4239 ISA chips. isapnp - ISA PnP detection - 0 = disable, 1 = enable (default) @@ -440,6 +417,9 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. The power-management is supported. + This module is aliased as snd-cs4232 since it provides the old + snd-cs4232 functionality, too. + Module snd-cs4281 ----------------- -- cgit v1.2.3-70-g09d2 From 83807400794a1d680a4fb70a610c5f486e734f45 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 17 Feb 2009 07:59:40 +0100 Subject: ALSA: au88x0 - Fix &&|| typo MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Fixed a typo of || and &&. As it's in a disabled code section, there is no behavior change, though. Reported-by: Jörg-Volker Peetz Signed-off-by: Takashi Iwai --- sound/pci/au88x0/au88x0_core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c index e6a04d037c1..3906f5afe27 100644 --- a/sound/pci/au88x0/au88x0_core.c +++ b/sound/pci/au88x0/au88x0_core.c @@ -2800,7 +2800,7 @@ vortex_translateformat(vortex_t * vortex, char bits, char nch, int encod) { int a, this_194; - if ((bits != 8) || (bits != 16)) + if ((bits != 8) && (bits != 16)) return -1; switch (encod) { -- cgit v1.2.3-70-g09d2 From b22f5d94c432e97df8d85151fcf3da16cee75f04 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 17 Feb 2009 08:02:16 +0100 Subject: sound: OSS: ad1848 - Fix another typo MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Fix another typo of || and &&. Reported-by: Jörg-Volker Peetz Signed-off-by: Takashi Iwai --- sound/oss/ad1848.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/oss/ad1848.c b/sound/oss/ad1848.c index a5b83568bdc..d12bd98a37b 100644 --- a/sound/oss/ad1848.c +++ b/sound/oss/ad1848.c @@ -280,7 +280,7 @@ static void wait_for_calibration(ad1848_info * devc) while (timeout > 0 && (ad_read(devc, 11) & 0x20)) timeout--; if (ad_read(devc, 11) & 0x20) - if ( (devc->model != MD_1845) || (devc->model != MD_1845_SSCAPE)) + if ((devc->model != MD_1845) && (devc->model != MD_1845_SSCAPE)) printk(KERN_WARNING "ad1848: Auto calibration timed out(3).\n"); } -- cgit v1.2.3-70-g09d2 From cda9043d56cee9fea39e4ee33fd605ae477a1950 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 17 Feb 2009 08:10:54 +0100 Subject: ALSA: cs4236 - Merge snd-cs4236-lib module into snd-cs4236 Since cs4232 and cs4236 drivers are merged, there is no reason to keep snd-cs4236-lib module separately. Let's merge it into the main driver as well. Signed-off-by: Takashi Iwai --- sound/isa/cs423x/Makefile | 6 +++--- sound/isa/cs423x/cs4236_lib.c | 24 ------------------------ 2 files changed, 3 insertions(+), 27 deletions(-) diff --git a/sound/isa/cs423x/Makefile b/sound/isa/cs423x/Makefile index 732f66cc036..6d397e8d54a 100644 --- a/sound/isa/cs423x/Makefile +++ b/sound/isa/cs423x/Makefile @@ -3,11 +3,11 @@ # Copyright (c) 2001 by Jaroslav Kysela # -snd-cs4236-lib-objs := cs4236_lib.o snd-cs4231-objs := cs4231.o -snd-cs4236-objs := cs4236.o +snd-cs4236-objs := cs4236.o cs4236_lib.o # Toplevel Module Dependency obj-$(CONFIG_SND_CS4231) += snd-cs4231.o -obj-$(CONFIG_SND_CS4236) += snd-cs4236.o snd-cs4236-lib.o +obj-$(CONFIG_SND_CS4236) += snd-cs4236.o + diff --git a/sound/isa/cs423x/cs4236_lib.c b/sound/isa/cs423x/cs4236_lib.c index 2406efdfd8d..38835f31298 100644 --- a/sound/isa/cs423x/cs4236_lib.c +++ b/sound/isa/cs423x/cs4236_lib.c @@ -88,10 +88,6 @@ #include #include -MODULE_AUTHOR("Jaroslav Kysela "); -MODULE_DESCRIPTION("Routines for control of CS4235/4236B/4237B/4238B/4239 chips"); -MODULE_LICENSE("GPL"); - /* * */ @@ -1022,23 +1018,3 @@ int snd_cs4236_mixer(struct snd_wss *chip) } return 0; } - -EXPORT_SYMBOL(snd_cs4236_create); -EXPORT_SYMBOL(snd_cs4236_pcm); -EXPORT_SYMBOL(snd_cs4236_mixer); - -/* - * INIT part - */ - -static int __init alsa_cs4236_init(void) -{ - return 0; -} - -static void __exit alsa_cs4236_exit(void) -{ -} - -module_init(alsa_cs4236_init) -module_exit(alsa_cs4236_exit) -- cgit v1.2.3-70-g09d2 From 31b59cf9cebb5bb675f49fe44814bbb7270374cc Mon Sep 17 00:00:00 2001 From: Paul Fertser Date: Mon, 16 Feb 2009 02:49:41 +0300 Subject: ASoC: Fix WM8753 DAIs unregistering WM8753 uses a tricky way to switch DAIs "on the fly", for that it registers 2 dummy DAIs and substitutes them depending on mixer control. List element of registered dummy DAIs should be preserved to allow unregistering of DAIs on module unload. Signed-off-by: Paul Fertser Signed-off-by: Mark Brown --- sound/soc/codecs/wm8753.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 6f9e6beabb1..dc6042c6424 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1451,30 +1451,35 @@ static void wm8753_set_dai_mode(struct snd_soc_codec *codec, unsigned int mode) if (mode < 4) { int playback_active, capture_active, codec_active, pop_wait; void *private_data; + struct list_head list; playback_active = wm8753_dai[0].playback.active; capture_active = wm8753_dai[0].capture.active; codec_active = wm8753_dai[0].active; private_data = wm8753_dai[0].private_data; pop_wait = wm8753_dai[0].pop_wait; + list = wm8753_dai[0].list; wm8753_dai[0] = wm8753_all_dai[mode << 1]; wm8753_dai[0].playback.active = playback_active; wm8753_dai[0].capture.active = capture_active; wm8753_dai[0].active = codec_active; wm8753_dai[0].private_data = private_data; wm8753_dai[0].pop_wait = pop_wait; + wm8753_dai[0].list = list; playback_active = wm8753_dai[1].playback.active; capture_active = wm8753_dai[1].capture.active; codec_active = wm8753_dai[1].active; private_data = wm8753_dai[1].private_data; pop_wait = wm8753_dai[1].pop_wait; + list = wm8753_dai[1].list; wm8753_dai[1] = wm8753_all_dai[(mode << 1) + 1]; wm8753_dai[1].playback.active = playback_active; wm8753_dai[1].capture.active = capture_active; wm8753_dai[1].active = codec_active; wm8753_dai[1].private_data = private_data; wm8753_dai[1].pop_wait = pop_wait; + wm8753_dai[1].list = list; } wm8753_dai[0].codec = codec; wm8753_dai[1].codec = codec; -- cgit v1.2.3-70-g09d2 From 7b317b692a03a870d7acda0a0bd4d211f1c064fe Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 16 Feb 2009 14:08:22 +0000 Subject: ASoC: Remove version display from the WM8731 driver It makes boot a bit more noisy and I never remember to update it. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8731.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 0150fe53a65..816e5bf9edc 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -29,8 +29,6 @@ #include "wm8731.h" -#define WM8731_VERSION "0.13" - struct snd_soc_codec_device soc_codec_dev_wm8731; /* codec private data */ @@ -702,8 +700,6 @@ static int wm8731_probe(struct platform_device *pdev) struct wm8731_priv *wm8731; int ret = 0; - pr_info("WM8731 Audio Codec %s", WM8731_VERSION); - setup = socdev->codec_data; codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); if (codec == NULL) -- cgit v1.2.3-70-g09d2 From 22d22ee5146ae823b1e93fe2887a7cba56015091 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 16 Feb 2009 19:20:15 +0000 Subject: ASoC: Clean up WM8731 bias level configuration The WM8731 bias level configuration function was written slightly obscurely - streamline the code a little and refresh the comments. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8731.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 816e5bf9edc..c6db6779377 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -396,21 +396,19 @@ static int wm8731_set_dai_fmt(struct snd_soc_dai *codec_dai, static int wm8731_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 reg = wm8731_read_reg_cache(codec, WM8731_PWR) & 0xff7f; + u16 reg; switch (level) { case SND_SOC_BIAS_ON: - /* vref/mid, osc on, dac unmute */ - wm8731_write(codec, WM8731_PWR, reg); break; case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: - /* everything off except vref/vmid, */ + /* Clear PWROFF, gate CLKOUT, everything else as-is */ + reg = wm8731_read_reg_cache(codec, WM8731_PWR) & 0xff7f; wm8731_write(codec, WM8731_PWR, reg | 0x0040); break; case SND_SOC_BIAS_OFF: - /* everything off, dac mute, inactive */ wm8731_write(codec, WM8731_ACTIVE, 0x0); wm8731_write(codec, WM8731_PWR, 0xffff); break; -- cgit v1.2.3-70-g09d2 From c16159123d5b3245e2b30023a207606c74032f9c Mon Sep 17 00:00:00 2001 From: Roel Kluin Date: Wed, 18 Feb 2009 10:15:00 +0100 Subject: sound: OSS: missing parentheses in pas2_card.c Add missing parentheses in pas2_card.c. Signed-off-by: Roel Kluin Signed-off-by: Takashi Iwai --- sound/oss/pas2_card.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/oss/pas2_card.c b/sound/oss/pas2_card.c index 25f3a22c52e..7f377ec3486 100644 --- a/sound/oss/pas2_card.c +++ b/sound/oss/pas2_card.c @@ -156,9 +156,7 @@ static int __init config_pas_hw(struct address_info *hw_config) * 0x80 */ , 0xB88); - pas_write(0x80 - | joystick?0x40:0 - ,0xF388); + pas_write(0x80 | (joystick ? 0x40 : 0), 0xF388); if (pas_irq < 0 || pas_irq > 15) { -- cgit v1.2.3-70-g09d2 From d6943541158985030108e4a0a483cdadc3c80ee1 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 16 Feb 2009 13:38:11 +0000 Subject: ASoC: Improve diagnostics for AT91SAM9G20-EK probe We should display an error by default if we fail to register. Signed-off-by: Mark Brown --- sound/soc/atmel/sam9g20_wm8731.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index 6ea04be911d..be3f923d3c4 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -273,6 +273,7 @@ static int __init at91sam9g20ek_init(void) */ ssc = ssc_request(0); if (IS_ERR(ssc)) { + printk(KERN_ERR "ASoC: Failed to request SSC 0\n"); ret = PTR_ERR(ssc); ssc = NULL; goto err_ssc; @@ -281,8 +282,7 @@ static int __init at91sam9g20ek_init(void) at91sam9g20ek_snd_device = platform_device_alloc("soc-audio", -1); if (!at91sam9g20ek_snd_device) { - printk(KERN_DEBUG - "platform device allocation failed\n"); + printk(KERN_ERR "ASoC: Platform device allocation failed\n"); ret = -ENOMEM; } @@ -292,8 +292,7 @@ static int __init at91sam9g20ek_init(void) ret = platform_device_add(at91sam9g20ek_snd_device); if (ret) { - printk(KERN_DEBUG - "platform device allocation failed\n"); + printk(KERN_ERR "ASoC: Platform device allocation failed\n"); platform_device_put(at91sam9g20ek_snd_device); } -- cgit v1.2.3-70-g09d2 From 40135ea07190316a789b2edfbf7c8131598bdf81 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 16 Feb 2009 16:04:05 +0000 Subject: ASoC: Check machine type before loading on AT91SAM9G20-EK Signed-off-by: Mark Brown --- sound/soc/atmel/sam9g20_wm8731.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index be3f923d3c4..b7efdc8119d 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -45,6 +45,7 @@ #include #include +#include #include #include @@ -268,6 +269,9 @@ static int __init at91sam9g20ek_init(void) struct ssc_device *ssc = NULL; int ret; + if (!machine_is_at91sam9g20ek()) + return -ENODEV; + /* * Request SSC device */ -- cgit v1.2.3-70-g09d2 From 5de7f9b20069257aa5f0bb74723c8603adc5841a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 16 Feb 2009 17:51:54 +0000 Subject: ASoC: Actively manage MCLK for AT91SAM9G20-EK We have software control of the MCLK for the WM8731 so save a bit of power by actively managing it within the machine driver, enabling it only while the codec is active. Once ASoC supports multiple boards and doesn't require the soc-audio device the initial clock setup should be pushed down into the arch/arm code but for now this reduces merge issues. Tested-by: Sedji Gaouaou Signed-off-by: Mark Brown --- sound/soc/atmel/sam9g20_wm8731.c | 68 ++++++++++++++++++++++++++++++++++++++-- 1 file changed, 65 insertions(+), 3 deletions(-) diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index b7efdc8119d..ab32514a858 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -53,6 +53,9 @@ #include "atmel-pcm.h" #include "atmel_ssc_dai.h" +#define MCLK_RATE 12000000 + +static struct clk *mclk; static int at91sam9g20ek_startup(struct snd_pcm_substream *substream) { @@ -60,11 +63,12 @@ static int at91sam9g20ek_startup(struct snd_pcm_substream *substream) struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; int ret; - /* codec system clock is supplied by PCK0, set to 12MHz */ ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, - 12000000, SND_SOC_CLOCK_IN); - if (ret < 0) + MCLK_RATE, SND_SOC_CLOCK_IN); + if (ret < 0) { + clk_disable(mclk); return ret; + } return 0; } @@ -190,6 +194,31 @@ static struct snd_soc_ops at91sam9g20ek_ops = { .shutdown = at91sam9g20ek_shutdown, }; +static int at91sam9g20ek_set_bias_level(struct snd_soc_card *card, + enum snd_soc_bias_level level) +{ + static int mclk_on; + int ret = 0; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + if (!mclk_on) + ret = clk_enable(mclk); + if (ret == 0) + mclk_on = 1; + break; + + case SND_SOC_BIAS_OFF: + case SND_SOC_BIAS_STANDBY: + if (mclk_on) + clk_disable(mclk); + mclk_on = 0; + break; + } + + return ret; +} static const struct snd_soc_dapm_widget at91sam9g20ek_dapm_widgets[] = { SND_SOC_DAPM_MIC("Int Mic", NULL), @@ -248,6 +277,7 @@ static struct snd_soc_card snd_soc_at91sam9g20ek = { .platform = &atmel_soc_platform, .dai_link = &at91sam9g20ek_dai, .num_links = 1, + .set_bias_level = at91sam9g20ek_set_bias_level, }; static struct wm8731_setup_data at91sam9g20ek_wm8731_setup = { @@ -267,11 +297,37 @@ static int __init at91sam9g20ek_init(void) { struct atmel_ssc_info *ssc_p = at91sam9g20ek_dai.cpu_dai->private_data; struct ssc_device *ssc = NULL; + struct clk *pllb; int ret; if (!machine_is_at91sam9g20ek()) return -ENODEV; + /* + * Codec MCLK is supplied by PCK0 - set it up. + */ + mclk = clk_get(NULL, "pck0"); + if (IS_ERR(mclk)) { + printk(KERN_ERR "ASoC: Failed to get MCLK\n"); + ret = PTR_ERR(mclk); + goto err; + } + + pllb = clk_get(NULL, "pllb"); + if (IS_ERR(mclk)) { + printk(KERN_ERR "ASoC: Failed to get PLLB\n"); + ret = PTR_ERR(mclk); + goto err_mclk; + } + ret = clk_set_parent(mclk, pllb); + clk_put(pllb); + if (ret != 0) { + printk(KERN_ERR "ASoC: Failed to set MCLK parent\n"); + goto err_mclk; + } + + clk_set_rate(mclk, MCLK_RATE); + /* * Request SSC device */ @@ -303,6 +359,10 @@ static int __init at91sam9g20ek_init(void) return ret; err_ssc: +err_mclk: + clk_put(mclk); + mclk = NULL; +err: return ret; } @@ -320,6 +380,8 @@ static void __exit at91sam9g20ek_exit(void) platform_device_unregister(at91sam9g20ek_snd_device); at91sam9g20ek_snd_device = NULL; + clk_put(mclk); + mclk = NULL; } module_init(at91sam9g20ek_init); -- cgit v1.2.3-70-g09d2 From 7ee753804185eb0a46ac964fd6a6564bd67290c9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 16 Feb 2009 18:00:58 +0000 Subject: ASoC: Rename AT91SAMG20-EK for applications This is a bit more idiomatic and makes identifying a configuration based on the board type work better. Signed-off-by: Mark Brown --- sound/soc/atmel/sam9g20_wm8731.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index ab32514a858..aa524235fd9 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -273,7 +273,7 @@ static struct snd_soc_dai_link at91sam9g20ek_dai = { }; static struct snd_soc_card snd_soc_at91sam9g20ek = { - .name = "WM8731", + .name = "AT91SAMG20-EK", .platform = &atmel_soc_platform, .dai_link = &at91sam9g20ek_dai, .num_links = 1, -- cgit v1.2.3-70-g09d2 From a8035c8f04477895207b92915b908344749be336 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 16 Feb 2009 19:35:43 +0000 Subject: ASoC: Shuffle WM8731 SPI and I2C device registration This is a pure code motion patch intended to improve reviewability of a following patch moving WM8731 to use more standard device registration. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8731.c | 215 ++++++++++++++++++++++++---------------------- 1 file changed, 112 insertions(+), 103 deletions(-) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index c6db6779377..3ff971aeba2 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -36,6 +36,11 @@ struct wm8731_priv { unsigned int sysclk; }; +#ifdef CONFIG_SPI_MASTER +static int wm8731_spi_write(struct spi_device *spi, const char *data, int len); +static struct spi_driver wm8731_spi_driver; +#endif + /* * wm8731 register cache * We can't read the WM8731 register space when we are @@ -544,54 +549,9 @@ pcm_err: static struct snd_soc_device *wm8731_socdev; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - -/* - * WM8731 2 wire address is determined by GPIO5 - * state during powerup. - * low = 0x1a - * high = 0x1b - */ - -static int wm8731_i2c_probe(struct i2c_client *i2c, - const struct i2c_device_id *id) -{ - struct snd_soc_device *socdev = wm8731_socdev; - struct snd_soc_codec *codec = socdev->card->codec; - int ret; - - i2c_set_clientdata(i2c, codec); - codec->control_data = i2c; - - ret = wm8731_init(socdev); - if (ret < 0) - pr_err("failed to initialise WM8731\n"); - - return ret; -} -static int wm8731_i2c_remove(struct i2c_client *client) -{ - struct snd_soc_codec *codec = i2c_get_clientdata(client); - kfree(codec->reg_cache); - return 0; -} - -static const struct i2c_device_id wm8731_i2c_id[] = { - { "wm8731", 0 }, - { } -}; -MODULE_DEVICE_TABLE(i2c, wm8731_i2c_id); - -static struct i2c_driver wm8731_i2c_driver = { - .driver = { - .name = "WM8731 I2C Codec", - .owner = THIS_MODULE, - }, - .probe = wm8731_i2c_probe, - .remove = wm8731_i2c_remove, - .id_table = wm8731_i2c_id, -}; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static struct i2c_driver wm8731_i2c_driver; static int wm8731_add_i2c_device(struct platform_device *pdev, const struct wm8731_setup_data *setup) @@ -634,62 +594,6 @@ err_driver: } #endif -#if defined(CONFIG_SPI_MASTER) -static int __devinit wm8731_spi_probe(struct spi_device *spi) -{ - struct snd_soc_device *socdev = wm8731_socdev; - struct snd_soc_codec *codec = socdev->card->codec; - int ret; - - codec->control_data = spi; - - ret = wm8731_init(socdev); - if (ret < 0) - dev_err(&spi->dev, "failed to initialise WM8731\n"); - - return ret; -} - -static int __devexit wm8731_spi_remove(struct spi_device *spi) -{ - return 0; -} - -static struct spi_driver wm8731_spi_driver = { - .driver = { - .name = "wm8731", - .bus = &spi_bus_type, - .owner = THIS_MODULE, - }, - .probe = wm8731_spi_probe, - .remove = __devexit_p(wm8731_spi_remove), -}; - -static int wm8731_spi_write(struct spi_device *spi, const char *data, int len) -{ - struct spi_transfer t; - struct spi_message m; - u8 msg[2]; - - if (len <= 0) - return 0; - - msg[0] = data[0]; - msg[1] = data[1]; - - spi_message_init(&m); - memset(&t, 0, (sizeof t)); - - t.tx_buf = &msg[0]; - t.len = len; - - spi_message_add_tail(&t, &m); - spi_sync(spi, &m); - - return len; -} -#endif /* CONFIG_SPI_MASTER */ - static int wm8731_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); @@ -772,6 +676,111 @@ struct snd_soc_codec_device soc_codec_dev_wm8731 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_wm8731); +#if defined(CONFIG_SPI_MASTER) +static int __devinit wm8731_spi_probe(struct spi_device *spi) +{ + struct snd_soc_device *socdev = wm8731_socdev; + struct snd_soc_codec *codec = socdev->card->codec; + int ret; + + codec->control_data = spi; + + ret = wm8731_init(socdev); + if (ret < 0) + dev_err(&spi->dev, "failed to initialise WM8731\n"); + + return ret; +} + +static int __devexit wm8731_spi_remove(struct spi_device *spi) +{ + return 0; +} + +static struct spi_driver wm8731_spi_driver = { + .driver = { + .name = "wm8731", + .bus = &spi_bus_type, + .owner = THIS_MODULE, + }, + .probe = wm8731_spi_probe, + .remove = __devexit_p(wm8731_spi_remove), +}; + +static int wm8731_spi_write(struct spi_device *spi, const char *data, int len) +{ + struct spi_transfer t; + struct spi_message m; + u8 msg[2]; + + if (len <= 0) + return 0; + + msg[0] = data[0]; + msg[1] = data[1]; + + spi_message_init(&m); + memset(&t, 0, (sizeof t)); + + t.tx_buf = &msg[0]; + t.len = len; + + spi_message_add_tail(&t, &m); + spi_sync(spi, &m); + + return len; +} +#endif /* CONFIG_SPI_MASTER */ + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +/* + * WM8731 2 wire address is determined by GPIO5 + * state during powerup. + * low = 0x1a + * high = 0x1b + */ + +static int wm8731_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct snd_soc_device *socdev = wm8731_socdev; + struct snd_soc_codec *codec = socdev->card->codec; + int ret; + + i2c_set_clientdata(i2c, codec); + codec->control_data = i2c; + + ret = wm8731_init(socdev); + if (ret < 0) + pr_err("failed to initialise WM8731\n"); + + return ret; +} + +static int wm8731_i2c_remove(struct i2c_client *client) +{ + struct snd_soc_codec *codec = i2c_get_clientdata(client); + kfree(codec->reg_cache); + return 0; +} + +static const struct i2c_device_id wm8731_i2c_id[] = { + { "wm8731", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8731_i2c_id); + +static struct i2c_driver wm8731_i2c_driver = { + .driver = { + .name = "WM8731 I2C Codec", + .owner = THIS_MODULE, + }, + .probe = wm8731_i2c_probe, + .remove = wm8731_i2c_remove, + .id_table = wm8731_i2c_id, +}; +#endif + static int __init wm8731_modinit(void) { return snd_soc_register_dai(&wm8731_dai); -- cgit v1.2.3-70-g09d2 From 4dd3a29f295799295eac819bbf540690fbe30c16 Mon Sep 17 00:00:00 2001 From: Yoichi Yuasa Date: Wed, 18 Feb 2009 19:09:23 +0900 Subject: sound: fix opensound URL in oss Introduction Signed-off-by: Yoichi Yuasa Signed-off-by: Takashi Iwai --- Documentation/sound/oss/Introduction | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/sound/oss/Introduction b/Documentation/sound/oss/Introduction index f04ba6bb739..75d967ff926 100644 --- a/Documentation/sound/oss/Introduction +++ b/Documentation/sound/oss/Introduction @@ -80,7 +80,7 @@ Notes: additional features. 2. The commercial OSS driver may be obtained from the site: - http://www/opensound.com. This may be used for cards that + http://www.opensound.com. This may be used for cards that are unsupported by the kernel driver, or may be used by other operating systems. -- cgit v1.2.3-70-g09d2 From 5998102b9095fdb7c67755812038612afea315c5 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 16 Feb 2009 20:49:16 +0000 Subject: ASoC: Refactor WM8731 device registration Move the WM8731 driver to use a more standard device registration scheme where the device can be registered independantly of the ASoC probe. As a transition measure push the current manual code for registering the WM8731 into the individual machine driver probes. This allows separate patches to update the relevant architecture files with less risk of merge issues. Signed-off-by: Mark Brown --- sound/soc/atmel/sam9g20_wm8731.c | 43 ++++- sound/soc/codecs/wm8731.c | 334 +++++++++++++++++---------------------- sound/soc/codecs/wm8731.h | 6 - sound/soc/pxa/corgi.c | 43 ++++- sound/soc/pxa/poodle.c | 41 ++++- 5 files changed, 256 insertions(+), 211 deletions(-) diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index aa524235fd9..173a239a541 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -36,6 +36,7 @@ #include #include #include +#include #include @@ -280,15 +281,41 @@ static struct snd_soc_card snd_soc_at91sam9g20ek = { .set_bias_level = at91sam9g20ek_set_bias_level, }; -static struct wm8731_setup_data at91sam9g20ek_wm8731_setup = { - .i2c_bus = 0, - .i2c_address = 0x1b, -}; +/* + * FIXME: This is a temporary bodge to avoid cross-tree merge issues. + * New drivers should register the wm8731 I2C device in the machine + * setup code (under arch/arm for ARM systems). + */ +static int wm8731_i2c_register(void) +{ + struct i2c_board_info info; + struct i2c_adapter *adapter; + struct i2c_client *client; + + memset(&info, 0, sizeof(struct i2c_board_info)); + info.addr = 0x1b; + strlcpy(info.type, "wm8731", I2C_NAME_SIZE); + + adapter = i2c_get_adapter(0); + if (!adapter) { + printk(KERN_ERR "can't get i2c adapter 0\n"); + return -ENODEV; + } + + client = i2c_new_device(adapter, &info); + i2c_put_adapter(adapter); + if (!client) { + printk(KERN_ERR "can't add i2c device at 0x%x\n", + (unsigned int)info.addr); + return -ENODEV; + } + + return 0; +} static struct snd_soc_device at91sam9g20ek_snd_devdata = { .card = &snd_soc_at91sam9g20ek, .codec_dev = &soc_codec_dev_wm8731, - .codec_data = &at91sam9g20ek_wm8731_setup, }; static struct platform_device *at91sam9g20ek_snd_device; @@ -340,6 +367,10 @@ static int __init at91sam9g20ek_init(void) } ssc_p->ssc = ssc; + ret = wm8731_i2c_register(); + if (ret != 0) + goto err_ssc; + at91sam9g20ek_snd_device = platform_device_alloc("soc-audio", -1); if (!at91sam9g20ek_snd_device) { printk(KERN_ERR "ASoC: Platform device allocation failed\n"); @@ -359,6 +390,8 @@ static int __init at91sam9g20ek_init(void) return ret; err_ssc: + ssc_free(ssc); + ssc_p->ssc = NULL; err_mclk: clk_put(mclk); mclk = NULL; diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 3ff971aeba2..a2c478e53d5 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -29,16 +29,18 @@ #include "wm8731.h" +static struct snd_soc_codec *wm8731_codec; struct snd_soc_codec_device soc_codec_dev_wm8731; /* codec private data */ struct wm8731_priv { + struct snd_soc_codec codec; + u16 reg_cache[WM8731_CACHEREGNUM]; unsigned int sysclk; }; #ifdef CONFIG_SPI_MASTER static int wm8731_spi_write(struct spi_device *spi, const char *data, int len); -static struct spi_driver wm8731_spi_driver; #endif /* @@ -485,55 +487,33 @@ static int wm8731_resume(struct platform_device *pdev) return 0; } -/* - * initialise the WM8731 driver - * register the mixer and dsp interfaces with the kernel - */ -static int wm8731_init(struct snd_soc_device *socdev) +static int wm8731_probe(struct platform_device *pdev) { - struct snd_soc_codec *codec = socdev->card->codec; - int reg, ret = 0; + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; - codec->name = "WM8731"; - codec->owner = THIS_MODULE; - codec->read = wm8731_read_reg_cache; - codec->write = wm8731_write; - codec->set_bias_level = wm8731_set_bias_level; - codec->dai = &wm8731_dai; - codec->num_dai = 1; - codec->reg_cache_size = ARRAY_SIZE(wm8731_reg); - codec->reg_cache = kmemdup(wm8731_reg, sizeof(wm8731_reg), GFP_KERNEL); - if (codec->reg_cache == NULL) - return -ENOMEM; + if (wm8731_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } - wm8731_reset(codec); + socdev->card->codec = wm8731_codec; + codec = wm8731_codec; /* register pcms */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) { - printk(KERN_ERR "wm8731: failed to create pcms\n"); + dev_err(codec->dev, "failed to create pcms: %d\n", ret); goto pcm_err; } - /* power on device */ - wm8731_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - /* set the update bits */ - reg = wm8731_read_reg_cache(codec, WM8731_LOUT1V); - wm8731_write(codec, WM8731_LOUT1V, reg & ~0x0100); - reg = wm8731_read_reg_cache(codec, WM8731_ROUT1V); - wm8731_write(codec, WM8731_ROUT1V, reg & ~0x0100); - reg = wm8731_read_reg_cache(codec, WM8731_LINVOL); - wm8731_write(codec, WM8731_LINVOL, reg & ~0x0100); - reg = wm8731_read_reg_cache(codec, WM8731_RINVOL); - wm8731_write(codec, WM8731_RINVOL, reg & ~0x0100); - snd_soc_add_controls(codec, wm8731_snd_controls, - ARRAY_SIZE(wm8731_snd_controls)); + ARRAY_SIZE(wm8731_snd_controls)); wm8731_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { - printk(KERN_ERR "wm8731: failed to register card\n"); + dev_err(codec->dev, "failed to register card: %d\n", ret); goto card_err; } @@ -543,104 +523,6 @@ card_err: snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); pcm_err: - kfree(codec->reg_cache); - return ret; -} - -static struct snd_soc_device *wm8731_socdev; - - -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) -static struct i2c_driver wm8731_i2c_driver; - -static int wm8731_add_i2c_device(struct platform_device *pdev, - const struct wm8731_setup_data *setup) -{ - struct i2c_board_info info; - struct i2c_adapter *adapter; - struct i2c_client *client; - int ret; - - ret = i2c_add_driver(&wm8731_i2c_driver); - if (ret != 0) { - dev_err(&pdev->dev, "can't add i2c driver\n"); - return ret; - } - - memset(&info, 0, sizeof(struct i2c_board_info)); - info.addr = setup->i2c_address; - strlcpy(info.type, "wm8731", I2C_NAME_SIZE); - - adapter = i2c_get_adapter(setup->i2c_bus); - if (!adapter) { - dev_err(&pdev->dev, "can't get i2c adapter %d\n", - setup->i2c_bus); - goto err_driver; - } - - client = i2c_new_device(adapter, &info); - i2c_put_adapter(adapter); - if (!client) { - dev_err(&pdev->dev, "can't add i2c device at 0x%x\n", - (unsigned int)info.addr); - goto err_driver; - } - - return 0; - -err_driver: - i2c_del_driver(&wm8731_i2c_driver); - return -ENODEV; -} -#endif - -static int wm8731_probe(struct platform_device *pdev) -{ - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct wm8731_setup_data *setup; - struct snd_soc_codec *codec; - struct wm8731_priv *wm8731; - int ret = 0; - - setup = socdev->codec_data; - codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); - if (codec == NULL) - return -ENOMEM; - - wm8731 = kzalloc(sizeof(struct wm8731_priv), GFP_KERNEL); - if (wm8731 == NULL) { - kfree(codec); - return -ENOMEM; - } - - codec->private_data = wm8731; - socdev->card->codec = codec; - mutex_init(&codec->mutex); - INIT_LIST_HEAD(&codec->dapm_widgets); - INIT_LIST_HEAD(&codec->dapm_paths); - - wm8731_socdev = socdev; - ret = -ENODEV; - -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - if (setup->i2c_address) { - codec->hw_write = (hw_write_t)i2c_master_send; - ret = wm8731_add_i2c_device(pdev, setup); - } -#endif -#if defined(CONFIG_SPI_MASTER) - if (setup->spi) { - codec->hw_write = (hw_write_t)wm8731_spi_write; - ret = spi_register_driver(&wm8731_spi_driver); - if (ret != 0) - printk(KERN_ERR "can't add spi driver"); - } -#endif - - if (ret != 0) { - kfree(codec->private_data); - kfree(codec); - } return ret; } @@ -648,22 +530,9 @@ static int wm8731_probe(struct platform_device *pdev) static int wm8731_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->card->codec; - - if (codec->control_data) - wm8731_set_bias_level(codec, SND_SOC_BIAS_OFF); snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - i2c_unregister_device(codec->control_data); - i2c_del_driver(&wm8731_i2c_driver); -#endif -#if defined(CONFIG_SPI_MASTER) - spi_unregister_driver(&wm8731_spi_driver); -#endif - kfree(codec->private_data); - kfree(codec); return 0; } @@ -676,37 +545,78 @@ struct snd_soc_codec_device soc_codec_dev_wm8731 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_wm8731); -#if defined(CONFIG_SPI_MASTER) -static int __devinit wm8731_spi_probe(struct spi_device *spi) +static int wm8731_register(struct wm8731_priv *wm8731) { - struct snd_soc_device *socdev = wm8731_socdev; - struct snd_soc_codec *codec = socdev->card->codec; int ret; + struct snd_soc_codec *codec = &wm8731->codec; + u16 reg; - codec->control_data = spi; + if (wm8731_codec) { + dev_err(codec->dev, "Another WM8731 is registered\n"); + return -EINVAL; + } - ret = wm8731_init(socdev); - if (ret < 0) - dev_err(&spi->dev, "failed to initialise WM8731\n"); + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); - return ret; -} + codec->private_data = wm8731; + codec->name = "WM8731"; + codec->owner = THIS_MODULE; + codec->read = wm8731_read_reg_cache; + codec->write = wm8731_write; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = wm8731_set_bias_level; + codec->dai = &wm8731_dai; + codec->num_dai = 1; + codec->reg_cache_size = WM8731_CACHEREGNUM; + codec->reg_cache = &wm8731->reg_cache; + + memcpy(codec->reg_cache, wm8731_reg, sizeof(wm8731_reg)); + + wm8731_dai.dev = codec->dev; + + wm8731_reset(codec); + wm8731_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + /* Latch the update bits */ + reg = wm8731_read_reg_cache(codec, WM8731_LOUT1V); + wm8731_write(codec, WM8731_LOUT1V, reg & ~0x0100); + reg = wm8731_read_reg_cache(codec, WM8731_ROUT1V); + wm8731_write(codec, WM8731_ROUT1V, reg & ~0x0100); + reg = wm8731_read_reg_cache(codec, WM8731_LINVOL); + wm8731_write(codec, WM8731_LINVOL, reg & ~0x0100); + reg = wm8731_read_reg_cache(codec, WM8731_RINVOL); + wm8731_write(codec, WM8731_RINVOL, reg & ~0x0100); + + wm8731_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + return ret; + } + + ret = snd_soc_register_dai(&wm8731_dai); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + snd_soc_unregister_codec(codec); + return ret; + } -static int __devexit wm8731_spi_remove(struct spi_device *spi) -{ return 0; } -static struct spi_driver wm8731_spi_driver = { - .driver = { - .name = "wm8731", - .bus = &spi_bus_type, - .owner = THIS_MODULE, - }, - .probe = wm8731_spi_probe, - .remove = __devexit_p(wm8731_spi_remove), -}; +static void wm8731_unregister(struct wm8731_priv *wm8731) +{ + wm8731_set_bias_level(&wm8731->codec, SND_SOC_BIAS_OFF); + snd_soc_unregister_dai(&wm8731_dai); + snd_soc_unregister_codec(&wm8731->codec); + kfree(wm8731); + wm8731_codec = NULL; +} +#if defined(CONFIG_SPI_MASTER) static int wm8731_spi_write(struct spi_device *spi, const char *data, int len) { struct spi_transfer t; @@ -730,37 +640,67 @@ static int wm8731_spi_write(struct spi_device *spi, const char *data, int len) return len; } + +static int __devinit wm8731_spi_probe(struct spi_device *spi) +{ + struct snd_soc_codec *codec; + struct wm8731_priv *wm8731; + + wm8731 = kzalloc(sizeof(struct wm8731_priv), GFP_KERNEL); + if (wm8731 == NULL) + return -ENOMEM; + + codec = &wm8731->codec; + codec->control_data = spi; + codec->hw_write = (hw_write_t)wm8731_spi_write; + codec->dev = &spi->dev; + + return wm8731_register(wm8731); +} + +static int __devexit wm8731_spi_remove(struct spi_device *spi) +{ + /* FIXME: This isn't actually implemented... */ + return 0; +} + +static struct spi_driver wm8731_spi_driver = { + .driver = { + .name = "wm8731", + .bus = &spi_bus_type, + .owner = THIS_MODULE, + }, + .probe = wm8731_spi_probe, + .remove = __devexit_p(wm8731_spi_remove), +}; #endif /* CONFIG_SPI_MASTER */ #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) -/* - * WM8731 2 wire address is determined by GPIO5 - * state during powerup. - * low = 0x1a - * high = 0x1b - */ - static int wm8731_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { - struct snd_soc_device *socdev = wm8731_socdev; - struct snd_soc_codec *codec = socdev->card->codec; - int ret; + struct wm8731_priv *wm8731; + struct snd_soc_codec *codec; - i2c_set_clientdata(i2c, codec); + wm8731 = kzalloc(sizeof(struct wm8731_priv), GFP_KERNEL); + if (wm8731 == NULL) + return -ENOMEM; + + codec = &wm8731->codec; + codec->hw_write = (hw_write_t)i2c_master_send; + + i2c_set_clientdata(i2c, wm8731); codec->control_data = i2c; - ret = wm8731_init(socdev); - if (ret < 0) - pr_err("failed to initialise WM8731\n"); + codec->dev = &i2c->dev; - return ret; + return wm8731_register(wm8731); } static int wm8731_i2c_remove(struct i2c_client *client) { - struct snd_soc_codec *codec = i2c_get_clientdata(client); - kfree(codec->reg_cache); + struct wm8731_priv *wm8731 = i2c_get_clientdata(client); + wm8731_unregister(wm8731); return 0; } @@ -783,13 +723,33 @@ static struct i2c_driver wm8731_i2c_driver = { static int __init wm8731_modinit(void) { - return snd_soc_register_dai(&wm8731_dai); + int ret; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + ret = i2c_add_driver(&wm8731_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register WM8731 I2C driver: %d\n", + ret); + } +#endif +#if defined(CONFIG_SPI_MASTER) + ret = spi_register_driver(&wm8731_spi_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register WM8731 SPI driver: %d\n", + ret); + } +#endif + return 0; } module_init(wm8731_modinit); static void __exit wm8731_exit(void) { - snd_soc_unregister_dai(&wm8731_dai); +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&wm8731_i2c_driver); +#endif +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&wm8731_spi_driver); +#endif } module_exit(wm8731_exit); diff --git a/sound/soc/codecs/wm8731.h b/sound/soc/codecs/wm8731.h index 95190e9c0c1..cd7b806e8ad 100644 --- a/sound/soc/codecs/wm8731.h +++ b/sound/soc/codecs/wm8731.h @@ -34,12 +34,6 @@ #define WM8731_SYSCLK 0 #define WM8731_DAI 0 -struct wm8731_setup_data { - int spi; - int i2c_bus; - unsigned short i2c_address; -}; - extern struct snd_soc_dai wm8731_dai; extern struct snd_soc_codec_device soc_codec_dev_wm8731; diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 0d41be33d57..eaa66915a32 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -317,19 +317,44 @@ static struct snd_soc_card snd_soc_corgi = { .num_links = 1, }; -/* corgi audio private data */ -static struct wm8731_setup_data corgi_wm8731_setup = { - .i2c_bus = 0, - .i2c_address = 0x1b, -}; - /* corgi audio subsystem */ static struct snd_soc_device corgi_snd_devdata = { .card = &snd_soc_corgi, .codec_dev = &soc_codec_dev_wm8731, - .codec_data = &corgi_wm8731_setup, }; +/* + * FIXME: This is a temporary bodge to avoid cross-tree merge issues. + * New drivers should register the wm8731 I2C device in the machine + * setup code (under arch/arm for ARM systems). + */ +static int wm8731_i2c_register(void) +{ + struct i2c_board_info info; + struct i2c_adapter *adapter; + struct i2c_client *client; + + memset(&info, 0, sizeof(struct i2c_board_info)); + info.addr = 0x1b; + strlcpy(info.type, "wm8731", I2C_NAME_SIZE); + + adapter = i2c_get_adapter(0); + if (!adapter) { + printk(KERN_ERR "can't get i2c adapter 0\n"); + return -ENODEV; + } + + client = i2c_new_device(adapter, &info); + i2c_put_adapter(adapter); + if (!client) { + printk(KERN_ERR "can't add i2c device at 0x%x\n", + (unsigned int)info.addr); + return -ENODEV; + } + + return 0; +} + static struct platform_device *corgi_snd_device; static int __init corgi_init(void) @@ -340,6 +365,10 @@ static int __init corgi_init(void) machine_is_husky())) return -ENODEV; + ret = wm8731_i2c_setup(); + if (ret != 0) + return ret; + corgi_snd_device = platform_device_alloc("soc-audio", -1); if (!corgi_snd_device) return -ENOMEM; diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index 3a62d4354ef..fd683a0b742 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -283,17 +283,42 @@ static struct snd_soc_card snd_soc_poodle = { .num_links = 1, }; -/* poodle audio private data */ -static struct wm8731_setup_data poodle_wm8731_setup = { - .i2c_bus = 0, - .i2c_address = 0x1b, -}; +/* + * FIXME: This is a temporary bodge to avoid cross-tree merge issues. + * New drivers should register the wm8731 I2C device in the machine + * setup code (under arch/arm for ARM systems). + */ +static int wm8731_i2c_register(void) +{ + struct i2c_board_info info; + struct i2c_adapter *adapter; + struct i2c_client *client; + + memset(&info, 0, sizeof(struct i2c_board_info)); + info.addr = 0x1b; + strlcpy(info.type, "wm8731", I2C_NAME_SIZE); + + adapter = i2c_get_adapter(0); + if (!adapter) { + printk(KERN_ERR "can't get i2c adapter 0\n"); + return -ENODEV; + } + + client = i2c_new_device(adapter, &info); + i2c_put_adapter(adapter); + if (!client) { + printk(KERN_ERR "can't add i2c device at 0x%x\n", + (unsigned int)info.addr); + return -ENODEV; + } + + return 0; +} /* poodle audio subsystem */ static struct snd_soc_device poodle_snd_devdata = { .card = &snd_soc_poodle, .codec_dev = &soc_codec_dev_wm8731, - .codec_data = &poodle_wm8731_setup, }; static struct platform_device *poodle_snd_device; @@ -305,6 +330,10 @@ static int __init poodle_init(void) if (!machine_is_poodle()) return -ENODEV; + ret = wm8731_i2c_setup(); + if (ret != 0) + return ret; + locomo_gpio_set_dir(&poodle_locomo_device.dev, POODLE_LOCOMO_GPIO_AMP_ON, 0); /* should we mute HP at startup - burning power ?*/ -- cgit v1.2.3-70-g09d2 From 59544d33ff3118f22a484d8be06cdf5cfc2fdca5 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 18 Feb 2009 11:36:44 +0000 Subject: ASoC: Remove version display from the WM8753 driver Signed-off-by: Mark Brown --- sound/soc/codecs/wm8753.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index dc6042c6424..31ff337f822 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -51,8 +51,6 @@ #include "wm8753.h" -#define WM8753_VERSION "0.16" - static int caps_charge = 2000; module_param(caps_charge, int, 0); MODULE_PARM_DESC(caps_charge, "WM8753 cap charge time (msecs)"); @@ -1778,8 +1776,6 @@ static int wm8753_probe(struct platform_device *pdev) struct wm8753_priv *wm8753; int ret = 0; - pr_info("WM8753 Audio Codec %s", WM8753_VERSION); - setup = socdev->codec_data; codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); if (codec == NULL) -- cgit v1.2.3-70-g09d2 From b3bdb30b6d1989129e297641fec791e9e555e4d8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 18 Feb 2009 13:16:26 +0100 Subject: ALSA: hda - Add quirk for Acer X3200 Acer X3200 needs model=auto, otherwise model=acer is pre-selected. Reference: Novell bnc#476268 https://bugzilla.novell.com/show_bug.cgi?id=476268 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 244de597c5b..192c92a5af3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8545,6 +8545,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { ALC888_ACER_ASPIRE_4930G), SND_PCI_QUIRK(0x1025, 0x013f, "Acer Aspire 5930G", ALC888_ACER_ASPIRE_4930G), + SND_PCI_QUIRK(0x1025, 0x0157, "Acer X3200", ALC883_AUTO), SND_PCI_QUIRK(0x1025, 0x0158, "Acer AX1700-U3700A", ALC883_AUTO), SND_PCI_QUIRK(0x1025, 0x015e, "Acer Aspire 6930G", ALC888_ACER_ASPIRE_4930G), -- cgit v1.2.3-70-g09d2 From fc9967576829a01c98e5388410dc12c61006f79f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 18 Feb 2009 12:34:53 +0000 Subject: ASoC: Fix build for corgi and poodle Signed-off-by: Mark Brown --- sound/soc/pxa/corgi.c | 3 ++- sound/soc/pxa/poodle.c | 3 ++- 2 files changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index eaa66915a32..146973ae097 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -16,6 +16,7 @@ #include #include #include +#include #include #include #include @@ -365,7 +366,7 @@ static int __init corgi_init(void) machine_is_husky())) return -ENODEV; - ret = wm8731_i2c_setup(); + ret = wm8731_i2c_register(); if (ret != 0) return ret; diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index fd683a0b742..fb17a0a5a09 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -17,6 +17,7 @@ #include #include #include +#include #include #include #include @@ -330,7 +331,7 @@ static int __init poodle_init(void) if (!machine_is_poodle()) return -ENODEV; - ret = wm8731_i2c_setup(); + ret = wm8731_i2c_register(); if (ret != 0) return ret; -- cgit v1.2.3-70-g09d2 From 93b760b7072ca6972c15c798e97af3f830d8bbba Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 18 Feb 2009 12:44:40 +0000 Subject: ASoC: Implement SPI device unregistration for WM8731 Completely untested. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8731.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index a2c478e53d5..4191bdb803b 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -655,12 +655,17 @@ static int __devinit wm8731_spi_probe(struct spi_device *spi) codec->hw_write = (hw_write_t)wm8731_spi_write; codec->dev = &spi->dev; + spi->dev.driver_data = wm8731; + return wm8731_register(wm8731); } static int __devexit wm8731_spi_remove(struct spi_device *spi) { - /* FIXME: This isn't actually implemented... */ + struct wm8731_priv *wm8731 = spi->dev.driver_data; + + wm8731_unregister(wm8731); + return 0; } -- cgit v1.2.3-70-g09d2 From 6bab83fd886564e96abcff62862732159535f600 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 18 Feb 2009 14:39:05 +0200 Subject: ASoC: TWL4030: Add digital loopback support This patch adds the digital loopback/bypass support for twl4030 codec. The digital loopback will let the digimic0 (routed in the TX1 capture path inside of TWL4030) data to be routed back to the RX2 playback path (I2S stereo). It can also route the analog capture date routed through the TX1 back to RX2. Effectively the digital loopback is routing the audio from the TX1 capture path to the RX2 playback path. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 56 +++++++++++++++++++++++++++++++++++++++++----- 1 file changed, 50 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index c26854b398d..535d8ce2c32 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -504,6 +504,25 @@ static const struct snd_kcontrol_new twl4030_dapm_abypassr2_control = static const struct snd_kcontrol_new twl4030_dapm_abypassl2_control = SOC_DAPM_SINGLE("Switch", TWL4030_REG_ARXL2_APGA_CTL, 2, 1, 0); +/* Digital bypass gain, 0 mutes the bypass */ +static const unsigned int twl4030_dapm_dbypass_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 3, TLV_DB_SCALE_ITEM(-2400, 0, 1), + 4, 7, TLV_DB_SCALE_ITEM(-1800, 600, 0), +}; + +/* Digital bypass left (TX1L -> RX2L) */ +static const struct snd_kcontrol_new twl4030_dapm_dbypassl_control = + SOC_DAPM_SINGLE_TLV("Volume", + TWL4030_REG_ATX2ARXPGA, 3, 7, 0, + twl4030_dapm_dbypass_tlv); + +/* Digital bypass right (TX1R -> RX2R) */ +static const struct snd_kcontrol_new twl4030_dapm_dbypassr_control = + SOC_DAPM_SINGLE_TLV("Volume", + TWL4030_REG_ATX2ARXPGA, 0, 7, 0, + twl4030_dapm_dbypass_tlv); + static int micpath_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -608,12 +627,22 @@ static int bypass_event(struct snd_soc_dapm_widget *w, unsigned char reg; reg = twl4030_read_reg_cache(w->codec, m->reg); - if (reg & (1 << m->shift)) - twl4030->bypass_state |= - (1 << (m->reg - TWL4030_REG_ARXL1_APGA_CTL)); - else - twl4030->bypass_state &= - ~(1 << (m->reg - TWL4030_REG_ARXL1_APGA_CTL)); + + if (m->reg <= TWL4030_REG_ARXR2_APGA_CTL) { + /* Analog bypass */ + if (reg & (1 << m->shift)) + twl4030->bypass_state |= + (1 << (m->reg - TWL4030_REG_ARXL1_APGA_CTL)); + else + twl4030->bypass_state &= + ~(1 << (m->reg - TWL4030_REG_ARXL1_APGA_CTL)); + } else { + /* Digital bypass */ + if (reg & (0x7 << m->shift)) + twl4030->bypass_state |= (1 << (m->shift ? 5 : 4)); + else + twl4030->bypass_state &= ~(1 << (m->shift ? 5 : 4)); + } if (w->codec->bias_level == SND_SOC_BIAS_STANDBY) { if (twl4030->bypass_state) @@ -934,6 +963,14 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { &twl4030_dapm_abypassl2_control, bypass_event, SND_SOC_DAPM_POST_REG), + /* Digital bypasses */ + SND_SOC_DAPM_SWITCH_E("Left Digital Loopback", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_dbypassl_control, bypass_event, + SND_SOC_DAPM_POST_REG), + SND_SOC_DAPM_SWITCH_E("Right Digital Loopback", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_dbypassr_control, bypass_event, + SND_SOC_DAPM_POST_REG), + SND_SOC_DAPM_MIXER("Analog R1 Playback Mixer", TWL4030_REG_AVDAC_CTL, 0, 0, NULL, 0), SND_SOC_DAPM_MIXER("Analog L1 Playback Mixer", TWL4030_REG_AVDAC_CTL, @@ -1118,6 +1155,13 @@ static const struct snd_soc_dapm_route intercon[] = { {"Analog R2 Playback Mixer", NULL, "Right2 Analog Loopback"}, {"Analog L2 Playback Mixer", NULL, "Left2 Analog Loopback"}, + /* Digital bypass routes */ + {"Right Digital Loopback", "Volume", "TX1 Capture Route"}, + {"Left Digital Loopback", "Volume", "TX1 Capture Route"}, + + {"Analog R2 Playback Mixer", NULL, "Right Digital Loopback"}, + {"Analog L2 Playback Mixer", NULL, "Left Digital Loopback"}, + }; static int twl4030_add_widgets(struct snd_soc_codec *codec) -- cgit v1.2.3-70-g09d2 From 519cf2df5fb50c6d24412b2421ce2d1ff0346163 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 18 Feb 2009 21:06:01 +0000 Subject: ASoC: Check for errors when writing WM8731 reset register Signed-off-by: Mark Brown --- sound/soc/codecs/wm8731.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 4191bdb803b..9c9fc3b5a6c 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -574,9 +574,14 @@ static int wm8731_register(struct wm8731_priv *wm8731) memcpy(codec->reg_cache, wm8731_reg, sizeof(wm8731_reg)); + ret = wm8731_reset(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to issue reset\n"); + return ret; + } + wm8731_dai.dev = codec->dev; - wm8731_reset(codec); wm8731_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Latch the update bits */ -- cgit v1.2.3-70-g09d2 From c6f2981170272cce2c192087a16dd74dbde25ed2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 18 Feb 2009 21:25:40 +0000 Subject: ASoC: Add device init/exit annotations to new-style Wolfson CODEC drivers Signed-off-by: Mark Brown --- sound/soc/codecs/wm8350.c | 2 +- sound/soc/codecs/wm8731.c | 8 ++++---- sound/soc/codecs/wm8900.c | 8 ++++---- sound/soc/codecs/wm8903.c | 8 ++++---- 4 files changed, 13 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index d3562788d42..359e5cc86f3 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1574,7 +1574,7 @@ struct snd_soc_codec_device soc_codec_dev_wm8350 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_wm8350); -static int wm8350_codec_probe(struct platform_device *pdev) +static __devinit int wm8350_codec_probe(struct platform_device *pdev) { struct wm8350 *wm8350 = platform_get_drvdata(pdev); struct wm8350_data *priv; diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 9c9fc3b5a6c..4cac3195bfa 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -686,8 +686,8 @@ static struct spi_driver wm8731_spi_driver = { #endif /* CONFIG_SPI_MASTER */ #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) -static int wm8731_i2c_probe(struct i2c_client *i2c, - const struct i2c_device_id *id) +static __devinit int wm8731_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) { struct wm8731_priv *wm8731; struct snd_soc_codec *codec; @@ -707,7 +707,7 @@ static int wm8731_i2c_probe(struct i2c_client *i2c, return wm8731_register(wm8731); } -static int wm8731_i2c_remove(struct i2c_client *client) +static __devexit int wm8731_i2c_remove(struct i2c_client *client) { struct wm8731_priv *wm8731 = i2c_get_clientdata(client); wm8731_unregister(wm8731); @@ -726,7 +726,7 @@ static struct i2c_driver wm8731_i2c_driver = { .owner = THIS_MODULE, }, .probe = wm8731_i2c_probe, - .remove = wm8731_i2c_remove, + .remove = __devexit_p(wm8731_i2c_remove), .id_table = wm8731_i2c_id, }; #endif diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 85c0f1bc676..da5ca64f89b 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -1272,8 +1272,8 @@ static int wm8900_resume(struct platform_device *pdev) static struct snd_soc_codec *wm8900_codec; -static int wm8900_i2c_probe(struct i2c_client *i2c, - const struct i2c_device_id *id) +static __devinit int wm8900_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) { struct wm8900_priv *wm8900; struct snd_soc_codec *codec; @@ -1372,7 +1372,7 @@ err: return ret; } -static int wm8900_i2c_remove(struct i2c_client *client) +static __devexit int wm8900_i2c_remove(struct i2c_client *client) { snd_soc_unregister_dai(&wm8900_dai); snd_soc_unregister_codec(wm8900_codec); @@ -1398,7 +1398,7 @@ static struct i2c_driver wm8900_i2c_driver = { .owner = THIS_MODULE, }, .probe = wm8900_i2c_probe, - .remove = wm8900_i2c_remove, + .remove = __devexit_p(wm8900_i2c_remove), .id_table = wm8900_i2c_id, }; diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index d36b2b1edf1..c6fa8a71b4d 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1562,8 +1562,8 @@ static int wm8903_resume(struct platform_device *pdev) static struct snd_soc_codec *wm8903_codec; -static int wm8903_i2c_probe(struct i2c_client *i2c, - const struct i2c_device_id *id) +static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) { struct wm8903_priv *wm8903; struct snd_soc_codec *codec; @@ -1669,7 +1669,7 @@ err: return ret; } -static int wm8903_i2c_remove(struct i2c_client *client) +static __devexit int wm8903_i2c_remove(struct i2c_client *client) { struct snd_soc_codec *codec = i2c_get_clientdata(client); @@ -1699,7 +1699,7 @@ static struct i2c_driver wm8903_i2c_driver = { .owner = THIS_MODULE, }, .probe = wm8903_i2c_probe, - .remove = wm8903_i2c_remove, + .remove = __devexit_p(wm8903_i2c_remove), .id_table = wm8903_i2c_id, }; -- cgit v1.2.3-70-g09d2 From 07eba61dd68678e30b24b4776f59798f625e089d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 19 Feb 2009 08:06:35 +0100 Subject: ALSA: hda - Don't enable beep for digital-only ALC262 When ALC262 codec is configured as digital-only, it's meaningless to add the digital beep input. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 13 ++++++++----- 1 file changed, 8 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 192c92a5af3..91da92259c8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -11051,10 +11051,12 @@ static int patch_alc262(struct hda_codec *codec) } } - err = snd_hda_attach_beep_device(codec, 0x1); - if (err < 0) { - alc_free(codec); - return err; + if (!spec->no_analog) { + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } } if (board_config != ALC262_AUTO) @@ -11087,7 +11089,8 @@ static int patch_alc262(struct hda_codec *codec) } if (!spec->cap_mixer && !spec->no_analog) set_capture_mixer(spec); - set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); + if (!spec->no_analog) + set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); spec->vmaster_nid = 0x0c; -- cgit v1.2.3-70-g09d2 From ab9fec099b796b002b6996c4c5845167d8fe6dbd Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 19 Feb 2009 08:13:26 +0100 Subject: ALSA: hda - Avoid doubly beep attachment in patch_alc268() Remove the doubly attachment in patch_alc268(). The input beep is attached conditionally only when needed. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 91da92259c8..df32f9353e7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12100,12 +12100,6 @@ static int patch_alc268(struct hda_codec *codec) } } - err = snd_hda_attach_beep_device(codec, 0x1); - if (err < 0) { - alc_free(codec); - return err; - } - if (board_config != ALC268_AUTO) setup_preset(spec, &alc268_presets[board_config]); -- cgit v1.2.3-70-g09d2 From 7e0e44d430281d398769f1d7864e161203252760 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 19 Feb 2009 08:15:49 +0100 Subject: ALSA: hda - Add digital-only mode for ALC268 ALC268 can be configured as digital-only, e.g. for HDMI, on some machines. Allow the parser to set up the digital-only mode. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 19 +++++++++++++------ 1 file changed, 13 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index df32f9353e7..169b3837af5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -11824,9 +11824,14 @@ static int alc268_parse_auto_config(struct hda_codec *codec) alc268_ignore); if (err < 0) return err; - if (!spec->autocfg.line_outs) + if (!spec->autocfg.line_outs) { + if (spec->autocfg.dig_outs || spec->autocfg.dig_in_pin) { + spec->multiout.max_channels = 2; + spec->no_analog = 1; + goto dig_only; + } return 0; /* can't find valid BIOS pin config */ - + } err = alc268_auto_create_multi_out_ctls(spec, &spec->autocfg); if (err < 0) return err; @@ -11836,10 +11841,12 @@ static int alc268_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = 2; + dig_only: /* digital only support output */ - if (spec->autocfg.dig_outs) + if (spec->autocfg.dig_outs) { spec->multiout.dig_out_nid = ALC268_DIGOUT_NID; - + spec->dig_out_type = spec->autocfg.dig_out_type[0]; + } if (spec->kctls.list) add_mixer(spec, spec->kctls.list); @@ -12140,7 +12147,7 @@ static int patch_alc268(struct hda_codec *codec) (0 << AC_AMPCAP_MUTE_SHIFT)); } - if (!spec->adc_nids && spec->input_mux) { + if (!spec->no_analog && !spec->adc_nids && spec->input_mux) { /* check whether NID 0x07 is valid */ unsigned int wcap = get_wcaps(codec, 0x07); int i; @@ -12764,7 +12771,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; - if (!spec->cap_mixer) + if (!spec->cap_mixer && !spec->no_analog) set_capture_mixer(spec); store_pin_configs(codec); -- cgit v1.2.3-70-g09d2 From bb71858853a5c9616eea98512f4075d4f081154d Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 19 Feb 2009 08:37:13 +0100 Subject: sound: oxygen: make the owner module a parameter of the probe function Move the owner field out of the oxygen_model structure and make it a parameter of oxygen_pci_probe(), because the actual owner module does not depend on the card model. Furthermore, moving it out of the model structure allows us to create the card structure before the actual model is known. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/hifier.c | 3 +-- sound/pci/oxygen/oxygen.c | 3 +-- sound/pci/oxygen/oxygen.h | 2 +- sound/pci/oxygen/oxygen_lib.c | 3 ++- sound/pci/oxygen/virtuoso.c | 5 +---- 5 files changed, 6 insertions(+), 10 deletions(-) diff --git a/sound/pci/oxygen/hifier.c b/sound/pci/oxygen/hifier.c index 1ab833f843e..cc98bad9916 100644 --- a/sound/pci/oxygen/hifier.c +++ b/sound/pci/oxygen/hifier.c @@ -151,7 +151,6 @@ static const struct oxygen_model model_hifier = { .shortname = "C-Media CMI8787", .longname = "C-Media Oxygen HD Audio", .chip = "CMI8788", - .owner = THIS_MODULE, .init = hifier_init, .control_filter = hifier_control_filter, .cleanup = hifier_cleanup, @@ -185,7 +184,7 @@ static int __devinit hifier_probe(struct pci_dev *pci, ++dev; return -ENOENT; } - err = oxygen_pci_probe(pci, index[dev], id[dev], &model_hifier, 0); + err = oxygen_pci_probe(pci, index[dev], id[dev], THIS_MODULE, &model_hifier, 0); if (err >= 0) ++dev; return err; diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index de999c6d6dd..12b6c2137d5 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -315,7 +315,6 @@ static const struct oxygen_model model_generic = { .shortname = "C-Media CMI8788", .longname = "C-Media Oxygen HD Audio", .chip = "CMI8788", - .owner = THIS_MODULE, .probe = generic_probe, .init = generic_init, .cleanup = generic_cleanup, @@ -353,7 +352,7 @@ static int __devinit generic_oxygen_probe(struct pci_dev *pci, ++dev; return -ENOENT; } - err = oxygen_pci_probe(pci, index[dev], id[dev], + err = oxygen_pci_probe(pci, index[dev], id[dev], THIS_MODULE, &model_generic, pci_id->driver_data); if (err >= 0) ++dev; diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index 19107c6307e..268bff4f29d 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -62,7 +62,6 @@ struct oxygen_model { const char *shortname; const char *longname; const char *chip; - struct module *owner; int (*probe)(struct oxygen *chip, unsigned long driver_data); void (*init)(struct oxygen *chip); int (*control_filter)(struct snd_kcontrol_new *template); @@ -134,6 +133,7 @@ struct oxygen { /* oxygen_lib.c */ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id, + struct module *owner, const struct oxygen_model *model, unsigned long driver_data); void oxygen_pci_remove(struct pci_dev *pci); diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index 9c81e0b0511..b5560fa5a5e 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -452,6 +452,7 @@ static void oxygen_card_free(struct snd_card *card) } int oxygen_pci_probe(struct pci_dev *pci, int index, char *id, + struct module *owner, const struct oxygen_model *model, unsigned long driver_data) { @@ -459,7 +460,7 @@ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id, struct oxygen *chip; int err; - err = snd_card_create(index, id, model->owner, + err = snd_card_create(index, id, owner, sizeof(*chip) + model->model_data_size, &card); if (err < 0) return err; diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 6c870c12a17..c05f7e7bdb3 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -816,7 +816,6 @@ static int xonar_model_probe(struct oxygen *chip, unsigned long driver_data) static const struct oxygen_model model_xonar_d2 = { .longname = "Asus Virtuoso 200", .chip = "AV200", - .owner = THIS_MODULE, .probe = xonar_model_probe, .init = xonar_d2_init, .control_filter = xonar_d2_control_filter, @@ -849,7 +848,6 @@ static const struct oxygen_model model_xonar_d2 = { static const struct oxygen_model model_xonar_d1 = { .longname = "Asus Virtuoso 100", .chip = "AV200", - .owner = THIS_MODULE, .probe = xonar_model_probe, .init = xonar_d1_init, .control_filter = xonar_d1_control_filter, @@ -878,7 +876,6 @@ static const struct oxygen_model model_xonar_d1 = { static const struct oxygen_model model_xonar_hdav = { .longname = "Asus Virtuoso 200", .chip = "AV200", - .owner = THIS_MODULE, .probe = xonar_model_probe, .init = xonar_hdav_init, .cleanup = xonar_hdav_cleanup, @@ -925,7 +922,7 @@ static int __devinit xonar_probe(struct pci_dev *pci, return -ENOENT; } BUG_ON(pci_id->driver_data >= ARRAY_SIZE(models)); - err = oxygen_pci_probe(pci, index[dev], id[dev], + err = oxygen_pci_probe(pci, index[dev], id[dev], THIS_MODULE, models[pci_id->driver_data], pci_id->driver_data); if (err >= 0) -- cgit v1.2.3-70-g09d2 From 6ed91157093c60e26bf0215b752f07af52935afc Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 19 Feb 2009 08:38:25 +0100 Subject: sound: oxygen: allocate model_data dynamically Allocate the model-specific data dynamically instead of including it in the memory block of the card structure. This will allow us to determine the actual model after the card creation. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/oxygen_lib.c | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index b5560fa5a5e..228f30800fd 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -446,6 +446,7 @@ static void oxygen_card_free(struct snd_card *card) free_irq(chip->irq, chip); flush_scheduled_work(); chip->model.cleanup(chip); + kfree(chip->model_data); mutex_destroy(&chip->mutex); pci_release_regions(chip->pci); pci_disable_device(chip->pci); @@ -460,8 +461,7 @@ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id, struct oxygen *chip; int err; - err = snd_card_create(index, id, owner, - sizeof(*chip) + model->model_data_size, &card); + err = snd_card_create(index, id, owner, sizeof(*chip), &card); if (err < 0) return err; @@ -470,7 +470,6 @@ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id, chip->pci = pci; chip->irq = -1; chip->model = *model; - chip->model_data = chip + 1; spin_lock_init(&chip->reg_lock); mutex_init(&chip->mutex); INIT_WORK(&chip->spdif_input_bits_work, @@ -496,6 +495,15 @@ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id, } chip->addr = pci_resource_start(pci, 0); + if (chip->model.model_data_size) { + chip->model_data = kmalloc(chip->model.model_data_size, + GFP_KERNEL); + if (!chip->model_data) { + err = -ENOMEM; + goto err_pci_regions; + } + } + pci_set_master(pci); snd_card_set_dev(card, &pci->dev); card->private_free = oxygen_card_free; -- cgit v1.2.3-70-g09d2 From a69bb3c3fe0881d986ec78e253cb8a6bb9c28230 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 19 Feb 2009 08:38:55 +0100 Subject: sound: oxygen: use static driver name When allocating resources, use a fixed name instead of reading it from the model structure. This allows us to allocate the resources before the actual model is known. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/oxygen_lib.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index 228f30800fd..516d94ad2bb 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -34,6 +34,7 @@ MODULE_AUTHOR("Clemens Ladisch "); MODULE_DESCRIPTION("C-Media CMI8788 helper library"); MODULE_LICENSE("GPL v2"); +#define DRIVER "oxygen" static inline int oxygen_uart_input_ready(struct oxygen *chip) { @@ -481,7 +482,7 @@ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id, if (err < 0) goto err_card; - err = pci_request_regions(pci, model->chip); + err = pci_request_regions(pci, DRIVER); if (err < 0) { snd_printk(KERN_ERR "cannot reserve PCI resources\n"); goto err_pci_enable; @@ -517,7 +518,7 @@ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id, chip->model.init(chip); err = request_irq(pci->irq, oxygen_interrupt, IRQF_SHARED, - chip->model.chip, chip); + DRIVER, chip); if (err < 0) { snd_printk(KERN_ERR "cannot grab interrupt %d\n", pci->irq); goto err_card; -- cgit v1.2.3-70-g09d2 From 30459d7b1843cbdea56ca120c8cac10dc5613e90 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 19 Feb 2009 08:42:44 +0100 Subject: sound: oxygen: handle cards with broken EEPROM Under as yet unknown circumstances, the first word of the sound card's EEPROM gets overwritten. When this has happened, we cannot rely on the subsystem IDs that the kernel reads from the PCI configuration registers. Instead, we read the IDs directly from the EEPROM and do the ID matching manually. Because the model-specific driver cannot determine the model before calling oxygen_pci_probe(), that function now gets a get_model() callback as parameter. The customizing of the model structure, which was formerly done by the probe() callback, also has moved into get_model(). Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/hifier.c | 11 +++- sound/pci/oxygen/oxygen.c | 44 +++++++------- sound/pci/oxygen/oxygen.h | 17 +++++- sound/pci/oxygen/oxygen_io.c | 15 +++++ sound/pci/oxygen/oxygen_lib.c | 51 +++++++++++++--- sound/pci/oxygen/virtuoso.c | 134 +++++++++++++++++++++++------------------- 6 files changed, 179 insertions(+), 93 deletions(-) diff --git a/sound/pci/oxygen/hifier.c b/sound/pci/oxygen/hifier.c index cc98bad9916..84ef1318341 100644 --- a/sound/pci/oxygen/hifier.c +++ b/sound/pci/oxygen/hifier.c @@ -45,6 +45,7 @@ MODULE_PARM_DESC(enable, "enable card"); static struct pci_device_id hifier_ids[] __devinitdata = { { OXYGEN_PCI_SUBID(0x14c3, 0x1710) }, { OXYGEN_PCI_SUBID(0x14c3, 0x1711) }, + { OXYGEN_PCI_SUBID_BROKEN_EEPROM }, { } }; MODULE_DEVICE_TABLE(pci, hifier_ids); @@ -172,6 +173,13 @@ static const struct oxygen_model model_hifier = { .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, }; +static int __devinit get_hifier_model(struct oxygen *chip, + const struct pci_device_id *id) +{ + chip->model = model_hifier; + return 0; +} + static int __devinit hifier_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { @@ -184,7 +192,8 @@ static int __devinit hifier_probe(struct pci_dev *pci, ++dev; return -ENOENT; } - err = oxygen_pci_probe(pci, index[dev], id[dev], THIS_MODULE, &model_hifier, 0); + err = oxygen_pci_probe(pci, index[dev], id[dev], THIS_MODULE, + hifier_ids, get_hifier_model); if (err >= 0) ++dev; return err; diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index 12b6c2137d5..f2c37f379d3 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -293,29 +293,10 @@ static void set_ak5385_params(struct oxygen *chip, static const DECLARE_TLV_DB_LINEAR(ak4396_db_scale, TLV_DB_GAIN_MUTE, 0); -static int generic_probe(struct oxygen *chip, unsigned long driver_data) -{ - if (driver_data == MODEL_MERIDIAN) { - chip->model.init = meridian_init; - chip->model.resume = meridian_resume; - chip->model.set_adc_params = set_ak5385_params; - chip->model.device_config = PLAYBACK_0_TO_I2S | - PLAYBACK_1_TO_SPDIF | - CAPTURE_0_FROM_I2S_2 | - CAPTURE_1_FROM_SPDIF; - } - if (driver_data == MODEL_MERIDIAN || driver_data == MODEL_HALO) { - chip->model.misc_flags = OXYGEN_MISC_MIDI; - chip->model.device_config |= MIDI_OUTPUT | MIDI_INPUT; - } - return 0; -} - static const struct oxygen_model model_generic = { .shortname = "C-Media CMI8788", .longname = "C-Media Oxygen HD Audio", .chip = "CMI8788", - .probe = generic_probe, .init = generic_init, .cleanup = generic_cleanup, .resume = generic_resume, @@ -340,6 +321,29 @@ static const struct oxygen_model model_generic = { .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, }; +static int __devinit get_oxygen_model(struct oxygen *chip, + const struct pci_device_id *id) +{ + chip->model = model_generic; + switch (id->driver_data) { + case MODEL_MERIDIAN: + chip->model.init = meridian_init; + chip->model.resume = meridian_resume; + chip->model.set_adc_params = set_ak5385_params; + chip->model.device_config = PLAYBACK_0_TO_I2S | + PLAYBACK_1_TO_SPDIF | + CAPTURE_0_FROM_I2S_2 | + CAPTURE_1_FROM_SPDIF; + break; + } + if (id->driver_data == MODEL_MERIDIAN || + id->driver_data == MODEL_HALO) { + chip->model.misc_flags = OXYGEN_MISC_MIDI; + chip->model.device_config |= MIDI_OUTPUT | MIDI_INPUT; + } + return 0; +} + static int __devinit generic_oxygen_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { @@ -353,7 +357,7 @@ static int __devinit generic_oxygen_probe(struct pci_dev *pci, return -ENOENT; } err = oxygen_pci_probe(pci, index[dev], id[dev], THIS_MODULE, - &model_generic, pci_id->driver_data); + oxygen_ids, get_oxygen_model); if (err >= 0) ++dev; return err; diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index 268bff4f29d..c500d48ea34 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -49,7 +49,13 @@ enum { .subvendor = sv, \ .subdevice = sd +#define BROKEN_EEPROM_DRIVER_DATA ((unsigned long)-1) +#define OXYGEN_PCI_SUBID_BROKEN_EEPROM \ + OXYGEN_PCI_SUBID(PCI_VENDOR_ID_CMEDIA, 0x8788), \ + .driver_data = BROKEN_EEPROM_DRIVER_DATA + struct pci_dev; +struct pci_device_id; struct snd_card; struct snd_pcm_substream; struct snd_pcm_hardware; @@ -62,7 +68,6 @@ struct oxygen_model { const char *shortname; const char *longname; const char *chip; - int (*probe)(struct oxygen *chip, unsigned long driver_data); void (*init)(struct oxygen *chip); int (*control_filter)(struct snd_kcontrol_new *template); int (*mixer_init)(struct oxygen *chip); @@ -82,6 +87,7 @@ struct oxygen_model { void (*ac97_switch)(struct oxygen *chip, unsigned int reg, unsigned int mute); const unsigned int *dac_tlv; + unsigned long private_data; size_t model_data_size; unsigned int device_config; u8 dac_channels; @@ -134,8 +140,11 @@ struct oxygen { int oxygen_pci_probe(struct pci_dev *pci, int index, char *id, struct module *owner, - const struct oxygen_model *model, - unsigned long driver_data); + const struct pci_device_id *ids, + int (*get_model)(struct oxygen *chip, + const struct pci_device_id *id + ) + ); void oxygen_pci_remove(struct pci_dev *pci); #ifdef CONFIG_PM int oxygen_pci_suspend(struct pci_dev *pci, pm_message_t state); @@ -180,6 +189,8 @@ void oxygen_write_i2c(struct oxygen *chip, u8 device, u8 map, u8 data); void oxygen_reset_uart(struct oxygen *chip); void oxygen_write_uart(struct oxygen *chip, u8 data); +u16 oxygen_read_eeprom(struct oxygen *chip, unsigned int index); + static inline void oxygen_set_bits8(struct oxygen *chip, unsigned int reg, u8 value) { diff --git a/sound/pci/oxygen/oxygen_io.c b/sound/pci/oxygen/oxygen_io.c index 3126c4b403d..05f48ef1a44 100644 --- a/sound/pci/oxygen/oxygen_io.c +++ b/sound/pci/oxygen/oxygen_io.c @@ -254,3 +254,18 @@ void oxygen_write_uart(struct oxygen *chip, u8 data) _write_uart(chip, 0, data); } EXPORT_SYMBOL(oxygen_write_uart); + +u16 oxygen_read_eeprom(struct oxygen *chip, unsigned int index) +{ + unsigned int timeout; + + oxygen_write8(chip, OXYGEN_EEPROM_CONTROL, + index | OXYGEN_EEPROM_DIR_READ); + for (timeout = 0; timeout < 100; ++timeout) { + udelay(1); + if (!(oxygen_read8(chip, OXYGEN_EEPROM_STATUS) + & OXYGEN_EEPROM_BUSY)) + break; + } + return oxygen_read16(chip, OXYGEN_EEPROM_DATA); +} diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index 516d94ad2bb..d83c3a95732 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -244,6 +244,34 @@ static void oxygen_proc_init(struct oxygen *chip) #define oxygen_proc_init(chip) #endif +static const struct pci_device_id * +oxygen_search_pci_id(struct oxygen *chip, const struct pci_device_id ids[]) +{ + u16 subdevice; + + /* + * Make sure the EEPROM pins are available, i.e., not used for SPI. + * (This function is called before we initialize or use SPI.) + */ + oxygen_clear_bits8(chip, OXYGEN_FUNCTION, + OXYGEN_FUNCTION_ENABLE_SPI_4_5); + /* + * Read the subsystem device ID directly from the EEPROM, because the + * chip didn't if the first EEPROM word was overwritten. + */ + subdevice = oxygen_read_eeprom(chip, 2); + /* + * We use only the subsystem device ID for searching because it is + * unique even without the subsystem vendor ID, which may have been + * overwritten in the EEPROM. + */ + for (; ids->vendor; ++ids) + if (ids->subdevice == subdevice && + ids->driver_data != BROKEN_EEPROM_DRIVER_DATA) + return ids; + return NULL; +} + static void oxygen_init(struct oxygen *chip) { unsigned int i; @@ -455,11 +483,15 @@ static void oxygen_card_free(struct snd_card *card) int oxygen_pci_probe(struct pci_dev *pci, int index, char *id, struct module *owner, - const struct oxygen_model *model, - unsigned long driver_data) + const struct pci_device_id *ids, + int (*get_model)(struct oxygen *chip, + const struct pci_device_id *id + ) + ) { struct snd_card *card; struct oxygen *chip; + const struct pci_device_id *pci_id; int err; err = snd_card_create(index, id, owner, sizeof(*chip), &card); @@ -470,7 +502,6 @@ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id, chip->card = card; chip->pci = pci; chip->irq = -1; - chip->model = *model; spin_lock_init(&chip->reg_lock); mutex_init(&chip->mutex); INIT_WORK(&chip->spdif_input_bits_work, @@ -496,6 +527,15 @@ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id, } chip->addr = pci_resource_start(pci, 0); + pci_id = oxygen_search_pci_id(chip, ids); + if (!pci_id) { + err = -ENODEV; + goto err_pci_regions; + } + err = get_model(chip, pci_id); + if (err < 0) + goto err_pci_regions; + if (chip->model.model_data_size) { chip->model_data = kmalloc(chip->model.model_data_size, GFP_KERNEL); @@ -509,11 +549,6 @@ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id, snd_card_set_dev(card, &pci->dev); card->private_free = oxygen_card_free; - if (chip->model.probe) { - err = chip->model.probe(chip, driver_data); - if (err < 0) - goto err_card; - } oxygen_init(chip); chip->model.init(chip); diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index c05f7e7bdb3..4ac49772da8 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -160,6 +160,7 @@ static struct pci_device_id xonar_ids[] __devinitdata = { { OXYGEN_PCI_SUBID(0x1043, 0x82b7), .driver_data = MODEL_D2X }, { OXYGEN_PCI_SUBID(0x1043, 0x8314), .driver_data = MODEL_HDAV }, { OXYGEN_PCI_SUBID(0x1043, 0x834f), .driver_data = MODEL_D1 }, + { OXYGEN_PCI_SUBID_BROKEN_EEPROM }, { } }; MODULE_DEVICE_TABLE(pci, xonar_ids); @@ -188,7 +189,6 @@ MODULE_DEVICE_TABLE(pci, xonar_ids); #define I2C_DEVICE_CS4362A 0x30 /* 001100, AD0=0, /W=0 */ struct xonar_data { - unsigned int model; unsigned int anti_pop_delay; unsigned int dacs; u16 output_enable_bit; @@ -334,15 +334,9 @@ static void xonar_d2_init(struct oxygen *chip) struct xonar_data *data = chip->model_data; data->anti_pop_delay = 300; + data->dacs = 4; data->output_enable_bit = GPIO_D2_OUTPUT_ENABLE; data->pcm1796_oversampling = PCM1796_OS_64; - if (data->model == MODEL_D2X) { - data->ext_power_reg = OXYGEN_GPIO_DATA; - data->ext_power_int_reg = OXYGEN_GPIO_INTERRUPT_MASK; - data->ext_power_bit = GPIO_D2X_EXT_POWER; - oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL, - GPIO_D2X_EXT_POWER); - } pcm1796_init(chip); @@ -355,6 +349,18 @@ static void xonar_d2_init(struct oxygen *chip) snd_component_add(chip->card, "CS5381"); } +static void xonar_d2x_init(struct oxygen *chip) +{ + struct xonar_data *data = chip->model_data; + + data->ext_power_reg = OXYGEN_GPIO_DATA; + data->ext_power_int_reg = OXYGEN_GPIO_INTERRUPT_MASK; + data->ext_power_bit = GPIO_D2X_EXT_POWER; + oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_D2X_EXT_POWER); + + xonar_d2_init(chip); +} + static void update_cs4362a_volumes(struct oxygen *chip) { u8 mute; @@ -422,11 +428,6 @@ static void xonar_d1_init(struct oxygen *chip) data->cs4398_fm = CS4398_FM_SINGLE | CS4398_DEM_NONE | CS4398_DIF_LJUST; data->cs4362a_fm = CS4362A_FM_SINGLE | CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L; - if (data->model == MODEL_DX) { - data->ext_power_reg = OXYGEN_GPI_DATA; - data->ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK; - data->ext_power_bit = GPI_DX_EXT_POWER; - } oxygen_write16(chip, OXYGEN_2WIRE_BUS_STATUS, OXYGEN_2WIRE_LENGTH_8 | @@ -447,6 +448,17 @@ static void xonar_d1_init(struct oxygen *chip) snd_component_add(chip->card, "CS5361"); } +static void xonar_dx_init(struct oxygen *chip) +{ + struct xonar_data *data = chip->model_data; + + data->ext_power_reg = OXYGEN_GPI_DATA; + data->ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK; + data->ext_power_bit = GPI_DX_EXT_POWER; + + xonar_d1_init(chip); +} + static void xonar_hdav_init(struct oxygen *chip) { struct xonar_data *data = chip->model_data; @@ -458,6 +470,7 @@ static void xonar_hdav_init(struct oxygen *chip) OXYGEN_2WIRE_SPEED_FAST); data->anti_pop_delay = 100; + data->dacs = chip->model.private_data == MODEL_HDAV_H6 ? 4 : 1; data->output_enable_bit = GPIO_DX_OUTPUT_ENABLE; data->ext_power_reg = OXYGEN_GPI_DATA; data->ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK; @@ -773,50 +786,9 @@ static int xonar_d1_mixer_init(struct oxygen *chip) return snd_ctl_add(chip->card, snd_ctl_new1(&front_panel_switch, chip)); } -static int xonar_model_probe(struct oxygen *chip, unsigned long driver_data) -{ - static const char *const names[] = { - [MODEL_D1] = "Xonar D1", - [MODEL_DX] = "Xonar DX", - [MODEL_D2] = "Xonar D2", - [MODEL_D2X] = "Xonar D2X", - [MODEL_HDAV] = "Xonar HDAV1.3", - [MODEL_HDAV_H6] = "Xonar HDAV1.3+H6", - }; - static const u8 dacs[] = { - [MODEL_D1] = 2, - [MODEL_DX] = 2, - [MODEL_D2] = 4, - [MODEL_D2X] = 4, - [MODEL_HDAV] = 1, - [MODEL_HDAV_H6] = 4, - }; - struct xonar_data *data = chip->model_data; - - data->model = driver_data; - if (data->model == MODEL_HDAV) { - oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL, - GPIO_HDAV_DB_MASK); - switch (oxygen_read16(chip, OXYGEN_GPIO_DATA) & - GPIO_HDAV_DB_MASK) { - case GPIO_HDAV_DB_H6: - data->model = MODEL_HDAV_H6; - break; - case GPIO_HDAV_DB_XX: - snd_printk(KERN_ERR "unknown daughterboard\n"); - return -ENODEV; - } - } - - data->dacs = dacs[data->model]; - chip->model.shortname = names[data->model]; - return 0; -} - static const struct oxygen_model model_xonar_d2 = { .longname = "Asus Virtuoso 200", .chip = "AV200", - .probe = xonar_model_probe, .init = xonar_d2_init, .control_filter = xonar_d2_control_filter, .mixer_init = xonar_d2_mixer_init, @@ -848,7 +820,6 @@ static const struct oxygen_model model_xonar_d2 = { static const struct oxygen_model model_xonar_d1 = { .longname = "Asus Virtuoso 100", .chip = "AV200", - .probe = xonar_model_probe, .init = xonar_d1_init, .control_filter = xonar_d1_control_filter, .mixer_init = xonar_d1_mixer_init, @@ -876,7 +847,6 @@ static const struct oxygen_model model_xonar_d1 = { static const struct oxygen_model model_xonar_hdav = { .longname = "Asus Virtuoso 200", .chip = "AV200", - .probe = xonar_model_probe, .init = xonar_hdav_init, .cleanup = xonar_hdav_cleanup, .suspend = xonar_hdav_suspend, @@ -902,8 +872,8 @@ static const struct oxygen_model model_xonar_hdav = { .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, }; -static int __devinit xonar_probe(struct pci_dev *pci, - const struct pci_device_id *pci_id) +static int __devinit get_xonar_model(struct oxygen *chip, + const struct pci_device_id *id) { static const struct oxygen_model *const models[] = { [MODEL_D1] = &model_xonar_d1, @@ -912,6 +882,50 @@ static int __devinit xonar_probe(struct pci_dev *pci, [MODEL_D2X] = &model_xonar_d2, [MODEL_HDAV] = &model_xonar_hdav, }; + static const char *const names[] = { + [MODEL_D1] = "Xonar D1", + [MODEL_DX] = "Xonar DX", + [MODEL_D2] = "Xonar D2", + [MODEL_D2X] = "Xonar D2X", + [MODEL_HDAV] = "Xonar HDAV1.3", + [MODEL_HDAV_H6] = "Xonar HDAV1.3+H6", + }; + unsigned int model = id->driver_data; + + if (model >= ARRAY_SIZE(models) || !models[model]) + return -EINVAL; + chip->model = *models[model]; + + switch (model) { + case MODEL_D2X: + chip->model.init = xonar_d2x_init; + break; + case MODEL_DX: + chip->model.init = xonar_dx_init; + break; + case MODEL_HDAV: + oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL, + GPIO_HDAV_DB_MASK); + switch (oxygen_read16(chip, OXYGEN_GPIO_DATA) & + GPIO_HDAV_DB_MASK) { + case GPIO_HDAV_DB_H6: + model = MODEL_HDAV_H6; + break; + case GPIO_HDAV_DB_XX: + snd_printk(KERN_ERR "unknown daughterboard\n"); + return -ENODEV; + } + break; + } + + chip->model.shortname = names[model]; + chip->model.private_data = model; + return 0; +} + +static int __devinit xonar_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) +{ static int dev; int err; @@ -921,10 +935,8 @@ static int __devinit xonar_probe(struct pci_dev *pci, ++dev; return -ENOENT; } - BUG_ON(pci_id->driver_data >= ARRAY_SIZE(models)); err = oxygen_pci_probe(pci, index[dev], id[dev], THIS_MODULE, - models[pci_id->driver_data], - pci_id->driver_data); + xonar_ids, get_xonar_model); if (err >= 0) ++dev; return err; -- cgit v1.2.3-70-g09d2 From 1275d6f608abda23d101ada17dc39940192d4bc4 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 19 Feb 2009 08:44:12 +0100 Subject: sound: oxygen: automatically restore overwritten EEPROM If the EEPROM was partially overwritten (which seems to happen before the OS is booted), restore its entire contents by deducing it from the remaining information. This does not have any effect on the Linux driver, which works even with incomplete information in the EEPROM, but it makes other drivers work again. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/oxygen.h | 3 +++ sound/pci/oxygen/oxygen_io.c | 16 ++++++++++++++++ sound/pci/oxygen/oxygen_lib.c | 29 +++++++++++++++++++++++++++++ 3 files changed, 48 insertions(+) diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index c500d48ea34..bd615dbffad 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -18,6 +18,8 @@ #define OXYGEN_IO_SIZE 0x100 +#define OXYGEN_EEPROM_ID 0x434d /* "CM" */ + /* model-specific configuration of outputs/inputs */ #define PLAYBACK_0_TO_I2S 0x0001 /* PLAYBACK_0_TO_AC97_0 not implemented */ @@ -190,6 +192,7 @@ void oxygen_reset_uart(struct oxygen *chip); void oxygen_write_uart(struct oxygen *chip, u8 data); u16 oxygen_read_eeprom(struct oxygen *chip, unsigned int index); +void oxygen_write_eeprom(struct oxygen *chip, unsigned int index, u16 value); static inline void oxygen_set_bits8(struct oxygen *chip, unsigned int reg, u8 value) diff --git a/sound/pci/oxygen/oxygen_io.c b/sound/pci/oxygen/oxygen_io.c index 05f48ef1a44..c1eb923f2ac 100644 --- a/sound/pci/oxygen/oxygen_io.c +++ b/sound/pci/oxygen/oxygen_io.c @@ -269,3 +269,19 @@ u16 oxygen_read_eeprom(struct oxygen *chip, unsigned int index) } return oxygen_read16(chip, OXYGEN_EEPROM_DATA); } + +void oxygen_write_eeprom(struct oxygen *chip, unsigned int index, u16 value) +{ + unsigned int timeout; + + oxygen_write16(chip, OXYGEN_EEPROM_DATA, value); + oxygen_write8(chip, OXYGEN_EEPROM_CONTROL, + index | OXYGEN_EEPROM_DIR_WRITE); + for (timeout = 0; timeout < 10; ++timeout) { + msleep(1); + if (!(oxygen_read8(chip, OXYGEN_EEPROM_STATUS) + & OXYGEN_EEPROM_BUSY)) + return; + } + snd_printk(KERN_ERR "EEPROM write timeout\n"); +} diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index d83c3a95732..6e1cdd2fd76 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -272,6 +272,34 @@ oxygen_search_pci_id(struct oxygen *chip, const struct pci_device_id ids[]) return NULL; } +static void oxygen_restore_eeprom(struct oxygen *chip, + const struct pci_device_id *id) +{ + if (oxygen_read_eeprom(chip, 0) != OXYGEN_EEPROM_ID) { + /* + * This function gets called only when a known card model has + * been detected, i.e., we know there is a valid subsystem + * product ID at index 2 in the EEPROM. Therefore, we have + * been able to deduce the correct subsystem vendor ID, and + * this is enough information to restore the original EEPROM + * contents. + */ + oxygen_write_eeprom(chip, 1, id->subvendor); + oxygen_write_eeprom(chip, 0, OXYGEN_EEPROM_ID); + + oxygen_set_bits8(chip, OXYGEN_MISC, + OXYGEN_MISC_WRITE_PCI_SUBID); + pci_write_config_word(chip->pci, PCI_SUBSYSTEM_VENDOR_ID, + id->subvendor); + pci_write_config_word(chip->pci, PCI_SUBSYSTEM_ID, + id->subdevice); + oxygen_clear_bits8(chip, OXYGEN_MISC, + OXYGEN_MISC_WRITE_PCI_SUBID); + + snd_printk(KERN_INFO "EEPROM ID restored\n"); + } +} + static void oxygen_init(struct oxygen *chip) { unsigned int i; @@ -532,6 +560,7 @@ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id, err = -ENODEV; goto err_pci_regions; } + oxygen_restore_eeprom(chip, pci_id); err = get_model(chip, pci_id); if (err < 0) goto err_pci_regions; -- cgit v1.2.3-70-g09d2 From eca985d28e1a8092ba2686ec5485fd688df5cfb3 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Wed, 18 Feb 2009 19:07:18 +0100 Subject: sound: Remove documentation for OSS CS4232 driver There is no OSS cs4232 driver in the kernel any more and this documentation does not contain any info useful for ALSA driver. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- Documentation/sound/oss/CS4232 | 23 ----------------------- 1 file changed, 23 deletions(-) delete mode 100644 Documentation/sound/oss/CS4232 diff --git a/Documentation/sound/oss/CS4232 b/Documentation/sound/oss/CS4232 deleted file mode 100644 index 7d6af7a5c1c..00000000000 --- a/Documentation/sound/oss/CS4232 +++ /dev/null @@ -1,23 +0,0 @@ -To configure the Crystal CS423x sound chip and activate its DSP functions, -modules may be loaded in this order: - - modprobe sound - insmod ad1848 - insmod uart401 - insmod cs4232 io=* irq=* dma=* dma2=* - -This is the meaning of the parameters: - - io--I/O address of the Windows Sound System (normally 0x534) - irq--IRQ of this device - dma and dma2--DMA channels (DMA2 may be 0) - -On some cards, the board attempts to do non-PnP setup, and fails. If you -have problems, use Linux' PnP facilities. - -To get MIDI facilities add - - insmod opl3 io=* - -where "io" is the I/O address of the OPL3 synthesizer. This will be shown -in /proc/sys/pnp and is normally 0x388. -- cgit v1.2.3-70-g09d2 From ce3bdaa8710c10eec5a6dae67aaf73088d0ced4f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 19 Feb 2009 14:29:49 +0000 Subject: ASoC: Disable WM8731 line bypass by default MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This avoids temporarily enabling the ouput stages during startup which can cause audible effets in the output stages. Reported-by: Fredrik RedgÃ¥rd Signed-off-by: Mark Brown --- sound/soc/codecs/wm8731.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 4cac3195bfa..9e7ebcc2c49 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -594,6 +594,10 @@ static int wm8731_register(struct wm8731_priv *wm8731) reg = wm8731_read_reg_cache(codec, WM8731_RINVOL); wm8731_write(codec, WM8731_RINVOL, reg & ~0x0100); + /* Disable bypass path by default */ + reg = wm8731_read_reg_cache(codec, WM8731_APANA); + wm8731_write(codec, WM8731_APANA, reg & ~0x4); + wm8731_codec = codec; ret = snd_soc_register_codec(codec); -- cgit v1.2.3-70-g09d2 From d91b424d6d7bda0773b6b6b606d48d089c4f5115 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 20 Feb 2009 09:31:14 +0100 Subject: sound: oxygen: handle AK5385 ADC on Claro halo cards The HT-Omega Claro halo's ADC is an AK5385 instead of a WM8785, so we should handle the ADC parameters as we do with the X-Meridian. Using the code for the wrong ADC does not seem to have any audible effects, and the Windows driver does it, but it is nonetheless a good idea to run the AK5385 with an oversampling ratio that is not outside the documented limits. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/oxygen.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index f2c37f379d3..1d8e2b29745 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -196,6 +196,12 @@ static void meridian_init(struct oxygen *chip) ak5385_init(chip); } +static void halo_init(struct oxygen *chip) +{ + ak4396_init(chip); + ak5385_init(chip); +} + static void generic_cleanup(struct oxygen *chip) { } @@ -211,6 +217,11 @@ static void meridian_resume(struct oxygen *chip) ak4396_registers_init(chip); } +static void halo_resume(struct oxygen *chip) +{ + ak4396_registers_init(chip); +} + static void set_ak4396_params(struct oxygen *chip, struct snd_pcm_hw_params *params) { @@ -335,6 +346,11 @@ static int __devinit get_oxygen_model(struct oxygen *chip, CAPTURE_0_FROM_I2S_2 | CAPTURE_1_FROM_SPDIF; break; + case MODEL_HALO: + chip->model.init = halo_init; + chip->model.resume = halo_resume; + chip->model.set_adc_params = set_ak5385_params; + break; } if (id->driver_data == MODEL_MERIDIAN || id->driver_data == MODEL_HALO) { -- cgit v1.2.3-70-g09d2 From eacbb9dba6b4c982a0217ea2c7d15db88d4fda37 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 20 Feb 2009 09:33:40 +0100 Subject: sound: virtuoso: increase minimum volume to -60 dB Use -60 dB as the minimum value of the master volume mixer control. While the DACs would support ranges down to about -120 dB, such attenuations are not useful in practice. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/virtuoso.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 4ac49772da8..00dc97806f1 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -758,8 +758,8 @@ static void xonar_line_mic_ac97_switch(struct oxygen *chip, } } -static const DECLARE_TLV_DB_SCALE(pcm1796_db_scale, -12000, 50, 0); -static const DECLARE_TLV_DB_SCALE(cs4362a_db_scale, -12700, 100, 0); +static const DECLARE_TLV_DB_SCALE(pcm1796_db_scale, -6000, 50, 0); +static const DECLARE_TLV_DB_SCALE(cs4362a_db_scale, -6000, 100, 0); static int xonar_d2_control_filter(struct snd_kcontrol_new *template) { @@ -808,8 +808,8 @@ static const struct oxygen_model model_xonar_d2 = { MIDI_OUTPUT | MIDI_INPUT, .dac_channels = 8, - .dac_volume_min = 0x0f, - .dac_volume_max = 0xff, + .dac_volume_min = 255 - 2*60, + .dac_volume_max = 255, .misc_flags = OXYGEN_MISC_MIDI, .function_flags = OXYGEN_FUNCTION_SPI | OXYGEN_FUNCTION_ENABLE_SPI_4_5, @@ -837,7 +837,7 @@ static const struct oxygen_model model_xonar_d1 = { PLAYBACK_1_TO_SPDIF | CAPTURE_0_FROM_I2S_2, .dac_channels = 8, - .dac_volume_min = 0, + .dac_volume_min = 127 - 60, .dac_volume_max = 127, .function_flags = OXYGEN_FUNCTION_2WIRE, .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, @@ -864,8 +864,8 @@ static const struct oxygen_model model_xonar_hdav = { PLAYBACK_1_TO_SPDIF | CAPTURE_0_FROM_I2S_2, .dac_channels = 8, - .dac_volume_min = 0x0f, - .dac_volume_max = 0xff, + .dac_volume_min = 255 - 2*60, + .dac_volume_max = 255, .misc_flags = OXYGEN_MISC_MIDI, .function_flags = OXYGEN_FUNCTION_2WIRE, .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, -- cgit v1.2.3-70-g09d2 From f3990e610a157e9c36af85a75bc66260dff31f40 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 20 Feb 2009 09:32:40 +0100 Subject: sound: usb-audio: remove MIN_PACKS_URB Remove the MIN_PACKS_URB symbol because other limits can force the number of packets down to one, regardless of the value of this symbol, and nobody has ever changed it anyway. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/usb/usbaudio.c | 14 ++++++-------- 1 file changed, 6 insertions(+), 8 deletions(-) diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index c69cc6e4f54..2b24496ddec 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -107,7 +107,6 @@ MODULE_PARM_DESC(ignore_ctl_error, #define MAX_PACKS_HS (MAX_PACKS * 8) /* in high speed mode */ #define MAX_URBS 8 #define SYNC_URBS 4 /* always four urbs for sync */ -#define MIN_PACKS_URB 1 /* minimum 1 packet per urb */ #define MAX_QUEUE 24 /* try not to exceed this queue length, in ms */ struct audioformat { @@ -1071,8 +1070,7 @@ static int init_substream_urbs(struct snd_usb_substream *subs, unsigned int peri subs->packs_per_ms = packs_per_ms; if (is_playback) { - urb_packs = nrpacks; - urb_packs = max(urb_packs, (unsigned int)MIN_PACKS_URB); + urb_packs = max(nrpacks, 1); urb_packs = min(urb_packs, (unsigned int)MAX_PACKS); } else urb_packs = 1; @@ -1093,9 +1091,9 @@ static int init_substream_urbs(struct snd_usb_substream *subs, unsigned int peri total_packs = (total_packs + packs_per_ms - 1) & ~(packs_per_ms - 1); /* we need at least two URBs for queueing */ - if (total_packs < 2 * MIN_PACKS_URB * packs_per_ms) - total_packs = 2 * MIN_PACKS_URB * packs_per_ms; - else { + if (total_packs < 2 * packs_per_ms) { + total_packs = 2 * packs_per_ms; + } else { /* and we don't want too long a queue either */ maxpacks = max((unsigned int)MAX_QUEUE, urb_packs * 2); if (total_packs > maxpacks * packs_per_ms) @@ -1909,7 +1907,7 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre * in the current code assume the 1ms period. */ snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_TIME, - 1000 * MIN_PACKS_URB, + 1000, /*(nrpacks * MAX_URBS) * 1000*/ UINT_MAX); err = check_hw_params_convention(subs); @@ -3753,7 +3751,7 @@ static int usb_audio_resume(struct usb_interface *intf) static int __init snd_usb_audio_init(void) { - if (nrpacks < MIN_PACKS_URB || nrpacks > MAX_PACKS) { + if (nrpacks < 1 || nrpacks > MAX_PACKS) { printk(KERN_WARNING "invalid nrpacks value.\n"); return -EINVAL; } -- cgit v1.2.3-70-g09d2 From 3be141494a080a9189b51fa78154c975ad8d9806 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 20 Feb 2009 14:11:16 +0100 Subject: ALSA: hda - Add generic pincfg initialization Added the generic pincfg cache and save/restore functions. Also introduced the pin-overriding via hwdep sysfs. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 151 +++++++++++++++++++++++++++++++++++++++++++--- sound/pci/hda/hda_codec.h | 15 +++++ sound/pci/hda/hda_hwdep.c | 66 ++++++++++++++++++++ 3 files changed, 223 insertions(+), 9 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 98884bc8f35..6fa871f66a7 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -682,11 +682,132 @@ static int read_widget_caps(struct hda_codec *codec, hda_nid_t fg_node) return 0; } +/* read all pin default configurations and save codec->init_pins */ +static int read_pin_defaults(struct hda_codec *codec) +{ + int i; + hda_nid_t nid = codec->start_nid; + + for (i = 0; i < codec->num_nodes; i++, nid++) { + struct hda_pincfg *pin; + unsigned int wcaps = get_wcaps(codec, nid); + unsigned int wid_type = (wcaps & AC_WCAP_TYPE) >> + AC_WCAP_TYPE_SHIFT; + if (wid_type != AC_WID_PIN) + continue; + pin = snd_array_new(&codec->init_pins); + if (!pin) + return -ENOMEM; + pin->nid = nid; + pin->cfg = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_CONFIG_DEFAULT, 0); + } + return 0; +} + +/* look up the given pin config list and return the item matching with NID */ +static struct hda_pincfg *look_up_pincfg(struct hda_codec *codec, + struct snd_array *array, + hda_nid_t nid) +{ + int i; + for (i = 0; i < array->used; i++) { + struct hda_pincfg *pin = snd_array_elem(array, i); + if (pin->nid == nid) + return pin; + } + return NULL; +} + +/* write a config value for the given NID */ +static void set_pincfg(struct hda_codec *codec, hda_nid_t nid, + unsigned int cfg) +{ + int i; + for (i = 0; i < 4; i++) { + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_CONFIG_DEFAULT_BYTES_0 + i, + cfg & 0xff); + cfg >>= 8; + } +} + +/* set the current pin config value for the given NID. + * the value is cached, and read via snd_hda_codec_get_pincfg() + */ +int snd_hda_add_pincfg(struct hda_codec *codec, struct snd_array *list, + hda_nid_t nid, unsigned int cfg) +{ + struct hda_pincfg *pin; + + pin = look_up_pincfg(codec, list, nid); + if (!pin) { + pin = snd_array_new(list); + if (!pin) + return -ENOMEM; + pin->nid = nid; + } + pin->cfg = cfg; + set_pincfg(codec, nid, cfg); + return 0; +} + +int snd_hda_codec_set_pincfg(struct hda_codec *codec, + hda_nid_t nid, unsigned int cfg) +{ + return snd_hda_add_pincfg(codec, &codec->cur_pins, nid, cfg); +} +EXPORT_SYMBOL_HDA(snd_hda_codec_set_pincfg); + +/* get the current pin config value of the given pin NID */ +unsigned int snd_hda_codec_get_pincfg(struct hda_codec *codec, hda_nid_t nid) +{ + struct hda_pincfg *pin; + + pin = look_up_pincfg(codec, &codec->cur_pins, nid); + if (pin) + return pin->cfg; +#ifdef CONFIG_SND_HDA_HWDEP + pin = look_up_pincfg(codec, &codec->override_pins, nid); + if (pin) + return pin->cfg; +#endif + pin = look_up_pincfg(codec, &codec->init_pins, nid); + if (pin) + return pin->cfg; + return 0; +} +EXPORT_SYMBOL_HDA(snd_hda_codec_get_pincfg); + +/* restore all current pin configs */ +static void restore_pincfgs(struct hda_codec *codec) +{ + int i; + for (i = 0; i < codec->init_pins.used; i++) { + struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i); + set_pincfg(codec, pin->nid, + snd_hda_codec_get_pincfg(codec, pin->nid)); + } +} static void init_hda_cache(struct hda_cache_rec *cache, unsigned int record_size); static void free_hda_cache(struct hda_cache_rec *cache); +/* restore the initial pin cfgs and release all pincfg lists */ +static void restore_init_pincfgs(struct hda_codec *codec) +{ + /* first free cur_pins and override_pins, then call restore_pincfg + * so that only the values in init_pins are restored + */ + snd_array_free(&codec->cur_pins); +#ifdef CONFIG_SND_HDA_HWDEP + snd_array_free(&codec->override_pins); +#endif + restore_pincfgs(codec); + snd_array_free(&codec->init_pins); +} + /* * codec destructor */ @@ -694,6 +815,7 @@ static void snd_hda_codec_free(struct hda_codec *codec) { if (!codec) return; + restore_init_pincfgs(codec); #ifdef CONFIG_SND_HDA_POWER_SAVE cancel_delayed_work(&codec->power_work); flush_workqueue(codec->bus->workq); @@ -751,6 +873,8 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr init_hda_cache(&codec->amp_cache, sizeof(struct hda_amp_info)); init_hda_cache(&codec->cmd_cache, sizeof(struct hda_cache_head)); snd_array_init(&codec->mixers, sizeof(struct snd_kcontrol *), 32); + snd_array_init(&codec->init_pins, sizeof(struct hda_pincfg), 16); + snd_array_init(&codec->cur_pins, sizeof(struct hda_pincfg), 16); if (codec->bus->modelname) { codec->modelname = kstrdup(codec->bus->modelname, GFP_KERNEL); if (!codec->modelname) { @@ -787,15 +911,18 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr setup_fg_nodes(codec); if (!codec->afg && !codec->mfg) { snd_printdd("hda_codec: no AFG or MFG node found\n"); - snd_hda_codec_free(codec); - return -ENODEV; + err = -ENODEV; + goto error; } - if (read_widget_caps(codec, codec->afg ? codec->afg : codec->mfg) < 0) { + err = read_widget_caps(codec, codec->afg ? codec->afg : codec->mfg); + if (err < 0) { snd_printk(KERN_ERR "hda_codec: cannot malloc\n"); - snd_hda_codec_free(codec); - return -ENOMEM; + goto error; } + err = read_pin_defaults(codec); + if (err < 0) + goto error; if (!codec->subsystem_id) { hda_nid_t nid = codec->afg ? codec->afg : codec->mfg; @@ -808,10 +935,8 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr if (do_init) { err = snd_hda_codec_configure(codec); - if (err < 0) { - snd_hda_codec_free(codec); - return err; - } + if (err < 0) + goto error; } snd_hda_codec_proc_new(codec); @@ -824,6 +949,10 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr if (codecp) *codecp = codec; return 0; + + error: + snd_hda_codec_free(codec); + return err; } EXPORT_SYMBOL_HDA(snd_hda_codec_new); @@ -1334,6 +1463,9 @@ void snd_hda_codec_reset(struct hda_codec *codec) free_hda_cache(&codec->cmd_cache); init_hda_cache(&codec->amp_cache, sizeof(struct hda_amp_info)); init_hda_cache(&codec->cmd_cache, sizeof(struct hda_cache_head)); + /* free only cur_pins so that init_pins + override_pins are restored */ + snd_array_free(&codec->cur_pins); + restore_pincfgs(codec); codec->num_pcms = 0; codec->pcm_info = NULL; codec->preset = NULL; @@ -2175,6 +2307,7 @@ static void hda_call_codec_resume(struct hda_codec *codec) hda_set_power_state(codec, codec->afg ? codec->afg : codec->mfg, AC_PWRST_D0); + restore_pincfgs(codec); /* restore all current pin configs */ hda_exec_init_verbs(codec); if (codec->patch_ops.resume) codec->patch_ops.resume(codec); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 09a332ada0c..6d01a8058f0 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -778,11 +778,14 @@ struct hda_codec { unsigned short spdif_ctls; /* SPDIF control bits */ unsigned int spdif_in_enable; /* SPDIF input enable? */ hda_nid_t *slave_dig_outs; /* optional digital out slave widgets */ + struct snd_array init_pins; /* initial (BIOS) pin configurations */ + struct snd_array cur_pins; /* current pin configurations */ #ifdef CONFIG_SND_HDA_HWDEP struct snd_hwdep *hwdep; /* assigned hwdep device */ struct snd_array init_verbs; /* additional init verbs */ struct snd_array hints; /* additional hints */ + struct snd_array override_pins; /* default pin configs to override */ #endif /* misc flags */ @@ -855,6 +858,18 @@ void snd_hda_codec_resume_cache(struct hda_codec *codec); #define snd_hda_sequence_write_cache snd_hda_sequence_write #endif +/* the struct for codec->pin_configs */ +struct hda_pincfg { + hda_nid_t nid; + unsigned int cfg; +}; + +unsigned int snd_hda_codec_get_pincfg(struct hda_codec *codec, hda_nid_t nid); +int snd_hda_codec_set_pincfg(struct hda_codec *codec, hda_nid_t nid, + unsigned int cfg); +int snd_hda_add_pincfg(struct hda_codec *codec, struct snd_array *list, + hda_nid_t nid, unsigned int cfg); /* for hwdep */ + /* * Mixer */ diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index 4ae51dcb81a..71039a6dec2 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -109,6 +109,7 @@ static void clear_hwdep_elements(struct hda_codec *codec) for (i = 0; i < codec->hints.used; i++, head++) kfree(*head); snd_array_free(&codec->hints); + snd_array_free(&codec->override_pins); } static void hwdep_free(struct snd_hwdep *hwdep) @@ -141,6 +142,7 @@ int /*__devinit*/ snd_hda_create_hwdep(struct hda_codec *codec) snd_array_init(&codec->init_verbs, sizeof(struct hda_verb), 32); snd_array_init(&codec->hints, sizeof(char *), 32); + snd_array_init(&codec->override_pins, sizeof(struct hda_pincfg), 16); return 0; } @@ -316,6 +318,67 @@ static ssize_t hints_store(struct device *dev, return count; } +static ssize_t pin_configs_show(struct hda_codec *codec, + struct snd_array *list, + char *buf) +{ + int i, len = 0; + for (i = 0; i < list->used; i++) { + struct hda_pincfg *pin = snd_array_elem(list, i); + len += sprintf(buf + len, "0x%02x 0x%08x\n", + pin->nid, pin->cfg); + } + return len; +} + +static ssize_t init_pin_configs_show(struct device *dev, + struct device_attribute *attr, + char *buf) +{ + struct snd_hwdep *hwdep = dev_get_drvdata(dev); + struct hda_codec *codec = hwdep->private_data; + return pin_configs_show(codec, &codec->init_pins, buf); +} + +static ssize_t override_pin_configs_show(struct device *dev, + struct device_attribute *attr, + char *buf) +{ + struct snd_hwdep *hwdep = dev_get_drvdata(dev); + struct hda_codec *codec = hwdep->private_data; + return pin_configs_show(codec, &codec->override_pins, buf); +} + +static ssize_t cur_pin_configs_show(struct device *dev, + struct device_attribute *attr, + char *buf) +{ + struct snd_hwdep *hwdep = dev_get_drvdata(dev); + struct hda_codec *codec = hwdep->private_data; + return pin_configs_show(codec, &codec->cur_pins, buf); +} + +#define MAX_PIN_CONFIGS 32 + +static ssize_t override_pin_configs_store(struct device *dev, + struct device_attribute *attr, + const char *buf, size_t count) +{ + struct snd_hwdep *hwdep = dev_get_drvdata(dev); + struct hda_codec *codec = hwdep->private_data; + int nid, cfg; + int err; + + if (sscanf(buf, "%i %i", &nid, &cfg) != 2) + return -EINVAL; + if (!nid) + return -EINVAL; + err = snd_hda_add_pincfg(codec, &codec->override_pins, nid, cfg); + if (err < 0) + return err; + return count; +} + #define CODEC_ATTR_RW(type) \ __ATTR(type, 0644, type##_show, type##_store) #define CODEC_ATTR_RO(type) \ @@ -333,6 +396,9 @@ static struct device_attribute codec_attrs[] = { CODEC_ATTR_RW(modelname), CODEC_ATTR_WO(init_verbs), CODEC_ATTR_WO(hints), + CODEC_ATTR_RO(init_pin_configs), + CODEC_ATTR_RW(override_pin_configs), + CODEC_ATTR_RO(cur_pin_configs), CODEC_ATTR_WO(reconfig), CODEC_ATTR_WO(clear), }; -- cgit v1.2.3-70-g09d2 From 0e8a21b59d48a63f45b3e6d2aca7fb91c5aec882 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 20 Feb 2009 14:13:06 +0100 Subject: ALSA: hda - Remove realtek codec-specific pin save/restore functions Now it's done in the common code. Also use the common access functions for pin defaults. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 78 ++----------------------------------------- 1 file changed, 3 insertions(+), 75 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 169b3837af5..d7f255e3b91 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -329,13 +329,6 @@ struct alc_spec { /* for PLL fix */ hda_nid_t pll_nid; unsigned int pll_coef_idx, pll_coef_bit; - -#ifdef SND_HDA_NEEDS_RESUME -#define ALC_MAX_PINS 16 - unsigned int num_pins; - hda_nid_t pin_nids[ALC_MAX_PINS]; - unsigned int pin_cfgs[ALC_MAX_PINS]; -#endif }; /* @@ -1009,8 +1002,7 @@ static void alc_subsystem_id(struct hda_codec *codec, nid = 0x1d; if (codec->vendor_id == 0x10ec0260) nid = 0x17; - ass = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONFIG_DEFAULT, 0); + ass = snd_hda_codec_get_pincfg(codec, nid); if (!(ass & 1) && !(ass & 0x100000)) return; if ((ass >> 30) != 1) /* no physical connection */ @@ -1184,16 +1176,8 @@ static void alc_fix_pincfg(struct hda_codec *codec, return; cfg = pinfix[quirk->value]; - for (; cfg->nid; cfg++) { - int i; - u32 val = cfg->val; - for (i = 0; i < 4; i++) { - snd_hda_codec_write(codec, cfg->nid, 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_0 + i, - val & 0xff); - val >>= 8; - } - } + for (; cfg->nid; cfg++) + snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val); } /* @@ -3215,61 +3199,13 @@ static void alc_free(struct hda_codec *codec) } #ifdef SND_HDA_NEEDS_RESUME -static void store_pin_configs(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - hda_nid_t nid, end_nid; - - end_nid = codec->start_nid + codec->num_nodes; - for (nid = codec->start_nid; nid < end_nid; nid++) { - unsigned int wid_caps = get_wcaps(codec, nid); - unsigned int wid_type = - (wid_caps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; - if (wid_type != AC_WID_PIN) - continue; - if (spec->num_pins >= ARRAY_SIZE(spec->pin_nids)) - break; - spec->pin_nids[spec->num_pins] = nid; - spec->pin_cfgs[spec->num_pins] = - snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONFIG_DEFAULT, 0); - spec->num_pins++; - } -} - -static void resume_pin_configs(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - int i; - - for (i = 0; i < spec->num_pins; i++) { - hda_nid_t pin_nid = spec->pin_nids[i]; - unsigned int pin_config = spec->pin_cfgs[i]; - snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_0, - pin_config & 0x000000ff); - snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_1, - (pin_config & 0x0000ff00) >> 8); - snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_2, - (pin_config & 0x00ff0000) >> 16); - snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, - pin_config >> 24); - } -} - static int alc_resume(struct hda_codec *codec) { - resume_pin_configs(codec); codec->patch_ops.init(codec); snd_hda_codec_resume_amp(codec); snd_hda_codec_resume_cache(codec); return 0; } -#else -#define store_pin_configs(codec) #endif /* @@ -4329,7 +4265,6 @@ static int alc880_parse_auto_config(struct hda_codec *codec) spec->num_mux_defs = 1; spec->input_mux = &spec->private_imux[0]; - store_pin_configs(codec); return 1; } @@ -5693,7 +5628,6 @@ static int alc260_parse_auto_config(struct hda_codec *codec) spec->num_mux_defs = 1; spec->input_mux = &spec->private_imux[0]; - store_pin_configs(codec); return 1; } @@ -10688,7 +10622,6 @@ static int alc262_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; - store_pin_configs(codec); return 1; } @@ -11861,7 +11794,6 @@ static int alc268_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; - store_pin_configs(codec); return 1; } @@ -12774,7 +12706,6 @@ static int alc269_parse_auto_config(struct hda_codec *codec) if (!spec->cap_mixer && !spec->no_analog) set_capture_mixer(spec); - store_pin_configs(codec); return 1; } @@ -13825,7 +13756,6 @@ static int alc861_parse_auto_config(struct hda_codec *codec) spec->num_adc_nids = ARRAY_SIZE(alc861_adc_nids); set_capture_mixer(spec); - store_pin_configs(codec); return 1; } @@ -14927,7 +14857,6 @@ static int alc861vd_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; - store_pin_configs(codec); return 1; } @@ -16737,7 +16666,6 @@ static int alc662_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; - store_pin_configs(codec); return 1; } -- cgit v1.2.3-70-g09d2 From 330ee9957910826a072c2ad5d4045182335f9963 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 20 Feb 2009 14:33:36 +0100 Subject: ALSA: hda - Remove IDT codec-specific pin save/restore functions Removed its own save/restore functions and replaced with the common code. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 255 +++++++++++------------------------------ 1 file changed, 65 insertions(+), 190 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index d00a211a813..da48d8c0b29 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -229,7 +229,6 @@ struct sigmatel_spec { /* pin widgets */ hda_nid_t *pin_nids; unsigned int num_pins; - unsigned int *pin_configs; /* codec specific stuff */ struct hda_verb *init; @@ -2272,101 +2271,19 @@ static struct snd_pci_quirk stac9205_cfg_tbl[] = { {} /* terminator */ }; -static int stac92xx_save_bios_config_regs(struct hda_codec *codec) +static void stac92xx_set_config_regs(struct hda_codec *codec, + unsigned int *pincfgs) { int i; struct sigmatel_spec *spec = codec->spec; - - kfree(spec->pin_configs); - spec->pin_configs = kcalloc(spec->num_pins, sizeof(*spec->pin_configs), - GFP_KERNEL); - if (!spec->pin_configs) - return -ENOMEM; - - for (i = 0; i < spec->num_pins; i++) { - hda_nid_t nid = spec->pin_nids[i]; - unsigned int pin_cfg; - - if (!nid) - continue; - pin_cfg = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONFIG_DEFAULT, 0x00); - snd_printdd(KERN_INFO "hda_codec: pin nid %2.2x bios pin config %8.8x\n", - nid, pin_cfg); - spec->pin_configs[i] = pin_cfg; - } - - return 0; -} -static void stac92xx_set_config_reg(struct hda_codec *codec, - hda_nid_t pin_nid, unsigned int pin_config) -{ - int i; - snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_0, - pin_config & 0x000000ff); - snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_1, - (pin_config & 0x0000ff00) >> 8); - snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_2, - (pin_config & 0x00ff0000) >> 16); - snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, - pin_config >> 24); - i = snd_hda_codec_read(codec, pin_nid, 0, - AC_VERB_GET_CONFIG_DEFAULT, - 0x00); - snd_printdd(KERN_INFO "hda_codec: pin nid %2.2x pin config %8.8x\n", - pin_nid, i); -} - -static void stac92xx_set_config_regs(struct hda_codec *codec) -{ - int i; - struct sigmatel_spec *spec = codec->spec; - - if (!spec->pin_configs) - return; + if (!pincfgs) + return; for (i = 0; i < spec->num_pins; i++) - if (spec->pin_nids[i] && spec->pin_configs[i]) - stac92xx_set_config_reg(codec, spec->pin_nids[i], - spec->pin_configs[i]); -} - -static int stac_save_pin_cfgs(struct hda_codec *codec, unsigned int *pins) -{ - struct sigmatel_spec *spec = codec->spec; - - if (!pins) - return stac92xx_save_bios_config_regs(codec); - - kfree(spec->pin_configs); - spec->pin_configs = kmemdup(pins, - spec->num_pins * sizeof(*pins), - GFP_KERNEL); - if (!spec->pin_configs) - return -ENOMEM; - - stac92xx_set_config_regs(codec); - return 0; -} - -static void stac_change_pin_config(struct hda_codec *codec, hda_nid_t nid, - unsigned int cfg) -{ - struct sigmatel_spec *spec = codec->spec; - int i; - - for (i = 0; i < spec->num_pins; i++) { - if (spec->pin_nids[i] == nid) { - spec->pin_configs[i] = cfg; - stac92xx_set_config_reg(codec, nid, cfg); - break; - } - } + if (spec->pin_nids[i] && pincfgs[i]) + snd_hda_codec_set_pincfg(codec, spec->pin_nids[i], + pincfgs[i]); } /* @@ -2853,8 +2770,7 @@ static hda_nid_t check_mic_out_switch(struct hda_codec *codec) mic_pin = AUTO_PIN_MIC; for (;;) { hda_nid_t nid = cfg->input_pins[mic_pin]; - def_conf = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONFIG_DEFAULT, 0); + def_conf = snd_hda_codec_get_pincfg(codec, nid); /* some laptops have an internal analog microphone * which can't be used as a output */ if (get_defcfg_connect(def_conf) != AC_JACK_PORT_FIXED) { @@ -3426,11 +3342,7 @@ static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec, unsigned int wcaps; unsigned int def_conf; - def_conf = snd_hda_codec_read(codec, - spec->dmic_nids[i], - 0, - AC_VERB_GET_CONFIG_DEFAULT, - 0); + def_conf = snd_hda_codec_get_pincfg(codec, spec->dmic_nids[i]); if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE) continue; @@ -3779,9 +3691,7 @@ static int stac9200_auto_create_lfe_ctls(struct hda_codec *codec, for (i = 0; i < spec->autocfg.line_outs && lfe_pin == 0x0; i++) { hda_nid_t pin = spec->autocfg.line_out_pins[i]; unsigned int defcfg; - defcfg = snd_hda_codec_read(codec, pin, 0, - AC_VERB_GET_CONFIG_DEFAULT, - 0x00); + defcfg = snd_hda_codec_get_pincfg(codec, pin); if (get_defcfg_device(defcfg) == AC_JACK_SPEAKER) { unsigned int wcaps = get_wcaps(codec, pin); wcaps &= (AC_WCAP_STEREO | AC_WCAP_OUT_AMP); @@ -3885,8 +3795,7 @@ static int stac92xx_add_jack(struct hda_codec *codec, #ifdef CONFIG_SND_JACK struct sigmatel_spec *spec = codec->spec; struct sigmatel_jack *jack; - int def_conf = snd_hda_codec_read(codec, nid, - 0, AC_VERB_GET_CONFIG_DEFAULT, 0); + int def_conf = snd_hda_codec_get_pincfg(codec, nid); int connectivity = get_defcfg_connect(def_conf); char name[32]; @@ -4066,8 +3975,7 @@ static int stac92xx_init(struct hda_codec *codec) pinctl); } } - conf = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONFIG_DEFAULT, 0); + conf = snd_hda_codec_get_pincfg(codec, nid); if (get_defcfg_connect(conf) != AC_JACK_PORT_FIXED) { enable_pin_detect(codec, nid, STAC_INSERT_EVENT); @@ -4108,8 +4016,7 @@ static int stac92xx_init(struct hda_codec *codec) stac_toggle_power_map(codec, nid, 1); continue; } - def_conf = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONFIG_DEFAULT, 0); + def_conf = snd_hda_codec_get_pincfg(codec, nid); def_conf = get_defcfg_connect(def_conf); /* skip any ports that don't have jacks since presence * detection is useless */ @@ -4163,7 +4070,6 @@ static void stac92xx_free(struct hda_codec *codec) if (! spec) return; - kfree(spec->pin_configs); stac92xx_free_jacks(codec); snd_array_free(&spec->events); @@ -4474,7 +4380,6 @@ static int stac92xx_resume(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; - stac92xx_set_config_regs(codec); stac92xx_init(codec); snd_hda_codec_resume_amp(codec); snd_hda_codec_resume_cache(codec); @@ -4523,16 +4428,11 @@ static int patch_stac9200(struct hda_codec *codec) spec->board_config = snd_hda_check_board_config(codec, STAC_9200_MODELS, stac9200_models, stac9200_cfg_tbl); - if (spec->board_config < 0) { + if (spec->board_config < 0) snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC9200, using BIOS defaults\n"); - err = stac92xx_save_bios_config_regs(codec); - } else - err = stac_save_pin_cfgs(codec, + else + stac92xx_set_config_regs(codec, stac9200_brd_tbl[spec->board_config]); - if (err < 0) { - stac92xx_free(codec); - return err; - } spec->multiout.max_channels = 2; spec->multiout.num_dacs = 1; @@ -4600,17 +4500,12 @@ static int patch_stac925x(struct hda_codec *codec) stac925x_models, stac925x_cfg_tbl); again: - if (spec->board_config < 0) { + if (spec->board_config < 0) snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC925x," "using BIOS defaults\n"); - err = stac92xx_save_bios_config_regs(codec); - } else - err = stac_save_pin_cfgs(codec, + else + stac92xx_set_config_regs(codec, stac925x_brd_tbl[spec->board_config]); - if (err < 0) { - stac92xx_free(codec); - return err; - } spec->multiout.max_channels = 2; spec->multiout.num_dacs = 1; @@ -4688,17 +4583,12 @@ static int patch_stac92hd73xx(struct hda_codec *codec) stac92hd73xx_models, stac92hd73xx_cfg_tbl); again: - if (spec->board_config < 0) { + if (spec->board_config < 0) snd_printdd(KERN_INFO "hda_codec: Unknown model for" " STAC92HD73XX, using BIOS defaults\n"); - err = stac92xx_save_bios_config_regs(codec); - } else - err = stac_save_pin_cfgs(codec, + else + stac92xx_set_config_regs(codec, stac92hd73xx_brd_tbl[spec->board_config]); - if (err < 0) { - stac92xx_free(codec); - return err; - } num_dacs = snd_hda_get_connections(codec, 0x0a, conn, STAC92HD73_DAC_COUNT + 2) - 1; @@ -4758,18 +4648,18 @@ again: spec->init = dell_m6_core_init; switch (spec->board_config) { case STAC_DELL_M6_AMIC: /* Analog Mics */ - stac92xx_set_config_reg(codec, 0x0b, 0x90A70170); + snd_hda_codec_set_pincfg(codec, 0x0b, 0x90A70170); spec->num_dmics = 0; spec->private_dimux.num_items = 1; break; case STAC_DELL_M6_DMIC: /* Digital Mics */ - stac92xx_set_config_reg(codec, 0x13, 0x90A60160); + snd_hda_codec_set_pincfg(codec, 0x13, 0x90A60160); spec->num_dmics = 1; spec->private_dimux.num_items = 2; break; case STAC_DELL_M6_BOTH: /* Both */ - stac92xx_set_config_reg(codec, 0x0b, 0x90A70170); - stac92xx_set_config_reg(codec, 0x13, 0x90A60160); + snd_hda_codec_set_pincfg(codec, 0x0b, 0x90A70170); + snd_hda_codec_set_pincfg(codec, 0x13, 0x90A60160); spec->num_dmics = 1; spec->private_dimux.num_items = 2; break; @@ -4865,17 +4755,12 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) stac92hd83xxx_models, stac92hd83xxx_cfg_tbl); again: - if (spec->board_config < 0) { + if (spec->board_config < 0) snd_printdd(KERN_INFO "hda_codec: Unknown model for" " STAC92HD83XXX, using BIOS defaults\n"); - err = stac92xx_save_bios_config_regs(codec); - } else - err = stac_save_pin_cfgs(codec, + else + stac92xx_set_config_regs(codec, stac92hd83xxx_brd_tbl[spec->board_config]); - if (err < 0) { - stac92xx_free(codec); - return err; - } switch (codec->vendor_id) { case 0x111d7604: @@ -4945,6 +4830,16 @@ static struct hda_input_mux stac92hd71bxx_dmux_amixer = { } }; +/* get the pin connection (fixed, none, etc) */ +static unsigned int stac_get_defcfg_connect(struct hda_codec *codec, int idx) +{ + struct sigmatel_spec *spec = codec->spec; + unsigned int cfg; + + cfg = snd_hda_codec_get_pincfg(codec, spec->pin_nids[idx]); + return get_defcfg_connect(cfg); +} + static int stac92hd71bxx_connected_ports(struct hda_codec *codec, hda_nid_t *nids, int num_nids) { @@ -4958,7 +4853,7 @@ static int stac92hd71bxx_connected_ports(struct hda_codec *codec, break; if (idx >= spec->num_pins) break; - def_conf = get_defcfg_connect(spec->pin_configs[idx]); + def_conf = stac_get_defcfg_connect(codec, idx); if (def_conf == AC_JACK_PORT_NONE) break; } @@ -4978,13 +4873,13 @@ static int stac92hd71bxx_connected_smuxes(struct hda_codec *codec, return 0; /* dig1pin case */ - if (get_defcfg_connect(spec->pin_configs[idx+1]) != AC_JACK_PORT_NONE) + if (stac_get_defcfg_connect(codec, idx + 1) != AC_JACK_PORT_NONE) return 2; /* dig0pin + dig2pin case */ - if (get_defcfg_connect(spec->pin_configs[idx+2]) != AC_JACK_PORT_NONE) + if (stac_get_defcfg_connect(codec, idx + 2) != AC_JACK_PORT_NONE) return 2; - if (get_defcfg_connect(spec->pin_configs[idx]) != AC_JACK_PORT_NONE) + if (stac_get_defcfg_connect(codec, idx) != AC_JACK_PORT_NONE) return 1; else return 0; @@ -5023,17 +4918,12 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) stac92hd71bxx_models, stac92hd71bxx_cfg_tbl); again: - if (spec->board_config < 0) { + if (spec->board_config < 0) snd_printdd(KERN_INFO "hda_codec: Unknown model for" " STAC92HD71BXX, using BIOS defaults\n"); - err = stac92xx_save_bios_config_regs(codec); - } else - err = stac_save_pin_cfgs(codec, + else + stac92xx_set_config_regs(codec, stac92hd71bxx_brd_tbl[spec->board_config]); - if (err < 0) { - stac92xx_free(codec); - return err; - } if (spec->board_config > STAC_92HD71BXX_REF) { /* GPIO0 = EAPD */ @@ -5097,8 +4987,8 @@ again: /* disable VSW */ spec->init = &stac92hd71bxx_analog_core_init[HD_DISABLE_PORTF]; unmute_init++; - stac_change_pin_config(codec, 0x0f, 0x40f000f0); - stac_change_pin_config(codec, 0x19, 0x40f000f3); + snd_hda_codec_set_pincfg(codec, 0x0f, 0x40f000f0); + snd_hda_codec_set_pincfg(codec, 0x19, 0x40f000f3); stac92hd71bxx_dmic_nids[STAC92HD71BXX_NUM_DMICS - 1] = 0; spec->num_dmics = stac92hd71bxx_connected_ports(codec, stac92hd71bxx_dmic_nids, @@ -5147,7 +5037,7 @@ again: switch (spec->board_config) { case STAC_HP_M4: /* enable internal microphone */ - stac_change_pin_config(codec, 0x0e, 0x01813040); + snd_hda_codec_set_pincfg(codec, 0x0e, 0x01813040); stac92xx_auto_set_pinctl(codec, 0x0e, AC_PINCTL_IN_EN | AC_PINCTL_VREF_80); /* fallthru */ @@ -5163,7 +5053,7 @@ again: spec->num_dmuxes = 0; break; case STAC_HP_DV5: - stac_change_pin_config(codec, 0x0d, 0x90170010); + snd_hda_codec_set_pincfg(codec, 0x0d, 0x90170010); stac92xx_auto_set_pinctl(codec, 0x0d, AC_PINCTL_OUT_EN); break; }; @@ -5247,17 +5137,12 @@ static int patch_stac922x(struct hda_codec *codec) } again: - if (spec->board_config < 0) { + if (spec->board_config < 0) snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC922x, " "using BIOS defaults\n"); - err = stac92xx_save_bios_config_regs(codec); - } else - err = stac_save_pin_cfgs(codec, + else + stac92xx_set_config_regs(codec, stac922x_brd_tbl[spec->board_config]); - if (err < 0) { - stac92xx_free(codec); - return err; - } spec->adc_nids = stac922x_adc_nids; spec->mux_nids = stac922x_mux_nids; @@ -5315,17 +5200,12 @@ static int patch_stac927x(struct hda_codec *codec) stac927x_models, stac927x_cfg_tbl); again: - if (spec->board_config < 0) { + if (spec->board_config < 0) snd_printdd(KERN_INFO "hda_codec: Unknown model for" "STAC927x, using BIOS defaults\n"); - err = stac92xx_save_bios_config_regs(codec); - } else - err = stac_save_pin_cfgs(codec, + else + stac92xx_set_config_regs(codec, stac927x_brd_tbl[spec->board_config]); - if (err < 0) { - stac92xx_free(codec); - return err; - } spec->digbeep_nid = 0x23; spec->adc_nids = stac927x_adc_nids; @@ -5354,15 +5234,15 @@ static int patch_stac927x(struct hda_codec *codec) case 0x10280209: case 0x1028022e: /* correct the device field to SPDIF out */ - stac_change_pin_config(codec, 0x21, 0x01442070); + snd_hda_codec_set_pincfg(codec, 0x21, 0x01442070); break; }; /* configure the analog microphone on some laptops */ - stac_change_pin_config(codec, 0x0c, 0x90a79130); + snd_hda_codec_set_pincfg(codec, 0x0c, 0x90a79130); /* correct the front output jack as a hp out */ - stac_change_pin_config(codec, 0x0f, 0x0227011f); + snd_hda_codec_set_pincfg(codec, 0x0f, 0x0227011f); /* correct the front input jack as a mic */ - stac_change_pin_config(codec, 0x0e, 0x02a79130); + snd_hda_codec_set_pincfg(codec, 0x0e, 0x02a79130); /* fallthru */ case STAC_DELL_3ST: /* GPIO2 High = Enable EAPD */ @@ -5447,16 +5327,11 @@ static int patch_stac9205(struct hda_codec *codec) stac9205_models, stac9205_cfg_tbl); again: - if (spec->board_config < 0) { + if (spec->board_config < 0) snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC9205, using BIOS defaults\n"); - err = stac92xx_save_bios_config_regs(codec); - } else - err = stac_save_pin_cfgs(codec, + else + stac92xx_set_config_regs(codec, stac9205_brd_tbl[spec->board_config]); - if (err < 0) { - stac92xx_free(codec); - return err; - } spec->digbeep_nid = 0x23; spec->adc_nids = stac9205_adc_nids; @@ -5484,8 +5359,8 @@ static int patch_stac9205(struct hda_codec *codec) switch (spec->board_config){ case STAC_9205_DELL_M43: /* Enable SPDIF in/out */ - stac_change_pin_config(codec, 0x1f, 0x01441030); - stac_change_pin_config(codec, 0x20, 0x1c410030); + snd_hda_codec_set_pincfg(codec, 0x1f, 0x01441030); + snd_hda_codec_set_pincfg(codec, 0x20, 0x1c410030); /* Enable unsol response for GPIO4/Dock HP connection */ err = stac_add_event(spec, codec->afg, STAC_VREF_EVENT, 0x01); -- cgit v1.2.3-70-g09d2 From 2f334f92cfb44d17b9f24a43f8998cca03f9a3dd Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 20 Feb 2009 14:37:42 +0100 Subject: ALSA: hda - Remove codec-specific pin save/restore functions Replace the accessor to pin defaults with the common code for caching. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 3 +-- sound/pci/hda/patch_cmedia.c | 12 ++++++------ sound/pci/hda/patch_via.c | 7 ++----- 3 files changed, 9 insertions(+), 13 deletions(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 2c58d7b05ab..53d0edaf04c 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1047,8 +1047,7 @@ static struct hda_amp_list ad1986a_loopbacks[] = { static int is_jack_available(struct hda_codec *codec, hda_nid_t nid) { - unsigned int conf = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONFIG_DEFAULT, 0); + unsigned int conf = snd_hda_codec_get_pincfg(codec, nid); return get_defcfg_connect(conf) != AC_JACK_PORT_NONE; } diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index f3ebe837f2d..c921264bbd7 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -680,13 +680,13 @@ static int patch_cmi9880(struct hda_codec *codec) struct auto_pin_cfg cfg; /* collect pin default configuration */ - port_e = snd_hda_codec_read(codec, 0x0f, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); - port_f = snd_hda_codec_read(codec, 0x10, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); + port_e = snd_hda_codec_get_pincfg(codec, 0x0f); + port_f = snd_hda_codec_get_pincfg(codec, 0x10); spec->front_panel = 1; if (get_defcfg_connect(port_e) == AC_JACK_PORT_NONE || get_defcfg_connect(port_f) == AC_JACK_PORT_NONE) { - port_g = snd_hda_codec_read(codec, 0x1f, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); - port_h = snd_hda_codec_read(codec, 0x20, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); + port_g = snd_hda_codec_get_pincfg(codec, 0x1f); + port_h = snd_hda_codec_get_pincfg(codec, 0x20); spec->channel_modes = cmi9880_channel_modes; /* no front panel */ if (get_defcfg_connect(port_g) == AC_JACK_PORT_NONE || @@ -703,8 +703,8 @@ static int patch_cmi9880(struct hda_codec *codec) spec->multiout.max_channels = cmi9880_channel_modes[0].channels; } else { spec->input_mux = &cmi9880_basic_mux; - port_spdifi = snd_hda_codec_read(codec, 0x13, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); - port_spdifo = snd_hda_codec_read(codec, 0x12, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); + port_spdifi = snd_hda_codec_get_pincfg(codec, 0x13); + port_spdifo = snd_hda_codec_get_pincfg(codec, 0x12); if (get_defcfg_connect(port_spdifo) != AC_JACK_PORT_NONE) spec->multiout.dig_out_nid = CMI_DIG_OUT_NID; if (get_defcfg_connect(port_spdifi) != AC_JACK_PORT_NONE) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 639b2ff510a..b25a5cc637d 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1308,16 +1308,13 @@ static void vt1708_set_pinconfig_connect(struct hda_codec *codec, hda_nid_t nid) unsigned int def_conf; unsigned char seqassoc; - def_conf = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONFIG_DEFAULT, 0); + def_conf = snd_hda_codec_get_pincfg(codec, nid); seqassoc = (unsigned char) get_defcfg_association(def_conf); seqassoc = (seqassoc << 4) | get_defcfg_sequence(def_conf); if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE) { if (seqassoc == 0xff) { def_conf = def_conf & (~(AC_JACK_PORT_BOTH << 30)); - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, - def_conf >> 24); + snd_hda_codec_set_pincfg(codec, nid, def_conf); } } -- cgit v1.2.3-70-g09d2 From f1085c4f319f1e43c95718045a235f276cc4b615 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 20 Feb 2009 14:50:35 +0100 Subject: ALSA: hda - Update documentation for pincfg sysfs entries Added the brief descriptions of new sysfs entries for pint default config values. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio.txt | 14 +++++++++++++- 1 file changed, 13 insertions(+), 1 deletion(-) diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt index 99f7fbbe3e6..9c51e104546 100644 --- a/Documentation/sound/alsa/HD-Audio.txt +++ b/Documentation/sound/alsa/HD-Audio.txt @@ -365,10 +365,22 @@ modelname:: to this file. init_verbs:: The extra verbs to execute at initialization. You can add a verb by - writing to this file. Pass tree numbers, nid, verb and parameter. + writing to this file. Pass three numbers: nid, verb and parameter. hints:: Shows hint strings for codec parsers for any use. Right now it's not used. +init_pin_configs:: + Shows the initial pin default config values set by BIOS. +override_pin_configs:: + Shows the pin default config values to override the BIOS setup. + Writing this (with two numbers, NID and value) appends the new + value. The given will be used instead of the initial BIOS value at + the next reconfiguration time. +cur_pin_configs:: + Shows the pin default values set by the codec parser explicitly. + This doesn't show all pin values but only the changed values by + the parser. That is, if the parser doesn't change the pin default + config values by itself, this will contain nothing. reconfig:: Triggers the codec re-configuration. When any value is written to this file, the driver re-initialize and parses the codec tree -- cgit v1.2.3-70-g09d2 From eeb1080b29a0fa00e426ba77eb96f3a157b335ab Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 22 Feb 2009 14:19:23 +0000 Subject: ASoC: Report I/O errors from WM8753 reset Signed-off-by: Mark Brown --- sound/soc/codecs/wm8753.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 31ff337f822..180ec94ad8a 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1561,7 +1561,11 @@ static int wm8753_init(struct snd_soc_device *socdev) wm8753_set_dai_mode(codec, 0); - wm8753_reset(codec); + ret = wm8753_reset(codec); + if (ret < 0) { + printk(KERN_ERR "wm8753: failed to reset device\n"); + return ret; + } /* register pcms */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); -- cgit v1.2.3-70-g09d2 From 93e051d2771e6bf70e86b8265bfbf296a457d044 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 22 Feb 2009 14:24:00 +0000 Subject: ASoC: Only unregister drivers we registered for WM8753 Signed-off-by: Mark Brown --- sound/soc/codecs/wm8753.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 180ec94ad8a..93c22c4f082 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1845,6 +1845,7 @@ static int wm8753_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; + struct wm8753_setup_data *setup = socdev->codec_data; if (codec->control_data) wm8753_set_bias_level(codec, SND_SOC_BIAS_OFF); @@ -1852,11 +1853,14 @@ static int wm8753_remove(struct platform_device *pdev) snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - i2c_unregister_device(codec->control_data); - i2c_del_driver(&wm8753_i2c_driver); + if (setup->i2c_address) { + i2c_unregister_device(codec->control_data); + i2c_del_driver(&wm8753_i2c_driver); + } #endif #if defined(CONFIG_SPI_MASTER) - spi_unregister_driver(&wm8753_spi_driver); + if (setup->spi) + spi_unregister_driver(&wm8753_spi_driver); #endif kfree(codec->private_data); kfree(codec); -- cgit v1.2.3-70-g09d2 From cc95948972576c3efa43c9ed05b4a265805a4c54 Mon Sep 17 00:00:00 2001 From: Michael Schwingen Date: Sun, 22 Feb 2009 18:58:45 +0100 Subject: ALSA: hda - add support for "Maxdata Favorit 100XS" (Intel HDA/ALC260) Signed-off-by: Michael Schwingen Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 130 ++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 130 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 169b3837af5..abddabc1efa 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -78,6 +78,7 @@ enum { ALC260_ACER, ALC260_WILL, ALC260_REPLACER_672V, + ALC260_FAVORIT100, #ifdef CONFIG_SND_DEBUG ALC260_TEST, #endif @@ -4537,6 +4538,26 @@ static struct hda_input_mux alc260_acer_capture_sources[2] = { }, }, }; + +/* Maxdata Favorit 100XS */ +static struct hda_input_mux alc260_favorit100_capture_sources[2] = { + { + .num_items = 2, + .items = { + { "Line/Mic", 0x0 }, + { "CD", 0x4 }, + }, + }, + { + .num_items = 3, + .items = { + { "Line/Mic", 0x0 }, + { "CD", 0x4 }, + { "Mixer", 0x5 }, + }, + }, +}; + /* * This is just place-holder, so there's something for alc_build_pcms to look * at when it calculates the maximum number of channels. ALC260 has no mixer @@ -4817,6 +4838,18 @@ static struct snd_kcontrol_new alc260_acer_mixer[] = { { } /* end */ }; +/* Maxdata Favorit 100XS: one output and one input (0x12) jack + */ +static struct snd_kcontrol_new alc260_favorit100_mixer[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Master Playback Switch", 0x08, 2, HDA_INPUT), + ALC_PIN_MODE("Output Jack Mode", 0x0f, ALC_PIN_DIR_INOUT), + HDA_CODEC_VOLUME("Line/Mic Playback Volume", 0x07, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Line/Mic Playback Switch", 0x07, 0x0, HDA_INPUT), + ALC_PIN_MODE("Line/Mic Jack Mode", 0x12, ALC_PIN_DIR_IN), + { } /* end */ +}; + /* Packard bell V7900 ALC260 pin usage: HP = 0x0f, Mic jack = 0x12, * Line In jack = 0x14, CD audio = 0x16, pc beep = 0x17. */ @@ -5188,6 +5221,89 @@ static struct hda_verb alc260_acer_init_verbs[] = { { } }; +/* Initialisation sequence for Maxdata Favorit 100XS + * (adapted from Acer init verbs). + */ +static struct hda_verb alc260_favorit100_init_verbs[] = { + /* GPIO 0 enables the output jack. + * Turn this on and rely on the standard mute + * methods whenever the user wants to turn these outputs off. + */ + {0x01, AC_VERB_SET_GPIO_MASK, 0x01}, + {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01}, + {0x01, AC_VERB_SET_GPIO_DATA, 0x01}, + /* Line/Mic input jack is connected to Mic1 pin */ + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50}, + /* Ensure all other unused pins are disabled and muted. */ + {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + /* Disable digital (SPDIF) pins */ + {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0}, + {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0}, + + /* Ensure Mic1 and Line1 pin widgets take input from the OUT1 sum + * bus when acting as outputs. + */ + {0x0b, AC_VERB_SET_CONNECT_SEL, 0}, + {0x0d, AC_VERB_SET_CONNECT_SEL, 0}, + + /* Start with output sum widgets muted and their output gains at min */ + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + + /* Unmute Line-out pin widget amp left and right + * (no equiv mixer ctrl) + */ + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Unmute Mic1 and Line1 pin widget input buffers since they start as + * inputs. If the pin mode is changed by the user the pin mode control + * will take care of enabling the pin's input/output buffers as needed. + * Therefore there's no need to enable the input buffer at this + * stage. + */ + {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + /* Mute capture amp left and right */ + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + /* Set ADC connection select to match default mixer setting - mic + * (on mic1 pin) + */ + {0x04, AC_VERB_SET_CONNECT_SEL, 0x00}, + + /* Do similar with the second ADC: mute capture input amp and + * set ADC connection to mic to match ALSA's default state. + */ + {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x05, AC_VERB_SET_CONNECT_SEL, 0x00}, + + /* Mute all inputs to mixer widget (even unconnected ones) */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */ + + { } +}; + static struct hda_verb alc260_will_verbs[] = { {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {0x0b, AC_VERB_SET_CONNECT_SEL, 0x00}, @@ -5730,6 +5846,7 @@ static const char *alc260_models[ALC260_MODEL_LAST] = { [ALC260_ACER] = "acer", [ALC260_WILL] = "will", [ALC260_REPLACER_672V] = "replacer", + [ALC260_FAVORIT100] = "favorit100", #ifdef CONFIG_SND_DEBUG [ALC260_TEST] = "test", #endif @@ -5739,6 +5856,7 @@ static const char *alc260_models[ALC260_MODEL_LAST] = { static struct snd_pci_quirk alc260_cfg_tbl[] = { SND_PCI_QUIRK(0x1025, 0x007b, "Acer C20x", ALC260_ACER), SND_PCI_QUIRK(0x1025, 0x008f, "Acer", ALC260_ACER), + SND_PCI_QUIRK(0x1509, 0x4540, "Favorit 100XS", ALC260_FAVORIT100), SND_PCI_QUIRK(0x103c, 0x2808, "HP d5700", ALC260_HP_3013), SND_PCI_QUIRK(0x103c, 0x280a, "HP d5750", ALC260_HP_3013), SND_PCI_QUIRK(0x103c, 0x3010, "HP", ALC260_HP_3013), @@ -5840,6 +5958,18 @@ static struct alc_config_preset alc260_presets[] = { .num_mux_defs = ARRAY_SIZE(alc260_acer_capture_sources), .input_mux = alc260_acer_capture_sources, }, + [ALC260_FAVORIT100] = { + .mixers = { alc260_favorit100_mixer }, + .init_verbs = { alc260_favorit100_init_verbs }, + .num_dacs = ARRAY_SIZE(alc260_dac_nids), + .dac_nids = alc260_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids), + .adc_nids = alc260_dual_adc_nids, + .num_channel_mode = ARRAY_SIZE(alc260_modes), + .channel_mode = alc260_modes, + .num_mux_defs = ARRAY_SIZE(alc260_favorit100_capture_sources), + .input_mux = alc260_favorit100_capture_sources, + }, [ALC260_WILL] = { .mixers = { alc260_will_mixer }, .init_verbs = { alc260_init_verbs, alc260_will_verbs }, -- cgit v1.2.3-70-g09d2 From ce71bfd1aa6d6a4069929eeceed254e13400ddf4 Mon Sep 17 00:00:00 2001 From: Andreas Mohr Date: Sun, 22 Feb 2009 20:33:41 +0100 Subject: ALSA: ALS4000, slight mixer improvements - add 8kHz / 20 kHz low-pass filter switch control - add ALS4000 Mono capture route control - add annotations to specs pages - improve ALS4000 PM saved regs selection (remove SB dummy register, add missing ones) - add some missing ALS4000 register defines - constify two variables Signed-off-by: Andreas Mohr Signed-off-by: Takashi Iwai --- include/sound/sb.h | 4 +- sound/isa/sb/sb_mixer.c | 156 ++++++++++++++++++++++++++++++++++++------------ 2 files changed, 121 insertions(+), 39 deletions(-) diff --git a/include/sound/sb.h b/include/sound/sb.h index 85f93c5fe1e..4e62ee1e411 100644 --- a/include/sound/sb.h +++ b/include/sound/sb.h @@ -249,6 +249,7 @@ struct snd_sb { #define SB_ALS4000_3D_AUTO_MUTE 0x52 #define SB_ALS4000_ANALOG_BLOCK_CTRL 0x53 #define SB_ALS4000_3D_DELAYLINE_PATTERN 0x54 +#define SB_ALS4000_CR3_CONFIGURATION 0xc3 /* bit 7 is Digital Loop Enable */ #define SB_ALS4000_QSOUND 0xdb /* IRQ setting bitmap */ @@ -330,7 +331,8 @@ enum { SB_MIX_DOUBLE, SB_MIX_INPUT_SW, SB_MIX_CAPTURE_PRO, - SB_MIX_CAPTURE_DT019X + SB_MIX_CAPTURE_DT019X, + SB_MIX_MONO_CAPTURE_ALS4K }; #define SB_MIXVAL_DOUBLE(left_reg, right_reg, left_shift, right_shift, mask) \ diff --git a/sound/isa/sb/sb_mixer.c b/sound/isa/sb/sb_mixer.c index 406a431af91..475220bbcc9 100644 --- a/sound/isa/sb/sb_mixer.c +++ b/sound/isa/sb/sb_mixer.c @@ -182,7 +182,7 @@ static int snd_sbmixer_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_ static int snd_dt019x_input_sw_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[5] = { + static const char *texts[5] = { "CD", "Mic", "Line", "Synth", "Master" }; @@ -268,13 +268,74 @@ static int snd_dt019x_input_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl return change; } +/* + * ALS4000 mono recording control switch + */ + +static int snd_als4k_mono_capture_route_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char *texts[3] = { + "L chan only", "R chan only", "L ch/2 + R ch/2" + }; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 3; + if (uinfo->value.enumerated.item > 2) + uinfo->value.enumerated.item = 2; + strcpy(uinfo->value.enumerated.name, + texts[uinfo->value.enumerated.item]); + return 0; +} + +static int snd_als4k_mono_capture_route_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_sb *sb = snd_kcontrol_chip(kcontrol); + unsigned long flags; + unsigned char oval; + + spin_lock_irqsave(&sb->mixer_lock, flags); + oval = snd_sbmixer_read(sb, SB_ALS4000_MONO_IO_CTRL); + spin_unlock_irqrestore(&sb->mixer_lock, flags); + oval >>= 6; + if (oval > 2) + oval = 2; + + ucontrol->value.enumerated.item[0] = oval; + return 0; +} + +static int snd_als4k_mono_capture_route_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_sb *sb = snd_kcontrol_chip(kcontrol); + unsigned long flags; + int change; + unsigned char nval, oval; + + if (ucontrol->value.enumerated.item[0] > 2) + return -EINVAL; + spin_lock_irqsave(&sb->mixer_lock, flags); + oval = snd_sbmixer_read(sb, SB_ALS4000_MONO_IO_CTRL); + + nval = (oval & ~(3 << 6)) + | (ucontrol->value.enumerated.item[0] << 6); + change = nval != oval; + if (change) + snd_sbmixer_write(sb, SB_ALS4000_MONO_IO_CTRL, nval); + spin_unlock_irqrestore(&sb->mixer_lock, flags); + return change; +} + /* * SBPRO input multiplexer */ static int snd_sb8mixer_info_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[3] = { + static const char *texts[3] = { "Mic", "CD", "Line" }; @@ -442,6 +503,12 @@ int snd_sbmixer_add_ctl(struct snd_sb *chip, const char *name, int index, int ty .get = snd_dt019x_input_sw_get, .put = snd_dt019x_input_sw_put, }, + [SB_MIX_MONO_CAPTURE_ALS4K] = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .info = snd_als4k_mono_capture_route_info, + .get = snd_als4k_mono_capture_route_get, + .put = snd_als4k_mono_capture_route_put, + }, }; struct snd_kcontrol *ctl; int err; @@ -636,6 +703,8 @@ static struct sbmix_elem snd_dt019x_ctl_capture_source = }; static struct sbmix_elem *snd_dt019x_controls[] = { + /* ALS4000 below has some parts which we might be lacking, + * e.g. snd_als4000_ctl_mono_playback_switch - check it! */ &snd_dt019x_ctl_master_play_vol, &snd_dt019x_ctl_pcm_play_vol, &snd_dt019x_ctl_synth_play_vol, @@ -666,18 +735,21 @@ static unsigned char snd_dt019x_init_values[][2] = { /* * ALS4000 specific mixer elements */ -/* FIXME: SB_ALS4000_MONO_IO_CTRL needs output select ctrl! */ static struct sbmix_elem snd_als4000_ctl_master_mono_playback_switch = SB_SINGLE("Master Mono Playback Switch", SB_ALS4000_MONO_IO_CTRL, 5, 1); -static struct sbmix_elem snd_als4000_ctl_master_mono_capture_route = - SB_SINGLE("Master Mono Capture Route", SB_ALS4000_MONO_IO_CTRL, 6, 0x03); -/* FIXME: mono playback switch also available on DT019X? */ +static struct sbmix_elem snd_als4k_ctl_master_mono_capture_route = { + .name = "Master Mono Capture Route", + .type = SB_MIX_MONO_CAPTURE_ALS4K + }; static struct sbmix_elem snd_als4000_ctl_mono_playback_switch = SB_SINGLE("Mono Playback Switch", SB_DT019X_OUTPUT_SW2, 0, 1); static struct sbmix_elem snd_als4000_ctl_mic_20db_boost = SB_SINGLE("Mic Boost (+20dB)", SB_ALS4000_MIC_IN_GAIN, 0, 0x03); -static struct sbmix_elem snd_als4000_ctl_mixer_loopback = - SB_SINGLE("Analog Loopback", SB_ALS4000_MIC_IN_GAIN, 7, 0x01); +static struct sbmix_elem snd_als4000_ctl_mixer_analog_loopback = + SB_SINGLE("Analog Loopback Switch", SB_ALS4000_MIC_IN_GAIN, 7, 0x01); +static struct sbmix_elem snd_als4000_ctl_mixer_digital_loopback = + SB_SINGLE("Digital Loopback Switch", + SB_ALS4000_CR3_CONFIGURATION, 7, 0x01); /* FIXME: functionality of 3D controls might be swapped, I didn't find * a description of how to identify what is supposed to be what */ static struct sbmix_elem snd_als4000_3d_control_switch = @@ -694,6 +766,9 @@ static struct sbmix_elem snd_als4000_3d_control_delay = SB_SINGLE("3D Control - Wide", SB_ALS4000_3D_TIME_DELAY, 0, 0x0f); static struct sbmix_elem snd_als4000_3d_control_poweroff_switch = SB_SINGLE("3D PowerOff Switch", SB_ALS4000_3D_TIME_DELAY, 4, 0x01); +static struct sbmix_elem snd_als4000_ctl_3db_freq_control_switch = + SB_SINGLE("Master Playback 8kHz / 20kHz LPF Switch", + SB_ALS4000_FMDAC, 5, 0x01); #ifdef NOT_AVAILABLE static struct sbmix_elem snd_als4000_ctl_fmdac = SB_SINGLE("FMDAC Switch (Option ?)", SB_ALS4000_FMDAC, 0, 0x01); @@ -702,35 +777,37 @@ static struct sbmix_elem snd_als4000_ctl_qsound = #endif static struct sbmix_elem *snd_als4000_controls[] = { - &snd_sb16_ctl_master_play_vol, - &snd_dt019x_ctl_pcm_play_switch, - &snd_sb16_ctl_pcm_play_vol, - &snd_sb16_ctl_synth_capture_route, - &snd_dt019x_ctl_synth_play_switch, - &snd_sb16_ctl_synth_play_vol, - &snd_sb16_ctl_cd_capture_route, - &snd_sb16_ctl_cd_play_switch, - &snd_sb16_ctl_cd_play_vol, - &snd_sb16_ctl_line_capture_route, - &snd_sb16_ctl_line_play_switch, - &snd_sb16_ctl_line_play_vol, - &snd_sb16_ctl_mic_capture_route, - &snd_als4000_ctl_mic_20db_boost, - &snd_sb16_ctl_auto_mic_gain, - &snd_sb16_ctl_mic_play_switch, - &snd_sb16_ctl_mic_play_vol, - &snd_sb16_ctl_pc_speaker_vol, - &snd_sb16_ctl_capture_vol, - &snd_sb16_ctl_play_vol, - &snd_als4000_ctl_master_mono_playback_switch, - &snd_als4000_ctl_master_mono_capture_route, - &snd_als4000_ctl_mono_playback_switch, - &snd_als4000_ctl_mixer_loopback, - &snd_als4000_3d_control_switch, - &snd_als4000_3d_control_ratio, - &snd_als4000_3d_control_freq, - &snd_als4000_3d_control_delay, - &snd_als4000_3d_control_poweroff_switch, + /* ALS4000a.PDF regs page */ + &snd_sb16_ctl_master_play_vol, /* MX30/31 12 */ + &snd_dt019x_ctl_pcm_play_switch, /* MX4C 16 */ + &snd_sb16_ctl_pcm_play_vol, /* MX32/33 12 */ + &snd_sb16_ctl_synth_capture_route, /* MX3D/3E 14 */ + &snd_dt019x_ctl_synth_play_switch, /* MX4C 16 */ + &snd_sb16_ctl_synth_play_vol, /* MX34/35 12/13 */ + &snd_sb16_ctl_cd_capture_route, /* MX3D/3E 14 */ + &snd_sb16_ctl_cd_play_switch, /* MX3C 14 */ + &snd_sb16_ctl_cd_play_vol, /* MX36/37 13 */ + &snd_sb16_ctl_line_capture_route, /* MX3D/3E 14 */ + &snd_sb16_ctl_line_play_switch, /* MX3C 14 */ + &snd_sb16_ctl_line_play_vol, /* MX38/39 13 */ + &snd_sb16_ctl_mic_capture_route, /* MX3D/3E 14 */ + &snd_als4000_ctl_mic_20db_boost, /* MX4D 16 */ + &snd_sb16_ctl_mic_play_switch, /* MX3C 14 */ + &snd_sb16_ctl_mic_play_vol, /* MX3A 13 */ + &snd_sb16_ctl_pc_speaker_vol, /* MX3B 14 */ + &snd_sb16_ctl_capture_vol, /* MX3F/40 15 */ + &snd_sb16_ctl_play_vol, /* MX41/42 15 */ + &snd_als4000_ctl_master_mono_playback_switch, /* MX4C 16 */ + &snd_als4k_ctl_master_mono_capture_route, /* MX4B 16 */ + &snd_als4000_ctl_mono_playback_switch, /* MX4C 16 */ + &snd_als4000_ctl_mixer_analog_loopback, /* MX4D 16 */ + &snd_als4000_ctl_mixer_digital_loopback, /* CR3 21 */ + &snd_als4000_3d_control_switch, /* MX50 17 */ + &snd_als4000_3d_control_ratio, /* MX50 17 */ + &snd_als4000_3d_control_freq, /* MX50 17 */ + &snd_als4000_3d_control_delay, /* MX51 18 */ + &snd_als4000_3d_control_poweroff_switch, /* MX51 18 */ + &snd_als4000_ctl_3db_freq_control_switch, /* MX4F 17 */ #ifdef NOT_AVAILABLE &snd_als4000_ctl_fmdac, &snd_als4000_ctl_qsound, @@ -905,13 +982,14 @@ static unsigned char dt019x_saved_regs[] = { }; static unsigned char als4000_saved_regs[] = { + /* please verify in dsheet whether regs to be added + are actually real H/W or just dummy */ SB_DSP4_MASTER_DEV, SB_DSP4_MASTER_DEV + 1, SB_DSP4_OUTPUT_SW, SB_DSP4_PCM_DEV, SB_DSP4_PCM_DEV + 1, SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, SB_DSP4_SYNTH_DEV, SB_DSP4_SYNTH_DEV + 1, SB_DSP4_CD_DEV, SB_DSP4_CD_DEV + 1, - SB_DSP4_MIC_AGC, SB_DSP4_MIC_DEV, SB_DSP4_SPEAKER_DEV, SB_DSP4_IGAIN_DEV, SB_DSP4_IGAIN_DEV + 1, @@ -919,8 +997,10 @@ static unsigned char als4000_saved_regs[] = { SB_DT019X_OUTPUT_SW2, SB_ALS4000_MONO_IO_CTRL, SB_ALS4000_MIC_IN_GAIN, + SB_ALS4000_FMDAC, SB_ALS4000_3D_SND_FX, SB_ALS4000_3D_TIME_DELAY, + SB_ALS4000_CR3_CONFIGURATION, }; static void save_mixer(struct snd_sb *chip, unsigned char *regs, int num_regs) -- cgit v1.2.3-70-g09d2 From e588ed8304f76cbb396ee85e657a58990298a675 Mon Sep 17 00:00:00 2001 From: Tim Blechmann Date: Fri, 20 Feb 2009 19:30:35 +0100 Subject: ALSA: hdsp - poll for iobox sleeping for 2 seconds before checking for the iobox is not enough on some systems. this patch increases the timeout, but polls the card during that time. it thus speeds up the module loading when the card has already been initialized, while being more robust on systems, which require a higher timeout than the predefined 2 seconds. Signed-off-by: Tim Blechmann Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdsp.c | 29 +++++++++++++++++++++++++---- 1 file changed, 25 insertions(+), 4 deletions(-) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index bacfdd12619..12c6b4305ec 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -653,7 +653,6 @@ static unsigned int hdsp_read(struct hdsp *hdsp, int reg) static int hdsp_check_for_iobox (struct hdsp *hdsp) { - if (hdsp->io_type == H9652 || hdsp->io_type == H9632) return 0; if (hdsp_read (hdsp, HDSP_statusRegister) & HDSP_ConfigError) { snd_printk ("Hammerfall-DSP: no Digiface or Multiface connected!