From dfb12eeb0f04b37e5eb3858864d074af4ecd2ac7 Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Sun, 27 Dec 2009 15:48:40 -0500 Subject: ALSA: atiixp: Specify codec for Foxconn RC4107MA-RS2 BugLink: https://bugs.launchpad.net/ubuntu/+bug/498863 This mainboard needs ac97_codec=0. Cc: stable@kernel.org Tested-by: Apoorv Parle Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/atiixp.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/pci') diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index d6752dff2a4..42b4fbbd8e2 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -297,6 +297,7 @@ static struct pci_device_id snd_atiixp_ids[] = { MODULE_DEVICE_TABLE(pci, snd_atiixp_ids); static struct snd_pci_quirk atiixp_quirks[] __devinitdata = { + SND_PCI_QUIRK(0x105b, 0x0c81, "Foxconn RC4107MA-RS2", 0), SND_PCI_QUIRK(0x15bd, 0x3100, "DFI RS482", 0), { } /* terminator */ }; -- cgit v1.2.3-70-g09d2 From 9980c6209ebc2a02eb3ca51a4fae1e17f645c868 Mon Sep 17 00:00:00 2001 From: Roel Kluin Date: Sun, 27 Dec 2009 22:26:47 +0100 Subject: ALSA: test off by one in setsamplerate() With `while (i++ < MAX_WRITE_RETRY)' i reaches MAX_WRITE_RETRY + 1 after the loop Signed-off-by: Roel Kluin Signed-off-by: Takashi Iwai --- sound/pci/riptide/riptide.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index b5ca02e2038..e66ef2b69b5 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -1058,7 +1058,7 @@ setsamplerate(struct cmdif *cif, unsigned char *intdec, unsigned int rate) rptr.retwords[2] != M && rptr.retwords[3] != N && i++ < MAX_WRITE_RETRY); - if (i == MAX_WRITE_RETRY) { + if (i > MAX_WRITE_RETRY) { snd_printdd("sent samplerate %d: %d failed\n", *intdec, rate); return -EIO; -- cgit v1.2.3-70-g09d2 From 9c0afc861a7228f718cb6a79fa7f9d46bf9ff300 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 12 Jan 2010 14:00:11 +0100 Subject: ALSA: hda - Fix ALC861-VD capture source mixer The capture source or input source mixer element wasn't created properly for ALC861-VD codec due to the wrong NID passed to alc_auto_create_input_ctls(). References: Novell bnc#568305 http://bugzilla.novell.com/show_bug.cgi?id=568305 Signed-off-by: Takashi Iwai Cc: --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c7465053d6b..e3caa78ccd5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -15493,7 +15493,7 @@ static struct alc_config_preset alc861vd_presets[] = { static int alc861vd_auto_create_input_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { - return alc_auto_create_input_ctls(codec, cfg, 0x15, 0x09, 0); + return alc_auto_create_input_ctls(codec, cfg, 0x15, 0x22, 0); } -- cgit v1.2.3-70-g09d2 From 4dee8baa18d611b6dc854e1cc193550ff6f687be Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 13 Jan 2010 17:20:08 +0100 Subject: ALSA: hda - Fix Toshiba NB20x quirk entry The alc664-mode4 model doesn't seem to fit with Toshiba NB205 correctly. NB205 uses the pin 0x17 connected with the mixer 0x0f for the speaker output, which isn't controlled by mode4 model at all. Rather model=auto works fine as is on the latest driver, so let it back again. Tested-by: Nickolas Lloyd Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e3caa78ccd5..bff60cea777 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -17251,7 +17251,7 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = { 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(0x1179, 0xff6e, "Toshiba NB200", ALC663_ASUS_MODE4), + SND_PCI_QUIRK(0x1179, 0xff6e, "Toshiba NB20x", ALC662_AUTO), SND_PCI_QUIRK(0x144d, 0xca00, "Samsung NC10", ALC272_SAMSUNG_NC10), SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L", ALC662_3ST_6ch_DIG), -- cgit v1.2.3-70-g09d2 From a76221d47ef2b73ff16c0fef00a784026308ea02 Mon Sep 17 00:00:00 2001 From: Alex Murray Date: Wed, 13 Jan 2010 23:15:03 +1030 Subject: ALSA: hda - Improved MacBook (Pro) 5,1 / 5,2 support This patch adds support for