From 2b203dbbcbac731b07bd0e27c3eda26a26aecb72 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 11 Feb 2011 12:17:30 +0100 Subject: ALSA: hda - Avoid cast with union data for HDMI audio infoframe Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 22 +++++++++++++--------- 1 file changed, 13 insertions(+), 9 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index a5876773672..bb4930484ec 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -110,6 +110,12 @@ struct dp_audio_infoframe { u8 LFEPBL01_LSV36_DM_INH7; }; +union audio_infoframe { + struct hdmi_audio_infoframe hdmi; + struct dp_audio_infoframe dp; + u8 bytes[0]; +}; + /* * CEA speaker placement: * @@ -620,8 +626,7 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, hda_nid_t nid, int channels = substream->runtime->channels; int ca; int i; - u8 ai[max(sizeof(struct hdmi_audio_infoframe), - sizeof(struct dp_audio_infoframe))]; + union audio_infoframe ai; ca = hdmi_channel_allocation(codec, nid, channels); @@ -633,11 +638,10 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, hda_nid_t nid, pin_nid = spec->pin[i]; - memset(ai, 0, sizeof(ai)); + memset(&ai, 0, sizeof(ai)); if (spec->sink_eld[i].conn_type == 0) { /* HDMI */ - struct hdmi_audio_infoframe *hdmi_ai; + struct hdmi_audio_infoframe *hdmi_ai = &ai.hdmi; - hdmi_ai = (struct hdmi_audio_infoframe *)ai; hdmi_ai->type = 0x84; hdmi_ai->ver = 0x01; hdmi_ai->len = 0x0a; @@ -645,9 +649,8 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, hda_nid_t nid, hdmi_ai->CA = ca; hdmi_checksum_audio_infoframe(hdmi_ai); } else if (spec->sink_eld[i].conn_type == 1) { /* DisplayPort */ - struct dp_audio_infoframe *dp_ai; + struct dp_audio_infoframe *dp_ai = &ai.dp; - dp_ai = (struct dp_audio_infoframe *)ai; dp_ai->type = 0x84; dp_ai->len = 0x1b; dp_ai->ver = 0x11 << 2; @@ -664,7 +667,8 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, hda_nid_t nid, * sizeof(*dp_ai) to avoid partial match/update problems when * the user switches between HDMI/DP monitors. */ - if (!hdmi_infoframe_uptodate(codec, pin_nid, ai, sizeof(ai))) { + if (!hdmi_infoframe_uptodate(codec, pin_nid, ai.bytes, + sizeof(ai))) { snd_printdd("hdmi_setup_audio_infoframe: " "cvt=%d pin=%d channels=%d\n", nid, pin_nid, @@ -672,7 +676,7 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, hda_nid_t nid, hdmi_setup_channel_mapping(codec, pin_nid, ca); hdmi_stop_infoframe_trans(codec, pin_nid); hdmi_fill_audio_infoframe(codec, pin_nid, - ai, sizeof(ai)); + ai.bytes, sizeof(ai)); hdmi_start_infoframe_trans(codec, pin_nid); } } -- cgit v1.2.3-70-g09d2 From 2822084607c41ca3a2eb70e804aebaddcfdbe5a6 Mon Sep 17 00:00:00 2001 From: Raymond Yau Date: Tue, 8 Feb 2011 19:58:25 +0800 Subject: ALSA: hda - simplify multistreaming playback model of ad1988 Signed-off-by: Raymond Yau Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 117 ++++--------------------------------------- 1 file changed, 10 insertions(+), 107 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 8dabab79868..734c6ee55d8 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -30,10 +30,10 @@ #include "hda_beep.h" struct ad198x_spec { - struct snd_kcontrol_new *mixers[5]; + struct snd_kcontrol_new *mixers[6]; int num_mixers; unsigned int beep_amp; /* beep amp value, set via set_beep_amp() */ - const struct hda_verb *init_verbs[5]; /* initialization verbs + const struct hda_verb *init_verbs[6]; /* initialization verbs * don't forget NULL termination! */ unsigned int num_init_verbs; @@ -331,36 +331,11 @@ static int ad198x_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout); } -static int ad198x_alt_playback_pcm_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) -{ - struct ad198x_spec *spec = codec->spec; - snd_hda_codec_setup_stream(codec, spec->alt_dac_nid[0], stream_tag, - 0, format); - return 0; -} - -static int ad198x_alt_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct ad198x_spec *spec = codec->spec; - snd_hda_codec_cleanup_stream(codec, spec->alt_dac_nid[0]); - return 0; -} - static struct hda_pcm_stream ad198x_pcm_analog_alt_playback = { .substreams = 1, .channels_min = 2, .channels_max = 2, /* NID is set in ad198x_build_pcms */ - .ops = { - .prepare = ad198x_alt_playback_pcm_prepare, - .cleanup = ad198x_alt_playback_pcm_cleanup - }, }; /* @@ -2239,29 +2214,6 @@ static struct snd_kcontrol_new ad1988_6stack_mixers2[] = { static struct snd_kcontrol_new ad1988_6stack_fp_mixers[] = { HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x29, 2, HDA_INPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x2a, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x27, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x27, 2, 2, HDA_INPUT), - HDA_BIND_MUTE("Side Playback Switch", 0x28, 2, HDA_INPUT), - HDA_BIND_MUTE("Headphone Playback Switch", 0x22, 2, HDA_INPUT), - HDA_BIND_MUTE("Mono Playback Switch", 0x1e, 2, HDA_INPUT), - - HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x6, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x6, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x4, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x4, HDA_INPUT), - - HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT), - - HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x39, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x3c, 0x0, HDA_OUTPUT), - { } /* end */ }; @@ -2545,11 +2497,6 @@ static struct hda_verb ad1988_6stack_init_verbs[] = { }; static struct hda_verb ad1988_6stack_fp_init_verbs[] = { - /* Front, Surround, CLFE, side DAC; unmute as default */ - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Headphone; unmute as default */ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Port-A front headphon path */ @@ -2558,50 +2505,6 @@ static struct hda_verb ad1988_6stack_fp_init_verbs[] = { {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - /* Port-D line-out path */ - {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - /* Port-F surround path */ - {0x2a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x2a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - /* Port-G CLFE path */ - {0x27, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x27, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x24, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - /* Port-H side path */ - {0x28, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x28, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x25, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x25, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - /* Mono out path */ - {0x36, AC_VERB_SET_CONNECT_SEL, 0x1}, /* DAC1:04h */ - {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb01f}, /* unmute, 