From 9fb83526a898f14adbd3f6f52fa7126f528f15ac Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 25 Jan 2012 15:19:20 +0000 Subject: ASoC: wm5100: Make sure we switch to button reporting mode When we have identified an accessory make sure we've flagged that we've done so in order to make sure we always report buttons and don't continue to polarity flip. Signed-off-by: Mark Brown --- sound/soc/codecs/wm5100.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 66f0611..3f8fd3c 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -2183,6 +2183,7 @@ static void wm5100_micd_irq(struct snd_soc_codec *codec) if (wm5100->jack_detecting) { dev_dbg(codec->dev, "Microphone detected\n"); wm5100->jack_mic = true; + wm5100->jack_detecting = false; snd_soc_jack_report(wm5100->jack, SND_JACK_HEADSET, SND_JACK_HEADSET | SND_JACK_BTN_0); @@ -2221,6 +2222,7 @@ static void wm5100_micd_irq(struct snd_soc_codec *codec) SND_JACK_BTN_0); } else if (wm5100->jack_detecting) { dev_dbg(codec->dev, "Headphone detected\n"); + wm5100->jack_detecting = false; snd_soc_jack_report(wm5100->jack, SND_JACK_HEADPHONE, SND_JACK_HEADPHONE); -- cgit v1.1 From a188fcba73837f83a78dc90a44998a978f50ac83 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 25 Jan 2012 17:57:16 +0000 Subject: ASoC: wm5100: Fix microphone configuration We need to write the configuration for each microphone to a different register. Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm5100.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 3f8fd3c..fb757af 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -2612,6 +2612,13 @@ static const struct regmap_config wm5100_regmap = { .cache_type = REGCACHE_RBTREE, }; +static const unsigned int wm5100_mic_ctrl_reg[] = { + WM5100_IN1L_CONTROL, + WM5100_IN2L_CONTROL, + WM5100_IN3L_CONTROL, + WM5100_IN4L_CONTROL, +}; + static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -2744,7 +2751,7 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, } for (i = 0; i < ARRAY_SIZE(wm5100->pdata.in_mode); i++) { - regmap_update_bits(wm5100->regmap, WM5100_IN1L_CONTROL, + regmap_update_bits(wm5100->regmap, wm5100_mic_ctrl_reg[i], WM5100_IN1_MODE_MASK | WM5100_IN1_DMIC_SUP_MASK, (wm5100->pdata.in_mode[i] << -- cgit v1.1 From 1b76d2ee4012f325ae14e0e71dad1a0835195906 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 25 Jan 2012 21:10:07 +0000 Subject: ASoC: wm8996: Mark register cache as dirty when regulators are disabled Otherwise we won't resync later. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8996.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 13aa2bd..61f7daa 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -108,7 +108,7 @@ static int wm8996_regulator_event_##n(struct notifier_block *nb, \ struct wm8996_priv *wm8996 = container_of(nb, struct wm8996_priv, \ disable_nb[n]); \ if (event & REGULATOR_EVENT_DISABLE) { \ - regcache_cache_only(wm8996->regmap, true); \ + regcache_mark_dirty(wm8996->regmap); \ } \ return 0; \ } -- cgit v1.1 From 5539a102882d5ddd1bb95ea9f6f43130a789cb7f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 25 Jan 2012 21:10:21 +0000 Subject: ASoC: wm8962: Mark register cache as dirty when regulators are disabled Otherwise we won't resync later. