From 9c4cc0bdede1c39bde60a0d5d9251aac71fbe719 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 15 Mar 2010 09:07:52 +0100 Subject: ALSA: hda - Fix secondary ADC of ALC260 basic model Fix adc_nids[] for ALC260 basic model to match with num_adc_nids. Otherwise you get an invalid NID in the secondary "Input Source" mixer element. Signed-off-by: Takashi Iwai Cc: --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 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 3a8371990d75..ba45868d5242 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6477,7 +6477,7 @@ static struct alc_config_preset alc260_presets[] = { .num_dacs = ARRAY_SIZE(alc260_dac_nids), .dac_nids = alc260_dac_nids, .num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids), - .adc_nids = alc260_adc_nids, + .adc_nids = alc260_dual_adc_nids, .num_channel_mode = ARRAY_SIZE(alc260_modes), .channel_mode = alc260_modes, .input_mux = &alc260_capture_source, -- cgit v1.2.3 From b43f6e5e258d67acae5961896d10bbe38c271070 Mon Sep 17 00:00:00 2001 From: Anisse Astier Date: Wed, 10 Mar 2010 19:17:46 +0100 Subject: ALSA: hda - Add PCI quirks for MSI NetOn AP1900 and Wind Top AE2220 This should make the speakers and jack detection work on MSI all-in-one computers NetOn AP1900 and Wind Top AE2220. Signed-off-by: Anisse Astier Cc: 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 ba45868d5242..07637c4aa46f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9195,6 +9195,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x1462, 0x4314, "MSI", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x4319, "MSI", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x4324, "MSI", ALC883_TARGA_DIG), + SND_PCI_QUIRK(0x1462, 0x4570, "MSI Wind Top AE2220", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x6510, "MSI GX620", ALC883_TARGA_8ch_DIG), SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7187, "MSI", ALC883_6ST_DIG), @@ -9204,6 +9205,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x1462, 0x7280, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7327, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7350, "MSI", ALC883_6ST_DIG), + SND_PCI_QUIRK(0x1462, 0x7437, "MSI NetOn AP1900", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0xa422, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x1462, 0xaa08, "MSI", ALC883_TARGA_2ch_DIG), -- cgit v1.2.3 From 572c0e3c73341755f3e7dfaaef6b26df12bd709c Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Sun, 14 Mar 2010 23:44:03 -0400 Subject: ALSA: hda: Use LPIB and 6stack-dig for eMachines T5212 BugLink: https://bugs.launchpad.net/bugs/538895 The OR has verified that both position_fix=1 and model=6stack-dig are necessary to have capture function properly. (The existing 3stack-6ch model quirk seems to be incorrect.) Reported-by: Reuben Bailey Tested-by: Reuben Bailey Cc: Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 1 + sound/pci/hda/patch_realtek.c | 2 +- 2 files changed, 2 insertions(+), 1 deletion(-) (limited to 'sound/pci/hda/patch_realtek.c') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 027d3f4c1c59..1766ad2926d6 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2271,6 +2271,7 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB), SND_PCI_QUIRK(0x1462, 0x1002, "MSI Wind U115", POS_FIX_LPIB), SND_PCI_QUIRK(0x1565, 0x820f, "Biostar Microtech", POS_FIX_LPIB), + SND_PCI_QUIRK(0x8086, 0xd601, "eMachines T5212", POS_FIX_LPIB), {} }; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 07637c4aa46f..4ec57633af88 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9237,7 +9237,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x8086, 0x0022, "DX58SO", ALC889_INTEL), SND_PCI_QUIRK(0x8086, 0x0021, "Intel IbexPeak", ALC889A_INTEL), SND_PCI_QUIRK(0x8086, 0x3b56, "Intel IbexPeak", ALC889A_INTEL), - SND_PCI_QUIRK(0x8086, 0xd601, "D102GGC", ALC883_3ST_6ch), + SND_PCI_QUIRK(0x8086, 0xd601, "D102GGC", ALC882_6ST_DIG), {} }; -- cgit v1.2.3 From bae84e70d66fe46c12231082cf1c4848ea22f3ef Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 22 Mar 2010 08:30:20 +0100 Subject: ALSA: hda - Fix access-after-free in patch_realtek.c alc_free_kctls() has to be called after all jobs done in alc_build_controls(). Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 5 +++-- 1 file changed, 3 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 4ec57633af88..053d53d8c8b2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2532,8 +2532,6 @@ static int alc_build_controls(struct hda_codec *codec) return err; } - alc_free_kctls(codec); /* no longer needed */ - /* assign Capture Source enums to NID */ kctl = snd_hda_find_mixer_ctl(codec, "Capture Source"); if (!kctl) @@ -2602,6 +2600,9 @@ static int alc_build_controls(struct hda_codec *codec) } } } + + alc_free_kctls(codec); /* no longer needed */ + return 0; } -- cgit v1.2.3 From 6a4f2ccb467e00281470cde2dee08fe5ecde62d1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 25 Mar 2010 15:00:15 +0100 Subject: ALSA: hda - Don't set invalid connection index in Realtek initialiaiton Skip initialization of connections of DAC widgets that aren't used, which resulted in invalid verb parameters. 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 053d53d8c8b2..9a23444e9e7a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10043,8 +10043,11 @@ static void alc882_auto_set_output_and_unmute(struct hda_codec *codec, alc_set_pin_output(codec, nid, pin_type); if (spec->multiout.dac_nids[dac_idx] == 0x25) idx = 4; - else + else { + if (spec->multiout.num_dacs >= dac_idx) + return; idx = spec->multiout.dac_nids[dac_idx] - 2; + } snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, idx); } -- cgit v1.2.3 From 5dbd5ec6e1cf2e49128025d80813a275744a7ac5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 29 Mar 2010 09:16:24 +0200 Subject: ALSA: hda - Fix invalid bit values passed to snd_hda_codec_amp_stereo() The mask and value parameters passed to snd_hda_codec_amp_stereo() should be 8-bit values for mute and volume. Passing AMP_IN_MUTE() is wrong, which is found in many places in patch_realtek.c as a left-over from the conversion to snd_hda_codec_amp_stereo(). Reported-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 52 +++++++++++++++++++++---------------------- 1 file changed, 26 insertions(+), 26 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 9a23444e9e7a..bc55c1e96df5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12459,11 +12459,11 @@ static void alc268_aspire_one_speaker_automute(struct hda_codec *codec) unsigned char bits; present = snd_hda_jack_detect(codec, 0x15); - bits = present ? AMP_IN_MUTE(0) : 0; + bits = present ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x0f, HDA_INPUT, 0, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); snd_hda_codec_amp_stereo(codec, 0x0f, HDA_INPUT, 1, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); } static void alc268_acer_lc_unsol_event(struct hda_codec *codec, @@ -13482,11 +13482,11 @@ static void alc269_quanta_fl1_speaker_automute(struct hda_codec *codec) unsigned char bits; present = snd_hda_jack_detect(codec, 0x15); - bits = present ? AMP_IN_MUTE(0) : 0; + bits = present ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 0x0c); @@ -13511,11 +13511,11 @@ static void alc269_lifebook_speaker_automute(struct hda_codec *codec) /* Check port replicator headphone socket */ present |= snd_hda_jack_detect(codec, 0x1a); - bits = present ? AMP_IN_MUTE(0) : 0; + bits = present ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 0x0c); @@ -13646,11 +13646,11 @@ static void alc269_speaker_automute(struct hda_codec *codec) unsigned char bits; present = snd_hda_jack_detect(codec, nid); - bits = present ? AMP_IN_MUTE(0) : 0; + bits = present ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); } /* unsolicited event for HP jack sensing */ @@ -17115,9 +17115,9 @@ static void alc663_m51va_speaker_automute(struct hda_codec *codec) present = snd_hda_jack_detect(codec, 0x21); bits = present ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); } static void alc663_21jd_two_speaker_automute(struct hda_codec *codec) @@ -17128,13 +17128,13 @@ static void alc663_21jd_two_speaker_automute(struct hda_codec *codec) present = snd_hda_jack_detect(codec, 0x21); bits = present ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 0, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 1, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); } static void alc663_15jd_two_speaker_automute(struct hda_codec *codec) @@ -17145,13 +17145,13 @@ static void alc663_15jd_two_speaker_automute(struct hda_codec *codec) present = snd_hda_jack_detect(codec, 0x15); bits = present ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 0, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 1, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); } static void