automatically muting the speakers when headphones are inserted, as well as relabelling the headphone widgets from the non-standard "HP" to the standard "Headphone" for the mb5 model. Signed-off-by: Alex Murray Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 28 ++++++++++++++++++++++++++-- 1 file changed, 26 insertions(+), 2 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index bff60cea777..11b989bacd3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7094,8 +7094,8 @@ static struct snd_kcontrol_new alc885_mb5_mixer[] = { HDA_BIND_MUTE ("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("LFE Playback Volume", 0x0e, 0x00, HDA_OUTPUT), HDA_BIND_MUTE ("LFE Playback Switch", 0x0e, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("HP Playback Volume", 0x0f, 0x00, HDA_OUTPUT), - HDA_BIND_MUTE ("HP Playback Switch", 0x0f, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0f, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("Headphone Playback Switch", 0x0f, 0x02, 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, 0x01, HDA_INPUT), @@ -7496,6 +7496,7 @@ static struct hda_verb alc885_mb5_init_verbs[] = { {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x14, AC_VERB_SET_CONNECT_SEL, 0x03}, + {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, /* Front Mic pin: input vref at 80% */ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, @@ -7680,6 +7681,27 @@ static void alc885_mbp3_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[0] = 0x14; } +static void alc885_mb5_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x14, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + snd_hda_codec_amp_stereo(codec, 0x18, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); + snd_hda_codec_amp_stereo(codec, 0x1a, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); + +} + +static void alc885_mb5_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + /* Headphone insertion or removal. */ + if ((res >> 26) == ALC880_HP_EVENT) + alc885_mb5_automute(codec); +} + static void alc885_imac91_automute(struct hda_codec *codec) { unsigned int present; @@ -9126,6 +9148,8 @@ static struct alc_config_preset alc882_presets[] = { .input_mux = &mb5_capture_source, .dig_out_nid = ALC882_DIGOUT_NID, .dig_in_nid = ALC882_DIGIN_NID, + .unsol_event = alc885_mb5_unsol_event, + .init_hook = alc885_mb5_automute, }, [ALC885_MACPRO] = { .mixers = { alc882_macpro_mixer }, -- cgit v1.2.3-70-g09d2 From c7a8eb103248a110cdbe0530d8c5ce987f099eee Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 14 Jan 2010 12:39:02 +0100 Subject: ALSA: hda - Fix missing capture mixer for ALC861/660 codecs The capture-related mixer elements are missing with ALC861/ALC660 codecs when quirks are present, due to missing call of set_capture_mixer(). Reference: Novell bnc#567340 http://bugzilla.novell.com/show_bug.cgi?id=567340 Signed-off-by: Takashi Iwai Cc: --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 11b989bacd3..abae1007cea 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -14879,6 +14879,8 @@ static int patch_alc861(struct hda_codec *codec) spec->stream_digital_playback = &alc861_pcm_digital_playback; spec->stream_digital_capture = &alc861_pcm_digital_capture; + if (!spec->cap_mixer) + set_capture_mixer(codec); set_beep_amp(spec, 0x23, 0, HDA_OUTPUT); spec->vmaster_nid = 0x03; -- cgit v1.2.3-70-g09d2 From d38cce7046cfd0011f69d5dcf6a22525438154f6 Mon Sep 17 00:00:00 2001 From: Kunal Gangakhedkar Date: Fri, 15 Jan 2010 21:01:47 +0530 Subject: ALSA: hda - Fix mute led GPIO on HP dv-series notebooks On my laptop (HP dv6-1110ax), there are no OEM strings in SMBIOS of type "HP_Mute_LED*". Hence, the GPIO for the mute button LED doesn't get set properly. I didn't find the strings in my cousin's laptop (HP dv9500t CTO) either. As per the documentation of find_mute_led_gpio(), these strings occur in HP B-series