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 296de4e..bda3da8 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -96,7 +96,7 @@ static int wm8962_regulator_event_##n(struct notifier_block *nb, \ struct wm8962_priv *wm8962 = container_of(nb, struct wm8962_priv, \ disable_nb[n]); \ if (event & REGULATOR_EVENT_DISABLE) { \ - regcache_cache_only(wm8962->regmap, true); \ + regcache_mark_dirty(wm8962->regmap); \ } \ return 0; \ } -- cgit v1.1 From 5c1b136b7bf702e550039cb0039ec9c790c48f99 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 25 Jan 2012 21:10:33 +0000 Subject: ASoC: wm5100: Mark register cache as dirty when regulators are disabled Otherwise we won't resync later. Signed-off-by: Mark Brown --- sound/soc/codecs/wm5100.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index fb757af..89f2af7 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -1405,6 +1405,7 @@ static int wm5100_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_OFF: regcache_cache_only(wm5100->regmap, true); + regcache_mark_dirty(wm5100->regmap); if (wm5100->pdata.ldo_ena) gpio_set_value_cansleep(wm5100->pdata.ldo_ena, 0); regulator_bulk_disable(ARRAY_SIZE(wm5100->core_supplies), -- cgit v1.1 From 77231abe55433aa17eca712718745275853fa66d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 20 Jan 2012 12:19:43 +0000 Subject: ASoC: wm_hubs: Enable line out VMID buffer for single ended line outputs For optimal performance the single ended line outputs require that the line output VMID buffer be enabled. Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm_hubs.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 2a61094..9ccc416 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -613,6 +613,8 @@ SND_SOC_DAPM_INPUT("IN2RP:VXRP"), SND_SOC_DAPM_SUPPLY("MICBIAS2", WM8993_POWER_MANAGEMENT_1, 5, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MICBIAS1", WM8993_POWER_MANAGEMENT_1, 4, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("LINEOUT_VMID_BUF", WM8993_ANTIPOP1, 7, 0, NULL, 0), + SND_SOC_DAPM_MIXER("IN1L PGA", WM8993_POWER_MANAGEMENT_2, 6, 0, in1l_pga, ARRAY_SIZE(in1l_pga)), SND_SOC_DAPM_MIXER("IN1R PGA", WM8993_POWER_MANAGEMENT_2, 4, 0, @@ -834,9 +836,11 @@ static const struct snd_soc_dapm_route lineout1_diff_routes[] = { }; static const struct snd_soc_dapm_route lineout1_se_routes[] = { + { "LINEOUT1N Mixer", NULL, "LINEOUT_VMID_BUF" }, { "LINEOUT1N Mixer", "Left Output Switch", "Left Output PGA" }, { "LINEOUT1N Mixer", "Right Output Switch", "Right Output PGA" }, + { "LINEOUT1P Mixer", NULL, "LINEOUT_VMID_BUF" }, { "LINEOUT1P Mixer", "Left Output Switch", "Left Output PGA" }, { "LINEOUT1N Driver", NULL, "LINEOUT1N Mixer" }, @@ -853,9 +857,11 @@ static const struct snd_soc_dapm_route lineout2_diff_routes[] = { }; static const struct snd_soc_dapm_route lineout2_se_routes[] = { + { "LINEOUT2N Mixer", NULL, "LINEOUT_VMID_BUF" }, { "LINEOUT2N Mixer", "Left Output Switch", "Left Output PGA" }, { "LINEOUT2N Mixer", "Right Output Switch", "Right Output PGA" }, + { "LINEOUT2P Mixer", NULL, "LINEOUT_VMID_BUF" }, { "LINEOUT2P Mixer", "Right Output Switch", "Right Output PGA" }, { "LINEOUT2N Driver", NULL, "LINEOUT2N Mixer" }, -- cgit v1.1 From a389d67cf9849aff1722ed73186a584e2196a873 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Fri, 27 Jan 2012 14:31:19 +0100 Subject: ALSA: HDA: Remove quirk for Asus N53Jq The user reports that he needs to add model=auto for audio to work properly. In fact, since node 0x15 is not even a pin node, the existing fixup is definitely wrong. Relevant information can be found in the buglink below. Cc: stable@kernel.org (3.2+) BugLink: