From 8f398ae72fc7e03356fc1ee6b54beef79ba6be6a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 23 Jul 2011 18:57:11 +0200 Subject: ALSA: hda - Fix DAC filling for multi-connection pins in Realtek parser Fix a regression in the DAC filling code in patch_realtek.c. The already filled DACs in multiout.dac_nids[] were ignored because of num_dacs=0, thus always pointed to the first DAC. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound/pci/hda/patch_realtek.c') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 52ce07534e5b..569d2aa4eeb5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2801,7 +2801,8 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec) int i; again: - spec->multiout.num_dacs = 0; + /* set num_dacs once to full for alc_auto_look_for_dac() */ + spec->multiout.num_dacs = cfg->line_outs; spec->multiout.hp_nid = 0; spec->multiout.extra_out_nid[0] = 0; memset(spec->private_dac_nids, 0, sizeof(spec->private_dac_nids)); @@ -2834,6 +2835,8 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec) } } + /* re-count num_dacs and squash invalid entries */ + spec->multiout.num_dacs = 0; for (i = 0; i < cfg->line_outs; i++) { if (spec->private_dac_nids[i]) spec->multiout.num_dacs++; -- cgit v1.2.3 From 2a43952a99072f43c92355882b7965c8762ae3f3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 26 Jul 2011 09:52:50 +0200 Subject: ALSA: hda - Make CONFIG_SND_HDA_POWER_SAVE depending on CONFIG_PM It makes little sense to enable power-saving without PM. This removes SND_HDA_NEEDS_RESUME define so that we can use CONFIG_PM in all places. Signed-off-by: Takashi Iwai --- sound/pci/hda/Kconfig | 1 + sound/pci/hda/hda_codec.c | 16 ++++++++-------- sound/pci/hda/hda_codec.h | 12 +++--------- sound/pci/hda/hda_local.h | 2 +- sound/pci/hda/patch_analog.c | 4 ++-- sound/pci/hda/patch_realtek.c | 10 +++++----- sound/pci/hda/patch_sigmatel.c | 6 +++--- sound/pci/hda/patch_via.c | 4 ++-- 8 files changed, 25 insertions(+), 30 deletions(-) (limited to 'sound/pci/hda/patch_realtek.c') diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 7489b4608551..bb7e102d6726 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -243,6 +243,7 @@ config SND_HDA_GENERIC config SND_HDA_POWER_SAVE bool "Aggressive power-saving on HD-audio" + depends on PM help Say Y here to enable more aggressive power-saving mode on HD-audio driver. The power-saving timeout can be configured diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index c0e83ed0b351..27b0c78abb5b 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1101,7 +1101,7 @@ void snd_hda_shutup_pins(struct hda_codec *codec) } EXPORT_SYMBOL_HDA(snd_hda_shutup_pins); -#ifdef SND_HDA_NEEDS_RESUME +#ifdef CONFIG_PM /* Restore the pin controls cleared previously via snd_hda_shutup_pins() */ static void restore_shutup_pins(struct hda_codec *codec) { @@ -1499,7 +1499,7 @@ static void purify_inactive_streams(struct hda_codec *codec) } } -#ifdef SND_HDA_NEEDS_RESUME +#ifdef CONFIG_PM /* clean up all streams; called from suspend */ static void hda_cleanup_all_streams(struct hda_codec *codec) { @@ -1838,7 +1838,7 @@ int snd_hda_codec_amp_stereo(struct hda_codec *codec, hda_nid_t nid, } EXPORT_SYMBOL_HDA(snd_hda_codec_amp_stereo); -#ifdef SND_HDA_NEEDS_RESUME +#ifdef CONFIG_PM /** * snd_hda_codec_resume_amp - Resume all AMP commands from the cache * @codec: HD-audio codec @@ -1868,7 +1868,7 @@ void snd_hda_codec_resume_amp(struct hda_codec *codec) } } EXPORT_SYMBOL_HDA(snd_hda_codec_resume_amp); -#endif /* SND_HDA_NEEDS_RESUME */ +#endif /* CONFIG_PM */ static u32 get_amp_max_value(struct hda_codec *codec, hda_nid_t nid, int dir, unsigned int ofs) @@ -3082,7 +3082,7 @@ int snd_hda_create_spdif_in_ctls(struct hda_codec *codec, hda_nid_t