systems - so, before scanning the SMBIOS strings, we need to check if we're dealing with a B-series system. Need to get confirmation from HP if this logic takes care of all the systems. I'm trying to poke a friend there. Signed-off-by: Kunal Gangakhedkar Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 61 +++++++++++++++++++++++++++++++----------- 1 file changed, 45 insertions(+), 16 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 2291a839681..799ba257090 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4730,6 +4730,26 @@ static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res) } } +static int hp_blike_system(u32 subsystem_id); + +static void set_hp_led_gpio(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + switch (codec->vendor_id) { + case 0x111d7608: + /* GPIO 0 */ + spec->gpio_led = 0x01; + break; + case 0x111d7600: + case 0x111d7601: + case 0x111d7602: + case 0x111d7603: + /* GPIO 3 */ + spec->gpio_led = 0x08; + break; + } +} + /* * This method searches for the mute LED GPIO configuration * provided as OEM string in SMBIOS. The format of that string @@ -4741,6 +4761,14 @@ static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res) * * So, HP B-series like systems may have HP_Mute_LED_0 (current models) * or HP_Mute_LED_0_3 (future models) OEM SMBIOS strings + * + * + * The dv-series laptops don't seem to have the HP_Mute_LED* strings in + * SMBIOS - at least the ones I have seen do not have them - which include + * my own system (HP Pavilion dv6-1110ax) and my cousin's + * HP Pavilion dv9500t CTO. + * Need more information on whether it is true across the entire series. + * -- kunal */ static int find_mute_led_gpio(struct hda_codec *codec) { @@ -4751,28 +4779,27 @@ static int find_mute_led_gpio(struct hda_codec *codec) while ((dev = dmi_find_device(DMI_DEV_TYPE_OEM_STRING, NULL, dev))) { if (sscanf(dev->name, "HP_Mute_LED_%d_%d", - &spec->gpio_led_polarity, - &spec->gpio_led) == 2) { + &spec->gpio_led_polarity, + &spec->gpio_led) == 2) { spec->gpio_led = 1 << spec->gpio_led; return 1; } if (sscanf(dev->name, "HP_Mute_LED_%d", - &spec->gpio_led_polarity) == 1) { - switch (codec->vendor_id) { - case 0x111d7608: - /* GPIO 0 */ - spec->gpio_led = 0x01; - return 1; - case 0x111d7600: - case 0x111d7601: - case 0x111d7602: - case 0x111d7603: - /* GPIO 3 */ - spec->gpio_led = 0x08; - return 1; - } + &spec->gpio_led_polarity) == 1) { + set_hp_led_gpio(codec); + return 1; } } + + /* + * Fallback case - if we don't find the DMI strings, + * we statically set the GPIO - if not a B-series system. + */ + if (!hp_blike_system(codec->subsystem_id)) { + set_hp_led_gpio(codec); + spec->gpio_led_polarity = 1; + return 1; + } } return 0; } @@ -5548,6 +5575,8 @@ again: spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids); spec->num_smuxes = stac92hd71bxx_connected_smuxes(codec, 0x1e); + snd_printdd("Found board config: %d\n", spec->board_config); + switch (spec->board_config) { case STAC_HP_M4: /* enable internal microphone */ -- cgit v1.2.3-70-g09d2 From eaa9b3a748539651f50e3a234c8854e1b42a839a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 17 Jan 2010 13:09:33 +0100 Subject: ALSA: hda - Fix capture on Sony VAIO with single input Sony VAIO VGN-P11G with ALC262 codec has only one input pin, and the recording doesn't work with model=auto because ALC262 parser sets the wrong cap NIDs to choose the route and the default route for the sole input pin wasn't initialized properly. This patch solves these issues. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 62 +++++++++++++++++++++++++++++++++++++------ 1 file changed, 54 insertions(+), 8 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index abae1007cea..3f92def752f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1230,6 +1230,8 @@ static void alc_init_auto_mic(struct hda_codec *codec) return; /* invalid entry */ } } + if (!ext || !fixed) + return; if (!