alc662_f5z_speaker_automute(struct hda_codec *codec) @@ -17190,14 +17190,14 @@ static void alc663_two_hp_m2_speaker_automute(struct hda_codec *codec) if (present1 || present2) { snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, - AMP_IN_MUTE(0), AMP_IN_MUTE(0)); + HDA_AMP_MUTE, HDA_AMP_MUTE); snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, - AMP_IN_MUTE(0), AMP_IN_MUTE(0)); + HDA_AMP_MUTE, HDA_AMP_MUTE); } else { snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, - AMP_IN_MUTE(0), 0); + HDA_AMP_MUTE, 0); snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, - AMP_IN_MUTE(0), 0); + HDA_AMP_MUTE, 0); } } -- cgit v1.2.3 From 6694635d3ae1b038d7a0e38b80637db867c7c8e2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 29 Mar 2010 17:21:45 +0200 Subject: ALSA: hda - Fix ADC/MUX assignment of ALC269 codec ALC269 codec has a few different variants, and each of them may have different ADC and MUX widgets. For example, one model has ADC 0x08 with MUX 0x23 while others has ADC 0x09 or ADC 0x07 with MUX 022 or 0x24. The difference of ADC appears usually as the capability of the digital mic pin (0x12), and the current driver sometimes misses the internal mic pin due to the mismatching ADC. This patch adds a bit more clever way to find the matching ADC instead of the static list. Now the driver checks all active input pins and fills only the ADC/MUX's that contain all of them. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 95 ++++++++++++++++++++++++++++++++++++------- 1 file changed, 80 insertions(+), 15 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 bc55c1e96df5..22aea7b089c6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4984,6 +4984,69 @@ static void set_capture_mixer(struct hda_codec *codec) } } +/* fill adc_nids (and capsrc_nids) containing all active input pins */ +static void fillup_priv_adc_nids(struct hda_codec *codec, hda_nid_t *nids, + int num_nids) +{ + struct alc_spec *spec = codec->spec; + int n; + hda_nid_t fallback_adc = 0, fallback_cap = 0; + + for (n = 0; n < num_nids; n++) { + hda_nid_t adc, cap; + hda_nid_t conn[HDA_MAX_NUM_INPUTS]; + int nconns, i, j; + + adc = nids[n]; + if (get_wcaps_type(get_wcaps(codec, adc)) != AC_WID_AUD_IN) + continue; + cap = adc; + nconns = snd_hda_get_connections(codec, cap, conn, + ARRAY_SIZE(conn)); + if (nconns == 1) { + cap = conn[0]; + nconns = snd_hda_get_connections(codec, cap, conn, + ARRAY_SIZE(conn)); + } + if (nconns <= 0) + continue; + if (!fallback_adc) { + fallback_adc = adc; + fallback_cap = cap; + } + for (i = 0; i < AUTO_PIN_LAST; i++) { + hda_nid_t nid = spec->autocfg.input_pins[i]; + if (!nid) + continue; + for (j = 0; j < nconns; j++) { + if (conn[j] == nid) + break; + } + if (j >= nconns) + break; + } + if (i >= AUTO_PIN_LAST) { + int num_adcs = spec->num_adc_nids; + spec->private_adc_nids[num_adcs] = adc; + spec->private_capsrc_nids[num_adcs] = cap; + spec->num_adc_nids++; + spec->adc_nids = spec->private_adc_nids; + if (adc != cap) + spec->capsrc_nids = spec->private_capsrc_nids; + } + } + if (!spec->num_adc_nids) { + printk(KERN_WARNING "hda_codec: %s: no valid ADC found;" + " using fallback 0x%x\n", fallback_adc); + spec->private_adc_nids[0] = fallback_adc; + spec->adc_nids = spec->private_adc_nids; + if (fallback_adc != fallback_cap) { + spec->private_capsrc_nids[0] = fallback_cap; + spec->capsrc_nids = spec->private_adc_nids; + } + } +} + #ifdef CONFIG_SND_HDA_INPUT_BEEP #define set_beep_amp(spec, nid, idx, dir) \ ((spec)->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 3, idx, dir)) @@ -13333,9 +13396,9 @@ static hda_nid_t alc269vb_capsrc_nids[1] = { 0x22, }; -/* NOTE: ADC2 (0x07) is connected from a recording *MIXER* (0x24), - * not a mux! - */ +static hda_nid_t alc269_adc_candidates[] = { + 0x08, 0x09, 0x07, +}; #define alc269_modes alc260_modes #define alc269_capture_source alc880_lg_lw_capture_source @@ -13842,7 +13905,6 @@ static int alc269_parse_auto_config(struct hda_codec *codec) struct alc_spec *spec = codec->spec; int err; static hda_nid_t alc269_ignore[] = { 0x1d, 0 }; - hda_nid_t real_capsrc_nids; err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, alc269_ignore); @@ -13866,18 +13928,19 @@ static int alc269_parse_auto_config(struct hda_codec *codec) if ((alc_read_coef_idx(codec, 0) & 0x00f0) == 0x0010) { add_verb(spec, alc269vb_init_verbs); - real_capsrc_nids = alc269vb_capsrc_nids[0]; alc_ssid_check(codec, 