From 254bba6a7e28c77d8b32d9eaeba02d4943ee844e Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Einar=20R=C3=BCnkaru?= Date: Wed, 16 Dec 2009 22:16:13 +0200 Subject: ALSA: hda - Fixed internal mic initialization for Dell Vostro 1015 MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Fixed initialization of internal mic and added internal mic boost control Renamed analog mic boost control to ext mic boost contol. Name pair analog/digital seems too confusing for a normal user. Signed-off-by: Einar Rünkaru Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 39 +++++++++++++++++++++++++++++++++------ 1 file changed, 33 insertions(+), 6 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index a09c03c3f62b..ca9ad9fddbf2 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -111,6 +111,7 @@ struct conexant_spec { unsigned int dell_automute; unsigned int port_d_mode; unsigned char ext_mic_bias; + unsigned int dell_vostro; }; static int conexant_playback_pcm_open(struct hda_pcm_stream *hinfo, @@ -2109,9 +2110,12 @@ static int cxt5066_mic_boost_mux_enum_get(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); int val; + hda_nid_t nid = kcontrol->private_value & 0xff; + int inout = (kcontrol->private_value & 0x100) ? + AC_AMP_GET_INPUT : AC_AMP_GET_OUTPUT; - val = snd_hda_codec_read(codec, 0x17, 0, - AC_VERB_GET_AMP_GAIN_MUTE, AC_AMP_GET_OUTPUT); + val = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_AMP_GAIN_MUTE, inout); ucontrol->value.enumerated.item[0] = val & AC_AMP_GAIN; return 0; @@ -2123,6 +2127,9 @@ static int cxt5066_mic_boost_mux_enum_put(struct snd_kcontrol *kcontrol, struct hda_codec *codec = snd_kcontrol_chip(kcontrol); const struct hda_input_mux *imux = &cxt5066_analog_mic_boost; unsigned int idx; + hda_nid_t nid = kcontrol->private_value & 0xff; + int inout = (kcontrol->private_value & 0x100) ? + AC_AMP_SET_INPUT : AC_AMP_SET_OUTPUT; if (!imux->num_items) return 0; @@ -2130,9 +2137,9 @@ static int cxt5066_mic_boost_mux_enum_put(struct snd_kcontrol *kcontrol, if (idx >= imux->num_items) idx = imux->num_items - 1; - snd_hda_codec_write_cache(codec, 0x17, 0, + snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, - AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT | AC_AMP_SET_OUTPUT | + AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT | inout | imux->items[idx].index); return 1; @@ -2201,10 +2208,11 @@ static struct snd_kcontrol_new cxt5066_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Analog Mic Boost Capture Enum", + .name = "Ext Mic Boost Capture Enum", .info = cxt5066_mic_boost_mux_enum_info, .get = cxt5066_mic_boost_mux_enum_get, .put = cxt5066_mic_boost_mux_enum_put, + .private_value = 0x17, }, HDA_BIND_VOL("Capture Volume", &cxt5066_bind_capture_vol_others), @@ -2212,6 +2220,18 @@ static struct snd_kcontrol_new cxt5066_mixers[] = { {} }; +static struct snd_kcontrol_new cxt5066_vostro_mixers[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Int Mic Boost Capture Enum", + .info = cxt5066_mic_boost_mux_enum_info, + .get = cxt5066_mic_boost_mux_enum_get, + .put = cxt5066_mic_boost_mux_enum_put, + .private_value = 0x23 | 0x100, + }, + {} +}; + static struct hda_verb cxt5066_init_verbs[] = { {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Port B */ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Port C */ @@ -2397,11 +2417,16 @@ static struct hda_verb cxt5066_init_verbs_portd_lo[] = { /* initialize jack-sensing, too */ static int cxt5066_init(struct hda_codec *codec) { + struct conexant_spec *spec = codec->spec; + snd_printdd("CXT5066: init\n"); conexant_init(codec); if (codec->patch_ops.unsol_event) { cxt5066_hp_automute(codec); - cxt5066_automic(codec); + if (spec->dell_vostro) + cxt5066_vostro_automic(codec); + else + cxt5066_automic(codec); } return 0; } @@ -2500,7 +2525,9 @@ static int patch_cxt5066(struct hda_codec *codec) spec->init_verbs[0] = cxt5066_init_verbs_vostro; spec->mixers[spec->num_mixers++] = cxt5066_mixer_master_olpc; spec->mixers[spec->num_mixers++] = cxt5066_mixers; + spec->mixers[spec->num_mixers++] = cxt5066_vostro_mixers; spec->port_d_mode = 0; + spec->dell_vostro = 1; /* no S/PDIF out */ spec->multiout.dig_out_nid = 0; -- cgit v1.2.2 From c0f8faf0c7cd497ec7c1d61e1e9424f4384c1f68 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Einar=20R=C3=BCnkaru?= Date: Wed, 16 Dec 2009 22:41:36 +0200 Subject: ALSA: hda - Make use of beep device found in Dell Vostro 1015n MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Conexant CX20583-10Z has digital beep device with volume control. Making use of them. Signed-off-by: Einar Rünkaru Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index ca9ad9fddbf2..c578c28f368e 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -29,6 +29,7 @@ #include "hda_codec.h" #include "hda_local.h" +#include "hda_beep.h" #define CXT_PIN_DIR_IN 0x00 #define CXT_PIN_DIR_OUT 0x01 @@ -477,6 +478,7 @@ static void conexant_free(struct hda_codec *codec) snd_array_free(&spec->jacks); } #endif + snd_hda_detach_beep_device(codec); kfree(codec->spec); } @@ -2229,6 +2231,7 @@ static struct snd_kcontrol_new cxt5066_vostro_mixers[] = { .put = cxt5066_mic_boost_mux_enum_put, .private_value = 0x23 | 0x100, }, + HDA_CODEC_VOLUME_MONO("Beep Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT), {} }; @@ -2528,6 +2531,7 @@ static int patch_cxt5066(struct hda_codec *codec) spec->mixers[spec->num_mixers++] = cxt5066_vostro_mixers; spec->port_d_mode = 0; spec->dell_vostro = 1; + snd_hda_attach_beep_device(codec, 0x13); /* no S/PDIF out */ spec->multiout.dig_out_nid = 0; -- cgit v1.2.2 From 035eb0cff0671ada49ba9f3e5c9e7b0cb950efea Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 17 Dec 2009 15:00:26 +0100 Subject: ALSA: hda - Fix missing capsrc_nids for ALC88x Some model quirks missed the corresponding capsrc_nids. This resulted in non-working capture source selection. Signed-off-by: Takashi Iwai Cc: --- sound/pci/hda/patch_realtek.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 287bb6019df9..d9a9f0c7cf5b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9238,6 +9238,8 @@ static struct alc_config_preset alc882_presets[] = { .dac_nids = alc883_dac_nids, .num_adc_nids = ARRAY_SIZE(alc889_adc_nids), .adc_nids = alc889_adc_nids, + .capsrc_nids = alc889_capsrc_nids, + .capsrc_nids = alc889_capsrc_nids, .dig_out_nid = ALC883_DIGOUT_NID, .dig_in_nid = ALC883_DIGIN_NID, .slave_dig_outs = alc883_slave_dig_outs, @@ -9284,6 +9286,7 @@ static struct alc_config_preset alc882_presets[] = { .dac_nids = alc883_dac_nids, .adc_nids = alc883_adc_nids_alt, .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_alt), + .capsrc_nids = alc883_capsrc_nids, .dig_out_nid = ALC883_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), .channel_mode = alc883_3ST_2ch_modes, @@ -9430,6 +9433,7 @@ static struct alc_config_preset alc882_presets[] = { .dac_nids = alc883_dac_nids, .adc_nids = alc883_adc_nids_alt, .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_alt), + .capsrc_nids = alc883_capsrc_nids, .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes), .channel_mode = alc883_sixstack_modes, .input_mux = &alc883_capture_source, @@ -9491,6 +9495,7 @@ static struct alc_config_preset alc882_presets[] = { .dac_nids = alc883_dac_nids, .adc_nids = alc883_adc_nids_alt, .