0dB */ - /* Port-B front mic-in path */ - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x39, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - /* Port-C line-in path */ - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x3a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x33, AC_VERB_SET_CONNECT_SEL, 0x0}, - /* Port-E mic-in path */ - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x3c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x34, AC_VERB_SET_CONNECT_SEL, 0x0}, - /* Analog CD Input */ - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - /* Analog Mix output amp */ - {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */ { } }; @@ -3316,20 +3219,20 @@ static int patch_ad1988(struct hda_codec *codec) spec->mixers[0] = ad1988_6stack_mixers1_rev2; else spec->mixers[0] = ad1988_6stack_mixers1; + spec->mixers[1] = ad1988_6stack_mixers2; + spec->num_init_verbs = 1; + spec->init_verbs[0] = ad1988_6stack_init_verbs; if (board_config == AD1988_6STACK_DIG_FP) { - spec->mixers[1] = ad1988_6stack_fp_mixers; + spec->num_mixers++; + spec->mixers[2] = ad1988_6stack_fp_mixers; + spec->num_init_verbs++; + spec->init_verbs[1] = ad1988_6stack_fp_init_verbs; spec->slave_vols = ad1988_6stack_fp_slave_vols; spec->slave_sws = ad1988_6stack_fp_slave_sws; spec->alt_dac_nid = ad1988_alt_dac_nid; spec->stream_analog_alt_playback = &ad198x_pcm_analog_alt_playback; - } else - spec->mixers[1] = ad1988_6stack_mixers2; - spec->num_init_verbs = 1; - if (board_config == AD1988_6STACK_DIG_FP) - spec->init_verbs[0] = ad1988_6stack_fp_init_verbs; - else - spec->init_verbs[0] = ad1988_6stack_init_verbs; + } if ((board_config == AD1988_6STACK_DIG) || (board_config == AD1988_6STACK_DIG_FP)) { spec->multiout.dig_out_nid = AD1988_SPDIF_OUT; -- cgit v1.2.3-70-g09d2 From 786c51f9168cfd2d49250c6e5e60035cbb2fd5a1 Mon Sep 17 00:00:00 2001 From: Łukasz Wojniłowicz Date: Thu, 24 Feb 2011 10:03:31 +0100 Subject: ALSA: hda - 4930g add internal lfe slider MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Lately I sent patch that switched lfe with side in mixer for acer-aspire-4930g. Then I connected 5.1 speaker system and noticed that lfe slider wasn't working and that old lfe slider worked. What I'm doing now is: - reverting old patch - adding internal lfe slider - removing side as it is superfluous (ALC888S-VC is 7.1 but in fact laptop can only do 5.1 and it is so in drivers for MS Windows) Signed-off-by: Łukasz Wojniłowicz Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 13 ++++++------- 1 file changed, 6 insertions(+), 7 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3328a259a24..a0f27b99992 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2295,13 +2295,13 @@ static struct snd_kcontrol_new alc888_acer_aspire_4930g_mixer[] = { HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0f, 2, 0x0, + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0f, 2, 2, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0f, 1, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0f, 1, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Side Playback Volume", 0x0e, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Side Playback Switch", 0x0e, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Internal LFE Playback Volume", 0x0f, 1, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Internal LFE Playback Switch", 0x0f, 1, 2, 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("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), @@ -2312,7 +2312,6 @@ static struct snd_kcontrol_new alc888_acer_aspire_4930g_mixer[] = { { } /* end */ }; - static struct snd_kcontrol_new alc889_acer_aspire_8930g_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), -- cgit v1.2.3-70-g09d2 From cd372fb3befde3bceef3fdcbc550dde50c894e36 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 3 Mar 2011 14:40:14 +0100 Subject: ALSA: hda - Make common input-jack helper functions Since multiple codec drivers already use the input-jack stuff, let's make common helper functions to reduce the duplicated codes. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 105 ++++++++++++++++++++++++++++++++++ sound/pci/hda/hda_codec.h | 5 ++ sound/pci/hda/hda_local.h | 24 ++++++++ sound/pci/hda/patch_conexant.c | 124 +++++------------------------------------ sound/pci/hda/patch_realtek.c | 94 ++++--------------------------- sound/pci/hda/patch_sigmatel.c | 85 ++-------------------------- 6 files changed, 165 insertions(+), 272 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index ae5c5d5e4b7..2c79e96d032 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -29,6 +29,7 @@ #include #include #include +#include #include "hda_local.h" #include "hda_beep.h" #include @@ -4959,5 +4960,109 @@ void snd_print_pcm_bits(int pcm, char *buf, int buflen) } EXPORT_SYMBOL_HDA(snd_print_pcm_bits); +#ifdef CONFIG_SND_HDA_INPUT_JACK +/* + * Input-jack notification support + */ +struct hda_jack_item { + hda_nid_t nid; + int type; + struct snd_jack *jack; +}; + +static const char *get_jack_default_name(struct hda_codec *codec, hda_nid_t nid, + int type) +{ + switch (type) { + case SND_JACK_HEADPHONE: + return "Headphone"; + case SND_JACK_MICROPHONE: + return "Mic"; + case SND_JACK_LINEOUT: + return "Line-out"; + case SND_JACK_HEADSET: + return "Headset"; + default: + return "Misc"; + } +} + +static void hda_free_jack_priv(struct snd_jack *jack) +{ + struct hda_jack_item *jacks = jack->private_data; + jacks->nid = 0; + jacks->jack = NULL; +} + +int snd_hda_input_jack_add(struct hda_codec *codec, hda_nid_t nid, int type, + const char *name) +{ + struct hda_jack_item *jack; + int err; + + snd_array_init(&codec->jacks, sizeof(*jack), 32); + jack = snd_array_new(&codec->jacks); + if (!jack) + return -ENOMEM; + + jack->nid = nid; + jack->type = type; + if (!name) + name = get_jack_default_name(codec, nid, type); + err = snd_jack_new(codec->bus->card, name, type, &jack->jack); + if (err < 0) { + jack->nid = 0; + return err; + } + jack->jack->private_data = jack; + jack->jack->private_free = hda_free_jack_priv; + return 0; +} +EXPORT_SYMBOL_HDA(snd_hda_input_jack_add); + +void snd_hda_input_jack_report(struct hda_codec *codec, hda_nid_t nid) +{ + struct hda_jack_item *jacks = codec->jacks.list; + int i; + + if (!jacks) + return; + + for (i = 0; i < codec->jacks.used; i++, jacks++) { + unsigned int pin_ctl; + unsigned int present; + int type; + + if (jacks->nid != nid) + continue; + present = snd_hda_jack_detect(codec, nid); + type = jacks->type; + if (type == (SND_JACK_HEADPHONE | SND_JACK_LINEOUT)) { + pin_ctl = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + type = (pin_ctl & AC_PINCTL_HP_EN) ? + SND_JACK_HEADPHONE : SND_JACK_LINEOUT; + } + snd_jack_report(jacks->jack, present ? type : 0); + } +} +EXPORT_SYMBOL_HDA(snd_hda_input_jack_report); + +/* free jack instances manually when clearing/reconfiguring */ +void snd_hda_input_jack_free(struct hda_codec *codec) +{ + if (!codec->bus->shutdown && codec->jacks.list) { + struct hda_jack_item *jacks = codec->jacks.list; + int i; + for (i = 0; i < codec->jacks.used; i++, jacks++) { + if (jacks->jack) + snd_device_free(codec->bus->card, jacks->jack); + } + } + snd_array_free(&codec->jacks); +} +EXPORT_SYMBOL_HDA(snd_hda_input_jack_free); +#endif /* CONFIG_SND_HDA_INPUT_JACK */ + MODULE_DESCRIPTION("HDA codec core"); MODULE_LICENSE("GPL"); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index fdf8d44f8b6..e46d5420a9f 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -866,6 +866,11 @@ struct hda_codec { /* codec-specific additional proc output */ void (*proc_widget_hook)(struct snd_info_buffer *buffer, struct hda_codec *codec, hda_nid_t nid); + +#ifdef CONFIG_SND_HDA_INPUT_JACK + /* jack detection */ + struct snd_array jacks; +#endif }; /* direction */ diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 3ab5e7a303d..ff5e2ac2239 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -656,4 +656,28 @@ static inline void snd_hda_eld_proc_free(struct hda_codec *codec, #define SND_PRINT_CHANNEL_ALLOCATION_ADVISED_BUFSIZE 80 void snd_print_channel_allocation(int spk_alloc, char *buf, int buflen); +/* + * Input-jack notification support + */ +#ifdef CONFIG_SND_HDA_INPUT_JACK +int snd_hda_input_jack_add(struct hda_codec *codec, hda_nid_t nid, int type, + const char *name); +void snd_hda_input_jack_report(struct hda_codec *codec, hda_nid_t nid); +void snd_hda_input_jack_free(struct hda_codec *codec); +#else /* CONFIG_SND_HDA_INPUT_JACK */ +static inline int snd_hda_input_jack_add(struct hda_codec *codec, + hda_nid_t nid, int type, + const char *name) +{ + return 0; +} +static inline void snd_hda_input_jack_report(struct hda_codec *codec, + hda_nid_t nid) +{ +} +static inline void snd_hda_input_jack_free(struct hda_codec *codec) +{ +} +#endif /* CONFIG_SND_HDA_INPUT_JACK */ + #endif /* __SOUND_HDA_LOCAL_H */ diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 4d5004e693f..d08cf31596f 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -49,14 +49,6 @@ #define AUTO_MIC_PORTB (1 << 1) #define AUTO_MIC_PORTC (1 << 2) -struct conexant_jack { - - hda_nid_t nid; - int type; - struct snd_jack *jack; - -}; - struct pin_dac_pair { hda_nid_t pin; hda_nid_t dac; @@ -111,9 +103,6 @@ struct conexant_spec { unsigned int spdif_route; - /* jack detection */ - struct snd_array jacks; - /* dynamic controls, init_verbs and input_mux */ struct auto_pin_cfg autocfg; struct hda_input_mux private_imux; @@ -393,71 +382,9 @@ static int conexant_mux_enum_put(struct snd_kcontrol *kcontrol, &spec->cur_mux[adc_idx]); } -#ifdef CONFIG_SND_HDA_INPUT_JACK -static void conexant_free_jack_priv(struct snd_jack *jack) -{ - struct conexant_jack *jacks = jack->private_data; - jacks->nid = 0; - jacks->jack = NULL; -} - -static int conexant_add_jack(struct hda_codec *codec, - hda_nid_t nid, int type) -{ - struct conexant_spec *spec; - struct conexant_jack *jack; - const char *name; - int i, err; - - spec = codec->spec; - snd_array_init(&spec->jacks, sizeof(*jack), 32); - - jack = spec->jacks.list; - for (i = 0; i < spec->jacks.used; i++, jack++) - if (jack->nid == nid) - return 0 ; /* already present */ - - jack = snd_array_new(&spec->jacks); - name = (type == SND_JACK_HEADPHONE) ? "Headphone" : "Mic" ; - - if (!jack) - return -ENOMEM; - - jack->nid = nid; - jack->type = type; - - err = snd_jack_new(codec->bus->card, name, type, &jack->jack); - if (err < 0) - return err; - jack->jack->private_data = jack; - jack->jack->private_free = conexant_free_jack_priv; - return 0; -} - -static void conexant_report_jack(struct hda_codec *codec, hda_nid_t nid) -{ - struct conexant_spec *spec = codec->spec; - struct conexant_jack *jacks = spec->jacks.list; - - if (jacks) { - int i; - for (i = 0; i < spec->jacks.used; i++) { - if (jacks->nid == nid) { - unsigned int present; - present = snd_hda_jack_detect(codec, nid); - - present = (present) ? jacks->type : 0 ; - - snd_jack_report(jacks->jack, - present); - } - jacks++; - } - } -} - static int conexant_init_jacks(struct hda_codec *codec) { +#ifdef CONFIG_SND_HDA_INPUT_JACK struct conexant_spec *spec = codec->spec; int i; @@ -469,15 +396,15 @@ static int conexant_init_jacks(struct hda_codec *codec) int err = 0; switch (hv->param ^ AC_USRSP_EN) { case CONEXANT_HP_EVENT: - err = conexant_add_jack(codec, hv->nid, - SND_JACK_HEADPHONE); - conexant_report_jack(codec, hv->nid); + err = snd_hda_input_jack_add(codec, hv->nid, + SND_JACK_HEADPHONE, NULL); + snd_hda_input_jack_report(codec, hv->nid); break; case CXT5051_PORTC_EVENT: case CONEXANT_MIC_EVENT: - err = conexant_add_jack(codec, hv->nid, - SND_JACK_MICROPHONE); - conexant_report_jack(codec, hv->nid); + err = snd_hda_input_jack_add(codec, hv->nid, + SND_JACK_MICROPHONE, NULL); + snd_hda_input_jack_report(codec, hv->nid); break; } if (err < 0) @@ -485,19 +412,9 @@ static int conexant_init_jacks(struct hda_codec *codec) ++hv; } } - return 0; - -} -#else -static inline void conexant_report_jack(struct hda_codec *codec, hda_nid_t nid) -{ -} - -static inline int conexant_init_jacks(struct hda_codec *codec) -{ +#endif /* CONFIG_SND_HDA_INPUT_JACK */ return 0; } -#endif static int conexant_init(struct hda_codec *codec) { @@ -511,18 +428,7 @@ static int conexant_init(struct hda_codec *codec) static void conexant_free(struct hda_codec *codec) { -#ifdef CONFIG_SND_HDA_INPUT_JACK - struct conexant_spec *spec = codec->spec; - if (spec->jacks.list) { - struct conexant_jack *jacks = spec->jacks.list; - int i; - for (i = 0; i < spec->jacks.used; i++, jacks++) { - if (jacks->jack) - snd_device_free(codec->bus->card, jacks->jack); - } - snd_array_free(&spec->jacks); - } -#endif + snd_hda_input_jack_free(codec); snd_hda_detach_beep_device(codec); kfree(codec->spec); } @@ -1787,7 +1693,7 @@ static void cxt5051_hp_unsol_event(struct hda_codec *codec, cxt5051_portc_automic(codec); break; } - conexant_report_jack(codec, nid); + snd_hda_input_jack_report(codec, nid); } static struct snd_kcontrol_new cxt5051_playback_mixers[] = { @@ -1959,10 +1865,8 @@ static void cxt5051_init_mic_port(struct hda_codec *codec, hda_nid_t nid, snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | event); -#ifdef CONFIG_SND_HDA_INPUT_JACK - conexant_add_jack(codec, nid, SND_JACK_MICROPHONE); - conexant_report_jack(codec, nid); -#endif + snd_hda_input_jack_add(codec, nid, SND_JACK_MICROPHONE, NULL); + snd_hda_input_jack_report(codec, nid); } static struct hda_verb cxt5051_ideapad_init_verbs[] = { @@ -3477,11 +3381,11 @@ static void cx_auto_unsol_event(struct hda_codec *codec, unsigned int res) switch (res >> 26) { case CONEXANT_HP_EVENT: cx_auto_hp_automute(codec); - conexant_report_jack(codec, nid); + snd_hda_input_jack_report(codec, nid); break; case CONEXANT_MIC_EVENT: cx_auto_automic(codec); - conexant_report_jack(codec, nid); + snd_hda_input_jack_report(codec, nid); break; } } diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 96cb442dc1d..f6c344126b8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -282,12 +282,6 @@ struct alc_mic_route { unsigned char amix_idx; }; -struct alc_jack { - hda_nid_t nid; - int type; - struct snd_jack *jack; -}; - #define MUX_IDX_UNDEF ((unsigned char)-1) struct alc_customize_define { @@ -366,9 +360,6 @@ struct alc_spec { /* PCM information */ struct hda_pcm pcm_rec[3]; /* used in alc_build_pcms() */ - /* jack detection */ - struct snd_array jacks; - /* dynamic controls, init_verbs and input_mux */ struct auto_pin_cfg autocfg; struct alc_customize_define cdefine; @@ -1032,94 +1023,32 @@ static void alc_fix_pll_init(struct hda_codec *codec, hda_nid_t nid, alc_fix_pll(codec); } -#ifdef CONFIG_SND_HDA_INPUT_JACK -static void alc_free_jack_priv(struct snd_jack *jack) -{ - struct alc_jack *jacks = jack->private_data; - jacks->nid = 0; - jacks->jack = NULL; -} - -static int alc_add_jack(struct hda_codec *codec, - hda_nid_t nid, int type) -{ - struct alc_spec *spec; - struct alc_jack *jack; - const char *name; - int err; - - spec = codec->spec; - snd_array_init(&spec->jacks, sizeof(*jack), 32); - jack = snd_array_new(&spec->jacks); - if (!jack) - return -ENOMEM; - - jack->nid = nid; - jack->type = type; - name = (type == SND_JACK_HEADPHONE) ? "Headphone" : "Mic" ; - - err = snd_jack_new(codec->bus->card, name, type, &jack->jack); - if (err < 0) - return err; - jack->jack->private_data = jack; - jack->jack->private_free = alc_free_jack_priv; - return 0; -} - -static void alc_report_jack(struct hda_codec *codec, hda_nid_t nid) -{ - struct alc_spec *spec = codec->spec; - struct alc_jack *jacks = spec->jacks.list; - - if (jacks) { - int i; - for (i = 0; i < spec->jacks.used; i++) { - if (jacks->nid == nid) { - unsigned int present; - present = snd_hda_jack_detect(codec, nid); - - present = (present) ? jacks->type : 0; - - snd_jack_report(jacks->jack, present); - } - jacks++; - } - } -} - static int alc_init_jacks(struct hda_codec *codec) { +#ifdef CONFIG_SND_HDA_INPUT_JACK struct alc_spec *spec = codec->spec; int err; unsigned int hp_nid = spec->autocfg.hp_pins[0]; unsigned int mic_nid = spec->ext_mic.pin; if (hp_nid) { - err = alc_add_jack(codec, hp_nid, SND_JACK_HEADPHONE); + err = snd_hda_input_jack_add(codec, hp_nid, + SND_JACK_HEADPHONE, NULL); if (err < 0) return err; - alc_report_jack(codec, hp_nid); + snd_hda_input_jack_report(codec, hp_nid); } if (mic_nid) { - err = alc_add_jack(codec, mic_nid, SND_JACK_MICROPHONE); + err = snd_hda_input_jack_add(codec, mic_nid, + SND_JACK_MICROPHONE, NULL); if (err < 0) return err; - alc_report_jack(codec, mic_nid); + snd_hda_input_jack_report(codec, mic_nid); } - - return 0; -} -#else -static inline void alc_report_jack(struct hda_codec *codec, hda_nid_t nid) -{ -} - -static inline int alc_init_jacks(struct hda_codec *codec) -{ +#endif /* CONFIG_SND_HDA_INPUT_JACK */ return 0; } -#endif static void alc_automute_speaker(struct hda_codec *codec, int pinctl) { @@ -1133,7 +1062,7 @@ static void alc_automute_speaker(struct hda_codec *codec, int pinctl) nid = spec->autocfg.hp_pins[i]; if (!nid) break; - alc_report_jack(codec, nid); + snd_hda_input_jack_report(codec, nid); spec->jack_present |= snd_hda_jack_detect(codec, nid); } @@ -1240,7 +1169,7 @@ static void alc_mic_automute(struct hda_codec *codec) AC_VERB_SET_CONNECT_SEL, alive->mux_idx); } - alc_report_jack(codec, spec->ext_mic.pin); + snd_hda_input_jack_report(codec, spec->ext_mic.pin); /* FIXME: analog mixer */ } @@ -4283,6 +4212,7 @@ static void alc_free(struct hda_codec *codec) return; alc_shutup(codec); + snd_hda_input_jack_free(codec); alc_free_kctls(codec); kfree(spec); snd_hda_detach_beep_device(codec); @@ -14494,7 +14424,7 @@ static void alc269_speaker_automute(struct hda_codec *codec) HDA_AMP_MUTE, bits); snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, HDA_AMP_MUTE, bits); - alc_report_jack(codec, nid); + snd_hda_input_jack_report(codec, nid); } /* unsolicited event for HP jack sensing */ diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index bd7b123f644..8fe5608c5f2 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -180,12 +180,6 @@ struct sigmatel_event { int data; }; -struct sigmatel_jack { - hda_nid_t nid; - int type; - struct snd_jack *jack; -}; - struct sigmatel_mic_route { hda_nid_t pin; signed char mux_idx; @@ -229,9 +223,6 @@ struct sigmatel_spec { hda_nid_t *pwr_nids; hda_nid_t *dac_list; - /* jack detection */ - struct snd_array jacks; - /* events */ struct snd_array events; @@ -4054,21 +4045,10 @@ static void stac_gpio_set(struct hda_codec *codec, unsigned int mask, AC_VERB_SET_GPIO_DATA, gpiostate); /* sync */ } -#ifdef CONFIG_SND_HDA_INPUT_JACK -static void stac92xx_free_jack_priv(struct snd_jack *jack) -{ - struct sigmatel_jack *jacks = jack->private_data; - jacks->nid = 0; - jacks->jack = NULL; -} -#endif - static int stac92xx_add_jack(struct hda_codec *codec, hda_nid_t nid, int type) { #ifdef CONFIG_SND_HDA_INPUT_JACK - struct sigmatel_spec *spec = codec->spec; - struct sigmatel_jack *jack; int def_conf = snd_hda_codec_get_pincfg(codec, nid); int connectivity = get_defcfg_connect(def_conf); char name[32]; @@ -4077,26 +4057,15 @@ static int stac92xx_add_jack(struct hda_codec *codec, if (connectivity && connectivity != AC_JACK_PORT_FIXED) return 0; - snd_array_init(&spec->jacks, sizeof(*jack), 32); - jack = snd_array_new(&spec->jacks); - if (!jack) - return -ENOMEM; - jack->nid = nid; - jack->type = type; - snprintf(name, sizeof(name), "%s at %s %s Jack", snd_hda_get_jack_type(def_conf), snd_hda_get_jack_connectivity(def_conf), snd_hda_get_jack_location(def_conf)); - err = snd_jack_new(codec->bus->card, name, type, &jack->jack); - if (err < 0) { - jack->nid = 0; + err = snd_hda_input_jack_add(codec, nid, type, name); + if (err < 0) return err; - } - jack->jack->private_data = jack; - jack->jack->private_free = stac92xx_free_jack_priv; -#endif +#endif /* CONFIG_SND_HDA_INPUT_JACK */ return 0; } @@ -4399,23 +4368,6 @@ static int stac92xx_init(struct hda_codec *codec) return 0; } -static void stac92xx_free_jacks(struct