0, 0x1b, 0x14, 0x21); } else { add_verb(spec, alc269_init_verbs); - real_capsrc_nids = alc269_capsrc_nids[0]; alc_ssid_check(codec, 0x15, 0x1b, 0x14, 0); } spec->num_mux_defs = 1; spec->input_mux = &spec->private_imux[0]; + fillup_priv_adc_nids(codec, alc269_adc_candidates, + sizeof(alc269_adc_candidates)); + /* set default input source */ - snd_hda_codec_write_cache(codec, real_capsrc_nids, + snd_hda_codec_write_cache(codec, spec->capsrc_nids[0], 0, AC_VERB_SET_CONNECT_SEL, spec->input_mux->items[0].index); @@ -14156,14 +14219,16 @@ static int patch_alc269(struct hda_codec *codec) spec->stream_digital_playback = &alc269_pcm_digital_playback; spec->stream_digital_capture = &alc269_pcm_digital_capture; - if (!is_alc269vb) { - spec->adc_nids = alc269_adc_nids; - spec->num_adc_nids = ARRAY_SIZE(alc269_adc_nids); - spec->capsrc_nids = alc269_capsrc_nids; - } else { - spec->adc_nids = alc269vb_adc_nids; - spec->num_adc_nids = ARRAY_SIZE(alc269vb_adc_nids); - spec->capsrc_nids = alc269vb_capsrc_nids; + if (!spec->adc_nids) { /* wasn't filled automatically? use default */ + if (!is_alc269vb) { + spec->adc_nids = alc269_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(alc269_adc_nids); + spec->capsrc_nids = alc269_capsrc_nids; + } else { + spec->adc_nids = alc269vb_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(alc269vb_adc_nids); + spec->capsrc_nids = alc269vb_capsrc_nids; + } } if (!spec->cap_mixer) -- cgit v1.2.3 From 1f85d72d2c9c9a1d6d32cf325936bc224ad5d591 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 30 Mar 2010 07:48:05 +0200 Subject: ALSA: hda - Add missing printk argument in previous patch Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 3 ++- 1 file changed, 2 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 22aea7b089c6..ca93c4cc144e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5037,7 +5037,8 @@ static void fillup_priv_adc_nids(struct hda_codec *codec, hda_nid_t *nids, } if (!spec->num_adc_nids) { printk(KERN_WARNING "hda_codec: %s: no valid ADC found;" - " using fallback 0x%x\n", fallback_adc); + " using fallback 0x%x\n", + codec->chip_name, fallback_adc); spec->private_adc_nids[0] = fallback_adc; spec->adc_nids = spec->private_adc_nids; if (fallback_adc != fallback_cap) { -- cgit v1.2.3 From d12841827a6de120199609dadb6ff4ec99bd90ea Mon Sep 17 00:00:00 2001 From: Tony Vroon Date: Mon, 5 Apr 2010 16:30:43 +0100 Subject: ALSA: hda - Enable amplifiers on Acer Inspire 6530G After more tests it appears that EAPD needs to be enabled on both the 0x14 and 0x15 NIDs to enable the main speaker and headphone amplifiers. The maximum volume setting is now equal to what the machine achieves under other operating systems. Disabling Front or LFE playback triggers EAPD and disables the amplifier. As such, these two playback switches have been removed from the mixer. Signed-off-by: Tony Vroon Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 9 +++++++-- 1 file changed, 7 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 ca93c4cc144e..547206296d7b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1621,6 +1621,11 @@ static struct hda_verb alc888_acer_aspire_4930g_verbs[] = { */ static struct hda_verb alc888_acer_aspire_6530g_verbs[] = { +/* Route to built-in subwoofer as well as speakers */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, /* Bias voltage on for external mic port */ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN | PIN_VREF80}, /* Front Mic: set to PIN_IN (empty by default) */ @@ -1632,10 +1637,12 @@ static struct hda_verb alc888_acer_aspire_6530g_verbs[] = { /* Enable speaker output */ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2}, /* Enable headphone output */ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | PIN_HP}, {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2}, { } }; @@ -8462,9 +8469,7 @@ static struct snd_kcontrol_new alc883_acer_aspire_mixer[] = { static struct snd_kcontrol_new alc888_acer_aspire_6530_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("LFE Playback Volume", 0x0f, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("LFE Playback Switch", 0x0f, 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("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), -- cgit v1.2.3 From f9700d5a4575e7fb343df10a1d29d425e4b81082 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 5 Apr 2010 23:25:13 +0200 Subject: ALSA: hda - Fix