\n"); @@ -661,7 +660,29 @@ static int hdsp_check_for_iobox (struct hdsp *hdsp) return -EIO; } return 0; +} +static int hdsp_wait_for_iobox(struct hdsp *hdsp, unsigned int loops, + unsigned int delay) +{ + unsigned int i; + + if (hdsp->io_type == H9652 || hdsp->io_type == H9632) + return 0; + + for (i = 0; i != loops; ++i) { + if (hdsp_read(hdsp, HDSP_statusRegister) & HDSP_ConfigError) + msleep(delay); + else { + snd_printd("Hammerfall-DSP: iobox found after %ums!\n", + i * delay); + return 0; + } + } + + snd_printk("Hammerfall-DSP: no Digiface or Multiface connected!\n"); + hdsp->state &= ~HDSP_FirmwareLoaded; + return -EIO; } static int snd_hdsp_load_firmware_from_cache(struct hdsp *hdsp) { @@ -5046,10 +5067,10 @@ static int __devinit snd_hdsp_create(struct snd_card *card, return err; if (!is_9652 && !is_9632) { - /* we wait 2 seconds to let freshly inserted cardbus cards do their hardware init */ - ssleep(2); + /* we wait a maximum of 10 seconds to let freshly + * inserted cardbus cards do their hardware init */ + err = hdsp_wait_for_iobox(hdsp, 1000, 10); - err = hdsp_check_for_iobox(hdsp); if (err < 0) return err; -- cgit v1.2.3-70-g09d2 From f9ffc5d6f0161b66202f2df9ecc42d1be241020d Mon Sep 17 00:00:00 2001 From: Tim Blechmann Date: Fri, 20 Feb 2009 19:38:16 +0100 Subject: ALSA: hdsp - whitespace cleanup Impact: remove trailing spaces Signed-off-by: Tim Blechmann Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdsp.c | 474 +++++++++++++++++++++++------------------------ 1 file changed, 237 insertions(+), 237 deletions(-) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 12c6b4305ec..dc65fe1c9c6 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -113,7 +113,7 @@ MODULE_FIRMWARE("digiface_firmware_rev11.bin"); /* the meters are regular i/o-mapped registers, but offset considerably from the rest. the peak registers are reset - when read; the least-significant 4 bits are full-scale counters; + when read; the least-significant 4 bits are full-scale counters; the actual peak value is in the most-significant 24 bits. */ @@ -131,7 +131,7 @@ MODULE_FIRMWARE("digiface_firmware_rev11.bin"); 26*3 values are read in ss mode 14*3 in ds mode, with no gap between values */ -#define HDSP_9652_peakBase 7164 +#define HDSP_9652_peakBase 7164 #define HDSP_9652_rmsBase 4096 /* c.f. the hdsp_9632_meters_t struct */ @@ -173,12 +173,12 @@ MODULE_FIRMWARE("digiface_firmware_rev11.bin"); #define HDSP_SPDIFEmphasis (1<<10) /* 0=none, 1=on */ #define HDSP_SPDIFNonAudio (1<<11) /* 0=off, 1=on */ #define HDSP_SPDIFOpticalOut (1<<12) /* 1=use 1st ADAT connector for SPDIF, 0=do not */ -#define HDSP_SyncRef2 (1<<13) -#define HDSP_SPDIFInputSelect0 (1<<14) -#define HDSP_SPDIFInputSelect1 (1<<15) -#define HDSP_SyncRef0 (1<<16) +#define HDSP_SyncRef2 (1<<13) +#define HDSP_SPDIFInputSelect0 (1<<14) +#define HDSP_SPDIFInputSelect1 (1<<15) +#define HDSP_SyncRef0 (1<<16) #define HDSP_SyncRef1 (1<<17) -#define HDSP_AnalogExtensionBoard (1<<18) /* For H9632 cards */ +#define HDSP_AnalogExtensionBoard (1<<18) /* For H9632 cards */ #define HDSP_XLRBreakoutCable (1<<20) /* For H9632 cards */ #define HDSP_Midi0InterruptEnable (1<<22) #define HDSP_Midi1InterruptEnable (1<<23) @@ -314,7 +314,7 @@ MODULE_FIRMWARE("digiface_firmware_rev11.bin"); #define HDSP_TimecodeSync (1<<27) #define HDSP_AEBO (1<<28) /* H9632 specific Analog Extension Boards */ #define HDSP_AEBI (1<<29) /* 0 = present, 1 = absent */ -#define HDSP_midi0IRQPending (1<<30) +#define HDSP_midi0IRQPending (1<<30) #define HDSP_midi1IRQPending (1<<31) #define HDSP_spdifFrequencyMask (HDSP_spdifFrequency0|HDSP_spdifFrequency1|HDSP_spdifFrequency2) @@ -391,7 +391,7 @@ MODULE_FIRMWARE("digiface_firmware_rev11.bin"); #define HDSP_CHANNEL_BUFFER_BYTES (4*HDSP_CHANNEL_BUFFER_SAMPLES) /* the size of the area we need to allocate for DMA transfers. the - size is the same regardless of the number of channels - the + size is the same regardless of the number of channels - the Multiface still uses the same memory area. Note that we allocate 1 more channel than is apparently needed @@ -460,7 +460,7 @@ struct hdsp { unsigned char qs_in_channels; /* quad speed mode for H9632 */ unsigned char ds_in_channels; unsigned char ss_in_channels; /* different for multiface/digiface */ - unsigned char qs_out_channels; + unsigned char qs_out_channels; unsigned char ds_out_channels; unsigned char ss_out_channels; @@ -502,9 +502,9 @@ static char channel_map_df_ss[HDSP_MAX_CHANNELS] = { static char channel_map_mf_ss[HDSP_MAX_CHANNELS] = { /* Multiface */ /* Analog */ - 0, 1, 2, 3, 4, 5, 6, 7, + 0, 1, 2, 3, 4, 5, 6, 7, /* ADAT 2 */ - 16, 17, 18, 19, 20, 21, 22, 23, + 16, 17, 18, 19, 20, 21, 22, 23, /* SPDIF */ 24, 25, -1, -1, -1, -1, -1, -1, -1, -1 @@ -525,11 +525,11 @@ static char channel_map_H9632_ss[HDSP_MAX_CHANNELS] = { /* SPDIF */ 8, 9, /* Analog */ - 10, 11, + 10, 11, /* AO4S-192 and AI4S-192 extension boards */ 12, 13, 14, 15, /* others don't exist */ - -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, -1, -1 }; @@ -539,7 +539,7 @@ static char channel_map_H9632_ds[HDSP_MAX_CHANNELS] = { /* SPDIF */ 8, 9, /* Analog */ - 10, 11, + 10, 11, /* AO4S-192 and AI4S-192 extension boards */ 12, 13, 14, 15, /* others don't exist */ @@ -587,7 +587,7 @@ static void snd_hammerfall_free_buffer(struct snd_dma_buffer *dmab, struct pci_d static struct pci_device_id snd_hdsp_ids[] = { { .vendor = PCI_VENDOR_ID_XILINX, - .device = PCI_DEVICE_ID_XILINX_HAMMERFALL_DSP, + .device = PCI_DEVICE_ID_XILINX_HAMMERFALL_DSP, .subvendor = PCI_ANY_ID, .subdevice = PCI_ANY_ID, }, /* RME Hammerfall-DSP */ @@ -691,19 +691,19 @@ static int snd_hdsp_load_firmware_from_cache(struct hdsp *hdsp) { unsigned long flags; if ((hdsp_read (hdsp, HDSP_statusRegister) & HDSP_DllError) != 0) { - + snd_printk ("Hammerfall-DSP: loading firmware\n"); hdsp_write (hdsp, HDSP_control2Reg, HDSP_S_PROGRAM); hdsp_write (hdsp, HDSP_fifoData, 0); - + if (hdsp_fifo_wait (hdsp, 0, HDSP_LONG_WAIT)) { snd_printk ("Hammerfall-DSP: timeout waiting for download preparation\n"); return -EIO; } - + hdsp_write (hdsp, HDSP_control2Reg, HDSP_S_LOAD); - + for (i = 0; i < 24413; ++i) { hdsp_write(hdsp, HDSP_fifoData, hdsp->firmware_cache[i]); if (hdsp_fifo_wait (hdsp, 127, HDSP_LONG_WAIT)) { @@ -713,7 +713,7 @@ static int snd_hdsp_load_firmware_from_cache(struct hdsp *hdsp) { } ssleep(3); - + if (hdsp_fifo_wait (hdsp, 0, HDSP_LONG_WAIT)) { snd_printk ("Hammerfall-DSP: timeout at end of firmware loading\n"); return -EIO; @@ -726,15 +726,15 @@ static int snd_hdsp_load_firmware_from_cache(struct hdsp *hdsp) { #endif hdsp_write (hdsp, HDSP_control2Reg, hdsp->control2_register); snd_printk ("Hammerfall-DSP: finished firmware loading\n"); - + } if (hdsp->state & HDSP_InitializationComplete) { snd_printk(KERN_INFO "Hammerfall-DSP: firmware loaded from cache, restoring defaults\n"); spin_lock_irqsave(&hdsp->lock, flags); snd_hdsp_set_defaults(hdsp); - spin_unlock_irqrestore(&hdsp->lock, flags); + spin_unlock_irqrestore(&hdsp->lock, flags); } - + hdsp->state |= HDSP_FirmwareLoaded; return 0; @@ -743,7 +743,7 @@ static int snd_hdsp_load_firmware_from_cache(struct hdsp *hdsp) { static int hdsp_get_iobox_version (struct hdsp *hdsp) { if ((hdsp_read (hdsp, HDSP_statusRegister) & HDSP_DllError) != 0) { - + hdsp_write (hdsp, HDSP_control2Reg, HDSP_PROGRAM); hdsp_write (hdsp, HDSP_fifoData, 0); if (hdsp_fifo_wait (hdsp, 0, HDSP_SHORT_WAIT) < 0) @@ -759,7 +759,7 @@ static int hdsp_get_iobox_version (struct hdsp *hdsp) hdsp_fifo_wait (hdsp, 0, HDSP_SHORT_WAIT); } else { hdsp->io_type = Digiface; - } + } } else { /* firmware was already loaded, get iobox type */ if (hdsp_read(hdsp, HDSP_status2Register) & HDSP_version1) @@ -807,13 +807,13 @@ static int hdsp_check_for_firmware (struct hdsp *hdsp, int load_on_demand) static int hdsp_fifo_wait(struct hdsp *hdsp, int count, int timeout) -{ +{ int i; /* the fifoStatus registers reports on how many words are available in the command FIFO. */ - + for (i = 0; i < timeout; i++) { if ((int)(hdsp_read (hdsp, HDSP_fifoStatus) & 0xff) <= count) @@ -845,11 +845,11 @@ static int hdsp_write_gain(struct hdsp *hdsp, unsigned int addr, unsigned short if (addr >= HDSP_MATRIX_MIXER_SIZE) return -1; - + if (hdsp->io_type == H9652 || hdsp->io_type == H9632) { /* from martin bjornsen: - + "You can only write dwords to the mixer memory which contain two mixer values in the low and high @@ -868,7 +868,7 @@ static int hdsp_write_gain(struct hdsp *hdsp, unsigned int addr, unsigned short hdsp->mixer_matrix[addr] = data; - + /* `addr' addresses a 16-bit wide address, but the address space accessed via hdsp_write uses byte offsets. put another way, addr @@ -877,17 +877,17 @@ static int hdsp_write_gain(struct hdsp *hdsp, unsigned int addr, unsigned short to access 0 to 2703 ... */ ad = addr/2; - - hdsp_write (hdsp, 4096 + (ad*4), - (hdsp->mixer_matrix[(addr&0x7fe)+1] << 16) + + + hdsp_write (hdsp, 4096 + (ad*4), + (hdsp->mixer_matrix[(addr&0x7fe)+1] << 16) + hdsp->mixer_matrix[addr&0x7fe]); - + return 0; } else { ad = (addr << 16) + data; - + if (hdsp_fifo_wait(hdsp, 127, HDSP_LONG_WAIT)) return -1; @@ -923,7 +923,7 @@ static int hdsp_spdif_sample_rate(struct hdsp *hdsp) if (status & HDSP_SPDIFErrorFlag) return 0; - + switch (rate_bits) { case HDSP_spdifFrequency32KHz: return 32000; case HDSP_spdifFrequency44_1KHz: return 44100; @@ -931,13 +931,13 @@ static int hdsp_spdif_sample_rate(struct hdsp *hdsp) case HDSP_spdifFrequency64KHz: return 64000; case HDSP_spdifFrequency88_2KHz: return 88200; case HDSP_spdifFrequency96KHz: return 96000; - case HDSP_spdifFrequency128KHz: + case HDSP_spdifFrequency128KHz: if (hdsp->io_type == H9632) return 128000; break; - case HDSP_spdifFrequency176_4KHz: + case HDSP_spdifFrequency176_4KHz: if (hdsp->io_type == H9632) return 176400; break; - case HDSP_spdifFrequency192KHz: + case HDSP_spdifFrequency192KHz: if (hdsp->io_type == H9632) return 192000; break; default: @@ -1048,7 +1048,7 @@ static void hdsp_set_dds_value(struct hdsp *hdsp, int rate) { u64 n; u32 r; - + if (rate >= 112000) rate /= 4; else if (rate >= 56000) @@ -1074,35 +1074,35 @@ static int hdsp_set_rate(struct hdsp *hdsp, int rate, int called_internally) there is no need for it (e.g. during module initialization). */ - - if (!(hdsp->control_register & HDSP_ClockModeMaster)) { + + if (!(hdsp->control_register & HDSP_ClockModeMaster)) { if (called_internally) { /* request from ctl or card initialization */ snd_printk(KERN_ERR "Hammerfall-DSP: device is not running as a clock master: cannot set sample rate.\n"); return -1; - } else { + } else { /* hw_param request while in AutoSync mode */ int external_freq = hdsp_external_sample_rate(hdsp); int spdif_freq = hdsp_spdif_sample_rate(hdsp); - + if ((spdif_freq == external_freq*2) && (hdsp_autosync_ref(hdsp) >= HDSP_AUTOSYNC_FROM_ADAT1)) snd_printk(KERN_INFO "Hammerfall-DSP: Detected ADAT in double speed mode\n"); else if (hdsp->io_type == H9632 && (spdif_freq == external_freq*4) && (hdsp_autosync_ref(hdsp) >= HDSP_AUTOSYNC_FROM_ADAT1)) - snd_printk(KERN_INFO "Hammerfall-DSP: Detected ADAT in quad speed mode\n"); + snd_printk(KERN_INFO "Hammerfall-DSP: Detected ADAT in quad speed mode\n"); else if (rate != external_freq) { snd_printk(KERN_INFO "Hammerfall-DSP: No AutoSync source for requested rate\n"); return -1; - } - } + } + } } current_rate = hdsp->system_sample_rate; /* Changing from a "single speed" to a "double speed" rate is not allowed if any substreams are open. This is because - such a change causes a shift in the location of + such a change causes a shift in the location of the DMA buffers and a reduction in the number of available - buffers. + buffers. Note that a similar but essentially insoluble problem exists for externally-driven rate changes. All we can do @@ -1110,7 +1110,7 @@ static int hdsp_set_rate(struct hdsp *hdsp, int rate, int called_internally) if (rate > 96000 && hdsp->io_type != H9632) return -EINVAL; - + switch (rate) { case 32000: if (current_rate > 48000) @@ -1200,7 +1200,7 @@ static int hdsp_set_rate(struct hdsp *hdsp, int rate, int called_internally) break; } } - + hdsp->system_sample_rate = rate; return 0; @@ -1266,16 +1266,16 @@ static int snd_hdsp_midi_output_write (struct hdsp_midi *hmidi) unsigned char buf[128]; /* Output is not interrupt driven */ - + spin_lock_irqsave (&hmidi->lock, flags); if (hmidi->output) { if (!snd_rawmidi_transmit_empty (hmidi->output)) { if ((n_pending = snd_hdsp_midi_output_possible (hmidi->hdsp, hmidi->id)) > 0) { if (n_pending > (int)sizeof (buf)) n_pending = sizeof (buf); - + if ((to_write = snd_rawmidi_transmit (hmidi->output, buf, n_pending)) > 0) { - for (i = 0; i < to_write; ++i) + for (i = 0; i < to_write; ++i) snd_hdsp_midi_write_byte (hmidi->hdsp, hmidi->id, buf[i]); } } @@ -1346,14 +1346,14 @@ static void snd_hdsp_midi_output_timer(unsigned long data) { struct hdsp_midi *hmidi = (struct hdsp_midi *) data; unsigned long flags; - + snd_hdsp_midi_output_write(hmidi); spin_lock_irqsave (&hmidi->lock, flags); /* this does not bump hmidi->istimer, because the kernel automatically removed the timer when it expired, and we are now adding it back, thus - leaving istimer wherever it was set before. + leaving istimer wherever it was set before. */ if (hmidi->istimer) { @@ -1522,7 +1522,7 @@ static int snd_hdsp_control_spdif_info(struct snd_kcontrol *kcontrol, struct snd static int snd_hdsp_control_spdif_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + snd_hdsp_convert_to_aes(&ucontrol->value.iec958, hdsp->creg_spdif); return 0; } @@ -1532,7 +1532,7 @@ static int snd_hdsp_control_spdif_put(struct snd_kcontrol *kcontrol, struct snd_ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; u32 val; - + val = snd_hdsp_convert_from_aes(&ucontrol->value.iec958); spin_lock_irq(&hdsp->lock); change = val != hdsp->creg_spdif; @@ -1551,7 +1551,7 @@ static int snd_hdsp_control_spdif_stream_info(struct snd_kcontrol *kcontrol, str static int snd_hdsp_control_spdif_stream_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + snd_hdsp_convert_to_aes(&ucontrol->value.iec958, hdsp->creg_spdif_stream); return 0; } @@ -1561,7 +1561,7 @@ static int snd_hdsp_control_spdif_stream_put(struct snd_kcontrol *kcontrol, stru struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; u32 val; - + val = snd_hdsp_convert_from_aes(&ucontrol->value.iec958); spin_lock_irq(&hdsp->lock); change = val != hdsp->creg_spdif_stream; @@ -1623,7 +1623,7 @@ static int snd_hdsp_info_spdif_in(struct snd_kcontrol *kcontrol, struct snd_ctl_ static int snd_hdsp_get_spdif_in(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_spdif_in(hdsp); return 0; } @@ -1633,7 +1633,7 @@ static int snd_hdsp_put_spdif_in(struct snd_kcontrol *kcontrol, struct snd_ctl_e struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.enumerated.item[0] % ((hdsp->io_type == H9632) ? 4 : 3); @@ -1670,7 +1670,7 @@ static int hdsp_set_spdif_output(struct hdsp *hdsp, int out) static int snd_hdsp_get_spdif_out(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.integer.value[0] = hdsp_spdif_out(hdsp); return 0; } @@ -1680,7 +1680,7 @@ static int snd_hdsp_put_spdif_out(struct snd_kcontrol *kcontrol, struct snd_ctl_ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -1714,7 +1714,7 @@ static int hdsp_set_spdif_professional(struct hdsp *hdsp, int val) static int snd_hdsp_get_spdif_professional(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.integer.value[0] = hdsp_spdif_professional(hdsp); return 0; } @@ -1724,7 +1724,7 @@ static int snd_hdsp_put_spdif_professional(struct snd_kcontrol *kcontrol, struct struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -1758,7 +1758,7 @@ static int hdsp_set_spdif_emphasis(struct hdsp *hdsp, int val) static int snd_hdsp_get_spdif_emphasis(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.integer.value[0] = hdsp_spdif_emphasis(hdsp); return 0; } @@ -1768,7 +1768,7 @@ static int snd_hdsp_put_spdif_emphasis(struct snd_kcontrol *kcontrol, struct snd struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -1802,7 +1802,7 @@ static int hdsp_set_spdif_nonaudio(struct hdsp *hdsp, int val) static int snd_hdsp_get_spdif_nonaudio(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.integer.value[0] = hdsp_spdif_nonaudio(hdsp); return 0; } @@ -1812,7 +1812,7 @@ static int snd_hdsp_put_spdif_nonaudio(struct snd_kcontrol *kcontrol, struct snd struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -1849,7 +1849,7 @@ static int snd_hdsp_info_spdif_sample_rate(struct snd_kcontrol *kcontrol, struct static int snd_hdsp_get_spdif_sample_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + switch (hdsp_spdif_sample_rate(hdsp)) { case 32000: ucontrol->value.enumerated.item[0] = 0; @@ -1879,7 +1879,7 @@ static int snd_hdsp_get_spdif_sample_rate(struct snd_kcontrol *kcontrol, struct ucontrol->value.enumerated.item[0] = 9; break; default: - ucontrol->value.enumerated.item[0] = 6; + ucontrol->value.enumerated.item[0] = 6; } return 0; } @@ -1903,7 +1903,7 @@ static int snd_hdsp_info_system_sample_rate(struct snd_kcontrol *kcontrol, struc static int snd_hdsp_get_system_sample_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp->system_sample_rate; return 0; } @@ -1920,7 +1920,7 @@ static int snd_hdsp_get_system_sample_rate(struct snd_kcontrol *kcontrol, struct static int snd_hdsp_info_autosync_sample_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - static char *texts[] = {"32000", "44100", "48000", "64000", "88200", "96000", "None", "128000", "176400", "192000"}; + static char *texts[] = {"32000", "44100", "48000", "64000", "88200", "96000", "None", "128000", "176400", "192000"}; uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = (hdsp->io_type == H9632) ? 10 : 7 ; @@ -1933,7 +1933,7 @@ static int snd_hdsp_info_autosync_sample_rate(struct snd_kcontrol *kcontrol, str static int snd_hdsp_get_autosync_sample_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + switch (hdsp_external_sample_rate(hdsp)) { case 32000: ucontrol->value.enumerated.item[0] = 0; @@ -1961,9 +1961,9 @@ static int snd_hdsp_get_autosync_sample_rate(struct snd_kcontrol *kcontrol, stru break; case 192000: ucontrol->value.enumerated.item[0] = 9; - break; + break; default: - ucontrol->value.enumerated.item[0] = 6; + ucontrol->value.enumerated.item[0] = 6; } return 0; } @@ -1989,7 +1989,7 @@ static int hdsp_system_clock_mode(struct hdsp *hdsp) static int snd_hdsp_info_system_clock_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static char *texts[] = {"Master", "Slave" }; - + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = 2; @@ -2002,7 +2002,7 @@ static int snd_hdsp_info_system_clock_mode(struct snd_kcontrol *kcontrol, struct static int snd_hdsp_get_system_clock_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_system_clock_mode(hdsp); return 0; } @@ -2039,7 +2039,7 @@ static int hdsp_clock_source(struct hdsp *hdsp) case 192000: return 9; default: - return 3; + return 3; } } else { return 0; @@ -2053,7 +2053,7 @@ static int hdsp_set_clock_source(struct hdsp *hdsp, int mode) case HDSP_CLOCK_SOURCE_AUTOSYNC: if (hdsp_external_sample_rate(hdsp) != 0) { if (!hdsp_set_rate(hdsp, hdsp_external_sample_rate(hdsp), 1)) { - hdsp->control_register &= ~HDSP_ClockModeMaster; + hdsp->control_register &= ~HDSP_ClockModeMaster; hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register); return 0; } @@ -2064,7 +2064,7 @@ static int hdsp_set_clock_source(struct hdsp *hdsp, int mode) break; case HDSP_CLOCK_SOURCE_INTERNAL_44_1KHZ: rate = 44100; - break; + break; case HDSP_CLOCK_SOURCE_INTERNAL_48KHZ: rate = 48000; break; @@ -2099,13 +2099,13 @@ static int snd_hdsp_info_clock_source(struct snd_kcontrol *kcontrol, struct snd_ { static char *texts[] = {"AutoSync", "Internal 32.0 kHz", "Internal 44.1 kHz", "Internal 48.0 kHz", "Internal 64.0 kHz", "Internal 88.2 kHz", "Internal 96.0 kHz", "Internal 128 kHz", "Internal 176.4 kHz", "Internal 192.0 KHz" }; struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; if (hdsp->io_type == H9632) uinfo->value.enumerated.items = 10; else - uinfo->value.enumerated.items = 7; + uinfo->value.enumerated.items = 7; if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); @@ -2115,7 +2115,7 @@ static int snd_hdsp_info_clock_source(struct snd_kcontrol *kcontrol, struct snd_ static int snd_hdsp_get_clock_source(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_clock_source(hdsp); return 0; } @@ -2125,7 +2125,7 @@ static int snd_hdsp_put_clock_source(struct snd_kcontrol *kcontrol, struct snd_c struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.enumerated.item[0]; @@ -2151,7 +2151,7 @@ static int snd_hdsp_put_clock_source(struct snd_kcontrol *kcontrol, struct snd_c static int snd_hdsp_get_clock_source_lock(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.integer.value[0] = hdsp->clock_source_locked; return 0; } @@ -2186,7 +2186,7 @@ static int hdsp_da_gain(struct hdsp *hdsp) case HDSP_DAGainMinus10dBV: return 2; default: - return 1; + return 1; } } @@ -2201,8 +2201,8 @@ static int hdsp_set_da_gain(struct hdsp *hdsp, int mode) hdsp->control_register |= HDSP_DAGainPlus4dBu; break; case 2: - hdsp->control_register |= HDSP_DAGainMinus10dBV; - break; + hdsp->control_register |= HDSP_DAGainMinus10dBV; + break; default: return -1; @@ -2214,7 +2214,7 @@ static int hdsp_set_da_gain(struct hdsp *hdsp, int mode) static int snd_hdsp_info_da_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static char *texts[] = {"Hi Gain", "+4 dBu", "-10 dbV"}; - + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = 3; @@ -2227,7 +2227,7 @@ static int snd_hdsp_info_da_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_e static int snd_hdsp_get_da_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_da_gain(hdsp); return 0; } @@ -2237,7 +2237,7 @@ static int snd_hdsp_put_da_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_el struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.enumerated.item[0]; @@ -2271,7 +2271,7 @@ static int hdsp_ad_gain(struct hdsp *hdsp) case HDSP_ADGainLowGain: return 2; default: - return 1; + return 1; } } @@ -2283,11 +2283,11 @@ static int hdsp_set_ad_gain(struct hdsp *hdsp, int mode) hdsp->control_register |= HDSP_ADGainMinus10dBV; break; case 1: - hdsp->control_register |= HDSP_ADGainPlus4dBu; + hdsp->control_register |= HDSP_ADGainPlus4dBu; break; case 2: - hdsp->control_register |= HDSP_ADGainLowGain; - break; + hdsp->control_register |= HDSP_ADGainLowGain; + break; default: return -1; @@ -2299,7 +2299,7 @@ static int hdsp_set_ad_gain(struct hdsp *hdsp, int mode) static int snd_hdsp_info_ad_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static char *texts[] = {"-10 dBV", "+4 dBu", "Lo Gain"}; - + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = 3; @@ -2312,7 +2312,7 @@ static int snd_hdsp_info_ad_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_e static int snd_hdsp_get_ad_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_ad_gain(hdsp); return 0; } @@ -2322,7 +2322,7 @@ static int snd_hdsp_put_ad_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_el struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.enumerated.item[0]; @@ -2356,7 +2356,7 @@ static int hdsp_phone_gain(struct hdsp *hdsp) case HDSP_PhoneGainMinus12dB: return 2; default: - return 0; + return 0; } } @@ -2368,11 +2368,11 @@ static int hdsp_set_phone_gain(struct hdsp *hdsp, int mode) hdsp->control_register |= HDSP_PhoneGain0dB; break; case 1: - hdsp->control_register |= HDSP_PhoneGainMinus6dB; + hdsp->control_register |= HDSP_PhoneGainMinus6dB; break; case 2: - hdsp->control_register |= HDSP_PhoneGainMinus12dB; - break; + hdsp->control_register |= HDSP_PhoneGainMinus12dB; + break; default: return -1; @@ -2384,7 +2384,7 @@ static int hdsp_set_phone_gain(struct hdsp *hdsp, int mode) static int snd_hdsp_info_phone_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static char *texts[] = {"0 dB", "-6 dB", "-12 dB"}; - + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = 3; @@ -2397,7 +2397,7 @@ static int snd_hdsp_info_phone_gain(struct snd_kcontrol *kcontrol, struct snd_ct static int snd_hdsp_get_phone_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_phone_gain(hdsp); return 0; } @@ -2407,7 +2407,7 @@ static int snd_hdsp_put_phone_gain(struct snd_kcontrol *kcontrol, struct snd_ctl struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.enumerated.item[0]; @@ -2453,7 +2453,7 @@ static int hdsp_set_xlr_breakout_cable(struct hdsp *hdsp, int mode) static int snd_hdsp_get_xlr_breakout_cable(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_xlr_breakout_cable(hdsp); return 0; } @@ -2463,7 +2463,7 @@ static int snd_hdsp_put_xlr_breakout_cable(struct snd_kcontrol *kcontrol, struct struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -2509,7 +2509,7 @@ static int hdsp_set_aeb(struct hdsp *hdsp, int mode) static int snd_hdsp_get_aeb(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_aeb(hdsp); return 0; } @@ -2519,7 +2519,7 @@ static int snd_hdsp_put_aeb(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_v struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -2597,7 +2597,7 @@ static int snd_hdsp_info_pref_sync_ref(struct snd_kcontrol *kcontrol, struct snd { static char *texts[] = {"Word", "IEC958", "ADAT1", "ADAT Sync", "ADAT2", "ADAT3" }; struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; @@ -2616,7 +2616,7 @@ static int snd_hdsp_info_pref_sync_ref(struct snd_kcontrol *kcontrol, struct snd uinfo->value.enumerated.items = 0; break; } - + if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); @@ -2626,7 +2626,7 @@ static int snd_hdsp_info_pref_sync_ref(struct snd_kcontrol *kcontrol, struct snd static int snd_hdsp_get_pref_sync_ref(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_pref_sync_ref(hdsp); return 0; } @@ -2636,7 +2636,7 @@ static int snd_hdsp_put_pref_sync_ref(struct snd_kcontrol *kcontrol, struct snd_ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change, max; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; @@ -2685,7 +2685,7 @@ static int hdsp_autosync_ref(struct hdsp *hdsp) case HDSP_SelSyncRef_SPDIF: return HDSP_AUTOSYNC_FROM_SPDIF; case HDSP_SelSyncRefMask: - return HDSP_AUTOSYNC_FROM_NONE; + return HDSP_AUTOSYNC_FROM_NONE; case HDSP_SelSyncRef_ADAT1: return HDSP_AUTOSYNC_FROM_ADAT1; case HDSP_SelSyncRef_ADAT2: @@ -2701,7 +2701,7 @@ static int hdsp_autosync_ref(struct hdsp *hdsp) static int snd_hdsp_info_autosync_ref(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static char *texts[] = {"Word", "ADAT Sync", "IEC958", "None", "ADAT1", "ADAT2", "ADAT3" }; - + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = 7; @@ -2714,7 +2714,7 @@ static int snd_hdsp_info_autosync_ref(struct snd_kcontrol *kcontrol, struct snd_ static int snd_hdsp_get_autosync_ref(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_autosync_ref(hdsp); return 0; } @@ -2748,7 +2748,7 @@ static int hdsp_set_line_output(struct hdsp *hdsp, int out) static int snd_hdsp_get_line_out(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + spin_lock_irq(&hdsp->lock); ucontrol->value.integer.value[0] = hdsp_line_out(hdsp); spin_unlock_irq(&hdsp->lock); @@ -2760,7 +2760,7 @@ static int snd_hdsp_put_line_out(struct snd_kcontrol *kcontrol, struct snd_ctl_e struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -2794,7 +2794,7 @@ static int hdsp_set_precise_pointer(struct hdsp *hdsp, int precise) static int snd_hdsp_get_precise_pointer(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + spin_lock_irq(&hdsp->lock); ucontrol->value.integer.value[0] = hdsp->precise_ptr; spin_unlock_irq(&hdsp->lock); @@ -2806,7 +2806,7 @@ static int snd_hdsp_put_precise_pointer(struct snd_kcontrol *kcontrol, struct sn struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -2840,7 +2840,7 @@ static int hdsp_set_use_midi_tasklet(struct hdsp *hdsp, int use_tasklet) static int snd_hdsp_get_use_midi_tasklet(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + spin_lock_irq(&hdsp->lock); ucontrol->value.integer.value[0] = hdsp->use_midi_tasklet; spin_unlock_irq(&hdsp->lock); @@ -2852,7 +2852,7 @@ static int snd_hdsp_put_use_midi_tasklet(struct snd_kcontrol *kcontrol, struct s struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -2894,12 +2894,12 @@ static int snd_hdsp_get_mixer(struct snd_kcontrol *kcontrol, struct snd_ctl_elem source = ucontrol->value.integer.value[0]; destination = ucontrol->value.integer.value[1]; - + if (source >= hdsp->max_channels) addr = hdsp_playback_to_output_key(hdsp,source-hdsp->max_channels,destination); else addr = hdsp_input_to_output_key(hdsp,source, destination); - + spin_lock_irq(&hdsp->lock); ucontrol->value.integer.value[2] = hdsp_read_gain (hdsp, addr); spin_unlock_irq(&hdsp->lock); @@ -2947,7 +2947,7 @@ static int snd_hdsp_put_mixer(struct snd_kcontrol *kcontrol, struct snd_ctl_elem static int snd_hdsp_info_sync_check(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = {"No Lock", "Lock", "Sync" }; + static char *texts[] = {"No Lock", "Lock", "Sync" }; uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = 3; @@ -2992,7 +2992,7 @@ static int hdsp_spdif_sync_check(struct hdsp *hdsp) int status = hdsp_read(hdsp, HDSP_statusRegister); if (status & HDSP_SPDIFErrorFlag) return 0; - else { + else { if (status & HDSP_SPDIFSync) return 2; else @@ -3028,7 +3028,7 @@ static int hdsp_adatsync_sync_check(struct hdsp *hdsp) return 1; } else return 0; -} +} static int snd_hdsp_get_adatsync_sync_check(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -3046,17 +3046,17 @@ static int snd_hdsp_get_adatsync_sync_check(struct snd_kcontrol *kcontrol, struc } static int hdsp_adat_sync_check(struct hdsp *hdsp, int idx) -{ +{ int status = hdsp_read(hdsp, HDSP_statusRegister); - + if (status & (HDSP_Lock0>>idx)) { if (status & (HDSP_Sync0>>idx)) return 2; else - return 1; + return 1; } else return 0; -} +} static int snd_hdsp_get_adat_sync_check(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -3074,7 +3074,7 @@ static int snd_hdsp_get_adat_sync_check(struct snd_kcontrol *kcontrol, struct sn break; case Multiface: case H9632: - if (offset >= 1) + if (offset >= 1) return -EINVAL; break; default: @@ -3136,7 +3136,7 @@ static int snd_hdsp_info_dds_offset(struct snd_kcontrol *kcontrol, struct snd_ct static int snd_hdsp_get_dds_offset(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_dds_offset(hdsp); return 0; } @@ -3146,7 +3146,7 @@ static int snd_hdsp_put_dds_offset(struct snd_kcontrol *kcontrol, struct snd_ctl struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.enumerated.item[0]; @@ -3191,7 +3191,7 @@ static struct snd_kcontrol_new snd_hdsp_controls[] = { .get = snd_hdsp_control_spdif_mask_get, .private_value = IEC958_AES0_NONAUDIO | IEC958_AES0_PROFESSIONAL | - IEC958_AES0_CON_EMPHASIS, + IEC958_AES0_CON_EMPHASIS, }, { .access = SNDRV_CTL_ELEM_ACCESS_READ, @@ -3209,7 +3209,7 @@ HDSP_SPDIF_OUT("IEC958 Output also on ADAT1", 0), HDSP_SPDIF_PROFESSIONAL("IEC958 Professional Bit", 0), HDSP_SPDIF_EMPHASIS("IEC958 Emphasis Bit", 0), HDSP_SPDIF_NON_AUDIO("IEC958 Non-audio Bit", 0), -/* 'Sample Clock Source' complies with the alsa control naming scheme */ +/* 'Sample Clock Source' complies with the alsa control naming scheme */ HDSP_CLOCK_SOURCE("Sample Clock Source", 0), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -3261,7 +3261,7 @@ static int snd_hdsp_create_controls(struct snd_card *card, struct hdsp *hdsp) return err; } } - + /* DA, AD and Phone gain and XLR breakout cable controls for H9632 cards */ if (hdsp->io_type == H9632) { for (idx = 0; idx < ARRAY_SIZE(snd_hdsp_9632_controls); idx++) { @@ -3280,7 +3280,7 @@ static int snd_hdsp_create_controls(struct snd_card *card, struct hdsp *hdsp) } /*------------------------------------------------------------ - /proc interface + /proc interface ------------------------------------------------------------*/ static void @@ -3319,7 +3319,7 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) } } } - + status = hdsp_read(hdsp, HDSP_statusRegister); status2 = hdsp_read(hdsp, HDSP_status2Register); @@ -3383,17 +3383,17 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) break; case HDSP_CLOCK_SOURCE_INTERNAL_192KHZ: clock_source = "Internal 192 kHz"; - break; + break; default: - clock_source = "Error"; + clock_source = "Error"; } snd_iprintf (buffer, "Sample Clock Source: %s\n", clock_source); - + if (hdsp_system_clock_mode(hdsp)) system_clock_mode = "Slave"; else system_clock_mode = "Master"; - + switch (hdsp_pref_sync_ref (hdsp)) { case HDSP_SYNC_FROM_WORD: pref_sync_ref = "Word Clock"; @@ -3418,7 +3418,7 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) break; } snd_iprintf (buffer, "Preferred Sync Reference: %s\n", pref_sync_ref); - + switch (hdsp_autosync_ref (hdsp)) { case HDSP_AUTOSYNC_FROM_WORD: autosync_ref = "Word Clock"; @@ -3431,7 +3431,7 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) break; case HDSP_AUTOSYNC_FROM_NONE: autosync_ref = "None"; - break; + break; case HDSP_AUTOSYNC_FROM_ADAT1: autosync_ref = "ADAT1"; break; @@ -3446,14 +3446,14 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) break; } snd_iprintf (buffer, "AutoSync Reference: %s\n", autosync_ref); - + snd_iprintf (buffer, "AutoSync Frequency: %d\n", hdsp_external_sample_rate(hdsp)); - + snd_iprintf (buffer, "System Clock Mode: %s\n", system_clock_mode); snd_iprintf (buffer, "System Clock Frequency: %d\n", hdsp->system_sample_rate); snd_iprintf (buffer, "System Clock Locked: %s\n", hdsp->clock_source_locked ? "Yes" : "No"); - + snd_iprintf(buffer, "\n"); switch (hdsp_spdif_in(hdsp)) { @@ -3473,7 +3473,7 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) snd_iprintf(buffer, "IEC958 input: ???\n"); break; } - + if (hdsp->control_register & HDSP_SPDIFOpticalOut) snd_iprintf(buffer, "IEC958 output: Coaxial & ADAT1\n"); else @@ -3531,13 +3531,13 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) snd_iprintf (buffer, "SPDIF: No Lock\n"); else snd_iprintf (buffer, "SPDIF: %s\n", x ? "Sync" : "Lock"); - + x = status2 & HDSP_wc_sync; if (status2 & HDSP_wc_lock) snd_iprintf (buffer, "Word Clock: %s\n", x ? "Sync" : "Lock"); else snd_iprintf (buffer, "Word Clock: No Lock\n"); - + x = status & HDSP_TimecodeSync; if (status & HDSP_TimecodeLock) snd_iprintf(buffer, "ADAT Sync: %s\n", x ? "Sync" : "Lock"); @@ -3545,11 +3545,11 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) snd_iprintf(buffer, "ADAT Sync: No Lock\n"); snd_iprintf(buffer, "\n"); - + /* Informations about H9632 specific controls */ if (hdsp->io_type == H9632) { char *tmp; - + switch (hdsp_ad_gain(hdsp)) { case 0: tmp = "-10 dBV"; @@ -3575,7 +3575,7 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) break; } snd_iprintf(buffer, "DA Gain : %s\n", tmp); - + switch (hdsp_phone_gain(hdsp)) { case 0: tmp = "0 dB"; @@ -3589,8 +3589,8 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) } snd_iprintf(buffer, "Phones Gain : %s\n", tmp); - snd_iprintf(buffer, "XLR Breakout Cable : %s\n", hdsp_xlr_breakout_cable(hdsp) ? "yes" : "no"); - + snd_iprintf(buffer, "XLR Breakout Cable : %s\n", hdsp_xlr_breakout_cable(hdsp) ? "yes" : "no"); + if (hdsp->control_register & HDSP_AnalogExtensionBoard) snd_iprintf(buffer, "AEB : on (ADAT1 internal)\n"); else @@ -3653,18 +3653,18 @@ static int snd_hdsp_set_defaults(struct hdsp *hdsp) /* set defaults: - SPDIF Input via Coax + SPDIF Input via Coax Master clock mode maximum latency (7 => 2^7 = 8192 samples, 64Kbyte buffer, which implies 2 4096 sample, 32Kbyte periods). - Enable line out. + Enable line out. */ - hdsp->control_register = HDSP_ClockModeMaster | - HDSP_SPDIFInputCoaxial | - hdsp_encode_latency(7) | + hdsp->control_register = HDSP_ClockModeMaster | + HDSP_SPDIFInputCoaxial | + hdsp_encode_latency(7) | HDSP_LineOut; - + hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register); @@ -3682,7 +3682,7 @@ static int snd_hdsp_set_defaults(struct hdsp *hdsp) hdsp_compute_period_size(hdsp); /* silence everything */ - + for (i = 0; i < HDSP_MATRIX_MIXER_SIZE; ++i) hdsp->mixer_matrix[i] = MINUS_INFINITY_GAIN; @@ -3690,7 +3690,7 @@ static int snd_hdsp_set_defaults(struct hdsp *hdsp) if (hdsp_write_gain (hdsp, i, MINUS_INFINITY_GAIN)) return -EIO; } - + /* H9632 specific defaults */ if (hdsp->io_type == H9632) { hdsp->control_register |= (HDSP_DAGainPlus4dBu | HDSP_ADGainPlus4dBu | HDSP_PhoneGain0dB); @@ -3708,12 +3708,12 @@ static int snd_hdsp_set_defaults(struct hdsp *hdsp) static void hdsp_midi_tasklet(unsigned long arg) { struct hdsp *hdsp = (struct hdsp *)arg; - + if (hdsp->midi[0].pending) snd_hdsp_midi_input_read (&hdsp->midi[0]); if (hdsp->midi[1].pending) snd_hdsp_midi_input_read (&hdsp->midi[1]); -} +} static irqreturn_t snd_hdsp_interrupt(int irq, void *dev_id) { @@ -3725,7 +3725,7 @@ static irqreturn_t snd_hdsp_interrupt(int irq, void *dev_id) unsigned int midi0status; unsigned int midi1status; int schedule = 0; - + status = hdsp_read(hdsp, HDSP_statusRegister); audio = status & HDSP_audioIRQPending; @@ -3739,15 +3739,15 @@ static irqreturn_t snd_hdsp_interrupt(int irq, void *dev_id) midi0status = hdsp_read (hdsp, HDSP_midiStatusIn0) & 0xff; midi1status = hdsp_read (hdsp, HDSP_midiStatusIn1) & 0xff; - + if (audio) { if (hdsp->capture_substream) snd_pcm_period_elapsed(hdsp->pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream); - + if (hdsp->playback_substream) snd_pcm_period_elapsed(hdsp->pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream); } - + if (midi0 && midi0status) { if (hdsp->use_midi_tasklet) { /* we disable interrupts for this input until processing is done */ @@ -3790,10 +3790,10 @@ static char *hdsp_channel_buffer_location(struct hdsp *hdsp, if (snd_BUG_ON(channel < 0 || channel >= hdsp->max_channels)) return NULL; - + if ((mapped_channel = hdsp->channel_map[channel]) < 0) return NULL; - + if (stream == SNDRV_PCM_STREAM_CAPTURE) return hdsp->capture_buffer + (mapped_channel * HDSP_CHANNEL_BUFFER_BYTES); else @@ -3986,7 +3986,7 @@ static int snd_hdsp_trigger(struct snd_pcm_substream *substream, int cmd) struct hdsp *hdsp = snd_pcm_substream_chip(substream); struct snd_pcm_substream *other; int running; - + if (hdsp_check_for_iobox (hdsp)) return -EIO; @@ -4080,10 +4080,10 @@ static struct snd_pcm_hardware snd_hdsp_playback_subinfo = .formats = SNDRV_PCM_FMTBIT_S32_LE, #endif .rates = (SNDRV_PCM_RATE_32000 | - SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_64000 | - SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_64000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000), .rate_min = 32000, .rate_max = 96000, @@ -4109,10 +4109,10 @@ static struct snd_pcm_hardware snd_hdsp_capture_subinfo = .formats = SNDRV_PCM_FMTBIT_S32_LE, #endif .rates = (SNDRV_PCM_RATE_32000 | - SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_64000 | - SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_64000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000), .rate_min = 32000, .rate_max = 96000, @@ -4191,7 +4191,7 @@ static int snd_hdsp_hw_rule_in_channels_rate(struct snd_pcm_hw_params *params, .max = hdsp->qs_in_channels, .integer = 1, }; - return snd_interval_refine(c, &t); + return snd_interval_refine(c, &t); } else if (r->min > 48000 && r->max <= 96000) { struct snd_interval t = { .min = hdsp->ds_in_channels, @@ -4222,7 +4222,7 @@ static int snd_hdsp_hw_rule_out_channels_rate(struct snd_pcm_hw_params *params, .max = hdsp->qs_out_channels, .integer = 1, }; - return snd_interval_refine(c, &t); + return snd_interval_refine(c, &t); } else if (r->min > 48000 && r->max <= 96000) { struct snd_interval t = { .min = hdsp->ds_out_channels, @@ -4339,8 +4339,8 @@ static int snd_hdsp_playback_open(struct snd_pcm_substream *substream) if (hdsp->io_type == H9632) { runtime->hw.channels_min = hdsp->qs_out_channels; runtime->hw.channels_max = hdsp->ss_out_channels; - } - + } + snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, snd_hdsp_hw_rule_out_channels, hdsp, SNDRV_PCM_HW_PARAM_CHANNELS, -1); @@ -4550,7 +4550,7 @@ static int hdsp_get_peak(struct hdsp *hdsp, struct hdsp_peak_rms __user *peak_rm hdsp->iobase + HDSP_playbackRmsLevel + i * 8 + 4, hdsp->iobase + HDSP_playbackRmsLevel + i * 8)) return -EFAULT; - if (copy_u64_le(&peak_rms->input_rms[i], + if (copy_u64_le(&peak_rms->input_rms[i], hdsp->iobase + HDSP_inputRmsLevel + i * 8 + 4, hdsp->iobase + HDSP_inputRmsLevel + i * 8)) return -EFAULT; @@ -4560,7 +4560,7 @@ static int hdsp_get_peak(struct hdsp *hdsp, struct hdsp_peak_rms __user *peak_rm static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigned int cmd, unsigned long arg) { - struct hdsp *hdsp = (struct hdsp *)hw->private_data; + struct hdsp *hdsp = (struct hdsp *)hw->private_data; void __user *argp = (void __user *)arg; int err; @@ -4594,7 +4594,7 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne struct hdsp_config_info info; unsigned long flags; int i; - + err = hdsp_check_for_iobox(hdsp); if (err < 0) return err; @@ -4628,7 +4628,7 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne info.ad_gain = (unsigned char)hdsp_ad_gain(hdsp); info.phone_gain = (unsigned char)hdsp_phone_gain(hdsp); info.xlr_breakout_cable = (unsigned char)hdsp_xlr_breakout_cable(hdsp); - + } if (hdsp->io_type == H9632 || hdsp->io_type == H9652) info.analog_extension_board = (unsigned char)hdsp_aeb(hdsp); @@ -4639,7 +4639,7 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne } case SNDRV_HDSP_IOCTL_GET_9632_AEB: { struct hdsp_9632_aeb h9632_aeb; - + if (hdsp->io_type != H9632) return -EINVAL; h9632_aeb.aebi = hdsp->ss_in_channels - H9632_SS_CHANNELS; h9632_aeb.aebo = hdsp->ss_out_channels - H9632_SS_CHANNELS; @@ -4650,7 +4650,7 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne case SNDRV_HDSP_IOCTL_GET_VERSION: { struct hdsp_version hdsp_version; int err; - + if (hdsp->io_type == H9652 || hdsp->io_type == H9632) return -EINVAL; if (hdsp->io_type == Undefined) { if ((err = hdsp_get_iobox_version(hdsp)) < 0) @@ -4666,7 +4666,7 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne struct hdsp_firmware __user *firmware; u32 __user *firmware_data; int err; - + if (hdsp->io_type == H9652 || hdsp->io_type == H9632) return -EINVAL; /* SNDRV_HDSP_IOCTL_GET_VERSION must have been called */ if (hdsp->io_type == Undefined) return -EINVAL; @@ -4679,25 +4679,25 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne if (get_user(firmware_data, &firmware->firmware_data)) return -EFAULT; - + if (hdsp_check_for_iobox (hdsp)) return -EIO; if (copy_from_user(hdsp->firmware_cache, firmware_data, sizeof(hdsp->firmware_cache)) != 0) return -EFAULT; - + hdsp->state |= HDSP_FirmwareCached; if ((err = snd_hdsp_load_firmware_from_cache(hdsp)) < 0) return err; - + if (!(hdsp->state & HDSP_InitializationComplete)) { if ((err = snd_hdsp_enable_io(hdsp)) < 0) return err; - - snd_hdsp_initialize_channels(hdsp); + + snd_hdsp_initialize_channels(hdsp); snd_hdsp_initialize_midi_flush(hdsp); - + if ((err = snd_hdsp_create_alsa_devices(hdsp->card, hdsp)) < 0) { snd_printk(KERN_ERR "Hammerfall-DSP: error creating alsa devices\n"); return err; @@ -4744,16 +4744,16 @@ static int snd_hdsp_create_hwdep(struct snd_card *card, struct hdsp *hdsp) { struct snd_hwdep *hw; int err; - + if ((err = snd_hwdep_new(card, "HDSP hwdep", 0, &hw)) < 0) return err; - + hdsp->hwdep = hw; hw->private_data = hdsp; strcpy(hw->name, "HDSP hwdep interface"); hw->ops.ioctl = snd_hdsp_hwdep_ioctl; - + return 0; } @@ -4786,24 +4786,24 @@ static void snd_hdsp_9652_enable_mixer (struct hdsp *hdsp) static int snd_hdsp_enable_io (struct hdsp *hdsp) { int i; - + if (hdsp_fifo_wait (hdsp, 0, 100)) { snd_printk(KERN_ERR "Hammerfall-DSP: enable_io fifo_wait failed\n"); return -EIO; } - + for (i = 0; i < hdsp->max_channels; ++i) { hdsp_write (hdsp, HDSP_inputEnable + (4 * i), 1); hdsp_write (hdsp, HDSP_outputEnable + (4 * i), 1); } - + return 0; } static void snd_hdsp_initialize_channels(struct hdsp *hdsp) { int status, aebi_channels, aebo_channels; - + switch (hdsp->io_type) { case Digiface: hdsp->card_name = "RME Hammerfall DSP + Digiface"; @@ -4816,7 +4816,7 @@ static void snd_hdsp_initialize_channels(struct hdsp *hdsp) hdsp->ss_in_channels = hdsp->ss_out_channels = H9652_SS_CHANNELS; hdsp->ds_in_channels = hdsp->ds_out_channels = H9652_DS_CHANNELS; break; - + case H9632: status = hdsp_read(hdsp, HDSP_statusRegister); /* HDSP_AEBx bits are low when AEB are connected */ @@ -4836,7 +4836,7 @@ static void snd_hdsp_initialize_channels(struct hdsp *hdsp) hdsp->ss_in_channels = hdsp->ss_out_channels = MULTIFACE_SS_CHANNELS; hdsp->ds_in_channels = hdsp->ds_out_channels = MULTIFACE_DS_CHANNELS; break; - + default: /* should never get here */ break; @@ -4852,12 +4852,12 @@ static void snd_hdsp_initialize_midi_flush (struct hdsp *hdsp) static int snd_hdsp_create_alsa_devices(struct snd_card *card, struct hdsp *hdsp) { int err; - + if ((err = snd_hdsp_create_pcm(card, hdsp)) < 0) { snd_printk(KERN_ERR "Hammerfall-DSP: Error creating pcm interface\n"); return err; } - + if ((err = snd_hdsp_create_midi(card, hdsp, 0)) < 0) { snd_printk(KERN_ERR "Hammerfall-DSP: Error creating first midi interface\n"); @@ -4888,19 +4888,19 @@ static int snd_hdsp_create_alsa_devices(struct snd_card *card, struct hdsp *hdsp snd_printk(KERN_ERR "Hammerfall-DSP: Error setting default values\n"); return err; } - + if (!(hdsp->state & HDSP_InitializationComplete)) { strcpy(card->shortname, "Hammerfall DSP"); - sprintf(card->longname, "%s at 0x%lx, irq %d", hdsp->card_name, + sprintf(card->longname, "%s at 0x%lx, irq %d", hdsp->card_name, hdsp->port, hdsp->irq); - + if ((err = snd_card_register(card)) < 0) { snd_printk(KERN_ERR "Hammerfall-DSP: error registering card\n"); return err; } hdsp->state |= HDSP_InitializationComplete; } - + return 0; } @@ -4911,7 +4911,7 @@ static int hdsp_request_fw_loader(struct hdsp *hdsp) const char *fwfile; const struct firmware *fw; int err; - + if (hdsp->io_type == H9652 || hdsp->io_type == H9632) return 0; if (hdsp->io_type == Undefined) { @@ -4920,7 +4920,7 @@ static int hdsp_request_fw_loader(struct hdsp *hdsp) if (hdsp->io_type == H9652 || hdsp->io_type == H9632) return 0; } - + /* caution: max length of firmware filename is 30! */ switch (hdsp->io_type) { case Multiface: @@ -4954,12 +4954,12 @@ static int hdsp_request_fw_loader(struct hdsp *hdsp) memcpy(hdsp->firmware_cache, fw->data, sizeof(hdsp->firmware_cache)); release_firmware(fw); - + hdsp->state |= HDSP_FirmwareCached; if ((err = snd_hdsp_load_firmware_from_cache(hdsp)) < 0) return err; - + if (!(hdsp->state & HDSP_InitializationComplete)) { if ((err = snd_hdsp_enable_io(hdsp)) < 0) return err; @@ -5006,14 +5006,14 @@ static int __devinit snd_hdsp_create(struct snd_card *card, hdsp->max_channels = 26; hdsp->card = card; - + spin_lock_init(&hdsp->lock); tasklet_init(&hdsp->midi_tasklet, hdsp_midi_tasklet, (unsigned long)hdsp); - + pci_read_config_word(hdsp->pci, PCI_CLASS_REVISION, &hdsp->firmware_rev); hdsp->firmware_rev &= 0xff; - + /* From Martin Bjoernsen : "It is important that the card's latency timer register in the PCI configuration space is set to a value much larger @@ -5022,7 +5022,7 @@ static int __devinit snd_hdsp_create(struct snd_card *card, to its maximum 255 to avoid problems with some computers." */ pci_write_config_byte(hdsp->pci, PCI_LATENCY_TIMER, 0xFF); - + strcpy(card->driver, "H-DSP"); strcpy(card->mixername, "Xilinx FPGA"); @@ -5036,7 +5036,7 @@ static int __devinit snd_hdsp_create(struct snd_card *card, } else { hdsp->card_name = "RME HDSP 9632"; hdsp->max_channels = 16; - is_9632 = 1; + is_9632 = 1; } if ((err = pci_enable_device(pci)) < 0) @@ -5065,7 +5065,7 @@ static int __devinit snd_hdsp_create(struct snd_card *card, if ((err = snd_hdsp_initialize_memory(hdsp)) < 0) return err; - + if (!is_9652 && !is_9632) { /* we wait a maximum of 10 seconds to let freshly * inserted cardbus cards do their hardware init */ @@ -5092,35 +5092,35 @@ static int __devinit snd_hdsp_create(struct snd_card *card, return err; return 0; } else { - snd_printk(KERN_INFO "Hammerfall-DSP: Firmware already present, initializing card.\n"); + snd_printk(KERN_INFO "Hammerfall-DSP: Firmware already present, initializing card.\n"); if (hdsp_read(hdsp, HDSP_status2Register) & HDSP_version1) hdsp->io_type = Multiface; - else + else hdsp->io_type = Digiface; } } - + if ((err = snd_hdsp_enable_io(hdsp)) != 0) return err; - + if (is_9652) hdsp->io_type = H9652; - + if (is_9632) hdsp->io_type = H9632; if ((err = snd_hdsp_create_hwdep(card, hdsp)) < 0) return err; - + snd_hdsp_initialize_channels(hdsp); snd_hdsp_initialize_midi_flush(hdsp); - hdsp->state |= HDSP_FirmwareLoaded; + hdsp->state |= HDSP_FirmwareLoaded; if ((err = snd_hdsp_create_alsa_devices(card, hdsp)) < 0) return err; - return 0; + return 0; } static int snd_hdsp_free(struct hdsp *hdsp) @@ -5136,13 +5136,13 @@ static int snd_hdsp_free(struct hdsp *hdsp) free_irq(hdsp->irq, (void *)hdsp); snd_hdsp_free_buffers(hdsp); - + if (hdsp->iobase) iounmap(hdsp->iobase); if (hdsp->port) pci_release_regions(hdsp->pci); - + pci_disable_device(hdsp->pci); return 0; } @@ -5187,7 +5187,7 @@ static int __devinit snd_hdsp_probe(struct pci_dev *pci, } strcpy(card->shortname, "Hammerfall DSP"); - sprintf(card->longname, "%s at 0x%lx, irq %d", hdsp->card_name, + sprintf(card->longname, "%s at 0x%lx, irq %d", hdsp->card_name, hdsp->port, hdsp->irq); if ((err = snd_card_register(card)) < 0) { -- cgit v1.2.3-70-g09d2 From c17a1abae2f29047a0f57324240b01609489261b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 23 Feb 2009 09:28:12 +0100 Subject: ALSA: hda - Use snd_hda_codec_get_pincfg() in the rest places Replace with snd_hda_codec_get_pincfg() in the places where available. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 3 +-- sound/pci/hda/hda_generic.c | 2 +- 2 files changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 6fa871f66a7..8ec2dfca9a6 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3488,8 +3488,7 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, if (ignore_nids && is_in_nid_list(nid, ignore_nids)) continue; - def_conf = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONFIG_DEFAULT, 0); + def_conf = snd_hda_codec_get_pincfg(codec, nid); if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE) continue; loc = get_defcfg_location(def_conf); diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 65745e96dc7..2c81a683e8f 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -146,7 +146,7 @@ static int add_new_node(struct hda_codec *codec, struct hda_gspec *spec, hda_nid if (node->type == AC_WID_PIN) { node->pin_caps = snd_hda_param_read(codec, node->nid, AC_PAR_PIN_CAP); node->pin_ctl = snd_hda_codec_read(codec, node->nid, 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); - node->def_cfg = snd_hda_codec_read(codec, node->nid, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); + node->def_cfg = snd_hda_codec_get_pincfg(codec, node->nid); } if (node->wid_caps & AC_WCAP_OUT_AMP) { -- cgit v1.2.3-70-g09d2 From 346ff70fdbe9093947b9494fe714c89cafcceade Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 23 Feb 2009 09:42:57 +0100 Subject: ALSA: hda - Rename {override,cur}_pin with {user,driver}_pin Rename from override_pin and cur_pin with user_pin and driver_pin, respectively, to be a bit more intuitive. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio.txt | 12 ++++++------ sound/pci/hda/hda_codec.c | 18 +++++++++--------- sound/pci/hda/hda_codec.h | 4 ++-- sound/pci/hda/hda_hwdep.c | 32 ++++++++++++++++---------------- 4 files changed, 33 insertions(+), 33 deletions(-) diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt index 9c51e104546..f590850c149 100644 --- a/Documentation/sound/alsa/HD-Audio.txt +++ b/Documentation/sound/alsa/HD-Audio.txt @@ -371,16 +371,16 @@ hints:: not used. init_pin_configs:: Shows the initial pin default config values set by BIOS. -override_pin_configs:: - Shows the pin default config values to override the BIOS setup. - Writing this (with two numbers, NID and value) appends the new - value. The given will be used instead of the initial BIOS value at - the next reconfiguration time. -cur_pin_configs:: +driver_pin_configs:: Shows the pin default values set by the codec parser explicitly. This doesn't show all pin values but only the changed values by the parser. That is, if the parser doesn't change the pin default config values by itself, this will contain nothing. +user_pin_configs:: + Shows the pin default config values to override the BIOS setup. + Writing this (with two numbers, NID and value) appends the new + value. The given will be used instead of the initial BIOS value at + the next reconfiguration time. reconfig:: Triggers the codec re-configuration. When any value is written to this file, the driver re-initialize and parses the codec tree diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 8ec2dfca9a6..df9453d0122 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -755,7 +755,7 @@ int snd_hda_add_pincfg(struct hda_codec *codec, struct snd_array *list, int snd_hda_codec_set_pincfg(struct hda_codec *codec, hda_nid_t nid, unsigned int cfg) { - return snd_hda_add_pincfg(codec, &codec->cur_pins, nid, cfg); + return snd_hda_add_pincfg(codec, &codec->driver_pins, nid, cfg); } EXPORT_SYMBOL_HDA(snd_hda_codec_set_pincfg); @@ -764,11 +764,11 @@ unsigned int snd_hda_codec_get_pincfg(struct hda_codec *codec, hda_nid_t nid) { struct hda_pincfg *pin; - pin = look_up_pincfg(codec, &codec->cur_pins, nid); + pin = look_up_pincfg(codec, &codec->driver_pins, nid); if (pin) return pin->cfg; #ifdef CONFIG_SND_HDA_HWDEP - pin = look_up_pincfg(codec, &codec->override_pins, nid); + pin = look_up_pincfg(codec, &codec->user_pins, nid); if (pin) return pin->cfg; #endif @@ -797,12 +797,12 @@ static void free_hda_cache(struct hda_cache_rec *cache); /* restore the initial pin cfgs and release all pincfg lists */ static void restore_init_pincfgs(struct hda_codec *codec) { - /* first free cur_pins and override_pins, then call restore_pincfg + /* first free driver_pins and user_pins, then call restore_pincfg * so that only the values in init_pins are restored */ - snd_array_free(&codec->cur_pins); + snd_array_free(&codec->driver_pins); #ifdef CONFIG_SND_HDA_HWDEP - snd_array_free(&codec->override_pins); + snd_array_free(&codec->user_pins); #endif restore_pincfgs(codec); snd_array_free(&codec->init_pins); @@ -874,7 +874,7 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr init_hda_cache(&codec->cmd_cache, sizeof(struct hda_cache_head)); snd_array_init(&codec->mixers, sizeof(struct snd_kcontrol *), 32); snd_array_init(&codec->init_pins, sizeof(struct hda_pincfg), 16); - snd_array_init(&codec->cur_pins, sizeof(struct hda_pincfg), 16); + snd_array_init(&codec->driver_pins, sizeof(struct hda_pincfg), 16); if (codec->bus->modelname) { codec->modelname = kstrdup(codec->bus->modelname, GFP_KERNEL); if (!codec->modelname) { @@ -1463,8 +1463,8 @@ void snd_hda_codec_reset(struct hda_codec *codec) free_hda_cache(&codec->cmd_cache); init_hda_cache(&codec->amp_cache, sizeof(struct hda_amp_info)); init_hda_cache(&codec->cmd_cache, sizeof(struct hda_cache_head)); - /* free only cur_pins so that init_pins + override_pins are restored */ - snd_array_free(&codec->cur_pins); + /* free only driver_pins so that init_pins + user_pins are restored */ + snd_array_free(&codec->driver_pins); restore_pincfgs(codec); codec->num_pcms = 0; codec->pcm_info = NULL; diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 6d01a8058f0..2ea628478a9 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -779,13 +779,13 @@ struct hda_codec { unsigned int spdif_in_enable; /* SPDIF input enable? */ hda_nid_t *slave_dig_outs; /* optional digital out slave widgets */ struct snd_array init_pins; /* initial (BIOS) pin configurations */ - struct snd_array cur_pins; /* current pin configurations */ + struct snd_array driver_pins; /* pin configs set by codec parser */ #ifdef CONFIG_SND_HDA_HWDEP struct snd_hwdep *hwdep; /* assigned hwdep device */ struct snd_array init_verbs; /* additional init verbs */ struct snd_array hints; /* additional hints */ - struct snd_array override_pins; /* default pin configs to override */ + struct snd_array user_pins; /* default pin configs to override */ #endif /* misc flags */ diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index 71039a6dec2..c660383ef38 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -109,7 +109,7 @@ static void clear_hwdep_elements(struct hda_codec *codec) for (i = 0; i < codec->hints.used; i++, head++) kfree(*head); snd_array_free(&codec->hints); - snd_array_free(&codec->override_pins); + snd_array_free(&codec->user_pins); } static void hwdep_free(struct snd_hwdep *hwdep) @@ -142,7 +142,7 @@ int /*__devinit*/ snd_hda_create_hwdep(struct hda_codec *codec) snd_array_init(&codec->init_verbs, sizeof(struct hda_verb), 32); snd_array_init(&codec->hints, sizeof(char *), 32); - snd_array_init(&codec->override_pins, sizeof(struct hda_pincfg), 16); + snd_array_init(&codec->user_pins, sizeof(struct hda_pincfg), 16); return 0; } @@ -340,29 +340,29 @@ static ssize_t init_pin_configs_show(struct device *dev, return pin_configs_show(codec, &codec->init_pins, buf); } -static ssize_t override_pin_configs_show(struct device *dev, - struct device_attribute *attr, - char *buf) +static ssize_t user_pin_configs_show(struct device *dev, + struct device_attribute *attr, + char *buf) { struct snd_hwdep *hwdep = dev_get_drvdata(dev); struct hda_codec *codec = hwdep->private_data; - return pin_configs_show(codec, &codec->override_pins, buf); + return pin_configs_show(codec, &codec->user_pins, buf); } -static ssize_t cur_pin_configs_show(struct device *dev, - struct device_attribute *attr, - char *buf) +static ssize_t driver_pin_configs_show(struct device *dev, + struct device_attribute *attr, + char *buf) { struct snd_hwdep *hwdep = dev_get_drvdata(dev); struct hda_codec *codec = hwdep->private_data; - return pin_configs_show(codec, &codec->cur_pins, buf); + return pin_configs_show(codec, &codec->driver_pins, buf); } #define MAX_PIN_CONFIGS 32 -static ssize_t override_pin_configs_store(struct device *dev, - struct device_attribute *attr, - const char *buf, size_t count) +static ssize_t user_pin_configs_store(struct device *dev, + struct device_attribute *attr, + const char *buf, size_t count) { struct snd_hwdep *hwdep = dev_get_drvdata(dev); struct hda_codec *codec = hwdep->private_data; @@ -373,7 +373,7 @@ static ssize_t override_pin_configs_store(struct device *dev, return -EINVAL; if (!nid) return -EINVAL; - err = snd_hda_add_pincfg(codec, &codec->override_pins, nid, cfg); + err = snd_hda_add_pincfg(codec, &codec->user_pins, nid, cfg); if (err < 0) return err; return count; @@ -397,8 +397,8 @@ static struct device_attribute codec_attrs[] = { CODEC_ATTR_WO(init_verbs), CODEC_ATTR_WO(hints), CODEC_ATTR_RO(init_pin_configs), - CODEC_ATTR_RW(override_pin_configs), - CODEC_ATTR_RO(cur_pin_configs), + CODEC_ATTR_RW(user_pin_configs), + CODEC_ATTR_RO(driver_pin_configs), CODEC_ATTR_WO(reconfig), CODEC_ATTR_WO(clear), }; -- cgit v1.2.3-70-g09d2 From 5e7b8e0d87091ae21b291588817b5359a5e00795 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 23 Feb 2009 09:45:59 +0100 Subject: ALSA: hda - Make user_pin overriding the driver setup Make user_pin overriding even the driver pincfg, e.g. the static / fixed pin config table in patch_sigmatel.c. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio.txt | 3 ++- sound/pci/hda/hda_codec.c | 16 ++++++++++++---- 2 files changed, 14 insertions(+), 5 deletions(-) diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt index f590850c149..a4e5ef87af6 100644 --- a/Documentation/sound/alsa/HD-Audio.txt +++ b/Documentation/sound/alsa/HD-Audio.txt @@ -380,7 +380,8 @@ user_pin_configs:: Shows the pin default config values to override the BIOS setup. Writing this (with two numbers, NID and value) appends the new value. The given will be used instead of the initial BIOS value at - the next reconfiguration time. + the next reconfiguration time. Note that this config will override + even the driver pin configs, too. reconfig:: Triggers the codec re-configuration. When any value is written to this file, the driver re-initialize and parses the codec tree diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index df9453d0122..a13480fa8e7 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -739,7 +739,9 @@ int snd_hda_add_pincfg(struct hda_codec *codec, struct snd_array *list, hda_nid_t nid, unsigned int cfg) { struct hda_pincfg *pin; + unsigned int oldcfg; + oldcfg = snd_hda_codec_get_pincfg(codec, nid); pin = look_up_pincfg(codec, list, nid); if (!pin) { pin = snd_array_new(list); @@ -748,7 +750,13 @@ int snd_hda_add_pincfg(struct hda_codec *codec, struct snd_array *list, pin->nid = nid; } pin->cfg = cfg; - set_pincfg(codec, nid, cfg); + + /* change only when needed; e.g. if the pincfg is already present + * in user_pins[], don't write it + */ + cfg = snd_hda_codec_get_pincfg(codec, nid); + if (oldcfg != cfg) + set_pincfg(codec, nid, cfg); return 0; } @@ -764,14 +772,14 @@ unsigned int snd_hda_codec_get_pincfg(struct hda_codec *codec, hda_nid_t nid) { struct hda_pincfg *pin; - pin = look_up_pincfg(codec, &codec->driver_pins, nid); - if (pin) - return pin->cfg; #ifdef CONFIG_SND_HDA_HWDEP pin = look_up_pincfg(codec, &codec->user_pins, nid); if (pin) return pin->cfg; #endif + pin = look_up_pincfg(codec, &codec->driver_pins, nid); + if (pin) + return pin->cfg; pin = look_up_pincfg(codec, &codec->init_pins, nid); if (pin) return pin->cfg; -- cgit v1.2.3-70-g09d2 From 13c989beba166b470b1e6b0fa117148bcbfa3dd1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 23 Feb 2009 11:33:34 +0100 Subject: ALSA: hda - Don't give over 0dB volume for AD1984A HP laptops Set the upper limit 0dB to the volume of mixer amp 0x20 for AD1984A HP laptops. The overloaded volume may damage the internal speaker. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 2c58d7b05ab..b1680284146 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -3986,6 +3986,14 @@ static int patch_ad1884a(struct hda_codec *codec) spec->multiout.dig_out_nid = 0; codec->patch_ops.unsol_event = ad1884a_hp_unsol_event; codec->patch_ops.init = ad1884a_hp_init; + /* set the upper-limit for mixer amp to 0dB for avoiding the + * possible damage by overloading + */ + snd_hda_override_amp_caps(codec, 0x20, HDA_INPUT, + (0x17 << AC_AMPCAP_OFFSET_SHIFT) | + (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) | + (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) | + (1 << AC_AMPCAP_MUTE_SHIFT)); break; case AD1884A_THINKPAD: spec->mixers[0] = ad1984a_thinkpad_mixers; -- cgit v1.2.3-70-g09d2 From 39c2871eeaeeddcbecee29ec905ec528a057ca52 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 23 Feb 2009 14:14:51 +0100 Subject: ALSA: hda - Add an example about pin reconfiguration Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio.txt | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt index a4e5ef87af6..99958be7b45 100644 --- a/Documentation/sound/alsa/HD-Audio.txt +++ b/Documentation/sound/alsa/HD-Audio.txt @@ -391,6 +391,14 @@ clear:: Resets the codec, removes the mixer elements and PCM stuff of the specified codec, and clear all init verbs and hints. +For example, when you want to change the pin default configuration +value of the pin widget 0x14 to 0x9993013f, and let the driver +re-configure based on that state, run like below: +------------------------------------------------------------------------ + # echo 0x14 0x9993013f > /sys/class/sound/hwC0D0/user_pin_configs + # echo 1 > /sys/class/sound/hwC0D0/reconfig +------------------------------------------------------------------------ + Power-Saving ~~~~~~~~~~~~ -- cgit v1.2.3-70-g09d2 From a65d629ceb4cff5e7d5edadfd6bf1f64c370a517 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 23 Feb 2009 16:57:04 +0100 Subject: ALSA: hda - Add pseudo device-locking for clear/reconfig Added the pseudo device-locking using card->shutdown flag to avoid the crash via clear/reconfig during operations. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 54 +++++++++++++++++++++++++++++++++++++++++++---- sound/pci/hda/hda_hwdep.c | 15 +++++++++++-- sound/pci/hda/hda_local.h | 2 +- 3 files changed, 64 insertions(+), 7 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index a13480fa8e7..5dceee8a113 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1445,9 +1445,52 @@ void snd_hda_ctls_clear(struct hda_codec *codec) snd_array_free(&codec->mixers); } -void snd_hda_codec_reset(struct hda_codec *codec) +/* pseudo device locking + * toggle card->shutdown to allow/disallow the device access (as a hack) + */ +static int hda_lock_devices(struct snd_card *card) { - int i; + spin_lock(&card->files_lock); + if (card->shutdown) { + spin_unlock(&card->files_lock); + return -EINVAL; + } + card->shutdown = 1; + spin_unlock(&card->files_lock); + return 0; +} + +static void hda_unlock_devices(struct snd_card *card) +{ + spin_lock(&card->files_lock); + card->shutdown = 0; + spin_unlock(&card->files_lock); +} + +int snd_hda_codec_reset(struct hda_codec *codec) +{ + struct snd_card *card = codec->bus->card; + int i, pcm; + + if (hda_lock_devices(card) < 0) + return -EBUSY; + /* check whether the codec isn't used by any mixer or PCM streams */ + if (!list_empty(&card->ctl_files)) { + hda_unlock_devices(card); + return -EBUSY; + } + for (pcm = 0; pcm < codec->num_pcms; pcm++) { + struct hda_pcm *cpcm = &codec->pcm_info[pcm]; + if (!cpcm->pcm) + continue; + if (cpcm->pcm->streams[0].substream_opened || + cpcm->pcm->streams[1].substream_opened) { + hda_unlock_devices(card); + return -EBUSY; + } + } + + /* OK, let it free */ #ifdef CONFIG_SND_HDA_POWER_SAVE cancel_delayed_work(&codec->power_work); @@ -1457,8 +1500,7 @@ void snd_hda_codec_reset(struct hda_codec *codec) /* relase PCMs */ for (i = 0; i < codec->num_pcms; i++) { if (codec->pcm_info[i].pcm) { - snd_device_free(codec->bus->card, - codec->pcm_info[i].pcm); + snd_device_free(card, codec->pcm_info[i].pcm); clear_bit(codec->pcm_info[i].device, codec->bus->pcm_dev_bits); } @@ -1479,6 +1521,10 @@ void snd_hda_codec_reset(struct hda_codec *codec) codec->preset = NULL; module_put(codec->owner); codec->owner = NULL; + + /* allow device access again */ + hda_unlock_devices(card); + return 0; } #endif /* CONFIG_SND_HDA_RECONFIG */ diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index c660383ef38..4af484b8240 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -155,7 +155,13 @@ int /*__devinit*/ snd_hda_create_hwdep(struct hda_codec *codec) static int clear_codec(struct hda_codec *codec) { - snd_hda_codec_reset(codec); + int err; + + err = snd_hda_codec_reset(codec); + if (err < 0) { + snd_printk(KERN_ERR "The codec is being used, can't free.\n"); + return err; + } clear_hwdep_elements(codec); return 0; } @@ -165,7 +171,12 @@ static int reconfig_codec(struct hda_codec *codec) int err; snd_printk(KERN_INFO "hda-codec: reconfiguring\n"); - snd_hda_codec_reset(codec); + err = snd_hda_codec_reset(codec); + if (err < 0) { + snd_printk(KERN_ERR + "The codec is being used, can't reconfigure.\n"); + return err; + } err = snd_hda_codec_configure(codec); if (err < 0) return err; diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 84e2cf644fd..4bd82a37a4c 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -98,7 +98,7 @@ struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec, const char *name); int snd_hda_add_vmaster(struct hda_codec *codec, char *name, unsigned int *tlv, const char **slaves); -void snd_hda_codec_reset(struct hda_codec *codec); +int snd_hda_codec_reset(struct hda_codec *codec); int snd_hda_codec_configure(struct hda_codec *codec); /* amp value bits */ -- cgit v1.2.3-70-g09d2 From 8056d9bbb57207854462b6b0a3a75d172300cce5 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 23 Jan 2009 14:44:54 +0000 Subject: ASoC: Improve WM9713 voice DAC shutdown procedure Signed-off-by: Mark Brown --- sound/soc/codecs/wm9713.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 54db9c52498..a93aea5c187 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -940,13 +940,14 @@ static void wm9713_voiceshutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; - u16 status; + u16 status, rate; /* Gracefully shut down the voice interface. */ status = ac97_read(codec, AC97_EXTENDED_STATUS) | 0x1000; - ac97_write(codec, AC97_HANDSET_RATE, 0x0280); + rate = ac97_read(codec, AC97_HANDSET_RATE) & 0xF0FF; + ac97_write(codec, AC97_HANDSET_RATE, rate | 0x0200); schedule_timeout_interruptible(msecs_to_jiffies(1)); - ac97_write(codec, AC97_HANDSET_RATE, 0x0F80); + ac97_write(codec, AC97_HANDSET_RATE, rate | 0x0F00); ac97_write(codec, AC97_EXTENDED_MID, status); } -- cgit v1.2.3-70-g09d2 From d3b894218441ecb1c83e47c682e2d6589ee37a8d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 23 Feb 2009 18:45:19 +0000 Subject: ASoC: Fix Zylonite voice interface stereo configurations We always run in the first timeslot of one. Signed-off-by: Mark Brown --- sound/soc/pxa/zylonite.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index ec2fb764b24..0140a250db2 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -127,9 +127,8 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_tdm_slot(cpu_dai, - params_channels(params), - params_channels(params)); + /* We're not really in network mode but the emulation wants this. */ + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 1); if (ret < 0) return ret; -- cgit v1.2.3-70-g09d2 From 69e169da5a69cc991d54bb4d54f236523145756c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 22 Feb 2009 14:39:03 +0000 Subject: ASoC: Shuffle WM8753 device registration code This patch should be pure code motion, separating that out from the functional changes to move to new style device registration. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8753.c | 209 +++++++++++++++++++++++----------------------- 1 file changed, 105 insertions(+), 104 deletions(-) diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 93c22c4f082..4b426888f98 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -51,6 +51,11 @@ #include "wm8753.h" +#ifdef CONFIG_SPI_MASTER +static struct spi_driver wm8753_spi_driver; +static int wm8753_spi_write(struct spi_device *spi, const char *data, int len); +#endif + static int caps_charge = 2000; module_param(caps_charge, int, 0); MODULE_PARM_DESC(caps_charge, "WM8753 cap charge time (msecs)"); @@ -1626,53 +1631,7 @@ pcm_err: static struct snd_soc_device *wm8753_socdev; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - -/* - * WM8753 2 wire address is determined by GPIO5 - * state during powerup. - * low = 0x1a - * high = 0x1b - */ - -static int wm8753_i2c_probe(struct i2c_client *i2c, - const struct i2c_device_id *id) -{ - struct snd_soc_device *socdev = wm8753_socdev; - struct snd_soc_codec *codec = socdev->card->codec; - int ret; - - i2c_set_clientdata(i2c, codec); - codec->control_data = i2c; - - ret = wm8753_init(socdev); - if (ret < 0) - pr_err("failed to initialise WM8753\n"); - - return ret; -} - -static int wm8753_i2c_remove(struct i2c_client *client) -{ - struct snd_soc_codec *codec = i2c_get_clientdata(client); - kfree(codec->reg_cache); - return 0; -} - -static const struct i2c_device_id wm8753_i2c_id[] = { - { "wm8753", 0 }, - { } -}; -MODULE_DEVICE_TABLE(i2c, wm8753_i2c_id); - -static struct i2c_driver wm8753_i2c_driver = { - .driver = { - .name = "WM8753 I2C Codec", - .owner = THIS_MODULE, - }, - .probe = wm8753_i2c_probe, - .remove = wm8753_i2c_remove, - .id_table = wm8753_i2c_id, -}; +static struct i2c_driver wm8753_i2c_driver; static int wm8753_add_i2c_device(struct platform_device *pdev, const struct wm8753_setup_data *setup) @@ -1715,63 +1674,6 @@ err_driver: } #endif -#if defined(CONFIG_SPI_MASTER) -static int __devinit wm8753_spi_probe(struct spi_device *spi) -{ - struct snd_soc_device *socdev = wm8753_socdev; - struct snd_soc_codec *codec = socdev->card->codec; - int ret; - - codec->control_data = spi; - - ret = wm8753_init(socdev); - if (ret < 0) - dev_err(&spi->dev, "failed to initialise WM8753\n"); - - return ret; -} - -static int __devexit wm8753_spi_remove(struct spi_device *spi) -{ - return 0; -} - -static struct spi_driver wm8753_spi_driver = { - .driver = { - .name = "wm8753", - .bus = &spi_bus_type, - .owner = THIS_MODULE, - }, - .probe = wm8753_spi_probe, - .remove = __devexit_p(wm8753_spi_remove), -}; - -static int wm8753_spi_write(struct spi_device *spi, const char *data, int len) -{ - struct spi_transfer t; - struct spi_message m; - u8 msg[2]; - - if (len <= 0) - return 0; - - msg[0] = data[0]; - msg[1] = data[1]; - - spi_message_init(&m); - memset(&t, 0, (sizeof t)); - - t.tx_buf = &msg[0]; - t.len = len; - - spi_message_add_tail(&t, &m); - spi_sync(spi, &m); - - return len; -} -#endif - - static int wm8753_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); @@ -1876,6 +1778,105 @@ struct snd_soc_codec_device soc_codec_dev_wm8753 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_wm8753); +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + +static int wm8753_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct snd_soc_device *socdev = wm8753_socdev; + struct snd_soc_codec *codec = socdev->card->codec; + int ret; + + i2c_set_clientdata(i2c, codec); + codec->control_data = i2c; + + ret = wm8753_init(socdev); + if (ret < 0) + pr_err("failed to initialise WM8753\n"); + + return ret; +} + +static int wm8753_i2c_remove(struct i2c_client *client) +{ + struct snd_soc_codec *codec = i2c_get_clientdata(client); + kfree(codec->reg_cache); + return 0; +} + +static const struct i2c_device_id wm8753_i2c_id[] = { + { "wm8753", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8753_i2c_id); + +static struct i2c_driver wm8753_i2c_driver = { + .driver = { + .name = "WM8753 I2C Codec", + .owner = THIS_MODULE, + }, + .probe = wm8753_i2c_probe, + .remove = wm8753_i2c_remove, + .id_table = wm8753_i2c_id, +}; +#endif + +#if defined(CONFIG_SPI_MASTER) +static int wm8753_spi_write(struct spi_device *spi, const char *data, int len) +{ + struct spi_transfer t; + struct spi_message m; + u8 msg[2]; + + if (len <= 0) + return 0; + + msg[0] = data[0]; + msg[1] = data[1]; + + spi_message_init(&m); + memset(&t, 0, (sizeof t)); + + t.tx_buf = &msg[0]; + t.len = len; + + spi_message_add_tail(&t, &m); + spi_sync(spi, &m); + + return len; +} + +static int __devinit wm8753_spi_probe(struct spi_device *spi) +{ + struct snd_soc_device *socdev = wm8753_socdev; + struct snd_soc_codec *codec = socdev->card->codec; + int ret; + + codec->control_data = spi; + + ret = wm8753_init(socdev); + if (ret < 0) + dev_err(&spi->dev, "failed to initialise WM8753\n"); + + return ret; +} + +static int __devexit wm8753_spi_remove(struct spi_device *spi) +{ + return 0; +} + +static struct spi_driver wm8753_spi_driver = { + .driver = { + .name = "wm8753", + .bus = &spi_bus_type, + .owner = THIS_MODULE, + }, + .probe = wm8753_spi_probe, + .remove = __devexit_p(wm8753_spi_remove), +}; +#endif + static int __init wm8753_modinit(void) { return snd_soc_register_dais(wm8753_dai, ARRAY_SIZE(wm8753_dai)); -- cgit v1.2.3-70-g09d2 From c2bac1606a937d4700f8fdd8e051a4c49593c41b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 24 Feb 2009 23:33:12 +0000 Subject: ASoC: Convert WM8753 to register via normal device probe The base support for the only in-tree user, the GTA01, is out of tree and will be updated separately. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8753.c | 369 +++++++++++++++++-------------------- sound/soc/codecs/wm8753.h | 6 - sound/soc/s3c24xx/neo1973_wm8753.c | 6 - 3 files changed, 171 insertions(+), 210 deletions(-) diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 4b426888f98..bc29558149e 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -63,12 +63,6 @@ MODULE_PARM_DESC(caps_charge, "WM8753 cap charge time (msecs)"); static void wm8753_set_dai_mode(struct snd_soc_codec *codec, unsigned int mode); -/* codec private data */ -struct wm8753_priv { - unsigned int sysclk; - unsigned int pcmclk; -}; - /* * wm8753 register cache * We can't read the WM8753 register space when we @@ -93,6 +87,14 @@ static const u16 wm8753_reg[] = { 0x0000, 0x0000 }; +/* codec private data */ +struct wm8753_priv { + unsigned int sysclk; + unsigned int pcmclk; + struct snd_soc_codec codec; + u16 reg_cache[ARRAY_SIZE(wm8753_reg)]; +}; + /* * read wm8753 register cache */ @@ -1542,36 +1544,24 @@ static int wm8753_resume(struct platform_device *pdev) return 0; } -/* - * initialise the WM8753 driver - * register the mixer and dsp interfaces with the kernel - */ -static int wm8753_init(struct snd_soc_device *socdev) +static struct snd_soc_codec *wm8753_codec; + +static int wm8753_probe(struct platform_device *pdev) { - struct snd_soc_codec *codec = socdev->card->codec; - int reg, ret = 0; + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; - codec->name = "WM8753"; - codec->owner = THIS_MODULE; - codec->read = wm8753_read_reg_cache; - codec->write = wm8753_write; - codec->set_bias_level = wm8753_set_bias_level; - codec->dai = wm8753_dai; - codec->num_dai = 2; - codec->reg_cache_size = ARRAY_SIZE(wm8753_reg); - codec->reg_cache = kmemdup(wm8753_reg, sizeof(wm8753_reg), GFP_KERNEL); + if (!wm8753_codec) { + dev_err(&pdev->dev, "WM8753 codec not yet registered\n"); + return -EINVAL; + } - if (codec->reg_cache == NULL) - return -ENOMEM; + socdev->card->codec = wm8753_codec; + codec = wm8753_codec; wm8753_set_dai_mode(codec, 0); - ret = wm8753_reset(codec); - if (ret < 0) { - printk(KERN_ERR "wm8753: failed to reset device\n"); - return ret; - } - /* register pcms */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) { @@ -1579,36 +1569,7 @@ static int wm8753_init(struct snd_soc_device *socdev) goto pcm_err; } - /* charge output caps */ - wm8753_set_bias_level(codec, SND_SOC_BIAS_PREPARE); - codec->bias_level = SND_SOC_BIAS_STANDBY; - schedule_delayed_work(&codec->delayed_work, - msecs_to_jiffies(caps_charge)); - - /* set the update bits */ - reg = wm8753_read_reg_cache(codec, WM8753_LDAC); - wm8753_write(codec, WM8753_LDAC, reg | 0x0100); - reg = wm8753_read_reg_cache(codec, WM8753_RDAC); - wm8753_write(codec, WM8753_RDAC, reg | 0x0100); - reg = wm8753_read_reg_cache(codec, WM8753_LADC); - wm8753_write(codec, WM8753_LADC, reg | 0x0100); - reg = wm8753_read_reg_cache(codec, WM8753_RADC); - wm8753_write(codec, WM8753_RADC, reg | 0x0100); - reg = wm8753_read_reg_cache(codec, WM8753_LOUT1V); - wm8753_write(codec, WM8753_LOUT1V, reg | 0x0100); - reg = wm8753_read_reg_cache(codec, WM8753_ROUT1V); - wm8753_write(codec, WM8753_ROUT1V, reg | 0x0100); - reg = wm8753_read_reg_cache(codec, WM8753_LOUT2V); - wm8753_write(codec, WM8753_LOUT2V, reg | 0x0100); - reg = wm8753_read_reg_cache(codec, WM8753_ROUT2V); - wm8753_write(codec, WM8753_ROUT2V, reg | 0x0100); - reg = wm8753_read_reg_cache(codec, WM8753_LINVOL); - wm8753_write(codec, WM8753_LINVOL, reg | 0x0100); - reg = wm8753_read_reg_cache(codec, WM8753_RINVOL); - wm8753_write(codec, WM8753_RINVOL, reg | 0x0100); - - snd_soc_add_controls(codec, wm8753_snd_controls, - ARRAY_SIZE(wm8753_snd_controls)); + wm8753_add_controls(codec); wm8753_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { @@ -1616,110 +1577,13 @@ static int wm8753_init(struct snd_soc_device *socdev) goto card_err; } - return ret; + return 0; card_err: snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); -pcm_err: - kfree(codec->reg_cache); - return ret; -} - -/* If the i2c layer weren't so broken, we could pass this kind of data - around */ -static struct snd_soc_device *wm8753_socdev; - -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) -static struct i2c_driver wm8753_i2c_driver; - -static int wm8753_add_i2c_device(struct platform_device *pdev, - const struct wm8753_setup_data *setup) -{ - struct i2c_board_info info; - struct i2c_adapter *adapter; - struct i2c_client *client; - int ret; - - ret = i2c_add_driver(&wm8753_i2c_driver); - if (ret != 0) { - dev_err(&pdev->dev, "can't add i2c driver\n"); - return ret; - } - - memset(&info, 0, sizeof(struct i2c_board_info)); - info.addr = setup->i2c_address; - strlcpy(info.type, "wm8753", I2C_NAME_SIZE); - - adapter = i2c_get_adapter(setup->i2c_bus); - if (!adapter) { - dev_err(&pdev->dev, "can't get i2c adapter %d\n", - setup->i2c_bus); - goto err_driver; - } - - client = i2c_new_device(adapter, &info); - i2c_put_adapter(adapter); - if (!client) { - dev_err(&pdev->dev, "can't add i2c device at 0x%x\n", - (unsigned int)info.addr); - goto err_driver; - } - - return 0; - -err_driver: - i2c_del_driver(&wm8753_i2c_driver); - return -ENODEV; -} -#endif - -static int wm8753_probe(struct platform_device *pdev) -{ - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct wm8753_setup_data *setup; - struct snd_soc_codec *codec; - struct wm8753_priv *wm8753; - int ret = 0; - - setup = socdev->codec_data; - codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); - if (codec == NULL) - return -ENOMEM; - - wm8753 = kzalloc(sizeof(struct wm8753_priv), GFP_KERNEL); - if (wm8753 == NULL) { - kfree(codec); - return -ENOMEM; - } - - codec->private_data = wm8753; - socdev->card->codec = codec; - mutex_init(&codec->mutex); - INIT_LIST_HEAD(&codec->dapm_widgets); - INIT_LIST_HEAD(&codec->dapm_paths); - wm8753_socdev = socdev; - INIT_DELAYED_WORK(&codec->delayed_work, wm8753_work); - -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - if (setup->i2c_address) { - codec->hw_write = (hw_write_t)i2c_master_send; - ret = wm8753_add_i2c_device(pdev, setup); - } -#endif -#if defined(CONFIG_SPI_MASTER) - if (setup->spi) { - codec->hw_write = (hw_write_t)wm8753_spi_write; - ret = spi_register_driver(&wm8753_spi_driver); - if (ret != 0) - printk(KERN_ERR "can't add spi driver"); - } -#endif - if (ret != 0) { - kfree(codec->private_data); - kfree(codec); - } +pcm_err: return ret; } @@ -1746,26 +1610,9 @@ static int run_delayed_work(struct delayed_work *dwork) static int wm8753_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->card->codec; - struct wm8753_setup_data *setup = socdev->codec_data; - if (codec->control_data) - wm8753_set_bias_level(codec, SND_SOC_BIAS_OFF); - run_delayed_work(&codec->delayed_work); snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - if (setup->i2c_address) { - i2c_unregister_device(codec->control_data); - i2c_del_driver(&wm8753_i2c_driver); - } -#endif -#if defined(CONFIG_SPI_MASTER) - if (setup->spi) - spi_unregister_driver(&wm8753_spi_driver); -#endif - kfree(codec->private_data); - kfree(codec); return 0; } @@ -1778,30 +1625,134 @@ struct snd_soc_codec_device soc_codec_dev_wm8753 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_wm8753); +static int wm8753_register(struct wm8753_priv *wm8753) +{ + int ret, i; + struct snd_soc_codec *codec = &wm8753->codec; + u16 reg; + + if (wm8753_codec) { + dev_err(codec->dev, "Multiple WM8753 devices not supported\n"); + ret = -EINVAL; + goto err; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->name = "WM8753"; + codec->owner = THIS_MODULE; + codec->read = wm8753_read_reg_cache; + codec->write = wm8753_write; + codec->bias_level = SND_SOC_BIAS_STANDBY; + codec->set_bias_level = wm8753_set_bias_level; + codec->dai = wm8753_dai; + codec->num_dai = 2; + codec->reg_cache_size = ARRAY_SIZE(wm8753->reg_cache); + codec->reg_cache = &wm8753->reg_cache; + codec->private_data = wm8753; + + memcpy(codec->reg_cache, wm8753_reg, sizeof(codec->reg_cache)); + INIT_DELAYED_WORK(&codec->delayed_work, wm8753_work); + + ret = wm8753_reset(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to issue reset\n"); + goto err; + } + + /* charge output caps */ + wm8753_set_bias_level(codec, SND_SOC_BIAS_PREPARE); + schedule_delayed_work(&codec->delayed_work, + msecs_to_jiffies(caps_charge)); + + /* set the update bits */ + reg = wm8753_read_reg_cache(codec, WM8753_LDAC); + wm8753_write(codec, WM8753_LDAC, reg | 0x0100); + reg = wm8753_read_reg_cache(codec, WM8753_RDAC); + wm8753_write(codec, WM8753_RDAC, reg | 0x0100); + reg = wm8753_read_reg_cache(codec, WM8753_LADC); + wm8753_write(codec, WM8753_LADC, reg | 0x0100); + reg = wm8753_read_reg_cache(codec, WM8753_RADC); + wm8753_write(codec, WM8753_RADC, reg | 0x0100); + reg = wm8753_read_reg_cache(codec, WM8753_LOUT1V); + wm8753_write(codec, WM8753_LOUT1V, reg | 0x0100); + reg = wm8753_read_reg_cache(codec, WM8753_ROUT1V); + wm8753_write(codec, WM8753_ROUT1V, reg | 0x0100); + reg = wm8753_read_reg_cache(codec, WM8753_LOUT2V); + wm8753_write(codec, WM8753_LOUT2V, reg | 0x0100); + reg = wm8753_read_reg_cache(codec, WM8753_ROUT2V); + wm8753_write(codec, WM8753_ROUT2V, reg | 0x0100); + reg = wm8753_read_reg_cache(codec, WM8753_LINVOL); + wm8753_write(codec, WM8753_LINVOL, reg | 0x0100); + reg = wm8753_read_reg_cache(codec, WM8753_RINVOL); + wm8753_write(codec, WM8753_RINVOL, reg | 0x0100); + + wm8753_codec = codec; + + for (i = 0; i < ARRAY_SIZE(wm8753_dai); i++) + wm8753_dai[i].dev = codec->dev; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + goto err; + } + + ret = snd_soc_register_dais(&wm8753_dai[0], ARRAY_SIZE(wm8753_dai)); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAIs: %d\n", ret); + goto err_codec; + } + + return 0; + +err_codec: + run_delayed_work(&codec->delayed_work); + snd_soc_unregister_codec(codec); +err: + kfree(wm8753); + return ret; +} + +static void wm8753_unregister(struct wm8753_priv *wm8753) +{ + wm8753_set_bias_level(&wm8753->codec, SND_SOC_BIAS_OFF); + run_delayed_work(&wm8753->codec.delayed_work); + snd_soc_unregister_dais(&wm8753_dai[0], ARRAY_SIZE(wm8753_dai)); + snd_soc_unregister_codec(&wm8753->codec); + kfree(wm8753); + wm8753_codec = NULL; +} + #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static int wm8753_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { - struct snd_soc_device *socdev = wm8753_socdev; - struct snd_soc_codec *codec = socdev->card->codec; - int ret; + struct snd_soc_codec *codec; + struct wm8753_priv *wm8753; - i2c_set_clientdata(i2c, codec); - codec->control_data = i2c; + wm8753 = kzalloc(sizeof(struct wm8753_priv), GFP_KERNEL); + if (wm8753 == NULL) + return -ENOMEM; - ret = wm8753_init(socdev); - if (ret < 0) - pr_err("failed to initialise WM8753\n"); + codec = &wm8753->codec; + codec->hw_write = (hw_write_t)i2c_master_send; + codec->control_data = i2c; + i2c_set_clientdata(i2c, wm8753); - return ret; + codec->dev = &i2c->dev; + + return wm8753_register(wm8753); } static int wm8753_i2c_remove(struct i2c_client *client) { - struct snd_soc_codec *codec = i2c_get_clientdata(client); - kfree(codec->reg_cache); - return 0; + struct wm8753_priv *wm8753 = i2c_get_clientdata(client); + wm8753_unregister(wm8753); + return 0; } static const struct i2c_device_id wm8753_i2c_id[] = { @@ -1812,7 +1763,7 @@ MODULE_DEVICE_TABLE(i2c, wm8753_i2c_id); static struct i2c_driver wm8753_i2c_driver = { .driver = { - .name = "WM8753 I2C Codec", + .name = "wm8753", .owner = THIS_MODULE, }, .probe = wm8753_i2c_probe, @@ -1848,21 +1799,27 @@ static int wm8753_spi_write(struct spi_device *spi, const char *data, int len) static int __devinit wm8753_spi_probe(struct spi_device *spi) { - struct snd_soc_device *socdev = wm8753_socdev; - struct snd_soc_codec *codec = socdev->card->codec; - int ret; + struct snd_soc_codec *codec; + struct wm8753_priv *wm8753; + + wm8753 = kzalloc(sizeof(struct wm8753_priv), GFP_KERNEL); + if (wm8753 == NULL) + return -ENOMEM; + codec = &wm8753->codec; codec->control_data = spi; + codec->hw_write = (hw_write_t)wm8753_spi_write; + codec->dev = &spi->dev; - ret = wm8753_init(socdev); - if (ret < 0) - dev_err(&spi->dev, "failed to initialise WM8753\n"); + spi->dev.driver_data = wm8753; - return ret; + return wm8753_register(wm8753); } static int __devexit wm8753_spi_remove(struct spi_device *spi) { + struct wm8753_priv *wm8753 = spi->dev.driver_data; + wm8753_unregister(wm8753); return 0; } @@ -1879,13 +1836,29 @@ static struct spi_driver wm8753_spi_driver = { static int __init wm8753_modinit(void) { - return snd_soc_register_dais(wm8753_dai, ARRAY_SIZE(wm8753_dai)); + int ret; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + ret = i2c_add_driver(&wm8753_i2c_driver); + if (ret != 0) + pr_err("Failed to register WM8753 I2C driver: %d\n", ret); +#endif +#if defined(CONFIG_SPI_MASTER) + ret = spi_register_driver(&wm8753_spi_driver); + if (ret != 0) + pr_err("Failed to register WM8753 SPI driver: %d\n", ret); +#endif + return 0; } module_init(wm8753_modinit); static void __exit wm8753_exit(void) { - snd_soc_unregister_dais(wm8753_dai, ARRAY_SIZE(wm8753_dai)); +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&wm8753_i2c_driver); +#endif +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&wm8753_spi_driver); +#endif } module_exit(wm8753_exit); diff --git a/sound/soc/codecs/wm8753.h b/sound/soc/codecs/wm8753.h index f55704ce931..57b2ba24404 100644 --- a/sound/soc/codecs/wm8753.h +++ b/sound/soc/codecs/wm8753.h @@ -77,12 +77,6 @@ #define WM8753_BIASCTL 0x3d #define WM8753_ADCTL2 0x3f -struct wm8753_setup_data { - int spi; - int i2c_bus; - unsigned short i2c_address; -}; - #define WM8753_PLL1 0 #define WM8753_PLL2 1 diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index 45bb12e8ea4..286e11ad50e 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -585,15 +585,9 @@ static struct snd_soc_card neo1973 = { .num_links = ARRAY_SIZE(neo1973_dai), }; -static struct wm8753_setup_data neo1973_wm8753_setup = { - .i2c_bus = 0, - .i2c_address = 0x1a, -}; - static struct snd_soc_device neo1973_snd_devdata = { .card = &neo1973, .codec_dev = &soc_codec_dev_wm8753, - .codec_data = &neo1973_wm8753_setup, }; static int lm4857_i2c_probe(struct i2c_client *client, -- cgit v1.2.3-70-g09d2 From e611bd82441130991d7f4600dfd4632cebd417c5 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 22 Feb 2009 20:04:41 +0000 Subject: ASoC: Only write back non-default registers when resuming WM8753 This will reduce the number of writes done on resume, allowing that to complete faster (especially on systems with very slow I2C like the current Samsung driver). Signed-off-by: Mark Brown --- sound/soc/codecs/wm8753.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index bc29558149e..2241204b515 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1526,6 +1526,11 @@ static int wm8753_resume(struct platform_device *pdev) for (i = 0; i < ARRAY_SIZE(wm8753_reg); i++) { if (i + 1 == WM8753_RESET) continue; + + /* No point in writing hardware default values back */ + if (cache[i] == wm8753_reg[i]) + continue; + data[0] = ((i + 1) << 1) | ((cache[i] >> 8) & 0x0001); data[1] = cache[i] & 0x00ff; codec->hw_write(codec->control_data, data, 2); -- cgit v1.2.3-70-g09d2 From 873dc78a8676b7ba6260b1d74c50d8ea5025ecbe Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Feb 2009 18:12:13 +0100 Subject: ALSA: hda - Clean up / fix quirks for HP laptops with AD1984A Use SND_PCI_QUIRK_MASK() to clean up / support better HP laptops with AD1984A codec. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 0253cb93aa7..5bb48ee8b6c 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -3923,9 +3923,8 @@ static struct snd_pci_quirk ad1884a_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x3037, "HP 2230s", AD1884A_LAPTOP), SND_PCI_QUIRK(0x103c, 0x3056, "HP", AD1884A_MOBILE), SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x3070, "HP", AD1884A_MOBILE), - SND_PCI_QUIRK(0x103c, 0x30e6, "HP 6730b", AD1884A_LAPTOP), - SND_PCI_QUIRK(0x103c, 0x30e7, "HP EliteBook 8530p", AD1884A_LAPTOP), - SND_PCI_QUIRK(0x103c, 0x3614, "HP 6730s", AD1884A_LAPTOP), + SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x30e0, "HP laptop", AD1884A_LAPTOP), + SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3600, "HP laptop", AD1884A_LAPTOP), SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD), {} }; -- cgit v1.2.3-70-g09d2 From f872a9194cb006994d47a58efc875218594e6072 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 26 Feb 2009 00:57:01 +0100 Subject: ALSA: hda - Clean up / fix quirk for Sony laptops with ALC262 Clean up / fix quirk entries for Sony laptops with ALC262 codec using NSD_PCI_QUIRK_MASK(). This also fixes the kernel bug #12780 http://bugme.linux-foundation.org/show_bug.cgi?id=12780 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 50ae8f33af5..d670d33cfa1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10824,10 +10824,8 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x104d, 0x1f00, "Sony ASSAMD", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x104d, 0x8203, "Sony UX-90", ALC262_HIPPO), SND_PCI_QUIRK(0x104d, 0x820f, "Sony ASSAMD", ALC262_SONY_ASSAMD), - SND_PCI_QUIRK(0x104d, 0x900e, "Sony ASSAMD", ALC262_SONY_ASSAMD), - SND_PCI_QUIRK(0x104d, 0x9015, "Sony 0x9015", ALC262_SONY_ASSAMD), - SND_PCI_QUIRK(0x104d, 0x9033, "Sony VAIO VGN-SR19XN", - ALC262_SONY_ASSAMD), + SND_PCI_QUIRK_MASK(0x104d, 0xff00, 0x9000, "Sony VAIO", + ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1", ALC262_TOSHIBA_RX1), SND_PCI_QUIRK(0x1179, 0xff7b, "Toshiba S06", ALC262_TOSHIBA_S06), -- cgit v1.2.3-70-g09d2 From 930738de602d2ceb0d1c1b368fe2a8d2a974ab72 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 26 Feb 2009 09:27:20 +0100 Subject: sound: virtuoso: add Xonar Essence STX support Add support for the Asus Xonar Essence STX sound card. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/ALSA-Configuration.txt | 2 +- sound/pci/Kconfig | 3 +- sound/pci/oxygen/virtuoso.c | 192 ++++++++++++++++++++++++ 3 files changed, 195 insertions(+), 2 deletions(-) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 841a9365d5f..1356d2a6772 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -1824,7 +1824,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. ------------------- Module for sound cards based on the Asus AV100/AV200 chips, - i.e., Xonar D1, DX, D2, D2X and HDAV1.3 (Deluxe). + i.e., Xonar D1, DX, D2, D2X, HDAV1.3 (Deluxe), and Essence STX. This module supports autoprobe and multiple cards. diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 82b9bddcdcd..21d117ada84 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -744,7 +744,8 @@ config SND_VIRTUOSO select SND_OXYGEN_LIB help Say Y here to include support for sound cards based on the - Asus AV100/AV200 chips, i.e., Xonar D1, DX, D2 and D2X. + Asus AV100/AV200 chips, i.e., Xonar D1, DX, D2, D2X, and + Essence STX. Support for the HDAV1.3 (Deluxe) is very experimental. To compile this driver as a module, choose M here: the module diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 00dc97806f1..bc5ce11c8b1 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -112,6 +112,34 @@ * CS4362A: AD0 <- 0 */ +/* + * Xonar Essence STX + * ----------------- + * + * CMI8788: + * + * I²C <-> PCM1792A + * + * GPI 0 <- external power present + * + * GPIO 0 -> enable output to speakers + * GPIO 1 -> route HP to front panel (0) or rear jack (1) + * GPIO 2 -> M0 of CS5381 + * GPIO 3 -> M1 of CS5381 + * GPIO 7 -> route output to speaker jacks (0) or HP (1) + * GPIO 8 -> route input jack to line-in (0) or mic-in (1) + * + * PCM1792A: + * + * AD0 <- 0 + * + * H6 daughterboard + * ---------------- + * + * GPIO 4 <- 0 + * GPIO 5 <- 0 + */ + #include #include #include @@ -152,6 +180,7 @@ enum { MODEL_DX, MODEL_HDAV, /* without daughterboard */ MODEL_HDAV_H6, /* with H6 daughterboard */ + MODEL_STX, }; static struct pci_device_id xonar_ids[] __devinitdata = { @@ -160,6 +189,7 @@ static struct pci_device_id xonar_ids[] __devinitdata = { { OXYGEN_PCI_SUBID(0x1043, 0x82b7), .driver_data = MODEL_D2X }, { OXYGEN_PCI_SUBID(0x1043, 0x8314), .driver_data = MODEL_HDAV }, { OXYGEN_PCI_SUBID(0x1043, 0x834f), .driver_data = MODEL_D1 }, + { OXYGEN_PCI_SUBID(0x1043, 0x835c), .driver_data = MODEL_STX }, { OXYGEN_PCI_SUBID_BROKEN_EEPROM }, { } }; @@ -184,6 +214,9 @@ MODULE_DEVICE_TABLE(pci, xonar_ids); #define GPIO_HDAV_DB_H6 0x0000 #define GPIO_HDAV_DB_XX 0x0020 +#define GPIO_ST_HP_REAR 0x0002 +#define GPIO_ST_HP 0x0080 + #define I2C_DEVICE_PCM1796(i) (0x98 + ((i) << 1)) /* 10011, ADx=i, /W=0 */ #define I2C_DEVICE_CS4398 0x9e /* 10011, AD1=1, AD0=1, /W=0 */ #define I2C_DEVICE_CS4362A 0x30 /* 001100, AD0=0, /W=0 */ @@ -497,6 +530,36 @@ static void xonar_hdav_init(struct oxygen *chip) snd_component_add(chip->card, "CS5381"); } +static void xonar_stx_init(struct oxygen *chip) +{ + struct xonar_data *data = chip->model_data; + + oxygen_write16(chip, OXYGEN_2WIRE_BUS_STATUS, + OXYGEN_2WIRE_LENGTH_8 | + OXYGEN_2WIRE_INTERRUPT_MASK | + OXYGEN_2WIRE_SPEED_FAST); + + data->anti_pop_delay = 100; + data->dacs = 1; + data->output_enable_bit = GPIO_DX_OUTPUT_ENABLE; + data->ext_power_reg = OXYGEN_GPI_DATA; + data->ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK; + data->ext_power_bit = GPI_DX_EXT_POWER; + data->pcm1796_oversampling = PCM1796_OS_64; + + pcm1796_init(chip); + + oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, + GPIO_DX_INPUT_ROUTE | GPIO_ST_HP_REAR | GPIO_ST_HP); + oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, + GPIO_DX_INPUT_ROUTE | GPIO_ST_HP_REAR | GPIO_ST_HP); + + xonar_common_init(chip); + + snd_component_add(chip->card, "PCM1792A"); + snd_component_add(chip->card, "CS5381"); +} + static void xonar_disable_output(struct oxygen *chip) { struct xonar_data *data = chip->model_data; @@ -524,6 +587,11 @@ static void xonar_hdav_cleanup(struct oxygen *chip) xonar_disable_output(chip); } +static void xonar_st_cleanup(struct oxygen *chip) +{ + xonar_disable_output(chip); +} + static void xonar_d2_suspend(struct oxygen *chip) { xonar_d2_cleanup(chip); @@ -540,6 +608,11 @@ static void xonar_hdav_suspend(struct oxygen *chip) msleep(2); } +static void xonar_st_suspend(struct oxygen *chip) +{ + xonar_st_cleanup(chip); +} + static void xonar_d2_resume(struct oxygen *chip) { pcm1796_init(chip); @@ -567,6 +640,12 @@ static void xonar_hdav_resume(struct oxygen *chip) xonar_enable_output(chip); } +static void xonar_st_resume(struct oxygen *chip) +{ + pcm1796_init(chip); + xonar_enable_output(chip); +} + static void xonar_hdav_pcm_hardware_filter(unsigned int channel, struct snd_pcm_hardware *hardware) { @@ -746,6 +825,72 @@ static const struct snd_kcontrol_new front_panel_switch = { .private_value = GPIO_DX_FRONT_PANEL, }; +static int st_output_switch_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + static const char *const names[3] = { + "Speakers", "Headphones", "FP Headphones" + }; + + info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + info->count = 1; + info->value.enumerated.items = 3; + if (info->value.enumerated.item >= 3) + info->value.enumerated.item = 2; + strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); + return 0; +} + +static int st_output_switch_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + u16 gpio; + + gpio = oxygen_read16(chip, OXYGEN_GPIO_DATA); + if (!(gpio & GPIO_ST_HP)) + value->value.enumerated.item[0] = 0; + else if (gpio & GPIO_ST_HP_REAR) + value->value.enumerated.item[0] = 1; + else + value->value.enumerated.item[0] = 2; + return 0; +} + + +static int st_output_switch_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + u16 gpio_old, gpio; + + mutex_lock(&chip->mutex); + gpio_old = oxygen_read16(chip, OXYGEN_GPIO_DATA); + gpio = gpio_old; + switch (value->value.enumerated.item[0]) { + case 0: + gpio &= ~(GPIO_ST_HP | GPIO_ST_HP_REAR); + break; + case 1: + gpio |= GPIO_ST_HP | GPIO_ST_HP_REAR; + break; + case 2: + gpio = (gpio | GPIO_ST_HP) & ~GPIO_ST_HP_REAR; + break; + } + oxygen_write16(chip, OXYGEN_GPIO_DATA, gpio); + mutex_unlock(&chip->mutex); + return gpio != gpio_old; +} + +static const struct snd_kcontrol_new st_output_switch = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog Output", + .info = st_output_switch_info, + .get = st_output_switch_get, + .put = st_output_switch_put, +}; + static void xonar_line_mic_ac97_switch(struct oxygen *chip, unsigned int reg, unsigned int mute) { @@ -776,6 +921,15 @@ static int xonar_d1_control_filter(struct snd_kcontrol_new *template) return 0; } +static int xonar_st_control_filter(struct snd_kcontrol_new *template) +{ + if (!strncmp(template->name, "CD Capture ", 11)) + return 1; /* no CD input */ + if (!strcmp(template->name, "Stereo Upmixing")) + return 1; /* stereo only - we don't need upmixing */ + return 0; +} + static int xonar_d2_mixer_init(struct oxygen *chip) { return snd_ctl_add(chip->card, snd_ctl_new1(&alt_switch, chip)); @@ -786,6 +940,11 @@ static int xonar_d1_mixer_init(struct oxygen *chip) return snd_ctl_add(chip->card, snd_ctl_new1(&front_panel_switch, chip)); } +static int xonar_st_mixer_init(struct oxygen *chip) +{ + return snd_ctl_add(chip->card, snd_ctl_new1(&st_output_switch, chip)); +} + static const struct oxygen_model model_xonar_d2 = { .longname = "Asus Virtuoso 200", .chip = "AV200", @@ -872,6 +1031,33 @@ static const struct oxygen_model model_xonar_hdav = { .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, }; +static const struct oxygen_model model_xonar_st = { + .longname = "Asus Virtuoso 100", + .chip = "AV200", + .init = xonar_stx_init, + .control_filter = xonar_st_control_filter, + .mixer_init = xonar_st_mixer_init, + .cleanup = xonar_st_cleanup, + .suspend = xonar_st_suspend, + .resume = xonar_st_resume, + .set_dac_params = set_pcm1796_params, + .set_adc_params = set_cs53x1_params, + .update_dac_volume = update_pcm1796_volume, + .update_dac_mute = update_pcm1796_mute, + .ac97_switch = xonar_line_mic_ac97_switch, + .dac_tlv = pcm1796_db_scale, + .model_data_size = sizeof(struct xonar_data), + .device_config = PLAYBACK_0_TO_I2S | + PLAYBACK_1_TO_SPDIF | + CAPTURE_0_FROM_I2S_2, + .dac_channels = 2, + .dac_volume_min = 255 - 2*60, + .dac_volume_max = 255, + .function_flags = OXYGEN_FUNCTION_2WIRE, + .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, + .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, +}; + static int __devinit get_xonar_model(struct oxygen *chip, const struct pci_device_id *id) { @@ -881,6 +1067,7 @@ static int __devinit get_xonar_model(struct oxygen *chip, [MODEL_D2] = &model_xonar_d2, [MODEL_D2X] = &model_xonar_d2, [MODEL_HDAV] = &model_xonar_hdav, + [MODEL_STX] = &model_xonar_st, }; static const char *const names[] = { [MODEL_D1] = "Xonar D1", @@ -889,6 +1076,7 @@ static int __devinit get_xonar_model(struct oxygen *chip, [MODEL_D2X] = "Xonar D2X", [MODEL_HDAV] = "Xonar HDAV1.3", [MODEL_HDAV_H6] = "Xonar HDAV1.3+H6", + [MODEL_STX] = "Xonar Essence STX", }; unsigned int model = id->driver_data; @@ -916,6 +1104,10 @@ static int __devinit get_xonar_model(struct oxygen *chip, return -ENODEV; } break; + case MODEL_STX: + oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL, + GPIO_HDAV_DB_MASK); + break; } chip->model.shortname = names[model]; -- cgit v1.2.3-70-g09d2 From 5d44aa4c7322e0cda6d71cc3f0dffaceea0daae5 Mon Sep 17 00:00:00 2001 From: Hannes Eder Date: Wed, 25 Feb 2009 22:29:29 +0100 Subject: sound/oss: fix sparse warnings: different signedness Impact: Change signature of 'set_volume_stereo' and 'set_volume_mono'. Fix this sparse warnings: sound/oss/pss.c:545:42: warning: incorrect type in argument 2 (different signedness) sound/oss/pss.c:546:42: warning: incorrect type in argument 3 (different signedness) sound/oss/pss.c:554:59: warning: incorrect type in argument 2 (different signedness) sound/oss/pss.c:560:59: warning: incorrect type in argument 2 (different signedness) sound/oss/pss.c:566:59: warning: incorrect type in argument 2 (different signedness) Signed-off-by: Hannes Eder Signed-off-by: Takashi Iwai --- sound/oss/pss.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/oss/pss.c b/sound/oss/pss.c index 16ed06950dc..16517a5a130 100644 --- a/sound/oss/pss.c +++ b/sound/oss/pss.c @@ -457,10 +457,9 @@ static void pss_mixer_reset(pss_confdata *devc) } } -static int set_volume_mono(unsigned __user *p, int *aleft) +static int set_volume_mono(unsigned __user *p, unsigned int *aleft) { - int left; - unsigned volume; + unsigned int left, volume; if (get_user(volume, p)) return -EFAULT; @@ -471,10 +470,11 @@ static int set_volume_mono(unsigned __user *p, int *aleft) return 0; } -static int set_volume_stereo(unsigned __user *p, int *aleft, int *aright) +static int set_volume_stereo(unsigned __user *p, + unsigned int *aleft, + unsigned int *aright) { - int left, right; - unsigned volume; + unsigned int left, right, volume; if (get_user(volume, p)) return -EFAULT; -- cgit v1.2.3-70-g09d2 From e5bf48437370f3fc603e2dce12e8d3fb1a6a2457 Mon Sep 17 00:00:00 2001 From: Hannes Eder Date: Wed, 25 Feb 2009 22:29:47 +0100 Subject: sound/oss: fix sparse warning: symbol shadows an earlier one Impact: Move variable to a more inner scope. Fix this sparse warning: sound/oss/sequencer.c:235:29: warning: symbol 'err' shadows an earlier one sound/oss/sequencer.c:215:13: originally declared here Signed-off-by: Hannes Eder Signed-off-by: Takashi Iwai --- sound/oss/sequencer.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/oss/sequencer.c b/sound/oss/sequencer.c index 5c215f787ca..c79874696be 100644 --- a/sound/oss/sequencer.c +++ b/sound/oss/sequencer.c @@ -212,7 +212,6 @@ int sequencer_write(int dev, struct file *file, const char __user *buf, int coun { unsigned char event_rec[EV_SZ], ev_code; int p = 0, c, ev_size; - int err; int mode = translate_mode(file); dev = dev >> 4; @@ -285,7 +284,7 @@ int sequencer_write(int dev, struct file *file, const char __user *buf, int coun { if (!midi_opened[event_rec[2]]) { - int mode; + int err, mode; int dev = event_rec[2]; if (dev >= max_mididev || midi_devs[dev]==NULL) -- cgit v1.2.3-70-g09d2 From 619389882ba37121d0f2f7b08e4944e47b379118 Mon Sep 17 00:00:00 2001 From: Hannes Eder Date: Wed, 25 Feb 2009 22:26:48 +0100 Subject: ALSA: sound/usb/usx2y: fix sparse warning: Should it be static? Impact: Move declaration to header file. Fix this sparse warning: sound/usb/usx2y/usx2yhwdeppcm.c:739:5: warning: symbol 'usX2Y_hwdep_pcm_new' was not declared. Should it be static? Signed-off-by: Hannes Eder Signed-off-by: Takashi Iwai --- sound/usb/usx2y/usX2Yhwdep.c | 3 --- sound/usb/usx2y/usx2yhwdeppcm.h | 2 ++ 2 files changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/usb/usx2y/usX2Yhwdep.c b/sound/usb/usx2y/usX2Yhwdep.c index 1558a5c4094..fc650c800af 100644 --- a/sound/usb/usx2y/usX2Yhwdep.c +++ b/sound/usb/usx2y/usX2Yhwdep.c @@ -30,9 +30,6 @@ #include "usbusx2y.h" #include "usX2Yhwdep.h" -int usX2Y_hwdep_pcm_new(struct snd_card *card); - - static int snd_us428ctls_vm_fault(struct vm_area_struct *area, struct vm_fault *vmf) { diff --git a/sound/usb/usx2y/usx2yhwdeppcm.h b/sound/usb/usx2y/usx2yhwdeppcm.h index c3382fdc386..9c4fb84b2aa 100644 --- a/sound/usb/usx2y/usx2yhwdeppcm.h +++ b/sound/usb/usx2y/usx2yhwdeppcm.h @@ -18,3 +18,5 @@ struct snd_usX2Y_hwdep_pcm_shm { volatile unsigned captured_iso_frames; int capture_iso_start; }; + +int usX2Y_hwdep_pcm_new(struct snd_card *card); -- cgit v1.2.3-70-g09d2 From 3a755ec2e8af0024a06a5adbcc81c012eae61782 Mon Sep 17 00:00:00 2001 From: Hannes Eder Date: Wed, 25 Feb 2009 22:28:26 +0100 Subject: ALSA: sound/usb/usx2y: fix sparse warning: do-while statement is not a compound ... Fix this sparse warning: sound/usb/usx2y/usbusx2y.c:231:33: warning: do-while statement is not a compound statement Signed-off-by: Hannes Eder Signed-off-by: Takashi Iwai --- sound/usb/usx2y/usbusx2y.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/usb/usx2y/usbusx2y.c b/sound/usb/usx2y/usbusx2y.c index 11639bd72a5..c545a02dee4 100644 --- a/sound/usb/usx2y/usbusx2y.c +++ b/sound/usb/usx2y/usbusx2y.c @@ -227,9 +227,9 @@ static void i_usX2Y_In04Int(struct urb *urb) if (usX2Y->US04) { if (0 == usX2Y->US04->submitted) - do + do { err = usb_submit_urb(usX2Y->US04->urb[usX2Y->US04->submitted++], GFP_ATOMIC); - while (!err && usX2Y->US04->submitted < usX2Y->US04->len); + } while (!err && usX2Y->US04->submitted < usX2Y->US04->len); } else if (us428ctls && us428ctls->p4outLast >= 0 && us428ctls->p4outLast < N_us428_p4out_BUFS) { if (us428ctls->p4outLast != us428ctls->p4outSent) { -- cgit v1.2.3-70-g09d2 From d73d341d3995ae3c63a4b4543b7c308ebd1e58ea Mon Sep 17 00:00:00 2001 From: Hannes Eder Date: Wed, 25 Feb 2009 22:29:15 +0100 Subject: ALSA: sound/drivers/vx: fix sparse warning: different signedness Fix this sparse warning: sound/drivers/vx/vx_uer.c:301:42: warning: incorrect type in argument 2 (different signedness) Signed-off-by: Hannes Eder Signed-off-by: Takashi Iwai --- sound/drivers/vx/vx_uer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/drivers/vx/vx_uer.c b/sound/drivers/vx/vx_uer.c index 0e1ba9b4790..b0560fec6bb 100644 --- a/sound/drivers/vx/vx_uer.c +++ b/sound/drivers/vx/vx_uer.c @@ -103,7 +103,7 @@ static void vx_write_one_cbit(struct vx_core *chip, int index, int val) * returns the frequency of UER, or 0 if not sync, * or a negative error code. */ -static int vx_read_uer_status(struct vx_core *chip, int *mode) +static int vx_read_uer_status(struct vx_core *chip, unsigned int *mode) { int val, freq; -- cgit v1.2.3-70-g09d2 From 730d45f9130f81fd49009301e9dfbd19fe2b3e1f Mon Sep 17 00:00:00 2001 From: Hannes Eder Date: Wed, 25 Feb 2009 22:28:59 +0100 Subject: ALSA: sound/pci/emu10k1: fix sparse warning: different signedness Fix this sparse warnings: sound/pci/emu10k1/emu10k1_main.c:723:66: warning: incorrect type in argument 3 (different signedness) sound/pci/emu10k1/emu10k1_main.c:724:68: warning: incorrect type in argument 3 (different signedness) sound/pci/emu10k1/emu10k1_main.c:748:74: warning: incorrect type in argument 3 (different signedness) sound/pci/emu10k1/emu10k1_main.c:751:66: warning: incorrect type in argument 3 (different signedness) sound/pci/emu10k1/emu10k1_main.c:759:73: warning: incorrect type in argument 3 (different signedness) sound/pci/emu10k1/emu10k1_main.c:760:73: warning: incorrect type in argument 3 (different signedness) sound/pci/emu10k1/emu10k1_main.c:837:50: warning: incorrect type in argument 3 (different signedness) sound/pci/emu10k1/emu10k1_main.c:845:50: warning: incorrect type in argument 3 (different signedness) sound/pci/emu10k1/emu10k1_main.c:881:50: warning: incorrect type in argument 3 (different signedness) sound/pci/emu10k1/emu10k1_main.c:889:57: warning: incorrect type in argument 3 (different signedness) sound/pci/emu10k1/emu10k1_main.c:890:57: warning: incorrect type in argument 3 (different signedness) sound/pci/emu10k1/emu10k1_main.c:895:60: warning: incorrect type in argument 3 (different signedness) sound/pci/emu10k1/emu10k1_main.c:897:60: warning: incorrect type in argument 3 (different signedness) sound/pci/emu10k1/emu10k1_main.c:899:60: warning: incorrect type in argument 3 (different signedness) sound/pci/emu10k1/emu10k1_main.c:910:56: warning: incorrect type in argument 3 (different signedness) sound/pci/emu10k1/emu10k1_main.c:914:57: warning: incorrect type in argument 3 (different signedness) sound/pci/emu10k1/emu10k1_main.c:918:56: warning: incorrect type in argument 3 (different signedness) sound/pci/emu10k1/emu10k1_main.c:922:57: warning: incorrect type in argument 3 (different signedness) sound/pci/emu10k1/emu10k1_main.c:924:58: warning: incorrect type in argument 3 (different signedness) sound/pci/emu10k1/emu10k1_main.c:936:60: warning: incorrect type in argument 3 (different signedness) sound/pci/emu10k1/emu10k1_main.c:1073:60: warning: incorrect type in argument 3 (different signedness) sound/pci/emu10k1/emu10k1_main.c:1088:60: warning: incorrect type in argument 3 (different signedness) sound/pci/emu10k1/emu10k1_main.c:1093:58: warning: incorrect type in argument 3 (different signedness) Signed-off-by: Hannes Eder Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emu10k1_main.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 8343aecbd25..e6836fc3388 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -711,8 +711,7 @@ static int snd_emu1010_load_firmware(struct snd_emu10k1 *emu, const char *filena static int emu1010_firmware_thread(void *data) { struct snd_emu10k1 *emu = data; - int tmp, tmp2; - int reg; + u32 tmp, tmp2, reg; int err; for (;;) { @@ -758,7 +757,7 @@ static int emu1010_firmware_thread(void *data) snd_printk(KERN_INFO "emu1010: Audio Dock Firmware loaded\n"); snd_emu1010_fpga_read(emu, EMU_DOCK_MAJOR_REV, &tmp); snd_emu1010_fpga_read(emu, EMU_DOCK_MINOR_REV, &tmp2); - snd_printk(KERN_INFO "Audio Dock ver:%d.%d\n", + snd_printk(KERN_INFO "Audio Dock ver: %u.%u\n", tmp, tmp2); /* Sync clocking between 1010 and Dock */ /* Allow DLL to settle */ @@ -805,8 +804,7 @@ static int emu1010_firmware_thread(void *data) static int snd_emu10k1_emu1010_init(struct snd_emu10k1 *emu) { unsigned int i; - int tmp, tmp2; - int reg; + u32 tmp, tmp2, reg; int err; const char *filename = NULL; @@ -888,7 +886,7 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 *emu) snd_printk(KERN_INFO "emu1010: Hana Firmware loaded\n"); snd_emu1010_fpga_read(emu, EMU_HANA_MAJOR_REV, &tmp); snd_emu1010_fpga_read(emu, EMU_HANA_MINOR_REV, &tmp2); - snd_printk(KERN_INFO "emu1010: Hana version: %d.%d\n", tmp, tmp2); + snd_printk(KERN_INFO "emu1010: Hana version: %u.%u\n", tmp, tmp2); /* Enable 48Volt power to Audio Dock */ snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, EMU_HANA_DOCK_PWR_ON); -- cgit v1.2.3-70-g09d2 From 5d9b6c07831456b7a7d90eac31c853d60eaf8ab6 Mon Sep 17 00:00:00 2001 From: Hannes Eder Date: Wed, 25 Feb 2009 22:28:45 +0100 Subject: ALSA: sound/pci/hda: fix sparse warning: different signedness Fix this sparse warning: sound/pci/hda/hda_codec.c:1544:19: warning: incorrect type in assignment (different signedness) sound/pci/hda/hda_codec.c:1544:19: expected unsigned long *vals sound/pci/hda/hda_codec.c:1544:19: got long * Signed-off-by: Hannes Eder Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_local.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 4bd82a37a4c..03ee9dd0491 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -136,7 +136,7 @@ extern struct hda_ctl_ops snd_hda_bind_sw; /* for bind-switch */ struct hda_bind_ctls { struct hda_ctl_ops *ops; - long values[]; + unsigned long values[]; }; int snd_hda_mixer_bind_ctls_info(struct snd_kcontrol *kcontrol, -- cgit v1.2.3-70-g09d2 From 6d5643455ced9ee45a4aa7403fe8056d826bde85 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 26 Feb 2009 11:29:58 +0100 Subject: ASoC: wm8753 - Fix build error sound/soc/codecs/wm8753.c: In function 'wm8753_probe': sound/soc/codecs/wm8753.c:1577: error: implicit declaration of function 'wm8753_add_controls' Acked-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/codecs/wm8753.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 2241204b515..7f353e935d7 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1574,7 +1574,8 @@ static int wm8753_probe(struct platform_device *pdev) goto pcm_err; } - wm8753_add_controls(codec); + snd_soc_add_controls(codec, wm8753_snd_controls, + ARRAY_SIZE(wm8753_snd_controls)); wm8753_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { -- cgit v1.2.3-70-g09d2 From 23f0c048ba59ad5c2f3fd85ed98360b631dbf6f8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 26 Feb 2009 13:03:58 +0100 Subject: ALSA: hda - Clean up the input pin setup in automatic mode Clean up the input-pin setup in automatic mode in patch_realtek.c. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 62 ++++++++++++++++++------------------------- 1 file changed, 26 insertions(+), 36 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d670d33cfa1..b3406302d06 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -760,6 +760,24 @@ static int alc_eapd_ctrl_put(struct snd_kcontrol *kcontrol, .private_value = nid | (mask<<16) } #endif /* CONFIG_SND_DEBUG */ +/* + * set up the input pin config (depending on the given auto-pin type) + */ +static void alc_set_input_pin(struct hda_codec *codec, hda_nid_t nid, + int auto_pin_type) +{ + unsigned int val = PIN_IN; + + if (auto_pin_type <= AUTO_PIN_FRONT_MIC) { + unsigned int pincap; + pincap = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); + pincap = (pincap & AC_PINCAP_VREF) >> AC_PINCAP_VREF_SHIFT; + if (pincap & AC_PINCAP_VREF_80) + val = PIN_VREF80; + } + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, val); +} + /* */ static void add_mixer(struct alc_spec *spec, struct snd_kcontrol_new *mix) @@ -4188,10 +4206,7 @@ static void alc880_auto_init_analog_input(struct hda_codec *codec) for (i = 0; i < AUTO_PIN_LAST; i++) { hda_nid_t nid = spec->autocfg.input_pins[i]; if (alc880_is_input_pin(nid)) { - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - i <= AUTO_PIN_FRONT_MIC ? - PIN_VREF80 : PIN_IN); + alc_set_input_pin(codec, nid, i); if (nid != ALC880_PIN_CD_NID) snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, @@ -5657,10 +5672,7 @@ static void alc260_auto_init_analog_input(struct hda_codec *codec) for (i = 0; i < AUTO_PIN_LAST; i++) { hda_nid_t nid = spec->autocfg.input_pins[i]; if (nid >= 0x12) { - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - i <= AUTO_PIN_FRONT_MIC ? - PIN_VREF80 : PIN_IN); + alc_set_input_pin(codec, nid, i); if (nid != ALC260_PIN_CD_NID) snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, @@ -7006,16 +7018,7 @@ static void alc882_auto_init_analog_input(struct hda_codec *codec) unsigned int vref; if (!nid) continue; - vref = PIN_IN; - if (1 /*i <= AUTO_PIN_FRONT_MIC*/) { - unsigned int pincap; - pincap = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); - if ((pincap >> AC_PINCAP_VREF_SHIFT) & - AC_PINCAP_VREF_80) - vref = PIN_VREF80; - } - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, vref); + alc_set_input_pin(codec, nid, AUTO_PIN_FRONT_MIC /*i*/); if (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP) snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, @@ -9100,10 +9103,7 @@ static void alc883_auto_init_analog_input(struct hda_codec *codec) for (i = 0; i < AUTO_PIN_LAST; i++) { hda_nid_t nid = spec->autocfg.input_pins[i]; if (alc883_is_input_pin(nid)) { - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - (i <= AUTO_PIN_FRONT_MIC ? - PIN_VREF80 : PIN_IN)); + alc_set_input_pin(codec, nid, i); if (nid != ALC883_PIN_CD_NID) snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, @@ -13831,12 +13831,8 @@ static void alc861_auto_init_analog_input(struct hda_codec *codec) for (i = 0; i < AUTO_PIN_LAST; i++) { hda_nid_t nid = spec->autocfg.input_pins[i]; - if (nid >= 0x0c && nid <= 0x11) { - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - i <= AUTO_PIN_FRONT_MIC ? - PIN_VREF80 : PIN_IN); - } + if (nid >= 0x0c && nid <= 0x11) + alc_set_input_pin(codec, nid, i); } } @@ -14803,10 +14799,7 @@ static void alc861vd_auto_init_analog_input(struct hda_codec *codec) for (i = 0; i < AUTO_PIN_LAST; i++) { hda_nid_t nid = spec->autocfg.input_pins[i]; if (alc861vd_is_input_pin(nid)) { - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - i <= AUTO_PIN_FRONT_MIC ? - PIN_VREF80 : PIN_IN); + alc_set_input_pin(codec, nid, i); if (nid != ALC861VD_PIN_CD_NID) snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, @@ -16732,10 +16725,7 @@ static void alc662_auto_init_analog_input(struct hda_codec *codec) for (i = 0; i < AUTO_PIN_LAST; i++) { hda_nid_t nid = spec->autocfg.input_pins[i]; if (alc662_is_input_pin(nid)) { - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - (i <= AUTO_PIN_FRONT_MIC ? - PIN_VREF80 : PIN_IN)); + alc_set_input_pin(codec, nid, i); if (nid != ALC662_PIN_CD_NID) snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, -- cgit v1.2.3-70-g09d2 From 1607b8ea0a4cc20752978fadb027daafc8a2d93c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 26 Feb 2009 16:50:43 +0100 Subject: ALSA: hda - Add model=auto for STAC/IDT codecs Added the model=auto to STAC/IDT codecs to use the BIOS default setup explicitly. It can be used to disable the device-specific model quirk in the driver. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 8 ++++++++ sound/pci/hda/patch_sigmatel.c | 16 ++++++++++++++++ 2 files changed, 24 insertions(+) diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 0e52d273ce9..a448bbefd48 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -280,6 +280,7 @@ STAC9200 gateway-m4 Gateway laptops with EAPD control gateway-m4-2 Gateway laptops with EAPD control panasonic Panasonic CF-74 + auto BIOS setup (default) STAC9205/9254 ============= @@ -288,6 +289,7 @@ STAC9205/9254 dell-m43 Dell Precision dell-m44 Dell Inspiron eapd Keep EAPD on (e.g. Gateway T1616) + auto BIOS setup (default) STAC9220/9221 ============= @@ -311,6 +313,7 @@ STAC9220/9221 dell-d82 Dell (unknown) dell-m81 Dell (unknown) dell-m82 Dell XPS M1210 + auto BIOS setup (default) STAC9202/9250/9251 ================== @@ -322,6 +325,7 @@ STAC9202/9250/9251 m3 Some Gateway MX series laptops m5 Some Gateway MX series laptops (MP6954) m6 Some Gateway NX series laptops + auto BIOS setup (default) STAC9227/9228/9229/927x ======================= @@ -331,6 +335,7 @@ STAC9227/9228/9229/927x 5stack D965 5stack + SPDIF dell-3stack Dell Dimension E520 dell-bios Fixes with Dell BIOS setup + auto BIOS setup (default) STAC92HD71B* ============ @@ -339,6 +344,7 @@ STAC92HD71B* dell-m4-2 Dell desktops dell-m4-3 Dell desktops hp-m4 HP dv laptops + auto BIOS setup (default) STAC92HD73* =========== @@ -348,11 +354,13 @@ STAC92HD73* dell-m6-dmic Dell desktops/laptops with digital mics dell-m6 Dell desktops/laptops with both type of mics dell-eq Dell desktops/laptops + auto BIOS setup (default) STAC92HD83* =========== ref Reference board mic-ref Reference board with power managment for ports + auto BIOS setup (default) STAC9872 ======== diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index da48d8c0b29..37ffd96a9ff 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -43,6 +43,7 @@ enum { }; enum { + STAC_AUTO, STAC_REF, STAC_9200_OQO, STAC_9200_DELL_D21, @@ -62,6 +63,7 @@ enum { }; enum { + STAC_9205_AUTO, STAC_9205_REF, STAC_9205_DELL_M42, STAC_9205_DELL_M43, @@ -71,6 +73,7 @@ enum { }; enum { + STAC_92HD73XX_AUTO, STAC_92HD73XX_NO_JD, /* no jack-detection */ STAC_92HD73XX_REF, STAC_DELL_M6_AMIC, @@ -81,6 +84,7 @@ enum { }; enum { + STAC_92HD83XXX_AUTO, STAC_92HD83XXX_REF, STAC_92HD83XXX_PWR_REF, STAC_DELL_S14, @@ -88,6 +92,7 @@ enum { }; enum { + STAC_92HD71BXX_AUTO, STAC_92HD71BXX_REF, STAC_DELL_M4_1, STAC_DELL_M4_2, @@ -98,6 +103,7 @@ enum { }; enum { + STAC_925x_AUTO, STAC_925x_REF, STAC_M1, STAC_M1_2, @@ -110,6 +116,7 @@ enum { }; enum { + STAC_922X_AUTO, STAC_D945_REF, STAC_D945GTP3, STAC_D945GTP5, @@ -137,6 +144,7 @@ enum { }; enum { + STAC_927X_AUTO, STAC_D965_REF_NO_JD, /* no jack-detection */ STAC_D965_REF, STAC_D965_3ST, @@ -1488,6 +1496,7 @@ static unsigned int *stac9200_brd_tbl[STAC_9200_MODELS] = { }; static const char *stac9200_models[STAC_9200_MODELS] = { + [STAC_AUTO] = "auto", [STAC_REF] = "ref", [STAC_9200_OQO] = "oqo", [STAC_9200_DELL_D21] = "dell-d21", @@ -1633,6 +1642,7 @@ static unsigned int *stac925x_brd_tbl[STAC_925x_MODELS] = { }; static const char *stac925x_models[STAC_925x_MODELS] = { + [STAC_925x_AUTO] = "auto", [STAC_REF] = "ref", [STAC_M1] = "m1", [STAC_M1_2] = "m1-2", @@ -1692,6 +1702,7 @@ static unsigned int *stac92hd73xx_brd_tbl[STAC_92HD73XX_MODELS] = { }; static const char *stac92hd73xx_models[STAC_92HD73XX_MODELS] = { + [STAC_92HD73XX_AUTO] = "auto", [STAC_92HD73XX_NO_JD] = "no-jd", [STAC_92HD73XX_REF] = "ref", [STAC_DELL_M6_AMIC] = "dell-m6-amic", @@ -1748,6 +1759,7 @@ static unsigned int *stac92hd83xxx_brd_tbl[STAC_92HD83XXX_MODELS] = { }; static const char *stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = { + [STAC_92HD83XXX_AUTO] = "auto", [STAC_92HD83XXX_REF] = "ref", [STAC_92HD83XXX_PWR_REF] = "mic-ref", [STAC_DELL_S14] = "dell-s14", @@ -1802,6 +1814,7 @@ static unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = { }; static const char *stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = { + [STAC_92HD71BXX_AUTO] = "auto", [STAC_92HD71BXX_REF] = "ref", [STAC_DELL_M4_1] = "dell-m4-1", [STAC_DELL_M4_2] = "dell-m4-2", @@ -1973,6 +1986,7 @@ static unsigned int *stac922x_brd_tbl[STAC_922X_MODELS] = { }; static const char *stac922x_models[STAC_922X_MODELS] = { + [STAC_922X_AUTO] = "auto", [STAC_D945_REF] = "ref", [STAC_D945GTP5] = "5stack", [STAC_D945GTP3] = "3stack", @@ -2125,6 +2139,7 @@ static unsigned int *stac927x_brd_tbl[STAC_927X_MODELS] = { }; static const char *stac927x_models[STAC_927X_MODELS] = { + [STAC_927X_AUTO] = "auto", [STAC_D965_REF_NO_JD] = "ref-no-jd", [STAC_D965_REF] = "ref", [STAC_D965_3ST] = "3stack", @@ -2222,6 +2237,7 @@ static unsigned int *stac9205_brd_tbl[STAC_9205_MODELS] = { }; static const char *stac9205_models[STAC_9205_MODELS] = { + [STAC_9205_AUTO] = "auto", [STAC_9205_REF] = "ref", [STAC_9205_DELL_M42] = "dell-m42", [STAC_9205_DELL_M43] = "dell-m43", -- cgit v1.2.3-70-g09d2 From bedfcebb4fb33fc9ebd395462e72afa103db0bec Mon Sep 17 00:00:00 2001 From: peerchen Date: Fri, 27 Feb 2009 17:03:19 +0800 Subject: ALSA: hda - Add the Device IDs for MCP89 and remove IDs of MCP7B Added the Device IDs for MCP89 HD audio controller. Removed the IDs of MCP7B cause this chipset had been cancelled. Signed-off-by: Peer Chen Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index c5a5dc5698a..47a5833feb7 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2454,10 +2454,10 @@ static struct pci_device_id azx_ids[] = { { PCI_DEVICE(0x10de, 0x0ac1), .driver_data = AZX_DRIVER_NVIDIA }, { PCI_DEVICE(0x10de, 0x0ac2), .driver_data = AZX_DRIVER_NVIDIA }, { PCI_DEVICE(0x10de, 0x0ac3), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0bd4), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0bd5), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0bd6), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0bd7), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x0d94), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x0d95), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x0d96), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x0d97), .driver_data = AZX_DRIVER_NVIDIA }, /* Teradici */ { PCI_DEVICE(0x6549, 0x1200), .driver_data = AZX_DRIVER_TERA }, /* AMD Generic, PCI class code and Vendor ID for HD Audio */ -- cgit v1.2.3-70-g09d2 From 82af308f658cf2193e5058bbbfd37c3437cfb4e7 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 27 Feb 2009 09:27:44 +0100 Subject: sound: oxygen: zero-initialize model data Model drivers assume that model_data is zeroed, so we better use kzalloc() (like we did before when it was allocated together with the card structure). Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/oxygen_lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index 6e1cdd2fd76..312251d3969 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -566,7 +566,7 @@ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id, goto err_pci_regions; if (chip->model.model_data_size) { - chip->model_data = kmalloc(chip->model.model_data_size, + chip->model_data = kzalloc(chip->model.model_data_size, GFP_KERNEL); if (!chip->model_data) { err = -ENOMEM; -- cgit v1.2.3-70-g09d2 From 53eff7e1e0de1cde8e8cbe619f401d2578dde946 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 27 Feb 2009 17:49:44 +0100 Subject: ALSA: hda - Match all 103c:17xx devices for HP BPC model Use SND_PCI_QUIRK_MASK() to match all devices with 103c:17xx for HP BPC model. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e72b74efc69..0b4afa0a351 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10807,7 +10807,8 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { ALC262_HP_BPC), SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1300, "HP xw series", ALC262_HP_BPC), - SND_PCI_QUIRK(0x103c, 0x170b, "HP xw*", ALC262_HP_BPC), + SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1700, "HP xw series", + ALC262_HP_BPC), SND_PCI_QUIRK(0x103c, 0x2800, "HP D7000", ALC262_HP_BPC_D7000_WL), SND_PCI_QUIRK(0x103c, 0x2801, "HP D7000", ALC262_HP_BPC_D7000_WF), SND_PCI_QUIRK(0x103c, 0x2802, "HP D7000", ALC262_HP_BPC_D7000_WL), -- cgit v1.2.3-70-g09d2 From c82c8abdeef53eb0bb0504becb4e91bbccceaee8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 27 Feb 2009 17:52:22 +0100 Subject: ALSA: hda - Fix an "unused variable" compile warning MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Forgot to remove an unused variable. sound/pci/hda/patch_realtek.c: In function ‘alc882_auto_init_analog_input’: sound/pci/hda/patch_realtek.c:7018: warning: unused variable ‘vref’ Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0b4afa0a351..1cc31ac0352 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7015,7 +7015,6 @@ static void alc882_auto_init_analog_input(struct hda_codec *codec) for (i = 0; i < AUTO_PIN_LAST; i++) { hda_nid_t nid = spec->autocfg.input_pins[i]; - unsigned int vref; if (!nid) continue; alc_set_input_pin(codec, nid, AUTO_PIN_FRONT_MIC /*i*/); -- cgit v1.2.3-70-g09d2 From c8efef1745d168b80c800e98cce48a59630dbbfc Mon Sep 17 00:00:00 2001 From: Ben Dooks Date: Sat, 28 Feb 2009 17:09:57 +0000 Subject: ASoC: Fix copyright statements on Simtec files Fix the copyright statements in two of the S3C24XX ASoC files that have (c) when we require the full word. Signed-off-by: Ben Dooks Signed-off-by: Mark Brown --- sound/soc/s3c24xx/s3c24xx-i2s.c | 2 +- sound/soc/s3c24xx/s3c24xx-pcm.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 6f4d439b57a..1c2b0549710 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -4,7 +4,7 @@ * (c) 2006 Wolfson Microelectronics PLC. * Graeme Gregory graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com * - * (c) 2004-2005 Simtec Electronics + * Copyright 2004-2005 Simtec Electronics * http://armlinux.simtec.co.uk/ * Ben Dooks * diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c index 7c64d31d067..ba1ae09dfae 100644 --- a/sound/soc/s3c24xx/s3c24xx-pcm.c +++ b/sound/soc/s3c24xx/s3c24xx-pcm.c @@ -4,7 +4,7 @@ * (c) 2006 Wolfson Microelectronics PLC. * Graeme Gregory graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com * - * (c) 2004-2005 Simtec Electronics + * Copyright 2004-2005 Simtec Electronics * http://armlinux.simtec.co.uk/ * Ben Dooks * -- cgit v1.2.3-70-g09d2 From 4eae080dda3a563160be2f642cfbda27ffc42178 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 25 Feb 2009 14:37:21 +0100 Subject: ASoC: Add cs4270 support for slave mode configurations Added support for scenarios where the Cirrus CS4270 audio codec is slave to the bitclk and lrclk. Mixed setups are unsupported. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- sound/soc/codecs/cs4270.c | 24 +++++++++++++++++++++++- 1 file changed, 23 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index cd4a9ee38e4..339e0f6b0fe 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -109,6 +109,7 @@ struct cs4270_private { u8 reg_cache[CS4270_NUMREGS]; unsigned int mclk; /* Input frequency of the MCLK pin */ unsigned int mode; /* The mode (I2S or left-justified) */ + unsigned int slave_mode; }; /** @@ -247,6 +248,7 @@ static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai, struct cs4270_private *cs4270 = codec->private_data; int ret = 0; + /* set DAI format */ switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: case SND_SOC_DAIFMT_LEFT_J: @@ -257,6 +259,21 @@ static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai, ret = -EINVAL; } + /* set master/slave audio interface */ + switch (format & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + cs4270->slave_mode = 1; + break; + case SND_SOC_DAIFMT_CBM_CFM: + cs4270->slave_mode = 0; + break; + case SND_SOC_DAIFMT_CBM_CFS: + /* unsupported - cs4270 can eigther be slave or master to + * both the bitclk and the lrclk. */ + default: + ret = -EINVAL; + } + return ret; } @@ -399,7 +416,12 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, reg = snd_soc_read(codec, CS4270_MODE); reg &= ~(CS4270_MODE_SPEED_MASK | CS4270_MODE_DIV_MASK); - reg |= cs4270_mode_ratios[i].speed_mode | cs4270_mode_ratios[i].mclk; + reg |= cs4270_mode_ratios[i].mclk; + + if (cs4270->slave_mode) + reg |= CS4270_MODE_SLAVE; + else + reg |= cs4270_mode_ratios[i].speed_mode; ret = snd_soc_write(codec, CS4270_MODE, reg); if (ret < 0) { -- cgit v1.2.3-70-g09d2 From 8b37dbd2a180667e51db0552383df18743239c25 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 28 Feb 2009 21:14:20 +0000 Subject: ASoC: Add SND_SOC_DAPM_PIN_SWITCH controls for exposing DAPM pins On some systems it is desirable for control for DAPM pins to be provided to user space. This is the case with things like GSM modems which are controlled primarily from user space, for example. Provide a helper which exposes the state of a DAPM pin to user space for use in cases like this. Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 12 +++++++++ sound/soc/soc-dapm.c | 70 ++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 82 insertions(+) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index bb3a863ad14..a7def6a9a03 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -192,6 +192,12 @@ .get = snd_soc_dapm_get_value_enum_double, \ .put = snd_soc_dapm_put_value_enum_double, \ .private_value = (unsigned long)&xenum } +#define SOC_DAPM_PIN_SWITCH(xname) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname " Switch", \ + .info = snd_soc_dapm_info_pin_switch, \ + .get = snd_soc_dapm_get_pin_switch, \ + .put = snd_soc_dapm_put_pin_switch, \ + .private_value = (unsigned long)xname } /* dapm stream operations */ #define SND_SOC_DAPM_STREAM_NOP 0x0 @@ -238,6 +244,12 @@ int snd_soc_dapm_get_value_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +int snd_soc_dapm_info_pin_switch(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo); +int snd_soc_dapm_get_pin_switch(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *uncontrol); +int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *uncontrol); int snd_soc_dapm_new_control(struct snd_soc_codec *codec, const struct snd_soc_dapm_widget *widget); int snd_soc_dapm_new_controls(struct snd_soc_codec *codec, diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index f4a8753c84c..4b8dbbfe2ef 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1420,6 +1420,76 @@ out: } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_value_enum_double); +/** + * snd_soc_dapm_info_pin_switch - Info for a pin switch + * + * @kcontrol: mixer control + * @uinfo: control element information + * + * Callback to provide information about a pin switch control. + */ +int snd_soc_dapm_info_pin_switch(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_info_pin_switch); + +/** + * snd_soc_dapm_get_pin_switch - Get information for a pin switch + * + * @kcontrol: mixer control + * @ucontrol: Value + */ +int snd_soc_dapm_get_pin_switch(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + const char *pin = (const char *)kcontrol->private_value; + + mutex_lock(&codec->mutex); + + ucontrol->value.integer.value[0] = + snd_soc_dapm_get_pin_status(codec, pin); + + mutex_unlock(&codec->mutex); + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_get_pin_switch); + +/** + * snd_soc_dapm_put_pin_switch - Set information for a pin switch + * + * @kcontrol: mixer control + * @ucontrol: Value + */ +int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + const char *pin = (const char *)kcontrol->private_value; + + mutex_lock(&codec->mutex); + + if (ucontrol->value.integer.value[0]) + snd_soc_dapm_enable_pin(codec, pin); + else + snd_soc_dapm_disable_pin(codec, pin); + + snd_soc_dapm_sync(codec); + + mutex_unlock(&codec->mutex); + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_put_pin_switch); + /** * snd_soc_dapm_new_control - create new dapm control * @codec: audio codec -- cgit v1.2.3-70-g09d2 From 892981ffbe9a5c4cbc9d75f423b145f32c765f9c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 2 Mar 2009 08:04:35 +0100 Subject: ALSA: hda - Don't create a beep control for digital-only ALC268 When an ALC268 codec is set up as the digital-only (as found in Toshiba laptops), it shouldn't contain any beep control that conflict with the primary codec. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1cc31ac0352..c60c86acd9b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -11915,7 +11915,7 @@ static int alc268_parse_auto_config(struct hda_codec *codec) if (spec->kctls.list) add_mixer(spec, spec->kctls.list); - if (spec->autocfg.speaker_pins[0] != 0x1d) + if (!spec->no_analog && spec->autocfg.speaker_pins[0] != 0x1d) add_mixer(spec, alc268_beep_mixer); add_verb(spec, alc268_volume_init_verbs); -- cgit v1.2.3-70-g09d2 From 4c4531d64dd0442813c7307b860bf40a2aec51bc Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 2 Mar 2009 08:06:11 +0100 Subject: ALSA: hda - Remove Toshiba probe_mask quirk Revert the Toshiba probe_mask quirk for 2.6.29 kernel (commit 38f1df27e3191d76e983cb9c6b4392582fd32fda). In the current tree, the digital-only codec is handled properly so no codec conflict should occur. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 68a128fb487..47a5833feb7 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2095,8 +2095,6 @@ static struct snd_pci_quirk probe_mask_list[] __devinitdata = { SND_PCI_QUIRK(0x1028, 0x20ac, "Dell Studio Desktop", 0x01), /* including bogus ALC268 in slot#2 that conflicts with ALC888 */ SND_PCI_QUIRK(0x17c0, 0x4085, "Medion MD96630", 0x01), - /* conflict of ALC268 in slot#3 (digital I/O); a temporary fix */ - SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba laptop", 0x03), /* forced codec slots */ SND_PCI_QUIRK(0x1046, 0x1262, "ASUS W5F", 0x103), {} -- cgit v1.2.3-70-g09d2 From d1f1af2dbf8207db590853a59bec465c4f68cfdc Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 2 Mar 2009 10:35:29 +0100 Subject: ALSA: hda - Intialize more codec fields in snd_hda_codec_reset() Initiailize forgotten fields in snd_hda_codec_reset(). Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 5dceee8a113..3b44c789f23 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1519,6 +1519,9 @@ int snd_hda_codec_reset(struct hda_codec *codec) codec->num_pcms = 0; codec->pcm_info = NULL; codec->preset = NULL; + memset(&codec->patch_ops, 0, sizeof(codec->patch_ops)); + codec->slave_dig_outs = NULL; + codec->spdif_status_reset = 0; module_put(codec->owner); codec->owner = NULL; -- cgit v1.2.3-70-g09d2 From f93d461bcde6ac3db542361c00a7e4167f88176d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 2 Mar 2009 10:44:15 +0100 Subject: ALSA: hda - Revert the codec probe at control-creation errors Revert the codec probe instead of returning the error to the driver when any error occurs at creating the control elements. The control element conflict can be non-fatal in many cases, especially if it comes from the digital-only codec. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 3b44c789f23..1be34ed9c0e 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1434,7 +1434,6 @@ int snd_hda_ctl_add(struct hda_codec *codec, struct snd_kcontrol *kctl) } EXPORT_SYMBOL_HDA(snd_hda_ctl_add); -#ifdef CONFIG_SND_HDA_RECONFIG /* Clear all controls assigned to the given codec */ void snd_hda_ctls_clear(struct hda_codec *codec) { @@ -1529,7 +1528,6 @@ int snd_hda_codec_reset(struct hda_codec *codec) hda_unlock_devices(card); return 0; } -#endif /* CONFIG_SND_HDA_RECONFIG */ /* create a virtual master control and add slaves */ int snd_hda_add_vmaster(struct hda_codec *codec, char *name, @@ -2392,8 +2390,16 @@ int /*__devinit*/ snd_hda_build_controls(struct hda_bus *bus) list_for_each_entry(codec, &bus->codec_list, list) { int err = snd_hda_codec_build_controls(codec); - if (err < 0) - return err; + if (err < 0) { + printk(KERN_ERR "hda_codec: cannot build controls" + "for #%d (error %d)\n", codec->addr, err); + err = snd_hda_codec_reset(codec); + if (err < 0) { + printk(KERN_ERR + "hda_codec: cannot revert codec\n"); + return err; + } + } } return 0; } -- cgit v1.2.3-70-g09d2 From 6e655bf21697d2594243098a14e0699e8d4a4059 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 2 Mar 2009 10:46:03 +0100 Subject: ALSA: hda - Don't return a fatal error at PCM-creation errors Don't return a fatal error to the driver but continue to probe when any error occurs at creating PCM streams for each codec. It's often non-fatal and keeping it would help debugging. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 22 +++++++++++++++++----- 1 file changed, 17 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 1be34ed9c0e..7c9ef5c18e7 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2833,8 +2833,16 @@ int snd_hda_codec_build_pcms(struct hda_codec *codec) if (!codec->patch_ops.build_pcms) return 0; err = codec->patch_ops.build_pcms(codec); - if (err < 0) - return err; + if (err < 0) { + printk(KERN_ERR "hda_codec: cannot build PCMs" + "for #%d (error %d)\n", codec->addr, err); + err = snd_hda_codec_reset(codec); + if (err < 0) { + printk(KERN_ERR + "hda_codec: cannot revert codec\n"); + return err; + } + } } for (pcm = 0; pcm < codec->num_pcms; pcm++) { struct hda_pcm *cpcm = &codec->pcm_info[pcm]; @@ -2846,11 +2854,15 @@ int snd_hda_codec_build_pcms(struct hda_codec *codec) if (!cpcm->pcm) { dev = get_empty_pcm_device(codec->bus, cpcm->pcm_type); if (dev < 0) - return 0; + continue; /* no fatal error */ cpcm->device = dev; err = snd_hda_attach_pcm(codec, cpcm); - if (err < 0) - return err; + if (err < 0) { + printk(KERN_ERR "hda_codec: cannot attach " + "PCM stream %d for codec #%d\n", + dev, codec->addr); + continue; /* no fatal error */ + } } } return 0; -- cgit v1.2.3-70-g09d2 From 1713c0d508fbbb42aa5f90039195e5ac31a50625 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Fri, 27 Feb 2009 21:41:40 +0100 Subject: ALSA: opl3sa2 fix irq releasing and short name of card Two simple fixes: 1. Use the same pointer for the free_irq() and the request_irq() calls. 