(get_wcaps(codec, ext) & AC_WCAP_UNSOL_CAP)) return; /* no unsol support */ snd_printdd("realtek: Enable auto-mic switch on NID 0x%x/0x%x\n", @@ -4812,6 +4814,49 @@ static void fixup_automic_adc(struct hda_codec *codec) spec->auto_mic = 0; /* disable auto-mic to be sure */ } +/* choose the ADC/MUX containing the input pin and initialize the setup */ +static void fixup_single_adc(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + hda_nid_t pin; + int i; + + /* search for the input pin; there must be only one */ + for (i = 0; i < AUTO_PIN_LAST; i++) { + if (spec->autocfg.input_pins[i]) { + pin = spec->autocfg.input_pins[i]; + break; + } + } + if (!pin) + return; + + /* set the default connection to that pin */ + for (i = 0; i < spec->num_adc_nids; i++) { + hda_nid_t cap = spec->capsrc_nids ? + spec->capsrc_nids[i] : spec->adc_nids[i]; + int idx; + + idx = get_connection_index(codec, cap, pin); + if (idx < 0) + continue; + /* use only this ADC */ + if (spec->capsrc_nids) + spec->capsrc_nids += i; + spec->adc_nids += i; + spec->num_adc_nids = 1; + /* select or unmute this route */ + if (get_wcaps_type(get_wcaps(codec, cap)) == AC_WID_AUD_MIX) { + snd_hda_codec_amp_stereo(codec, cap, HDA_INPUT, idx, + HDA_AMP_MUTE, 0); + } else { + snd_hda_codec_write_cache(codec, cap, 0, + AC_VERB_SET_CONNECT_SEL, idx); + } + return; + } +} + static void set_capture_mixer(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -4824,14 +4869,15 @@ static void set_capture_mixer(struct hda_codec *codec) alc_capture_mixer3 }, }; if (spec->num_adc_nids > 0 && spec->num_adc_nids <= 3) { - int mux; - if (spec->auto_mic) { - mux = 0; + int mux = 0; + if (spec->auto_mic) fixup_automic_adc(codec); - } else if (spec->input_mux && spec->input_mux->num_items > 1) - mux = 1; - else - mux = 0; + else if (spec->input_mux) { + if (spec->input_mux->num_items > 1) + mux = 1; + else if (spec->input_mux->num_items == 1) + fixup_single_adc(codec); + } spec->cap_mixer = caps[mux][spec->num_adc_nids - 1]; } } @@ -11203,7 +11249,7 @@ static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec, } #define alc262_auto_create_input_ctls \ - alc880_auto_create_input_ctls + alc882_auto_create_input_ctls /* * generic initialization of ADC, input mixers and output mixers -- cgit v1.2.3-70-g09d2 From 4feabefe53eb3742f0b2773a43200d1686f3a288 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 19 Jan 2010 15:38:44 +0100 Subject: ALSA: hda - Fix parsing pin node 0x21 on ALC259 ALC259 has a widget NID 0x21 for the output pin, but it wasn't handled properly in alc268_new_analog_output(). Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3f92def752f..79cdae324c5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12541,6 +12541,7 @@ static int alc268_new_analog_output(struct alc_spec *spec, hda_nid_t nid, dac = 0x02; break; case 0x15: + case 0x21: dac = 0x03; break; default: -- cgit v1.2.3-70-g09d2 From 3fb4a508b8e7957aa899f32cd6d9d462e102c7ca Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 19 Jan 2010 15:46:37 +0100 Subject: ALSA: hda - Turn on EAPD only if available for Realtek codecs Some codecs disable widgets used for output pins and reserve as vendor- spec widgets. Thus we need to check the widget type and pin cap before actually sending SET_EAPD verbs in the auto-configuration mode. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 29 +++++++++++++++++------------ 1 file changed, 17 insertions(+), 12 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 79cdae324c5..6ae610c0111 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1093,6 +1093,16 @@ static void alc889_coef_init(struct hda_codec *codec) snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_PROC_COEF, tmp|0x2010); } +/* turn on/off EAPD control (only if available) */ +static void set_eapd(struct hda_codec *codec, hda_nid_t nid, int on) +{ + if (get_wcaps_type(get_wcaps(codec, nid)) != AC_WID_PIN) + return; + if (snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_EAPD) + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_EAPD_BTLENABLE, + on ? 