a wrong array range check in patch_realtek.c The commit 6a4f2ccb467e00281470cde2dee08fe5ecde62d1 introduced a wrong comparision for the array range check, which effectively skips the whole initialization of DAC connections. Fixed now. Reference: bko#15689 https://bugzilla.kernel.org/show_bug.cgi?id=15689 Reported-by: Adrian Ulrich Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 7 +++---- 1 file changed, 3 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 547206296d7b..c7730dbb9ddb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10110,13 +10110,12 @@ static void alc882_auto_set_output_and_unmute(struct hda_codec *codec, int idx; alc_set_pin_output(codec, nid, pin_type); + if (dac_idx >= spec->multiout.num_dacs) + return; if (spec->multiout.dac_nids[dac_idx] == 0x25) idx = 4; - else { - if (spec->multiout.num_dacs >= dac_idx) - return; + else idx = spec->multiout.dac_nids[dac_idx] - 2; - } snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, idx); } -- cgit v1.2.3 From 7ad7b218f4aae4f395b3b4cef261572556bbd20a Mon Sep 17 00:00:00 2001 From: Maurus Cuelenaere Date: Tue, 6 Apr 2010 18:12:52 +0200 Subject: ALSA: hda: Add support for Medion WIM2160 This adds support for the Medion WIM2160 soundcard. There's no PCI quirk added because it has the same PCI id as the Medion MD2. Signed-off-by: Maurus Cuelenaere Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 53 +++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 53 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 c7730dbb9ddb..2971e48e50ad 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -230,6 +230,7 @@ enum { ALC888_ACER_ASPIRE_7730G, ALC883_MEDION, ALC883_MEDION_MD2, + ALC883_MEDION_WIM2160, ALC883_LAPTOP_EAPD, ALC883_LENOVO_101E_2ch, ALC883_LENOVO_NB0763, @@ -8455,6 +8456,42 @@ static struct snd_kcontrol_new alc883_medion_md2_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc883_medion_wim2160_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_MUTE("Speaker Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x08, 0x0, HDA_INPUT), + { } /* end */ +}; + +static struct hda_verb alc883_medion_wim2160_verbs[] = { + /* Unmute front mixer */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + + /* Set speaker pin to front mixer */ + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, + + /* Init headphone pin */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + + { } /* end */ +}; + +/* toggle speaker-output according to the hp-jack state */ +static void alc883_medion_wim2160_setup(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x1a; + spec->autocfg.speaker_pins[0] = 0x15; +} + static struct snd_kcontrol_new alc883_acer_aspire_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -9164,6 +9201,7 @@ static const char *alc882_models[ALC882_MODEL_LAST] = { [ALC888_ACER_ASPIRE_7730G] = "acer-aspire-7730g", [ALC883_MEDION] = "medion", [ALC883_MEDION_MD2] = "medion-md2", + [ALC883_MEDION_WIM2160] = "medion-wim2160", [ALC883_LAPTOP_EAPD] = "laptop-eapd", [ALC883_LENOVO_101E_2ch] = "lenovo-101e", [ALC883_LENOVO_NB0763] = "lenovo-nb0763", @@ -9818,6 +9856,21 @@ static struct alc_config_preset alc882_presets[] = { .setup = alc883_medion_md2_setup, .init_hook = alc_automute_amp, }, + [ALC883_MEDION_WIM2160] = { + .mixers = { alc883_medion_wim2160_mixer }, + .init_verbs = { alc883_init_verbs, alc883_medion_wim2160_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .dig_out_nid = ALC883_DIGOUT_NID, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), + .adc_nids = alc883_adc_nids, + .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), + .channel_mode = alc883_3ST_2ch_modes, + .input_mux = &alc883_capture_source, + .unsol_event = alc_automute_amp_unsol_event, + .setup = alc883_medion_wim2160_setup, + .init_hook = alc_automute_amp, + }, [ALC883_LAPTOP_EAPD] = { .mixers = { alc883_base_mixer }, .init_verbs = { alc883_init_verbs, alc882_eapd_verbs }, -- cgit v1.2.3 From 531d8791accf1464bc6854ff69d08dd866189d17 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Fri, 9 Apr 2010 10:57:33 +0200 Subject: ALSA: hda - Fix auto-parser of ALC269vb for HP pin NID 0x21 ALC269vb has an alternative HP pin 0x21 in addition. Fix the parser to recognize it. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (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 2971e48e50ad..fbbdfbc8a1ca 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12869,6 +12869,7 @@ static int alc268_new_analog_output(struct alc_spec *spec, hda_nid_t nid, dac = 0x02; break; case 0x15: + case 0x21: /* ALC269vb has this pin, too */ dac = 0x03; break; default: -- cgit v1.2.3 From 226b1ec8c18bcb6d1aa448a29b2c8aeae1946228 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Fri, 9 Apr 2010 11:01:20 +0200 Subject: ALSA: hda - Fix setup for ALC269vb amic and dmic models Corrected HP and mic pins for ALC269vb amic and dmic models. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 28 ++++++++++++++++++++-------- 1 file changed, 20 insertions(+), 8 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 fbbdfbc8a1ca..9b58f29833e6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -13789,19 +13789,19 @@ static void alc269_laptop_unsol_event(struct hda_codec *codec, } } -static void alc269_laptop_dmic_setup(struct hda_codec *codec) +static void alc269_laptop_amic_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; - spec->int_mic.pin = 0x12; - spec->int_mic.mux_idx = 5; + spec->int_mic.pin = 0x19; + spec->int_mic.mux_idx = 1; spec->auto_mic = 1; } -static void alc269vb_laptop_dmic_setup(struct hda_codec *codec) +static void alc269_laptop_dmic_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x15; @@ -13809,14 +13809,14 @@ static void alc269vb_laptop_dmic_setup(struct hda_codec *codec) spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; spec->int_mic.pin = 0x12; - spec->int_mic.mux_idx = 6; + spec->int_mic.mux_idx = 5; spec->auto_mic = 1; } -static void alc269_laptop_amic_setup(struct hda_codec *codec) +static void alc269vb_laptop_amic_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - spec->autocfg.hp_pins[0] = 0x15; + spec->autocfg.hp_pins[0] = 0x21; spec->autocfg.speaker_pins[0] = 0x14; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; @@ -13825,6 +13825,18 @@ static void alc269_laptop_amic_setup(struct hda_codec *codec) spec->auto_mic = 1; } +static void alc269vb_laptop_dmic_setup(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + spec->autocfg.hp_pins[0] = 0x21; + spec->autocfg.speaker_pins[0] = 0x14; + spec->ext_mic.pin = 0x18; + spec->ext_mic.mux_idx = 0; + spec->int_mic.pin = 0x12; + spec->int_mic.mux_idx = 6; + spec->auto_mic = 1; +} + static void alc269_laptop_inithook(struct hda_codec *codec) { alc269_speaker_automute(codec); @@ -14162,7 +14174,7 @@ static struct alc_config_preset alc269_presets[] = { .num_channel_mode = ARRAY_SIZE(alc269_modes), .channel_mode = alc269_modes, .unsol_event = alc269_laptop_unsol_event, - .setup = alc269_laptop_amic_setup, + .setup = alc269vb_laptop_amic_setup, .init_hook = alc269_laptop_inithook, }, [ALC269VB_DMIC] = { -- cgit v1.2.3 From 7f311a46916a3be00a1a8e3f1bdf461d08f1d263 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 9 Apr 2010 17:32:23 +0200 Subject: ALSA: hda - Fix initial capture source connections of ALC880/260 The widget connections of ADC of ALC880 and ALC2260 aren't initialized, thus it might point to invalid pin. This can be a problem when mode=auto and there is only one input pin. Then user can't change the connection at all. This patch adds the code to initialize the input pin connection of these codecs. Reference: Novell bnc#594363 https://bugzilla.novell.com/show_bug.cgi?id=594363 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 23 +++++++++++++++++++++++ 1 file changed, 23 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 9b58f29833e6..8d60b1f25ce1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4809,6 +4809,25 @@ static void alc880_auto_init_analog_input(struct hda_codec *codec) } } +static void alc880_auto_init_input_src(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + int c; + + for (c = 0; c < spec->num_adc_nids; c++) { + unsigned int mux_idx; + const struct hda_input_mux *imux; + mux_idx = c >= spec->num_mux_defs ? 