2. A short name of card is appended with '2' or '3' character depending on a detected chip. Remove the '2' character from the short name. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/opl3sa2.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/isa/opl3sa2.c b/sound/isa/opl3sa2.c index 06810dfb9d9..19b2d0420a2 100644 --- a/sound/isa/opl3sa2.c +++ b/sound/isa/opl3sa2.c @@ -617,7 +617,7 @@ static void snd_opl3sa2_free(struct snd_card *card) { struct snd_opl3sa2 *chip = card->private_data; if (chip->irq >= 0) - free_irq(chip->irq, (void *)chip); + free_irq(chip->irq, card); release_and_free_resource(chip->res_port); } @@ -630,7 +630,7 @@ static struct snd_card *snd_opl3sa2_card_new(int dev) if (card == NULL) return NULL; strcpy(card->driver, "OPL3SA2"); - strcpy(card->shortname, "Yamaha OPL3-SA2"); + strcpy(card->shortname, "Yamaha OPL3-SA"); chip = card->private_data; spin_lock_init(&chip->reg_lock); chip->irq = -1; -- cgit v1.2.3-70-g09d2 From eab2b553c3d3ed20698c4a9c7e049a60b804e2f5 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 2 Mar 2009 11:45:50 +0100 Subject: sound: usb-audio: fix rules check for 32-channel devices When storing the channel numbers used by a format, and if the device happens to support 32 channels, the code would try to store 1<<32 in a 32-bit value. Since no valid format can have zero channels, we can use 1<<(channels-1) instead of 1< Signed-off-by: Takashi Iwai --- sound/usb/usbaudio.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 2b24496ddec..f853b627cf4 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -1783,7 +1783,7 @@ static int check_hw_params_convention(struct snd_usb_substream *subs) if (rates[f->format] && rates[f->format] != f->rates) goto __out; } - channels[f->format] |= (1 << f->channels); + channels[f->format] |= 1 << (f->channels - 1); rates[f->format] |= f->rates; /* needs knot? */ if (f->rates & SNDRV_PCM_RATE_KNOT) @@ -1810,7 +1810,7 @@ static int check_hw_params_convention(struct snd_usb_substream *subs) continue; for (i = 0; i < 32; i++) { if (f->rates & (1 << i)) - channels[i] |= (1 << f->channels); + channels[i] |= 1 << (f->channels - 1); } } cmaster = 0; -- cgit v1.2.3-70-g09d2 From b1c86bb807448701400abc6eb8e958475ab5424b Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 2 Mar 2009 12:06:28 +0100 Subject: sound: usb-audio: fix queue length check for high speed devices When checking for the maximum queue length, we have to take into account that MAX_QUEUE is measured in milliseconds (i.e., frames) while the unit of urb_packs is whatever data packet interval the device uses (possibly less than one frame when using high speed devices). Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/usb/usbaudio.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index f853b627cf4..defe9913cbb 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -1095,9 +1095,8 @@ static int init_substream_urbs(struct snd_usb_substream *subs, unsigned int peri total_packs = 2 * packs_per_ms; } else { /* and we don't want too long a queue either */ - maxpacks = max((unsigned int)MAX_QUEUE, urb_packs * 2); - if (total_packs > maxpacks * packs_per_ms) - total_packs = maxpacks * packs_per_ms; + maxpacks = max(MAX_QUEUE * packs_per_ms, urb_packs * 2); + total_packs = min(total_packs, maxpacks); } } else { total_packs = MAX_URBS * urb_packs; -- cgit v1.2.3-70-g09d2 From ff09d49ad0176a5f52a398c137a7ff5f669d6be4 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Sat, 28 Feb 2009 13:21:03 +0100 Subject: ASoC: fix typo and removed unneeded switch case for cs4270 This removes a misspelled comment and got rid of superfluous switch case. Signed-off-by: Daniel Mack Acked-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/codecs/cs4270.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 339e0f6b0fe..f86f33cc179 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -267,10 +267,8 @@ static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai, case SND_SOC_DAIFMT_CBM_CFM: cs4270->slave_mode = 0; break; - case SND_SOC_DAIFMT_CBM_CFS: - /* unsupported - cs4270 can eigther be slave or master to - * both the bitclk and the lrclk. */ default: + /* all other modes are unsupported by the hardware */ ret = -EINVAL; } -- cgit v1.2.3-70-g09d2 From 43b62713f67d9f0655f3a61f5bd14d6297ddd3ce Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 2 Mar 2009 14:25:17 +0100 Subject: ALSA: hda - Add hint string helper functions Added snd_hda_get_hint() and snd_hda_get_bool_hint() helper functions to retrieve a hint value. Internally, the hint is stored in a pair of two strings, key and val. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_hwdep.c | 112 ++++++++++++++++++++++++++++++++++++++++------ sound/pci/hda/hda_local.h | 17 +++++++ 2 files changed, 116 insertions(+), 13 deletions(-) diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index 4af484b8240..5e554de9cd9 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -30,6 +30,12 @@ #include #include +/* hint string pair */ +struct hda_hint { + const char *key; + const char *val; /* contained in the same alloc as key */ +}; + /* * write/read an out-of-bound verb */ @@ -99,15 +105,15 @@ static int hda_hwdep_open(struct snd_hwdep *hw, struct file *file) static void clear_hwdep_elements(struct hda_codec *codec) { - char **head; int i; /* clear init verbs */ snd_array_free(&codec->init_verbs); /* clear hints */ - head = codec->hints.list; - for (i = 0; i < codec->hints.used; i++, head++) - kfree(*head); + for (i = 0; i < codec->hints.used; i++) { + struct hda_hint *hint = snd_array_elem(&codec->hints, i); + kfree(hint->key); /* we don't need to free hint->val */ + } snd_array_free(&codec->hints); snd_array_free(&codec->user_pins); } @@ -141,7 +147,7 @@ int /*__devinit*/ snd_hda_create_hwdep(struct hda_codec *codec) #endif snd_array_init(&codec->init_verbs, sizeof(struct hda_verb), 32); - snd_array_init(&codec->hints, sizeof(char *), 32); + snd_array_init(&codec->hints, sizeof(struct hda_hint), 32); snd_array_init(&codec->user_pins, sizeof(struct hda_pincfg), 16); return 0; @@ -306,26 +312,81 @@ static ssize_t init_verbs_store(struct device *dev, return count; } +static struct hda_hint *get_hint(struct hda_codec *codec, const char *key) +{ + int i; + + for (i = 0; i < codec->hints.used; i++) { + struct hda_hint *hint = snd_array_elem(&codec->hints, i); + if (!strcmp(hint->key, key)) + return hint; + } + return NULL; +} + +static void remove_trail_spaces(char *str) +{ + char *p; + if (!*str) + return; + p = str + strlen(str) - 1; + for (; isspace(*p); p--) { + *p = 0; + if (p == str) + return; + } +} + +#define MAX_HINTS 1024 + static ssize_t hints_store(struct device *dev, struct device_attribute *attr, const char *buf, size_t count) { struct snd_hwdep *hwdep = dev_get_drvdata(dev); struct hda_codec *codec = hwdep->private_data; - char *p; - char **hint; + char *key, *val; + struct hda_hint *hint; - if (!*buf || isspace(*buf) || *buf == '#' || *buf == '\n') + while (isspace(*buf)) + buf++; + if (!*buf || *buf == '#' || *buf == '\n') return count; - p = kstrndup_noeol(buf, 1024); - if (!p) + if (*buf == '=') + return -EINVAL; + key = kstrndup_noeol(buf, 1024); + if (!key) return -ENOMEM; - hint = snd_array_new(&codec->hints); + /* extract key and val */ + val = strchr(key, '='); + if (!val) { + kfree(key); + return -EINVAL; + } + *val++ = 0; + while (isspace(*val)) + val++; + remove_trail_spaces(key); + remove_trail_spaces(val); + hint = get_hint(codec, key); + if (hint) { + /* replace */ + kfree(hint->key); + hint->key = key; + hint->val = val; + return count; + } + /* allocate a new hint entry */ + if (codec->hints.used >= MAX_HINTS) + hint = NULL; + else + hint = snd_array_new(&codec->hints); if (!hint) { - kfree(p); + kfree(key); return -ENOMEM; } - *hint = p; + hint->key = key; + hint->val = val; return count; } @@ -428,4 +489,29 @@ int snd_hda_hwdep_add_sysfs(struct hda_codec *codec) return 0; } +/* + * Look for hint string + */ +const char *snd_hda_get_hint(struct hda_codec *codec, const char *key) +{ + struct hda_hint *hint = get_hint(codec, key); + return hint ? hint->val : NULL; +} +EXPORT_SYMBOL_HDA(snd_hda_get_hint); + +int snd_hda_get_bool_hint(struct hda_codec *codec, const char *key) +{ + const char *p = snd_hda_get_hint(codec, key); + if (!p || !*p) + return -ENOENT; + switch (toupper(*p)) { + case 'T': /* true */ + case 'Y': /* yes */ + case '1': + return 1; + } + return 0; +} +EXPORT_SYMBOL_HDA(snd_hda_get_bool_hint); + #endif /* CONFIG_SND_HDA_RECONFIG */ diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 03ee9dd0491..27428c718fd 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -433,6 +433,23 @@ static inline int snd_hda_hwdep_add_sysfs(struct hda_codec *codec) } #endif +#ifdef CONFIG_SND_HDA_RECONFIG +const char *snd_hda_get_hint(struct hda_codec *codec, const char *key); +int snd_hda_get_bool_hint(struct hda_codec *codec, const char *key); +#else +static inline +const char *snd_hda_get_hint(struct hda_codec *codec, const char *key) +{ + return NULL; +} + +static inline +int snd_hda_get_bool_hint(struct hda_codec *codec, const char *key) +{ + return -ENOENT; +} +#endif + /* * power-management */ -- cgit v1.2.3-70-g09d2 From ab1726f920275b52991b2eff7538ac6d313bf9a2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 2 Mar 2009 17:09:25 +0100 Subject: ALSA: hda - Add show for init_verbs and hints sysfs entries Added the show method for init_verbs and hints hwdep sysfs entries. They show the current values. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_hwdep.c | 35 +++++++++++++++++++++++++++++++++-- 1 file changed, 33 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index 5e554de9cd9..1e3ccc740af 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -290,6 +290,22 @@ static ssize_t type##_store(struct device *dev, \ CODEC_ACTION_STORE(reconfig); CODEC_ACTION_STORE(clear); +static ssize_t init_verbs_show(struct device *dev, + struct device_attribute *attr, + char *buf) +{ + struct snd_hwdep *hwdep = dev_get_drvdata(dev); + struct hda_codec *codec = hwdep->private_data; + int i, len = 0; + for (i = 0; i < codec->init_verbs.used; i++) { + struct hda_verb *v = snd_array_elem(&codec->init_verbs, i); + len += snprintf(buf + len, PAGE_SIZE - len, + "0x%02x 0x%03x 0x%04x\n", + v->nid, v->verb, v->param); + } + return len; +} + static ssize_t init_verbs_store(struct device *dev, struct device_attribute *attr, const char *buf, size_t count) @@ -312,6 +328,21 @@ static ssize_t init_verbs_store(struct device *dev, return count; } +static ssize_t hints_show(struct device *dev, + struct device_attribute *attr, + char *buf) +{ + struct snd_hwdep *hwdep = dev_get_drvdata(dev); + struct hda_codec *codec = hwdep->private_data; + int i, len = 0; + for (i = 0; i < codec->hints.used; i++) { + struct hda_hint *hint = snd_array_elem(&codec->hints, i); + len += snprintf(buf + len, PAGE_SIZE - len, + "%s = %s\n", hint->key, hint->val); + } + return len; +} + static struct hda_hint *get_hint(struct hda_codec *codec, const char *key) { int i; @@ -466,8 +497,8 @@ static struct device_attribute codec_attrs[] = { CODEC_ATTR_RO(mfg), CODEC_ATTR_RW(name), CODEC_ATTR_RW(modelname), - CODEC_ATTR_WO(init_verbs), - CODEC_ATTR_WO(hints), + CODEC_ATTR_RW(init_verbs), + CODEC_ATTR_RW(hints), CODEC_ATTR_RO(init_pin_configs), CODEC_ATTR_RW(user_pin_configs), CODEC_ATTR_RO(driver_pin_configs), -- cgit v1.2.3-70-g09d2 From d78d7a90adf793943cc29a414b6f4364a700aad5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 2 Mar 2009 14:26:25 +0100 Subject: ALSA: hda - Create "Analog Loopback" controls optionally Don't create "Analog Loopback" controls as default since these controls are usually more harmful than useful for normal users. Only created when "loopback = yes" hint is given. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 56 ++++++++++++++++++++++++++++++++---------- 1 file changed, 43 insertions(+), 13 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 13056429aa6..7381325b98f 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -190,6 +190,7 @@ struct sigmatel_spec { unsigned int stream_delay; /* analog loopback */ + struct snd_kcontrol_new *aloopback_ctl; unsigned char aloopback_mask; unsigned char aloopback_shift; @@ -1013,8 +1014,6 @@ static struct snd_kcontrol_new stac92hd73xx_6ch_mixer[] = { HDA_CODEC_VOLUME("DAC Mixer Capture Volume", 0x1d, 0x3, HDA_INPUT), HDA_CODEC_MUTE("DAC Mixer Capture Switch", 0x1d, 0x3, HDA_INPUT), - STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 3), - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), @@ -1024,9 +1023,22 @@ static struct snd_kcontrol_new stac92hd73xx_6ch_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new stac92hd73xx_8ch_mixer[] = { +static struct snd_kcontrol_new stac92hd73xx_6ch_loopback[] = { + STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 3), + {} +}; + +static struct snd_kcontrol_new stac92hd73xx_8ch_loopback[] = { STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 4), + {} +}; +static struct snd_kcontrol_new stac92hd73xx_10ch_loopback[] = { + STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 5), + {} +}; + +static struct snd_kcontrol_new stac92hd73xx_8ch_mixer[] = { HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), @@ -1051,8 +1063,6 @@ static struct snd_kcontrol_new stac92hd73xx_8ch_mixer[] = { }; static struct snd_kcontrol_new stac92hd73xx_10ch_mixer[] = { - STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 5), - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), @@ -1104,8 +1114,6 @@ static struct snd_kcontrol_new stac92hd83xxx_mixer[] = { }; static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { - STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2), - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), @@ -1131,9 +1139,11 @@ static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new stac92hd71bxx_mixer[] = { - STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2), +static struct snd_kcontrol_new stac92hd71bxx_loopback[] = { + STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2) +}; +static struct snd_kcontrol_new stac92hd71bxx_mixer[] = { HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), @@ -1151,8 +1161,6 @@ static struct snd_kcontrol_new stac925x_mixer[] = { }; static struct snd_kcontrol_new stac9205_mixer[] = { - STAC_ANALOG_LOOPBACK(0xFE0, 0x7E0, 1), - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1b, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1d, 0x0, HDA_OUTPUT), @@ -1161,6 +1169,11 @@ static struct snd_kcontrol_new stac9205_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new stac9205_loopback[] = { + STAC_ANALOG_LOOPBACK(0xFE0, 0x7E0, 1), + {} +}; + /* This needs to be generated dynamically based on sequence */ static struct snd_kcontrol_new stac922x_mixer[] = { HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x17, 0x0, HDA_INPUT), @@ -1173,8 +1186,6 @@ static struct snd_kcontrol_new stac922x_mixer[] = { static struct snd_kcontrol_new stac927x_mixer[] = { - STAC_ANALOG_LOOPBACK(0xFEB, 0x7EB, 1), - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x18, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1b, 0x0, HDA_OUTPUT), @@ -1186,6 +1197,11 @@ static struct snd_kcontrol_new stac927x_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new stac927x_loopback[] = { + STAC_ANALOG_LOOPBACK(0xFEB, 0x7EB, 1), + {} +}; + static struct snd_kcontrol_new stac_dmux_mixer = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Digital Input Source", @@ -1312,6 +1328,13 @@ static int stac92xx_build_controls(struct hda_codec *codec) return err; } + if (spec->aloopback_ctl && + snd_hda_get_bool_hint(codec, "loopback") == 1) { + err = snd_hda_add_new_ctls(codec, spec->aloopback_ctl); + if (err < 0) + return err; + } + stac92xx_free_kctls(codec); /* no longer needed */ /* create jack input elements */ @@ -4618,14 +4641,18 @@ again: case 0x3: /* 6 Channel */ spec->mixer = stac92hd73xx_6ch_mixer; spec->init = stac92hd73xx_6ch_core_init; + spec->aloopback_ctl = stac92hd73xx_6ch_loopback; break; case 0x4: /* 8 Channel */ spec->mixer = stac92hd73xx_8ch_mixer; spec->init = stac92hd73xx_8ch_core_init; + spec->aloopback_ctl = stac92hd73xx_8ch_loopback; break; case 0x5: /* 10 Channel */ spec->mixer = stac92hd73xx_10ch_mixer; spec->init = stac92hd73xx_10ch_core_init; + spec->aloopback_ctl = stac92hd73xx_10ch_loopback; + break; } spec->multiout.dac_nids = spec->dac_nids; @@ -5036,6 +5063,7 @@ again: if (get_wcaps(codec, 0xa) & AC_WCAP_IN_AMP) snd_hda_sequence_write_cache(codec, unmute_init); + spec->aloopback_ctl = stac92hd71bxx_loopback; spec->aloopback_mask = 0x50; spec->aloopback_shift = 0; @@ -5285,6 +5313,7 @@ static int patch_stac927x(struct hda_codec *codec) } spec->num_pwrs = 0; + spec->aloopback_ctl = stac927x_loopback; spec->aloopback_mask = 0x40; spec->aloopback_shift = 0; spec->eapd_switch = 1; @@ -5364,6 +5393,7 @@ static int patch_stac9205(struct hda_codec *codec) spec->init = stac9205_core_init; spec->mixer = stac9205_mixer; + spec->aloopback_ctl = stac9205_loopback; spec->aloopback_mask = 0x40; spec->aloopback_shift = 0; -- cgit v1.2.3-70-g09d2 From 6565e4faca257fc51a4c55199d72e2701ba7e819 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 2 Mar 2009 14:38:35 +0100 Subject: ALSA: hda - Add more hint options for IDT/Sigmatel codecs Allow more options to be set/reset via hwdep hint entry. hp_detect, gpio_mask, gpio_dir, gpio_data, eapd_mask and eapd_switch can be checked. For example, to disable hp_detect on the fly, # echo "hp_detect=0" > /sys/class/sound/hwC0D0/hints Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 33 +++++++++++++++++++++++++++++++++ 1 file changed, 33 insertions(+) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 7381325b98f..e9331561a48 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -3949,6 +3949,36 @@ static void stac92xx_power_down(struct hda_codec *codec) static void stac_toggle_power_map(struct hda_codec *codec, hda_nid_t nid, int enable); +/* override some hints from the hwdep entry */ +static void stac_store_hints(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + const char *p; + int val; + + val = snd_hda_get_bool_hint(codec, "hp_detect"); + if (val >= 0) + spec->hp_detect = val; + p = snd_hda_get_hint(codec, "gpio_mask"); + if (p) { + spec->gpio_mask = simple_strtoul(p, NULL, 0); + spec->eapd_mask = spec->gpio_dir = spec->gpio_data = + spec->gpio_mask; + } + p = snd_hda_get_hint(codec, "gpio_dir"); + if (p) + spec->gpio_dir = simple_strtoul(p, NULL, 0) & spec->gpio_mask; + p = snd_hda_get_hint(codec, "gpio_data"); + if (p) + spec->gpio_data = simple_strtoul(p, NULL, 0) & spec->gpio_mask; + p = snd_hda_get_hint(codec, "eapd_mask"); + if (p) + spec->eapd_mask = simple_strtoul(p, NULL, 0) & spec->gpio_mask; + val = snd_hda_get_bool_hint(codec, "eapd_switch"); + if (val >= 0) + spec->eapd_switch = val; +} + static int stac92xx_init(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; @@ -3965,6 +3995,9 @@ static int stac92xx_init(struct hda_codec *codec) spec->adc_nids[i], 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + /* override some hints */ + stac_store_hints(codec); + /* set up GPIO */ gpio = spec->gpio_data; /* turn on EAPD statically when spec->eapd_switch isn't set. -- cgit v1.2.3-70-g09d2 From d02b1f3910f12cfe377a31afebcbbde4f5664b74 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 2 Mar 2009 17:34:51 +0100 Subject: ALSA: hda - Update documetation for hints sysfs entry Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio.txt | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt index 99958be7b45..c5948f2f9a2 100644 --- a/Documentation/sound/alsa/HD-Audio.txt +++ b/Documentation/sound/alsa/HD-Audio.txt @@ -365,10 +365,13 @@ modelname:: to this file. init_verbs:: The extra verbs to execute at initialization. You can add a verb by - writing to this file. Pass three numbers: nid, verb and parameter. + writing to this file. Pass three numbers: nid, verb and parameter + (separated with a space). hints:: - Shows hint strings for codec parsers for any use. Right now it's - not used. + Shows / stores hint strings for codec parsers for any use. + Its format is `key = value`. For example, passing `hp_detect = yes` + to IDT/STAC codec parser will result in the disablement of the + headphone detection. init_pin_configs:: Shows the initial pin default config values set by BIOS. driver_pin_configs:: -- cgit v1.2.3-70-g09d2 From 82ad39f9391fca1d3177bd9f6a5264eff5b5346a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 3 Mar 2009 15:00:35 +0100 Subject: ALSA: hda - Fix gcc compile warning MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit It's false positive, but annoying. sound/pci/hda/hda_codec.c: In function ‘get_empty_pcm_device’: sound/pci/hda/hda_codec.c:2772: warning: ‘dev’ may be used uninitialized in this function Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 7c9ef5c18e7..04cb1251e3e 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2776,13 +2776,10 @@ static int get_empty_pcm_device(struct hda_bus *bus, int type) for (i = 0; i < ARRAY_SIZE(audio_idx); i++) { dev = audio_idx[i]; if (!test_bit(dev, bus->pcm_dev_bits)) - break; - } - if (i >= ARRAY_SIZE(audio_idx)) { - snd_printk(KERN_WARNING "Too many audio devices\n"); - return -EAGAIN; + goto ok; } - break; + snd_printk(KERN_WARNING "Too many audio devices\n"); + return -EAGAIN; case HDA_PCM_TYPE_SPDIF: case HDA_PCM_TYPE_HDMI: case HDA_PCM_TYPE_MODEM: @@ -2797,6 +2794,7 @@ static int get_empty_pcm_device(struct hda_bus *bus, int type) snd_printk(KERN_WARNING "Invalid PCM type %d\n", type); return -EINVAL; } + ok: set_bit(dev, bus->pcm_dev_bits); return dev; } -- cgit v1.2.3-70-g09d2 From a3c7729e6c5d41bbeb3e13befbcf8e4ef76e55dc Mon Sep 17 00:00:00 2001 From: Philipp Zabel Date: Tue, 3 Mar 2009 16:10:53 +0100 Subject: ASoC: Remove version display from the UDA1380 driver Signed-off-by: Philipp Zabel Signed-off-by: Mark Brown --- sound/soc/codecs/uda1380.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 5242b8156b3..8686a554536 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -35,8 +35,6 @@ #include "uda1380.h" -#define UDA1380_VERSION "0.6" - /* * uda1380 register cache */ @@ -826,8 +824,6 @@ static int uda1380_probe(struct platform_device *pdev) struct snd_soc_codec *codec; int ret; - pr_info("UDA1380 Audio Codec %s", UDA1380_VERSION); - setup = socdev->codec_data; codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); if (codec == NULL) -- cgit v1.2.3-70-g09d2 From ef9e5e5c31cb2c6254760611289ac13e4e41b964 Mon Sep 17 00:00:00 2001 From: Philipp Zabel Date: Tue, 3 Mar 2009 16:10:54 +0100 Subject: ASoC: UDA1380: change decimator/interpolator register handling If the UDA1380's interpolator or decimator are set to be clocked from the WSPLL (which syncs to the WSI signal), the DAI link must be running to change the interpolator/decimator registers (which include volume controls and digital mute setting). * Queue work in the alsa PCM_START .trigger to flush registers as soon as the link is running. This replaces the .prepare and .digital_mute callbacks. * Use the SILENCE override instead of MTM for muting and remove its alsa control to avoid confusion. Signed-off-by: Philipp Zabel Signed-off-by: Mark Brown --- sound/soc/codecs/uda1380.c | 102 +++++++++++++++++++++------------------------ 1 file changed, 48 insertions(+), 54 deletions(-) diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 8686a554536..1c9d2a75add 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -25,6 +25,7 @@ #include #include #include +#include #include #include #include @@ -35,6 +36,9 @@ #include "uda1380.h" +static struct work_struct uda1380_work; +static struct snd_soc_codec *uda1380_codec; + /* * uda1380 register cache */ @@ -50,6 +54,8 @@ static const u16 uda1380_reg[UDA1380_CACHEREGNUM] = { 0x0000, 0x8000, 0x0002, 0x0000, }; +static unsigned long uda1380_cache_dirty; + /* * read uda1380 register cache */ @@ -71,8 +77,11 @@ static inline void uda1380_write_reg_cache(struct snd_soc_codec *codec, u16 reg, unsigned int value) { u16 *cache = codec->reg_cache; + if (reg >= UDA1380_CACHEREGNUM) return; + if ((reg >= 0x10) && (cache[reg] != value)) + set_bit(reg - 0x10, &uda1380_cache_dirty); cache[reg] = value; } @@ -111,6 +120,8 @@ static int uda1380_write(struct snd_soc_codec *codec, unsigned int reg, (data[0]<<8) | data[1]); return -EIO; } + if (reg >= 0x10) + clear_bit(reg - 0x10, &uda1380_cache_dirty); return 0; } else return -EIO; @@ -118,6 +129,20 @@ static int uda1380_write(struct snd_soc_codec *codec, unsigned int reg, #define uda1380_reset(c) uda1380_write(c, UDA1380_RESET, 0) +static void uda1380_flush_work(struct work_struct *work) +{ + int bit, reg; + + for_each_bit(bit, &uda1380_cache_dirty, UDA1380_CACHEREGNUM - 0x10) { + reg = 0x10 + bit; + pr_debug("uda1380: flush reg %x val %x:\n", reg, + uda1380_read_reg_cache(uda1380_codec, reg)); + uda1380_write(uda1380_codec, reg, + uda1380_read_reg_cache(uda1380_codec, reg)); + clear_bit(bit, &uda1380_cache_dirty); + } +} + /* declarations of ALSA reg_elem_REAL controls */ static const char *uda1380_deemp[] = { "None", @@ -252,7 +277,6 @@ static const struct snd_kcontrol_new uda1380_snd_controls[] = { SOC_SINGLE("DAC Polarity inverting Switch", UDA1380_MIXER, 15, 1, 0), /* DA_POL_INV */ SOC_ENUM("Noise Shaper", uda1380_sel_ns_enum), /* SEL_NS */ SOC_ENUM("Digital Mixer Signal Control", uda1380_mix_enum), /* MIX_POS, MIX */ - SOC_SINGLE("Silence Switch", UDA1380_MIXER, 7, 1, 0), /* SILENCE, force DAC output to silence */ SOC_SINGLE("Silence Detector Switch", UDA1380_MIXER, 6, 1, 0), /* SDET_ON */ SOC_ENUM("Silence Detector Setting", uda1380_sdet_enum), /* SD_VALUE */ SOC_ENUM("Oversampling Input", uda1380_os_enum), /* OS */ @@ -438,41 +462,28 @@ static int uda1380_set_dai_fmt_capture(struct snd_soc_dai *codec_dai, return 0; } -/* - * Flush reg cache - * We can only write the interpolator and decimator registers - * when the DAI is being clocked by the CPU DAI. It's up to the - * machine and cpu DAI driver to do this before we are called. - */ -static int uda1380_pcm_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) +static int uda1380_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->card->codec; - int reg, reg_start, reg_end, clk; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - reg_start = UDA1380_MVOL; - reg_end = UDA1380_MIXER; - } else { - reg_start = UDA1380_DEC; - reg_end = UDA1380_AGC; - } - - /* FIXME disable DAC_CLK */ - clk = uda1380_read_reg_cache(codec, UDA1380_CLK); - uda1380_write(codec, UDA1380_CLK, clk & ~R00_DAC_CLK); - - for (reg = reg_start; reg <= reg_end; reg++) { - pr_debug("uda1380: flush reg %x val %x:", reg, - uda1380_read_reg_cache(codec, reg)); - uda1380_write(codec, reg, uda1380_read_reg_cache(codec, reg)); + int mixer = uda1380_read_reg_cache(codec, UDA1380_MIXER); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + uda1380_write_reg_cache(codec, UDA1380_MIXER, + mixer & ~R14_SILENCE); + schedule_work(&uda1380_work); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + uda1380_write_reg_cache(codec, UDA1380_MIXER, + mixer | R14_SILENCE); + schedule_work(&uda1380_work); + break; } - - /* FIXME restore DAC_CLK */ - uda1380_write(codec, UDA1380_CLK, clk); - return 0; } @@ -538,24 +549,6 @@ static void uda1380_pcm_shutdown(struct snd_pcm_substream *substream, uda1380_write(codec, UDA1380_CLK, clk); } -static int uda1380_mute(struct snd_soc_dai *codec_dai, int mute) -{ - struct snd_soc_codec *codec = codec_dai->codec; - u16 mute_reg = uda1380_read_reg_cache(codec, UDA1380_DEEMP) & ~R13_MTM; - - /* FIXME: mute(codec,0) is called when the magician clock is already - * set to WSPLL, but for some unknown reason writing to interpolator - * registers works only when clocked by SYSCLK */ - u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK); - uda1380_write(codec, UDA1380_CLK, ~R00_DAC_CLK & clk); - if (mute) - uda1380_write(codec, UDA1380_DEEMP, mute_reg | R13_MTM); - else - uda1380_write(codec, UDA1380_DEEMP, mute_reg); - uda1380_write(codec, UDA1380_CLK, clk); - return 0; -} - static int uda1380_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -597,10 +590,9 @@ struct snd_soc_dai uda1380_dai[] = { .rates = UDA1380_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, .ops = { + .trigger = uda1380_trigger, .hw_params = uda1380_pcm_hw_params, .shutdown = uda1380_pcm_shutdown, - .prepare = uda1380_pcm_prepare, - .digital_mute = uda1380_mute, .set_fmt = uda1380_set_dai_fmt_both, }, }, @@ -614,10 +606,9 @@ struct snd_soc_dai uda1380_dai[] = { .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .ops = { + .trigger = uda1380_trigger, .hw_params = uda1380_pcm_hw_params, .shutdown = uda1380_pcm_shutdown, - .prepare = uda1380_pcm_prepare, - .digital_mute = uda1380_mute, .set_fmt = uda1380_set_dai_fmt_playback, }, }, @@ -631,9 +622,9 @@ struct snd_soc_dai uda1380_dai[] = { .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .ops = { + .trigger = uda1380_trigger, .hw_params = uda1380_pcm_hw_params, .shutdown = uda1380_pcm_shutdown, - .prepare = uda1380_pcm_prepare, .set_fmt = uda1380_set_dai_fmt_capture, }, }, @@ -692,6 +683,9 @@ static int uda1380_init(struct snd_soc_device *socdev, int dac_clk) codec->reg_cache_step = 1; uda1380_reset(codec); + uda1380_codec = codec; + INIT_WORK(&uda1380_work, uda1380_flush_work); + /* register pcms */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) { -- cgit v1.2.3-70-g09d2 From aa4ef01de5f2e7ed948b88f9f1cfc93c8e0c3f25 Mon Sep 17 00:00:00 2001 From: Philipp Zabel Date: Tue, 3 Mar 2009 16:10:51 +0100 Subject: ASoC: Use network mode with 2 slots for 16-bit stereo in pxa-ssp/Zylonite For consistency with 24-bit and 32-bit modes, don't send 16-bit stereo in one 32-bit transfer. Use 2 slots instead on Zylonite. It should result in exactly the same behaviour. Now it is possible to use 16-bit single slot transfers in pxa-ssp, which are needed for Magician to get two frame clock pulses per sample (one for each channel). Signed-off-by: Philipp Zabel Tested-by: Mark Brown Signed-off-by: Mark Brown --- sound/soc/pxa/pxa-ssp.c | 3 +-- sound/soc/pxa/zylonite.c | 7 +++++-- 2 files changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 4a973ab710b..c49bb12b0a6 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -644,8 +644,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, sscr0 |= SSCR0_FPCKE; #endif sscr0 |= SSCR0_DataSize(16); - if (params_channels(params) > 1) - sscr0 |= SSCR0_EDSS; + /* use network mode (2 slots) for 16 bit stereo */ break; case SNDRV_PCM_FORMAT_S24_LE: sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(8)); diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index 0140a250db2..9f6116edbb8 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -127,8 +127,11 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - /* We're not really in network mode but the emulation wants this. */ - ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 1); + /* Use network mode for stereo, one slot per channel. */ + if (params_channels(params) > 1) + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 2); + else + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 1); if (ret < 0) return ret; -- cgit v1.2.3-70-g09d2 From 5f2a9384a9291d898b4bf85c4fbf497eef582977 Mon Sep 17 00:00:00 2001 From: Philipp Zabel Date: Tue, 3 Mar 2009 16:10:52 +0100 Subject: ASoC: UDA1380: DATAI is slave only Only allow SND_SOC_DAIFMT_CBS_CFS for the playback DAI. Signed-off-by: Philipp Zabel Signed-off-by: Mark Brown --- sound/soc/codecs/uda1380.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 1c9d2a75add..1b10f488328 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -399,8 +399,9 @@ static int uda1380_set_dai_fmt_both(struct snd_soc_dai *codec_dai, iface |= R01_SFORI_MSB | R01_SFORO_MSB; } - if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) == SND_SOC_DAIFMT_CBM_CFM) - iface |= R01_SIM; + /* DATAI is slave only, so in single-link mode, this has to be slave */ + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) + return -EINVAL; uda1380_write(codec, UDA1380_IFACE, iface); @@ -428,6 +429,10 @@ static int uda1380_set_dai_fmt_playback(struct snd_soc_dai *codec_dai, iface |= R01_SFORI_MSB; } + /* DATAI is slave only, so this has to be slave */ + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) + return -EINVAL; + uda1380_write(codec, UDA1380_IFACE, iface); return 0; -- cgit v1.2.3-70-g09d2 From 79d7d5333b598e9a559bf27833f0ad2b8bf6ad2c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 4 Mar 2009 09:03:50 +0100 Subject: ALSA: hda - Fix HP dv6736 mic input Fix the mic input of HP dv6736 with Conexant 5051 codec chip. This laptop seems have no mic-switching per jack connection. A new model hp-dv6736 is introduced to match with the h/w implementation. Reference: Novell bnc#480753 https://bugzilla.novell.com/show_bug.cgi?id=480753 Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 + sound/pci/hda/patch_conexant.c | 63 +++++++++++++++++++++++++--- 2 files changed, 59 insertions(+), 5 deletions(-) diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index a448bbefd48..80b796e4a80 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -262,6 +262,7 @@ Conexant 5051 ============= laptop Basic Laptop config (default) hp HP Spartan laptop + hp-dv6736 HP dv6736 lenovo-x200 Lenovo X200 laptop STAC9200 diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index b8de73ecfde..1938e92e1f0 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -72,6 +72,7 @@ struct conexant_spec { */ unsigned int cur_eapd; unsigned int hp_present; + unsigned int no_auto_mic; unsigned int need_dac_fix; /* capture */ @@ -1665,8 +1666,11 @@ static int cxt5051_hp_master_sw_put(struct snd_kcontrol *kcontrol, /* toggle input of built-in and mic jack appropriately */ static void cxt5051_portb_automic(struct hda_codec *codec) { + struct conexant_spec *spec = codec->spec; unsigned int present; + if (spec->no_auto_mic) + return; present = snd_hda_codec_read(codec, 0x17, 0, AC_VERB_GET_PIN_SENSE, 0) & AC_PINSENSE_PRESENCE; @@ -1682,6 +1686,8 @@ static void cxt5051_portc_automic(struct hda_codec *codec) unsigned int present; hda_nid_t new_adc; + if (spec->no_auto_mic) + return; present = snd_hda_codec_read(codec, 0x18, 0, AC_VERB_GET_PIN_SENSE, 0) & AC_PINSENSE_PRESENCE; @@ -1768,6 +1774,22 @@ static struct snd_kcontrol_new cxt5051_hp_mixers[] = { {} }; +static struct snd_kcontrol_new cxt5051_hp_dv6736_mixers[] = { + HDA_CODEC_VOLUME("Mic Volume", 0x14, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("Mic Switch", 0x14, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = cxt_eapd_info, + .get = cxt_eapd_get, + .put = cxt5051_hp_master_sw_put, + .private_value = 0x1a, + }, + + {} +}; + static struct hda_verb cxt5051_init_verbs[] = { /* Line in, Mic */ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, @@ -1798,6 +1820,32 @@ static struct hda_verb cxt5051_init_verbs[] = { { } /* end */ }; +static struct hda_verb cxt5051_hp_dv6736_init_verbs[] = { + /* Line in, Mic */ + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0}, + {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0}, + /* SPK */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* HP, Amp */ + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x16, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* DAC1 */ + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Record selector: Int mic */ + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1) | 0x44}, + {0x14, AC_VERB_SET_CONNECT_SEL, 0x1}, + /* SPDIF route: PCM */ + {0x1c, AC_VERB_SET_CONNECT_SEL, 0x0}, + /* EAPD */ + {0x1a, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ + {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT}, + {0x17, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CXT5051_PORTB_EVENT}, + { } /* end */ +}; + static struct hda_verb cxt5051_lenovo_x200_init_verbs[] = { /* Line in, Mic */ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, @@ -1849,6 +1897,7 @@ static int cxt5051_init(struct hda_codec *codec) enum { CXT5051_LAPTOP, /* Laptops w/ EAPD support */ CXT5051_HP, /* no docking */ + CXT5051_HP_DV6736, /* HP without mic switch */ CXT5051_LENOVO_X200, /* Lenovo X200 laptop */ CXT5051_MODELS }; @@ -1856,10 +1905,12 @@ enum { static const char *cxt5051_models[CXT5051_MODELS] = { [CXT5051_LAPTOP] = "laptop", [CXT5051_HP] = "hp", + [CXT5051_HP_DV6736] = "hp-dv6736", [CXT5051_LENOVO_X200] = "lenovo-x200", }; static struct snd_pci_quirk cxt5051_cfg_tbl[] = { + SND_PCI_QUIRK(0x103c, 0x30cf, "HP DV6736", CXT5051_HP_DV6736), SND_PCI_QUIRK(0x14f1, 0x0101, "Conexant Reference board", CXT5051_LAPTOP), SND_PCI_QUIRK(0x14f1, 0x5051, "HP Spartan 1.1", CXT5051_HP), @@ -1896,20 +1947,22 @@ static int patch_cxt5051(struct hda_codec *codec) spec->cur_adc = 0; spec->cur_adc_idx = 0; + codec->patch_ops.unsol_event = cxt5051_hp_unsol_event; + board_config = snd_hda_check_board_config(codec, CXT5051_MODELS, cxt5051_models, cxt5051_cfg_tbl); switch (board_config) { case CXT5051_HP: - codec->patch_ops.unsol_event = cxt5051_hp_unsol_event; spec->mixers[0] = cxt5051_hp_mixers; break; + case CXT5051_HP_DV6736: + spec->init_verbs[0] = cxt5051_hp_dv6736_init_verbs; + spec->mixers[0] = cxt5051_hp_dv6736_mixers; + spec->no_auto_mic = 1; + break; case CXT5051_LENOVO_X200: spec->init_verbs[0] = cxt5051_lenovo_x200_init_verbs; - /* fallthru */ - default: - case CXT5051_LAPTOP: - codec->patch_ops.unsol_event = cxt5051_hp_unsol_event; break; } -- cgit v1.2.3-70-g09d2 From bd6afe3f34d41ed81e0c62a5a2181bb7bd51aebf Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 4 Mar 2009 11:30:25 +0100 Subject: ALSA: hda - Fix conflict of mixer controls on Sony VAIO VGN-AR71S The recent update enabled the model=sony-assamd for all ALC262 with PCI SSID 104d:90xx. But this includes the VAIO VGN-AR* that has the primary codec of STAC92xx and the secondary ALC262 as a slave digital-only codec. For this device, the model=auto must be chosen to work properly. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c60c86acd9b..8c02f789e4f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10825,6 +10825,7 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x104d, 0x1f00, "Sony ASSAMD", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x104d, 0x8203, "Sony UX-90", ALC262_HIPPO), SND_PCI_QUIRK(0x104d, 0x820f, "Sony ASSAMD", ALC262_SONY_ASSAMD), + SND_PCI_QUIRK(0x104d, 0x9016, "Sony VAIO", ALC262_AUTO), /* dig-only */ SND_PCI_QUIRK_MASK(0x104d, 0xff00, 0x9000, "Sony VAIO", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1", -- cgit v1.2.3-70-g09d2 From ec67624d33d5639bcc6ee6918cb1fc0bd1bac3a8 Mon Sep 17 00:00:00 2001 From: "Lopez Cruz, Misael" Date: Tue, 3 Mar 2009 15:25:04 -0600 Subject: ASoC: Add GPIO support for jack reporting interface Add GPIO support to jack reporting framework in ASoC using gpiolib calls. The gpio support exports two new functions: snd_soc_jack_add_gpios and snd_soc_jack_free_gpios. Client drivers using gpio feature must pass an array of jack_gpio pins belonging to a specific jack to the snd_soc_jack_add_gpios function. The framework will request the gpios, set the data direction and request irq. The framework will update power status of related jack_pins when an event on the gpio pins comes according to the reporting bits defined for each gpio. All gpio resources allocated when adding jack_gpio pins can be released using snd_soc_jack_free_gpios function. Signed-off-by: Misael Lopez Cruz Signed-off-by: Mark Brown --- include/sound/soc.h | 32 +++++++++++++ sound/soc/soc-jack.c | 129 +++++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 161 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index 0e773526416..a40bc6f316f 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -16,6 +16,8 @@ #include #include #include +#include +#include #include #include #include @@ -168,6 +170,9 @@ struct soc_enum; struct snd_soc_ac97_ops; struct snd_soc_jack; struct snd_soc_jack_pin; +#ifdef CONFIG_GPIOLIB +struct snd_soc_jack_gpio; +#endif typedef int (*hw_write_t)(void *,const char* ,int); typedef int (*hw_read_t)(void *,char* ,int); @@ -194,6 +199,12 @@ int snd_soc_jack_new(struct snd_soc_card *card, const char *id, int type, void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask); int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count, struct snd_soc_jack_pin *pins); +#ifdef CONFIG_GPIOLIB +int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, + struct snd_soc_jack_gpio *gpios); +void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count, + struct snd_soc_jack_gpio *gpios); +#endif /* codec IO */ #define snd_soc_read(codec, reg) codec->read(codec, reg) @@ -264,6 +275,27 @@ struct snd_soc_jack_pin { bool invert; }; +/** + * struct snd_soc_jack_gpio - Describes a gpio pin for jack detection + * + * @gpio: gpio number + * @name: gpio name + * @report: value to report when jack detected + * @invert: report presence in low state + * @debouce_time: debouce time in ms + */ +#ifdef CONFIG_GPIOLIB +struct snd_soc_jack_gpio { + unsigned int gpio; + const char *name; + int report; + int invert; + int debounce_time; + struct snd_soc_jack *jack; + struct work_struct work; +}; +#endif + struct snd_soc_jack { struct snd_jack *jack; struct snd_soc_card *card; diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index ab64a30bedd..bdf2484c222 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -14,6 +14,10 @@ #include #include #include +#include +#include +#include +#include /** * snd_soc_jack_new - Create a new jack @@ -136,3 +140,128 @@ int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count, return 0; } EXPORT_SYMBOL_GPL(snd_soc_jack_add_pins); + +#ifdef CONFIG_GPIOLIB +/* gpio detect */ +void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio) +{ + struct snd_soc_jack *jack = gpio->jack; + int enable; + int report; + + if (gpio->debounce_time > 0) + mdelay(gpio->debounce_time); + + enable = gpio_get_value(gpio->gpio); + if (gpio->invert) + enable = !enable; + + if (enable) + report = gpio->report; + else + report = 0; + + snd_soc_jack_report(jack, report, gpio->report); +} + +/* irq handler for gpio pin */ +static irqreturn_t gpio_handler(int irq, void *data) +{ + struct snd_soc_jack_gpio *gpio = data; + + schedule_work(&gpio->work); + + return IRQ_HANDLED; +} + +/* gpio work */ +static void gpio_work(struct work_struct *work) +{ + struct snd_soc_jack_gpio *gpio; + + gpio = container_of(work, struct snd_soc_jack_gpio, work); + snd_soc_jack_gpio_detect(gpio); +} + +/** + * snd_soc_jack_add_gpios - Associate GPIO pins with an ASoC jack + * + * @jack: ASoC jack + * @count: number of pins + * @gpios: array of gpio pins + * + * This function will request gpio, set data direction and request irq + * for each gpio in the array. + */ +int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, + struct snd_soc_jack_gpio *gpios) +{ + int i, ret; + + for (i = 0; i < count; i++) { + if (!gpio_is_valid(gpios[i].gpio)) { + printk(KERN_ERR "Invalid gpio %d\n", + gpios[i].gpio); + ret = -EINVAL; + goto undo; + } + if (!gpios[i].name) { + printk(KERN_ERR "No name for gpio %d\n", + gpios[i].gpio); + ret = -EINVAL; + goto undo; + } + + ret = gpio_request(gpios[i].gpio, gpios[i].name); + if (ret) + goto undo; + + ret = gpio_direction_input(gpios[i].gpio); + if (ret) + goto err; + + ret = request_irq(gpio_to_irq(gpios[i].gpio), + gpio_handler, + IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING, + jack->card->dev->driver->name, + &gpios[i]); + if (ret) + goto err; + + INIT_WORK(&gpios[i].work, gpio_work); + gpios[i].jack = jack; + } + + return 0; + +err: + gpio_free(gpios[i].gpio); +undo: + snd_soc_jack_free_gpios(jack, i, gpios); + + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_jack_add_gpios); + +/** + * snd_soc_jack_free_gpios - Release GPIO pins' resources of an ASoC jack + * + * @jack: ASoC jack + * @count: number of pins + * @gpios: array of gpio pins + * + * Release gpio and irq resources for gpio pins associated with an ASoC jack. + */ +void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count, + struct snd_soc_jack_gpio *gpios) +{ + int i; + + for (i = 0; i < count; i++) { + free_irq(gpio_to_irq(gpios[i].gpio), &gpios[i]); + gpio_free(gpios[i].gpio); + gpios[i].jack = NULL; + } +} +EXPORT_SYMBOL_GPL(snd_soc_jack_free_gpios); +#endif /* CONFIG_GPIOLIB */ -- cgit v1.2.3-70-g09d2 From 86027ae78c9294bb450b76eec28cfb431a8fb3ee Mon Sep 17 00:00:00 2001 From: Jonas Andersson Date: Wed, 4 Mar 2009 08:24:26 +0100 Subject: ASoC: wm8510 pll settings When setting WM8510_MCLKDIV the pll was turned off. When setting pll frequency you got twice the expected freq, because the code calculated with postscaler of 8, but the hardware divide by 4. Signed-off-by: Jonas Andersson Signed-off-by: Mark Brown --- sound/soc/atmel/playpaq_wm8510.c | 24 ++++++++++++------------ sound/soc/codecs/wm8510.c | 4 ++-- 2 files changed, 14 insertions(+), 14 deletions(-) diff --git a/sound/soc/atmel/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c index 43dd8cee83c..70657534e6b 100644 --- a/sound/soc/atmel/playpaq_wm8510.c +++ b/sound/soc/atmel/playpaq_wm8510.c @@ -164,38 +164,38 @@ static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream, */ switch (params_rate(params)) { case 48000: - pll_out = 12288000; - mclk_div = WM8510_MCLKDIV_1; + pll_out = 24576000; + mclk_div = WM8510_MCLKDIV_2; bclk = WM8510_BCLKDIV_8; break; case 44100: - pll_out = 11289600; - mclk_div = WM8510_MCLKDIV_1; + pll_out = 22579200; + mclk_div = WM8510_MCLKDIV_2; bclk = WM8510_BCLKDIV_8; break; case 22050: - pll_out = 11289600; - mclk_div = WM8510_MCLKDIV_2; + pll_out = 22579200; + mclk_div = WM8510_MCLKDIV_4; bclk = WM8510_BCLKDIV_8; break; case 16000: - pll_out = 12288000; - mclk_div = WM8510_MCLKDIV_3; + pll_out = 24576000; + mclk_div = WM8510_MCLKDIV_6; bclk = WM8510_BCLKDIV_8; break; case 11025: - pll_out = 11289600; - mclk_div = WM8510_MCLKDIV_4; + pll_out = 22579200; + mclk_div = WM8510_MCLKDIV_8; bclk = WM8510_BCLKDIV_8; break; case 8000: - pll_out = 12288000; - mclk_div = WM8510_MCLKDIV_6; + pll_out = 24576000; + mclk_div = WM8510_MCLKDIV_12; bclk = WM8510_BCLKDIV_8; break; diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index f01078cfbd7..6d4ef71e919 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -336,7 +336,7 @@ static int wm8510_set_dai_pll(struct snd_soc_dai *codec_dai, return 0; } - pll_factors(freq_out*8, freq_in); + pll_factors(freq_out*4, freq_in); wm8510_write(codec, WM8510_PLLN, (pll_div.pre_div << 4) | pll_div.n); wm8510_write(codec, WM8510_PLLK1, pll_div.k >> 18); @@ -367,7 +367,7 @@ static int wm8510_set_dai_clkdiv(struct snd_soc_dai *codec_dai, wm8510_write(codec, WM8510_GPIO, reg | div); break; case WM8510_MCLKDIV: - reg = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x1f; + reg = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x11f; wm8510_write(codec, WM8510_CLOCK, reg | div); break; case WM8510_ADCCLK: -- cgit v1.2.3-70-g09d2 From 3ea0d7cf472c6118bb8c0842d606f5436251e179 Mon Sep 17 00:00:00 2001 From: Herton Ronaldo Krzesinski Date: Wed, 4 Mar 2009 14:22:50 -0300 Subject: ALSA: hda - Add 4 channel mode for 3stack-hp model (ALC888) Add additional 4 channel mode for 3stack-hp models. Signed-off-by: Herton Ronaldo Krzesinski Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 27 ++++++++++++++++++++++++--- 1 file changed, 24 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8c02f789e4f..3696ff31838 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8068,24 +8068,45 @@ static struct hda_verb alc888_6st_dell_verbs[] = { { } }; +/* + * 2ch mode + */ static struct hda_verb alc888_3st_hp_2ch_init[] = { { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, { 0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, - { } + { } /* end */ }; +/* + * 4ch mode + */ +static struct hda_verb alc888_3st_hp_4ch_init[] = { + { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, + { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x16, AC_VERB_SET_CONNECT_SEL, 0x01 }, + { } /* end */ +}; + +/* + * 6ch mode + */ static struct hda_verb alc888_3st_hp_6ch_init[] = { { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 }, { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, { 0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { } + { 0x16, AC_VERB_SET_CONNECT_SEL, 0x01 }, + { } /* end */ }; -static struct hda_channel_mode alc888_3st_hp_modes[2] = { +static struct hda_channel_mode alc888_3st_hp_modes[3] = { { 2, alc888_3st_hp_2ch_init }, + { 4, alc888_3st_hp_4ch_init }, { 6, alc888_3st_hp_6ch_init }, }; -- cgit v1.2.3-70-g09d2 From 8718b700ccbcc3c6016d38a75e005293c3660f1c Mon Sep 17 00:00:00 2001 From: Herton Ronaldo Krzesinski Date: Wed, 4 Mar 2009 14:22:51 -0300 Subject: ALSA: hda - Add headphone automute support for 3stack-hp model (ALC888) Mute speaker outputs on headphone insertion for machines that use 3stack-hp model. Signed-off-by: Herton Ronaldo Krzesinski Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 38 +++++++++++++++++++++++++++++++++----- 1 file changed, 33 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3696ff31838..251647d8b5b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8056,16 +8056,42 @@ static struct hda_verb alc888_lenovo_sky_verbs[] = { { } /* end */ }; +static struct hda_verb alc888_6st_dell_verbs[] = { + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + { } +}; + +static void alc888_3st_hp_front_automute(struct hda_codec *codec) +{ + unsigned int present, bits; + + present = snd_hda_codec_read(codec, 0x1b, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + bits = present ? HDA_AMP_MUTE : 0; + snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); + snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); + snd_hda_codec_amp_stereo(codec, 0x18, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); +} + +static void alc888_3st_hp_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + switch (res >> 26) { + case ALC880_HP_EVENT: + alc888_3st_hp_front_automute(codec); + break; + } +} + static struct hda_verb alc888_3st_hp_verbs[] = { {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front: output 0 (0x0c) */ {0x16, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Rear : output 1 (0x0d) */ {0x18, AC_VERB_SET_CONNECT_SEL, 0x02}, /* CLFE : output 2 (0x0e) */ - { } -}; - -static struct hda_verb alc888_6st_dell_verbs[] = { {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, - { } + { } /* end */ }; /* @@ -8950,6 +8976,8 @@ static struct alc_config_preset alc883_presets[] = { .channel_mode = alc888_3st_hp_modes, .need_dac_fix = 1, .input_mux = &alc883_capture_source, + .unsol_event = alc888_3st_hp_unsol_event, + .init_hook = alc888_3st_hp_front_automute, }, [ALC888_6ST_DELL] = { .mixers = { alc883_base_mixer, alc883_chmode_mixer }, -- cgit v1.2.3-70-g09d2 From 7ec30f0e7768985ab2ef6334840e3fc8fa253421 Mon Sep 17 00:00:00 2001 From: Herton Ronaldo Krzesinski Date: Wed, 4 Mar 2009 14:22:52 -0300 Subject: ALSA: hda - Map 3stack-hp model (ALC888) for HP Educ.ar Added model=3stack-hp for HP Educ.ar desktop machine (103c:2a72). Signed-off-by: Herton Ronaldo Krzesinski Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 251647d8b5b..91ef9f27b12 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8673,6 +8673,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_3ST_HP), SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC883_6ST_DIG), SND_PCI_QUIRK(0x103c, 0x2a66, "HP Acacia", ALC888_3ST_HP), + SND_PCI_QUIRK(0x103c, 0x2a72, "HP Educ.ar", ALC888_3ST_HP), SND_PCI_QUIRK(0x1043, 0x1873, "Asus M90V", ALC888_ASUS_M90V), SND_PCI_QUIRK(0x1043, 0x8249, "Asus M2A-VM HDMI", ALC883_3ST_6ch_DIG), SND_PCI_QUIRK(0x1043, 0x8284, "Asus Z37E", ALC883_6ST_DIG), -- cgit v1.2.3-70-g09d2 From 6335d05548eece40092000aa91b64a50310d69d5 Mon Sep 17 00:00:00 2001 From: Eric Miao Date: Tue, 3 Mar 2009 09:41:00 +0800 Subject: ASoC: make ops a pointer in 'struct snd_soc_dai' Considering the fact that most cpu_dai or codec_dai are using a same 'snd_soc_dai_ops' for several similar interfaces, 'ops' would be better made a pointer instead, to make sharing easier and code a bit cleaner. The patch below is rather preliminary since the asoc tree is being actively developed, and this touches almost every piece of code, (and possibly many others in development need to be changed as well). Building of all codecs are OK, yet to every SoC, I didn't test that. Signed-off-by: Eric Miao Acked-by: Timur Tabi Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 2 +- sound/soc/atmel/atmel_ssc_dai.c | 33 +++++-------- sound/soc/au1x/psc-ac97.c | 10 ++-- sound/soc/au1x/psc-i2s.c | 12 +++-- sound/soc/blackfin/bf5xx-i2s.c | 14 +++--- sound/soc/codecs/ac97.c | 7 ++- sound/soc/codecs/ak4535.c | 14 +++--- sound/soc/codecs/cs4270.c | 14 +++--- sound/soc/codecs/ssm2602.c | 20 ++++---- sound/soc/codecs/tlv320aic23.c | 18 ++++--- sound/soc/codecs/tlv320aic26.c | 14 +++--- sound/soc/codecs/tlv320aic3x.c | 14 +++--- sound/soc/codecs/uda134x.c | 18 ++++--- sound/soc/codecs/uda1380.c | 46 ++++++++++-------- sound/soc/codecs/wm8350.c | 20 ++++---- sound/soc/codecs/wm8510.c | 16 +++--- sound/soc/codecs/wm8580.c | 30 +++++++----- sound/soc/codecs/wm8728.c | 12 +++-- sound/soc/codecs/wm8731.c | 18 ++++--- sound/soc/codecs/wm8750.c | 14 +++--- sound/soc/codecs/wm8753.c | 90 +++++++++++++++++++--------------- sound/soc/codecs/wm8900.c | 16 +++--- sound/soc/codecs/wm8903.c | 18 ++++--- sound/soc/codecs/wm8971.c | 14 +++--- sound/soc/codecs/wm8990.c | 18 ++++--- sound/soc/codecs/wm9705.c | 8 +-- sound/soc/codecs/wm9712.c | 14 ++++-- sound/soc/codecs/wm9713.c | 40 +++++++++------ sound/soc/davinci/davinci-i2s.c | 14 +++--- sound/soc/fsl/fsl_ssi.c | 18 ++++--- sound/soc/fsl/mpc5200_psc_i2s.c | 20 ++++---- sound/soc/omap/omap-mcbsp.c | 20 ++++---- sound/soc/pxa/pxa-ssp.c | 65 +++++++------------------ sound/soc/pxa/pxa2xx-ac97.c | 13 ++--- sound/soc/pxa/pxa2xx-i2s.c | 18 ++++--- sound/soc/s3c24xx/s3c2412-i2s.c | 16 +++--- sound/soc/s3c24xx/s3c2443-ac97.c | 13 ++--- sound/soc/s3c24xx/s3c24xx-i2s.c | 16 +++--- sound/soc/sh/ssi.c | 30 +++++------- sound/soc/soc-core.c | 102 +++++++++++++++++++++------------------ 40 files changed, 481 insertions(+), 428 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 24247f76360..13676472ddf 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -203,7 +203,7 @@ struct snd_soc_dai { int (*resume)(struct snd_soc_dai *dai); /* ops */ - struct snd_soc_dai_ops ops; + struct snd_soc_dai_ops *ops; /* DAI capabilities */ struct snd_soc_pcm_stream capture; diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index ff0054b7650..e588e63f18d 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -697,6 +697,15 @@ static int atmel_ssc_resume(struct snd_soc_dai *cpu_dai) #define ATMEL_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) +static struct snd_soc_dai_ops atmel_ssc_dai_ops = { + .startup = atmel_ssc_startup, + .shutdown = atmel_ssc_shutdown, + .prepare = atmel_ssc_prepare, + .hw_params = atmel_ssc_hw_params, + .set_fmt = atmel_ssc_set_dai_fmt, + .set_clkdiv = atmel_ssc_set_dai_clkdiv, +}; + struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = { { .name = "atmel-ssc0", .id = 0, @@ -712,13 +721,7 @@ struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = { .channels_max = 2, .rates = ATMEL_SSC_RATES, .formats = ATMEL_SSC_FORMATS,}, - .ops = { - .startup = atmel_ssc_startup, - .shutdown = atmel_ssc_shutdown, - .prepare = atmel_ssc_prepare, - .hw_params = atmel_ssc_hw_params, - .set_fmt = atmel_ssc_set_dai_fmt, - .set_clkdiv = atmel_ssc_set_dai_clkdiv,}, + .ops = &atmel_ssc_dai_ops, .private_data = &ssc_info[0], }, #if NUM_SSC_DEVICES == 3 @@ -736,13 +739,7 @@ struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = { .channels_max = 2, .rates = ATMEL_SSC_RATES, .formats = ATMEL_SSC_FORMATS,}, - .ops = { - .startup = atmel_ssc_startup, - .shutdown = atmel_ssc_shutdown, - .prepare = atmel_ssc_prepare, - .hw_params = atmel_ssc_hw_params, - .set_fmt = atmel_ssc_set_dai_fmt, - .set_clkdiv = atmel_ssc_set_dai_clkdiv,}, + .ops = &atmel_ssc_dai_ops, .private_data = &ssc_info[1], }, { .name = "atmel-ssc2", @@ -759,13 +756,7 @@ struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = { .channels_max = 2, .rates = ATMEL_SSC_RATES, .formats = ATMEL_SSC_FORMATS,}, - .ops = { - .startup = atmel_ssc_startup, - .shutdown = atmel_ssc_shutdown, - .prepare = atmel_ssc_prepare, - .hw_params = atmel_ssc_hw_params, - .set_fmt = atmel_ssc_set_dai_fmt, - .set_clkdiv = atmel_ssc_set_dai_clkdiv,}, + .ops = &atmel_ssc_dai_ops, .private_data = &ssc_info[2], }, #endif diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index f0e30aec7f2..479d7bdf186 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -342,6 +342,11 @@ static int au1xpsc_ac97_resume(struct snd_soc_dai *dai) return 0; } +static struct snd_soc_dai_ops au1xpsc_ac97_dai_ops = { + .trigger = au1xpsc_ac97_trigger, + .hw_params = au1xpsc_ac97_hw_params, +}; + struct snd_soc_dai au1xpsc_ac97_dai = { .name = "au1xpsc_ac97", .ac97_control = 1, @@ -361,10 +366,7 @@ struct snd_soc_dai au1xpsc_ac97_dai = { .channels_min = 2, .channels_max = 2, }, - .ops = { - .trigger = au1xpsc_ac97_trigger, - .hw_params = au1xpsc_ac97_hw_params, - }, + .ops = &au1xpsc_ac97_dai_ops, }; EXPORT_SYMBOL_GPL(au1xpsc_ac97_dai); diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index f916de4400e..bb589327ee3 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -367,6 +367,12 @@ static int au1xpsc_i2s_resume(struct snd_soc_dai *cpu_dai) return 0; } +static struct snd_soc_dai_ops au1xpsc_i2s_dai_ops = { + .trigger = au1xpsc_i2s_trigger, + .hw_params = au1xpsc_i2s_hw_params, + .set_fmt = au1xpsc_i2s_set_fmt, +}; + struct snd_soc_dai au1xpsc_i2s_dai = { .name = "au1xpsc_i2s", .probe = au1xpsc_i2s_probe, @@ -385,11 +391,7 @@ struct snd_soc_dai au1xpsc_i2s_dai = { .channels_min = 2, .channels_max = 8, /* 2 without external help */ }, - .ops = { - .trigger = au1xpsc_i2s_trigger, - .hw_params = au1xpsc_i2s_hw_params, - .set_fmt = au1xpsc_i2s_set_fmt, - }, + .ops = &au1xpsc_i2s_dai_ops, }; EXPORT_SYMBOL(au1xpsc_i2s_dai); diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c index d1d95d2393f..96482441967 100644 --- a/sound/soc/blackfin/bf5xx-i2s.c +++ b/sound/soc/blackfin/bf5xx-i2s.c @@ -287,6 +287,13 @@ static int bf5xx_i2s_resume(struct platform_device *pdev, #define BF5XX_I2S_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |\ SNDRV_PCM_FMTBIT_S32_LE) +static struct snd_soc_dai_ops bf5xx_i2s_dai_ops = { + .startup = bf5xx_i2s_startup, + .shutdown = bf5xx_i2s_shutdown, + .hw_params = bf5xx_i2s_hw_params, + .set_fmt = bf5xx_i2s_set_dai_fmt, +}; + struct snd_soc_dai bf5xx_i2s_dai = { .name = "bf5xx-i2s", .id = 0, @@ -304,12 +311,7 @@ struct snd_soc_dai bf5xx_i2s_dai = { .channels_max = 2, .rates = BF5XX_I2S_RATES, .formats = BF5XX_I2S_FORMATS,}, - .ops = { - .startup = bf5xx_i2s_startup, - .shutdown = bf5xx_i2s_shutdown, - .hw_params = bf5xx_i2s_hw_params, - .set_fmt = bf5xx_i2s_set_dai_fmt, - }, + .ops = &bf5xx_i2s_dai_ops, }; EXPORT_SYMBOL_GPL(bf5xx_i2s_dai); diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index 11f84b6e5cb..b0d4af145b8 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -41,6 +41,10 @@ static int ac97_prepare(struct snd_pcm_substream *substream, SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\ SNDRV_PCM_RATE_48000) +static struct snd_soc_dai_ops ac97_dai_ops = { + .prepare = ac97_prepare, +}; + struct snd_soc_dai ac97_dai = { .name = "AC97 HiFi", .ac97_control = 1, @@ -56,8 +60,7 @@ struct snd_soc_dai ac97_dai = { .channels_max = 2, .rates = STD_AC97_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .prepare = ac97_prepare,}, + .ops = &ac97_dai_ops, }; EXPORT_SYMBOL_GPL(ac97_dai); diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index d56e6bb1fed..1f63d387a2f 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -421,6 +421,13 @@ static int ak4535_set_bias_level(struct snd_soc_codec *codec, SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) +static struct snd_soc_dai_ops ak4535_dai_ops = { + .hw_params = ak4535_hw_params, + .set_fmt = ak4535_set_dai_fmt, + .digital_mute = ak4535_mute, + .set_sysclk = ak4535_set_dai_sysclk, +}; + struct snd_soc_dai ak4535_dai = { .name = "AK4535", .playback = { @@ -435,12 +442,7 @@ struct snd_soc_dai ak4535_dai = { .channels_max = 2, .rates = AK4535_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .hw_params = ak4535_hw_params, - .set_fmt = ak4535_set_dai_fmt, - .digital_mute = ak4535_mute, - .set_sysclk = ak4535_set_dai_sysclk, - }, + .ops = &ak4535_dai_ops, }; EXPORT_SYMBOL_GPL(ak4535_dai); diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index f86f33cc179..7ae3d6520e3 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -503,6 +503,13 @@ static const struct snd_kcontrol_new cs4270_snd_controls[] = { */ static struct snd_soc_codec *cs4270_codec; +static struct snd_soc_dai_ops cs4270_dai_ops = { + .hw_params = cs4270_hw_params, + .set_sysclk = cs4270_set_dai_sysclk, + .set_fmt = cs4270_set_dai_fmt, + .digital_mute = cs4270_mute, +}; + struct snd_soc_dai cs4270_dai = { .name = "cs4270", .playback = { @@ -519,12 +526,7 @@ struct snd_soc_dai cs4270_dai = { .rates = 0, .formats = CS4270_FORMATS, }, - .ops = { - .hw_params = cs4270_hw_params, - .set_sysclk = cs4270_set_dai_sysclk, - .set_fmt = cs4270_set_dai_fmt, - .digital_mute = cs4270_mute, - }, + .ops = &cs4270_dai_ops, }; EXPORT_SYMBOL_GPL(cs4270_dai); diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 58e225dadc7..87f606c7682 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -506,6 +506,16 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec, #define SSM2602_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) +static struct snd_soc_dai_ops ssm2602_dai_ops = { + .startup = ssm2602_startup, + .prepare = ssm2602_pcm_prepare, + .hw_params = ssm2602_hw_params, + .shutdown = ssm2602_shutdown, + .digital_mute = ssm2602_mute, + .set_sysclk = ssm2602_set_dai_sysclk, + .set_fmt = ssm2602_set_dai_fmt, +}; + struct snd_soc_dai ssm2602_dai = { .name = "SSM2602", .playback = { @@ -520,15 +530,7 @@ struct snd_soc_dai ssm2602_dai = { .channels_max = 2, .rates = SSM2602_RATES, .formats = SSM2602_FORMATS,}, - .ops = { - .startup = ssm2602_startup, - .prepare = ssm2602_pcm_prepare, - .hw_params = ssm2602_hw_params, - .shutdown = ssm2602_shutdown, - .digital_mute = ssm2602_mute, - .set_sysclk = ssm2602_set_dai_sysclk, - .set_fmt = ssm2602_set_dai_fmt, - } + .ops = &ssm2602_dai_ops, }; EXPORT_SYMBOL_GPL(ssm2602_dai); diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 8b20c360adf..c3f4afb5d01 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -580,6 +580,15 @@ static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec, #define AIC23_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) +static struct snd_soc_dai_ops tlv320aic23_dai_ops = { + .prepare = tlv320aic23_pcm_prepare, + .hw_params = tlv320aic23_hw_params, + .shutdown = tlv320aic23_shutdown, + .digital_mute = tlv320aic23_mute, + .set_fmt = tlv320aic23_set_dai_fmt, + .set_sysclk = tlv320aic23_set_dai_sysclk, +}; + struct snd_soc_dai tlv320aic23_dai = { .name = "tlv320aic23", .playback = { @@ -594,14 +603,7 @@ struct snd_soc_dai tlv320aic23_dai = { .channels_max = 2, .rates = AIC23_RATES, .formats = AIC23_FORMATS,}, - .ops = { - .prepare = tlv320aic23_pcm_prepare, - .hw_params = tlv320aic23_hw_params, - .shutdown = tlv320aic23_shutdown, - .digital_mute = tlv320aic23_mute, - .set_fmt = tlv320aic23_set_dai_fmt, - .set_sysclk = tlv320aic23_set_dai_sysclk, - } + .ops = &tlv320aic23_dai_ops, }; EXPORT_SYMBOL_GPL(tlv320aic23_dai); diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index 229e464cf71..a7f333fc579 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -270,6 +270,13 @@ static int aic26_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) #define AIC26_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE |\ SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE) +static struct snd_soc_dai_ops aic26_dai_ops = { + .hw_params = aic26_hw_params, + .digital_mute = aic26_mute, + .set_sysclk = aic26_set_sysclk, + .set_fmt = aic26_set_fmt, +}; + struct snd_soc_dai aic26_dai = { .name = "tlv320aic26", .playback = { @@ -286,12 +293,7 @@ struct snd_soc_dai aic26_dai = { .rates = AIC26_RATES, .formats = AIC26_FORMATS, }, - .ops = { - .hw_params = aic26_hw_params, - .digital_mute = aic26_mute, - .set_sysclk = aic26_set_sysclk, - .set_fmt = aic26_set_fmt, - }, + .ops = &aic26_dai_ops, }; EXPORT_SYMBOL_GPL(aic26_dai); diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index d638e3f0728..ab099f48248 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1088,6 +1088,13 @@ EXPORT_SYMBOL_GPL(aic3x_button_pressed); #define AIC3X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) +static struct snd_soc_dai_ops aic3x_dai_ops = { + .hw_params = aic3x_hw_params, + .digital_mute = aic3x_mute, + .set_sysclk = aic3x_set_dai_sysclk, + .set_fmt = aic3x_set_dai_fmt, +}; + struct snd_soc_dai aic3x_dai = { .name = "tlv320aic3x", .playback = { @@ -1102,12 +1109,7 @@ struct snd_soc_dai aic3x_dai = { .channels_max = 2, .rates = AIC3X_RATES, .formats = AIC3X_FORMATS,}, - .ops = { - .hw_params = aic3x_hw_params, - .digital_mute = aic3x_mute, - .set_sysclk = aic3x_set_dai_sysclk, - .set_fmt = aic3x_set_dai_fmt, - } + .ops = &aic3x_dai_ops, }; EXPORT_SYMBOL_GPL(aic3x_dai); diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index 661599295ca..ddefb8f8014 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -431,6 +431,15 @@ SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]), SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0), }; +static struct snd_soc_dai_ops uda134x_dai_ops = { + .startup = uda134x_startup, + .shutdown = uda134x_shutdown, + .hw_params = uda134x_hw_params, + .digital_mute = uda134x_mute, + .set_sysclk = uda134x_set_dai_sysclk, + .set_fmt = uda134x_set_dai_fmt, +}; + struct snd_soc_dai uda134x_dai = { .name = "UDA134X", /* playback capabilities */ @@ -450,14 +459,7 @@ struct snd_soc_dai uda134x_dai = { .formats = UDA134X_FORMATS, }, /* pcm operations */ - .ops = { - .startup = uda134x_startup, - .shutdown = uda134x_shutdown, - .hw_params = uda134x_hw_params, - .digital_mute = uda134x_mute, - .set_sysclk = uda134x_set_dai_sysclk, - .set_fmt = uda134x_set_dai_fmt, - } + .ops = &uda134x_dai_ops, }; EXPORT_SYMBOL(uda134x_dai); diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 5242b8156b3..cafa7684c0e 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -583,6 +583,29 @@ static int uda1380_set_bias_level(struct snd_soc_codec *codec, SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) +static struct snd_soc_dai_ops uda1380_dai_ops = { + .hw_params = uda1380_pcm_hw_params, + .shutdown = uda1380_pcm_shutdown, + .prepare = uda1380_pcm_prepare, + .digital_mute = uda1380_mute, + .set_fmt = uda1380_set_dai_fmt_both, +}; + +static struct snd_soc_dai_ops uda1380_dai_ops_playback = { + .hw_params = uda1380_pcm_hw_params, + .shutdown = uda1380_pcm_shutdown, + .prepare = uda1380_pcm_prepare, + .digital_mute = uda1380_mute, + .set_fmt = uda1380_set_dai_fmt_playback, +}; + +static struct snd_soc_dai_ops uda1380_dai_ops_capture = { + .hw_params = uda1380_pcm_hw_params, + .shutdown = uda1380_pcm_shutdown, + .prepare = uda1380_pcm_prepare, + .set_fmt = uda1380_set_dai_fmt_capture, +}; + struct snd_soc_dai uda1380_dai[] = { { .name = "UDA1380", @@ -598,13 +621,7 @@ struct snd_soc_dai uda1380_dai[] = { .channels_max = 2, .rates = UDA1380_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .hw_params = uda1380_pcm_hw_params, - .shutdown = uda1380_pcm_shutdown, - .prepare = uda1380_pcm_prepare, - .digital_mute = uda1380_mute, - .set_fmt = uda1380_set_dai_fmt_both, - }, + .ops = &uda1380_dai_ops, }, { /* playback only - dual interface */ .name = "UDA1380", @@ -615,13 +632,7 @@ struct snd_soc_dai uda1380_dai[] = { .rates = UDA1380_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, - .ops = { - .hw_params = uda1380_pcm_hw_params, - .shutdown = uda1380_pcm_shutdown, - .prepare = uda1380_pcm_prepare, - .digital_mute = uda1380_mute, - .set_fmt = uda1380_set_dai_fmt_playback, - }, + .ops = &uda1380_dai_ops_playback, }, { /* capture only - dual interface*/ .name = "UDA1380", @@ -632,12 +643,7 @@ struct snd_soc_dai uda1380_dai[] = { .rates = UDA1380_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, - .ops = { - .hw_params = uda1380_pcm_hw_params, - .shutdown = uda1380_pcm_shutdown, - .prepare = uda1380_pcm_prepare, - .set_fmt = uda1380_set_dai_fmt_capture, - }, + .ops = &uda1380_dai_ops_capture, }, }; EXPORT_SYMBOL_GPL(uda1380_dai); diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 359e5cc86f3..3b1d0993bed 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1538,6 +1538,16 @@ static int wm8350_remove(struct platform_device *pdev) SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) +static struct snd_soc_dai_ops wm8350_dai_ops = { + .hw_params = wm8350_pcm_hw_params, + .digital_mute = wm8350_mute, + .trigger = wm8350_pcm_trigger, + .set_fmt = wm8350_set_dai_fmt, + .set_sysclk = wm8350_set_dai_sysclk, + .set_pll = wm8350_set_fll, + .set_clkdiv = wm8350_set_clkdiv, +}; + struct snd_soc_dai wm8350_dai = { .name = "WM8350", .playback = { @@ -1554,15 +1564,7 @@ struct snd_soc_dai wm8350_dai = { .rates = WM8350_RATES, .formats = WM8350_FORMATS, }, - .ops = { - .hw_params = wm8350_pcm_hw_params, - .digital_mute = wm8350_mute, - .trigger = wm8350_pcm_trigger, - .set_fmt = wm8350_set_dai_fmt, - .set_sysclk = wm8350_set_dai_sysclk, - .set_pll = wm8350_set_fll, - .set_clkdiv = wm8350_set_clkdiv, - }, + .ops = &wm8350_dai_ops, }; EXPORT_SYMBOL_GPL(wm8350_dai); diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index f01078cfbd7..cc975a62fa5 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -554,6 +554,14 @@ static int wm8510_set_bias_level(struct snd_soc_codec *codec, #define WM8510_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) +static struct snd_soc_dai_ops wm8510_dai_ops = { + .hw_params = wm8510_pcm_hw_params, + .digital_mute = wm8510_mute, + .set_fmt = wm8510_set_dai_fmt, + .set_clkdiv = wm8510_set_dai_clkdiv, + .set_pll = wm8510_set_dai_pll, +}; + struct snd_soc_dai wm8510_dai = { .name = "WM8510 HiFi", .playback = { @@ -568,13 +576,7 @@ struct snd_soc_dai wm8510_dai = { .channels_max = 2, .rates = WM8510_RATES, .formats = WM8510_FORMATS,}, - .ops = { - .hw_params = wm8510_pcm_hw_params, - .digital_mute = wm8510_mute, - .set_fmt = wm8510_set_dai_fmt, - .set_clkdiv = wm8510_set_dai_clkdiv, - .set_pll = wm8510_set_dai_pll, - }, + .ops = &wm8510_dai_ops, }; EXPORT_SYMBOL_GPL(wm8510_dai); diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index d3c51ba5e6f..ee0af23a1ac 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -771,6 +771,21 @@ static int wm8580_set_bias_level(struct snd_soc_codec *codec, #define WM8580_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) +static struct snd_soc_dai_ops wm8580_dai_ops_playback = { + .hw_params = wm8580_paif_hw_params, + .set_fmt = wm8580_set_paif_dai_fmt, + .set_clkdiv = wm8580_set_dai_clkdiv, + .set_pll = wm8580_set_dai_pll, + .digital_mute = wm8580_digital_mute, +}; + +static struct snd_soc_dai_ops wm8580_dai_ops_capture = { + .hw_params = wm8580_paif_hw_params, + .set_fmt = wm8580_set_paif_dai_fmt, + .set_clkdiv = wm8580_set_dai_clkdiv, + .set_pll = wm8580_set_dai_pll, +}; + struct snd_soc_dai wm8580_dai[] = { { .name = "WM8580 PAIFRX", @@ -782,13 +797,7 @@ struct snd_soc_dai wm8580_dai[] = { .rates = SNDRV_PCM_RATE_8000_192000, .formats = WM8580_FORMATS, }, - .ops = { - .hw_params = wm8580_paif_hw_params, - .set_fmt = wm8580_set_paif_dai_fmt, - .set_clkdiv = wm8580_set_dai_clkdiv, - .set_pll = wm8580_set_dai_pll, - .digital_mute = wm8580_digital_mute, - }, + .ops = &wm8580_dai_ops_playback, }, { .name = "WM8580 PAIFTX", @@ -800,12 +809,7 @@ struct snd_soc_dai wm8580_dai[] = { .rates = SNDRV_PCM_RATE_8000_192000, .formats = WM8580_FORMATS, }, - .ops = { - .hw_params = wm8580_paif_hw_params, - .set_fmt = wm8580_set_paif_dai_fmt, - .set_clkdiv = wm8580_set_dai_clkdiv, - .set_pll = wm8580_set_dai_pll, - }, + .ops = &wm8580_dai_ops_capture, }, }; EXPORT_SYMBOL_GPL(wm8580_dai); diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index f8363b30889..e7ff2121ede 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -244,6 +244,12 @@ static int wm8728_set_bias_level(struct snd_soc_codec *codec, #define WM8728_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) +static struct snd_soc_dai_ops wm8728_dai_ops = { + .hw_params = wm8728_hw_params, + .digital_mute = wm8728_mute, + .set_fmt = wm8728_set_dai_fmt, +}; + struct snd_soc_dai wm8728_dai = { .name = "WM8728", .playback = { @@ -253,11 +259,7 @@ struct snd_soc_dai wm8728_dai = { .rates = WM8728_RATES, .formats = WM8728_FORMATS, }, - .ops = { - .hw_params = wm8728_hw_params, - .digital_mute = wm8728_mute, - .set_fmt = wm8728_set_dai_fmt, - } + .ops = &wm8728_dai_ops, }; EXPORT_SYMBOL_GPL(wm8728_dai); diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 9e7ebcc2c49..e043e3f6000 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -433,6 +433,15 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, #define WM8731_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) +static struct snd_soc_dai_ops wm8731_dai_ops = { + .prepare = wm8731_pcm_prepare, + .hw_params = wm8731_hw_params, + .shutdown = wm8731_shutdown, + .digital_mute = wm8731_mute, + .set_sysclk = wm8731_set_dai_sysclk, + .set_fmt = wm8731_set_dai_fmt, +}; + struct snd_soc_dai wm8731_dai = { .name = "WM8731", .playback = { @@ -447,14 +456,7 @@ struct snd_soc_dai wm8731_dai = { .channels_max = 2, .rates = WM8731_RATES, .formats = WM8731_FORMATS,}, - .ops = { - .prepare = wm8731_pcm_prepare, - .hw_params = wm8731_hw_params, - .shutdown = wm8731_shutdown, - .digital_mute = wm8731_mute, - .set_sysclk = wm8731_set_dai_sysclk, - .set_fmt = wm8731_set_dai_fmt, - } + .ops = &wm8731_dai_ops, }; EXPORT_SYMBOL_GPL(wm8731_dai); diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 96afb86addc..b64509b01a4 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -679,6 +679,13 @@ static int wm8750_set_bias_level(struct snd_soc_codec *codec, #define WM8750_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) +static struct snd_soc_dai_ops wm8750_dai_ops = { + .hw_params = wm8750_pcm_hw_params, + .digital_mute = wm8750_mute, + .set_fmt = wm8750_set_dai_fmt, + .set_sysclk = wm8750_set_dai_sysclk, +}; + struct snd_soc_dai wm8750_dai = { .name = "WM8750", .playback = { @@ -693,12 +700,7 @@ struct snd_soc_dai wm8750_dai = { .channels_max = 2, .rates = WM8750_RATES, .formats = WM8750_FORMATS,}, - .ops = { - .hw_params = wm8750_pcm_hw_params, - .digital_mute = wm8750_mute, - .set_fmt = wm8750_set_dai_fmt, - .set_sysclk = wm8750_set_dai_sysclk, - }, + .ops = &wm8750_dai_ops, }; EXPORT_SYMBOL_GPL(wm8750_dai); diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 7f353e935d7..cc6e57f9acf 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1306,6 +1306,51 @@ static int wm8753_set_bias_level(struct snd_soc_codec *codec, * 3. Voice disabled - HIFI over HIFI * 4. Voice disabled - HIFI over HIFI, uses voice DAI LRC for capture */ +static struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode1 = { + .hw_params = wm8753_i2s_hw_params, + .digital_mute = wm8753_mute, + .set_fmt = wm8753_mode1h_set_dai_fmt, + .set_clkdiv = wm8753_set_dai_clkdiv, + .set_pll = wm8753_set_dai_pll, + .set_sysclk = wm8753_set_dai_sysclk, +}; + +static struct snd_soc_dai_ops wm8753_dai_ops_voice_mode1 = { + .hw_params = wm8753_pcm_hw_params, + .digital_mute = wm8753_mute, + .set_fmt = wm8753_mode1v_set_dai_fmt, + .set_clkdiv = wm8753_set_dai_clkdiv, + .set_pll = wm8753_set_dai_pll, + .set_sysclk = wm8753_set_dai_sysclk, +}; + +static struct snd_soc_dai_ops wm8753_dai_ops_voice_mode2 = { + .hw_params = wm8753_pcm_hw_params, + .digital_mute = wm8753_mute, + .set_fmt = wm8753_mode2_set_dai_fmt, + .set_clkdiv = wm8753_set_dai_clkdiv, + .set_pll = wm8753_set_dai_pll, + .set_sysclk = wm8753_set_dai_sysclk, +}; + +static struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode3 = { + .hw_params = wm8753_i2s_hw_params, + .digital_mute = wm8753_mute, + .set_fmt = wm8753_mode3_4_set_dai_fmt, + .set_clkdiv = wm8753_set_dai_clkdiv, + .set_pll = wm8753_set_dai_pll, + .set_sysclk = wm8753_set_dai_sysclk, +}; + +static struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode4 = { + .hw_params = wm8753_i2s_hw_params, + .digital_mute = wm8753_mute, + .set_fmt = wm8753_mode3_4_set_dai_fmt, + .set_clkdiv = wm8753_set_dai_clkdiv, + .set_pll = wm8753_set_dai_pll, + .set_sysclk = wm8753_set_dai_sysclk, +}; + static const struct snd_soc_dai wm8753_all_dai[] = { /* DAI HiFi mode 1 */ { .name = "WM8753 HiFi", @@ -1322,14 +1367,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = { .channels_max = 2, .rates = WM8753_RATES, .formats = WM8753_FORMATS}, - .ops = { - .hw_params = wm8753_i2s_hw_params, - .digital_mute = wm8753_mute, - .set_fmt = wm8753_mode1h_set_dai_fmt, - .set_clkdiv = wm8753_set_dai_clkdiv, - .set_pll = wm8753_set_dai_pll, - .set_sysclk = wm8753_set_dai_sysclk, - }, + .ops = &wm8753_dai_ops_hifi_mode1, }, /* DAI Voice mode 1 */ { .name = "WM8753 Voice", @@ -1346,14 +1384,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = { .channels_max = 2, .rates = WM8753_RATES, .formats = WM8753_FORMATS,}, - .ops = { - .hw_params = wm8753_pcm_hw_params, - .digital_mute = wm8753_mute, - .set_fmt = wm8753_mode1v_set_dai_fmt, - .set_clkdiv = wm8753_set_dai_clkdiv, - .set_pll = wm8753_set_dai_pll, - .set_sysclk = wm8753_set_dai_sysclk, - }, + .ops = &wm8753_dai_ops_voice_mode1, }, /* DAI HiFi mode 2 - dummy */ { .name = "WM8753 HiFi", @@ -1374,14 +1405,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = { .channels_max = 2, .rates = WM8753_RATES, .formats = WM8753_FORMATS,}, - .ops = { - .hw_params = wm8753_pcm_hw_params, - .digital_mute = wm8753_mute, - .set_fmt = wm8753_mode2_set_dai_fmt, - .set_clkdiv = wm8753_set_dai_clkdiv, - .set_pll = wm8753_set_dai_pll, - .set_sysclk = wm8753_set_dai_sysclk, - }, + .ops = &wm8753_dai_ops_voice_mode2, }, /* DAI HiFi mode 3 */ { .name = "WM8753 HiFi", @@ -1398,14 +1422,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = { .channels_max = 2, .rates = WM8753_RATES, .formats = WM8753_FORMATS,}, - .ops = { - .hw_params = wm8753_i2s_hw_params, - .digital_mute = wm8753_mute, - .set_fmt = wm8753_mode3_4_set_dai_fmt, - .set_clkdiv = wm8753_set_dai_clkdiv, - .set_pll = wm8753_set_dai_pll, - .set_sysclk = wm8753_set_dai_sysclk, - }, + .ops = &wm8753_dai_ops_hifi_mode3, }, /* DAI Voice mode 3 - dummy */ { .name = "WM8753 Voice", @@ -1426,14 +1443,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = { .channels_max = 2, .rates = WM8753_RATES, .formats = WM8753_FORMATS,}, - .ops = { - .hw_params = wm8753_i2s_hw_params, - .digital_mute = wm8753_mute, - .set_fmt = wm8753_mode3_4_set_dai_fmt, - .set_clkdiv = wm8753_set_dai_clkdiv, - .set_pll = wm8753_set_dai_pll, - .set_sysclk = wm8753_set_dai_sysclk, - }, + .ops = &wm8753_dai_ops_hifi_mode4, }, /* DAI Voice mode 4 - dummy */ { .name = "WM8753 Voice", diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index da5ca64f89b..46c5ea1ff92 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -1088,6 +1088,14 @@ static int wm8900_digital_mute(struct snd_soc_dai *codec_dai, int mute) (SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \ SNDRV_PCM_FORMAT_S24_LE) +static struct snd_soc_dai_ops wm8900_dai_ops = { + .hw_params = wm8900_hw_params, + .set_clkdiv = wm8900_set_dai_clkdiv, + .set_pll = wm8900_set_dai_pll, + .set_fmt = wm8900_set_dai_fmt, + .digital_mute = wm8900_digital_mute, +}; + struct snd_soc_dai wm8900_dai = { .name = "WM8900 HiFi", .playback = { @@ -1104,13 +1112,7 @@ struct snd_soc_dai wm8900_dai = { .rates = WM8900_RATES, .formats = WM8900_PCM_FORMATS, }, - .ops = { - .hw_params = wm8900_hw_params, - .set_clkdiv = wm8900_set_dai_clkdiv, - .set_pll = wm8900_set_dai_pll, - .set_fmt = wm8900_set_dai_fmt, - .digital_mute = wm8900_digital_mute, - }, + .ops = &wm8900_dai_ops, }; EXPORT_SYMBOL_GPL(wm8900_dai); diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index c6fa8a71b4d..8cf571f1a80 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1497,6 +1497,15 @@ static int wm8903_hw_params(struct snd_pcm_substream *substream, SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) +static struct snd_soc_dai_ops wm8903_dai_ops = { + .startup = wm8903_startup, + .shutdown = wm8903_shutdown, + .hw_params = wm8903_hw_params, + .digital_mute = wm8903_digital_mute, + .set_fmt = wm8903_set_dai_fmt, + .set_sysclk = wm8903_set_dai_sysclk, +}; + struct snd_soc_dai wm8903_dai = { .name = "WM8903", .playback = { @@ -1513,14 +1522,7 @@ struct snd_soc_dai wm8903_dai = { .rates = WM8903_CAPTURE_RATES, .formats = WM8903_FORMATS, }, - .ops = { - .startup = wm8903_startup, - .shutdown = wm8903_shutdown, - .hw_params = wm8903_hw_params, - .digital_mute = wm8903_digital_mute, - .set_fmt = wm8903_set_dai_fmt, - .set_sysclk = wm8903_set_dai_sysclk - } + .ops = &wm8903_dai_ops, }; EXPORT_SYMBOL_GPL(wm8903_dai); diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 24d4c905a01..032dca22dbd 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -604,6 +604,13 @@ static int wm8971_set_bias_level(struct snd_soc_codec *codec, #define WM8971_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) +static struct snd_soc_dai_ops wm8971_dai_ops = { + .hw_params = wm8971_pcm_hw_params, + .digital_mute = wm8971_mute, + .set_fmt = wm8971_set_dai_fmt, + .set_sysclk = wm8971_set_dai_sysclk, +}; + struct snd_soc_dai wm8971_dai = { .name = "WM8971", .playback = { @@ -618,12 +625,7 @@ struct snd_soc_dai wm8971_dai = { .channels_max = 2, .rates = WM8971_RATES, .formats = WM8971_FORMATS,}, - .ops = { - .hw_params = wm8971_pcm_hw_params, - .digital_mute = wm8971_mute, - .set_fmt = wm8971_set_dai_fmt, - .set_sysclk = wm8971_set_dai_sysclk, - }, + .ops = &wm8971_dai_ops, }; EXPORT_SYMBOL_GPL(wm8971_dai); diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 1a38421f759..c518c3e5aa3 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1332,6 +1332,15 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, * 1. ADC/DAC on Primary Interface * 2. ADC on Primary Interface/DAC on secondary */ +static struct snd_soc_dai_ops wm8990_dai_ops = { + .hw_params = wm8990_hw_params, + .digital_mute = wm8990_mute, + .set_fmt = wm8990_set_dai_fmt, + .set_clkdiv = wm8990_set_dai_clkdiv, + .set_pll = wm8990_set_dai_pll, + .set_sysclk = wm8990_set_dai_sysclk, +}; + struct snd_soc_dai wm8990_dai = { /* ADC/DAC on primary */ .name = "WM8990 ADC/DAC Primary", @@ -1348,14 +1357,7 @@ struct snd_soc_dai wm8990_dai = { .channels_max = 2, .rates = WM8990_RATES, .formats = WM8990_FORMATS,}, - .ops = { - .hw_params = wm8990_hw_params, - .digital_mute = wm8990_mute, - .set_fmt = wm8990_set_dai_fmt, - .set_clkdiv = wm8990_set_dai_clkdiv, - .set_pll = wm8990_set_dai_pll, - .set_sysclk = wm8990_set_dai_sysclk, - }, + .ops = &wm8990_dai_ops, }; EXPORT_SYMBOL_GPL(wm8990_dai); diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index 2e9e06b2daa..3265817c5c2 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -269,6 +269,10 @@ static int ac97_prepare(struct snd_pcm_substream *substream, SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000) +static struct snd_soc_dai_ops wm9705_dai_ops = { + .prepare = ac97_prepare, +}; + struct snd_soc_dai wm9705_dai[] = { { .name = "AC97 HiFi", @@ -287,9 +291,7 @@ struct snd_soc_dai wm9705_dai[] = { .rates = WM9705_AC97_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, - .ops = { - .prepare = ac97_prepare, - }, + .ops = &wm9705_dai_ops, }, { .name = "AC97 Aux", diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index b3a8be77676..765cf1e7369 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -517,6 +517,14 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream, SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\ SNDRV_PCM_RATE_48000) +static struct snd_soc_dai_ops wm9712_dai_ops_hifi = { + .prepare = ac97_prepare, +}; + +static struct snd_soc_dai_ops wm9712_dai_ops_aux = { + .prepare = ac97_aux_prepare, +}; + struct snd_soc_dai wm9712_dai[] = { { .name = "AC97 HiFi", @@ -533,8 +541,7 @@ struct snd_soc_dai wm9712_dai[] = { .channels_max = 2, .rates = WM9712_AC97_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .prepare = ac97_prepare,}, + .ops = &wm9712_dai_ops_hifi, }, { .name = "AC97 Aux", @@ -544,8 +551,7 @@ struct snd_soc_dai wm9712_dai[] = { .channels_max = 1, .rates = WM9712_AC97_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .prepare = ac97_aux_prepare,}, + .ops = &wm9712_dai_ops_aux, } }; EXPORT_SYMBOL_GPL(wm9712_dai); diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index a93aea5c187..523bad077fa 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1005,6 +1005,27 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream, (SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \ SNDRV_PCM_FORMAT_S24_LE) +static struct snd_soc_dai_ops wm9713_dai_ops_hifi = { + .prepare = ac97_hifi_prepare, + .set_clkdiv = wm9713_set_dai_clkdiv, + .set_pll = wm9713_set_dai_pll, +}; + +static struct snd_soc_dai_ops wm9713_dai_ops_aux = { + .prepare = ac97_aux_prepare, + .set_clkdiv = wm9713_set_dai_clkdiv, + .set_pll = wm9713_set_dai_pll, +}; + +static struct snd_soc_dai_ops wm9713_dai_ops_voice = { + .hw_params = wm9713_pcm_hw_params, + .shutdown = wm9713_voiceshutdown, + .set_clkdiv = wm9713_set_dai_clkdiv, + .set_pll = wm9713_set_dai_pll, + .set_fmt = wm9713_set_dai_fmt, + .set_tristate = wm9713_set_dai_tristate, +}; + struct snd_soc_dai wm9713_dai[] = { { .name = "AC97 HiFi", @@ -1021,10 +1042,7 @@ struct snd_soc_dai wm9713_dai[] = { .channels_max = 2, .rates = WM9713_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .prepare = ac97_hifi_prepare, - .set_clkdiv = wm9713_set_dai_clkdiv, - .set_pll = wm9713_set_dai_pll,}, + .ops = &wm9713_dai_ops_hifi, }, { .name = "AC97 Aux", @@ -1034,10 +1052,7 @@ struct snd_soc_dai wm9713_dai[] = { .channels_max = 1, .rates = WM9713_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .prepare = ac97_aux_prepare, - .set_clkdiv = wm9713_set_dai_clkdiv, - .set_pll = wm9713_set_dai_pll,}, + .ops = &wm9713_dai_ops_aux, }, { .name = "WM9713 Voice", @@ -1053,14 +1068,7 @@ struct snd_soc_dai wm9713_dai[] = { .channels_max = 2, .rates = WM9713_PCM_RATES, .formats = WM9713_PCM_FORMATS,}, - .ops = { - .hw_params = wm9713_pcm_hw_params, - .shutdown = wm9713_voiceshutdown, - .set_clkdiv = wm9713_set_dai_clkdiv, - .set_pll = wm9713_set_dai_pll, - .set_fmt = wm9713_set_dai_fmt, - .set_tristate = wm9713_set_dai_tristate, - }, + .ops = &wm9713_dai_ops_voice, }, }; EXPORT_SYMBOL_GPL(wm9713_dai); diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 0fee779e3c7..ffdb9439d3d 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -499,6 +499,13 @@ static void davinci_i2s_remove(struct platform_device *pdev, #define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000 +static struct snd_soc_dai_ops davinci_i2s_dai_ops = { + .startup = davinci_i2s_startup, + .trigger = davinci_i2s_trigger, + .hw_params = davinci_i2s_hw_params, + .set_fmt = davinci_i2s_set_dai_fmt, +}; + struct snd_soc_dai davinci_i2s_dai = { .name = "davinci-i2s", .id = 0, @@ -514,12 +521,7 @@ struct snd_soc_dai davinci_i2s_dai = { .channels_max = 2, .rates = DAVINCI_I2S_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .startup = davinci_i2s_startup, - .trigger = davinci_i2s_trigger, - .hw_params = davinci_i2s_hw_params, - .set_fmt = davinci_i2s_set_dai_fmt, - }, + .ops = &davinci_i2s_dai_ops, }; EXPORT_SYMBOL_GPL(davinci_i2s_dai); diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 6844009833d..0fddd437a7c 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -562,6 +562,15 @@ static int fsl_ssi_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int format) /** * fsl_ssi_dai_template: template CPU DAI for the SSI */ +static struct snd_soc_dai_ops fsl_ssi_dai_ops = { + .startup = fsl_ssi_startup, + .hw_params = fsl_ssi_hw_params, + .shutdown = fsl_ssi_shutdown, + .trigger = fsl_ssi_trigger, + .set_sysclk = fsl_ssi_set_sysclk, + .set_fmt = fsl_ssi_set_fmt, +}; + static struct snd_soc_dai fsl_ssi_dai_template = { .playback = { /* The SSI does not support monaural audio. */ @@ -576,14 +585,7 @@ static struct snd_soc_dai fsl_ssi_dai_template = { .rates = FSLSSI_I2S_RATES, .formats = FSLSSI_I2S_FORMATS, }, - .ops = { - .startup = fsl_ssi_startup, - .hw_params = fsl_ssi_hw_params, - .shutdown = fsl_ssi_shutdown, - .trigger = fsl_ssi_trigger, - .set_sysclk = fsl_ssi_set_sysclk, - .set_fmt = fsl_ssi_set_fmt, - }, + .ops = &fsl_ssi_dai_ops, }; /** diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c index 9eb1ce185bd..3aa729df27b 100644 --- a/sound/soc/fsl/mpc5200_psc_i2s.c +++ b/sound/soc/fsl/mpc5200_psc_i2s.c @@ -468,6 +468,16 @@ static int psc_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int format) /** * psc_i2s_dai_template: template CPU Digital Audio Interface */ +static struct snd_soc_dai_ops psc_i2s_dai_ops = { + .startup = psc_i2s_startup, + .hw_params = psc_i2s_hw_params, + .hw_free = psc_i2s_hw_free, + .shutdown = psc_i2s_shutdown, + .trigger = psc_i2s_trigger, + .set_sysclk = psc_i2s_set_sysclk, + .set_fmt = psc_i2s_set_fmt, +}; + static struct snd_soc_dai psc_i2s_dai_template = { .playback = { .channels_min = 2, @@ -481,15 +491,7 @@ static struct snd_soc_dai psc_i2s_dai_template = { .rates = PSC_I2S_RATES, .formats = PSC_I2S_FORMATS, }, - .ops = { - .startup = psc_i2s_startup, - .hw_params = psc_i2s_hw_params, - .hw_free = psc_i2s_hw_free, - .shutdown = psc_i2s_shutdown, - .trigger = psc_i2s_trigger, - .set_sysclk = psc_i2s_set_sysclk, - .set_fmt = psc_i2s_set_fmt, - }, + .ops = &psc_i2s_dai_ops, }; /* --------------------------------------------------------------------- diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 05dd5abcddf..d6882be3345 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -461,6 +461,16 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, return err; } +static struct snd_soc_dai_ops omap_mcbsp_dai_ops = { + .startup = omap_mcbsp_dai_startup, + .shutdown = omap_mcbsp_dai_shutdown, + .trigger = omap_mcbsp_dai_trigger, + .hw_params = omap_mcbsp_dai_hw_params, + .set_fmt = omap_mcbsp_dai_set_dai_fmt, + .set_clkdiv = omap_mcbsp_dai_set_clkdiv, + .set_sysclk = omap_mcbsp_dai_set_dai_sysclk, +}; + #define OMAP_MCBSP_DAI_BUILDER(link_id) \ { \ .name = "omap-mcbsp-dai-"#link_id, \ @@ -477,15 +487,7 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, .rates = OMAP_MCBSP_RATES, \ .formats = SNDRV_PCM_FMTBIT_S16_LE, \ }, \ - .ops = { \ - .startup = omap_mcbsp_dai_startup, \ - .shutdown = omap_mcbsp_dai_shutdown, \ - .trigger = omap_mcbsp_dai_trigger, \ - .hw_params = omap_mcbsp_dai_hw_params, \ - .set_fmt = omap_mcbsp_dai_set_dai_fmt, \ - .set_clkdiv = omap_mcbsp_dai_set_clkdiv, \ - .set_sysclk = omap_mcbsp_dai_set_dai_sysclk, \ - }, \ + .ops = &omap_mcbsp_dai_ops, \ .private_data = &mcbsp_data[(link_id)].bus_id, \ } diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 4a973ab710b..3e18064e86b 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -784,6 +784,19 @@ static void pxa_ssp_remove(struct platform_device *pdev, SNDRV_PCM_FMTBIT_S24_LE | \ SNDRV_PCM_FMTBIT_S32_LE) +static struct snd_soc_dai_ops pxa_ssp_dai_ops = { + .startup = pxa_ssp_startup, + .shutdown = pxa_ssp_shutdown, + .trigger = pxa_ssp_trigger, + .hw_params = pxa_ssp_hw_params, + .set_sysclk = pxa_ssp_set_dai_sysclk, + .set_clkdiv = pxa_ssp_set_dai_clkdiv, + .set_pll = pxa_ssp_set_dai_pll, + .set_fmt = pxa_ssp_set_dai_fmt, + .set_tdm_slot = pxa_ssp_set_dai_tdm_slot, + .set_tristate = pxa_ssp_set_dai_tristate, +}; + struct snd_soc_dai pxa_ssp_dai[] = { { .name = "pxa2xx-ssp1", @@ -804,18 +817,7 @@ struct snd_soc_dai pxa_ssp_dai[] = { .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, - .ops = { - .startup = pxa_ssp_startup, - .shutdown = pxa_ssp_shutdown, - .trigger = pxa_ssp_trigger, - .hw_params = pxa_ssp_hw_params, - .set_sysclk = pxa_ssp_set_dai_sysclk, - .set_clkdiv = pxa_ssp_set_dai_clkdiv, - .set_pll = pxa_ssp_set_dai_pll, - .set_fmt = pxa_ssp_set_dai_fmt, - .set_tdm_slot = pxa_ssp_set_dai_tdm_slot, - .set_tristate = pxa_ssp_set_dai_tristate, - }, + .ops = &pxa_ssp_dai_ops, }, { .name = "pxa2xx-ssp2", .id = 1, @@ -835,18 +837,7 @@ struct snd_soc_dai pxa_ssp_dai[] = { .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, - .ops = { - .startup = pxa_ssp_startup, - .shutdown = pxa_ssp_shutdown, - .trigger = pxa_ssp_trigger, - .hw_params = pxa_ssp_hw_params, - .set_sysclk = pxa_ssp_set_dai_sysclk, - .set_clkdiv = pxa_ssp_set_dai_clkdiv, - .set_pll = pxa_ssp_set_dai_pll, - .set_fmt = pxa_ssp_set_dai_fmt, - .set_tdm_slot = pxa_ssp_set_dai_tdm_slot, - .set_tristate = pxa_ssp_set_dai_tristate, - }, + .ops = &pxa_ssp_dai_ops, }, { .name = "pxa2xx-ssp3", @@ -867,18 +858,7 @@ struct snd_soc_dai pxa_ssp_dai[] = { .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, - .ops = { - .startup = pxa_ssp_startup, - .shutdown = pxa_ssp_shutdown, - .trigger = pxa_ssp_trigger, - .hw_params = pxa_ssp_hw_params, - .set_sysclk = pxa_ssp_set_dai_sysclk, - .set_clkdiv = pxa_ssp_set_dai_clkdiv, - .set_pll = pxa_ssp_set_dai_pll, - .set_fmt = pxa_ssp_set_dai_fmt, - .set_tdm_slot = pxa_ssp_set_dai_tdm_slot, - .set_tristate = pxa_ssp_set_dai_tristate, - }, + .ops = &pxa_ssp_dai_ops, }, { .name = "pxa2xx-ssp4", @@ -899,18 +879,7 @@ struct snd_soc_dai pxa_ssp_dai[] = { .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, - .ops = { - .startup = pxa_ssp_startup, - .shutdown = pxa_ssp_shutdown, - .trigger = pxa_ssp_trigger, - .hw_params = pxa_ssp_hw_params, - .set_sysclk = pxa_ssp_set_dai_sysclk, - .set_clkdiv = pxa_ssp_set_dai_clkdiv, - .set_pll = pxa_ssp_set_dai_pll, - .set_fmt = pxa_ssp_set_dai_fmt, - .set_tdm_slot = pxa_ssp_set_dai_tdm_slot, - .set_tristate = pxa_ssp_set_dai_tristate, - }, + .ops = &pxa_ssp_dai_ops, }, }; EXPORT_SYMBOL_GPL(pxa_ssp_dai); diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 812c2b4d3e0..11cd0f289c1 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -164,6 +164,10 @@ static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream, SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000) +static struct snd_soc_dai_ops pxa_ac97_dai_ops = { + .hw_params = pxa2xx_ac97_hw_params, +}; + /* * There is only 1 physical AC97 interface for pxa2xx, but it * has extra fifo's that can be used for aux DACs and ADCs. @@ -189,8 +193,7 @@ struct snd_soc_dai pxa_ac97_dai[] = { .channels_max = 2, .rates = PXA2XX_AC97_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .hw_params = pxa2xx_ac97_hw_params,}, + .ops = &pxa_ac97_dai_ops, }, { .name = "pxa2xx-ac97-aux", @@ -208,8 +211,7 @@ struct snd_soc_dai pxa_ac97_dai[] = { .channels_max = 1, .rates = PXA2XX_AC97_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .hw_params = pxa2xx_ac97_hw_aux_params,}, + .ops = &pxa_ac97_dai_ops, }, { .name = "pxa2xx-ac97-mic", @@ -221,8 +223,7 @@ struct snd_soc_dai pxa_ac97_dai[] = { .channels_max = 1, .rates = PXA2XX_AC97_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .hw_params = pxa2xx_ac97_hw_mic_params,}, + .ops = &pxa_ac97_dai_ops, }, }; diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 83b59d7fe96..e6c24408c5f 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -304,6 +304,15 @@ static int pxa2xx_i2s_resume(struct snd_soc_dai *dai) SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000) +static struct snd_soc_dai_ops pxa_i2s_dai_ops = { + .startup = pxa2xx_i2s_startup, + .shutdown = pxa2xx_i2s_shutdown, + .trigger = pxa2xx_i2s_trigger, + .hw_params = pxa2xx_i2s_hw_params, + .set_fmt = pxa2xx_i2s_set_dai_fmt, + .set_sysclk = pxa2xx_i2s_set_dai_sysclk, +}; + struct snd_soc_dai pxa_i2s_dai = { .name = "pxa2xx-i2s", .id = 0, @@ -319,14 +328,7 @@ struct snd_soc_dai pxa_i2s_dai = { .channels_max = 2, .rates = PXA2XX_I2S_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .startup = pxa2xx_i2s_startup, - .shutdown = pxa2xx_i2s_shutdown, - .trigger = pxa2xx_i2s_trigger, - .hw_params = pxa2xx_i2s_hw_params, - .set_fmt = pxa2xx_i2s_set_dai_fmt, - .set_sysclk = pxa2xx_i2s_set_dai_sysclk, - }, + .ops = &pxa_i2s_dai_ops, }; EXPORT_SYMBOL_GPL(pxa_i2s_dai); diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c index f3fc0aba0aa..382d7eee53e 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.c +++ b/sound/soc/s3c24xx/s3c2412-i2s.c @@ -708,6 +708,14 @@ static int s3c2412_i2s_resume(struct snd_soc_dai *dai) SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) +static struct snd_soc_dai_ops s3c2412_i2s_dai_ops = { + .trigger = s3c2412_i2s_trigger, + .hw_params = s3c2412_i2s_hw_params, + .set_fmt = s3c2412_i2s_set_fmt, + .set_clkdiv = s3c2412_i2s_set_clkdiv, + .set_sysclk = s3c2412_i2s_set_sysclk, +}; + struct snd_soc_dai s3c2412_i2s_dai = { .name = "s3c2412-i2s", .id = 0, @@ -726,13 +734,7 @@ struct snd_soc_dai s3c2412_i2s_dai = { .rates = S3C2412_I2S_RATES, .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE, }, - .ops = { - .trigger = s3c2412_i2s_trigger, - .hw_params = s3c2412_i2s_hw_params, - .set_fmt = s3c2412_i2s_set_fmt, - .set_clkdiv = s3c2412_i2s_set_clkdiv, - .set_sysclk = s3c2412_i2s_set_sysclk, - }, + .ops = &s3c2412_i2s_dai_ops, }; EXPORT_SYMBOL_GPL(s3c2412_i2s_dai); diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c index 5822d2dd49b..83ea623234e 100644 --- a/sound/soc/s3c24xx/s3c2443-ac97.c +++ b/sound/soc/s3c24xx/s3c2443-ac97.c @@ -355,6 +355,11 @@ static int s3c2443_ac97_mic_trigger(struct snd_pcm_substream *substream, SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) +static struct snd_soc_dai_ops s3c2443_ac97_dai_ops = { + .hw_params = s3c2443_ac97_hw_params, + .trigger = s3c2443_ac97_trigger, +}; + struct snd_soc_dai s3c2443_ac97_dai[] = { { .name = "s3c2443-ac97", @@ -374,9 +379,7 @@ struct snd_soc_dai s3c2443_ac97_dai[] = { .channels_max = 2, .rates = s3c2443_AC97_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .hw_params = s3c2443_ac97_hw_params, - .trigger = s3c2443_ac97_trigger}, + .ops = &s3c2443_ac97_dai_ops, }, { .name = "pxa2xx-ac97-mic", @@ -388,9 +391,7 @@ struct snd_soc_dai s3c2443_ac97_dai[] = { .channels_max = 1, .rates = s3c2443_AC97_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .hw_params = s3c2443_ac97_hw_mic_params, - .trigger = s3c2443_ac97_mic_trigger,}, + .ops = &s3c2443_ac97_dai_ops, }, }; EXPORT_SYMBOL_GPL(s3c2443_ac97_dai); diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 1c2b0549710..4473fb584c4 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -456,6 +456,14 @@ static int s3c24xx_i2s_resume(struct snd_soc_dai *cpu_dai) SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) +static struct snd_soc_dai_ops s3c24xx_i2s_dai_ops = { + .trigger = s3c24xx_i2s_trigger, + .hw_params = s3c24xx_i2s_hw_params, + .set_fmt = s3c24xx_i2s_set_fmt, + .set_clkdiv = s3c24xx_i2s_set_clkdiv, + .set_sysclk = s3c24xx_i2s_set_sysclk, +}; + struct snd_soc_dai s3c24xx_i2s_dai = { .name = "s3c24xx-i2s", .id = 0, @@ -472,13 +480,7 @@ struct snd_soc_dai s3c24xx_i2s_dai = { .channels_max = 2, .rates = S3C24XX_I2S_RATES, .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .trigger = s3c24xx_i2s_trigger, - .hw_params = s3c24xx_i2s_hw_params, - .set_fmt = s3c24xx_i2s_set_fmt, - .set_clkdiv = s3c24xx_i2s_set_clkdiv, - .set_sysclk = s3c24xx_i2s_set_sysclk, - }, + .ops = &s3c24xx_i2s_dai_ops, }; EXPORT_SYMBOL_GPL(s3c24xx_i2s_dai); diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c index d1e5390fdde..56fa0872abb 100644 --- a/sound/soc/sh/ssi.c +++ b/sound/soc/sh/ssi.c @@ -336,6 +336,16 @@ static int ssi_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_U24_3LE | \ SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_LE) +static struct snd_soc_dai_ops ssi_dai_ops = { + .startup = ssi_startup, + .shutdown = ssi_shutdown, + .trigger = ssi_trigger, + .hw_params = ssi_hw_params, + .set_sysclk = ssi_set_sysclk, + .set_clkdiv = ssi_set_clkdiv, + .set_fmt = ssi_set_fmt, +}; + struct snd_soc_dai sh4_ssi_dai[] = { { .name = "SSI0", @@ -352,15 +362,7 @@ struct snd_soc_dai sh4_ssi_dai[] = { .channels_min = 2, .channels_max = 8, }, - .ops = { - .startup = ssi_startup, - .shutdown = ssi_shutdown, - .trigger = ssi_trigger, - .hw_params = ssi_hw_params, - .set_sysclk = ssi_set_sysclk, - .set_clkdiv = ssi_set_clkdiv, - .set_fmt = ssi_set_fmt, - }, + .ops = &ssi_dai_ops, }, #ifdef CONFIG_CPU_SUBTYPE_SH7760 { @@ -378,15 +380,7 @@ struct snd_soc_dai sh4_ssi_dai[] = { .channels_min = 2, .channels_max = 8, }, - .ops = { - .startup = ssi_startup, - .shutdown = ssi_shutdown, - .trigger = ssi_trigger, - .hw_params = ssi_hw_params, - .set_sysclk = ssi_set_sysclk, - .set_clkdiv = ssi_set_clkdiv, - .set_fmt = ssi_set_fmt, - }, + .ops = &ssi_dai_ops, }, #endif }; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d4b90d82a09..16518329f6b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -133,8 +133,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) mutex_lock(&pcm_mutex); /* startup the audio subsystem */ - if (cpu_dai->ops.startup) { - ret = cpu_dai->ops.startup(substream, cpu_dai); + if (cpu_dai->ops->startup) { + ret = cpu_dai->ops->startup(substream, cpu_dai); if (ret < 0) { printk(KERN_ERR "asoc: can't open interface %s\n", cpu_dai->name); @@ -150,8 +150,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) } } - if (codec_dai->ops.startup) { - ret = codec_dai->ops.startup(substream, codec_dai); + if (codec_dai->ops->startup) { + ret = codec_dai->ops->startup(substream, codec_dai); if (ret < 0) { printk(KERN_ERR "asoc: can't open codec %s\n", codec_dai->name); @@ -247,8 +247,8 @@ codec_dai_err: platform->pcm_ops->close(substream); platform_err: - if (cpu_dai->ops.shutdown) - cpu_dai->ops.shutdown(substream, cpu_dai); + if (cpu_dai->ops->shutdown) + cpu_dai->ops->shutdown(substream, cpu_dai); out: mutex_unlock(&pcm_mutex); return ret; @@ -340,11 +340,11 @@ static int soc_codec_close(struct snd_pcm_substream *substream) if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) snd_soc_dai_digital_mute(codec_dai, 1); - if (cpu_dai->ops.shutdown) - cpu_dai->ops.shutdown(substream, cpu_dai); + if (cpu_dai->ops->shutdown) + cpu_dai->ops->shutdown(substream, cpu_dai); - if (codec_dai->ops.shutdown) - codec_dai->ops.shutdown(substream, codec_dai); + if (codec_dai->ops->shutdown) + codec_dai->ops->shutdown(substream, codec_dai); if (machine->ops && machine->ops->shutdown) machine->ops->shutdown(substream); @@ -408,16 +408,16 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) } } - if (codec_dai->ops.prepare) { - ret = codec_dai->ops.prepare(substream, codec_dai); + if (codec_dai->ops->prepare) { + ret = codec_dai->ops->prepare(substream, codec_dai); if (ret < 0) { printk(KERN_ERR "asoc: codec DAI prepare error\n"); goto out; } } - if (cpu_dai->ops.prepare) { - ret = cpu_dai->ops.prepare(substream, cpu_dai); + if (cpu_dai->ops->prepare) { + ret = cpu_dai->ops->prepare(substream, cpu_dai); if (ret < 0) { printk(KERN_ERR "asoc: cpu DAI prepare error\n"); goto out; @@ -494,8 +494,8 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, } } - if (codec_dai->ops.hw_params) { - ret = codec_dai->ops.hw_params(substream, params, codec_dai); + if (codec_dai->ops->hw_params) { + ret = codec_dai->ops->hw_params(substream, params, codec_dai); if (ret < 0) { printk(KERN_ERR "asoc: can't set codec %s hw params\n", codec_dai->name); @@ -503,8 +503,8 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, } } - if (cpu_dai->ops.hw_params) { - ret = cpu_dai->ops.hw_params(substream, params, cpu_dai); + if (cpu_dai->ops->hw_params) { + ret = cpu_dai->ops->hw_params(substream, params, cpu_dai); if (ret < 0) { printk(KERN_ERR "asoc: interface %s hw params failed\n", cpu_dai->name); @@ -526,12 +526,12 @@ out: return ret; platform_err: - if (cpu_dai->ops.hw_free) - cpu_dai->ops.hw_free(substream, cpu_dai); + if (cpu_dai->ops->hw_free) + cpu_dai->ops->hw_free(substream, cpu_dai); interface_err: - if (codec_dai->ops.hw_free) - codec_dai->ops.hw_free(substream, codec_dai); + if (codec_dai->ops->hw_free) + codec_dai->ops->hw_free(substream, codec_dai); codec_err: if (machine->ops && machine->ops->hw_free) @@ -570,11 +570,11 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) platform->pcm_ops->hw_free(substream); /* now free hw params for the DAI's */ - if (codec_dai->ops.hw_free) - codec_dai->ops.hw_free(substream, codec_dai); + if (codec_dai->ops->hw_free) + codec_dai->ops->hw_free(substream, codec_dai); - if (cpu_dai->ops.hw_free) - cpu_dai->ops.hw_free(substream, cpu_dai); + if (cpu_dai->ops->hw_free) + cpu_dai->ops->hw_free(substream, cpu_dai); mutex_unlock(&pcm_mutex); return 0; @@ -591,8 +591,8 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) struct snd_soc_dai *codec_dai = machine->codec_dai; int ret; - if (codec_dai->ops.trigger) { - ret = codec_dai->ops.trigger(substream, cmd, codec_dai); + if (codec_dai->ops->trigger) { + ret = codec_dai->ops->trigger(substream, cmd, codec_dai); if (ret < 0) return ret; } @@ -603,8 +603,8 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) return ret; } - if (cpu_dai->ops.trigger) { - ret = cpu_dai->ops.trigger(substream, cmd, cpu_dai); + if (cpu_dai->ops->trigger) { + ret = cpu_dai->ops->trigger(substream, cmd, cpu_dai); if (ret < 0) return ret; } @@ -645,8 +645,8 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) /* mute any active DAC's */ for (i = 0; i < card->num_links; i++) { struct snd_soc_dai *dai = card->dai_link[i].codec_dai; - if (dai->ops.digital_mute && dai->playback.active) - dai->ops.digital_mute(dai, 1); + if (dai->ops->digital_mute && dai->playback.active) + dai->ops->digital_mute(dai, 1); } /* suspend all pcms */ @@ -741,8 +741,8 @@ static void soc_resume_deferred(struct work_struct *work) /* unmute any active DACs */ for (i = 0; i < card->num_links; i++) { struct snd_soc_dai *dai = card->dai_link[i].codec_dai; - if (dai->ops.digital_mute && dai->playback.active) - dai->ops.digital_mute(dai, 0); + if (dai->ops->digital_mute && dai->playback.active) + dai->ops->digital_mute(dai, 0); } for (i = 0; i < card->num_links; i++) { @@ -2051,8 +2051,8 @@ EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8); int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int freq, int dir) { - if (dai->ops.set_sysclk) - return dai->ops.set_sysclk(dai, clk_id, freq, dir); + if (dai->ops->set_sysclk) + return dai->ops->set_sysclk(dai, clk_id, freq, dir); else return -EINVAL; } @@ -2071,8 +2071,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk); int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) { - if (dai->ops.set_clkdiv) - return dai->ops.set_clkdiv(dai, div_id, div); + if (dai->ops->set_clkdiv) + return dai->ops->set_clkdiv(dai, div_id, div); else return -EINVAL; } @@ -2090,8 +2090,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_clkdiv); int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int pll_id, unsigned int freq_in, unsigned int freq_out) { - if (dai->ops.set_pll) - return dai->ops.set_pll(dai, pll_id, freq_in, freq_out); + if (dai->ops->set_pll) + return dai->ops->set_pll(dai, pll_id, freq_in, freq_out); else return -EINVAL; } @@ -2106,8 +2106,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll); */ int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { - if (dai->ops.set_fmt) - return dai->ops.set_fmt(dai, fmt); + if (dai->ops->set_fmt) + return dai->ops->set_fmt(dai, fmt); else return -EINVAL; } @@ -2125,8 +2125,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt); int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, unsigned int mask, int slots) { - if (dai->ops.set_sysclk) - return dai->ops.set_tdm_slot(dai, mask, slots); + if (dai->ops->set_sysclk) + return dai->ops->set_tdm_slot(dai, mask, slots); else return -EINVAL; } @@ -2141,8 +2141,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot); */ int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate) { - if (dai->ops.set_sysclk) - return dai->ops.set_tristate(dai, tristate); + if (dai->ops->set_sysclk) + return dai->ops->set_tristate(dai, tristate); else return -EINVAL; } @@ -2157,8 +2157,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_tristate); */ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute) { - if (dai->ops.digital_mute) - return dai->ops.digital_mute(dai, mute); + if (dai->ops->digital_mute) + return dai->ops->digital_mute(dai, mute); else return -EINVAL; } @@ -2211,6 +2211,9 @@ static int snd_soc_unregister_card(struct snd_soc_card *card) return 0; } +static struct snd_soc_dai_ops null_dai_ops = { +}; + /** * snd_soc_register_dai - Register a DAI with the ASoC core * @@ -2225,6 +2228,9 @@ int snd_soc_register_dai(struct snd_soc_dai *dai) if (!dai->dev) printk(KERN_WARNING "No device for DAI %s\n", dai->name); + if (!dai->ops) + dai->ops = &null_dai_ops; + INIT_LIST_HEAD(&dai->list); mutex_lock(&client_mutex); -- cgit v1.2.3-70-g09d2 From c2503cd3be9eacb1dd06ec5b6fba8bb06aac12a8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 5 Mar 2009 09:37:40 +0100 Subject: ALSA: hdsp - Ignore MIDI and PCM events in interrupts until initialized Ignore MIDI and PCM events in the interrupt handler until the device gets initialized properly. Otherwise you may get kernel panic by the access to uninitialized devices via hotplugging. Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdsp.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index dc65fe1c9c6..314e73531bd 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -3740,6 +3740,9 @@ static irqreturn_t snd_hdsp_interrupt(int irq, void *dev_id) midi0status = hdsp_read (hdsp, HDSP_midiStatusIn0) & 0xff; midi1status = hdsp_read (hdsp, HDSP_midiStatusIn1) & 0xff; + if (!(hdsp->state & HDSP_InitializationComplete)) + return IRQ_HANDLED; + if (audio) { if (hdsp->capture_substream) snd_pcm_period_elapsed(hdsp->pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream); -- cgit v1.2.3-70-g09d2 From 37db623ae2a7bde234a8ed683d0d13d6f939199c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 5 Mar 2009 09:40:16 +0100 Subject: ALSA: hda - Fix check of ALC888S-VC in alc888_coef_init() Fixed the wrong bits check to identify ALC888S-VC model in alc888_coef_init(). Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 91ef9f27b12..6325ea43cf0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -982,7 +982,7 @@ static void alc888_coef_init(struct hda_codec *codec) snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 0); tmp = snd_hda_codec_read(codec, 0x20, 0, AC_VERB_GET_PROC_COEF, 0); snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 7); - if ((tmp & 0xf0) == 2) + if ((tmp & 0xf0) == 0x20) /* alc888S-VC */ snd_hda_codec_read(codec, 0x20, 0, AC_VERB_SET_PROC_COEF, 0x830); -- cgit v1.2.3-70-g09d2 From 8150bc886be5ce3cc301a2baca1fcf2cf7bd7f39 Mon Sep 17 00:00:00 2001 From: Ben Dooks Date: Wed, 4 Mar 2009 00:49:26 +0000 Subject: S3C24XX: Move and update IIS headers Move the IIS headers to their correct place. Signed-off-by: Ben Dooks Signed-off-by: Mark Brown --- arch/arm/mach-s3c2410/dma.c | 2 +- arch/arm/mach-s3c2410/include/mach/hardware.h | 3 - arch/arm/mach-s3c2410/include/mach/io.h | 2 +- arch/arm/mach-s3c2412/dma.c | 4 +- arch/arm/mach-s3c2440/dma.c | 2 +- arch/arm/mach-s3c2443/dma.c | 2 +- arch/arm/mach-shark/include/mach/io.h | 2 +- arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h | 72 +++++++++++++++++++++ arch/arm/plat-s3c24xx/clock-dclk.c | 1 + arch/arm/plat-s3c24xx/include/plat/regs-iis.h | 77 +++++++++++++++++++++++ include/asm-arm/plat-s3c24xx/regs-iis.h | 77 ----------------------- include/asm-arm/plat-s3c24xx/regs-s3c2412-iis.h | 72 --------------------- sound/soc/s3c24xx/neo1973_wm8753.c | 2 +- sound/soc/s3c24xx/s3c2412-i2s.c | 6 +- sound/soc/s3c24xx/s3c24xx-i2s.c | 2 +- sound/soc/s3c24xx/s3c24xx_uda134x.c | 2 +- 16 files changed, 162 insertions(+), 166 deletions(-) create mode 100644 arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h create mode 100644 arch/arm/plat-s3c24xx/include/plat/regs-iis.h delete mode 100644 include/asm-arm/plat-s3c24xx/regs-iis.h delete mode 100644 include/asm-arm/plat-s3c24xx/regs-s3c2412-iis.h diff --git a/arch/arm/mach-s3c2410/dma.c b/arch/arm/mach-s3c2410/dma.c index 552b4c778fd..440c014e24b 100644 --- a/arch/arm/mach-s3c2410/dma.c +++ b/arch/arm/mach-s3c2410/dma.c @@ -28,7 +28,7 @@ #include #include #include -#include +#include #include static struct s3c24xx_dma_map __initdata s3c2410_dma_mappings[] = { diff --git a/arch/arm/mach-s3c2410/include/mach/hardware.h b/arch/arm/mach-s3c2410/include/mach/hardware.h index 74d5a1a4024..db72beb61d7 100644 --- a/arch/arm/mach-s3c2410/include/mach/hardware.h +++ b/arch/arm/mach-s3c2410/include/mach/hardware.h @@ -131,7 +131,4 @@ extern int s3c2412_gpio_set_sleepcfg(unsigned int pin, unsigned int state); /* machine specific hardware definitions should go after this */ -/* currently here until moved into config (todo) */ -#define CONFIG_NO_MULTIWORD_IO - #endif /* __ASM_ARCH_HARDWARE_H */ diff --git a/arch/arm/mach-s3c2410/include/mach/io.h b/arch/arm/mach-s3c2410/include/mach/io.h index 9813dbf2ae4..c477771c092 100644 --- a/arch/arm/mach-s3c2410/include/mach/io.h +++ b/arch/arm/mach-s3c2410/include/mach/io.h @@ -9,7 +9,7 @@ #ifndef __ASM_ARM_ARCH_IO_H #define __ASM_ARM_ARCH_IO_H -#include +#include #define IO_SPACE_LIMIT 0xffffffff diff --git a/arch/arm/mach-s3c2412/dma.c b/arch/arm/mach-s3c2412/dma.c index 919856c9433..9e3478506c6 100644 --- a/arch/arm/mach-s3c2412/dma.c +++ b/arch/arm/mach-s3c2412/dma.c @@ -29,8 +29,8 @@ #include #include #include -#include -#include +#include +#include #include #define MAP(x) { (x)| DMA_CH_VALID, (x)| DMA_CH_VALID, (x)| DMA_CH_VALID, (x)| DMA_CH_VALID } diff --git a/arch/arm/mach-s3c2440/dma.c b/arch/arm/mach-s3c2440/dma.c index 5b5ee0b8f4e..69b6cf34df4 100644 --- a/arch/arm/mach-s3c2440/dma.c +++ b/arch/arm/mach-s3c2440/dma.c @@ -28,7 +28,7 @@ #include #include #include -#include +#include #include static struct s3c24xx_dma_map __initdata s3c2440_dma_mappings[] = { diff --git a/arch/arm/mach-s3c2443/dma.c b/arch/arm/mach-s3c2443/dma.c index 2a58a4d5aa5..8430e582918 100644 --- a/arch/arm/mach-s3c2443/dma.c +++ b/arch/arm/mach-s3c2443/dma.c @@ -29,7 +29,7 @@ #include #include #include -#include +#include #include #define MAP(x) { \ diff --git a/arch/arm/mach-shark/include/mach/io.h b/arch/arm/mach-shark/include/mach/io.h index c5cee829fc8..8ca7d7f09bd 100644 --- a/arch/arm/mach-shark/include/mach/io.h +++ b/arch/arm/mach-shark/include/mach/io.h @@ -14,7 +14,7 @@ #define PCIO_BASE 0xe0000000 #define IO_SPACE_LIMIT 0xffffffff -#define __io(a) ((void __iomem *)(PCIO_BASE + (a))) +#define __io(a) __typesafe_io(PCIO_BASE + (a)) #define __mem_pci(addr) (addr) #endif diff --git a/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h b/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h new file mode 100644 index 00000000000..25d4058bcfe --- /dev/null +++ b/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h @@ -0,0 +1,72 @@ +/* linux/include/asm-arm/plat-s3c24xx/regs-s3c2412-iis.h + * + * Copyright 2007 Simtec Electronics + * http://armlinux.simtec.co.uk/ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * S3C2412 IIS register definition +*/ + +#ifndef __ASM_ARCH_REGS_S3C2412_IIS_H +#define __ASM_ARCH_REGS_S3C2412_IIS_H + +#define S3C2412_IISCON (0x00) +#define S3C2412_IISMOD (0x04) +#define S3C2412_IISFIC (0x08) +#define S3C2412_IISPSR (0x0C) +#define S3C2412_IISTXD (0x10) +#define S3C2412_IISRXD (0x14) + +#define S3C2412_IISCON_LRINDEX (1 << 11) +#define S3C2412_IISCON_TXFIFO_EMPTY (1 << 10) +#define S3C2412_IISCON_RXFIFO_EMPTY (1 << 9) +#define S3C2412_IISCON_TXFIFO_FULL (1 << 8) +#define S3C2412_IISCON_RXFIFO_FULL (1 << 7) +#define S3C2412_IISCON_TXDMA_PAUSE (1 << 6) +#define S3C2412_IISCON_RXDMA_PAUSE (1 << 5) +#define S3C2412_IISCON_TXCH_PAUSE (1 << 4) +#define S3C2412_IISCON_RXCH_PAUSE (1 << 3) +#define S3C2412_IISCON_TXDMA_ACTIVE (1 << 2) +#define S3C2412_IISCON_RXDMA_ACTIVE (1 << 1) +#define S3C2412_IISCON_IIS_ACTIVE (1 << 0) + +#define S3C2412_IISMOD_MASTER_INTERNAL (0 << 10) +#define S3C2412_IISMOD_MASTER_EXTERNAL (1 << 10) +#define S3C2412_IISMOD_SLAVE (2 << 10) +#define S3C2412_IISMOD_MASTER_MASK (3 << 10) +#define S3C2412_IISMOD_MODE_TXONLY (0 << 8) +#define S3C2412_IISMOD_MODE_RXONLY (1 << 8) +#define S3C2412_IISMOD_MODE_TXRX (2 << 8) +#define S3C2412_IISMOD_MODE_MASK (3 << 8) +#define S3C2412_IISMOD_LR_LLOW (0 << 7) +#define S3C2412_IISMOD_LR_RLOW (1 << 7) +#define S3C2412_IISMOD_SDF_IIS (0 << 5) +#define S3C2412_IISMOD_SDF_MSB (0 << 5) +#define S3C2412_IISMOD_SDF_LSB (0 << 5) +#define S3C2412_IISMOD_SDF_MASK (3 << 5) +#define S3C2412_IISMOD_RCLK_256FS (0 << 3) +#define S3C2412_IISMOD_RCLK_512FS (1 << 3) +#define S3C2412_IISMOD_RCLK_384FS (2 << 3) +#define S3C2412_IISMOD_RCLK_768FS (3 << 3) +#define S3C2412_IISMOD_RCLK_MASK (3 << 3) +#define S3C2412_IISMOD_BCLK_32FS (0 << 1) +#define S3C2412_IISMOD_BCLK_48FS (1 << 1) +#define S3C2412_IISMOD_BCLK_16FS (2 << 1) +#define S3C2412_IISMOD_BCLK_24FS (3 << 1) +#define S3C2412_IISMOD_BCLK_MASK (3 << 1) +#define S3C2412_IISMOD_8BIT (1 << 0) + +#define S3C2412_IISPSR_PSREN (1 << 15) + +#define S3C2412_IISFIC_TXFLUSH (1 << 15) +#define S3C2412_IISFIC_RXFLUSH (1 << 7) +#define S3C2412_IISFIC_TXCOUNT(x) (((x) >> 8) & 0xf) +#define S3C2412_IISFIC_RXCOUNT(x) (((x) >> 0) & 0xf) + + + +#endif /* __ASM_ARCH_REGS_S3C2412_IIS_H */ + diff --git a/arch/arm/plat-s3c24xx/clock-dclk.c b/arch/arm/plat-s3c24xx/clock-dclk.c index 5b75a797b5a..35219dcf9f0 100644 --- a/arch/arm/plat-s3c24xx/clock-dclk.c +++ b/arch/arm/plat-s3c24xx/clock-dclk.c @@ -18,6 +18,7 @@ #include #include +#include #include #include diff --git a/arch/arm/plat-s3c24xx/include/plat/regs-iis.h b/arch/arm/plat-s3c24xx/include/plat/regs-iis.h new file mode 100644 index 00000000000..a6f1d5df13b --- /dev/null +++ b/arch/arm/plat-s3c24xx/include/plat/regs-iis.h @@ -0,0 +1,77 @@ +/* arch/arm/mach-s3c2410/include/mach/regs-iis.h + * + * Copyright (c) 2003 Simtec Electronics + * http://www.simtec.co.uk/products/SWLINUX/ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * S3C2410 IIS register definition +*/ + +#ifndef __ASM_ARCH_REGS_IIS_H +#define __ASM_ARCH_REGS_IIS_H + +#define S3C2410_IISCON (0x00) + +#define S3C2410_IISCON_LRINDEX (1<<8) +#define S3C2410_IISCON_TXFIFORDY (1<<7) +#define S3C2410_IISCON_RXFIFORDY (1<<6) +#define S3C2410_IISCON_TXDMAEN (1<<5) +#define S3C2410_IISCON_RXDMAEN (1<<4) +#define S3C2410_IISCON_TXIDLE (1<<3) +#define S3C2410_IISCON_RXIDLE (1<<2) +#define S3C2410_IISCON_PSCEN (1<<1) +#define S3C2410_IISCON_IISEN (1<<0) + +#define S3C2410_IISMOD (0x04) + +#define S3C2440_IISMOD_MPLL (1<<9) +#define S3C2410_IISMOD_SLAVE (1<<8) +#define S3C2410_IISMOD_NOXFER (0<<6) +#define S3C2410_IISMOD_RXMODE (1<<6) +#define S3C2410_IISMOD_TXMODE (2<<6) +#define S3C2410_IISMOD_TXRXMODE (3<<6) +#define S3C2410_IISMOD_LR_LLOW (0<<5) +#define S3C2410_IISMOD_LR_RLOW (1<<5) +#define S3C2410_IISMOD_IIS (0<<4) +#define S3C2410_IISMOD_MSB (1<<4) +#define S3C2410_IISMOD_8BIT (0<<3) +#define S3C2410_IISMOD_16BIT (1<<3) +#define S3C2410_IISMOD_BITMASK (1<<3) +#define S3C2410_IISMOD_256FS (0<<2) +#define S3C2410_IISMOD_384FS (1<<2) +#define S3C2410_IISMOD_16FS (0<<0) +#define S3C2410_IISMOD_32FS (1<<0) +#define S3C2410_IISMOD_48FS (2<<0) +#define S3C2410_IISMOD_FS_MASK (3<<0) + +#define S3C2410_IISPSR (0x08) +#define S3C2410_IISPSR_INTMASK (31<<5) +#define S3C2410_IISPSR_INTSHIFT (5) +#define S3C2410_IISPSR_EXTMASK (31<<0) +#define S3C2410_IISPSR_EXTSHFIT (0) + +#define S3C2410_IISFCON (0x0c) + +#define S3C2410_IISFCON_TXDMA (1<<15) +#define S3C2410_IISFCON_RXDMA (1<<14) +#define S3C2410_IISFCON_TXENABLE (1<<13) +#define S3C2410_IISFCON_RXENABLE (1<<12) +#define S3C2410_IISFCON_TXMASK (0x3f << 6) +#define S3C2410_IISFCON_TXSHIFT (6) +#define S3C2410_IISFCON_RXMASK (0x3f) +#define S3C2410_IISFCON_RXSHIFT (0) + +#define S3C2400_IISFCON_TXDMA (1<<11) +#define S3C2400_IISFCON_RXDMA (1<<10) +#define S3C2400_IISFCON_TXENABLE (1<<9) +#define S3C2400_IISFCON_RXENABLE (1<<8) +#define S3C2400_IISFCON_TXMASK (0x07 << 4) +#define S3C2400_IISFCON_TXSHIFT (4) +#define S3C2400_IISFCON_RXMASK (0x07) +#define S3C2400_IISFCON_RXSHIFT (0) + +#define S3C2410_IISFIFO (0x10) +#endif /* __ASM_ARCH_REGS_IIS_H */ diff --git a/include/asm-arm/plat-s3c24xx/regs-iis.h b/include/asm-arm/plat-s3c24xx/regs-iis.h deleted file mode 100644 index a6f1d5df13b..00000000000 --- a/include/asm-arm/plat-s3c24xx/regs-iis.h +++ /dev/null @@ -1,77 +0,0 @@ -/* arch/arm/mach-s3c2410/include/mach/regs-iis.h - * - * Copyright (c) 2003 Simtec Electronics - * http://www.simtec.co.uk/products/SWLINUX/ - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - * - * S3C2410 IIS register definition -*/ - -#ifndef __ASM_ARCH_REGS_IIS_H -#define __ASM_ARCH_REGS_IIS_H - -#define S3C2410_IISCON (0x00) - -#define S3C2410_IISCON_LRINDEX (1<<8) -#define S3C2410_IISCON_TXFIFORDY (1<<7) -#define S3C2410_IISCON_RXFIFORDY (1<<6) -#define S3C2410_IISCON_TXDMAEN (1<<5) -#define S3C2410_IISCON_RXDMAEN (1<<4) -#define S3C2410_IISCON_TXIDLE (1<<3) -#define S3C2410_IISCON_RXIDLE (1<<2) -#define S3C2410_IISCON_PSCEN (1<<1) -#define S3C2410_IISCON_IISEN (1<<0) - -#define S3C2410_IISMOD (0x04) - -#define S3C2440_IISMOD_MPLL (1<<9) -#define S3C2410_IISMOD_SLAVE (1<<8) -#define S3C2410_IISMOD_NOXFER (0<<6) -#define S3C2410_IISMOD_RXMODE (1<<6) -#define S3C2410_IISMOD_TXMODE (2<<6) -#define S3C2410_IISMOD_TXRXMODE (3<<6) -#define S3C2410_IISMOD_LR_LLOW (0<<5) -#define S3C2410_IISMOD_LR_RLOW (1<<5) -#define S3C2410_IISMOD_IIS (0<<4) -#define S3C2410_IISMOD_MSB (1<<4) -#define S3C2410_IISMOD_8BIT (0<<3) -#define S3C2410_IISMOD_16BIT (1<<3) -#define S3C2410_IISMOD_BITMASK (1<<3) -#define S3C2410_IISMOD_256FS (0<<2) -#define S3C2410_IISMOD_384FS (1<<2) -#define S3C2410_IISMOD_16FS (0<<0) -#define S3C2410_IISMOD_32FS (1<<0) -#define S3C2410_IISMOD_48FS (2<<0) -#define S3C2410_IISMOD_FS_MASK (3<<0) - -#define S3C2410_IISPSR (0x08) -#define S3C2410_IISPSR_INTMASK (31<<5) -#define S3C2410_IISPSR_INTSHIFT (5) -#define S3C2410_IISPSR_EXTMASK (31<<0) -#define S3C2410_IISPSR_EXTSHFIT (0) - -#define S3C2410_IISFCON (0x0c) - -#define S3C2410_IISFCON_TXDMA (1<<15) -#define S3C2410_IISFCON_RXDMA (1<<14) -#define S3C2410_IISFCON_TXENABLE (1<<13) -#define S3C2410_IISFCON_RXENABLE (1<<12) -#define S3C2410_IISFCON_TXMASK (0x3f << 6) -#define S3C2410_IISFCON_TXSHIFT (6) -#define S3C2410_IISFCON_RXMASK (0x3f) -#define S3C2410_IISFCON_RXSHIFT (0) - -#define S3C2400_IISFCON_TXDMA (1<<11) -#define S3C2400_IISFCON_RXDMA (1<<10) -#define S3C2400_IISFCON_TXENABLE (1<<9) -#define S3C2400_IISFCON_RXENABLE (1<<8) -#define S3C2400_IISFCON_TXMASK (0x07 << 4) -#define S3C2400_IISFCON_TXSHIFT (4) -#define S3C2400_IISFCON_RXMASK (0x07) -#define S3C2400_IISFCON_RXSHIFT (0) - -#define S3C2410_IISFIFO (0x10) -#endif /* __ASM_ARCH_REGS_IIS_H */ diff --git a/include/asm-arm/plat-s3c24xx/regs-s3c2412-iis.h b/include/asm-arm/plat-s3c24xx/regs-s3c2412-iis.h deleted file mode 100644 index 25d4058bcfe..00000000000 --- a/include/asm-arm/plat-s3c24xx/regs-s3c2412-iis.h +++ /dev/null @@ -1,72 +0,0 @@ -/* linux/include/asm-arm/plat-s3c24xx/regs-s3c2412-iis.h - * - * Copyright 2007 Simtec Electronics - * http://armlinux.simtec.co.uk/ - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - * - * S3C2412 IIS register definition -*/ - -#ifndef __ASM_ARCH_REGS_S3C2412_IIS_H -#define __ASM_ARCH_REGS_S3C2412_IIS_H - -#define S3C2412_IISCON (0x00) -#define S3C2412_IISMOD (0x04) -#define S3C2412_IISFIC (0x08) -#define S3C2412_IISPSR (0x0C) -#define S3C2412_IISTXD (0x10) -#define S3C2412_IISRXD (0x14) - -#define S3C2412_IISCON_LRINDEX (1 << 11) -#define S3C2412_IISCON_TXFIFO_EMPTY (1 << 10) -#define S3C2412_IISCON_RXFIFO_EMPTY (1 << 9) -#define S3C2412_IISCON_TXFIFO_FULL (1 << 8) -#define S3C2412_IISCON_RXFIFO_FULL (1 << 7) -#define S3C2412_IISCON_TXDMA_PAUSE (1 << 6) -#define S3C2412_IISCON_RXDMA_PAUSE (1 << 5) -#define S3C2412_IISCON_TXCH_PAUSE (1 << 4) -#define S3C2412_IISCON_RXCH_PAUSE (1 << 3) -#define S3C2412_IISCON_TXDMA_ACTIVE (1 << 2) -#define S3C2412_IISCON_RXDMA_ACTIVE (1 << 1) -#define S3C2412_IISCON_IIS_ACTIVE (1 << 0) - -#define S3C2412_IISMOD_MASTER_INTERNAL (0 << 10) -#define S3C2412_IISMOD_MASTER_EXTERNAL (1 << 10) -#define S3C2412_IISMOD_SLAVE (2 << 10) -#define S3C2412_IISMOD_MASTER_MASK (3 << 10) -#define S3C2412_IISMOD_MODE_TXONLY (0 << 8) -#define S3C2412_IISMOD_MODE_RXONLY (1 << 8) -#define S3C2412_IISMOD_MODE_TXRX (2 << 8) -#define S3C2412_IISMOD_MODE_MASK (3 << 8) -#define S3C2412_IISMOD_LR_LLOW (0 << 7) -#define S3C2412_IISMOD_LR_RLOW (1 << 7) -#define S3C2412_IISMOD_SDF_IIS (0 << 5) -#define S3C2412_IISMOD_SDF_MSB (0 << 5) -#define S3C2412_IISMOD_SDF_LSB (0 << 5) -#define S3C2412_IISMOD_SDF_MASK (3 << 5) -#define S3C2412_IISMOD_RCLK_256FS (0 << 3) -#define S3C2412_IISMOD_RCLK_512FS (1 << 3) -#define S3C2412_IISMOD_RCLK_384FS (2 << 3) -#define S3C2412_IISMOD_RCLK_768FS (3 << 3) -#define S3C2412_IISMOD_RCLK_MASK (3 << 3) -#define S3C2412_IISMOD_BCLK_32FS (0 << 1) -#define S3C2412_IISMOD_BCLK_48FS (1 << 1) -#define S3C2412_IISMOD_BCLK_16FS (2 << 1) -#define S3C2412_IISMOD_BCLK_24FS (3 << 1) -#define S3C2412_IISMOD_BCLK_MASK (3 << 1) -#define S3C2412_IISMOD_8BIT (1 << 0) - -#define S3C2412_IISPSR_PSREN (1 << 15) - -#define S3C2412_IISFIC_TXFLUSH (1 << 15) -#define S3C2412_IISFIC_RXFLUSH (1 << 7) -#define S3C2412_IISFIC_TXCOUNT(x) (((x) >> 8) & 0xf) -#define S3C2412_IISFIC_RXCOUNT(x) (((x) >> 0) & 0xf) - - - -#endif /* __ASM_ARCH_REGS_S3C2412_IIS_H */ - diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index 45bb12e8ea4..286ff4497fd 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -33,7 +33,7 @@ #include #include -#include +#include #include "../codecs/wm8753.h" #include "lm4857.h" diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c index f3fc0aba0aa..36b927d3f54 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.c +++ b/sound/soc/s3c24xx/s3c2412-i2s.c @@ -22,6 +22,7 @@ #include #include #include +#include #include #include @@ -30,10 +31,7 @@ #include #include -#include -#include - -#include +#include #include #include diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 6f4d439b57a..2569b910b9b 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -34,7 +34,7 @@ #include #include -#include +#include #include "s3c24xx-pcm.h" #include "s3c24xx-i2s.h" diff --git a/sound/soc/s3c24xx/s3c24xx_uda134x.c b/sound/soc/s3c24xx/s3c24xx_uda134x.c index a0a4d1832a1..8e79a416db5 100644 --- a/sound/soc/s3c24xx/s3c24xx_uda134x.c +++ b/sound/soc/s3c24xx/s3c24xx_uda134x.c @@ -22,7 +22,7 @@ #include #include -#include +#include #include "s3c24xx-pcm.h" #include "s3c24xx-i2s.h" -- cgit v1.2.3-70-g09d2 From 899e6cf5e6d83a91d2e257f7a4e8ca98db3831cc Mon Sep 17 00:00:00 2001 From: Ben Dooks Date: Wed, 4 Mar 2009 00:49:28 +0000 Subject: S3C: Move to The file needs to be common to both ARCH_S3C2410 and ARCH_S3C64XX as they share common driver code, so move it to . Signed-off-by: Ben Dooks Signed-off-by: Mark Brown --- arch/arm/mach-s3c2410/include/mach/audio.h | 45 ------------------------------ arch/arm/plat-s3c/include/plat/audio.h | 45 ++++++++++++++++++++++++++++++ sound/soc/s3c24xx/neo1973_wm8753.c | 2 +- sound/soc/s3c24xx/s3c2412-i2s.c | 2 +- sound/soc/s3c24xx/s3c2443-ac97.c | 2 +- sound/soc/s3c24xx/s3c24xx-i2s.c | 2 +- sound/soc/s3c24xx/s3c24xx-pcm.c | 2 +- 7 files changed, 50 insertions(+), 50 deletions(-) delete mode 100644 arch/arm/mach-s3c2410/include/mach/audio.h create mode 100644 arch/arm/plat-s3c/include/plat/audio.h diff --git a/arch/arm/mach-s3c2410/include/mach/audio.h b/arch/arm/mach-s3c2410/include/mach/audio.h deleted file mode 100644 index de0e8da48bc..00000000000 --- a/arch/arm/mach-s3c2410/include/mach/audio.h +++ /dev/null @@ -1,45 +0,0 @@ -/* arch/arm/mach-s3c2410/include/mach/audio.h - * - * Copyright (c) 2004-2005 Simtec Electronics - * http://www.simtec.co.uk/products/SWLINUX/ - * Ben Dooks - * - * S3C24XX - Audio platfrom_device info - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. -*/ - -#ifndef __ASM_ARCH_AUDIO_H -#define __ASM_ARCH_AUDIO_H __FILE__ - -/* struct s3c24xx_iis_ops - * - * called from the s3c24xx audio core to deal with the architecture - * or the codec's setup and control. - * - * the pointer to itself is passed through in case the caller wants to - * embed this in an larger structure for easy reference to it's context. -*/ - -struct s3c24xx_iis_ops { - struct module *owner; - - int (*startup)(struct s3c24xx_iis_ops *me); - void (*shutdown)(struct s3c24xx_iis_ops *me); - int (*suspend)(struct s3c24xx_iis_ops *me); - int (*resume)(struct s3c24xx_iis_ops *me); - - int (*open)(struct s3c24xx_iis_ops *me, struct snd_pcm_substream *strm); - int (*close)(struct s3c24xx_iis_ops *me, struct snd_pcm_substream *strm); - int (*prepare)(struct s3c24xx_iis_ops *me, struct snd_pcm_substream *strm, struct snd_pcm_runtime *rt); -}; - -struct s3c24xx_platdata_iis { - const char *codec_clk; - struct s3c24xx_iis_ops *ops; - int (*match_dev)(struct device *dev); -}; - -#endif /* __ASM_ARCH_AUDIO_H */ diff --git a/arch/arm/plat-s3c/include/plat/audio.h b/arch/arm/plat-s3c/include/plat/audio.h new file mode 100644 index 00000000000..de0e8da48bc --- /dev/null +++ b/arch/arm/plat-s3c/include/plat/audio.h @@ -0,0 +1,45 @@ +/* arch/arm/mach-s3c2410/include/mach/audio.h + * + * Copyright (c) 2004-2005 Simtec Electronics + * http://www.simtec.co.uk/products/SWLINUX/ + * Ben Dooks + * + * S3C24XX - Audio platfrom_device info + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. +*/ + +#ifndef __ASM_ARCH_AUDIO_H +#define __ASM_ARCH_AUDIO_H __FILE__ + +/* struct s3c24xx_iis_ops + * + * called from the s3c24xx audio core to deal with the architecture + * or the codec's setup and control. + * + * the pointer to itself is passed through in case the caller wants to + * embed this in an larger structure for easy reference to it's context. +*/ + +struct s3c24xx_iis_ops { + struct module *owner; + + int (*startup)(struct s3c24xx_iis_ops *me); + void (*shutdown)(struct s3c24xx_iis_ops *me); + int (*suspend)(struct s3c24xx_iis_ops *me); + int (*resume)(struct s3c24xx_iis_ops *me); + + int (*open)(struct s3c24xx_iis_ops *me, struct snd_pcm_substream *strm); + int (*close)(struct s3c24xx_iis_ops *me, struct snd_pcm_substream *strm); + int (*prepare)(struct s3c24xx_iis_ops *me, struct snd_pcm_substream *strm, struct snd_pcm_runtime *rt); +}; + +struct s3c24xx_platdata_iis { + const char *codec_clk; + struct s3c24xx_iis_ops *ops; + int (*match_dev)(struct device *dev); +}; + +#endif /* __ASM_ARCH_AUDIO_H */ diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index 286ff4497fd..40530fc7fba 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -29,7 +29,7 @@ #include #include #include -#include +#include #include #include diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c index 36b927d3f54..3297698ff29 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.c +++ b/sound/soc/s3c24xx/s3c2412-i2s.c @@ -34,7 +34,7 @@ #include #include -#include +#include #include #include "s3c24xx-pcm.h" diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c index 5822d2dd49b..5c7f18a2264 100644 --- a/sound/soc/s3c24xx/s3c2443-ac97.c +++ b/sound/soc/s3c24xx/s3c2443-ac97.c @@ -31,7 +31,7 @@ #include #include #include -#include +#include #include #include diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 2569b910b9b..a7312e4fe4a 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -30,7 +30,7 @@ #include #include #include -#include +#include #include #include diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c index 7c64d31d067..bfea13f3ba7 100644 --- a/sound/soc/s3c24xx/s3c24xx-pcm.c +++ b/sound/soc/s3c24xx/s3c24xx-pcm.c @@ -29,7 +29,7 @@ #include #include #include -#include +#include #include "s3c24xx-pcm.h" -- cgit v1.2.3-70-g09d2 From f03d3115a6bcb814019d945c50c2ef91e5f14477 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 5 Mar 2009 14:18:16 +0100 Subject: ALSA: Fix sample rate of Lenovo Ideapad to 44.1kHz Noises can be heard on analog outputs of (some model of) Lenovo Ideapad due to the hardware problem, and the only workaround right now is to fix the sample rate to 44.1kHz. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 34 +++++++++++++++++++++++++++++++--- 1 file changed, 31 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6325ea43cf0..b794cba494c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12845,6 +12845,27 @@ static int alc269_auto_create_analog_input_ctls(struct alc_spec *spec, #define alc269_pcm_digital_playback alc880_pcm_digital_playback #define alc269_pcm_digital_capture alc880_pcm_digital_capture +static struct hda_pcm_stream alc269_44k_pcm_analog_playback = { + .substreams = 1, + .channels_min = 2, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_44100, /* fixed rate */ + /* NID is set in alc_build_pcms */ + .ops = { + .open = alc880_playback_pcm_open, + .prepare = alc880_playback_pcm_prepare, + .cleanup = alc880_playback_pcm_cleanup + }, +}; + +static struct hda_pcm_stream alc269_44k_pcm_analog_capture = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_44100, /* fixed rate */ + /* NID is set in alc_build_pcms */ +}; + /* * BIOS auto configuration */ @@ -13060,9 +13081,16 @@ static int patch_alc269(struct hda_codec *codec) setup_preset(spec, &alc269_presets[board_config]); spec->stream_name_analog = "ALC269 Analog"; - spec->stream_analog_playback = &alc269_pcm_analog_playback; - spec->stream_analog_capture = &alc269_pcm_analog_capture; - + if (codec->subsystem_id == 0x17aa3bf8) { + /* Due to a hardware problem on Lenovo Ideadpad, we need to + * fix the sample rate of analog I/O to 44.1kHz + */ + spec->stream_analog_playback = &alc269_44k_pcm_analog_playback; + spec->stream_analog_capture = &alc269_44k_pcm_analog_capture; + } else { + spec->stream_analog_playback = &alc269_pcm_analog_playback; + spec->stream_analog_capture = &alc269_pcm_analog_capture; + } spec->stream_name_digital = "ALC269 Digital"; spec->stream_digital_playback = &alc269_pcm_digital_playback; spec->stream_digital_capture = &alc269_pcm_digital_capture; -- cgit v1.2.3-70-g09d2 From dc04d1b4d2043e2fca2d94d6d5542b930f2bc5b3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 6 Mar 2009 10:00:05 +0100 Subject: ALSA: hda - Create output controls according to pin types for IDT/STAC Improve the parser to pick up more intuitive control names for the outputs judging from the pin type, instead of fixed names assigned to channels. Also, revive the multi-HP workaround since this change fixes the problem with the multi-HP detection. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 141 +++++++++++++++++++++-------------------- 1 file changed, 72 insertions(+), 69 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 2e0a599f8c1..edd2ed7ebb4 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -3039,35 +3039,33 @@ static int add_spec_extra_dacs(struct sigmatel_spec *spec, hda_nid_t nid) return 1; } -static int is_unique_dac(struct sigmatel_spec *spec, hda_nid_t nid) -{ - int i; - - if (spec->autocfg.line_outs != 1) - return 0; - if (spec->multiout.hp_nid == nid) - return 0; - for (i = 0; i < ARRAY_SIZE(spec->multiout.extra_out_nid); i++) - if (spec->multiout.extra_out_nid[i] == nid) - return 0; - return 1; -} - -/* add playback controls from the parsed DAC table */ -static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec, - const struct auto_pin_cfg *cfg) +/* Create output controls + * The mixer elements are named depending on the given type (AUTO_PIN_XXX_OUT) + */ +static int create_multi_out_ctls(struct hda_codec *codec, int num_outs, + const hda_nid_t *pins, + const hda_nid_t *dac_nids, + int type) { struct sigmatel_spec *spec = codec->spec; static const char *chname[4] = { "Front", "Surround", NULL /*CLFE*/, "Side" }; - hda_nid_t nid = 0; + static const char *hp_pfxs[] = { + "Headphone", "Headphone2", "Headphone3", "Headphone4" + }; + static const char *speaker_pfxs[] = { + "Speaker", "External Speaker", "Speaker2", "Speaker3" + }; + hda_nid_t nid; int i, err; unsigned int wid_caps; - for (i = 0; i < cfg->line_outs && spec->multiout.dac_nids[i]; i++) { - nid = spec->multiout.dac_nids[i]; - if (i == 2) { + for (i = 0; i < num_outs && i < ARRAY_SIZE(chname); i++) { + nid = dac_nids[i]; + if (!nid) + continue; + if (type != AUTO_PIN_HP_OUT && i == 2) { /* Center/LFE */ err = create_controls(codec, "Center", nid, 1); if (err < 0) @@ -3088,23 +3086,43 @@ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec, } } else { - const char *name = chname[i]; - /* if it's a single DAC, assign a better name */ - if (!i && is_unique_dac(spec, nid)) { - switch (cfg->line_out_type) { - case AUTO_PIN_HP_OUT: - name = "Headphone"; - break; - case AUTO_PIN_SPEAKER_OUT: - name = "Speaker"; - break; - } + const char *name; + switch (type) { + case AUTO_PIN_HP_OUT: + name = hp_pfxs[i]; + break; + case AUTO_PIN_SPEAKER_OUT: + name = speaker_pfxs[i]; + break; + default: + name = chname[i]; + break; } err = create_controls(codec, name, nid, 3); if (err < 0) return err; + if (type == AUTO_PIN_HP_OUT && !spec->hp_detect) { + wid_caps = get_wcaps(codec, pins[i]); + if (wid_caps & AC_WCAP_UNSOL_CAP) + spec->hp_detect = 1; + } } } + return 0; +} + +/* add playback controls from the parsed DAC table */ +static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec, + const struct auto_pin_cfg *cfg) +{ + struct sigmatel_spec *spec = codec->spec; + int err; + + err = create_multi_out_ctls(codec, cfg->line_outs, cfg->line_out_pins, + spec->multiout.dac_nids, + cfg->line_out_type); + if (err < 0) + return err; if (cfg->hp_outs > 1 && cfg->line_out_type == AUTO_PIN_LINE_OUT) { err = stac92xx_add_control(spec, @@ -3139,40 +3157,18 @@ static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec, struct auto_pin_cfg *cfg) { struct sigmatel_spec *spec = codec->spec; - hda_nid_t nid; - int i, err, nums; + int err; + + err = create_multi_out_ctls(codec, cfg->hp_outs, cfg->hp_pins, + spec->hp_dacs, AUTO_PIN_HP_OUT); + if (err < 0) + return err; + + err = create_multi_out_ctls(codec, cfg->speaker_outs, cfg->speaker_pins, + spec->speaker_dacs, AUTO_PIN_SPEAKER_OUT); + if (err < 0) + return err; - nums = 0; - for (i = 0; i < cfg->hp_outs; i++) { - static const char *pfxs[] = { - "Headphone", "Headphone2", "Headphone3", - }; - unsigned int wid_caps = get_wcaps(codec, cfg->hp_pins[i]); - if (wid_caps & AC_WCAP_UNSOL_CAP) - spec->hp_detect = 1; - if (nums >= ARRAY_SIZE(pfxs)) - continue; - nid = spec->hp_dacs[i]; - if (!nid) - continue; - err = create_controls(codec, pfxs[nums++], nid, 3); - if (err < 0) - return err; - } - nums = 0; - for (i = 0; i < cfg->speaker_outs; i++) { - static const char *pfxs[] = { - "Speaker", "External Speaker", "Speaker2", - }; - if (nums >= ARRAY_SIZE(pfxs)) - continue; - nid = spec->speaker_dacs[i]; - if (!nid) - continue; - err = create_controls(codec, pfxs[nums++], nid, 3); - if (err < 0) - return err; - } return 0; } @@ -3505,6 +3501,7 @@ static void stac92xx_auto_init_hp_out(struct hda_codec *codec) static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out, hda_nid_t dig_in) { struct sigmatel_spec *spec = codec->spec; + int hp_swap = 0; int err; if ((err = snd_hda_parse_pin_def_config(codec, @@ -3514,7 +3511,6 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out if (! spec->autocfg.line_outs) return 0; /* can't find valid pin config */ -#if 0 /* FIXME: temporarily disabled */ /* If we have no real line-out pin and multiple hp-outs, HPs should * be set up as multi-channel outputs. */ @@ -3533,8 +3529,8 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out spec->autocfg.line_outs = spec->autocfg.hp_outs; spec->autocfg.line_out_type = AUTO_PIN_HP_OUT; spec->autocfg.hp_outs = 0; + hp_swap = 1; } -#endif /* FIXME: temporarily disabled */ if (spec->autocfg.mono_out_pin) { int dir = get_wcaps(codec, spec->autocfg.mono_out_pin) & (AC_WCAP_OUT_AMP | AC_WCAP_IN_AMP); @@ -3627,12 +3623,19 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out #endif err = stac92xx_auto_create_hp_ctls(codec, &spec->autocfg); - if (err < 0) return err; - err = stac92xx_auto_create_analog_input_ctls(codec, &spec->autocfg); + /* All output parsing done, now restore the swapped hp pins */ + if (hp_swap) { + memcpy(spec->autocfg.hp_pins, spec->autocfg.line_out_pins, + sizeof(spec->autocfg.hp_pins)); + spec->autocfg.hp_outs = spec->autocfg.line_outs; + spec->autocfg.line_out_type = AUTO_PIN_HP_OUT; + spec->autocfg.line_outs = 0; + } + err = stac92xx_auto_create_analog_input_ctls(codec, &spec->autocfg); if (err < 0) return err; -- cgit v1.2.3-70-g09d2 From 7a411ee01bf3114ba2a2ae013eaae4e3c41f8eb5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 6 Mar 2009 10:08:14 +0100 Subject: ALSA: hda - Allow slave controls with non-zero indices Fix snd_hda_add_vmaster() to check the non-zero indices of slave controls. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 21 +++++++++++++-------- 1 file changed, 13 insertions(+), 8 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 04cb1251e3e..1885e764910 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1552,15 +1552,20 @@ int snd_hda_add_vmaster(struct hda_codec *codec, char *name, for (s = slaves; *s; s++) { struct snd_kcontrol *sctl; - - sctl = snd_hda_find_mixer_ctl(codec, *s); - if (!sctl) { - snd_printdd("Cannot find slave %s, skipped\n", *s); - continue; + int i = 0; + for (;;) { + sctl = _snd_hda_find_mixer_ctl(codec, *s, i); + if (!sctl) { + if (!i) + snd_printdd("Cannot find slave %s, " + "skipped\n", *s); + break; + } + err = snd_ctl_add_slave(kctl, sctl); + if (err < 0) + return err; + i++; } - err = snd_ctl_add_slave(kctl, sctl); - if (err < 0) - return err; } return 0; } -- cgit v1.2.3-70-g09d2 From 668b9652be33510a2a42b290dd335d34d38e2068 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 6 Mar 2009 10:13:24 +0100 Subject: ALSA: hda - Create multiple HP / speaker controls with index Create multiple "Headphone" and "Speaker" controls with non-zero index numbers instead of "Headphone2", etc. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 33 ++++++++++++++------------------- 1 file changed, 14 insertions(+), 19 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index edd2ed7ebb4..d19090fd2d1 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1227,10 +1227,7 @@ static const char *slave_vols[] = { "LFE Playback Volume", "Side Playback Volume", "Headphone Playback Volume", - "Headphone2 Playback Volume", "Speaker Playback Volume", - "External Speaker Playback Volume", - "Speaker2 Playback Volume", NULL }; @@ -1241,10 +1238,7 @@ static const char *slave_sws[] = { "LFE Playback Switch", "Side Playback Switch", "Headphone Playback Switch", - "Headphone2 Playback Switch", "Speaker Playback Switch", - "External Speaker Playback Switch", - "Speaker2 Playback Switch", "IEC958 Playback Switch", NULL }; @@ -2976,8 +2970,8 @@ static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec) } /* create volume control/switch for the given prefx type */ -static int create_controls(struct hda_codec *codec, const char *pfx, - hda_nid_t nid, int chs) +static int create_controls_idx(struct hda_codec *codec, const char *pfx, + int idx, hda_nid_t nid, int chs) { struct sigmatel_spec *spec = codec->spec; char name[32]; @@ -3001,19 +2995,22 @@ static int create_controls(struct hda_codec *codec, const char *pfx, } sprintf(name, "%s Playback Volume", pfx); - err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, name, + err = stac92xx_add_control_idx(spec, STAC_CTL_WIDGET_VOL, idx, name, HDA_COMPOSE_AMP_VAL_OFS(nid, chs, 0, HDA_OUTPUT, spec->volume_offset)); if (err < 0) return err; sprintf(name, "%s Playback Switch", pfx); - err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, name, + err = stac92xx_add_control_idx(spec, STAC_CTL_WIDGET_MUTE, idx, name, HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT)); if (err < 0) return err; return 0; } +#define create_controls(codec, pfx, nid, chs) \ + create_controls_idx(codec, pfx, 0, nid, chs) + static int add_spec_dacs(struct sigmatel_spec *spec, hda_nid_t nid) { if (spec->multiout.num_dacs > 4) { @@ -3051,12 +3048,6 @@ static int create_multi_out_ctls(struct hda_codec *codec, int num_outs, static const char *chname[4] = { "Front", "Surround", NULL /*CLFE*/, "Side" }; - static const char *hp_pfxs[] = { - "Headphone", "Headphone2", "Headphone3", "Headphone4" - }; - static const char *speaker_pfxs[] = { - "Speaker", "External Speaker", "Speaker2", "Speaker3" - }; hda_nid_t nid; int i, err; unsigned int wid_caps; @@ -3087,18 +3078,22 @@ static int create_multi_out_ctls(struct hda_codec *codec, int num_outs, } else { const char *name; + int idx; switch (type) { case AUTO_PIN_HP_OUT: - name = hp_pfxs[i]; + name = "Headphone"; + idx = i; break; case AUTO_PIN_SPEAKER_OUT: - name = speaker_pfxs[i]; + name = "Speaker"; + idx = i; break; default: name = chname[i]; + idx = 0; break; } - err = create_controls(codec, name, nid, 3); + err = create_controls_idx(codec, name, idx, nid, 3); if (err < 0) return err; if (type == AUTO_PIN_HP_OUT && !spec->hp_detect) { -- cgit v1.2.3-70-g09d2 From ee58a7ca21b2acf0d7ad0e1eb2f8d916ecf9fadc Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 6 Mar 2009 12:00:24 +0100 Subject: ALSA: hda - Connect to primary DAC if no individual DAC is available In stac92xx_auto_fill_dac_nids[], connect to the primary DAC if no individual DAC is available for each pin. This ensures that the pin works somehow at least. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index d19090fd2d1..ee119259183 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2871,6 +2871,16 @@ static hda_nid_t get_unassigned_dac(struct hda_codec *codec, hda_nid_t nid) return conn[j]; } } + /* if all DACs are already assigned, connect to the primary DAC */ + if (conn_len > 1) { + for (j = 0; j < conn_len; j++) { + if (conn[j] == spec->multiout.dac_nids[0]) { + snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_CONNECT_SEL, j); + break; + } + } + } return 0; } -- cgit v1.2.3-70-g09d2 From 139e071b0ff37800ed0a68b10c4bb325f51786eb Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 6 Mar 2009 12:10:41 +0100 Subject: ALSA: hda - Assign HP and speaker DACs before mic/line-in Assign DACs to HP and speaker before mic-in/line-in shared outputs. This improves the usability as it results in more intuitive mixer names. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 40 ++++++++++++++++++++-------------------- 1 file changed, 20 insertions(+), 20 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index ee119259183..123bcf7c3b2 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2921,6 +2921,26 @@ static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec) add_spec_dacs(spec, dac); } + for (i = 0; i < cfg->hp_outs; i++) { + nid = cfg->hp_pins[i]; + dac = get_unassigned_dac(codec, nid); + if (dac) { + if (!spec->multiout.hp_nid) + spec->multiout.hp_nid = dac; + else + add_spec_extra_dacs(spec, dac); + } + spec->hp_dacs[i] = dac; + } + + for (i = 0; i < cfg->speaker_outs; i++) { + nid = cfg->speaker_pins[i]; + dac = get_unassigned_dac(codec, nid); + if (dac) + add_spec_extra_dacs(spec, dac); + spec->speaker_dacs[i] = dac; + } + /* add line-in as output */ nid = check_line_out_switch(codec); if (nid) { @@ -2948,26 +2968,6 @@ static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec) } } - for (i = 0; i < cfg->hp_outs; i++) { - nid = cfg->hp_pins[i]; - dac = get_unassigned_dac(codec, nid); - if (dac) { - if (!spec->multiout.hp_nid) - spec->multiout.hp_nid = dac; - else - add_spec_extra_dacs(spec, dac); - } - spec->hp_dacs[i] = dac; - } - - for (i = 0; i < cfg->speaker_outs; i++) { - nid = cfg->speaker_pins[i]; - dac = get_unassigned_dac(codec, nid); - if (dac) - add_spec_extra_dacs(spec, dac); - spec->speaker_dacs[i] = dac; - } - snd_printd("stac92xx: dac_nids=%d (0x%x/0x%x/0x%x/0x%x/0x%x)\n", spec->multiout.num_dacs, spec->multiout.dac_nids[0], -- cgit v1.2.3-70-g09d2 From 90f349d96e1dc05b1f7916958282c30760eeacd6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 6 Mar 2009 14:30:08 +0100 Subject: ALSA: ac97 - Add patch entry for Conexant CX20468-31 chip Added the patch entry for Conexant CX20468-31 chip (4358:5430). Reference: Novell bnc#471265 https://bugzilla.novell.com/show_bug.cgi?id=471265 Signed-off-by: Takashi Iwai --- sound/pci/ac97/ac97_codec.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index bc707b60385..44f2381b0ae 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -143,6 +143,7 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = { { 0x43525970, 0xfffffff8, "CS4202", NULL, NULL }, { 0x43585421, 0xffffffff, "HSD11246", NULL, NULL }, // SmartMC II { 0x43585428, 0xfffffff8, "Cx20468", patch_conexant, NULL }, // SmartAMC fixme: the mask might be different +{ 0x43585430, 0xffffffff, "Cx20468-31", patch_conexant, NULL }, { 0x43585431, 0xffffffff, "Cx20551", patch_cx20551, NULL }, { 0x44543031, 0xfffffff0, "DT0398", NULL, NULL }, { 0x454d4328, 0xffffffff, "EM28028", NULL, NULL }, // same as TR28028? -- cgit v1.2.3-70-g09d2 From 979c036e090332cd3a090ce8b4eb50c3aa28dea0 Mon Sep 17 00:00:00 2001 From: "Lopez Cruz, Misael" Date: Wed, 4 Mar 2009 11:39:07 -0600 Subject: ASoC: Add DAPM machine widgets to SDP3430 driver Add DAPM machine domain widgets to SDP3430 machine driver. Interconnection: * Ext Mic: MAINMIC, SUBMIC * Ext Spk: HFL, HFR * Headset Jack: HSMIC, HSOL, HSOR Signed-off-by: Misael Lopez Cruz Signed-off-by: Mark Brown --- sound/soc/omap/sdp3430.c | 64 ++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 64 insertions(+) diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index e226fa75669..4eab4b491de 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -81,12 +81,76 @@ static struct snd_soc_ops sdp3430_ops = { .hw_params = sdp3430_hw_params, }; +/* SDP3430 machine DAPM */ +static const struct snd_soc_dapm_widget sdp3430_twl4030_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Ext Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_HP("Headset Jack", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* External Mics: MAINMIC, SUBMIC with bias*/ + {"MAINMIC", NULL, "Mic Bias 1"}, + {"SUBMIC", NULL, "Mic Bias 2"}, + {"Mic Bias 1", NULL, "Ext Mic"}, + {"Mic Bias 2", NULL, "Ext Mic"}, + + /* External Speakers: HFL, HFR */ + {"Ext Spk", NULL, "HFL"}, + {"Ext Spk", NULL, "HFR"}, + + /* Headset: HSMIC (with bias), HSOL, HSOR */ + {"Headset Jack", NULL, "HSOL"}, + {"Headset Jack", NULL, "HSOR"}, + {"HSMIC", NULL, "Headset Mic Bias"}, + {"Headset Mic Bias", NULL, "Headset Jack"}, +}; + +static int sdp3430_twl4030_init(struct snd_soc_codec *codec) +{ + int ret; + + /* Add SDP3430 specific widgets */ + ret = snd_soc_dapm_new_controls(codec, sdp3430_twl4030_dapm_widgets, + ARRAY_SIZE(sdp3430_twl4030_dapm_widgets)); + if (ret) + return ret; + + /* Set up SDP3430 specific audio path audio_map */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + /* SDP3430 connected pins */ + snd_soc_dapm_enable_pin(codec, "Ext Mic"); + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + snd_soc_dapm_enable_pin(codec, "Headset Jack"); + + /* TWL4030 not connected pins */ + snd_soc_dapm_nc_pin(codec, "AUXL"); + snd_soc_dapm_nc_pin(codec, "AUXR"); + snd_soc_dapm_nc_pin(codec, "CARKITMIC"); + snd_soc_dapm_nc_pin(codec, "DIGIMIC0"); + snd_soc_dapm_nc_pin(codec, "DIGIMIC1"); + + snd_soc_dapm_nc_pin(codec, "OUTL"); + snd_soc_dapm_nc_pin(codec, "OUTR"); + snd_soc_dapm_nc_pin(codec, "EARPIECE"); + snd_soc_dapm_nc_pin(codec, "PREDRIVEL"); + snd_soc_dapm_nc_pin(codec, "PREDRIVER"); + snd_soc_dapm_nc_pin(codec, "CARKITL"); + snd_soc_dapm_nc_pin(codec, "CARKITR"); + + ret = snd_soc_dapm_sync(codec); + + return ret; +} + /* Digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link sdp3430_dai = { .name = "TWL4030", .stream_name = "TWL4030", .cpu_dai = &omap_mcbsp_dai[0], .codec_dai = &twl4030_dai, + .init = sdp3430_twl4030_init, .ops = &sdp3430_ops, }; -- cgit v1.2.3-70-g09d2 From 3093e48c48b69ccc06a1f78ffe7ece7ee2ee09ef Mon Sep 17 00:00:00 2001 From: Ben Dooks Date: Wed, 4 Mar 2009 00:49:27 +0000 Subject: ASoC: Add JIVE audio support Add support for the Jive's WM8750 codec attached via the S3C2412 IIS. Signed-off-by: Ben Dooks Signed-off-by: Mark Brown --- sound/soc/s3c24xx/Kconfig | 9 ++ sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/jive_wm8750.c | 216 ++++++++++++++++++++++++++++++++++++++++ 3 files changed, 227 insertions(+) create mode 100644 sound/soc/s3c24xx/jive_wm8750.c diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index e05a71084c3..7803218d190 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -26,6 +26,15 @@ config SND_S3C24XX_SOC_NEO1973_WM8753 Say Y if you want to add support for SoC audio on smdk2440 with the WM8753. +config SND_S3C24XX_SOC_JIVE_WM8750 + tristate "SoC I2S Audio support for Jive" + depends on SND_S3C24XX_SOC && MACH_JIVE + select SND_SOC_WM8750 + select SND_SOC_WM8750_SPI + select SND_S3C2412_SOC_I2S + help + Sat Y if you want to add support for SoC audio on the Jive. + config SND_S3C24XX_SOC_SMDK2443_WM9710 tristate "SoC AC97 Audio support for SMDK2443 - WM9710" depends on SND_S3C24XX_SOC && MACH_SMDK2443 diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 96b3f3f617d..bc11976e269 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -10,11 +10,13 @@ obj-$(CONFIG_SND_S3C2443_SOC_AC97) += snd-soc-s3c2443-ac97.o obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o # S3C24XX Machine Support +snd-soc-jive-wm8750-objs := jive_wm8750.o snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o snd-soc-smdk2443-wm9710-objs := smdk2443_wm9710.o snd-soc-ln2440sbc-alc650-objs := ln2440sbc_alc650.o snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.o +obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o obj-$(CONFIG_SND_S3C24XX_SOC_SMDK2443_WM9710) += snd-soc-smdk2443-wm9710.o obj-$(CONFIG_SND_S3C24XX_SOC_LN2440SBC_ALC650) += snd-soc-ln2440sbc-alc650.o diff --git a/sound/soc/s3c24xx/jive_wm8750.c b/sound/soc/s3c24xx/jive_wm8750.c new file mode 100644 index 00000000000..fe466c1f71e --- /dev/null +++ b/sound/soc/s3c24xx/jive_wm8750.c @@ -0,0 +1,216 @@ +/* sound/soc/s3c24xx/jive_wm8750.c + * + * Copyright 2007,2008 Simtec Electronics + * + * Based on sound/soc/pxa/spitz.c + * Copyright 2005 Wolfson Microelectronics PLC. + * Copyright 2005 Openedhand Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. +*/ + +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include + +#include + +#include "s3c24xx-pcm.h" +#include "s3c2412-i2s.h" + +#include "../codecs/wm8750.h" + +static const struct snd_soc_dapm_route audio_map[] = { + { "Headphone Jack", NULL, "LOUT1" }, + { "Headphone Jack", NULL, "ROUT1" }, + { "Internal Speaker", NULL, "LOUT2" }, + { "Internal Speaker", NULL, "ROUT2" }, + { "LINPUT1", NULL, "Line Input" }, + { "RINPUT1", NULL, "Line Input" }, +}; + +static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Internal Speaker", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), +}; + +static int jive_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->socdev->codec; + + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Internal Speaker"); + snd_soc_dapm_enable_pin(codec, "Line In"); + + snd_soc_dapm_sync(codec); + + return 0; +} + +static int jive_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct s3c2412_rate_calc div; + unsigned int clk = 0; + int ret = 0; + + switch (params_rate(params)) { + case 8000: + case 16000: + case 48000: + case 96000: + clk = 12288000; + break; + case 11025: + case 22050: + case 44100: + clk = 11289600; + break; + } + + s3c2412_iis_calc_rate(&div, NULL, params_rate(params), + s3c2412_get_iisclk()); + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8750_SYSCLK, clk, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C2412_DIV_RCLK, div.fs_div); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C2412_DIV_PRESCALER, + div.clk_div - 1); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops jive_ops = { + .startup = jive_startup, + .hw_params = jive_hw_params, +}; + +static int jive_wm8750_init(struct snd_soc_codec *codec) +{ + int err; + + /* These endpoints are not being used. */ + snd_soc_dapm_disable_pin(codec, "LINPUT2"); + snd_soc_dapm_disable_pin(codec, "RINPUT2"); + snd_soc_dapm_disable_pin(codec, "LINPUT3"); + snd_soc_dapm_disable_pin(codec, "RINPUT3"); + snd_soc_dapm_disable_pin(codec, "OUT3"); + snd_soc_dapm_disable_pin(codec, "MONO"); + + /* Add jive specific widgets */ + err = snd_soc_dapm_new_controls(codec, wm8750_dapm_widgets, + ARRAY_SIZE(wm8750_dapm_widgets)); + if (err) { + printk(KERN_ERR "%s: failed to add widgets (%d)\n", + __func__, err); + return err; + } + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link jive_dai = { + .name = "wm8750", + .stream_name = "WM8750", + .cpu_dai = &s3c2412_i2s_dai, + .codec_dai = &wm8750_dai, + .init = jive_wm8750_init, + .ops = &jive_ops, +}; + +/* jive audio machine driver */ +static struct snd_soc_machine snd_soc_machine_jive = { + .name = "Jive", + .dai_link = &jive_dai, + .num_links = 1, +}; + +/* jive audio private data */ +static struct wm8750_setup_data jive_wm8750_setup = { +}; + +/* jive audio subsystem */ +static struct snd_soc_device jive_snd_devdata = { + .machine = &snd_soc_machine_jive, + .platform = &s3c24xx_soc_platform, + .codec_dev = &soc_codec_dev_wm8750_spi, + .codec_data = &jive_wm8750_setup, +}; + +static struct platform_device *jive_snd_device; + +static int __init jive_init(void) +{ + int ret; + + if (!machine_is_jive()) + return 0; + + printk("JIVE WM8750 Audio support\n"); + + jive_snd_device = platform_device_alloc("soc-audio", -1); + if (!jive_snd_device) + return -ENOMEM; + + platform_set_drvdata(jive_snd_device, &jive_snd_devdata); + jive_snd_devdata.dev = &jive_snd_device->dev; + ret = platform_device_add(jive_snd_device); + + if (ret) + platform_device_put(jive_snd_device); + + return ret; +} + +static void __exit jive_exit(void) +{ + platform_device_unregister(jive_snd_device); +} + +module_init(jive_init); +module_exit(jive_exit); + +MODULE_AUTHOR("Ben Dooks "); +MODULE_DESCRIPTION("ALSA SoC Jive Audio support"); +MODULE_LICENSE("GPL"); -- cgit v1.2.3-70-g09d2 From dc85447b196a683784eb85654c436fd58c3e2ed1 Mon Sep 17 00:00:00 2001 From: Ben Dooks Date: Wed, 4 Mar 2009 00:49:30 +0000 Subject: ASoC: Split s3c2412-i2s.c into core and SoC specific parts The S3C2412 I2S (IIS) interface is replicated on further Samsung SoC parts in a broadly compatible way, so split the common code out into a core called s3c-i2s-v2.[ch] so that the newer SoCs such as the S3C6410 can make use of it. As such, all the original s3c2412 functions are currently being left with their original names, and will be renamed later in the series. Signed-off-by: Ben Dooks Signed-off-by: Mark Brown --- sound/soc/s3c24xx/Kconfig | 4 + sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/jive_wm8750.c | 6 +- sound/soc/s3c24xx/s3c-i2s-v2.c | 645 ++++++++++++++++++++++++++++++++++++++++ sound/soc/s3c24xx/s3c-i2s-v2.h | 90 ++++++ sound/soc/s3c24xx/s3c2412-i2s.c | 596 ++----------------------------------- sound/soc/s3c24xx/s3c2412-i2s.h | 17 +- 7 files changed, 767 insertions(+), 593 deletions(-) create mode 100644 sound/soc/s3c24xx/s3c-i2s-v2.c create mode 100644 sound/soc/s3c24xx/s3c-i2s-v2.h diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 7803218d190..bd98adaa5b0 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -9,8 +9,12 @@ config SND_S3C24XX_SOC config SND_S3C24XX_SOC_I2S tristate +config SND_S3C_I2SV2_SOC + tristate + config SND_S3C2412_SOC_I2S tristate + select SND_S3C_I2SV2_SOC config SND_S3C2443_SOC_AC97 tristate diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index bc11976e269..848981e18ac 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -3,11 +3,13 @@ snd-soc-s3c24xx-objs := s3c24xx-pcm.o snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o snd-soc-s3c2443-ac97-objs := s3c2443-ac97.o +snd-soc-s3c-i2s-v2-objs := s3c-i2s-v2.o obj-$(CONFIG_SND_S3C24XX_SOC) += snd-soc-s3c24xx.o obj-$(CONFIG_SND_S3C24XX_SOC_I2S) += snd-soc-s3c24xx-i2s.o obj-$(CONFIG_SND_S3C2443_SOC_AC97) += snd-soc-s3c2443-ac97.o obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o +obj-$(CONFIG_SND_S3C_I2SV2_SOC) += snd-soc-s3c-i2s-v2.o # S3C24XX Machine Support snd-soc-jive-wm8750-objs := jive_wm8750.o diff --git a/sound/soc/s3c24xx/jive_wm8750.c b/sound/soc/s3c24xx/jive_wm8750.c index fe466c1f71e..7dfe26ea8f4 100644 --- a/sound/soc/s3c24xx/jive_wm8750.c +++ b/sound/soc/s3c24xx/jive_wm8750.c @@ -65,7 +65,7 @@ static int jive_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - struct s3c2412_rate_calc div; + struct s3c_i2sv2_rate_calc div; unsigned int clk = 0; int ret = 0; @@ -83,8 +83,8 @@ static int jive_hw_params(struct snd_pcm_substream *substream, break; } - s3c2412_iis_calc_rate(&div, NULL, params_rate(params), - s3c2412_get_iisclk()); + s3c_i2sv2_calc_rate(&div, NULL, params_rate(params), + s3c2412_get_iisclk()); /* set codec DAI configuration */ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c new file mode 100644 index 00000000000..43262e1e8f9 --- /dev/null +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -0,0 +1,645 @@ +/* sound/soc/s3c24xx/s3c-i2c-v2.c + * + * ALSA Soc Audio Layer - I2S core for newer Samsung SoCs. + * + * Copyright (c) 2006 Wolfson Microelectronics PLC. + * Graeme Gregory graeme.gregory@wolfsonmicro.com + * linux@wolfsonmicro.com + * + * Copyright (c) 2008, 2007, 2004-2005 Simtec Electronics + * http://armlinux.simtec.co.uk/ + * Ben Dooks + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include + +#include + +#include +#include + +#include "s3c-i2s-v2.h" + +#define S3C2412_I2S_DEBUG_CON 0 +#define S3C2412_I2S_DEBUG 0 + +#if S3C2412_I2S_DEBUG +#define DBG(x...) printk(KERN_INFO x) +#else +#define DBG(x...) do { } while (0) +#endif + +static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai) +{ + return cpu_dai->private_data; +} + +#define bit_set(v, b) (((v) & (b)) ? 1 : 0) + +#if S3C2412_I2S_DEBUG_CON +static void dbg_showcon(const char *fn, u32 con) +{ + printk(KERN_DEBUG "%s: LRI=%d, TXFEMPT=%d, RXFEMPT=%d, TXFFULL=%d, RXFFULL=%d\n", fn, + bit_set(con, S3C2412_IISCON_LRINDEX), + bit_set(con, S3C2412_IISCON_TXFIFO_EMPTY), + bit_set(con, S3C2412_IISCON_RXFIFO_EMPTY), + bit_set(con, S3C2412_IISCON_TXFIFO_FULL), + bit_set(con, S3C2412_IISCON_RXFIFO_FULL)); + + printk(KERN_DEBUG "%s: PAUSE: TXDMA=%d, RXDMA=%d, TXCH=%d, RXCH=%d\n", + fn, + bit_set(con, S3C2412_IISCON_TXDMA_PAUSE), + bit_set(con, S3C2412_IISCON_RXDMA_PAUSE), + bit_set(con, S3C2412_IISCON_TXCH_PAUSE), + bit_set(con, S3C2412_IISCON_RXCH_PAUSE)); + printk(KERN_DEBUG "%s: ACTIVE: TXDMA=%d, RXDMA=%d, IIS=%d\n", fn, + bit_set(con, S3C2412_IISCON_TXDMA_ACTIVE), + bit_set(con, S3C2412_IISCON_RXDMA_ACTIVE), + bit_set(con, S3C2412_IISCON_IIS_ACTIVE)); +} +#else +static inline void dbg_showcon(const char *fn, u32 con) +{ +} +#endif + + +/* Turn on or off the transmission path. */ +void s3c2412_snd_txctrl(struct s3c_i2sv2_info *i2s, int on) +{ + void __iomem *regs = i2s->regs; + u32 fic, con, mod; + + DBG("%s(%d)\n", __func__, on); + + fic = readl(regs + S3C2412_IISFIC); + con = readl(regs + S3C2412_IISCON); + mod = readl(regs + S3C2412_IISMOD); + + DBG("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); + + if (on) { + con |= S3C2412_IISCON_TXDMA_ACTIVE | S3C2412_IISCON_IIS_ACTIVE; + con &= ~S3C2412_IISCON_TXDMA_PAUSE; + con &= ~S3C2412_IISCON_TXCH_PAUSE; + + switch (mod & S3C2412_IISMOD_MODE_MASK) { + case S3C2412_IISMOD_MODE_TXONLY: + case S3C2412_IISMOD_MODE_TXRX: + /* do nothing, we are in the right mode */ + break; + + case S3C2412_IISMOD_MODE_RXONLY: + mod &= ~S3C2412_IISMOD_MODE_MASK; + mod |= S3C2412_IISMOD_MODE_TXRX; + break; + + default: + dev_err(i2s->dev, "TXEN: Invalid MODE in IISMOD\n"); + } + + writel(con, regs + S3C2412_IISCON); + writel(mod, regs + S3C2412_IISMOD); + } else { + /* Note, we do not have any indication that the FIFO problems + * tha the S3C2410/2440 had apply here, so we should be able + * to disable the DMA and TX without resetting the FIFOS. + */ + + con |= S3C2412_IISCON_TXDMA_PAUSE; + con |= S3C2412_IISCON_TXCH_PAUSE; + con &= ~S3C2412_IISCON_TXDMA_ACTIVE; + + switch (mod & S3C2412_IISMOD_MODE_MASK) { + case S3C2412_IISMOD_MODE_TXRX: + mod &= ~S3C2412_IISMOD_MODE_MASK; + mod |= S3C2412_IISMOD_MODE_RXONLY; + break; + + case S3C2412_IISMOD_MODE_TXONLY: + mod &= ~S3C2412_IISMOD_MODE_MASK; + con &= ~S3C2412_IISCON_IIS_ACTIVE; + break; + + default: + dev_err(i2s->dev, "TXDIS: Invalid MODE in IISMOD\n"); + } + + writel(mod, regs + S3C2412_IISMOD); + writel(con, regs + S3C2412_IISCON); + } + + fic = readl(regs + S3C2412_IISFIC); + dbg_showcon(__func__, con); + DBG("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); +} +EXPORT_SYMBOL_GPL(s3c2412_snd_txctrl); + +void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on) +{ + void __iomem *regs = i2s->regs; + u32 fic, con, mod; + + DBG("%s(%d)\n", __func__, on); + + fic = readl(regs + S3C2412_IISFIC); + con = readl(regs + S3C2412_IISCON); + mod = readl(regs + S3C2412_IISMOD); + + DBG("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); + + if (on) { + con |= S3C2412_IISCON_RXDMA_ACTIVE | S3C2412_IISCON_IIS_ACTIVE; + con &= ~S3C2412_IISCON_RXDMA_PAUSE; + con &= ~S3C2412_IISCON_RXCH_PAUSE; + + switch (mod & S3C2412_IISMOD_MODE_MASK) { + case S3C2412_IISMOD_MODE_TXRX: + case S3C2412_IISMOD_MODE_RXONLY: + /* do nothing, we are in the right mode */ + break; + + case S3C2412_IISMOD_MODE_TXONLY: + mod &= ~S3C2412_IISMOD_MODE_MASK; + mod |= S3C2412_IISMOD_MODE_TXRX; + break; + + default: + dev_err(i2s->dev, "RXEN: Invalid MODE in IISMOD\n"); + } + + writel(mod, regs + S3C2412_IISMOD); + writel(con, regs + S3C2412_IISCON); + } else { + /* See txctrl notes on FIFOs. */ + + con &= ~S3C2412_IISCON_RXDMA_ACTIVE; + con |= S3C2412_IISCON_RXDMA_PAUSE; + con |= S3C2412_IISCON_RXCH_PAUSE; + + switch (mod & S3C2412_IISMOD_MODE_MASK) { + case S3C2412_IISMOD_MODE_RXONLY: + con &= ~S3C2412_IISCON_IIS_ACTIVE; + mod &= ~S3C2412_IISMOD_MODE_MASK; + break; + + case S3C2412_IISMOD_MODE_TXRX: + mod &= ~S3C2412_IISMOD_MODE_MASK; + mod |= S3C2412_IISMOD_MODE_TXONLY; + break; + + default: + dev_err(i2s->dev, "RXEN: Invalid MODE in IISMOD\n"); + } + + writel(con, regs + S3C2412_IISCON); + writel(mod, regs + S3C2412_IISMOD); + } + + fic = readl(regs + S3C2412_IISFIC); + DBG("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); +} +EXPORT_SYMBOL_GPL(s3c2412_snd_rxctrl); + +/* + * Wait for the LR signal to allow synchronisation to the L/R clock + * from the codec. May only be needed for slave mode. + */ +static int s3c2412_snd_lrsync(struct s3c_i2sv2_info *i2s) +{ + u32 iiscon; + unsigned long timeout = jiffies + msecs_to_jiffies(5); + + DBG("Entered %s\n", __func__); + + while (1) { + iiscon = readl(i2s->regs + S3C2412_IISCON); + if (iiscon & S3C2412_IISCON_LRINDEX) + break; + + if (timeout < jiffies) { + printk(KERN_ERR "%s: timeout\n", __func__); + return -ETIMEDOUT; + } + } + + return 0; +} + +/* + * Set S3C2412 I2S DAI format + */ +static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct s3c_i2sv2_info *i2s = to_info(cpu_dai); + u32 iismod; + + DBG("Entered %s\n", __func__); + + iismod = readl(i2s->regs + S3C2412_IISMOD); + DBG("hw_params r: IISMOD: %x \n", iismod); + +#if defined(CONFIG_CPU_S3C2412) || defined(CONFIG_CPU_S3C2413) +#define IISMOD_MASTER_MASK S3C2412_IISMOD_MASTER_MASK +#define IISMOD_SLAVE S3C2412_IISMOD_SLAVE +#define IISMOD_MASTER S3C2412_IISMOD_MASTER_INTERNAL +#endif + +#if defined(CONFIG_PLAT_S3C64XX) +/* From Rev1.1 datasheet, we have two master and two slave modes: + * IMS[11:10]: + * 00 = master mode, fed from PCLK + * 01 = master mode, fed from CLKAUDIO + * 10 = slave mode, using PCLK + * 11 = slave mode, using I2SCLK + */ +#define IISMOD_MASTER_MASK (1 << 11) +#define IISMOD_SLAVE (1 << 11) +#define IISMOD_MASTER (0x0) +#endif + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + i2s->master = 0; + iismod &= ~IISMOD_MASTER_MASK; + iismod |= IISMOD_SLAVE; + break; + case SND_SOC_DAIFMT_CBS_CFS: + i2s->master = 1; + iismod &= ~IISMOD_MASTER_MASK; + iismod |= IISMOD_MASTER; + break; + default: + DBG("unknwon master/slave format\n"); + return -EINVAL; + } + + iismod &= ~S3C2412_IISMOD_SDF_MASK; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_RIGHT_J: + iismod |= S3C2412_IISMOD_SDF_MSB; + break; + case SND_SOC_DAIFMT_LEFT_J: + iismod |= S3C2412_IISMOD_SDF_LSB; + break; + case SND_SOC_DAIFMT_I2S: + iismod |= S3C2412_IISMOD_SDF_IIS; + break; + default: + DBG("Unknown data format\n"); + return -EINVAL; + } + + writel(iismod, i2s->regs + S3C2412_IISMOD); + DBG("hw_params w: IISMOD: %x \n", iismod); + return 0; +} + +static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *socdai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai_link *dai = rtd->dai; + struct s3c_i2sv2_info *i2s = to_info(dai->cpu_dai); + u32 iismod; + + DBG("Entered %s\n", __func__); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dai->cpu_dai->dma_data = i2s->dma_playback; + else + dai->cpu_dai->dma_data = i2s->dma_capture; + + /* Working copies of register */ + iismod = readl(i2s->regs + S3C2412_IISMOD); + DBG("%s: r: IISMOD: %x\n", __func__, iismod); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + iismod |= S3C2412_IISMOD_8BIT; + break; + case SNDRV_PCM_FORMAT_S16_LE: + iismod &= ~S3C2412_IISMOD_8BIT; + break; + } + + writel(iismod, i2s->regs + S3C2412_IISMOD); + DBG("%s: w: IISMOD: %x\n", __func__, iismod); + return 0; +} + +static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct s3c_i2sv2_info *i2s = to_info(rtd->dai->cpu_dai); + int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); + unsigned long irqs; + int ret = 0; + + DBG("Entered %s\n", __func__); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + /* On start, ensure that the FIFOs are cleared and reset. */ + + writel(capture ? S3C2412_IISFIC_RXFLUSH : S3C2412_IISFIC_TXFLUSH, + i2s->regs + S3C2412_IISFIC); + + /* clear again, just in case */ + writel(0x0, i2s->regs + S3C2412_IISFIC); + + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (!i2s->master) { + ret = s3c2412_snd_lrsync(i2s); + if (ret) + goto exit_err; + } + + local_irq_save(irqs); + + if (capture) + s3c2412_snd_rxctrl(i2s, 1); + else + s3c2412_snd_txctrl(i2s, 1); + + local_irq_restore(irqs); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + local_irq_save(irqs); + + if (capture) + s3c2412_snd_rxctrl(i2s, 0); + else + s3c2412_snd_txctrl(i2s, 0); + + local_irq_restore(irqs); + break; + default: + ret = -EINVAL; + break; + } + +exit_err: + return ret; +} + +/* + * Set S3C2412 Clock dividers + */ +static int s3c2412_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + struct s3c_i2sv2_info *i2s = to_info(cpu_dai); + u32 reg; + + DBG("%s(%p, %d, %d)\n", __func__, cpu_dai, div_id, div); + + switch (div_id) { + case S3C_I2SV2_DIV_BCLK: + reg = readl(i2s->regs + S3C2412_IISMOD); + reg &= ~S3C2412_IISMOD_BCLK_MASK; + writel(reg | div, i2s->regs + S3C2412_IISMOD); + + DBG("%s: MOD=%08x\n", __func__, readl(i2s->regs + S3C2412_IISMOD)); + break; + + case S3C_I2SV2_DIV_RCLK: + if (div > 3) { + /* convert value to bit field */ + + switch (div) { + case 256: + div = S3C2412_IISMOD_RCLK_256FS; + break; + + case 384: + div = S3C2412_IISMOD_RCLK_384FS; + break; + + case 512: + div = S3C2412_IISMOD_RCLK_512FS; + break; + + case 768: + div = S3C2412_IISMOD_RCLK_768FS; + break; + + default: + return -EINVAL; + } + } + + reg = readl(i2s->regs + S3C2412_IISMOD); + reg &= ~S3C2412_IISMOD_RCLK_MASK; + writel(reg | div, i2s->regs + S3C2412_IISMOD); + DBG("%s: MOD=%08x\n", __func__, readl(i2s->regs + S3C2412_IISMOD)); + break; + + case S3C_I2SV2_DIV_PRESCALER: + if (div >= 0) { + writel((div << 8) | S3C2412_IISPSR_PSREN, + i2s->regs + S3C2412_IISPSR); + } else { + writel(0x0, i2s->regs + S3C2412_IISPSR); + } + DBG("%s: PSR=%08x\n", __func__, readl(i2s->regs + S3C2412_IISPSR)); + break; + + default: + return -EINVAL; + } + + return 0; +} + +/* default table of all avaialable root fs divisors */ +static unsigned int iis_fs_tab[] = { 256, 512, 384, 768 }; + +int s3c2412_iis_calc_rate(struct s3c_i2sv2_rate_calc *info, + unsigned int *fstab, + unsigned int rate, struct clk *clk) +{ + unsigned long clkrate = clk_get_rate(clk); + unsigned int div; + unsigned int fsclk; + unsigned int actual; + unsigned int fs; + unsigned int fsdiv; + signed int deviation = 0; + unsigned int best_fs = 0; + unsigned int best_div = 0; + unsigned int best_rate = 0; + unsigned int best_deviation = INT_MAX; + + if (fstab == NULL) + fstab = iis_fs_tab; + + for (fs = 0; fs < ARRAY_SIZE(iis_fs_tab); fs++) { + fsdiv = iis_fs_tab[fs]; + + fsclk = clkrate / fsdiv; + div = fsclk / rate; + + if ((fsclk % rate) > (rate / 2)) + div++; + + if (div <= 1) + continue; + + actual = clkrate / (fsdiv * div); + deviation = actual - rate; + + printk(KERN_DEBUG "%dfs: div %d => result %d, deviation %d\n", + fsdiv, div, actual, deviation); + + deviation = abs(deviation); + + if (deviation < best_deviation) { + best_fs = fsdiv; + best_div = div; + best_rate = actual; + best_deviation = deviation; + } + + if (deviation == 0) + break; + } + + printk(KERN_DEBUG "best: fs=%d, div=%d, rate=%d\n", + best_fs, best_div, best_rate); + + info->fs_div = best_fs; + info->clk_div = best_div; + + return 0; +} +EXPORT_SYMBOL_GPL(s3c2412_iis_calc_rate); + +int s3c_i2sv2_probe(struct platform_device *pdev, + struct snd_soc_dai *dai, + struct s3c_i2sv2_info *i2s, + unsigned long base) +{ + struct device *dev = &pdev->dev; + + i2s->dev = dev; + + /* record our i2s structure for later use in the callbacks */ + dai->private_data = i2s; + + i2s->regs = ioremap(base, 0x100); + if (i2s->regs == NULL) { + dev_err(dev, "cannot ioremap registers\n"); + return -ENXIO; + } + + i2s->iis_pclk = clk_get(dev, "iis"); + if (i2s->iis_pclk == NULL) { + DBG("failed to get iis_clock\n"); + iounmap(i2s->regs); + return -ENOENT; + } + + clk_enable(i2s->iis_pclk); + + s3c2412_snd_txctrl(i2s, 0); + s3c2412_snd_rxctrl(i2s, 0); + + return 0; +} + +EXPORT_SYMBOL_GPL(s3c_i2sv2_probe); + +#ifdef CONFIG_PM +static int s3c2412_i2s_suspend(struct snd_soc_dai *dai) +{ + struct s3c_i2sv2_info *i2s = to_info(dai); + u32 iismod; + + if (dai->active) { + i2s->suspend_iismod = readl(i2s->regs + S3C2412_IISMOD); + i2s->suspend_iiscon = readl(i2s->regs + S3C2412_IISCON); + i2s->suspend_iispsr = readl(i2s->regs + S3C2412_IISPSR); + + /* some basic suspend checks */ + + iismod = readl(i2s->regs + S3C2412_IISMOD); + + if (iismod & S3C2412_IISCON_RXDMA_ACTIVE) + pr_warning("%s: RXDMA active?\n", __func__); + + if (iismod & S3C2412_IISCON_TXDMA_ACTIVE) + pr_warning("%s: TXDMA active?\n", __func__); + + if (iismod & S3C2412_IISCON_IIS_ACTIVE) + pr_warning("%s: IIS active\n", __func__); + } + + return 0; +} + +static int s3c2412_i2s_resume(struct snd_soc_dai *dai) +{ + struct s3c_i2sv2_info *i2s = to_info(dai); + + pr_info("dai_active %d, IISMOD %08x, IISCON %08x\n", + dai->active, i2s->suspend_iismod, i2s->suspend_iiscon); + + if (dai->active) { + writel(i2s->suspend_iiscon, i2s->regs + S3C2412_IISCON); + writel(i2s->suspend_iismod, i2s->regs + S3C2412_IISMOD); + writel(i2s->suspend_iispsr, i2s->regs + S3C2412_IISPSR); + + writel(S3C2412_IISFIC_RXFLUSH | S3C2412_IISFIC_TXFLUSH, + i2s->regs + S3C2412_IISFIC); + + ndelay(250); + writel(0x0, i2s->regs + S3C2412_IISFIC); + } + + return 0; +} +#else +#define s3c2412_i2s_suspend NULL +#define s3c2412_i2s_resume NULL +#endif + +int s3c_i2sv2_register_dai(struct snd_soc_dai *dai) +{ + dai->ops.trigger = s3c2412_i2s_trigger; + dai->ops.hw_params = s3c2412_i2s_hw_params; + dai->ops.set_fmt = s3c2412_i2s_set_fmt; + dai->ops.set_clkdiv = s3c2412_i2s_set_clkdiv; + + dai->suspend = s3c2412_i2s_suspend; + dai->resume = s3c2412_i2s_resume; + + return snd_soc_register_dai(dai); +} + +EXPORT_SYMBOL_GPL(s3c_i2sv2_register_dai); diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.h b/sound/soc/s3c24xx/s3c-i2s-v2.h new file mode 100644 index 00000000000..f66854a77fb --- /dev/null +++ b/sound/soc/s3c24xx/s3c-i2s-v2.h @@ -0,0 +1,90 @@ +/* sound/soc/s3c24xx/s3c-i2s-v2.h + * + * ALSA Soc Audio Layer - S3C_I2SV2 I2S driver + * + * Copyright (c) 2007 Simtec Electronics + * http://armlinux.simtec.co.uk/ + * Ben Dooks + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. +*/ + +/* This code is the core support for the I2S block found in a number of + * Samsung SoC devices which is unofficially named I2S-V2. Currently the + * S3C2412 and the S3C64XX series use this block to provide 1 or 2 I2S + * channels via configurable GPIO. + */ + +#ifndef __SND_SOC_S3C24XX_S3C_I2SV2_I2S_H +#define __SND_SOC_S3C24XX_S3C_I2SV2_I2S_H __FILE__ + +#define S3C_I2SV2_DIV_BCLK (1) +#define S3C_I2SV2_DIV_RCLK (2) +#define S3C_I2SV2_DIV_PRESCALER (3) + +/** + * struct s3c_i2sv2_info - S3C I2S-V2 information + * @dev: The parent device passed to use from the probe. + * @regs: The pointer to the device registe block. + * @master: True if the I2S core is the I2S bit clock master. + * @dma_playback: DMA information for playback channel. + * @dma_capture: DMA information for capture channel. + * @suspend_iismod: PM save for the IISMOD register. + * @suspend_iiscon: PM save for the IISCON register. + * @suspend_iispsr: PM save for the IISPSR register. + * + * This is the private codec state for the hardware associated with an + * I2S channel such as the register mappings and clock sources. + */ +struct s3c_i2sv2_info { + struct device *dev; + void __iomem *regs; + + struct clk *iis_pclk; + struct clk *iis_cclk; + struct clk *iis_clk; + + unsigned char master; + + struct s3c24xx_pcm_dma_params *dma_playback; + struct s3c24xx_pcm_dma_params *dma_capture; + + u32 suspend_iismod; + u32 suspend_iiscon; + u32 suspend_iispsr; +}; + +struct s3c_i2sv2_rate_calc { + unsigned int clk_div; /* for prescaler */ + unsigned int fs_div; /* for root frame clock */ +}; + +extern int s3c_i2sv2_iis_calc_rate(struct s3c_i2sv2_rate_calc *info, + unsigned int *fstab, + unsigned int rate, struct clk *clk); + +/** + * s3c_i2sv2_probe - probe for i2s device helper + * @pdev: The platform device supplied to the original probe. + * @dai: The ASoC DAI structure supplied to the original probe. + * @i2s: Our local i2s structure to fill in. + * @base: The base address for the registers. + */ +extern int s3c_i2sv2_probe(struct platform_device *pdev, + struct snd_soc_dai *dai, + struct s3c_i2sv2_info *i2s, + unsigned long base); + +/** + * s3c_i2sv2_register_dai - register dai with soc core + * @dai: The snd_soc_dai structure to register + * + * Fill in any missing fields and then register the given dai with the + * soc core. + */ +extern int s3c_i2sv2_register_dai(struct snd_soc_dai *dai); + +#endif /* __SND_SOC_S3C24XX_S3C_I2SV2_I2S_H */ diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c index 3297698ff29..5099d939667 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.c +++ b/sound/soc/s3c24xx/s3c2412-i2s.c @@ -33,7 +33,7 @@ #include -#include +#include #include #include @@ -41,7 +41,6 @@ #include "s3c2412-i2s.h" #define S3C2412_I2S_DEBUG 0 -#define S3C2412_I2S_DEBUG_CON 0 #if S3C2412_I2S_DEBUG #define DBG(x...) printk(KERN_INFO x) @@ -71,431 +70,7 @@ static struct s3c24xx_pcm_dma_params s3c2412_i2s_pcm_stereo_in = { .dma_size = 4, }; -struct s3c2412_i2s_info { - struct device *dev; - void __iomem *regs; - struct clk *iis_clk; - struct clk *iis_pclk; - struct clk *iis_cclk; - - u32 suspend_iismod; - u32 suspend_iiscon; - u32 suspend_iispsr; -}; - -static struct s3c2412_i2s_info s3c2412_i2s; - -#define bit_set(v, b) (((v) & (b)) ? 1 : 0) - -#if S3C2412_I2S_DEBUG_CON -static void dbg_showcon(const char *fn, u32 con) -{ - printk(KERN_DEBUG "%s: LRI=%d, TXFEMPT=%d, RXFEMPT=%d, TXFFULL=%d, RXFFULL=%d\n", fn, - bit_set(con, S3C2412_IISCON_LRINDEX), - bit_set(con, S3C2412_IISCON_TXFIFO_EMPTY), - bit_set(con, S3C2412_IISCON_RXFIFO_EMPTY), - bit_set(con, S3C2412_IISCON_TXFIFO_FULL), - bit_set(con, S3C2412_IISCON_RXFIFO_FULL)); - - printk(KERN_DEBUG "%s: PAUSE: TXDMA=%d, RXDMA=%d, TXCH=%d, RXCH=%d\n", - fn, - bit_set(con, S3C2412_IISCON_TXDMA_PAUSE), - bit_set(con, S3C2412_IISCON_RXDMA_PAUSE), - bit_set(con, S3C2412_IISCON_TXCH_PAUSE), - bit_set(con, S3C2412_IISCON_RXCH_PAUSE)); - printk(KERN_DEBUG "%s: ACTIVE: TXDMA=%d, RXDMA=%d, IIS=%d\n", fn, - bit_set(con, S3C2412_IISCON_TXDMA_ACTIVE), - bit_set(con, S3C2412_IISCON_RXDMA_ACTIVE), - bit_set(con, S3C2412_IISCON_IIS_ACTIVE)); -} -#else -static inline void dbg_showcon(const char *fn, u32 con) -{ -} -#endif - -/* Turn on or off the transmission path. */ -static void s3c2412_snd_txctrl(int on) -{ - struct s3c2412_i2s_info *i2s = &s3c2412_i2s; - void __iomem *regs = i2s->regs; - u32 fic, con, mod; - - DBG("%s(%d)\n", __func__, on); - - fic = readl(regs + S3C2412_IISFIC); - con = readl(regs + S3C2412_IISCON); - mod = readl(regs + S3C2412_IISMOD); - - DBG("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); - - if (on) { - con |= S3C2412_IISCON_TXDMA_ACTIVE | S3C2412_IISCON_IIS_ACTIVE; - con &= ~S3C2412_IISCON_TXDMA_PAUSE; - con &= ~S3C2412_IISCON_TXCH_PAUSE; - - switch (mod & S3C2412_IISMOD_MODE_MASK) { - case S3C2412_IISMOD_MODE_TXONLY: - case S3C2412_IISMOD_MODE_TXRX: - /* do nothing, we are in the right mode */ - break; - - case S3C2412_IISMOD_MODE_RXONLY: - mod &= ~S3C2412_IISMOD_MODE_MASK; - mod |= S3C2412_IISMOD_MODE_TXRX; - break; - - default: - dev_err(i2s->dev, "TXEN: Invalid MODE in IISMOD\n"); - } - - writel(con, regs + S3C2412_IISCON); - writel(mod, regs + S3C2412_IISMOD); - } else { - /* Note, we do not have any indication that the FIFO problems - * tha the S3C2410/2440 had apply here, so we should be able - * to disable the DMA and TX without resetting the FIFOS. - */ - - con |= S3C2412_IISCON_TXDMA_PAUSE; - con |= S3C2412_IISCON_TXCH_PAUSE; - con &= ~S3C2412_IISCON_TXDMA_ACTIVE; - - switch (mod & S3C2412_IISMOD_MODE_MASK) { - case S3C2412_IISMOD_MODE_TXRX: - mod &= ~S3C2412_IISMOD_MODE_MASK; - mod |= S3C2412_IISMOD_MODE_RXONLY; - break; - - case S3C2412_IISMOD_MODE_TXONLY: - mod &= ~S3C2412_IISMOD_MODE_MASK; - con &= ~S3C2412_IISCON_IIS_ACTIVE; - break; - - default: - dev_err(i2s->dev, "TXDIS: Invalid MODE in IISMOD\n"); - } - - writel(mod, regs + S3C2412_IISMOD); - writel(con, regs + S3C2412_IISCON); - } - - fic = readl(regs + S3C2412_IISFIC); - dbg_showcon(__func__, con); - DBG("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); -} - -static void s3c2412_snd_rxctrl(int on) -{ - struct s3c2412_i2s_info *i2s = &s3c2412_i2s; - void __iomem *regs = i2s->regs; - u32 fic, con, mod; - - DBG("%s(%d)\n", __func__, on); - - fic = readl(regs + S3C2412_IISFIC); - con = readl(regs + S3C2412_IISCON); - mod = readl(regs + S3C2412_IISMOD); - - DBG("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); - - if (on) { - con |= S3C2412_IISCON_RXDMA_ACTIVE | S3C2412_IISCON_IIS_ACTIVE; - con &= ~S3C2412_IISCON_RXDMA_PAUSE; - con &= ~S3C2412_IISCON_RXCH_PAUSE; - - switch (mod & S3C2412_IISMOD_MODE_MASK) { - case S3C2412_IISMOD_MODE_TXRX: - case S3C2412_IISMOD_MODE_RXONLY: - /* do nothing, we are in the right mode */ - break; - - case S3C2412_IISMOD_MODE_TXONLY: - mod &= ~S3C2412_IISMOD_MODE_MASK; - mod |= S3C2412_IISMOD_MODE_TXRX; - break; - - default: - dev_err(i2s->dev, "RXEN: Invalid MODE in IISMOD\n"); - } - - writel(mod, regs + S3C2412_IISMOD); - writel(con, regs + S3C2412_IISCON); - } else { - /* See txctrl notes on FIFOs. */ - - con &= ~S3C2412_IISCON_RXDMA_ACTIVE; - con |= S3C2412_IISCON_RXDMA_PAUSE; - con |= S3C2412_IISCON_RXCH_PAUSE; - - switch (mod & S3C2412_IISMOD_MODE_MASK) { - case S3C2412_IISMOD_MODE_RXONLY: - con &= ~S3C2412_IISCON_IIS_ACTIVE; - mod &= ~S3C2412_IISMOD_MODE_MASK; - break; - - case S3C2412_IISMOD_MODE_TXRX: - mod &= ~S3C2412_IISMOD_MODE_MASK; - mod |= S3C2412_IISMOD_MODE_TXONLY; - break; - - default: - dev_err(i2s->dev, "RXEN: Invalid MODE in IISMOD\n"); - } - - writel(con, regs + S3C2412_IISCON); - writel(mod, regs + S3C2412_IISMOD); - } - - fic = readl(regs + S3C2412_IISFIC); - DBG("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); -} - - -/* - * Wait for the LR signal to allow synchronisation to the L/R clock - * from the codec. May only be needed for slave mode. - */ -static int s3c2412_snd_lrsync(void) -{ - u32 iiscon; - unsigned long timeout = jiffies + msecs_to_jiffies(5); - - DBG("Entered %s\n", __func__); - - while (1) { - iiscon = readl(s3c2412_i2s.regs + S3C2412_IISCON); - if (iiscon & S3C2412_IISCON_LRINDEX) - break; - - if (timeout < jiffies) { - printk(KERN_ERR "%s: timeout\n", __func__); - return -ETIMEDOUT; - } - } - - return 0; -} - -/* - * Check whether CPU is the master or slave - */ -static inline int s3c2412_snd_is_clkmaster(void) -{ - u32 iismod = readl(s3c2412_i2s.regs + S3C2412_IISMOD); - - DBG("Entered %s\n", __func__); - - iismod &= S3C2412_IISMOD_MASTER_MASK; - return !(iismod == S3C2412_IISMOD_SLAVE); -} - -/* - * Set S3C2412 I2S DAI format - */ -static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai, - unsigned int fmt) -{ - u32 iismod; - - - DBG("Entered %s\n", __func__); - - iismod = readl(s3c2412_i2s.regs + S3C2412_IISMOD); - DBG("hw_params r: IISMOD: %x \n", iismod); - - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - iismod &= ~S3C2412_IISMOD_MASTER_MASK; - iismod |= S3C2412_IISMOD_SLAVE; - break; - case SND_SOC_DAIFMT_CBS_CFS: - iismod &= ~S3C2412_IISMOD_MASTER_MASK; - iismod |= S3C2412_IISMOD_MASTER_INTERNAL; - break; - default: - DBG("unknwon master/slave format\n"); - return -EINVAL; - } - - iismod &= ~S3C2412_IISMOD_SDF_MASK; - - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_RIGHT_J: - iismod |= S3C2412_IISMOD_SDF_MSB; - break; - case SND_SOC_DAIFMT_LEFT_J: - iismod |= S3C2412_IISMOD_SDF_LSB; - break; - case SND_SOC_DAIFMT_I2S: - iismod |= S3C2412_IISMOD_SDF_IIS; - break; - default: - DBG("Unknown data format\n"); - return -EINVAL; - } - - writel(iismod, s3c2412_i2s.regs + S3C2412_IISMOD); - DBG("hw_params w: IISMOD: %x \n", iismod); - return 0; -} - -static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - u32 iismod; - - DBG("Entered %s\n", __func__); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - rtd->dai->cpu_dai->dma_data = &s3c2412_i2s_pcm_stereo_out; - else - rtd->dai->cpu_dai->dma_data = &s3c2412_i2s_pcm_stereo_in; - - /* Working copies of register */ - iismod = readl(s3c2412_i2s.regs + S3C2412_IISMOD); - DBG("%s: r: IISMOD: %x\n", __func__, iismod); - - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S8: - iismod |= S3C2412_IISMOD_8BIT; - break; - case SNDRV_PCM_FORMAT_S16_LE: - iismod &= ~S3C2412_IISMOD_8BIT; - break; - } - - writel(iismod, s3c2412_i2s.regs + S3C2412_IISMOD); - DBG("%s: w: IISMOD: %x\n", __func__, iismod); - return 0; -} - -static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd, - struct snd_soc_dai *dai) -{ - int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); - unsigned long irqs; - int ret = 0; - - DBG("Entered %s\n", __func__); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - /* On start, ensure that the FIFOs are cleared and reset. */ - - writel(capture ? S3C2412_IISFIC_RXFLUSH : S3C2412_IISFIC_TXFLUSH, - s3c2412_i2s.regs + S3C2412_IISFIC); - - /* clear again, just in case */ - writel(0x0, s3c2412_i2s.regs + S3C2412_IISFIC); - - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (!s3c2412_snd_is_clkmaster()) { - ret = s3c2412_snd_lrsync(); - if (ret) - goto exit_err; - } - - local_irq_save(irqs); - - if (capture) - s3c2412_snd_rxctrl(1); - else - s3c2412_snd_txctrl(1); - - local_irq_restore(irqs); - break; - - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - local_irq_save(irqs); - - if (capture) - s3c2412_snd_rxctrl(0); - else - s3c2412_snd_txctrl(0); - - local_irq_restore(irqs); - break; - default: - ret = -EINVAL; - break; - } - -exit_err: - return ret; -} - -/* default table of all avaialable root fs divisors */ -static unsigned int s3c2412_iis_fs[] = { 256, 512, 384, 768, 0 }; - -int s3c2412_iis_calc_rate(struct s3c2412_rate_calc *info, - unsigned int *fstab, - unsigned int rate, struct clk *clk) -{ - unsigned long clkrate = clk_get_rate(clk); - unsigned int div; - unsigned int fsclk; - unsigned int actual; - unsigned int fs; - unsigned int fsdiv; - signed int deviation = 0; - unsigned int best_fs = 0; - unsigned int best_div = 0; - unsigned int best_rate = 0; - unsigned int best_deviation = INT_MAX; - - - if (fstab == NULL) - fstab = s3c2412_iis_fs; - - for (fs = 0;; fs++) { - fsdiv = s3c2412_iis_fs[fs]; - - if (fsdiv == 0) - break; - - fsclk = clkrate / fsdiv; - div = fsclk / rate; - - if ((fsclk % rate) > (rate / 2)) - div++; - - if (div <= 1) - continue; - - actual = clkrate / (fsdiv * div); - deviation = actual - rate; - - printk(KERN_DEBUG "%dfs: div %d => result %d, deviation %d\n", - fsdiv, div, actual, deviation); - - deviation = abs(deviation); - - if (deviation < best_deviation) { - best_fs = fsdiv; - best_div = div; - best_rate = actual; - best_deviation = deviation; - } - - if (deviation == 0) - break; - } - - printk(KERN_DEBUG "best: fs=%d, div=%d, rate=%d\n", - best_fs, best_div, best_rate); - - info->fs_div = best_fs; - info->clk_div = best_div; - - return 0; -} -EXPORT_SYMBOL_GPL(s3c2412_iis_calc_rate); +static struct s3c_i2sv2_info s3c2412_i2s; /* * Set S3C2412 Clock source @@ -510,10 +85,12 @@ static int s3c2412_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, switch (clk_id) { case S3C2412_CLKSRC_PCLK: + s3c2412_i2s.master = 1; iismod &= ~S3C2412_IISMOD_MASTER_MASK; iismod |= S3C2412_IISMOD_MASTER_INTERNAL; break; case S3C2412_CLKSRC_I2SCLK: + s3c2412_i2s.master = 0; iismod &= ~S3C2412_IISMOD_MASTER_MASK; iismod |= S3C2412_IISMOD_MASTER_EXTERNAL; break; @@ -525,74 +102,6 @@ static int s3c2412_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, return 0; } -/* - * Set S3C2412 Clock dividers - */ -static int s3c2412_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai, - int div_id, int div) -{ - struct s3c2412_i2s_info *i2s = &s3c2412_i2s; - u32 reg; - - DBG("%s(%p, %d, %d)\n", __func__, cpu_dai, div_id, div); - - switch (div_id) { - case S3C2412_DIV_BCLK: - reg = readl(i2s->regs + S3C2412_IISMOD); - reg &= ~S3C2412_IISMOD_BCLK_MASK; - writel(reg | div, i2s->regs + S3C2412_IISMOD); - - DBG("%s: MOD=%08x\n", __func__, readl(i2s->regs + S3C2412_IISMOD)); - break; - - case S3C2412_DIV_RCLK: - if (div > 3) { - /* convert value to bit field */ - - switch (div) { - case 256: - div = S3C2412_IISMOD_RCLK_256FS; - break; - - case 384: - div = S3C2412_IISMOD_RCLK_384FS; - break; - - case 512: - div = S3C2412_IISMOD_RCLK_512FS; - break; - - case 768: - div = S3C2412_IISMOD_RCLK_768FS; - break; - - default: - return -EINVAL; - } - } - - reg = readl(s3c2412_i2s.regs + S3C2412_IISMOD); - reg &= ~S3C2412_IISMOD_RCLK_MASK; - writel(reg | div, i2s->regs + S3C2412_IISMOD); - DBG("%s: MOD=%08x\n", __func__, readl(i2s->regs + S3C2412_IISMOD)); - break; - - case S3C2412_DIV_PRESCALER: - if (div >= 0) { - writel((div << 8) | S3C2412_IISPSR_PSREN, - i2s->regs + S3C2412_IISPSR); - } else { - writel(0x0, i2s->regs + S3C2412_IISPSR); - } - DBG("%s: PSR=%08x\n", __func__, readl(i2s->regs + S3C2412_IISPSR)); - break; - - default: - return -EINVAL; - } - - return 0; -} struct clk *s3c2412_get_iisclk(void) { @@ -604,20 +113,16 @@ EXPORT_SYMBOL_GPL(s3c2412_get_iisclk); static int s3c2412_i2s_probe(struct platform_device *pdev, struct snd_soc_dai *dai) { - DBG("Entered %s\n", __func__); + int ret; - s3c2412_i2s.dev = &pdev->dev; + DBG("Entered %s\n", __func__); - s3c2412_i2s.regs = ioremap(S3C2410_PA_IIS, 0x100); - if (s3c2412_i2s.regs == NULL) - return -ENXIO; + ret = s3c_i2sv2_probe(pdev, dai, &s3c2412_i2s, S3C2410_PA_IIS); + if (ret) + return ret; - s3c2412_i2s.iis_pclk = clk_get(&pdev->dev, "iis"); - if (s3c2412_i2s.iis_pclk == NULL) { - DBG("failed to get iis_clock\n"); - iounmap(s3c2412_i2s.regs); - return -ENODEV; - } + s3c2412_i2s.dma_capture = &s3c2412_i2s_pcm_stereo_in; + s3c2412_i2s.dma_playback = &s3c2412_i2s_pcm_stereo_out; s3c2412_i2s.iis_cclk = clk_get(&pdev->dev, "i2sclk"); if (s3c2412_i2s.iis_cclk == NULL) { @@ -626,12 +131,12 @@ static int s3c2412_i2s_probe(struct platform_device *pdev, return -ENODEV; } - clk_set_parent(s3c2412_i2s.iis_cclk, clk_get(NULL, "mpll")); + /* Set MPLL as the source for IIS CLK */ - clk_enable(s3c2412_i2s.iis_pclk); + clk_set_parent(s3c2412_i2s.iis_cclk, clk_get(NULL, "mpll")); clk_enable(s3c2412_i2s.iis_cclk); - s3c2412_i2s.iis_clk = s3c2412_i2s.iis_pclk; + s3c2412_i2s.iis_cclk = s3c2412_i2s.iis_pclk; /* Configure the I2S pins in correct mode */ s3c2410_gpio_cfgpin(S3C2410_GPE0, S3C2410_GPE0_I2SLRCK); @@ -640,78 +145,18 @@ static int s3c2412_i2s_probe(struct platform_device *pdev, s3c2410_gpio_cfgpin(S3C2410_GPE3, S3C2410_GPE3_I2SSDI); s3c2410_gpio_cfgpin(S3C2410_GPE4, S3C2410_GPE4_I2SSDO); - s3c2412_snd_txctrl(0); - s3c2412_snd_rxctrl(0); - - return 0; -} - -#ifdef CONFIG_PM -static int s3c2412_i2s_suspend(struct snd_soc_dai *dai) -{ - struct s3c2412_i2s_info *i2s = &s3c2412_i2s; - u32 iismod; - - if (dai->active) { - i2s->suspend_iismod = readl(i2s->regs + S3C2412_IISMOD); - i2s->suspend_iiscon = readl(i2s->regs + S3C2412_IISCON); - i2s->suspend_iispsr = readl(i2s->regs + S3C2412_IISPSR); - - /* some basic suspend checks */ - - iismod = readl(i2s->regs + S3C2412_IISMOD); - - if (iismod & S3C2412_IISCON_RXDMA_ACTIVE) - pr_warning("%s: RXDMA active?\n", __func__); - - if (iismod & S3C2412_IISCON_TXDMA_ACTIVE) - pr_warning("%s: TXDMA active?\n", __func__); - - if (iismod & S3C2412_IISCON_IIS_ACTIVE) - pr_warning("%s: IIS active\n", __func__); - } - return 0; } -static int s3c2412_i2s_resume(struct snd_soc_dai *dai) -{ - struct s3c2412_i2s_info *i2s = &s3c2412_i2s; - - pr_info("dai_active %d, IISMOD %08x, IISCON %08x\n", - dai->active, i2s->suspend_iismod, i2s->suspend_iiscon); - - if (dai->active) { - writel(i2s->suspend_iiscon, i2s->regs + S3C2412_IISCON); - writel(i2s->suspend_iismod, i2s->regs + S3C2412_IISMOD); - writel(i2s->suspend_iispsr, i2s->regs + S3C2412_IISPSR); - - writel(S3C2412_IISFIC_RXFLUSH | S3C2412_IISFIC_TXFLUSH, - i2s->regs + S3C2412_IISFIC); - - ndelay(250); - writel(0x0, i2s->regs + S3C2412_IISFIC); - - } - - return 0; -} -#else -#define s3c2412_i2s_suspend NULL -#define s3c2412_i2s_resume NULL -#endif /* CONFIG_PM */ - #define S3C2412_I2S_RATES \ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) struct snd_soc_dai s3c2412_i2s_dai = { - .name = "s3c2412-i2s", - .id = 0, - .probe = s3c2412_i2s_probe, - .suspend = s3c2412_i2s_suspend, - .resume = s3c2412_i2s_resume, + .name = "s3c2412-i2s", + .id = 0, + .probe = s3c2412_i2s_probe, .playback = { .channels_min = 2, .channels_max = 2, @@ -725,10 +170,6 @@ struct snd_soc_dai s3c2412_i2s_dai = { .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE, }, .ops = { - .trigger = s3c2412_i2s_trigger, - .hw_params = s3c2412_i2s_hw_params, - .set_fmt = s3c2412_i2s_set_fmt, - .set_clkdiv = s3c2412_i2s_set_clkdiv, .set_sysclk = s3c2412_i2s_set_sysclk, }, }; @@ -736,7 +177,7 @@ EXPORT_SYMBOL_GPL(s3c2412_i2s_dai); static int __init s3c2412_i2s_init(void) { - return snd_soc_register_dai(&s3c2412_i2s_dai); + return s3c_i2sv2_register_dai(&s3c2412_i2s_dai); } module_init(s3c2412_i2s_init); @@ -746,7 +187,6 @@ static void __exit s3c2412_i2s_exit(void) } module_exit(s3c2412_i2s_exit); - /* Module information */ MODULE_AUTHOR("Ben Dooks, "); MODULE_DESCRIPTION("S3C2412 I2S SoC Interface"); diff --git a/sound/soc/s3c24xx/s3c2412-i2s.h b/sound/soc/s3c24xx/s3c2412-i2s.h index aac08a25e54..92848e54be1 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.h +++ b/sound/soc/s3c24xx/s3c2412-i2s.h @@ -15,9 +15,11 @@ #ifndef __SND_SOC_S3C24XX_S3C2412_I2S_H #define __SND_SOC_S3C24XX_S3C2412_I2S_H __FILE__ -#define S3C2412_DIV_BCLK (1) -#define S3C2412_DIV_RCLK (2) -#define S3C2412_DIV_PRESCALER (3) +#include "s3c-i2s-v2.h" + +#define S3C2412_DIV_BCLK S3C_I2SV2_DIV_BCLK +#define S3C2412_DIV_RCLK S3C_I2SV2_DIV_RCLK +#define S3C2412_DIV_PRESCALER S3C_I2SV2_DIV_PRESCALER #define S3C2412_CLKSRC_PCLK (0) #define S3C2412_CLKSRC_I2SCLK (1) @@ -26,13 +28,4 @@ extern struct clk *s3c2412_get_iisclk(void); extern struct snd_soc_dai s3c2412_i2s_dai; -struct s3c2412_rate_calc { - unsigned int clk_div; /* for prescaler */ - unsigned int fs_div; /* for root frame clock */ -}; - -extern int s3c2412_iis_calc_rate(struct s3c2412_rate_calc *info, - unsigned int *fstab, - unsigned int rate, struct clk *clk); - #endif /* __SND_SOC_S3C24XX_S3C2412_I2S_H */ -- cgit v1.2.3-70-g09d2 From f8cf8176c7fc2c790e900595755b93e30633754d Mon Sep 17 00:00:00 2001 From: Ben Dooks Date: Wed, 4 Mar 2009 00:49:31 +0000 Subject: ASoC: Add s3c64xx-i2s support Add the initial code to support the S3C64XX I2S hardware using the s3c-i2s-v2 core code. Signed-off-by: Ben Dooks Signed-off-by: Mark Brown --- arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h | 3 + sound/soc/s3c24xx/Kconfig | 6 +- sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/s3c64xx-i2s.c | 220 ++++++++++++++++++++++ sound/soc/s3c24xx/s3c64xx-i2s.h | 31 +++ 5 files changed, 261 insertions(+), 1 deletion(-) create mode 100644 sound/soc/s3c24xx/s3c64xx-i2s.c create mode 100644 sound/soc/s3c24xx/s3c64xx-i2s.h diff --git a/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h b/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h index 25d4058bcfe..a5600b381d4 100644 --- a/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h +++ b/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h @@ -33,6 +33,9 @@ #define S3C2412_IISCON_RXDMA_ACTIVE (1 << 1) #define S3C2412_IISCON_IIS_ACTIVE (1 << 0) +#define S3C64XX_IISMOD_IMS_PCLK (0 << 10) +#define S3C64XX_IISMOD_IMS_SYSMUX (1 << 10) + #define S3C2412_IISMOD_MASTER_INTERNAL (0 << 10) #define S3C2412_IISMOD_MASTER_EXTERNAL (1 << 10) #define S3C2412_IISMOD_SLAVE (2 << 10) diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index bd98adaa5b0..036a4965c95 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -1,6 +1,6 @@ config SND_S3C24XX_SOC tristate "SoC Audio for the Samsung S3C24XX chips" - depends on ARCH_S3C2410 + depends on ARCH_S3C2410 || ARCH_S3C64XX help Say Y or M if you want to add support for codecs attached to the S3C24XX AC97, I2S or SSP interface. You will also need @@ -16,6 +16,10 @@ config SND_S3C2412_SOC_I2S tristate select SND_S3C_I2SV2_SOC +config SND_S3C64XX_SOC_I2S + tristate + select SND_S3C_I2SV2_SOC + config SND_S3C2443_SOC_AC97 tristate select AC97_BUS diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 848981e18ac..07a93a2ebe5 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -2,6 +2,7 @@ snd-soc-s3c24xx-objs := s3c24xx-pcm.o snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o +snd-soc-s3c64xx-i2s-objs := s3c64xx-i2s.o snd-soc-s3c2443-ac97-objs := s3c2443-ac97.o snd-soc-s3c-i2s-v2-objs := s3c-i2s-v2.o @@ -9,6 +10,7 @@ obj-$(CONFIG_SND_S3C24XX_SOC) += snd-soc-s3c24xx.o obj-$(CONFIG_SND_S3C24XX_SOC_I2S) += snd-soc-s3c24xx-i2s.o obj-$(CONFIG_SND_S3C2443_SOC_AC97) += snd-soc-s3c2443-ac97.o obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o +obj-$(CONFIG_SND_S3C64XX_SOC_I2S) += snd-soc-s3c64xx-i2s.o obj-$(CONFIG_SND_S3C_I2SV2_SOC) += snd-soc-s3c-i2s-v2.o # S3C24XX Machine Support diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c new file mode 100644 index 00000000000..6e1e85dc1ff --- /dev/null +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c @@ -0,0 +1,220 @@ +/* sound/soc/s3c24xx/s3c64xx-i2s.c + * + * ALSA SoC Audio Layer - S3C64XX I2S driver + * + * Copyright 2008 Openmoko, Inc. + * Copyright 2008 Simtec Electronics + * Ben Dooks + * http://armlinux.simtec.co.uk/ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include + +#include +#include + +#include "s3c24xx-pcm.h" +#include "s3c64xx-i2s.h" + +static struct s3c2410_dma_client s3c64xx_dma_client_out = { + .name = "I2S PCM Stereo out" +}; + +static struct s3c2410_dma_client s3c64xx_dma_client_in = { + .name = "I2S PCM Stereo in" +}; + +static struct s3c24xx_pcm_dma_params s3c64xx_i2s_pcm_stereo_out[2] = { + [0] = { + .channel = DMACH_I2S0_OUT, + .client = &s3c64xx_dma_client_out, + .dma_addr = S3C64XX_PA_IIS0 + S3C2412_IISTXD, + .dma_size = 4, + }, + [1] = { + .channel = DMACH_I2S1_OUT, + .client = &s3c64xx_dma_client_out, + .dma_addr = S3C64XX_PA_IIS1 + S3C2412_IISTXD, + .dma_size = 4, + }, +}; + +static struct s3c24xx_pcm_dma_params s3c64xx_i2s_pcm_stereo_in[2] = { + [0] = { + .channel = DMACH_I2S0_IN, + .client = &s3c64xx_dma_client_in, + .dma_addr = S3C64XX_PA_IIS0 + S3C2412_IISRXD, + .dma_size = 4, + }, + [1] = { + .channel = DMACH_I2S1_IN, + .client = &s3c64xx_dma_client_in, + .dma_addr = S3C64XX_PA_IIS1 + S3C2412_IISRXD, + .dma_size = 4, + }, +}; + +static struct s3c_i2sv2_info s3c64xx_i2s[2]; + +static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai) +{ + return cpu_dai->private_data; +} + +static int s3c64xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + struct s3c_i2sv2_info *i2s = to_info(cpu_dai); + u32 iismod = readl(i2s->regs + S3C2412_IISMOD); + + switch (clk_id) { + case S3C64XX_CLKSRC_PCLK: + iismod &= ~S3C64XX_IISMOD_IMS_SYSMUX; + break; + + case S3C64XX_CLKSRC_MUX: + iismod |= S3C64XX_IISMOD_IMS_SYSMUX; + break; + + default: + return -EINVAL; + } + + writel(iismod, i2s->regs + S3C2412_IISMOD); + + return 0; +} + + +unsigned long s3c64xx_i2s_get_clockrate(struct snd_soc_dai *dai) +{ + struct s3c_i2sv2_info *i2s = to_info(dai); + + return clk_get_rate(i2s->iis_cclk); +} +EXPORT_SYMBOL_GPL(s3c64xx_i2s_get_clockrate); + +static int s3c64xx_i2s_probe(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + struct device *dev = &pdev->dev; + struct s3c_i2sv2_info *i2s; + int ret; + + dev_dbg(dev, "%s: probing dai %d\n", __func__, pdev->id); + + if (pdev->id < 0 || pdev->id > ARRAY_SIZE(s3c64xx_i2s)) { + dev_err(dev, "id %d out of range\n", pdev->id); + return -EINVAL; + } + + i2s = &s3c64xx_i2s[pdev->id]; + + ret = s3c_i2sv2_probe(pdev, dai, i2s, + pdev->id ? S3C64XX_PA_IIS1 : S3C64XX_PA_IIS0); + if (ret) + return ret; + + i2s->dma_capture = &s3c64xx_i2s_pcm_stereo_in[pdev->id]; + i2s->dma_playback = &s3c64xx_i2s_pcm_stereo_out[pdev->id]; + + i2s->iis_cclk = clk_get(dev, "audio-bus"); + if (IS_ERR(i2s->iis_cclk)) { + dev_err(dev, "failed to get audio-bus"); + iounmap(i2s->regs); + return -ENODEV; + } + + /* configure GPIO for i2s port */ + switch (pdev->id) { + case 0: + s3c_gpio_cfgpin(S3C64XX_GPD(0), S3C64XX_GPD0_I2S0_CLK); + s3c_gpio_cfgpin(S3C64XX_GPD(1), S3C64XX_GPD1_I2S0_CDCLK); + s3c_gpio_cfgpin(S3C64XX_GPD(2), S3C64XX_GPD2_I2S0_LRCLK); + s3c_gpio_cfgpin(S3C64XX_GPD(3), S3C64XX_GPD3_I2S0_DI); + s3c_gpio_cfgpin(S3C64XX_GPD(4), S3C64XX_GPD4_I2S0_D0); + break; + case 1: + s3c_gpio_cfgpin(S3C64XX_GPE(0), S3C64XX_GPE0_I2S1_CLK); + s3c_gpio_cfgpin(S3C64XX_GPE(1), S3C64XX_GPE1_I2S1_CDCLK); + s3c_gpio_cfgpin(S3C64XX_GPE(2), S3C64XX_GPE2_I2S1_LRCLK); + s3c_gpio_cfgpin(S3C64XX_GPE(3), S3C64XX_GPE3_I2S1_DI); + s3c_gpio_cfgpin(S3C64XX_GPE(4), S3C64XX_GPE4_I2S1_D0); + } + + return 0; +} + + +#define S3C64XX_I2S_RATES \ + (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) + +#define S3C64XX_I2S_FMTS \ + (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE) + +struct snd_soc_dai s3c64xx_i2s_dai = { + .name = "s3c64xx-i2s", + .id = 0, + .probe = s3c64xx_i2s_probe, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = S3C64XX_I2S_RATES, + .formats = S3C64XX_I2S_FMTS, + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = S3C64XX_I2S_RATES, + .formats = S3C64XX_I2S_FMTS, + }, + .ops = { + .set_sysclk = s3c64xx_i2s_set_sysclk, + }, +}; +EXPORT_SYMBOL_GPL(s3c64xx_i2s_dai); + +static int __init s3c64xx_i2s_init(void) +{ + return s3c_i2sv2_register_dai(&s3c64xx_i2s_dai); +} +module_init(s3c64xx_i2s_init); + +static void __exit s3c64xx_i2s_exit(void) +{ + snd_soc_unregister_dai(&s3c64xx_i2s_dai); +} +module_exit(s3c64xx_i2s_exit); + +/* Module information */ +MODULE_AUTHOR("Ben Dooks, "); +MODULE_DESCRIPTION("S3C64XX I2S SoC Interface"); +MODULE_LICENSE("GPL"); + + + diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.h b/sound/soc/s3c24xx/s3c64xx-i2s.h new file mode 100644 index 00000000000..b7ffe3c38b6 --- /dev/null +++ b/sound/soc/s3c24xx/s3c64xx-i2s.h @@ -0,0 +1,31 @@ +/* sound/soc/s3c24xx/s3c64xx-i2s.h + * + * ALSA SoC Audio Layer - S3C64XX I2S driver + * + * Copyright 2008 Openmoko, Inc. + * Copyright 2008 Simtec Electronics + * Ben Dooks + * http://armlinux.simtec.co.uk/ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __SND_SOC_S3C24XX_S3C64XX_I2S_H +#define __SND_SOC_S3C24XX_S3C64XX_I2S_H __FILE__ + +#include "s3c-i2s-v2.h" + +#define S3C64XX_DIV_BCLK S3C_I2SV2_DIV_BCLK +#define S3C64XX_DIV_RCLK S3C_I2SV2_DIV_RCLK +#define S3C64XX_DIV_PRESCALER S3C_I2SV2_DIV_PRESCALER + +#define S3C64XX_CLKSRC_PCLK (0) +#define S3C64XX_CLKSRC_MUX (1) + +extern struct snd_soc_dai s3c64xx_i2s_dai; + +extern unsigned long s3c64xx_i2s_get_clockrate(struct snd_soc_dai *cpu_dai); + +#endif /* __SND_SOC_S3C24XX_S3C64XX_I2S_H */ -- cgit v1.2.3-70-g09d2 From c36623a7543e7a23ceeafbeb7b34a9e020100898 Mon Sep 17 00:00:00 2001 From: Ben Dooks Date: Wed, 4 Mar 2009 00:49:34 +0000 Subject: ASoC: Select DMA if I2S is configured Select the relevant DMA implementation when the sound driver is selected. Signed-off-by: Ben Dooks Signed-off-by: Mark Brown --- sound/soc/s3c24xx/Kconfig | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 036a4965c95..f22dbeec48a 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -8,6 +8,7 @@ config SND_S3C24XX_SOC config SND_S3C24XX_SOC_I2S tristate + select S3C2410_DMA config SND_S3C_I2SV2_SOC tristate @@ -15,13 +16,16 @@ config SND_S3C_I2SV2_SOC config SND_S3C2412_SOC_I2S tristate select SND_S3C_I2SV2_SOC + select S3C2410_DMA config SND_S3C64XX_SOC_I2S tristate select SND_S3C_I2SV2_SOC + select S3C64XX_DMA config SND_S3C2443_SOC_AC97 tristate + select S3C2410_DMA select AC97_BUS select SND_SOC_AC97_BUS -- cgit v1.2.3-70-g09d2 From a1b3eaeb146937e954cdb6dcb67f8761b73e2eef Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 4 Mar 2009 20:17:48 +0000 Subject: ASoC: Refresh JIVE driver Remove uneeded startup callback and use snd_soc_dapm_nc_pin() Signed-off-by: Mark Brown --- sound/soc/s3c24xx/jive_wm8750.c | 27 ++++++--------------------- 1 file changed, 6 insertions(+), 21 deletions(-) diff --git a/sound/soc/s3c24xx/jive_wm8750.c b/sound/soc/s3c24xx/jive_wm8750.c index 7dfe26ea8f4..32063790d95 100644 --- a/sound/soc/s3c24xx/jive_wm8750.c +++ b/sound/soc/s3c24xx/jive_wm8750.c @@ -45,20 +45,6 @@ static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = { SND_SOC_DAPM_LINE("Line In", NULL), }; -static int jive_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->socdev->codec; - - snd_soc_dapm_enable_pin(codec, "Headphone Jack"); - snd_soc_dapm_enable_pin(codec, "Internal Speaker"); - snd_soc_dapm_enable_pin(codec, "Line In"); - - snd_soc_dapm_sync(codec); - - return 0; -} - static int jive_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -119,7 +105,6 @@ static int jive_hw_params(struct snd_pcm_substream *substream, } static struct snd_soc_ops jive_ops = { - .startup = jive_startup, .hw_params = jive_hw_params, }; @@ -128,12 +113,12 @@ static int jive_wm8750_init(struct snd_soc_codec *codec) int err; /* These endpoints are not being used. */ - snd_soc_dapm_disable_pin(codec, "LINPUT2"); - snd_soc_dapm_disable_pin(codec, "RINPUT2"); - snd_soc_dapm_disable_pin(codec, "LINPUT3"); - snd_soc_dapm_disable_pin(codec, "RINPUT3"); - snd_soc_dapm_disable_pin(codec, "OUT3"); - snd_soc_dapm_disable_pin(codec, "MONO"); + snd_soc_dapm_nc_pin(codec, "LINPUT2"); + snd_soc_dapm_nc_pin(codec, "RINPUT2"); + snd_soc_dapm_nc_pin(codec, "LINPUT3"); + snd_soc_dapm_nc_pin(codec, "RINPUT3"); + snd_soc_dapm_nc_pin(codec, "OUT3"); + snd_soc_dapm_nc_pin(codec, "MONO"); /* Add jive specific widgets */ err = snd_soc_dapm_new_controls(codec, wm8750_dapm_widgets, -- cgit v1.2.3-70-g09d2 From 89492be88616aa20b3a6c3eb512f83c0c7d0c8a3 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 5 Mar 2009 12:48:49 +0200 Subject: ASoC: TWL4030: Make the HS ramp delay configurable Enum type for selecting the desired ramp delay for the headset output. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 18 ++++++++++++++---- 1 file changed, 14 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 535d8ce2c32..86bb15cc82c 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -584,12 +584,11 @@ static int headsetl_event(struct snd_soc_dapm_widget *w, /* Save the current volume */ hs_gain = twl4030_read_reg_cache(w->codec, TWL4030_REG_HS_GAIN_SET); + hs_pop = twl4030_read_reg_cache(w->codec, TWL4030_REG_HS_POPN_SET); switch (event) { case SND_SOC_DAPM_POST_PMU: /* Do the anti-pop/bias ramp enable according to the TRM */ - hs_pop = TWL4030_RAMP_DELAY_645MS; - twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop); hs_pop |= TWL4030_VMID_EN; twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop); /* Is this needed? Can we just use whatever gain here? */ @@ -603,8 +602,6 @@ static int headsetl_event(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_POST_PMD: /* Do the anti-pop/bias ramp disable according to the TRM */ - hs_pop = twl4030_read_reg_cache(w->codec, - TWL4030_REG_HS_POPN_SET); hs_pop &= ~TWL4030_RAMP_EN; twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop); /* Bypass the reg_cache to mute the headset */ @@ -847,6 +844,17 @@ static DECLARE_TLV_DB_SCALE(digital_capture_tlv, 0, 100, 0); */ static DECLARE_TLV_DB_SCALE(input_gain_tlv, 0, 600, 0); +static const char *twl4030_rampdelay_texts[] = { + "27/20/14 ms", "55/40/27 ms", "109/81/55 ms", "218/161/109 ms", + "437/323/218 ms", "874/645/437 ms", "1748/1291/874 ms", + "3495/2581/1748 ms" +}; + +static const struct soc_enum twl4030_rampdelay_enum = + SOC_ENUM_SINGLE(TWL4030_REG_HS_POPN_SET, 2, + ARRAY_SIZE(twl4030_rampdelay_texts), + twl4030_rampdelay_texts); + static const struct snd_kcontrol_new twl4030_snd_controls[] = { /* Common playback gain controls */ SOC_DOUBLE_R_TLV("DAC1 Digital Fine Playback Volume", @@ -901,6 +909,8 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = { SOC_DOUBLE_TLV("Analog Capture Volume", TWL4030_REG_ANAMIC_GAIN, 0, 3, 5, 0, input_gain_tlv), + + SOC_ENUM("HS ramp delay", twl4030_rampdelay_enum), }; static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { -- cgit v1.2.3-70-g09d2 From 20a41eac4fbaa22d051d0fbaeaf3315d2d8c4860 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 4 Mar 2009 21:16:57 +0100 Subject: ASoC: Fix name of register bit in pxa-ssp A bit in PXA's SSCR0 register was erroneously named ADC but its name is in fact ACS (audio clock select). Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- arch/arm/mach-pxa/include/mach/regs-ssp.h | 2 +- sound/soc/pxa/pxa-ssp.c | 6 +++--- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/arch/arm/mach-pxa/include/mach/regs-ssp.h b/arch/arm/mach-pxa/include/mach/regs-ssp.h index 3c04cde2cf1..cacdcae451e 100644 --- a/arch/arm/mach-pxa/include/mach/regs-ssp.h +++ b/arch/arm/mach-pxa/include/mach/regs-ssp.h @@ -47,7 +47,7 @@ #define SSCR0_TUM (1 << 23) /* Transmit FIFO underrun interrupt mask */ #define SSCR0_FRDC (0x07000000) /* Frame rate divider control (mask) */ #define SSCR0_SlotsPerFrm(x) (((x) - 1) << 24) /* Time slots per frame [1..8] */ -#define SSCR0_ADC (1 << 30) /* Audio clock select */ +#define SSCR0_ACS (1 << 30) /* Audio clock select */ #define SSCR0_MOD (1 << 31) /* Mode (normal or network) */ #endif diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index c49bb12b0a6..7fc13f03d1d 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -300,7 +300,7 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, int val; u32 sscr0 = ssp_read_reg(ssp, SSCR0) & - ~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ADC); + ~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); dev_dbg(&ssp->pdev->dev, "pxa_ssp_set_dai_sysclk id: %d, clk_id %d, freq %d\n", @@ -328,7 +328,7 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, case PXA_SSP_CLK_AUDIO: priv->sysclk = 0; ssp_set_scr(&priv->dev, 1); - sscr0 |= SSCR0_ADC; + sscr0 |= SSCR0_ACS; break; default: return -ENODEV; @@ -524,7 +524,7 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, /* reset port settings */ sscr0 = ssp_read_reg(ssp, SSCR0) & - (SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ADC); + (SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); sscr1 = SSCR1_RxTresh(8) | SSCR1_TxTresh(7); sspsp = 0; -- cgit v1.2.3-70-g09d2 From 42aa3418ebd7b79be0e1ee7515e365c1574114f9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 1 Mar 2009 19:21:10 +0000 Subject: ASoC: Factor out DAPM widget power check into separate function Essentially simple code motion to facilitate refactoring of the power decisions. Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 258 +++++++++++++++++++++++++++------------------------ 1 file changed, 137 insertions(+), 121 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 4b8dbbfe2ef..7da6d0db40f 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -522,6 +522,137 @@ int dapm_reg_event(struct snd_soc_dapm_widget *w, } EXPORT_SYMBOL_GPL(dapm_reg_event); +/* + * Scan a single DAPM widget for a complete audio path and update the + * power status appropriately. + */ +static int dapm_power_widget(struct snd_soc_codec *codec, int event, + struct snd_soc_dapm_widget *w) +{ + int in, out, power_change, power, ret; + + /* vmid - no action */ + if (w->id == snd_soc_dapm_vmid) + return 0; + + /* active ADC */ + if (w->id == snd_soc_dapm_adc && w->active) { + in = is_connected_input_ep(w); + dapm_clear_walk(w->codec); + w->power = (in != 0) ? 1 : 0; + dapm_update_bits(w); + return 0; + } + + /* active DAC */ + if (w->id == snd_soc_dapm_dac && w->active) { + out = is_connected_output_ep(w); + dapm_clear_walk(w->codec); + w->power = (out != 0) ? 1 : 0; + dapm_update_bits(w); + return 0; + } + + /* pre and post event widgets */ + if (w->id == snd_soc_dapm_pre) { + if (!w->event) + return 0; + + if (event == SND_SOC_DAPM_STREAM_START) { + ret = w->event(w, + NULL, SND_SOC_DAPM_PRE_PMU); + if (ret < 0) + return ret; + } else if (event == SND_SOC_DAPM_STREAM_STOP) { + ret = w->event(w, + NULL, SND_SOC_DAPM_PRE_PMD); + if (ret < 0) + return ret; + } + return 0; + } + if (w->id == snd_soc_dapm_post) { + if (!w->event) + return 0; + + if (event == SND_SOC_DAPM_STREAM_START) { + ret = w->event(w, + NULL, SND_SOC_DAPM_POST_PMU); + if (ret < 0) + return ret; + } else if (event == SND_SOC_DAPM_STREAM_STOP) { + ret = w->event(w, + NULL, SND_SOC_DAPM_POST_PMD); + if (ret < 0) + return ret; + } + return 0; + } + + /* all other widgets */ + in = is_connected_input_ep(w); + dapm_clear_walk(w->codec); + out = is_connected_output_ep(w); + dapm_clear_walk(w->codec); + power = (out != 0 && in != 0) ? 1 : 0; + power_change = (w->power == power) ? 0 : 1; + w->power = power; + + if (!power_change) + return 0; + + /* call any power change event handlers */ + if (w->event) + pr_debug("power %s event for %s flags %x\n", + w->power ? "on" : "off", + w->name, w->event_flags); + + /* power up pre event */ + if (power && w->event && + (w->event_flags & SND_SOC_DAPM_PRE_PMU)) { + ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMU); + if (ret < 0) + return ret; + } + + /* power down pre event */ + if (!power && w->event && + (w->event_flags & SND_SOC_DAPM_PRE_PMD)) { + ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMD); + if (ret < 0) + return ret; + } + + /* Lower PGA volume to reduce pops */ + if (w->id == snd_soc_dapm_pga && !power) + dapm_set_pga(w, power); + + dapm_update_bits(w); + + /* Raise PGA volume to reduce pops */ + if (w->id == snd_soc_dapm_pga && power) + dapm_set_pga(w, power); + + /* power up post event */ + if (power && w->event && + (w->event_flags & SND_SOC_DAPM_POST_PMU)) { + ret = w->event(w, + NULL, SND_SOC_DAPM_POST_PMU); + if (ret < 0) + return ret; + } + + /* power down post event */ + if (!power && w->event && + (w->event_flags & SND_SOC_DAPM_POST_PMD)) { + ret = w->event(w, NULL, SND_SOC_DAPM_POST_PMD); + if (ret < 0) + return ret; + } + + return 0; +} + /* * Scan each dapm widget for complete audio path. * A complete path is a route that has valid endpoints i.e.:- @@ -534,7 +665,7 @@ EXPORT_SYMBOL_GPL(dapm_reg_event); static int dapm_power_widgets(struct snd_soc_codec *codec, int event) { struct snd_soc_dapm_widget *w; - int in, out, i, c = 1, *seq = NULL, ret = 0, power_change, power; + int i, c = 1, *seq = NULL, ret = 0; /* do we have a sequenced stream event */ if (event == SND_SOC_DAPM_STREAM_START) { @@ -545,135 +676,20 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) seq = dapm_down_seq; } - for(i = 0; i < c; i++) { + for (i = 0; i < c; i++) { list_for_each_entry(w, &codec->dapm_widgets, list) { /* is widget in stream order */ if (seq && seq[i] && w->id != seq[i]) continue; - /* vmid - no action */ - if (w->id == snd_soc_dapm_vmid) - continue; - - /* active ADC */ - if (w->id == snd_soc_dapm_adc && w->active) { - in = is_connected_input_ep(w); - dapm_clear_walk(w->codec); - w->power = (in != 0) ? 1 : 0; - dapm_update_bits(w); - continue; - } - - /* active DAC */ - if (w->id == snd_soc_dapm_dac && w->active) { - out = is_connected_output_ep(w); - dapm_clear_walk(w->codec); - w->power = (out != 0) ? 1 : 0; - dapm_update_bits(w); - continue; - } - - /* pre and post event widgets */ - if (w->id == snd_soc_dapm_pre) { - if (!w->event) - continue; - - if (event == SND_SOC_DAPM_STREAM_START) { - ret = w->event(w, - NULL, SND_SOC_DAPM_PRE_PMU); - if (ret < 0) - return ret; - } else if (event == SND_SOC_DAPM_STREAM_STOP) { - ret = w->event(w, - NULL, SND_SOC_DAPM_PRE_PMD); - if (ret < 0) - return ret; - } - continue; - } - if (w->id == snd_soc_dapm_post) { - if (!w->event) - continue; - - if (event == SND_SOC_DAPM_STREAM_START) { - ret = w->event(w, - NULL, SND_SOC_DAPM_POST_PMU); - if (ret < 0) - return ret; - } else if (event == SND_SOC_DAPM_STREAM_STOP) { - ret = w->event(w, - NULL, SND_SOC_DAPM_POST_PMD); - if (ret < 0) - return ret; - } - continue; - } - - /* all other widgets */ - in = is_connected_input_ep(w); - dapm_clear_walk(w->codec); - out = is_connected_output_ep(w); - dapm_clear_walk(w->codec); - power = (out != 0 && in != 0) ? 1 : 0; - power_change = (w->power == power) ? 0: 1; - w->power = power; - - if (!power_change) - continue; - - /* call any power change event handlers */ - if (w->event) - pr_debug("power %s event for %s flags %x\n", - w->power ? "on" : "off", - w->name, w->event_flags); - - /* power up pre event */ - if (power && w->event && - (w->event_flags & SND_SOC_DAPM_PRE_PMU)) { - ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMU); - if (ret < 0) - return ret; - } - - /* power down pre event */ - if (!power && w->event && - (w->event_flags & SND_SOC_DAPM_PRE_PMD)) { - ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMD); - if (ret < 0) - return ret; - } - - /* Lower PGA volume to reduce pops */ - if (w->id == snd_soc_dapm_pga && !power) - dapm_set_pga(w, power); - - dapm_update_bits(w); - - /* Raise PGA volume to reduce pops */ - if (w->id == snd_soc_dapm_pga && power) - dapm_set_pga(w, power); - - /* power up post event */ - if (power && w->event && - (w->event_flags & SND_SOC_DAPM_POST_PMU)) { - ret = w->event(w, - NULL, SND_SOC_DAPM_POST_PMU); - if (ret < 0) - return ret; - } - - /* power down post event */ - if (!power && w->event && - (w->event_flags & SND_SOC_DAPM_POST_PMD)) { - ret = w->event(w, NULL, SND_SOC_DAPM_POST_PMD); - if (ret < 0) - return ret; - } + ret = dapm_power_widget(codec, event, w); + if (ret != 0) + return ret; } } - return ret; + return 0; } #ifdef DEBUG -- cgit v1.2.3-70-g09d2 From b0c5033f02182d1e9634edc737df88b82264e820 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Thu, 5 Mar 2009 14:21:26 +0100 Subject: ASoC: add two more bitfields for PXA SSP Add two more bitfields for the PSP register. As they seem to exist for PXA3xx only, define them conditionally. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- arch/arm/mach-pxa/include/mach/regs-ssp.h | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/arch/arm/mach-pxa/include/mach/regs-ssp.h b/arch/arm/mach-pxa/include/mach/regs-ssp.h index cacdcae451e..f43905a2773 100644 --- a/arch/arm/mach-pxa/include/mach/regs-ssp.h +++ b/arch/arm/mach-pxa/include/mach/regs-ssp.h @@ -106,6 +106,11 @@ #define SSSR_TINT (1 << 19) /* Receiver Time-out Interrupt */ #define SSSR_PINT (1 << 18) /* Peripheral Trailing Byte Interrupt */ +#if defined(CONFIG_PXA3xx) +#define SSPSP_EDMYSTOP(x) ((x) << 28) /* Extended Dummy Stop */ +#define SSPSP_EDMYSTRT(x) ((x) << 26) /* Extended Dummy Start */ +#endif + #define SSPSP_FSRT (1 << 25) /* Frame Sync Relative Timing */ #define SSPSP_DMYSTOP(x) ((x) << 23) /* Dummy Stop */ #define SSPSP_SFRMWDTH(x) ((x) << 16) /* Serial Frame Width */ -- cgit v1.2.3-70-g09d2 From 07495f3e5af3a472f0f49957692cac15168fa528 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 5 Mar 2009 17:06:23 +0000 Subject: ASoC: Fix memory allocation for snd_soc_dapm_switch names snd_soc_dapm_switch ends up ends up in dapm_new_mixer() (since a switch is a special case of a mixer with only one input) but this wasn't correctly handled in the code. Also fix the coding style for the switch below while we're here. Reported-by: Joonyoung Shim Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7da6d0db40f..735903a7467 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -332,7 +332,7 @@ static int dapm_new_mixer(struct snd_soc_codec *codec, * kcontrol name. */ name_len = strlen(w->kcontrols[i].name) + 1; - if (w->id == snd_soc_dapm_mixer) + if (w->id != snd_soc_dapm_mixer_named_ctl) name_len += 1 + strlen(w->name); path->long_name = kmalloc(name_len, GFP_KERNEL); @@ -341,15 +341,14 @@ static int dapm_new_mixer(struct snd_soc_codec *codec, return -ENOMEM; switch (w->id) { - case snd_soc_dapm_mixer: default: snprintf(path->long_name, name_len, "%s %s", w->name, w->kcontrols[i].name); - break; + break; case snd_soc_dapm_mixer_named_ctl: snprintf(path->long_name, name_len, "%s", w->kcontrols[i].name); - break; + break; } path->long_name[name_len - 1] = '\0'; -- cgit v1.2.3-70-g09d2 From 499d8f4a528f1ebd0c19d89174fdc67130090c89 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 5 Mar 2009 17:26:15 +0000 Subject: ASoC: Update Kconfig for Samsung CPUs to reflect S3C64xx support We now support the 64xx series as well as the 24xx series - make sure people using Kconfig know this. Signed-off-by: Mark Brown --- sound/soc/s3c24xx/Kconfig | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index f22dbeec48a..78d01ff487c 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -1,10 +1,10 @@ config SND_S3C24XX_SOC - tristate "SoC Audio for the Samsung S3C24XX chips" + tristate "SoC Audio for the Samsung S3CXXXX chips" depends on ARCH_S3C2410 || ARCH_S3C64XX help Say Y or M if you want to add support for codecs attached to - the S3C24XX AC97, I2S or SSP interface. You will also need - to select the audio interfaces to support below. + the S3C24XX and S3C64XX AC97, I2S or SSP interface. You will + also need to select the audio interfaces to support below. config SND_S3C24XX_SOC_I2S tristate -- cgit v1.2.3-70-g09d2 From a454dad19e78388d9f140ad0dfa6a849c57d385d Mon Sep 17 00:00:00 2001 From: Timur Tabi Date: Thu, 5 Mar 2009 17:23:37 -0600 Subject: ASoC: add support for SSI asynchronous mode to the Freescale SSI drivers Add a new device tree property for the SSI node: "fsl,ssi-asynchronous". If defined, the SSI is programmed into asynchronous mode, otherwise it is programmed into synchronous mode. In asynchronous mode, pin SRCK must be connected to the same clock source as STFS, and pin SRFS must be connected to the same signal as STFS. Asynchronous mode allows playback and capture to use different sample sizes. It also technically allows different sample rates, but the driver does not support that. Signed-off-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 42 ++++++++++++++++++++++++++++++++---------- sound/soc/fsl/fsl_ssi.h | 2 ++ sound/soc/fsl/mpc8610_hpcd.c | 5 +++++ 3 files changed, 39 insertions(+), 10 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 6844009833d..8cb6bcf2c00 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -72,6 +72,7 @@ * @dev: struct device pointer * @playback: the number of playback streams opened * @capture: the number of capture streams opened + * @asynchronous: 0=synchronous mode, 1=asynchronous mode * @cpu_dai: the CPU DAI for this device * @dev_attr: the sysfs device attribute structure * @stats: SSI statistics @@ -86,6 +87,7 @@ struct fsl_ssi_private { struct device *dev; unsigned int playback; unsigned int capture; + int asynchronous; struct snd_soc_dai cpu_dai; struct device_attribute dev_attr; @@ -301,9 +303,10 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, * * FIXME: Little-endian samples require a different shift dir */ - clrsetbits_be32(&ssi->scr, CCSR_SSI_SCR_I2S_MODE_MASK, - CCSR_SSI_SCR_TFR_CLK_DIS | - CCSR_SSI_SCR_I2S_MODE_SLAVE | CCSR_SSI_SCR_SYN); + clrsetbits_be32(&ssi->scr, + CCSR_SSI_SCR_I2S_MODE_MASK | CCSR_SSI_SCR_SYN, + CCSR_SSI_SCR_TFR_CLK_DIS | CCSR_SSI_SCR_I2S_MODE_SLAVE + | (ssi_private->asynchronous ? 0 : CCSR_SSI_SCR_SYN)); out_be32(&ssi->stcr, CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFEN0 | @@ -382,10 +385,15 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, SNDRV_PCM_HW_PARAM_RATE, first_runtime->rate, first_runtime->rate); - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_SAMPLE_BITS, - first_runtime->sample_bits, - first_runtime->sample_bits); + /* If we're in synchronous mode, then we need to constrain + * the sample size as well. We don't support independent sample + * rates in asynchronous mode. + */ + if (!ssi_private->asynchronous) + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + first_runtime->sample_bits, + first_runtime->sample_bits); ssi_private->second_stream = substream; } @@ -421,13 +429,18 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, struct ccsr_ssi __iomem *ssi = ssi_private->ssi; unsigned int sample_size = snd_pcm_format_width(params_format(hw_params)); - u32 wl; + u32 wl = CCSR_SSI_SxCCR_WL(sample_size); /* The SSI should always be disabled at this points (SSIEN=0) */ - wl = CCSR_SSI_SxCCR_WL(sample_size); /* In synchronous mode, the SSI uses STCCR for capture */ - clrsetbits_be32(&ssi->stccr, CCSR_SSI_SxCCR_WL_MASK, wl); + if ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) || + !ssi_private->asynchronous) + clrsetbits_be32(&ssi->stccr, + CCSR_SSI_SxCCR_WL_MASK, wl); + else + clrsetbits_be32(&ssi->srccr, + CCSR_SSI_SxCCR_WL_MASK, wl); } return 0; @@ -653,6 +666,7 @@ struct snd_soc_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info) ssi_private->ssi_phys = ssi_info->ssi_phys; ssi_private->irq = ssi_info->irq; ssi_private->dev = ssi_info->dev; + ssi_private->asynchronous = ssi_info->asynchronous; ssi_private->dev->driver_data = fsl_ssi_dai; @@ -703,6 +717,14 @@ void fsl_ssi_destroy_dai(struct snd_soc_dai *fsl_ssi_dai) } EXPORT_SYMBOL_GPL(fsl_ssi_destroy_dai); +static int __init fsl_ssi_init(void) +{ + printk(KERN_INFO "Freescale Synchronous Serial Interface (SSI) ASoC Driver\n"); + + return 0; +} +module_init(fsl_ssi_init); + MODULE_AUTHOR("Timur Tabi "); MODULE_DESCRIPTION("Freescale Synchronous Serial Interface (SSI) ASoC Driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/fsl/fsl_ssi.h b/sound/soc/fsl/fsl_ssi.h index 83b44d700e3..eade01feaab 100644 --- a/sound/soc/fsl/fsl_ssi.h +++ b/sound/soc/fsl/fsl_ssi.h @@ -208,6 +208,7 @@ struct ccsr_ssi { * ssi_phys: physical address of the SSI registers * irq: IRQ of this SSI * dev: struct device, used to create the sysfs statistics file + * asynchronous: 0=synchronous mode, 1=asynchronous mode */ struct fsl_ssi_info { unsigned int id; @@ -215,6 +216,7 @@ struct fsl_ssi_info { dma_addr_t ssi_phys; unsigned int irq; struct device *dev; + int asynchronous; }; struct snd_soc_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info); diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index acf39a646b2..ef67d1cdffe 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -353,6 +353,11 @@ static int mpc8610_hpcd_probe(struct of_device *ofdev, } ssi_info.irq = machine_data->ssi_irq; + /* Do we want to use asynchronous mode? */ + ssi_info.asynchronous = + of_find_property(np, "fsl,ssi-asynchronous", NULL) ? 1 : 0; + if (ssi_info.asynchronous) + dev_info(&ofdev->dev, "using asynchronous mode\n"); /* Map the global utilities registers. */ guts_np = of_find_compatible_node(NULL, NULL, "fsl,mpc8610-guts"); -- cgit v1.2.3-70-g09d2 From de0b988828a45f4fefc96ff2fbed3ba2184319b9 Mon Sep 17 00:00:00 2001 From: "Lopez Cruz, Misael" Date: Thu, 5 Mar 2009 11:32:31 -0600 Subject: ASoC: Add headset jack detection for SDP3430 machine driver Add headset jack detection for SDP3430 boards using SoC jack reporting interface. Headset detection on SDP3430 board is achieved through TWL4030 GPIO_2 pin. Signed-off-by: Misael Lopez Cruz Signed-off-by: Mark Brown --- sound/soc/omap/sdp3430.c | 43 +++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 41 insertions(+), 2 deletions(-) diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index 4eab4b491de..715c648203a 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -28,6 +28,7 @@ #include #include #include +#include #include #include @@ -122,7 +123,7 @@ static int sdp3430_twl4030_init(struct snd_soc_codec *codec) /* SDP3430 connected pins */ snd_soc_dapm_enable_pin(codec, "Ext Mic"); snd_soc_dapm_enable_pin(codec, "Ext Spk"); - snd_soc_dapm_enable_pin(codec, "Headset Jack"); + snd_soc_dapm_disable_pin(codec, "Headset Jack"); /* TWL4030 not connected pins */ snd_soc_dapm_nc_pin(codec, "AUXL"); @@ -144,6 +145,27 @@ static int sdp3430_twl4030_init(struct snd_soc_codec *codec) return ret; } +/* Headset jack */ +static struct snd_soc_jack hs_jack; + +/* Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin hs_jack_pins[] = { + { + .pin = "Headset Jack", + .mask = SND_JACK_HEADSET, + }, +}; + +/* Headset jack detection gpios */ +static struct snd_soc_jack_gpio hs_jack_gpios[] = { + { + .gpio = (OMAP_MAX_GPIO_LINES + 2), + .name = "hsdet-gpio", + .report = SND_JACK_HEADSET, + .debounce_time = 200, + }, +}; + /* Digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link sdp3430_dai = { .name = "TWL4030", @@ -194,7 +216,21 @@ static int __init sdp3430_soc_init(void) if (ret) goto err1; - return 0; + /* Headset jack detection */ + ret = snd_soc_jack_new(&snd_soc_sdp3430, "SDP3430 headset jack", + SND_JACK_HEADSET, &hs_jack); + if (ret) + return ret; + + ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), + hs_jack_pins); + if (ret) + return ret; + + ret = snd_soc_jack_add_gpios(&hs_jack, ARRAY_SIZE(hs_jack_gpios), + hs_jack_gpios); + + return ret; err1: printk(KERN_ERR "Unable to add platform device\n"); @@ -206,6 +242,9 @@ module_init(sdp3430_soc_init); static void __exit sdp3430_soc_exit(void) { + snd_soc_jack_free_gpios(&hs_jack, ARRAY_SIZE(hs_jack_gpios), + hs_jack_gpios); + platform_device_unregister(sdp3430_snd_device); } module_exit(sdp3430_soc_exit); -- cgit v1.2.3-70-g09d2 From 3465d93a128acce836249e3de40932d2ed25cac6 Mon Sep 17 00:00:00 2001 From: Mike Frysinger Date: Fri, 6 Mar 2009 15:53:28 +0800 Subject: ASoC: Blackfin: move gpio_err behind the define that is only user of it Signed-off-by: Mike Frysinger Signed-off-by: Bryan Wu Signed-off-by: Mark Brown --- sound/soc/blackfin/bf5xx-ac97.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c index 5885702c78f..8a935f2d176 100644 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ b/sound/soc/blackfin/bf5xx-ac97.c @@ -357,8 +357,8 @@ sport_config_err: sport_err: #ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM); -#endif gpio_err: +#endif peripheral_free_list(sport_req[sport_num]); peripheral_err: free_page((unsigned long)cmd_count); -- cgit v1.2.3-70-g09d2 From 67a9c573b5bf39bc6b40c322c58640687c1b79fe Mon Sep 17 00:00:00 2001 From: Mike Frysinger Date: Fri, 6 Mar 2009 15:53:30 +0800 Subject: ASoC: Blackfin: fix typo in MUTE definition Reported-by: Rob Maris Signed-off-by: Mike Frysinger Signed-off-by: Bryan Wu Signed-off-by: Mark Brown --- sound/soc/codecs/ad73311.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/ad73311.h b/sound/soc/codecs/ad73311.h index 507ce0c30ed..569573d2d4d 100644 --- a/sound/soc/codecs/ad73311.h +++ b/sound/soc/codecs/ad73311.h @@ -70,7 +70,7 @@ #define REGD_IGS(x) (x & 0x7) #define REGD_RMOD (1 << 3) #define REGD_OGS(x) ((x & 0x7) << 4) -#define REGD_MUTE (x << 7) +#define REGD_MUTE (1 << 7) /* Control register E */ #define CTRL_REG_E (4 << 8) -- cgit v1.2.3-70-g09d2 From 26bd7b496cabc232fcff9ae0249828420c52b5af Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 6 Mar 2009 11:32:17 +0000 Subject: ASoC: Staticise workqueue function for GPIO jack detection Signed-off-by: Mark Brown --- sound/soc/soc-jack.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index bdf2484c222..28346fb2e70 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -143,7 +143,7 @@ EXPORT_SYMBOL_GPL(snd_soc_jack_add_pins); #ifdef CONFIG_GPIOLIB /* gpio detect */ -void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio) +static void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio) { struct snd_soc_jack *jack = gpio->jack; int enable; -- cgit v1.2.3-70-g09d2 From ee7d476714464206317d4420d67e3bfa0308448d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 6 Mar 2009 18:04:34 +0000 Subject: ASoC: Re-remove hand-rolled pr_debug() macros The recent set of S3C64xx patches re-added a lot of uses of DBG() that had previously been removed - revert this so the standard pr_debug() macro is used. Signed-off-by: Mark Brown --- sound/soc/s3c24xx/neo1973_wm8753.c | 44 ++++++++++++++-------------------- sound/soc/s3c24xx/s3c-i2s-v2.c | 49 ++++++++++++++++---------------------- sound/soc/s3c24xx/s3c2412-i2s.c | 12 +++------- sound/soc/s3c24xx/s3c24xx-i2s.c | 49 ++++++++++++++++---------------------- sound/soc/s3c24xx/s3c24xx-pcm.c | 45 +++++++++++++++------------------- 5 files changed, 82 insertions(+), 117 deletions(-) diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index 74573352718..5f6aeec0437 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -40,14 +40,6 @@ #include "s3c24xx-pcm.h" #include "s3c24xx-i2s.h" -/* Debugging stuff */ -#define S3C24XX_SOC_NEO1973_WM8753_DEBUG 0 -#if S3C24XX_SOC_NEO1973_WM8753_DEBUG -#define DBG(x...) printk(KERN_DEBUG "s3c24xx-soc-neo1973-wm8753: " x) -#else -#define DBG(x...) -#endif - /* define the scenarios */ #define NEO_AUDIO_OFF 0 #define NEO_GSM_CALL_AUDIO_HANDSET 1 @@ -72,7 +64,7 @@ static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream, int ret = 0; unsigned long iis_clkrate; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); iis_clkrate = s3c24xx_i2s_get_clockrate(); @@ -158,7 +150,7 @@ static int neo1973_hifi_hw_free(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); /* disable the PLL */ return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0); @@ -181,7 +173,7 @@ static int neo1973_voice_hw_params(struct snd_pcm_substream *substream, int ret = 0; unsigned long iis_clkrate; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); iis_clkrate = s3c24xx_i2s_get_clockrate(); @@ -224,7 +216,7 @@ static int neo1973_voice_hw_free(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); /* disable the PLL */ return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0); @@ -246,7 +238,7 @@ static int neo1973_get_scenario(struct snd_kcontrol *kcontrol, static int set_scenario_endpoints(struct snd_soc_codec *codec, int scenario) { - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); switch (neo1973_scenario) { case NEO_AUDIO_OFF: @@ -330,7 +322,7 @@ static int neo1973_set_scenario(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); if (neo1973_scenario == ucontrol->value.integer.value[0]) return 0; @@ -344,7 +336,7 @@ static u8 lm4857_regs[4] = {0x00, 0x40, 0x80, 0xC0}; static void lm4857_write_regs(void) { - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); if (i2c_master_send(i2c, lm4857_regs, 4) != 4) printk(KERN_ERR "lm4857: i2c write failed\n"); @@ -357,7 +349,7 @@ static int lm4857_get_reg(struct snd_kcontrol *kcontrol, int shift = (kcontrol->private_value >> 8) & 0x0F; int mask = (kcontrol->private_value >> 16) & 0xFF; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); ucontrol->value.integer.value[0] = (lm4857_regs[reg] >> shift) & mask; return 0; @@ -385,7 +377,7 @@ static int lm4857_get_mode(struct snd_kcontrol *kcontrol, { u8 value = lm4857_regs[LM4857_CTRL] & 0x0F; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); if (value) value -= 5; @@ -399,7 +391,7 @@ static int lm4857_set_mode(struct snd_kcontrol *kcontrol, { u8 value = ucontrol->value.integer.value[0]; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); if (value) value += 5; @@ -508,7 +500,7 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec) { int i, err; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); /* set up NC codec pins */ snd_soc_dapm_nc_pin(codec, "LOUT2"); @@ -593,7 +585,7 @@ static struct snd_soc_device neo1973_snd_devdata = { static int lm4857_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); i2c = client; @@ -603,7 +595,7 @@ static int lm4857_i2c_probe(struct i2c_client *client, static int lm4857_i2c_remove(struct i2c_client *client) { - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); i2c = NULL; @@ -614,7 +606,7 @@ static u8 lm4857_state; static int lm4857_suspend(struct i2c_client *dev, pm_message_t state) { - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); dev_dbg(&dev->dev, "lm4857_suspend\n"); lm4857_state = lm4857_regs[LM4857_CTRL] & 0xf; @@ -627,7 +619,7 @@ static int lm4857_suspend(struct i2c_client *dev, pm_message_t state) static int lm4857_resume(struct i2c_client *dev) { - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); if (lm4857_state) { lm4857_regs[LM4857_CTRL] |= (lm4857_state & 0x0f); @@ -638,7 +630,7 @@ static int lm4857_resume(struct i2c_client *dev) static void lm4857_shutdown(struct i2c_client *dev) { - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); dev_dbg(&dev->dev, "lm4857_shutdown\n"); lm4857_regs[LM4857_CTRL] &= 0xf0; @@ -669,7 +661,7 @@ static int __init neo1973_init(void) { int ret; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); if (!machine_is_neo1973_gta01()) { printk(KERN_INFO @@ -700,7 +692,7 @@ static int __init neo1973_init(void) static void __exit neo1973_exit(void) { - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); i2c_del_driver(&lm4857_i2c_driver); platform_device_unregister(neo1973_snd_device); diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index 43262e1e8f9..b7b8e47ae36 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -38,13 +38,6 @@ #include "s3c-i2s-v2.h" #define S3C2412_I2S_DEBUG_CON 0 -#define S3C2412_I2S_DEBUG 0 - -#if S3C2412_I2S_DEBUG -#define DBG(x...) printk(KERN_INFO x) -#else -#define DBG(x...) do { } while (0) -#endif static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai) { @@ -87,13 +80,13 @@ void s3c2412_snd_txctrl(struct s3c_i2sv2_info *i2s, int on) void __iomem *regs = i2s->regs; u32 fic, con, mod; - DBG("%s(%d)\n", __func__, on); + pr_debug("%s(%d)\n", __func__, on); fic = readl(regs + S3C2412_IISFIC); con = readl(regs + S3C2412_IISCON); mod = readl(regs + S3C2412_IISMOD); - DBG("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); + pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); if (on) { con |= S3C2412_IISCON_TXDMA_ACTIVE | S3C2412_IISCON_IIS_ACTIVE; @@ -148,7 +141,7 @@ void s3c2412_snd_txctrl(struct s3c_i2sv2_info *i2s, int on) fic = readl(regs + S3C2412_IISFIC); dbg_showcon(__func__, con); - DBG("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); + pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); } EXPORT_SYMBOL_GPL(s3c2412_snd_txctrl); @@ -157,13 +150,13 @@ void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on) void __iomem *regs = i2s->regs; u32 fic, con, mod; - DBG("%s(%d)\n", __func__, on); + pr_debug("%s(%d)\n", __func__, on); fic = readl(regs + S3C2412_IISFIC); con = readl(regs + S3C2412_IISCON); mod = readl(regs + S3C2412_IISMOD); - DBG("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); + pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); if (on) { con |= S3C2412_IISCON_RXDMA_ACTIVE | S3C2412_IISCON_IIS_ACTIVE; @@ -214,7 +207,7 @@ void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on) } fic = readl(regs + S3C2412_IISFIC); - DBG("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); + pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); } EXPORT_SYMBOL_GPL(s3c2412_snd_rxctrl); @@ -227,7 +220,7 @@ static int s3c2412_snd_lrsync(struct s3c_i2sv2_info *i2s) u32 iiscon; unsigned long timeout = jiffies + msecs_to_jiffies(5); - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); while (1) { iiscon = readl(i2s->regs + S3C2412_IISCON); @@ -252,10 +245,10 @@ static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai, struct s3c_i2sv2_info *i2s = to_info(cpu_dai); u32 iismod; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); iismod = readl(i2s->regs + S3C2412_IISMOD); - DBG("hw_params r: IISMOD: %x \n", iismod); + pr_debug("hw_params r: IISMOD: %x \n", iismod); #if defined(CONFIG_CPU_S3C2412) || defined(CONFIG_CPU_S3C2413) #define IISMOD_MASTER_MASK S3C2412_IISMOD_MASTER_MASK @@ -288,7 +281,7 @@ static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai, iismod |= IISMOD_MASTER; break; default: - DBG("unknwon master/slave format\n"); + pr_debug("unknwon master/slave format\n"); return -EINVAL; } @@ -305,12 +298,12 @@ static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai, iismod |= S3C2412_IISMOD_SDF_IIS; break; default: - DBG("Unknown data format\n"); + pr_debug("Unknown data format\n"); return -EINVAL; } writel(iismod, i2s->regs + S3C2412_IISMOD); - DBG("hw_params w: IISMOD: %x \n", iismod); + pr_debug("hw_params w: IISMOD: %x \n", iismod); return 0; } @@ -323,7 +316,7 @@ static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream, struct s3c_i2sv2_info *i2s = to_info(dai->cpu_dai); u32 iismod; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) dai->cpu_dai->dma_data = i2s->dma_playback; @@ -332,7 +325,7 @@ static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream, /* Working copies of register */ iismod = readl(i2s->regs + S3C2412_IISMOD); - DBG("%s: r: IISMOD: %x\n", __func__, iismod); + pr_debug("%s: r: IISMOD: %x\n", __func__, iismod); switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: @@ -344,7 +337,7 @@ static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream, } writel(iismod, i2s->regs + S3C2412_IISMOD); - DBG("%s: w: IISMOD: %x\n", __func__, iismod); + pr_debug("%s: w: IISMOD: %x\n", __func__, iismod); return 0; } @@ -357,7 +350,7 @@ static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd, unsigned long irqs; int ret = 0; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -417,7 +410,7 @@ static int s3c2412_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai, struct s3c_i2sv2_info *i2s = to_info(cpu_dai); u32 reg; - DBG("%s(%p, %d, %d)\n", __func__, cpu_dai, div_id, div); + pr_debug("%s(%p, %d, %d)\n", __func__, cpu_dai, div_id, div); switch (div_id) { case S3C_I2SV2_DIV_BCLK: @@ -425,7 +418,7 @@ static int s3c2412_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai, reg &= ~S3C2412_IISMOD_BCLK_MASK; writel(reg | div, i2s->regs + S3C2412_IISMOD); - DBG("%s: MOD=%08x\n", __func__, readl(i2s->regs + S3C2412_IISMOD)); + pr_debug("%s: MOD=%08x\n", __func__, readl(i2s->regs + S3C2412_IISMOD)); break; case S3C_I2SV2_DIV_RCLK: @@ -457,7 +450,7 @@ static int s3c2412_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai, reg = readl(i2s->regs + S3C2412_IISMOD); reg &= ~S3C2412_IISMOD_RCLK_MASK; writel(reg | div, i2s->regs + S3C2412_IISMOD); - DBG("%s: MOD=%08x\n", __func__, readl(i2s->regs + S3C2412_IISMOD)); + pr_debug("%s: MOD=%08x\n", __func__, readl(i2s->regs + S3C2412_IISMOD)); break; case S3C_I2SV2_DIV_PRESCALER: @@ -467,7 +460,7 @@ static int s3c2412_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai, } else { writel(0x0, i2s->regs + S3C2412_IISPSR); } - DBG("%s: PSR=%08x\n", __func__, readl(i2s->regs + S3C2412_IISPSR)); + pr_debug("%s: PSR=%08x\n", __func__, readl(i2s->regs + S3C2412_IISPSR)); break; default: @@ -560,7 +553,7 @@ int s3c_i2sv2_probe(struct platform_device *pdev, i2s->iis_pclk = clk_get(dev, "iis"); if (i2s->iis_pclk == NULL) { - DBG("failed to get iis_clock\n"); + pr_debug("failed to get iis_clock\n"); iounmap(i2s->regs); return -ENOENT; } diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c index 5099d939667..1ceae690d01 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.c +++ b/sound/soc/s3c24xx/s3c2412-i2s.c @@ -42,12 +42,6 @@ #define S3C2412_I2S_DEBUG 0 -#if S3C2412_I2S_DEBUG -#define DBG(x...) printk(KERN_INFO x) -#else -#define DBG(x...) do { } while (0) -#endif - static struct s3c2410_dma_client s3c2412_dma_client_out = { .name = "I2S PCM Stereo out" }; @@ -80,7 +74,7 @@ static int s3c2412_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, { u32 iismod = readl(s3c2412_i2s.regs + S3C2412_IISMOD); - DBG("%s(%p, %d, %u, %d)\n", __func__, cpu_dai, clk_id, + pr_debug("%s(%p, %d, %u, %d)\n", __func__, cpu_dai, clk_id, freq, dir); switch (clk_id) { @@ -115,7 +109,7 @@ static int s3c2412_i2s_probe(struct platform_device *pdev, { int ret; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); ret = s3c_i2sv2_probe(pdev, dai, &s3c2412_i2s, S3C2410_PA_IIS); if (ret) @@ -126,7 +120,7 @@ static int s3c2412_i2s_probe(struct platform_device *pdev, s3c2412_i2s.iis_cclk = clk_get(&pdev->dev, "i2sclk"); if (s3c2412_i2s.iis_cclk == NULL) { - DBG("failed to get i2sclk clock\n"); + pr_debug("failed to get i2sclk clock\n"); iounmap(s3c2412_i2s.regs); return -ENODEV; } diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index c0c6ec1536b..2fbead33b7c 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -39,13 +39,6 @@ #include "s3c24xx-pcm.h" #include "s3c24xx-i2s.h" -#define S3C24XX_I2S_DEBUG 0 -#if S3C24XX_I2S_DEBUG -#define DBG(x...) printk(KERN_DEBUG "s3c24xx-i2s: " x) -#else -#define DBG(x...) -#endif - static struct s3c2410_dma_client s3c24xx_dma_client_out = { .name = "I2S PCM Stereo out" }; @@ -84,13 +77,13 @@ static void s3c24xx_snd_txctrl(int on) u32 iiscon; u32 iismod; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON); iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); - DBG("r: IISCON: %lx IISMOD: %lx IISFCON: %lx\n", iiscon, iismod, iisfcon); + pr_debug("r: IISCON: %lx IISMOD: %lx IISFCON: %lx\n", iiscon, iismod, iisfcon); if (on) { iisfcon |= S3C2410_IISFCON_TXDMA | S3C2410_IISFCON_TXENABLE; @@ -120,7 +113,7 @@ static void s3c24xx_snd_txctrl(int on) writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); } - DBG("w: IISCON: %lx IISMOD: %lx IISFCON: %lx\n", iiscon, iismod, iisfcon); + pr_debug("w: IISCON: %lx IISMOD: %lx IISFCON: %lx\n", iiscon, iismod, iisfcon); } static void s3c24xx_snd_rxctrl(int on) @@ -129,13 +122,13 @@ static void s3c24xx_snd_rxctrl(int on) u32 iiscon; u32 iismod; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON); iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); - DBG("r: IISCON: %lx IISMOD: %lx IISFCON: %lx\n", iiscon, iismod, iisfcon); + pr_debug("r: IISCON: %lx IISMOD: %lx IISFCON: %lx\n", iiscon, iismod, iisfcon); if (on) { iisfcon |= S3C2410_IISFCON_RXDMA | S3C2410_IISFCON_RXENABLE; @@ -165,7 +158,7 @@ static void s3c24xx_snd_rxctrl(int on) writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); } - DBG("w: IISCON: %lx IISMOD: %lx IISFCON: %lx\n", iiscon, iismod, iisfcon); + pr_debug("w: IISCON: %lx IISMOD: %lx IISFCON: %lx\n", iiscon, iismod, iisfcon); } /* @@ -177,7 +170,7 @@ static int s3c24xx_snd_lrsync(void) u32 iiscon; int timeout = 50; /* 5ms */ - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); while (1) { iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); @@ -197,7 +190,7 @@ static int s3c24xx_snd_lrsync(void) */ static inline int s3c24xx_snd_is_clkmaster(void) { - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); return (readl(s3c24xx_i2s.regs + S3C2410_IISMOD) & S3C2410_IISMOD_SLAVE) ? 0:1; } @@ -210,10 +203,10 @@ static int s3c24xx_i2s_set_fmt(struct snd_soc_dai *cpu_dai, { u32 iismod; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); - DBG("hw_params r: IISMOD: %lx \n", iismod); + pr_debug("hw_params r: IISMOD: %lx \n", iismod); switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: @@ -238,7 +231,7 @@ static int s3c24xx_i2s_set_fmt(struct snd_soc_dai *cpu_dai, } writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); - DBG("hw_params w: IISMOD: %lx \n", iismod); + pr_debug("hw_params w: IISMOD: %lx \n", iismod); return 0; } @@ -249,7 +242,7 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; u32 iismod; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) rtd->dai->cpu_dai->dma_data = &s3c24xx_i2s_pcm_stereo_out; @@ -258,7 +251,7 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, /* Working copies of register */ iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); - DBG("hw_params r: IISMOD: %lx\n", iismod); + pr_debug("hw_params r: IISMOD: %lx\n", iismod); switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: @@ -276,7 +269,7 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, } writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); - DBG("hw_params w: IISMOD: %lx\n", iismod); + pr_debug("hw_params w: IISMOD: %lx\n", iismod); return 0; } @@ -285,7 +278,7 @@ static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, { int ret = 0; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -327,7 +320,7 @@ static int s3c24xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, { u32 iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); iismod &= ~S3C2440_IISMOD_MPLL; @@ -353,7 +346,7 @@ static int s3c24xx_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai, { u32 reg; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); switch (div_id) { case S3C24XX_DIV_BCLK: @@ -389,7 +382,7 @@ EXPORT_SYMBOL_GPL(s3c24xx_i2s_get_clockrate); static int s3c24xx_i2s_probe(struct platform_device *pdev, struct snd_soc_dai *dai) { - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); s3c24xx_i2s.regs = ioremap(S3C2410_PA_IIS, 0x100); if (s3c24xx_i2s.regs == NULL) @@ -397,7 +390,7 @@ static int s3c24xx_i2s_probe(struct platform_device *pdev, s3c24xx_i2s.iis_clk = clk_get(&pdev->dev, "iis"); if (s3c24xx_i2s.iis_clk == NULL) { - DBG("failed to get iis_clock\n"); + pr_debug("failed to get iis_clock\n"); iounmap(s3c24xx_i2s.regs); return -ENODEV; } @@ -421,7 +414,7 @@ static int s3c24xx_i2s_probe(struct platform_device *pdev, #ifdef CONFIG_PM static int s3c24xx_i2s_suspend(struct snd_soc_dai *cpu_dai) { - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); s3c24xx_i2s.iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); s3c24xx_i2s.iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); @@ -435,7 +428,7 @@ static int s3c24xx_i2s_suspend(struct snd_soc_dai *cpu_dai) static int s3c24xx_i2s_resume(struct snd_soc_dai *cpu_dai) { - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); clk_enable(s3c24xx_i2s.iis_clk); writel(s3c24xx_i2s.iiscon, s3c24xx_i2s.regs + S3C2410_IISCON); diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c index 5c96c1ed629..269f5c8e7ca 100644 --- a/sound/soc/s3c24xx/s3c24xx-pcm.c +++ b/sound/soc/s3c24xx/s3c24xx-pcm.c @@ -33,13 +33,6 @@ #include "s3c24xx-pcm.h" -#define S3C24XX_PCM_DEBUG 0 -#if S3C24XX_PCM_DEBUG -#define DBG(x...) printk(KERN_DEBUG "s3c24xx-pcm: " x) -#else -#define DBG(x...) -#endif - static const struct snd_pcm_hardware s3c24xx_pcm_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | @@ -84,16 +77,16 @@ static void s3c24xx_pcm_enqueue(struct snd_pcm_substream *substream) dma_addr_t pos = prtd->dma_pos; int ret; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); while (prtd->dma_loaded < prtd->dma_limit) { unsigned long len = prtd->dma_period; - DBG("dma_loaded: %d\n", prtd->dma_loaded); + pr_debug("dma_loaded: %d\n", prtd->dma_loaded); if ((pos + len) > prtd->dma_end) { len = prtd->dma_end - pos; - DBG(KERN_DEBUG "%s: corrected dma len %ld\n", + pr_debug(KERN_DEBUG "%s: corrected dma len %ld\n", __func__, len); } @@ -119,7 +112,7 @@ static void s3c24xx_audio_buffdone(struct s3c2410_dma_chan *channel, struct snd_pcm_substream *substream = dev_id; struct s3c24xx_runtime_data *prtd; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); if (result == S3C2410_RES_ABORT || result == S3C2410_RES_ERR) return; @@ -148,7 +141,7 @@ static int s3c24xx_pcm_hw_params(struct snd_pcm_substream *substream, unsigned long totbytes = params_buffer_bytes(params); int ret = 0; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); /* return if this is a bufferless transfer e.g. * codec <--> BT codec or GSM modem -- lg FIXME */ @@ -161,14 +154,14 @@ static int s3c24xx_pcm_hw_params(struct snd_pcm_substream *substream, /* prepare DMA */ prtd->params = dma; - DBG("params %p, client %p, channel %d\n", prtd->params, + pr_debug("params %p, client %p, channel %d\n", prtd->params, prtd->params->client, prtd->params->channel); ret = s3c2410_dma_request(prtd->params->channel, prtd->params->client, NULL); if (ret < 0) { - DBG(KERN_ERR "failed to get dma channel\n"); + pr_debug(KERN_ERR "failed to get dma channel\n"); return ret; } } @@ -196,7 +189,7 @@ static int s3c24xx_pcm_hw_free(struct snd_pcm_substream *substream) { struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); /* TODO - do we need to ensure DMA flushed */ snd_pcm_set_runtime_buffer(substream, NULL); @@ -214,7 +207,7 @@ static int s3c24xx_pcm_prepare(struct snd_pcm_substream *substream) struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; int ret = 0; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); /* return if this is a bufferless transfer e.g. * codec <--> BT codec or GSM modem -- lg FIXME */ @@ -259,7 +252,7 @@ static int s3c24xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; int ret = 0; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); spin_lock(&prtd->lock); @@ -297,7 +290,7 @@ s3c24xx_pcm_pointer(struct snd_pcm_substream *substream) unsigned long res; dma_addr_t src, dst; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); spin_lock(&prtd->lock); s3c2410_dma_getposition(prtd->params->channel, &src, &dst); @@ -309,7 +302,7 @@ s3c24xx_pcm_pointer(struct snd_pcm_substream *substream) spin_unlock(&prtd->lock); - DBG("Pointer %x %x\n", src, dst); + pr_debug("Pointer %x %x\n", src, dst); /* we seem to be getting the odd error from the pcm library due * to out-of-bounds pointers. this is maybe due to the dma engine @@ -330,7 +323,7 @@ static int s3c24xx_pcm_open(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct s3c24xx_runtime_data *prtd; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); snd_soc_set_runtime_hwparams(substream, &s3c24xx_pcm_hardware); @@ -349,10 +342,10 @@ static int s3c24xx_pcm_close(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct s3c24xx_runtime_data *prtd = runtime->private_data; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); if (!prtd) - DBG("s3c24xx_pcm_close called with prtd == NULL\n"); + pr_debug("s3c24xx_pcm_close called with prtd == NULL\n"); kfree(prtd); @@ -364,7 +357,7 @@ static int s3c24xx_pcm_mmap(struct snd_pcm_substream *substream, { struct snd_pcm_runtime *runtime = substream->runtime; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); return dma_mmap_writecombine(substream->pcm->card->dev, vma, runtime->dma_area, @@ -390,7 +383,7 @@ static int s3c24xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) struct snd_dma_buffer *buf = &substream->dma_buffer; size_t size = s3c24xx_pcm_hardware.buffer_bytes_max; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); buf->dev.type = SNDRV_DMA_TYPE_DEV; buf->dev.dev = pcm->card->dev; @@ -409,7 +402,7 @@ static void s3c24xx_pcm_free_dma_buffers(struct snd_pcm *pcm) struct snd_dma_buffer *buf; int stream; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); for (stream = 0; stream < 2; stream++) { substream = pcm->streams[stream].substream; @@ -433,7 +426,7 @@ static int s3c24xx_pcm_new(struct snd_card *card, { int ret = 0; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); if (!card->dev->dma_mask) card->dev->dma_mask = &s3c24xx_pcm_dmamask; -- cgit v1.2.3-70-g09d2 From b52a5195efd6173c06107ca5beb44389130596dc Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 6 Mar 2009 18:13:43 +0000 Subject: ASoC: Fix logging severity for some S3C error messages Upgrade the severity of some failure messages from debug level so they're displayed by default. Signed-off-by: Mark Brown --- sound/soc/s3c24xx/s3c-i2s-v2.c | 2 +- sound/soc/s3c24xx/s3c24xx-i2s.c | 2 +- sound/soc/s3c24xx/s3c24xx-pcm.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index b7b8e47ae36..295a4c91026 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -553,7 +553,7 @@ int s3c_i2sv2_probe(struct platform_device *pdev, i2s->iis_pclk = clk_get(dev, "iis"); if (i2s->iis_pclk == NULL) { - pr_debug("failed to get iis_clock\n"); + dev_err(dev, "failed to get iis_clock\n"); iounmap(i2s->regs); return -ENOENT; } diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 2fbead33b7c..580cfed71cc 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -390,7 +390,7 @@ static int s3c24xx_i2s_probe(struct platform_device *pdev, s3c24xx_i2s.iis_clk = clk_get(&pdev->dev, "iis"); if (s3c24xx_i2s.iis_clk == NULL) { - pr_debug("failed to get iis_clock\n"); + pr_err("failed to get iis_clock\n"); iounmap(s3c24xx_i2s.regs); return -ENODEV; } diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c index 269f5c8e7ca..a9d68fa2b34 100644 --- a/sound/soc/s3c24xx/s3c24xx-pcm.c +++ b/sound/soc/s3c24xx/s3c24xx-pcm.c @@ -161,7 +161,7 @@ static int s3c24xx_pcm_hw_params(struct snd_pcm_substream *substream, prtd->params->client, NULL); if (ret < 0) { - pr_debug(KERN_ERR "failed to get dma channel\n"); + printk(KERN_ERR "failed to get dma channel\n"); return ret; } } -- cgit v1.2.3-70-g09d2 From 96deff6baf55da68b4b9b4dfe8ef572c6f1835fd Mon Sep 17 00:00:00 2001 From: Hugo Villeneuve Date: Fri, 6 Mar 2009 15:56:53 -0500 Subject: ASoC: Davinci: Fix incorrect machine type for SFFSDR board Signed-off-by: Hugo Villeneuve Signed-off-by: Mark Brown --- sound/soc/davinci/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index b502741692d..bd7392c9657 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -20,7 +20,7 @@ config SND_DAVINCI_SOC_EVM config SND_DAVINCI_SOC_SFFSDR tristate "SoC Audio support for SFFSDR" - depends on SND_DAVINCI_SOC && MACH_DAVINCI_SFFSDR + depends on SND_DAVINCI_SOC && MACH_SFFSDR select SND_DAVINCI_SOC_I2S select SND_SOC_PCM3008 select SFFSDR_FPGA -- cgit v1.2.3-70-g09d2 From 3a638ff272744247aad4a75b1fac174ac5746114 Mon Sep 17 00:00:00 2001 From: Timur Tabi Date: Fri, 6 Mar 2009 18:39:34 -0600 Subject: ASoC: Improve pause/unpause performance in Freescale 8610 drivers Add support for true pause and unpause. Without this, mplayer will drop some audio (less than one second, but still noticeable) when pausing playback. Remove support for PM suspend and resume from the trigger function, since the driver doesn't support PM anyway. Optimize the delay after starting capture. Instead of delaying 1ms, the driver now polls the hardware. The new delay is shorter by over 90% yet still effective. Signed-off-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_dma.c | 3 ++- sound/soc/fsl/fsl_ssi.c | 23 ++++++++++++++--------- 2 files changed, 16 insertions(+), 10 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 58a3fa49750..b3eb8570cd7 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -142,7 +142,8 @@ static const struct snd_pcm_hardware fsl_dma_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_JOINT_DUPLEX, + SNDRV_PCM_INFO_JOINT_DUPLEX | + SNDRV_PCM_INFO_PAUSE, .formats = FSLDMA_PCM_FORMATS, .rates = FSLDMA_PCM_RATES, .rate_min = 5512, diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 8cb6bcf2c00..b7733e6be19 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -464,28 +464,33 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, switch (cmd) { case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: + clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); setbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE); } else { - clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + long timeout = jiffies + 10; + setbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_RE); - /* - * I think we need this delay to allow time for the SSI - * to put data into its FIFO. Without it, ALSA starts - * to complain about overruns. + /* Wait until the SSI has filled its FIFO. Without this + * delay, ALSA complains about overruns. When the FIFO + * is full, the DMA controller initiates its first + * transfer. Until then, however, the DMA's DAR + * register is zero, which translates to an + * out-of-bounds pointer. This makes ALSA think an + * overrun has occurred. */ - mdelay(1); + while (!(in_be32(&ssi->sisr) & CCSR_SSI_SISR_RFF0) && + (jiffies < timeout)); + if (!(in_be32(&ssi->sisr) & CCSR_SSI_SISR_RFF0)) + return -EIO; } break; case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) clrbits32(&ssi->scr, CCSR_SSI_SCR_TE); -- cgit v1.2.3-70-g09d2 From b191f63c4fe9fbcfe583180228080d02b8dcdebc Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Sun, 8 Mar 2009 17:51:52 +0100 Subject: ASoC: bring cs4270 feature/limitations list in sync Removes numbers from the list of features/limitations and makes it reflect recent changes to the code. Signed-off-by: Daniel Mack Acked-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/codecs/cs4270.c | 15 +++++++-------- 1 file changed, 7 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index f86f33cc179..0e0c23ee6af 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -12,14 +12,13 @@ * * Current features/limitations: * - * 1) Software mode is supported. Stand-alone mode is not supported. - * 2) Only I2C is supported, not SPI - * 3) Only Master mode is supported, not Slave. - * 4) The machine driver's 'startup' function must call - * cs4270_set_dai_sysclk() with the value of MCLK. - * 5) Only I2S and left-justified modes are supported - * 6) Power management is not supported - * 7) The only supported control is volume and hardware mute (if enabled) + * - Software mode is supported. Stand-alone mode is not supported. + * - Only I2C is supported, not SPI + * - Support for master and slave mode + * - The machine driver's 'startup' function must call + * cs4270_set_dai_sysclk() with the value of MCLK. + * - Only I2S and left-justified modes are supported + * - Power management is not supported */ #include -- cgit v1.2.3-70-g09d2 From 055a49b0c92c6282e7db22e9e6ebcae6cb74ebb4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 8 Mar 2009 18:57:34 +0000 Subject: ASoC: Remove unneeded forward reference to WM8753 SPI implementation Signed-off-by: Mark Brown --- sound/soc/codecs/wm8753.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 7f353e935d7..1d5eca89de6 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -51,11 +51,6 @@ #include "wm8753.h" -#ifdef CONFIG_SPI_MASTER -static struct spi_driver wm8753_spi_driver; -static int wm8753_spi_write(struct spi_device *spi, const char *data, int len); -#endif - static int caps_charge = 2000; module_param(caps_charge, int, 0); MODULE_PARM_DESC(caps_charge, "WM8753 cap charge time (msecs)"); -- cgit v1.2.3-70-g09d2 From f271fa28fbaf947d9c79f188dd149176da727dd5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 9 Mar 2009 00:52:17 +0100 Subject: ASoC: Fix Kconfig dependency of CONFIG_SND_S3C24XX_SOC_JIVE_WM8750 Remove a non-existing Kconfig CONFIG_SND_SOC_WM8750_SPI. Signed-off-by: Takashi Iwai --- sound/soc/s3c24xx/Kconfig | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 78d01ff487c..2f3a21eee05 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -42,7 +42,6 @@ config SND_S3C24XX_SOC_JIVE_WM8750 tristate "SoC I2S Audio support for Jive" depends on SND_S3C24XX_SOC && MACH_JIVE select SND_SOC_WM8750 - select SND_SOC_WM8750_SPI select SND_S3C2412_SOC_I2S help Sat Y if you want to add support for SoC audio on the Jive. -- cgit v1.2.3-70-g09d2 From 873591db59e66434fd0a484c92f69fc21100b33d Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 9 Mar 2009 09:12:55 +0100 Subject: sound: oxygen: enable headphone output on Claro cards On the HT-Omega Claro (halo) sound cards, the headphone amplifier must be enabled explicitly by setting a GPIO bit. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/oxygen.c | 63 +++++++++++++++++++++++++++++++++++++++-------- 1 file changed, 53 insertions(+), 10 deletions(-) diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index 1d8e2b29745..72db4c39007 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -1,5 +1,5 @@ /* - * C-Media CMI8788 driver for C-Media's reference design and for the X-Meridian + * C-Media CMI8788 driver for C-Media's reference design and similar models * * Copyright (c) Clemens Ladisch * @@ -26,6 +26,7 @@ * * GPIO 0 -> DFS0 of AK5385 * GPIO 1 -> DFS1 of AK5385 + * GPIO 8 -> enable headphone amplifier on HT-Omega models */ #include @@ -61,7 +62,8 @@ MODULE_PARM_DESC(enable, "enable card"); enum { MODEL_CMEDIA_REF, /* C-Media's reference design */ MODEL_MERIDIAN, /* AuzenTech X-Meridian */ - MODEL_HALO, /* HT-Omega Claro halo */ + MODEL_CLARO, /* HT-Omega Claro */ + MODEL_CLARO_HALO, /* HT-Omega Claro halo */ }; static struct pci_device_id oxygen_ids[] __devinitdata = { @@ -74,8 +76,8 @@ static struct pci_device_id oxygen_ids[] __devinitdata = { { OXYGEN_PCI_SUBID(0x147a, 0xa017), .driver_data = MODEL_CMEDIA_REF }, { OXYGEN_PCI_SUBID(0x1a58, 0x0910), .driver_data = MODEL_CMEDIA_REF }, { OXYGEN_PCI_SUBID(0x415a, 0x5431), .driver_data = MODEL_MERIDIAN }, - { OXYGEN_PCI_SUBID(0x7284, 0x9761), .driver_data = MODEL_CMEDIA_REF }, - { OXYGEN_PCI_SUBID(0x7284, 0x9781), .driver_data = MODEL_HALO }, + { OXYGEN_PCI_SUBID(0x7284, 0x9761), .driver_data = MODEL_CLARO }, + { OXYGEN_PCI_SUBID(0x7284, 0x9781), .driver_data = MODEL_CLARO_HALO }, { } }; MODULE_DEVICE_TABLE(pci, oxygen_ids); @@ -86,6 +88,8 @@ MODULE_DEVICE_TABLE(pci, oxygen_ids); #define GPIO_AK5385_DFS_DOUBLE 0x0001 #define GPIO_AK5385_DFS_QUAD 0x0002 +#define GPIO_CLARO_HP 0x0100 + struct generic_data { u8 ak4396_ctl2; u16 saved_wm8785_registers[2]; @@ -196,16 +200,46 @@ static void meridian_init(struct oxygen *chip) ak5385_init(chip); } -static void halo_init(struct oxygen *chip) +static void claro_enable_hp(struct oxygen *chip) +{ + msleep(300); + oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_CLARO_HP); + oxygen_set_bits16(chip, OXYGEN_GPIO_DATA, GPIO_CLARO_HP); +} + +static void claro_init(struct oxygen *chip) +{ + ak4396_init(chip); + wm8785_init(chip); + claro_enable_hp(chip); +} + +static void claro_halo_init(struct oxygen *chip) { ak4396_init(chip); ak5385_init(chip); + claro_enable_hp(chip); } static void generic_cleanup(struct oxygen *chip) { } +static void claro_disable_hp(struct oxygen *chip) +{ + oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, GPIO_CLARO_HP); +} + +static void claro_cleanup(struct oxygen *chip) +{ + claro_disable_hp(chip); +} + +static void claro_suspend(struct oxygen *chip) +{ + claro_disable_hp(chip); +} + static void generic_resume(struct oxygen *chip) { ak4396_registers_init(chip); @@ -217,9 +251,10 @@ static void meridian_resume(struct oxygen *chip) ak4396_registers_init(chip); } -static void halo_resume(struct oxygen *chip) +static void claro_resume(struct oxygen *chip) { ak4396_registers_init(chip); + claro_enable_hp(chip); } static void set_ak4396_params(struct oxygen *chip, @@ -346,14 +381,22 @@ static int __devinit get_oxygen_model(struct oxygen *chip, CAPTURE_0_FROM_I2S_2 | CAPTURE_1_FROM_SPDIF; break; - case MODEL_HALO: - chip->model.init = halo_init; - chip->model.resume = halo_resume; + case MODEL_CLARO: + chip->model.init = claro_init; + chip->model.cleanup = claro_cleanup; + chip->model.suspend = claro_suspend; + chip->model.resume = claro_resume; + break; + case MODEL_CLARO_HALO: + chip->model.init = claro_halo_init; + chip->model.cleanup = claro_cleanup; + chip->model.suspend = claro_suspend; + chip->model.resume = claro_resume; chip->model.set_adc_params = set_ak5385_params; break; } if (id->driver_data == MODEL_MERIDIAN || - id->driver_data == MODEL_HALO) { + id->driver_data == MODEL_CLARO_HALO) { chip->model.misc_flags = OXYGEN_MISC_MIDI; chip->model.device_config |= MIDI_OUTPUT | MIDI_INPUT; } -- cgit v1.2.3-70-g09d2 From a381934e5f9c0c3c292d780d61f5be9c22b2ef54 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 9 Mar 2009 02:13:17 +0100 Subject: ASoC: Add a driver for AK4104 S/PDIF transmitter This adds a driver for the SPI connected AK4104 S/PDIF transmitter device. Its features are fairly simple, but as there is need to set up certain bits in the IEC958 information, this better goes into a real driver. Signed-off-by: Daniel Mack Cc: Mark Brown Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak4104.c | 363 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ak4104.h | 7 + 4 files changed, 376 insertions(+) create mode 100644 sound/soc/codecs/ak4104.c create mode 100644 sound/soc/codecs/ak4104.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 628a591c728..a1af311e7f0 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -14,6 +14,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS select SND_SOC_AD1980 if SND_SOC_AC97_BUS select SND_SOC_AD73311 if I2C + select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_PCM3008 @@ -60,6 +61,9 @@ config SND_SOC_AD1980 config SND_SOC_AD73311 tristate +config SND_SOC_AK4104 + tristate + config SND_SOC_AK4535 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 3664cdc300b..4717c3c9904 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -1,6 +1,7 @@ snd-soc-ac97-objs := ac97.o snd-soc-ad1980-objs := ad1980.o snd-soc-ad73311-objs := ad73311.o +snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o snd-soc-cs4270-objs := cs4270.o snd-soc-l3-objs := l3.o @@ -30,6 +31,7 @@ snd-soc-wm9713-objs := wm9713.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o +obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c new file mode 100644 index 00000000000..338381f4fe1 --- /dev/null +++ b/sound/soc/codecs/ak4104.c @@ -0,0 +1,363 @@ +/* + * AK4104 ALSA SoC (ASoC) driver + * + * Copyright (c) 2009 Daniel Mack + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include +#include +#include +#include +#include +#include + +#include "ak4104.h" + +/* AK4104 registers addresses */ +#define AK4104_REG_CONTROL1 0x00 +#define AK4104_REG_RESERVED 0x01 +#define AK4104_REG_CONTROL2 0x02 +#define AK4104_REG_TX 0x03 +#define AK4104_REG_CHN_STATUS(x) ((x) + 0x04) +#define AK4104_NUM_REGS 10 + +#define AK4104_REG_MASK 0x1f +#define AK4104_READ 0xc0 +#define AK4104_WRITE 0xe0 +#define AK4104_RESERVED_VAL 0x5b + +/* Bit masks for AK4104 registers */ +#define AK4104_CONTROL1_RSTN (1 << 0) +#define AK4104_CONTROL1_PW (1 << 1) +#define AK4104_CONTROL1_DIF0 (1 << 2) +#define AK4104_CONTROL1_DIF1 (1 << 3) + +#define AK4104_CONTROL2_SEL0 (1 << 0) +#define AK4104_CONTROL2_SEL1 (1 << 1) +#define AK4104_CONTROL2_MODE (1 << 2) + +#define AK4104_TX_TXE (1 << 0) +#define AK4104_TX_V (1 << 1) + +#define DRV_NAME "ak4104" + +struct ak4104_private { + struct snd_soc_codec codec; + u8 reg_cache[AK4104_NUM_REGS]; +}; + +static int ak4104_fill_cache(struct snd_soc_codec *codec) +{ + int i; + u8 *reg_cache = codec->reg_cache; + struct spi_device *spi = codec->control_data; + + for (i = 0; i < codec->reg_cache_size; i++) { + int ret = spi_w8r8(spi, i | AK4104_READ); + if (ret < 0) { + dev_err(&spi->dev, "SPI write failure\n"); + return ret; + } + + reg_cache[i] = ret; + } + + return 0; +} + +static unsigned int ak4104_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 *reg_cache = codec->reg_cache; + + if (reg >= codec->reg_cache_size) + return -EINVAL; + + return reg_cache[reg]; +} + +static int ak4104_spi_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 *cache = codec->reg_cache; + struct spi_device *spi = codec->control_data; + + if (reg >= codec->reg_cache_size) + return -EINVAL; + + reg &= AK4104_REG_MASK; + reg |= AK4104_WRITE; + + /* only write to the hardware if value has changed */ + if (cache[reg] != value) { + u8 tmp[2] = { reg, value }; + if (spi_write(spi, tmp, sizeof(tmp))) { + dev_err(&spi->dev, "SPI write failed\n"); + return -EIO; + } + + cache[reg] = value; + } + + return 0; +} + +static int ak4104_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + int val = 0; + + val = ak4104_read_reg_cache(codec, AK4104_REG_CONTROL1); + if (val < 0) + return val; + + val &= ~(AK4104_CONTROL1_DIF0 | AK4104_CONTROL1_DIF1); + + /* set DAI format */ + switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + val |= AK4104_CONTROL1_DIF0; + break; + case SND_SOC_DAIFMT_I2S: + val |= AK4104_CONTROL1_DIF0 | AK4104_CONTROL1_DIF1; + break; + default: + dev_err(codec->dev, "invalid dai format\n"); + return -EINVAL; + } + + /* This device can only be slave */ + if ((format & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) + return -EINVAL; + + return ak4104_spi_write(codec, AK4104_REG_CONTROL1, val); +} + +static int ak4104_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + int val = 0; + + /* set the IEC958 bits: consumer mode, no copyright bit */ + val |= IEC958_AES0_CON_NOT_COPYRIGHT; + ak4104_spi_write(codec, AK4104_REG_CHN_STATUS(0), val); + + val = 0; + + switch (params_rate(params)) { + case 44100: + val |= IEC958_AES3_CON_FS_44100; + break; + case 48000: + val |= IEC958_AES3_CON_FS_48000; + break; + case 32000: + val |= IEC958_AES3_CON_FS_32000; + break; + default: + dev_err(codec->dev, "unsupported sampling rate\n"); + return -EINVAL; + } + + return ak4104_spi_write(codec, AK4104_REG_CHN_STATUS(3), val); +} + +struct snd_soc_dai ak4104_dai = { + .name = DRV_NAME, + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_32000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_3LE | + SNDRV_PCM_FMTBIT_S24_LE + }, + .ops = { + .hw_params = ak4104_hw_params, + .set_fmt = ak4104_set_dai_fmt, + } +}; + +static struct snd_soc_codec *ak4104_codec; + +static int ak4104_spi_probe(struct spi_device *spi) +{ + struct snd_soc_codec *codec; + struct ak4104_private *ak4104; + int ret, val; + + spi->bits_per_word = 8; + spi->mode = SPI_MODE_0; + ret = spi_setup(spi); + if (ret < 0) + return ret; + + ak4104 = kzalloc(sizeof(struct ak4104_private), GFP_KERNEL); + if (!ak4104) { + dev_err(&spi->dev, "could not allocate codec\n"); + return -ENOMEM; + } + + codec = &ak4104->codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->dev = &spi->dev; + codec->name = DRV_NAME; + codec->owner = THIS_MODULE; + codec->dai = &ak4104_dai; + codec->num_dai = 1; + codec->private_data = ak4104; + codec->control_data = spi; + codec->reg_cache = ak4104->reg_cache; + codec->reg_cache_size = AK4104_NUM_REGS; + + /* read all regs and fill the cache */ + ret = ak4104_fill_cache(codec); + if (ret < 0) { + dev_err(&spi->dev, "failed to fill register cache\n"); + return ret; + } + + /* read the 'reserved' register - according to the datasheet, it + * should contain 0x5b. Not a good way to verify the presence of + * the device, but there is no hardware ID register. */ + if (ak4104_read_reg_cache(codec, AK4104_REG_RESERVED) != + AK4104_RESERVED_VAL) { + ret = -ENODEV; + goto error_free_codec; + } + + /* set power-up and non-reset bits */ + val = ak4104_read_reg_cache(codec, AK4104_REG_CONTROL1); + val |= AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN; + ret = ak4104_spi_write(codec, AK4104_REG_CONTROL1, val); + if (ret < 0) + goto error_free_codec; + + /* enable transmitter */ + val = ak4104_read_reg_cache(codec, AK4104_REG_TX); + val |= AK4104_TX_TXE; + ret = ak4104_spi_write(codec, AK4104_REG_TX, val); + if (ret < 0) + goto error_free_codec; + + ak4104_codec = codec; + ret = snd_soc_register_dai(&ak4104_dai); + if (ret < 0) { + dev_err(&spi->dev, "failed to register DAI\n"); + goto error_free_codec; + } + + spi_set_drvdata(spi, ak4104); + dev_info(&spi->dev, "SPI device initialized\n"); + return 0; + +error_free_codec: + kfree(ak4104); + ak4104_dai.dev = NULL; + return ret; +} + +static int __devexit ak4104_spi_remove(struct spi_device *spi) +{ + int ret, val; + struct ak4104_private *ak4104 = spi_get_drvdata(spi); + + val = ak4104_read_reg_cache(&ak4104->codec, AK4104_REG_CONTROL1); + if (val < 0) + return val; + + /* clear power-up and non-reset bits */ + val &= ~(AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN); + ret = ak4104_spi_write(&ak4104->codec, AK4104_REG_CONTROL1, val); + if (ret < 0) + return ret; + + ak4104_codec = NULL; + kfree(ak4104); + return 0; +} + +static int ak4104_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = ak4104_codec; + int ret; + + /* Connect the codec to the socdev. snd_soc_new_pcms() needs this. */ + socdev->card->codec = codec; + + /* Register PCMs */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms\n"); + return ret; + } + + /* Register the socdev */ + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(codec->dev, "failed to register card\n"); + snd_soc_free_pcms(socdev); + return ret; + } + + return 0; +} + +static int ak4104_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + snd_soc_free_pcms(socdev); + return 0; +}; + +struct snd_soc_codec_device soc_codec_device_ak4104 = { + .probe = ak4104_probe, + .remove = ak4104_remove +}; +EXPORT_SYMBOL_GPL(soc_codec_device_ak4104); + +static struct spi_driver ak4104_spi_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + }, + .probe = ak4104_spi_probe, + .remove = __devexit_p(ak4104_spi_remove), +}; + +static int __init ak4104_init(void) +{ + pr_info("Asahi Kasei AK4104 ALSA SoC Codec Driver\n"); + return spi_register_driver(&ak4104_spi_driver); +} +module_init(ak4104_init); + +static void __exit ak4104_exit(void) +{ + spi_unregister_driver(&ak4104_spi_driver); +} +module_exit(ak4104_exit); + +MODULE_AUTHOR("Daniel Mack "); +MODULE_DESCRIPTION("Asahi Kasei AK4104 ALSA SoC driver"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/codecs/ak4104.h b/sound/soc/codecs/ak4104.h new file mode 100644 index 00000000000..eb88fe7e4de --- /dev/null +++ b/sound/soc/codecs/ak4104.h @@ -0,0 +1,7 @@ +#ifndef _AK4104_H +#define _AK4104_H + +extern struct snd_soc_dai ak4104_dai; +extern struct snd_soc_codec_device soc_codec_device_ak4104; + +#endif -- cgit v1.2.3-70-g09d2 From ed3da3d9a0ef13c6fe1414ec73c9c1be12747b62 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 3 Mar 2009 17:00:15 +0100 Subject: ALSA: Rewrite hw_ptr updaters Clean up and improve snd_pcm_update_hw_ptr*() functions. snd_pcm_update_hw_ptr() tries to detect the unexpected hwptr jumps more strictly to avoid the position mess-up, which often results in the bad quality I/O with pulseaudio. The hw-ptr skip error messages are printed when xrun proc is set to non-zero. Signed-off-by: Takashi Iwai --- sound/core/pcm_lib.c | 128 +++++++++++++++++++++++++++++++++------------------ 1 file changed, 83 insertions(+), 45 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 921691080f3..86ac9ae9460 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -125,19 +125,27 @@ void snd_pcm_playback_silence(struct snd_pcm_substream *substream, snd_pcm_ufram } } +#ifdef CONFIG_SND_PCM_XRUN_DEBUG +#define xrun_debug(substream) ((substream)->pstr->xrun_debug) +#else +#define xrun_debug(substream) 0 +#endif + +#define dump_stack_on_xrun(substream) do { \ + if (xrun_debug(substream) > 1) \ + dump_stack(); \ + } while (0) + static void xrun(struct snd_pcm_substream *substream) { snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); -#ifdef CONFIG_SND_PCM_XRUN_DEBUG - if (substream->pstr->xrun_debug) { + if (xrun_debug(substream)) { snd_printd(KERN_DEBUG "XRUN: pcmC%dD%d%c\n", substream->pcm->card->number, substream->pcm->device, substream->stream ? 'c' : 'p'); - if (substream->pstr->xrun_debug > 1) - dump_stack(); + dump_stack_on_xrun(substream); } -#endif } static inline snd_pcm_uframes_t snd_pcm_update_hw_ptr_pos(struct snd_pcm_substream *substream, @@ -182,11 +190,21 @@ static inline int snd_pcm_update_hw_ptr_post(struct snd_pcm_substream *substream return 0; } +#define hw_ptr_error(substream, fmt, args...) \ + do { \ + if (xrun_debug(substream)) { \ + if (printk_ratelimit()) { \ + snd_printd("hda_codec: " fmt, ##args); \ + } \ + dump_stack_on_xrun(substream); \ + } \ + } while (0) + static inline int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_uframes_t pos; - snd_pcm_uframes_t new_hw_ptr, hw_ptr_interrupt; + snd_pcm_uframes_t new_hw_ptr, hw_ptr_interrupt, hw_base; snd_pcm_sframes_t delta; pos = snd_pcm_update_hw_ptr_pos(substream, runtime); @@ -194,36 +212,47 @@ static inline int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *subs xrun(substream); return -EPIPE; } - if (runtime->period_size == runtime->buffer_size) - goto __next_buf; - new_hw_ptr = runtime->hw_ptr_base + pos; + hw_base = runtime->hw_ptr_base; + new_hw_ptr = hw_base + pos; hw_ptr_interrupt = runtime->hw_ptr_interrupt + runtime->period_size; - - delta = hw_ptr_interrupt - new_hw_ptr; - if (delta > 0) { - if ((snd_pcm_uframes_t)delta < runtime->buffer_size / 2) { -#ifdef CONFIG_SND_PCM_XRUN_DEBUG - if (runtime->periods > 1 && substream->pstr->xrun_debug) { - snd_printd(KERN_ERR "Unexpected hw_pointer value [1] (stream = %i, delta: -%ld, max jitter = %ld): wrong interrupt acknowledge?\n", substream->stream, (long) delta, runtime->buffer_size / 2); - if (substream->pstr->xrun_debug > 1) - dump_stack(); - } -#endif - return 0; + delta = new_hw_ptr - hw_ptr_interrupt; + if (hw_ptr_interrupt == runtime->boundary) + hw_ptr_interrupt = 0; + if (delta < 0) { + delta += runtime->buffer_size; + if (delta < 0) { + hw_ptr_error(substream, + "Unexpected hw_pointer value " + "(stream=%i, pos=%ld, intr_ptr=%ld)\n", + substream->stream, (long)pos, + (long)hw_ptr_interrupt); + /* rebase to interrupt position */ + hw_base = new_hw_ptr = hw_ptr_interrupt; + delta = 0; + } else { + hw_base += runtime->buffer_size; + if (hw_base == runtime->boundary) + hw_base = 0; + new_hw_ptr = hw_base + pos; } - __next_buf: - runtime->hw_ptr_base += runtime->buffer_size; - if (runtime->hw_ptr_base == runtime->boundary) - runtime->hw_ptr_base = 0; - new_hw_ptr = runtime->hw_ptr_base + pos; } - + if (delta > runtime->period_size) { + hw_ptr_error(substream, + "Lost interrupts? " + "(stream=%i, delta=%ld, intr_ptr=%ld)\n", + substream->stream, (long)delta, + (long)hw_ptr_interrupt); + /* rebase hw_ptr_interrupt */ + hw_ptr_interrupt = + new_hw_ptr - new_hw_ptr % runtime->period_size; + } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) snd_pcm_playback_silence(substream, new_hw_ptr); + runtime->hw_ptr_base = hw_base; runtime->status->hw_ptr = new_hw_ptr; - runtime->hw_ptr_interrupt = new_hw_ptr - new_hw_ptr % runtime->period_size; + runtime->hw_ptr_interrupt = hw_ptr_interrupt; return snd_pcm_update_hw_ptr_post(substream, runtime); } @@ -233,7 +262,7 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_uframes_t pos; - snd_pcm_uframes_t old_hw_ptr, new_hw_ptr; + snd_pcm_uframes_t old_hw_ptr, new_hw_ptr, hw_base; snd_pcm_sframes_t delta; old_hw_ptr = runtime->status->hw_ptr; @@ -242,29 +271,38 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream) xrun(substream); return -EPIPE; } - new_hw_ptr = runtime->hw_ptr_base + pos; - - delta = old_hw_ptr - new_hw_ptr; - if (delta > 0) { - if ((snd_pcm_uframes_t)delta < runtime->buffer_size / 2) { -#ifdef CONFIG_SND_PCM_XRUN_DEBUG - if (runtime->periods > 2 && substream->pstr->xrun_debug) { - snd_printd(KERN_ERR "Unexpected hw_pointer value [2] (stream = %i, delta: -%ld, max jitter = %ld): wrong interrupt acknowledge?\n", substream->stream, (long) delta, runtime->buffer_size / 2); - if (substream->pstr->xrun_debug > 1) - dump_stack(); - } -#endif + hw_base = runtime->hw_ptr_base; + new_hw_ptr = hw_base + pos; + + delta = new_hw_ptr - old_hw_ptr; + if (delta < 0) { + delta += runtime->buffer_size; + if (delta < 0) { + hw_ptr_error(substream, + "Unexpected hw_pointer value [2] " + "(stream=%i, pos=%ld, old_ptr=%ld)\n", + substream->stream, (long)pos, + (long)old_hw_ptr); return 0; } - runtime->hw_ptr_base += runtime->buffer_size; - if (runtime->hw_ptr_base == runtime->boundary) - runtime->hw_ptr_base = 0; - new_hw_ptr = runtime->hw_ptr_base + pos; + hw_base += runtime->buffer_size; + if (hw_base == runtime->boundary) + hw_base = 0; + new_hw_ptr = hw_base + pos; + } + if (delta > runtime->period_size && runtime->periods > 1) { + hw_ptr_error(substream, + "hw_ptr skipping! " + "(pos=%ld, delta=%ld, period=%ld)\n", + (long)pos, (long)delta, + (long)runtime->period_size); + return 0; } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) snd_pcm_playback_silence(substream, new_hw_ptr); + runtime->hw_ptr_base = hw_base; runtime->status->hw_ptr = new_hw_ptr; return snd_pcm_update_hw_ptr_post(substream, runtime); -- cgit v1.2.3-70-g09d2 From 85122ea40c4fc82af5b66b8683f525c2c4a36d1a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 6 Mar 2009 16:30:07 +0100 Subject: ALSA: Remove unneeded snd_pcm_substream.timer_lock The timer callbacks are called in the protected status by the lock of the timer instance, so there is no need for an extra lock in the PCM substream. Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 1 - sound/core/pcm.c | 1 - sound/core/pcm_timer.c | 6 ------ 3 files changed, 8 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 40c5a6fa6bc..e4f60076e6c 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -364,7 +364,6 @@ struct snd_pcm_substream { /* -- timer section -- */ struct snd_timer *timer; /* timer */ unsigned timer_running: 1; /* time is running */ - spinlock_t timer_lock; /* -- next substream -- */ struct snd_pcm_substream *next; /* -- linked substreams -- */ diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 192a433a240..37f567a68ef 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -667,7 +667,6 @@ int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count) spin_lock_init(&substream->self_group.lock); INIT_LIST_HEAD(&substream->self_group.substreams); list_add_tail(&substream->link_list, &substream->self_group.substreams); - spin_lock_init(&substream->timer_lock); atomic_set(&substream->mmap_count, 0); prev = substream; } diff --git a/sound/core/pcm_timer.c b/sound/core/pcm_timer.c index 2c89c04f291..ca8068b63d6 100644 --- a/sound/core/pcm_timer.c +++ b/sound/core/pcm_timer.c @@ -85,25 +85,19 @@ static unsigned long snd_pcm_timer_resolution(struct snd_timer * timer) static int snd_pcm_timer_start(struct snd_timer * timer) { - unsigned long flags; struct snd_pcm_substream *substream; substream = snd_timer_chip(timer); - spin_lock_irqsave(&substream->timer_lock, flags); substream->timer_running = 1; - spin_unlock_irqrestore(&substream->timer_lock, flags); return 0; } static int snd_pcm_timer_stop(struct snd_timer * timer) { - unsigned long flags; struct snd_pcm_substream *substream; substream = snd_timer_chip(timer); - spin_lock_irqsave(&substream->timer_lock, flags); substream->timer_running = 0; - spin_unlock_irqrestore(&substream->timer_lock, flags); return 0; } -- cgit v1.2.3-70-g09d2 From f5b1db634280ecaf3147ee996f26aad0ed4828c4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 16 Jan 2009 18:15:22 +0100 Subject: ALSA: add snd_ctl_add_slave_uncached() Added snd_ctl_add_slave_uncached() function to add a slave element with volatile controls. The values of normal slave elements are supposed to be cachable, i.e. they are changed only via the put callbacks. OTOH, when a slave element is volatile and its values may be changed by other reason (e.g. hardware status change), the values will get inconsistent. The new function allows the slave elements with volatile changes. When the slave is tied with this call, the native get callback is issued at each time so that the values are always updated. Signed-off-by: Takashi Iwai --- include/sound/control.h | 20 ++++++++++++++++++-- sound/core/vmaster.c | 46 +++++++++++++++++++++++++++++----------------- 2 files changed, 47 insertions(+), 19 deletions(-) diff --git a/include/sound/control.h b/include/sound/control.h index 4721b4bba05..4cf8f7aaa13 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -171,6 +171,22 @@ int snd_ctl_boolean_stereo_info(struct snd_kcontrol *kcontrol, */ struct snd_kcontrol *snd_ctl_make_virtual_master(char *name, const unsigned int *tlv); -int snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave); - +int _snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave, + unsigned int flags); +/* optional flags for slave */ +#define SND_CTL_SLAVE_NEED_UPDATE (1 << 0) + +static inline int +snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave) +{ + return _snd_ctl_add_slave(master, slave, 0); +} + +static inline int +snd_ctl_add_slave_uncached(struct snd_kcontrol *master, + struct snd_kcontrol *slave) +{ + return _snd_ctl_add_slave(master, slave, SND_CTL_SLAVE_NEED_UPDATE); +} + #endif /* __SOUND_CONTROL_H */ diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c index 4cc57f902e2..d51b198d06d 100644 --- a/sound/core/vmaster.c +++ b/sound/core/vmaster.c @@ -50,18 +50,38 @@ struct link_slave { struct link_master *master; struct link_ctl_info info; int vals[2]; /* current values */ + unsigned int flags; struct snd_kcontrol slave; /* the copy of original control entry */ }; +static int slave_update(struct link_slave *slave) +{ + struct snd_ctl_elem_value *uctl; + int err, ch; + + uctl = kmalloc(sizeof(*uctl), GFP_KERNEL); + if (!uctl) + return -ENOMEM; + uctl->id = slave->slave.id; + err = slave->slave.get(&slave->slave, uctl); + for (ch = 0; ch < slave->info.count; ch++) + slave->vals[ch] = uctl->value.integer.value[ch]; + kfree(uctl); + return 0; +} + /* get the slave ctl info and save the initial values */ static int slave_init(struct link_slave *slave) { struct snd_ctl_elem_info *uinfo; - struct snd_ctl_elem_value *uctl; - int err, ch; + int err; - if (slave->info.count) - return 0; /* already initialized */ + if (slave->info.count) { + /* already initialized */ + if (slave->flags & SND_CTL_SLAVE_NEED_UPDATE) + return slave_update(slave); + return 0; + } uinfo = kmalloc(sizeof(*uinfo), GFP_KERNEL); if (!uinfo) @@ -85,15 +105,7 @@ static int slave_init(struct link_slave *slave) slave->info.max_val = uinfo->value.integer.max; kfree(uinfo); - uctl = kmalloc(sizeof(*uctl), GFP_KERNEL); - if (!uctl) - return -ENOMEM; - uctl->id = slave->slave.id; - err = slave->slave.get(&slave->slave, uctl); - for (ch = 0; ch < slave->info.count; ch++) - slave->vals[ch] = uctl->value.integer.value[ch]; - kfree(uctl); - return 0; + return slave_update(slave); } /* initialize master volume */ @@ -229,7 +241,8 @@ static void slave_free(struct snd_kcontrol *kcontrol) * - logarithmic volume control (dB level), no linear volume * - master can only attenuate the volume, no gain */ -int snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave) +int _snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave, + unsigned int flags) { struct link_master *master_link = snd_kcontrol_chip(master); struct link_slave *srec; @@ -241,6 +254,7 @@ int snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave) srec->slave = *slave; memcpy(srec->slave.vd, slave->vd, slave->count * sizeof(*slave->vd)); srec->master = master_link; + srec->flags = flags; /* override callbacks */ slave->info = slave_info; @@ -254,8 +268,7 @@ int snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave) list_add_tail(&srec->list, &master_link->slaves); return 0; } - -EXPORT_SYMBOL(snd_ctl_add_slave); +EXPORT_SYMBOL(_snd_ctl_add_slave); /* * ctl callbacks for master controls @@ -367,5 +380,4 @@ struct snd_kcontrol *snd_ctl_make_virtual_master(char *name, return kctl; } - EXPORT_SYMBOL(snd_ctl_make_virtual_master); -- cgit v1.2.3-70-g09d2 From b0a8a8fd1b3bd6fbbb4b599191b859d41e12a002 Mon Sep 17 00:00:00 2001 From: Risto Suominen Date: Tue, 20 Jan 2009 22:01:13 +0200 Subject: ALSA: powermac - Correct HP detection and input selectors for PMac 5500 Correct headphone detection and input selectors for PowerMac 5500 (AWACS). Signed-off-by: Risto Suominen Signed-off-by: Takashi Iwai --- sound/ppc/awacs.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/ppc/awacs.c b/sound/ppc/awacs.c index 7bd33e6552a..0258ccb8f43 100644 --- a/sound/ppc/awacs.c +++ b/sound/ppc/awacs.c @@ -767,6 +767,7 @@ static void snd_pmac_awacs_resume(struct snd_pmac *chip) #endif /* CONFIG_PM */ #define IS_PM7500 (machine_is_compatible("AAPL,7500")) +#define IS_PM5500 (machine_is_compatible("AAPL,e411")) #define IS_BEIGE (machine_is_compatible("AAPL,Gossamer")) #define IS_IMAC1 (machine_is_compatible("PowerMac2,1")) #define IS_IMAC2 (machine_is_compatible("PowerMac2,2") \ @@ -858,6 +859,7 @@ int __init snd_pmac_awacs_init(struct snd_pmac *chip) { int pm7500 = IS_PM7500; + int pm5500 = IS_PM5500; int beige = IS_BEIGE; int g4agp = IS_G4AGP; int imac; @@ -915,7 +917,7 @@ snd_pmac_awacs_init(struct snd_pmac *chip) /* set headphone-jack detection bit */ switch (chip->model) { case PMAC_AWACS: - chip->hp_stat_mask = pm7500 ? MASK_HDPCONN + chip->hp_stat_mask = pm7500 || pm5500 ? MASK_HDPCONN : MASK_LOCONN; break; case PMAC_SCREAMER: @@ -954,7 +956,7 @@ snd_pmac_awacs_init(struct snd_pmac *chip) return err; if (beige || g4agp) ; - else if (chip->model == PMAC_SCREAMER) + else if (chip->model == PMAC_SCREAMER || pm5500) err = build_mixers(chip, ARRAY_SIZE(snd_pmac_screamer_mixers2), snd_pmac_screamer_mixers2); else if (!pm7500) -- cgit v1.2.3-70-g09d2 From 573934bc038b0f47d17a5608e74b79dcd7c191ea Mon Sep 17 00:00:00 2001 From: Risto Suominen Date: Tue, 20 Jan 2009 22:01:14 +0200 Subject: ALSA: powermac - Correct volume controls for PowerBook G3 Lombard Correct volume controls for PowerBook G3 Lombard (Screamer). Signed-off-by: Risto Suominen Signed-off-by: Takashi Iwai --- sound/ppc/awacs.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/ppc/awacs.c b/sound/ppc/awacs.c index 0258ccb8f43..d89c23e135d 100644 --- a/sound/ppc/awacs.c +++ b/sound/ppc/awacs.c @@ -773,6 +773,7 @@ static void snd_pmac_awacs_resume(struct snd_pmac *chip) #define IS_IMAC2 (machine_is_compatible("PowerMac2,2") \ || machine_is_compatible("PowerMac4,1")) #define IS_G4AGP (machine_is_compatible("PowerMac3,1")) +#define IS_LOMBARD (machine_is_compatible("PowerBook1,1")) static int imac1, imac2; @@ -862,6 +863,7 @@ snd_pmac_awacs_init(struct snd_pmac *chip) int pm5500 = IS_PM5500; int beige = IS_BEIGE; int g4agp = IS_G4AGP; + int lombard = IS_LOMBARD; int imac; int err, vol; @@ -972,7 +974,7 @@ snd_pmac_awacs_init(struct snd_pmac *chip) err = build_mixers(chip, ARRAY_SIZE(snd_pmac_screamer_mixers_beige), snd_pmac_screamer_mixers_beige); - else if (imac) + else if (imac || lombard) err = build_mixers(chip, ARRAY_SIZE(snd_pmac_screamer_mixers_imac), snd_pmac_screamer_mixers_imac); @@ -986,7 +988,7 @@ snd_pmac_awacs_init(struct snd_pmac *chip) snd_pmac_awacs_mixers_pmac); if (err < 0) return err; - chip->master_sw_ctl = snd_ctl_new1((pm7500 || imac || g4agp) + chip->master_sw_ctl = snd_ctl_new1((pm7500 || imac || g4agp || lombard) ? &snd_pmac_awacs_master_sw_imac : &snd_pmac_awacs_master_sw, chip); err = snd_ctl_add(chip->card, chip->master_sw_ctl); -- cgit v1.2.3-70-g09d2 From 4d9e93b1adf2923c0a0cbc545d6e78dec3334faf Mon Sep 17 00:00:00 2001 From: Risto Suominen Date: Tue, 20 Jan 2009 22:01:15 +0200 Subject: ALSA: powermac - Correct volume controls and HP detection for PMac 8500/9500 Correct volume controls and headphone detection for PowerMac 8500/9500 (AWACS). Signed-off-by: Risto Suominen Signed-off-by: Takashi Iwai --- sound/ppc/awacs.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/ppc/awacs.c b/sound/ppc/awacs.c index d89c23e135d..9abbf645eb6 100644 --- a/sound/ppc/awacs.c +++ b/sound/ppc/awacs.c @@ -766,7 +766,9 @@ static void snd_pmac_awacs_resume(struct snd_pmac *chip) } #endif /* CONFIG_PM */ -#define IS_PM7500 (machine_is_compatible("AAPL,7500")) +#define IS_PM7500 (machine_is_compatible("AAPL,7500") \ + || machine_is_compatible("AAPL,8500") \ + || machine_is_compatible("AAPL,9500")) #define IS_PM5500 (machine_is_compatible("AAPL,e411")) #define IS_BEIGE (machine_is_compatible("AAPL,Gossamer")) #define IS_IMAC1 (machine_is_compatible("PowerMac2,1")) -- cgit v1.2.3-70-g09d2 From ed336d3404a8fdeda1e3f1c189b5f83186675448 Mon Sep 17 00:00:00 2001 From: Risto Suominen Date: Tue, 20 Jan 2009 22:01:16 +0200 Subject: ALSA: powermac - Allow input from mic in iBook G3 Dual-USB Allow input from microphone on iBook G3 Dual-USB (Tumbler). Signed-off-by: Risto Suominen Signed-off-by: Takashi Iwai --- sound/ppc/pmac.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/ppc/pmac.c b/sound/ppc/pmac.c index af76ee862d2..bd8f92b1c22 100644 --- a/sound/ppc/pmac.c +++ b/sound/ppc/pmac.c @@ -1033,7 +1033,8 @@ static int __init snd_pmac_detect(struct snd_pmac *chip) } if (of_device_is_compatible(sound, "tumbler")) { chip->model = PMAC_TUMBLER; - chip->can_capture = machine_is_compatible("PowerMac4,2"); + chip->can_capture = machine_is_compatible("PowerMac4,2") + || machine_is_compatible("PowerBook4,1"); chip->can_duplex = 0; // chip->can_byte_swap = 0; /* FIXME: check this */ chip->num_freqs = ARRAY_SIZE(tumbler_freqs); -- cgit v1.2.3-70-g09d2 From dca7c74172fee0cf6ee1e303df093c31b5561039 Mon Sep 17 00:00:00 2001 From: Risto Suominen Date: Tue, 20 Jan 2009 22:01:17 +0200 Subject: ALSA: Add vmaster controls for Pmac 5500, iMac G3 SL, and PBook G3 Lombard Add virtual master controls for PowerMac 5500 (AWACS) and iMac G3 Slot-loading and PowerBook G3 Lombard (Screamer). Signed-off-by: Risto Suominen Signed-off-by: Takashi Iwai --- sound/ppc/Kconfig | 1 + sound/ppc/awacs.c | 74 ++++++++++++++++++++++++++++++++++++++++++++++++++----- 2 files changed, 69 insertions(+), 6 deletions(-) diff --git a/sound/ppc/Kconfig b/sound/ppc/Kconfig index 777de2b1717..bd2338ab2ce 100644 --- a/sound/ppc/Kconfig +++ b/sound/ppc/Kconfig @@ -13,6 +13,7 @@ config SND_POWERMAC tristate "PowerMac (AWACS, DACA, Burgundy, Tumbler, Keywest)" depends on I2C && INPUT && PPC_PMAC select SND_PCM + select SND_VMASTER help Say Y here to include support for the integrated sound device. diff --git a/sound/ppc/awacs.c b/sound/ppc/awacs.c index 9abbf645eb6..80df9b1f651 100644 --- a/sound/ppc/awacs.c +++ b/sound/ppc/awacs.c @@ -608,9 +608,12 @@ static struct snd_kcontrol_new snd_pmac_screamer_mixers_beige[] __initdata = { AWACS_SWITCH("CD Capture Switch", 0, SHIFT_MUX_LINE, 0), }; -static struct snd_kcontrol_new snd_pmac_screamer_mixers_imac[] __initdata = { +static struct snd_kcontrol_new snd_pmac_screamer_mixers_lo[] __initdata = { AWACS_VOLUME("Line out Playback Volume", 2, 6, 1), - AWACS_VOLUME("Master Playback Volume", 5, 6, 1), +}; + +static struct snd_kcontrol_new snd_pmac_screamer_mixers_imac[] __initdata = { + AWACS_VOLUME("Play-through Playback Volume", 5, 6, 1), AWACS_SWITCH("CD Capture Switch", 0, SHIFT_MUX_CD, 0), }; @@ -627,6 +630,10 @@ static struct snd_kcontrol_new snd_pmac_awacs_mixers_pmac7500[] __initdata = { AWACS_SWITCH("Line Capture Switch", 0, SHIFT_MUX_MIC, 0), }; +static struct snd_kcontrol_new snd_pmac_awacs_mixers_pmac5500[] __initdata = { + AWACS_VOLUME("Headphone Playback Volume", 2, 6, 1), +}; + static struct snd_kcontrol_new snd_pmac_awacs_mixers_pmac[] __initdata = { AWACS_VOLUME("Master Playback Volume", 2, 6, 1), AWACS_SWITCH("CD Capture Switch", 0, SHIFT_MUX_CD, 0), @@ -645,12 +652,19 @@ static struct snd_kcontrol_new snd_pmac_screamer_mixers2[] __initdata = { AWACS_SWITCH("Mic Capture Switch", 0, SHIFT_MUX_LINE, 0), }; +static struct snd_kcontrol_new snd_pmac_awacs_mixers2_pmac5500[] __initdata = { + AWACS_SWITCH("CD Capture Switch", 0, SHIFT_MUX_CD, 0), +}; + static struct snd_kcontrol_new snd_pmac_awacs_master_sw __initdata = AWACS_SWITCH("Master Playback Switch", 1, SHIFT_HDMUTE, 1); static struct snd_kcontrol_new snd_pmac_awacs_master_sw_imac __initdata = AWACS_SWITCH("Line out Playback Switch", 1, SHIFT_HDMUTE, 1); +static struct snd_kcontrol_new snd_pmac_awacs_master_sw_pmac5500 __initdata = +AWACS_SWITCH("Headphone Playback Switch", 1, SHIFT_HDMUTE, 1); + static struct snd_kcontrol_new snd_pmac_awacs_mic_boost[] __initdata = { AWACS_SWITCH("Mic Boost Capture Switch", 0, SHIFT_GAINLINE, 0), }; @@ -868,6 +882,8 @@ snd_pmac_awacs_init(struct snd_pmac *chip) int lombard = IS_LOMBARD; int imac; int err, vol; + struct snd_kcontrol *vmaster_sw, *vmaster_vol; + struct snd_kcontrol *master_vol, *speaker_vol; imac1 = IS_IMAC1; imac2 = IS_IMAC2; @@ -968,19 +984,35 @@ snd_pmac_awacs_init(struct snd_pmac *chip) snd_pmac_awacs_mixers2); if (err < 0) return err; + if (pm5500) { + err = build_mixers(chip, + ARRAY_SIZE(snd_pmac_awacs_mixers2_pmac5500), + snd_pmac_awacs_mixers2_pmac5500); + if (err < 0) + return err; + } if (pm7500) err = build_mixers(chip, ARRAY_SIZE(snd_pmac_awacs_mixers_pmac7500), snd_pmac_awacs_mixers_pmac7500); + else if (pm5500) + err = snd_ctl_add(chip->card, + (master_vol = snd_ctl_new1(snd_pmac_awacs_mixers_pmac5500, + chip))); else if (beige) err = build_mixers(chip, ARRAY_SIZE(snd_pmac_screamer_mixers_beige), snd_pmac_screamer_mixers_beige); - else if (imac || lombard) + else if (imac || lombard) { + err = snd_ctl_add(chip->card, + (master_vol = snd_ctl_new1(snd_pmac_screamer_mixers_lo, + chip))); + if (err < 0) + return err; err = build_mixers(chip, ARRAY_SIZE(snd_pmac_screamer_mixers_imac), snd_pmac_screamer_mixers_imac); - else if (g4agp) + } else if (g4agp) err = build_mixers(chip, ARRAY_SIZE(snd_pmac_screamer_mixers_g4agp), snd_pmac_screamer_mixers_g4agp); @@ -992,6 +1024,8 @@ snd_pmac_awacs_init(struct snd_pmac *chip) return err; chip->master_sw_ctl = snd_ctl_new1((pm7500 || imac || g4agp || lombard) ? &snd_pmac_awacs_master_sw_imac + : pm5500 + ? &snd_pmac_awacs_master_sw_pmac5500 : &snd_pmac_awacs_master_sw, chip); err = snd_ctl_add(chip->card, chip->master_sw_ctl); if (err < 0) @@ -1023,8 +1057,9 @@ snd_pmac_awacs_init(struct snd_pmac *chip) #endif /* PMAC_AMP_AVAIL */ { /* route A = headphone, route C = speaker */ - err = build_mixers(chip, ARRAY_SIZE(snd_pmac_awacs_speaker_vol), - snd_pmac_awacs_speaker_vol); + err = snd_ctl_add(chip->card, + (speaker_vol = snd_ctl_new1(snd_pmac_awacs_speaker_vol, + chip))); if (err < 0) return err; chip->speaker_sw_ctl = snd_ctl_new1(imac1 @@ -1037,6 +1072,33 @@ snd_pmac_awacs_init(struct snd_pmac *chip) return err; } + if (pm5500 || imac || lombard) { + vmaster_sw = snd_ctl_make_virtual_master( + "Master Playback Switch", (unsigned int *) NULL); + err = snd_ctl_add_slave_uncached(vmaster_sw, + chip->master_sw_ctl); + if (err < 0) + return err; + err = snd_ctl_add_slave_uncached(vmaster_sw, + chip->speaker_sw_ctl); + if (err < 0) + return err; + err = snd_ctl_add(chip->card, vmaster_sw); + if (err < 0) + return err; + vmaster_vol = snd_ctl_make_virtual_master( + "Master Playback Volume", (unsigned int *) NULL); + err = snd_ctl_add_slave(vmaster_vol, master_vol); + if (err < 0) + return err; + err = snd_ctl_add_slave(vmaster_vol, speaker_vol); + if (err < 0) + return err; + err = snd_ctl_add(chip->card, vmaster_vol); + if (err < 0) + return err; + } + if (beige || g4agp) err = build_mixers(chip, ARRAY_SIZE(snd_pmac_screamer_mic_boost_beige), -- cgit v1.2.3-70-g09d2 From 6da6711385165eff76411b77974eec13c5ef6486 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 5 Feb 2009 16:02:46 +0100 Subject: ALSA: powermac - Add missing KERN_* prefix to printk Signed-off-by: Takashi Iwai --- sound/ppc/daca.c | 2 +- sound/ppc/pmac.c | 8 ++++---- sound/ppc/powermac.c | 2 +- sound/ppc/tumbler.c | 13 +++++++------ 4 files changed, 13 insertions(+), 12 deletions(-) diff --git a/sound/ppc/daca.c b/sound/ppc/daca.c index 8a5b2903193..f8d478c2da6 100644 --- a/sound/ppc/daca.c +++ b/sound/ppc/daca.c @@ -82,7 +82,7 @@ static int daca_set_volume(struct pmac_daca *mix) data[1] |= mix->deemphasis ? 0x40 : 0; if (i2c_smbus_write_block_data(mix->i2c.client, DACA_REG_AVOL, 2, data) < 0) { - snd_printk("failed to set volume \n"); + snd_printk(KERN_ERR "failed to set volume \n"); return -EINVAL; } return 0; diff --git a/sound/ppc/pmac.c b/sound/ppc/pmac.c index bd8f92b1c22..9b4e9c31669 100644 --- a/sound/ppc/pmac.c +++ b/sound/ppc/pmac.c @@ -299,7 +299,7 @@ static int snd_pmac_pcm_trigger(struct snd_pmac *chip, struct pmac_stream *rec, case SNDRV_PCM_TRIGGER_SUSPEND: spin_lock(&chip->reg_lock); rec->running = 0; - /*printk("stopped!!\n");*/ + /*printk(KERN_DEBUG "stopped!!\n");*/ snd_pmac_dma_stop(rec); for (i = 0, cp = rec->cmd.cmds; i < rec->nperiods; i++, cp++) out_le16(&cp->command, DBDMA_STOP); @@ -334,7 +334,7 @@ static snd_pcm_uframes_t snd_pmac_pcm_pointer(struct snd_pmac *chip, } #endif count += rec->cur_period * rec->period_size; - /*printk("pointer=%d\n", count);*/ + /*printk(KERN_DEBUG "pointer=%d\n", count);*/ return bytes_to_frames(subs->runtime, count); } @@ -486,7 +486,7 @@ static void snd_pmac_pcm_update(struct snd_pmac *chip, struct pmac_stream *rec) if (! (stat & ACTIVE)) break; - /*printk("update frag %d\n", rec->cur_period);*/ + /*printk(KERN_DEBUG "update frag %d\n", rec->cur_period);*/ st_le16(&cp->xfer_status, 0); st_le16(&cp->req_count, rec->period_size); /*st_le16(&cp->res_count, 0);*/ @@ -806,7 +806,7 @@ snd_pmac_ctrl_intr(int irq, void *devid) struct snd_pmac *chip = devid; int ctrl = in_le32(&chip->awacs->control); - /*printk("pmac: control interrupt.. 0x%x\n", ctrl);*/ + /*printk(KERN_DEBUG "pmac: control interrupt.. 0x%x\n", ctrl);*/ if (ctrl & MASK_PORTCHG) { /* do something when headphone is plugged/unplugged? */ if (chip->update_automute) diff --git a/sound/ppc/powermac.c b/sound/ppc/powermac.c index c936225771b..e9b02d97435 100644 --- a/sound/ppc/powermac.c +++ b/sound/ppc/powermac.c @@ -110,7 +110,7 @@ static int __init snd_pmac_probe(struct platform_device *devptr) goto __error; break; default: - snd_printk("unsupported hardware %d\n", chip->model); + snd_printk(KERN_ERR "unsupported hardware %d\n", chip->model); err = -EINVAL; goto __error; } diff --git a/sound/ppc/tumbler.c b/sound/ppc/tumbler.c index 3eb22338541..40222fcc087 100644 --- a/sound/ppc/tumbler.c +++ b/sound/ppc/tumbler.c @@ -41,7 +41,7 @@ #undef DEBUG #ifdef DEBUG -#define DBG(fmt...) printk(fmt) +#define DBG(fmt...) printk(KERN_DEBUG fmt) #else #define DBG(fmt...) #endif @@ -240,7 +240,7 @@ static int tumbler_set_master_volume(struct pmac_tumbler *mix) if (i2c_smbus_write_i2c_block_data(mix->i2c.client, TAS_REG_VOL, 6, block) < 0) { - snd_printk("failed to set volume \n"); + snd_printk(KERN_ERR "failed to set volume \n"); return -EINVAL; } return 0; @@ -350,7 +350,7 @@ static int tumbler_set_drc(struct pmac_tumbler *mix) if (i2c_smbus_write_i2c_block_data(mix->i2c.client, TAS_REG_DRC, 2, val) < 0) { - snd_printk("failed to set DRC\n"); + snd_printk(KERN_ERR "failed to set DRC\n"); return -EINVAL; } return 0; @@ -386,7 +386,7 @@ static int snapper_set_drc(struct pmac_tumbler *mix) if (i2c_smbus_write_i2c_block_data(mix->i2c.client, TAS_REG_DRC, 6, val) < 0) { - snd_printk("failed to set DRC\n"); + snd_printk(KERN_ERR "failed to set DRC\n"); return -EINVAL; } return 0; @@ -506,7 +506,8 @@ static int tumbler_set_mono_volume(struct pmac_tumbler *mix, block[i] = (vol >> ((info->bytes - i - 1) * 8)) & 0xff; if (i2c_smbus_write_i2c_block_data(mix->i2c.client, info->reg, info->bytes, block) < 0) { - snd_printk("failed to set mono volume %d\n", info->index); + snd_printk(KERN_ERR "failed to set mono volume %d\n", + info->index); return -EINVAL; } return 0; @@ -643,7 +644,7 @@ static int snapper_set_mix_vol1(struct pmac_tumbler *mix, int idx, int ch, int r } if (i2c_smbus_write_i2c_block_data(mix->i2c.client, reg, 9, block) < 0) { - snd_printk("failed to set mono volume %d\n", reg); + snd_printk(KERN_ERR "failed to set mono volume %d\n", reg); return -EINVAL; } return 0; -- cgit v1.2.3-70-g09d2 From 39661758631da37efbc961e57a4ddefad573cc52 Mon Sep 17 00:00:00 2001 From: Roel Kluin Date: Wed, 25 Feb 2009 13:40:26 +0100 Subject: ALSA: snd-powermac: timeout reaches -1 If unsuccessful, timeout reaches -1 after the loop. Signed-off-by: Roel Kluin Signed-off-by: Takashi Iwai --- sound/ppc/burgundy.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/ppc/burgundy.c b/sound/ppc/burgundy.c index f860d39af36..45a76297c38 100644 --- a/sound/ppc/burgundy.c +++ b/sound/ppc/burgundy.c @@ -35,7 +35,7 @@ snd_pmac_burgundy_busy_wait(struct snd_pmac *chip) int timeout = 50; while ((in_le32(&chip->awacs->codec_ctrl) & MASK_NEWECMD) && timeout--) udelay(1); - if (! timeout) + if (timeout < 0) printk(KERN_DEBUG "burgundy_busy_wait: timeout\n"); } -- cgit v1.2.3-70-g09d2 From 79c7cdd5441f5d3900c1632adcc8cd2bee35c8da Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 9 Feb 2009 14:47:19 +0100 Subject: ALSA: Add kernel-doc comments to vmaster stuff Signed-off-by: Takashi Iwai --- .../sound/alsa/DocBook/alsa-driver-api.tmpl | 4 +++ include/sound/control.h | 32 ++++++++++++++++++++++ sound/core/vmaster.c | 16 +++++++++-- 3 files changed, 50 insertions(+), 2 deletions(-) diff --git a/Documentation/sound/alsa/DocBook/alsa-driver-api.tmpl b/Documentation/sound/alsa/DocBook/alsa-driver-api.tmpl index 9d644f7e241..115962827c8 100644 --- a/Documentation/sound/alsa/DocBook/alsa-driver-api.tmpl +++ b/Documentation/sound/alsa/DocBook/alsa-driver-api.tmpl @@ -71,6 +71,10 @@ !Esound/pci/ac97/ac97_codec.c !Esound/pci/ac97/ac97_pcm.c + Virtual Master Control API +!Esound/core/vmaster.c +!Iinclude/sound/control.h + MIDI API Raw MIDI API diff --git a/include/sound/control.h b/include/sound/control.h index 4cf8f7aaa13..ef96f07aa03 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -176,12 +176,44 @@ int _snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave, /* optional flags for slave */ #define SND_CTL_SLAVE_NEED_UPDATE (1 << 0) +/** + * snd_ctl_add_slave - Add a virtual slave control + * @master: vmaster element + * @slave: slave element to add + * + * Add a virtual slave control to the given master element created via + * snd_ctl_create_virtual_master() beforehand. + * Returns zero if successful or a negative error code. + * + * All slaves must be the same type (returning the same information + * via info callback). The fucntion doesn't check it, so it's your + * responsibility. + * + * Also, some additional limitations: + * at most two channels, + * logarithmic volume control (dB level) thus no linear volume, + * master can only attenuate the volume without gain + */ static inline int snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave) { return _snd_ctl_add_slave(master, slave, 0); } +/** + * snd_ctl_add_slave_uncached - Add a virtual slave control + * @master: vmaster element + * @slave: slave element to add + * + * Add a virtual slave control to the given master. + * Unlike snd_ctl_add_slave(), the element added via this function + * is supposed to have volatile values, and get callback is called + * at each time quried from the master. + * + * When the control peeks the hardware values directly and the value + * can be changed by other means than the put callback of the element, + * this function should be used to keep the value always up-to-date. + */ static inline int snd_ctl_add_slave_uncached(struct snd_kcontrol *master, struct snd_kcontrol *slave) diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c index d51b198d06d..257624bd199 100644 --- a/sound/core/vmaster.c +++ b/sound/core/vmaster.c @@ -340,8 +340,20 @@ static void master_free(struct snd_kcontrol *kcontrol) } -/* - * Create a virtual master control with the given name +/** + * snd_ctl_make_virtual_master - Create a virtual master control + * @name: name string of the control element to create + * @tlv: optional TLV int array for dB information + * + * Creates a virtual matster control with the given name string. + * Returns the created control element, or NULL for errors (ENOMEM). + * + * After creating a vmaster element, you can add the slave controls + * via snd_ctl_add_slave() or snd_ctl_add_slave_uncached(). + * + * The optional argument @tlv can be used to specify the TLV information + * for dB scale of the master control. It should be a single element + * with #SNDRV_CTL_TLVT_DB_SCALE type, and should be the max 0dB. */ struct snd_kcontrol *snd_ctl_make_virtual_master(char *name, const unsigned int *tlv) -- cgit v1.2.3-70-g09d2 From 662c319ae4b4fb60001816dfe1dde5fdfc7a2af9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 9 Feb 2009 08:53:50 +0100 Subject: ALSA: Add sound/core/jack.c to driver-API docbook entry Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/DocBook/alsa-driver-api.tmpl | 3 +++ 1 file changed, 3 insertions(+) diff --git a/Documentation/sound/alsa/DocBook/alsa-driver-api.tmpl b/Documentation/sound/alsa/DocBook/alsa-driver-api.tmpl index 9d644f7e241..37b006cdf2f 100644 --- a/Documentation/sound/alsa/DocBook/alsa-driver-api.tmpl +++ b/Documentation/sound/alsa/DocBook/alsa-driver-api.tmpl @@ -88,6 +88,9 @@ Miscellaneous Functions Hardware-Dependent Devices API !Esound/core/hwdep.c + + Jack Abstraction Layer API +!Esound/core/jack.c ISA DMA Helpers !Esound/core/isadma.c -- cgit v1.2.3-70-g09d2 From 118dd6bfe7e0cddc8ab417ead19cc76000e92773 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 23 Feb 2009 16:35:21 +0100 Subject: ALSA: Clean up snd_monitor_file management Use the standard linked list for snd_monitor_file management. Also, move the list deletion of shutdown_list element into snd_disconnect_release() (for simplification). Signed-off-by: Takashi Iwai --- include/sound/core.h | 6 +++--- sound/core/init.c | 42 +++++++++++++++--------------------------- 2 files changed, 18 insertions(+), 30 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index f632484bc74..bd4529e0c27 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -97,9 +97,9 @@ struct snd_device { struct snd_monitor_file { struct file *file; - struct snd_monitor_file *next; const struct file_operations *disconnected_f_op; - struct list_head shutdown_list; + struct list_head shutdown_list; /* still need to shutdown */ + struct list_head list; /* link of monitor files */ }; /* main structure for soundcard */ @@ -134,7 +134,7 @@ struct snd_card { struct snd_info_entry *proc_id; /* the card id */ struct proc_dir_entry *proc_root_link; /* number link to real id */ - struct snd_monitor_file *files; /* all files associated to this card */ + struct list_head files_list; /* all files associated to this card */ struct snd_shutdown_f_ops *s_f_ops; /* file operations in the shutdown state */ spinlock_t files_lock; /* lock the files for this card */ diff --git a/sound/core/init.c b/sound/core/init.c index 0d5520c415d..05c6da554cb 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -195,6 +195,7 @@ struct snd_card *snd_card_new(int idx, const char *xid, INIT_LIST_HEAD(&card->controls); INIT_LIST_HEAD(&card->ctl_files); spin_lock_init(&card->files_lock); + INIT_LIST_HEAD(&card->files_list); init_waitqueue_head(&card->shutdown_sleep); #ifdef CONFIG_PM mutex_init(&card->power_lock); @@ -259,6 +260,7 @@ static int snd_disconnect_release(struct inode *inode, struct file *file) list_for_each_entry(_df, &shutdown_files, shutdown_list) { if (_df->file == file) { df = _df; + list_del_init(&df->shutdown_list); break; } } @@ -347,8 +349,7 @@ int snd_card_disconnect(struct snd_card *card) /* phase 2: replace file->f_op with special dummy operations */ spin_lock(&card->files_lock); - mfile = card->files; - while (mfile) { + list_for_each_entry(mfile, &card->files_list, list) { file = mfile->file; /* it's critical part, use endless loop */ @@ -361,8 +362,6 @@ int snd_card_disconnect(struct snd_card *card) mfile->file->f_op = &snd_shutdown_f_ops; fops_get(mfile->file->f_op); - - mfile = mfile->next; } spin_unlock(&card->files_lock); @@ -442,7 +441,7 @@ int snd_card_free_when_closed(struct snd_card *card) return ret; spin_lock(&card->files_lock); - if (card->files == NULL) + if (list_empty(&card->files_list)) free_now = 1; else card->free_on_last_close = 1; @@ -462,7 +461,7 @@ int snd_card_free(struct snd_card *card) return ret; /* wait, until all devices are ready for the free operation */ - wait_event(card->shutdown_sleep, card->files == NULL); + wait_event(card->shutdown_sleep, list_empty(&card->files_list)); snd_card_do_free(card); return 0; } @@ -809,15 +808,13 @@ int snd_card_file_add(struct snd_card *card, struct file *file) return -ENOMEM; mfile->file = file; mfile->disconnected_f_op = NULL; - mfile->next = NULL; spin_lock(&card->files_lock); if (card->shutdown) { spin_unlock(&card->files_lock); kfree(mfile); return -ENODEV; } - mfile->next = card->files; - card->files = mfile; + list_add(&mfile->list, &card->files_list); spin_unlock(&card->files_lock); return 0; } @@ -839,29 +836,20 @@ EXPORT_SYMBOL(snd_card_file_add); */ int snd_card_file_remove(struct snd_card *card, struct file *file) { - struct snd_monitor_file *mfile, *pfile = NULL; + struct snd_monitor_file *mfile, *found = NULL; int last_close = 0; spin_lock(&card->files_lock); - mfile = card->files; - while (mfile) { + list_for_each_entry(mfile, &card->files_list, list) { if (mfile->file == file) { - if (pfile) - pfile->next = mfile->next; - else - card->files = mfile->next; + list_del(&mfile->list); + if (mfile->disconnected_f_op) + fops_put(mfile->disconnected_f_op); + found = mfile; break; } - pfile = mfile; - mfile = mfile->next; - } - if (mfile && mfile->disconnected_f_op) { - fops_put(mfile->disconnected_f_op); - spin_lock(&shutdown_lock); - list_del(&mfile->shutdown_list); - spin_unlock(&shutdown_lock); } - if (card->files == NULL) + if (list_empty(&card->files_list)) last_close = 1; spin_unlock(&card->files_lock); if (last_close) { @@ -869,11 +857,11 @@ int snd_card_file_remove(struct snd_card *card, struct file *file) if (card->free_on_last_close) snd_card_do_free(card); } - if (!mfile) { + if (!found) { snd_printk(KERN_ERR "ALSA card file remove problem (%p)\n", file); return -ENOENT; } - kfree(mfile); + kfree(found); return 0; } -- cgit v1.2.3-70-g09d2 From f9d202833d0beac09ef1c6a41305151da4fe5d4c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 11 Feb 2009 14:55:59 +0100 Subject: ALSA: rawmidi - Fix possible race in open The module refcount should be handled in the register_mutex to avoid possible races with module unloading. Signed-off-by: Takashi Iwai --- sound/core/rawmidi.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 002777ba336..60f33e9412a 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -237,15 +237,16 @@ int snd_rawmidi_kernel_open(struct snd_card *card, int device, int subdevice, rfile->input = rfile->output = NULL; mutex_lock(®ister_mutex); rmidi = snd_rawmidi_search(card, device); - mutex_unlock(®ister_mutex); if (rmidi == NULL) { - err = -ENODEV; - goto __error1; + mutex_unlock(®ister_mutex); + return -ENODEV; } if (!try_module_get(rmidi->card->module)) { - err = -EFAULT; - goto __error1; + mutex_unlock(®ister_mutex); + return -ENXIO; } + mutex_unlock(®ister_mutex); + if (!(mode & SNDRV_RAWMIDI_LFLG_NOOPENLOCK)) mutex_lock(&rmidi->open_mutex); if (mode & SNDRV_RAWMIDI_LFLG_INPUT) { @@ -370,10 +371,9 @@ int snd_rawmidi_kernel_open(struct snd_card *card, int device, int subdevice, snd_rawmidi_runtime_free(sinput); if (output != NULL) snd_rawmidi_runtime_free(soutput); - module_put(rmidi->card->module); if (!(mode & SNDRV_RAWMIDI_LFLG_NOOPENLOCK)) mutex_unlock(&rmidi->open_mutex); - __error1: + module_put(rmidi->card->module); return err; } -- cgit v1.2.3-70-g09d2 From 9a1b64caac82aa02cb74587ffc798e6f42c6170a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 11 Feb 2009 17:03:49 +0100 Subject: ALSA: rawmidi - Refactor rawmidi open/close codes Refactor rawmidi open/close code messes. Signed-off-by: Takashi Iwai --- include/sound/rawmidi.h | 1 - sound/core/rawmidi.c | 377 +++++++++++++++++++++++++----------------------- 2 files changed, 194 insertions(+), 184 deletions(-) diff --git a/include/sound/rawmidi.h b/include/sound/rawmidi.h index b550a416d07..c23c2658570 100644 --- a/include/sound/rawmidi.h +++ b/include/sound/rawmidi.h @@ -42,7 +42,6 @@ #define SNDRV_RAWMIDI_LFLG_INPUT (1<<1) #define SNDRV_RAWMIDI_LFLG_OPEN (3<<0) #define SNDRV_RAWMIDI_LFLG_APPEND (1<<2) -#define SNDRV_RAWMIDI_LFLG_NOOPENLOCK (1<<3) struct snd_rawmidi; struct snd_rawmidi_substream; diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 60f33e9412a..473247c8e6d 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -224,156 +224,143 @@ int snd_rawmidi_drain_input(struct snd_rawmidi_substream *substream) return 0; } -int snd_rawmidi_kernel_open(struct snd_card *card, int device, int subdevice, - int mode, struct snd_rawmidi_file * rfile) +/* look for an available substream for the given stream direction; + * if a specific subdevice is given, try to assign it + */ +static int assign_substream(struct snd_rawmidi *rmidi, int subdevice, + int stream, int mode, + struct snd_rawmidi_substream **sub_ret) { - struct snd_rawmidi *rmidi; - struct list_head *list1, *list2; - struct snd_rawmidi_substream *sinput = NULL, *soutput = NULL; - struct snd_rawmidi_runtime *input = NULL, *output = NULL; - int err; + struct snd_rawmidi_substream *substream; + struct snd_rawmidi_str *s = &rmidi->streams[stream]; + static unsigned int info_flags[2] = { + [SNDRV_RAWMIDI_STREAM_OUTPUT] = SNDRV_RAWMIDI_INFO_OUTPUT, + [SNDRV_RAWMIDI_STREAM_INPUT] = SNDRV_RAWMIDI_INFO_INPUT, + }; - if (rfile) - rfile->input = rfile->output = NULL; - mutex_lock(®ister_mutex); - rmidi = snd_rawmidi_search(card, device); - if (rmidi == NULL) { - mutex_unlock(®ister_mutex); - return -ENODEV; - } - if (!try_module_get(rmidi->card->module)) { - mutex_unlock(®ister_mutex); + if (!(rmidi->info_flags & info_flags[stream])) return -ENXIO; + if (subdevice >= 0 && subdevice >= s->substream_count) + return -ENODEV; + if (s->substream_opened >= s->substream_count) + return -EAGAIN; + + list_for_each_entry(substream, &s->substreams, list) { + if (substream->opened) { + if (stream == SNDRV_RAWMIDI_STREAM_INPUT || + !(mode & SNDRV_RAWMIDI_LFLG_APPEND)) + continue; + } + if (subdevice < 0 || subdevice == substream->number) { + *sub_ret = substream; + return 0; + } } - mutex_unlock(®ister_mutex); + return -EAGAIN; +} - if (!(mode & SNDRV_RAWMIDI_LFLG_NOOPENLOCK)) - mutex_lock(&rmidi->open_mutex); +/* open and do ref-counting for the given substream */ +static int open_substream(struct snd_rawmidi *rmidi, + struct snd_rawmidi_substream *substream, + int mode) +{ + int err; + + err = snd_rawmidi_runtime_create(substream); + if (err < 0) + return err; + err = substream->ops->open(substream); + if (err < 0) + return err; + substream->opened = 1; + if (substream->use_count++ == 0) + substream->active_sensing = 1; + if (mode & SNDRV_RAWMIDI_LFLG_APPEND) + substream->append = 1; + rmidi->streams[substream->stream].substream_opened++; + return 0; +} + +static void close_substream(struct snd_rawmidi *rmidi, + struct snd_rawmidi_substream *substream, + int cleanup); + +static int rawmidi_open_priv(struct snd_rawmidi *rmidi, int subdevice, int mode, + struct snd_rawmidi_file *rfile) +{ + struct snd_rawmidi_substream *sinput = NULL, *soutput = NULL; + int err; + + rfile->input = rfile->output = NULL; if (mode & SNDRV_RAWMIDI_LFLG_INPUT) { - if (!(rmidi->info_flags & SNDRV_RAWMIDI_INFO_INPUT)) { - err = -ENXIO; - goto __error; - } - if (subdevice >= 0 && (unsigned int)subdevice >= rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT].substream_count) { - err = -ENODEV; - goto __error; - } - if (rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT].substream_opened >= - rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT].substream_count) { - err = -EAGAIN; + err = assign_substream(rmidi, subdevice, + SNDRV_RAWMIDI_STREAM_INPUT, + mode, &sinput); + if (err < 0) goto __error; - } } if (mode & SNDRV_RAWMIDI_LFLG_OUTPUT) { - if (!(rmidi->info_flags & SNDRV_RAWMIDI_INFO_OUTPUT)) { - err = -ENXIO; - goto __error; - } - if (subdevice >= 0 && (unsigned int)subdevice >= rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substream_count) { - err = -ENODEV; - goto __error; - } - if (rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substream_opened >= - rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substream_count) { - err = -EAGAIN; + err = assign_substream(rmidi, subdevice, + SNDRV_RAWMIDI_STREAM_OUTPUT, + mode, &soutput); + if (err < 0) goto __error; - } - } - list1 = rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT].substreams.next; - while (1) { - if (list1 == &rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT].substreams) { - sinput = NULL; - if (mode & SNDRV_RAWMIDI_LFLG_INPUT) { - err = -EAGAIN; - goto __error; - } - break; - } - sinput = list_entry(list1, struct snd_rawmidi_substream, list); - if ((mode & SNDRV_RAWMIDI_LFLG_INPUT) && sinput->opened) - goto __nexti; - if (subdevice < 0 || (subdevice >= 0 && subdevice == sinput->number)) - break; - __nexti: - list1 = list1->next; - } - list2 = rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substreams.next; - while (1) { - if (list2 == &rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substreams) { - soutput = NULL; - if (mode & SNDRV_RAWMIDI_LFLG_OUTPUT) { - err = -EAGAIN; - goto __error; - } - break; - } - soutput = list_entry(list2, struct snd_rawmidi_substream, list); - if (mode & SNDRV_RAWMIDI_LFLG_OUTPUT) { - if (mode & SNDRV_RAWMIDI_LFLG_APPEND) { - if (soutput->opened && !soutput->append) - goto __nexto; - } else { - if (soutput->opened) - goto __nexto; - } - } - if (subdevice < 0 || (subdevice >= 0 && subdevice == soutput->number)) - break; - __nexto: - list2 = list2->next; } - if (mode & SNDRV_RAWMIDI_LFLG_INPUT) { - if ((err = snd_rawmidi_runtime_create(sinput)) < 0) - goto __error; - input = sinput->runtime; - if ((err = sinput->ops->open(sinput)) < 0) + + if (sinput) { + err = open_substream(rmidi, sinput, mode); + if (err < 0) goto __error; - sinput->opened = 1; - rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT].substream_opened++; - } else { - sinput = NULL; } - if (mode & SNDRV_RAWMIDI_LFLG_OUTPUT) { - if (soutput->opened) - goto __skip_output; - if ((err = snd_rawmidi_runtime_create(soutput)) < 0) { - if (mode & SNDRV_RAWMIDI_LFLG_INPUT) - sinput->ops->close(sinput); - goto __error; - } - output = soutput->runtime; - if ((err = soutput->ops->open(soutput)) < 0) { - if (mode & SNDRV_RAWMIDI_LFLG_INPUT) - sinput->ops->close(sinput); + if (soutput) { + err = open_substream(rmidi, soutput, mode); + if (err < 0) { + if (sinput) + close_substream(rmidi, sinput, 0); goto __error; } - __skip_output: - soutput->opened = 1; - if (mode & SNDRV_RAWMIDI_LFLG_APPEND) - soutput->append = 1; - if (soutput->use_count++ == 0) - soutput->active_sensing = 1; - rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substream_opened++; - } else { - soutput = NULL; - } - if (!(mode & SNDRV_RAWMIDI_LFLG_NOOPENLOCK)) - mutex_unlock(&rmidi->open_mutex); - if (rfile) { - rfile->rmidi = rmidi; - rfile->input = sinput; - rfile->output = soutput; } + + rfile->rmidi = rmidi; + rfile->input = sinput; + rfile->output = soutput; return 0; __error: - if (input != NULL) + if (sinput && sinput->runtime) snd_rawmidi_runtime_free(sinput); - if (output != NULL) + if (soutput && soutput->runtime) snd_rawmidi_runtime_free(soutput); - if (!(mode & SNDRV_RAWMIDI_LFLG_NOOPENLOCK)) - mutex_unlock(&rmidi->open_mutex); - module_put(rmidi->card->module); + return err; +} + +/* called from sound/core/seq/seq_midi.c */ +int snd_rawmidi_kernel_open(struct snd_card *card, int device, int subdevice, + int mode, struct snd_rawmidi_file * rfile) +{ + struct snd_rawmidi *rmidi; + int err; + + if (snd_BUG_ON(!rfile)) + return -EINVAL; + + mutex_lock(®ister_mutex); + rmidi = snd_rawmidi_search(card, device); + if (rmidi == NULL) { + mutex_unlock(®ister_mutex); + return -ENODEV; + } + if (!try_module_get(rmidi->card->module)) { + mutex_unlock(®ister_mutex); + return -ENXIO; + } + mutex_unlock(®ister_mutex); + + mutex_lock(&rmidi->open_mutex); + err = rawmidi_open_priv(rmidi, subdevice, mode, rfile); + mutex_unlock(&rmidi->open_mutex); + if (err < 0) + module_put(rmidi->card->module); return err; } @@ -385,10 +372,13 @@ static int snd_rawmidi_open(struct inode *inode, struct file *file) unsigned short fflags; int err; struct snd_rawmidi *rmidi; - struct snd_rawmidi_file *rawmidi_file; + struct snd_rawmidi_file *rawmidi_file = NULL; wait_queue_t wait; struct snd_ctl_file *kctl; + if ((file->f_flags & O_APPEND) && !(file->f_flags & O_NONBLOCK)) + return -EINVAL; /* invalid combination */ + if (maj == snd_major) { rmidi = snd_lookup_minor_data(iminor(inode), SNDRV_DEVICE_TYPE_RAWMIDI); @@ -402,24 +392,25 @@ static int snd_rawmidi_open(struct inode *inode, struct file *file) if (rmidi == NULL) return -ENODEV; - if ((file->f_flags & O_APPEND) && !(file->f_flags & O_NONBLOCK)) - return -EINVAL; /* invalid combination */ + + if (!try_module_get(rmidi->card->module)) + return -ENXIO; + + mutex_lock(&rmidi->open_mutex); card = rmidi->card; err = snd_card_file_add(card, file); if (err < 0) - return -ENODEV; + goto __error_card; fflags = snd_rawmidi_file_flags(file); if ((file->f_flags & O_APPEND) || maj == SOUND_MAJOR) /* OSS emul? */ fflags |= SNDRV_RAWMIDI_LFLG_APPEND; - fflags |= SNDRV_RAWMIDI_LFLG_NOOPENLOCK; rawmidi_file = kmalloc(sizeof(*rawmidi_file), GFP_KERNEL); if (rawmidi_file == NULL) { - snd_card_file_remove(card, file); - return -ENOMEM; + err = -ENOMEM; + goto __error; } init_waitqueue_entry(&wait, current); add_wait_queue(&rmidi->open_wait, &wait); - mutex_lock(&rmidi->open_mutex); while (1) { subdevice = -1; read_lock(&card->ctl_files_rwlock); @@ -431,8 +422,7 @@ static int snd_rawmidi_open(struct inode *inode, struct file *file) } } read_unlock(&card->ctl_files_rwlock); - err = snd_rawmidi_kernel_open(rmidi->card, rmidi->device, - subdevice, fflags, rawmidi_file); + err = rawmidi_open_priv(rmidi, subdevice, fflags, rawmidi_file); if (err >= 0) break; if (err == -EAGAIN) { @@ -451,67 +441,89 @@ static int snd_rawmidi_open(struct inode *inode, struct file *file) break; } } + remove_wait_queue(&rmidi->open_wait, &wait); + if (err < 0) { + kfree(rawmidi_file); + goto __error; + } #ifdef CONFIG_SND_OSSEMUL if (rawmidi_file->input && rawmidi_file->input->runtime) rawmidi_file->input->runtime->oss = (maj == SOUND_MAJOR); if (rawmidi_file->output && rawmidi_file->output->runtime) rawmidi_file->output->runtime->oss = (maj == SOUND_MAJOR); #endif - remove_wait_queue(&rmidi->open_wait, &wait); - if (err >= 0) { - file->private_data = rawmidi_file; - } else { - snd_card_file_remove(card, file); - kfree(rawmidi_file); - } + file->private_data = rawmidi_file; mutex_unlock(&rmidi->open_mutex); + return 0; + + __error: + snd_card_file_remove(card, file); + __error_card: + mutex_unlock(&rmidi->open_mutex); + module_put(rmidi->card->module); return err; } -int snd_rawmidi_kernel_release(struct snd_rawmidi_file * rfile) +static void close_substream(struct snd_rawmidi *rmidi, + struct snd_rawmidi_substream *substream, + int cleanup) { - struct snd_rawmidi *rmidi; - struct snd_rawmidi_substream *substream; - struct snd_rawmidi_runtime *runtime; + rmidi->streams[substream->stream].substream_opened--; + if (--substream->use_count) + return; - if (snd_BUG_ON(!rfile)) - return -ENXIO; - rmidi = rfile->rmidi; - mutex_lock(&rmidi->open_mutex); - if (rfile->input != NULL) { - substream = rfile->input; - rfile->input = NULL; - runtime = substream->runtime; - snd_rawmidi_input_trigger(substream, 0); - substream->ops->close(substream); - if (runtime->private_free != NULL) - runtime->private_free(substream); - snd_rawmidi_runtime_free(substream); - substream->opened = 0; - rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT].substream_opened--; - } - if (rfile->output != NULL) { - substream = rfile->output; - rfile->output = NULL; - if (--substream->use_count == 0) { - runtime = substream->runtime; + if (cleanup) { + if (substream->stream == SNDRV_RAWMIDI_STREAM_INPUT) + snd_rawmidi_input_trigger(substream, 0); + else { if (substream->active_sensing) { unsigned char buf = 0xfe; - /* sending single active sensing message to shut the device up */ + /* sending single active sensing message + * to shut the device up + */ snd_rawmidi_kernel_write(substream, &buf, 1); } if (snd_rawmidi_drain_output(substream) == -ERESTARTSYS) snd_rawmidi_output_trigger(substream, 0); - substream->ops->close(substream); - if (runtime->private_free != NULL) - runtime->private_free(substream); - snd_rawmidi_runtime_free(substream); - substream->opened = 0; - substream->append = 0; } - rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substream_opened--; } + substream->ops->close(substream); + if (substream->runtime->private_free) + substream->runtime->private_free(substream); + snd_rawmidi_runtime_free(substream); + substream->opened = 0; + substream->append = 0; +} + +static void rawmidi_release_priv(struct snd_rawmidi_file *rfile) +{ + struct snd_rawmidi *rmidi; + + rmidi = rfile->rmidi; + mutex_lock(&rmidi->open_mutex); + if (rfile->input) { + close_substream(rmidi, rfile->input, 1); + rfile->input = NULL; + } + if (rfile->output) { + close_substream(rmidi, rfile->output, 1); + rfile->output = NULL; + } + rfile->rmidi = NULL; mutex_unlock(&rmidi->open_mutex); + wake_up(&rmidi->open_wait); +} + +/* called from sound/core/seq/seq_midi.c */ +int snd_rawmidi_kernel_release(struct snd_rawmidi_file *rfile) +{ + struct snd_rawmidi *rmidi; + + if (snd_BUG_ON(!rfile)) + return -ENXIO; + + rmidi = rfile->rmidi; + rawmidi_release_priv(rfile); module_put(rmidi->card->module); return 0; } @@ -520,15 +532,14 @@ static int snd_rawmidi_release(struct inode *inode, struct file *file) { struct snd_rawmidi_file *rfile; struct snd_rawmidi *rmidi; - int err; rfile = file->private_data; - err = snd_rawmidi_kernel_release(rfile); rmidi = rfile->rmidi; - wake_up(&rmidi->open_wait); + rawmidi_release_priv(rfile); kfree(rfile); snd_card_file_remove(rmidi->card, file); - return err; + module_put(rmidi->card->module); + return 0; } static int snd_rawmidi_info(struct snd_rawmidi_substream *substream, -- cgit v1.2.3-70-g09d2 From 5f8206c04857965cc2ff6c395633c4fdd977dd77 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 9 Feb 2009 08:50:43 +0100 Subject: ALSA: Fix DocBook headers Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/DocBook/alsa-driver-api.tmpl | 10 ++++++---- Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl | 8 ++++---- 2 files changed, 10 insertions(+), 8 deletions(-) diff --git a/Documentation/sound/alsa/DocBook/alsa-driver-api.tmpl b/Documentation/sound/alsa/DocBook/alsa-driver-api.tmpl index 90f163c4bde..0230a96f056 100644 --- a/Documentation/sound/alsa/DocBook/alsa-driver-api.tmpl +++ b/Documentation/sound/alsa/DocBook/alsa-driver-api.tmpl @@ -1,11 +1,11 @@ - - - - + + + The ALSA Driver API @@ -35,6 +35,8 @@ + + Management of Cards and Devices Card Management !Esound/core/init.c diff --git a/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl b/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl index 320384c1791..46b08fef374 100644 --- a/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl +++ b/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl @@ -1,11 +1,11 @@ - - - - + + + Writing an ALSA Driver -- cgit v1.2.3-70-g09d2 From e776ec19a47a325ee1d9ece2d983526dcd626c53 Mon Sep 17 00:00:00 2001 From: Randy Dunlap Date: Sat, 28 Feb 2009 17:40:18 +0100 Subject: ALSA: Move ALSA docbooks to be with the rest of the kernel docbooks Move ALSA docbooks to be with the rest of the kernel docbooks and add them to the Makefile so that they build. Latter required a few minor changes to alsa .tmpl files. (I did not remove all of the trailing whitespace in the .tmpl files.) Fixes kernel bugzilla #12726: http://bugzilla.kernel.org/show_bug.cgi?id=12726 Signed-off-by: Randy Dunlap Cc: documentation_man-pages@kernel-bugs.osdl.org Cc: Nicola Soranzo Signed-off-by: Takashi Iwai --- Documentation/DocBook/Makefile | 3 +- Documentation/DocBook/alsa-driver-api.tmpl | 109 + Documentation/DocBook/writing-an-alsa-driver.tmpl | 6216 ++++++++++++++++++++ .../sound/alsa/DocBook/alsa-driver-api.tmpl | 109 - .../sound/alsa/DocBook/writing-an-alsa-driver.tmpl | 6216 -------------------- 5 files changed, 6327 insertions(+), 6326 deletions(-) create mode 100644 Documentation/DocBook/alsa-driver-api.tmpl create mode 100644 Documentation/DocBook/writing-an-alsa-driver.tmpl delete mode 100644 Documentation/sound/alsa/DocBook/alsa-driver-api.tmpl delete mode 100644 Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl diff --git a/Documentation/DocBook/Makefile b/Documentation/DocBook/Makefile index 1462ed86d40..a3a83d38f96 100644 --- a/Documentation/DocBook/Makefile +++ b/Documentation/DocBook/Makefile @@ -12,7 +12,8 @@ DOCBOOKS := z8530book.xml mcabook.xml device-drivers.xml \ kernel-api.xml filesystems.xml lsm.xml usb.xml kgdb.xml \ gadget.xml libata.xml mtdnand.xml librs.xml rapidio.xml \ genericirq.xml s390-drivers.xml uio-howto.xml scsi.xml \ - mac80211.xml debugobjects.xml sh.xml regulator.xml + mac80211.xml debugobjects.xml sh.xml regulator.xml \ + alsa-driver-api.xml writing-an-alsa-driver.xml ### # The build process is as follows (targets): diff --git a/Documentation/DocBook/alsa-driver-api.tmpl b/Documentation/DocBook/alsa-driver-api.tmpl new file mode 100644 index 00000000000..0230a96f056 --- /dev/null +++ b/Documentation/DocBook/alsa-driver-api.tmpl @@ -0,0 +1,109 @@ + + + + + + + + + The ALSA Driver API + + + + This document is free; you can redistribute it and/or modify it + under the terms of the GNU General Public License as published by + the Free Software Foundation; either version 2 of the License, or + (at your option) any later version. + + + + This document is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the + implied warranty of MERCHANTABILITY or FITNESS FOR A + PARTICULAR PURPOSE. See the GNU General Public License + for more details. + + + + You should have received a copy of the GNU General Public + License along with this program; if not, write to the Free + Software Foundation, Inc., 59 Temple Place, Suite 330, Boston, + MA 02111-1307 USA + + + + + + + + Management of Cards and Devices + Card Management +!Esound/core/init.c + + Device Components +!Esound/core/device.c + + Module requests and Device File Entries +!Esound/core/sound.c + + Memory Management Helpers +!Esound/core/memory.c +!Esound/core/memalloc.c + + + PCM API + PCM Core +!Esound/core/pcm.c +!Esound/core/pcm_lib.c +!Esound/core/pcm_native.c + + PCM Format Helpers +!Esound/core/pcm_misc.c + + PCM Memory Management +!Esound/core/pcm_memory.c + + + Control/Mixer API + General Control Interface +!Esound/core/control.c + + AC97 Codec API +!Esound/pci/ac97/ac97_codec.c +!Esound/pci/ac97/ac97_pcm.c + + Virtual Master Control API +!Esound/core/vmaster.c +!Iinclude/sound/control.h + + + MIDI API + Raw MIDI API +!Esound/core/rawmidi.c + + MPU401-UART API +!Esound/drivers/mpu401/mpu401_uart.c + + + Proc Info API + Proc Info Interface +!Esound/core/info.c + + + Miscellaneous Functions + Hardware-Dependent Devices API +!Esound/core/hwdep.c + + Jack Abstraction Layer API +!Esound/core/jack.c + + ISA DMA Helpers +!Esound/core/isadma.c + + Other Helper Macros +!Iinclude/sound/core.h + + + + diff --git a/Documentation/DocBook/writing-an-alsa-driver.tmpl b/Documentation/DocBook/writing-an-alsa-driver.tmpl new file mode 100644 index 00000000000..46b08fef374 --- /dev/null +++ b/Documentation/DocBook/writing-an-alsa-driver.tmpl @@ -0,0 +1,6216 @@ + + + + + + + + + Writing an ALSA Driver + + Takashi + Iwai + +
+ tiwai@suse.de +
+ + + + Oct 15, 2007 + 0.3.7 + + + + This document describes how to write an ALSA (Advanced Linux + Sound Architecture) driver. + + + + + + Copyright (c) 2002-2005 Takashi Iwai tiwai@suse.de + + + + This document is free; you can redistribute it and/or modify it + under the terms of the GNU General Public License as published by + the Free Software Foundation; either version 2 of the License, or + (at your option) any later version. + + + + This document is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the + implied warranty of MERCHANTABILITY or FITNESS FOR A + PARTICULAR PURPOSE. See the GNU General Public License + for more details. + + + + You should have received a copy of the GNU General Public + License along with this program; if not, write to the Free + Software Foundation, Inc., 59 Temple Place, Suite 330, Boston, + MA 02111-1307 USA + + + + + + + + + + Preface + + This document describes how to write an + + ALSA (Advanced Linux Sound Architecture) + driver. The document focuses mainly on PCI soundcards. + In the case of other device types, the API might + be different, too. However, at least the ALSA kernel API is + consistent, and therefore it would be still a bit help for + writing them. + + + + This document targets people who already have enough + C language skills and have basic linux kernel programming + knowledge. This document doesn't explain the general + topic of linux kernel coding and doesn't cover low-level + driver implementation details. It only describes + the standard way to write a PCI sound driver on ALSA. + + + + If you are already familiar with the older ALSA ver.0.5.x API, you + can check the drivers such as sound/pci/es1938.c or + sound/pci/maestro3.c which have also almost the same + code-base in the ALSA 0.5.x tree, so you can compare the differences. + + + + This document is still a draft version. Any feedback and + corrections, please!! + + + + + + + + + File Tree Structure + +
+ General + + The ALSA drivers are provided in two ways. + + + + One is the trees provided as a tarball or via cvs from the + ALSA's ftp site, and another is the 2.6 (or later) Linux kernel + tree. To synchronize both, the ALSA driver tree is split into + two different trees: alsa-kernel and alsa-driver. The former + contains purely the source code for the Linux 2.6 (or later) + tree. This tree is designed only for compilation on 2.6 or + later environment. The latter, alsa-driver, contains many subtle + files for compiling ALSA drivers outside of the Linux kernel tree, + wrapper functions for older 2.2 and 2.4 kernels, to adapt the latest kernel API, + and additional drivers which are still in development or in + tests. The drivers in alsa-driver tree will be moved to + alsa-kernel (and eventually to the 2.6 kernel tree) when they are + finished and confirmed to work fine. + + + + The file tree structure of ALSA driver is depicted below. Both + alsa-kernel and alsa-driver have almost the same file + structure, except for core directory. It's + named as acore in alsa-driver tree. + + + ALSA File Tree Structure + + sound + /core + /oss + /seq + /oss + /instr + /ioctl32 + /include + /drivers + /mpu401 + /opl3 + /i2c + /l3 + /synth + /emux + /pci + /(cards) + /isa + /(cards) + /arm + /ppc + /sparc + /usb + /pcmcia /(cards) + /oss + + + +
+ +
+ core directory + + This directory contains the middle layer which is the heart + of ALSA drivers. In this directory, the native ALSA modules are + stored. The sub-directories contain different modules and are + dependent upon the kernel config. + + +
+ core/oss + + + The codes for PCM and mixer OSS emulation modules are stored + in this directory. The rawmidi OSS emulation is included in + the ALSA rawmidi code since it's quite small. The sequencer + code is stored in core/seq/oss directory (see + + below). + +
+ +
+ core/ioctl32 + + + This directory contains the 32bit-ioctl wrappers for 64bit + architectures such like x86-64, ppc64 and sparc64. For 32bit + and alpha architectures, these are not compiled. + +
+ +
+ core/seq + + This directory and its sub-directories are for the ALSA + sequencer. This directory contains the sequencer core and + primary sequencer modules such like snd-seq-midi, + snd-seq-virmidi, etc. They are compiled only when + CONFIG_SND_SEQUENCER is set in the kernel + config. + +
+ +
+ core/seq/oss + + This contains the OSS sequencer emulation codes. + +
+ +
+ core/seq/instr + + This directory contains the modules for the sequencer + instrument layer. + +
+
+ +
+ include directory + + This is the place for the public header files of ALSA drivers, + which are to be exported to user-space, or included by + several files at different directories. Basically, the private + header files should not be placed in this directory, but you may + still find files there, due to historical reasons :) + +
+ +
+ drivers directory + + This directory contains code shared among different drivers + on different architectures. They are hence supposed not to be + architecture-specific. + For example, the dummy pcm driver and the serial MIDI + driver are found in this directory. In the sub-directories, + there is code for components which are independent from + bus and cpu architectures. + + +
+ drivers/mpu401 + + The MPU401 and MPU401-UART modules are stored here. + +
+ +
+ drivers/opl3 and opl4 + + The OPL3 and OPL4 FM-synth stuff is found here. + +
+
+ +
+ i2c directory + + This contains the ALSA i2c components. + + + + Although there is a standard i2c layer on Linux, ALSA has its + own i2c code for some cards, because the soundcard needs only a + simple operation and the standard i2c API is too complicated for + such a purpose. + + +
+ i2c/l3 + + This is a sub-directory for ARM L3 i2c. + +
+
+ +
+ synth directory + + This contains the synth middle-level modules. + + + + So far, there is only Emu8000/Emu10k1 synth driver under + the synth/emux sub-directory. + +
+ +
+ pci directory + + This directory and its sub-directories hold the top-level card modules + for PCI soundcards and the code specific to the PCI BUS. + + + + The drivers compiled from a single file are stored directly + in the pci directory, while the drivers with several source files are + stored on their own sub-directory (e.g. emu10k1, ice1712). + +
+ +
+ isa directory + + This directory and its sub-directories hold the top-level card modules + for ISA soundcards. + +
+ +
+ arm, ppc, and sparc directories + + They are used for top-level card modules which are + specific to one of these architectures. + +
+ +
+ usb directory + + This directory contains the USB-audio driver. In the latest version, the + USB MIDI driver is integrated in the usb-audio driver. + +
+ +
+ pcmcia directory + + The PCMCIA, especially PCCard drivers will go here. CardBus + drivers will be in the pci directory, because their API is identical + to that of standard PCI cards. + +
+ +
+ oss directory + + The OSS/Lite source files are stored here in Linux 2.6 (or + later) tree. In the ALSA driver tarball, this directory is empty, + of course :) + +
+ + + + + + + + Basic Flow for PCI Drivers + +
+ Outline + + The minimum flow for PCI soundcards is as follows: + + + define the PCI ID table (see the section + PCI Entries + ). + create probe() callback. + create remove() callback. + create a pci_driver structure + containing the three pointers above. + create an init() function just calling + the pci_register_driver() to register the pci_driver table + defined above. + create an exit() function to call + the pci_unregister_driver() function. + + +
+ +
+ Full Code Example + + The code example is shown below. Some parts are kept + unimplemented at this moment but will be filled in the + next sections. The numbers in the comment lines of the + snd_mychip_probe() function + refer to details explained in the following section. + + + Basic Flow for PCI Drivers - Example + + + #include + #include + #include + #include + + /* module parameters (see "Module Parameters") */ + /* SNDRV_CARDS: maximum number of cards supported by this module */ + static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; + static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; + static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; + + /* definition of the chip-specific record */ + struct mychip { + struct snd_card *card; + /* the rest of the implementation will be in section + * "PCI Resource Management" + */ + }; + + /* chip-specific destructor + * (see "PCI Resource Management") + */ + static int snd_mychip_free(struct mychip *chip) + { + .... /* will be implemented later... */ + } + + /* component-destructor + * (see "Management of Cards and Components") + */ + static int snd_mychip_dev_free(struct snd_device *device) + { + return snd_mychip_free(device->device_data); + } + + /* chip-specific constructor + * (see "Management of Cards and Components") + */ + static int __devinit snd_mychip_create(struct snd_card *card, + struct pci_dev *pci, + struct mychip **rchip) + { + struct mychip *chip; + int err; + static struct snd_device_ops ops = { + .dev_free = snd_mychip_dev_free, + }; + + *rchip = NULL; + + /* check PCI availability here + * (see "PCI Resource Management") + */ + .... + + /* allocate a chip-specific data with zero filled */ + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + if (chip == NULL) + return -ENOMEM; + + chip->card = card; + + /* rest of initialization here; will be implemented + * later, see "PCI Resource Management" + */ + .... + + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) { + snd_mychip_free(chip); + return err; + } + + snd_card_set_dev(card, &pci->dev); + + *rchip = chip; + return 0; + } + + /* constructor -- see "Constructor" sub-section */ + static int __devinit snd_mychip_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) + { + static int dev; + struct snd_card *card; + struct mychip *chip; + int err; + + /* (1) */ + if (dev >= SNDRV_CARDS) + return -ENODEV; + if (!enable[dev]) { + dev++; + return -ENOENT; + } + + /* (2) */ + err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); + if (err < 0) + return err; + + /* (3) */ + err = snd_mychip_create(card, pci, &chip); + if (err < 0) { + snd_card_free(card); + return err; + } + + /* (4) */ + strcpy(card->driver, "My Chip"); + strcpy(card->shortname, "My Own Chip 123"); + sprintf(card->longname, "%s at 0x%lx irq %i", + card->shortname, chip->ioport, chip->irq); + + /* (5) */ + .... /* implemented later */ + + /* (6) */ + err = snd_card_register(card); + if (err < 0) { + snd_card_free(card); + return err; + } + + /* (7) */ + pci_set_drvdata(pci, card); + dev++; + return 0; + } + + /* destructor -- see the "Destructor" sub-section */ + static void __devexit snd_mychip_remove(struct pci_dev *pci) + { + snd_card_free(pci_get_drvdata(pci)); + pci_set_drvdata(pci, NULL); + } +]]> + + + +
+ +
+ Constructor + + The real constructor of PCI drivers is the probe callback. + The probe callback and other component-constructors which are called + from the probe callback should be defined with + the __devinit prefix. You + cannot use the __init prefix for them, + because any PCI device could be a hotplug device. + + + + In the probe callback, the following scheme is often used. + + +
+ 1) Check and increment the device index. + + + += SNDRV_CARDS) + return -ENODEV; + if (!enable[dev]) { + dev++; + return -ENOENT; + } +]]> + + + + where enable[dev] is the module option. + + + + Each time the probe callback is called, check the + availability of the device. If not available, simply increment + the device index and returns. dev will be incremented also + later (step + 7). + +
+ +
+ 2) Create a card instance + + + + + + + + + + The details will be explained in the section + + Management of Cards and Components. + +
+ +
+ 3) Create a main component + + In this part, the PCI resources are allocated. + + + + + + + + The details will be explained in the section PCI Resource + Management. + +
+ +
+ 4) Set the driver ID and name strings. + + + +driver, "My Chip"); + strcpy(card->shortname, "My Own Chip 123"); + sprintf(card->longname, "%s at 0x%lx irq %i", + card->shortname, chip->ioport, chip->irq); +]]> + + + + The driver field holds the minimal ID string of the + chip. This is used by alsa-lib's configurator, so keep it + simple but unique. + Even the same driver can have different driver IDs to + distinguish the functionality of each chip type. + + + + The shortname field is a string shown as more verbose + name. The longname field contains the information + shown in /proc/asound/cards. + +
+ +
+ 5) Create other components, such as mixer, MIDI, etc. + + Here you define the basic components such as + PCM, + mixer (e.g. AC97), + MIDI (e.g. MPU-401), + and other interfaces. + Also, if you want a proc + file, define it here, too. + +
+ +
+ 6) Register the card instance. + + + + + + + + + + Will be explained in the section Management + of Cards and Components, too. + +
+ +
+ 7) Set the PCI driver data and return zero. + + + + + + + + In the above, the card record is stored. This pointer is + used in the remove callback and power-management + callbacks, too. + +
+
+ +
+ Destructor + + The destructor, remove callback, simply releases the card + instance. Then the ALSA middle layer will release all the + attached components automatically. + + + + It would be typically like the following: + + + + + + + + The above code assumes that the card pointer is set to the PCI + driver data. + +
+ +
+ Header Files + + For the above example, at least the following include files + are necessary. + + + + + #include + #include + #include + #include +]]> + + + + where the last one is necessary only when module options are + defined in the source file. If the code is split into several + files, the files without module options don't need them. + + + + In addition to these headers, you'll need + <linux/interrupt.h> for interrupt + handling, and <asm/io.h> for I/O + access. If you use the mdelay() or + udelay() functions, you'll need to include + <linux/delay.h> too. + + + + The ALSA interfaces like the PCM and control APIs are defined in other + <sound/xxx.h> header files. + They have to be included after + <sound/core.h>. + + +
+ + + + + + + + Management of Cards and Components + +
+ Card Instance + + For each soundcard, a card record must be allocated. + + + + A card record is the headquarters of the soundcard. It manages + the whole list of devices (components) on the soundcard, such as + PCM, mixers, MIDI, synthesizer, and so on. Also, the card + record holds the ID and the name strings of the card, manages + the root of proc files, and controls the power-management states + and hotplug disconnections. The component list on the card + record is used to manage the correct release of resources at + destruction. + + + + As mentioned above, to create a card instance, call + snd_card_create(). + + + + + + + + + + The function takes five arguments, the card-index number, the + id string, the module pointer (usually + THIS_MODULE), + the size of extra-data space, and the pointer to return the + card instance. The extra_size argument is used to + allocate card->private_data for the + chip-specific data. Note that these data + are allocated by snd_card_create(). + +
+ +
+ Components + + After the card is created, you can attach the components + (devices) to the card instance. In an ALSA driver, a component is + represented as a struct snd_device object. + A component can be a PCM instance, a control interface, a raw + MIDI interface, etc. Each such instance has one component + entry. + + + + A component can be created via + snd_device_new() function. + + + + + + + + + + This takes the card pointer, the device-level + (SNDRV_DEV_XXX), the data pointer, and the + callback pointers (&ops). The + device-level defines the type of components and the order of + registration and de-registration. For most components, the + device-level is already defined. For a user-defined component, + you can use SNDRV_DEV_LOWLEVEL. + + + + This function itself doesn't allocate the data space. The data + must be allocated manually beforehand, and its pointer is passed + as the argument. This pointer is used as the + (chip identifier in the above example) + for the instance. + + + + Each pre-defined ALSA component such as ac97 and pcm calls + snd_device_new() inside its + constructor. The destructor for each component is defined in the + callback pointers. Hence, you don't need to take care of + calling a destructor for such a component. + + + + If you wish to create your own component, you need to + set the destructor function to the dev_free callback in + the ops, so that it can be released + automatically via snd_card_free(). + The next example will show an implementation of chip-specific + data. + +
+ +
+ Chip-Specific Data + + Chip-specific information, e.g. the I/O port address, its + resource pointer, or the irq number, is stored in the + chip-specific record. + + + + + + + + + + In general, there are two ways of allocating the chip record. + + +
+ 1. Allocating via <function>snd_card_create()</function>. + + As mentioned above, you can pass the extra-data-length + to the 4th argument of snd_card_create(), i.e. + + + + + + + + struct mychip is the type of the chip record. + + + + In return, the allocated record can be accessed as + + + +private_data; +]]> + + + + With this method, you don't have to allocate twice. + The record is released together with the card instance. + +
+ +
+ 2. Allocating an extra device. + + + After allocating a card instance via + snd_card_create() (with + 0 on the 4th arg), call + kzalloc(). + + + + + + + + + + The chip record should have the field to hold the card + pointer at least, + + + + + + + + + + Then, set the card pointer in the returned chip instance. + + + +card = card; +]]> + + + + + + Next, initialize the fields, and register this chip + record as a low-level device with a specified + ops, + + + + + + + + snd_mychip_dev_free() is the + device-destructor function, which will call the real + destructor. + + + + + +device_data); + } +]]> + + + + where snd_mychip_free() is the real destructor. + +
+
+ +
+ Registration and Release + + After all components are assigned, register the card instance + by calling snd_card_register(). Access + to the device files is enabled at this point. That is, before + snd_card_register() is called, the + components are safely inaccessible from external side. If this + call fails, exit the probe function after releasing the card via + snd_card_free(). + + + + For releasing the card instance, you can call simply + snd_card_free(). As mentioned earlier, all + components are released automatically by this call. + + + + As further notes, the destructors (both + snd_mychip_dev_free and + snd_mychip_free) cannot be defined with + the __devexit prefix, because they may be + called from the constructor, too, at the false path. + + + + For a device which allows hotplugging, you can use + snd_card_free_when_closed. This one will + postpone the destruction until all devices are closed. + + +
+ + + + + + + + + PCI Resource Management + +
+ Full Code Example + + In this section, we'll complete the chip-specific constructor, + destructor and PCI entries. Example code is shown first, + below. + + + PCI Resource Management Example + +irq >= 0) + free_irq(chip->irq, chip); + /* release the I/O ports & memory */ + pci_release_regions(chip->pci); + /* disable the PCI entry */ + pci_disable_device(chip->pci); + /* release the data */ + kfree(chip); + return 0; + } + + /* chip-specific constructor */ + static int __devinit snd_mychip_create(struct snd_card *card, + struct pci_dev *pci, + struct mychip **rchip) + { + struct mychip *chip; + int err; + static struct snd_device_ops ops = { + .dev_free = snd_mychip_dev_free, + }; + + *rchip = NULL; + + /* initialize the PCI entry */ + err = pci_enable_device(pci); + if (err < 0) + return err; + /* check PCI availability (28bit DMA) */ + if (pci_set_dma_mask(pci, DMA_28BIT_MASK) < 0 || + pci_set_consistent_dma_mask(pci, DMA_28BIT_MASK) < 0) { + printk(KERN_ERR "error to set 28bit mask DMA\n"); + pci_disable_device(pci); + return -ENXIO; + } + + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + if (chip == NULL) { + pci_disable_device(pci); + return -ENOMEM; + } + + /* initialize the stuff */ + chip->card = card; + chip->pci = pci; + chip->irq = -1; + + /* (1) PCI resource allocation */ + err = pci_request_regions(pci, "My Chip"); + if (err < 0) { + kfree(chip); + pci_disable_device(pci); + return err; + } + chip->port = pci_resource_start(pci, 0); + if (request_irq(pci->irq, snd_mychip_interrupt, + IRQF_SHARED, "My Chip", chip)) { + printk(KERN_ERR "cannot grab irq %d\n", pci->irq); + snd_mychip_free(chip); + return -EBUSY; + } + chip->irq = pci->irq; + + /* (2) initialization of the chip hardware */ + .... /* (not implemented in this document) */ + + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) { + snd_mychip_free(chip); + return err; + } + + snd_card_set_dev(card, &pci->dev); + + *rchip = chip; + return 0; + } + + /* PCI IDs */ + static struct pci_device_id snd_mychip_ids[] = { + { PCI_VENDOR_ID_FOO, PCI_DEVICE_ID_BAR, + PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, + .... + { 0, } + }; + MODULE_DEVICE_TABLE(pci, snd_mychip_ids); + + /* pci_driver definition */ + static struct pci_driver driver = { + .name = "My Own Chip", + .id_table = snd_mychip_ids, + .probe = snd_mychip_probe, + .remove = __devexit_p(snd_mychip_remove), + }; + + /* module initialization */ + static int __init alsa_card_mychip_init(void) + { + return pci_register_driver(&driver); + } + + /* module clean up */ + static void __exit alsa_card_mychip_exit(void) + { + pci_unregister_driver(&driver); + } + + module_init(alsa_card_mychip_init) + module_exit(alsa_card_mychip_exit) + + EXPORT_NO_SYMBOLS; /* for old kernels only */ +]]> + + + +
+ +
+ Some Hafta's + + The allocation of PCI resources is done in the + probe() function, and usually an extra + xxx_create() function is written for this + purpose. + + + + In the case of PCI devices, you first have to call + the pci_enable_device() function before + allocating resources. Also, you need to set the proper PCI DMA + mask to limit the accessed I/O range. In some cases, you might + need to call pci_set_master() function, + too. + + + + Suppose the 28bit mask, and the code to be added would be like: + + + + + + + +
+ +
+ Resource Allocation + + The allocation of I/O ports and irqs is done via standard kernel + functions. Unlike ALSA ver.0.5.x., there are no helpers for + that. And these resources must be released in the destructor + function (see below). Also, on ALSA 0.9.x, you don't need to + allocate (pseudo-)DMA for PCI like in ALSA 0.5.x. + + + + Now assume that the PCI device has an I/O port with 8 bytes + and an interrupt. Then struct mychip will have the + following fields: + + + + + + + + + + For an I/O port (and also a memory region), you need to have + the resource pointer for the standard resource management. For + an irq, you have to keep only the irq number (integer). But you + need to initialize this number as -1 before actual allocation, + since irq 0 is valid. The port address and its resource pointer + can be initialized as null by + kzalloc() automatically, so you + don't have to take care of resetting them. + + + + The allocation of an I/O port is done like this: + + + +port = pci_resource_start(pci, 0); +]]> + + + + + + + It will reserve the I/O port region of 8 bytes of the given + PCI device. The returned value, chip->res_port, is allocated + via kmalloc() by + request_region(). The pointer must be + released via kfree(), but there is a + problem with this. This issue will be explained later. + + + + The allocation of an interrupt source is done like this: + + + +irq, snd_mychip_interrupt, + IRQF_SHARED, "My Chip", chip)) { + printk(KERN_ERR "cannot grab irq %d\n", pci->irq); + snd_mychip_free(chip); + return -EBUSY; + } + chip->irq = pci->irq; +]]> + + + + where snd_mychip_interrupt() is the + interrupt handler defined later. + Note that chip->irq should be defined + only when request_irq() succeeded. + + + + On the PCI bus, interrupts can be shared. Thus, + IRQF_SHARED is used as the interrupt flag of + request_irq(). + + + + The last argument of request_irq() is the + data pointer passed to the interrupt handler. Usually, the + chip-specific record is used for that, but you can use what you + like, too. + + + + I won't give details about the interrupt handler at this + point, but at least its appearance can be explained now. The + interrupt handler looks usually like the following: + + + + + + + + + + Now let's write the corresponding destructor for the resources + above. The role of destructor is simple: disable the hardware + (if already activated) and release the resources. So far, we + have no hardware part, so the disabling code is not written here. + + + + To release the resources, the check-and-release + method is a safer way. For the interrupt, do like this: + + + +irq >= 0) + free_irq(chip->irq, chip); +]]> + + + + Since the irq number can start from 0, you should initialize + chip->irq with a negative value (e.g. -1), so that you can + check the validity of the irq number as above. + + + + When you requested I/O ports or memory regions via + pci_request_region() or + pci_request_regions() like in this example, + release the resource(s) using the corresponding function, + pci_release_region() or + pci_release_regions(). + + + +pci); +]]> + + + + + + When you requested manually via request_region() + or request_mem_region, you can release it via + release_resource(). Suppose that you keep + the resource pointer returned from request_region() + in chip->res_port, the release procedure looks like: + + + +res_port); +]]> + + + + + + Don't forget to call pci_disable_device() + before the end. + + + + And finally, release the chip-specific record. + + + + + + + + + + Again, remember that you cannot + use the __devexit prefix for this destructor. + + + + We didn't implement the hardware disabling part in the above. + If you need to do this, please note that the destructor may be + called even before the initialization of the chip is completed. + It would be better to have a flag to skip hardware disabling + if the hardware was not initialized yet. + + + + When the chip-data is assigned to the card using + snd_device_new() with + SNDRV_DEV_LOWLELVEL , its destructor is + called at the last. That is, it is assured that all other + components like PCMs and controls have already been released. + You don't have to stop PCMs, etc. explicitly, but just + call low-level hardware stopping. + + + + The management of a memory-mapped region is almost as same as + the management of an I/O port. You'll need three fields like + the following: + + + + + + + + and the allocation would be like below: + + + +iobase_phys = pci_resource_start(pci, 0); + chip->iobase_virt = ioremap_nocache(chip->iobase_phys, + pci_resource_len(pci, 0)); +]]> + + + + and the corresponding destructor would be: + + + +iobase_virt) + iounmap(chip->iobase_virt); + .... + pci_release_regions(chip->pci); + .... + } +]]> + + + + +
+ +
+ Registration of Device Struct + + At some point, typically after calling snd_device_new(), + you need to register the struct device of the chip + you're handling for udev and co. ALSA provides a macro for compatibility with + older kernels. Simply call like the following: + + +dev); +]]> + + + so that it stores the PCI's device pointer to the card. This will be + referred by ALSA core functions later when the devices are registered. + + + In the case of non-PCI, pass the proper device struct pointer of the BUS + instead. (In the case of legacy ISA without PnP, you don't have to do + anything.) + +
+ +
+ PCI Entries + + So far, so good. Let's finish the missing PCI + stuff. At first, we need a + pci_device_id table for this + chipset. It's a table of PCI vendor/device ID number, and some + masks. + + + + For example, + + + + + + + + + + The first and second fields of + the pci_device_id structure are the vendor and + device IDs. If you have no reason to filter the matching + devices, you can leave the remaining fields as above. The last + field of the pci_device_id struct contains + private data for this entry. You can specify any value here, for + example, to define specific operations for supported device IDs. + Such an example is found in the intel8x0 driver. + + + + The last entry of this list is the terminator. You must + specify this all-zero entry. + + + + Then, prepare the pci_driver record: + + + + + + + + + + The probe and + remove functions have already + been defined in the previous sections. + The remove function should + be defined with the + __devexit_p() macro, so that it's not + defined for built-in (and non-hot-pluggable) case. The + name + field is the name string of this device. Note that you must not + use a slash / in this string. + + + + And at last, the module entries: + + + + + + + + + + Note that these module entries are tagged with + __init and + __exit prefixes, not + __devinit nor + __devexit. + + + + Oh, one thing was forgotten. If you have no exported symbols, + you need to declare it in 2.2 or 2.4 kernels (it's not necessary in 2.6 kernels). + + + + + + + + That's all! + +
+ + + + + + + + PCM Interface + +
+ General + + The PCM middle layer of ALSA is quite powerful and it is only + necessary for each driver to implement the low-level functions + to access its hardware. + + + + For accessing to the PCM layer, you need to include + <sound/pcm.h> first. In addition, + <sound/pcm_params.h> might be needed + if you access to some functions related with hw_param. + + + + Each card device can have up to four pcm instances. A pcm + instance corresponds to a pcm device file. The limitation of + number of instances comes only from the available bit size of + the Linux's device numbers. Once when 64bit device number is + used, we'll have more pcm instances available. + + + + A pcm instance consists of pcm playback and capture streams, + and each pcm stream consists of one or more pcm substreams. Some + soundcards support multiple playback functions. For example, + emu10k1 has a PCM playback of 32 stereo substreams. In this case, at + each open, a free substream is (usually) automatically chosen + and opened. Meanwhile, when only one substream exists and it was + already opened, the successful open will either block + or error with EAGAIN according to the + file open mode. But you don't have to care about such details in your + driver. The PCM middle layer will take care of such work. + +
+ +
+ Full Code Example + + The example code below does not include any hardware access + routines but shows only the skeleton, how to build up the PCM + interfaces. + + + PCM Example Code + + + .... + + /* hardware definition */ + static struct snd_pcm_hardware snd_mychip_playback_hw = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID), + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rates = SNDRV_PCM_RATE_8000_48000, + .rate_min = 8000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 32768, + .period_bytes_min = 4096, + .period_bytes_max = 32768, + .periods_min = 1, + .periods_max = 1024, + }; + + /* hardware definition */ + static struct snd_pcm_hardware snd_mychip_capture_hw = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID), + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rates = SNDRV_PCM_RATE_8000_48000, + .rate_min = 8000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 32768, + .period_bytes_min = 4096, + .period_bytes_max = 32768, + .periods_min = 1, + .periods_max = 1024, + }; + + /* open callback */ + static int snd_mychip_playback_open(struct snd_pcm_substream *substream) + { + struct mychip *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw = snd_mychip_playback_hw; + /* more hardware-initialization will be done here */ + .... + return 0; + } + + /* close callback */ + static int snd_mychip_playback_close(struct snd_pcm_substream *substream) + { + struct mychip *chip = snd_pcm_substream_chip(substream); + /* the hardware-specific codes will be here */ + .... + return 0; + + } + + /* open callback */ + static int snd_mychip_capture_open(struct snd_pcm_substream *substream) + { + struct mychip *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw = snd_mychip_capture_hw; + /* more hardware-initialization will be done here */ + .... + return 0; + } + + /* close callback */ + static int snd_mychip_capture_close(struct snd_pcm_substream *substream) + { + struct mychip *chip = snd_pcm_substream_chip(substream); + /* the hardware-specific codes will be here */ + .... + return 0; + + } + + /* hw_params callback */ + static int snd_mychip_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) + { + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + } + + /* hw_free callback */ + static int snd_mychip_pcm_hw_free(struct snd_pcm_substream *substream) + { + return snd_pcm_lib_free_pages(substream); + } + + /* prepare callback */ + static int snd_mychip_pcm_prepare(struct snd_pcm_substream *substream) + { + struct mychip *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + /* set up the hardware with the current configuration + * for example... + */ + mychip_set_sample_format(chip, runtime->format); + mychip_set_sample_rate(chip, runtime->rate); + mychip_set_channels(chip, runtime->channels); + mychip_set_dma_setup(chip, runtime->dma_addr, + chip->buffer_size, + chip->period_size); + return 0; + } + + /* trigger callback */ + static int snd_mychip_pcm_trigger(struct snd_pcm_substream *substream, + int cmd) + { + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + /* do something to start the PCM engine */ + .... + break; + case SNDRV_PCM_TRIGGER_STOP: + /* do something to stop the PCM engine */ + .... + break; + default: + return -EINVAL; + } + } + + /* pointer callback */ + static snd_pcm_uframes_t + snd_mychip_pcm_pointer(struct snd_pcm_substream *substream) + { + struct mychip *chip = snd_pcm_substream_chip(substream); + unsigned int current_ptr; + + /* get the current hardware pointer */ + current_ptr = mychip_get_hw_pointer(chip); + return current_ptr; + } + + /* operators */ + static struct snd_pcm_ops snd_mychip_playback_ops = { + .open = snd_mychip_playback_open, + .close = snd_mychip_playback_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_mychip_pcm_hw_params, + .hw_free = snd_mychip_pcm_hw_free, + .prepare = snd_mychip_pcm_prepare, + .trigger = snd_mychip_pcm_trigger, + .pointer = snd_mychip_pcm_pointer, + }; + + /* operators */ + static struct snd_pcm_ops snd_mychip_capture_ops = { + .open = snd_mychip_capture_open, + .close = snd_mychip_capture_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_mychip_pcm_hw_params, + .hw_free = snd_mychip_pcm_hw_free, + .prepare = snd_mychip_pcm_prepare, + .trigger = snd_mychip_pcm_trigger, + .pointer = snd_mychip_pcm_pointer, + }; + + /* + * definitions of capture are omitted here... + */ + + /* create a pcm device */ + static int __devinit snd_mychip_new_pcm(struct mychip *chip) + { + struct snd_pcm *pcm; + int err; + + err = snd_pcm_new(chip->card, "My Chip", 0, 1, 1, &pcm); + if (err < 0) + return err; + pcm->private_data = chip; + strcpy(pcm->name, "My Chip"); + chip->pcm = pcm; + /* set operators */ + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &snd_mychip_playback_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, + &snd_mychip_capture_ops); + /* pre-allocation of buffers */ + /* NOTE: this may fail */ + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(chip->pci), + 64*1024, 64*1024); + return 0; + } +]]> + + + +
+ +
+ Constructor + + A pcm instance is allocated by the snd_pcm_new() + function. It would be better to create a constructor for pcm, + namely, + + + +card, "My Chip", 0, 1, 1, &pcm); + if (err < 0) + return err; + pcm->private_data = chip; + strcpy(pcm->name, "My Chip"); + chip->pcm = pcm; + .... + return 0; + } +]]> + + + + + + The snd_pcm_new() function takes four + arguments. The first argument is the card pointer to which this + pcm is assigned, and the second is the ID string. + + + + The third argument (index, 0 in the + above) is the index of this new pcm. It begins from zero. If + you create more than one pcm instances, specify the + different numbers in this argument. For example, + index = 1 for the second PCM device. + + + + The fourth and fifth arguments are the number of substreams + for playback and capture, respectively. Here 1 is used for + both arguments. When no playback or capture substreams are available, + pass 0 to the corresponding argument. + + + + If a chip supports multiple playbacks or captures, you can + specify more numbers, but they must be handled properly in + open/close, etc. callbacks. When you need to know which + substream you are referring to, then it can be obtained from + struct snd_pcm_substream data passed to each callback + as follows: + + + +number; +]]> + + + + + + After the pcm is created, you need to set operators for each + pcm stream. + + + + + + + + + + The operators are defined typically like this: + + + + + + + + All the callbacks are described in the + + Operators subsection. + + + + After setting the operators, you probably will want to + pre-allocate the buffer. For the pre-allocation, simply call + the following: + + + +pci), + 64*1024, 64*1024); +]]> + + + + It will allocate a buffer up to 64kB as default. + Buffer management details will be described in the later section Buffer and Memory + Management. + + + + Additionally, you can set some extra information for this pcm + in pcm->info_flags. + The available values are defined as + SNDRV_PCM_INFO_XXX in + <sound/asound.h>, which is used for + the hardware definition (described later). When your soundchip + supports only half-duplex, specify like this: + + + +info_flags = SNDRV_PCM_INFO_HALF_DUPLEX; +]]> + + + +
+ +
+ ... And the Destructor? + + The destructor for a pcm instance is not always + necessary. Since the pcm device will be released by the middle + layer code automatically, you don't have to call the destructor + explicitly. + + + + The destructor would be necessary if you created + special records internally and needed to release them. In such a + case, set the destructor function to + pcm->private_free: + + + PCM Instance with a Destructor + +my_private_pcm_data); + /* do what you like else */ + .... + } + + static int __devinit snd_mychip_new_pcm(struct mychip *chip) + { + struct snd_pcm *pcm; + .... + /* allocate your own data */ + chip->my_private_pcm_data = kmalloc(...); + /* set the destructor */ + pcm->private_data = chip; + pcm->private_free = mychip_pcm_free; + .... + } +]]> + + + +
+ +
+ Runtime Pointer - The Chest of PCM Information + + When the PCM substream is opened, a PCM runtime instance is + allocated and assigned to the substream. This pointer is + accessible via substream->runtime. + This runtime pointer holds most information you need + to control the PCM: the copy of hw_params and sw_params configurations, the buffer + pointers, mmap records, spinlocks, etc. + + + + The definition of runtime instance is found in + <sound/pcm.h>. Here are + the contents of this file: + + + + + + + + + For the operators (callbacks) of each sound driver, most of + these records are supposed to be read-only. Only the PCM + middle-layer changes / updates them. The exceptions are + the hardware description (hw), interrupt callbacks + (transfer_ack_xxx), DMA buffer information, and the private + data. Besides, if you use the standard buffer allocation + method via snd_pcm_lib_malloc_pages(), + you don't need to set the DMA buffer information by yourself. + + + + In the sections below, important records are explained. + + +
+ Hardware Description + + The hardware descriptor (struct snd_pcm_hardware) + contains the definitions of the fundamental hardware + configuration. Above all, you'll need to define this in + + the open callback. + Note that the runtime instance holds the copy of the + descriptor, not the pointer to the existing descriptor. That + is, in the open callback, you can modify the copied descriptor + (runtime->hw) as you need. For example, if the maximum + number of channels is 1 only on some chip models, you can + still use the same hardware descriptor and change the + channels_max later: + + +runtime; + ... + runtime->hw = snd_mychip_playback_hw; /* common definition */ + if (chip->model == VERY_OLD_ONE) + runtime->hw.channels_max = 1; +]]> + + + + + + Typically, you'll have a hardware descriptor as below: + + + + + + + + + + + The info field contains the type and + capabilities of this pcm. The bit flags are defined in + <sound/asound.h> as + SNDRV_PCM_INFO_XXX. Here, at least, you + have to specify whether the mmap is supported and which + interleaved format is supported. + When the is supported, add the + SNDRV_PCM_INFO_MMAP flag here. When the + hardware supports the interleaved or the non-interleaved + formats, SNDRV_PCM_INFO_INTERLEAVED or + SNDRV_PCM_INFO_NONINTERLEAVED flag must + be set, respectively. If both are supported, you can set both, + too. + + + + In the above example, MMAP_VALID and + BLOCK_TRANSFER are specified for the OSS mmap + mode. Usually both are set. Of course, + MMAP_VALID is set only if the mmap is + really supported. + + + + The other possible flags are + SNDRV_PCM_INFO_PAUSE and + SNDRV_PCM_INFO_RESUME. The + PAUSE bit means that the pcm supports the + pause operation, while the + RESUME bit means that the pcm supports + the full suspend/resume operation. + If the PAUSE flag is set, + the trigger callback below + must handle the corresponding (pause push/release) commands. + The suspend/resume trigger commands can be defined even without + the RESUME flag. See + Power Management section for details. + + + + When the PCM substreams can be synchronized (typically, + synchronized start/stop of a playback and a capture streams), + you can give SNDRV_PCM_INFO_SYNC_START, + too. In this case, you'll need to check the linked-list of + PCM substreams in the trigger callback. This will be + described in the later section. + + + + + + formats field contains the bit-flags + of supported formats (SNDRV_PCM_FMTBIT_XXX). + If the hardware supports more than one format, give all or'ed + bits. In the example above, the signed 16bit little-endian + format is specified. + + + + + + rates field contains the bit-flags of + supported rates (SNDRV_PCM_RATE_XXX). + When the chip supports continuous rates, pass + CONTINUOUS bit additionally. + The pre-defined rate bits are provided only for typical + rates. If your chip supports unconventional rates, you need to add + the KNOT bit and set up the hardware + constraint manually (explained later). + + + + + + rate_min and + rate_max define the minimum and + maximum sample rate. This should correspond somehow to + rates bits. + + + + + + channel_min and + channel_max + define, as you might already expected, the minimum and maximum + number of channels. + + + + + + buffer_bytes_max defines the + maximum buffer size in bytes. There is no + buffer_bytes_min field, since + it can be calculated from the minimum period size and the + minimum number of periods. + Meanwhile, period_bytes_min and + define the minimum and maximum size of the period in bytes. + periods_max and + periods_min define the maximum and + minimum number of periods in the buffer. + + + + The period is a term that corresponds to + a fragment in the OSS world. The period defines the size at + which a PCM interrupt is generated. This size strongly + depends on the hardware. + Generally, the smaller period size will give you more + interrupts, that is, more controls. + In the case of capture, this size defines the input latency. + On the other hand, the whole buffer size defines the + output latency for the playback direction. + + + + + + There is also a field fifo_size. + This specifies the size of the hardware FIFO, but currently it + is neither used in the driver nor in the alsa-lib. So, you + can ignore this field. + + + + +
+ +
+ PCM Configurations + + Ok, let's go back again to the PCM runtime records. + The most frequently referred records in the runtime instance are + the PCM configurations. + The PCM configurations are stored in the runtime instance + after the application sends hw_params data via + alsa-lib. There are many fields copied from hw_params and + sw_params structs. For example, + format holds the format type + chosen by the application. This field contains the enum value + SNDRV_PCM_FORMAT_XXX. + + + + One thing to be noted is that the configured buffer and period + sizes are stored in frames in the runtime. + In the ALSA world, 1 frame = channels * samples-size. + For conversion between frames and bytes, you can use the + frames_to_bytes() and + bytes_to_frames() helper functions. + + +period_size); +]]> + + + + + + Also, many software parameters (sw_params) are + stored in frames, too. Please check the type of the field. + snd_pcm_uframes_t is for the frames as unsigned + integer while snd_pcm_sframes_t is for the frames + as signed integer. + +
+ +
+ DMA Buffer Information + + The DMA buffer is defined by the following four fields, + dma_area, + dma_addr, + dma_bytes and + dma_private. + The dma_area holds the buffer + pointer (the logical address). You can call + memcpy from/to + this pointer. Meanwhile, dma_addr + holds the physical address of the buffer. This field is + specified only when the buffer is a linear buffer. + dma_bytes holds the size of buffer + in bytes. dma_private is used for + the ALSA DMA allocator. + + + + If you use a standard ALSA function, + snd_pcm_lib_malloc_pages(), for + allocating the buffer, these fields are set by the ALSA middle + layer, and you should not change them by + yourself. You can read them but not write them. + On the other hand, if you want to allocate the buffer by + yourself, you'll need to manage it in hw_params callback. + At least, dma_bytes is mandatory. + dma_area is necessary when the + buffer is mmapped. If your driver doesn't support mmap, this + field is not necessary. dma_addr + is also optional. You can use + dma_private as you like, too. + +
+ +
+ Running Status + + The running status can be referred via runtime->status. + This is the pointer to the struct snd_pcm_mmap_status + record. For example, you can get the current DMA hardware + pointer via runtime->status->hw_ptr. + + + + The DMA application pointer can be referred via + runtime->control, which points to the + struct snd_pcm_mmap_control record. + However, accessing directly to this value is not recommended. + +
+ +
+ Private Data + + You can allocate a record for the substream and store it in + runtime->private_data. Usually, this + is done in + + the open callback. + Don't mix this with pcm->private_data. + The pcm->private_data usually points to the + chip instance assigned statically at the creation of PCM, while the + runtime->private_data points to a dynamic + data structure created at the PCM open callback. + + + +runtime->private_data = data; + .... + } +]]> + + + + + + The allocated object must be released in + + the close callback. + +
+ +
+ Interrupt Callbacks + + The field transfer_ack_begin and + transfer_ack_end are called at + the beginning and at the end of + snd_pcm_period_elapsed(), respectively. + +
+ +
+ +
+ Operators + + OK, now let me give details about each pcm callback + (ops). In general, every callback must + return 0 if successful, or a negative error number + such as -EINVAL. To choose an appropriate + error number, it is advised to check what value other parts of + the kernel return when the same kind of request fails. + + + + The callback function takes at least the argument with + snd_pcm_substream pointer. To retrieve + the chip record from the given substream instance, you can use the + following macro. + + + + + + + + The macro reads substream->private_data, + which is a copy of pcm->private_data. + You can override the former if you need to assign different data + records per PCM substream. For example, the cmi8330 driver assigns + different private_data for playback and capture directions, + because it uses two different codecs (SB- and AD-compatible) for + different directions. + + +
+ open callback + + + + + + + + This is called when a pcm substream is opened. + + + + At least, here you have to initialize the runtime->hw + record. Typically, this is done by like this: + + + +runtime; + + runtime->hw = snd_mychip_playback_hw; + return 0; + } +]]> + + + + where snd_mychip_playback_hw is the + pre-defined hardware description. + + + + You can allocate a private data in this callback, as described + in + Private Data section. + + + + If the hardware configuration needs more constraints, set the + hardware constraints here, too. + See + Constraints for more details. + +
+ +
+ close callback + + + + + + + + Obviously, this is called when a pcm substream is closed. + + + + Any private instance for a pcm substream allocated in the + open callback will be released here. + + + +runtime->private_data); + .... + } +]]> + + + +
+ +
+ ioctl callback + + This is used for any special call to pcm ioctls. But + usually you can pass a generic ioctl callback, + snd_pcm_lib_ioctl. + +
+ +
+ hw_params callback + + + + + + + + + + This is called when the hardware parameter + (hw_params) is set + up by the application, + that is, once when the buffer size, the period size, the + format, etc. are defined for the pcm substream. + + + + Many hardware setups should be done in this callback, + including the allocation of buffers. + + + + Parameters to be initialized are retrieved by + params_xxx() macros. To allocate + buffer, you can call a helper function, + + + + + + + + snd_pcm_lib_malloc_pages() is available + only when the DMA buffers have been pre-allocated. + See the section + Buffer Types for more details. + + + + Note that this and prepare callbacks + may be called multiple times per initialization. + For example, the OSS emulation may + call these callbacks at each change via its ioctl. + + + + Thus, you need to be careful not to allocate the same buffers + many times, which will lead to memory leaks! Calling the + helper function above many times is OK. It will release the + previous buffer automatically when it was already allocated. + + + + Another note is that this callback is non-atomic + (schedulable). This is important, because the + trigger callback + is atomic (non-schedulable). That is, mutexes or any + schedule-related functions are not available in + trigger callback. + Please see the subsection + + Atomicity for details. + +
+ +
+ hw_free callback + + + + + + + + + + This is called to release the resources allocated via + hw_params. For example, releasing the + buffer via + snd_pcm_lib_malloc_pages() is done by + calling the following: + + + + + + + + + + This function is always called before the close callback is called. + Also, the callback may be called multiple times, too. + Keep track whether the resource was already released. + +
+ +
+ prepare callback + + + + + + + + + + This callback is called when the pcm is + prepared. You can set the format type, sample + rate, etc. here. The difference from + hw_params is that the + prepare callback will be called each + time + snd_pcm_prepare() is called, i.e. when + recovering after underruns, etc. + + + + Note that this callback is now non-atomic. + You can use schedule-related functions safely in this callback. + + + + In this and the following callbacks, you can refer to the + values via the runtime record, + substream->runtime. + For example, to get the current + rate, format or channels, access to + runtime->rate, + runtime->format or + runtime->channels, respectively. + The physical address of the allocated buffer is set to + runtime->dma_area. The buffer and period sizes are + in runtime->buffer_size and runtime->period_size, + respectively. + + + + Be careful that this callback will be called many times at + each setup, too. + +
+ +
+ trigger callback + + + + + + + + This is called when the pcm is started, stopped or paused. + + + + Which action is specified in the second argument, + SNDRV_PCM_TRIGGER_XXX in + <sound/pcm.h>. At least, + the START and STOP + commands must be defined in this callback. + + + + + + + + + + When the pcm supports the pause operation (given in the info + field of the hardware table), the PAUSE_PUSE + and PAUSE_RELEASE commands must be + handled here, too. The former is the command to pause the pcm, + and the latter to restart the pcm again. + + + + When the pcm supports the suspend/resume operation, + regardless of full or partial suspend/resume support, + the SUSPEND and RESUME + commands must be handled, too. + These commands are issued when the power-management status is + changed. Obviously, the SUSPEND and + RESUME commands + suspend and resume the pcm substream, and usually, they + are identical to the STOP and + START commands, respectively. + See the + Power Management section for details. + + + + As mentioned, this callback is atomic. You cannot call + functions which may sleep. + The trigger callback should be as minimal as possible, + just really triggering the DMA. The other stuff should be + initialized hw_params and prepare callbacks properly + beforehand. + +
+ +
+ pointer callback + + + + + + + + This callback is called when the PCM middle layer inquires + the current hardware position on the buffer. The position must + be returned in frames, + ranging from 0 to buffer_size - 1. + + + + This is called usually from the buffer-update routine in the + pcm middle layer, which is invoked when + snd_pcm_period_elapsed() is called in the + interrupt routine. Then the pcm middle layer updates the + position and calculates the available space, and wakes up the + sleeping poll threads, etc. + + + + This callback is also atomic. + +
+ +
+ copy and silence callbacks + + These callbacks are not mandatory, and can be omitted in + most cases. These callbacks are used when the hardware buffer + cannot be in the normal memory space. Some chips have their + own buffer on the hardware which is not mappable. In such a + case, you have to transfer the data manually from the memory + buffer to the hardware buffer. Or, if the buffer is + non-contiguous on both physical and virtual memory spaces, + these callbacks must be defined, too. + + + + If these two callbacks are defined, copy and set-silence + operations are done by them. The detailed will be described in + the later section Buffer and Memory + Management. + +
+ +
+ ack callback + + This callback is also not mandatory. This callback is called + when the appl_ptr is updated in read or write operations. + Some drivers like emu10k1-fx and cs46xx need to track the + current appl_ptr for the internal buffer, and this callback + is useful only for such a purpose. + + + This callback is atomic. + +
+ +
+ page callback + + + This callback is optional too. This callback is used + mainly for non-contiguous buffers. The mmap calls this + callback to get the page address. Some examples will be + explained in the later section Buffer and Memory + Management, too. + +
+
+ +
+ Interrupt Handler + + The rest of pcm stuff is the PCM interrupt handler. The + role of PCM interrupt handler in the sound driver is to update + the buffer position and to tell the PCM middle layer when the + buffer position goes across the prescribed period size. To + inform this, call the snd_pcm_period_elapsed() + function. + + + + There are several types of sound chips to generate the interrupts. + + +
+ Interrupts at the period (fragment) boundary + + This is the most frequently found type: the hardware + generates an interrupt at each period boundary. + In this case, you can call + snd_pcm_period_elapsed() at each + interrupt. + + + + snd_pcm_period_elapsed() takes the + substream pointer as its argument. Thus, you need to keep the + substream pointer accessible from the chip instance. For + example, define substream field in the chip record to hold the + current running substream pointer, and set the pointer value + at open callback (and reset at close callback). + + + + If you acquire a spinlock in the interrupt handler, and the + lock is used in other pcm callbacks, too, then you have to + release the lock before calling + snd_pcm_period_elapsed(), because + snd_pcm_period_elapsed() calls other pcm + callbacks inside. + + + + Typical code would be like: + + + Interrupt Handler Case #1 + +lock); + .... + if (pcm_irq_invoked(chip)) { + /* call updater, unlock before it */ + spin_unlock(&chip->lock); + snd_pcm_period_elapsed(chip->substream); + spin_lock(&chip->lock); + /* acknowledge the interrupt if necessary */ + } + .... + spin_unlock(&chip->lock); + return IRQ_HANDLED; + } +]]> + + + +
+ +
+ High frequency timer interrupts + + This happense when the hardware doesn't generate interrupts + at the period boundary but issues timer interrupts at a fixed + timer rate (e.g. es1968 or ymfpci drivers). + In this case, you need to check the current hardware + position and accumulate the processed sample length at each + interrupt. When the accumulated size exceeds the period + size, call + snd_pcm_period_elapsed() and reset the + accumulator. + + + + Typical code would be like the following. + + + Interrupt Handler Case #2 + +lock); + .... + if (pcm_irq_invoked(chip)) { + unsigned int last_ptr, size; + /* get the current hardware pointer (in frames) */ + last_ptr = get_hw_ptr(chip); + /* calculate the processed frames since the + * last update + */ + if (last_ptr < chip->last_ptr) + size = runtime->buffer_size + last_ptr + - chip->last_ptr; + else + size = last_ptr - chip->last_ptr; + /* remember the last updated point */ + chip->last_ptr = last_ptr; + /* accumulate the size */ + chip->size += size; + /* over the period boundary? */ + if (chip->size >= runtime->period_size) { + /* reset the accumulator */ + chip->size %= runtime->period_size; + /* call updater */ + spin_unlock(&chip->lock); + snd_pcm_period_elapsed(substream); + spin_lock(&chip->lock); + } + /* acknowledge the interrupt if necessary */ + } + .... + spin_unlock(&chip->lock); + return IRQ_HANDLED; + } +]]> + + + +
+ +
+ On calling <function>snd_pcm_period_elapsed()</function> + + In both cases, even if more than one period are elapsed, you + don't have to call + snd_pcm_period_elapsed() many times. Call + only once. And the pcm layer will check the current hardware + pointer and update to the latest status. + +
+
+ +
+ Atomicity + + One of the most important (and thus difficult to debug) problems + in kernel programming are race conditions. + In the Linux kernel, they are usually avoided via spin-locks, mutexes + or semaphores. In general, if a race condition can happen + in an interrupt handler, it has to be managed atomically, and you + have to use a spinlock to protect the critical session. If the + critical section is not in interrupt handler code and + if taking a relatively long time to execute is acceptable, you + should use mutexes or semaphores instead. + + + + As already seen, some pcm callbacks are atomic and some are + not. For example, the hw_params callback is + non-atomic, while trigger callback is + atomic. This means, the latter is called already in a spinlock + held by the PCM middle layer. Please take this atomicity into + account when you choose a locking scheme in the callbacks. + + + + In the atomic callbacks, you cannot use functions which may call + schedule or go to + sleep. Semaphores and mutexes can sleep, + and hence they cannot be used inside the atomic callbacks + (e.g. trigger callback). + To implement some delay in such a callback, please use + udelay() or mdelay(). + + + + All three atomic callbacks (trigger, pointer, and ack) are + called with local interrupts disabled. + + +
+
+ Constraints + + If your chip supports unconventional sample rates, or only the + limited samples, you need to set a constraint for the + condition. + + + + For example, in order to restrict the sample rates in the some + supported values, use + snd_pcm_hw_constraint_list(). + You need to call this function in the open callback. + + + Example of Hardware Constraints + +runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + if (err < 0) + return err; + .... + } +]]> + + + + + + There are many different constraints. + Look at sound/pcm.h for a complete list. + You can even define your own constraint rules. + For example, let's suppose my_chip can manage a substream of 1 channel + if and only if the format is S16_LE, otherwise it supports any format + specified in the snd_pcm_hardware structure (or in any + other constraint_list). You can build a rule like this: + + + Example of Hardware Constraints for Channels + +min < 2) { + fmt.bits[0] &= SNDRV_PCM_FMTBIT_S16_LE; + return snd_mask_refine(f, &fmt); + } + return 0; + } +]]> + + + + + + Then you need to call this function to add your rule: + + + +runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + hw_rule_channels_by_format, 0, SNDRV_PCM_HW_PARAM_FORMAT, + -1); +]]> + + + + + + The rule function is called when an application sets the number of + channels. But an application can set the format before the number of + channels. Thus you also need to define the inverse rule: + + + Example of Hardware Constraints for Channels + +bits[0] == SNDRV_PCM_FMTBIT_S16_LE) { + ch.min = ch.max = 1; + ch.integer = 1; + return snd_interval_refine(c, &ch); + } + return 0; + } +]]> + + + + + + ...and in the open callback: + + +runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, + hw_rule_format_by_channels, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + -1); +]]> + + + + + + I won't give more details here, rather I + would like to say, Luke, use the source. + +
+ + + + + + + + + Control Interface + +
+ General + + The control interface is used widely for many switches, + sliders, etc. which are accessed from user-space. Its most + important use is the mixer interface. In other words, since ALSA + 0.9.x, all the mixer stuff is implemented on the control kernel API. + + + + ALSA has a well-defined AC97 control module. If your chip + supports only the AC97 and nothing else, you can skip this + section. + + + + The control API is defined in + <sound/control.h>. + Include this file if you want to add your own controls. + +
+ +
+ Definition of Controls + + To create a new control, you need to define the + following three + callbacks: info, + get and + put. Then, define a + struct snd_kcontrol_new record, such as: + + + Definition of a Control + + + + + + + + Most likely the control is created via + snd_ctl_new1(), and in such a case, you can + add the __devinitdata prefix to the + definition as above. + + + + The iface field specifies the control + type, SNDRV_CTL_ELEM_IFACE_XXX, which + is usually MIXER. + Use CARD for global controls that are not + logically part of the mixer. + If the control is closely associated with some specific device on + the sound card, use HWDEP, + PCM, RAWMIDI, + TIMER, or SEQUENCER, and + specify the device number with the + device and + subdevice fields. + + + + The name is the name identifier + string. Since ALSA 0.9.x, the control name is very important, + because its role is classified from its name. There are + pre-defined standard control names. The details are described in + the + Control Names subsection. + + + + The index field holds the index number + of this control. If there are several different controls with + the same name, they can be distinguished by the index + number. This is the case when + several codecs exist on the card. If the index is zero, you can + omit the definition above. + + + + The access field contains the access + type of this control. Give the combination of bit masks, + SNDRV_CTL_ELEM_ACCESS_XXX, there. + The details will be explained in + the + Access Flags subsection. + + + + The private_value field contains + an arbitrary long integer value for this record. When using + the generic info, + get and + put callbacks, you can pass a value + through this field. If several small numbers are necessary, you can + combine them in bitwise. Or, it's possible to give a pointer + (casted to unsigned long) of some record to this field, too. + + + + The tlv field can be used to provide + metadata about the control; see the + + Metadata subsection. + + + + The other three are + + callback functions. + +
+ +
+ Control Names + + There are some standards to define the control names. A + control is usually defined from the three parts as + SOURCE DIRECTION FUNCTION. + + + + The first, SOURCE, specifies the source + of the control, and is a string such as Master, + PCM, CD and + Line. There are many pre-defined sources. + + + + The second, DIRECTION, is one of the + following strings according to the direction of the control: + Playback, Capture, Bypass + Playback and Bypass Capture. Or, it can + be omitted, meaning both playback and capture directions. + + + + The third, FUNCTION, is one of the + following strings according to the function of the control: + Switch, Volume and + Route. + + + + The example of control names are, thus, Master Capture + Switch or PCM Playback Volume. + + + + There are some exceptions: + + +
+ Global capture and playback + + Capture Source, Capture Switch + and Capture Volume are used for the global + capture (input) source, switch and volume. Similarly, + Playback Switch and Playback + Volume are used for the global output gain switch and + volume. + +
+ +
+ Tone-controls + + tone-control switch and volumes are specified like + Tone Control - XXX, e.g. Tone Control - + Switch, Tone Control - Bass, + Tone Control - Center. + +
+ +
+ 3D controls + + 3D-control switches and volumes are specified like 3D + Control - XXX, e.g. 3D Control - + Switch, 3D Control - Center, 3D + Control - Space. + +
+ +
+ Mic boost + + Mic-boost switch is set as Mic Boost or + Mic Boost (6dB). + + + + More precise information can be found in + Documentation/sound/alsa/ControlNames.txt. + +
+
+ +
+ Access Flags + + + The access flag is the bitmask which specifies the access type + of the given control. The default access type is + SNDRV_CTL_ELEM_ACCESS_READWRITE, + which means both read and write are allowed to this control. + When the access flag is omitted (i.e. = 0), it is + considered as READWRITE access as default. + + + + When the control is read-only, pass + SNDRV_CTL_ELEM_ACCESS_READ instead. + In this case, you don't have to define + the put callback. + Similarly, when the control is write-only (although it's a rare + case), you can use the WRITE flag instead, and + you don't need the get callback. + + + + If the control value changes frequently (e.g. the VU meter), + VOLATILE flag should be given. This means + that the control may be changed without + + notification. Applications should poll such + a control constantly. + + + + When the control is inactive, set + the INACTIVE flag, too. + There are LOCK and + OWNER flags to change the write + permissions. + + +
+ +
+ Callbacks + +
+ info callback + + The info callback is used to get + detailed information on this control. This must store the + values of the given struct snd_ctl_elem_info + object. For example, for a boolean control with a single + element: + + + Example of info callback + +type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; + } +]]> + + + + + + The type field specifies the type + of the control. There are BOOLEAN, + INTEGER, ENUMERATED, + BYTES, IEC958 and + INTEGER64. The + count field specifies the + number of elements in this control. For example, a stereo + volume would have count = 2. The + value field is a union, and + the values stored are depending on the type. The boolean and + integer types are identical. + + + + The enumerated type is a bit different from others. You'll + need to set the string for the currently given item index. + + + +type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 4; + if (uinfo->value.enumerated.item > 3) + uinfo->value.enumerated.item = 3; + strcpy(uinfo->value.enumerated.name, + texts[uinfo->value.enumerated.item]); + return 0; + } +]]> + + + + + + Some common info callbacks are available for your convenience: + snd_ctl_boolean_mono_info() and + snd_ctl_boolean_stereo_info(). + Obviously, the former is an info callback for a mono channel + boolean item, just like snd_myctl_mono_info + above, and the latter is for a stereo channel boolean item. + + +
+ +
+ get callback + + + This callback is used to read the current value of the + control and to return to user-space. + + + + For example, + + + Example of get callback + +value.integer.value[0] = get_some_value(chip); + return 0; + } +]]> + + + + + + The value field depends on + the type of control as well as on the info callback. For example, + the sb driver uses this field to store the register offset, + the bit-shift and the bit-mask. The + private_value field is set as follows: + + + + + + and is retrieved in callbacks like + + +private_value & 0xff; + int shift = (kcontrol->private_value >> 16) & 0xff; + int mask = (kcontrol->private_value >> 24) & 0xff; + .... + } +]]> + + + + + + In the get callback, + you have to fill all the elements if the + control has more than one elements, + i.e. count > 1. + In the example above, we filled only one element + (value.integer.value[0]) since it's + assumed as count = 1. + +
+ +
+ put callback + + + This callback is used to write a value from user-space. + + + + For example, + + + Example of put callback + +current_value != + ucontrol->value.integer.value[0]) { + change_current_value(chip, + ucontrol->value.integer.value[0]); + changed = 1; + } + return changed; + } +]]> + + + + As seen above, you have to return 1 if the value is + changed. If the value is not changed, return 0 instead. + If any fatal error happens, return a negative error code as + usual. + + + + As in the get callback, + when the control has more than one elements, + all elements must be evaluated in this callback, too. + +
+ +
+ Callbacks are not atomic + + All these three callbacks are basically not atomic. + +
+
+ +
+ Constructor + + When everything is ready, finally we can create a new + control. To create a control, there are two functions to be + called, snd_ctl_new1() and + snd_ctl_add(). + + + + In the simplest way, you can do like this: + + + + + + + + where my_control is the + struct snd_kcontrol_new object defined above, and chip + is the object pointer to be passed to + kcontrol->private_data + which can be referred to in callbacks. + + + + snd_ctl_new1() allocates a new + snd_kcontrol instance (that's why the definition + of my_control can be with + the __devinitdata + prefix), and snd_ctl_add assigns the given + control component to the card. + +
+ +
+ Change Notification + + If you need to change and update a control in the interrupt + routine, you can call snd_ctl_notify(). For + example, + + + + + + + + This function takes the card pointer, the event-mask, and the + control id pointer for the notification. The event-mask + specifies the types of notification, for example, in the above + example, the change of control values is notified. + The id pointer is the pointer of struct snd_ctl_elem_id + to be notified. + You can find some examples in es1938.c or + es1968.c for hardware volume interrupts. + +
+ +
+ Metadata + + To provide information about the dB values of a mixer control, use + on of the DECLARE_TLV_xxx macros from + <sound/tlv.h> to define a variable + containing this information, set thetlv.p + field to point to this variable, and include the + SNDRV_CTL_ELEM_ACCESS_TLV_READ flag in the + access field; like this: + + + + + + + + + The DECLARE_TLV_DB_SCALE macro defines + information about a mixer control where each step in the control's + value changes the dB value by a constant dB amount. + The first parameter is the name of the variable to be defined. + The second parameter is the minimum value, in units of 0.01 dB. + The third parameter is the step size, in units of 0.01 dB. + Set the fourth parameter to 1 if the minimum value actually mutes + the control. + + + + The DECLARE_TLV_DB_LINEAR macro defines + information about a mixer control where the control's value affects + the output linearly. + The first parameter is the name of the variable to be defined. + The second parameter is the minimum value, in units of 0.01 dB. + The third parameter is the maximum value, in units of 0.01 dB. + If the minimum value mutes the control, set the second parameter to + TLV_DB_GAIN_MUTE. + +
+ + + + + + + + + API for AC97 Codec + +
+ General + + The ALSA AC97 codec layer is a well-defined one, and you don't + have to write much code to control it. Only low-level control + routines are necessary. The AC97 codec API is defined in + <sound/ac97_codec.h>. + +
+ +
+ Full Code Example + + + Example of AC97 Interface + +private_data; + .... + /* read a register value here from the codec */ + return the_register_value; + } + + static void snd_mychip_ac97_write(struct snd_ac97 *ac97, + unsigned short reg, unsigned short val) + { + struct mychip *chip = ac97->private_data; + .... + /* write the given register value to the codec */ + } + + static int snd_mychip_ac97(struct mychip *chip) + { + struct snd_ac97_bus *bus; + struct snd_ac97_template ac97; + int err; + static struct snd_ac97_bus_ops ops = { + .write = snd_mychip_ac97_write, + .read = snd_mychip_ac97_read, + }; + + err = snd_ac97_bus(chip->card, 0, &ops, NULL, &bus); + if (err < 0) + return err; + memset(&ac97, 0, sizeof(ac97)); + ac97.private_data = chip; + return snd_ac97_mixer(bus, &ac97, &chip->ac97); + } + +]]> + + + +
+ +
+ Constructor + + To create an ac97 instance, first call snd_ac97_bus + with an ac97_bus_ops_t record with callback functions. + + + + + + + + The bus record is shared among all belonging ac97 instances. + + + + And then call snd_ac97_mixer() with an + struct snd_ac97_template + record together with the bus pointer created above. + + + +ac97); +]]> + + + + where chip->ac97 is a pointer to a newly created + ac97_t instance. + In this case, the chip pointer is set as the private data, so that + the read/write callback functions can refer to this chip instance. + This instance is not necessarily stored in the chip + record. If you need to change the register values from the + driver, or need the suspend/resume of ac97 codecs, keep this + pointer to pass to the corresponding functions. + +
+ +
+ Callbacks + + The standard callbacks are read and + write. Obviously they + correspond to the functions for read and write accesses to the + hardware low-level codes. + + + + The read callback returns the + register value specified in the argument. + + + +private_data; + .... + return the_register_value; + } +]]> + + + + Here, the chip can be cast from ac97->private_data. + + + + Meanwhile, the write callback is + used to set the register value. + + + + + + + + + + These callbacks are non-atomic like the control API callbacks. + + + + There are also other callbacks: + reset, + wait and + init. + + + + The reset callback is used to reset + the codec. If the chip requires a special kind of reset, you can + define this callback. + + + + The wait callback is used to + add some waiting time in the standard initialization of the codec. If the + chip requires the extra waiting time, define this callback. + + + + The init callback is used for + additional initialization of the codec. + +
+ +
+ Updating Registers in The Driver + + If you need to access to the codec from the driver, you can + call the following functions: + snd_ac97_write(), + snd_ac97_read(), + snd_ac97_update() and + snd_ac97_update_bits(). + + + + Both snd_ac97_write() and + snd_ac97_update() functions are used to + set a value to the given register + (AC97_XXX). The difference between them is + that snd_ac97_update() doesn't write a + value if the given value has been already set, while + snd_ac97_write() always rewrites the + value. + + + + + + + + + + snd_ac97_read() is used to read the value + of the given register. For example, + + + + + + + + + + snd_ac97_update_bits() is used to update + some bits in the given register. + + + + + + + + + + Also, there is a function to change the sample rate (of a + given register such as + AC97_PCM_FRONT_DAC_RATE) when VRA or + DRA is supported by the codec: + snd_ac97_set_rate(). + + + + + + + + + + The following registers are available to set the rate: + AC97_PCM_MIC_ADC_RATE, + AC97_PCM_FRONT_DAC_RATE, + AC97_PCM_LR_ADC_RATE, + AC97_SPDIF. When + AC97_SPDIF is specified, the register is + not really changed but the corresponding IEC958 status bits will + be updated. + +
+ +
+ Clock Adjustment + + In some chips, the clock of the codec isn't 48000 but using a + PCI clock (to save a quartz!). In this case, change the field + bus->clock to the corresponding + value. For example, intel8x0 + and es1968 drivers have their own function to read from the clock. + +
+ +
+ Proc Files + + The ALSA AC97 interface will create a proc file such as + /proc/asound/card0/codec97#0/ac97#0-0 and + ac97#0-0+regs. You can refer to these files to + see the current status and registers of the codec. + +
+ +
+ Multiple Codecs + + When there are several codecs on the same card, you need to + call snd_ac97_mixer() multiple times with + ac97.num=1 or greater. The num field + specifies the codec number. + + + + If you set up multiple codecs, you either need to write + different callbacks for each codec or check + ac97->num in the callback routines. + +
+ + + + + + + + + MIDI (MPU401-UART) Interface + +
+ General + + Many soundcards have built-in MIDI (MPU401-UART) + interfaces. When the soundcard supports the standard MPU401-UART + interface, most likely you can use the ALSA MPU401-UART API. The + MPU401-UART API is defined in + <sound/mpu401.h>. + + + + Some soundchips have a similar but slightly different + implementation of mpu401 stuff. For example, emu10k1 has its own + mpu401 routines. + +
+ +
+ Constructor + + To create a rawmidi object, call + snd_mpu401_uart_new(). + + + + + + + + + + The first argument is the card pointer, and the second is the + index of this component. You can create up to 8 rawmidi + devices. + + + + The third argument is the type of the hardware, + MPU401_HW_XXX. If it's not a special one, + you can use MPU401_HW_MPU401. + + + + The 4th argument is the I/O port address. Many + backward-compatible MPU401 have an I/O port such as 0x330. Or, it + might be a part of its own PCI I/O region. It depends on the + chip design. + + + + The 5th argument is a bitflag for additional information. + When the I/O port address above is part of the PCI I/O + region, the MPU401 I/O port might have been already allocated + (reserved) by the driver itself. In such a case, pass a bit flag + MPU401_INFO_INTEGRATED, + and the mpu401-uart layer will allocate the I/O ports by itself. + + + + When the controller supports only the input or output MIDI stream, + pass the MPU401_INFO_INPUT or + MPU401_INFO_OUTPUT bitflag, respectively. + Then the rawmidi instance is created as a single stream. + + + + MPU401_INFO_MMIO bitflag is used to change + the access method to MMIO (via readb and writeb) instead of + iob and outb. In this case, you have to pass the iomapped address + to snd_mpu401_uart_new(). + + + + When MPU401_INFO_TX_IRQ is set, the output + stream isn't checked in the default interrupt handler. The driver + needs to call snd_mpu401_uart_interrupt_tx() + by itself to start processing the output stream in the irq handler. + + + + Usually, the port address corresponds to the command port and + port + 1 corresponds to the data port. If not, you may change + the cport field of + struct snd_mpu401 manually + afterward. However, snd_mpu401 pointer is not + returned explicitly by + snd_mpu401_uart_new(). You need to cast + rmidi->private_data to + snd_mpu401 explicitly, + + + +private_data; +]]> + + + + and reset the cport as you like: + + + +cport = my_own_control_port; +]]> + + + + + + The 6th argument specifies the irq number for UART. If the irq + is already allocated, pass 0 to the 7th argument + (irq_flags). Otherwise, pass the flags + for irq allocation + (SA_XXX bits) to it, and the irq will be + reserved by the mpu401-uart layer. If the card doesn't generate + UART interrupts, pass -1 as the irq number. Then a timer + interrupt will be invoked for polling. + +
+ +
+ Interrupt Handler + + When the interrupt is allocated in + snd_mpu401_uart_new(), the private + interrupt handler is used, hence you don't have anything else to do + than creating the mpu401 stuff. Otherwise, you have to call + snd_mpu401_uart_interrupt() explicitly when + a UART interrupt is invoked and checked in your own interrupt + handler. + + + + In this case, you need to pass the private_data of the + returned rawmidi object from + snd_mpu401_uart_new() as the second + argument of snd_mpu401_uart_interrupt(). + + + +private_data, regs); +]]> + + + +
+ + + + + + + + + RawMIDI Interface + +
+ Overview + + + The raw MIDI interface is used for hardware MIDI ports that can + be accessed as a byte stream. It is not used for synthesizer + chips that do not directly understand MIDI. + + + + ALSA handles file and buffer management. All you have to do is + to write some code to move data between the buffer and the + hardware. + + + + The rawmidi API is defined in + <sound/rawmidi.h>. + +
+ +
+ Constructor + + + To create a rawmidi device, call the + snd_rawmidi_new function: + + +card, "MyMIDI", 0, outs, ins, &rmidi); + if (err < 0) + return err; + rmidi->private_data = chip; + strcpy(rmidi->name, "My MIDI"); + rmidi->info_flags = SNDRV_RAWMIDI_INFO_OUTPUT | + SNDRV_RAWMIDI_INFO_INPUT | + SNDRV_RAWMIDI_INFO_DUPLEX; +]]> + + + + + + The first argument is the card pointer, the second argument is + the ID string. + + + + The third argument is the index of this component. You can + create up to 8 rawmidi devices. + + + + The fourth and fifth arguments are the number of output and + input substreams, respectively, of this device (a substream is + the equivalent of a MIDI port). + + + + Set the info_flags field to specify + the capabilities of the device. + Set SNDRV_RAWMIDI_INFO_OUTPUT if there is + at least one output port, + SNDRV_RAWMIDI_INFO_INPUT if there is at + least one input port, + and SNDRV_RAWMIDI_INFO_DUPLEX if the device + can handle output and input at the same time. + + + + After the rawmidi device is created, you need to set the + operators (callbacks) for each substream. There are helper + functions to set the operators for all the substreams of a device: + + + + + + + + + The operators are usually defined like this: + + + + + + These callbacks are explained in the Callbacks + section. + + + + If there are more than one substream, you should give a + unique name to each of them: + + +streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substreams, + list { + sprintf(substream->name, "My MIDI Port %d", substream->number + 1); + } + /* same for SNDRV_RAWMIDI_STREAM_INPUT */ +]]> + + + +
+ +
+ Callbacks + + + In all the callbacks, the private data that you've set for the + rawmidi device can be accessed as + substream->rmidi->private_data. + + + + + If there is more than one port, your callbacks can determine the + port index from the struct snd_rawmidi_substream data passed to each + callback: + + +number; +]]> + + + + +
+ <function>open</function> callback + + + + + + + + + This is called when a substream is opened. + You can initialize the hardware here, but you shouldn't + start transmitting/receiving data yet. + +
+ +
+ <function>close</function> callback + + + + + + + + + Guess what. + + + + The open and close + callbacks of a rawmidi device are serialized with a mutex, + and can sleep. + +
+ +
+ <function>trigger</function> callback for output + substreams + + + + + + + + + This is called with a nonzero up + parameter when there is some data in the substream buffer that + must be transmitted. + + + + To read data from the buffer, call + snd_rawmidi_transmit_peek. It will + return the number of bytes that have been read; this will be + less than the number of bytes requested when there are no more + data in the buffer. + After the data have been transmitted successfully, call + snd_rawmidi_transmit_ack to remove the + data from the substream buffer: + + + + + + + + + If you know beforehand that the hardware will accept data, you + can use the snd_rawmidi_transmit function + which reads some data and removes them from the buffer at once: + + + + + + + + + If you know beforehand how many bytes you can accept, you can + use a buffer size greater than one with the + snd_rawmidi_transmit* functions. + + + + The trigger callback must not sleep. If + the hardware FIFO is full before the substream buffer has been + emptied, you have to continue transmitting data later, either + in an interrupt handler, or with a timer if the hardware + doesn't have a MIDI transmit interrupt. + + + + The trigger callback is called with a + zero up parameter when the transmission + of data should be aborted. + +
+ +
+ <function>trigger</function> callback for input + substreams + + + + + + + + + This is called with a nonzero up + parameter to enable receiving data, or with a zero + up parameter do disable receiving data. + + + + The trigger callback must not sleep; the + actual reading of data from the device is usually done in an + interrupt handler. + + + + When data reception is enabled, your interrupt handler should + call snd_rawmidi_receive for all received + data: + + + + + + +
+ +
+ <function>drain</function> callback + + + + + + + + + This is only used with output substreams. This function should wait + until all data read from the substream buffer have been transmitted. + This ensures that the device can be closed and the driver unloaded + without losing data. + + + + This callback is optional. If you do not set + drain in the struct snd_rawmidi_ops + structure, ALSA will simply wait for 50 milliseconds + instead. + +
+
+ + + + + + + + + Miscellaneous Devices + +
+ FM OPL3 + + The FM OPL3 is still used in many chips (mainly for backward + compatibility). ALSA has a nice OPL3 FM control layer, too. The + OPL3 API is defined in + <sound/opl3.h>. + + + + FM registers can be directly accessed through the direct-FM API, + defined in <sound/asound_fm.h>. In + ALSA native mode, FM registers are accessed through + the Hardware-Dependant Device direct-FM extension API, whereas in + OSS compatible mode, FM registers can be accessed with the OSS + direct-FM compatible API in /dev/dmfmX device. + + + + To create the OPL3 component, you have two functions to + call. The first one is a constructor for the opl3_t + instance. + + + + + + + + + + The first argument is the card pointer, the second one is the + left port address, and the third is the right port address. In + most cases, the right port is placed at the left port + 2. + + + + The fourth argument is the hardware type. + + + + When the left and right ports have been already allocated by + the card driver, pass non-zero to the fifth argument + (integrated). Otherwise, the opl3 module will + allocate the specified ports by itself. + + + + When the accessing the hardware requires special method + instead of the standard I/O access, you can create opl3 instance + separately with snd_opl3_new(). + + + + + + + + + + Then set command, + private_data and + private_free for the private + access function, the private data and the destructor. + The l_port and r_port are not necessarily set. Only the + command must be set properly. You can retrieve the data + from the opl3->private_data field. + + + + After creating the opl3 instance via snd_opl3_new(), + call snd_opl3_init() to initialize the chip to the + proper state. Note that snd_opl3_create() always + calls it internally. + + + + If the opl3 instance is created successfully, then create a + hwdep device for this opl3. + + + + + + + + + + The first argument is the opl3_t instance you + created, and the second is the index number, usually 0. + + + + The third argument is the index-offset for the sequencer + client assigned to the OPL3 port. When there is an MPU401-UART, + give 1 for here (UART always takes 0). + +
+ +
+ Hardware-Dependent Devices + + Some chips need user-space access for special + controls or for loading the micro code. In such a case, you can + create a hwdep (hardware-dependent) device. The hwdep API is + defined in <sound/hwdep.h>. You can + find examples in opl3 driver or + isa/sb/sb16_csp.c. + + + + The creation of the hwdep instance is done via + snd_hwdep_new(). + + + + + + + + where the third argument is the index number. + + + + You can then pass any pointer value to the + private_data. + If you assign a private data, you should define the + destructor, too. The destructor function is set in + the private_free field. + + + +private_data = p; + hw->private_free = mydata_free; +]]> + + + + and the implementation of the destructor would be: + + + +private_data; + kfree(p); + } +]]> + + + + + + The arbitrary file operations can be defined for this + instance. The file operators are defined in + the ops table. For example, assume that + this chip needs an ioctl. + + + +ops.open = mydata_open; + hw->ops.ioctl = mydata_ioctl; + hw->ops.release = mydata_release; +]]> + + + + And implement the callback functions as you like. + +
+ +
+ IEC958 (S/PDIF) + + Usually the controls for IEC958 devices are implemented via + the control interface. There is a macro to compose a name string for + IEC958 controls, SNDRV_CTL_NAME_IEC958() + defined in <include/asound.h>. + + + + There are some standard controls for IEC958 status bits. These + controls use the type SNDRV_CTL_ELEM_TYPE_IEC958, + and the size of element is fixed as 4 bytes array + (value.iec958.status[x]). For the info + callback, you don't specify + the value field for this type (the count field must be set, + though). + + + + IEC958 Playback Con Mask is used to return the + bit-mask for the IEC958 status bits of consumer mode. Similarly, + IEC958 Playback Pro Mask returns the bitmask for + professional mode. They are read-only controls, and are defined + as MIXER controls (iface = + SNDRV_CTL_ELEM_IFACE_MIXER). + + + + Meanwhile, IEC958 Playback Default control is + defined for getting and setting the current default IEC958 + bits. Note that this one is usually defined as a PCM control + (iface = SNDRV_CTL_ELEM_IFACE_PCM), + although in some places it's defined as a MIXER control. + + + + In addition, you can define the control switches to + enable/disable or to set the raw bit mode. The implementation + will depend on the chip, but the control should be named as + IEC958 xxx, preferably using + the SNDRV_CTL_NAME_IEC958() macro. + + + + You can find several cases, for example, + pci/emu10k1, + pci/ice1712, or + pci/cmipci.c. + +
+ + + + + + + + + Buffer and Memory Management + +
+ Buffer Types + + ALSA provides several different buffer allocation functions + depending on the bus and the architecture. All these have a + consistent API. The allocation of physically-contiguous pages is + done via + snd_malloc_xxx_pages() function, where xxx + is the bus type. + + + + The allocation of pages with fallback is + snd_malloc_xxx_pages_fallback(). This + function tries to allocate the specified pages but if the pages + are not available, it tries to reduce the page sizes until + enough space is found. + + + + The release the pages, call + snd_free_xxx_pages() function. + + + + Usually, ALSA drivers try to allocate and reserve + a large contiguous physical space + at the time the module is loaded for the later use. + This is called pre-allocation. + As already written, you can call the following function at + pcm instance construction time (in the case of PCI bus). + + + + + + + + where size is the byte size to be + pre-allocated and the max is the maximum + size to be changed via the prealloc proc file. + The allocator will try to get an area as large as possible + within the given size. + + + + The second argument (type) and the third argument (device pointer) + are dependent on the bus. + In the case of the ISA bus, pass snd_dma_isa_data() + as the third argument with SNDRV_DMA_TYPE_DEV type. + For the continuous buffer unrelated to the bus can be pre-allocated + with SNDRV_DMA_TYPE_CONTINUOUS type and the + snd_dma_continuous_data(GFP_KERNEL) device pointer, + where GFP_KERNEL is the kernel allocation flag to + use. + For the PCI scatter-gather buffers, use + SNDRV_DMA_TYPE_DEV_SG with + snd_dma_pci_data(pci) + (see the + Non-Contiguous Buffers + section). + + + + Once the buffer is pre-allocated, you can use the + allocator in the hw_params callback: + + + + + + + + Note that you have to pre-allocate to use this function. + +
+ +