2 : 0); +} + static void alc_auto_init_amp(struct hda_codec *codec, int type) { unsigned int tmp; @@ -1110,25 +1120,22 @@ static void alc_auto_init_amp(struct hda_codec *codec, int type) case ALC_INIT_DEFAULT: switch (codec->vendor_id) { case 0x10ec0260: - snd_hda_codec_write(codec, 0x0f, 0, - AC_VERB_SET_EAPD_BTLENABLE, 2); - snd_hda_codec_write(codec, 0x10, 0, - AC_VERB_SET_EAPD_BTLENABLE, 2); + set_eapd(codec, 0x0f, 1); + set_eapd(codec, 0x10, 1); break; case 0x10ec0262: case 0x10ec0267: case 0x10ec0268: case 0x10ec0269: + case 0x10ec0270: case 0x10ec0272: case 0x10ec0660: case 0x10ec0662: case 0x10ec0663: case 0x10ec0862: case 0x10ec0889: - snd_hda_codec_write(codec, 0x14, 0, - AC_VERB_SET_EAPD_BTLENABLE, 2); - snd_hda_codec_write(codec, 0x15, 0, - AC_VERB_SET_EAPD_BTLENABLE, 2); + set_eapd(codec, 0x14, 1); + set_eapd(codec, 0x15, 1); break; } switch (codec->vendor_id) { @@ -1836,10 +1843,8 @@ static void alc889_acer_aspire_8930g_setup(struct hda_codec *codec) #ifdef CONFIG_SND_HDA_POWER_SAVE static void alc889_power_eapd(struct hda_codec *codec, int power) { - snd_hda_codec_write(codec, 0x14, 0, - AC_VERB_SET_EAPD_BTLENABLE, power ? 2 : 0); - snd_hda_codec_write(codec, 0x15, 0, - AC_VERB_SET_EAPD_BTLENABLE, power ? 2 : 0); + set_eapd(codec, 0x14, power); + set_eapd(codec, 0x15, power); } #endif -- cgit v1.2.3-70-g09d2 From dc99be47667c56046555e89e62f1ac17fa06329a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 20 Jan 2010 08:35:06 +0100 Subject: ALSA: hda - Fix HP T5735 automute This patch fixes the aut-mute setup on HP T5735 with ALC262 codec. Instead of wrong amp, use pin control toggling for muting the speaker now. Tested-by: Lee Trager Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6ae610c0111..d00e6d1da08 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10382,7 +10382,7 @@ static void alc262_hp_t5735_setup(struct hda_codec *codec) struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x15; - spec->autocfg.speaker_pins[0] = 0x0c; /* HACK: not actually a pin */ + spec->autocfg.speaker_pins[0] = 0x14; } static struct snd_kcontrol_new alc262_hp_t5735_mixer[] = { @@ -11793,9 +11793,9 @@ static struct alc_config_preset alc262_presets[] = { .num_channel_mode = ARRAY_SIZE(alc262_modes), .channel_mode = alc262_modes, .input_mux = &alc262_capture_source, - .unsol_event = alc_automute_amp_unsol_event, + .unsol_event = alc_sku_unsol_event, .setup = alc262_hp_t5735_setup, - .init_hook = alc_automute_amp, + .init_hook = alc_inithook, }, [ALC262_HP_RP5700] = { .mixers = { alc262_hp_rp5700_mixer }, -- cgit v1.2.3-70-g09d2 From 973b8cb0ead3e0b1dd3ee7b2df52e4dff1ffc707 Mon Sep 17 00:00:00 2001 From: Łukasz Wojniłowicz Date: Sun, 24 Jan 2010 14:12:37 +0100 Subject: ALSA: hda - add possibility to choose speakers configuration for 4930g MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Now one can choose speaker configuration in e.g. PulseAudio mixer Signed-off-by: Łukasz Wojniłowicz Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d00e6d1da08..da34095c707 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9478,6 +9478,7 @@ static struct alc_config_preset alc882_presets[] = { .