0 : c; + imux = &spec->input_mux[mux_idx]; + if (!imux->num_items && mux_idx > 0) + imux = &spec->input_mux[0]; + if (imux) + snd_hda_codec_write(codec, spec->adc_nids[c], 0, + AC_VERB_SET_CONNECT_SEL, + imux->items[0].index); + } +} + /* parse the BIOS configuration and set up the alc_spec */ /* return 1 if successful, 0 if the proper config is not found, * or a negative error code @@ -4887,6 +4906,7 @@ static void alc880_auto_init(struct hda_codec *codec) alc880_auto_init_multi_out(codec); alc880_auto_init_extra_out(codec); alc880_auto_init_analog_input(codec); + alc880_auto_init_input_src(codec); if (spec->unsol_event) alc_inithook(codec); } @@ -6398,6 +6418,8 @@ static void alc260_auto_init_analog_input(struct hda_codec *codec) } } +#define alc260_auto_init_input_src alc880_auto_init_input_src + /* * generic initialization of ADC, input mixers and output mixers */ @@ -6484,6 +6506,7 @@ static void alc260_auto_init(struct hda_codec *codec) struct alc_spec *spec = codec->spec; alc260_auto_init_multi_out(codec); alc260_auto_init_analog_input(codec); + alc260_auto_init_input_src(codec); if (spec->unsol_event) alc_inithook(codec); } -- cgit v1.2.3 From 7fa90e873f520dad5ec58f47340996cda083e875 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 12 Apr 2010 08:49:00 +0200 Subject: ALSA: hda - Enhance fix-up table for Realtek codecs A few enhancement / fixes for fix-up table of some Realtek codecs: - Apply fix-ups only for the auto model - Apply additional verbs after normal init verbs - Add a debug print to show the fix-up application This is basically a preliminary work for the next fix for Sony VAIO. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 35 ++++++++++++++++++++++++++++------- 1 file changed, 28 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 8d60b1f25ce1..cff57710d1fb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1390,22 +1390,31 @@ struct alc_fixup { static void alc_pick_fixup(struct hda_codec *codec, const struct snd_pci_quirk *quirk, - const struct alc_fixup *fix) + const struct alc_fixup *fix, + int pre_init) { const struct alc_pincfg *cfg; quirk = snd_pci_quirk_lookup(codec->bus->pci, quirk); if (!quirk) return; - fix += quirk->value; cfg = fix->pins; - if (cfg) { + if (pre_init && cfg) { +#ifdef CONFIG_SND_DEBUG_VERBOSE + snd_printdd(KERN_INFO "hda_codec: %s: Apply pincfg for %s\n", + codec->chip_name, quirk->name); +#endif for (; cfg->nid; cfg++) snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val); } - if (fix->verbs) + if (!pre_init && fix->verbs) { +#ifdef CONFIG_SND_DEBUG_VERBOSE + snd_printdd(KERN_INFO "hda_codec: %s: Apply fix-verbs for %s\n", + codec->chip_name, quirk->name); +#endif add_verb(codec->spec, fix->verbs); + } } static int alc_read_coef_idx(struct hda_codec *codec, @@ -10439,7 +10448,8 @@ static int patch_alc882(struct hda_codec *codec) board_config = ALC882_AUTO; } - alc_pick_fixup(codec, alc882_fixup_tbl, alc882_fixups); + if (board_config == ALC882_AUTO) + alc_pick_fixup(codec, alc882_fixup_tbl, alc882_fixups, 1); if (board_config == ALC882_AUTO) { /* automatic parse from the BIOS config */ @@ -10512,6 +10522,9 @@ static int patch_alc882(struct hda_codec *codec) set_capture_mixer(codec); set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); + if (board_config == ALC882_AUTO) + alc_pick_fixup(codec, alc882_fixup_tbl, alc882_fixups, 0); + spec->vmaster_nid = 0x0c; codec->patch_ops = alc_patch_ops; @@ -15417,7 +15430,8 @@ static int patch_alc861(struct hda_codec *codec) board_config = ALC861_AUTO; } - alc_pick_fixup(codec, alc861_fixup_tbl, alc861_fixups); + if (board_config == ALC861_AUTO) + alc_pick_fixup(codec, alc861_fixup_tbl, alc861_fixups, 1); if (board_config == ALC861_AUTO) { /* automatic parse from the BIOS config */ @@ -15454,6 +15468,9 @@ static int patch_alc861(struct hda_codec *codec) spec->vmaster_nid = 0x03; + if (board_config == ALC861_AUTO) + alc_pick_fixup(codec, alc861_fixup_tbl, alc861_fixups, 0); + codec->patch_ops = alc_patch_ops; if (board_config == ALC861_AUTO) { spec->init_hook = alc861_auto_init; @@ -16388,7 +16405,8 @@ static int patch_alc861vd(struct hda_codec *codec) board_config = ALC861VD_AUTO; } - alc_pick_fixup(codec, alc861vd_fixup_tbl, alc861vd_fixups); + if (board_config == ALC861VD_AUTO) + alc_pick_fixup(codec, alc861vd_fixup_tbl, alc861vd_fixups, 1); if (board_config == ALC861VD_AUTO) { /* automatic parse from the BIOS config */ @@ -16436,6 +16454,9 @@ static int patch_alc861vd(struct hda_codec *codec) spec->vmaster_nid = 0x02; + if (board_config == ALC861VD_AUTO) + alc_pick_fixup(codec, alc861vd_fixup_tbl, alc861vd_fixups, 0); + codec->patch_ops = alc_patch_ops; if (board_config == ALC861VD_AUTO) -- cgit v1.2.3 From ff818c24c2af370153646d302d831b69b023816f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 12 Apr 2010 08:59:25 +0200 Subject: ALSA: hda - Add fix-up for Sony VAIO with ALC269 Sony VAIO models with ALC269 need to initialize the pin 0x19 to VREF ground or Hi-Z to make the headphone working. Other than that, model=auto works fine, so let's use model=auto with a specific fix-up table. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 29 ++++++++++++++++++++++++++++- 1 file changed, 28 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 cff57710d1fb..4b35176d3454 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -14077,6 +14077,27 @@ static void alc269_auto_init(struct hda_codec *codec) alc_inithook(codec); } +enum { + ALC269_FIXUP_SONY_VAIO, +}; + +const static struct hda_verb alc269_sony_vaio_fixup_verbs[] = { + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREFGRD}, + {} +}; + +static const struct alc_fixup alc269_fixups[] = { + [ALC269_FIXUP_SONY_VAIO] = { + .verbs = alc269_sony_vaio_fixup_verbs + }, +}; + +static struct snd_pci_quirk alc269_fixup_tbl[] = { + SND_PCI_QUIRK(0x104d, 0x9071, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), + {} +}; + + /* * configuration and preset */ @@ -14136,7 +14157,7 @@ static struct snd_pci_quirk alc269_cfg_tbl[] = { ALC269_DMIC), SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005HA", ALC269_DMIC), SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005HA", ALC269_DMIC), - SND_PCI_QUIRK(0x104d, 0x9071, "SONY XTB", ALC269_DMIC), + SND_PCI_QUIRK(0x104d, 0x9071, "Sony VAIO", ALC269_AUTO), SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook ICH9M-based", ALC269_LIFEBOOK), SND_PCI_QUIRK(0x152d, 0x1778, "Quanta ON1", ALC269_DMIC), SND_PCI_QUIRK(0x1734, 0x115d, "FSC Amilo", ALC269_FUJITSU), @@ -14290,6 +14311,9 @@ static int patch_alc269(struct hda_codec *codec) board_config = ALC269_AUTO; } + if (board_config == ALC269_AUTO) + alc_pick_fixup(codec, alc269_fixup_tbl, alc269_fixups, 1); + if (board_config == ALC269_AUTO) { /* automatic parse from the BIOS config */ err = alc269_parse_auto_config(codec); @@ -14342,6 +14366,9 @@ static int patch_alc269(struct hda_codec *codec) set_capture_mixer(codec); set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT); + if (board_config == ALC269_AUTO) + alc_pick_fixup(codec, alc269_fixup_tbl, alc269_fixups, 0); + spec->vmaster_nid = 0x02; codec->patch_ops = alc_patch_ops; -- cgit v1.2.3 From d1501ea844eefdf925f6b711875b4b2b928fddf8 Mon Sep 17 00:00:00 2001 From: Joerg Schirottke Date: Thu, 15 Apr 2010 08:37:41 +0200 Subject: ALSA: hda - add a quirk for Clevo M570U laptop Added the matching model for Clevo laptop M570U. Signed-off-by: Joerg Schirottke Tested-by: Maximilian Gerhard Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (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 4b35176d3454..aad1627f56f1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9350,6 +9350,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x1462, 0xaa08, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x147b, 0x1083, "Abit IP35-PRO", ALC883_6ST_DIG), + SND_PCI_QUIRK(0x1558, 0x0571, "Clevo laptop M570U", ALC883_3ST_6ch_DIG), SND_PCI_QUIRK(0x1558, 0x0721, "Clevo laptop M720R", ALC883_CLEVO_M720), SND_PCI_QUIRK(0x1558, 0x0722, "Clevo laptop M720SR", ALC883_CLEVO_M720), SND_PCI_QUIRK(0x1558, 0x5409, "Clevo laptop M540R", ALC883_CLEVO_M540R), -- cgit v1.2.3 From 3353541fe533350a22a03e2fb7dc085b35912575 Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Thu, 22 Apr 2010 07:15:26 -0400 Subject: ALSA: hda: Use ALC880_F1734 quirk for Fujitsu Siemens AMILO Xi 1526 BugLink: https://launchpad.net/bugs/567494 The OR has verified that the existing model quirk, ALC880_UNIWILL, is insufficient for audible playback and capture by default. Instead, the ALC880_F1734 model quirk needs to be used. This change is necessary for both 2.6.32.11 and 2.6.33.2. Reported-by: Arnaud Malpeyre Tested-by: Arnaud Malpeyre Cc: Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 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 aad1627f56f1..7404dba16f83 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4143,7 +4143,7 @@ static struct snd_pci_quirk alc880_cfg_tbl[] = { SND_PCI_QUIRK(0x1695, 0x4012, "EPox EP-5LDA", ALC880_5ST_DIG), SND_PCI_QUIRK(0x1734, 0x107c, "FSC F1734", ALC880_F1734), SND_PCI_QUIRK(0x1734, 0x1094, "FSC Amilo M1451G", ALC880_FUJITSU), - SND_PCI_QUIRK(0x1734, 0x10ac, "FSC", ALC880_UNIWILL), + SND_PCI_QUIRK(0x1734, 0x10ac, "FSC AMILO Xi 1526", ALC880_F1734), SND_PCI_QUIRK(0x1734, 0x10b0, "Fujitsu", ALC880_FUJITSU), SND_PCI_QUIRK(0x1854, 0x0018, "LG LW20", ALC880_LG_LW), SND_PCI_QUIRK(0x1854, 0x003b, "LG", ALC880_LG), -- cgit v1.2.3