+ External Hardware Buffers + + Some chips have their own hardware buffers and the DMA + transfer from the host memory is not available. In such a case, + you need to either 1) copy/set the audio data directly to the + external hardware buffer, or 2) make an intermediate buffer and + copy/set the data from it to the external hardware buffer in + interrupts (or in tasklets, preferably). + + + + The first case works fine if the external hardware buffer is large + enough. This method doesn't need any extra buffers and thus is + more effective. You need to define the + copy and + silence callbacks for + the data transfer. However, there is a drawback: it cannot + be mmapped. The examples are GUS's GF1 PCM or emu8000's + wavetable PCM. + + + + The second case allows for mmap on the buffer, although you have + to handle an interrupt or a tasklet to transfer the data + from the intermediate buffer to the hardware buffer. You can find an + example in the vxpocket driver. + + + + Another case is when the chip uses a PCI memory-map + region for the buffer instead of the host memory. In this case, + mmap is available only on certain architectures like the Intel one. + In non-mmap mode, the data cannot be transferred as in the normal + way. Thus you need to define the copy and + silence callbacks as well, + as in the cases above. The examples are found in + rme32.c and rme96.c. + + + + The implementation of the copy and + silence callbacks depends upon + whether the hardware supports interleaved or non-interleaved + samples. The copy callback is + defined like below, a bit + differently depending whether the direction is playback or + capture: + + + + + + + + + + In the case of interleaved samples, the second argument + (channel) is not used. The third argument + (pos) points the + current position offset in frames. + + + + The meaning of the fourth argument is different between + playback and capture. For playback, it holds the source data + pointer, and for capture, it's the destination data pointer. + + + + The last argument is the number of frames to be copied. + + + + What you have to do in this callback is again different + between playback and capture directions. In the + playback case, you copy the given amount of data + (count) at the specified pointer + (src) to the specified offset + (pos) on the hardware buffer. When + coded like memcpy-like way, the copy would be like: + + + + + + + + + + For the capture direction, you copy the given amount of + data (count) at the specified offset + (pos) on the hardware buffer to the + specified pointer (dst). + + + + + + + + Note that both the position and the amount of data are given + in frames. + + + + In the case of non-interleaved samples, the implementation + will be a bit more complicated. + + + + You need to check the channel argument, and if it's -1, copy + the whole channels. Otherwise, you have to copy only the + specified channel. Please check + isa/gus/gus_pcm.c as an example. + + + + The silence callback is also + implemented in a similar way. + + + + + + + + + + The meanings of arguments are the same as in the + copy + callback, although there is no src/dst + argument. In the case of interleaved samples, the channel + argument has no meaning, as well as on + copy callback. + + + + The role of silence callback is to + set the given amount + (count) of silence data at the + specified offset (pos) on the hardware + buffer. Suppose that the data format is signed (that is, the + silent-data is 0), and the implementation using a memset-like + function would be like: + + + + + + + + + + In the case of non-interleaved samples, again, the + implementation becomes a bit more complicated. See, for example, + isa/gus/gus_pcm.c. + +
+ +
+ Non-Contiguous Buffers + + If your hardware supports the page table as in emu10k1 or the + buffer descriptors as in via82xx, you can use the scatter-gather + (SG) DMA. ALSA provides an interface for handling SG-buffers. + The API is provided in <sound/pcm.h>. + + + + For creating the SG-buffer handler, call + snd_pcm_lib_preallocate_pages() or + snd_pcm_lib_preallocate_pages_for_all() + with SNDRV_DMA_TYPE_DEV_SG + in the PCM constructor like other PCI pre-allocator. + You need to pass snd_dma_pci_data(pci), + where pci is the struct pci_dev pointer + of the chip as well. + The struct snd_sg_buf instance is created as + substream->dma_private. You can cast + the pointer like: + + + +dma_private; +]]> + + + + + + Then call snd_pcm_lib_malloc_pages() + in the hw_params callback + as well as in the case of normal PCI buffer. + The SG-buffer handler will allocate the non-contiguous kernel + pages of the given size and map them onto the virtually contiguous + memory. The virtual pointer is addressed in runtime->dma_area. + The physical address (runtime->dma_addr) is set to zero, + because the buffer is physically non-contigous. + The physical address table is set up in sgbuf->table. + You can get the physical address at a certain offset via + snd_pcm_sgbuf_get_addr(). + + + + When a SG-handler is used, you need to set + snd_pcm_sgbuf_ops_page as + the page callback. + (See + page callback section.) + + + + To release the data, call + snd_pcm_lib_free_pages() in the + hw_free callback as usual. + +
+ +
+ Vmalloc'ed Buffers + + It's possible to use a buffer allocated via + vmalloc, for example, for an intermediate + buffer. Since the allocated pages are not contiguous, you need + to set the page callback to obtain + the physical address at every offset. + + + + The implementation of page callback + would be like this: + + + + + + /* get the physical page pointer on the given offset */ + static struct page *mychip_page(struct snd_pcm_substream *substream, + unsigned long offset) + { + void *pageptr = substream->runtime->dma_area + offset; + return vmalloc_to_page(pageptr); + } +]]> + + + +
+ + + + + + + + + Proc Interface + + ALSA provides an easy interface for procfs. The proc files are + very useful for debugging. I recommend you set up proc files if + you write a driver and want to get a running status or register + dumps. The API is found in + <sound/info.h>. + + + + To create a proc file, call + snd_card_proc_new(). + + + + + + + + where the second argument specifies the name of the proc file to be + created. The above example will create a file + my-file under the card directory, + e.g. /proc/asound/card0/my-file. + + + + Like other components, the proc entry created via + snd_card_proc_new() will be registered and + released automatically in the card registration and release + functions. + + + + When the creation is successful, the function stores a new + instance in the pointer given in the third argument. + It is initialized as a text proc file for read only. To use + this proc file as a read-only text file as it is, set the read + callback with a private data via + snd_info_set_text_ops(). + + + + + + + + where the second argument (chip) is the + private data to be used in the callbacks. The third parameter + specifies the read buffer size and the fourth + (my_proc_read) is the callback function, which + is defined like + + + + + + + + + + + In the read callback, use snd_iprintf() for + output strings, which works just like normal + printf(). For example, + + + +private_data; + + snd_iprintf(buffer, "This is my chip!\n"); + snd_iprintf(buffer, "Port = %ld\n", chip->port); + } +]]> + + + + + + The file permissions can be changed afterwards. As default, it's + set as read only for all users. If you want to add write + permission for the user (root as default), do as follows: + + + +mode = S_IFREG | S_IRUGO | S_IWUSR; +]]> + + + + and set the write buffer size and the callback + + + +c.text.write = my_proc_write; +]]> + + + + + + For the write callback, you can use + snd_info_get_line() to get a text line, and + snd_info_get_str() to retrieve a string from + the line. Some examples are found in + core/oss/mixer_oss.c, core/oss/and + pcm_oss.c. + + + + For a raw-data proc-file, set the attributes as follows: + + + +content = SNDRV_INFO_CONTENT_DATA; + entry->private_data = chip; + entry->c.ops = &my_file_io_ops; + entry->size = 4096; + entry->mode = S_IFREG | S_IRUGO; +]]> + + + + + + The callback is much more complicated than the text-file + version. You need to use a low-level I/O functions such as + copy_from/to_user() to transfer the + data. + + + + local_max_size) + size = local_max_size - pos; + if (copy_to_user(buf, local_data + pos, size)) + return -EFAULT; + return size; + } +]]> + + + + + + + + + + + + Power Management + + If the chip is supposed to work with suspend/resume + functions, you need to add power-management code to the + driver. The additional code for power-management should be + ifdef'ed with + CONFIG_PM. + + + + If the driver fully supports suspend/resume + that is, the device can be + properly resumed to its state when suspend was called, + you can set the SNDRV_PCM_INFO_RESUME flag + in the pcm info field. Usually, this is possible when the + registers of the chip can be safely saved and restored to + RAM. If this is set, the trigger callback is called with + SNDRV_PCM_TRIGGER_RESUME after the resume + callback completes. + + + + Even if the driver doesn't support PM fully but + partial suspend/resume is still possible, it's still worthy to + implement suspend/resume callbacks. In such a case, applications + would reset the status by calling + snd_pcm_prepare() and restart the stream + appropriately. Hence, you can define suspend/resume callbacks + below but don't set SNDRV_PCM_INFO_RESUME + info flag to the PCM. + + + + Note that the trigger with SUSPEND can always be called when + snd_pcm_suspend_all is called, + regardless of the SNDRV_PCM_INFO_RESUME flag. + The RESUME flag affects only the behavior + of snd_pcm_resume(). + (Thus, in theory, + SNDRV_PCM_TRIGGER_RESUME isn't needed + to be handled in the trigger callback when no + SNDRV_PCM_INFO_RESUME flag is set. But, + it's better to keep it for compatibility reasons.) + + + In the earlier version of ALSA drivers, a common + power-management layer was provided, but it has been removed. + The driver needs to define the suspend/resume hooks according to + the bus the device is connected to. In the case of PCI drivers, the + callbacks look like below: + + + + + + + + + + The scheme of the real suspend job is as follows. + + + Retrieve the card and the chip data. + Call snd_power_change_state() with + SNDRV_CTL_POWER_D3hot to change the + power status. + Call snd_pcm_suspend_all() to suspend the running PCM streams. + If AC97 codecs are used, call + snd_ac97_suspend() for each codec. + Save the register values if necessary. + Stop the hardware if necessary. + Disable the PCI device by calling + pci_disable_device(). Then, call + pci_save_state() at last. + + + + + A typical code would be like: + + + +private_data; + /* (2) */ + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + /* (3) */ + snd_pcm_suspend_all(chip->pcm); + /* (4) */ + snd_ac97_suspend(chip->ac97); + /* (5) */ + snd_mychip_save_registers(chip); + /* (6) */ + snd_mychip_stop_hardware(chip); + /* (7) */ + pci_disable_device(pci); + pci_save_state(pci); + return 0; + } +]]> + + + + + + The scheme of the real resume job is as follows. + + + Retrieve the card and the chip data. + Set up PCI. First, call pci_restore_state(). + Then enable the pci device again by calling pci_enable_device(). + Call pci_set_master() if necessary, too. + Re-initialize the chip. + Restore the saved registers if necessary. + Resume the mixer, e.g. calling + snd_ac97_resume(). + Restart the hardware (if any). + Call snd_power_change_state() with + SNDRV_CTL_POWER_D0 to notify the processes. + + + + + A typical code would be like: + + + +private_data; + /* (2) */ + pci_restore_state(pci); + pci_enable_device(pci); + pci_set_master(pci); + /* (3) */ + snd_mychip_reinit_chip(chip); + /* (4) */ + snd_mychip_restore_registers(chip); + /* (5) */ + snd_ac97_resume(chip->ac97); + /* (6) */ + snd_mychip_restart_chip(chip); + /* (7) */ + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + return 0; + } +]]> + + + + + + As shown in the above, it's better to save registers after + suspending the PCM operations via + snd_pcm_suspend_all() or + snd_pcm_suspend(). It means that the PCM + streams are already stoppped when the register snapshot is + taken. But, remember that you don't have to restart the PCM + stream in the resume callback. It'll be restarted via + trigger call with SNDRV_PCM_TRIGGER_RESUME + when necessary. + + + + OK, we have all callbacks now. Let's set them up. In the + initialization of the card, make sure that you can get the chip + data from the card instance, typically via + private_data field, in case you + created the chip data individually. + + + +private_data = chip; + .... + } +]]> + + + + When you created the chip data with + snd_card_create(), it's anyway accessible + via private_data field. + + + +private_data; + .... + } +]]> + + + + + + + If you need a space to save the registers, allocate the + buffer for it here, too, since it would be fatal + if you cannot allocate a memory in the suspend phase. + The allocated buffer should be released in the corresponding + destructor. + + + + And next, set suspend/resume callbacks to the pci_driver. + + + + + + + + + + + + + + + + Module Parameters + + There are standard module options for ALSA. At least, each + module should have the index, + id and enable + options. + + + + If the module supports multiple cards (usually up to + 8 = SNDRV_CARDS cards), they should be + arrays. The default initial values are defined already as + constants for easier programming: + + + + + + + + + + If the module supports only a single card, they could be single + variables, instead. enable option is not + always necessary in this case, but it would be better to have a + dummy option for compatibility. + + + + The module parameters must be declared with the standard + module_param()(), + module_param_array()() and + MODULE_PARM_DESC() macros. + + + + The typical coding would be like below: + + + + + + + + + + Also, don't forget to define the module description, classes, + license and devices. Especially, the recent modprobe requires to + define the module license as GPL, etc., otherwise the system is + shown as tainted. + + + + + + + + + + + + + + + + How To Put Your Driver Into ALSA Tree +
+ General + + So far, you've learned how to write the driver codes. + And you might have a question now: how to put my own + driver into the ALSA driver tree? + Here (finally :) the standard procedure is described briefly. + + + + Suppose that you create a new PCI driver for the card + xyz. The card module name would be + snd-xyz. The new driver is usually put into the alsa-driver + tree, alsa-driver/pci directory in + the case of PCI cards. + Then the driver is evaluated, audited and tested + by developers and users. After a certain time, the driver + will go to the alsa-kernel tree (to the corresponding directory, + such as alsa-kernel/pci) and eventually + will be integrated into the Linux 2.6 tree (the directory would be + linux/sound/pci). + + + + In the following sections, the driver code is supposed + to be put into alsa-driver tree. The two cases are covered: + a driver consisting of a single source file and one consisting + of several source files. + +
+ +
+ Driver with A Single Source File + + + + + Modify alsa-driver/pci/Makefile + + + + Suppose you have a file xyz.c. Add the following + two lines + + + + + + + + + + + Create the Kconfig entry + + + + Add the new entry of Kconfig for your xyz driver. + + + + + + + the line, select SND_PCM, specifies that the driver xyz supports + PCM. In addition to SND_PCM, the following components are + supported for select command: + SND_RAWMIDI, SND_TIMER, SND_HWDEP, SND_MPU401_UART, + SND_OPL3_LIB, SND_OPL4_LIB, SND_VX_LIB, SND_AC97_CODEC. + Add the select command for each supported component. + + + + Note that some selections imply the lowlevel selections. + For example, PCM includes TIMER, MPU401_UART includes RAWMIDI, + AC97_CODEC includes PCM, and OPL3_LIB includes HWDEP. + You don't need to give the lowlevel selections again. + + + + For the details of Kconfig script, refer to the kbuild + documentation. + + + + + + + Run cvscompile script to re-generate the configure script and + build the whole stuff again. + + + + +
+ +
+ Drivers with Several Source Files + + Suppose that the driver snd-xyz have several source files. + They are located in the new subdirectory, + pci/xyz. + + + + + Add a new directory (xyz) in + alsa-driver/pci/Makefile as below + + + + + + + + + + + + Under the directory xyz, create a Makefile + + + Sample Makefile for a driver xyz + + + + + + + + + + Create the Kconfig entry + + + + This procedure is as same as in the last section. + + + + + + Run cvscompile script to re-generate the configure script and + build the whole stuff again. + + + + +
+ + + + + + + + Useful Functions + +
+ <function>snd_printk()</function> and friends + + ALSA provides a verbose version of the + printk() function. If a kernel config + CONFIG_SND_VERBOSE_PRINTK is set, this + function prints the given message together with the file name + and the line of the caller. The KERN_XXX + prefix is processed as + well as the original printk() does, so it's + recommended to add this prefix, e.g. + + + + + + + + + + There are also printk()'s for + debugging. snd_printd() can be used for + general debugging purposes. If + CONFIG_SND_DEBUG is set, this function is + compiled, and works just like + snd_printk(). If the ALSA is compiled + without the debugging flag, it's ignored. + + + + snd_printdd() is compiled in only when + CONFIG_SND_DEBUG_VERBOSE is set. Please note + that CONFIG_SND_DEBUG_VERBOSE is not set as default + even if you configure the alsa-driver with + option. You need to give + explicitly option instead. + +
+ +
+ <function>snd_BUG()</function> + + It shows the BUG? message and + stack trace as well as snd_BUG_ON at the point. + It's useful to show that a fatal error happens there. + + + When no debug flag is set, this macro is ignored. + +
+ +
+ <function>snd_BUG_ON()</function> + + snd_BUG_ON() macro is similar with + WARN_ON() macro. For example, + + + + + + + + or it can be used as the condition, + + + + + + + + + + The macro takes an conditional expression to evaluate. + When CONFIG_SND_DEBUG, is set, the + expression is actually evaluated. If it's non-zero, it shows + the warning message such as + BUG? (xxx) + normally followed by stack trace. It returns the evaluated + value. + When no CONFIG_SND_DEBUG is set, this + macro always returns zero. + + +
+ + + + + + + + + Acknowledgments + + I would like to thank Phil Kerr for his help for improvement and + corrections of this document. + + + Kevin Conder reformatted the original plain-text to the + DocBook format. + + + Giuliano Pochini corrected typos and contributed the example codes + in the hardware constraints section. + + + diff --git a/Documentation/sound/alsa/DocBook/alsa-driver-api.tmpl b/Documentation/sound/alsa/DocBook/alsa-driver-api.tmpl deleted file mode 100644 index 0230a96f056..00000000000 --- a/Documentation/sound/alsa/DocBook/alsa-driver-api.tmpl +++ /dev/null @@ -1,109 +0,0 @@ - - - - - - - - - The ALSA Driver API - - - - This document is free; you can redistribute it and/or modify it - under the terms of the GNU General Public License as published by - the Free Software Foundation; either version 2 of the License, or - (at your option) any later version. - - - - This document is distributed in the hope that it will be useful, - but WITHOUT ANY WARRANTY; without even the - implied warranty of MERCHANTABILITY or FITNESS FOR A - PARTICULAR PURPOSE. See the GNU General Public License - for more details. - - - - You should have received a copy of the GNU General Public - License along with this program; if not, write to the Free - Software Foundation, Inc., 59 Temple Place, Suite 330, Boston, - MA 02111-1307 USA - - - - - - - - Management of Cards and Devices - Card Management -!Esound/core/init.c - - Device Components -!Esound/core/device.c - - Module requests and Device File Entries -!Esound/core/sound.c - - Memory Management Helpers -!Esound/core/memory.c -!Esound/core/memalloc.c - - - PCM API - PCM Core -!Esound/core/pcm.c -!Esound/core/pcm_lib.c -!Esound/core/pcm_native.c - - PCM Format Helpers -!Esound/core/pcm_misc.c - - PCM Memory Management -!Esound/core/pcm_memory.c - - - Control/Mixer API - General Control Interface -!Esound/core/control.c - - AC97 Codec API -!Esound/pci/ac97/ac97_codec.c -!Esound/pci/ac97/ac97_pcm.c - - Virtual Master Control API -!Esound/core/vmaster.c -!Iinclude/sound/control.h - - - MIDI API - Raw MIDI API -!Esound/core/rawmidi.c - - MPU401-UART API -!Esound/drivers/mpu401/mpu401_uart.c - - - Proc Info API - Proc Info Interface -!Esound/core/info.c - - - Miscellaneous Functions - Hardware-Dependent Devices API -!Esound/core/hwdep.c - - Jack Abstraction Layer API -!Esound/core/jack.c - - ISA DMA Helpers -!Esound/core/isadma.c - - Other Helper Macros -!Iinclude/sound/core.h - - - - diff --git a/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl b/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl deleted file mode 100644 index 46b08fef374..00000000000 --- a/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl +++ /dev/null @@ -1,6216 +0,0 @@ - - - - - - - - - Writing an ALSA Driver - - Takashi - Iwai - -
- tiwai@suse.de -
- - - - Oct 15, 2007 - 0.3.7 - - - - This document describes how to write an ALSA (Advanced Linux - Sound Architecture) driver. - - - - - - Copyright (c) 2002-2005 Takashi Iwai tiwai@suse.de - - - - This document is free; you can redistribute it and/or modify it - under the terms of the GNU General Public License as published by - the Free Software Foundation; either version 2 of the License, or - (at your option) any later version. - - - - This document is distributed in the hope that it will be useful, - but WITHOUT ANY WARRANTY; without even the - implied warranty of MERCHANTABILITY or FITNESS FOR A - PARTICULAR PURPOSE. See the GNU General Public License - for more details. - - - - You should have received a copy of the GNU General Public - License along with this program; if not, write to the Free - Software Foundation, Inc., 59 Temple Place, Suite 330, Boston, - MA 02111-1307 USA - - - - - - - - - - Preface - - This document describes how to write an - - ALSA (Advanced Linux Sound Architecture) - driver. The document focuses mainly on PCI soundcards. - In the case of other device types, the API might - be different, too. However, at least the ALSA kernel API is - consistent, and therefore it would be still a bit help for - writing them. - - - - This document targets people who already have enough - C language skills and have basic linux kernel programming - knowledge. This document doesn't explain the general - topic of linux kernel coding and doesn't cover low-level - driver implementation details. It only describes - the standard way to write a PCI sound driver on ALSA. - - - - If you are already familiar with the older ALSA ver.0.5.x API, you - can check the drivers such as sound/pci/es1938.c or - sound/pci/maestro3.c which have also almost the same - code-base in the ALSA 0.5.x tree, so you can compare the differences. - - - - This document is still a draft version. Any feedback and - corrections, please!! - - - - - - - - - File Tree Structure - -
- General - - The ALSA drivers are provided in two ways. - - - - One is the trees provided as a tarball or via cvs from the - ALSA's ftp site, and another is the 2.6 (or later) Linux kernel - tree. To synchronize both, the ALSA driver tree is split into - two different trees: alsa-kernel and alsa-driver. The former - contains purely the source code for the Linux 2.6 (or later) - tree. This tree is designed only for compilation on 2.6 or - later environment. The latter, alsa-driver, contains many subtle - files for compiling ALSA drivers outside of the Linux kernel tree, - wrapper functions for older 2.2 and 2.4 kernels, to adapt the latest kernel API, - and additional drivers which are still in development or in - tests. The drivers in alsa-driver tree will be moved to - alsa-kernel (and eventually to the 2.6 kernel tree) when they are - finished and confirmed to work fine. - - - - The file tree structure of ALSA driver is depicted below. Both - alsa-kernel and alsa-driver have almost the same file - structure, except for core directory. It's - named as acore in alsa-driver tree. - - - ALSA File Tree Structure - - sound - /core - /oss - /seq - /oss - /instr - /ioctl32 - /include - /drivers - /mpu401 - /opl3 - /i2c - /l3 - /synth - /emux - /pci - /(cards) - /isa - /(cards) - /arm - /ppc - /sparc - /usb - /pcmcia /(cards) - /oss - - - -
- -
- core directory - - This directory contains the middle layer which is the heart - of ALSA drivers. In this directory, the native ALSA modules are - stored. The sub-directories contain different modules and are - dependent upon the kernel config. - - -
- core/oss - - - The codes for PCM and mixer OSS emulation modules are stored - in this directory. The rawmidi OSS emulation is included in - the ALSA rawmidi code since it's quite small. The sequencer - code is stored in core/seq/oss directory (see - - below). - -
- -
- core/ioctl32 - - - This directory contains the 32bit-ioctl wrappers for 64bit - architectures such like x86-64, ppc64 and sparc64. For 32bit - and alpha architectures, these are not compiled. - -
- -
- core/seq - - This directory and its sub-directories are for the ALSA - sequencer. This directory contains the sequencer core and - primary sequencer modules such like snd-seq-midi, - snd-seq-virmidi, etc. They are compiled only when - CONFIG_SND_SEQUENCER is set in the kernel - config. - -
- -
- core/seq/oss - - This contains the OSS sequencer emulation codes. - -
- -
- core/seq/instr - - This directory contains the modules for the sequencer - instrument layer. - -
-
- -
- include directory - - This is the place for the public header files of ALSA drivers, - which are to be exported to user-space, or included by - several files at different directories. Basically, the private - header files should not be placed in this directory, but you may - still find files there, due to historical reasons :) - -
- -
- drivers directory - - This directory contains code shared among different drivers - on different architectures. They are hence supposed not to be - architecture-specific. - For example, the dummy pcm driver and the serial MIDI - driver are found in this directory. In the sub-directories, - there is code for components which are independent from - bus and cpu architectures. - - -
- drivers/mpu401 - - The MPU401 and MPU401-UART modules are stored here. - -
- -
- drivers/opl3 and opl4 - - The OPL3 and OPL4 FM-synth stuff is found here. - -
-
- -
- i2c directory - - This contains the ALSA i2c components. - - - - Although there is a standard i2c layer on Linux, ALSA has its - own i2c code for some cards, because the soundcard needs only a - simple operation and the standard i2c API is too complicated for - such a purpose. - - -
- i2c/l3 - - This is a sub-directory for ARM L3 i2c. - -
-
- -
- synth directory - - This contains the synth middle-level modules. - - - - So far, there is only Emu8000/Emu10k1 synth driver under - the synth/emux sub-directory. - -
- -
- pci directory - - This directory and its sub-directories hold the top-level card modules - for PCI soundcards and the code specific to the PCI BUS. - - - - The drivers compiled from a single file are stored directly - in the pci directory, while the drivers with several source files are - stored on their own sub-directory (e.g. emu10k1, ice1712). - -
- -
- isa directory - - This directory and its sub-directories hold the top-level card modules - for ISA soundcards. - -
- -
- arm, ppc, and sparc directories - - They are used for top-level card modules which are - specific to one of these architectures. - -
- -
- usb directory - - This directory contains the USB-audio driver. In the latest version, the - USB MIDI driver is integrated in the usb-audio driver. - -
- -
- pcmcia directory - - The PCMCIA, especially PCCard drivers will go here. CardBus - drivers will be in the pci directory, because their API is identical - to that of standard PCI cards. - -
- -
- oss directory - - The OSS/Lite source files are stored here in Linux 2.6 (or - later) tree. In the ALSA driver tarball, this directory is empty, - of course :) - -
- - - - - - - - Basic Flow for PCI Drivers - -
- Outline - - The minimum flow for PCI soundcards is as follows: - - - define the PCI ID table (see the section - PCI Entries - ). - create probe() callback. - create remove() callback. - create a pci_driver structure - containing the three pointers above. - create an init() function just calling - the pci_register_driver() to register the pci_driver table - defined above. - create an exit() function to call - the pci_unregister_driver() function. - - -
- -
- Full Code Example - - The code example is shown below. Some parts are kept - unimplemented at this moment but will be filled in the - next sections. The numbers in the comment lines of the - snd_mychip_probe() function - refer to details explained in the following section. - - - Basic Flow for PCI Drivers - Example - - - #include - #include - #include - #include - - /* module parameters (see "Module Parameters") */ - /* SNDRV_CARDS: maximum number of cards supported by this module */ - static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; - static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; - static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; - - /* definition of the chip-specific record */ - struct mychip { - struct snd_card *card; - /* the rest of the implementation will be in section - * "PCI Resource Management" - */ - }; - - /* chip-specific destructor - * (see "PCI Resource Management") - */ - static int snd_mychip_free(struct mychip *chip) - { - .... /* will be implemented later... */ - } - - /* component-destructor - * (see "Management of Cards and Components") - */ - static int snd_mychip_dev_free(struct snd_device *device) - { - return snd_mychip_free(device->device_data); - } - - /* chip-specific constructor - * (see "Management of Cards and Components") - */ - static int __devinit snd_mychip_create(struct snd_card *card, - struct pci_dev *pci, - struct mychip **rchip) - { - struct mychip *chip; - int err; - static struct snd_device_ops ops = { - .dev_free = snd_mychip_dev_free, - }; - - *rchip = NULL; - - /* check PCI availability here - * (see "PCI Resource Management") - */ - .... - - /* allocate a chip-specific data with zero filled */ - chip = kzalloc(sizeof(*chip), GFP_KERNEL); - if (chip == NULL) - return -ENOMEM; - - chip->card = card; - - /* rest of initialization here; will be implemented - * later, see "PCI Resource Management" - */ - .... - - err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); - if (err < 0) { - snd_mychip_free(chip); - return err; - } - - snd_card_set_dev(card, &pci->dev); - - *rchip = chip; - return 0; - } - - /* constructor -- see "Constructor" sub-section */ - static int __devinit snd_mychip_probe(struct pci_dev *pci, - const struct pci_device_id *pci_id) - { - static int dev; - struct snd_card *card; - struct mychip *chip; - int err; - - /* (1) */ - if (dev >= SNDRV_CARDS) - return -ENODEV; - if (!enable[dev]) { - dev++; - return -ENOENT; - } - - /* (2) */ - err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); - if (err < 0) - return err; - - /* (3) */ - err = snd_mychip_create(card, pci, &chip); - if (err < 0) { - snd_card_free(card); - return err; - } - - /* (4) */ - strcpy(card->driver, "My Chip"); - strcpy(card->shortname, "My Own Chip 123"); - sprintf(card->longname, "%s at 0x%lx irq %i", - card->shortname, chip->ioport, chip->irq); - - /* (5) */ - .... /* implemented later */ - - /* (6) */ - err = snd_card_register(card); - if (err < 0) { - snd_card_free(card); - return err; - } - - /* (7) */ - pci_set_drvdata(pci, card); - dev++; - return 0; - } - - /* destructor -- see the "Destructor" sub-section */ - static void __devexit snd_mychip_remove(struct pci_dev *pci) - { - snd_card_free(pci_get_drvdata(pci)); - pci_set_drvdata(pci, NULL); - } -]]> - - - -
- -
- Constructor - - The real constructor of PCI drivers is the probe callback. - The probe callback and other component-constructors which are called - from the probe callback should be defined with - the __devinit prefix. You - cannot use the __init prefix for them, - because any PCI device could be a hotplug device. - - - - In the probe callback, the following scheme is often used. - - -
- 1) Check and increment the device index. - - - -= SNDRV_CARDS) - return -ENODEV; - if (!enable[dev]) { - dev++; - return -ENOENT; - } -]]> - - - - where enable[dev] is the module option. - - - - Each time the probe callback is called, check the - availability of the device. If not available, simply increment - the device index and returns. dev will be incremented also - later (step - 7). - -
- -
- 2) Create a card instance - - - - - - - - - - The details will be explained in the section - - Management of Cards and Components. - -
- -
- 3) Create a main component - - In this part, the PCI resources are allocated. - - - - - - - - The details will be explained in the section PCI Resource - Management. - -
- -
- 4) Set the driver ID and name strings. - - - -driver, "My Chip"); - strcpy(card->shortname, "My Own Chip 123"); - sprintf(card->longname, "%s at 0x%lx irq %i", - card->shortname, chip->ioport, chip->irq); -]]> - - - - The driver field holds the minimal ID string of the - chip. This is used by alsa-lib's configurator, so keep it - simple but unique. - Even the same driver can have different driver IDs to - distinguish the functionality of each chip type. - - - - The shortname field is a string shown as more verbose - name. The longname field contains the information - shown in /proc/asound/cards. - -
- -
- 5) Create other components, such as mixer, MIDI, etc. - - Here you define the basic components such as - PCM, - mixer (e.g. AC97), - MIDI (e.g. MPU-401), - and other interfaces. - Also, if you want a proc - file, define it here, too. - -
- -
- 6) Register the card instance. - - - - - - - - - - Will be explained in the section Management - of Cards and Components, too. - -
- -
- 7) Set the PCI driver data and return zero. - - - - - - - - In the above, the card record is stored. This pointer is - used in the remove callback and power-management - callbacks, too. - -
-
- -
- Destructor - - The destructor, remove callback, simply releases the card - instance. Then the ALSA middle layer will release all the - attached components automatically. - - - - It would be typically like the following: - - - - - - - - The above code assumes that the card pointer is set to the PCI - driver data. - -
- -
- Header Files - - For the above example, at least the following include files - are necessary. - - - - - #include - #include - #include - #include -]]> - - - - where the last one is necessary only when module options are - defined in the source file. If the code is split into several - files, the files without module options don't need them. - - - - In addition to these headers, you'll need - <linux/interrupt.h> for interrupt - handling, and <asm/io.h> for I/O - access. If you use the mdelay() or - udelay() functions, you'll need to include - <linux/delay.h> too. - - - - The ALSA interfaces like the PCM and control APIs are defined in other - <sound/xxx.h> header files. - They have to be included after - <sound/core.h>. - - -
- - - - - - - - Management of Cards and Components - -
- Card Instance - - For each soundcard, a card record must be allocated. - - - - A card record is the headquarters of the soundcard. It manages - the whole list of devices (components) on the soundcard, such as - PCM, mixers, MIDI, synthesizer, and so on. Also, the card - record holds the ID and the name strings of the card, manages - the root of proc files, and controls the power-management states - and hotplug disconnections. The component list on the card - record is used to manage the correct release of resources at - destruction. - - - - As mentioned above, to create a card instance, call - snd_card_create(). - - - - - - - - - - The function takes five arguments, the card-index number, the - id string, the module pointer (usually - THIS_MODULE), - the size of extra-data space, and the pointer to return the - card instance. The extra_size argument is used to - allocate card->private_data for the - chip-specific data. Note that these data - are allocated by snd_card_create(). - -
- -
- Components - - After the card is created, you can attach the components - (devices) to the card instance. In an ALSA driver, a component is - represented as a struct snd_device object. - A component can be a PCM instance, a control interface, a raw - MIDI interface, etc. Each such instance has one component - entry. - - - - A component can be created via - snd_device_new() function. - - - - - - - - - - This takes the card pointer, the device-level - (SNDRV_DEV_XXX), the data pointer, and the - callback pointers (&ops). The - device-level defines the type of components and the order of - registration and de-registration. For most components, the - device-level is already defined. For a user-defined component, - you can use SNDRV_DEV_LOWLEVEL. - - - - This function itself doesn't allocate the data space. The data - must be allocated manually beforehand, and its pointer is passed - as the argument. This pointer is used as the - (chip identifier in the above example) - for the instance. - - - - Each pre-defined ALSA component such as ac97 and pcm calls - snd_device_new() inside its - constructor. The destructor for each component is defined in the - callback pointers. Hence, you don't need to take care of - calling a destructor for such a component. - - - - If you wish to create your own component, you need to - set the destructor function to the dev_free callback in - the ops, so that it can be released - automatically via snd_card_free(). - The next example will show an implementation of chip-specific - data. - -
- -
- Chip-Specific Data - - Chip-specific information, e.g. the I/O port address, its - resource pointer, or the irq number, is stored in the - chip-specific record. - - - - - - - - - - In general, there are two ways of allocating the chip record. - - -
- 1. Allocating via <function>snd_card_create()</function>. - - As mentioned above, you can pass the extra-data-length - to the 4th argument of snd_card_create(), i.e. - - - - - - - - struct mychip is the type of the chip record. - - - - In return, the allocated record can be accessed as - - - -private_data; -]]> - - - - With this method, you don't have to allocate twice. - The record is released together with the card instance. - -
- -
- 2. Allocating an extra device. - - - After allocating a card instance via - snd_card_create() (with - 0 on the 4th arg), call - kzalloc(). - - - - - - - - - - The chip record should have the field to hold the card - pointer at least, - - - - - - - - - - Then, set the card pointer in the returned chip instance. - - - -card = card; -]]> - - - - - - Next, initialize the fields, and register this chip - record as a low-level device with a specified - ops, - - - - - - - - snd_mychip_dev_free() is the - device-destructor function, which will call the real - destructor. - - - - - -device_data); - } -]]> - - - - where snd_mychip_free() is the real destructor. - -
-
- -
- Registration and Release - - After all components are assigned, register the card instance - by calling snd_card_register(). Access - to the device files is enabled at this point. That is, before - snd_card_register() is called, the - components are safely inaccessible from external side. If this - call fails, exit the probe function after releasing the card via - snd_card_free(). - - - - For releasing the card instance, you can call simply - snd_card_free(). As mentioned earlier, all - components are released automatically by this call. - - - - As further notes, the destructors (both - snd_mychip_dev_free and - snd_mychip_free) cannot be defined with - the __devexit prefix, because they may be - called from the constructor, too, at the false path. - - - - For a device which allows hotplugging, you can use - snd_card_free_when_closed. This one will - postpone the destruction until all devices are closed. - - -
- - - - - - - - - PCI Resource Management - -
- Full Code Example - - In this section, we'll complete the chip-specific constructor, - destructor and PCI entries. Example code is shown first, - below. - - - PCI Resource Management Example - -irq >= 0) - free_irq(chip->irq, chip); - /* release the I/O ports & memory */ - pci_release_regions(chip->pci); - /* disable the PCI entry */ - pci_disable_device(chip->pci); - /* release the data */ - kfree(chip); - return 0; - } - - /* chip-specific constructor */ - static int __devinit snd_mychip_create(struct snd_card *card, - struct pci_dev *pci, - struct mychip **rchip) - { - struct mychip *chip; - int err; - static struct snd_device_ops ops = { - .dev_free = snd_mychip_dev_free, - }; - - *rchip = NULL; - - /* initialize the PCI entry */ - err = pci_enable_device(pci); - if (err < 0) - return err; - /* check PCI availability (28bit DMA) */ - if (pci_set_dma_mask(pci, DMA_28BIT_MASK) < 0 || - pci_set_consistent_dma_mask(pci, DMA_28BIT_MASK) < 0) { - printk(KERN_ERR "error to set 28bit mask DMA\n"); - pci_disable_device(pci); - return -ENXIO; - } - - chip = kzalloc(sizeof(*chip), GFP_KERNEL); - if (chip == NULL) { - pci_disable_device(pci); - return -ENOMEM; - } - - /* initialize the stuff */ - chip->card = card; - chip->pci = pci; - chip->irq = -1; - - /* (1) PCI resource allocation */ - err = pci_request_regions(pci, "My Chip"); - if (err < 0) { - kfree(chip); - pci_disable_device(pci); - return err; - } - chip->port = pci_resource_start(pci, 0); - if (request_irq(pci->irq, snd_mychip_interrupt, - IRQF_SHARED, "My Chip", chip)) { - printk(KERN_ERR "cannot grab irq %d\n", pci->irq); - snd_mychip_free(chip); - return -EBUSY; - } - chip->irq = pci->irq; - - /* (2) initialization of the chip hardware */ - .... /* (not implemented in this document) */ - - err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); - if (err < 0) { - snd_mychip_free(chip); - return err; - } - - snd_card_set_dev(card, &pci->dev); - - *rchip = chip; - return 0; - } - - /* PCI IDs */ - static struct pci_device_id snd_mychip_ids[] = { - { PCI_VENDOR_ID_FOO, PCI_DEVICE_ID_BAR, - PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, - .... - { 0, } - }; - MODULE_DEVICE_TABLE(pci, snd_mychip_ids); - - /* pci_driver definition */ - static struct pci_driver driver = { - .name = "My Own Chip", - .id_table = snd_mychip_ids, - .probe = snd_mychip_probe, - .remove = __devexit_p(snd_mychip_remove), - }; - - /* module initialization */ - static int __init alsa_card_mychip_init(void) - { - return pci_register_driver(&driver); - } - - /* module clean up */ - static void __exit alsa_card_mychip_exit(void) - { - pci_unregister_driver(&driver); - } - - module_init(alsa_card_mychip_init) - module_exit(alsa_card_mychip_exit) - - EXPORT_NO_SYMBOLS; /* for old kernels only */ -]]> - - - -
- -
- Some Hafta's - - The allocation of PCI resources is done in the - probe() function, and usually an extra - xxx_create() function is written for this - purpose. - - - - In the case of PCI devices, you first have to call - the pci_enable_device() function before - allocating resources. Also, you need to set the proper PCI DMA - mask to limit the accessed I/O range. In some cases, you might - need to call pci_set_master() function, - too. - - - - Suppose the 28bit mask, and the code to be added would be like: - - - - - - - -
- -
- Resource Allocation - - The allocation of I/O ports and irqs is done via standard kernel - functions. Unlike ALSA ver.0.5.x., there are no helpers for - that. And these resources must be released in the destructor - function (see below). Also, on ALSA 0.9.x, you don't need to - allocate (pseudo-)DMA for PCI like in ALSA 0.5.x. - - - - Now assume that the PCI device has an I/O port with 8 bytes - and an interrupt. Then struct mychip will have the - following fields: - - - - - - - - - - For an I/O port (and also a memory region), you need to have - the resource pointer for the standard resource management. For - an irq, you have to keep only the irq number (integer). But you - need to initialize this number as -1 before actual allocation, - since irq 0 is valid. The port address and its resource pointer - can be initialized as null by - kzalloc() automatically, so you - don't have to take care of resetting them. - - - - The allocation of an I/O port is done like this: - - - -port = pci_resource_start(pci, 0); -]]> - - - - - - - It will reserve the I/O port region of 8 bytes of the given - PCI device. The returned value, chip->res_port, is allocated - via kmalloc() by - request_region(). The pointer must be - released via kfree(), but there is a - problem with this. This issue will be explained later. - - - - The allocation of an interrupt source is done like this: - - - -irq, snd_mychip_interrupt, - IRQF_SHARED, "My Chip", chip)) { - printk(KERN_ERR "cannot grab irq %d\n", pci->irq); - snd_mychip_free(chip); - return -EBUSY; - } - chip->irq = pci->irq; -]]> - - - - where snd_mychip_interrupt() is the - interrupt handler defined later. - Note that chip->irq should be defined - only when request_irq() succeeded. - - - - On the PCI bus, interrupts can be shared. Thus, - IRQF_SHARED is used as the interrupt flag of - request_irq(). - - - - The last argument of request_irq() is the - data pointer passed to the interrupt handler. Usually, the - chip-specific record is used for that, but you can use what you - like, too. - - - - I won't give details about the interrupt handler at this - point, but at least its appearance can be explained now. The - interrupt handler looks usually like the following: - - - - - - - - - - Now let's write the corresponding destructor for the resources - above. The role of destructor is simple: disable the hardware - (if already activated) and release the resources. So far, we - have no hardware part, so the disabling code is not written here. - - - - To release the resources, the check-and-release - method is a safer way. For the interrupt, do like this: - - - -irq >= 0) - free_irq(chip->irq, chip); -]]> - - - - Since the irq number can start from 0, you should initialize - chip->irq with a negative value (e.g. -1), so that you can - check the validity of the irq number as above. - - - - When you requested I/O ports or memory regions via - pci_request_region() or - pci_request_regions() like in this example, - release the resource(s) using the corresponding function, - pci_release_region() or - pci_release_regions(). - - - -pci); -]]> - - - - - - When you requested manually via request_region() - or request_mem_region, you can release it via - release_resource(). Suppose that you keep - the resource pointer returned from request_region() - in chip->res_port, the release procedure looks like: - - - -res_port); -]]> - - - - - - Don't forget to call pci_disable_device() - before the end. - - - - And finally, release the chip-specific record. - - - - - - - - - - Again, remember that you cannot - use the __devexit prefix for this destructor. - - - - We didn't implement the hardware disabling part in the above. - If you need to do this, please note that the destructor may be - called even before the initialization of the chip is completed. - It would be better to have a flag to skip hardware disabling - if the hardware was not initialized yet. - - - - When the chip-data is assigned to the card using - snd_device_new() with - SNDRV_DEV_LOWLELVEL , its destructor is - called at the last. That is, it is assured that all other - components like PCMs and controls have already been released. - You don't have to stop PCMs, etc. explicitly, but just - call low-level hardware stopping. - - - - The management of a memory-mapped region is almost as same as - the management of an I/O port. You'll need three fields like - the following: - - - - - - - - and the allocation would be like below: - - - -iobase_phys = pci_resource_start(pci, 0); - chip->iobase_virt = ioremap_nocache(chip->iobase_phys, - pci_resource_len(pci, 0)); -]]> - - - - and the corresponding destructor would be: - - - -iobase_virt) - iounmap(chip->iobase_virt); - .... - pci_release_regions(chip->pci); - .... - } -]]> - - - - -
- -
- Registration of Device Struct - - At some point, typically after calling snd_device_new(), - you need to register the struct device of the chip - you're handling for udev and co. ALSA provides a macro for compatibility with - older kernels. Simply call like the following: - - -dev); -]]> - - - so that it stores the PCI's device pointer to the card. This will be - referred by ALSA core functions later when the devices are registered. - - - In the case of non-PCI, pass the proper device struct pointer of the BUS - instead. (In the case of legacy ISA without PnP, you don't have to do - anything.) - -
- -
- PCI Entries - - So far, so good. Let's finish the missing PCI - stuff. At first, we need a - pci_device_id table for this - chipset. It's a table of PCI vendor/device ID number, and some - masks. - - - - For example, - - - - - - - - - - The first and second fields of - the pci_device_id structure are the vendor and - device IDs. If you have no reason to filter the matching - devices, you can leave the remaining fields as above. The last - field of the pci_device_id struct contains - private data for this entry. You can specify any value here, for - example, to define specific operations for supported device IDs. - Such an example is found in the intel8x0 driver. - - - - The last entry of this list is the terminator. You must - specify this all-zero entry. - - - - Then, prepare the pci_driver record: - - - - - - - - - - The probe and - remove functions have already - been defined in the previous sections. - The remove function should - be defined with the - __devexit_p() macro, so that it's not - defined for built-in (and non-hot-pluggable) case. The - name - field is the name string of this device. Note that you must not - use a slash / in this string. - - - - And at last, the module entries: - - - - - - - - - - Note that these module entries are tagged with - __init and - __exit prefixes, not - __devinit nor - __devexit. - - - - Oh, one thing was forgotten. If you have no exported symbols, - you need to declare it in 2.2 or 2.4 kernels (it's not necessary in 2.6 kernels). - - - - - - - - That's all! - -
- - - - - - - - PCM Interface - -
- General - - The PCM middle layer of ALSA is quite powerful and it is only - necessary for each driver to implement the low-level functions - to access its hardware. - - - - For accessing to the PCM layer, you need to include - <sound/pcm.h> first. In addition, - <sound/pcm_params.h> might be needed - if you access to some functions related with hw_param. - - - - Each card device can have up to four pcm instances. A pcm - instance corresponds to a pcm device file. The limitation of - number of instances comes only from the available bit size of - the Linux's device numbers. Once when 64bit device number is - used, we'll have more pcm instances available. - - - - A pcm instance consists of pcm playback and capture streams, - and each pcm stream consists of one or more pcm substreams. Some - soundcards support multiple playback functions. For example, - emu10k1 has a PCM playback of 32 stereo substreams. In this case, at - each open, a free substream is (usually) automatically chosen - and opened. Meanwhile, when only one substream exists and it was - already opened, the successful open will either block - or error with EAGAIN according to the - file open mode. But you don't have to care about such details in your - driver. The PCM middle layer will take care of such work. - -
- -
- Full Code Example - - The example code below does not include any hardware access - routines but shows only the skeleton, how to build up the PCM - interfaces. - - - PCM Example Code - - - .... - - /* hardware definition */ - static struct snd_pcm_hardware snd_mychip_playback_hw = { - .info = (SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP_VALID), - .formats = SNDRV_PCM_FMTBIT_S16_LE, - .rates = SNDRV_PCM_RATE_8000_48000, - .rate_min = 8000, - .rate_max = 48000, - .channels_min = 2, - .channels_max = 2, - .buffer_bytes_max = 32768, - .period_bytes_min = 4096, - .period_bytes_max = 32768, - .periods_min = 1, - .periods_max = 1024, - }; - - /* hardware definition */ - static struct snd_pcm_hardware snd_mychip_capture_hw = { - .info = (SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP_VALID), - .formats = SNDRV_PCM_FMTBIT_S16_LE, - .rates = SNDRV_PCM_RATE_8000_48000, - .rate_min = 8000, - .rate_max = 48000, - .channels_min = 2, - .channels_max = 2, - .buffer_bytes_max = 32768, - .period_bytes_min = 4096, - .period_bytes_max = 32768, - .periods_min = 1, - .periods_max = 1024, - }; - - /* open callback */ - static int snd_mychip_playback_open(struct snd_pcm_substream *substream) - { - struct mychip *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - - runtime->hw = snd_mychip_playback_hw; - /* more hardware-initialization will be done here */ - .... - return 0; - } - - /* close callback */ - static int snd_mychip_playback_close(struct snd_pcm_substream *substream) - { - struct mychip *chip = snd_pcm_substream_chip(substream); - /* the hardware-specific codes will be here */ - .... - return 0; - - } - - /* open callback */ - static int snd_mychip_capture_open(struct snd_pcm_substream *substream) - { - struct mychip *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - - runtime->hw = snd_mychip_capture_hw; - /* more hardware-initialization will be done here */ - .... - return 0; - } - - /* close callback */ - static int snd_mychip_capture_close(struct snd_pcm_substream *substream) - { - struct mychip *chip = snd_pcm_substream_chip(substream); - /* the hardware-specific codes will be here */ - .... - return 0; - - } - - /* hw_params callback */ - static int snd_mychip_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *hw_params) - { - return snd_pcm_lib_malloc_pages(substream, - params_buffer_bytes(hw_params)); - } - - /* hw_free callback */ - static int snd_mychip_pcm_hw_free(struct snd_pcm_substream *substream) - { - return snd_pcm_lib_free_pages(substream); - } - - /* prepare callback */ - static int snd_mychip_pcm_prepare(struct snd_pcm_substream *substream) - { - struct mychip *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - - /* set up the hardware with the current configuration - * for example... - */ - mychip_set_sample_format(chip, runtime->format); - mychip_set_sample_rate(chip, runtime->rate); - mychip_set_channels(chip, runtime->channels); - mychip_set_dma_setup(chip, runtime->dma_addr, - chip->buffer_size, - chip->period_size); - return 0; - } - - /* trigger callback */ - static int snd_mychip_pcm_trigger(struct snd_pcm_substream *substream, - int cmd) - { - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - /* do something to start the PCM engine */ - .... - break; - case SNDRV_PCM_TRIGGER_STOP: - /* do something to stop the PCM engine */ - .... - break; - default: - return -EINVAL; - } - } - - /* pointer callback */ - static snd_pcm_uframes_t - snd_mychip_pcm_pointer(struct snd_pcm_substream *substream) - { - struct mychip *chip = snd_pcm_substream_chip(substream); - unsigned int current_ptr; - - /* get the current hardware pointer */ - current_ptr = mychip_get_hw_pointer(chip); - return current_ptr; - } - - /* operators */ - static struct snd_pcm_ops snd_mychip_playback_ops = { - .open = snd_mychip_playback_open, - .close = snd_mychip_playback_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = snd_mychip_pcm_hw_params, - .hw_free = snd_mychip_pcm_hw_free, - .prepare = snd_mychip_pcm_prepare, - .trigger = snd_mychip_pcm_trigger, - .pointer = snd_mychip_pcm_pointer, - }; - - /* operators */ - static struct snd_pcm_ops snd_mychip_capture_ops = { - .open = snd_mychip_capture_open, - .close = snd_mychip_capture_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = snd_mychip_pcm_hw_params, - .hw_free = snd_mychip_pcm_hw_free, - .prepare = snd_mychip_pcm_prepare, - .trigger = snd_mychip_pcm_trigger, - .pointer = snd_mychip_pcm_pointer, - }; - - /* - * definitions of capture are omitted here... - */ - - /* create a pcm device */ - static int __devinit snd_mychip_new_pcm(struct mychip *chip) - { - struct snd_pcm *pcm; - int err; - - err = snd_pcm_new(chip->card, "My Chip", 0, 1, 1, &pcm); - if (err < 0) - return err; - pcm->private_data = chip; - strcpy(pcm->name, "My Chip"); - chip->pcm = pcm; - /* set operators */ - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, - &snd_mychip_playback_ops); - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, - &snd_mychip_capture_ops); - /* pre-allocation of buffers */ - /* NOTE: this may fail */ - snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(chip->pci), - 64*1024, 64*1024); - return 0; - } -]]> - - - -
- -
- Constructor - - A pcm instance is allocated by the snd_pcm_new() - function. It would be better to create a constructor for pcm, - namely, - - - -card, "My Chip", 0, 1, 1, &pcm); - if (err < 0) - return err; - pcm->private_data = chip; - strcpy(pcm->name, "My Chip"); - chip->pcm = pcm; - .... - return 0; - } -]]> - - - - - - The snd_pcm_new() function takes four - arguments. The first argument is the card pointer to which this - pcm is assigned, and the second is the ID string. - - - - The third argument (index, 0 in the - above) is the index of this new pcm. It begins from zero. If - you create more than one pcm instances, specify the - different numbers in this argument. For example, - index = 1 for the second PCM device. - - - - The fourth and fifth arguments are the number of substreams - for playback and capture, respectively. Here 1 is used for - both arguments. When no playback or capture substreams are available, - pass 0 to the corresponding argument. - - - - If a chip supports multiple playbacks or captures, you can - specify more numbers, but they must be handled properly in - open/close, etc. callbacks. When you need to know which - substream you are referring to, then it can be obtained from - struct snd_pcm_substream data passed to each callback - as follows: - - - -number; -]]> - - - - - - After the pcm is created, you need to set operators for each - pcm stream. - - - - - - - - - - The operators are defined typically like this: - - - - - - - - All the callbacks are described in the - - Operators subsection. - - - - After setting the operators, you probably will want to - pre-allocate the buffer. For the pre-allocation, simply call - the following: - - - -pci), - 64*1024, 64*1024); -]]> - - - - It will allocate a buffer up to 64kB as default. - Buffer management details will be described in the later section Buffer and Memory - Management. - - - - Additionally, you can set some extra information for this pcm - in pcm->info_flags. - The available values are defined as - SNDRV_PCM_INFO_XXX in - <sound/asound.h>, which is used for - the hardware definition (described later). When your soundchip - supports only half-duplex, specify like this: - - - -info_flags = SNDRV_PCM_INFO_HALF_DUPLEX; -]]> - - - -
- -
- ... And the Destructor? - - The destructor for a pcm instance is not always - necessary. Since the pcm device will be released by the middle - layer code automatically, you don't have to call the destructor - explicitly. - - - - The destructor would be necessary if you created - special records internally and needed to release them. In such a - case, set the destructor function to - pcm->private_free: - - - PCM Instance with a Destructor - -my_private_pcm_data); - /* do what you like else */ - .... - } - - static int __devinit snd_mychip_new_pcm(struct mychip *chip) - { - struct snd_pcm *pcm; - .... - /* allocate your own data */ - chip->my_private_pcm_data = kmalloc(...); - /* set the destructor */ - pcm->private_data = chip; - pcm->private_free = mychip_pcm_free; - .... - } -]]> - - - -
- -
- Runtime Pointer - The Chest of PCM Information - - When the PCM substream is opened, a PCM runtime instance is - allocated and assigned to the substream. This pointer is - accessible via substream->runtime. - This runtime pointer holds most information you need - to control the PCM: the copy of hw_params and sw_params configurations, the buffer - pointers, mmap records, spinlocks, etc. - - - - The definition of runtime instance is found in - <sound/pcm.h>. Here are - the contents of this file: - - - - - - - - - For the operators (callbacks) of each sound driver, most of - these records are supposed to be read-only. Only the PCM - middle-layer changes / updates them. The exceptions are - the hardware description (hw), interrupt callbacks - (transfer_ack_xxx), DMA buffer information, and the private - data. Besides, if you use the standard buffer allocation - method via snd_pcm_lib_malloc_pages(), - you don't need to set the DMA buffer information by yourself. - - - - In the sections below, important records are explained. - - -
- Hardware Description - - The hardware descriptor (struct snd_pcm_hardware) - contains the definitions of the fundamental hardware - configuration. Above all, you'll need to define this in - - the open callback. - Note that the runtime instance holds the copy of the - descriptor, not the pointer to the existing descriptor. That - is, in the open callback, you can modify the copied descriptor - (runtime->hw) as you need. For example, if the maximum - number of channels is 1 only on some chip models, you can - still use the same hardware descriptor and change the - channels_max later: - - -runtime; - ... - runtime->hw = snd_mychip_playback_hw; /* common definition */ - if (chip->model == VERY_OLD_ONE) - runtime->hw.channels_max = 1; -]]> - - - - - - Typically, you'll have a hardware descriptor as below: - - - - - - - - - - - The info field contains the type and - capabilities of this pcm. The bit flags are defined in - <sound/asound.h> as - SNDRV_PCM_INFO_XXX. Here, at least, you - have to specify whether the mmap is supported and which - interleaved format is supported. - When the is supported, add the - SNDRV_PCM_INFO_MMAP flag here. When the - hardware supports the interleaved or the non-interleaved - formats, SNDRV_PCM_INFO_INTERLEAVED or - SNDRV_PCM_INFO_NONINTERLEAVED flag must - be set, respectively. If both are supported, you can set both, - too. - - - - In the above example, MMAP_VALID and - BLOCK_TRANSFER are specified for the OSS mmap - mode. Usually both are set. Of course, - MMAP_VALID is set only if the mmap is - really supported. - - - - The other possible flags are - SNDRV_PCM_INFO_PAUSE and - SNDRV_PCM_INFO_RESUME. The - PAUSE bit means that the pcm supports the - pause operation, while the - RESUME bit means that the pcm supports - the full suspend/resume operation. - If the PAUSE flag is set, - the trigger callback below - must handle the corresponding (pause push/release) commands. - The suspend/resume trigger commands can be defined even without - the RESUME flag. See - Power Management section for details. - - - - When the PCM substreams can be synchronized (typically, - synchronized start/stop of a playback and a capture streams), - you can give SNDRV_PCM_INFO_SYNC_START, - too. In this case, you'll need to check the linked-list of - PCM substreams in the trigger callback. This will be - described in the later section. - - - - - - formats field contains the bit-flags - of supported formats (SNDRV_PCM_FMTBIT_XXX). - If the hardware supports more than one format, give all or'ed - bits. In the example above, the signed 16bit little-endian - format is specified. - - - - - - rates field contains the bit-flags of - supported rates (SNDRV_PCM_RATE_XXX). - When the chip supports continuous rates, pass - CONTINUOUS bit additionally. - The pre-defined rate bits are provided only for typical - rates. If your chip supports unconventional rates, you need to add - the KNOT bit and set up the hardware - constraint manually (explained later). - - - - - - rate_min and - rate_max define the minimum and - maximum sample rate. This should correspond somehow to - rates bits. - - - - - - channel_min and - channel_max - define, as you might already expected, the minimum and maximum - number of channels. - - - - - - buffer_bytes_max defines the - maximum buffer size in bytes. There is no - buffer_bytes_min field, since - it can be calculated from the minimum period size and the - minimum number of periods. - Meanwhile, period_bytes_min and - define the minimum and maximum size of the period in bytes. - periods_max and - periods_min define the maximum and - minimum number of periods in the buffer. - - - - The period is a term that corresponds to - a fragment in the OSS world. The period defines the size at - which a PCM interrupt is generated. This size strongly - depends on the hardware. - Generally, the smaller period size will give you more - interrupts, that is, more controls. - In the case of capture, this size defines the input latency. - On the other hand, the whole buffer size defines the - output latency for the playback direction. - - - - - - There is also a field fifo_size. - This specifies the size of the hardware FIFO, but currently it - is neither used in the driver nor in the alsa-lib. So, you - can ignore this field. - - - - -
- -
- PCM Configurations - - Ok, let's go back again to the PCM runtime records. - The most frequently referred records in the runtime instance are - the PCM configurations. - The PCM configurations are stored in the runtime instance - after the application sends hw_params data via - alsa-lib. There are many fields copied from hw_params and - sw_params structs. For example, - format holds the format type - chosen by the application. This field contains the enum value - SNDRV_PCM_FORMAT_XXX. - - - - One thing to be noted is that the configured buffer and period - sizes are stored in frames in the runtime. - In the ALSA world, 1 frame = channels * samples-size. - For conversion between frames and bytes, you can use the - frames_to_bytes() and - bytes_to_frames() helper functions. - - -period_size); -]]> - - - - - - Also, many software parameters (sw_params) are - stored in frames, too. Please check the type of the field. - snd_pcm_uframes_t is for the frames as unsigned - integer while snd_pcm_sframes_t is for the frames - as signed integer. - -
- -
- DMA Buffer Information - - The DMA buffer is defined by the following four fields, - dma_area, - dma_addr, - dma_bytes and - dma_private. - The dma_area holds the buffer - pointer (the logical address). You can call - memcpy from/to - this pointer. Meanwhile, dma_addr - holds the physical address of the buffer. This field is - specified only when the buffer is a linear buffer. - dma_bytes holds the size of buffer - in bytes. dma_private is used for - the ALSA DMA allocator. - - - - If you use a standard ALSA function, - snd_pcm_lib_malloc_pages(), for - allocating the buffer, these fields are set by the ALSA middle - layer, and you should not change them by - yourself. You can read them but not write them. - On the other hand, if you want to allocate the buffer by - yourself, you'll need to manage it in hw_params callback. - At least, dma_bytes is mandatory. - dma_area is necessary when the - buffer is mmapped. If your driver doesn't support mmap, this - field is not necessary. dma_addr - is also optional. You can use - dma_private as you like, too. - -
- -
- Running Status - - The running status can be referred via runtime->status. - This is the pointer to the struct snd_pcm_mmap_status - record. For example, you can get the current DMA hardware - pointer via runtime->status->hw_ptr. - - - - The DMA application pointer can be referred via - runtime->control, which points to the - struct snd_pcm_mmap_control record. - However, accessing directly to this value is not recommended. - -
- -
- Private Data - - You can allocate a record for the substream and store it in - runtime->private_data. Usually, this - is done in - - the open callback. - Don't mix this with pcm->private_data. - The pcm->private_data usually points to the - chip instance assigned statically at the creation of PCM, while the - runtime->private_data points to a dynamic - data structure created at the PCM open callback. - - - -runtime->private_data = data; - .... - } -]]> - - - - - - The allocated object must be released in - - the close callback. - -
- -
- Interrupt Callbacks - - The field transfer_ack_begin and - transfer_ack_end are called at - the beginning and at the end of - snd_pcm_period_elapsed(), respectively. - -
- -
- -
- Operators - - OK, now let me give details about each pcm callback - (ops). In general, every callback must - return 0 if successful, or a negative error number - such as -EINVAL. To choose an appropriate - error number, it is advised to check what value other parts of - the kernel return when the same kind of request fails. - - - - The callback function takes at least the argument with - snd_pcm_substream pointer. To retrieve - the chip record from the given substream instance, you can use the - following macro. - - - - - - - - The macro reads substream->private_data, - which is a copy of pcm->private_data. - You can override the former if you need to assign different data - records per PCM substream. For example, the cmi8330 driver assigns - different private_data for playback and capture directions, - because it uses two different codecs (SB- and AD-compatible) for - different directions. - - -
- open callback - - - - - - - - This is called when a pcm substream is opened. - - - - At least, here you have to initialize the runtime->hw - record. Typically, this is done by like this: - - - -runtime; - - runtime->hw = snd_mychip_playback_hw; - return 0; - } -]]> - - - - where snd_mychip_playback_hw is the - pre-defined hardware description. - - - - You can allocate a private data in this callback, as described - in - Private Data section. - - - - If the hardware configuration needs more constraints, set the - hardware constraints here, too. - See - Constraints for more details. - -
- -
- close callback - - - - - - - - Obviously, this is called when a pcm substream is closed. - - - - Any private instance for a pcm substream allocated in the - open callback will be released here. - - - -runtime->private_data); - .... - } -]]> - - - -
- -
- ioctl callback - - This is used for any special call to pcm ioctls. But - usually you can pass a generic ioctl callback, - snd_pcm_lib_ioctl. - -
- -
- hw_params callback - - - - - - - - - - This is called when the hardware parameter - (hw_params) is set - up by the application, - that is, once when the buffer size, the period size, the - format, etc. are defined for the pcm substream. - - - - Many hardware setups should be done in this callback, - including the allocation of buffers. - - - - Parameters to be initialized are retrieved by - params_xxx() macros. To allocate - buffer, you can call a helper function, - - - - - - - - snd_pcm_lib_malloc_pages() is available - only when the DMA buffers have been pre-allocated. - See the section - Buffer Types for more details. - - - - Note that this and prepare callbacks - may be called multiple times per initialization. - For example, the OSS emulation may - call these callbacks at each change via its ioctl. - - - - Thus, you need to be careful not to allocate the same buffers - many times, which will lead to memory leaks! Calling the - helper function above many times is OK. It will release the - previous buffer automatically when it was already allocated. - - - - Another note is that this callback is non-atomic - (schedulable). This is important, because the - trigger callback - is atomic (non-schedulable). That is, mutexes or any - schedule-related functions are not available in - trigger callback. - Please see the subsection - - Atomicity for details. - -
- -
- hw_free callback - - - - - - - - - - This is called to release the resources allocated via - hw_params. For example, releasing the - buffer via - snd_pcm_lib_malloc_pages() is done by - calling the following: - - - - - - - - - - This function is always called before the close callback is called. - Also, the callback may be called multiple times, too. - Keep track whether the resource was already released. - -
- -
- prepare callback - - - - - - - - - - This callback is called when the pcm is - prepared. You can set the format type, sample - rate, etc. here. The difference from - hw_params is that the - prepare callback will be called each - time - snd_pcm_prepare() is called, i.e. when - recovering after underruns, etc. - - - - Note that this callback is now non-atomic. - You can use schedule-related functions safely in this callback. - - - - In this and the following callbacks, you can refer to the - values via the runtime record, - substream->runtime. - For example, to get the current - rate, format or channels, access to - runtime->rate, - runtime->format or - runtime->channels, respectively. - The physical address of the allocated buffer is set to - runtime->dma_area. The buffer and period sizes are - in runtime->buffer_size and runtime->period_size, - respectively. - - - - Be careful that this callback will be called many times at - each setup, too. - -
- -
- trigger callback - - - - - - - - This is called when the pcm is started, stopped or paused. - - - - Which action is specified in the second argument, - SNDRV_PCM_TRIGGER_XXX in - <sound/pcm.h>. At least, - the START and STOP - commands must be defined in this callback. - - - - - - - - - - When the pcm supports the pause operation (given in the info - field of the hardware table), the PAUSE_PUSE - and PAUSE_RELEASE commands must be - handled here, too. The former is the command to pause the pcm, - and the latter to restart the pcm again. - - - - When the pcm supports the suspend/resume operation, - regardless of full or partial suspend/resume support, - the SUSPEND and RESUME - commands must be handled, too. - These commands are issued when the power-management status is - changed. Obviously, the SUSPEND and - RESUME commands - suspend and resume the pcm substream, and usually, they - are identical to the STOP and - START commands, respectively. - See the - Power Management section for details. - - - - As mentioned, this callback is atomic. You cannot call - functions which may sleep. - The trigger callback should be as minimal as possible, - just really triggering the DMA. The other stuff should be - initialized hw_params and prepare callbacks properly - beforehand. - -
- -
- pointer callback - - - - - - - - This callback is called when the PCM middle layer inquires - the current hardware position on the buffer. The position must - be returned in frames, - ranging from 0 to buffer_size - 1. - - - - This is called usually from the buffer-update routine in the - pcm middle layer, which is invoked when - snd_pcm_period_elapsed() is called in the - interrupt routine. Then the pcm middle layer updates the - position and calculates the available space, and wakes up the - sleeping poll threads, etc. - - - - This callback is also atomic. - -
- -
- copy and silence callbacks - - These callbacks are not mandatory, and can be omitted in - most cases. These callbacks are used when the hardware buffer - cannot be in the normal memory space. Some chips have their - own buffer on the hardware which is not mappable. In such a - case, you have to transfer the data manually from the memory - buffer to the hardware buffer. Or, if the buffer is - non-contiguous on both physical and virtual memory spaces, - these callbacks must be defined, too. - - - - If these two callbacks are defined, copy and set-silence - operations are done by them. The detailed will be described in - the later section Buffer and Memory - Management. - -
- -
- ack callback - - This callback is also not mandatory. This callback is called - when the appl_ptr is updated in read or write operations. - Some drivers like emu10k1-fx and cs46xx need to track the - current appl_ptr for the internal buffer, and this callback - is useful only for such a purpose. - - - This callback is atomic. - -
- -
- page callback - - - This callback is optional too. This callback is used - mainly for non-contiguous buffers. The mmap calls this - callback to get the page address. Some examples will be - explained in the later section Buffer and Memory - Management, too. - -
-
- -
- Interrupt Handler - - The rest of pcm stuff is the PCM interrupt handler. The - role of PCM interrupt handler in the sound driver is to update - the buffer position and to tell the PCM middle layer when the - buffer position goes across the prescribed period size. To - inform this, call the snd_pcm_period_elapsed() - function. - - - - There are several types of sound chips to generate the interrupts. - - -
- Interrupts at the period (fragment) boundary - - This is the most frequently found type: the hardware - generates an interrupt at each period boundary. - In this case, you can call - snd_pcm_period_elapsed() at each - interrupt. - - - - snd_pcm_period_elapsed() takes the - substream pointer as its argument. Thus, you need to keep the - substream pointer accessible from the chip instance. For - example, define substream field in the chip record to hold the - current running substream pointer, and set the pointer value - at open callback (and reset at close callback). - - - - If you acquire a spinlock in the interrupt handler, and the - lock is used in other pcm callbacks, too, then you have to - release the lock before calling - snd_pcm_period_elapsed(), because - snd_pcm_period_elapsed() calls other pcm - callbacks inside. - - - - Typical code would be like: - - - Interrupt Handler Case #1 - -lock); - .... - if (pcm_irq_invoked(chip)) { - /* call updater, unlock before it */ - spin_unlock(&chip->lock); - snd_pcm_period_elapsed(chip->substream); - spin_lock(&chip->lock); - /* acknowledge the interrupt if necessary */ - } - .... - spin_unlock(&chip->lock); - return IRQ_HANDLED; - } -]]> - - - -
- -
- High frequency timer interrupts - - This happense when the hardware doesn't generate interrupts - at the period boundary but issues timer interrupts at a fixed - timer rate (e.g. es1968 or ymfpci drivers). - In this case, you need to check the current hardware - position and accumulate the processed sample length at each - interrupt. When the accumulated size exceeds the period - size, call - snd_pcm_period_elapsed() and reset the - accumulator. - - - - Typical code would be like the following. - - - Interrupt Handler Case #2 - -lock); - .... - if (pcm_irq_invoked(chip)) { - unsigned int last_ptr, size; - /* get the current hardware pointer (in frames) */ - last_ptr = get_hw_ptr(chip); - /* calculate the processed frames since the - * last update - */ - if (last_ptr < chip->last_ptr) - size = runtime->buffer_size + last_ptr - - chip->last_ptr; - else - size = last_ptr - chip->last_ptr; - /* remember the last updated point */ - chip->last_ptr = last_ptr; - /* accumulate the size */ - chip->size += size; - /* over the period boundary? */ - if (chip->size >= runtime->period_size) { - /* reset the accumulator */ - chip->size %= runtime->period_size; - /* call updater */ - spin_unlock(&chip->lock); - snd_pcm_period_elapsed(substream); - spin_lock(&chip->lock); - } - /* acknowledge the interrupt if necessary */ - } - .... - spin_unlock(&chip->lock); - return IRQ_HANDLED; - } -]]> - - - -
- -
- On calling <function>snd_pcm_period_elapsed()</function> - - In both cases, even if more than one period are elapsed, you - don't have to call - snd_pcm_period_elapsed() many times. Call - only once. And the pcm layer will check the current hardware - pointer and update to the latest status. - -
-
- -
- Atomicity - - One of the most important (and thus difficult to debug) problems - in kernel programming are race conditions. - In the Linux kernel, they are usually avoided via spin-locks, mutexes - or semaphores. In general, if a race condition can happen - in an interrupt handler, it has to be managed atomically, and you - have to use a spinlock to protect the critical session. If the - critical section is not in interrupt handler code and - if taking a relatively long time to execute is acceptable, you - should use mutexes or semaphores instead. - - - - As already seen, some pcm callbacks are atomic and some are - not. For example, the hw_params callback is - non-atomic, while trigger callback is - atomic. This means, the latter is called already in a spinlock - held by the PCM middle layer. Please take this atomicity into - account when you choose a locking scheme in the callbacks. - - - - In the atomic callbacks, you cannot use functions which may call - schedule or go to - sleep. Semaphores and mutexes can sleep, - and hence they cannot be used inside the atomic callbacks - (e.g. trigger callback). - To implement some delay in such a callback, please use - udelay() or mdelay(). - - - - All three atomic callbacks (trigger, pointer, and ack) are - called with local interrupts disabled. - - -
-
- Constraints - - If your chip supports unconventional sample rates, or only the - limited samples, you need to set a constraint for the - condition. - - - - For example, in order to restrict the sample rates in the some - supported values, use - snd_pcm_hw_constraint_list(). - You need to call this function in the open callback. - - - Example of Hardware Constraints - -runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - &constraints_rates); - if (err < 0) - return err; - .... - } -]]> - - - - - - There are many different constraints. - Look at sound/pcm.h for a complete list. - You can even define your own constraint rules. - For example, let's suppose my_chip can manage a substream of 1 channel - if and only if the format is S16_LE, otherwise it supports any format - specified in the snd_pcm_hardware structure (or in any - other constraint_list). You can build a rule like this: - - - Example of Hardware Constraints for Channels - -min < 2) { - fmt.bits[0] &= SNDRV_PCM_FMTBIT_S16_LE; - return snd_mask_refine(f, &fmt); - } - return 0; - } -]]> - - - - - - Then you need to call this function to add your rule: - - - -runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - hw_rule_channels_by_format, 0, SNDRV_PCM_HW_PARAM_FORMAT, - -1); -]]> - - - - - - The rule function is called when an application sets the number of - channels. But an application can set the format before the number of - channels. Thus you also need to define the inverse rule: - - - Example of Hardware Constraints for Channels - -bits[0] == SNDRV_PCM_FMTBIT_S16_LE) { - ch.min = ch.max = 1; - ch.integer = 1; - return snd_interval_refine(c, &ch); - } - return 0; - } -]]> - - - - - - ...and in the open callback: - - -runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, - hw_rule_format_by_channels, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - -1); -]]> - - - - - - I won't give more details here, rather I - would like to say, Luke, use the source. - -
- - - - - - - - - Control Interface - -
- General - - The control interface is used widely for many switches, - sliders, etc. which are accessed from user-space. Its most - important use is the mixer interface. In other words, since ALSA - 0.9.x, all the mixer stuff is implemented on the control kernel API. - - - - ALSA has a well-defined AC97 control module. If your chip - supports only the AC97 and nothing else, you can skip this - section. - - - - The control API is defined in - <sound/control.h>. - Include this file if you want to add your own controls. - -
- -
- Definition of Controls - - To create a new control, you need to define the - following three - callbacks: info, - get and - put. Then, define a - struct snd_kcontrol_new record, such as: - - - Definition of a Control - - - - - - - - Most likely the control is created via - snd_ctl_new1(), and in such a case, you can - add the __devinitdata prefix to the - definition as above. - - - - The iface field specifies the control - type, SNDRV_CTL_ELEM_IFACE_XXX, which - is usually MIXER. - Use CARD for global controls that are not - logically part of the mixer. - If the control is closely associated with some specific device on - the sound card, use HWDEP, - PCM, RAWMIDI, - TIMER, or SEQUENCER, and - specify the device number with the - device and - subdevice fields. - - - - The name is the name identifier - string. Since ALSA 0.9.x, the control name is very important, - because its role is classified from its name. There are - pre-defined standard control names. The details are described in - the - Control Names subsection. - - - - The index field holds the index number - of this control. If there are several different controls with - the same name, they can be distinguished by the index - number. This is the case when - several codecs exist on the card. If the index is zero, you can - omit the definition above. - - - - The access field contains the access - type of this control. Give the combination of bit masks, - SNDRV_CTL_ELEM_ACCESS_XXX, there. - The details will be explained in - the - Access Flags subsection. - - - - The private_value field contains - an arbitrary long integer value for this record. When using - the generic info, - get and - put callbacks, you can pass a value - through this field. If several small numbers are necessary, you can - combine them in bitwise. Or, it's possible to give a pointer - (casted to unsigned long) of some record to this field, too. - - - - The tlv field can be used to provide - metadata about the control; see the - - Metadata subsection. - - - - The other three are - - callback functions. - -
- -
- Control Names - - There are some standards to define the control names. A - control is usually defined from the three parts as - SOURCE DIRECTION FUNCTION. - - - - The first, SOURCE, specifies the source - of the control, and is a string such as Master, - PCM, CD and - Line. There are many pre-defined sources. - - - - The second, DIRECTION, is one of the - following strings according to the direction of the control: - Playback, Capture, Bypass - Playback and Bypass Capture. Or, it can - be omitted, meaning both playback and capture directions. - - - - The third, FUNCTION, is one of the - following strings according to the function of the control: - Switch, Volume and - Route. - - - - The example of control names are, thus, Master Capture - Switch or PCM Playback Volume. - - - - There are some exceptions: - - -
- Global capture and playback - - Capture Source, Capture Switch - and Capture Volume are used for the global - capture (input) source, switch and volume. Similarly, - Playback Switch and Playback - Volume are used for the global output gain switch and - volume. - -
- -
- Tone-controls - - tone-control switch and volumes are specified like - Tone Control - XXX, e.g. Tone Control - - Switch, Tone Control - Bass, - Tone Control - Center. - -
- -
- 3D controls - - 3D-control switches and volumes are specified like 3D - Control - XXX, e.g. 3D Control - - Switch, 3D Control - Center, 3D - Control - Space. - -
- -
- Mic boost - - Mic-boost switch is set as Mic Boost or - Mic Boost (6dB). - - - - More precise information can be found in - Documentation/sound/alsa/ControlNames.txt. - -
-
- -
- Access Flags - - - The access flag is the bitmask which specifies the access type - of the given control. The default access type is - SNDRV_CTL_ELEM_ACCESS_READWRITE, - which means both read and write are allowed to this control. - When the access flag is omitted (i.e. = 0), it is - considered as READWRITE access as default. - - - - When the control is read-only, pass - SNDRV_CTL_ELEM_ACCESS_READ instead. - In this case, you don't have to define - the put callback. - Similarly, when the control is write-only (although it's a rare - case), you can use the WRITE flag instead, and - you don't need the get callback. - - - - If the control value changes frequently (e.g. the VU meter), - VOLATILE flag should be given. This means - that the control may be changed without - - notification. Applications should poll such - a control constantly. - - - - When the control is inactive, set - the INACTIVE flag, too. - There are LOCK and - OWNER flags to change the write - permissions. - - -
- -
- Callbacks - -
- info callback - - The info callback is used to get - detailed information on this control. This must store the - values of the given struct snd_ctl_elem_info - object. For example, for a boolean control with a single - element: - - - Example of info callback - -type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; - } -]]> - - - - - - The type field specifies the type - of the control. There are BOOLEAN, - INTEGER, ENUMERATED, - BYTES, IEC958 and - INTEGER64. The - count field specifies the - number of elements in this control. For example, a stereo - volume would have count = 2. The - value field is a union, and - the values stored are depending on the type. The boolean and - integer types are identical. - - - - The enumerated type is a bit different from others. You'll - need to set the string for the currently given item index. - - - -type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 4; - if (uinfo->value.enumerated.item > 3) - uinfo->value.enumerated.item = 3; - strcpy(uinfo->value.enumerated.name, - texts[uinfo->value.enumerated.item]); - return 0; - } -]]> - - - - - - Some common info callbacks are available for your convenience: - snd_ctl_boolean_mono_info() and - snd_ctl_boolean_stereo_info(). - Obviously, the former is an info callback for a mono channel - boolean item, just like snd_myctl_mono_info - above, and the latter is for a stereo channel boolean item. - - -
- -
- get callback - - - This callback is used to read the current value of the - control and to return to user-space. - - - - For example, - - - Example of get callback - -value.integer.value[0] = get_some_value(chip); - return 0; - } -]]> - - - - - - The value field depends on - the type of control as well as on the info callback. For example, - the sb driver uses this field to store the register offset, - the bit-shift and the bit-mask. The - private_value field is set as follows: - - - - - - and is retrieved in callbacks like - - -private_value & 0xff; - int shift = (kcontrol->private_value >> 16) & 0xff; - int mask = (kcontrol->private_value >> 24) & 0xff; - .... - } -]]> - - - - - - In the get callback, - you have to fill all the elements if the - control has more than one elements, - i.e. count > 1. - In the example above, we filled only one element - (value.integer.value[0]) since it's - assumed as count = 1. - -
- -
- put callback - - - This callback is used to write a value from user-space. - - - - For example, - - - Example of put callback - -current_value != - ucontrol->value.integer.value[0]) { - change_current_value(chip, - ucontrol->value.integer.value[0]); - changed = 1; - } - return changed; - } -]]> - - - - As seen above, you have to return 1 if the value is - changed. If the value is not changed, return 0 instead. - If any fatal error happens, return a negative error code as - usual. - - - - As in the get callback, - when the control has more than one elements, - all elements must be evaluated in this callback, too. - -
- -
- Callbacks are not atomic - - All these three callbacks are basically not atomic. - -
-
- -
- Constructor - - When everything is ready, finally we can create a new - control. To create a control, there are two functions to be - called, snd_ctl_new1() and - snd_ctl_add(). - - - - In the simplest way, you can do like this: - - - - - - - - where my_control is the - struct snd_kcontrol_new object defined above, and chip - is the object pointer to be passed to - kcontrol->private_data - which can be referred to in callbacks. - - - - snd_ctl_new1() allocates a new - snd_kcontrol instance (that's why the definition - of my_control can be with - the __devinitdata - prefix), and snd_ctl_add assigns the given - control component to the card. - -
- -
- Change Notification - - If you need to change and update a control in the interrupt - routine, you can call snd_ctl_notify(). For - example, - - - - - - - - This function takes the card pointer, the event-mask, and the - control id pointer for the notification. The event-mask - specifies the types of notification, for example, in the above - example, the change of control values is notified. - The id pointer is the pointer of struct snd_ctl_elem_id - to be notified. - You can find some examples in es1938.c or - es1968.c for hardware volume interrupts. - -
- -
- Metadata - - To provide information about the dB values of a mixer control, use - on of the DECLARE_TLV_xxx macros from - <sound/tlv.h> to define a variable - containing this information, set thetlv.p - field to point to this variable, and include the - SNDRV_CTL_ELEM_ACCESS_TLV_READ flag in the - access field; like this: - - - - - - - - - The DECLARE_TLV_DB_SCALE macro defines - information about a mixer control where each step in the control's - value changes the dB value by a constant dB amount. - The first parameter is the name of the variable to be defined. - The second parameter is the minimum value, in units of 0.01 dB. - The third parameter is the step size, in units of 0.01 dB. - Set the fourth parameter to 1 if the minimum value actually mutes - the control. - - - - The DECLARE_TLV_DB_LINEAR macro defines - information about a mixer control where the control's value affects - the output linearly. - The first parameter is the name of the variable to be defined. - The second parameter is the minimum value, in units of 0.01 dB. - The third parameter is the maximum value, in units of 0.01 dB. - If the minimum value mutes the control, set the second parameter to - TLV_DB_GAIN_MUTE. - -
- - - - - - - - - API for AC97 Codec - -
- General - - The ALSA AC97 codec layer is a well-defined one, and you don't - have to write much code to control it. Only low-level control - routines are necessary. The AC97 codec API is defined in - <sound/ac97_codec.h>. - -
- -
- Full Code Example - - - Example of AC97 Interface - -private_data; - .... - /* read a register value here from the codec */ - return the_register_value; - } - - static void snd_mychip_ac97_write(struct snd_ac97 *ac97, - unsigned short reg, unsigned short val) - { - struct mychip *chip = ac97->private_data; - .... - /* write the given register value to the codec */ - } - - static int snd_mychip_ac97(struct mychip *chip) - { - struct snd_ac97_bus *bus; - struct snd_ac97_template ac97; - int err; - static struct snd_ac97_bus_ops ops = { - .write = snd_mychip_ac97_write, - .read = snd_mychip_ac97_read, - }; - - err = snd_ac97_bus(chip->card, 0, &ops, NULL, &bus); - if (err < 0) - return err; - memset(&ac97, 0, sizeof(ac97)); - ac97.private_data = chip; - return snd_ac97_mixer(bus, &ac97, &chip->ac97); - } - -]]> - - - -
- -
- Constructor - - To create an ac97 instance, first call snd_ac97_bus - with an ac97_bus_ops_t record with callback functions. - - - - - - - - The bus record is shared among all belonging ac97 instances. - - - - And then call snd_ac97_mixer() with an - struct snd_ac97_template - record together with the bus pointer created above. - - - -ac97); -]]> - - - - where chip->ac97 is a pointer to a newly created - ac97_t instance. - In this case, the chip pointer is set as the private data, so that - the read/write callback functions can refer to this chip instance. - This instance is not necessarily stored in the chip - record. If you need to change the register values from the - driver, or need the suspend/resume of ac97 codecs, keep this - pointer to pass to the corresponding functions. - -
- -
- Callbacks - - The standard callbacks are read and - write. Obviously they - correspond to the functions for read and write accesses to the - hardware low-level codes. - - - - The read callback returns the - register value specified in the argument. - - - -private_data; - .... - return the_register_value; - } -]]> - - - - Here, the chip can be cast from ac97->private_data. - - - - Meanwhile, the write callback is - used to set the register value. - - - - - - - - - - These callbacks are non-atomic like the control API callbacks. - - - - There are also other callbacks: - reset, - wait and - init. - - - - The reset callback is used to reset - the codec. If the chip requires a special kind of reset, you can - define this callback. - - - - The wait callback is used to - add some waiting time in the standard initialization of the codec. If the - chip requires the extra waiting time, define this callback. - - - - The init callback is used for - additional initialization of the codec. - -
- -
- Updating Registers in The Driver - - If you need to access to the codec from the driver, you can - call the following functions: - snd_ac97_write(), - snd_ac97_read(), - snd_ac97_update() and - snd_ac97_update_bits(). - - - - Both snd_ac97_write() and - snd_ac97_update() functions are used to - set a value to the given register - (AC97_XXX). The difference between them is - that snd_ac97_update() doesn't write a - value if the given value has been already set, while - snd_ac97_write() always rewrites the - value. - - - - - - - - - - snd_ac97_read() is used to read the value - of the given register. For example, - - - - - - - - - - snd_ac97_update_bits() is used to update - some bits in the given register. - - - - - - - - - - Also, there is a function to change the sample rate (of a - given register such as - AC97_PCM_FRONT_DAC_RATE) when VRA or - DRA is supported by the codec: - snd_ac97_set_rate(). - - - - - - - - - - The following registers are available to set the rate: - AC97_PCM_MIC_ADC_RATE, - AC97_PCM_FRONT_DAC_RATE, - AC97_PCM_LR_ADC_RATE, - AC97_SPDIF. When - AC97_SPDIF is specified, the register is - not really changed but the corresponding IEC958 status bits will - be updated. - -
- -
- Clock Adjustment - - In some chips, the clock of the codec isn't 48000 but using a - PCI clock (to save a quartz!). In this case, change the field - bus->clock to the corresponding - value. For example, intel8x0 - and es1968 drivers have their own function to read from the clock. - -
- -
- Proc Files - - The ALSA AC97 interface will create a proc file such as - /proc/asound/card0/codec97#0/ac97#0-0 and - ac97#0-0+regs. You can refer to these files to - see the current status and registers of the codec. - -
- -
- Multiple Codecs - - When there are several codecs on the same card, you need to - call snd_ac97_mixer() multiple times with - ac97.num=1 or greater. The num field - specifies the codec number. - - - - If you set up multiple codecs, you either need to write - different callbacks for each codec or check - ac97->num in the callback routines. - -
- - - - - - - - - MIDI (MPU401-UART) Interface - -
- General - - Many soundcards have built-in MIDI (MPU401-UART) - interfaces. When the soundcard supports the standard MPU401-UART - interface, most likely you can use the ALSA MPU401-UART API. The - MPU401-UART API is defined in - <sound/mpu401.h>. - - - - Some soundchips have a similar but slightly different - implementation of mpu401 stuff. For example, emu10k1 has its own - mpu401 routines. - -
- -
- Constructor - - To create a rawmidi object, call - snd_mpu401_uart_new(). - - - - - - - - - - The first argument is the card pointer, and the second is the - index of this component. You can create up to 8 rawmidi - devices. - - - - The third argument is the type of the hardware, - MPU401_HW_XXX. If it's not a special one, - you can use MPU401_HW_MPU401. - - - - The 4th argument is the I/O port address. Many - backward-compatible MPU401 have an I/O port such as 0x330. Or, it - might be a part of its own PCI I/O region. It depends on the - chip design. - - - - The 5th argument is a bitflag for additional information. - When the I/O port address above is part of the PCI I/O - region, the MPU401 I/O port might have been already allocated - (reserved) by the driver itself. In such a case, pass a bit flag - MPU401_INFO_INTEGRATED, - and the mpu401-uart layer will allocate the I/O ports by itself. - - - - When the controller supports only the input or output MIDI stream, - pass the MPU401_INFO_INPUT or - MPU401_INFO_OUTPUT bitflag, respectively. - Then the rawmidi instance is created as a single stream. - - - - MPU401_INFO_MMIO bitflag is used to change - the access method to MMIO (via readb and writeb) instead of - iob and outb. In this case, you have to pass the iomapped address - to snd_mpu401_uart_new(). - - - - When MPU401_INFO_TX_IRQ is set, the output - stream isn't checked in the default interrupt handler. The driver - needs to call snd_mpu401_uart_interrupt_tx() - by itself to start processing the output stream in the irq handler. - - - - Usually, the port address corresponds to the command port and - port + 1 corresponds to the data port. If not, you may change - the cport field of - struct snd_mpu401 manually - afterward. However, snd_mpu401 pointer is not - returned explicitly by - snd_mpu401_uart_new(). You need to cast - rmidi->private_data to - snd_mpu401 explicitly, - - - -private_data; -]]> - - - - and reset the cport as you like: - - - -cport = my_own_control_port; -]]> - - - - - - The 6th argument specifies the irq number for UART. If the irq - is already allocated, pass 0 to the 7th argument - (irq_flags). Otherwise, pass the flags - for irq allocation - (SA_XXX bits) to it, and the irq will be - reserved by the mpu401-uart layer. If the card doesn't generate - UART interrupts, pass -1 as the irq number. Then a timer - interrupt will be invoked for polling. - -
- -
- Interrupt Handler - - When the interrupt is allocated in - snd_mpu401_uart_new(), the private - interrupt handler is used, hence you don't have anything else to do - than creating the mpu401 stuff. Otherwise, you have to call - snd_mpu401_uart_interrupt() explicitly when - a UART interrupt is invoked and checked in your own interrupt - handler. - - - - In this case, you need to pass the private_data of the - returned rawmidi object from - snd_mpu401_uart_new() as the second - argument of snd_mpu401_uart_interrupt(). - - - -private_data, regs); -]]> - - - -
- - - - - - - - - RawMIDI Interface - -
- Overview - - - The raw MIDI interface is used for hardware MIDI ports that can - be accessed as a byte stream. It is not used for synthesizer - chips that do not directly understand MIDI. - - - - ALSA handles file and buffer management. All you have to do is - to write some code to move data between the buffer and the - hardware. - - - - The rawmidi API is defined in - <sound/rawmidi.h>. - -
- -
- Constructor - - - To create a rawmidi device, call the - snd_rawmidi_new function: - - -card, "MyMIDI", 0, outs, ins, &rmidi); - if (err < 0) - return err; - rmidi->private_data = chip; - strcpy(rmidi->name, "My MIDI"); - rmidi->info_flags = SNDRV_RAWMIDI_INFO_OUTPUT | - SNDRV_RAWMIDI_INFO_INPUT | - SNDRV_RAWMIDI_INFO_DUPLEX; -]]> - - - - - - The first argument is the card pointer, the second argument is - the ID string. - - - - The third argument is the index of this component. You can - create up to 8 rawmidi devices. - - - - The fourth and fifth arguments are the number of output and - input substreams, respectively, of this device (a substream is - the equivalent of a MIDI port). - - - - Set the info_flags field to specify - the capabilities of the device. - Set SNDRV_RAWMIDI_INFO_OUTPUT if there is - at least one output port, - SNDRV_RAWMIDI_INFO_INPUT if there is at - least one input port, - and SNDRV_RAWMIDI_INFO_DUPLEX if the device - can handle output and input at the same time. - - - - After the rawmidi device is created, you need to set the - operators (callbacks) for each substream. There are helper - functions to set the operators for all the substreams of a device: - - - - - - - - - The operators are usually defined like this: - - - - - - These callbacks are explained in the Callbacks - section. - - - - If there are more than one substream, you should give a - unique name to each of them: - - -streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substreams, - list { - sprintf(substream->name, "My MIDI Port %d", substream->number + 1); - } - /* same for SNDRV_RAWMIDI_STREAM_INPUT */ -]]> - - - -
- -
- Callbacks - - - In all the callbacks, the private data that you've set for the - rawmidi device can be accessed as - substream->rmidi->private_data. - - - - - If there is more than one port, your callbacks can determine the - port index from the struct snd_rawmidi_substream data passed to each - callback: - - -number; -]]> - - - - -
- <function>open</function> callback - - - - - - - - - This is called when a substream is opened. - You can initialize the hardware here, but you shouldn't - start transmitting/receiving data yet. - -
- -
- <function>close</function> callback - - - - - - - - - Guess what. - - - - The open and close - callbacks of a rawmidi device are serialized with a mutex, - and can sleep. - -
- -
- <function>trigger</function> callback for output - substreams - - - - - - - - - This is called with a nonzero up - parameter when there is some data in the substream buffer that - must be transmitted. - - - - To read data from the buffer, call - snd_rawmidi_transmit_peek. It will - return the number of bytes that have been read; this will be - less than the number of bytes requested when there are no more - data in the buffer. - After the data have been transmitted successfully, call - snd_rawmidi_transmit_ack to remove the - data from the substream buffer: - - - - - - - - - If you know beforehand that the hardware will accept data, you - can use the snd_rawmidi_transmit function - which reads some data and removes them from the buffer at once: - - - - - - - - - If you know beforehand how many bytes you can accept, you can - use a buffer size greater than one with the - snd_rawmidi_transmit* functions. - - - - The trigger callback must not sleep. If - the hardware FIFO is full before the substream buffer has been - emptied, you have to continue transmitting data later, either - in an interrupt handler, or with a timer if the hardware - doesn't have a MIDI transmit interrupt. - - - - The trigger callback is called with a - zero up parameter when the transmission - of data should be aborted. - -
- -
- <function>trigger</function> callback for input - substreams - - - - - - - - - This is called with a nonzero up - parameter to enable receiving data, or with a zero - up parameter do disable receiving data. - - - - The trigger callback must not sleep; the - actual reading of data from the device is usually done in an - interrupt handler. - - - - When data reception is enabled, your interrupt handler should - call snd_rawmidi_receive for all received - data: - - - - - - -
- -
- <function>drain</function> callback - - - - - - - - - This is only used with output substreams. This function should wait - until all data read from the substream buffer have been transmitted. - This ensures that the device can be closed and the driver unloaded - without losing data. - - - - This callback is optional. If you do not set - drain in the struct snd_rawmidi_ops - structure, ALSA will simply wait for 50 milliseconds - instead. - -
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- - - - - - - - - Miscellaneous Devices - -
- FM OPL3 - - The FM OPL3 is still used in many chips (mainly for backward - compatibility). ALSA has a nice OPL3 FM control layer, too. The - OPL3 API is defined in - <sound/opl3.h>. - - - - FM registers can be directly accessed through the direct-FM API, - defined in <sound/asound_fm.h>. In - ALSA native mode, FM registers are accessed through - the Hardware-Dependant Device direct-FM extension API, whereas in - OSS compatible mode, FM registers can be accessed with the OSS - direct-FM compatible API in /dev/dmfmX device. - - - - To create the OPL3 component, you have two functions to - call. The first one is a constructor for the opl3_t - instance. - - - - - - - - - - The first argument is the card pointer, the second one is the - left port address, and the third is the right port address. In - most cases, the right port is placed at the left port + 2. - - - - The fourth argument is the hardware type. - - - - When the left and right ports have been already allocated by - the card driver, pass non-zero to the fifth argument - (integrated). Otherwise, the opl3 module will - allocate the specified ports by itself. - - - - When the accessing the hardware requires special method - instead of the standard I/O access, you can create opl3 instance - separately with snd_opl3_new(). - - - - - - - - - - Then set command, - private_data and - private_free for the private - access function, the private data and the destructor. - The l_port and r_port are not necessarily set. Only the - command must be set properly. You can retrieve the data - from the opl3->private_data field. - - - - After creating the opl3 instance via snd_opl3_new(), - call snd_opl3_init() to initialize the chip to the - proper state. Note that snd_opl3_create() always - calls it internally. - - - - If the opl3 instance is created successfully, then create a - hwdep device for this opl3. - - - - - - - - - - The first argument is the opl3_t instance you - created, and the second is the index number, usually 0. - - - - The third argument is the index-offset for the sequencer - client assigned to the OPL3 port. When there is an MPU401-UART, - give 1 for here (UART always takes 0). - -
- -
- Hardware-Dependent Devices - - Some chips need user-space access for special - controls or for loading the micro code. In such a case, you can - create a hwdep (hardware-dependent) device. The hwdep API is - defined in <sound/hwdep.h>. You can - find examples in opl3 driver or - isa/sb/sb16_csp.c. - - - - The creation of the hwdep instance is done via - snd_hwdep_new(). - - - - - - - - where the third argument is the index number. - - - - You can then pass any pointer value to the - private_data. - If you assign a private data, you should define the - destructor, too. The destructor function is set in - the private_free field. - - - -private_data = p; - hw->private_free = mydata_free; -]]> - - - - and the implementation of the destructor would be: - - - -private_data; - kfree(p); - } -]]> - - - - - - The arbitrary file operations can be defined for this - instance. The file operators are defined in - the ops table. For example, assume that - this chip needs an ioctl. - - - -ops.open = mydata_open; - hw->ops.ioctl = mydata_ioctl; - hw->ops.release = mydata_release; -]]> - - - - And implement the callback functions as you like. - -
- -
- IEC958 (S/PDIF) - - Usually the controls for IEC958 devices are implemented via - the control interface. There is a macro to compose a name string for - IEC958 controls, SNDRV_CTL_NAME_IEC958() - defined in <include/asound.h>. - - - - There are some standard controls for IEC958 status bits. These - controls use the type SNDRV_CTL_ELEM_TYPE_IEC958, - and the size of element is fixed as 4 bytes array - (value.iec958.status[x]). For the info - callback, you don't specify - the value field for this type (the count field must be set, - though). - - - - IEC958 Playback Con Mask is used to return the - bit-mask for the IEC958 status bits of consumer mode. Similarly, - IEC958 Playback Pro Mask returns the bitmask for - professional mode. They are read-only controls, and are defined - as MIXER controls (iface = - SNDRV_CTL_ELEM_IFACE_MIXER). - - - - Meanwhile, IEC958 Playback Default control is - defined for getting and setting the current default IEC958 - bits. Note that this one is usually defined as a PCM control - (iface = SNDRV_CTL_ELEM_IFACE_PCM), - although in some places it's defined as a MIXER control. - - - - In addition, you can define the control switches to - enable/disable or to set the raw bit mode. The implementation - will depend on the chip, but the control should be named as - IEC958 xxx, preferably using - the SNDRV_CTL_NAME_IEC958() macro. - - - - You can find several cases, for example, - pci/emu10k1, - pci/ice1712, or - pci/cmipci.c. - -
- - - - - - - - - Buffer and Memory Management - -
- Buffer Types - - ALSA provides several different buffer allocation functions - depending on the bus and the architecture. All these have a - consistent API. The allocation of physically-contiguous pages is - done via - snd_malloc_xxx_pages() function, where xxx - is the bus type. - - - - The allocation of pages with fallback is - snd_malloc_xxx_pages_fallback(). This - function tries to allocate the specified pages but if the pages - are not available, it tries to reduce the page sizes until - enough space is found. - - - - The release the pages, call - snd_free_xxx_pages() function. - - - - Usually, ALSA drivers try to allocate and reserve - a large contiguous physical space - at the time the module is loaded for the later use. - This is called pre-allocation. - As already written, you can call the following function at - pcm instance construction time (in the case of PCI bus). - - - - - - - - where size is the byte size to be - pre-allocated and the max is the maximum - size to be changed via the prealloc proc file. - The allocator will try to get an area as large as possible - within the given size. - - - - The second argument (type) and the third argument (device pointer) - are dependent on the bus. - In the case of the ISA bus, pass snd_dma_isa_data() - as the third argument with SNDRV_DMA_TYPE_DEV type. - For the continuous buffer unrelated to the bus can be pre-allocated - with SNDRV_DMA_TYPE_CONTINUOUS type and the - snd_dma_continuous_data(GFP_KERNEL) device pointer, - where GFP_KERNEL is the kernel allocation flag to - use. - For the PCI scatter-gather buffers, use - SNDRV_DMA_TYPE_DEV_SG with - snd_dma_pci_data(pci) - (see the - Non-Contiguous Buffers - section). - - - - Once the buffer is pre-allocated, you can use the - allocator in the hw_params callback: - - - - - - - - Note that you have to pre-allocate to use this function. - -
- -
- External Hardware Buffers - - Some chips have their own hardware buffers and the DMA - transfer from the host memory is not available. In such a case, - you need to either 1) copy/set the audio data directly to the - external hardware buffer, or 2) make an intermediate buffer and - copy/set the data from it to the external hardware buffer in - interrupts (or in tasklets, preferably). - - - - The first case works fine if the external hardware buffer is large - enough. This method doesn't need any extra buffers and thus is - more effective. You need to define the - copy and - silence callbacks for - the data transfer. However, there is a drawback: it cannot - be mmapped. The examples are GUS's GF1 PCM or emu8000's - wavetable PCM. - - - - The second case allows for mmap on the buffer, although you have - to handle an interrupt or a tasklet to transfer the data - from the intermediate buffer to the hardware buffer. You can find an - example in the vxpocket driver. - - - - Another case is when the chip uses a PCI memory-map - region for the buffer instead of the host memory. In this case, - mmap is available only on certain architectures like the Intel one. - In non-mmap mode, the data cannot be transferred as in the normal - way. Thus you need to define the copy and - silence callbacks as well, - as in the cases above. The examples are found in - rme32.c and rme96.c. - - - - The implementation of the copy and - silence callbacks depends upon - whether the hardware supports interleaved or non-interleaved - samples. The copy callback is - defined like below, a bit - differently depending whether the direction is playback or - capture: - - - - - - - - - - In the case of interleaved samples, the second argument - (channel) is not used. The third argument - (pos) points the - current position offset in frames. - - - - The meaning of the fourth argument is different between - playback and capture. For playback, it holds the source data - pointer, and for capture, it's the destination data pointer. - - - - The last argument is the number of frames to be copied. - - - - What you have to do in this callback is again different - between playback and capture directions. In the - playback case, you copy the given amount of data - (count) at the specified pointer - (src) to the specified offset - (pos) on the hardware buffer. When - coded like memcpy-like way, the copy would be like: - - - - - - - - - - For the capture direction, you copy the given amount of - data (count) at the specified offset - (pos) on the hardware buffer to the - specified pointer (dst). - - - - - - - - Note that both the position and the amount of data are given - in frames. - - - - In the case of non-interleaved samples, the implementation - will be a bit more complicated. - - - - You need to check the channel argument, and if it's -1, copy - the whole channels. Otherwise, you have to copy only the - specified channel. Please check - isa/gus/gus_pcm.c as an example. - - - - The silence callback is also - implemented in a similar way. - - - - - - - - - - The meanings of arguments are the same as in the - copy - callback, although there is no src/dst - argument. In the case of interleaved samples, the channel - argument has no meaning, as well as on - copy callback. - - - - The role of silence callback is to - set the given amount - (count) of silence data at the - specified offset (pos) on the hardware - buffer. Suppose that the data format is signed (that is, the - silent-data is 0), and the implementation using a memset-like - function would be like: - - - - - - - - - - In the case of non-interleaved samples, again, the - implementation becomes a bit more complicated. See, for example, - isa/gus/gus_pcm.c. - -
- -
- Non-Contiguous Buffers - - If your hardware supports the page table as in emu10k1 or the - buffer descriptors as in via82xx, you can use the scatter-gather - (SG) DMA. ALSA provides an interface for handling SG-buffers. - The API is provided in <sound/pcm.h>. - - - - For creating the SG-buffer handler, call - snd_pcm_lib_preallocate_pages() or - snd_pcm_lib_preallocate_pages_for_all() - with SNDRV_DMA_TYPE_DEV_SG - in the PCM constructor like other PCI pre-allocator. - You need to pass snd_dma_pci_data(pci), - where pci is the struct pci_dev pointer - of the chip as well. - The struct snd_sg_buf instance is created as - substream->dma_private. You can cast - the pointer like: - - - -dma_private; -]]> - - - - - - Then call snd_pcm_lib_malloc_pages() - in the hw_params callback - as well as in the case of normal PCI buffer. - The SG-buffer handler will allocate the non-contiguous kernel - pages of the given size and map them onto the virtually contiguous - memory. The virtual pointer is addressed in runtime->dma_area. - The physical address (runtime->dma_addr) is set to zero, - because the buffer is physically non-contigous. - The physical address table is set up in sgbuf->table. - You can get the physical address at a certain offset via - snd_pcm_sgbuf_get_addr(). - - - - When a SG-handler is used, you need to set - snd_pcm_sgbuf_ops_page as - the page callback. - (See - page callback section.) - - - - To release the data, call - snd_pcm_lib_free_pages() in the - hw_free callback as usual. - -
- -
- Vmalloc'ed Buffers - - It's possible to use a buffer allocated via - vmalloc, for example, for an intermediate - buffer. Since the allocated pages are not contiguous, you need - to set the page callback to obtain - the physical address at every offset. - - - - The implementation of page callback - would be like this: - - - - - - /* get the physical page pointer on the given offset */ - static struct page *mychip_page(struct snd_pcm_substream *substream, - unsigned long offset) - { - void *pageptr = substream->runtime->dma_area + offset; - return vmalloc_to_page(pageptr); - } -]]> - - - -
- - - - - - - - - Proc Interface - - ALSA provides an easy interface for procfs. The proc files are - very useful for debugging. I recommend you set up proc files if - you write a driver and want to get a running status or register - dumps. The API is found in - <sound/info.h>. - - - - To create a proc file, call - snd_card_proc_new(). - - - - - - - - where the second argument specifies the name of the proc file to be - created. The above example will create a file - my-file under the card directory, - e.g. /proc/asound/card0/my-file. - - - - Like other components, the proc entry created via - snd_card_proc_new() will be registered and - released automatically in the card registration and release - functions. - - - - When the creation is successful, the function stores a new - instance in the pointer given in the third argument. - It is initialized as a text proc file for read only. To use - this proc file as a read-only text file as it is, set the read - callback with a private data via - snd_info_set_text_ops(). - - - - - - - - where the second argument (chip) is the - private data to be used in the callbacks. The third parameter - specifies the read buffer size and the fourth - (my_proc_read) is the callback function, which - is defined like - - - - - - - - - - - In the read callback, use snd_iprintf() for - output strings, which works just like normal - printf(). For example, - - - -private_data; - - snd_iprintf(buffer, "This is my chip!\n"); - snd_iprintf(buffer, "Port = %ld\n", chip->port); - } -]]> - - - - - - The file permissions can be changed afterwards. As default, it's - set as read only for all users. If you want to add write - permission for the user (root as default), do as follows: - - - -mode = S_IFREG | S_IRUGO | S_IWUSR; -]]> - - - - and set the write buffer size and the callback - - - -c.text.write = my_proc_write; -]]> - - - - - - For the write callback, you can use - snd_info_get_line() to get a text line, and - snd_info_get_str() to retrieve a string from - the line. Some examples are found in - core/oss/mixer_oss.c, core/oss/and - pcm_oss.c. - - - - For a raw-data proc-file, set the attributes as follows: - - - -content = SNDRV_INFO_CONTENT_DATA; - entry->private_data = chip; - entry->c.ops = &my_file_io_ops; - entry->size = 4096; - entry->mode = S_IFREG | S_IRUGO; -]]> - - - - - - The callback is much more complicated than the text-file - version. You need to use a low-level I/O functions such as - copy_from/to_user() to transfer the - data. - - - - local_max_size) - size = local_max_size - pos; - if (copy_to_user(buf, local_data + pos, size)) - return -EFAULT; - return size; - } -]]> - - - - - - - - - - - - Power Management - - If the chip is supposed to work with suspend/resume - functions, you need to add power-management code to the - driver. The additional code for power-management should be - ifdef'ed with - CONFIG_PM. - - - - If the driver fully supports suspend/resume - that is, the device can be - properly resumed to its state when suspend was called, - you can set the SNDRV_PCM_INFO_RESUME flag - in the pcm info field. Usually, this is possible when the - registers of the chip can be safely saved and restored to - RAM. If this is set, the trigger callback is called with - SNDRV_PCM_TRIGGER_RESUME after the resume - callback completes. - - - - Even if the driver doesn't support PM fully but - partial suspend/resume is still possible, it's still worthy to - implement suspend/resume callbacks. In such a case, applications - would reset the status by calling - snd_pcm_prepare() and restart the stream - appropriately. Hence, you can define suspend/resume callbacks - below but don't set SNDRV_PCM_INFO_RESUME - info flag to the PCM. - - - - Note that the trigger with SUSPEND can always be called when - snd_pcm_suspend_all is called, - regardless of the SNDRV_PCM_INFO_RESUME flag. - The RESUME flag affects only the behavior - of snd_pcm_resume(). - (Thus, in theory, - SNDRV_PCM_TRIGGER_RESUME isn't needed - to be handled in the trigger callback when no - SNDRV_PCM_INFO_RESUME flag is set. But, - it's better to keep it for compatibility reasons.) - - - In the earlier version of ALSA drivers, a common - power-management layer was provided, but it has been removed. - The driver needs to define the suspend/resume hooks according to - the bus the device is connected to. In the case of PCI drivers, the - callbacks look like below: - - - - - - - - - - The scheme of the real suspend job is as follows. - - - Retrieve the card and the chip data. - Call snd_power_change_state() with - SNDRV_CTL_POWER_D3hot to change the - power status. - Call snd_pcm_suspend_all() to suspend the running PCM streams. - If AC97 codecs are used, call - snd_ac97_suspend() for each codec. - Save the register values if necessary. - Stop the hardware if necessary. - Disable the PCI device by calling - pci_disable_device(). Then, call - pci_save_state() at last. - - - - - A typical code would be like: - - - -private_data; - /* (2) */ - snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); - /* (3) */ - snd_pcm_suspend_all(chip->pcm); - /* (4) */ - snd_ac97_suspend(chip->ac97); - /* (5) */ - snd_mychip_save_registers(chip); - /* (6) */ - snd_mychip_stop_hardware(chip); - /* (7) */ - pci_disable_device(pci); - pci_save_state(pci); - return 0; - } -]]> - - - - - - The scheme of the real resume job is as follows. - - - Retrieve the card and the chip data. - Set up PCI. First, call pci_restore_state(). - Then enable the pci device again by calling pci_enable_device(). - Call pci_set_master() if necessary, too. - Re-initialize the chip. - Restore the saved registers if necessary. - Resume the mixer, e.g. calling - snd_ac97_resume(). - Restart the hardware (if any). - Call snd_power_change_state() with - SNDRV_CTL_POWER_D0 to notify the processes. - - - - - A typical code would be like: - - - -private_data; - /* (2) */ - pci_restore_state(pci); - pci_enable_device(pci); - pci_set_master(pci); - /* (3) */ - snd_mychip_reinit_chip(chip); - /* (4) */ - snd_mychip_restore_registers(chip); - /* (5) */ - snd_ac97_resume(chip->ac97); - /* (6) */ - snd_mychip_restart_chip(chip); - /* (7) */ - snd_power_change_state(card, SNDRV_CTL_POWER_D0); - return 0; - } -]]> - - - - - - As shown in the above, it's better to save registers after - suspending the PCM operations via - snd_pcm_suspend_all() or - snd_pcm_suspend(). It means that the PCM - streams are already stoppped when the register snapshot is - taken. But, remember that you don't have to restart the PCM - stream in the resume callback. It'll be restarted via - trigger call with SNDRV_PCM_TRIGGER_RESUME - when necessary. - - - - OK, we have all callbacks now. Let's set them up. In the - initialization of the card, make sure that you can get the chip - data from the card instance, typically via - private_data field, in case you - created the chip data individually. - - - -private_data = chip; - .... - } -]]> - - - - When you created the chip data with - snd_card_create(), it's anyway accessible - via private_data field. - - - -private_data; - .... - } -]]> - - - - - - - If you need a space to save the registers, allocate the - buffer for it here, too, since it would be fatal - if you cannot allocate a memory in the suspend phase. - The allocated buffer should be released in the corresponding - destructor. - - - - And next, set suspend/resume callbacks to the pci_driver. - - - - - - - - - - - - - - - - Module Parameters - - There are standard module options for ALSA. At least, each - module should have the index, - id and enable - options. - - - - If the module supports multiple cards (usually up to - 8 = SNDRV_CARDS cards), they should be - arrays. The default initial values are defined already as - constants for easier programming: - - - - - - - - - - If the module supports only a single card, they could be single - variables, instead. enable option is not - always necessary in this case, but it would be better to have a - dummy option for compatibility. - - - - The module parameters must be declared with the standard - module_param()(), - module_param_array()() and - MODULE_PARM_DESC() macros. - - - - The typical coding would be like below: - - - - - - - - - - Also, don't forget to define the module description, classes, - license and devices. Especially, the recent modprobe requires to - define the module license as GPL, etc., otherwise the system is - shown as tainted. - - - - - - - - - - - - - - - - How To Put Your Driver Into ALSA Tree -
- General - - So far, you've learned how to write the driver codes. - And you might have a question now: how to put my own - driver into the ALSA driver tree? - Here (finally :) the standard procedure is described briefly. - - - - Suppose that you create a new PCI driver for the card - xyz. The card module name would be - snd-xyz. The new driver is usually put into the alsa-driver - tree, alsa-driver/pci directory in - the case of PCI cards. - Then the driver is evaluated, audited and tested - by developers and users. After a certain time, the driver - will go to the alsa-kernel tree (to the corresponding directory, - such as alsa-kernel/pci) and eventually - will be integrated into the Linux 2.6 tree (the directory would be - linux/sound/pci). - - - - In the following sections, the driver code is supposed - to be put into alsa-driver tree. The two cases are covered: - a driver consisting of a single source file and one consisting - of several source files. - -
- -
- Driver with A Single Source File - - - - - Modify alsa-driver/pci/Makefile - - - - Suppose you have a file xyz.c. Add the following - two lines - - - - - - - - - - - Create the Kconfig entry - - - - Add the new entry of Kconfig for your xyz driver. - - - - - - - the line, select SND_PCM, specifies that the driver xyz supports - PCM. In addition to SND_PCM, the following components are - supported for select command: - SND_RAWMIDI, SND_TIMER, SND_HWDEP, SND_MPU401_UART, - SND_OPL3_LIB, SND_OPL4_LIB, SND_VX_LIB, SND_AC97_CODEC. - Add the select command for each supported component. - - - - Note that some selections imply the lowlevel selections. - For example, PCM includes TIMER, MPU401_UART includes RAWMIDI, - AC97_CODEC includes PCM, and OPL3_LIB includes HWDEP. - You don't need to give the lowlevel selections again. - - - - For the details of Kconfig script, refer to the kbuild - documentation. - - - - - - - Run cvscompile script to re-generate the configure script and - build the whole stuff again. - - - - -
- -
- Drivers with Several Source Files - - Suppose that the driver snd-xyz have several source files. - They are located in the new subdirectory, - pci/xyz. - - - - - Add a new directory (xyz) in - alsa-driver/pci/Makefile as below - - - - - - - - - - - - Under the directory xyz, create a Makefile - - - Sample Makefile for a driver xyz - - - - - - - - - - Create the Kconfig entry - - - - This procedure is as same as in the last section. - - - - - - Run cvscompile script to re-generate the configure script and - build the whole stuff again. - - - - -
- - - - - - - - Useful Functions - -
- <function>snd_printk()</function> and friends - - ALSA provides a verbose version of the - printk() function. If a kernel config - CONFIG_SND_VERBOSE_PRINTK is set, this - function prints the given message together with the file name - and the line of the caller. The KERN_XXX - prefix is processed as - well as the original printk() does, so it's - recommended to add this prefix, e.g. - - - - - - - - - - There are also printk()'s for - debugging. snd_printd() can be used for - general debugging purposes. If - CONFIG_SND_DEBUG is set, this function is - compiled, and works just like - snd_printk(). If the ALSA is compiled - without the debugging flag, it's ignored. - - - - snd_printdd() is compiled in only when - CONFIG_SND_DEBUG_VERBOSE is set. Please note - that CONFIG_SND_DEBUG_VERBOSE is not set as default - even if you configure the alsa-driver with - option. You need to give - explicitly option instead. - -
- -
- <function>snd_BUG()</function> - - It shows the BUG? message and - stack trace as well as snd_BUG_ON at the point. - It's useful to show that a fatal error happens there. - - - When no debug flag is set, this macro is ignored. - -
- -
- <function>snd_BUG_ON()</function> - - snd_BUG_ON() macro is similar with - WARN_ON() macro. For example, - - - - - - - - or it can be used as the condition, - - - - - - - - - - The macro takes an conditional expression to evaluate. - When CONFIG_SND_DEBUG, is set, the - expression is actually evaluated. If it's non-zero, it shows - the warning message such as - BUG? (xxx) - normally followed by stack trace. It returns the evaluated - value. - When no CONFIG_SND_DEBUG is set, this - macro always returns zero. - - -