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes), .channel_mode = alc883_3ST_6ch_modes, .need_dac_fix = 1, + .const_channel_count = 6, .num_mux_defs = ARRAY_SIZE(alc888_2_capture_sources), .input_mux = alc888_2_capture_sources, -- cgit v1.2.3-70-g09d2 From 8ce28d6abff34886d3797b25324c940471b99164 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 27 Jan 2010 20:26:08 +0100 Subject: ALSA: hda - Add an ASUS mobo to MSI blacklist Sid Boyce reported that his machine locks up without enable_msi=0 option. This looks like another ASUS mobo with Nvidia combo. Reported-by: Sid Boyce Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index ec9c348336c..565de38a3fc 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2332,6 +2332,7 @@ static void __devinit check_probe_mask(struct azx *chip, int dev) */ static struct snd_pci_quirk msi_black_list[] __devinitdata = { SND_PCI_QUIRK(0x1043, 0x81f2, "ASUS", 0), /* Athlon64 X2 + nvidia */ + SND_PCI_QUIRK(0x1043, 0x829c, "ASUS", 0), /* nvidia */ {} }; -- cgit v1.2.3-70-g09d2 From 1eb6dc7dabcb4aa762d96f4f6978f3ef86321d68 Mon Sep 17 00:00:00 2001 From: Maxim Levitsky Date: Thu, 4 Feb 2010 22:21:47 +0200 Subject: ALSA: hda - Delay switching to polling mode if an interrupt was missing My sound codec seems sometimes (very rarely) to omit interrupts (ALC268) However, interrupt mode still works. Thus if we get timeout, poll the codec once. If we get 3 such polls in a row, then switch to polling mode. This patch is maybe an bandaid, but this might be a workaround for hardware bug. Signed-off-by: Maxim Levitsky Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 19 +++++++++++++++++-- 1 file changed, 17 insertions(+), 2 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 565de38a3fc..d853e2c33bb 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -426,6 +426,7 @@ struct azx { /* flags */ int position_fix; + int poll_count; unsigned int running :1; unsigned int initialized :1; unsigned int single_cmd :1; @@ -506,7 +507,7 @@ static char *driver_short_names[] __devinitdata = { #define get_azx_dev(substream) (substream->runtime->private_data) static int azx_acquire_irq(struct azx *chip, int do_disconnect); - +static int azx_send_cmd(struct hda_bus *bus, unsigned int val); /* * Interface for HD codec */ @@ -664,11 +665,12 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus, { struct azx *chip = bus->private_data; unsigned long timeout; + int do_poll = 0; again: timeout = jiffies + msecs_to_jiffies(1000); for (;;) { - if (chip->polling_mode) { + if (chip->polling_mode || do_poll) { spin_lock_irq(&chip->reg_lock); azx_update_rirb(chip); spin_unlock_irq(&chip->reg_lock); @@ -676,6 +678,9 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus, if (!chip->rirb.cmds[addr]) { smp_rmb(); bus->rirb_error = 0; + + if (!do_poll) + chip->poll_count = 0; return chip->rirb.res[addr]; /* the last value */ } if (time_after(jiffies, timeout)) @@ -688,6 +693,16 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus, } } + if (!chip->polling_mode && chip->poll_count < 2) { + snd_printdd(SFX "azx_get_response timeout, " + "polling the codec once: last cmd=0x%08x\n", + chip->last_cmd[addr]); + do_poll = 1; + chip->poll_count++; + goto again; + } + + if (!chip->polling_mode) { snd_printk(KERN_WARNING SFX "azx_get_response timeout, " "switching to polling mode: last cmd=0x%08x\n", -- cgit v1.2.3-70-g09d2 From 9492837a6f54b069e13e40e3c89898bb8837a386 Mon Sep 17 00:00:00 2001 From: Maxim Levitsky Date: Thu, 4 Feb 2010 22:26:37 +0200 Subject: ALSA: cosmetic: make hda intel interrupt name consistent with others This renames the interrupt name in /proc/interrupt. HDA Intel -> hda_intel This also eliminates space from the name, probably helping some parsers. Don't think anybody depends on this name in userspace Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index d853e2c33bb..b8faa6dc5ab 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2058,7 +2058,7 @@ static int azx_acquire_irq(struct azx *chip, int do_disconnect) { if (request_irq(chip->pci->irq, azx_interrupt, chip->msi ? 