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_alt), + .capsrc_nids = alc883_capsrc_nids, .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_lenovo_101e_capture_source, @@ -9670,6 +9675,7 @@ static struct alc_config_preset alc882_presets[] = { alc880_gpio1_init_verbs }, .adc_nids = alc883_adc_nids, .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), + .capsrc_nids = alc883_capsrc_nids, .dac_nids = alc883_dac_nids, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .channel_mode = alc889A_mb31_6ch_modes, -- cgit v1.2.2 From 2fef62c825f09e29d2f52dc187ddf6f99e28c7f1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 18 Dec 2009 08:48:42 +0100 Subject: ALSA: hda - Fix quirk for Maxdata obook4-1 Works fine with the auto-parser. Reference: Novell bnc#564940 https://bugzilla.novell.com/show_bug.cgi?id=564940 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 d9a9f0c7cf5b..84bc2c7c4421 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8921,7 +8921,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x1462, 0x040d, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x1462, 0x0579, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x1462, 0x28fb, "Targa T8", ALC882_TARGA), /* MSI-1049 T8 */ - SND_PCI_QUIRK(0x1462, 0x2fb3, "MSI", ALC883_TARGA_2ch_DIG), + SND_PCI_QUIRK(0x1462, 0x2fb3, "MSI", ALC882_AUTO), SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC882_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x3729, "MSI S420", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x3783, "NEC S970", ALC883_TARGA_DIG), -- cgit v1.2.2 From 440b004cf953bec2bc8cd91c64ae707fd7e25327 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Sun, 20 Dec 2009 12:04:08 +0100 Subject: ALSA: hda/realtek: Remove extra .capsrc_nids initialization for ALC889_INTEL Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_realtek.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8b375771b3ab..2d3f4f893ef3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9238,8 +9238,6 @@ static struct alc_config_preset alc882_presets[] = { .dac_nids = alc883_dac_nids, .num_adc_nids = ARRAY_SIZE(alc889_adc_nids), .adc_nids = alc889_adc_nids, - .capsrc_nids = alc889_capsrc_nids, - .capsrc_nids = alc889_capsrc_nids, .dig_out_nid = ALC883_DIGOUT_NID, .dig_in_nid = ALC883_DIGIN_NID, .slave_dig_outs = alc883_slave_dig_outs, -- cgit v1.2.2 From e2595322a3a353a59cecd7f57e7aa421ecb02d12 Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Sat, 19 Dec 2009 18:19:02 -0500 Subject: ALSA: hda: Set Front Mic to input vref 50% for Lenovo 3000 Y410 BugLink: https://bugs.launchpad.net/bugs/479373 The OR has verified with hda-verb that the internal microphone needs VREF50 set for audible capture. Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 84bc2c7c4421..1554c3a6fd2e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10686,6 +10686,13 @@ static struct hda_verb alc262_lenovo_3000_unsol_verbs[] = { {} }; +static struct hda_verb alc262_lenovo_3000_init_verbs[] = { + /* Front Mic pin: input vref at 50% */ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {} +}; + static struct hda_input_mux alc262_fujitsu_capture_source = { .num_items = 3, .items = { @@ -11728,7 +11735,8 @@ static struct alc_config_preset alc262_presets[] = { [ALC262_LENOVO_3000] = { .mixers = { alc262_lenovo_3000_mixer }, .init_verbs = { alc262_init_verbs, alc262_EAPD_verbs, - alc262_lenovo_3000_unsol_verbs }, + alc262_lenovo_3000_unsol_verbs, + alc262_lenovo_3000_init_verbs }, .num_dacs = ARRAY_SIZE(alc262_dac_nids), .dac_nids = alc262_dac_nids, .hp_nid = 0x03, -- cgit v1.2.2 From 0f86a228f4a4639b3142ce0dad208433b2db377a Mon Sep 17 00:00:00 2001 From: Hector Martin Date: Sun, 20 Dec 2009 22:51:18 +0100 Subject: ALSA: HDA: simplify Aspire 8930G verb array This patch just simplifies the 8930G verb array a bit. Just use the common ALC889 EAPD verb array to make things more consistent. The file is already huge enough already. Signed-off-by: Hector Martin Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1554c3a6fd2e..cb97323acc17 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1665,9 +1665,6 @@ static struct hda_verb alc889_acer_aspire_8930g_verbs[] = { /* some bit here disables the other DACs. Init=0x4900 */ {0x20, AC_VERB_SET_COEF_INDEX, 0x08}, {0x20, AC_VERB_SET_PROC_COEF, 0x0000}, -/* Enable amplifiers */ - {0x14, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, - {0x15, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, /* DMIC fix * This laptop has a stereo digital microphone. The mics are only 1cm apart * which makes the stereo useless. However, either the mic or the ALC889 @@ -9386,7 +9383,8 @@ static struct alc_config_preset alc882_presets[] = { .mixers = { alc888_base_mixer, alc883_chmode_mixer }, .init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs, - alc889_acer_aspire_8930g_verbs }, + alc889_acer_aspire_8930g_verbs, + alc889_eapd_verbs}, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .num_adc_nids = ARRAY_SIZE(alc889_adc_nids), -- cgit v1.2.2 From 556eea9a926bff8f014b4f80522b4de97ae84213 Mon Sep 17 00:00:00 2001 From: Hector Martin Date: Sun, 20 Dec 2009 22:51:23 +0100 Subject: ALSA: HDA: remove useless mixers on Aspire 8930G This patch removes some extra mixers that do nothing on the Acer Aspire 8930G. The CD mixer is useless because the SATA DVD/Blu-Ray drive has no analog audio output, and the Side mixer is useless because we max out at 6ch anyway. Signed-off-by: Hector Martin Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 21 ++++++++++++++++++++- 1 file changed, 20 insertions(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index cb97323acc17..faeb74f28207 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1777,6 +1777,25 @@ static struct snd_kcontrol_new alc888_base_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), + 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", 0x0e, 1, 0x0, + HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + { } /* end */ +}; + + static void alc888_acer_aspire_4930g_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -9380,7 +9399,7 @@ static struct alc_config_preset alc882_presets[] = { .init_hook = alc_automute_amp, }, [ALC888_ACER_ASPIRE_8930G] = { - .mixers = { alc888_base_mixer, + .mixers = { alc889_acer_aspire_8930g_mixer, alc883_chmode_mixer }, .init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs, alc889_acer_aspire_8930g_verbs, -- cgit v1.2.2 From f5de24b06aa46427500d0fdbe8616b73a71d8c28 Mon Sep 17 00:00:00 2001 From: Hector Martin Date: Sun, 20 Dec 2009 22:51:31 +0100 Subject: ALSA: HDA: add powersaving hook for Realtek The current Realtek code makes no specific provision for turning stuff off. The codec chip is placed into low-power mode generically, but this doesn't turn off any external hardware connected to it, in particular external amplifiers. This patch creates a hook function that is called by the codec suspend/resume functions. It ought to disable any external hardware in a device-specific way. I've implemented a generic