hda_codec *codec) -{ -#ifdef CONFIG_SND_HDA_INPUT_JACK - /* free jack instances manually when clearing/reconfiguring */ - struct sigmatel_spec *spec = codec->spec; - if (!codec->bus->shutdown && spec->jacks.list) { - struct sigmatel_jack *jacks = spec->jacks.list; - int i; - for (i = 0; i < spec->jacks.used; i++, jacks++) { - if (jacks->jack) - snd_device_free(codec->bus->card, jacks->jack); - } - } - snd_array_free(&spec->jacks); -#endif -} - static void stac92xx_free_kctls(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; @@ -4449,7 +4401,7 @@ static void stac92xx_free(struct hda_codec *codec) return; stac92xx_shutup(codec); - stac92xx_free_jacks(codec); + snd_hda_input_jack_free(codec); snd_array_free(&spec->events); kfree(spec); @@ -4667,33 +4619,6 @@ static void stac92xx_pin_sense(struct hda_codec *codec, hda_nid_t nid) stac_toggle_power_map(codec, nid, get_pin_presence(codec, nid)); } -static void stac92xx_report_jack(struct hda_codec *codec, hda_nid_t nid) -{ - struct sigmatel_spec *spec = codec->spec; - struct sigmatel_jack *jacks = spec->jacks.list; - - if (jacks) { - int i; - for (i = 0; i < spec->jacks.used; i++) { - if (jacks->nid == nid) { - unsigned int pin_ctl = - snd_hda_codec_read(codec, nid, - 0, AC_VERB_GET_PIN_WIDGET_CONTROL, - 0x00); - int type = jacks->type; - if (type == (SND_JACK_LINEOUT - | SND_JACK_HEADPHONE)) - type = (pin_ctl & AC_PINCTL_HP_EN) - ? SND_JACK_HEADPHONE : SND_JACK_LINEOUT; - snd_jack_report(jacks->jack, - get_pin_presence(codec, nid) - ? type : 0); - } - jacks++; - } - } -} - /* get the pin connection (fixed, none, etc) */ static unsigned int stac_get_defcfg_connect(struct hda_codec *codec, int idx) { @@ -4782,7 +4707,7 @@ static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res) case STAC_PWR_EVENT: if (spec->num_pwrs > 0) stac92xx_pin_sense(codec, event->nid); - stac92xx_report_jack(codec, event->nid); + snd_hda_input_jack_report(codec, event->nid); switch (codec->subsystem_id) { case 0x103c308f: -- cgit v1.2.3-70-g09d2 From 32eea3884debb65ec1da633bc5df5aee23879865 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Fri, 4 Mar 2011 13:37:50 +0100 Subject: ALSA: HDA: Enable surround and subwoofer on Lenovo Ideapad Y530 MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The pin config values would change the association instead of the sequence, this commit fixes that up. Tested-by: Bartłomiej Żogała Cc: Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 10 ++++++++++ 1 file changed, 10 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f6c344126b8..b4a22dd775a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10677,6 +10677,7 @@ static struct alc_config_preset alc882_presets[] = { */ enum { PINFIX_ABIT_AW9D_MAX, + PINFIX_LENOVO_Y530, PINFIX_PB_M5210, PINFIX_ACER_ASPIRE_7736, }; @@ -10691,6 +10692,14 @@ static const struct alc_fixup alc882_fixups[] = { { } } }, + [PINFIX_LENOVO_Y530] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x15, 0x99130112 }, /* rear int speakers */ + { 0x16, 0x99130111 }, /* subwoofer */ + { } + } + }, [PINFIX_PB_M5210] = { .type = ALC_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -10706,6 +10715,7 @@ static const struct alc_fixup alc882_fixups[] = { static struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0155, "Packard-Bell M5120", PINFIX_PB_M5210), + SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Y530", PINFIX_LENOVO_Y530), SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", PINFIX_ABIT_AW9D_MAX), SND_PCI_QUIRK(0x1025, 0x0296, "Acer Aspire 7736z", PINFIX_ACER_ASPIRE_7736), {} -- cgit v1.2.3-70-g09d2 From ebbeb3d6aa22433c218da6f29fd7b3ebc89b87ea Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Fri, 4 Mar 2011 14:08:30 +0100 Subject: ALSA: HDA: Fix volume control naming for surround speakers on Realtek auto-parser MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit When more than one pair of internal speakers is present, allow names according to their channels. Tested-by: Bartłomiej Żogała Cc: Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b4a22dd775a..6f59bccb104 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5080,7 +5080,9 @@ static const char *alc_get_line_out_pfx(const struct auto_pin_cfg *cfg, switch (cfg->line_out_type) { case AUTO_PIN_SPEAKER_OUT: - return "Speaker"; + if (cfg->line_outs == 1) + return "Speaker"; + break; case AUTO_PIN_HP_OUT: return "Headphone"; default: -- cgit v1.2.3-70-g09d2 From 7e59e097c09b82760bb0fe08b0fa2b704d76c3f4 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Fri, 4 Mar 2011 14:22:25 +0100 Subject: ALSA: HDA: Fixup unnecessary volume control index on Realtek ALC88x MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Without this change, a volume control named "Surround" or "Side" would get an unnecessary index, causing it to be ignored by the vmaster and PulseAudio. Tested-by: Bartłomiej Żogała Cc: Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6f59bccb104..24bbc47dcea 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5136,16 +5136,19 @@ static int alc880_auto_create_multi_out_ctls(struct alc_spec *spec, return err; } else { const char *name = pfx; - if (!name) + int index = i; + if (!name) { name = chname[i]; + index = 0; + } err = __add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, - name, i, + name, index, HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT)); if (err < 0) return err; err = __add_pb_sw_ctrl(spec, ALC_CTL_BIND_MUTE, - name, i, + name, index, HDA_COMPOSE_AMP_VAL(nid, 3, 2, HDA_INPUT)); if (err < 0) -- cgit v1.2.3-70-g09d2 From 0a3fabe30e1a3b2037a12b863b8c45fffce38ee9 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Fri, 4 Mar 2011 16:54:52 +0100 Subject: ALSA: HDA: Realtek ALC88x: Do not over-initialize speakers and hp that are primary outputs MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Do not initialize again the what has already been initialized as multi outs, as this breaks surround speakers. Tested-by: Bartłomiej Żogała Cc: Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 37 +++++++++++++++++++++---------------- 1 file changed, 21 insertions(+), 16 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 24bbc47dcea..d403ee825ef 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10773,23 +10773,28 @@ static void alc882_auto_init_hp_out(struct hda_codec *codec) hda_nid_t pin, dac; int i; - for (i = 0; i < ARRAY_SIZE(spec->autocfg.hp_pins); i++) { - pin = spec->autocfg.hp_pins[i]; - if (!pin) - break; - dac = spec->multiout.hp_nid; - if (!dac) - dac = spec->multiout.dac_nids[0]; /* to front */ - alc882_auto_set_output_and_unmute(codec, pin, PIN_HP, dac); + if (spec->autocfg.line_out_type != AUTO_PIN_HP_OUT) { + for (i = 0; i < ARRAY_SIZE(spec->autocfg.hp_pins); i++) { + pin = spec->autocfg.hp_pins[i]; + if (!pin) + break; + dac = spec->multiout.hp_nid; + if (!dac) + dac = spec->multiout.dac_nids[0]; /* to front */ + alc882_auto_set_output_and_unmute(codec, pin, PIN_HP, dac); + } } - for (i = 0; i < ARRAY_SIZE(spec->autocfg.speaker_pins); i++) { - pin = spec->autocfg.speaker_pins[i]; - if (!pin) - break; - dac = spec->multiout.extra_out_nid[0]; - if (!dac) - dac = spec->multiout.dac_nids[0]; /* to front */ - alc882_auto_set_output_and_unmute(codec, pin, PIN_OUT, dac); + + if (spec->autocfg.line_out_type != AUTO_PIN_SPEAKER_OUT) { + for (i = 0; i < ARRAY_SIZE(spec->autocfg.speaker_pins); i++) { + pin = spec->autocfg.speaker_pins[i]; + if (!pin) + break; + dac = spec->multiout.extra_out_nid[0]; + if (!dac) + dac = spec->multiout.dac_nids[0]; /* to front */ + alc882_auto_set_output_and_unmute(codec, pin, PIN_OUT, dac); + } } } -- cgit v1.2.3-70-g09d2 From a09e89f67ca56d6fa7634bd0738d64fa61bc3c39 Mon Sep 17 00:00:00 2001 From: Adam Lackorzynski Date: Thu, 10 Mar 2011 17:41:56 +0100 Subject: ALSA: hda: Prevent writing ICH6_PCIREG_TCSEL on AMD systems azx_init_pci() always writes PCI config register ICH6_PCIREG_TCSEL although this looks to be only defined on Intel systems and has a different meaning on AMD systems. On AMD systems the PCI interrupt pin control register is modified instead. Since the meaning of offset 0x44 in device specific configuration space is unknown for devices by other vendors, we only exclude AMD systems to retain the current behaviour. Signed-off-by: Adam Lackorzynski Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index fcedad9a5fe..70a9d32f0e9 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1052,9 +1052,12 @@ static void azx_init_pci(struct azx *chip) /* Clear bits 0-2 of PCI register TCSEL (at offset 0x44) * TCSEL == Traffic Class Select Register, which sets PCI express QOS * Ensuring these bits are 0 clears playback static on some HD Audio - * codecs + * codecs. + * The PCI register TCSEL is defined in the Intel manuals. */ - update_pci_byte(chip->pci, ICH6_PCIREG_TCSEL, 0x07, 0); + if (chip->driver_type != AZX_DRIVER_ATI && + chip->driver_type != AZX_DRIVER_ATIHDMI) + update_pci_byte(chip->pci, ICH6_PCIREG_TCSEL, 0x07, 0); switch (chip->driver_type) { case AZX_DRIVER_ATI: -- cgit v1.2.3-70-g09d2 From 584c0c4c359bdac37d94157f8d7fc513d26c8328 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 10 Mar 2011 12:51:11 +0100 Subject: ALSA: hda - Initialize special cases for input src in init phase Currently some special handling for the unusual case like dual-ADCs or a single-input-src is done in the tree-parse time in set_capture_mixer(). But this setup could be overwritten by static init verbs. This patch moves the initialization into the init phase so that such input-src setup won't be lost. Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 19 ++++++++++++++++--- 1 file changed, 16 insertions(+), 3 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d403ee825ef..dcc455f098a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -385,6 +385,7 @@ struct alc_spec { /* other flags */ unsigned int no_analog :1; /* digital I/O only */ unsigned int dual_adc_switch:1; /* switch ADCs (for ALC275) */ + unsigned int single_input_src:1; int init_amp; int codec_variant; /* flag for other variants */ @@ -3847,6 +3848,8 @@ static struct hda_amp_list alc880_lg_loopbacks[] = { * Common callbacks */ +static void alc_init_special_input_src(struct hda_codec *codec); + static int alc_init(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -3857,6 +3860,7 @@ static int alc_init(struct hda_codec *codec) for (i = 0; i < spec->num_init_verbs; i++) snd_hda_sequence_write(codec, spec->init_verbs[i]); + alc_init_special_input_src(codec); if (spec->init_hook) spec->init_hook(codec); @@ -5519,6 +5523,7 @@ static void fixup_single_adc(struct hda_codec *codec) spec->capsrc_nids += i; spec->adc_nids += i; spec->num_adc_nids = 1; + spec->single_input_src = 1; } } @@ -5530,6 +5535,16 @@ static void fixup_dual_adc_switch(struct hda_codec *codec) init_capsrc_for_pin(codec, spec->int_mic.pin); } +/* initialize some special cases for input sources */ +static void alc_init_special_input_src(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + if (spec->dual_adc_switch) + fixup_dual_adc_switch(codec); + else if (spec->single_input_src) + init_capsrc_for_pin(codec, spec->autocfg.inputs[0].pin); +} + static void set_capture_mixer(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -5545,7 +5560,7 @@ static void set_capture_mixer(struct hda_codec *codec) int mux = 0; int num_adcs = spec->num_adc_nids; if (spec->dual_adc_switch) - fixup_dual_adc_switch(codec); + num_adcs = 1; else if (spec->auto_mic) fixup_automic_adc(codec); else if (spec->input_mux) { @@ -5554,8 +5569,6 @@ static void set_capture_mixer(struct hda_codec *codec) else if (spec->input_mux->num_items == 1) fixup_single_adc(codec); } - if (spec->dual_adc_switch) - num_adcs = 1; spec->cap_mixer = caps[mux][num_adcs - 1]; } } -- cgit v1.2.3-70-g09d2 From ae0ebbf70afe2889b39f575e800e7292abd259d6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 10 Mar 2011 14:11:59 +0100 Subject: ALSA: hda - Move default input-src selection to init part Move the default input-src selection code for alc268/269 to the init part instead of the parser. The input-src selection might be overwritten by init verbs. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 17 +++++------------ 1 file changed, 5 insertions(+), 12 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index dcc455f098a..11a1380b882 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -13758,6 +13758,7 @@ static int alc268_parse_auto_config(struct hda_codec *codec) } #define alc268_auto_init_analog_input alc882_auto_init_analog_input +#define alc268_auto_init_input_src alc882_auto_init_input_src /* init callback for auto-configuration model -- overriding the default init */ static void alc268_auto_init(struct hda_codec *codec) @@ -13767,6 +13768,7 @@ static void alc268_auto_init(struct hda_codec *codec) alc268_auto_init_hp_out(codec); alc268_auto_init_mono_speaker_out(codec); alc268_auto_init_analog_input(codec); + alc268_auto_init_input_src(codec); alc_auto_init_digital(codec); if (spec->unsol_event) alc_inithook(codec); @@ -14074,13 +14076,6 @@ static int patch_alc268(struct hda_codec *codec) spec->num_adc_nids = ARRAY_SIZE(alc268_adc_nids); add_mixer(spec, alc268_capture_mixer); } - /* set default input source */ - for (i = 0; i < spec->num_adc_nids; i++) - snd_hda_codec_write_cache(codec, alc268_capsrc_nids[i], - 