nid) } EXPORT_SYMBOL_HDA(snd_hda_create_spdif_in_ctls); -#ifdef SND_HDA_NEEDS_RESUME +#ifdef CONFIG_PM /* * command cache */ @@ -3199,7 +3199,7 @@ void snd_hda_sequence_write_cache(struct hda_codec *codec, seq->param); } EXPORT_SYMBOL_HDA(snd_hda_sequence_write_cache); -#endif /* SND_HDA_NEEDS_RESUME */ +#endif /* CONFIG_PM */ /* * set power state of the codec @@ -3274,7 +3274,7 @@ static void hda_exec_init_verbs(struct hda_codec *codec) static inline void hda_exec_init_verbs(struct hda_codec *codec) {} #endif -#ifdef SND_HDA_NEEDS_RESUME +#ifdef CONFIG_PM /* * call suspend and power-down; used both from PM and power-save */ @@ -3315,7 +3315,7 @@ static void hda_call_codec_resume(struct hda_codec *codec) snd_hda_codec_resume_cache(codec); } } -#endif /* SND_HDA_NEEDS_RESUME */ +#endif /* CONFIG_PM */ /** diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 82161466d3b0..663aa4fc384a 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -26,10 +26,6 @@ #include #include -#if defined(CONFIG_PM) || defined(CONFIG_SND_HDA_POWER_SAVE) -#define SND_HDA_NEEDS_RESUME /* resume control code is required */ -#endif - /* * nodes */ @@ -704,17 +700,15 @@ struct hda_codec_ops { int (*init)(struct hda_codec *codec); void (*free)(struct hda_codec *codec); void (*unsol_event)(struct hda_codec *codec, unsigned int res); -#ifdef SND_HDA_NEEDS_RESUME +#ifdef CONFIG_PM int (*suspend)(struct hda_codec *codec, pm_message_t state); + int (*pre_resume)(struct hda_codec *codec); int (*resume)(struct hda_codec *codec); #endif #ifdef CONFIG_SND_HDA_POWER_SAVE int (*check_power_status)(struct hda_codec *codec, hda_nid_t nid); #endif void (*reboot_notify)(struct hda_codec *codec); -#ifdef SND_HDA_NEEDS_RESUME - int (*pre_resume)(struct hda_codec *codec); -#endif }; /* record for amp information cache */ @@ -930,7 +924,7 @@ void snd_hda_sequence_write(struct hda_codec *codec, int snd_hda_queue_unsol_event(struct hda_bus *bus, u32 res, u32 res_ex); /* cached write */ -#ifdef SND_HDA_NEEDS_RESUME +#ifdef CONFIG_PM int snd_hda_codec_write_cache(struct hda_codec *codec, hda_nid_t nid, int direct, unsigned int verb, unsigned int parm); void snd_hda_sequence_write_cache(struct hda_codec *codec, diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 88b277e97409..2e7ac31afa8d 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -131,7 +131,7 @@ int snd_hda_codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch, int direction, int idx, int mask, int val); int snd_hda_codec_amp_stereo(struct hda_codec *codec, hda_nid_t nid, int dir, int idx, int mask, int val); -#ifdef SND_HDA_NEEDS_RESUME +#ifdef CONFIG_PM void snd_hda_codec_resume_amp(struct hda_codec *codec); #endif diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 1362c8ba4d1f..8648917acffb 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -563,7 +563,7 @@ static void ad198x_free(struct hda_codec *codec) snd_hda_detach_beep_device(codec); } -#ifdef SND_HDA_NEEDS_RESUME +#ifdef CONFIG_PM static int ad198x_suspend(struct hda_codec *codec, pm_message_t state) { ad198x_shutup(codec); @@ -579,7 +579,7 @@ static const struct hda_codec_ops ad198x_patch_ops = { #ifdef CONFIG_SND_HDA_POWER_SAVE .check_power_status = ad198x_check_power_status, #endif -#ifdef SND_HDA_NEEDS_RESUME +#ifdef CONFIG_PM .suspend = ad198x_suspend, #endif .reboot_notify = ad198x_shutup, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 569d2aa4eeb5..694327ae8b71 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2386,7 +2386,7 @@ static int alc_suspend(struct hda_codec *codec, pm_message_t state) } #endif -#ifdef SND_HDA_NEEDS_RESUME +#ifdef CONFIG_PM static int alc_resume(struct hda_codec *codec) { msleep(150); /* to avoid pop noise */ @@ -2406,7 +2406,7 @@ static const struct hda_codec_ops