https://bugs.launchpad.net/bugs/918254 Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0db1dc4..a7f17be 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5377,7 +5377,6 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x8330, "ASUS Eeepc P703 P900A", ALC269_FIXUP_AMIC), SND_PCI_QUIRK(0x1043, 0x1013, "ASUS N61Da", ALC269_FIXUP_AMIC), - SND_PCI_QUIRK(0x1043, 0x1113, "ASUS N63Jn", ALC269_FIXUP_AMIC), SND_PCI_QUIRK(0x1043, 0x1143, "ASUS B53f", ALC269_FIXUP_AMIC), SND_PCI_QUIRK(0x1043, 0x1133, "ASUS UJ20ft", ALC269_FIXUP_AMIC), SND_PCI_QUIRK(0x1043, 0x1183, "ASUS K72DR", ALC269_FIXUP_AMIC), -- cgit v1.1 From 114395c61ad2eb5a7a5cd163fcadb2414e48245a Mon Sep 17 00:00:00 2001 From: UK KIM Date: Sat, 28 Jan 2012 01:52:22 +0900 Subject: ASoC: wm_hubs: fix wrong bits for LINEOUT2 N/P mixer Signed-off-by: UK KIM Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm_hubs.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 9ccc416..ea26724 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -592,8 +592,8 @@ SOC_DAPM_SINGLE("Output Switch", WM8993_LINE_MIXER2, 0, 1, 0), }; static const struct snd_kcontrol_new line2n_mix[] = { -SOC_DAPM_SINGLE("Left Output Switch", WM8993_LINE_MIXER2, 6, 1, 0), -SOC_DAPM_SINGLE("Right Output Switch", WM8993_LINE_MIXER2, 5, 1, 0), +SOC_DAPM_SINGLE("Left Output Switch", WM8993_LINE_MIXER2, 5, 1, 0), +SOC_DAPM_SINGLE("Right Output Switch", WM8993_LINE_MIXER2, 6, 1, 0), }; static const struct snd_kcontrol_new line2p_mix[] = { -- cgit v1.1 From 1ae5cbc52e7c6619a3f44b87809fd25370df31bb Mon Sep 17 00:00:00 2001 From: Denis 'GNUtoo' Carikli Date: Mon, 30 Jan 2012 00:31:47 +0100 Subject: ASoC: neo1973_wm8753: remove references to the neo1973-gta01 machine MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The Openmoko GTA01 machine has been removed from the machine ID database, so we need to remove references to it as well. Without that fix we have: sound/soc/samsung/neo1973_wm8753.c: In function ‘neo1973_wm8753_init’: sound/soc/samsung/neo1973_wm8753.c:325:2: error: implicit declaration of function ‘machine_is_neo1973_gta01’ Signed-off-by: Denis 'GNUtoo' Carikli Signed-off-by: Mark Brown --- sound/soc/samsung/neo1973_wm8753.c | 65 +------------------------------------- 1 file changed, 1 insertion(+), 64 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index 7ac0ba2..c6012ff 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -230,8 +230,6 @@ static const struct snd_kcontrol_new neo1973_wm8753_controls[] = { /* GTA02 specific routes and controls */ -#ifdef CONFIG_MACH_NEO1973_GTA02 - static int gta02_speaker_enabled; static int lm4853_set_spk(struct snd_kcontrol *kcontrol, @@ -311,10 +309,6 @@ static int neo1973_gta02_wm8753_init(struct snd_soc_codec *codec) return 0; } -#else -static int neo1973_gta02_wm8753_init(struct snd_soc_code *codec) { return 0; } -#endif - static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; @@ -322,10 +316,6 @@ static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd) int ret; /* set up NC codec pins */ - if (machine_is_neo1973_gta01()) { - snd_soc_dapm_nc_pin(dapm, "LOUT2"); - snd_soc_dapm_nc_pin(dapm, "ROUT2"); - } snd_soc_dapm_nc_pin(dapm, "OUT3"); snd_soc_dapm_nc_pin(dapm, "OUT4"); snd_soc_dapm_nc_pin(dapm, "LINE1"); @@ -370,50 +360,6 @@ static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd) return 0; } -/* GTA01 specific controls */ - -#ifdef CONFIG_MACH_NEO1973_GTA01 - -static const struct