ALC889 function that sets the EAPD pin properly, and used it for the Acer Aspire 8930G which can benefit from this feature. On my laptop, this results in ~0.5W extra savings. Signed-off-by: Hector Martin Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 36 ++++++++++++++++++++++++++++++++++++ 1 file changed, 36 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index faeb74f28207..b3abe9ca826d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -337,6 +337,9 @@ struct alc_spec { /* hooks */ void (*init_hook)(struct hda_codec *codec); void (*unsol_event)(struct hda_codec *codec, unsigned int res); +#ifdef CONFIG_SND_HDA_POWER_SAVE + void (*power_hook)(struct hda_codec *codec, int power); +#endif /* for pin sensing */ unsigned int sense_updated: 1; @@ -388,6 +391,7 @@ struct alc_config_preset { void (*init_hook)(struct hda_codec *); #ifdef CONFIG_SND_HDA_POWER_SAVE struct hda_amp_list *loopbacks; + void (*power_hook)(struct hda_codec *codec, int power); #endif }; @@ -900,6 +904,7 @@ static void setup_preset(struct hda_codec *codec, spec->unsol_event = preset->unsol_event; spec->init_hook = preset->init_hook; #ifdef CONFIG_SND_HDA_POWER_SAVE + spec->power_hook = preset->power_hook; spec->loopback.amplist = preset->loopbacks; #endif @@ -1826,6 +1831,16 @@ static void alc889_acer_aspire_8930g_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[2] = 0x1b; } +#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); +} +#endif + /* * ALC880 3-stack model * @@ -3619,12 +3634,29 @@ static void alc_free(struct hda_codec *codec) snd_hda_detach_beep_device(codec); } +#ifdef CONFIG_SND_HDA_POWER_SAVE +static int alc_suspend(struct hda_codec *codec, pm_message_t state) +{ + struct alc_spec *spec = codec->spec; + if (spec && spec->power_hook) + spec->power_hook(codec, 0); + return 0; +} +#endif + #ifdef SND_HDA_NEEDS_RESUME static int alc_resume(struct hda_codec *codec) { +#ifdef CONFIG_SND_HDA_POWER_SAVE + struct alc_spec *spec = codec->spec; +#endif codec->patch_ops.init(codec); snd_hda_codec_resume_amp(codec); snd_hda_codec_resume_cache(codec); +#ifdef CONFIG_SND_HDA_POWER_SAVE + if (spec && spec->power_hook) + spec->power_hook(codec, 1); +#endif return 0; } #endif @@ -3641,6 +3673,7 @@ static struct hda_codec_ops alc_patch_ops = { .resume = alc_resume, #endif #ifdef CONFIG_SND_HDA_POWER_SAVE + .suspend = alc_suspend, .check_power_status = alc_check_power_status, #endif }; @@ -9420,6 +9453,9 @@ static struct alc_config_preset alc882_presets[] = { .unsol_event = alc_automute_amp_unsol_event, .setup = alc889_acer_aspire_8930g_setup, .init_hook = alc_automute_amp, +#ifdef CONFIG_SND_HDA_POWER_SAVE + .power_hook = alc889_power_eapd, +#endif }, [ALC888_ACER_ASPIRE_7730G] = { .mixers = { alc883_3ST_6ch_mixer, -- cgit v1.2.2 From d8d881dd2c814e1500558889d800cf78d11cf898 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 22 Dec 2009 07:52:49 +0100 Subject: ALSA: hda - Fix NULL dereference with enable_beep=0 option Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 19 ++++++++++--------- 1 file changed, 10 insertions(+), 9 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 3d59f8325848..417fb22ae83c 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -3779,15 +3779,16 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out err = snd_hda_attach_beep_device(codec, nid); if (err < 0) return err; - /* IDT/STAC codecs have linear beep tone parameter */ - codec->beep->linear_tone = 1; - /* if no beep switch is available, make its own one */ - caps = query_amp_caps(codec, nid, HDA_OUTPUT); - if (codec->beep && - !