alc_patch_ops = { .init = alc_init, .free = alc_free, .unsol_event = alc_unsol_event, -#ifdef SND_HDA_NEEDS_RESUME +#ifdef CONFIG_PM .resume = alc_resume, #endif #ifdef CONFIG_SND_HDA_POWER_SAVE @@ -4413,7 +4413,7 @@ static void alc269_shutup(struct hda_codec *codec) } } -#ifdef SND_HDA_NEEDS_RESUME +#ifdef CONFIG_PM static int alc269_resume(struct hda_codec *codec) { if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x018) { @@ -4436,7 +4436,7 @@ static int alc269_resume(struct hda_codec *codec) hda_call_check_power_status(codec, 0x01); return 0; } -#endif /* SND_HDA_NEEDS_RESUME */ +#endif /* CONFIG_PM */ static void alc269_fixup_hweq(struct hda_codec *codec, const struct alc_fixup *fix, int action) @@ -4728,7 +4728,7 @@ static int patch_alc269(struct hda_codec *codec) spec->vmaster_nid = 0x02; codec->patch_ops = alc_patch_ops; -#ifdef SND_HDA_NEEDS_RESUME +#ifdef CONFIG_PM codec->patch_ops.resume = alc269_resume; #endif if (board_config == ALC_MODEL_AUTO) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 8f80796c366f..fcf4c7142103 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4934,7 +4934,7 @@ static void stac927x_proc_hook(struct snd_info_buffer *buffer, #define stac927x_proc_hook NULL #endif -#ifdef SND_HDA_NEEDS_RESUME +#ifdef CONFIG_PM static int stac92xx_pre_resume(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; @@ -5030,7 +5030,7 @@ static int stac92xx_suspend(struct hda_codec *codec, pm_message_t state) stac92xx_shutup(codec); return 0; } -#endif +#endif /* CONFIG_PM */ static const struct hda_codec_ops stac92xx_patch_ops = { .build_controls = stac92xx_build_controls, @@ -5038,7 +5038,7 @@ static const struct hda_codec_ops stac92xx_patch_ops = { .init = stac92xx_init, .free = stac92xx_free, .unsol_event = stac92xx_unsol_event, -#ifdef SND_HDA_NEEDS_RESUME +#ifdef CONFIG_PM .suspend = stac92xx_suspend, .resume = stac92xx_resume, .pre_resume = stac92xx_pre_resume, diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index f38160b00e16..84d8798bf33a 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1708,7 +1708,7 @@ static void via_unsol_event(struct hda_codec *codec, via_gpio_control(codec); } -#ifdef SND_HDA_NEEDS_RESUME +#ifdef CONFIG_PM static int via_suspend(struct hda_codec *codec, pm_message_t state) { struct via_spec *spec = codec->spec; @@ -1736,7 +1736,7 @@ static const struct hda_codec_ops via_patch_ops = { .init = via_init, .free = via_free, .unsol_event = via_unsol_event, -#ifdef SND_HDA_NEEDS_RESUME +#ifdef CONFIG_PM .suspend = via_suspend, #endif #ifdef CONFIG_SND_HDA_POWER_SAVE -- cgit v1.2.3 From 60a6a8425a84fa46a3831ce79197640b8224311b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 27 Jul 2011 14:01:24 +0200 Subject: ALSA: hda - Fix Oops with Realtek quirks with NULL adc_nids Somce quirk models don't set adc_nids but let the parser filling it. But the recent code has unnecessary NULL-checks of spec->input_mux, and it resulted in NULL dereferences. This patch fixes that regression. Reported-and-tested-by: Oliver Neukum Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) (limited to 'sound/pci/hda/patch_realtek.c') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 694327ae8b71..0383dd8a11df 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1911,7 +1911,7 @@ static int alc_build_controls(struct hda_codec *codec) return err; } } - if (spec->cap_mixer) { + if (spec->cap_mixer && spec->adc_nids) { const char *kname = kctl ? kctl->id.name : NULL; for (knew = spec->cap_mixer; knew->name; knew++) { if (kname && strcmp(knew->name, kname) == 0) @@ -3677,7 +3677,7 @@ static int patch_alc880(struct hda_codec *codec) if (board_config != ALC_MODEL_AUTO) setup_preset(codec, &alc880_presets[board_config]); - if (!spec->no_analog && !spec->adc_nids && spec->input_mux) { + if (!spec->no_analog && !spec->adc_nids) { alc_auto_fill_adc_caps(codec); alc_rebuild_imux_for_auto_mic(codec); alc_remove_invalid_adc_nids(codec); @@ -3804,7 +3804,7 @@ static int patch_alc260(struct hda_codec *codec) if (board_config != ALC_MODEL_AUTO) setup_preset(codec, &alc260_presets[board_config]); - if (!spec->no_analog && !spec->adc_nids && spec->input_mux) { + if (!spec->no_analog && !spec->adc_nids) { alc_auto_fill_adc_caps(codec); alc_rebuild_imux_for_auto_mic(codec); alc_remove_invalid_adc_nids(codec); @@ -3983,7 +3983,7 @@ static int patch_alc882(struct hda_codec *codec) if (board_config != ALC_MODEL_AUTO) setup_preset(codec, &alc882_presets[board_config]); - if (!spec->no_analog && !spec->adc_nids && spec->input_mux) { + if (!spec->no_analog && !spec->adc_nids) { alc_auto_fill_adc_caps(codec); alc_rebuild_imux_for_auto_mic(codec); alc_remove_invalid_adc_nids(codec); @@ -4137,7 +4137,7 @@ static int patch_alc262(struct hda_codec *codec) if (board_config != ALC_MODEL_AUTO) setup_preset(codec, &alc262_presets[board_config]); - if (!spec->no_analog && !spec->adc_nids && spec->input_mux) { + if (!spec->no_analog && !spec->adc_nids) { alc_auto_fill_adc_caps(codec); alc_rebuild_imux_for_auto_mic(codec); alc_remove_invalid_adc_nids(codec); @@ -4293,7 +4293,7 @@ static int patch_alc268(struct hda_codec *codec) (0 << AC_AMPCAP_MUTE_SHIFT)); } - if (!spec->no_analog && !spec->adc_nids && spec->input_mux) { + if (!spec->no_analog && !spec->adc_nids) { alc_auto_fill_adc_caps(codec); alc_rebuild_imux_for_auto_mic(codec); alc_remove_invalid_adc_nids(codec); @@ -4705,7 +4705,7 @@ static int patch_alc269(struct hda_codec *codec) if (board_config != ALC_MODEL_AUTO) setup_preset(codec, &alc269_presets[board_config]); - if (!spec->no_analog && !spec->adc_nids && spec->input_mux) { + if (!spec->no_analog && !spec->adc_nids) { alc_auto_fill_adc_caps(codec); alc_rebuild_imux_for_auto_mic(codec); alc_remove_invalid_adc_nids(codec); @@ -4843,7 +4843,7 @@ static int patch_alc861(struct hda_codec *codec) if (board_config != ALC_MODEL_AUTO) setup_preset(codec, &alc861_presets[board_config]); - if (!spec->no_analog && !spec->adc_nids && spec->input_mux) { + if (!spec->no_analog && !spec->adc_nids) { alc_auto_fill_adc_caps(codec); alc_rebuild_imux_for_auto_mic(codec); alc_remove_invalid_adc_nids(codec); @@ -4984,7 +4984,7 @@ static int patch_alc861vd(struct hda_codec *codec) add_verb(spec, alc660vd_eapd_verbs); } - if (!spec->no_analog && !spec->adc_nids && spec->input_mux) { + if (!spec->no_analog && !spec->adc_nids) { alc_auto_fill_adc_caps(codec); alc_rebuild_imux_for_auto_mic(codec); alc_remove_invalid_adc_nids(codec); @@ -5200,7 +5200,7 @@ static int patch_alc662(struct hda_codec *codec) if (board_config != ALC_MODEL_AUTO) setup_preset(codec, &alc662_presets[board_config]); - if (!spec->no_analog && !spec->adc_nids && spec->input_mux) { + if (!spec->no_analog && !spec->adc_nids) { alc_auto_fill_adc_caps(codec); alc_rebuild_imux_for_auto_mic(codec); alc_remove_invalid_adc_nids(codec); @@ -5336,7 +5336,7 @@ static int patch_alc680(struct hda_codec *codec) #endif } - if (!spec->no_analog && !spec->adc_nids && spec->input_mux) { + if (!spec->no_analog && !spec->adc_nids) { alc_auto_fill_adc_caps(codec); alc_rebuild_imux_for_auto_mic(codec); alc_remove_invalid_adc_nids(codec); -- cgit v1.2.3 From c48a8fb0d31d6147d8d76b8e2ad7f51a2fbb5c4d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 27 Jul 2011 16:41:57 +0200 Subject: ALSA: hda - Fix duplicated DAC assignments for Realtek Copying hp_pins and speaker_pins from line_out_pins may confuse the parser, and it can lead to duplicated initializations for the same pin with a wrong DAC assignment. The problem appears in 