- - - - - - - - - Acknowledgments - - I would like to thank Phil Kerr for his help for improvement and - corrections of this document. - - - Kevin Conder reformatted the original plain-text to the - DocBook format. - - - Giuliano Pochini corrected typos and contributed the example codes - in the hardware constraints section. - - - -- cgit v1.2.3-70-g09d2 From 6b849bcff0004aa5dd216b4d3eb56f51c9df8a72 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 9 Mar 2009 18:18:33 +0000 Subject: ASoC: Convert PXA AC97 driver to probe with the platform device This will break any boards that don't register the AC97 controller device due to using ASoC. Signed-off-by: Mark Brown --- sound/soc/pxa/pxa2xx-ac97.c | 38 ++++++++++++++++++++++++++++++++++---- 1 file changed, 34 insertions(+), 4 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 812c2b4d3e0..49a2810ca58 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -106,13 +106,13 @@ static int pxa2xx_ac97_resume(struct snd_soc_dai *dai) static int pxa2xx_ac97_probe(struct platform_device *pdev, struct snd_soc_dai *dai) { - return pxa2xx_ac97_hw_probe(pdev); + return pxa2xx_ac97_hw_probe(to_platform_device(dai->dev)); } static void pxa2xx_ac97_remove(struct platform_device *pdev, struct snd_soc_dai *dai) { - pxa2xx_ac97_hw_remove(pdev); + pxa2xx_ac97_hw_remove(to_platform_device(dai->dev)); } static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream, @@ -229,15 +229,45 @@ struct snd_soc_dai pxa_ac97_dai[] = { EXPORT_SYMBOL_GPL(pxa_ac97_dai); EXPORT_SYMBOL_GPL(soc_ac97_ops); -static int __init pxa_ac97_init(void) +static int __devinit pxa2xx_ac97_dev_probe(struct platform_device *pdev) { + int i; + + for (i = 0; i < ARRAY_SIZE(pxa_ac97_dai); i++) + pxa_ac97_dai[i].dev = &pdev->dev; + + /* Punt most of the init to the SoC probe; we may need the machine + * driver to do interesting things with the clocking to get us up + * and running. + */ return snd_soc_register_dais(pxa_ac97_dai, ARRAY_SIZE(pxa_ac97_dai)); } + +static int __devexit pxa2xx_ac97_dev_remove(struct platform_device *pdev) +{ + snd_soc_unregister_dais(pxa_ac97_dai, ARRAY_SIZE(pxa_ac97_dai)); + + return 0; +} + +static struct platform_driver pxa2xx_ac97_driver = { + .probe = pxa2xx_ac97_dev_probe, + .remove = __devexit_p(pxa2xx_ac97_dev_remove), + .driver = { + .name = "pxa2xx-ac97", + .owner = THIS_MODULE, + }, +}; + +static int __init pxa_ac97_init(void) +{ + return platform_driver_register(&pxa2xx_ac97_driver); +} module_init(pxa_ac97_init); static void __exit pxa_ac97_exit(void) { - snd_soc_unregister_dais(pxa_ac97_dai, ARRAY_SIZE(pxa_ac97_dai)); + platform_driver_unregister(&pxa2xx_ac97_driver); } module_exit(pxa_ac97_exit); -- cgit v1.2.3-70-g09d2 From eac84739721857f4d5be3d9127f4644f16a9bea4 Mon Sep 17 00:00:00 2001 From: Ben Dooks Date: Mon, 9 Mar 2009 17:47:13 +0000 Subject: ASoC: Fix Samsung S3C2412_IISMOD_SDF_{MSB,LSB} definitions The definitions of S3C2412_IISMOD_SDF_MSB and S3C2412_IISMOD_SDF_LSB are incorrect, being the same S3C2412_IISMOD_SDF_IIS which is the only correct one in this series. Signed-off-by: Ben Dooks Signed-off-by: Mark Brown --- arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h b/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h index a5600b381d4..0fad7571030 100644 --- a/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h +++ b/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h @@ -47,8 +47,8 @@ #define S3C2412_IISMOD_LR_LLOW (0 << 7) #define S3C2412_IISMOD_LR_RLOW (1 << 7) #define S3C2412_IISMOD_SDF_IIS (0 << 5) -#define S3C2412_IISMOD_SDF_MSB (0 << 5) -#define S3C2412_IISMOD_SDF_LSB (0 << 5) +#define S3C2412_IISMOD_SDF_MSB (1 << 5) +#define S3C2412_IISMOD_SDF_LSB (2 << 5) #define S3C2412_IISMOD_SDF_MASK (3 << 5) #define S3C2412_IISMOD_RCLK_256FS (0 << 3) #define S3C2412_IISMOD_RCLK_512FS (1 << 3) -- cgit v1.2.3-70-g09d2 From ae6241fbf5c8863631532e8069037bae460607be Mon Sep 17 00:00:00 2001 From: Christoph Plattner Date: Sun, 8 Mar 2009 23:19:05 +0100 Subject: ALSA: hda - Added HP HDX16/HDX18 notebook support for HDA codecs (82HD71) Added codec recognition of HP HDX platforms and added support of the MUTE LED (orange/white). For this feature the CONFIG_SND_HDA_POWER_SAVE is needed to use event handling for mute control. Signed-off-by: Christoph Plattner Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 57 ++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 57 insertions(+) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 123bcf7c3b2..fb9f4ccba88 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -99,6 +99,7 @@ enum { STAC_DELL_M4_3, STAC_HP_M4, STAC_HP_DV5, + STAC_HP_HDX, STAC_92HD71BXX_MODELS }; @@ -1828,6 +1829,7 @@ static unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = { [STAC_DELL_M4_3] = dell_m4_3_pin_configs, [STAC_HP_M4] = NULL, [STAC_HP_DV5] = NULL, + [STAC_HP_HDX] = NULL, }; static const char *stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = { @@ -1838,6 +1840,7 @@ static const char *stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = { [STAC_DELL_M4_3] = "dell-m4-3", [STAC_HP_M4] = "hp-m4", [STAC_HP_DV5] = "hp-dv5", + [STAC_HP_HDX] = "hp-hdx", }; static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { @@ -1852,6 +1855,10 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { "HP dv4-7", STAC_HP_DV5), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x361a, "HP mini 1000", STAC_HP_M4), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x361b, + "HP HDX", STAC_HP_HDX), /* HDX16 */ + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3610, + "HP HDX", STAC_HP_HDX), /* HDX18 */ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0233, "unknown Dell", STAC_DELL_M4_1), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0234, @@ -4472,6 +4479,41 @@ static int stac92xx_resume(struct hda_codec *codec) return 0; } + +/* + * using power check for controlling mute led of HP HDX notebooks + * check for mute state only on Speakers (nid = 0x10) + * + * For this feature CONFIG_SND_HDA_POWER_SAVE is needed, otherwise + * the LED is NOT working properly ! + */ + +#ifdef CONFIG_SND_HDA_POWER_SAVE +static int stac92xx_check_power_status (struct hda_codec * codec, hda_nid_t nid) +{ + struct sigmatel_spec *spec = codec->spec; + + /* only handle on HP HDX */ + if (spec->board_config != STAC_HP_HDX) + return 0; + + if (nid == 0x10) + { + if (snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0) & + HDA_AMP_MUTE) + spec->gpio_data &= ~0x08; /* orange */ + else + spec->gpio_data |= 0x08; /* white */ + + stac_gpio_set(codec, spec->gpio_mask, + spec->gpio_dir, + spec->gpio_data); + } + + return 0; +} +#endif + static int stac92xx_suspend(struct hda_codec *codec, pm_message_t state) { struct sigmatel_spec *spec = codec->spec; @@ -4493,6 +4535,9 @@ static struct hda_codec_ops stac92xx_patch_ops = { .suspend = stac92xx_suspend, .resume = stac92xx_resume, #endif +#ifdef CONFIG_SND_HDA_POWER_SAVE + .check_power_status = stac92xx_check_power_status, +#endif }; static int patch_stac9200(struct hda_codec *codec) @@ -5089,6 +5134,13 @@ again: /* no output amps */ spec->num_pwrs = 0; /* fallthru */ + case 0x111d76b2: /* Codec of HP HDX16/HDX18 */ + + /* orange/white mute led on GPIO3, orange=0, white=1 */ + spec->gpio_mask |= 0x08; + spec->gpio_dir |= 0x08; + spec->gpio_data |= 0x08; /* set to white */ + /* fallthru */ default: memcpy(&spec->private_dimux, &stac92hd71bxx_dmux_amixer, sizeof(stac92hd71bxx_dmux_amixer)); @@ -5143,6 +5195,11 @@ again: snd_hda_codec_set_pincfg(codec, 0x0d, 0x90170010); stac92xx_auto_set_pinctl(codec, 0x0d, AC_PINCTL_OUT_EN); break; + case STAC_HP_HDX: + spec->num_dmics = 1; + spec->num_dmuxes = 1; + spec->num_smuxes = 1; + break; }; spec->multiout.dac_nids = spec->dac_nids; -- cgit v1.2.3-70-g09d2 From 443e26d014c242623dd70cda054cc6e5ebf7993d Mon Sep 17 00:00:00 2001 From: Christoph Plattner Date: Tue, 10 Mar 2009 00:05:56 +0100 Subject: ALSA: hda - Rework on patch_sigmatel.c for HP HDX16/HDX18 Code rework, comments of mail tiwai@suse.de (2009-03-09) incorporated. Code tested on HP HDX16 (not tested on HDX18 yet). Signed-off-by: Christoph Plattner Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 30 +++++++++++++++--------------- 1 file changed, 15 insertions(+), 15 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index fb9f4ccba88..d119feed42c 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4489,14 +4489,10 @@ static int stac92xx_resume(struct hda_codec *codec) */ #ifdef CONFIG_SND_HDA_POWER_SAVE -static int stac92xx_check_power_status (struct hda_codec * codec, hda_nid_t nid) +static int stac92xx_hp_hdx_check_power_status (struct hda_codec * codec, hda_nid_t nid) { struct sigmatel_spec *spec = codec->spec; - /* only handle on HP HDX */ - if (spec->board_config != STAC_HP_HDX) - return 0; - if (nid == 0x10) { if (snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0) & @@ -4535,9 +4531,6 @@ static struct hda_codec_ops stac92xx_patch_ops = { .suspend = stac92xx_suspend, .resume = stac92xx_resume, #endif -#ifdef CONFIG_SND_HDA_POWER_SAVE - .check_power_status = stac92xx_check_power_status, -#endif }; static int patch_stac9200(struct hda_codec *codec) @@ -5134,13 +5127,6 @@ again: /* no output amps */ spec->num_pwrs = 0; /* fallthru */ - case 0x111d76b2: /* Codec of HP HDX16/HDX18 */ - - /* orange/white mute led on GPIO3, orange=0, white=1 */ - spec->gpio_mask |= 0x08; - spec->gpio_dir |= 0x08; - spec->gpio_data |= 0x08; /* set to white */ - /* fallthru */ default: memcpy(&spec->private_dimux, &stac92hd71bxx_dmux_amixer, sizeof(stac92hd71bxx_dmux_amixer)); @@ -5199,6 +5185,20 @@ again: spec->num_dmics = 1; spec->num_dmuxes = 1; spec->num_smuxes = 1; + /* + * For controlling MUTE LED on HP HDX16/HDX18 notebooks, + * the CONFIG_SND_HDA_POWER_SAVE is needed to be set. + */ +#ifdef CONFIG_SND_HDA_POWER_SAVE + /* orange/white mute led on GPIO3, orange=0, white=1 */ + spec->gpio_mask |= 0x08; + spec->gpio_dir |= 0x08; + spec->gpio_data |= 0x08; /* set to white */ + + /* register check_power_status callback. */ + codec->patch_ops.check_power_status = + stac92xx_hp_hdx_check_power_status; +#endif break; }; -- cgit v1.2.3-70-g09d2 From 6fce61aeaf0dc1dfa306092539397ab903a9afc4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Mar 2009 07:48:57 +0100 Subject: ALSA: hda - Fix coding style issues in last two patches Also re-ordered the quirk entries per SSID. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 26 +++++++++++++------------- 1 file changed, 13 insertions(+), 13 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index d119feed42c..72c87aa20bd 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1853,12 +1853,12 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { "HP dv4-7", STAC_HP_DV5), SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x3600, "HP dv4-7", STAC_HP_DV5), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3610, + "HP HDX", STAC_HP_HDX), /* HDX18 */ SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x361a, "HP mini 1000", STAC_HP_M4), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x361b, - "HP HDX", STAC_HP_HDX), /* HDX16 */ - SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3610, - "HP HDX", STAC_HP_HDX), /* HDX18 */ + "HP HDX", STAC_HP_HDX), /* HDX16 */ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0233, "unknown Dell", STAC_DELL_M4_1), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0234, @@ -4489,20 +4489,20 @@ static int stac92xx_resume(struct hda_codec *codec) */ #ifdef CONFIG_SND_HDA_POWER_SAVE -static int stac92xx_hp_hdx_check_power_status (struct hda_codec * codec, hda_nid_t nid) +static int stac92xx_hp_hdx_check_power_status(struct hda_codec *codec, + hda_nid_t nid) { struct sigmatel_spec *spec = codec->spec; - - if (nid == 0x10) - { - if (snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0) & + + if (nid == 0x10) { + if (snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0) & HDA_AMP_MUTE) spec->gpio_data &= ~0x08; /* orange */ else spec->gpio_data |= 0x08; /* white */ - - stac_gpio_set(codec, spec->gpio_mask, - spec->gpio_dir, + + stac_gpio_set(codec, spec->gpio_mask, + spec->gpio_dir, spec->gpio_data); } @@ -5185,7 +5185,7 @@ again: spec->num_dmics = 1; spec->num_dmuxes = 1; spec->num_smuxes = 1; - /* + /* * For controlling MUTE LED on HP HDX16/HDX18 notebooks, * the CONFIG_SND_HDA_POWER_SAVE is needed to be set. */ @@ -5196,7 +5196,7 @@ again: spec->gpio_data |= 0x08; /* set to white */ /* register check_power_status callback. */ - codec->patch_ops.check_power_status = + codec->patch_ops.check_power_status = stac92xx_hp_hdx_check_power_status; #endif break; -- cgit v1.2.3-70-g09d2 From 6dfc0d2c4b9a5455c60e0b9ee95bbf22fc516cef Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Mar 2009 07:54:20 +0100 Subject: ALSA: hda - Add missing models to documentation Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 80b796e4a80..f9253ea3c19 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -344,7 +344,9 @@ STAC92HD71B* dell-m4-1 Dell desktops dell-m4-2 Dell desktops dell-m4-3 Dell desktops - hp-m4 HP dv laptops + hp-m4 HP mini 1000 + hp-dv5 HP dv series + hp-hdx HP HDX series auto BIOS setup (default) STAC92HD73* @@ -361,6 +363,7 @@ STAC92HD83* =========== ref Reference board mic-ref Reference board with power managment for ports + dell-s14 Dell laptop auto BIOS setup (default) STAC9872 -- cgit v1.2.3-70-g09d2 From dd5746a85cb21ea5b3afca0b569586a05aa56846 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Mar 2009 14:30:40 +0100 Subject: ALSA: hda - Create vmaster for conexant codecs Instead of binding volumes, create a virtual master volume for Conexant codecs. This allows separate HP and speaker volume controls. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 47 ++++++++++++++++++++++++++++++++---------- 1 file changed, 36 insertions(+), 11 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 1938e92e1f0..e1476d6d8b3 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -58,6 +58,7 @@ struct conexant_spec { struct snd_kcontrol_new *mixers[5]; int num_mixers; + hda_nid_t vmaster_nid; const struct hda_verb *init_verbs[5]; /* initialization verbs * don't forget NULL @@ -462,6 +463,18 @@ static void conexant_free(struct hda_codec *codec) kfree(codec->spec); } +static const char *slave_vols[] = { + "Headphone Playback Volume", + "Speaker Playback Volume", + NULL +}; + +static const char *slave_sws[] = { + "Headphone Playback Switch", + "Speaker Playback Switch", + NULL +}; + static int conexant_build_controls(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; @@ -489,6 +502,26 @@ static int conexant_build_controls(struct hda_codec *codec) if (err < 0) return err; } + + /* if we have no master control, let's create it */ + if (spec->vmaster_nid && + !snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) { + unsigned int vmaster_tlv[4]; + snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid, + HDA_OUTPUT, vmaster_tlv); + err = snd_hda_add_vmaster(codec, "Master Playback Volume", + vmaster_tlv, slave_vols); + if (err < 0) + return err; + } + if (spec->vmaster_nid && + !snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) { + err = snd_hda_add_vmaster(codec, "Master Playback Switch", + NULL, slave_sws); + if (err < 0) + return err; + } + return 0; } @@ -1182,16 +1215,6 @@ static int cxt5047_hp_master_sw_put(struct snd_kcontrol *kcontrol, return 1; } -/* bind volumes of both NID 0x13 (Headphones) and 0x1d (Speakers) */ -static struct hda_bind_ctls cxt5047_bind_master_vol = { - .ops = &snd_hda_bind_vol, - .values = { - HDA_COMPOSE_AMP_VAL(0x13, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(0x1d, 3, 0, HDA_OUTPUT), - 0 - }, -}; - /* mute internal speaker if HP is plugged */ static void cxt5047_hp_automute(struct hda_codec *codec) { @@ -1311,7 +1334,8 @@ static struct snd_kcontrol_new cxt5047_toshiba_mixers[] = { HDA_CODEC_MUTE("Capture Switch", 0x12, 0x03, HDA_INPUT), HDA_CODEC_VOLUME("PCM Volume", 0x10, 0x00, HDA_OUTPUT), HDA_CODEC_MUTE("PCM Switch", 0x10, 0x00, HDA_OUTPUT), - HDA_BIND_VOL("Master Playback Volume", &cxt5047_bind_master_vol), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x13, 0x00, HDA_OUTPUT), + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x1d, 0x00, HDA_OUTPUT), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", @@ -1631,6 +1655,7 @@ static int patch_cxt5047(struct hda_codec *codec) codec->patch_ops.unsol_event = cxt5047_hp_unsol_event; #endif } + spec->vmaster_nid = 0x13; return 0; } -- cgit v1.2.3-70-g09d2 From b880c74adf7e79b97de710a152ea82f292f9abc7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Mar 2009 14:41:05 +0100 Subject: ALSA: hda - Create "Capture Source" control dynamically in patch_conexant.c Create "Capture Source" control dynamically for Conexant codecs. If only one capture item is available, don't create such a control since it's just useless. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 61 ++++++++++++------------------------------ 1 file changed, 17 insertions(+), 44 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index e1476d6d8b3..d5d736ff7c6 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -463,6 +463,17 @@ static void conexant_free(struct hda_codec *codec) kfree(codec->spec); } +static struct snd_kcontrol_new cxt_capture_mixers[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .info = conexant_mux_enum_info, + .get = conexant_mux_enum_get, + .put = conexant_mux_enum_put + }, + {} +}; + static const char *slave_vols[] = { "Headphone Playback Volume", "Speaker Playback Volume", @@ -522,6 +533,12 @@ static int conexant_build_controls(struct hda_codec *codec) return err; } + if (spec->input_mux) { + err = snd_hda_add_new_ctls(codec, cxt_capture_mixers); + if (err < 0) + return err; + } + return 0; } @@ -753,13 +770,6 @@ static void cxt5045_hp_unsol_event(struct hda_codec *codec, } static struct snd_kcontrol_new cxt5045_mixers[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = conexant_mux_enum_info, - .get = conexant_mux_enum_get, - .put = conexant_mux_enum_put - }, HDA_CODEC_VOLUME("Int Mic Capture Volume", 0x1a, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Int Mic Capture Switch", 0x1a, 0x01, HDA_INPUT), HDA_CODEC_VOLUME("Ext Mic Capture Volume", 0x1a, 0x02, HDA_INPUT), @@ -793,13 +803,6 @@ static struct snd_kcontrol_new cxt5045_benq_mixers[] = { }; static struct snd_kcontrol_new cxt5045_mixers_hp530[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = conexant_mux_enum_info, - .get = conexant_mux_enum_get, - .put = conexant_mux_enum_put - }, HDA_CODEC_VOLUME("Int Mic Capture Volume", 0x1a, 0x02, HDA_INPUT), HDA_CODEC_MUTE("Int Mic Capture Switch", 0x1a, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Ext Mic Capture Volume", 0x1a, 0x01, HDA_INPUT), @@ -1170,20 +1173,6 @@ static struct hda_channel_mode cxt5047_modes[1] = { { 2, NULL }, }; -static struct hda_input_mux cxt5047_capture_source = { - .num_items = 1, - .items = { - { "Mic", 0x2 }, - } -}; - -static struct hda_input_mux cxt5047_hp_capture_source = { - .num_items = 1, - .items = { - { "ExtMic", 0x2 }, - } -}; - static struct hda_input_mux cxt5047_toshiba_capture_source = { .num_items = 2, .items = { @@ -1321,13 +1310,6 @@ static struct snd_kcontrol_new cxt5047_mixers[] = { }; static struct snd_kcontrol_new cxt5047_toshiba_mixers[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = conexant_mux_enum_info, - .get = conexant_mux_enum_get, - .put = conexant_mux_enum_put - }, HDA_CODEC_VOLUME("Mic Bypass Capture Volume", 0x19, 0x02, HDA_INPUT), HDA_CODEC_MUTE("Mic Bypass Capture Switch", 0x19, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x03, HDA_INPUT), @@ -1349,13 +1331,6 @@ static struct snd_kcontrol_new cxt5047_toshiba_mixers[] = { }; static struct snd_kcontrol_new cxt5047_hp_mixers[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = conexant_mux_enum_info, - .get = conexant_mux_enum_get, - .put = conexant_mux_enum_put - }, HDA_CODEC_VOLUME("Mic Bypass Capture Volume", 0x19, 0x02, HDA_INPUT), HDA_CODEC_MUTE("Mic Bypass Capture Switch", 0x19,0x02,HDA_INPUT), HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x03, HDA_INPUT), @@ -1614,7 +1589,6 @@ static int patch_cxt5047(struct hda_codec *codec) spec->num_adc_nids = 1; spec->adc_nids = cxt5047_adc_nids; spec->capsrc_nids = cxt5047_capsrc_nids; - spec->input_mux = &cxt5047_capture_source; spec->num_mixers = 1; spec->mixers[0] = cxt5047_mixers; spec->num_init_verbs = 1; @@ -1633,7 +1607,6 @@ static int patch_cxt5047(struct hda_codec *codec) codec->patch_ops.unsol_event = cxt5047_hp2_unsol_event; break; case CXT5047_LAPTOP_HP: - spec->input_mux = &cxt5047_hp_capture_source; spec->num_init_verbs = 2; spec->init_verbs[1] = cxt5047_hp_init_verbs; spec->mixers[0] = cxt5047_hp_mixers; -- cgit v1.2.3-70-g09d2 From 3b628867f328cfe1ad4811d63961579874f87041 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Mar 2009 14:53:54 +0100 Subject: ALSA: hda - Remove superfluous verbs for Cxt5047 laptop-eapd model Remove superfluous verbs from cxt5047_toshiba_init_verbs[]. Also fix comments and minor coding style issues. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index d5d736ff7c6..e9e47574c61 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1377,12 +1377,9 @@ static struct hda_verb cxt5047_init_verbs[] = { /* configuration for Toshiba Laptops */ static struct hda_verb cxt5047_toshiba_init_verbs[] = { - {0x13, AC_VERB_SET_EAPD_BTLENABLE, 0x0 }, /* default on */ - /* pin sensing on HP and Mic jacks */ - {0x13, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, - {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT}, + {0x13, AC_VERB_SET_EAPD_BTLENABLE, 0x0}, /* default off */ /* Speaker routing */ - {0x1d, AC_VERB_SET_CONNECT_SEL,0x1}, + {0x1d, AC_VERB_SET_CONNECT_SEL, 0x1}, {} }; -- cgit v1.2.3-70-g09d2 From 5b3a7440cbabdda07cfb3dcf4a07e0115a3dff9a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Mar 2009 15:10:55 +0100 Subject: ALSA: hda - Fix / clean up init verbs for Cxt5047 codec Fix the initial connections of output pins 0x13 and 0x1d for Conexant 5047 codec to point to the mixer amp properly. Removed unneeded (doubly) verbs from arrays, also removed the unneeded changing of widget 0x1c, which is now completely unused. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 36 +++--------------------------------- 1 file changed, 3 insertions(+), 33 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index e9e47574c61..71822140294 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1165,7 +1165,7 @@ static int patch_cxt5045(struct hda_codec *codec) /* Conexant 5047 specific */ #define CXT5047_SPDIF_OUT 0x11 -static hda_nid_t cxt5047_dac_nids[2] = { 0x10, 0x1c }; +static hda_nid_t cxt5047_dac_nids[1] = { 0x10 }; /* 0x1c */ static hda_nid_t cxt5047_adc_nids[1] = { 0x12 }; static hda_nid_t cxt5047_capsrc_nids[1] = { 0x1a }; @@ -1216,9 +1216,6 @@ static void cxt5047_hp_automute(struct hda_codec *codec) bits = (spec->hp_present || !spec->cur_eapd) ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0, HDA_AMP_MUTE, bits); - /* Mute/Unmute PCM 2 for good measure - some systems need this */ - snd_hda_codec_amp_stereo(codec, 0x1c, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); } /* mute internal speaker if HP is plugged */ @@ -1233,9 +1230,6 @@ static void cxt5047_hp2_automute(struct hda_codec *codec) bits = spec->hp_present ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0, HDA_AMP_MUTE, bits); - /* Mute/Unmute PCM 2 for good measure - some systems need this */ - snd_hda_codec_amp_stereo(codec, 0x1c, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); } /* toggle input of built-in and mic jack appropriately */ @@ -1299,8 +1293,6 @@ static struct snd_kcontrol_new cxt5047_mixers[] = { HDA_CODEC_MUTE("Capture Switch", 0x12, 0x03, HDA_INPUT), HDA_CODEC_VOLUME("PCM Volume", 0x10, 0x00, HDA_OUTPUT), HDA_CODEC_MUTE("PCM Switch", 0x10, 0x00, HDA_OUTPUT), - HDA_CODEC_VOLUME("PCM-2 Volume", 0x1c, 0x00, HDA_OUTPUT), - HDA_CODEC_MUTE("PCM-2 Switch", 0x1c, 0x00, HDA_OUTPUT), HDA_CODEC_VOLUME("Speaker Playback Volume", 0x1d, 0x00, HDA_OUTPUT), HDA_CODEC_MUTE("Speaker Playback Switch", 0x1d, 0x00, HDA_OUTPUT), HDA_CODEC_VOLUME("Headphone Playback Volume", 0x13, 0x00, HDA_OUTPUT), @@ -1356,8 +1348,8 @@ static struct hda_verb cxt5047_init_verbs[] = { {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_50 }, /* HP, Speaker */ {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, - {0x13, AC_VERB_SET_CONNECT_SEL,0x1}, - {0x1d, AC_VERB_SET_CONNECT_SEL,0x0}, + {0x13, AC_VERB_SET_CONNECT_SEL, 0x0}, /* mixer(0x19) */ + {0x1d, AC_VERB_SET_CONNECT_SEL, 0x1}, /* mixer(0x19) */ /* Record selector: Mic */ {0x12, AC_VERB_SET_CONNECT_SEL,0x03}, {0x19, AC_VERB_SET_AMP_GAIN_MUTE, @@ -1378,26 +1370,6 @@ static struct hda_verb cxt5047_init_verbs[] = { /* configuration for Toshiba Laptops */ static struct hda_verb cxt5047_toshiba_init_verbs[] = { {0x13, AC_VERB_SET_EAPD_BTLENABLE, 0x0}, /* default off */ - /* Speaker routing */ - {0x1d, AC_VERB_SET_CONNECT_SEL, 0x1}, - {} -}; - -/* configuration for HP Laptops */ -static struct hda_verb cxt5047_hp_init_verbs[] = { - /* pin sensing on HP jack */ - {0x13, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, - /* 0x13 is actually shared by both HP and speaker; - * setting the connection to 0 (=0x19) makes the master volume control - * working mysteriouslly... - */ - {0x13, AC_VERB_SET_CONNECT_SEL, 0x0}, - /* Record selector: Ext Mic */ - {0x12, AC_VERB_SET_CONNECT_SEL,0x03}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, - AC_AMP_SET_INPUT|AC_AMP_SET_RIGHT|AC_AMP_SET_LEFT|0x17}, - /* Speaker routing */ - {0x1d, AC_VERB_SET_CONNECT_SEL,0x1}, {} }; @@ -1604,8 +1576,6 @@ static int patch_cxt5047(struct hda_codec *codec) codec->patch_ops.unsol_event = cxt5047_hp2_unsol_event; break; case CXT5047_LAPTOP_HP: - spec->num_init_verbs = 2; - spec->init_verbs[1] = cxt5047_hp_init_verbs; spec->mixers[0] = cxt5047_hp_mixers; codec->patch_ops.unsol_event = cxt5047_hp_unsol_event; codec->patch_ops.init = cxt5047_hp_init; -- cgit v1.2.3-70-g09d2 From df481e41b963b7fc3d7e3543a0c7bb140a682146 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Mar 2009 15:35:35 +0100 Subject: ALSA: hda - Clean up Cxt5047 parser Clean up Conexant 5047 pareser code: - Split mixer elements to separate arrays to reduce the duplicated entires - Fix mixer element names to the standard ones - Remove unneeded cxt5047_hp2_unsol_event; the normal unsol_event handler works fine. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 89 +++++++++--------------------------------- 1 file changed, 19 insertions(+), 70 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 71822140294..d60ccb5bb12 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1218,20 +1218,6 @@ static void cxt5047_hp_automute(struct hda_codec *codec) HDA_AMP_MUTE, bits); } -/* mute internal speaker if HP is plugged */ -static void cxt5047_hp2_automute(struct hda_codec *codec) -{ - struct conexant_spec *spec = codec->spec; - unsigned int bits; - - spec->hp_present = snd_hda_codec_read(codec, 0x13, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - - bits = spec->hp_present ? HDA_AMP_MUTE : 0; - snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); -} - /* toggle input of built-in and mic jack appropriately */ static void cxt5047_hp_automic(struct hda_codec *codec) { @@ -1269,47 +1255,14 @@ static void cxt5047_hp_unsol_event(struct hda_codec *codec, } } -/* unsolicited event for HP jack sensing - non-EAPD systems */ -static void cxt5047_hp2_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - res >>= 26; - switch (res) { - case CONEXANT_HP_EVENT: - cxt5047_hp2_automute(codec); - break; - case CONEXANT_MIC_EVENT: - cxt5047_hp_automic(codec); - break; - } -} - -static struct snd_kcontrol_new cxt5047_mixers[] = { - HDA_CODEC_VOLUME("Mic Bypass Capture Volume", 0x19, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Mic Bypass Capture Switch", 0x19, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Gain Volume", 0x1a, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Mic Gain Switch", 0x1a, 0x0, HDA_OUTPUT), +static struct snd_kcontrol_new cxt5047_base_mixers[] = { + HDA_CODEC_VOLUME("Mic Playback Volume", 0x19, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x19, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x1a, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x03, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x12, 0x03, HDA_INPUT), HDA_CODEC_VOLUME("PCM Volume", 0x10, 0x00, HDA_OUTPUT), HDA_CODEC_MUTE("PCM Switch", 0x10, 0x00, HDA_OUTPUT), - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x1d, 0x00, HDA_OUTPUT), - HDA_CODEC_MUTE("Speaker Playback Switch", 0x1d, 0x00, HDA_OUTPUT), - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x13, 0x00, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x13, 0x00, HDA_OUTPUT), - - {} -}; - -static struct snd_kcontrol_new cxt5047_toshiba_mixers[] = { - HDA_CODEC_VOLUME("Mic Bypass Capture Volume", 0x19, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Mic Bypass Capture Switch", 0x19, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x03, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x12, 0x03, HDA_INPUT), - HDA_CODEC_VOLUME("PCM Volume", 0x10, 0x00, HDA_OUTPUT), - HDA_CODEC_MUTE("PCM Switch", 0x10, 0x00, HDA_OUTPUT), - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x13, 0x00, HDA_OUTPUT), - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x1d, 0x00, HDA_OUTPUT), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", @@ -1322,22 +1275,14 @@ static struct snd_kcontrol_new cxt5047_toshiba_mixers[] = { {} }; -static struct snd_kcontrol_new cxt5047_hp_mixers[] = { - HDA_CODEC_VOLUME("Mic Bypass Capture Volume", 0x19, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Mic Bypass Capture Switch", 0x19,0x02,HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x03, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x12, 0x03, HDA_INPUT), - HDA_CODEC_VOLUME("PCM Volume", 0x10, 0x00, HDA_OUTPUT), - HDA_CODEC_MUTE("PCM Switch", 0x10, 0x00, HDA_OUTPUT), +static struct snd_kcontrol_new cxt5047_hp_spk_mixers[] = { + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x1d, 0x00, HDA_OUTPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x13, 0x00, HDA_OUTPUT), + {} +}; + +static struct snd_kcontrol_new cxt5047_hp_only_mixers[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x13, 0x00, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Switch", - .info = cxt_eapd_info, - .get = cxt_eapd_get, - .put = cxt5047_hp_master_sw_put, - .private_value = 0x13, - }, { } /* end */ }; @@ -1559,7 +1504,7 @@ static int patch_cxt5047(struct hda_codec *codec) spec->adc_nids = cxt5047_adc_nids; spec->capsrc_nids = cxt5047_capsrc_nids; spec->num_mixers = 1; - spec->mixers[0] = cxt5047_mixers; + spec->mixers[0] = cxt5047_base_mixers; spec->num_init_verbs = 1; spec->init_verbs[0] = cxt5047_init_verbs; spec->spdif_route = 0; @@ -1573,18 +1518,22 @@ static int patch_cxt5047(struct hda_codec *codec) cxt5047_cfg_tbl); switch (board_config) { case CXT5047_LAPTOP: - codec->patch_ops.unsol_event = cxt5047_hp2_unsol_event; + spec->num_mixers = 2; + spec->mixers[1] = cxt5047_hp_spk_mixers; + codec->patch_ops.unsol_event = cxt5047_hp_unsol_event; break; case CXT5047_LAPTOP_HP: - spec->mixers[0] = cxt5047_hp_mixers; + spec->num_mixers = 2; + spec->mixers[1] = cxt5047_hp_only_mixers; codec->patch_ops.unsol_event = cxt5047_hp_unsol_event; codec->patch_ops.init = cxt5047_hp_init; break; case CXT5047_LAPTOP_EAPD: spec->input_mux = &cxt5047_toshiba_capture_source; + spec->num_mixers = 2; + spec->mixers[1] = cxt5047_hp_spk_mixers; spec->num_init_verbs = 2; spec->init_verbs[1] = cxt5047_toshiba_init_verbs; - spec->mixers[0] = cxt5047_toshiba_mixers; codec->patch_ops.unsol_event = cxt5047_hp_unsol_event; break; #ifdef CONFIG_SND_DEBUG -- cgit v1.2.3-70-g09d2 From 14cbba89ae967d2e9106a80b270b078d7699109a Mon Sep 17 00:00:00 2001 From: Hugo Villeneuve Date: Mon, 9 Mar 2009 23:32:07 -0400 Subject: ALSA: ASoC: Davinci: Replaced DAI format RIGHT_J by DSP_B for SFFSDR Signed-off-by: Hugo Villeneuve Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-sffsdr.c | 17 ++++++++++------- 1 file changed, 10 insertions(+), 7 deletions(-) diff --git a/sound/soc/davinci/davinci-sffsdr.c b/sound/soc/davinci/davinci-sffsdr.c index 0bf81abba8c..a1ae3736a5d 100644 --- a/sound/soc/davinci/davinci-sffsdr.c +++ b/sound/soc/davinci/davinci-sffsdr.c @@ -36,6 +36,14 @@ #include "davinci-pcm.h" #include "davinci-i2s.h" +/* + * CLKX and CLKR are the inputs for the Sample Rate Generator. + * FSX and FSR are outputs, driven by the sample Rate Generator. + */ +#define AUDIO_FORMAT (SND_SOC_DAIFMT_DSP_B | \ + SND_SOC_DAIFMT_CBM_CFS | \ + SND_SOC_DAIFMT_IB_NF) + static int sffsdr_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -56,13 +64,8 @@ static int sffsdr_hw_params(struct snd_pcm_substream *substream, } #endif - /* Set cpu DAI configuration: - * CLKX and CLKR are the inputs for the Sample Rate Generator. - * FSX and FSR are outputs, driven by the sample Rate Generator. */ - ret = snd_soc_dai_set_fmt(cpu_dai, - SND_SOC_DAIFMT_RIGHT_J | - SND_SOC_DAIFMT_CBM_CFS | - SND_SOC_DAIFMT_IB_NF); + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, AUDIO_FORMAT); if (ret < 0) return ret; -- cgit v1.2.3-70-g09d2 From 090cec81ae9b4ff0c1d301b722f0e6c5fb72d8f9 Mon Sep 17 00:00:00 2001 From: Hugo Villeneuve Date: Mon, 9 Mar 2009 23:32:08 -0400 Subject: ALSA: ASoC: Davinci: Updated sffsdr_hw_params() function to new format Signed-off-by: Hugo Villeneuve Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-sffsdr.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/davinci/davinci-sffsdr.c b/sound/soc/davinci/davinci-sffsdr.c index a1ae3736a5d..40eccfe9e35 100644 --- a/sound/soc/davinci/davinci-sffsdr.c +++ b/sound/soc/davinci/davinci-sffsdr.c @@ -45,8 +45,7 @@ SND_SOC_DAIFMT_IB_NF) static int sffsdr_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) + struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; -- cgit v1.2.3-70-g09d2 From cbf1146d5ee113152c5cdeb54ff9d4b2f0c91736 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Tue, 10 Mar 2009 16:41:00 +0100 Subject: ASoC: don't touch pxa-ssp registers when stream is running In pxa_ssp_set_dai_fmt(), check whether there is anything to do at all. If there would be but the SSP port is in use already, bail out. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- sound/soc/pxa/pxa-ssp.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 7fc13f03d1d..52d97c4b82b 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -522,6 +522,17 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, u32 sscr1; u32 sspsp; + /* check if we need to change anything at all */ + if (priv->dai_fmt == fmt) + return 0; + + /* we can only change the settings if the port is not in use */ + if (ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) { + dev_err(&ssp->pdev->dev, + "can't change hardware dai format: stream is in use"); + return -EINVAL; + } + /* reset port settings */ sscr0 = ssp_read_reg(ssp, SSCR0) & (SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); -- cgit v1.2.3-70-g09d2 From f455dfb106916d855d59686fe16575c2ceb2cb2a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 10 Mar 2009 19:51:07 +0000 Subject: ASoC: Fix up merge with the ARM tree The same change has been made with the final lines in slightly differnet orders. Signed-off-by: Mark Brown --- arch/arm/mach-shark/include/mach/io.h | 1 + 1 file changed, 1 insertion(+) diff --git a/arch/arm/mach-shark/include/mach/io.h b/arch/arm/mach-shark/include/mach/io.h index 8ca7d7f09bd..568daea93fb 100644 --- a/arch/arm/mach-shark/include/mach/io.h +++ b/arch/arm/mach-shark/include/mach/io.h @@ -15,6 +15,7 @@ #define IO_SPACE_LIMIT 0xffffffff #define __io(a) __typesafe_io(PCIO_BASE + (a)) + #define __mem_pci(addr) (addr) #endif -- cgit v1.2.3-70-g09d2 From 47e78ecc2adb778c7d2b54924e90433a0182a6ba Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 11 Mar 2009 09:50:19 +0100 Subject: ALSA: Remove obsolete snd_xferv struct and ioctls Removed obsleted snd_xferv struct and ioctls that are no longer used in the current codebase. Signed-off-by: Takashi Iwai --- include/sound/asound.h | 14 -------------- 1 file changed, 14 deletions(-) diff --git a/include/sound/asound.h b/include/sound/asound.h index 1c02ed1d7c4..b6e01e6b3f8 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -919,18 +919,4 @@ struct snd_ctl_event { #define SNDRV_CTL_NAME_IEC958_PCM_STREAM "PCM Stream" #define SNDRV_CTL_NAME_IEC958(expl,direction,what) "IEC958 " expl SNDRV_CTL_NAME_##direction SNDRV_CTL_NAME_IEC958_##what -/* - * - */ - -struct snd_xferv { - const struct iovec *vector; - unsigned long count; -}; - -enum { - SNDRV_IOCTL_READV = _IOW('K', 0x00, struct snd_xferv), - SNDRV_IOCTL_WRITEV = _IOW('K', 0x01, struct snd_xferv), -}; - #endif /* __SOUND_ASOUND_H */ -- cgit v1.2.3-70-g09d2 From 78a05b522044a50dc2a6811d10b9ee3f7c3e78f8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 11 Mar 2009 09:52:28 +0100 Subject: ALSA: Use define for ioctl definitions Use define instead of enum for ioctl definitions since strace can't parse ioctls defined via enum properly. Signed-off-by: Takashi Iwai --- include/sound/asound.h | 171 +++++++++++++++++++++------------------------- include/sound/sfnt_info.h | 14 ++-- 2 files changed, 85 insertions(+), 100 deletions(-) diff --git a/include/sound/asound.h b/include/sound/asound.h index b6e01e6b3f8..fad3e0c7b93 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -126,12 +126,10 @@ struct snd_hwdep_dsp_image { unsigned long driver_data; /* W: driver-specific data */ }; -enum { - SNDRV_HWDEP_IOCTL_PVERSION = _IOR ('H', 0x00, int), - SNDRV_HWDEP_IOCTL_INFO = _IOR ('H', 0x01, struct snd_hwdep_info), - SNDRV_HWDEP_IOCTL_DSP_STATUS = _IOR('H', 0x02, struct snd_hwdep_dsp_status), - SNDRV_HWDEP_IOCTL_DSP_LOAD = _IOW('H', 0x03, struct snd_hwdep_dsp_image) -}; +#define SNDRV_HWDEP_IOCTL_PVERSION _IOR ('H', 0x00, int) +#define SNDRV_HWDEP_IOCTL_INFO _IOR ('H', 0x01, struct snd_hwdep_info) +#define SNDRV_HWDEP_IOCTL_DSP_STATUS _IOR('H', 0x02, struct snd_hwdep_dsp_status) +#define SNDRV_HWDEP_IOCTL_DSP_LOAD _IOW('H', 0x03, struct snd_hwdep_dsp_image) /***************************************************************************** * * @@ -451,40 +449,35 @@ enum { SNDRV_PCM_TSTAMP_TYPE_LAST = SNDRV_PCM_TSTAMP_TYPE_MONOTONIC, }; -enum { - SNDRV_PCM_IOCTL_PVERSION = _IOR('A', 0x00, int), - SNDRV_PCM_IOCTL_INFO = _IOR('A', 0x01, struct snd_pcm_info), - SNDRV_PCM_IOCTL_TSTAMP = _IOW('A', 0x02, int), - SNDRV_PCM_IOCTL_TTSTAMP = _IOW('A', 0x03, int), - SNDRV_PCM_IOCTL_HW_REFINE = _IOWR('A', 0x10, struct snd_pcm_hw_params), - SNDRV_PCM_IOCTL_HW_PARAMS = _IOWR('A', 0x11, struct snd_pcm_hw_params), - SNDRV_PCM_IOCTL_HW_FREE = _IO('A', 0x12), - SNDRV_PCM_IOCTL_SW_PARAMS = _IOWR('A', 0x13, struct snd_pcm_sw_params), - SNDRV_PCM_IOCTL_STATUS = _IOR('A', 0x20, struct snd_pcm_status), - SNDRV_PCM_IOCTL_DELAY = _IOR('A', 0x21, snd_pcm_sframes_t), - SNDRV_PCM_IOCTL_HWSYNC = _IO('A', 0x22), - SNDRV_PCM_IOCTL_SYNC_PTR = _IOWR('A', 0x23, struct snd_pcm_sync_ptr), - SNDRV_PCM_IOCTL_CHANNEL_INFO = _IOR('A', 0x32, struct snd_pcm_channel_info), - SNDRV_PCM_IOCTL_PREPARE = _IO('A', 0x40), - SNDRV_PCM_IOCTL_RESET = _IO('A', 0x41), - SNDRV_PCM_IOCTL_START = _IO('A', 0x42), - SNDRV_PCM_IOCTL_DROP = _IO('A', 0x43), - SNDRV_PCM_IOCTL_DRAIN = _IO('A', 0x44), - SNDRV_PCM_IOCTL_PAUSE = _IOW('A', 0x45, int), - SNDRV_PCM_IOCTL_REWIND = _IOW('A', 0x46, snd_pcm_uframes_t), - SNDRV_PCM_IOCTL_RESUME = _IO('A', 0x47), - SNDRV_PCM_IOCTL_XRUN = _IO('A', 0x48), - SNDRV_PCM_IOCTL_FORWARD = _IOW('A', 0x49, snd_pcm_uframes_t), - SNDRV_PCM_IOCTL_WRITEI_FRAMES = _IOW('A', 0x50, struct snd_xferi), - SNDRV_PCM_IOCTL_READI_FRAMES = _IOR('A', 0x51, struct snd_xferi), - SNDRV_PCM_IOCTL_WRITEN_FRAMES = _IOW('A', 0x52, struct snd_xfern), - SNDRV_PCM_IOCTL_READN_FRAMES = _IOR('A', 0x53, struct snd_xfern), - SNDRV_PCM_IOCTL_LINK = _IOW('A', 0x60, int), - SNDRV_PCM_IOCTL_UNLINK = _IO('A', 0x61), -}; - -/* Trick to make alsa-lib/acinclude.m4 happy */ -#define SNDRV_PCM_IOCTL_REWIND SNDRV_PCM_IOCTL_REWIND +#define SNDRV_PCM_IOCTL_PVERSION _IOR('A', 0x00, int) +#define SNDRV_PCM_IOCTL_INFO _IOR('A', 0x01, struct snd_pcm_info) +#define SNDRV_PCM_IOCTL_TSTAMP _IOW('A', 0x02, int) +#define SNDRV_PCM_IOCTL_TTSTAMP _IOW('A', 0x03, int) +#define SNDRV_PCM_IOCTL_HW_REFINE _IOWR('A', 0x10, struct snd_pcm_hw_params) +#define SNDRV_PCM_IOCTL_HW_PARAMS _IOWR('A', 0x11, struct snd_pcm_hw_params) +#define SNDRV_PCM_IOCTL_HW_FREE _IO('A', 0x12) +#define SNDRV_PCM_IOCTL_SW_PARAMS _IOWR('A', 0x13, struct snd_pcm_sw_params) +#define SNDRV_PCM_IOCTL_STATUS _IOR('A', 0x20, struct snd_pcm_status) +#define SNDRV_PCM_IOCTL_DELAY _IOR('A', 0x21, snd_pcm_sframes_t) +#define SNDRV_PCM_IOCTL_HWSYNC _IO('A', 0x22) +#define SNDRV_PCM_IOCTL_SYNC_PTR _IOWR('A', 0x23, struct snd_pcm_sync_ptr) +#define SNDRV_PCM_IOCTL_CHANNEL_INFO _IOR('A', 0x32, struct snd_pcm_channel_info) +#define SNDRV_PCM_IOCTL_PREPARE _IO('A', 0x40) +#define SNDRV_PCM_IOCTL_RESET _IO('A', 0x41) +#define SNDRV_PCM_IOCTL_START _IO('A', 0x42) +#define SNDRV_PCM_IOCTL_DROP _IO('A', 0x43) +#define SNDRV_PCM_IOCTL_DRAIN _IO('A', 0x44) +#define SNDRV_PCM_IOCTL_PAUSE _IOW('A', 0x45, int) +#define SNDRV_PCM_IOCTL_REWIND _IOW('A', 0x46, snd_pcm_uframes_t) +#define SNDRV_PCM_IOCTL_RESUME _IO('A', 0x47) +#define SNDRV_PCM_IOCTL_XRUN _IO('A', 0x48) +#define SNDRV_PCM_IOCTL_FORWARD _IOW('A', 0x49, snd_pcm_uframes_t) +#define SNDRV_PCM_IOCTL_WRITEI_FRAMES _IOW('A', 0x50, struct snd_xferi) +#define SNDRV_PCM_IOCTL_READI_FRAMES _IOR('A', 0x51, struct snd_xferi) +#define SNDRV_PCM_IOCTL_WRITEN_FRAMES _IOW('A', 0x52, struct snd_xfern) +#define SNDRV_PCM_IOCTL_READN_FRAMES _IOR('A', 0x53, struct snd_xfern) +#define SNDRV_PCM_IOCTL_LINK _IOW('A', 0x60, int) +#define SNDRV_PCM_IOCTL_UNLINK _IO('A', 0x61) /***************************************************************************** * * @@ -538,14 +531,12 @@ struct snd_rawmidi_status { unsigned char reserved[16]; /* reserved for future use */ }; -enum { - SNDRV_RAWMIDI_IOCTL_PVERSION = _IOR('W', 0x00, int), - SNDRV_RAWMIDI_IOCTL_INFO = _IOR('W', 0x01, struct snd_rawmidi_info), - SNDRV_RAWMIDI_IOCTL_PARAMS = _IOWR('W', 0x10, struct snd_rawmidi_params), - SNDRV_RAWMIDI_IOCTL_STATUS = _IOWR('W', 0x20, struct snd_rawmidi_status), - SNDRV_RAWMIDI_IOCTL_DROP = _IOW('W', 0x30, int), - SNDRV_RAWMIDI_IOCTL_DRAIN = _IOW('W', 0x31, int), -}; +#define SNDRV_RAWMIDI_IOCTL_PVERSION _IOR('W', 0x00, int) +#define SNDRV_RAWMIDI_IOCTL_INFO _IOR('W', 0x01, struct snd_rawmidi_info) +#define SNDRV_RAWMIDI_IOCTL_PARAMS _IOWR('W', 0x10, struct snd_rawmidi_params) +#define SNDRV_RAWMIDI_IOCTL_STATUS _IOWR('W', 0x20, struct snd_rawmidi_status) +#define SNDRV_RAWMIDI_IOCTL_DROP _IOW('W', 0x30, int) +#define SNDRV_RAWMIDI_IOCTL_DRAIN _IOW('W', 0x31, int) /* * Timer section - /dev/snd/timer @@ -654,23 +645,21 @@ struct snd_timer_status { unsigned char reserved[64]; /* reserved */ }; -enum { - SNDRV_TIMER_IOCTL_PVERSION = _IOR('T', 0x00, int), - SNDRV_TIMER_IOCTL_NEXT_DEVICE = _IOWR('T', 0x01, struct snd_timer_id), - SNDRV_TIMER_IOCTL_TREAD = _IOW('T', 0x02, int), - SNDRV_TIMER_IOCTL_GINFO = _IOWR('T', 0x03, struct snd_timer_ginfo), - SNDRV_TIMER_IOCTL_GPARAMS = _IOW('T', 0x04, struct snd_timer_gparams), - SNDRV_TIMER_IOCTL_GSTATUS = _IOWR('T', 0x05, struct snd_timer_gstatus), - SNDRV_TIMER_IOCTL_SELECT = _IOW('T', 0x10, struct snd_timer_select), - SNDRV_TIMER_IOCTL_INFO = _IOR('T', 0x11, struct snd_timer_info), - SNDRV_TIMER_IOCTL_PARAMS = _IOW('T', 0x12, struct snd_timer_params), - SNDRV_TIMER_IOCTL_STATUS = _IOR('T', 0x14, struct snd_timer_status), - /* The following four ioctls are changed since 1.0.9 due to confliction */ - SNDRV_TIMER_IOCTL_START = _IO('T', 0xa0), - SNDRV_TIMER_IOCTL_STOP = _IO('T', 0xa1), - SNDRV_TIMER_IOCTL_CONTINUE = _IO('T', 0xa2), - SNDRV_TIMER_IOCTL_PAUSE = _IO('T', 0xa3), -}; +#define SNDRV_TIMER_IOCTL_PVERSION _IOR('T', 0x00, int) +#define SNDRV_TIMER_IOCTL_NEXT_DEVICE _IOWR('T', 0x01, struct snd_timer_id) +#define SNDRV_TIMER_IOCTL_TREAD _IOW('T', 0x02, int) +#define SNDRV_TIMER_IOCTL_GINFO _IOWR('T', 0x03, struct snd_timer_ginfo) +#define SNDRV_TIMER_IOCTL_GPARAMS _IOW('T', 0x04, struct snd_timer_gparams) +#define SNDRV_TIMER_IOCTL_GSTATUS _IOWR('T', 0x05, struct snd_timer_gstatus) +#define SNDRV_TIMER_IOCTL_SELECT _IOW('T', 0x10, struct snd_timer_select) +#define SNDRV_TIMER_IOCTL_INFO _IOR('T', 0x11, struct snd_timer_info) +#define SNDRV_TIMER_IOCTL_PARAMS _IOW('T', 0x12, struct snd_timer_params) +#define SNDRV_TIMER_IOCTL_STATUS _IOR('T', 0x14, struct snd_timer_status) +/* The following four ioctls are changed since 1.0.9 due to confliction */ +#define SNDRV_TIMER_IOCTL_START _IO('T', 0xa0) +#define SNDRV_TIMER_IOCTL_STOP _IO('T', 0xa1) +#define SNDRV_TIMER_IOCTL_CONTINUE _IO('T', 0xa2) +#define SNDRV_TIMER_IOCTL_PAUSE _IO('T', 0xa3) struct snd_timer_read { unsigned int resolution; @@ -847,33 +836,31 @@ struct snd_ctl_tlv { unsigned int tlv[0]; /* first TLV */ }; -enum { - SNDRV_CTL_IOCTL_PVERSION = _IOR('U', 0x00, int), - SNDRV_CTL_IOCTL_CARD_INFO = _IOR('U', 0x01, struct snd_ctl_card_info), - SNDRV_CTL_IOCTL_ELEM_LIST = _IOWR('U', 0x10, struct snd_ctl_elem_list), - SNDRV_CTL_IOCTL_ELEM_INFO = _IOWR('U', 0x11, struct snd_ctl_elem_info), - SNDRV_CTL_IOCTL_ELEM_READ = _IOWR('U', 0x12, struct snd_ctl_elem_value), - SNDRV_CTL_IOCTL_ELEM_WRITE = _IOWR('U', 0x13, struct snd_ctl_elem_value), - SNDRV_CTL_IOCTL_ELEM_LOCK = _IOW('U', 0x14, struct snd_ctl_elem_id), - SNDRV_CTL_IOCTL_ELEM_UNLOCK = _IOW('U', 0x15, struct snd_ctl_elem_id), - SNDRV_CTL_IOCTL_SUBSCRIBE_EVENTS = _IOWR('U', 0x16, int), - SNDRV_CTL_IOCTL_ELEM_ADD = _IOWR('U', 0x17, struct snd_ctl_elem_info), - SNDRV_CTL_IOCTL_ELEM_REPLACE = _IOWR('U', 0x18, struct snd_ctl_elem_info), - SNDRV_CTL_IOCTL_ELEM_REMOVE = _IOWR('U', 0x19, struct snd_ctl_elem_id), - SNDRV_CTL_IOCTL_TLV_READ = _IOWR('U', 0x1a, struct snd_ctl_tlv), - SNDRV_CTL_IOCTL_TLV_WRITE = _IOWR('U', 0x1b, struct snd_ctl_tlv), - SNDRV_CTL_IOCTL_TLV_COMMAND = _IOWR('U', 0x1c, struct snd_ctl_tlv), - SNDRV_CTL_IOCTL_HWDEP_NEXT_DEVICE = _IOWR('U', 0x20, int), - SNDRV_CTL_IOCTL_HWDEP_INFO = _IOR('U', 0x21, struct snd_hwdep_info), - SNDRV_CTL_IOCTL_PCM_NEXT_DEVICE = _IOR('U', 0x30, int), - SNDRV_CTL_IOCTL_PCM_INFO = _IOWR('U', 0x31, struct snd_pcm_info), - SNDRV_CTL_IOCTL_PCM_PREFER_SUBDEVICE = _IOW('U', 0x32, int), - SNDRV_CTL_IOCTL_RAWMIDI_NEXT_DEVICE = _IOWR('U', 0x40, int), - SNDRV_CTL_IOCTL_RAWMIDI_INFO = _IOWR('U', 0x41, struct snd_rawmidi_info), - SNDRV_CTL_IOCTL_RAWMIDI_PREFER_SUBDEVICE = _IOW('U', 0x42, int), - SNDRV_CTL_IOCTL_POWER = _IOWR('U', 0xd0, int), - SNDRV_CTL_IOCTL_POWER_STATE = _IOR('U', 0xd1, int), -}; +#define SNDRV_CTL_IOCTL_PVERSION _IOR('U', 0x00, int) +#define SNDRV_CTL_IOCTL_CARD_INFO _IOR('U', 0x01, struct snd_ctl_card_info) +#define SNDRV_CTL_IOCTL_ELEM_LIST _IOWR('U', 0x10, struct snd_ctl_elem_list) +#define SNDRV_CTL_IOCTL_ELEM_INFO _IOWR('U', 0x11, struct snd_ctl_elem_info) +#define SNDRV_CTL_IOCTL_ELEM_READ _IOWR('U', 0x12, struct snd_ctl_elem_value) +#define SNDRV_CTL_IOCTL_ELEM_WRITE _IOWR('U', 0x13, struct snd_ctl_elem_value) +#define SNDRV_CTL_IOCTL_ELEM_LOCK _IOW('U', 0x14, struct snd_ctl_elem_id) +#define SNDRV_CTL_IOCTL_ELEM_UNLOCK _IOW('U', 0x15, struct snd_ctl_elem_id) +#define SNDRV_CTL_IOCTL_SUBSCRIBE_EVENTS _IOWR('U', 0x16, int) +#define SNDRV_CTL_IOCTL_ELEM_ADD _IOWR('U', 0x17, struct snd_ctl_elem_info) +#define SNDRV_CTL_IOCTL_ELEM_REPLACE _IOWR('U', 0x18, struct snd_ctl_elem_info) +#define SNDRV_CTL_IOCTL_ELEM_REMOVE _IOWR('U', 0x19, struct snd_ctl_elem_id) +#define SNDRV_CTL_IOCTL_TLV_READ _IOWR('U', 0x1a, struct snd_ctl_tlv) +#define SNDRV_CTL_IOCTL_TLV_WRITE _IOWR('U', 0x1b, struct snd_ctl_tlv) +#define SNDRV_CTL_IOCTL_TLV_COMMAND _IOWR('U', 0x1c, struct snd_ctl_tlv) +#define SNDRV_CTL_IOCTL_HWDEP_NEXT_DEVICE _IOWR('U', 0x20, int) +#define SNDRV_CTL_IOCTL_HWDEP_INFO _IOR('U', 0x21, struct snd_hwdep_info) +#define SNDRV_CTL_IOCTL_PCM_NEXT_DEVICE _IOR('U', 0x30, int) +#define SNDRV_CTL_IOCTL_PCM_INFO _IOWR('U', 0x31, struct snd_pcm_info) +#define SNDRV_CTL_IOCTL_PCM_PREFER_SUBDEVICE _IOW('U', 0x32, int) +#define SNDRV_CTL_IOCTL_RAWMIDI_NEXT_DEVICE _IOWR('U', 0x40, int) +#define SNDRV_CTL_IOCTL_RAWMIDI_INFO _IOWR('U', 0x41, struct snd_rawmidi_info) +#define SNDRV_CTL_IOCTL_RAWMIDI_PREFER_SUBDEVICE _IOW('U', 0x42, int) +#define SNDRV_CTL_IOCTL_POWER _IOWR('U', 0xd0, int) +#define SNDRV_CTL_IOCTL_POWER_STATE _IOR('U', 0xd1, int) /* * Read interface. diff --git a/include/sound/sfnt_info.h b/include/sound/sfnt_info.h index 5d1ab9c4950..1bce7fd1725 100644 --- a/include/sound/sfnt_info.h +++ b/include/sound/sfnt_info.h @@ -202,13 +202,11 @@ struct snd_emux_misc_mode { int value2; /* reserved */ }; -enum { - SNDRV_EMUX_IOCTL_VERSION = _IOR('H', 0x80, unsigned int), - SNDRV_EMUX_IOCTL_LOAD_PATCH = _IOWR('H', 0x81, struct soundfont_patch_info), - SNDRV_EMUX_IOCTL_RESET_SAMPLES = _IO('H', 0x82), - SNDRV_EMUX_IOCTL_REMOVE_LAST_SAMPLES = _IO('H', 0x83), - SNDRV_EMUX_IOCTL_MEM_AVAIL = _IOW('H', 0x84, int), - SNDRV_EMUX_IOCTL_MISC_MODE = _IOWR('H', 0x84, struct snd_emux_misc_mode), -}; +#define SNDRV_EMUX_IOCTL_VERSION _IOR('H', 0x80, unsigned int) +#define SNDRV_EMUX_IOCTL_LOAD_PATCH _IOWR('H', 0x81, struct soundfont_patch_info) +#define SNDRV_EMUX_IOCTL_RESET_SAMPLES _IO('H', 0x82) +#define SNDRV_EMUX_IOCTL_REMOVE_LAST_SAMPLES _IO('H', 0x83) +#define SNDRV_EMUX_IOCTL_MEM_AVAIL _IOW('H', 0x84, int) +#define SNDRV_EMUX_IOCTL_MISC_MODE _IOWR('H', 0x84, struct snd_emux_misc_mode) #endif /* __SOUND_SFNT_INFO_H */ -- cgit v1.2.3-70-g09d2 From a2b03461cb072eb6097a55ec0289294b09382284 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 11 Mar 2009 11:02:33 +0000 Subject: [ARM] Revert extraneous changes from the S3C audio header move These changes were included in the S3C audio header move but are not directly related to it. Signed-off-by: Mark Brown --- arch/arm/mach-s3c2410/include/mach/hardware.h | 3 +++ arch/arm/mach-shark/include/mach/io.h | 2 +- 2 files changed, 4 insertions(+), 1 deletion(-) diff --git a/arch/arm/mach-s3c2410/include/mach/hardware.h b/arch/arm/mach-s3c2410/include/mach/hardware.h index db72beb61d7..74d5a1a4024 100644 --- a/arch/arm/mach-s3c2410/include/mach/hardware.h +++ b/arch/arm/mach-s3c2410/include/mach/hardware.h @@ -131,4 +131,7 @@ extern int s3c2412_gpio_set_sleepcfg(unsigned int pin, unsigned int state); /* machine specific hardware definitions should go after this */ +/* currently here until moved into config (todo) */ +#define CONFIG_NO_MULTIWORD_IO + #endif /* __ASM_ARCH_HARDWARE_H */ diff --git a/arch/arm/mach-shark/include/mach/io.h b/arch/arm/mach-shark/include/mach/io.h index 8ca7d7f09bd..c5cee829fc8 100644 --- a/arch/arm/mach-shark/include/mach/io.h +++ b/arch/arm/mach-shark/include/mach/io.h @@ -14,7 +14,7 @@ #define PCIO_BASE 0xe0000000 #define IO_SPACE_LIMIT 0xffffffff -#define __io(a) __typesafe_io(PCIO_BASE + (a)) +#define __io(a) ((void __iomem *)(PCIO_BASE + (a))) #define __mem_pci(addr) (addr) #endif -- cgit v1.2.3-70-g09d2 From 5706d5013212c8afcb9fe5332ee6442488280c66 Mon Sep 17 00:00:00 2001 From: David Brownell Date: Wed, 11 Mar 2009 02:37:25 -0800 Subject: ASoC: buildfix for OSK Buildfix: CC sound/soc/omap/osk5912.o sound/soc/omap/osk5912.c: In function 'osk_soc_init': sound/soc/omap/osk5912.c:189: error: implicit declaration of function 'clk_get_usecount' make[3]: *** [sound/soc/omap/osk5912.o] Error 1 There's no such (standard) clock interface. Signed-off-by: David Brownell Signed-off-by: Mark Brown --- sound/soc/omap/osk5912.c | 12 ++---------- 1 file changed, 2 insertions(+), 10 deletions(-) diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c index cd41a948df7..a952a4eb336 100644 --- a/sound/soc/omap/osk5912.c +++ b/sound/soc/omap/osk5912.c @@ -186,13 +186,6 @@ static int __init osk_soc_init(void) return -ENODEV; } - if (clk_get_usecount(tlv320aic23_mclk) > 0) { - /* MCLK is already in use */ - printk(KERN_WARNING - "MCLK in use at %d Hz. We change it to %d Hz\n", - (uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK); - } - /* * Configure 12 MHz output on MCLK. */ @@ -205,9 +198,8 @@ static int __init osk_soc_init(void) } } - printk(KERN_INFO "MCLK = %d [%d], usecount = %d\n", - (uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK, - clk_get_usecount(tlv320aic23_mclk)); + printk(KERN_INFO "MCLK = %d [%d]\n", + (uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK); return 0; err1: -- cgit v1.2.3-70-g09d2 From aaf1e176fa9a96fe1eea33b710684bba066aedc1 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 10 Mar 2009 10:55:15 +0000 Subject: ASoC: Add initial driver for the WM8400 CODEC The WM8400 is a highly integrated audio CODEC and power management unit intended for mobile multimedia application. This driver supports the primary audio CODEC features, including: - 1W speaker driver - Fully differential headphone output - Up to 4 differential microphone inputs Signed-off-by: Mark Brown --- include/linux/mfd/wm8400-audio.h | 1 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8400.c | 1479 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8400.h | 62 ++ 5 files changed, 1548 insertions(+) create mode 100644 sound/soc/codecs/wm8400.c create mode 100644 sound/soc/codecs/wm8400.h diff --git a/include/linux/mfd/wm8400-audio.h b/include/linux/mfd/wm8400-audio.h index b6640e01804..e06ed3eb1d0 100644 --- a/include/linux/mfd/wm8400-audio.h +++ b/include/linux/mfd/wm8400-audio.h @@ -1181,6 +1181,7 @@ #define WM8400_FLL_OUTDIV_SHIFT 0 /* FLL_OUTDIV - [2:0] */ #define WM8400_FLL_OUTDIV_WIDTH 3 /* FLL_OUTDIV - [2:0] */ +struct wm8400; void wm8400_reset_codec_reg_cache(struct wm8400 *wm8400); #endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index a1af311e7f0..b6c7f7a01cb 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -26,6 +26,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_UDA134X select SND_SOC_UDA1380 if I2C select SND_SOC_WM8350 if MFD_WM8350 + select SND_SOC_WM8400 if MFD_WM8400 select SND_SOC_WM8510 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8580 if I2C select SND_SOC_WM8728 if SND_SOC_I2C_AND_SPI @@ -110,6 +111,9 @@ config SND_SOC_UDA1380 config SND_SOC_WM8350 tristate +config SND_SOC_WM8400 + tristate + config SND_SOC_WM8510 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 4717c3c9904..030d2454725 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -14,6 +14,7 @@ snd-soc-twl4030-objs := twl4030.o snd-soc-uda134x-objs := uda134x.o snd-soc-uda1380-objs := uda1380.o snd-soc-wm8350-objs := wm8350.o +snd-soc-wm8400-objs := wm8400.o snd-soc-wm8510-objs := wm8510.o snd-soc-wm8580-objs := wm8580.o snd-soc-wm8728-objs := wm8728.o @@ -44,6 +45,7 @@ obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o obj-$(CONFIG_SND_SOC_WM8350) += snd-soc-wm8350.o +obj-$(CONFIG_SND_SOC_WM8400) += snd-soc-wm8400.o obj-$(CONFIG_SND_SOC_WM8510) += snd-soc-wm8510.o obj-$(CONFIG_SND_SOC_WM8580) += snd-soc-wm8580.o obj-$(CONFIG_SND_SOC_WM8728) += snd-soc-wm8728.o diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c new file mode 100644 index 00000000000..9cb73d9d023 --- /dev/null +++ b/sound/soc/codecs/wm8400.c @@ -0,0 +1,1479 @@ +/* + * wm8400.c -- WM8400 ALSA Soc Audio driver + * + * Copyright 2008, 2009 Wolfson Microelectronics PLC. + * Author: Mark Brown + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "wm8400.h" + +/* Fake register for internal state */ +#define WM8400_INTDRIVBITS (WM8400_REGISTER_COUNT + 1) +#define WM8400_INMIXL_PWR 0 +#define WM8400_AINLMUX_PWR 1 +#define WM8400_INMIXR_PWR 2 +#define WM8400_AINRMUX_PWR 3 + +static struct regulator_bulk_data power[] = { + { + .supply = "I2S1VDD", + }, + { + .supply = "I2S2VDD", + }, + { + .supply = "DCVDD", + }, + { + .supply = "FLLVDD", + }, + { + .supply = "HPVDD", + }, + { + .supply = "SPKVDD", + }, +}; + +/* codec private data */ +struct wm8400_priv { + struct snd_soc_codec codec; + struct wm8400 *wm8400; + u16 fake_register; + unsigned int sysclk; + unsigned int pcmclk; + struct work_struct work; +}; + +static inline unsigned int wm8400_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + struct wm8400_priv *wm8400 = codec->private_data; + + if (reg == WM8400_INTDRIVBITS) + return wm8400->fake_register; + else + return wm8400_reg_read(wm8400->wm8400, reg); +} + +/* + * write to the wm8400 register space + */ +static int wm8400_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + struct wm8400_priv *wm8400 = codec->private_data; + + if (reg == WM8400_INTDRIVBITS) { + wm8400->fake_register = value; + return 0; + } else + return wm8400_set_bits(wm8400->wm8400, reg, 0xffff, value); +} + +static void wm8400_codec_reset(struct snd_soc_codec *codec) +{ + struct wm8400_priv *wm8400 = codec->private_data; + + wm8400_reset_codec_reg_cache(wm8400->wm8400); +} + +static const DECLARE_TLV_DB_LINEAR(rec_mix_tlv, -1500, 600); + +static const DECLARE_TLV_DB_LINEAR(in_pga_tlv, -1650, 3000); + +static const DECLARE_TLV_DB_LINEAR(out_mix_tlv, -2100, 0); + +static const DECLARE_TLV_DB_LINEAR(out_pga_tlv, -7300, 600); + +static const DECLARE_TLV_DB_LINEAR(out_omix_tlv, -600, 0); + +static const DECLARE_TLV_DB_LINEAR(out_dac_tlv, -7163, 0); + +static const DECLARE_TLV_DB_LINEAR(in_adc_tlv, -7163, 1763); + +static const DECLARE_TLV_DB_LINEAR(out_sidetone_tlv, -3600, 0); + +static int wm8400_outpga_put_volsw_vu(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int reg = mc->reg; + int ret; + u16 val; + + ret = snd_soc_put_volsw(kcontrol, ucontrol); + if (ret < 0) + return ret; + + /* now hit the volume update bits (always bit 8) */ + val = wm8400_read(codec, reg); + return wm8400_write(codec, reg, val | 0x0100); +} + +#define WM8400_OUTPGA_SINGLE_R_TLV(xname, reg, shift, max, invert, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ + SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw, \ + .get = snd_soc_get_volsw, .put = wm8400_outpga_put_volsw_vu, \ + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } + + +static const char *wm8400_digital_sidetone[] = + {"None", "Left ADC", "Right ADC", "Reserved"}; + +static const struct soc_enum wm8400_left_digital_sidetone_enum = +SOC_ENUM_SINGLE(WM8400_DIGITAL_SIDE_TONE, + WM8400_ADC_TO_DACL_SHIFT, 2, wm8400_digital_sidetone); + +static const struct soc_enum wm8400_right_digital_sidetone_enum = +SOC_ENUM_SINGLE(WM8400_DIGITAL_SIDE_TONE, + WM8400_ADC_TO_DACR_SHIFT, 2, wm8400_digital_sidetone); + +static const char *wm8400_adcmode[] = + {"Hi-fi mode", "Voice mode 1", "Voice mode 2", "Voice mode 3"}; + +static const struct soc_enum wm8400_right_adcmode_enum = +SOC_ENUM_SINGLE(WM8400_ADC_CTRL, WM8400_ADC_HPF_CUT_SHIFT, 3, wm8400_adcmode); + +static const struct snd_kcontrol_new wm8400_snd_controls[] = { +/* INMIXL */ +SOC_SINGLE("LIN12 PGA Boost", WM8400_INPUT_MIXER3, WM8400_L12MNBST_SHIFT, + 1, 0), +SOC_SINGLE("LIN34 PGA Boost", WM8400_INPUT_MIXER3, WM8400_L34MNBST_SHIFT, + 1, 0), +/* INMIXR */ +SOC_SINGLE("RIN12 PGA Boost", WM8400_INPUT_MIXER3, WM8400_R12MNBST_SHIFT, + 1, 0), +SOC_SINGLE("RIN34 PGA Boost", WM8400_INPUT_MIXER3, WM8400_R34MNBST_SHIFT, + 1, 0), + +/* LOMIX */ +SOC_SINGLE_TLV("LOMIX LIN3 Bypass Volume", WM8400_OUTPUT_MIXER3, + WM8400_LLI3LOVOL_SHIFT, 7, 0, out_mix_tlv), +SOC_SINGLE_TLV("LOMIX RIN12 PGA Bypass Volume", WM8400_OUTPUT_MIXER3, + WM8400_LR12LOVOL_SHIFT, 7, 0, out_mix_tlv), +SOC_SINGLE_TLV("LOMIX LIN12 PGA Bypass Volume", WM8400_OUTPUT_MIXER3, + WM8400_LL12LOVOL_SHIFT, 7, 0, out_mix_tlv), +SOC_SINGLE_TLV("LOMIX RIN3 Bypass Volume", WM8400_OUTPUT_MIXER5, + WM8400_LRI3LOVOL_SHIFT, 7, 0, out_mix_tlv), +SOC_SINGLE_TLV("LOMIX AINRMUX Bypass Volume", WM8400_OUTPUT_MIXER5, + WM8400_LRBLOVOL_SHIFT, 7, 0, out_mix_tlv), +SOC_SINGLE_TLV("LOMIX AINLMUX Bypass Volume", WM8400_OUTPUT_MIXER5, + WM8400_LRBLOVOL_SHIFT, 7, 0, out_mix_tlv), + +/* ROMIX */ +SOC_SINGLE_TLV("ROMIX RIN3 Bypass Volume", WM8400_OUTPUT_MIXER4, + WM8400_RRI3ROVOL_SHIFT, 7, 0, out_mix_tlv), +SOC_SINGLE_TLV("ROMIX LIN12 PGA Bypass Volume", WM8400_OUTPUT_MIXER4, + WM8400_RL12ROVOL_SHIFT, 7, 0, out_mix_tlv), +SOC_SINGLE_TLV("ROMIX RIN12 PGA Bypass Volume", WM8400_OUTPUT_MIXER4, + WM8400_RR12ROVOL_SHIFT, 7, 0, out_mix_tlv), +SOC_SINGLE_TLV("ROMIX LIN3 Bypass Volume", WM8400_OUTPUT_MIXER6, + WM8400_RLI3ROVOL_SHIFT, 7, 0, out_mix_tlv), +SOC_SINGLE_TLV("ROMIX AINLMUX Bypass Volume", WM8400_OUTPUT_MIXER6, + WM8400_RLBROVOL_SHIFT, 7, 0, out_mix_tlv), +SOC_SINGLE_TLV("ROMIX AINRMUX Bypass Volume", WM8400_OUTPUT_MIXER6, + WM8400_RRBROVOL_SHIFT, 7, 0, out_mix_tlv), + +/* LOUT */ +WM8400_OUTPGA_SINGLE_R_TLV("LOUT Volume", WM8400_LEFT_OUTPUT_VOLUME, + WM8400_LOUTVOL_SHIFT, WM8400_LOUTVOL_MASK, 0, out_pga_tlv), +SOC_SINGLE("LOUT ZC", WM8400_LEFT_OUTPUT_VOLUME, WM8400_LOZC_SHIFT, 1, 0), + +/* ROUT */ +WM8400_OUTPGA_SINGLE_R_TLV("ROUT Volume", WM8400_RIGHT_OUTPUT_VOLUME, + WM8400_ROUTVOL_SHIFT, WM8400_ROUTVOL_MASK, 0, out_pga_tlv), +SOC_SINGLE("ROUT ZC", WM8400_RIGHT_OUTPUT_VOLUME, WM8400_ROZC_SHIFT, 1, 0), + +/* LOPGA */ +WM8400_OUTPGA_SINGLE_R_TLV("LOPGA Volume", WM8400_LEFT_OPGA_VOLUME, + WM8400_LOPGAVOL_SHIFT, WM8400_LOPGAVOL_MASK, 0, out_pga_tlv), +SOC_SINGLE("LOPGA ZC Switch", WM8400_LEFT_OPGA_VOLUME, + WM8400_LOPGAZC_SHIFT, 1, 0), + +/* ROPGA */ +WM8400_OUTPGA_SINGLE_R_TLV("ROPGA Volume", WM8400_RIGHT_OPGA_VOLUME, + WM8400_ROPGAVOL_SHIFT, WM8400_ROPGAVOL_MASK, 0, out_pga_tlv), +SOC_SINGLE("ROPGA ZC Switch", WM8400_RIGHT_OPGA_VOLUME, + WM8400_ROPGAZC_SHIFT, 1, 0), + +SOC_SINGLE("LON Mute Switch", WM8400_LINE_OUTPUTS_VOLUME, + WM8400_LONMUTE_SHIFT, 1, 0), +SOC_SINGLE("LOP Mute Switch", WM8400_LINE_OUTPUTS_VOLUME, + WM8400_LOPMUTE_SHIFT, 1, 0), +SOC_SINGLE("LOP Attenuation Switch", WM8400_LINE_OUTPUTS_VOLUME, + WM8400_LOATTN_SHIFT, 1, 0), +SOC_SINGLE("RON Mute Switch", WM8400_LINE_OUTPUTS_VOLUME, + WM8400_RONMUTE_SHIFT, 1, 0), +SOC_SINGLE("ROP Mute Switch", WM8400_LINE_OUTPUTS_VOLUME, + WM8400_ROPMUTE_SHIFT, 1, 0), +SOC_SINGLE("ROP Attenuation Switch", WM8400_LINE_OUTPUTS_VOLUME, + WM8400_ROATTN_SHIFT, 1, 0), + +SOC_SINGLE("OUT3 Mute Switch", WM8400_OUT3_4_VOLUME, + WM8400_OUT3MUTE_SHIFT, 1, 0), +SOC_SINGLE("OUT3 Attenuation Switch", WM8400_OUT3_4_VOLUME, + WM8400_OUT3ATTN_SHIFT, 1, 0), + +SOC_SINGLE("OUT4 Mute Switch", WM8400_OUT3_4_VOLUME, + WM8400_OUT4MUTE_SHIFT, 1, 0), +SOC_SINGLE("OUT4 Attenuation Switch", WM8400_OUT3_4_VOLUME, + WM8400_OUT4ATTN_SHIFT, 1, 0), + +SOC_SINGLE("Speaker Mode Switch", WM8400_CLASSD1, + WM8400_CDMODE_SHIFT, 1, 0), + +SOC_SINGLE("Speaker Output Attenuation Volume", WM8400_SPEAKER_VOLUME, + WM8400_SPKATTN_SHIFT, WM8400_SPKATTN_MASK, 0), +SOC_SINGLE("Speaker DC Boost Volume", WM8400_CLASSD3, + WM8400_DCGAIN_SHIFT, 6, 0), +SOC_SINGLE("Speaker AC Boost Volume", WM8400_CLASSD3, + WM8400_ACGAIN_SHIFT, 6, 0), + +WM8400_OUTPGA_SINGLE_R_TLV("Left DAC Digital Volume", + WM8400_LEFT_DAC_DIGITAL_VOLUME, WM8400_DACL_VOL_SHIFT, + 127, 0, out_dac_tlv), + +WM8400_OUTPGA_SINGLE_R_TLV("Right DAC Digital Volume", + WM8400_RIGHT_DAC_DIGITAL_VOLUME, WM8400_DACR_VOL_SHIFT, + 127, 0, out_dac_tlv), + +SOC_ENUM("Left Digital Sidetone", wm8400_left_digital_sidetone_enum), +SOC_ENUM("Right Digital Sidetone", wm8400_right_digital_sidetone_enum), + +SOC_SINGLE_TLV("Left Digital Sidetone Volume", WM8400_DIGITAL_SIDE_TONE, + WM8400_ADCL_DAC_SVOL_SHIFT, 15, 0, out_sidetone_tlv), +SOC_SINGLE_TLV("Right Digital Sidetone Volume", WM8400_DIGITAL_SIDE_TONE, + WM8400_ADCR_DAC_SVOL_SHIFT, 15, 0, out_sidetone_tlv), + +SOC_SINGLE("ADC Digital High Pass Filter Switch", WM8400_ADC_CTRL, + WM8400_ADC_HPF_ENA_SHIFT, 1, 0), + +SOC_ENUM("ADC HPF Mode", wm8400_right_adcmode_enum), + +WM8400_OUTPGA_SINGLE_R_TLV("Left ADC Digital Volume", + WM8400_LEFT_ADC_DIGITAL_VOLUME, + WM8400_ADCL_VOL_SHIFT, + WM8400_ADCL_VOL_MASK, + 0, + in_adc_tlv), + +WM8400_OUTPGA_SINGLE_R_TLV("Right ADC Digital Volume", + WM8400_RIGHT_ADC_DIGITAL_VOLUME, + WM8400_ADCR_VOL_SHIFT, + WM8400_ADCR_VOL_MASK, + 0, + in_adc_tlv), + +WM8400_OUTPGA_SINGLE_R_TLV("LIN12 Volume", + WM8400_LEFT_LINE_INPUT_1_2_VOLUME, + WM8400_LIN12VOL_SHIFT, + WM8400_LIN12VOL_MASK, + 0, + in_pga_tlv), + +SOC_SINGLE("LIN12 ZC Switch", WM8400_LEFT_LINE_INPUT_1_2_VOLUME, + WM8400_LI12ZC_SHIFT, 1, 0), + +SOC_SINGLE("LIN12 Mute Switch", WM8400_LEFT_LINE_INPUT_1_2_VOLUME, + WM8400_LI12MUTE_SHIFT, 1, 0), + +WM8400_OUTPGA_SINGLE_R_TLV("LIN34 Volume", + WM8400_LEFT_LINE_INPUT_3_4_VOLUME, + WM8400_LIN34VOL_SHIFT, + WM8400_LIN34VOL_MASK, + 0, + in_pga_tlv), + +SOC_SINGLE("LIN34 ZC Switch", WM8400_LEFT_LINE_INPUT_3_4_VOLUME, + WM8400_LI34ZC_SHIFT, 1, 0), + +SOC_SINGLE("LIN34 Mute Switch", WM8400_LEFT_LINE_INPUT_3_4_VOLUME, + WM8400_LI34MUTE_SHIFT, 1, 0), + +WM8400_OUTPGA_SINGLE_R_TLV("RIN12 Volume", + WM8400_RIGHT_LINE_INPUT_1_2_VOLUME, + WM8400_RIN12VOL_SHIFT, + WM8400_RIN12VOL_MASK, + 0, + in_pga_tlv), + +SOC_SINGLE("RIN12 ZC Switch", WM8400_RIGHT_LINE_INPUT_1_2_VOLUME, + WM8400_RI12ZC_SHIFT, 1, 0), + +SOC_SINGLE("RIN12 Mute Switch", WM8400_RIGHT_LINE_INPUT_1_2_VOLUME, + WM8400_RI12MUTE_SHIFT, 1, 0), + +WM8400_OUTPGA_SINGLE_R_TLV("RIN34 Volume", + WM8400_RIGHT_LINE_INPUT_3_4_VOLUME, + WM8400_RIN34VOL_SHIFT, + WM8400_RIN34VOL_MASK, + 0, + in_pga_tlv), + +SOC_SINGLE("RIN34 ZC Switch", WM8400_RIGHT_LINE_INPUT_3_4_VOLUME, + WM8400_RI34ZC_SHIFT, 1, 0), + +SOC_SINGLE("RIN34 Mute Switch", WM8400_RIGHT_LINE_INPUT_3_4_VOLUME, + WM8400_RI34MUTE_SHIFT, 1, 0), + +}; + +/* add non dapm controls */ +static int wm8400_add_controls(struct snd_soc_codec *codec) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(wm8400_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&wm8400_snd_controls[i],codec, + NULL)); + if (err < 0) + return err; + } + return 0; +} + +/* + * _DAPM_ Controls + */ + +static int inmixer_event (struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + u16 reg, fakepower; + + reg = wm8400_read(w->codec, WM8400_POWER_MANAGEMENT_2); + fakepower = wm8400_read(w->codec, WM8400_INTDRIVBITS); + + if (fakepower & ((1 << WM8400_INMIXL_PWR) | + (1 << WM8400_AINLMUX_PWR))) { + reg |= WM8400_AINL_ENA; + } else { + reg &= ~WM8400_AINL_ENA; + } + + if (fakepower & ((1 << WM8400_INMIXR_PWR) | + (1 << WM8400_AINRMUX_PWR))) { + reg |= WM8400_AINR_ENA; + } else { + reg &= ~WM8400_AINL_ENA; + } + wm8400_write(w->codec, WM8400_POWER_MANAGEMENT_2, reg); + + return 0; +} + +static int outmixer_event (struct snd_soc_dapm_widget *w, + struct snd_kcontrol * kcontrol, int event) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + u32 reg_shift = mc->shift; + int ret = 0; + u16 reg; + + switch (reg_shift) { + case WM8400_SPEAKER_MIXER | (WM8400_LDSPK << 8) : + reg = wm8400_read(w->codec, WM8400_OUTPUT_MIXER1); + if (reg & WM8400_LDLO) { + printk(KERN_WARNING + "Cannot set as Output Mixer 1 LDLO Set\n"); + ret = -1; + } + break; + case WM8400_SPEAKER_MIXER | (WM8400_RDSPK << 8): + reg = wm8400_read(w->codec, WM8400_OUTPUT_MIXER2); + if (reg & WM8400_RDRO) { + printk(KERN_WARNING + "Cannot set as Output Mixer 2 RDRO Set\n"); + ret = -1; + } + break; + case WM8400_OUTPUT_MIXER1 | (WM8400_LDLO << 8): + reg = wm8400_read(w->codec, WM8400_SPEAKER_MIXER); + if (reg & WM8400_LDSPK) { + printk(KERN_WARNING + "Cannot set as Speaker Mixer LDSPK Set\n"); + ret = -1; + } + break; + case WM8400_OUTPUT_MIXER2 | (WM8400_RDRO << 8): + reg = wm8400_read(w->codec, WM8400_SPEAKER_MIXER); + if (reg & WM8400_RDSPK) { + printk(KERN_WARNING + "Cannot set as Speaker Mixer RDSPK Set\n"); + ret = -1; + } + break; + } + + return ret; +} + +/* INMIX dB values */ +static const unsigned int in_mix_tlv[] = { + TLV_DB_RANGE_HEAD(1), + 0,7, TLV_DB_LINEAR_ITEM(-1200, 600), +}; + +/* Left In PGA Connections */ +static const struct snd_kcontrol_new wm8400_dapm_lin12_pga_controls[] = { +SOC_DAPM_SINGLE("LIN1 Switch", WM8400_INPUT_MIXER2, WM8400_LMN1_SHIFT, 1, 0), +SOC_DAPM_SINGLE("LIN2 Switch", WM8400_INPUT_MIXER2, WM8400_LMP2_SHIFT, 1, 0), +}; + +static const struct snd_kcontrol_new wm8400_dapm_lin34_pga_controls[] = { +SOC_DAPM_SINGLE("LIN3 Switch", WM8400_INPUT_MIXER2, WM8400_LMN3_SHIFT, 1, 0), +SOC_DAPM_SINGLE("LIN4 Switch", WM8400_INPUT_MIXER2, WM8400_LMP4_SHIFT, 1, 0), +}; + +/* Right In PGA Connections */ +static const struct snd_kcontrol_new wm8400_dapm_rin12_pga_controls[] = { +SOC_DAPM_SINGLE("RIN1 Switch", WM8400_INPUT_MIXER2, WM8400_RMN1_SHIFT, 1, 0), +SOC_DAPM_SINGLE("RIN2 Switch", WM8400_INPUT_MIXER2, WM8400_RMP2_SHIFT, 1, 0), +}; + +static const struct snd_kcontrol_new wm8400_dapm_rin34_pga_controls[] = { +SOC_DAPM_SINGLE("RIN3 Switch", WM8400_INPUT_MIXER2, WM8400_RMN3_SHIFT, 1, 0), +SOC_DAPM_SINGLE("RIN4 Switch", WM8400_INPUT_MIXER2, WM8400_RMP4_SHIFT, 1, 0), +}; + +/* INMIXL */ +static const struct snd_kcontrol_new wm8400_dapm_inmixl_controls[] = { +SOC_DAPM_SINGLE_TLV("Record Left Volume", WM8400_INPUT_MIXER3, + WM8400_LDBVOL_SHIFT, WM8400_LDBVOL_MASK, 0, in_mix_tlv), +SOC_DAPM_SINGLE_TLV("LIN2 Volume", WM8400_INPUT_MIXER5, WM8400_LI2BVOL_SHIFT, + 7, 0, in_mix_tlv), +SOC_DAPM_SINGLE("LINPGA12 Switch", WM8400_INPUT_MIXER3, WM8400_L12MNB_SHIFT, + 1, 0), +SOC_DAPM_SINGLE("LINPGA34 Switch", WM8400_INPUT_MIXER3, WM8400_L34MNB_SHIFT, + 1, 0), +}; + +/* INMIXR */ +static const struct snd_kcontrol_new wm8400_dapm_inmixr_controls[] = { +SOC_DAPM_SINGLE_TLV("Record Right Volume", WM8400_INPUT_MIXER4, + WM8400_RDBVOL_SHIFT, WM8400_RDBVOL_MASK, 0, in_mix_tlv), +SOC_DAPM_SINGLE_TLV("RIN2 Volume", WM8400_INPUT_MIXER6, WM8400_RI2BVOL_SHIFT, + 7, 0, in_mix_tlv), +SOC_DAPM_SINGLE("RINPGA12 Switch", WM8400_INPUT_MIXER3, WM8400_L12MNB_SHIFT, + 1, 0), +SOC_DAPM_SINGLE("RINPGA34 Switch", WM8400_INPUT_MIXER3, WM8400_L34MNB_SHIFT, + 1, 0), +}; + +/* AINLMUX */ +static const char *wm8400_ainlmux[] = + {"INMIXL Mix", "RXVOICE Mix", "DIFFINL Mix"}; + +static const struct soc_enum wm8400_ainlmux_enum = +SOC_ENUM_SINGLE( WM8400_INPUT_MIXER1, WM8400_AINLMODE_SHIFT, + ARRAY_SIZE(wm8400_ainlmux), wm8400_ainlmux); + +static const struct snd_kcontrol_new wm8400_dapm_ainlmux_controls = +SOC_DAPM_ENUM("Route", wm8400_ainlmux_enum); + +/* DIFFINL */ + +/* AINRMUX */ +static const char *wm8400_ainrmux[] = + {"INMIXR Mix", "RXVOICE Mix", "DIFFINR Mix"}; + +static const struct soc_enum wm8400_ainrmux_enum = +SOC_ENUM_SINGLE( WM8400_INPUT_MIXER1, WM8400_AINRMODE_SHIFT, + ARRAY_SIZE(wm8400_ainrmux), wm8400_ainrmux); + +static const struct snd_kcontrol_new wm8400_dapm_ainrmux_controls = +SOC_DAPM_ENUM("Route", wm8400_ainrmux_enum); + +/* RXVOICE */ +static const struct snd_kcontrol_new wm8400_dapm_rxvoice_controls[] = { +SOC_DAPM_SINGLE_TLV("LIN4/RXN", WM8400_INPUT_MIXER5, WM8400_LR4BVOL_SHIFT, + WM8400_LR4BVOL_MASK, 0, in_mix_tlv), +SOC_DAPM_SINGLE_TLV("RIN4/RXP", WM8400_INPUT_MIXER6, WM8400_RL4BVOL_SHIFT, + WM8400_RL4BVOL_MASK, 0, in_mix_tlv), +}; + +/* LOMIX */ +static const struct snd_kcontrol_new wm8400_dapm_lomix_controls[] = { +SOC_DAPM_SINGLE("LOMIX Right ADC Bypass Switch", WM8400_OUTPUT_MIXER1, + WM8400_LRBLO_SHIFT, 1, 0), +SOC_DAPM_SINGLE("LOMIX Left ADC Bypass Switch", WM8400_OUTPUT_MIXER1, + WM8400_LLBLO_SHIFT, 1, 0), +SOC_DAPM_SINGLE("LOMIX RIN3 Bypass Switch", WM8400_OUTPUT_MIXER1, + WM8400_LRI3LO_SHIFT, 1, 0), +SOC_DAPM_SINGLE("LOMIX LIN3 Bypass Switch", WM8400_OUTPUT_MIXER1, + WM8400_LLI3LO_SHIFT, 1, 0), +SOC_DAPM_SINGLE("LOMIX RIN12 PGA Bypass Switch", WM8400_OUTPUT_MIXER1, + WM8400_LR12LO_SHIFT, 1, 0), +SOC_DAPM_SINGLE("LOMIX LIN12 PGA Bypass Switch", WM8400_OUTPUT_MIXER1, + WM8400_LL12LO_SHIFT, 1, 0), +SOC_DAPM_SINGLE("LOMIX Left DAC Switch", WM8400_OUTPUT_MIXER1, + WM8400_LDLO_SHIFT, 1, 0), +}; + +/* ROMIX */ +static const struct snd_kcontrol_new wm8400_dapm_romix_controls[] = { +SOC_DAPM_SINGLE("ROMIX Left ADC Bypass Switch", WM8400_OUTPUT_MIXER2, + WM8400_RLBRO_SHIFT, 1, 0), +SOC_DAPM_SINGLE("ROMIX Right ADC Bypass Switch", WM8400_OUTPUT_MIXER2, + WM8400_RRBRO_SHIFT, 1, 0), +SOC_DAPM_SINGLE("ROMIX LIN3 Bypass Switch", WM8400_OUTPUT_MIXER2, + WM8400_RLI3RO_SHIFT, 1, 0), +SOC_DAPM_SINGLE("ROMIX RIN3 Bypass Switch", WM8400_OUTPUT_MIXER2, + WM8400_RRI3RO_SHIFT, 1, 0), +SOC_DAPM_SINGLE("ROMIX LIN12 PGA Bypass Switch", WM8400_OUTPUT_MIXER2, + WM8400_RL12RO_SHIFT, 1, 0), +SOC_DAPM_SINGLE("ROMIX RIN12 PGA Bypass Switch", WM8400_OUTPUT_MIXER2, + WM8400_RR12RO_SHIFT, 1, 0), +SOC_DAPM_SINGLE("ROMIX Right DAC Switch", WM8400_OUTPUT_MIXER2, + WM8400_RDRO_SHIFT, 1, 0), +}; + +/* LONMIX */ +static const struct snd_kcontrol_new wm8400_dapm_lonmix_controls[] = { +SOC_DAPM_SINGLE("LONMIX Left Mixer PGA Switch", WM8400_LINE_MIXER1, + WM8400_LLOPGALON_SHIFT, 1, 0), +SOC_DAPM_SINGLE("LONMIX Right Mixer PGA Switch", WM8400_LINE_MIXER1, + WM8400_LROPGALON_SHIFT, 1, 0), +SOC_DAPM_SINGLE("LONMIX Inverted LOP Switch", WM8400_LINE_MIXER1, + WM8400_LOPLON_SHIFT, 1, 0), +}; + +/* LOPMIX */ +static const struct snd_kcontrol_new wm8400_dapm_lopmix_controls[] = { +SOC_DAPM_SINGLE("LOPMIX Right Mic Bypass Switch", WM8400_LINE_MIXER1, + WM8400_LR12LOP_SHIFT, 1, 0), +SOC_DAPM_SINGLE("LOPMIX Left Mic Bypass Switch", WM8400_LINE_MIXER1, + WM8400_LL12LOP_SHIFT, 1, 0), +SOC_DAPM_SINGLE("LOPMIX Left Mixer PGA Switch", WM8400_LINE_MIXER1, + WM8400_LLOPGALOP_SHIFT, 1, 0), +}; + +/* RONMIX */ +static const struct snd_kcontrol_new wm8400_dapm_ronmix_controls[] = { +SOC_DAPM_SINGLE("RONMIX Right Mixer PGA Switch", WM8400_LINE_MIXER2, + WM8400_RROPGARON_SHIFT, 1, 0), +SOC_DAPM_SINGLE("RONMIX Left Mixer PGA Switch", WM8400_LINE_MIXER2, + WM8400_RLOPGARON_SHIFT, 1, 0), +SOC_DAPM_SINGLE("RONMIX Inverted ROP Switch", WM8400_LINE_MIXER2, + WM8400_ROPRON_SHIFT, 1, 0), +}; + +/* ROPMIX */ +static const struct snd_kcontrol_new wm8400_dapm_ropmix_controls[] = { +SOC_DAPM_SINGLE("ROPMIX Left Mic Bypass Switch", WM8400_LINE_MIXER2, + WM8400_RL12ROP_SHIFT, 1, 0), +SOC_DAPM_SINGLE("ROPMIX Right Mic Bypass Switch", WM8400_LINE_MIXER2, + WM8400_RR12ROP_SHIFT, 1, 0), +SOC_DAPM_SINGLE("ROPMIX Right Mixer PGA Switch", WM8400_LINE_MIXER2, + WM8400_RROPGAROP_SHIFT, 1, 0), +}; + +/* OUT3MIX */ +static const struct snd_kcontrol_new wm8400_dapm_out3mix_controls[] = { +SOC_DAPM_SINGLE("OUT3MIX LIN4/RXP Bypass Switch", WM8400_OUT3_4_MIXER, + WM8400_LI4O3_SHIFT, 1, 0), +SOC_DAPM_SINGLE("OUT3MIX Left Out PGA Switch", WM8400_OUT3_4_MIXER, + WM8400_LPGAO3_SHIFT, 1, 0), +}; + +/* OUT4MIX */ +static const struct snd_kcontrol_new wm8400_dapm_out4mix_controls[] = { +SOC_DAPM_SINGLE("OUT4MIX Right Out PGA Switch", WM8400_OUT3_4_MIXER, + WM8400_RPGAO4_SHIFT, 1, 0), +SOC_DAPM_SINGLE("OUT4MIX RIN4/RXP Bypass Switch", WM8400_OUT3_4_MIXER, + WM8400_RI4O4_SHIFT, 1, 0), +}; + +/* SPKMIX */ +static const struct snd_kcontrol_new wm8400_dapm_spkmix_controls[] = { +SOC_DAPM_SINGLE("SPKMIX LIN2 Bypass Switch", WM8400_SPEAKER_MIXER, + WM8400_LI2SPK_SHIFT, 1, 0), +SOC_DAPM_SINGLE("SPKMIX LADC Bypass Switch", WM8400_SPEAKER_MIXER, + WM8400_LB2SPK_SHIFT, 1, 0), +SOC_DAPM_SINGLE("SPKMIX Left Mixer PGA Switch", WM8400_SPEAKER_MIXER, + WM8400_LOPGASPK_SHIFT, 1, 0), +SOC_DAPM_SINGLE("SPKMIX Left DAC Switch", WM8400_SPEAKER_MIXER, + WM8400_LDSPK_SHIFT, 1, 0), +SOC_DAPM_SINGLE("SPKMIX Right DAC Switch", WM8400_SPEAKER_MIXER, + WM8400_RDSPK_SHIFT, 1, 0), +SOC_DAPM_SINGLE("SPKMIX Right Mixer PGA Switch", WM8400_SPEAKER_MIXER, + WM8400_ROPGASPK_SHIFT, 1, 0), +SOC_DAPM_SINGLE("SPKMIX RADC Bypass Switch", WM8400_SPEAKER_MIXER, + WM8400_RL12ROP_SHIFT, 1, 0), +SOC_DAPM_SINGLE("SPKMIX RIN2 Bypass Switch", WM8400_SPEAKER_MIXER, + WM8400_RI2SPK_SHIFT, 1, 0), +}; + +static const struct snd_soc_dapm_widget wm8400_dapm_widgets[] = { +/* Input Side */ +/* Input Lines */ +SND_SOC_DAPM_INPUT("LIN1"), +SND_SOC_DAPM_INPUT("LIN2"), +SND_SOC_DAPM_INPUT("LIN3"), +SND_SOC_DAPM_INPUT("LIN4/RXN"), +SND_SOC_DAPM_INPUT("RIN3"), +SND_SOC_DAPM_INPUT("RIN4/RXP"), +SND_SOC_DAPM_INPUT("RIN1"), +SND_SOC_DAPM_INPUT("RIN2"), +SND_SOC_DAPM_INPUT("Internal ADC Source"), + +/* DACs */ +SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8400_POWER_MANAGEMENT_2, + WM8400_ADCL_ENA_SHIFT, 0), +SND_SOC_DAPM_ADC("Right ADC", "Right Capture", WM8400_POWER_MANAGEMENT_2, + WM8400_ADCR_ENA_SHIFT, 0), + +/* Input PGAs */ +SND_SOC_DAPM_MIXER("LIN12 PGA", WM8400_POWER_MANAGEMENT_2, + WM8400_LIN12_ENA_SHIFT, + 0, &wm8400_dapm_lin12_pga_controls[0], + ARRAY_SIZE(wm8400_dapm_lin12_pga_controls)), +SND_SOC_DAPM_MIXER("LIN34 PGA", WM8400_POWER_MANAGEMENT_2, + WM8400_LIN34_ENA_SHIFT, + 0, &wm8400_dapm_lin34_pga_controls[0], + ARRAY_SIZE(wm8400_dapm_lin34_pga_controls)), +SND_SOC_DAPM_MIXER("RIN12 PGA", WM8400_POWER_MANAGEMENT_2, + WM8400_RIN12_ENA_SHIFT, + 0, &wm8400_dapm_rin12_pga_controls[0], + ARRAY_SIZE(wm8400_dapm_rin12_pga_controls)), +SND_SOC_DAPM_MIXER("RIN34 PGA", WM8400_POWER_MANAGEMENT_2, + WM8400_RIN34_ENA_SHIFT, + 0, &wm8400_dapm_rin34_pga_controls[0], + ARRAY_SIZE(wm8400_dapm_rin34_pga_controls)), + +/* INMIXL */ +SND_SOC_DAPM_MIXER_E("INMIXL", WM8400_INTDRIVBITS, WM8400_INMIXL_PWR, 0, + &wm8400_dapm_inmixl_controls[0], + ARRAY_SIZE(wm8400_dapm_inmixl_controls), + inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + +/* AINLMUX */ +SND_SOC_DAPM_MUX_E("AILNMUX", WM8400_INTDRIVBITS, WM8400_AINLMUX_PWR, 0, + &wm8400_dapm_ainlmux_controls, inmixer_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + +/* INMIXR */ +SND_SOC_DAPM_MIXER_E("INMIXR", WM8400_INTDRIVBITS, WM8400_INMIXR_PWR, 0, + &wm8400_dapm_inmixr_controls[0], + ARRAY_SIZE(wm8400_dapm_inmixr_controls), + inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + +/* AINRMUX */ +SND_SOC_DAPM_MUX_E("AIRNMUX", WM8400_INTDRIVBITS, WM8400_AINRMUX_PWR, 0, + &wm8400_dapm_ainrmux_controls, inmixer_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + +/* Output Side */ +/* DACs */ +SND_SOC_DAPM_DAC("Left DAC", "Left Playback", WM8400_POWER_MANAGEMENT_3, + WM8400_DACL_ENA_SHIFT, 0), +SND_SOC_DAPM_DAC("Right DAC", "Right Playback", WM8400_POWER_MANAGEMENT_3, + WM8400_DACR_ENA_SHIFT, 0), + +/* LOMIX */ +SND_SOC_DAPM_MIXER_E("LOMIX", WM8400_POWER_MANAGEMENT_3, + WM8400_LOMIX_ENA_SHIFT, + 0, &wm8400_dapm_lomix_controls[0], + ARRAY_SIZE(wm8400_dapm_lomix_controls), + outmixer_event, SND_SOC_DAPM_PRE_REG), + +/* LONMIX */ +SND_SOC_DAPM_MIXER("LONMIX", WM8400_POWER_MANAGEMENT_3, WM8400_LON_ENA_SHIFT, + 0, &wm8400_dapm_lonmix_controls[0], + ARRAY_SIZE(wm8400_dapm_lonmix_controls)), + +/* LOPMIX */ +SND_SOC_DAPM_MIXER("LOPMIX", WM8400_POWER_MANAGEMENT_3, WM8400_LOP_ENA_SHIFT, + 0, &wm8400_dapm_lopmix_controls[0], + ARRAY_SIZE(wm8400_dapm_lopmix_controls)), + +/* OUT3MIX */ +SND_SOC_DAPM_MIXER("OUT3MIX", WM8400_POWER_MANAGEMENT_1, WM8400_OUT3_ENA_SHIFT, + 0, &wm8400_dapm_out3mix_controls[0], + ARRAY_SIZE(wm8400_dapm_out3mix_controls)), + +/* SPKMIX */ +SND_SOC_DAPM_MIXER_E("SPKMIX", WM8400_POWER_MANAGEMENT_1, WM8400_SPK_ENA_SHIFT, + 0, &wm8400_dapm_spkmix_controls[0], + ARRAY_SIZE(wm8400_dapm_spkmix_controls), outmixer_event, + SND_SOC_DAPM_PRE_REG), + +/* OUT4MIX */ +SND_SOC_DAPM_MIXER("OUT4MIX", WM8400_POWER_MANAGEMENT_1, WM8400_OUT4_ENA_SHIFT, + 0, &wm8400_dapm_out4mix_controls[0], + ARRAY_SIZE(wm8400_dapm_out4mix_controls)), + +/* ROPMIX */ +SND_SOC_DAPM_MIXER("ROPMIX", WM8400_POWER_MANAGEMENT_3, WM8400_ROP_ENA_SHIFT, + 0, &wm8400_dapm_ropmix_controls[0], + ARRAY_SIZE(wm8400_dapm_ropmix_controls)), + +/* RONMIX */ +SND_SOC_DAPM_MIXER("RONMIX", WM8400_POWER_MANAGEMENT_3, WM8400_RON_ENA_SHIFT, + 0, &wm8400_dapm_ronmix_controls[0], + ARRAY_SIZE(wm8400_dapm_ronmix_controls)), + +/* ROMIX */ +SND_SOC_DAPM_MIXER_E("ROMIX", WM8400_POWER_MANAGEMENT_3, + WM8400_ROMIX_ENA_SHIFT, + 0, &wm8400_dapm_romix_controls[0], + ARRAY_SIZE(wm8400_dapm_romix_controls), + outmixer_event, SND_SOC_DAPM_PRE_REG), + +/* LOUT PGA */ +SND_SOC_DAPM_PGA("LOUT PGA", WM8400_POWER_MANAGEMENT_1, WM8400_LOUT_ENA_SHIFT, + 0, NULL, 0), + +/* ROUT PGA */ +SND_SOC_DAPM_PGA("ROUT PGA", WM8400_POWER_MANAGEMENT_1, WM8400_ROUT_ENA_SHIFT, + 0, NULL, 0), + +/* LOPGA */ +SND_SOC_DAPM_PGA("LOPGA", WM8400_POWER_MANAGEMENT_3, WM8400_LOPGA_ENA_SHIFT, 0, + NULL, 0), + +/* ROPGA */ +SND_SOC_DAPM_PGA("ROPGA", WM8400_POWER_MANAGEMENT_3, WM8400_ROPGA_ENA_SHIFT, 0, + NULL, 0), + +/* MICBIAS */ +SND_SOC_DAPM_MICBIAS("MICBIAS", WM8400_POWER_MANAGEMENT_1, + WM8400_MIC1BIAS_ENA_SHIFT, 0), + +SND_SOC_DAPM_OUTPUT("LON"), +SND_SOC_DAPM_OUTPUT("LOP"), +SND_SOC_DAPM_OUTPUT("OUT3"), +SND_SOC_DAPM_OUTPUT("LOUT"), +SND_SOC_DAPM_OUTPUT("SPKN"), +SND_SOC_DAPM_OUTPUT("SPKP"), +SND_SOC_DAPM_OUTPUT("ROUT"), +SND_SOC_DAPM_OUTPUT("OUT4"), +SND_SOC_DAPM_OUTPUT("ROP"), +SND_SOC_DAPM_OUTPUT("RON"), + +SND_SOC_DAPM_OUTPUT("Internal DAC Sink"), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* Make DACs turn on when playing even if not mixed into any outputs */ + {"Internal DAC Sink", NULL, "Left DAC"}, + {"Internal DAC Sink", NULL, "Right DAC"}, + + /* Make ADCs turn on when recording + * even if not mixed from any inputs */ + {"Left ADC", NULL, "Internal ADC Source"}, + {"Right ADC", NULL, "Internal ADC Source"}, + + /* Input Side */ + /* LIN12 PGA */ + {"LIN12 PGA", "LIN1 Switch", "LIN1"}, + {"LIN12 PGA", "LIN2 Switch", "LIN2"}, + /* LIN34 PGA */ + {"LIN34 PGA", "LIN3 Switch", "LIN3"}, + {"LIN34 PGA", "LIN4 Switch", "LIN4/RXN"}, + /* INMIXL */ + {"INMIXL", "Record Left Volume", "LOMIX"}, + {"INMIXL", "LIN2 Volume", "LIN2"}, + {"INMIXL", "LINPGA12 Switch", "LIN12 PGA"}, + {"INMIXL", "LINPGA34 Switch", "LIN34 PGA"}, + /* AILNMUX */ + {"AILNMUX", "INMIXL Mix", "INMIXL"}, + {"AILNMUX", "DIFFINL Mix", "LIN12 PGA"}, + {"AILNMUX", "DIFFINL Mix", "LIN34 PGA"}, + {"AILNMUX", "RXVOICE Mix", "LIN4/RXN"}, + {"AILNMUX", "RXVOICE Mix", "RIN4/RXP"}, + /* ADC */ + {"Left ADC", NULL, "AILNMUX"}, + + /* RIN12 PGA */ + {"RIN12 PGA", "RIN1 Switch", "RIN1"}, + {"RIN12 PGA", "RIN2 Switch", "RIN2"}, + /* RIN34 PGA */ + {"RIN34 PGA", "RIN3 Switch", "RIN3"}, + {"RIN34 PGA", "RIN4 Switch", "RIN4/RXP"}, + /* INMIXL */ + {"INMIXR", "Record Right Volume", "ROMIX"}, + {"INMIXR", "RIN2 Volume", "RIN2"}, + {"INMIXR", "RINPGA12 Switch", "RIN12 PGA"}, + {"INMIXR", "RINPGA34 Switch", "RIN34 PGA"}, + /* AIRNMUX */ + {"AIRNMUX", "INMIXR Mix", "INMIXR"}, + {"AIRNMUX", "DIFFINR Mix", "RIN12 PGA"}, + {"AIRNMUX", "DIFFINR Mix", "RIN34 PGA"}, + {"AIRNMUX", "RXVOICE Mix", "LIN4/RXN"}, + {"AIRNMUX", "RXVOICE Mix", "RIN4/RXP"}, + /* ADC */ + {"Right ADC", NULL, "AIRNMUX"}, + + /* LOMIX */ + {"LOMIX", "LOMIX RIN3 Bypass Switch", "RIN3"}, + {"LOMIX", "LOMIX LIN3 Bypass Switch", "LIN3"}, + {"LOMIX", "LOMIX LIN12 PGA Bypass Switch", "LIN12 PGA"}, + {"LOMIX", "LOMIX RIN12 PGA Bypass Switch", "RIN12 PGA"}, + {"LOMIX", "LOMIX Right ADC Bypass Switch", "AIRNMUX"}, + {"LOMIX", "LOMIX Left ADC Bypass Switch", "AILNMUX"}, + {"LOMIX", "LOMIX Left DAC Switch", "Left DAC"}, + + /* ROMIX */ + {"ROMIX", "ROMIX RIN3 Bypass Switch", "RIN3"}, + {"ROMIX", "ROMIX LIN3 Bypass Switch", "LIN3"}, + {"ROMIX", "ROMIX LIN12 PGA Bypass Switch", "LIN12 PGA"}, + {"ROMIX", "ROMIX RIN12 PGA Bypass Switch", "RIN12 PGA"}, + {"ROMIX", "ROMIX Right ADC Bypass Switch", "AIRNMUX"}, + {"ROMIX", "ROMIX Left ADC Bypass Switch", "AILNMUX"}, + {"ROMIX", "ROMIX Right DAC Switch", "Right DAC"}, + + /* SPKMIX */ + {"SPKMIX", "SPKMIX LIN2 Bypass Switch", "LIN2"}, + {"SPKMIX", "SPKMIX RIN2 Bypass Switch", "RIN2"}, + {"SPKMIX", "SPKMIX LADC Bypass Switch", "AILNMUX"}, + {"SPKMIX", "SPKMIX RADC Bypass Switch", "AIRNMUX"}, + {"SPKMIX", "SPKMIX Left Mixer PGA Switch", "LOPGA"}, + {"SPKMIX", "SPKMIX Right Mixer PGA Switch", "ROPGA"}, + {"SPKMIX", "SPKMIX Right DAC Switch", "Right DAC"}, + {"SPKMIX", "SPKMIX Left DAC Switch", "Right DAC"}, + + /* LONMIX */ + {"LONMIX", "LONMIX Left Mixer PGA Switch", "LOPGA"}, + {"LONMIX", "LONMIX Right Mixer PGA Switch", "ROPGA"}, + {"LONMIX", "LONMIX Inverted LOP Switch", "LOPMIX"}, + + /* LOPMIX */ + {"LOPMIX", "LOPMIX Right Mic Bypass Switch", "RIN12 PGA"}, + {"LOPMIX", "LOPMIX Left Mic Bypass Switch", "LIN12 PGA"}, + {"LOPMIX", "LOPMIX Left Mixer PGA Switch", "LOPGA"}, + + /* OUT3MIX */ + {"OUT3MIX", "OUT3MIX LIN4/RXP Bypass Switch", "LIN4/RXN"}, + {"OUT3MIX", "OUT3MIX Left Out PGA Switch", "LOPGA"}, + + /* OUT4MIX */ + {"OUT4MIX", "OUT4MIX Right Out PGA Switch", "ROPGA"}, + {"OUT4MIX", "OUT4MIX RIN4/RXP Bypass Switch", "RIN4/RXP"}, + + /* RONMIX */ + {"RONMIX", "RONMIX Right Mixer PGA Switch", "ROPGA"}, + {"RONMIX", "RONMIX Left Mixer PGA Switch", "LOPGA"}, + {"RONMIX", "RONMIX Inverted ROP Switch", "ROPMIX"}, + + /* ROPMIX */ + {"ROPMIX", "ROPMIX Left Mic Bypass Switch", "LIN12 PGA"}, + {"ROPMIX", "ROPMIX Right Mic Bypass Switch", "RIN12 PGA"}, + {"ROPMIX", "ROPMIX Right Mixer PGA Switch", "ROPGA"}, + + /* Out Mixer PGAs */ + {"LOPGA", NULL, "LOMIX"}, + {"ROPGA", NULL, "ROMIX"}, + + {"LOUT PGA", NULL, "LOMIX"}, + {"ROUT PGA", NULL, "ROMIX"}, + + /* Output Pins */ + {"LON", NULL, "LONMIX"}, + {"LOP", NULL, "LOPMIX"}, + {"OUT3", NULL, "OUT3MIX"}, + {"LOUT", NULL, "LOUT PGA"}, + {"SPKN", NULL, "SPKMIX"}, + {"ROUT", NULL, "ROUT PGA"}, + {"OUT4", NULL, "OUT4MIX"}, + {"ROP", NULL, "ROPMIX"}, + {"RON", NULL, "RONMIX"}, +}; + +static int wm8400_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, wm8400_dapm_widgets, + ARRAY_SIZE(wm8400_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +/* + * Clock after FLL and dividers + */ +static int wm8400_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8400_priv *wm8400 = codec->private_data; + + wm8400->sysclk = freq; + return 0; +} + +/* + * Sets ADC and Voice DAC format. + */ +static int wm8400_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 audio1, audio3; + + audio1 = wm8400_read(codec, WM8400_AUDIO_INTERFACE_1); + audio3 = wm8400_read(codec, WM8400_AUDIO_INTERFACE_3); + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + audio3 &= ~WM8400_AIF_MSTR1; + break; + case SND_SOC_DAIFMT_CBM_CFM: + audio3 |= WM8400_AIF_MSTR1; + break; + default: + return -EINVAL; + } + + audio1 &= ~WM8400_AIF_FMT_MASK; + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + audio1 |= WM8400_AIF_FMT_I2S; + audio1 &= ~WM8400_AIF_LRCLK_INV; + break; + case SND_SOC_DAIFMT_RIGHT_J: + audio1 |= WM8400_AIF_FMT_RIGHTJ; + audio1 &= ~WM8400_AIF_LRCLK_INV; + break; + case SND_SOC_DAIFMT_LEFT_J: + audio1 |= WM8400_AIF_FMT_LEFTJ; + audio1 &= ~WM8400_AIF_LRCLK_INV; + break; + case SND_SOC_DAIFMT_DSP_A: + audio1 |= WM8400_AIF_FMT_DSP; + audio1 &= ~WM8400_AIF_LRCLK_INV; + break; + case SND_SOC_DAIFMT_DSP_B: + audio1 |= WM8400_AIF_FMT_DSP | WM8400_AIF_LRCLK_INV; + break; + default: + return -EINVAL; + } + + wm8400_write(codec, WM8400_AUDIO_INTERFACE_1, audio1); + wm8400_write(codec, WM8400_AUDIO_INTERFACE_3, audio3); + return 0; +} + +static int wm8400_set_dai_clkdiv(struct snd_soc_dai *codec_dai, + int div_id, int div) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 reg; + + switch (div_id) { + case WM8400_MCLK_DIV: + reg = wm8400_read(codec, WM8400_CLOCKING_2) & + ~WM8400_MCLK_DIV_MASK; + wm8400_write(codec, WM8400_CLOCKING_2, reg | div); + break; + case WM8400_DACCLK_DIV: + reg = wm8400_read(codec, WM8400_CLOCKING_2) & + ~WM8400_DAC_CLKDIV_MASK; + wm8400_write(codec, WM8400_CLOCKING_2, reg | div); + break; + case WM8400_ADCCLK_DIV: + reg = wm8400_read(codec, WM8400_CLOCKING_2) & + ~WM8400_ADC_CLKDIV_MASK; + wm8400_write(codec, WM8400_CLOCKING_2, reg | div); + break; + case WM8400_BCLK_DIV: + reg = wm8400_read(codec, WM8400_CLOCKING_1) & + ~WM8400_BCLK_DIV_MASK; + wm8400_write(codec, WM8400_CLOCKING_1, reg | div); + break; + default: + return -EINVAL; + } + + return 0; +} + +/* + * Set PCM DAI bit size and sample rate. + */ +static int wm8400_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + u16 audio1 = wm8400_read(codec, WM8400_AUDIO_INTERFACE_1); + + audio1 &= ~WM8400_AIF_WL_MASK; + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + audio1 |= WM8400_AIF_WL_20BITS; + break; + case SNDRV_PCM_FORMAT_S24_LE: + audio1 |= WM8400_AIF_WL_24BITS; + break; + case SNDRV_PCM_FORMAT_S32_LE: + audio1 |= WM8400_AIF_WL_32BITS; + break; + } + + wm8400_write(codec, WM8400_AUDIO_INTERFACE_1, audio1); + return 0; +} + +static int wm8400_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 val = wm8400_read(codec, WM8400_DAC_CTRL) & ~WM8400_DAC_MUTE; + + if (mute) + wm8400_write(codec, WM8400_DAC_CTRL, val | WM8400_DAC_MUTE); + else + wm8400_write(codec, WM8400_DAC_CTRL, val); + + return 0; +} + +/* TODO: set bias for best performance at standby */ +static int wm8400_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct wm8400_priv *wm8400 = codec->private_data; + u16 val; + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + /* VMID=2*50k */ + val = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1) & + ~WM8400_VMID_MODE_MASK; + wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val | 0x2); + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->bias_level == SND_SOC_BIAS_OFF) { + ret = regulator_bulk_enable(ARRAY_SIZE(power), + &power[0]); + if (ret != 0) { + dev_err(wm8400->wm8400->dev, + "Failed to enable regulators: %d\n", + ret); + return ret; + } + + wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, + WM8400_CODEC_ENA | WM8400_SYSCLK_ENA); + + /* Enable all output discharge bits */ + wm8400_write(codec, WM8400_ANTIPOP1, WM8400_DIS_LLINE | + WM8400_DIS_RLINE | WM8400_DIS_OUT3 | + WM8400_DIS_OUT4 | WM8400_DIS_LOUT | + WM8400_DIS_ROUT); + + /* Enable POBCTRL, SOFT_ST, VMIDTOG and BUFDCOPEN */ + wm8400_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST | + WM8400_BUFDCOPEN | WM8400_POBCTRL); + + msleep(500); + + /* Enable outputs */ + val = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1); + val |= WM8400_SPK_ENA | WM8400_OUT3_ENA | + WM8400_OUT4_ENA | WM8400_LOUT_ENA | + WM8400_ROUT_ENA; + wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val); + + /* disable all output discharge bits */ + wm8400_write(codec, WM8400_ANTIPOP1, 0); + + /* Enable VREF & VMID at 2x50k */ + val |= 0x2 | WM8400_VREF_ENA; + wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val); + + msleep(600); + + /* Enable BUFIOEN */ + wm8400_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST | + WM8400_BUFDCOPEN | WM8400_POBCTRL | + WM8400_BUFIOEN); + + /* Disable outputs */ + val &= ~(WM8400_SPK_ENA | WM8400_OUT3_ENA | + WM8400_OUT4_ENA | WM8400_LOUT_ENA | + WM8400_ROUT_ENA); + wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val); + + /* disable POBCTRL, SOFT_ST and BUFDCOPEN */ + wm8400_write(codec, WM8400_ANTIPOP2, WM8400_BUFIOEN); + } + + /* VMID=2*300k */ + val = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1) & + ~WM8400_VMID_MODE_MASK; + wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val | 0x4); + break; + + case SND_SOC_BIAS_OFF: + /* Enable POBCTRL and SOFT_ST */ + wm8400_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST | + WM8400_POBCTRL | WM8400_BUFIOEN); + + /* Enable POBCTRL, SOFT_ST and BUFDCOPEN */ + wm8400_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST | + WM8400_BUFDCOPEN | WM8400_POBCTRL | + WM8400_BUFIOEN); + + /* mute DAC */ + val = wm8400_read(codec, WM8400_DAC_CTRL); + wm8400_write(codec, WM8400_DAC_CTRL, val | WM8400_DAC_MUTE); + + /* Enable any disabled outputs */ + val = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1); + val |= WM8400_SPK_ENA | WM8400_OUT3_ENA | + WM8400_OUT4_ENA | WM8400_LOUT_ENA | + WM8400_ROUT_ENA; + wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val); + + /* Disable VMID */ + val &= ~WM8400_VMID_MODE_MASK; + wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val); + + msleep(300); + + /* Enable all output discharge bits */ + wm8400_write(codec, WM8400_ANTIPOP1, WM8400_DIS_LLINE | + WM8400_DIS_RLINE | WM8400_DIS_OUT3 | + WM8400_DIS_OUT4 | WM8400_DIS_LOUT | + WM8400_DIS_ROUT); + + /* Disable VREF */ + val &= ~WM8400_VREF_ENA; + wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val); + + /* disable POBCTRL, SOFT_ST and BUFDCOPEN */ + wm8400_write(codec, WM8400_ANTIPOP2, 0x0); + + ret = regulator_bulk_disable(ARRAY_SIZE(power), + &power[0]); + if (ret != 0) + return ret; + + break; + } + + codec->bias_level = level; + return 0; +} + +#define WM8400_RATES SNDRV_PCM_RATE_8000_96000 + +#define WM8400_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +/* + * The WM8400 supports 2 different and mutually exclusive DAI + * configurations. + * + * 1. ADC/DAC on Primary Interface + * 2. ADC on Primary Interface/DAC on secondary + */ +struct snd_soc_dai wm8400_dai = { +/* ADC/DAC on primary */ + .name = "WM8400 ADC/DAC Primary", + .id = 1, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM8400_RATES, + .formats = WM8400_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM8400_RATES, + .formats = WM8400_FORMATS, + }, + .ops = { + .hw_params = wm8400_hw_params, + .digital_mute = wm8400_mute, + .set_fmt = wm8400_set_dai_fmt, + .set_clkdiv = wm8400_set_dai_clkdiv, + .set_sysclk = wm8400_set_dai_sysclk, + }, +}; +EXPORT_SYMBOL_GPL(wm8400_dai); + +static int wm8400_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + wm8400_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int wm8400_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + wm8400_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} + +static struct snd_soc_codec *wm8400_codec; + +static int wm8400_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret; + + if (!wm8400_codec) { + dev_err(&pdev->dev, "wm8400 not yet discovered\n"); + return -ENODEV; + } + codec = wm8400_codec; + + socdev->card->codec = codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(&pdev->dev, "failed to create pcms\n"); + goto pcm_err; + } + + wm8400_add_controls(codec); + wm8400_add_widgets(codec); + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(&pdev->dev, "failed to register card\n"); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + return ret; +} + +/* power down chip */ +static int wm8400_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8400 = { + .probe = wm8400_probe, + .remove = wm8400_remove, + .suspend = wm8400_suspend, + .resume = wm8400_resume, +}; + +static void wm8400_probe_deferred(struct work_struct *work) +{ + struct wm8400_priv *priv = container_of(work, struct wm8400_priv, + work); + struct snd_soc_codec *codec = &priv->codec; + int ret; + + /* charge output caps */ + wm8400_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + /* We're done, tell the subsystem. */ + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(priv->wm8400->dev, + "Failed to register codec: %d\n", ret); + goto err; + } + + ret = snd_soc_register_dai(&wm8400_dai); + if (ret != 0) { + dev_err(priv->wm8400->dev, + "Failed to register DAI: %d\n", ret); + goto err_codec; + } + + return; + +err_codec: + snd_soc_unregister_codec(codec); +err: + wm8400_set_bias_level(codec, SND_SOC_BIAS_OFF); +} + +static int wm8400_codec_probe(struct platform_device *dev) +{ + struct wm8400_priv *priv; + int ret; + u16 reg; + struct snd_soc_codec *codec; + + priv = kzalloc(sizeof(struct wm8400_priv), GFP_KERNEL); + if (priv == NULL) + return -ENOMEM; + + codec = &priv->codec; + codec->private_data = priv; + codec->control_data = dev->dev.driver_data; + priv->wm8400 = dev->dev.driver_data; + + ret = regulator_bulk_get(priv->wm8400->dev, + ARRAY_SIZE(power), &power[0]); + if (ret != 0) { + dev_err(&dev->dev, "Failed to get regulators: %d\n", ret); + goto err; + } + + codec->dev = &dev->dev; + wm8400_dai.dev = &dev->dev; + + codec->name = "WM8400"; + codec->owner = THIS_MODULE; + codec->read = wm8400_read; + codec->write = wm8400_write; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = wm8400_set_bias_level; + codec->dai = &wm8400_dai; + codec->num_dai = 1; + codec->reg_cache_size = WM8400_REGISTER_COUNT; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + INIT_WORK(&priv->work, wm8400_probe_deferred); + + wm8400_codec_reset(codec); + + reg = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1); + wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, reg | WM8400_CODEC_ENA); + + /* Latch volume update bits */ + reg = wm8400_read(codec, WM8400_LEFT_LINE_INPUT_1_2_VOLUME); + wm8400_write(codec, WM8400_LEFT_LINE_INPUT_1_2_VOLUME, + reg & WM8400_IPVU); + reg = wm8400_read(codec, WM8400_RIGHT_LINE_INPUT_1_2_VOLUME); + wm8400_write(codec, WM8400_RIGHT_LINE_INPUT_1_2_VOLUME, + reg & WM8400_IPVU); + + wm8400_write(codec, WM8400_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8)); + wm8400_write(codec, WM8400_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8)); + + wm8400_codec = codec; + + if (!schedule_work(&priv->work)) { + ret = -EINVAL; + goto err_regulator; + } + + return 0; + +err_regulator: + wm8400_codec = NULL; + regulator_bulk_free(ARRAY_SIZE(power), power); +err: + kfree(priv); + return ret; +} + +static int __exit wm8400_codec_remove(struct platform_device *dev) +{ + struct wm8400_priv *priv = wm8400_codec->private_data; + u16 reg; + + snd_soc_unregister_dai(&wm8400_dai); + snd_soc_unregister_codec(wm8400_codec); + + reg = wm8400_read(wm8400_codec, WM8400_POWER_MANAGEMENT_1); + wm8400_write(wm8400_codec, WM8400_POWER_MANAGEMENT_1, + reg & (~WM8400_CODEC_ENA)); + + regulator_bulk_free(ARRAY_SIZE(power), power); + kfree(priv); + + wm8400_codec = NULL; + + return 0; +} + +static struct platform_driver wm8400_codec_driver = { + .driver = { + .name = "wm8400-codec", + .owner = THIS_MODULE, + }, + .probe = wm8400_codec_probe, + .remove = __exit_p(wm8400_codec_remove), +}; + +static int __init wm8400_codec_init(void) +{ + return platform_driver_register(&wm8400_codec_driver); +} +module_init(wm8400_codec_init); + +static void __exit wm8400_codec_exit(void) +{ + platform_driver_unregister(&wm8400_codec_driver); +} +module_exit(wm8400_codec_exit); + +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8400); + +MODULE_DESCRIPTION("ASoC WM8400 driver"); +MODULE_AUTHOR("Mark Brown"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:wm8400-codec"); diff --git a/sound/soc/codecs/wm8400.h b/sound/soc/codecs/wm8400.h new file mode 100644 index 00000000000..79c5934d477 --- /dev/null +++ b/sound/soc/codecs/wm8400.h @@ -0,0 +1,62 @@ +/* + * wm8400.h -- audio driver for WM8400 + * + * Copyright 2008 Wolfson Microelectronics PLC. + * Author: Mark Brown + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#ifndef _WM8400_CODEC_H +#define _WM8400_CODEC_H + +#define WM8400_MCLK_DIV 0 +#define WM8400_DACCLK_DIV 1 +#define WM8400_ADCCLK_DIV 2 +#define WM8400_BCLK_DIV 3 + +#define WM8400_MCLK_DIV_1 0x400 +#define WM8400_MCLK_DIV_2 0x800 + +#define WM8400_DAC_CLKDIV_1 0x00 +#define WM8400_DAC_CLKDIV_1_5 0x04 +#define WM8400_DAC_CLKDIV_2 0x08 +#define WM8400_DAC_CLKDIV_3 0x0c +#define WM8400_DAC_CLKDIV_4 0x10 +#define WM8400_DAC_CLKDIV_5_5 0x14 +#define WM8400_DAC_CLKDIV_6 0x18 + +#define WM8400_ADC_CLKDIV_1 0x00 +#define WM8400_ADC_CLKDIV_1_5 0x20 +#define WM8400_ADC_CLKDIV_2 0x40 +#define WM8400_ADC_CLKDIV_3 0x60 +#define WM8400_ADC_CLKDIV_4 0x80 +#define WM8400_ADC_CLKDIV_5_5 0xa0 +#define WM8400_ADC_CLKDIV_6 0xc0 + + +#define WM8400_BCLK_DIV_1 (0x0 << 1) +#define WM8400_BCLK_DIV_1_5 (0x1 << 1) +#define WM8400_BCLK_DIV_2 (0x2 << 1) +#define WM8400_BCLK_DIV_3 (0x3 << 1) +#define WM8400_BCLK_DIV_4 (0x4 << 1) +#define WM8400_BCLK_DIV_5_5 (0x5 << 1) +#define WM8400_BCLK_DIV_6 (0x6 << 1) +#define WM8400_BCLK_DIV_8 (0x7 << 1) +#define WM8400_BCLK_DIV_11 (0x8 << 1) +#define WM8400_BCLK_DIV_12 (0x9 << 1) +#define WM8400_BCLK_DIV_16 (0xA << 1) +#define WM8400_BCLK_DIV_22 (0xB << 1) +#define WM8400_BCLK_DIV_24 (0xC << 1) +#define WM8400_BCLK_DIV_32 (0xD << 1) +#define WM8400_BCLK_DIV_44 (0xE << 1) +#define WM8400_BCLK_DIV_48 (0xF << 1) + +extern struct snd_soc_dai wm8400_dai; +extern struct snd_soc_codec_device soc_codec_dev_wm8400; + +#endif -- cgit v1.2.3-70-g09d2 From 02b7cbc3994622900e8fc201f5f229b591c43628 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 11 Mar 2009 14:12:28 +0000 Subject: ASoC: Remove version display from WM8580 driver Signed-off-by: Mark Brown --- sound/soc/codecs/wm8580.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index d3c51ba5e6f..6cab82a9c9d 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -35,8 +35,6 @@ #include "wm8580.h" -#define WM8580_VERSION "0.1" - struct pll_state { unsigned int in; unsigned int out; @@ -972,8 +970,6 @@ static int wm8580_probe(struct platform_device *pdev) struct wm8580_priv *wm8580; int ret = 0; - pr_info("WM8580 Audio Codec %s\n", WM8580_VERSION); - setup = socdev->codec_data; codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); if (codec == NULL) -- cgit v1.2.3-70-g09d2 From 5314adc3612d893c7cc526b3312d124805e45bc3 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 11 Mar 2009 16:28:29 +0000 Subject: ASoC: Fix formats for s3c24xx-i2s register prints The register values are all u32 so don't need the long format. Signed-off-by: Mark Brown --- sound/soc/s3c24xx/s3c24xx-i2s.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 580cfed71cc..407ccd7180f 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -83,7 +83,7 @@ static void s3c24xx_snd_txctrl(int on) iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); - pr_debug("r: IISCON: %lx IISMOD: %lx IISFCON: %lx\n", iiscon, iismod, iisfcon); + pr_debug("r: IISCON: %x IISMOD: %x IISFCON: %x\n", iiscon, iismod, iisfcon); if (on) { iisfcon |= S3C2410_IISFCON_TXDMA | S3C2410_IISFCON_TXENABLE; @@ -113,7 +113,7 @@ static void s3c24xx_snd_txctrl(int on) writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); } - pr_debug("w: IISCON: %lx IISMOD: %lx IISFCON: %lx\n", iiscon, iismod, iisfcon); + pr_debug("w: IISCON: %x IISMOD: %x IISFCON: %x\n", iiscon, iismod, iisfcon); } static void s3c24xx_snd_rxctrl(int on) @@ -128,7 +128,7 @@ static void s3c24xx_snd_rxctrl(int on) iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); - pr_debug("r: IISCON: %lx IISMOD: %lx IISFCON: %lx\n", iiscon, iismod, iisfcon); + pr_debug("r: IISCON: %x IISMOD: %x IISFCON: %x\n", iiscon, iismod, iisfcon); if (on) { iisfcon |= S3C2410_IISFCON_RXDMA | S3C2410_IISFCON_RXENABLE; @@ -158,7 +158,7 @@ static void s3c24xx_snd_rxctrl(int on) writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); } - pr_debug("w: IISCON: %lx IISMOD: %lx IISFCON: %lx\n", iiscon, iismod, iisfcon); + pr_debug("w: IISCON: %x IISMOD: %x IISFCON: %x\n", iiscon, iismod, iisfcon); } /* @@ -206,7 +206,7 @@ static int s3c24xx_i2s_set_fmt(struct snd_soc_dai *cpu_dai, pr_debug("Entered %s\n", __func__); iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); - pr_debug("hw_params r: IISMOD: %lx \n", iismod); + pr_debug("hw_params r: IISMOD: %x \n", iismod); switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: @@ -231,7 +231,7 @@ static int s3c24xx_i2s_set_fmt(struct snd_soc_dai *cpu_dai, } writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); - pr_debug("hw_params w: IISMOD: %lx \n", iismod); + pr_debug("hw_params w: IISMOD: %x \n", iismod); return 0; } @@ -251,7 +251,7 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, /* Working copies of register */ iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); - pr_debug("hw_params r: IISMOD: %lx\n", iismod); + pr_debug("hw_params r: IISMOD: %x\n", iismod); switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: @@ -269,7 +269,7 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, } writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); - pr_debug("hw_params w: IISMOD: %lx\n", iismod); + pr_debug("hw_params w: IISMOD: %x\n", iismod); return 0; } -- cgit v1.2.3-70-g09d2 From 603b6fd5b8d313a109d3739d8706ee51962ff402 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 11 Mar 2009 18:28:24 +0000 Subject: [ARM] Revert futher extraneous changes from the S3C header move Can't see any immediate need for these; build tested. Signed-off-by: Mark Brown --- arch/arm/mach-s3c2410/include/mach/io.h | 2 +- arch/arm/plat-s3c24xx/clock-dclk.c | 1 - 2 files changed, 1 insertion(+), 2 deletions(-) diff --git a/arch/arm/mach-s3c2410/include/mach/io.h b/arch/arm/mach-s3c2410/include/mach/io.h index c477771c092..9813dbf2ae4 100644 --- a/arch/arm/mach-s3c2410/include/mach/io.h +++ b/arch/arm/mach-s3c2410/include/mach/io.h @@ -9,7 +9,7 @@ #ifndef __ASM_ARM_ARCH_IO_H #define __ASM_ARM_ARCH_IO_H -#include +#include #define IO_SPACE_LIMIT 0xffffffff diff --git a/arch/arm/plat-s3c24xx/clock-dclk.c b/arch/arm/plat-s3c24xx/clock-dclk.c index 35219dcf9f0..5b75a797b5a 100644 --- a/arch/arm/plat-s3c24xx/clock-dclk.c +++ b/arch/arm/plat-s3c24xx/clock-dclk.c @@ -18,7 +18,6 @@ #include #include -#include #include #include -- cgit v1.2.3-70-g09d2 From 5d75bc557859805f00eeddb09d7cc8ffc7e5334e Mon Sep 17 00:00:00 2001 From: Gregorio Guidi Date: Thu, 12 Mar 2009 16:41:51 +0100 Subject: ALSA: hda - fix headphone settings and master volume (Conexant CX20551) Update the places where the 0x1d widget is used for Conexant 5047, fixing mismatch introduced after changing the connection. Signed-off-by: Gregorio Guidi Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index d60ccb5bb12..6cb184e9c2f 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1196,7 +1196,7 @@ static int cxt5047_hp_master_sw_put(struct snd_kcontrol *kcontrol, * the headphone jack */ bits = (!spec->hp_present && spec->cur_eapd) ? 0 : HDA_AMP_MUTE; - snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0, + snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0x01, HDA_AMP_MUTE, bits); bits = spec->cur_eapd ? 0 : HDA_AMP_MUTE; snd_hda_codec_amp_stereo(codec, 0x13, HDA_OUTPUT, 0, @@ -1214,7 +1214,7 @@ static void cxt5047_hp_automute(struct hda_codec *codec) AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; bits = (spec->hp_present || !spec->cur_eapd) ? HDA_AMP_MUTE : 0; - snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0, + snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0x01, HDA_AMP_MUTE, bits); } @@ -1276,7 +1276,7 @@ static struct snd_kcontrol_new cxt5047_base_mixers[] = { }; static struct snd_kcontrol_new cxt5047_hp_spk_mixers[] = { - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x1d, 0x00, HDA_OUTPUT), + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x1d, 0x01, HDA_OUTPUT), HDA_CODEC_VOLUME("Headphone Playback Volume", 0x13, 0x00, HDA_OUTPUT), {} }; -- cgit v1.2.3-70-g09d2 From 6f7cb44ba1a5195bf719f9ba1d57bd79e13262c1 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 11 Mar 2009 18:31:08 +0000 Subject: ASoC: Move WM8580 to normal I2C device probe Refactor the WM8580 device registration to probe via standard I2C device registration, registering the DAIs once the device has probed via I2C. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8580.c | 326 ++++++++++++++++++++++------------------------ sound/soc/codecs/wm8580.h | 5 - 2 files changed, 158 insertions(+), 173 deletions(-) diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 27f9e231bf6..442ea6f160f 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -1,7 +1,7 @@ /* * wm8580.c -- WM8580 ALSA Soc Audio driver * - * Copyright 2008 Wolfson Microelectronics PLC. + * Copyright 2008, 2009 Wolfson Microelectronics PLC. * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the @@ -35,17 +35,6 @@ #include "wm8580.h" -struct pll_state { - unsigned int in; - unsigned int out; -}; - -/* codec private data */ -struct wm8580_priv { - struct pll_state a; - struct pll_state b; -}; - /* WM8580 register space */ #define WM8580_PLLA1 0x00 #define WM8580_PLLA2 0x01 @@ -100,6 +89,8 @@ struct wm8580_priv { #define WM8580_READBACK 0x34 #define WM8580_RESET 0x35 +#define WM8580_MAX_REGISTER 0x35 + /* PLLB4 (register 7h) */ #define WM8580_PLLB4_MCLKOUTSRC_MASK 0x60 #define WM8580_PLLB4_MCLKOUTSRC_PLLA 0x20 @@ -191,6 +182,20 @@ static const u16 wm8580_reg[] = { 0x0000, 0x0000 /*R53*/ }; +struct pll_state { + unsigned int in; + unsigned int out; +}; + +/* codec private data */ +struct wm8580_priv { + struct snd_soc_codec codec; + u16 reg_cache[WM8580_MAX_REGISTER + 1]; + struct pll_state a; + struct pll_state b; +}; + + /* * read wm8580 register cache */ @@ -755,8 +760,22 @@ static int wm8580_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (codec->bias_level == SND_SOC_BIAS_OFF) { + /* Power up and get individual control of the DACs */ + reg = wm8580_read(codec, WM8580_PWRDN1); + reg &= ~(WM8580_PWRDN1_PWDN | WM8580_PWRDN1_ALLDACPD); + wm8580_write(codec, WM8580_PWRDN1, reg); + + /* Make VMID high impedence */ + reg = wm8580_read(codec, WM8580_ADC_CONTROL1); + reg &= ~0x100; + wm8580_write(codec, WM8580_ADC_CONTROL1, reg); + } break; + case SND_SOC_BIAS_OFF: reg = wm8580_read(codec, WM8580_PWRDN1); wm8580_write(codec, WM8580_PWRDN1, reg | WM8580_PWRDN1_PWDN); @@ -812,100 +831,163 @@ struct snd_soc_dai wm8580_dai[] = { }; EXPORT_SYMBOL_GPL(wm8580_dai); -/* - * initialise the WM8580 driver - * register the mixer and dsp interfaces with the kernel - */ -static int wm8580_init(struct snd_soc_device *socdev) +static struct snd_soc_codec *wm8580_codec; + +static int wm8580_probe(struct platform_device *pdev) { - struct snd_soc_codec *codec = socdev->card->codec; + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; int ret = 0; - codec->name = "WM8580"; - codec->owner = THIS_MODULE; - codec->read = wm8580_read_reg_cache; - codec->write = wm8580_write; - codec->set_bias_level = wm8580_set_bias_level; - codec->dai = wm8580_dai; - codec->num_dai = ARRAY_SIZE(wm8580_dai); - codec->reg_cache_size = ARRAY_SIZE(wm8580_reg); - codec->reg_cache = kmemdup(wm8580_reg, sizeof(wm8580_reg), - GFP_KERNEL); - - if (codec->reg_cache == NULL) - return -ENOMEM; - - /* Get the codec into a known state */ - wm8580_write(codec, WM8580_RESET, 0); - - /* Power up and get individual control of the DACs */ - wm8580_write(codec, WM8580_PWRDN1, wm8580_read(codec, WM8580_PWRDN1) & - ~(WM8580_PWRDN1_PWDN | WM8580_PWRDN1_ALLDACPD)); + if (wm8580_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } - /* Make VMID high impedence */ - wm8580_write(codec, WM8580_ADC_CONTROL1, - wm8580_read(codec, WM8580_ADC_CONTROL1) & ~0x100); + socdev->card->codec = wm8580_codec; + codec = wm8580_codec; /* register pcms */ - ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, - SNDRV_DEFAULT_STR1); + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) { - printk(KERN_ERR "wm8580: failed to create pcms\n"); + dev_err(codec->dev, "failed to create pcms: %d\n", ret); goto pcm_err; } snd_soc_add_controls(codec, wm8580_snd_controls, - ARRAY_SIZE(wm8580_snd_controls)); + ARRAY_SIZE(wm8580_snd_controls)); wm8580_add_widgets(codec); - ret = snd_soc_init_card(socdev); if (ret < 0) { - printk(KERN_ERR "wm8580: failed to register card\n"); + dev_err(codec->dev, "failed to register card: %d\n", ret); goto card_err; } + return ret; card_err: snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); pcm_err: - kfree(codec->reg_cache); return ret; } -/* If the i2c layer weren't so broken, we could pass this kind of data - around */ -static struct snd_soc_device *wm8580_socdev; +/* power down chip */ +static int wm8580_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); -/* - * WM8580 2 wire address is determined by GPIO5 - * state during powerup. - * low = 0x1a - * high = 0x1b - */ + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8580 = { + .probe = wm8580_probe, + .remove = wm8580_remove, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8580); + +static int wm8580_register(struct wm8580_priv *wm8580) +{ + int ret, i; + struct snd_soc_codec *codec = &wm8580->codec; + + if (wm8580_codec) { + dev_err(codec->dev, "Another WM8580 is registered\n"); + ret = -EINVAL; + goto err; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = wm8580; + codec->name = "WM8580"; + codec->owner = THIS_MODULE; + codec->read = wm8580_read_reg_cache; + codec->write = wm8580_write; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = wm8580_set_bias_level; + codec->dai = wm8580_dai; + codec->num_dai = ARRAY_SIZE(wm8580_dai); + codec->reg_cache_size = ARRAY_SIZE(wm8580->reg_cache); + codec->reg_cache = &wm8580->reg_cache; + + memcpy(codec->reg_cache, wm8580_reg, sizeof(wm8580_reg)); + + /* Get the codec into a known state */ + ret = wm8580_write(codec, WM8580_RESET, 0); + if (ret != 0) { + dev_err(codec->dev, "Failed to reset codec: %d\n", ret); + goto err; + } + + for (i = 0; i < ARRAY_SIZE(wm8580_dai); i++) + wm8580_dai[i].dev = codec->dev; + + wm8580_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + wm8580_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + goto err; + } + + ret = snd_soc_register_dais(wm8580_dai, ARRAY_SIZE(wm8580_dai)); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + goto err_codec; + } + + return 0; + +err_codec: + snd_soc_unregister_codec(codec); +err: + kfree(wm8580); + return ret; +} + +static void wm8580_unregister(struct wm8580_priv *wm8580) +{ + wm8580_set_bias_level(&wm8580->codec, SND_SOC_BIAS_OFF); + snd_soc_unregister_dais(wm8580_dai, ARRAY_SIZE(wm8580_dai)); + snd_soc_unregister_codec(&wm8580->codec); + kfree(wm8580); + wm8580_codec = NULL; +} + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static int wm8580_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { - struct snd_soc_device *socdev = wm8580_socdev; - struct snd_soc_codec *codec = socdev->card->codec; - int ret; + struct wm8580_priv *wm8580; + struct snd_soc_codec *codec; + + wm8580 = kzalloc(sizeof(struct wm8580_priv), GFP_KERNEL); + if (wm8580 == NULL) + return -ENOMEM; + + codec = &wm8580->codec; + codec->hw_write = (hw_write_t)i2c_master_send; - i2c_set_clientdata(i2c, codec); + i2c_set_clientdata(i2c, wm8580); codec->control_data = i2c; - ret = wm8580_init(socdev); - if (ret < 0) - dev_err(&i2c->dev, "failed to initialise WM8580\n"); - return ret; + codec->dev = &i2c->dev; + + return wm8580_register(wm8580); } static int wm8580_i2c_remove(struct i2c_client *client) { - struct snd_soc_codec *codec = i2c_get_clientdata(client); - kfree(codec->reg_cache); + struct wm8580_priv *wm8580 = i2c_get_clientdata(client); + wm8580_unregister(wm8580); return 0; } @@ -917,127 +999,35 @@ MODULE_DEVICE_TABLE(i2c, wm8580_i2c_id); static struct i2c_driver wm8580_i2c_driver = { .driver = { - .name = "WM8580 I2C Codec", + .name = "wm8580", .owner = THIS_MODULE, }, .probe = wm8580_i2c_probe, .remove = wm8580_i2c_remove, .id_table = wm8580_i2c_id, }; +#endif -static int wm8580_add_i2c_device(struct platform_device *pdev, - const struct wm8580_setup_data *setup) +static int __init wm8580_modinit(void) { - struct i2c_board_info info; - struct i2c_adapter *adapter; - struct i2c_client *client; int ret; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) ret = i2c_add_driver(&wm8580_i2c_driver); if (ret != 0) { - dev_err(&pdev->dev, "can't add i2c driver\n"); - return ret; - } - - memset(&info, 0, sizeof(struct i2c_board_info)); - info.addr = setup->i2c_address; - strlcpy(info.type, "wm8580", I2C_NAME_SIZE); - - adapter = i2c_get_adapter(setup->i2c_bus); - if (!adapter) { - dev_err(&pdev->dev, "can't get i2c adapter %d\n", - setup->i2c_bus); - goto err_driver; - } - - client = i2c_new_device(adapter, &info); - i2c_put_adapter(adapter); - if (!client) { - dev_err(&pdev->dev, "can't add i2c device at 0x%x\n", - (unsigned int)info.addr); - goto err_driver; - } - - return 0; - -err_driver: - i2c_del_driver(&wm8580_i2c_driver); - return -ENODEV; -} -#endif - -static int wm8580_probe(struct platform_device *pdev) -{ - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct wm8580_setup_data *setup; - struct snd_soc_codec *codec; - struct wm8580_priv *wm8580; - int ret = 0; - - setup = socdev->codec_data; - codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); - if (codec == NULL) - return -ENOMEM; - - wm8580 = kzalloc(sizeof(struct wm8580_priv), GFP_KERNEL); - if (wm8580 == NULL) { - kfree(codec); - return -ENOMEM; + pr_err("Failed to register WM8580 I2C driver: %d\n", ret); } - - codec->private_data = wm8580; - socdev->card->codec = codec; - mutex_init(&codec->mutex); - INIT_LIST_HEAD(&codec->dapm_widgets); - INIT_LIST_HEAD(&codec->dapm_paths); - wm8580_socdev = socdev; - -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - if (setup->i2c_address) { - codec->hw_write = (hw_write_t)i2c_master_send; - ret = wm8580_add_i2c_device(pdev, setup); - } -#else - /* Add other interfaces here */ -#endif - return ret; -} - -/* power down chip */ -static int wm8580_remove(struct platform_device *pdev) -{ - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->card->codec; - - if (codec->control_data) - wm8580_set_bias_level(codec, SND_SOC_BIAS_OFF); - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - i2c_unregister_device(codec->control_data); - i2c_del_driver(&wm8580_i2c_driver); #endif - kfree(codec->private_data); - kfree(codec); return 0; } - -struct snd_soc_codec_device soc_codec_dev_wm8580 = { - .probe = wm8580_probe, - .remove = wm8580_remove, -}; -EXPORT_SYMBOL_GPL(soc_codec_dev_wm8580); - -static int __init wm8580_modinit(void) -{ - return snd_soc_register_dais(wm8580_dai, ARRAY_SIZE(wm8580_dai)); -} module_init(wm8580_modinit); static void __exit wm8580_exit(void) { - snd_soc_unregister_dais(wm8580_dai, ARRAY_SIZE(wm8580_dai)); +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&wm8580_i2c_driver); +#endif } module_exit(wm8580_exit); diff --git a/sound/soc/codecs/wm8580.h b/sound/soc/codecs/wm8580.h index 09e4422f6f2..0dfb5ddde6a 100644 --- a/sound/soc/codecs/wm8580.h +++ b/sound/soc/codecs/wm8580.h @@ -28,11 +28,6 @@ #define WM8580_CLKSRC_OSC 4 #define WM8580_CLKSRC_NONE 5 -struct wm8580_setup_data { - int i2c_bus; - unsigned short i2c_address; -}; - #define WM8580_DAI_PAIFRX 0 #define WM8580_DAI_PAIFTX 1 -- cgit v1.2.3-70-g09d2 From eb5f6d753e337834c7ceb07824ee472e43d9a7a2 Mon Sep 17 00:00:00 2001 From: Philipp Zabel Date: Thu, 12 Mar 2009 11:07:54 +0100 Subject: ASoC: Replace remaining uses of snd_soc_cnew with snd_soc_add_controls. The drivers are basically duplicating the same code over and over. As snd_soc_cnew is going to be made static some time after the next merge window, we might as well convert them now. Signed-off-by: Philipp Zabel Signed-off-by: Mark Brown --- sound/soc/codecs/cs4270.c | 23 +++++------------------ sound/soc/codecs/tlv320aic26.c | 11 ++++------- sound/soc/codecs/wm8400.c | 12 ++---------- sound/soc/omap/n810.c | 12 +++++------- sound/soc/pxa/corgi.c | 12 +++++------- sound/soc/pxa/palm27x.c | 13 +++++-------- sound/soc/pxa/poodle.c | 12 +++++------- sound/soc/pxa/spitz.c | 12 +++++------- sound/soc/pxa/tosa.c | 12 +++++------- sound/soc/s3c24xx/neo1973_wm8753.c | 13 +++++-------- 10 files changed, 46 insertions(+), 86 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 2137670c9b7..7fa09a38762 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -540,7 +540,6 @@ static int cs4270_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = cs4270_codec; - unsigned int i; int ret; /* Connect the codec to the socdev. snd_soc_new_pcms() needs this. */ @@ -554,23 +553,11 @@ static int cs4270_probe(struct platform_device *pdev) } /* Add the non-DAPM controls */ - for (i = 0; i < ARRAY_SIZE(cs4270_snd_controls); i++) { - struct snd_kcontrol *kctrl; - - kctrl = snd_soc_cnew(&cs4270_snd_controls[i], codec, NULL); - if (!kctrl) { - dev_err(codec->dev, "error creating control '%s'\n", - cs4270_snd_controls[i].name); - ret = -ENOMEM; - goto error_free_pcms; - } - - ret = snd_ctl_add(codec->card, kctrl); - if (ret < 0) { - dev_err(codec->dev, "error adding control '%s'\n", - cs4270_snd_controls[i].name); - goto error_free_pcms; - } + ret = snd_soc_add_controls(codec, cs4270_snd_controls, + ARRAY_SIZE(cs4270_snd_controls)); + if (ret < 0) { + dev_err(codec->dev, "failed to add controls\n"); + goto error_free_pcms; } /* And finally, register the socdev */ diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index a7f333fc579..3387d9e736e 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -324,9 +324,8 @@ static int aic26_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec; - struct snd_kcontrol *kcontrol; struct aic26 *aic26; - int i, ret, err; + int ret, err; dev_info(&pdev->dev, "Probing AIC26 SoC CODEC driver\n"); dev_dbg(&pdev->dev, "socdev=%p\n", socdev); @@ -353,11 +352,9 @@ static int aic26_probe(struct platform_device *pdev) /* register controls */ dev_dbg(&pdev->dev, "Registering controls\n"); - for (i = 0; i < ARRAY_SIZE(aic26_snd_controls); i++) { - kcontrol = snd_soc_cnew(&aic26_snd_controls[i], codec, NULL); - err = snd_ctl_add(codec->card, kcontrol); - WARN_ON(err < 0); - } + err = snd_soc_add_controls(codec, aic26_snd_controls, + ARRAY_SIZE(aic26_snd_controls)); + WARN_ON(err < 0); /* CODEC is setup, we can register the card now */ dev_dbg(&pdev->dev, "Registering card\n"); diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index 4e1cefff848..744e0dc73be 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -351,16 +351,8 @@ SOC_SINGLE("RIN34 Mute Switch", WM8400_RIGHT_LINE_INPUT_3_4_VOLUME, /* add non dapm controls */ static int wm8400_add_controls(struct snd_soc_codec *codec) { - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm8400_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8400_snd_controls[i],codec, - NULL)); - if (err < 0) - return err; - } - return 0; + return snd_soc_add_controls(codec, wm8400_snd_controls, + ARRAY_SIZE(wm8400_snd_controls)); } /* diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 9f037cd0191..86471fd6340 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -248,7 +248,7 @@ static const struct snd_kcontrol_new aic33_n810_controls[] = { static int n810_aic33_init(struct snd_soc_codec *codec) { - int i, err; + int err; /* Not connected */ snd_soc_dapm_nc_pin(codec, "MONO_LOUT"); @@ -256,12 +256,10 @@ static int n810_aic33_init(struct snd_soc_codec *codec) snd_soc_dapm_nc_pin(codec, "HPRCOM"); /* Add N810 specific controls */ - for (i = 0; i < ARRAY_SIZE(aic33_n810_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&aic33_n810_controls[i], codec, NULL)); - if (err < 0) - return err; - } + err = snd_soc_add_controls(codec, aic33_n810_controls, + ARRAY_SIZE(aic33_n810_controls)); + if (err < 0) + return err; /* Add N810 specific widgets */ snd_soc_dapm_new_controls(codec, aic33_dapm_widgets, diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 146973ae097..02263e5d8f0 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -276,18 +276,16 @@ static const struct snd_kcontrol_new wm8731_corgi_controls[] = { */ static int corgi_wm8731_init(struct snd_soc_codec *codec) { - int i, err; + int err; snd_soc_dapm_nc_pin(codec, "LLINEIN"); snd_soc_dapm_nc_pin(codec, "RLINEIN"); /* Add corgi specific controls */ - for (i = 0; i < ARRAY_SIZE(wm8731_corgi_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8731_corgi_controls[i], codec, NULL)); - if (err < 0) - return err; - } + err = snd_soc_add_controls(codec, wm8731_corgi_controls, + ARRAY_SIZE(wm8731_corgi_controls)); + if (err < 0) + return err; /* Add corgi specific widgets */ snd_soc_dapm_new_controls(codec, wm8731_dapm_widgets, diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c index 29958cd9dae..48a73f64500 100644 --- a/sound/soc/pxa/palm27x.c +++ b/sound/soc/pxa/palm27x.c @@ -146,19 +146,16 @@ static const struct snd_kcontrol_new palm27x_controls[] = { static int palm27x_ac97_init(struct snd_soc_codec *codec) { - int i, err; + int err; snd_soc_dapm_nc_pin(codec, "OUT3"); snd_soc_dapm_nc_pin(codec, "MONOOUT"); /* add palm27x specific controls */ - for (i = 0; i < ARRAY_SIZE(palm27x_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&palm27x_controls[i], - codec, NULL)); - if (err < 0) - return err; - } + err = snd_soc_add_controls(codec, palm27x_controls, + ARRAY_SIZE(palm27x_controls)); + if (err < 0) + return err; /* add palm27x specific widgets */ snd_soc_dapm_new_controls(codec, palm27x_dapm_widgets, diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index fb17a0a5a09..ef7c6c8dc8f 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -241,19 +241,17 @@ static const struct snd_kcontrol_new wm8731_poodle_controls[] = { */ static int poodle_wm8731_init(struct snd_soc_codec *codec) { - int i, err; + int err; snd_soc_dapm_nc_pin(codec, "LLINEIN"); snd_soc_dapm_nc_pin(codec, "RLINEIN"); snd_soc_dapm_enable_pin(codec, "MICIN"); /* Add poodle specific controls */ - for (i = 0; i < ARRAY_SIZE(wm8731_poodle_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8731_poodle_controls[i], codec, NULL)); - if (err < 0) - return err; - } + err = snd_soc_add_controls(codec, wm8731_poodle_controls, + ARRAY_SIZE(wm8731_poodle_controls)); + if (err < 0) + return err; /* Add poodle specific widgets */ snd_soc_dapm_new_controls(codec, wm8731_dapm_widgets, diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 1aafd8c645a..6ca9f53080c 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -278,7 +278,7 @@ static const struct snd_kcontrol_new wm8750_spitz_controls[] = { */ static int spitz_wm8750_init(struct snd_soc_codec *codec) { - int i, err; + int err; /* NC codec pins */ snd_soc_dapm_nc_pin(codec, "RINPUT1"); @@ -290,12 +290,10 @@ static int spitz_wm8750_init(struct snd_soc_codec *codec) snd_soc_dapm_nc_pin(codec, "MONO1"); /* Add spitz specific controls */ - for (i = 0; i < ARRAY_SIZE(wm8750_spitz_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8750_spitz_controls[i], codec, NULL)); - if (err < 0) - return err; - } + err = snd_soc_add_controls(codec, wm8750_spitz_controls, + ARRAY_SIZE(wm8750_spitz_controls)); + if (err < 0) + return err; /* Add spitz specific widgets */ snd_soc_dapm_new_controls(codec, wm8750_dapm_widgets, diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index 09b5bada03b..fc781374b1b 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -188,18 +188,16 @@ static const struct snd_kcontrol_new tosa_controls[] = { static int tosa_ac97_init(struct snd_soc_codec *codec) { - int i, err; + int err; snd_soc_dapm_nc_pin(codec, "OUT3"); snd_soc_dapm_nc_pin(codec, "MONOOUT"); /* add tosa specific controls */ - for (i = 0; i < ARRAY_SIZE(tosa_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&tosa_controls[i],codec, NULL)); - if (err < 0) - return err; - } + err = snd_soc_add_controls(codec, tosa_controls, + ARRAY_SIZE(tosa_controls)); + if (err < 0) + return err; /* add tosa specific widgets */ snd_soc_dapm_new_controls(codec, tosa_dapm_widgets, diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index 5f6aeec0437..289fadf60b1 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -498,7 +498,7 @@ static const struct snd_kcontrol_new wm8753_neo1973_controls[] = { */ static int neo1973_wm8753_init(struct snd_soc_codec *codec) { - int i, err; + int err; pr_debug("Entered %s\n", __func__); @@ -518,13 +518,10 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec) set_scenario_endpoints(codec, NEO_AUDIO_OFF); /* add neo1973 specific controls */ - for (i = 0; i < ARRAY_SIZE(wm8753_neo1973_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8753_neo1973_controls[i], - codec, NULL)); - if (err < 0) - return err; - } + err = snd_soc_add_controls(codec, wm8753_neo1973_controls, + ARRAY_SIZE(8753_neo1973_controls)); + if (err < 0) + return err; /* set up neo1973 specific audio routes */ err = snd_soc_dapm_add_routes(codec, dapm_routes, -- cgit v1.2.3-70-g09d2 From 3b7523fc828e41b2988feb400704e01b67859d78 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 12 Mar 2009 16:45:01 +0100 Subject: ALSA: hda - Add comments for the previous fix for conexant codecs Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 6cb184e9c2f..bc016fade19 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1196,6 +1196,10 @@ static int cxt5047_hp_master_sw_put(struct snd_kcontrol *kcontrol, * the headphone jack */ bits = (!spec->hp_present && spec->cur_eapd) ? 0 : HDA_AMP_MUTE; + /* NOTE: Conexat codec needs the index for *OUTPUT* amp of + * pin widgets unlike other codecs. In this case, we need to + * set index 0x01 for the volume from the mixer amp 0x19. + */ snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0x01, HDA_AMP_MUTE, bits); bits = spec->cur_eapd ? 0 : HDA_AMP_MUTE; @@ -1214,6 +1218,7 @@ static void cxt5047_hp_automute(struct hda_codec *codec) AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; bits = (spec->hp_present || !spec->cur_eapd) ? HDA_AMP_MUTE : 0; + /* See the note in cxt5047_hp_master_sw_put */ snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0x01, HDA_AMP_MUTE, bits); } @@ -1276,6 +1281,7 @@ static struct snd_kcontrol_new cxt5047_base_mixers[] = { }; static struct snd_kcontrol_new cxt5047_hp_spk_mixers[] = { + /* See the note in cxt5047_hp_master_sw_put */ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x1d, 0x01, HDA_OUTPUT), HDA_CODEC_VOLUME("Headphone Playback Volume", 0x13, 0x00, HDA_OUTPUT), {} -- cgit v1.2.3-70-g09d2 From 9421f9543b3a0a870499f64498406003de8214b4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 12 Mar 2009 17:06:07 +0100 Subject: ALSA: hda - Print multiple out-amp values of pin widgets on Conext codecs Add a flag to work around the non-standard amp-value handling on Conexant codecs. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.h | 3 +++ sound/pci/hda/hda_proc.c | 10 ++++++++-- sound/pci/hda/patch_conexant.c | 3 +++ 3 files changed, 14 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 2ea628478a9..079e1ab718d 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -793,6 +793,9 @@ struct hda_codec { * status change * (e.g. Realtek codecs) */ + unsigned int pin_amp_workaround:1; /* pin out-amp takes index + * (e.g. Conexant codecs) + */ #ifdef CONFIG_SND_HDA_POWER_SAVE unsigned int power_on :1; /* current (global) power-state */ unsigned int power_transition :1; /* power-state in transition */ diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 144b85276d5..93b25ba4d00 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -554,8 +554,14 @@ static void print_codec_info(struct snd_info_entry *entry, snd_iprintf(buffer, " Amp-Out caps: "); print_amp_caps(buffer, codec, nid, HDA_OUTPUT); snd_iprintf(buffer, " Amp-Out vals: "); - print_amp_vals(buffer, codec, nid, HDA_OUTPUT, - wid_caps & AC_WCAP_STEREO, 1); + if (wid_type == AC_WID_PIN && + codec->pin_amp_workaround) + print_amp_vals(buffer, codec, nid, HDA_OUTPUT, + wid_caps & AC_WCAP_STEREO, + conn_len); + else + print_amp_vals(buffer, codec, nid, HDA_OUTPUT, + wid_caps & AC_WCAP_STEREO, 1); } switch (wid_type) { diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index bc016fade19..1f2ad76ca94 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1066,6 +1066,7 @@ static int patch_cxt5045(struct hda_codec *codec) if (!spec) return -ENOMEM; codec->spec = spec; + codec->pin_amp_workaround = 1; spec->multiout.max_channels = 2; spec->multiout.num_dacs = ARRAY_SIZE(cxt5045_dac_nids); @@ -1501,6 +1502,7 @@ static int patch_cxt5047(struct hda_codec *codec) if (!spec) return -ENOMEM; codec->spec = spec; + codec->pin_amp_workaround = 1; spec->multiout.max_channels = 2; spec->multiout.num_dacs = ARRAY_SIZE(cxt5047_dac_nids); @@ -1847,6 +1849,7 @@ static int patch_cxt5051(struct hda_codec *codec) if (!spec) return -ENOMEM; codec->spec = spec; + codec->pin_amp_workaround = 1; codec->patch_ops = conexant_patch_ops; codec->patch_ops.init = cxt5051_init; -- cgit v1.2.3-70-g09d2 From 307282c8990c5658604b9fda8a64a9a07079b850 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 12 Mar 2009 18:17:58 +0100 Subject: ALSA: hda - Add model=vaio for STAC9872 Add the default pin config for model=vaio (in case of broken BIOS). Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 3 ++- sound/pci/hda/patch_sigmatel.c | 33 ++++++++++++++++++++++++++-- 2 files changed, 33 insertions(+), 3 deletions(-) diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index f9253ea3c19..8eec05bc079 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -368,4 +368,5 @@ STAC92HD83* STAC9872 ======== - N/A + vaio VAIO laptop without SPDIF + auto BIOS setup (default) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 72c87aa20bd..e06fc7decd3 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -155,6 +155,12 @@ enum { STAC_927X_MODELS }; +enum { + STAC_9872_AUTO, + STAC_9872_VAIO, + STAC_9872_MODELS +}; + struct sigmatel_event { hda_nid_t nid; unsigned char type; @@ -5588,6 +5594,25 @@ static hda_nid_t stac9872_mux_nids[] = { 0x15 }; +static unsigned int stac9872_vaio_pin_configs[9] = { + 0x03211020, 0x411111f0, 0x411111f0, 0x03a15030, + 0x411111f0, 0x90170110, 0x411111f0, 0x411111f0, + 0x90a7013e +}; + +static const char *stac9872_models[STAC_9872_MODELS] = { + [STAC_9872_AUTO] = "auto", + [STAC_9872_VAIO] = "vaio", +}; + +static unsigned int *stac9872_brd_tbl[STAC_9872_MODELS] = { + [STAC_9872_VAIO] = stac9872_vaio_pin_configs, +}; + +static struct snd_pci_quirk stac9872_cfg_tbl[] = { + {} /* terminator */ +}; + static int patch_stac9872(struct hda_codec *codec) { struct sigmatel_spec *spec; @@ -5598,11 +5623,15 @@ static int patch_stac9872(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; -#if 0 /* no model right now */ spec->board_config = snd_hda_check_board_config(codec, STAC_9872_MODELS, stac9872_models, stac9872_cfg_tbl); -#endif + if (spec->board_config < 0) + snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC9872, " + "using BIOS defaults\n"); + else + stac92xx_set_config_regs(codec, + stac9872_brd_tbl[spec->board_config]); spec->num_pins = ARRAY_SIZE(stac9872_pin_nids); spec->pin_nids = stac9872_pin_nids; -- cgit v1.2.3-70-g09d2 From bb6ac72fb19c6676eb8bafa8e3b8bf970a2294a2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 13 Mar 2009 09:02:42 +0100 Subject: ALSA: hda - power up before codec initialization Change the power state of each widget before starting the initialization work so that all verbs are executed properly. Also, keep power-up during hwdep reconfiguration. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 15 ++++++++------- sound/pci/hda/hda_hwdep.c | 14 +++++++++----- 2 files changed, 17 insertions(+), 12 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 1885e764910..cf6339436de 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -842,6 +842,9 @@ static void snd_hda_codec_free(struct hda_codec *codec) kfree(codec); } +static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, + unsigned int power_state); + /** * snd_hda_codec_new - create a HDA codec * @bus: the bus to assign @@ -941,6 +944,11 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr if (bus->modelname) codec->modelname = kstrdup(bus->modelname, GFP_KERNEL); + /* power-up all before initialization */ + hda_set_power_state(codec, + codec->afg ? codec->afg : codec->mfg, + AC_PWRST_D0); + if (do_init) { err = snd_hda_codec_configure(codec); if (err < 0) @@ -2413,19 +2421,12 @@ EXPORT_SYMBOL_HDA(snd_hda_build_controls); int snd_hda_codec_build_controls(struct hda_codec *codec) { int err = 0; - /* fake as if already powered-on */ - hda_keep_power_on(codec); - /* then fire up */ - hda_set_power_state(codec, - codec->afg ? codec->afg : codec->mfg, - AC_PWRST_D0); hda_exec_init_verbs(codec); /* continue to initialize... */ if (codec->patch_ops.init) err = codec->patch_ops.init(codec); if (!err && codec->patch_ops.build_controls) err = codec->patch_ops.build_controls(codec); - snd_hda_power_down(codec); if (err < 0) return err; return 0; diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index 1e3ccc740af..1c57505c287 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -176,25 +176,29 @@ static int reconfig_codec(struct hda_codec *codec) { int err; + snd_hda_power_up(codec); snd_printk(KERN_INFO "hda-codec: reconfiguring\n"); err = snd_hda_codec_reset(codec); if (err < 0) { snd_printk(KERN_ERR "The codec is being used, can't reconfigure.\n"); - return err; + goto error; } err = snd_hda_codec_configure(codec); if (err < 0) - return err; + goto error; /* rebuild PCMs */ err = snd_hda_codec_build_pcms(codec); if (err < 0) - return err; + goto error; /* rebuild mixers */ err = snd_hda_codec_build_controls(codec); if (err < 0) - return err; - return snd_card_register(codec->bus->card); + goto error; + err = snd_card_register(codec->bus->card); + error: + snd_hda_power_down(codec); + return err; } /* -- cgit v1.2.3-70-g09d2 From 77dd7e17b86bd81b3638e01d784a72652071508b Mon Sep 17 00:00:00 2001 From: "Lopez Cruz, Misael" Date: Thu, 12 Mar 2009 21:45:27 -0500 Subject: ASoC: Move headset jack registration to device initialization for SDP3430 Move headset jack registration to the codec/machine specific initialization. Having the jack registration in machine init causes that the jack device gets initialized but not registered since the sound card is registered before the jack. Moving jack registration to device initialization will register the jack device along with all other devices associated to the card when the card is registed. As a consequence of jack device registered properly, the jack is detected as an input device. Signed-off-by: Misael Lopez Cruz Signed-off-by: Mark Brown --- sound/soc/omap/sdp3430.c | 74 +++++++++++++++++++++++++----------------------- 1 file changed, 39 insertions(+), 35 deletions(-) diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index 715c648203a..0a41de677e7 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -39,6 +39,8 @@ #include "omap-pcm.h" #include "../codecs/twl4030.h" +static struct snd_soc_card snd_soc_sdp3430; + static int sdp3430_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -82,6 +84,27 @@ static struct snd_soc_ops sdp3430_ops = { .hw_params = sdp3430_hw_params, }; +/* Headset jack */ +static struct snd_soc_jack hs_jack; + +/* Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin hs_jack_pins[] = { + { + .pin = "Headset Jack", + .mask = SND_JACK_HEADSET, + }, +}; + +/* Headset jack detection gpios */ +static struct snd_soc_jack_gpio hs_jack_gpios[] = { + { + .gpio = (OMAP_MAX_GPIO_LINES + 2), + .name = "hsdet-gpio", + .report = SND_JACK_HEADSET, + .debounce_time = 200, + }, +}; + /* SDP3430 machine DAPM */ static const struct snd_soc_dapm_widget sdp3430_twl4030_dapm_widgets[] = { SND_SOC_DAPM_MIC("Ext Mic", NULL), @@ -141,30 +164,25 @@ static int sdp3430_twl4030_init(struct snd_soc_codec *codec) snd_soc_dapm_nc_pin(codec, "CARKITR"); ret = snd_soc_dapm_sync(codec); + if (ret) + return ret; - return ret; -} + /* Headset jack detection */ + ret = snd_soc_jack_new(&snd_soc_sdp3430, "Headset Jack", + SND_JACK_HEADSET, &hs_jack); + if (ret) + return ret; -/* Headset jack */ -static struct snd_soc_jack hs_jack; + ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), + hs_jack_pins); + if (ret) + return ret; -/* Headset jack detection DAPM pins */ -static struct snd_soc_jack_pin hs_jack_pins[] = { - { - .pin = "Headset Jack", - .mask = SND_JACK_HEADSET, - }, -}; + ret = snd_soc_jack_add_gpios(&hs_jack, ARRAY_SIZE(hs_jack_gpios), + hs_jack_gpios); -/* Headset jack detection gpios */ -static struct snd_soc_jack_gpio hs_jack_gpios[] = { - { - .gpio = (OMAP_MAX_GPIO_LINES + 2), - .name = "hsdet-gpio", - .report = SND_JACK_HEADSET, - .debounce_time = 200, - }, -}; + return ret; +} /* Digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link sdp3430_dai = { @@ -216,21 +234,7 @@ static int __init sdp3430_soc_init(void) if (ret) goto err1; - /* Headset jack detection */ - ret = snd_soc_jack_new(&snd_soc_sdp3430, "SDP3430 headset jack", - SND_JACK_HEADSET, &hs_jack); - if (ret) - return ret; - - ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), - hs_jack_pins); - if (ret) - return ret; - - ret = snd_soc_jack_add_gpios(&hs_jack, ARRAY_SIZE(hs_jack_gpios), - hs_jack_gpios); - - return ret; + return 0; err1: printk(KERN_ERR "Unable to add platform device\n"); -- cgit v1.2.3-70-g09d2 From 72d7466468b471f99cefae3c5f4a414bbbaa0bdd Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Thu, 12 Mar 2009 11:27:49 +0100 Subject: ASoC: switch PXA SSP driver from network mode to PSP This switches the pxa ssp port usage from network mode to PSP mode. Removed some comments and checks for configured TDM channels. A special case is added to support configuration where BCLK = 64fs. We need to do some black magic in this case which doesn't look nice but there is unfortunately no other option than that. Diagnosed-by: Tim Ruetz Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- sound/soc/pxa/pxa-ssp.c | 44 +++++++++++++++++++++++++++++++++----------- 1 file changed, 33 insertions(+), 11 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index d3fa6357a9f..4dd0d7c5722 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -558,18 +558,17 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - sscr0 |= SSCR0_MOD | SSCR0_PSP; + sscr0 |= SSCR0_PSP; sscr1 |= SSCR1_RWOT | SSCR1_TRAIL; switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: - sspsp |= SSPSP_FSRT; break; case SND_SOC_DAIFMT_NB_IF: - sspsp |= SSPSP_SFRMP | SSPSP_FSRT; + sspsp |= SSPSP_SFRMP; break; case SND_SOC_DAIFMT_IB_IF: - sspsp |= SSPSP_SFRMP; + sspsp |= SSPSP_SFRMP | SSPSP_SCMODE(3); break; default: return -EINVAL; @@ -655,33 +654,56 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, sscr0 |= SSCR0_FPCKE; #endif sscr0 |= SSCR0_DataSize(16); - /* use network mode (2 slots) for 16 bit stereo */ break; case SNDRV_PCM_FORMAT_S24_LE: sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(8)); - /* we must be in network mode (2 slots) for 24 bit stereo */ break; case SNDRV_PCM_FORMAT_S32_LE: sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(16)); - /* we must be in network mode (2 slots) for 32 bit stereo */ break; } ssp_write_reg(ssp, SSCR0, sscr0); switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - /* Cleared when the DAI format is set */ - sspsp = ssp_read_reg(ssp, SSPSP) | SSPSP_SFRMWDTH(width); + sspsp = ssp_read_reg(ssp, SSPSP); + + if (((sscr0 & SSCR0_SCR) == SSCR0_SerClkDiv(4)) && + (width == 16)) { + /* This is a special case where the bitclk is 64fs + * and we're not dealing with 2*32 bits of audio + * samples. + * + * The SSP values used for that are all found out by + * trying and failing a lot; some of the registers + * needed for that mode are only available on PXA3xx. + */ + +#ifdef CONFIG_PXA3xx + if (!cpu_is_pxa3xx()) + return -EINVAL; + + sspsp |= SSPSP_SFRMWDTH(width * 2); + sspsp |= SSPSP_SFRMDLY(width * 4); + sspsp |= SSPSP_EDMYSTOP(3); + sspsp |= SSPSP_DMYSTOP(3); + sspsp |= SSPSP_DMYSTRT(1); +#else + return -EINVAL; +#endif + } else + sspsp |= SSPSP_SFRMWDTH(width); + ssp_write_reg(ssp, SSPSP, sspsp); break; default: break; } - /* We always use a network mode so we always require TDM slots + /* When we use a network mode, we always require TDM slots * - complain loudly and fail if they've not been set up yet. */ - if (!(ssp_read_reg(ssp, SSTSA) & 0xf)) { + if ((sscr0 & SSCR0_MOD) && !(ssp_read_reg(ssp, SSTSA) & 0xf)) { dev_err(&ssp->pdev->dev, "No TDM timeslot configured\n"); return -EINVAL; } -- cgit v1.2.3-70-g09d2 From 58d8395b74f78a2f4225c5faea8b5bffb8af1cf9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 13 Mar 2009 17:04:34 +0100 Subject: ALSA: hda - Add another HP model with IDT92HD71bx codec HP laptops require GPIO0 on as EAPD. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index e06fc7decd3..4da72403fc8 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1855,6 +1855,8 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { "DFI LanParty", STAC_92HD71BXX_REF), SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, "DFI LanParty", STAC_92HD71BXX_REF), + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x3080, + "HP", STAC_HP_DV5), SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x30f0, "HP dv4-7", STAC_HP_DV5), SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x3600, -- cgit v1.2.3-70-g09d2 From 0ce36c5f7f87632f26c8fbefe68b5116eda152d2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 13 Mar 2009 14:26:08 +0000 Subject: ASoC: Fix non-networked I2S mode for PXA SSP Two issues are fixed here: - I2S transmits the left frame with the clock low but I don't seem to get LRCLK out without SFRMDLY being set so invert SFRMP and set a delay. - I2S has a clock cycle prior to the first data byte in each channel so we need to delay the data by one cycle. Tested-by: Daniel Mack Signed-off-by: Mark Brown --- sound/soc/pxa/pxa-ssp.c | 19 ++++++++++++++----- 1 file changed, 14 insertions(+), 5 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 4dd0d7c5722..b0bf40973d5 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -1,4 +1,3 @@ -#define DEBUG /* * pxa-ssp.c -- ALSA Soc Audio Layer * @@ -561,14 +560,15 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, sscr0 |= SSCR0_PSP; sscr1 |= SSCR1_RWOT | SSCR1_TRAIL; + /* See hw_params() */ switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: + sspsp |= SSPSP_SFRMP; break; case SND_SOC_DAIFMT_NB_IF: - sspsp |= SSPSP_SFRMP; break; case SND_SOC_DAIFMT_IB_IF: - sspsp |= SSPSP_SFRMP | SSPSP_SCMODE(3); + sspsp |= SSPSP_SCMODE(3); break; default: return -EINVAL; @@ -691,8 +691,17 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, #else return -EINVAL; #endif - } else - sspsp |= SSPSP_SFRMWDTH(width); + } else { + /* The frame width is the width the LRCLK is + * asserted for; the delay is expressed in + * half cycle units. We need the extra cycle + * because the data starts clocking out one BCLK + * after LRCLK changes polarity. + */ + sspsp |= SSPSP_SFRMWDTH(width + 1); + sspsp |= SSPSP_SFRMDLY((width + 1) * 2); + sspsp |= SSPSP_DMYSTRT(1); + } ssp_write_reg(ssp, SSPSP, sspsp); break; -- cgit v1.2.3-70-g09d2 From 85fab7802a4bc00cc752f430e22a0d9fc41fe199 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 13 Mar 2009 14:27:08 +0000 Subject: ASoC: Fix Zylonite for non-networked SSP mode This also simplifies the code a bit. Signed-off-by: Mark Brown --- sound/soc/pxa/zylonite.c | 55 ++++++++++++++++++++++-------------------------- 1 file changed, 25 insertions(+), 30 deletions(-) diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index 9f6116edbb8..9a386b4c4ed 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -96,42 +96,35 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; unsigned int pll_out = 0; - unsigned int acds = 0; unsigned int wm9713_div = 0; int ret = 0; - - switch (params_rate(params)) { + int rate = params_rate(params); + int width = snd_pcm_format_physical_width(params_format(params)); + + /* Only support ratios that we can generate neatly from the AC97 + * based master clock - in particular, this excludes 44.1kHz. + * In most applications the voice DAC will be used for telephony + * data so multiples of 8kHz will be the common case. + */ + switch (rate) { case 8000: wm9713_div = 12; - pll_out = 2048000; break; case 16000: wm9713_div = 6; - pll_out = 4096000; break; case 48000: - default: wm9713_div = 2; - pll_out = 12288000; - acds = 1; break; + default: + /* Don't support OSS emulation */ + return -EINVAL; } - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; + /* Add 1 to the width for the leading clock cycle */ + pll_out = rate * (width + 1) * 8; - /* Use network mode for stereo, one slot per channel. */ - if (params_channels(params) > 1) - ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 2); - else - ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 1); + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0, 1); if (ret < 0) return ret; @@ -139,14 +132,6 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_AUDIO_DIV_ACDS, acds); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0, 1); - if (ret < 0) - return ret; - if (clk_pout) ret = snd_soc_dai_set_clkdiv(codec_dai, WM9713_PCMCLK_PLL_DIV, WM9713_PCMDIV(wm9713_div)); @@ -156,6 +141,16 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + return 0; } -- cgit v1.2.3-70-g09d2 From 26ade896b6ba3fd017ef4a26e71e7b7569222cb6 Mon Sep 17 00:00:00 2001 From: Robert Jarzmik Date: Sun, 15 Mar 2009 14:10:54 +0100 Subject: ASoC: Allow choice of ac97 gpio reset line As the PXA27x series allow 2 gpios to reset the ac97 bus, allow through platform data configuration the definition of the correct gpio which will reset the AC97 bus. This comes from a silicon defect on the PXA27x series, where the gpio must be manually controlled in warm reset cases. Signed-off-by: Robert Jarzmik Signed-off-by: Mark Brown --- include/sound/pxa2xx-lib.h | 15 ++++++++++ sound/arm/pxa2xx-ac97-lib.c | 71 +++++++++++++++++++++++++++++++++++++++++---- 2 files changed, 81 insertions(+), 5 deletions(-) diff --git a/include/sound/pxa2xx-lib.h b/include/sound/pxa2xx-lib.h index 2fd3d251d9a..2c894b600e5 100644 --- a/include/sound/pxa2xx-lib.h +++ b/include/sound/pxa2xx-lib.h @@ -42,4 +42,19 @@ extern int pxa2xx_ac97_hw_resume(void); extern int pxa2xx_ac97_hw_probe(struct platform_device *dev); extern void pxa2xx_ac97_hw_remove(struct platform_device *dev); +/* AC97 platform_data */ +/** + * struct pxa2xx_ac97_platform_data - pxa ac97 platform data + * @reset_gpio: AC97 reset gpio (normally gpio113 or gpio95) + * a -1 value means no gpio will be used for reset + * + * Platform data should only be specified for pxa27x CPUs where a silicon bug + * prevents correct operation of the reset line. If not specified, the default + * behaviour is to consider gpio 113 as the AC97 reset line, which is the + * default on most boards. + */ +struct pxa2xx_ac97_platform_data { + int reset_gpio; +}; + #endif diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index 35afd0c33be..d721ea7cae8 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -31,6 +31,7 @@ static DECLARE_WAIT_QUEUE_HEAD(gsr_wq); static volatile long gsr_bits; static struct clk *ac97_clk; static struct clk *ac97conf_clk; +static int reset_gpio; /* * Beware PXA27x bugs: @@ -42,6 +43,45 @@ static struct clk *ac97conf_clk; * 1 jiffy timeout if interrupt never comes). */ +enum { + RESETGPIO_FORCE_HIGH, + RESETGPIO_FORCE_LOW, + RESETGPIO_NORMAL_ALTFUNC +}; + +/** + * set_resetgpio_mode - computes and sets the AC97_RESET gpio mode on PXA + * @mode: chosen action + * + * As the PXA27x CPUs suffer from a AC97 bug, a manual control of the reset line + * must be done to insure proper work of AC97 reset line. This function + * computes the correct gpio_mode for further use by reset functions, and + * applied the change through pxa_gpio_mode. + */ +static void set_resetgpio_mode(int resetgpio_action) +{ + int mode = 0; + + if (reset_gpio) + switch (resetgpio_action) { + case RESETGPIO_NORMAL_ALTFUNC: + if (reset_gpio == 113) + mode = 113 | GPIO_OUT | GPIO_DFLT_LOW; + if (reset_gpio == 95) + mode = 95 | GPIO_ALT_FN_1_OUT; + break; + case RESETGPIO_FORCE_LOW: + mode = reset_gpio | GPIO_OUT | GPIO_DFLT_LOW; + break; + case RESETGPIO_FORCE_HIGH: + mode = reset_gpio | GPIO_OUT | GPIO_DFLT_HIGH; + break; + }; + + if (mode) + pxa_gpio_mode(mode); +} + unsigned short pxa2xx_ac97_read(struct snd_ac97 *ac97, unsigned short reg) { unsigned short val = -1; @@ -137,10 +177,10 @@ static inline void pxa_ac97_warm_pxa27x(void) /* warm reset broken on Bulverde, so manually keep AC97 reset high */ - pxa_gpio_mode(113 | GPIO_OUT | GPIO_DFLT_HIGH); + set_resetgpio_mode(RESETGPIO_FORCE_HIGH); udelay(10); GCR |= GCR_WARM_RST; - pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); + set_resetgpio_mode(RESETGPIO_NORMAL_ALTFUNC); udelay(500); } @@ -308,8 +348,8 @@ int pxa2xx_ac97_hw_resume(void) pxa_gpio_mode(GPIO29_SDATA_IN_AC97_MD); } if (cpu_is_pxa27x()) { - /* Use GPIO 113 as AC97 Reset on Bulverde */ - pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); + /* Use GPIO 113 or 95 as AC97 Reset on Bulverde */ + set_resetgpio_mode(RESETGPIO_NORMAL_ALTFUNC); } clk_enable(ac97_clk); return 0; @@ -320,6 +360,27 @@ EXPORT_SYMBOL_GPL(pxa2xx_ac97_hw_resume); int __devinit pxa2xx_ac97_hw_probe(struct platform_device *dev) { int ret; + struct pxa2xx_ac97_platform_data *pdata = dev->dev.platform_data; + + if (pdata) { + switch (pdata->reset_gpio) { + case 95: + case 113: + reset_gpio = pdata->reset_gpio; + break; + case 0: + reset_gpio = 113; + break; + case -1: + break; + default: + dev_err(dev, "Invalid reset GPIO %d\n", + pdata->reset_gpio); + } + } else { + if (cpu_is_pxa27x()) + reset_gpio = 113; + } if (cpu_is_pxa25x() || cpu_is_pxa27x()) { pxa_gpio_mode(GPIO31_SYNC_AC97_MD); @@ -330,7 +391,7 @@ int __devinit pxa2xx_ac97_hw_probe(struct platform_device *dev) if (cpu_is_pxa27x()) { /* Use GPIO 113 as AC97 Reset on Bulverde */ - pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); + set_resetgpio_mode(RESETGPIO_NORMAL_ALTFUNC); ac97conf_clk = clk_get(&dev->dev, "AC97CONFCLK"); if (IS_ERR(ac97conf_clk)) { ret = PTR_ERR(ac97conf_clk); -- cgit v1.2.3-70-g09d2 From 9f5d790d1b0af8e3705df12fd5d49a1df2a45c47 Mon Sep 17 00:00:00 2001 From: Giuliano Pochini Date: Sun, 15 Mar 2009 21:33:34 +0100 Subject: ALSA: echoaudio: remove line-out volume from vmixer cards There is a long standing bug in the drivers for cards with a vmixer because I overlooked a detail in the c++ generic driver by echoaudio. Those cards do not have a line-out volume control. It is a virtual control provided by the generic driver. The bug is harmless because the DSP just ignores the command to change the volume. *NB:* It breaks alsa-tools/echomixer. A patch for it will follow. This patch removes the line-out volume control from vmixer-equipped cards. Signed-off-by: Giuliano Pochini Signed-off-by: Takashi Iwai --- sound/pci/echoaudio/echoaudio.c | 17 +++-------------- 1 file changed, 3 insertions(+), 14 deletions(-) diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 8dbc5c4ba42..4b70ea1e4c9 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -950,6 +950,8 @@ static int __devinit snd_echo_new_pcm(struct echoaudio *chip) Control interface ******************************************************************************/ +#ifndef ECHOCARD_HAS_VMIXER + /******************* PCM output volume *******************/ static int snd_echo_output_gain_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) @@ -1001,18 +1003,6 @@ static int snd_echo_output_gain_put(struct snd_kcontrol *kcontrol, return changed; } -#ifdef ECHOCARD_HAS_VMIXER -/* On Vmixer cards this one controls the line-out volume */ -static struct snd_kcontrol_new snd_echo_line_output_gain __devinitdata = { - .name = "Line Playback Volume", - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, - .info = snd_echo_output_gain_info, - .get = snd_echo_output_gain_get, - .put = snd_echo_output_gain_put, - .tlv = {.p = db_scale_output_gain}, -}; -#else static struct snd_kcontrol_new snd_echo_pcm_output_gain __devinitdata = { .name = "PCM Playback Volume", .