0, AC_VERB_SET_CONNECT_SEL, - i < spec->num_mux_defs ? - spec->input_mux[i].items[0].index : - spec->input_mux->items[0].index); } spec->vmaster_nid = 0x02; @@ -14769,11 +14764,6 @@ static int alc269_parse_auto_config(struct hda_codec *codec) fillup_priv_adc_nids(codec, alc269_adc_candidates, sizeof(alc269_adc_candidates)); - /* set default input source */ - if (!spec->dual_adc_switch) - select_or_unmute_capsrc(codec, spec->capsrc_nids[0], - spec->input_mux->items[0].index); - err = alc_auto_add_mic_boost(codec); if (err < 0) return err; @@ -14787,6 +14777,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec) #define alc269_auto_init_multi_out alc268_auto_init_multi_out #define alc269_auto_init_hp_out alc268_auto_init_hp_out #define alc269_auto_init_analog_input alc882_auto_init_analog_input +#define alc269_auto_init_input_src alc882_auto_init_input_src /* init callback for auto-configuration model -- overriding the default init */ @@ -14796,6 +14787,8 @@ static void alc269_auto_init(struct hda_codec *codec) alc269_auto_init_multi_out(codec); alc269_auto_init_hp_out(codec); alc269_auto_init_analog_input(codec); + if (!spec->dual_adc_switch) + alc269_auto_init_input_src(codec); alc_auto_init_digital(codec); if (spec->unsol_event) alc_inithook(codec); -- cgit v1.2.3-70-g09d2 From 094a42452abd5564429045e210281c6d22e67fca Mon Sep 17 00:00:00 2001 From: Vitaliy Kulikov Date: Wed, 9 Mar 2011 19:47:43 -0600 Subject: ALSA: hda - fix digital mic selection in mixer on 92HD8X codecs When the mux for digital mic is different from the mux for other mics, the current auto-parser doesn't handle them in a right way but provides only one mic. This patch fixes the issue. Signed-off-by: Vitaliy Kulikov Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 8fe5608c5f2..32f744d47da 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -748,7 +748,7 @@ static int stac92xx_mux_enum_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e struct sigmatel_spec *spec = codec->spec; unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); const struct hda_input_mux *imux = spec->input_mux; - unsigned int idx, prev_idx; + unsigned int idx, prev_idx, didx; idx = ucontrol->value.enumerated.item[0]; if (idx >= imux->num_items) @@ -760,7 +760,8 @@ static int stac92xx_mux_enum_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e snd_hda_codec_write_cache(codec, spec->mux_nids[adc_idx], 0, AC_VERB_SET_CONNECT_SEL, imux->items[idx].index); - if (prev_idx >= spec->num_analog_muxes) { + if (prev_idx >= spec->num_analog_muxes && + spec->mux_nids[adc_idx] != spec->dmux_nids[adc_idx]) { imux = spec->dinput_mux; /* 0 = analog */ snd_hda_codec_write_cache(codec, @@ -770,9 +771,13 @@ static int stac92xx_mux_enum_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e } } else { imux = spec->dinput_mux; + /* first dimux item is hardcoded to select analog imux, + * so lets skip it + */ + didx = idx - spec->num_analog_muxes + 1; snd_hda_codec_write_cache(codec, spec->dmux_nids[adc_idx], 0, AC_VERB_SET_CONNECT_SEL, - imux->items[idx - 1].index); + imux->items[didx].index); } spec->cur_mux[adc_idx] = idx; return 1; -- cgit v1.2.3-70-g09d2 From 699d899560cd7e72da39231e584412e7ac8114a4 Mon Sep 17 00:00:00 2001 From: Vitaliy Kulikov Date: Thu, 10 Mar 2011 13:43:35 -0600 Subject: ALSA: hda - pin-adc-mux-dmic auto-configuration of 92HD8X codecs This patch replaces use of the harcoded arrays of pins, muxes, digital mics and adcs with the auto-generated ones using codec parsing and auto-discovers all actually connected digital mic pins on 92HD8X-like codecs This patch also adds the support for d-mic on pin 0x20. Signed-off-by: Vitaliy Kulikov Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 222 +++++++++++++++++++++++++++-------------- 1 file changed, 149 insertions(+), 73 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 32f744d47da..05fcd60cc46 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -186,6 +186,10 @@ struct sigmatel_mic_route { signed char dmux_idx; }; +#define MAX_PINS_NUM 16 +#define MAX_ADCS_NUM 4 +#define MAX_DMICS_NUM 4 + struct sigmatel_spec { struct snd_kcontrol_new *mixers[4]; unsigned int num_mixers; @@ -300,6 +304,17 @@ struct sigmatel_spec { struct hda_input_mux private_imux; struct hda_input_mux private_smux; struct hda_input_mux private_mono_mux; + + /* auto spec */ + unsigned auto_pin_cnt; + hda_nid_t auto_pin_nids[MAX_PINS_NUM]; + unsigned auto_adc_cnt; + hda_nid_t auto_adc_nids[MAX_ADCS_NUM]; + hda_nid_t auto_mux_nids[MAX_ADCS_NUM]; + hda_nid_t auto_dmux_nids[MAX_ADCS_NUM]; + unsigned long auto_capvols[MAX_ADCS_NUM]; + unsigned auto_dmic_cnt; + hda_nid_t auto_dmic_nids[MAX_DMICS_NUM]; }; static hda_nid_t stac9200_adc_nids[1] = { @@ -355,14 +370,6 @@ static unsigned long stac92hd73xx_capvols[] = { #define STAC92HD83_DAC_COUNT 3 -static hda_nid_t stac92hd83xxx_mux_nids[2] = { - 0x17, 0x18, -}; - -static hda_nid_t stac92hd83xxx_adc_nids[2] = { - 0x15, 0x16, -}; - static hda_nid_t stac92hd83xxx_pwr_nids[4] = { 0xa, 0xb, 0xd, 0xe, }; @@ -375,25 +382,9 @@ static unsigned int stac92hd83xxx_pwr_mapping[4] = { 0x03, 0x0c, 0x20, 0x40, }; -#define STAC92HD83XXX_NUM_DMICS 2 -static hda_nid_t stac92hd83xxx_dmic_nids[STAC92HD83XXX_NUM_DMICS + 1] = { - 0x11, 0x20, 0 -}; - -#define STAC92HD88XXX_NUM_DMICS STAC92HD83XXX_NUM_DMICS -#define stac92hd88xxx_dmic_nids stac92hd83xxx_dmic_nids - -#define STAC92HD87B_NUM_DMICS 1 -static hda_nid_t stac92hd87b_dmic_nids[STAC92HD87B_NUM_DMICS + 1] = { - 0x11, 0 -}; - -#define STAC92HD83XXX_NUM_CAPS 2 -static unsigned long stac92hd83xxx_capvols[] = { - HDA_COMPOSE_AMP_VAL(0x17, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(0x18, 3, 0, HDA_OUTPUT), +static hda_nid_t stac92hd83xxx_dmic_nids[] = { + 0x11, 0x20, }; -#define stac92hd83xxx_capsws stac92hd83xxx_capvols static hda_nid_t stac92hd71bxx_pwr_nids[3] = { 0x0a, 0x0d, 0x0f @@ -572,21 +563,6 @@ static hda_nid_t stac92hd73xx_pin_nids[13] = { 0x14, 0x22, 0x23 }; -static hda_nid_t stac92hd83xxx_pin_nids[10] = { - 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, - 0x0f, 0x10, 0x11, 0x1f, 0x20, -}; - -static hda_nid_t stac92hd87xxx_pin_nids[6] = { - 0x0a, 0x0b, 0x0c, 0x0d, - 0x0f, 0x11, -}; - -static hda_nid_t stac92hd88xxx_pin_nids[8] = { - 0x0a, 0x0b, 0x0c, 0x0d, - 0x0f, 0x11, 0x1f, 0x20, -}; - #define STAC92HD71BXX_NUM_PINS 13 static hda_nid_t stac92hd71bxx_pin_nids_4port[STAC92HD71BXX_NUM_PINS] = { 0x0a, 0x0b, 0x0c, 0x0d, 0x00, @@ -3415,6 +3391,17 @@ static const char * const stac92xx_dmic_labels[5] = { "Digital Mic 3", "Digital Mic 4" }; +static hda_nid_t get_connected_node(struct hda_codec *codec, hda_nid_t mux, + int idx) +{ + hda_nid_t