3.0 kernel code. Cc: (for 3.0) Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) (limited to 'sound/pci/hda/patch_realtek.c') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0383dd8a11df..e125c60fe352 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -895,13 +895,15 @@ static void alc_init_auto_hp(struct hda_codec *codec) if (present == 3) spec->automute_hp_lo = 1; /* both HP and LO automute */ - if (!cfg->speaker_pins[0]) { + if (!cfg->speaker_pins[0] && + cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) { memcpy(cfg->speaker_pins, cfg->line_out_pins, sizeof(cfg->speaker_pins)); cfg->speaker_outs = cfg->line_outs; } - if (!cfg->hp_pins[0]) { + if (!cfg->hp_pins[0] && + cfg->line_out_type == AUTO_PIN_HP_OUT) { memcpy(cfg->hp_pins, cfg->line_out_pins, sizeof(cfg->hp_pins)); cfg->hp_outs = cfg->line_outs; @@ -920,6 +922,7 @@ static void alc_init_auto_hp(struct hda_codec *codec) spec->automute_mode = ALC_AUTOMUTE_PIN; } if (spec->automute && cfg->line_out_pins[0] && + cfg->speaker_pins[0] && cfg->line_out_pins[0] != cfg->hp_pins[0] && cfg->line_out_pins[0] != cfg->speaker_pins[0]) { for (i = 0; i < cfg->line_outs; i++) { -- cgit v1.2.3 From adabb3ec8b0bcbd2ca81973d33c3da726b939c7c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 3 Aug 2011 07:48:37 +0200 Subject: ALSA: hda - Fix digital-mic mono recording on ASUS Eee PC The digital-mic unit on ASUS Eee PC gives PDM signals instead of the normal stereo PCM, thus you can't record a mono stream from the stereo stream as is; the summed stereo signal results in almost zero level, and you'll hear only soft noise. As a workaround, use ALC269-specific COEF to manipulate the dmic route for mono, like used for ALC271x. This is implemented as a fix-up, thus it works only with model=auto or without REALTEK_QUIRKS Kconfig. Reported-and-tested-by: Pavel Roskin Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 26 ++++++++++++++++++++++++++ 1 file changed, 26 insertions(+) (limited to 'sound/pci/hda/patch_realtek.c') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e125c60fe352..9a1aa09f47fe 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4484,6 +4484,22 @@ static void alc269_fixup_pcm_44k(struct hda_codec *codec, spec->stream_analog_capture = &alc269_44k_pcm_analog_capture; } +static void alc269_fixup_stereo_dmic(struct hda_codec *codec, + const struct alc_fixup *fix, int action) +{ + int coef; + + if (action != ALC_FIXUP_ACT_INIT) + return; + /* The digital-mic unit sends PDM (differential signal) instead of + * the standard PCM, thus you can't record a valid mono stream as is. + * Below is a workaround specific to ALC269 to control the dmic + * signal source as mono. + */ + coef = alc_read_coef_idx(codec, 0x07); + alc_write_coef_idx(codec, 0x07, coef | 0x80); +} + enum { ALC269_FIXUP_SONY_VAIO, ALC275_FIXUP_SONY_VAIO_GPIO2, @@ -4494,6 +4510,7 @@ enum { ALC275_FIXUP_SONY_HWEQ, ALC271_FIXUP_DMIC, ALC269_FIXUP_PCM_44K, + ALC269_FIXUP_STEREO_DMIC, }; static const struct alc_fixup alc269_fixups[] = { @@ -4556,10 +4573,19 @@ static const struct alc_fixup alc269_fixups[] = { .type = ALC_FIXUP_FUNC, .v.func = alc269_fixup_pcm_44k, }, + [ALC269_FIXUP_STEREO_DMIC] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc269_fixup_stereo_dmic, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), + SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), + SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), + SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC), + SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC), + SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x104d, 0x9073, "Sony VAIO", ALC275_FIXUP_SONY_VAIO_GPIO2), SND_PCI_QUIRK(0x104d, 0x907b, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), -- cgit