0 : IRQF_SHARED, - "HDA Intel", chip)) { + "hda_intel", chip)) { printk(KERN_ERR "hda-intel: unable to grab IRQ %d, " "disabling device\n", chip->pci->irq); if (do_disconnect) -- cgit v1.2.3-70-g09d2 From fed08d036f2aabd8d0c684439de37f8ebec2bbc2 Mon Sep 17 00:00:00 2001 From: Jody Bruchon Date: Sat, 6 Feb 2010 10:46:26 -0500 Subject: ALSA: hda-intel: Avoid divide by zero crash On my AMD780V chipset, hda_intel.c can crash the kernel with a divide by zero for as-yet unknown reasons. A simple check for zero prevents it, though the problem that causes it remains. Since the workaround is harmless and won't affect anyone except victims of this bug, it should be safe; moreover, because this crash can be triggered by a user-mode application, there are denial of service implications on the systems affected by the bug without the patch. Signed-off-by: Jody Bruchon Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index b8faa6dc5ab..e767c3f395a 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1893,6 +1893,12 @@ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev) if (!bdl_pos_adj[chip->dev_index]) return 1; /* no delayed ack */ + if (azx_dev->period_bytes == 0) { + printk(KERN_WARNING + "hda-intel: Divide by zero was avoided " + "in azx_dev->period_bytes.\n"); + return 0; + } if (pos % azx_dev->period_bytes > azx_dev->period_bytes / 2) return 0; /* NG - it's below the period boundary */ return 1; /* OK, it's fine */ -- cgit v1.2.3-70-g09d2 From d6d8bf549393484e906913f02fa3c9518a2819b6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 12 Feb 2010 18:17:06 +0100 Subject: ALSA: hda - use WARN_ON_ONCE() for zero-division detection Replace the zero-division warning message with WARN_ON_ONCE() per the advice by Linus. This shouldn't happen, but if it happens, it's possible that the bug happens often due to buggy IRQs. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index e767c3f395a..3600e9cc9bc 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1893,12 +1893,9 @@ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev) if (!bdl_pos_adj[chip->dev_index]) return 1; /* no delayed ack */ - if (azx_dev->period_bytes == 0) { - printk(KERN_WARNING - "hda-intel: Divide by zero was avoided " - "in azx_dev->period_bytes.\n"); - return 0; - } + if (WARN_ONCE(!azx_dev->period_bytes, + "hda-intel: zero azx_dev->period_bytes")) + return 0; /* this shouldn't happen! */ if (pos % azx_dev->period_bytes > azx_dev->period_bytes / 2) return 0; /* NG - it's below the period boundary */ return 1; /* OK, it's fine */ -- cgit v1.2.3-70-g09d2 From b721e68bdc5b39c51bf6a1469f8d3663fbe03243 Mon Sep 17 00:00:00 2001 From: Giuliano Pochini Date: Wed, 17 Feb 2010 00:57:44 +0100 Subject: ALSA: Echoaudio, fix Guru Meditation #00000005.48454C50 This patch fixes a division by zero error in the irq handler. There is a small window between the hw_params() callback and when runtime->frame_bits is set by ALSA middle layer. When another substream is already running, if an interrupt is delivered during that window the irq handler calls pcm_pointer() which does a division by zero. The patch below makes the irq handler skip substreams that are initialized but not started yet. Cc to Clemens Ladisch because he proposed an alternate fix. For more information, please read the original thread in the linux-kernel mailing list: http://lkml.org/lkml/2010/2/2/187 Signed-off-by: Giuliano Pochini Signed-off-by: Takashi Iwai --- sound/pci/echoaudio/echoaudio.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 1305f7ca02c..641d7f07392 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -1821,7 +1821,9 @@ static irqreturn_t snd_echo_interrupt(int irq, void *dev_id) /* The hardware doesn't tell us which substream caused the irq, thus we have to check all running substreams. */ for (ss = 0; ss < DSP_MAXPIPES; ss++) { - if ((substream = chip->substream[ss])) { + substream = chip->substream[ss]; + if (substream && ((struct audiopipe *)substream->runtime-> + private_data)->state == PIPE_STATE_STARTED) { period = pcm_pointer(substream) / substream->runtime->period_size; if (period != chip->last_period[ss]) { -- cgit v1.2.3-70-g09d2