conn[HDA_MAX_NUM_INPUTS]; + int nums; + nums = snd_hda_get_connections(codec, mux, conn, ARRAY_SIZE(conn)); + if (idx >= 0 && idx < nums) + return conn[idx]; + return 0; +} + static int get_connection_index(struct hda_codec *codec, hda_nid_t mux, hda_nid_t nid) { @@ -3425,6 +3412,15 @@ static int get_connection_index(struct hda_codec *codec, hda_nid_t mux, for (i = 0; i < nums; i++) if (conn[i] == nid) return i; + + for (i = 0; i < nums; i++) { + unsigned int wid_caps = get_wcaps(codec, conn[i]); + unsigned int wid_type = get_wcaps_type(wid_caps); + + if (wid_type != AC_WID_PIN && wid_type != AC_WID_AUD_MIX) + if (get_connection_index(codec, conn[i], nid) >= 0) + return i; + } return -1; } @@ -3497,6 +3493,16 @@ static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec, type_idx, HDA_OUTPUT); if (err < 0) return err; + if (!err) { + nid = get_connected_node(codec, + spec->dmux_nids[0], index); + if (nid) + err = create_elem_capture_vol(codec, + nid, label, + type_idx, HDA_INPUT); + if (err < 0) + return err; + } } } @@ -5308,6 +5314,105 @@ static int hp_bnb2011_with_dock(struct hda_codec *codec) return 0; } +static void stac92hd8x_add_pin(struct hda_codec *codec, hda_nid_t nid) +{ + struct sigmatel_spec *spec = codec->spec; + unsigned int def_conf = snd_hda_codec_get_pincfg(codec, nid); + int i; + + spec->auto_pin_nids[spec->auto_pin_cnt] = nid; + spec->auto_pin_cnt++; + + if (get_defcfg_device(def_conf) == AC_JACK_MIC_IN && + get_defcfg_connect(def_conf) != AC_JACK_PORT_NONE) { + for (i = 0; i < ARRAY_SIZE(stac92hd83xxx_dmic_nids); i++) { + if (nid == stac92hd83xxx_dmic_nids[i]) { + spec->auto_dmic_nids[spec->auto_dmic_cnt] = nid; + spec->auto_dmic_cnt++; + } + } + } +} + +static void stac92hd8x_add_adc(struct hda_codec *codec, hda_nid_t nid) +{ + struct sigmatel_spec *spec = codec->spec; + + spec->auto_adc_nids[spec->auto_adc_cnt] = nid; + spec->auto_adc_cnt++; +} + +static void stac92hd8x_add_mux(struct hda_codec *codec, hda_nid_t nid) +{ + int i, j; + struct sigmatel_spec *spec = codec->spec; + + for (i = 0; i < spec->auto_adc_cnt; i++) { + if (get_connection_index(codec, + spec->auto_adc_nids[i], nid) >= 0) { + /* mux and volume for adc_nids[i] */ + if (!spec->auto_mux_nids[i]) { + spec->auto_mux_nids[i] = nid; + /* 92hd codecs capture volume is in mux */ + spec->auto_capvols[i] = HDA_COMPOSE_AMP_VAL(nid, + 3, 0, HDA_OUTPUT); + } + for (j = 0; j < spec->auto_dmic_cnt; j++) { + if (get_connection_index(codec, nid, + spec->auto_dmic_nids[j]) >= 0) { + /* dmux for adc_nids[i] */ + if (!spec->auto_dmux_nids[i]) + spec->auto_dmux_nids[i] = nid; + break; + } + } + break; + } + } +} + +static void stac92hd8x_fill_auto_spec(struct hda_codec *codec) +{ + hda_nid_t nid, end_nid; + unsigned int wid_caps, wid_type; + struct sigmatel_spec *spec = codec->spec; + + end_nid = codec->start_nid + codec->num_nodes; + + for (nid = codec->start_nid; nid < end_nid; nid++) { + wid_caps = get_wcaps(codec, nid); + wid_type = get_wcaps_type(wid_caps); + + if (wid_type == AC_WID_PIN) + stac92hd8x_add_pin(codec, nid); + + if (wid_type == AC_WID_AUD_IN && !(wid_caps & AC_WCAP_DIGITAL)) + stac92hd8x_add_adc(codec, nid); + } + + for (nid = codec->start_nid; nid < end_nid; nid++) { + wid_caps = get_wcaps(codec, nid); + wid_type = get_wcaps_type(wid_caps); + + if (wid_type == AC_WID_AUD_SEL) + stac92hd8x_add_mux(codec, nid); + } + + spec->pin_nids = spec->auto_pin_nids; + spec->num_pins = spec->auto_pin_cnt; + spec->adc_nids = spec->auto_adc_nids; + spec->num_adcs = spec->auto_adc_cnt; + spec->capvols = spec->auto_capvols; + spec->capsws = spec->auto_capvols; + spec->num_caps = spec->auto_adc_cnt; + spec->mux_nids = spec->auto_mux_nids; + spec->num_muxes = spec->auto_adc_cnt; + spec->dmux_nids = spec->auto_dmux_nids; + spec->num_dmuxes = spec->auto_adc_cnt; + spec->dmic_nids = spec->auto_dmic_nids; + spec->num_dmics = spec->auto_dmic_cnt; +} + static int patch_stac92hd83xxx(struct hda_codec *codec) { struct sigmatel_spec *spec; @@ -5329,26 +5434,17 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) snd_hda_codec_write_cache(codec, codec->afg, 0, 0x7ED, 0); codec->no_trigger_sense = 1; codec->spec = spec; + + stac92hd8x_fill_auto_spec(codec); + spec->linear_tone_beep = 0; codec->slave_dig_outs = stac92hd83xxx_slave_dig_outs; spec->digbeep_nid = 0x21; - spec->dmic_nids = stac92hd83xxx_dmic_nids; - spec->dmux_nids = stac92hd83xxx_mux_nids; - spec->mux_nids = stac92hd83xxx_mux_nids; - spec->num_muxes = ARRAY_SIZE(stac92hd83xxx_mux_nids); - spec->adc_nids = stac92hd83xxx_adc_nids; - spec->num_adcs = ARRAY_SIZE(stac92hd83xxx_adc_nids); spec->pwr_nids = stac92hd83xxx_pwr_nids; spec->pwr_mapping = stac92hd83xxx_pwr_mapping; spec->num_pwrs = ARRAY_SIZE(stac92hd83xxx_pwr_nids); spec->multiout.dac_nids = spec->dac_nids; - spec->init = stac92hd83xxx_core_init; - spec->num_pins = ARRAY_SIZE(stac92hd83xxx_pin_nids); - spec->pin_nids = stac92hd83xxx_pin_nids; - spec->num_caps = STAC92HD83XXX_NUM_CAPS; - spec->capvols = stac92hd83xxx_capvols; - spec->capsws = stac92hd83xxx_capsws; spec->board_config = snd_hda_check_board_config(codec, STAC_92HD83XXX_MODELS, @@ -5366,28 +5462,11 @@ again: case 0x111d76d1: case 0x111d76d9: case 0x111d76e5: - spec->dmic_nids = stac92hd87b_dmic_nids; - spec->num_dmics = stac92xx_connected_ports(codec, - stac92hd87b_dmic_nids, - STAC92HD87B_NUM_DMICS); - spec->num_pins = ARRAY_SIZE(stac92hd87xxx_pin_nids); - spec->pin_nids = stac92hd87xxx_pin_nids; - spec->mono_nid = 0; - spec->num_pwrs = 0; - break; case 0x111d7666: case 0x111d7667: case 0x111d7668: case 0x111d7669: case 0x111d76e3: - spec->num_dmics = stac92xx_connected_ports(codec, - stac92hd88xxx_dmic_nids, - STAC92HD88XXX_NUM_DMICS); - spec->num_pins = ARRAY_SIZE(stac92hd88xxx_pin_nids); - spec->pin_nids = stac92hd88xxx_pin_nids; - spec->mono_nid = 0; - spec->num_pwrs = 0; - break; case 0x111d7604: case 0x111d76d4: case 0x111d7605: @@ -5396,9 +5475,6 @@ again: if (spec->board_config == STAC_92HD83XXX_PWR_REF) break; spec->num_pwrs = 0; - spec->num_dmics = stac92xx_connected_ports(codec, - stac92hd83xxx_dmic_nids, - STAC92HD83XXX_NUM_DMICS); break; } -- cgit v1.2.3-70-g09d2 From cc90fd725e14020607c5a6ba3ea02a0ddec5655f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 14 Mar 2011 15:53:15 +0100 Subject: ALSA: hda - Remove an unused variable in patch_realtek.c Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 11a1380b882..639d2746713 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -14056,7 +14056,6 @@ static int patch_alc268(struct hda_codec *codec) 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; spec->capsrc_nids = alc268_capsrc_nids; /* get type */ -- cgit v1.2.3-70-g09d2