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -1022,6 +1012,7 @@ static struct snd_kcontrol_new snd_echo_pcm_output_gain __devinitdata = { .put = snd_echo_output_gain_put, .tlv = {.p = db_scale_output_gain}, }; + #endif @@ -2037,8 +2028,6 @@ static int __devinit snd_echo_probe(struct pci_dev *pci, #ifdef ECHOCARD_HAS_VMIXER snd_echo_vmixer.count = num_pipes_out(chip) * num_busses_out(chip); - if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_line_output_gain, chip))) < 0) - goto ctl_error; if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_vmixer, chip))) < 0) goto ctl_error; #else -- cgit v1.2.3-70-g09d2 From 4c55bb0149b604901e4989d1ee0fddc53df8eb0c Mon Sep 17 00:00:00 2001 From: Giuliano Pochini Date: Sun, 15 Mar 2009 21:33:55 +0100 Subject: ALSA: echoaudio: remove line-out volume from vmixer cards With this patch the drivers do not set the vmixer volume anymore at startup because it is actually the output volume of the voices and ALSA mandates that the volume must be 0 by default. Signed-off-by: Giuliano Pochini Signed-off-by: Takashi Iwai --- sound/pci/echoaudio/indigo_dsp.c | 12 ------------ sound/pci/echoaudio/indigodj_dsp.c | 12 ------------ sound/pci/echoaudio/indigoio_dsp.c | 12 ------------ sound/pci/echoaudio/mia_dsp.c | 12 ------------ 4 files changed, 48 deletions(-) diff --git a/sound/pci/echoaudio/indigo_dsp.c b/sound/pci/echoaudio/indigo_dsp.c index f05e39f7aad..0b2cd9c8627 100644 --- a/sound/pci/echoaudio/indigo_dsp.c +++ b/sound/pci/echoaudio/indigo_dsp.c @@ -63,18 +63,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) if ((err = init_line_levels(chip)) < 0) return err; - /* Default routing of the virtual channels: all vchannels are routed - to the stereo output */ - set_vmixer_gain(chip, 0, 0, 0); - set_vmixer_gain(chip, 1, 1, 0); - set_vmixer_gain(chip, 0, 2, 0); - set_vmixer_gain(chip, 1, 3, 0); - set_vmixer_gain(chip, 0, 4, 0); - set_vmixer_gain(chip, 1, 5, 0); - set_vmixer_gain(chip, 0, 6, 0); - set_vmixer_gain(chip, 1, 7, 0); - err = update_vmixer_level(chip); - DE_INIT(("init_hw done\n")); return err; } diff --git a/sound/pci/echoaudio/indigodj_dsp.c b/sound/pci/echoaudio/indigodj_dsp.c index 90730a5ecb4..08392916691 100644 --- a/sound/pci/echoaudio/indigodj_dsp.c +++ b/sound/pci/echoaudio/indigodj_dsp.c @@ -63,18 +63,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) if ((err = init_line_levels(chip)) < 0) return err; - /* Default routing of the virtual channels: vchannels 0-3 and - vchannels 4-7 are routed to real channels 0-4 */ - set_vmixer_gain(chip, 0, 0, 0); - set_vmixer_gain(chip, 1, 1, 0); - set_vmixer_gain(chip, 2, 2, 0); - set_vmixer_gain(chip, 3, 3, 0); - set_vmixer_gain(chip, 0, 4, 0); - set_vmixer_gain(chip, 1, 5, 0); - set_vmixer_gain(chip, 2, 6, 0); - set_vmixer_gain(chip, 3, 7, 0); - err = update_vmixer_level(chip); - DE_INIT(("init_hw done\n")); return err; } diff --git a/sound/pci/echoaudio/indigoio_dsp.c b/sound/pci/echoaudio/indigoio_dsp.c index a7e09ec2107..0604c8a8522 100644 --- a/sound/pci/echoaudio/indigoio_dsp.c +++ b/sound/pci/echoaudio/indigoio_dsp.c @@ -63,18 +63,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) if ((err = init_line_levels(chip)) < 0) return err; - /* Default routing of the virtual channels: all vchannels are routed - to the stereo output */ - set_vmixer_gain(chip, 0, 0, 0); - set_vmixer_gain(chip, 1, 1, 0); - set_vmixer_gain(chip, 0, 2, 0); - set_vmixer_gain(chip, 1, 3, 0); - set_vmixer_gain(chip, 0, 4, 0); - set_vmixer_gain(chip, 1, 5, 0); - set_vmixer_gain(chip, 0, 6, 0); - set_vmixer_gain(chip, 1, 7, 0); - err = update_vmixer_level(chip); - DE_INIT(("init_hw done\n")); return err; } diff --git a/sound/pci/echoaudio/mia_dsp.c b/sound/pci/echoaudio/mia_dsp.c index 227386602f9..f7abe1b60a1 100644 --- a/sound/pci/echoaudio/mia_dsp.c +++ b/sound/pci/echoaudio/mia_dsp.c @@ -69,18 +69,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) if ((err = init_line_levels(chip))) return err; - /* Default routing of the virtual channels: vchannels 0-3 go to analog - outputs and vchannels 4-7 go to S/PDIF outputs */ - set_vmixer_gain(chip, 0, 0, 0); - set_vmixer_gain(chip, 1, 1, 0); - set_vmixer_gain(chip, 0, 2, 0); - set_vmixer_gain(chip, 1, 3, 0); - set_vmixer_gain(chip, 2, 4, 0); - set_vmixer_gain(chip, 3, 5, 0); - set_vmixer_gain(chip, 2, 6, 0); - set_vmixer_gain(chip, 3, 7, 0); - err = update_vmixer_level(chip); - DE_INIT(("init_hw done\n")); return err; } -- cgit v1.2.3-70-g09d2 From 10d9e3d99ee8332bb73a3d7f12a8cd8ffab8b136 Mon Sep 17 00:00:00 2001 From: Joonyoung Shim Date: Mon, 16 Mar 2009 21:23:35 +0900 Subject: ASoC: twl4030 - Fix build error CC sound/soc/codecs/twl4030.o sound/soc/codecs/twl4030.c:1400: warning: braces around scalar initializer sound/soc/codecs/twl4030.c:1400: warning: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1401: error: field name not in record or union initializer sound/soc/codecs/twl4030.c:1401: error: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1401: warning: initialization from incompatible pointer type sound/soc/codecs/twl4030.c:1402: error: field name not in record or union initializer sound/soc/codecs/twl4030.c:1402: error: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1402: warning: excess elements in scalar initializer sound/soc/codecs/twl4030.c:1402: warning: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1403: error: field name not in record or union initializer sound/soc/codecs/twl4030.c:1403: error: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1403: warning: excess elements in scalar initializer sound/soc/codecs/twl4030.c:1403: warning: (near initialization for 'twl4030_dai.ops') Signed-off-by: Joonyoung Shim Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 86bb15cc82c..97738e2ece0 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1383,6 +1383,12 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, #define TWL4030_RATES (SNDRV_PCM_RATE_8000_48000) #define TWL4030_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE) +static struct snd_soc_dai_ops twl4030_dai_ops = { + .hw_params = twl4030_hw_params, + .set_sysclk = twl4030_set_dai_sysclk, + .set_fmt = twl4030_set_dai_fmt, +}; + struct snd_soc_dai twl4030_dai = { .name = "twl4030", .playback = { @@ -1397,11 +1403,7 @@ struct snd_soc_dai twl4030_dai = { .channels_max = 2, .rates = TWL4030_RATES, .formats = TWL4030_FORMATS,}, - .ops = { - .hw_params = twl4030_hw_params, - .set_sysclk = twl4030_set_dai_sysclk, - .set_fmt = twl4030_set_dai_fmt, - } + .ops = &twl4030_dai_ops, }; EXPORT_SYMBOL_GPL(twl4030_dai); -- cgit v1.2.3-70-g09d2 From f2a5d6a2ea2fa24573a8ce7ea7a7a2cce42e3588 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 16 Mar 2009 14:02:07 +0000 Subject: ASoC: Fix some missing dai_ops conversions Signed-off-by: Mark Brown --- sound/soc/s3c24xx/s3c64xx-i2s.c | 8 +++++--- sound/soc/sh/hac.c | 12 ++++++------ 2 files changed, 11 insertions(+), 9 deletions(-) diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c index 6e1e85dc1ff..33c5de7e255 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c @@ -177,6 +177,10 @@ static int s3c64xx_i2s_probe(struct platform_device *pdev, #define S3C64XX_I2S_FMTS \ (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE) +static struct snd_soc_dai_ops s3c64xx_i2s_dai_ops = { + .set_sysclk = s3c64xx_i2s_set_sysclk, +}; + struct snd_soc_dai s3c64xx_i2s_dai = { .name = "s3c64xx-i2s", .id = 0, @@ -193,9 +197,7 @@ struct snd_soc_dai s3c64xx_i2s_dai = { .rates = S3C64XX_I2S_RATES, .formats = S3C64XX_I2S_FMTS, }, - .ops = { - .set_sysclk = s3c64xx_i2s_set_sysclk, - }, + .ops = &s3c64xx_i2s_dai_ops, }; EXPORT_SYMBOL_GPL(s3c64xx_i2s_dai); diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c index eab31838bad..41db75af3c6 100644 --- a/sound/soc/sh/hac.c +++ b/sound/soc/sh/hac.c @@ -267,6 +267,10 @@ static int hac_hw_params(struct snd_pcm_substream *substream, #define AC97_FMTS \ SNDRV_PCM_FMTBIT_S16_LE +static struct snd_soc_dai_ops hac_dai_ops = { + .hw_params = hac_hw_params, +}; + struct snd_soc_dai sh4_hac_dai[] = { { .name = "HAC0", @@ -284,9 +288,7 @@ struct snd_soc_dai sh4_hac_dai[] = { .channels_min = 2, .channels_max = 2, }, - .ops = { - .hw_params = hac_hw_params, - }, + .ops = &hac_dai_ops, }, #ifdef CONFIG_CPU_SUBTYPE_SH7760 { @@ -305,9 +307,7 @@ struct snd_soc_dai sh4_hac_dai[] = { .channels_min = 2, .channels_max = 2, }, - .ops = { - .hw_params = hac_hw_params, - }, + .ops = &hac_dai_ops, }, #endif -- cgit v1.2.3-70-g09d2 From 852fd9e50f62b4ea7afe26eee0710464de4801b8 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 16 Mar 2009 14:13:12 +0000 Subject: ASoC: Each PXA AC97 DAI needs a separate ops Signed-off-by: Mark Brown --- sound/soc/pxa/pxa2xx-ac97.c | 16 ++++++++++++---- 1 file changed, 12 insertions(+), 4 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index cf809049272..01c21c6cdbb 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -164,10 +164,18 @@ static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream, SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000) -static struct snd_soc_dai_ops pxa_ac97_dai_ops = { +static struct snd_soc_dai_ops pxa_ac97_hifi_dai_ops = { .hw_params = pxa2xx_ac97_hw_params, }; +static struct snd_soc_dai_ops pxa_ac97_aux_dai_ops = { + .hw_params = pxa2xx_ac97_hw_aux_params, +}; + +static struct snd_soc_dai_ops pxa_ac97_mic_dai_ops = { + .hw_params = pxa2xx_ac97_hw_mic_params, +}; + /* * There is only 1 physical AC97 interface for pxa2xx, but it * has extra fifo's that can be used for aux DACs and ADCs. @@ -193,7 +201,7 @@ struct snd_soc_dai pxa_ac97_dai[] = { .channels_max = 2, .rates = PXA2XX_AC97_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = &pxa_ac97_dai_ops, + .ops = &pxa_ac97_hifi_dai_ops, }, { .name = "pxa2xx-ac97-aux", @@ -211,7 +219,7 @@ struct snd_soc_dai pxa_ac97_dai[] = { .channels_max = 1, .rates = PXA2XX_AC97_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = &pxa_ac97_dai_ops, + .ops = &pxa_ac97_aux_dai_ops, }, { .name = "pxa2xx-ac97-mic", @@ -223,7 +231,7 @@ struct snd_soc_dai pxa_ac97_dai[] = { .channels_max = 1, .rates = PXA2XX_AC97_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = &pxa_ac97_dai_ops, + .ops = &pxa_ac97_mic_dai_ops, }, }; -- cgit v1.2.3-70-g09d2 From b8dbed0f095263b9ced5bd2e6d54743a7fa13f1b Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Mon, 16 Mar 2009 14:56:58 +0100 Subject: ALSA: snd-hda-intel: Fix ALC662/ALC663 Beep Amplifier Index ALC662/663 codecs have Beep Amplifier Index 0x04 not 0x05 in 0x0b NID. Confirmed by testing on real hardware. Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b794cba494c..672103d84ff 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -16951,7 +16951,7 @@ static int patch_alc662(struct hda_codec *codec) if (!spec->cap_mixer) set_capture_mixer(spec); - set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); + set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT); spec->vmaster_nid = 0x02; -- cgit v1.2.3-70-g09d2 From b9591448e5160ccd353d8547ade018cfdf2b3e09 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 16 Mar 2009 15:25:00 +0100 Subject: ALSA: hda - Fix ALC662 beep again The previous commit breaks the (digital-) beep on ALC662. ALC662 has the connection index 0x05 while ALC662 and ALC272 have the index 0x04 for the beep widget. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 672103d84ff..5ad0f8d72dd 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -16951,7 +16951,10 @@ static int patch_alc662(struct hda_codec *codec) if (!spec->cap_mixer) set_capture_mixer(spec); - set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT); + if (codec->vendor_id == 0x10ec0662) + set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); + else + set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT); spec->vmaster_nid = 0x02; -- cgit v1.2.3-70-g09d2 From d2314e0e27566f8830ebed3587cc049e07e6a4ee Mon Sep 17 00:00:00 2001 From: Atsushi Nemoto Date: Mon, 16 Mar 2009 23:26:20 +0900 Subject: ASoC: Only deregister AC97 dev if it's name was not "AC97" The commit 14fa43f53ff3a9c3d8b9662574b7369812a31a97 ("ASoC: Only register AC97 bus if it's not done already") added a condition for calling of soc_ac97_dev_register() but not added for calling of soc_ac97_dev_unregister(). This patch adds same condition for soc_ac97_dev_unregister(). Without this fix, kernel crashes when unloading an asoc driver. Signed-off-by: Atsushi Nemoto Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 16518329f6b..6e710f705a7 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1432,7 +1432,8 @@ void snd_soc_free_pcms(struct snd_soc_device *socdev) #ifdef CONFIG_SND_SOC_AC97_BUS for (i = 0; i < codec->num_dai; i++) { codec_dai = &codec->dai[i]; - if (codec_dai->ac97_control && codec->ac97) { + if (codec_dai->ac97_control && codec->ac97 && + strcmp(codec->name, "AC97") != 0) { soc_ac97_dev_unregister(codec); goto free_card; } -- cgit v1.2.3-70-g09d2 From 323a59613e5be6094c93261486de48a08d3a53f2 Mon Sep 17 00:00:00 2001 From: Dmitry Artamonow Date: Fri, 13 Mar 2009 01:03:49 +0100 Subject: ALSA: drop outdated and broken sa11xx-uda1341 driver It depends on L3 support from 2.4 kernel (CONFIG_L3) that never got merged into mainline. Since there's no way to use it on any of supported machines (iPaq h3100 or h3600), better drop it for now. It can be reimplemented later using ASoC infrastructure (there's already a driver for uda1341 codec in mainline, so only CPU and machine parts need to be written). Signed-off-by: Dmitry Artamonow Cc: Russell King Signed-off-by: Takashi Iwai --- include/sound/uda1341.h | 126 ------ sound/arm/Kconfig | 11 - sound/arm/Makefile | 3 - sound/arm/sa11xx-uda1341.c | 984 --------------------------------------------- sound/i2c/Makefile | 2 - sound/i2c/l3/Makefile | 8 - sound/i2c/l3/uda1341.c | 935 ------------------------------------------ 7 files changed, 2069 deletions(-) delete mode 100644 include/sound/uda1341.h delete mode 100644 sound/arm/sa11xx-uda1341.c delete mode 100644 sound/i2c/l3/Makefile delete mode 100644 sound/i2c/l3/uda1341.c diff --git a/include/sound/uda1341.h b/include/sound/uda1341.h deleted file mode 100644 index 110d5dc3a2b..00000000000 --- a/include/sound/uda1341.h +++ /dev/null @@ -1,126 +0,0 @@ -/* - * linux/include/linux/l3/uda1341.h - * - * Philips UDA1341 mixer device driver for ALSA - * - * Copyright (c) 2002 Tomas Kasparek - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License. - * - * History: - * - * 2002-03-13 Tomas Kasparek Initial release - based on uda1341.h from OSS - * 2002-03-30 Tomas Kasparek Proc filesystem support, complete mixer and DSP - * features support - */ - -#define UDA1341_ALSA_NAME "snd-uda1341" - -/* - * Default rate set after inicialization - */ -#define AUDIO_RATE_DEFAULT 44100 - -/* - * UDA1341 L3 address and command types - */ -#define UDA1341_L3ADDR 5 -#define UDA1341_DATA0 (UDA1341_L3ADDR << 2 | 0) -#define UDA1341_DATA1 (UDA1341_L3ADDR << 2 | 1) -#define UDA1341_STATUS (UDA1341_L3ADDR << 2 | 2) - -enum uda1341_onoff { - OFF=0, - ON, -}; - -enum uda1341_format { - I2S=0, - LSB16, - LSB18, - LSB20, - MSB, - LSB16MSB, - LSB18MSB, - LSB20MSB, -}; - -enum uda1341_fs { - F512=0, - F384, - F256, - Funused, -}; - -enum uda1341_peak { - BEFORE=0, - AFTER, -}; - -enum uda1341_filter { - FLAT=0, - MIN, - MIN2, - MAX, -}; - -enum uda1341_mixer { - DOUBLE, - LINE, - MIC, - MIXER, -}; - -enum uda1341_deemp { - NONE, - D32, - D44, - D48, -}; - -enum uda1341_config { - CMD_READ_REG = 0, - CMD_RESET, - CMD_FS, - CMD_FORMAT, - CMD_OGAIN, - CMD_IGAIN, - CMD_DAC, - CMD_ADC, - CMD_VOLUME, - CMD_BASS, - CMD_TREBBLE, - CMD_PEAK, - CMD_DEEMP, - CMD_MUTE, - CMD_FILTER, - CMD_CH1, - CMD_CH2, - CMD_MIC, - CMD_MIXER, - CMD_AGC, - CMD_IG, - CMD_AGC_TIME, - CMD_AGC_LEVEL, -#ifdef CONFIG_PM - CMD_SUSPEND, - CMD_RESUME, -#endif - CMD_LAST, -}; - -enum write_through { - //used in update_bits (write_cfg) to avoid l3_write - just update local copy of regs. - REGS_ONLY=0, - //update local regs and write value to uda1341 - do l3_write - FLUSH, -}; - -int __init snd_chip_uda1341_mixer_new(struct snd_card *card, struct l3_client **clnt); - -/* - * Local variables: - * indent-tabs-mode: t - * End: - */ diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig index f8e6de48d81..885683a3b0b 100644 --- a/sound/arm/Kconfig +++ b/sound/arm/Kconfig @@ -11,17 +11,6 @@ menuconfig SND_ARM if SND_ARM -config SND_SA11XX_UDA1341 - tristate "SA11xx UDA1341TS driver (iPaq H3600)" - depends on ARCH_SA1100 && L3 - select SND_PCM - help - Say Y here if you have a Compaq iPaq H3x00 handheld computer - and want to use its Philips UDA 1341 audio chip. - - To compile this driver as a module, choose M here: the module - will be called snd-sa11xx-uda1341. - config SND_ARMAACI tristate "ARM PrimeCell PL041 AC Link support" depends on ARM_AMBA diff --git a/sound/arm/Makefile b/sound/arm/Makefile index 2054de11de8..5a549ed6c8a 100644 --- a/sound/arm/Makefile +++ b/sound/arm/Makefile @@ -2,9 +2,6 @@ # Makefile for ALSA # -obj-$(CONFIG_SND_SA11XX_UDA1341) += snd-sa11xx-uda1341.o -snd-sa11xx-uda1341-objs := sa11xx-uda1341.o - obj-$(CONFIG_SND_ARMAACI) += snd-aaci.o snd-aaci-objs := aaci.o devdma.o diff --git a/sound/arm/sa11xx-uda1341.c b/sound/arm/sa11xx-uda1341.c deleted file mode 100644 index 7101d3d8bae..00000000000 --- a/sound/arm/sa11xx-uda1341.c +++ /dev/null @@ -1,984 +0,0 @@ -/* - * Driver for Philips UDA1341TS on Compaq iPAQ H3600 soundcard - * Copyright (C) 2002 Tomas Kasparek - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License. - * - * History: - * - * 2002-03-13 Tomas Kasparek initial release - based on h3600-uda1341.c from OSS - * 2002-03-20 Tomas Kasparek playback over ALSA is working - * 2002-03-28 Tomas Kasparek playback over OSS emulation is working - * 2002-03-29 Tomas Kasparek basic capture is working (native ALSA) - * 2002-03-29 Tomas Kasparek capture is working (OSS emulation) - * 2002-04-04 Tomas Kasparek better rates handling (allow non-standard rates) - * 2003-02-14 Brian Avery fixed full duplex mode, other updates - * 2003-02-20 Tomas Kasparek merged updates by Brian (except HAL) - * 2003-04-19 Jaroslav Kysela recoded DMA stuff to follow 2.4.18rmk3-hh24 kernel - * working suspend and resume - * 2003-04-28 Tomas Kasparek updated work by Jaroslav to compile it under 2.5.x again - * merged HAL layer (patches from Brian) - */ - -/*************************************************************************************************** -* -* To understand what Alsa Drivers should be doing look at "Writing an Alsa Driver" by Takashi Iwai -* available in the Alsa doc section on the website -* -* A few notes to make things clearer. The UDA1341 is hooked up to Serial port 4 on the SA1100. -* We are using SSP mode to talk to the UDA1341. The UDA1341 bit & wordselect clocks are generated -* by this UART. Unfortunately, the clock only runs if the transmit buffer has something in it. -* So, if we are just recording, we feed the transmit DMA stream a bunch of 0x0000 so that the -* transmit buffer is full and the clock keeps going. The zeroes come from FLUSH_BASE_PHYS which -* is a mem loc that always decodes to 0's w/ no off chip access. -* -* Some alsa terminology: -* frame => num_channels * sample_size e.g stereo 16 bit is 2 * 16 = 32 bytes -* period => the least number of bytes that will generate an interrupt e.g. we have a 1024 byte -* buffer and 4 periods in the runtime structure this means we'll get an int every 256 -* bytes or 4 times per buffer. -* A number of the sizes are in frames rather than bytes, use frames_to_bytes and -* bytes_to_frames to convert. The easiest way to tell the units is to look at the -* type i.e. runtime-> buffer_size is in frames and its type is snd_pcm_uframes_t -* -* Notes about the pointer fxn: -* The pointer fxn needs to return the offset into the dma buffer in frames. -* Interrupts must be blocked before calling the dma_get_pos fxn to avoid race with interrupts. -* -* Notes about pause/resume -* Implementing this would be complicated so it's skipped. The problem case is: -* A full duplex connection is going, then play is paused. At this point you need to start xmitting -* 0's to keep the record active which means you cant just freeze the dma and resume it later you'd -* need to save off the dma info, and restore it properly on a resume. Yeach! -* -* Notes about transfer methods: -* The async write calls fail. I probably need to implement something else to support them? -* -***************************************************************************************************/ - -#include -#include -#include -#include -#include -#include -#include -#include -#include - -#ifdef CONFIG_PM -#include -#endif - -#include -#include -#include -#include - -#include -#include -#include - -#include - -#undef DEBUG_MODE -#undef DEBUG_FUNCTION_NAMES -#include - -/* - * FIXME: Is this enough as autodetection of 2.4.X-rmkY-hhZ kernels? - * We use DMA stuff from 2.4.18-rmk3-hh24 here to be able to compile this - * module for Familiar 0.6.1 - */ - -/* {{{ Type definitions */ - -MODULE_AUTHOR("Tomas Kasparek "); -MODULE_LICENSE("GPL"); -MODULE_DESCRIPTION("SA1100/SA1111 + UDA1341TS driver for ALSA"); -MODULE_SUPPORTED_DEVICE("{{UDA1341,iPAQ H3600 UDA1341TS}}"); - -static char *id; /* ID for this card */ - -module_param(id, charp, 0444); -MODULE_PARM_DESC(id, "ID string for SA1100/SA1111 + UDA1341TS soundcard."); - -struct audio_stream { - char *id; /* identification string */ - int stream_id; /* numeric identification */ - dma_device_t dma_dev; /* device identifier for DMA */ -#ifdef HH_VERSION - dmach_t dmach; /* dma channel identification */ -#else - dma_regs_t *dma_regs; /* points to our DMA registers */ -#endif - unsigned int active:1; /* we are using this stream for transfer now */ - int period; /* current transfer period */ - int periods; /* current count of periods registerd in the DMA engine */ - int tx_spin; /* are we recoding - flag used to do DMA trans. for sync */ - unsigned int old_offset; - spinlock_t dma_lock; /* for locking in DMA operations (see dma-sa1100.c in the kernel) */ - struct snd_pcm_substream *stream; -}; - -struct sa11xx_uda1341 { - struct snd_card *card; - struct l3_client *uda1341; - struct snd_pcm *pcm; - long samplerate; - struct audio_stream s[2]; /* playback & capture */ -}; - -static unsigned int rates[] = { - 8000, 10666, 10985, 14647, - 16000, 21970, 22050, 24000, - 29400, 32000, 44100, 48000, -}; - -static struct snd_pcm_hw_constraint_list hw_constraints_rates = { - .count = ARRAY_SIZE(rates), - .list = rates, - .mask = 0, -}; - -static struct platform_device *device; - -/* }}} */ - -/* {{{ Clock and sample rate stuff */ - -/* - * Stop-gap solution until rest of hh.org HAL stuff is merged. - */ -#define GPIO_H3600_CLK_SET0 GPIO_GPIO (12) -#define GPIO_H3600_CLK_SET1 GPIO_GPIO (13) - -#ifdef CONFIG_SA1100_H3XXX -#define clr_sa11xx_uda1341_egpio(x) clr_h3600_egpio(x) -#define set_sa11xx_uda1341_egpio(x) set_h3600_egpio(x) -#else -#error This driver could serve H3x00 handhelds only! -#endif - -static void sa11xx_uda1341_set_audio_clock(long val) -{ - switch (val) { - case 24000: case 32000: case 48000: /* 00: 12.288 MHz */ - GPCR = GPIO_H3600_CLK_SET0 | GPIO_H3600_CLK_SET1; - break; - - case 22050: case 29400: case 44100: /* 01: 11.2896 MHz */ - GPSR = GPIO_H3600_CLK_SET0; - GPCR = GPIO_H3600_CLK_SET1; - break; - - case 8000: case 10666: case 16000: /* 10: 4.096 MHz */ - GPCR = GPIO_H3600_CLK_SET0; - GPSR = GPIO_H3600_CLK_SET1; - break; - - case 10985: case 14647: case 21970: /* 11: 5.6245 MHz */ - GPSR = GPIO_H3600_CLK_SET0 | GPIO_H3600_CLK_SET1; - break; - } -} - -static void sa11xx_uda1341_set_samplerate(struct sa11xx_uda1341 *sa11xx_uda1341, long rate) -{ - int clk_div = 0; - int clk=0; - - /* We don't want to mess with clocks when frames are in flight */ - Ser4SSCR0 &= ~SSCR0_SSE; - /* wait for any frame to complete */ - udelay(125); - - /* - * We have the following clock sources: - * 4.096 MHz, 5.6245 MHz, 11.2896 MHz, 12.288 MHz - * Those can be divided either by 256, 384 or 512. - * This makes up 12 combinations for the following samplerates... - */ - if (rate >= 48000) - rate = 48000; - else if (rate >= 44100) - rate = 44100; - else if (rate >= 32000) - rate = 32000; - else if (rate >= 29400) - rate = 29400; - else if (rate >= 24000) - rate = 24000; - else if (rate >= 22050) - rate = 22050; - else if (rate >= 21970) - rate = 21970; - else if (rate >= 16000) - rate = 16000; - else if (rate >= 14647) - rate = 14647; - else if (rate >= 10985) - rate = 10985; - else if (rate >= 10666) - rate = 10666; - else - rate = 8000; - - /* Set the external clock generator */ - - sa11xx_uda1341_set_audio_clock(rate); - - /* Select the clock divisor */ - switch (rate) { - case 8000: - case 10985: - case 22050: - case 24000: - clk = F512; - clk_div = SSCR0_SerClkDiv(16); - break; - case 16000: - case 21970: - case 44100: - case 48000: - clk = F256; - clk_div = SSCR0_SerClkDiv(8); - break; - case 10666: - case 14647: - case 29400: - case 32000: - clk = F384; - clk_div = SSCR0_SerClkDiv(12); - break; - } - - /* FMT setting should be moved away when other FMTs are added (FIXME) */ - l3_command(sa11xx_uda1341->uda1341, CMD_FORMAT, (void *)LSB16); - - l3_command(sa11xx_uda1341->uda1341, CMD_FS, (void *)clk); - Ser4SSCR0 = (Ser4SSCR0 & ~0xff00) + clk_div + SSCR0_SSE; - sa11xx_uda1341->samplerate = rate; -} - -/* }}} */ - -/* {{{ HW init and shutdown */ - -static void sa11xx_uda1341_audio_init(struct sa11xx_uda1341 *sa11xx_uda1341) -{ - unsigned long flags; - - /* Setup DMA stuff */ - sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].id = "UDA1341 out"; - sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].stream_id = SNDRV_PCM_STREAM_PLAYBACK; - sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].dma_dev = DMA_Ser4SSPWr; - - sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].id = "UDA1341 in"; - sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].stream_id = SNDRV_PCM_STREAM_CAPTURE; - sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].dma_dev = DMA_Ser4SSPRd; - - /* Initialize the UDA1341 internal state */ - - /* Setup the uarts */ - local_irq_save(flags); - GAFR |= (GPIO_SSP_CLK); - GPDR &= ~(GPIO_SSP_CLK); - Ser4SSCR0 = 0; - Ser4SSCR0 = SSCR0_DataSize(16) + SSCR0_TI + SSCR0_SerClkDiv(8); - Ser4SSCR1 = SSCR1_SClkIactL + SSCR1_SClk1P + SSCR1_ExtClk; - Ser4SSCR0 |= SSCR0_SSE; - local_irq_restore(flags); - - /* Enable the audio power */ - - clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET); - set_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON); - set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE); - - /* Wait for the UDA1341 to wake up */ - mdelay(1); //FIXME - was removed by Perex - Why? - - /* Initialize the UDA1341 internal state */ - l3_open(sa11xx_uda1341->uda1341); - - /* external clock configuration (after l3_open - regs must be initialized */ - sa11xx_uda1341_set_samplerate(sa11xx_uda1341, sa11xx_uda1341->samplerate); - - /* Wait for the UDA1341 to wake up */ - set_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET); - mdelay(1); - - /* make the left and right channels unswapped (flip the WS latch) */ - Ser4SSDR = 0; - - clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE); -} - -static void sa11xx_uda1341_audio_shutdown(struct sa11xx_uda1341 *sa11xx_uda1341) -{ - /* mute on */ - set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE); - - /* disable the audio power and all signals leading to the audio chip */ - l3_close(sa11xx_uda1341->uda1341); - Ser4SSCR0 = 0; - clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET); - - /* power off and mute off */ - /* FIXME - is muting off necesary??? */ - - clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON); - clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE); -} - -/* }}} */ - -/* {{{ DMA staff */ - -/* - * these are the address and sizes used to fill the xmit buffer - * so we can get a clock in record only mode - */ -#define FORCE_CLOCK_ADDR (dma_addr_t)FLUSH_BASE_PHYS -#define FORCE_CLOCK_SIZE 4096 // was 2048 - -// FIXME Why this value exactly - wrote comment -#define DMA_BUF_SIZE 8176 /* <= MAX_DMA_SIZE from asm/arch-sa1100/dma.h */ - -#ifdef HH_VERSION - -static int audio_dma_request(struct audio_stream *s, void (*callback)(void *, int)) -{ - int ret; - - ret = sa1100_request_dma(&s->dmach, s->id, s->dma_dev); - if (ret < 0) { - printk(KERN_ERR "unable to grab audio dma 0x%x\n", s->dma_dev); - return ret; - } - sa1100_dma_set_callback(s->dmach, callback); - return 0; -} - -static inline void audio_dma_free(struct audio_stream *s) -{ - sa1100_free_dma(s->dmach); - s->dmach = -1; -} - -#else - -static int audio_dma_request(struct audio_stream *s, void (*callback)(void *)) -{ - int ret; - - ret = sa1100_request_dma(s->dma_dev, s->id, callback, s, &s->dma_regs); - if (ret < 0) - printk(KERN_ERR "unable to grab audio dma 0x%x\n", s->dma_dev); - return ret; -} - -static void audio_dma_free(struct audio_stream *s) -{ - sa1100_free_dma(s->dma_regs); - s->dma_regs = 0; -} - -#endif - -static u_int audio_get_dma_pos(struct audio_stream *s) -{ - struct snd_pcm_substream *substream = s->stream; - struct snd_pcm_runtime *runtime = substream->runtime; - unsigned int offset; - unsigned long flags; - dma_addr_t addr; - - // this must be called w/ interrupts locked out see dma-sa1100.c in the kernel - spin_lock_irqsave(&s->dma_lock, flags); -#ifdef HH_VERSION - sa1100_dma_get_current(s->dmach, NULL, &addr); -#else - addr = sa1100_get_dma_pos((s)->dma_regs); -#endif - offset = addr - runtime->dma_addr; - spin_unlock_irqrestore(&s->dma_lock, flags); - - offset = bytes_to_frames(runtime,offset); - if (offset >= runtime->buffer_size) - offset = 0; - - return offset; -} - -/* - * this stops the dma and clears the dma ptrs - */ -static void audio_stop_dma(struct audio_stream *s) -{ - unsigned long flags; - - spin_lock_irqsave(&s->dma_lock, flags); - s->active = 0; - s->period = 0; - /* this stops the dma channel and clears the buffer ptrs */ -#ifdef HH_VERSION - sa1100_dma_flush_all(s->dmach); -#else - sa1100_clear_dma(s->dma_regs); -#endif - spin_unlock_irqrestore(&s->dma_lock, flags); -} - -static void audio_process_dma(struct audio_stream *s) -{ - struct snd_pcm_substream *substream = s->stream; - struct snd_pcm_runtime *runtime; - unsigned int dma_size; - unsigned int offset; - int ret; - - /* we are requested to process synchronization DMA transfer */ - if (s->tx_spin) { - if (snd_BUG_ON(s->stream_id != SNDRV_PCM_STREAM_PLAYBACK)) - return; - /* fill the xmit dma buffers and return */ -#ifdef HH_VERSION - sa1100_dma_set_spin(s->dmach, FORCE_CLOCK_ADDR, FORCE_CLOCK_SIZE); -#else - while (1) { - ret = sa1100_start_dma(s->dma_regs, FORCE_CLOCK_ADDR, FORCE_CLOCK_SIZE); - if (ret) - return; - } -#endif - return; - } - - /* must be set here - only valid for running streams, not for forced_clock dma fills */ - runtime = substream->runtime; - while (s->active && s->periods < runtime->periods) { - dma_size = frames_to_bytes(runtime, runtime->period_size); - if (s->old_offset) { - /* a little trick, we need resume from old position */ - offset = frames_to_bytes(runtime, s->old_offset - 1); - s->old_offset = 0; - s->periods = 0; - s->period = offset / dma_size; - offset %= dma_size; - dma_size = dma_size - offset; - if (!dma_size) - continue; /* special case */ - } else { - offset = dma_size * s->period; - snd_BUG_ON(dma_size > DMA_BUF_SIZE); - } -#ifdef HH_VERSION - ret = sa1100_dma_queue_buffer(s->dmach, s, runtime->dma_addr + offset, dma_size); - if (ret) - return; //FIXME -#else - ret = sa1100_start_dma((s)->dma_regs, runtime->dma_addr + offset, dma_size); - if (ret) { - printk(KERN_ERR "audio_process_dma: cannot queue DMA buffer (%i)\n", ret); - return; - } -#endif - - s->period++; - s->period %= runtime->periods; - s->periods++; - } -} - -#ifdef HH_VERSION -static void audio_dma_callback(void *data, int size) -#else -static void audio_dma_callback(void *data) -#endif -{ - struct audio_stream *s = data; - - /* - * If we are getting a callback for an active stream then we inform - * the PCM middle layer we've finished a period - */ - if (s->active) - snd_pcm_period_elapsed(s->stream); - - spin_lock(&s->dma_lock); - if (!s->tx_spin && s->periods > 0) - s->periods--; - audio_process_dma(s); - spin_unlock(&s->dma_lock); -} - -/* }}} */ - -/* {{{ PCM setting */ - -/* {{{ trigger & timer */ - -static int snd_sa11xx_uda1341_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream); - int stream_id = substream->pstr->stream; - struct audio_stream *s = &chip->s[stream_id]; - struct audio_stream *s1 = &chip->s[stream_id ^ 1]; - int err = 0; - - /* note local interrupts are already disabled in the midlevel code */ - spin_lock(&s->dma_lock); - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - /* now we need to make sure a record only stream has a clock */ - if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) { - /* we need to force fill the xmit DMA with zeros */ - s1->tx_spin = 1; - audio_process_dma(s1); - } - /* this case is when you were recording then you turn on a - * playback stream so we stop (also clears it) the dma first, - * clear the sync flag and then we let it turned on - */ - else { - s->tx_spin = 0; - } - - /* requested stream startup */ - s->active = 1; - audio_process_dma(s); - break; - case SNDRV_PCM_TRIGGER_STOP: - /* requested stream shutdown */ - audio_stop_dma(s); - - /* - * now we need to make sure a record only stream has a clock - * so if we're stopping a playback with an active capture - * we need to turn the 0 fill dma on for the xmit side - */ - if (stream_id == SNDRV_PCM_STREAM_PLAYBACK && s1->active) { - /* we need to force fill the xmit DMA with zeros */ - s->tx_spin = 1; - audio_process_dma(s); - } - /* - * we killed a capture only stream, so we should also kill - * the zero fill transmit - */ - else { - if (s1->tx_spin) { - s1->tx_spin = 0; - audio_stop_dma(s1); - } - } - - break; - case SNDRV_PCM_TRIGGER_SUSPEND: - s->active = 0; -#ifdef HH_VERSION - sa1100_dma_stop(s->dmach); -#else - //FIXME - DMA API -#endif - s->old_offset = audio_get_dma_pos(s) + 1; -#ifdef HH_VERSION - sa1100_dma_flush_all(s->dmach); -#else - //FIXME - DMA API -#endif - s->periods = 0; - break; - case SNDRV_PCM_TRIGGER_RESUME: - s->active = 1; - s->tx_spin = 0; - audio_process_dma(s); - if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) { - s1->tx_spin = 1; - audio_process_dma(s1); - } - break; - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: -#ifdef HH_VERSION - sa1100_dma_stop(s->dmach); -#else - //FIXME - DMA API -#endif - s->active = 0; - if (stream_id == SNDRV_PCM_STREAM_PLAYBACK) { - if (s1->active) { - s->tx_spin = 1; - s->old_offset = audio_get_dma_pos(s) + 1; -#ifdef HH_VERSION - sa1100_dma_flush_all(s->dmach); -#else - //FIXME - DMA API -#endif - audio_process_dma(s); - } - } else { - if (s1->tx_spin) { - s1->tx_spin = 0; -#ifdef HH_VERSION - sa1100_dma_flush_all(s1->dmach); -#else - //FIXME - DMA API -#endif - } - } - break; - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - s->active = 1; - if (s->old_offset) { - s->tx_spin = 0; - audio_process_dma(s); - break; - } - if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) { - s1->tx_spin = 1; - audio_process_dma(s1); - } -#ifdef HH_VERSION - sa1100_dma_resume(s->dmach); -#else - //FIXME - DMA API -#endif - break; - default: - err = -EINVAL; - break; - } - spin_unlock(&s->dma_lock); - return err; -} - -static int snd_sa11xx_uda1341_prepare(struct snd_pcm_substream *substream) -{ - struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - struct audio_stream *s = &chip->s[substream->pstr->stream]; - - /* set requested samplerate */ - sa11xx_uda1341_set_samplerate(chip, runtime->rate); - - /* set requestd format when available */ - /* set FMT here !!! FIXME */ - - s->period = 0; - s->periods = 0; - - return 0; -} - -static snd_pcm_uframes_t snd_sa11xx_uda1341_pointer(struct snd_pcm_substream *substream) -{ - struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream); - return audio_get_dma_pos(&chip->s[substream->pstr->stream]); -} - -/* }}} */ - -static struct snd_pcm_hardware snd_sa11xx_uda1341_capture = -{ - .info = (SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME), - .formats = SNDRV_PCM_FMTBIT_S16_LE, - .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ - SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |\ - SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ - SNDRV_PCM_RATE_KNOT), - .rate_min = 8000, - .rate_max = 48000, - .channels_min = 2, - .channels_max = 2, - .buffer_bytes_max = 64*1024, - .period_bytes_min = 64, - .period_bytes_max = DMA_BUF_SIZE, - .periods_min = 2, - .periods_max = 255, - .fifo_size = 0, -}; - -static struct snd_pcm_hardware snd_sa11xx_uda1341_playback = -{ - .info = (SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME), - .formats = SNDRV_PCM_FMTBIT_S16_LE, - .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ - SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |\ - SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ - SNDRV_PCM_RATE_KNOT), - .rate_min = 8000, - .rate_max = 48000, - .channels_min = 2, - .channels_max = 2, - .buffer_bytes_max = 64*1024, - .period_bytes_min = 64, - .period_bytes_max = DMA_BUF_SIZE, - .periods_min = 2, - .periods_max = 255, - .fifo_size = 0, -}; - -static int snd_card_sa11xx_uda1341_open(struct snd_pcm_substream *substream) -{ - struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - int stream_id = substream->pstr->stream; - int err; - - chip->s[stream_id].stream = substream; - - if (stream_id == SNDRV_PCM_STREAM_PLAYBACK) - runtime->hw = snd_sa11xx_uda1341_playback; - else - runtime->hw = snd_sa11xx_uda1341_capture; - if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) - return err; - if ((err = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates)) < 0) - return err; - - return 0; -} - -static int snd_card_sa11xx_uda1341_close(struct snd_pcm_substream *substream) -{ - struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream); - - chip->s[substream->pstr->stream].stream = NULL; - return 0; -} - -/* {{{ HW params & free */ - -static int snd_sa11xx_uda1341_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *hw_params) -{ - - return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); -} - -static int snd_sa11xx_uda1341_hw_free(struct snd_pcm_substream *substream) -{ - return snd_pcm_lib_free_pages(substream); -} - -/* }}} */ - -static struct snd_pcm_ops snd_card_sa11xx_uda1341_playback_ops = { - .open = snd_card_sa11xx_uda1341_open, - .close = snd_card_sa11xx_uda1341_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = snd_sa11xx_uda1341_hw_params, - .hw_free = snd_sa11xx_uda1341_hw_free, - .prepare = snd_sa11xx_uda1341_prepare, - .trigger = snd_sa11xx_uda1341_trigger, - .pointer = snd_sa11xx_uda1341_pointer, -}; - -static struct snd_pcm_ops snd_card_sa11xx_uda1341_capture_ops = { - .open = snd_card_sa11xx_uda1341_open, - .close = snd_card_sa11xx_uda1341_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = snd_sa11xx_uda1341_hw_params, - .hw_free = snd_sa11xx_uda1341_hw_free, - .prepare = snd_sa11xx_uda1341_prepare, - .trigger = snd_sa11xx_uda1341_trigger, - .pointer = snd_sa11xx_uda1341_pointer, -}; - -static int __init snd_card_sa11xx_uda1341_pcm(struct sa11xx_uda1341 *sa11xx_uda1341, int device) -{ - struct snd_pcm *pcm; - int err; - - if ((err = snd_pcm_new(sa11xx_uda1341->card, "UDA1341 PCM", device, 1, 1, &pcm)) < 0) - return err; - - /* - * this sets up our initial buffers and sets the dma_type to isa. - * isa works but I'm not sure why (or if) it's the right choice - * this may be too large, trying it for now - */ - snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_isa_data(), - 64*1024, 64*1024); - - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_card_sa11xx_uda1341_playback_ops); - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_card_sa11xx_uda1341_capture_ops); - pcm->private_data = sa11xx_uda1341; - pcm->info_flags = 0; - strcpy(pcm->name, "UDA1341 PCM"); - - sa11xx_uda1341_audio_init(sa11xx_uda1341); - - /* setup DMA controller */ - audio_dma_request(&sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK], audio_dma_callback); - audio_dma_request(&sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE], audio_dma_callback); - - sa11xx_uda1341->pcm = pcm; - - return 0; -} - -/* }}} */ - -/* {{{ module init & exit */ - -#ifdef CONFIG_PM - -static int snd_sa11xx_uda1341_suspend(struct platform_device *devptr, - pm_message_t state) -{ - struct snd_card *card = platform_get_drvdata(devptr); - struct sa11xx_uda1341 *chip = card->private_data; - - snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); - snd_pcm_suspend_all(chip->pcm); -#ifdef HH_VERSION - sa1100_dma_sleep(chip->s[SNDRV_PCM_STREAM_PLAYBACK].dmach); - sa1100_dma_sleep(chip->s[SNDRV_PCM_STREAM_CAPTURE].dmach); -#else - //FIXME -#endif - l3_command(chip->uda1341, CMD_SUSPEND, NULL); - sa11xx_uda1341_audio_shutdown(chip); - - return 0; -} - -static int snd_sa11xx_uda1341_resume(struct platform_device *devptr) -{ - struct snd_card *card = platform_get_drvdata(devptr); - struct sa11xx_uda1341 *chip = card->private_data; - - sa11xx_uda1341_audio_init(chip); - l3_command(chip->uda1341, CMD_RESUME, NULL); -#ifdef HH_VERSION - sa1100_dma_wakeup(chip->s[SNDRV_PCM_STREAM_PLAYBACK].dmach); - sa1100_dma_wakeup(chip->s[SNDRV_PCM_STREAM_CAPTURE].dmach); -#else - //FIXME -#endif - snd_power_change_state(card, SNDRV_CTL_POWER_D0); - return 0; -} -#endif /* COMFIG_PM */ - -void snd_sa11xx_uda1341_free(struct snd_card *card) -{ - struct sa11xx_uda1341 *chip = card->private_data; - - audio_dma_free(&chip->s[SNDRV_PCM_STREAM_PLAYBACK]); - audio_dma_free(&chip->s[SNDRV_PCM_STREAM_CAPTURE]); -} - -static int __devinit sa11xx_uda1341_probe(struct platform_device *devptr) -{ - int err; - struct snd_card *card; - struct sa11xx_uda1341 *chip; - - /* register the soundcard */ - err = snd_card_create(-1, id, THIS_MODULE, - sizeof(struct sa11xx_uda1341), &card); - if (err < 0) - return err; - - chip = card->private_data; - spin_lock_init(&chip->s[0].dma_lock); - spin_lock_init(&chip->s[1].dma_lock); - - card->private_free = snd_sa11xx_uda1341_free; - chip->card = card; - chip->samplerate = AUDIO_RATE_DEFAULT; - - // mixer - if ((err = snd_chip_uda1341_mixer_new(card, &chip->uda1341))) - goto nodev; - - // PCM - if ((err = snd_card_sa11xx_uda1341_pcm(chip, 0)) < 0) - goto nodev; - - strcpy(card->driver, "UDA1341"); - strcpy(card->shortname, "H3600 UDA1341TS"); - sprintf(card->longname, "Compaq iPAQ H3600 with Philips UDA1341TS"); - - snd_card_set_dev(card, &devptr->dev); - - if ((err = snd_card_register(card)) == 0) { - printk(KERN_INFO "iPAQ audio support initialized\n"); - platform_set_drvdata(devptr, card); - return 0; - } - - nodev: - snd_card_free(card); - return err; -} - -static int __devexit sa11xx_uda1341_remove(struct platform_device *devptr) -{ - snd_card_free(platform_get_drvdata(devptr)); - platform_set_drvdata(devptr, NULL); - return 0; -} - -#define SA11XX_UDA1341_DRIVER "sa11xx_uda1341" - -static struct platform_driver sa11xx_uda1341_driver = { - .probe = sa11xx_uda1341_probe, - .remove = __devexit_p(sa11xx_uda1341_remove), -#ifdef CONFIG_PM - .suspend = snd_sa11xx_uda1341_suspend, - .resume = snd_sa11xx_uda1341_resume, -#endif - .driver = { - .name = SA11XX_UDA1341_DRIVER, - }, -}; - -static int __init sa11xx_uda1341_init(void) -{ - int err; - - if (!machine_is_h3xxx()) - return -ENODEV; - if ((err = platform_driver_register(&sa11xx_uda1341_driver)) < 0) - return err; - device = platform_device_register_simple(SA11XX_UDA1341_DRIVER, -1, NULL, 0); - if (!IS_ERR(device)) { - if (platform_get_drvdata(device)) - return 0; - platform_device_unregister(device); - err = -ENODEV; - } else - err = PTR_ERR(device); - platform_driver_unregister(&sa11xx_uda1341_driver); - return err; -} - -static void __exit sa11xx_uda1341_exit(void) -{ - platform_device_unregister(device); - platform_driver_unregister(&sa11xx_uda1341_driver); -} - -module_init(sa11xx_uda1341_init); -module_exit(sa11xx_uda1341_exit); - -/* }}} */ - -/* - * Local variables: - * indent-tabs-mode: t - * End: - */ diff --git a/sound/i2c/Makefile b/sound/i2c/Makefile index 37970666a45..36879bf8870 100644 --- a/sound/i2c/Makefile +++ b/sound/i2c/Makefile @@ -7,8 +7,6 @@ snd-i2c-objs := i2c.o snd-cs8427-objs := cs8427.o snd-tea6330t-objs := tea6330t.o -obj-$(CONFIG_L3) += l3/ - obj-$(CONFIG_SND) += other/ # Toplevel Module Dependency diff --git a/sound/i2c/l3/Makefile b/sound/i2c/l3/Makefile deleted file mode 100644 index 49455b8dcc0..00000000000 --- a/sound/i2c/l3/Makefile +++ /dev/null @@ -1,8 +0,0 @@ -# -# Makefile for ALSA -# - -snd-uda1341-objs := uda1341.o - -# Module Dependency -obj-$(CONFIG_SND_SA11XX_UDA1341) += snd-uda1341.o diff --git a/sound/i2c/l3/uda1341.c b/sound/i2c/l3/uda1341.c deleted file mode 100644 index 9840eb43648..00000000000 --- a/sound/i2c/l3/uda1341.c +++ /dev/null @@ -1,935 +0,0 @@ -/* - * Philips UDA1341 mixer device driver - * Copyright (c) 2002 Tomas Kasparek - * - * Portions are Copyright (C) 2000 Lernout & Hauspie Speech Products, N.V. - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License. - * - * History: - * - * 2002-03-13 Tomas Kasparek initial release - based on uda1341.c from OSS - * 2002-03-28 Tomas Kasparek basic mixer is working (volume, bass, treble) - * 2002-03-30 Tomas Kasparek proc filesystem support, complete mixer and DSP - * features support - * 2002-04-12 Tomas Kasparek proc interface update, code cleanup - * 2002-05-12 Tomas Kasparek another code cleanup - */ - -#include -#include -#include -#include -#include -#include - -#include - -#include -#include -#include -#include - -#include - -#include - -/* {{{ HW regs definition */ - -#define STAT0 0x00 -#define STAT1 0x80 -#define STAT_MASK 0x80 - -#define DATA0_0 0x00 -#define DATA0_1 0x40 -#define DATA0_2 0x80 -#define DATA_MASK 0xc0 - -#define IS_DATA0(x) ((x) >= data0_0 && (x) <= data0_2) -#define IS_DATA1(x) ((x) == data1) -#define IS_STATUS(x) ((x) == stat0 || (x) == stat1) -#define IS_EXTEND(x) ((x) >= ext0 && (x) <= ext6) - -/* }}} */ - - -static const char *peak_names[] = { - "before", - "after", -}; - -static const char *filter_names[] = { - "flat", - "min", - "min", - "max", -}; - -static const char *mixer_names[] = { - "double differential", - "input channel 1 (line in)", - "input channel 2 (microphone)", - "digital mixer", -}; - -static const char *deemp_names[] = { - "none", - "32 kHz", - "44.1 kHz", - "48 kHz", -}; - -enum uda1341_regs_names { - stat0, - stat1, - data0_0, - data0_1, - data0_2, - data1, - ext0, - ext1, - ext2, - empty, - ext4, - ext5, - ext6, - uda1341_reg_last, -}; - -static const char *uda1341_reg_names[] = { - "stat 0 ", - "stat 1 ", - "data 00", - "data 01", - "data 02", - "data 1 ", - "ext 0", - "ext 1", - "ext 2", - "empty", - "ext 4", - "ext 5", - "ext 6", -}; - -static const int uda1341_enum_items[] = { - 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, - 2, //peak - before/after - 4, //deemp - none/32/44.1/48 - 0, - 4, //filter - flat/min/min/max - 0, 0, 0, - 4, //mixer - differ/line/mic/mixer - 0, 0, 0, 0, 0, -}; - -static const char ** uda1341_enum_names[] = { - NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, - peak_names, //peak - before/after - deemp_names, //deemp - none/32/44.1/48 - NULL, - filter_names, //filter - flat/min/min/max - NULL, NULL, NULL, - mixer_names, //mixer - differ/line/mic/mixer - NULL, NULL, NULL, NULL, NULL, -}; - -typedef int uda1341_cfg[CMD_LAST]; - -struct uda1341 { - int (*write) (struct l3_client *uda1341, unsigned short reg, unsigned short val); - int (*read) (struct l3_client *uda1341, unsigned short reg); - unsigned char regs[uda1341_reg_last]; - int active; - spinlock_t reg_lock; - struct snd_card *card; - uda1341_cfg cfg; -#ifdef CONFIG_PM - unsigned char suspend_regs[uda1341_reg_last]; - uda1341_cfg suspend_cfg; -#endif -}; - -/* transfer 8bit integer into string with binary representation */ -static void int2str_bin8(uint8_t val, char *buf) -{ - const int size = sizeof(val) * 8; - int i; - - for (i= 0; i < size; i++){ - *(buf++) = (val >> (size - 1)) ? '1' : '0'; - val <<= 1; - } - *buf = '\0'; //end the string with zero -} - -/* {{{ HW manipulation routines */ - -static int snd_uda1341_codec_write(struct l3_client *clnt, unsigned short reg, unsigned short val) -{ - struct uda1341 *uda = clnt->driver_data; - unsigned char buf[2] = { 0xc0, 0xe0 }; // for EXT addressing - int err = 0; - - uda->regs[reg] = val; - - if (uda->active) { - if (IS_DATA0(reg)) { - err = l3_write(clnt, UDA1341_DATA0, (const unsigned char *)&val, 1); - } else if (IS_DATA1(reg)) { - err = l3_write(clnt, UDA1341_DATA1, (const unsigned char *)&val, 1); - } else if (IS_STATUS(reg)) { - err = l3_write(clnt, UDA1341_STATUS, (const unsigned char *)&val, 1); - } else if (IS_EXTEND(reg)) { - buf[0] |= (reg - ext0) & 0x7; //EXT address - buf[1] |= val; //EXT data - err = l3_write(clnt, UDA1341_DATA0, (const unsigned char *)buf, 2); - } - } else - printk(KERN_ERR "UDA1341 codec not active!\n"); - return err; -} - -static int snd_uda1341_codec_read(struct l3_client *clnt, unsigned short reg) -{ - unsigned char val; - int err; - - err = l3_read(clnt, reg, &val, 1); - if (err == 1) - // use just 6bits - the rest is address of the reg - return val & 63; - return err < 0 ? err : -EIO; -} - -static inline int snd_uda1341_valid_reg(struct l3_client *clnt, unsigned short reg) -{ - return reg < uda1341_reg_last; -} - -static int snd_uda1341_update_bits(struct l3_client *clnt, unsigned short reg, - unsigned short mask, unsigned short shift, - unsigned short value, int flush) -{ - int change; - unsigned short old, new; - struct uda1341 *uda = clnt->driver_data; - -#if 0 - printk(KERN_DEBUG "update_bits: reg: %s mask: %d shift: %d val: %d\n", - uda1341_reg_names[reg], mask, shift, value); -#endif - - if (!snd_uda1341_valid_reg(clnt, reg)) - return -EINVAL; - spin_lock(&uda->reg_lock); - old = uda->regs[reg]; - new = (old & ~(mask << shift)) | (value << shift); - change = old != new; - if (change) { - if (flush) uda->write(clnt, reg, new); - uda->regs[reg] = new; - } - spin_unlock(&uda->reg_lock); - return change; -} - -static int snd_uda1341_cfg_write(struct l3_client *clnt, unsigned short what, - unsigned short value, int flush) -{ - struct uda1341 *uda = clnt->driver_data; - int ret = 0; -#ifdef CONFIG_PM - int reg; -#endif - -#if 0 - printk(KERN_DEBUG "cfg_write what: %d value: %d\n", what, value); -#endif - - uda->cfg[what] = value; - - switch(what) { - case CMD_RESET: - ret = snd_uda1341_update_bits(clnt, data0_2, 1, 2, 1, flush); // MUTE - ret = snd_uda1341_update_bits(clnt, stat0, 1, 6, 1, flush); // RESET - ret = snd_uda1341_update_bits(clnt, stat0, 1, 6, 0, flush); // RESTORE - uda->cfg[CMD_RESET]=0; - break; - case CMD_FS: - ret = snd_uda1341_update_bits(clnt, stat0, 3, 4, value, flush); - break; - case CMD_FORMAT: - ret = snd_uda1341_update_bits(clnt, stat0, 7, 1, value, flush); - break; - case CMD_OGAIN: - ret = snd_uda1341_update_bits(clnt, stat1, 1, 6, value, flush); - break; - case CMD_IGAIN: - ret = snd_uda1341_update_bits(clnt, stat1, 1, 5, value, flush); - break; - case CMD_DAC: - ret = snd_uda1341_update_bits(clnt, stat1, 1, 0, value, flush); - break; - case CMD_ADC: - ret = snd_uda1341_update_bits(clnt, stat1, 1, 1, value, flush); - break; - case CMD_VOLUME: - ret = snd_uda1341_update_bits(clnt, data0_0, 63, 0, value, flush); - break; - case CMD_BASS: - ret = snd_uda1341_update_bits(clnt, data0_1, 15, 2, value, flush); - break; - case CMD_TREBBLE: - ret = snd_uda1341_update_bits(clnt, data0_1, 3, 0, value, flush); - break; - case CMD_PEAK: - ret = snd_uda1341_update_bits(clnt, data0_2, 1, 5, value, flush); - break; - case CMD_DEEMP: - ret = snd_uda1341_update_bits(clnt, data0_2, 3, 3, value, flush); - break; - case CMD_MUTE: - ret = snd_uda1341_update_bits(clnt, data0_2, 1, 2, value, flush); - break; - case CMD_FILTER: - ret = snd_uda1341_update_bits(clnt, data0_2, 3, 0, value, flush); - break; - case CMD_CH1: - ret = snd_uda1341_update_bits(clnt, ext0, 31, 0, value, flush); - break; - case CMD_CH2: - ret = snd_uda1341_update_bits(clnt, ext1, 31, 0, value, flush); - break; - case CMD_MIC: - ret = snd_uda1341_update_bits(clnt, ext2, 7, 2, value, flush); - break; - case CMD_MIXER: - ret = snd_uda1341_update_bits(clnt, ext2, 3, 0, value, flush); - break; - case CMD_AGC: - ret = snd_uda1341_update_bits(clnt, ext4, 1, 4, value, flush); - break; - case CMD_IG: - ret = snd_uda1341_update_bits(clnt, ext4, 3, 0, value & 0x3, flush); - ret = snd_uda1341_update_bits(clnt, ext5, 31, 0, value >> 2, flush); - break; - case CMD_AGC_TIME: - ret = snd_uda1341_update_bits(clnt, ext6, 7, 2, value, flush); - break; - case CMD_AGC_LEVEL: - ret = snd_uda1341_update_bits(clnt, ext6, 3, 0, value, flush); - break; -#ifdef CONFIG_PM - case CMD_SUSPEND: - for (reg = stat0; reg < uda1341_reg_last; reg++) - uda->suspend_regs[reg] = uda->regs[reg]; - for (reg = 0; reg < CMD_LAST; reg++) - uda->suspend_cfg[reg] = uda->cfg[reg]; - break; - case CMD_RESUME: - for (reg = stat0; reg < uda1341_reg_last; reg++) - snd_uda1341_codec_write(clnt, reg, uda->suspend_regs[reg]); - for (reg = 0; reg < CMD_LAST; reg++) - uda->cfg[reg] = uda->suspend_cfg[reg]; - break; -#endif - default: - ret = -EINVAL; - break; - } - - if (!uda->active) - printk(KERN_ERR "UDA1341 codec not active!\n"); - return ret; -} - -/* }}} */ - -/* {{{ Proc interface */ -#ifdef CONFIG_PROC_FS - -static const char *format_names[] = { - "I2S-bus", - "LSB 16bits", - "LSB 18bits", - "LSB 20bits", - "MSB", - "in LSB 16bits/out MSB", - "in LSB 18bits/out MSB", - "in LSB 20bits/out MSB", -}; - -static const char *fs_names[] = { - "512*fs", - "384*fs", - "256*fs", - "Unused - bad value!", -}; - -static const char* bass_values[][16] = { - {"0 dB", "0 dB", "0 dB", "0 dB", "0 dB", "0 dB", "0 dB", "0 dB", "0 dB", "0 dB", "0 dB", - "0 dB", "0 dB", "0 dB", "0 dB", "undefined", }, //flat - {"0 dB", "2 dB", "4 dB", "6 dB", "8 dB", "10 dB", "12 dB", "14 dB", "16 dB", "18 dB", "18 dB", - "18 dB", "18 dB", "18 dB", "18 dB", "undefined",}, // min - {"0 dB", "2 dB", "4 dB", "6 dB", "8 dB", "10 dB", "12 dB", "14 dB", "16 dB", "18 dB", "18 dB", - "18 dB", "18 dB", "18 dB", "18 dB", "undefined",}, // min - {"0 dB", "2 dB", "4 dB", "6 dB", "8 dB", "10 dB", "12 dB", "14 dB", "16 dB", "18 dB", "20 dB", - "22 dB", "24 dB", "24 dB", "24 dB", "undefined",}, // max -}; - -static const char *mic_sens_value[] = { - "-3 dB", "0 dB", "3 dB", "9 dB", "15 dB", "21 dB", "27 dB", "not used", -}; - -static const unsigned short AGC_atime[] = { - 11, 16, 11, 16, 21, 11, 16, 21, -}; - -static const unsigned short AGC_dtime[] = { - 100, 100, 200, 200, 200, 400, 400, 400, -}; - -static const char *AGC_level[] = { - "-9.0", "-11.5", "-15.0", "-17.5", -}; - -static const char *ig_small_value[] = { - "-3.0", "-2.5", "-2.0", "-1.5", "-1.0", "-0.5", -}; - -/* - * this was computed as peak_value[i] = pow((63-i)*1.42,1.013) - * - * UDA1341 datasheet on page 21: Peak value (dB) = (Peak level - 63.5)*5*log2 - * There is an table with these values [level]=value: [3]=-90.31, [7]=-84.29 - * [61]=-2.78, [62] = -1.48, [63] = 0.0 - * I tried to compute it, but using but even using logarithm with base either 10 or 2 - * i was'n able to get values in the table from the formula. So I constructed another - * formula (see above) to interpolate the values as good as possible. If there is some - * mistake, please contact me on tomas.kasparek@seznam.cz. Thanks. - * UDA1341TS datasheet is available at: - * http://www-us9.semiconductors.com/acrobat/datasheets/UDA1341TS_3.pdf - */ -static const char *peak_value[] = { - "-INF dB", "N.A.", "N.A", "90.31 dB", "N.A.", "N.A.", "N.A.", "-84.29 dB", - "-82.65 dB", "-81.13 dB", "-79.61 dB", "-78.09 dB", "-76.57 dB", "-75.05 dB", "-73.53 dB", - "-72.01 dB", "-70.49 dB", "-68.97 dB", "-67.45 dB", "-65.93 dB", "-64.41 dB", "-62.90 dB", - "-61.38 dB", "-59.86 dB", "-58.35 dB", "-56.83 dB", "-55.32 dB", "-53.80 dB", "-52.29 dB", - "-50.78 dB", "-49.26 dB", "-47.75 dB", "-46.24 dB", "-44.73 dB", "-43.22 dB", "-41.71 dB", - "-40.20 dB", "-38.69 dB", "-37.19 dB", "-35.68 dB", "-34.17 dB", "-32.67 dB", "-31.17 dB", - "-29.66 dB", "-28.16 dB", "-26.66 dB", "-25.16 dB", "-23.66 dB", "-22.16 dB", "-20.67 dB", - "-19.17 dB", "-17.68 dB", "-16.19 dB", "-14.70 dB", "-13.21 dB", "-11.72 dB", "-10.24 dB", - "-8.76 dB", "-7.28 dB", "-5.81 dB", "-4.34 dB", "-2.88 dB", "-1.43 dB", "0.00 dB", -}; - -static void snd_uda1341_proc_read(struct snd_info_entry *entry, - struct snd_info_buffer *buffer) -{ - struct l3_client *clnt = entry->private_data; - struct uda1341 *uda = clnt->driver_data; - int peak; - - peak = snd_uda1341_codec_read(clnt, UDA1341_DATA1); - if (peak < 0) - peak = 0; - - snd_iprintf(buffer, "%s\n\n", uda->card->longname); - - // for information about computed values see UDA1341TS datasheet pages 15 - 21 - snd_iprintf(buffer, "DAC power : %s\n", uda->cfg[CMD_DAC] ? "on" : "off"); - snd_iprintf(buffer, "ADC power : %s\n", uda->cfg[CMD_ADC] ? "on" : "off"); - snd_iprintf(buffer, "Clock frequency : %s\n", fs_names[uda->cfg[CMD_FS]]); - snd_iprintf(buffer, "Data format : %s\n\n", format_names[uda->cfg[CMD_FORMAT]]); - - snd_iprintf(buffer, "Filter mode : %s\n", filter_names[uda->cfg[CMD_FILTER]]); - snd_iprintf(buffer, "Mixer mode : %s\n", mixer_names[uda->cfg[CMD_MIXER]]); - snd_iprintf(buffer, "De-emphasis : %s\n", deemp_names[uda->cfg[CMD_DEEMP]]); - snd_iprintf(buffer, "Peak detection pos. : %s\n", uda->cfg[CMD_PEAK] ? "after" : "before"); - snd_iprintf(buffer, "Peak value : %s\n\n", peak_value[peak]); - - snd_iprintf(buffer, "Automatic Gain Ctrl : %s\n", uda->cfg[CMD_AGC] ? "on" : "off"); - snd_iprintf(buffer, "AGC attack time : %d ms\n", AGC_atime[uda->cfg[CMD_AGC_TIME]]); - snd_iprintf(buffer, "AGC decay time : %d ms\n", AGC_dtime[uda->cfg[CMD_AGC_TIME]]); - snd_iprintf(buffer, "AGC output level : %s dB\n\n", AGC_level[uda->cfg[CMD_AGC_LEVEL]]); - - snd_iprintf(buffer, "Mute : %s\n", uda->cfg[CMD_MUTE] ? "on" : "off"); - - if (uda->cfg[CMD_VOLUME] == 0) - snd_iprintf(buffer, "Volume : 0 dB\n"); - else if (uda->cfg[CMD_VOLUME] < 62) - snd_iprintf(buffer, "Volume : %d dB\n", -1*uda->cfg[CMD_VOLUME] +1); - else - snd_iprintf(buffer, "Volume : -INF dB\n"); - snd_iprintf(buffer, "Bass : %s\n", bass_values[uda->cfg[CMD_FILTER]][uda->cfg[CMD_BASS]]); - snd_iprintf(buffer, "Trebble : %d dB\n", uda->cfg[CMD_FILTER] ? 2*uda->cfg[CMD_TREBBLE] : 0); - snd_iprintf(buffer, "Input Gain (6dB) : %s\n", uda->cfg[CMD_IGAIN] ? "on" : "off"); - snd_iprintf(buffer, "Output Gain (6dB) : %s\n", uda->cfg[CMD_OGAIN] ? "on" : "off"); - snd_iprintf(buffer, "Mic sensitivity : %s\n", mic_sens_value[uda->cfg[CMD_MIC]]); - - - if(uda->cfg[CMD_CH1] < 31) - snd_iprintf(buffer, "Mixer gain channel 1: -%d.%c dB\n", - ((uda->cfg[CMD_CH1] >> 1) * 3) + (uda->cfg[CMD_CH1] & 1), - uda->cfg[CMD_CH1] & 1 ? '5' : '0'); - else - snd_iprintf(buffer, "Mixer gain channel 1: -INF dB\n"); - if(uda->cfg[CMD_CH2] < 31) - snd_iprintf(buffer, "Mixer gain channel 2: -%d.%c dB\n", - ((uda->cfg[CMD_CH2] >> 1) * 3) + (uda->cfg[CMD_CH2] & 1), - uda->cfg[CMD_CH2] & 1 ? '5' : '0'); - else - snd_iprintf(buffer, "Mixer gain channel 2: -INF dB\n"); - - if(uda->cfg[CMD_IG] > 5) - snd_iprintf(buffer, "Input Amp. Gain ch 2: %d.%c dB\n", - (uda->cfg[CMD_IG] >> 1) -3, uda->cfg[CMD_IG] & 1 ? '5' : '0'); - else - snd_iprintf(buffer, "Input Amp. Gain ch 2: %s dB\n", ig_small_value[uda->cfg[CMD_IG]]); -} - -static void snd_uda1341_proc_regs_read(struct snd_info_entry *entry, - struct snd_info_buffer *buffer) -{ - struct l3_client *clnt = entry->private_data; - struct uda1341 *uda = clnt->driver_data; - int reg; - char buf[12]; - - for (reg = 0; reg < uda1341_reg_last; reg ++) { - if (reg == empty) - continue; - int2str_bin8(uda->regs[reg], buf); - snd_iprintf(buffer, "%s = %s\n", uda1341_reg_names[reg], buf); - } - - int2str_bin8(snd_uda1341_codec_read(clnt, UDA1341_DATA1), buf); - snd_iprintf(buffer, "DATA1 = %s\n", buf); -} -#endif /* CONFIG_PROC_FS */ - -static void __devinit snd_uda1341_proc_init(struct snd_card *card, struct l3_client *clnt) -{ - struct snd_info_entry *entry; - - if (! snd_card_proc_new(card, "uda1341", &entry)) - snd_info_set_text_ops(entry, clnt, snd_uda1341_proc_read); - if (! snd_card_proc_new(card, "uda1341-regs", &entry)) - snd_info_set_text_ops(entry, clnt, snd_uda1341_proc_regs_read); -} - -/* }}} */ - -/* {{{ Mixer controls setting */ - -/* {{{ UDA1341 single functions */ - -#define UDA1341_SINGLE(xname, where, reg, shift, mask, invert) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .info = snd_uda1341_info_single, \ - .get = snd_uda1341_get_single, .put = snd_uda1341_put_single, \ - .private_value = where | (reg << 5) | (shift << 9) | (mask << 12) | (invert << 18) \ -} - -static int snd_uda1341_info_single(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - int mask = (kcontrol->private_value >> 12) & 63; - - uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = mask; - return 0; -} - -static int snd_uda1341_get_single(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct l3_client *clnt = snd_kcontrol_chip(kcontrol); - struct uda1341 *uda = clnt->driver_data; - int where = kcontrol->private_value & 31; - int mask = (kcontrol->private_value >> 12) & 63; - int invert = (kcontrol->private_value >> 18) & 1; - - ucontrol->value.integer.value[0] = uda->cfg[where]; - if (invert) - ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0]; - - return 0; -} - -static int snd_uda1341_put_single(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct l3_client *clnt = snd_kcontrol_chip(kcontrol); - struct uda1341 *uda = clnt->driver_data; - int where = kcontrol->private_value & 31; - int reg = (kcontrol->private_value >> 5) & 15; - int shift = (kcontrol->private_value >> 9) & 7; - int mask = (kcontrol->private_value >> 12) & 63; - int invert = (kcontrol->private_value >> 18) & 1; - unsigned short val; - - val = (ucontrol->value.integer.value[0] & mask); - if (invert) - val = mask - val; - - uda->cfg[where] = val; - return snd_uda1341_update_bits(clnt, reg, mask, shift, val, FLUSH); -} - -/* }}} */ - -/* {{{ UDA1341 enum functions */ - -#define UDA1341_ENUM(xname, where, reg, shift, mask, invert) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .info = snd_uda1341_info_enum, \ - .get = snd_uda1341_get_enum, .put = snd_uda1341_put_enum, \ - .private_value = where | (reg << 5) | (shift << 9) | (mask << 12) | (invert << 18) \ -} - -static int snd_uda1341_info_enum(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - int where = kcontrol->private_value & 31; - const char **texts; - - // this register we don't handle this way - if (!uda1341_enum_items[where]) - return -EINVAL; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = uda1341_enum_items[where]; - - if (uinfo->value.enumerated.item >= uda1341_enum_items[where]) - uinfo->value.enumerated.item = uda1341_enum_items[where] - 1; - - texts = uda1341_enum_names[where]; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; -} - -static int snd_uda1341_get_enum(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct l3_client *clnt = snd_kcontrol_chip(kcontrol); - struct uda1341 *uda = clnt->driver_data; - int where = kcontrol->private_value & 31; - - ucontrol->value.enumerated.item[0] = uda->cfg[where]; - return 0; -} - -static int snd_uda1341_put_enum(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct l3_client *clnt = snd_kcontrol_chip(kcontrol); - struct uda1341 *uda = clnt->driver_data; - int where = kcontrol->private_value & 31; - int reg = (kcontrol->private_value >> 5) & 15; - int shift = (kcontrol->private_value >> 9) & 7; - int mask = (kcontrol->private_value >> 12) & 63; - - uda->cfg[where] = (ucontrol->value.enumerated.item[0] & mask); - - return snd_uda1341_update_bits(clnt, reg, mask, shift, uda->cfg[where], FLUSH); -} - -/* }}} */ - -/* {{{ UDA1341 2regs functions */ - -#define UDA1341_2REGS(xname, where, reg_1, reg_2, shift_1, shift_2, mask_1, mask_2, invert) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), .info = snd_uda1341_info_2regs, \ - .get = snd_uda1341_get_2regs, .put = snd_uda1341_put_2regs, \ - .private_value = where | (reg_1 << 5) | (reg_2 << 9) | (shift_1 << 13) | (shift_2 << 16) | \ - (mask_1 << 19) | (mask_2 << 25) | (invert << 31) \ -} - - -static int snd_uda1341_info_2regs(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - int mask_1 = (kcontrol->private_value >> 19) & 63; - int mask_2 = (kcontrol->private_value >> 25) & 63; - int mask; - - mask = (mask_2 + 1) * (mask_1 + 1) - 1; - uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = mask; - return 0; -} - -static int snd_uda1341_get_2regs(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct l3_client *clnt = snd_kcontrol_chip(kcontrol); - struct uda1341 *uda = clnt->driver_data; - int where = kcontrol->private_value & 31; - int mask_1 = (kcontrol->private_value >> 19) & 63; - int mask_2 = (kcontrol->private_value >> 25) & 63; - int invert = (kcontrol->private_value >> 31) & 1; - int mask; - - mask = (mask_2 + 1) * (mask_1 + 1) - 1; - - ucontrol->value.integer.value[0] = uda->cfg[where]; - if (invert) - ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0]; - return 0; -} - -static int snd_uda1341_put_2regs(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct l3_client *clnt = snd_kcontrol_chip(kcontrol); - struct uda1341 *uda = clnt->driver_data; - int where = kcontrol->private_value & 31; - int reg_1 = (kcontrol->private_value >> 5) & 15; - int reg_2 = (kcontrol->private_value >> 9) & 15; - int shift_1 = (kcontrol->private_value >> 13) & 7; - int shift_2 = (kcontrol->private_value >> 16) & 7; - int mask_1 = (kcontrol->private_value >> 19) & 63; - int mask_2 = (kcontrol->private_value >> 25) & 63; - int invert = (kcontrol->private_value >> 31) & 1; - int mask; - unsigned short val1, val2, val; - - val = ucontrol->value.integer.value[0]; - - mask = (mask_2 + 1) * (mask_1 + 1) - 1; - - val1 = val & mask_1; - val2 = (val / (mask_1 + 1)) & mask_2; - - if (invert) { - val1 = mask_1 - val1; - val2 = mask_2 - val2; - } - - uda->cfg[where] = invert ? mask - val : val; - - //FIXME - return value - snd_uda1341_update_bits(clnt, reg_1, mask_1, shift_1, val1, FLUSH); - return snd_uda1341_update_bits(clnt, reg_2, mask_2, shift_2, val2, FLUSH); -} - -/* }}} */ - -static struct snd_kcontrol_new snd_uda1341_controls[] = { - UDA1341_SINGLE("Master Playback Switch", CMD_MUTE, data0_2, 2, 1, 1), - UDA1341_SINGLE("Master Playback Volume", CMD_VOLUME, data0_0, 0, 63, 1), - - UDA1341_SINGLE("Bass Playback Volume", CMD_BASS, data0_1, 2, 15, 0), - UDA1341_SINGLE("Treble Playback Volume", CMD_TREBBLE, data0_1, 0, 3, 0), - - UDA1341_SINGLE("Input Gain Switch", CMD_IGAIN, stat1, 5, 1, 0), - UDA1341_SINGLE("Output Gain Switch", CMD_OGAIN, stat1, 6, 1, 0), - - UDA1341_SINGLE("Mixer Gain Channel 1 Volume", CMD_CH1, ext0, 0, 31, 1), - UDA1341_SINGLE("Mixer Gain Channel 2 Volume", CMD_CH2, ext1, 0, 31, 1), - - UDA1341_SINGLE("Mic Sensitivity Volume", CMD_MIC, ext2, 2, 7, 0), - - UDA1341_SINGLE("AGC Output Level", CMD_AGC_LEVEL, ext6, 0, 3, 0), - UDA1341_SINGLE("AGC Time Constant", CMD_AGC_TIME, ext6, 2, 7, 0), - UDA1341_SINGLE("AGC Time Constant Switch", CMD_AGC, ext4, 4, 1, 0), - - UDA1341_SINGLE("DAC Power", CMD_DAC, stat1, 0, 1, 0), - UDA1341_SINGLE("ADC Power", CMD_ADC, stat1, 1, 1, 0), - - UDA1341_ENUM("Peak detection", CMD_PEAK, data0_2, 5, 1, 0), - UDA1341_ENUM("De-emphasis", CMD_DEEMP, data0_2, 3, 3, 0), - UDA1341_ENUM("Mixer mode", CMD_MIXER, ext2, 0, 3, 0), - UDA1341_ENUM("Filter mode", CMD_FILTER, data0_2, 0, 3, 0), - - UDA1341_2REGS("Gain Input Amplifier Gain (channel 2)", CMD_IG, ext4, ext5, 0, 0, 3, 31, 0), -}; - -static void uda1341_free(struct l3_client *clnt) -{ - l3_detach_client(clnt); // calls kfree for driver_data (struct uda1341) - kfree(clnt); -} - -static int uda1341_dev_free(struct snd_device *device) -{ - struct l3_client *clnt = device->device_data; - uda1341_free(clnt); - return 0; -} - -int __init snd_chip_uda1341_mixer_new(struct snd_card *card, struct l3_client **clntp) -{ - static struct snd_device_ops ops = { - .dev_free = uda1341_dev_free, - }; - struct l3_client *clnt; - int idx, err; - - if (snd_BUG_ON(!card)) - return -EINVAL; - - clnt = kzalloc(sizeof(*clnt), GFP_KERNEL); - if (clnt == NULL) - return -ENOMEM; - - if ((err = l3_attach_client(clnt, "l3-bit-sa1100-gpio", UDA1341_ALSA_NAME))) { - kfree(clnt); - return err; - } - - for (idx = 0; idx < ARRAY_SIZE(snd_uda1341_controls); idx++) { - if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_uda1341_controls[idx], clnt))) < 0) { - uda1341_free(clnt); - return err; - } - } - - if ((err = snd_device_new(card, SNDRV_DEV_CODEC, clnt, &ops)) < 0) { - uda1341_free(clnt); - return err; - } - - *clntp = clnt; - strcpy(card->mixername, "UDA1341TS Mixer"); - ((struct uda1341 *)clnt->driver_data)->card = card; - - snd_uda1341_proc_init(card, clnt); - - return 0; -} - -/* }}} */ - -/* {{{ L3 operations */ - -static int uda1341_attach(struct l3_client *clnt) -{ - struct uda1341 *uda; - - uda = kzalloc(sizeof(*uda), 0, GFP_KERNEL); - if (!uda) - return -ENOMEM; - - /* init fixed parts of my copy of registers */ - uda->regs[stat0] = STAT0; - uda->regs[stat1] = STAT1; - - uda->regs[data0_0] = DATA0_0; - uda->regs[data0_1] = DATA0_1; - uda->regs[data0_2] = DATA0_2; - - uda->write = snd_uda1341_codec_write; - uda->read = snd_uda1341_codec_read; - - spin_lock_init(&uda->reg_lock); - - clnt->driver_data = uda; - return 0; -} - -static void uda1341_detach(struct l3_client *clnt) -{ - kfree(clnt->driver_data); -} - -static int -uda1341_command(struct l3_client *clnt, int cmd, void *arg) -{ - if (cmd != CMD_READ_REG) - return snd_uda1341_cfg_write(clnt, cmd, (int) arg, FLUSH); - - return snd_uda1341_codec_read(clnt, (int) arg); -} - -static int uda1341_open(struct l3_client *clnt) -{ - struct uda1341 *uda = clnt->driver_data; - - uda->active = 1; - - /* init default configuration */ - snd_uda1341_cfg_write(clnt, CMD_RESET, 0, REGS_ONLY); - snd_uda1341_cfg_write(clnt, CMD_FS, F256, FLUSH); // unknown state after reset - snd_uda1341_cfg_write(clnt, CMD_FORMAT, LSB16, FLUSH); // unknown state after reset - snd_uda1341_cfg_write(clnt, CMD_OGAIN, ON, FLUSH); // default off after reset - snd_uda1341_cfg_write(clnt, CMD_IGAIN, ON, FLUSH); // default off after reset - snd_uda1341_cfg_write(clnt, CMD_DAC, ON, FLUSH); // ??? default value after reset - snd_uda1341_cfg_write(clnt, CMD_ADC, ON, FLUSH); // ??? default value after reset - snd_uda1341_cfg_write(clnt, CMD_VOLUME, 20, FLUSH); // default 0dB after reset - snd_uda1341_cfg_write(clnt, CMD_BASS, 0, REGS_ONLY); // default value after reset - snd_uda1341_cfg_write(clnt, CMD_TREBBLE, 0, REGS_ONLY); // default value after reset - snd_uda1341_cfg_write(clnt, CMD_PEAK, AFTER, REGS_ONLY);// default value after reset - snd_uda1341_cfg_write(clnt, CMD_DEEMP, NONE, REGS_ONLY);// default value after reset - //at this moment should be QMUTED by h3600_audio_init - snd_uda1341_cfg_write(clnt, CMD_MUTE, OFF, REGS_ONLY); // default value after reset - snd_uda1341_cfg_write(clnt, CMD_FILTER, MAX, FLUSH); // defaul flat after reset - snd_uda1341_cfg_write(clnt, CMD_CH1, 31, FLUSH); // default value after reset - snd_uda1341_cfg_write(clnt, CMD_CH2, 4, FLUSH); // default value after reset - snd_uda1341_cfg_write(clnt, CMD_MIC, 4, FLUSH); // default 0dB after reset - snd_uda1341_cfg_write(clnt, CMD_MIXER, MIXER, FLUSH); // default doub.dif.mode - snd_uda1341_cfg_write(clnt, CMD_AGC, OFF, FLUSH); // default value after reset - snd_uda1341_cfg_write(clnt, CMD_IG, 0, FLUSH); // unknown state after reset - snd_uda1341_cfg_write(clnt, CMD_AGC_TIME, 0, FLUSH); // default value after reset - snd_uda1341_cfg_write(clnt, CMD_AGC_LEVEL, 0, FLUSH); // default value after reset - - return 0; -} - -static void uda1341_close(struct l3_client *clnt) -{ - struct uda1341 *uda = clnt->driver_data; - - uda->active = 0; -} - -/* }}} */ - -/* {{{ Module and L3 initialization */ - -static struct l3_ops uda1341_ops = { - .open = uda1341_open, - .command = uda1341_command, - .close = uda1341_close, -}; - -static struct l3_driver uda1341_driver = { - .name = UDA1341_ALSA_NAME, - .attach_client = uda1341_attach, - .detach_client = uda1341_detach, - .ops = &uda1341_ops, - .owner = THIS_MODULE, -}; - -static int __init uda1341_init(void) -{ - return l3_add_driver(&uda1341_driver); -} - -static void __exit uda1341_exit(void) -{ - l3_del_driver(&uda1341_driver); -} - -module_init(uda1341_init); -module_exit(uda1341_exit); - -MODULE_AUTHOR("Tomas Kasparek "); -MODULE_LICENSE("GPL"); -MODULE_DESCRIPTION("Philips UDA1341 CODEC driver for ALSA"); -MODULE_SUPPORTED_DEVICE("{{UDA1341,UDA1341TS}}"); - -EXPORT_SYMBOL(snd_chip_uda1341_mixer_new); - -/* }}} */ - -/* - * Local variables: - * indent-tabs-mode: t - * End: - */ -- cgit v1.2.3-70-g09d2 From ee5047102cf632351c418060bfbe3b6eb5c42e7b Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 17 Mar 2009 14:30:31 +0100 Subject: ALSA: snd-hda-intel - add checks for invalid values to *query_supported_pcm() If ratesp or formatsp values are zero, wrong values are passed to ALSA's the PCM midlevel code. The bug is showed more later than expected. Also, clean a bit the code. Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 33 +++++++++++++++++++++++++-------- 1 file changed, 25 insertions(+), 8 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index cf6339436de..b90a2400f53 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2539,12 +2539,11 @@ EXPORT_SYMBOL_HDA(snd_hda_calc_stream_format); static int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid, u32 *ratesp, u64 *formatsp, unsigned int *bpsp) { - int i; - unsigned int val, streams; + unsigned int i, val, wcaps; val = 0; - if (nid != codec->afg && - (get_wcaps(codec, nid) & AC_WCAP_FORMAT_OVRD)) { + wcaps = get_wcaps(codec, nid); + if (nid != codec->afg && (wcaps & AC_WCAP_FORMAT_OVRD)) { val = snd_hda_param_read(codec, nid, AC_PAR_PCM); if (val == -1) return -EIO; @@ -2558,15 +2557,20 @@ static int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid, if (val & (1 << i)) rates |= rate_bits[i].alsa_bits; } + if (rates == 0) { + snd_printk(KERN_ERR "hda_codec: rates == 0 " + "(nid=0x%x, val=0x%x, ovrd=%i)\n", + nid, val, + (wcaps & AC_WCAP_FORMAT_OVRD) ? 1 : 0); + return -EIO; + } *ratesp = rates; } if (formatsp || bpsp) { u64 formats = 0; - unsigned int bps; - unsigned int wcaps; + unsigned int streams, bps; - wcaps = get_wcaps(codec, nid); streams = snd_hda_param_read(codec, nid, AC_PAR_STREAM); if (streams == -1) return -EIO; @@ -2619,6 +2623,15 @@ static int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid, formats |= SNDRV_PCM_FMTBIT_U8; bps = 8; } + if (formats == 0) { + snd_printk(KERN_ERR "hda_codec: formats == 0 " + "(nid=0x%x, val=0x%x, ovrd=%i, " + "streams=0x%x)\n", + nid, val, + (wcaps & AC_WCAP_FORMAT_OVRD) ? 1 : 0, + streams); + return -EIO; + } if (formatsp) *formatsp = formats; if (bpsp) @@ -2734,12 +2747,16 @@ static int hda_pcm_default_cleanup(struct hda_pcm_stream *hinfo, static int set_pcm_default_values(struct hda_codec *codec, struct hda_pcm_stream *info) { + int err; + /* query support PCM information from the given NID */ if (info->nid && (!info->rates || !info->formats)) { - snd_hda_query_supported_pcm(codec, info->nid, + err = snd_hda_query_supported_pcm(codec, info->nid, info->rates ? NULL : &info->rates, info->formats ? NULL : &info->formats, info->maxbps ? NULL : &info->maxbps); + if (err < 0) + return err; } if (info->ops.open == NULL) info->ops.open = hda_pcm_default_open_close; -- cgit v1.2.3-70-g09d2 From 1313e7041480f523a09dedc7ef2185d8ee94c163 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 18 Mar 2009 11:03:53 +0100 Subject: ALSA: snd-usb-caiaq: only warn once on streaming errors Limit the number of printed warnings to one in case of streaming errors. printk() happens to be expensive, especially in code called as often as here. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/caiaq/caiaq-audio.c | 4 +++- sound/usb/caiaq/caiaq-device.h | 2 +- 2 files changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/usb/caiaq/caiaq-audio.c b/sound/usb/caiaq/caiaq-audio.c index fc6d571eeac..577b1129de0 100644 --- a/sound/usb/caiaq/caiaq-audio.c +++ b/sound/usb/caiaq/caiaq-audio.c @@ -114,6 +114,7 @@ static int stream_start(struct snd_usb_caiaqdev *dev) dev->output_panic = 0; dev->first_packet = 1; dev->streaming = 1; + dev->warned = 0; for (i = 0; i < N_URBS; i++) { ret = usb_submit_urb(dev->data_urbs_in[i], GFP_ATOMIC); @@ -406,10 +407,11 @@ static void read_in_urb(struct snd_usb_caiaqdev *dev, break; } - if (dev->input_panic || dev->output_panic) { + if ((dev->input_panic || dev->output_panic) && !dev->warned) { debug("streaming error detected %s %s\n", dev->input_panic ? "(input)" : "", dev->output_panic ? "(output)" : ""); + dev->warned = 1; } } diff --git a/sound/usb/caiaq/caiaq-device.h b/sound/usb/caiaq/caiaq-device.h index 0560c327d99..098b194f725 100644 --- a/sound/usb/caiaq/caiaq-device.h +++ b/sound/usb/caiaq/caiaq-device.h @@ -89,7 +89,7 @@ struct snd_usb_caiaqdev { int audio_out_buf_pos[MAX_STREAMS]; int period_in_count[MAX_STREAMS]; int period_out_count[MAX_STREAMS]; - int input_panic, output_panic; + int input_panic, output_panic, warned; char *audio_in_buf, *audio_out_buf; unsigned int samplerates; -- cgit v1.2.3-70-g09d2 From 9311c9b4f12218b588e51806c44d290cfec678a3 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 18 Mar 2009 11:03:54 +0100 Subject: ALSA: snd-usb-caiaq: drop bogus iso packets Drop inbound packets that are smaller than expected. This has been observed at the very beginning of the streaming transaction. And when the hardware is in panic mode (which can only very rarely happen in case of massive EMI chaos), mute the input channels. Signed-off-by: Daniel Mack Tested-by: Mark Hills Signed-off-by: Takashi Iwai --- sound/usb/caiaq/caiaq-audio.c | 6 ++++++ sound/usb/caiaq/caiaq-device.c | 2 ++ sound/usb/caiaq/caiaq-device.h | 2 +- 3 files changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/usb/caiaq/caiaq-audio.c b/sound/usb/caiaq/caiaq-audio.c index 577b1129de0..08d51e0c9fe 100644 --- a/sound/usb/caiaq/caiaq-audio.c +++ b/sound/usb/caiaq/caiaq-audio.c @@ -377,6 +377,9 @@ static void read_in_urb_mode2(struct snd_usb_caiaqdev *dev, for (stream = 0; stream < dev->n_streams; stream++, i++) { sub = dev->sub_capture[stream]; + if (dev->input_panic) + usb_buf[i] = 0; + if (sub) { struct snd_pcm_runtime *rt = sub->runtime; char *audio_buf = rt->dma_area; @@ -398,6 +401,9 @@ static void read_in_urb(struct snd_usb_caiaqdev *dev, if (!dev->streaming) return; + if (iso->actual_length < dev->bpp) + return; + switch (dev->spec.data_alignment) { case 0: read_in_urb_mode0(dev, urb, iso); diff --git a/sound/usb/caiaq/caiaq-device.c b/sound/usb/caiaq/caiaq-device.c index 5736669df2d..336a93de0b3 100644 --- a/sound/usb/caiaq/caiaq-device.c +++ b/sound/usb/caiaq/caiaq-device.c @@ -251,6 +251,8 @@ int snd_usb_caiaq_set_audio_params (struct snd_usb_caiaqdev *dev, if (dev->audio_parm_answer != 1) debug("unable to set the device's audio params\n"); + else + dev->bpp = bpp; return dev->audio_parm_answer == 1 ? 0 : -EINVAL; } diff --git a/sound/usb/caiaq/caiaq-device.h b/sound/usb/caiaq/caiaq-device.h index 098b194f725..4cce1ad7493 100644 --- a/sound/usb/caiaq/caiaq-device.h +++ b/sound/usb/caiaq/caiaq-device.h @@ -91,7 +91,7 @@ struct snd_usb_caiaqdev { int period_out_count[MAX_STREAMS]; int input_panic, output_panic, warned; char *audio_in_buf, *audio_out_buf; - unsigned int samplerates; + unsigned int samplerates, bpp; struct snd_pcm_substream *sub_playback[MAX_STREAMS]; struct snd_pcm_substream *sub_capture[MAX_STREAMS]; -- cgit v1.2.3-70-g09d2 From 28514fe5bbbdbc0f7c9700569378d55cafd061ea Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 18 Mar 2009 11:03:55 +0100 Subject: ALSA: snd-usb-caiaq: bump version number Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/caiaq/caiaq-device.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/caiaq/caiaq-device.c b/sound/usb/caiaq/caiaq-device.c index 336a93de0b3..771c523b3fc 100644 --- a/sound/usb/caiaq/caiaq-device.c +++ b/sound/usb/caiaq/caiaq-device.c @@ -42,7 +42,7 @@ #endif MODULE_AUTHOR("Daniel Mack "); -MODULE_DESCRIPTION("caiaq USB audio, version 1.3.12"); +MODULE_DESCRIPTION("caiaq USB audio, version 1.3.13"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," "{Native Instruments, RigKontrol3}," -- cgit v1.2.3-70-g09d2 From e3598f6e4218d1aad3369c97217266b2375e6aca Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 18 Mar 2009 15:19:10 +0000 Subject: ASoC: Further optimise WM8400 bias configuration sequence The active discharge does not bring sufficient benefit to justify the lengthy times involved so don't do that. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8400.c | 27 ++------------------------- 1 file changed, 2 insertions(+), 25 deletions(-) diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index 744e0dc73be..462f8b0d9ac 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -1096,45 +1096,22 @@ static int wm8400_set_bias_level(struct snd_soc_codec *codec, wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, WM8400_CODEC_ENA | WM8400_SYSCLK_ENA); - /* Enable all output discharge bits */ - wm8400_write(codec, WM8400_ANTIPOP1, WM8400_DIS_LLINE | - WM8400_DIS_RLINE | WM8400_DIS_OUT3 | - WM8400_DIS_OUT4 | WM8400_DIS_LOUT | - WM8400_DIS_ROUT); - /* Enable POBCTRL, SOFT_ST, VMIDTOG and BUFDCOPEN */ wm8400_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST | WM8400_BUFDCOPEN | WM8400_POBCTRL); - msleep(500); - - /* Enable outputs */ - val = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1); - val |= WM8400_SPK_ENA | WM8400_OUT3_ENA | - WM8400_OUT4_ENA | WM8400_LOUT_ENA | - WM8400_ROUT_ENA; - wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val); - - /* disable all output discharge bits */ - wm8400_write(codec, WM8400_ANTIPOP1, 0); + msleep(50); /* Enable VREF & VMID at 2x50k */ + val = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1); val |= 0x2 | WM8400_VREF_ENA; wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val); - msleep(600); - /* Enable BUFIOEN */ wm8400_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST | WM8400_BUFDCOPEN | WM8400_POBCTRL | WM8400_BUFIOEN); - /* Disable outputs */ - val &= ~(WM8400_SPK_ENA | WM8400_OUT3_ENA | - WM8400_OUT4_ENA | WM8400_LOUT_ENA | - WM8400_ROUT_ENA); - wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val); - /* disable POBCTRL, SOFT_ST and BUFDCOPEN */ wm8400_write(codec, WM8400_ANTIPOP2, WM8400_BUFIOEN); } -- cgit v1.2.3-70-g09d2 From 24a51029fc3055f33684e650b5e3a59f77c9b05c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 18 Mar 2009 15:19:48 +0000 Subject: ASoC: Add separate AVDD for WM8400 There is an AVDD supply as well, normally one or more of the other upplies would be tied to it. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8400.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index 462f8b0d9ac..b7350c25b61 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -48,6 +48,9 @@ static struct regulator_bulk_data power[] = { { .supply = "DCVDD", }, + { + .supply = "AVDD", + }, { .supply = "FLLVDD", }, -- cgit v1.2.3-70-g09d2 From a2328d0249fce44381289525bd580b37d2105963 Mon Sep 17 00:00:00 2001 From: Giuliano Pochini Date: Thu, 19 Mar 2009 00:09:03 +0100 Subject: ALSA: Echoaudio: add support for Indigo express cards This patch adds support for IndigoIOx and IndigoDJx. Signed-off-by: Giuliano Pochini Signed-off-by: Takashi Iwai --- sound/pci/Kconfig | 20 ++++++ sound/pci/echoaudio/Makefile | 4 ++ sound/pci/echoaudio/echoaudio.h | 3 + sound/pci/echoaudio/echoaudio_dsp.h | 9 ++- sound/pci/echoaudio/indigo_express_dsp.c | 119 +++++++++++++++++++++++++++++++ sound/pci/echoaudio/indigodjx.c | 107 +++++++++++++++++++++++++++ sound/pci/echoaudio/indigodjx_dsp.c | 68 ++++++++++++++++++ sound/pci/echoaudio/indigoiox.c | 109 ++++++++++++++++++++++++++++ sound/pci/echoaudio/indigoiox_dsp.c | 68 ++++++++++++++++++ 9 files changed, 505 insertions(+), 2 deletions(-) create mode 100644 sound/pci/echoaudio/indigo_express_dsp.c create mode 100644 sound/pci/echoaudio/indigodjx.c create mode 100644 sound/pci/echoaudio/indigodjx_dsp.c create mode 100644 sound/pci/echoaudio/indigoiox.c create mode 100644 sound/pci/echoaudio/indigoiox_dsp.c diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 82b9bddcdcd..9387ab00a41 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -400,6 +400,26 @@ config SND_INDIGODJ To compile this driver as a module, choose M here: the module will be called snd-indigodj +config SND_INDIGOIOX + tristate "(Echoaudio) Indigo IOx" + select FW_LOADER + select SND_PCM + help + Say 'Y' or 'M' to include support for Echoaudio Indigo IOx. + + To compile this driver as a module, choose M here: the module + will be called snd-indigoiox + +config SND_INDIGODJX + tristate "(Echoaudio) Indigo DJx" + select FW_LOADER + select SND_PCM + help + Say 'Y' or 'M' to include support for Echoaudio Indigo DJx. + + To compile this driver as a module, choose M here: the module + will be called snd-indigodjx + config SND_EMU10K1 tristate "Emu10k1 (SB Live!, Audigy, E-mu APS)" select FW_LOADER diff --git a/sound/pci/echoaudio/Makefile b/sound/pci/echoaudio/Makefile index 7b576aeb3f8..1361de77e0c 100644 --- a/sound/pci/echoaudio/Makefile +++ b/sound/pci/echoaudio/Makefile @@ -15,6 +15,8 @@ snd-echo3g-objs := echo3g.o snd-indigo-objs := indigo.o snd-indigoio-objs := indigoio.o snd-indigodj-objs := indigodj.o +snd-indigoiox-objs := indigoiox.o +snd-indigodjx-objs := indigodjx.o obj-$(CONFIG_SND_DARLA20) += snd-darla20.o obj-$(CONFIG_SND_GINA20) += snd-gina20.o @@ -28,3 +30,5 @@ obj-$(CONFIG_SND_ECHO3G) += snd-echo3g.o obj-$(CONFIG_SND_INDIGO) += snd-indigo.o obj-$(CONFIG_SND_INDIGOIO) += snd-indigoio.o obj-$(CONFIG_SND_INDIGODJ) += snd-indigodj.o +obj-$(CONFIG_SND_INDIGOIOX) += snd-indigoiox.o +obj-$(CONFIG_SND_INDIGODJX) += snd-indigodjx.o diff --git a/sound/pci/echoaudio/echoaudio.h b/sound/pci/echoaudio/echoaudio.h index 1c88e051abf..f9490ae36c2 100644 --- a/sound/pci/echoaudio/echoaudio.h +++ b/sound/pci/echoaudio/echoaudio.h @@ -189,6 +189,9 @@ #define INDIGO 0x0090 #define INDIGO_IO 0x00a0 #define INDIGO_DJ 0x00b0 +#define DC8 0x00c0 +#define INDIGO_IOX 0x00d0 +#define INDIGO_DJX 0x00e0 #define ECHO3G 0x0100 diff --git a/sound/pci/echoaudio/echoaudio_dsp.h b/sound/pci/echoaudio/echoaudio_dsp.h index e352f3ae292..cb7d75a0a50 100644 --- a/sound/pci/echoaudio/echoaudio_dsp.h +++ b/sound/pci/echoaudio/echoaudio_dsp.h @@ -576,8 +576,13 @@ SET_LAYLA24_FREQUENCY_REG command. #define E3G_ASIC_NOT_LOADED 0xffff #define E3G_BOX_TYPE_MASK 0xf0 -#define EXT_3GBOX_NC 0x01 -#define EXT_3GBOX_NOT_SET 0x02 +/* Indigo express control register values */ +#define INDIGO_EXPRESS_32000 0x02 +#define INDIGO_EXPRESS_44100 0x01 +#define INDIGO_EXPRESS_48000 0x00 +#define INDIGO_EXPRESS_DOUBLE_SPEED 0x10 +#define INDIGO_EXPRESS_QUAD_SPEED 0x04 +#define INDIGO_EXPRESS_CLOCK_MASK 0x17 /* diff --git a/sound/pci/echoaudio/indigo_express_dsp.c b/sound/pci/echoaudio/indigo_express_dsp.c new file mode 100644 index 00000000000..9ab625e1565 --- /dev/null +++ b/sound/pci/echoaudio/indigo_express_dsp.c @@ -0,0 +1,119 @@ +/************************************************************************ + +This file is part of Echo Digital Audio's generic driver library. +Copyright Echo Digital Audio Corporation (c) 1998 - 2005 +All rights reserved +www.echoaudio.com + +This library is free software; you can redistribute it and/or +modify it under the terms of the GNU Lesser General Public +License as published by the Free Software Foundation; either +version 2.1 of the License, or (at your option) any later version. + +This library is distributed in the hope that it will be useful, +but WITHOUT ANY WARRANTY; without even the implied warranty of +MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +Lesser General Public License for more details. + +You should have received a copy of the GNU Lesser General Public +License along with this library; if not, write to the Free Software +Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + +************************************************************************* + + Translation from C++ and adaptation for use in ALSA-Driver + were made by Giuliano Pochini + +*************************************************************************/ + +static int set_sample_rate(struct echoaudio *chip, u32 rate) +{ + u32 clock, control_reg, old_control_reg; + + if (wait_handshake(chip)) + return -EIO; + + old_control_reg = le32_to_cpu(chip->comm_page->control_register); + control_reg = old_control_reg & ~INDIGO_EXPRESS_CLOCK_MASK; + + switch (rate) { + case 32000: + clock = INDIGO_EXPRESS_32000; + break; + case 44100: + clock = INDIGO_EXPRESS_44100; + break; + case 48000: + clock = INDIGO_EXPRESS_48000; + break; + case 64000: + clock = INDIGO_EXPRESS_32000|INDIGO_EXPRESS_DOUBLE_SPEED; + break; + case 88200: + clock = INDIGO_EXPRESS_44100|INDIGO_EXPRESS_DOUBLE_SPEED; + break; + case 96000: + clock = INDIGO_EXPRESS_48000|INDIGO_EXPRESS_DOUBLE_SPEED; + break; + default: + return -EINVAL; + } + + control_reg |= clock; + if (control_reg != old_control_reg) { + chip->comm_page->control_register = cpu_to_le32(control_reg); + chip->sample_rate = rate; + clear_handshake(chip); + return send_vector(chip, DSP_VC_UPDATE_CLOCKS); + } + return 0; +} + + + +/* This function routes the sound from a virtual channel to a real output */ +static int set_vmixer_gain(struct echoaudio *chip, u16 output, u16 pipe, + int gain) +{ + int index; + + if (snd_BUG_ON(pipe >= num_pipes_out(chip) || + output >= num_busses_out(chip))) + return -EINVAL; + + if (wait_handshake(chip)) + return -EIO; + + chip->vmixer_gain[output][pipe] = gain; + index = output * num_pipes_out(chip) + pipe; + chip->comm_page->vmixer[index] = gain; + + DE_ACT(("set_vmixer_gain: pipe %d, out %d = %d\n", pipe, output, gain)); + return 0; +} + + + +/* Tell the DSP to read and update virtual mixer levels in comm page. */ +static int update_vmixer_level(struct echoaudio *chip) +{ + if (wait_handshake(chip)) + return -EIO; + clear_handshake(chip); + return send_vector(chip, DSP_VC_SET_VMIXER_GAIN); +} + + + +static u32 detect_input_clocks(const struct echoaudio *chip) +{ + return ECHO_CLOCK_BIT_INTERNAL; +} + + + +/* The IndigoIO has no ASIC. Just do nothing */ +static int load_asic(struct echoaudio *chip) +{ + return 0; +} diff --git a/sound/pci/echoaudio/indigodjx.c b/sound/pci/echoaudio/indigodjx.c new file mode 100644 index 00000000000..3482ef69f49 --- /dev/null +++ b/sound/pci/echoaudio/indigodjx.c @@ -0,0 +1,107 @@ +/* + * ALSA driver for Echoaudio soundcards. + * Copyright (C) 2009 Giuliano Pochini + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. + */ + +#define INDIGO_FAMILY +#define ECHOCARD_INDIGO_DJX +#define ECHOCARD_NAME "Indigo DJx" +#define ECHOCARD_HAS_SUPER_INTERLEAVE +#define ECHOCARD_HAS_VMIXER +#define ECHOCARD_HAS_STEREO_BIG_ENDIAN32 + +/* Pipe indexes */ +#define PX_ANALOG_OUT 0 /* 8 */ +#define PX_DIGITAL_OUT 8 /* 0 */ +#define PX_ANALOG_IN 8 /* 0 */ +#define PX_DIGITAL_IN 8 /* 0 */ +#define PX_NUM 8 + +/* Bus indexes */ +#define BX_ANALOG_OUT 0 /* 4 */ +#define BX_DIGITAL_OUT 4 /* 0 */ +#define BX_ANALOG_IN 4 /* 0 */ +#define BX_DIGITAL_IN 4 /* 0 */ +#define BX_NUM 4 + + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "echoaudio.h" + +MODULE_FIRMWARE("ea/loader_dsp.fw"); +MODULE_FIRMWARE("ea/indigo_djx_dsp.fw"); + +#define FW_361_LOADER 0 +#define FW_INDIGO_DJX_DSP 1 + +static const struct firmware card_fw[] = { + {0, "loader_dsp.fw"}, + {0, "indigo_djx_dsp.fw"} +}; + +static struct pci_device_id snd_echo_ids[] = { + {0x1057, 0x3410, 0xECC0, 0x00E0, 0, 0, 0}, /* Indigo DJx*/ + {0,} +}; + +static struct snd_pcm_hardware pcm_hardware_skel = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_SYNC_START, + .formats = SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_3LE | + SNDRV_PCM_FMTBIT_S32_LE | + SNDRV_PCM_FMTBIT_S32_BE, + .rates = SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 32000, + .rate_max = 96000, + .channels_min = 1, + .channels_max = 4, + .buffer_bytes_max = 262144, + .period_bytes_min = 32, + .period_bytes_max = 131072, + .periods_min = 2, + .periods_max = 220, +}; + +#include "indigodjx_dsp.c" +#include "indigo_express_dsp.c" +#include "echoaudio_dsp.c" +#include "echoaudio.c" diff --git a/sound/pci/echoaudio/indigodjx_dsp.c b/sound/pci/echoaudio/indigodjx_dsp.c new file mode 100644 index 00000000000..f591fc2ed96 --- /dev/null +++ b/sound/pci/echoaudio/indigodjx_dsp.c @@ -0,0 +1,68 @@ +/************************************************************************ + +This file is part of Echo Digital Audio's generic driver library. +Copyright Echo Digital Audio Corporation (c) 1998 - 2005 +All rights reserved +www.echoaudio.com + +This library is free software; you can redistribute it and/or +modify it under the terms of the GNU Lesser General Public +License as published by the Free Software Foundation; either +version 2.1 of the License, or (at your option) any later version. + +This library is distributed in the hope that it will be useful, +but WITHOUT ANY WARRANTY; without even the implied warranty of +MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +Lesser General Public License for more details. + +You should have received a copy of the GNU Lesser General Public +License along with this library; if not, write to the Free Software +Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + +************************************************************************* + + Translation from C++ and adaptation for use in ALSA-Driver + were made by Giuliano Pochini + +*************************************************************************/ + +static int update_vmixer_level(struct echoaudio *chip); +static int set_vmixer_gain(struct echoaudio *chip, u16 output, + u16 pipe, int gain); + + +static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) +{ + int err; + + DE_INIT(("init_hw() - Indigo DJx\n")); + if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_DJX)) + return -ENODEV; + + err = init_dsp_comm_page(chip); + if (err < 0) { + DE_INIT(("init_hw - could not initialize DSP comm page\n")); + return err; + } + + chip->device_id = device_id; + chip->subdevice_id = subdevice_id; + chip->bad_board = TRUE; + chip->dsp_code_to_load = &card_fw[FW_INDIGO_DJX_DSP]; + /* Since this card has no ASIC, mark it as loaded so everything + works OK */ + chip->asic_loaded = TRUE; + chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL; + + err = load_firmware(chip); + if (err < 0) + return err; + chip->bad_board = FALSE; + + err = init_line_levels(chip); + if (err < 0) + return err; + + DE_INIT(("init_hw done\n")); + return err; +} diff --git a/sound/pci/echoaudio/indigoiox.c b/sound/pci/echoaudio/indigoiox.c new file mode 100644 index 00000000000..aebee27a40f --- /dev/null +++ b/sound/pci/echoaudio/indigoiox.c @@ -0,0 +1,109 @@ +/* + * ALSA driver for Echoaudio soundcards. + * Copyright (C) 2009 Giuliano Pochini + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. + */ + +#define INDIGO_FAMILY +#define ECHOCARD_INDIGO_IOX +#define ECHOCARD_NAME "Indigo IOx" +#define ECHOCARD_HAS_MONITOR +#define ECHOCARD_HAS_SUPER_INTERLEAVE +#define ECHOCARD_HAS_VMIXER +#define ECHOCARD_HAS_STEREO_BIG_ENDIAN32 + +/* Pipe indexes */ +#define PX_ANALOG_OUT 0 /* 8 */ +#define PX_DIGITAL_OUT 8 /* 0 */ +#define PX_ANALOG_IN 8 /* 2 */ +#define PX_DIGITAL_IN 10 /* 0 */ +#define PX_NUM 10 + +/* Bus indexes */ +#define BX_ANALOG_OUT 0 /* 2 */ +#define BX_DIGITAL_OUT 2 /* 0 */ +#define BX_ANALOG_IN 2 /* 2 */ +#define BX_DIGITAL_IN 4 /* 0 */ +#define BX_NUM 4 + + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "echoaudio.h" + +MODULE_FIRMWARE("ea/loader_dsp.fw"); +MODULE_FIRMWARE("ea/indigo_iox_dsp.fw"); + +#define FW_361_LOADER 0 +#define FW_INDIGO_IOX_DSP 1 + +static const struct firmware card_fw[] = { + {0, "loader_dsp.fw"}, + {0, "indigo_iox_dsp.fw"} +}; + +static struct pci_device_id snd_echo_ids[] = { + {0x1057, 0x3410, 0xECC0, 0x00D0, 0, 0, 0}, /* Indigo IOx */ + {0,} +}; + +static struct snd_pcm_hardware pcm_hardware_skel = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_SYNC_START, + .formats = SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_3LE | + SNDRV_PCM_FMTBIT_S32_LE | + SNDRV_PCM_FMTBIT_S32_BE, + .rates = SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 32000, + .rate_max = 96000, + .channels_min = 1, + .channels_max = 8, + .buffer_bytes_max = 262144, + .period_bytes_min = 32, + .period_bytes_max = 131072, + .periods_min = 2, + .periods_max = 220, +}; + +#include "indigoiox_dsp.c" +#include "indigo_express_dsp.c" +#include "echoaudio_dsp.c" +#include "echoaudio.c" + diff --git a/sound/pci/echoaudio/indigoiox_dsp.c b/sound/pci/echoaudio/indigoiox_dsp.c new file mode 100644 index 00000000000..f357521c79e --- /dev/null +++ b/sound/pci/echoaudio/indigoiox_dsp.c @@ -0,0 +1,68 @@ +/************************************************************************ + +This file is part of Echo Digital Audio's generic driver library. +Copyright Echo Digital Audio Corporation (c) 1998 - 2005 +All rights reserved +www.echoaudio.com + +This library is free software; you can redistribute it and/or +modify it under the terms of the GNU Lesser General Public +License as published by the Free Software Foundation; either +version 2.1 of the License, or (at your option) any later version. + +This library is distributed in the hope that it will be useful, +but WITHOUT ANY WARRANTY; without even the implied warranty of +MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +Lesser General Public License for more details. + +You should have received a copy of the GNU Lesser General Public +License along with this library; if not, write to the Free Software +Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + +************************************************************************* + + Translation from C++ and adaptation for use in ALSA-Driver + were made by Giuliano Pochini + +*************************************************************************/ + +static int update_vmixer_level(struct echoaudio *chip); +static int set_vmixer_gain(struct echoaudio *chip, u16 output, + u16 pipe, int gain); + + +static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) +{ + int err; + + DE_INIT(("init_hw() - Indigo IOx\n")); + if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_IOX)) + return -ENODEV; + + err = init_dsp_comm_page(chip); + if (err < 0) { + DE_INIT(("init_hw - could not initialize DSP comm page\n")); + return err; + } + + chip->device_id = device_id; + chip->subdevice_id = subdevice_id; + chip->bad_board = TRUE; + chip->dsp_code_to_load = &card_fw[FW_INDIGO_IOX_DSP]; + /* Since this card has no ASIC, mark it as loaded so everything + works OK */ + chip->asic_loaded = TRUE; + chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL; + + err = load_firmware(chip); + if (err < 0) + return err; + chip->bad_board = FALSE; + + err = init_line_levels(chip); + if (err < 0) + return err; + + DE_INIT(("init_hw done\n")); + return err; +} -- cgit v1.2.3-70-g09d2 From cad377acf3d6af6279622048e96680e79e352183 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 19 Mar 2009 09:55:15 +0100 Subject: ALSA: pcm - Fix a typo in error messages Fix a typo in error messages; forgotten after a copy&paste error. Signed-off-by: Takashi Iwai --- sound/core/pcm_lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 86ac9ae9460..2ff25ed4d83 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -194,7 +194,7 @@ static inline int snd_pcm_update_hw_ptr_post(struct snd_pcm_substream *substream do { \ if (xrun_debug(substream)) { \ if (printk_ratelimit()) { \ - snd_printd("hda_codec: " fmt, ##args); \ + snd_printd("PCM: " fmt, ##args); \ } \ dump_stack_on_xrun(substream); \ } \ -- cgit v1.2.3-70-g09d2 From 98204646f2b15d368701265e4194b773a6f94600 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 19 Mar 2009 09:59:21 +0100 Subject: ALSA: pcm - avoid unnecessary inline Remove unnecessary explicit inlininig of internal functions. Let compiler optimize. Signed-off-by: Takashi Iwai --- sound/core/pcm_lib.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 2ff25ed4d83..302654769fa 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -148,8 +148,9 @@ static void xrun(struct snd_pcm_substream *substream) } } -static inline snd_pcm_uframes_t snd_pcm_update_hw_ptr_pos(struct snd_pcm_substream *substream, - struct snd_pcm_runtime *runtime) +static snd_pcm_uframes_t +snd_pcm_update_hw_ptr_pos(struct snd_pcm_substream *substream, + struct snd_pcm_runtime *runtime) { snd_pcm_uframes_t pos; @@ -167,8 +168,8 @@ static inline snd_pcm_uframes_t snd_pcm_update_hw_ptr_pos(struct snd_pcm_substre return pos; } -static inline int snd_pcm_update_hw_ptr_post(struct snd_pcm_substream *substream, - struct snd_pcm_runtime *runtime) +static int snd_pcm_update_hw_ptr_post(struct snd_pcm_substream *substream, + struct snd_pcm_runtime *runtime) { snd_pcm_uframes_t avail; @@ -200,7 +201,7 @@ static inline int snd_pcm_update_hw_ptr_post(struct snd_pcm_substream *substream } \ } while (0) -static inline int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) +static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_uframes_t pos; -- cgit v1.2.3-70-g09d2 From 5f513e1197f27e9a0bcfec0feaac59f976f4a37e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 19 Mar 2009 10:01:47 +0100 Subject: ALSA: pcm - Reset invalid position even without debug option Always reset the invalind hw_ptr position returned by the pointer callback. The behavior should be consitent independently from the debug option. Also, add the printk_ratelimit() check to avoid flooding debug prints. Signed-off-by: Takashi Iwai --- sound/core/pcm_lib.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 302654769fa..92ed6d81922 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -159,11 +159,15 @@ snd_pcm_update_hw_ptr_pos(struct snd_pcm_substream *substream, pos = substream->ops->pointer(substream); if (pos == SNDRV_PCM_POS_XRUN) return pos; /* XRUN */ -#ifdef CONFIG_SND_DEBUG if (pos >= runtime->buffer_size) { - snd_printk(KERN_ERR "BUG: stream = %i, pos = 0x%lx, buffer size = 0x%lx, period size = 0x%lx\n", substream->stream, pos, runtime->buffer_size, runtime->period_size); + if (printk_ratelimit()) { + snd_printd(KERN_ERR "BUG: stream = %i, pos = 0x%lx, " + "buffer size = 0x%lx, period size = 0x%lx\n", + substream->stream, pos, runtime->buffer_size, + runtime->period_size); + } + pos = 0; } -#endif pos -= pos % runtime->min_align; return pos; } -- cgit v1.2.3-70-g09d2 From ded652f7024bc2d7b6118b561a44187af30841b0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 19 Mar 2009 10:08:49 +0100 Subject: ALSA: pcm - Fix delta calculation at boundary overlap When the hw_ptr_interrupt reaches the boundary, it must check whether the hw_base was already lapped and corret the delta value appropriately. Also, rebasing the hw_ptr needs a correction because buffer_size isn't always aligned to period_size. Signed-off-by: Takashi Iwai --- sound/core/pcm_lib.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 92ed6d81922..063c675177a 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -221,8 +221,11 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) new_hw_ptr = hw_base + pos; hw_ptr_interrupt = runtime->hw_ptr_interrupt + runtime->period_size; delta = new_hw_ptr - hw_ptr_interrupt; - if (hw_ptr_interrupt == runtime->boundary) - hw_ptr_interrupt = 0; + if (hw_ptr_interrupt >= runtime->boundary) { + hw_ptr_interrupt %= runtime->boundary; + if (!hw_base) /* hw_base was already lapped; recalc delta */ + delta = new_hw_ptr - hw_ptr_interrupt; + } if (delta < 0) { delta += runtime->buffer_size; if (delta < 0) { @@ -233,6 +236,8 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) (long)hw_ptr_interrupt); /* rebase to interrupt position */ hw_base = new_hw_ptr = hw_ptr_interrupt; + /* align hw_base to buffer_size */ + hw_base -= hw_base % runtime->buffer_size; delta = 0; } else { hw_base += runtime->buffer_size; -- cgit v1.2.3-70-g09d2 From 97b71c94d691728b82052e9c4d6286fbc9965d7f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 18 Mar 2009 15:09:13 +0100 Subject: ALSA: hda - Don't reset BDL unnecessarily So far, the prepare callback is called multiple times, BDL entries are reset and re-programmed at each time. This patch adds the check to avoid the reset of BDL entries when the same parameters are used. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 46 ++++++++++++++++++++++++++++++++-------------- 1 file changed, 32 insertions(+), 14 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 6bcf5af6edc..ba97795d89c 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1076,8 +1076,7 @@ static int azx_setup_periods(struct azx *chip, azx_sd_writel(azx_dev, SD_BDLPL, 0); azx_sd_writel(azx_dev, SD_BDLPU, 0); - period_bytes = snd_pcm_lib_period_bytes(substream); - azx_dev->period_bytes = period_bytes; + period_bytes = azx_dev->period_bytes; periods = azx_dev->bufsize / period_bytes; /* program the initial BDL entries */ @@ -1124,9 +1123,6 @@ static int azx_setup_periods(struct azx *chip, error: snd_printk(KERN_ERR "Too many BDL entries: buffer=%d, period=%d\n", azx_dev->bufsize, period_bytes); - /* reset */ - azx_sd_writel(azx_dev, SD_BDLPL, 0); - azx_sd_writel(azx_dev, SD_BDLPU, 0); return -EINVAL; } @@ -1429,6 +1425,11 @@ static int azx_pcm_close(struct snd_pcm_substream *substream) static int azx_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { + struct azx_dev *azx_dev = get_azx_dev(substream); + + azx_dev->bufsize = 0; + azx_dev->period_bytes = 0; + azx_dev->format_val = 0; return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); } @@ -1443,6 +1444,9 @@ static int azx_pcm_hw_free(struct snd_pcm_substream *substream) azx_sd_writel(azx_dev, SD_BDLPL, 0); azx_sd_writel(azx_dev, SD_BDLPU, 0); azx_sd_writel(azx_dev, SD_CTL, 0); + azx_dev->bufsize = 0; + azx_dev->period_bytes = 0; + azx_dev->format_val = 0; hinfo->ops.cleanup(hinfo, apcm->codec, substream); @@ -1456,23 +1460,37 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) struct azx_dev *azx_dev = get_azx_dev(substream); struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream]; struct snd_pcm_runtime *runtime = substream->runtime; + unsigned int bufsize, period_bytes, format_val; + int err; - azx_dev->bufsize = snd_pcm_lib_buffer_bytes(substream); - azx_dev->format_val = snd_hda_calc_stream_format(runtime->rate, - runtime->channels, - runtime->format, - hinfo->maxbps); - if (!azx_dev->format_val) { + format_val = snd_hda_calc_stream_format(runtime->rate, + runtime->channels, + runtime->format, + hinfo->maxbps); + if (!format_val) { snd_printk(KERN_ERR SFX "invalid format_val, rate=%d, ch=%d, format=%d\n", runtime->rate, runtime->channels, runtime->format); return -EINVAL; } + bufsize = snd_pcm_lib_buffer_bytes(substream); + period_bytes = snd_pcm_lib_period_bytes(substream); + snd_printdd("azx_pcm_prepare: bufsize=0x%x, format=0x%x\n", - azx_dev->bufsize, azx_dev->format_val); - if (azx_setup_periods(chip, substream, azx_dev) < 0) - return -EINVAL; + bufsize, format_val); + + if (bufsize != azx_dev->bufsize || + period_bytes != azx_dev->period_bytes || + format_val != azx_dev->format_val) { + azx_dev->bufsize = bufsize; + azx_dev->period_bytes = period_bytes; + azx_dev->format_val = format_val; + err = azx_setup_periods(chip, substream, azx_dev); + if (err < 0) + return err; + } + azx_setup_controller(chip, azx_dev); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) azx_dev->fifo_size = azx_sd_readw(azx_dev, SD_FIFOSIZE) + 1; -- cgit v1.2.3-70-g09d2 From 1dddab400b7ad028b21d7d5b060e4a068d6d3cd9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 18 Mar 2009 15:15:37 +0100 Subject: ALSA: hda - Don't reset stream at each prepare callback Don't reset the stream at each prepare callback but do it only once after the open. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 33 ++++++++++++++++++++++----------- 1 file changed, 22 insertions(+), 11 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index ba97795d89c..8b2e4160de8 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -859,13 +859,18 @@ static void azx_stream_start(struct azx *chip, struct azx_dev *azx_dev) SD_CTL_DMA_START | SD_INT_MASK); } -/* stop a stream */ -static void azx_stream_stop(struct azx *chip, struct azx_dev *azx_dev) +/* stop DMA */ +static void azx_stream_clear(struct azx *chip, struct azx_dev *azx_dev) { - /* stop DMA */ azx_sd_writeb(azx_dev, SD_CTL, azx_sd_readb(azx_dev, SD_CTL) & ~(SD_CTL_DMA_START | SD_INT_MASK)); azx_sd_writeb(azx_dev, SD_STS, SD_INT_MASK); /* to be sure */ +} + +/* stop a stream */ +static void azx_stream_stop(struct azx *chip, struct azx_dev *azx_dev) +{ + azx_stream_clear(chip, azx_dev); /* disable SIE */ azx_writeb(chip, INTCTL, azx_readb(chip, INTCTL) & ~(1 << azx_dev->index)); @@ -1126,18 +1131,14 @@ static int azx_setup_periods(struct azx *chip, return -EINVAL; } -/* - * set up the SD for streaming - */ -static int azx_setup_controller(struct azx *chip, struct azx_dev *azx_dev) +/* reset stream */ +static void azx_stream_reset(struct azx *chip, struct azx_dev *azx_dev) { unsigned char val; int timeout; - /* make sure the run bit is zero for SD */ - azx_sd_writeb(azx_dev, SD_CTL, azx_sd_readb(azx_dev, SD_CTL) & - ~SD_CTL_DMA_START); - /* reset stream */ + azx_stream_clear(chip, azx_dev); + azx_sd_writeb(azx_dev, SD_CTL, azx_sd_readb(azx_dev, SD_CTL) | SD_CTL_STREAM_RESET); udelay(3); @@ -1154,7 +1155,15 @@ static int azx_setup_controller(struct azx *chip, struct azx_dev *azx_dev) while (((val = azx_sd_readb(azx_dev, SD_CTL)) & SD_CTL_STREAM_RESET) && --timeout) ; +} +/* + * set up the SD for streaming + */ +static int azx_setup_controller(struct azx *chip, struct azx_dev *azx_dev) +{ + /* make sure the run bit is zero for SD */ + azx_stream_clear(chip, azx_dev); /* program the stream_tag */ azx_sd_writel(azx_dev, SD_CTL, (azx_sd_readl(azx_dev, SD_CTL) & ~SD_CTL_STREAM_TAG_MASK)| @@ -1399,6 +1408,8 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) runtime->private_data = azx_dev; snd_pcm_set_sync(substream); mutex_unlock(&chip->open_mutex); + + azx_stream_reset(chip, azx_dev); return 0; } -- cgit v1.2.3-70-g09d2 From e8523b641cddedec754ae5e44ec579dbceea5ef4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 18 Mar 2009 18:28:01 +0000 Subject: ASoC: Add FLL support for WM8400 Signed-off-by: Mark Brown --- sound/soc/codecs/wm8400.c | 129 ++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 129 insertions(+) diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index b7350c25b61..510efa60400 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -70,6 +70,7 @@ struct wm8400_priv { unsigned int sysclk; unsigned int pcmclk; struct work_struct work; + int fll_in, fll_out; }; static inline unsigned int wm8400_read(struct snd_soc_codec *codec, @@ -931,6 +932,133 @@ static int wm8400_set_dai_sysclk(struct snd_soc_dai *codec_dai, return 0; } +struct fll_factors { + u16 n; + u16 k; + u16 outdiv; + u16 fratio; + u16 freq_ref; +}; + +#define FIXED_FLL_SIZE ((1 << 16) * 10) + +static int fll_factors(struct wm8400_priv *wm8400, struct fll_factors *factors, + unsigned int Fref, unsigned int Fout) +{ + u64 Kpart; + unsigned int K, Nmod, target; + + factors->outdiv = 2; + while (Fout * factors->outdiv < 90000000 || + Fout * factors->outdiv > 100000000) { + factors->outdiv *= 2; + if (factors->outdiv > 32) { + dev_err(wm8400->wm8400->dev, + "Unsupported FLL output frequency %dHz\n", + Fout); + return -EINVAL; + } + } + target = Fout * factors->outdiv; + factors->outdiv = factors->outdiv >> 2; + + if (Fref < 48000) + factors->freq_ref = 1; + else + factors->freq_ref = 0; + + if (Fref < 1000000) + factors->fratio = 9; + else + factors->fratio = 0; + + /* Ensure we have a fractional part */ + do { + if (Fref < 1000000) + factors->fratio--; + else + factors->fratio++; + + if (factors->fratio < 1 || factors->fratio > 8) { + dev_err(wm8400->wm8400->dev, + "Unable to calculate FRATIO\n"); + return -EINVAL; + } + + factors->n = target / (Fref * factors->fratio); + Nmod = target % (Fref * factors->fratio); + } while (Nmod == 0); + + /* Calculate fractional part - scale up so we can round. */ + Kpart = FIXED_FLL_SIZE * (long long)Nmod; + + do_div(Kpart, (Fref * factors->fratio)); + + K = Kpart & 0xFFFFFFFF; + + if ((K % 10) >= 5) + K += 5; + + /* Move down to proper range now rounding is done */ + factors->k = K / 10; + + dev_dbg(wm8400->wm8400->dev, + "FLL: Fref=%d Fout=%d N=%x K=%x, FRATIO=%x OUTDIV=%x\n", + Fref, Fout, + factors->n, factors->k, factors->fratio, factors->outdiv); + + return 0; +} + +static int wm8400_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8400_priv *wm8400 = codec->private_data; + struct fll_factors factors; + int ret; + u16 reg; + + if (freq_in == wm8400->fll_in && freq_out == wm8400->fll_out) + return 0; + + if (freq_out != 0) { + ret = fll_factors(wm8400, &factors, freq_in, freq_out); + if (ret != 0) + return ret; + } + + wm8400->fll_out = freq_out; + wm8400->fll_in = freq_in; + + /* We *must* disable the FLL before any changes */ + reg = wm8400_read(codec, WM8400_POWER_MANAGEMENT_2); + reg &= ~WM8400_FLL_ENA; + wm8400_write(codec, WM8400_POWER_MANAGEMENT_2, reg); + + reg = wm8400_read(codec, WM8400_FLL_CONTROL_1); + reg &= ~WM8400_FLL_OSC_ENA; + wm8400_write(codec, WM8400_FLL_CONTROL_1, reg); + + if (freq_out == 0) + return 0; + + reg &= ~(WM8400_FLL_REF_FREQ | WM8400_FLL_FRATIO_MASK); + reg |= WM8400_FLL_FRAC | factors.fratio; + reg |= factors.freq_ref << WM8400_FLL_REF_FREQ_SHIFT; + wm8400_write(codec, WM8400_FLL_CONTROL_1, reg); + + wm8400_write(codec, WM8400_FLL_CONTROL_2, factors.k); + wm8400_write(codec, WM8400_FLL_CONTROL_3, factors.n); + + reg = wm8400_read(codec, WM8400_FLL_CONTROL_4); + reg &= WM8400_FLL_OUTDIV_MASK; + reg |= factors.outdiv; + wm8400_write(codec, WM8400_FLL_CONTROL_4, reg); + + return 0; +} + /* * Sets ADC and Voice DAC format. */ @@ -1188,6 +1316,7 @@ static struct snd_soc_dai_ops wm8400_dai_ops = { .set_fmt = wm8400_set_dai_fmt, .set_clkdiv = wm8400_set_dai_clkdiv, .set_sysclk = wm8400_set_dai_sysclk, + .set_pll = wm8400_set_dai_pll, }; /* -- cgit v1.2.3-70-g09d2 From 13b9d2ab5921d77df5217a2104b687a1c729d521 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Wed, 18 Mar 2009 16:46:53 +0200 Subject: ASoC: OMAP: N810: Mark not connected input pins Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/omap/n810.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 86471fd6340..f203221d9cb 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -254,6 +254,11 @@ static int n810_aic33_init(struct snd_soc_codec *codec) snd_soc_dapm_nc_pin(codec, "MONO_LOUT"); snd_soc_dapm_nc_pin(codec, "HPLCOM"); snd_soc_dapm_nc_pin(codec, "HPRCOM"); + snd_soc_dapm_nc_pin(codec, "MIC3L"); + snd_soc_dapm_nc_pin(codec, "MIC3R"); + snd_soc_dapm_nc_pin(codec, "LINE1R"); + snd_soc_dapm_nc_pin(codec, "LINE2L"); + snd_soc_dapm_nc_pin(codec, "LINE2R"); /* Add N810 specific controls */ err = snd_soc_add_controls(codec, aic33_n810_controls, -- cgit v1.2.3-70-g09d2 From f8d5fc924b1bec95f47b0a9e8f9a6e22fa1c7b05 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Wed, 18 Mar 2009 16:46:54 +0200 Subject: ASoC: OMAP: N810: Add more jack functions Add functions "Headset" and "Mic" to the control "Jack Function" for activating and de-activating codec input pin LINE1L which is connected to the mic pin of 4-pole Nokia AV connecter. Note there is no mic bias voltage management here since bias is coming from Nokia ASIC and driver for it is not in mainline. Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/omap/n810.c | 28 ++++++++++++++++++++++++++-- 1 file changed, 26 insertions(+), 2 deletions(-) diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index f203221d9cb..a6d1178ce12 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -40,6 +40,13 @@ #define N810_HEADSET_AMP_GPIO 10 #define N810_SPEAKER_AMP_GPIO 101 +enum { + N810_JACK_DISABLED, + N810_JACK_HP, + N810_JACK_HS, + N810_JACK_MIC, +}; + static struct clk *sys_clkout2; static struct clk *sys_clkout2_src; static struct clk *func96m_clk; @@ -50,15 +57,32 @@ static int n810_dmic_func; static void n810_ext_control(struct snd_soc_codec *codec) { + int hp = 0, line1l = 0; + + switch (n810_jack_func) { + case N810_JACK_HS: + line1l = 1; + case N810_JACK_HP: + hp = 1; + break; + case N810_JACK_MIC: + line1l = 1; + break; + } + if (n810_spk_func) snd_soc_dapm_enable_pin(codec, "Ext Spk"); else snd_soc_dapm_disable_pin(codec, "Ext Spk"); - if (n810_jack_func) + if (hp) snd_soc_dapm_enable_pin(codec, "Headphone Jack"); else snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + if (line1l) + snd_soc_dapm_enable_pin(codec, "LINE1L"); + else + snd_soc_dapm_disable_pin(codec, "LINE1L"); if (n810_dmic_func) snd_soc_dapm_enable_pin(codec, "DMic"); @@ -229,7 +253,7 @@ static const struct snd_soc_dapm_route audio_map[] = { }; static const char *spk_function[] = {"Off", "On"}; -static const char *jack_function[] = {"Off", "Headphone"}; +static const char *jack_function[] = {"Off", "Headphone", "Headset", "Mic"}; static const char *input_function[] = {"ADC", "Digital Mic"}; static const struct soc_enum n810_enum[] = { SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(spk_function), spk_function), -- cgit v1.2.3-70-g09d2 From 632087748c3795a54d5631e640df65592774e045 Mon Sep 17 00:00:00 2001 From: "Lopez Cruz, Misael" Date: Thu, 19 Mar 2009 01:07:34 -0500 Subject: ASoC: Declare Headset as Mic and Headphone widgets for SDP3430 Headset was declared previously as a Headphone widget connecting HSMIC and HSOL/HSOR pins of TWL4030 codec in SDP430 machine driver. The capture path becomes invalid as the Headphone widget is not a valid input endpoint. Instead of that, the Headset is declared as separate Microphone and Headphone widgets. Current patch modifies audio map: - Headset Mic: HSMIC with bias - Headset Stereophone: HSOL, HSOR Signed-off-by: Misael Lopez Cruz Signed-off-by: Mark Brown --- sound/soc/omap/sdp3430.c | 24 ++++++++++++++++-------- 1 file changed, 16 insertions(+), 8 deletions(-) diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index 0a41de677e7..10f1c867f11 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -90,8 +90,12 @@ static struct snd_soc_jack hs_jack; /* Headset jack detection DAPM pins */ static struct snd_soc_jack_pin hs_jack_pins[] = { { - .pin = "Headset Jack", - .mask = SND_JACK_HEADSET, + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Headset Stereophone", + .mask = SND_JACK_HEADPHONE, }, }; @@ -109,7 +113,8 @@ static struct snd_soc_jack_gpio hs_jack_gpios[] = { static const struct snd_soc_dapm_widget sdp3430_twl4030_dapm_widgets[] = { SND_SOC_DAPM_MIC("Ext Mic", NULL), SND_SOC_DAPM_SPK("Ext Spk", NULL), - SND_SOC_DAPM_HP("Headset Jack", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_HP("Headset Stereophone", NULL), }; static const struct snd_soc_dapm_route audio_map[] = { @@ -123,11 +128,13 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Ext Spk", NULL, "HFL"}, {"Ext Spk", NULL, "HFR"}, - /* Headset: HSMIC (with bias), HSOL, HSOR */ - {"Headset Jack", NULL, "HSOL"}, - {"Headset Jack", NULL, "HSOR"}, + /* Headset Mic: HSMIC with bias */ {"HSMIC", NULL, "Headset Mic Bias"}, - {"Headset Mic Bias", NULL, "Headset Jack"}, + {"Headset Mic Bias", NULL, "Headset Mic"}, + + /* Headset Stereophone (Headphone): HSOL, HSOR */ + {"Headset Stereophone", NULL, "HSOL"}, + {"Headset Stereophone", NULL, "HSOR"}, }; static int sdp3430_twl4030_init(struct snd_soc_codec *codec) @@ -146,7 +153,8 @@ static int sdp3430_twl4030_init(struct snd_soc_codec *codec) /* SDP3430 connected pins */ snd_soc_dapm_enable_pin(codec, "Ext Mic"); snd_soc_dapm_enable_pin(codec, "Ext Spk"); - snd_soc_dapm_disable_pin(codec, "Headset Jack"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Headset Stereophone"); /* TWL4030 not connected pins */ snd_soc_dapm_nc_pin(codec, "AUXL"); -- cgit v1.2.3-70-g09d2 From 1f2186951e02f2a5bcda9459f63136918932385a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 19 Mar 2009 14:08:58 +0100 Subject: ALSA: Fix wrong pointer to dev_err() in arm/pxa2xx-ac97-lib.c Fix the wrong device pointer passed to dev_err(). Signed-off-by: Takashi Iwai --- sound/arm/pxa2xx-ac97-lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index d721ea7cae8..2e6355f4cbb 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -374,7 +374,7 @@ int __devinit pxa2xx_ac97_hw_probe(struct platform_device *dev) case -1: break; default: - dev_err(dev, "Invalid reset GPIO %d\n", + dev_err(&dev->dev, "Invalid reset GPIO %d\n", pdata->reset_gpio); } } else { -- cgit v1.2.3-70-g09d2 From 07a1e81355245ca65ab16c7b4ae2332e52ed7acd Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 19 Mar 2009 17:08:19 +0100 Subject: ALSA: hda - Don't show the current connection for power widgets The power-widgets have no connection selection, so skip the check in proc output, too. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_proc.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 93b25ba4d00..639cf0edaa9 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -399,8 +399,10 @@ static void print_conn_list(struct snd_info_buffer *buffer, { int c, curr = -1; - if (conn_len > 1 && wid_type != AC_WID_AUD_MIX && - wid_type != AC_WID_VOL_KNB) + if (conn_len > 1 && + wid_type != AC_WID_AUD_MIX && + wid_type != AC_WID_VOL_KNB && + wid_type != AC_WID_POWER) curr = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONNECT_SEL, 0); snd_iprintf(buffer, " Connection: %d\n", conn_len); -- cgit v1.2.3-70-g09d2 From c468ac29e63b9927275a94379d00b367f0f97c43 Mon Sep 17 00:00:00 2001 From: Wolfram Sang Date: Fri, 20 Mar 2009 10:08:11 +0100 Subject: ALSA: sound/ali5451: typo: s/resouces/resources/ Signed-off-by: Wolfram Sang Signed-off-by: Takashi Iwai --- sound/pci/ali5451/ali5451.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index 1a0fd65ec28..9069c78c2dc 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -2142,7 +2142,7 @@ static int __devinit snd_ali_resources(struct snd_ali *codec) { int err; - snd_ali_printk("resouces allocation ...\n"); + snd_ali_printk("resources allocation ...\n"); err = pci_request_regions(codec->pci, "ALI 5451"); if (err < 0) return err; @@ -2154,7 +2154,7 @@ static int __devinit snd_ali_resources(struct snd_ali *codec) return -EBUSY; } codec->irq = codec->pci->irq; - snd_ali_printk("resouces allocated.\n"); + snd_ali_printk("resources allocated.\n"); return 0; } static int snd_ali_dev_free(struct snd_device *device) -- cgit v1.2.3-70-g09d2 From 2d864c499a77129dc6aa4f7552ddf2885e4a9c47 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 20 Mar 2009 12:52:47 +0100 Subject: ALSA: hda - Detect digital-mic inputs on ALC663 / ALC272 Fix the detection of digital-mic inputs on ALC663 / ALC272 codecs in the auto-detection mode. The automatic mic switch via plugging isn't implemented yet, though. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 63 ++++++++++++++++++++++++++++++++----------- 1 file changed, 47 insertions(+), 16 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 5ad0f8d72dd..b69d9864f6f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -16725,26 +16725,58 @@ static int alc662_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin, return 0; } +/* return the index of the src widget from the connection list of the nid. + * return -1 if not found + */ +static int alc662_input_pin_idx(struct hda_codec *codec, hda_nid_t nid, + hda_nid_t src) +{ + hda_nid_t conn_list[HDA_MAX_CONNECTIONS]; + int i, conns; + + conns = snd_hda_get_connections(codec, nid, conn_list, + ARRAY_SIZE(conn_list)); + if (conns < 0) + return -1; + for (i = 0; i < conns; i++) + if (conn_list[i] == src) + return i; + return -1; +} + +static int alc662_is_input_pin(struct hda_codec *codec, hda_nid_t nid) +{ + unsigned int pincap = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); + return (pincap & AC_PINCAP_IN) != 0; +} + /* create playback/capture controls for input pins */ -static int alc662_auto_create_analog_input_ctls(struct alc_spec *spec, +static int alc662_auto_create_analog_input_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { + struct alc_spec *spec = codec->spec; struct hda_input_mux *imux = &spec->private_imux[0]; int i, err, idx; for (i = 0; i < AUTO_PIN_LAST; i++) { - if (alc880_is_input_pin(cfg->input_pins[i])) { - idx = alc880_input_pin_idx(cfg->input_pins[i]); - err = new_analog_input(spec, cfg->input_pins[i], - auto_pin_cfg_labels[i], - idx, 0x0b); - if (err < 0) - return err; - imux->items[imux->num_items].label = - auto_pin_cfg_labels[i]; - imux->items[imux->num_items].index = - alc880_input_pin_idx(cfg->input_pins[i]); - imux->num_items++; + if (alc662_is_input_pin(codec, cfg->input_pins[i])) { + idx = alc662_input_pin_idx(codec, 0x0b, + cfg->input_pins[i]); + if (idx >= 0) { + err = new_analog_input(spec, cfg->input_pins[i], + auto_pin_cfg_labels[i], + idx, 0x0b); + if (err < 0) + return err; + } + idx = alc662_input_pin_idx(codec, 0x22, + cfg->input_pins[i]); + if (idx >= 0) { + imux->items[imux->num_items].label = + auto_pin_cfg_labels[i]; + imux->items[imux->num_items].index = idx; + imux->num_items++; + } } } return 0; @@ -16794,7 +16826,6 @@ static void alc662_auto_init_hp_out(struct hda_codec *codec) alc662_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0); } -#define alc662_is_input_pin(nid) alc880_is_input_pin(nid) #define ALC662_PIN_CD_NID ALC880_PIN_CD_NID static void alc662_auto_init_analog_input(struct hda_codec *codec) @@ -16804,7 +16835,7 @@ static void alc662_auto_init_analog_input(struct hda_codec *codec) for (i = 0; i < AUTO_PIN_LAST; i++) { hda_nid_t nid = spec->autocfg.input_pins[i]; - if (alc662_is_input_pin(nid)) { + if (alc662_is_input_pin(codec, nid)) { alc_set_input_pin(codec, nid, i); if (nid != ALC662_PIN_CD_NID) snd_hda_codec_write(codec, nid, 0, @@ -16844,7 +16875,7 @@ static int alc662_parse_auto_config(struct hda_codec *codec) "Headphone"); if (err < 0) return err; - err = alc662_auto_create_analog_input_ctls(spec, &spec->autocfg); + err = alc662_auto_create_analog_input_ctls(codec, &spec->autocfg); if (err < 0) return err; -- cgit v1.2.3-70-g09d2 From 8b22d943c34b616eefbd6d2f8f197a53b1f29fd0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 20 Mar 2009 16:26:15 +0100 Subject: ALSA: pcm - Safer boundary checks Make the boundary checks a bit safer. These caese are rare or theoretically won't happen, but nothing bad to keep the checks safer... Signed-off-by: Takashi Iwai --- sound/core/pcm_lib.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 063c675177a..fbb2e391591 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -222,8 +222,9 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) hw_ptr_interrupt = runtime->hw_ptr_interrupt + runtime->period_size; delta = new_hw_ptr - hw_ptr_interrupt; if (hw_ptr_interrupt >= runtime->boundary) { - hw_ptr_interrupt %= runtime->boundary; - if (!hw_base) /* hw_base was already lapped; recalc delta */ + hw_ptr_interrupt -= runtime->boundary; + if (hw_base < runtime->boundary / 2) + /* hw_base was already lapped; recalc delta */ delta = new_hw_ptr - hw_ptr_interrupt; } if (delta < 0) { @@ -241,7 +242,7 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) delta = 0; } else { hw_base += runtime->buffer_size; - if (hw_base == runtime->boundary) + if (hw_base >= runtime->boundary) hw_base = 0; new_hw_ptr = hw_base + pos; } @@ -296,7 +297,7 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream) return 0; } hw_base += runtime->buffer_size; - if (hw_base == runtime->boundary) + if (hw_base >= runtime->boundary) hw_base = 0; new_hw_ptr = hw_base + pos; } -- cgit v1.2.3-70-g09d2 From 234b4346a064f8a2a488da10b3c1e640fb778a17 Mon Sep 17 00:00:00 2001 From: Pascal de Bruijn Date: Mon, 23 Mar 2009 11:15:59 +0100 Subject: ALSA: hda - Add function id to proc output This patch does two things: Output Intel HDA Function Id in /proc/asound/cardX/codec#X Align Vendor/Subsystem/Revision Ids to 8 characters, front-padded with zeros Before: Vendor Id: 0x11d41884 Subsystem Id: 0x103c281a Revision Id: 0x100100 After: Function Id: 0x1 Vendor Id: 0x11d41884 Subsystem Id: 0x103c281a Revision Id: 0x0100100 As report on the Kernel Bugzilla #12888 Signed-off-by: Pascal de Bruijn Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 6 +++--- sound/pci/hda/hda_codec.h | 1 + sound/pci/hda/hda_proc.c | 5 +++-- 3 files changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index b90a2400f53..1b5575ecb0a 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -647,9 +647,9 @@ static void /*__devinit*/ setup_fg_nodes(struct hda_codec *codec) total_nodes = snd_hda_get_sub_nodes(codec, AC_NODE_ROOT, &nid); for (i = 0; i < total_nodes; i++, nid++) { - unsigned int func; - func = snd_hda_param_read(codec, nid, AC_PAR_FUNCTION_TYPE); - switch (func & 0xff) { + codec->function_id = snd_hda_param_read(codec, nid, + AC_PAR_FUNCTION_TYPE) & 0xff; + switch (codec->function_id) { case AC_GRP_AUDIO_FUNCTION: codec->afg = nid; break; diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 079e1ab718d..2fdecf4b0eb 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -739,6 +739,7 @@ struct hda_codec { hda_nid_t mfg; /* MFG node id */ /* ids */ + u32 function_id; u32 vendor_id; u32 subsystem_id; u32 revision_id; diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 639cf0edaa9..93d7499350c 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -469,8 +469,9 @@ static void print_codec_info(struct snd_info_entry *entry, snd_iprintf(buffer, "Codec: %s\n", codec->name ? codec->name : "Not Set"); snd_iprintf(buffer, "Address: %d\n", codec->addr); - snd_iprintf(buffer, "Vendor Id: 0x%x\n", codec->vendor_id); - snd_iprintf(buffer, "Subsystem Id: 0x%x\n", codec->subsystem_id); + snd_iprintf(buffer, "Function Id: 0x%x\n", codec->function_id); + snd_iprintf(buffer, "Vendor Id: 0x%08x\n", codec->vendor_id); + snd_iprintf(buffer, "Subsystem Id: 0x%08x\n", codec->subsystem_id); snd_iprintf(buffer, "Revision Id: 0x%x\n", codec->revision_id); if (codec->mfg) -- cgit v1.2.3-70-g09d2 From 52ca15b7c0c711eb37f5e4b769e8488e5c516d43 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 23 Mar 2009 12:51:55 +0100 Subject: ALSA: hda - Avoid output amp manipulation to digital mic pins Don't set amp-out values to pins without PINCAP_OUT capability, which are usually assigned for digital mics on ALC663/ALC272. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b69d9864f6f..965a531d2fb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -16750,6 +16750,12 @@ static int alc662_is_input_pin(struct hda_codec *codec, hda_nid_t nid) return (pincap & AC_PINCAP_IN) != 0; } +static int alc662_is_output_pin(struct hda_codec *codec, hda_nid_t nid) +{ + unsigned int pincap = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); + return (pincap & AC_PINCAP_OUT) != 0; +} + /* create playback/capture controls for input pins */ static int alc662_auto_create_analog_input_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) @@ -16837,7 +16843,8 @@ static void alc662_auto_init_analog_input(struct hda_codec *codec) hda_nid_t nid = spec->autocfg.input_pins[i]; if (alc662_is_input_pin(codec, nid)) { alc_set_input_pin(codec, nid, i); - if (nid != ALC662_PIN_CD_NID) + if (nid != ALC662_PIN_CD_NID && + alc662_is_output_pin(codec, nid)) snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); -- cgit v1.2.3-70-g09d2 From 1327a32b878b5ed2113c63557b6f4f949f821857 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 23 Mar 2009 13:07:47 +0100 Subject: ALSA: hda - Cache pin-cap values Added snd_hda_query_pin_caps() to read and cache pin-cap values to avoid too frequently issuing the same verbs. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 16 ++++++++++++++++ sound/pci/hda/hda_generic.c | 2 +- sound/pci/hda/hda_local.h | 1 + sound/pci/hda/patch_realtek.c | 6 +++--- sound/pci/hda/patch_sigmatel.c | 7 +++---- 5 files changed, 24 insertions(+), 8 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 1b5575ecb0a..0f70d2d102e 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1052,6 +1052,7 @@ EXPORT_SYMBOL_HDA(snd_hda_codec_cleanup_stream); /* FIXME: more better hash key? */ #define HDA_HASH_KEY(nid,dir,idx) (u32)((nid) + ((idx) << 16) + ((dir) << 24)) +#define HDA_HASH_PINCAP_KEY(nid) (u32)((nid) + (0x02 << 24)) #define INFO_AMP_CAPS (1<<0) #define INFO_AMP_VOL(ch) (1 << (1 + (ch))) @@ -1142,6 +1143,21 @@ int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir, } EXPORT_SYMBOL_HDA(snd_hda_override_amp_caps); +u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid) +{ + struct hda_amp_info *info; + + info = get_alloc_amp_hash(codec, HDA_HASH_PINCAP_KEY(nid)); + if (!info) + return 0; + if (!info->head.val) { + info->amp_caps = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); + info->head.val |= INFO_AMP_CAPS; + } + return info->amp_caps; +} +EXPORT_SYMBOL_HDA(snd_hda_query_pin_caps); + /* * read the current volume to info * if the cache exists, read the cache value. diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 2c81a683e8f..1d5797a9668 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -144,7 +144,7 @@ static int add_new_node(struct hda_codec *codec, struct hda_gspec *spec, hda_nid node->type = (node->wid_caps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; if (node->type == AC_WID_PIN) { - node->pin_caps = snd_hda_param_read(codec, node->nid, AC_PAR_PIN_CAP); + node->pin_caps = snd_hda_query_pin_caps(codec, node->nid); node->pin_ctl = snd_hda_codec_read(codec, node->nid, 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); node->def_cfg = snd_hda_codec_get_pincfg(codec, node->nid); } diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 27428c718fd..83349013b4d 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -411,6 +411,7 @@ static inline u32 get_wcaps(struct hda_codec *codec, hda_nid_t nid) u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction); int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir, unsigned int caps); +u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid); int snd_hda_ctl_add(struct hda_codec *codec, struct snd_kcontrol *kctl); void snd_hda_ctls_clear(struct hda_codec *codec); diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 965a531d2fb..bf7e64e2c46 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -770,7 +770,7 @@ static void alc_set_input_pin(struct hda_codec *codec, hda_nid_t nid, if (auto_pin_type <= AUTO_PIN_FRONT_MIC) { unsigned int pincap; - pincap = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); + pincap = snd_hda_query_pin_caps(codec, nid); pincap = (pincap & AC_PINCAP_VREF) >> AC_PINCAP_VREF_SHIFT; if (pincap & AC_PINCAP_VREF_80) val = PIN_VREF80; @@ -16746,13 +16746,13 @@ static int alc662_input_pin_idx(struct hda_codec *codec, hda_nid_t nid, static int alc662_is_input_pin(struct hda_codec *codec, hda_nid_t nid) { - unsigned int pincap = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); + unsigned int pincap = snd_hda_query_pin_caps(codec, nid); return (pincap & AC_PINCAP_IN) != 0; } static int alc662_is_output_pin(struct hda_codec *codec, hda_nid_t nid) { - unsigned int pincap = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); + unsigned int pincap = snd_hda_query_pin_caps(codec, nid); return (pincap & AC_PINCAP_OUT) != 0; } diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 4da72403fc8..b1c180a9e9b 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2537,8 +2537,7 @@ static int stac92xx_build_pcms(struct hda_codec *codec) static unsigned int stac92xx_get_vref(struct hda_codec *codec, hda_nid_t nid) { - unsigned int pincap = snd_hda_param_read(codec, nid, - AC_PAR_PIN_CAP); + unsigned int pincap = snd_hda_query_pin_caps(codec, nid); pincap = (pincap & AC_PINCAP_VREF) >> AC_PINCAP_VREF_SHIFT; if (pincap & AC_PINCAP_VREF_100) return AC_PINCTL_VREF_100; @@ -2799,7 +2798,7 @@ static hda_nid_t check_line_out_switch(struct hda_codec *codec) if (cfg->line_out_type != AUTO_PIN_LINE_OUT) return 0; nid = cfg->input_pins[AUTO_PIN_LINE]; - pincap = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); + pincap = snd_hda_query_pin_caps(codec, nid); if (pincap & AC_PINCAP_OUT) return nid; return 0; @@ -2822,7 +2821,7 @@ static hda_nid_t check_mic_out_switch(struct hda_codec *codec) /* some laptops have an internal analog microphone * which can't be used as a output */ if (get_defcfg_connect(def_conf) != AC_JACK_PORT_FIXED) { - pincap = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); + pincap = snd_hda_query_pin_caps(codec, nid); if (pincap & AC_PINCAP_OUT) return nid; } -- cgit v1.2.3-70-g09d2 From e82c025b501a1ca62dec40989817dbb17c0b9167 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 23 Mar 2009 15:17:38 +0100 Subject: ALSA: hda - Fix the wrong pin-cap check in patch_realtek.c The check for the amp-output must be done for widget-caps rather than pin-caps as implemented in the recent change... Simply a thinko. Also, add the similar checks to all places that put output-amp mutes in the initialization. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 20 +++++++++----------- 1 file changed, 9 insertions(+), 11 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index bf7e64e2c46..8dcbb04e57b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4207,7 +4207,8 @@ static void alc880_auto_init_analog_input(struct hda_codec *codec) hda_nid_t nid = spec->autocfg.input_pins[i]; if (alc880_is_input_pin(nid)) { alc_set_input_pin(codec, nid, i); - if (nid != ALC880_PIN_CD_NID) + if (nid != ALC880_PIN_CD_NID && + (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP)) snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); @@ -5673,7 +5674,8 @@ static void alc260_auto_init_analog_input(struct hda_codec *codec) hda_nid_t nid = spec->autocfg.input_pins[i]; if (nid >= 0x12) { alc_set_input_pin(codec, nid, i); - if (nid != ALC260_PIN_CD_NID) + if (nid != ALC260_PIN_CD_NID && + (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP)) snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); @@ -9153,7 +9155,8 @@ static void alc883_auto_init_analog_input(struct hda_codec *codec) hda_nid_t nid = spec->autocfg.input_pins[i]; if (alc883_is_input_pin(nid)) { alc_set_input_pin(codec, nid, i); - if (nid != ALC883_PIN_CD_NID) + if (nid != ALC883_PIN_CD_NID && + (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP)) snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); @@ -14880,7 +14883,8 @@ static void alc861vd_auto_init_analog_input(struct hda_codec *codec) hda_nid_t nid = spec->autocfg.input_pins[i]; if (alc861vd_is_input_pin(nid)) { alc_set_input_pin(codec, nid, i); - if (nid != ALC861VD_PIN_CD_NID) + if (nid != ALC861VD_PIN_CD_NID && + (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP)) snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); @@ -16750,12 +16754,6 @@ static int alc662_is_input_pin(struct hda_codec *codec, hda_nid_t nid) return (pincap & AC_PINCAP_IN) != 0; } -static int alc662_is_output_pin(struct hda_codec *codec, hda_nid_t nid) -{ - unsigned int pincap = snd_hda_query_pin_caps(codec, nid); - return (pincap & AC_PINCAP_OUT) != 0; -} - /* create playback/capture controls for input pins */ static int alc662_auto_create_analog_input_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) @@ -16844,7 +16842,7 @@ static void alc662_auto_init_analog_input(struct hda_codec *codec) if (alc662_is_input_pin(codec, nid)) { alc_set_input_pin(codec, nid, i); if (nid != ALC662_PIN_CD_NID && - alc662_is_output_pin(codec, nid)) + (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP)) snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); -- cgit v1.2.3-70-g09d2 From a23b688f4d5c2490a50677b30011a677d8edf3d0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 23 Mar 2009 15:21:36 +0100 Subject: ALSA: hda - Don't create empty/single-item input source In patch_realtek.c, don't create empty or single-item "Input Source" control elements that are simply superfluous. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 47 ++++++++++++++++++++++++++++++++----------- 1 file changed, 35 insertions(+), 12 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8dcbb04e57b..7a3c6db6d5b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1595,8 +1595,7 @@ static int alc_cap_sw_put(struct snd_kcontrol *kcontrol, snd_hda_mixer_amp_switch_put); } -#define DEFINE_CAPMIX(num) \ -static struct snd_kcontrol_new alc_capture_mixer ## num[] = { \ +#define _DEFINE_CAPMIX(num) \ { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = "Capture Switch", \ @@ -1617,7 +1616,9 @@ static struct snd_kcontrol_new alc_capture_mixer ## num[] = { \ .get = alc_cap_vol_get, \ .put = alc_cap_vol_put, \ .tlv = { .c = alc_cap_vol_tlv }, \ - }, \ + } + +#define _DEFINE_CAPSRC(num) \ { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ /* .name = "Capture Source", */ \ @@ -1626,15 +1627,28 @@ static struct snd_kcontrol_new alc_capture_mixer ## num[] = { \ .info = alc_mux_enum_info, \ .get = alc_mux_enum_get, \ .put = alc_mux_enum_put, \ - }, \ - { } /* end */ \ + } + +#define DEFINE_CAPMIX(num) \ +static struct snd_kcontrol_new alc_capture_mixer ## num[] = { \ + _DEFINE_CAPMIX(num), \ + _DEFINE_CAPSRC(num), \ + { } /* end */ \ +} + +#define DEFINE_CAPMIX_NOSRC(num) \ +static struct snd_kcontrol_new alc_capture_mixer_nosrc ## num[] = { \ + _DEFINE_CAPMIX(num), \ + { } /* end */ \ } /* up to three ADCs */ DEFINE_CAPMIX(1); DEFINE_CAPMIX(2); DEFINE_CAPMIX(3); - +DEFINE_CAPMIX_NOSRC(1); +DEFINE_CAPMIX_NOSRC(2); +DEFINE_CAPMIX_NOSRC(3); /* * ALC880 5-stack model @@ -4298,13 +4312,22 @@ static void alc880_auto_init(struct hda_codec *codec) static void set_capture_mixer(struct alc_spec *spec) { - static struct snd_kcontrol_new *caps[3] = { - alc_capture_mixer1, - alc_capture_mixer2, - alc_capture_mixer3, + static struct snd_kcontrol_new *caps[2][3] = { + { alc_capture_mixer_nosrc1, + alc_capture_mixer_nosrc2, + alc_capture_mixer_nosrc3 }, + { alc_capture_mixer1, + alc_capture_mixer2, + alc_capture_mixer3 }, }; - if (spec->num_adc_nids > 0 && spec->num_adc_nids <= 3) - spec->cap_mixer = caps[spec->num_adc_nids - 1]; + if (spec->num_adc_nids > 0 && spec->num_adc_nids <= 3) { + int mux; + if (spec->input_mux && spec->input_mux->num_items > 1) + mux = 1; + else + mux = 0; + spec->cap_mixer = caps[mux][spec->num_adc_nids - 1]; + } } #define set_beep_amp(spec, nid, idx, dir) \ -- cgit v1.2.3-70-g09d2 From 14bafe3278e5da952a6586a5a9a9d286566049ed Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 23 Mar 2009 16:35:39 +0100 Subject: ALSA: hda - Use cached calls to get widget caps and pin caps Replace with the standard function calls to use caches for reading the widget caps and pin caps. hda_proc.c is still using the direct verbs to get raw values as much as possible. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 3 +-- sound/pci/hda/patch_sigmatel.c | 3 +-- 2 files changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 0f70d2d102e..a4e5e595211 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2321,8 +2321,7 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, * don't power down the widget if it controls * eapd and EAPD_BTLENABLE is set. */ - pincap = snd_hda_param_read(codec, nid, - AC_PAR_PIN_CAP); + pincap = snd_hda_query_pin_caps(codec, nid); if (pincap & AC_PINCAP_EAPD) { int eapd = snd_hda_codec_read(codec, nid, 0, diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index b1c180a9e9b..b5e108aa8f6 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2869,8 +2869,7 @@ static hda_nid_t get_unassigned_dac(struct hda_codec *codec, hda_nid_t nid) conn_len = snd_hda_get_connections(codec, nid, conn, HDA_MAX_CONNECTIONS); for (j = 0; j < conn_len; j++) { - wcaps = snd_hda_param_read(codec, conn[j], - AC_PAR_AUDIO_WIDGET_CAP); + wcaps = get_wcaps(codec, conn[j]); wtype = (wcaps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; /* we check only analog outputs */ if (wtype != AC_WID_AUD_OUT || (wcaps & AC_WCAP_DIGITAL)) -- cgit v1.2.3-70-g09d2 From 9b6682ff4c69484b6955f89f7902e3dde2481bed Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 23 Mar 2009 22:50:52 +0100 Subject: ALSA: hda - Add quirk for Acer Ferrari 5000 Add a quirk model=acer-aspire for Acer Ferrari 5000 with ALC883 codec. Note that model=auto doesn't work for this laptop because of broken BIOS (that doesn't set the subsystem id properly). Tested-by: Russ Dill Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7a3c6db6d5b..82097790f6f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8677,6 +8677,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1019, 0x6668, "ECS", ALC883_3ST_6ch_DIG), SND_PCI_QUIRK(0x1025, 0x006c, "Acer Aspire 9810", ALC883_ACER_ASPIRE), SND_PCI_QUIRK(0x1025, 0x0090, "Acer Aspire", ALC883_ACER_ASPIRE), + SND_PCI_QUIRK(0x1025, 0x010a, "Acer Ferrari 5000", ALC883_ACER_ASPIRE), SND_PCI_QUIRK(0x1025, 0x0110, "Acer Aspire", ALC883_ACER_ASPIRE), SND_PCI_QUIRK(0x1025, 0x0112, "Acer Aspire 9303", ALC883_ACER_ASPIRE), SND_PCI_QUIRK(0x1025, 0x0121, "Acer Aspire 5920G", ALC883_ACER_ASPIRE), -- cgit v1.2.3-70-g09d2