v1.2.3 From 3fe45aeaf2033c9eaa5028ed5ba68b466008876f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 18 Aug 2011 15:13:17 +0200 Subject: ALSA: hda - Add "PCM" volume to vmaster slave list The new parser may use "PCM" volume, but it was missing the vmaster slave list, thus "Master" volume didn't control it. Reference: https://bugzilla.kernel.org/show_bug.cgi?id=41342 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/pci/hda/patch_realtek.c') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9a1aa09f47fe..fcb11af9ad24 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1784,6 +1784,7 @@ static const char * const alc_slave_vols[] = { "Speaker Playback Volume", "Mono Playback Volume", "Line-Out Playback Volume", + "PCM Playback Volume", NULL, }; @@ -1798,6 +1799,7 @@ static const char * const alc_slave_sws[] = { "Mono Playback Switch", "IEC958 Playback Switch", "Line-Out Playback Switch", + "PCM Playback Switch", NULL, }; -- cgit v1.2.3 From 675c1aa3c4a7290e537e854d0af7cdf9692bd396 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 23 Aug 2011 12:36:28 +0200 Subject: ALSA: hda - Fix output-path initialization for Realtek auto-parser When the headphone or speaker output has no own DAC, initialize the path using the primary DAC. Otherwise the path won't be set properly and can result in the silence. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 20 +++++++++++++------- 1 file changed, 13 insertions(+), 7 deletions(-) (limited to 'sound/pci/hda/patch_realtek.c') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fcb11af9ad24..0fefc1088d11 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3083,16 +3083,22 @@ static void alc_auto_init_multi_out(struct hda_codec *codec) static void alc_auto_init_extra_out(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - hda_nid_t pin; + hda_nid_t pin, dac; pin = spec->autocfg.hp_pins[0]; - if (pin) - alc_auto_set_output_and_unmute(codec, pin, PIN_HP, - spec->multiout.hp_nid); + if (pin) { + dac = spec->multiout.hp_nid; + if (!dac) + dac = spec->multiout.dac_nids[0]; + alc_auto_set_output_and_unmute(codec, pin, PIN_HP, dac); + } pin = spec->autocfg.speaker_pins[0]; - if (pin) - alc_auto_set_output_and_unmute(codec, pin, PIN_OUT, - spec->multiout.extra_out_nid[0]); + if (pin) { + dac = spec->multiout.extra_out_nid[0]; + if (!dac) + dac = spec->multiout.dac_nids[0]; + alc_auto_set_output_and_unmute(codec, pin, PIN_OUT, dac); + } } /* -- cgit v1.2.3 From 3c715a98844f72cec0fa3ef2b68232b8f751468b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 23 Aug 2011 12:41:09 +0200 Subject: ALSA: hda - Update jack-sense info even when no automute is set The internal states, jack_present and line_jack_present should be updated upon unsolicited events even if no automute is set. Otherwise the wrong state is referred when the automute behavior is changed by the mixer control. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound/pci/hda/patch_realtek.c') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0fefc1088d11..7cabd7317163 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -565,11 +565,11 @@ static void alc_hp_automute(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - if (!spec->automute) - return; spec->jack_present = detect_jacks(codec, ARRAY_SIZE(spec->autocfg.hp_pins), spec->autocfg.hp_pins); + if (!spec->automute) + return; update_speakers(codec); } @@ -578,11 +578,11 @@ static void alc_line_automute(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - if (!spec->automute || !spec->detect_line) - return; spec->line_jack_present = detect_jacks(codec, ARRAY_SIZE(spec->autocfg.line_out_pins), spec->autocfg.line_out_pins); + if (!spec->automute || !spec->detect_line) + return; update_speakers(codec); } -- cgit v1.2.3