((caps & AC_AMPCAP_MUTE) >> AC_AMPCAP_MUTE_SHIFT)) { - err = stac92xx_beep_switch_ctl(codec); - if (err < 0) - return err; + if (codec->beep) { + /* IDT/STAC codecs have linear beep tone parameter */ + codec->beep->linear_tone = 1; + /* if no beep switch is available, make its own one */ + caps = query_amp_caps(codec, nid, HDA_OUTPUT); + if (!(caps & AC_AMPCAP_MUTE)) { + err = stac92xx_beep_switch_ctl(codec); + if (err < 0) + return err; + } } } #endif -- cgit v1.2.2 From 1a5ba2e9fc7999b8de2a71c7e7b9f58d752c05e4 Mon Sep 17 00:00:00 2001 From: Rafael Avila de Espindola Date: Tue, 22 Dec 2009 07:59:37 +0100 Subject: ALSA: hda - Add support for the new 27 inch IMacs With the attached patch I am able to use the sound on a new IMac 27. What works: *) Internal speakers *) Internal microphone *) Headphone I don't have an external mic or a SPDIF device to test the rest. Signed-off-by: Rafael Avila de Espindola Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cirrus.c | 22 +++++++++++++++++++++- 1 file changed, 21 insertions(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 4b200da1bd18..fe0423c39598 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -66,6 +66,7 @@ struct cs_spec { /* available models */ enum { CS420X_MBP55, + CS420X_IMAC27, CS420X_AUTO, CS420X_MODELS }; @@ -827,7 +828,8 @@ static void cs_automute(struct hda_codec *codec) AC_VERB_SET_PIN_WIDGET_CONTROL, hp_present ? 0 : PIN_OUT); } - if (spec->board_config == CS420X_MBP55) { + if (spec->board_config == CS420X_MBP55 || + spec->board_config == CS420X_IMAC27) { unsigned int gpio = hp_present ? 0x02 : 0x08; snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, gpio); @@ -1069,12 +1071,14 @@ static int cs_parse_auto_config(struct hda_codec *codec) static const char *cs420x_models[CS420X_MODELS] = { [CS420X_MBP55] = "mbp55", + [CS420X_IMAC27] = "imac27", [CS420X_AUTO] = "auto", }; static struct snd_pci_quirk cs420x_cfg_tbl[] = { SND_PCI_QUIRK(0x10de, 0xcb79, "MacBookPro 5,5", CS420X_MBP55), + SND_PCI_QUIRK(0x8086, 0x7270, "IMac 27 Inch", CS420X_IMAC27), {} /* terminator */ }; @@ -1097,8 +1101,23 @@ static struct cs_pincfg mbp55_pincfgs[] = { {} /* terminator */ }; +static struct cs_pincfg imac27_pincfgs[] = { + { 0x09, 0x012b4050 }, + { 0x0a, 0x90100140 }, + { 0x0b, 0x90100142 }, + { 0x0c, 0x018b3020 }, + { 0x0d, 0x90a00110 }, + { 0x0e, 0x400000f0 }, + { 0x0f, 0x01cbe030 }, + { 0x10, 0x014be060 }, + { 0x12, 0x01ab9070 }, + { 0x15, 0x400000f0 }, + {} /* terminator */ +}; + static struct cs_pincfg *cs_pincfgs[CS420X_MODELS] = { [CS420X_MBP55] = mbp55_pincfgs, + [CS420X_IMAC27] = imac27_pincfgs, }; static void fix_pincfg(struct hda_codec *codec, int model) @@ -1128,6 +1147,7 @@ static int patch_cs420x(struct hda_codec *codec) fix_pincfg(codec, spec->board_config); switch (spec->board_config) { + case CS420X_IMAC27: case CS420X_MBP55: /* GPIO1 = headphones */ /* GPIO3 = speakers */ -- cgit v1.2.2 From 9dc8398bab52931435fce403ce2eaf5822f28e58 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 22 Dec 2009 08:15:01 +0100 Subject: ALSA: hda - Add MSI blacklist A machine with AMD CPU with Nvidia board doesn't work with MSI. Reported-by: Robert J. King 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 9b56f937913e..ff8ad46cc50e 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2322,6 +2322,7 @@ static void __devinit check_probe_mask(struct azx *chip, int dev) * white/black-list for enable_msi */ static struct snd_pci_quirk msi_black_list[] __devinitdata = { + SND_PCI_QUIRK(0x1043, 0x81f2, "ASUS", 0), /* Athlon64 X2 + nvidia */ {} }; -- cgit v1.2.2