snd_soc_dapm_route neo1973_lm4857_routes[] = { - {"Amp IN", NULL, "ROUT1"}, - {"Amp IN", NULL, "LOUT1"}, - - {"Handset Spk", NULL, "Amp EP"}, - {"Stereo Out", NULL, "Amp LS"}, - {"Headphone", NULL, "Amp HP"}, -}; - -static const struct snd_soc_dapm_widget neo1973_lm4857_dapm_widgets[] = { - SND_SOC_DAPM_SPK("Handset Spk", NULL), - SND_SOC_DAPM_SPK("Stereo Out", NULL), - SND_SOC_DAPM_HP("Headphone", NULL), -}; - -static int neo1973_lm4857_init(struct snd_soc_dapm_context *dapm) -{ - int ret; - - ret = snd_soc_dapm_new_controls(dapm, neo1973_lm4857_dapm_widgets, - ARRAY_SIZE(neo1973_lm4857_dapm_widgets)); - if (ret) - return ret; - - ret = snd_soc_dapm_add_routes(dapm, neo1973_lm4857_routes, - ARRAY_SIZE(neo1973_lm4857_routes)); - if (ret) - return ret; - - snd_soc_dapm_ignore_suspend(dapm, "Stereo Out"); - snd_soc_dapm_ignore_suspend(dapm, "Handset Spk"); - snd_soc_dapm_ignore_suspend(dapm, "Headphone"); - - return 0; -} - -#else -static int neo1973_lm4857_init(struct snd_soc_dapm_context *dapm) { return 0; }; -#endif - static struct snd_soc_dai_link neo1973_dai[] = { { /* Hifi Playback - for similatious use with voice below */ .name = "WM8753", @@ -440,11 +386,6 @@ static struct snd_soc_aux_dev neo1973_aux_devs[] = { .name = "dfbmcs320", .codec_name = "dfbmcs320.0", }, - { - .name = "lm4857", - .codec_name = "lm4857.0-007c", - .init = neo1973_lm4857_init, - }, }; static struct snd_soc_codec_conf neo1973_codec_conf[] = { @@ -454,14 +395,10 @@ static struct snd_soc_codec_conf neo1973_codec_conf[] = { }, }; -#ifdef CONFIG_MACH_NEO1973_GTA02 static const struct gpio neo1973_gta02_gpios[] = { { GTA02_GPIO_HP_IN, GPIOF_OUT_INIT_HIGH, "GTA02_HP_IN" }, { GTA02_GPIO_AMP_SHUT, GPIOF_OUT_INIT_HIGH, "GTA02_AMP_SHUT" }, }; -#else -static const struct gpio neo1973_gta02_gpios[] = {}; -#endif static struct snd_soc_card neo1973 = { .name = "neo1973", @@ -480,7 +417,7 @@ static int __init neo1973_init(void) { int ret; - if (!machine_is_neo1973_gta01() && !machine_is_neo1973_gta02()) + if (!machine_is_neo1973_gta02()) return -ENODEV; if (machine_is_neo1973_gta02()) { -- cgit v1.1 From 31150f2327cbb66363f38e13ca1be973d2f9203a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 30 Jan 2012 10:54:08 +0100 Subject: ALSA: hda - Apply 0x0f-VREF fix to all ASUS laptops with ALC861/660 It turned out that other ASUS laptops require the similar fix to enable the VREF on the pin 0x0f for the secret output amp, not only ASUS A6Rp. Moreover, it's required even when the pin is being used as the output. Thus, writing a fixed value doesn't work always. This patch applies the VREF-fix for all ASUS laptops with ALC861/660 in a fixup function that checks the current value and turns on only the VREF value no matter whether input or output direction is set. The automute function is modified as well to keep the pin VREF upon muting/unmuting via pin-control; otherwise the pin VREF is reset at plugging/unplugging a jack. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=42588 Cc: [v3.2+] Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 43 +++++++++++++++++++++++++++++++++++-------- 1 file changed, 35 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a7f17be..42b6a01 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -177,6 +177,7 @@ struct alc_spec { unsigned int detect_lo:1; /* Line-out detection enabled */ unsigned int automute_speaker_possible:1; /* there are speakers and either LO or HP */ unsigned int automute_lo_possible:1; /* there are line outs and HP */ + unsigned int keep_vref_in_automute:1; /* Don't clear VREF in automute */ /* other flags */ unsigned int no_analog :1; /* digital I/O only */ @@ -495,13 +496,24 @@ static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins, for (i = 0; i < num_pins; i++) { hda_nid_t nid = pins[i]; + unsigned int val; if (!nid) break; switch (spec->automute_mode) { case ALC_AUTOMUTE_PIN: + /* don't reset VREF value in case it's controlling + * the amp (see alc861_fixup_asus_amp_vref_0f()) + */ + if (spec->keep_vref_in_automute) { + val = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + val &= ~PIN_HP; + } else + val = 0; + val |= pin_bits; snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - pin_bits); + val); break; case ALC_AUTOMUTE_AMP: snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0, @@ -5588,6 +5600,25 @@ enum { PINFIX_ASUS_A6RP, }; +/* On some laptops, VREF of pin 0x0f is abused for controlling the main amp */ +static void alc861_fixup_asus_amp_vref_0f(struct hda_codec *codec, + const struct alc_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + unsigned int val; + + if (action != ALC_FIXUP_ACT_INIT) + return; + val = snd_hda_codec_read(codec, 0x0f, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + if (!(val & (AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN))) + val |= AC_PINCTL_IN_EN; + val |= AC_PINCTL_VREF_50; + snd_hda_codec_write(codec, 0x0f, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, val); + spec->keep_vref_in_automute = 1; +} + static const struct alc_fixup alc861_fixups[] = { [PINFIX_FSC_AMILO_PI1505] = { .type = ALC_FIXUP_PINS, @@ -5598,17 +5629,13 @@ static const struct alc_fixup alc861_fixups[] = { } }, [PINFIX_ASUS_A6RP] = { - .type = ALC_FIXUP_VERBS, - .v.verbs = (const struct hda_verb[]) { - /* node 0x0f VREF seems controlling the master output */ - { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50 }, - { } - }, + .type = ALC_FIXUP_FUNC, + .v.func = alc861_fixup_asus_amp_vref_0f, }, }; static const struct snd_pci_quirk alc861_fixup_tbl[] = { - SND_PCI_QUIRK(0x1043, 0x1393, "ASUS A6Rp", PINFIX_ASUS_A6RP), + SND_PCI_QUIRK_VENDOR(0x1043, "ASUS laptop", PINFIX_ASUS_A6RP), SND_PCI_QUIRK(0x1584, 0x2b01, "Haier W18", PINFIX_ASUS_A6RP), SND_PCI_QUIRK(0x1734, 0x10c7, "FSC Amilo Pi1505", PINFIX_FSC_AMILO_PI1505), {} -- cgit v1.1 From ee76744c51ec342df9822b4a85dbbfc3887b6d60 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 31 Jan 2012 11:55:32 +0000 Subject: ASoC: wm_hubs: Fix routing of input PGAs to line output mixer IN1L/R is routed to both line output mixers, we don't route IN1 to LINEOUT1 and IN2 to LINEOUT2. Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm_hubs.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index ea26724..c1a3f8c 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -586,8 +586,8 @@ SOC_DAPM_SINGLE("Left Output Switch", WM8993_LINE_MIXER1, 0, 1, 0), }; static const struct snd_kcontrol_new line2_mix[] = { -SOC_DAPM_SINGLE("IN2R Switch", WM8993_LINE_MIXER2, 2, 1, 0), -SOC_DAPM_SINGLE("IN2L Switch", WM8993_LINE_MIXER2, 1, 1, 0), +SOC_DAPM_SINGLE("IN1R Switch", WM8993_LINE_MIXER2, 2, 1, 0), +SOC_DAPM_SINGLE("IN1L Switch", WM8993_LINE_MIXER2, 1, 1, 0), SOC_DAPM_SINGLE("Output Switch", WM8993_LINE_MIXER2, 0, 1, 0), }; @@ -848,8 +848,8 @@ static const struct snd_soc_dapm_route lineout1_se_routes[] = { }; static const struct snd_soc_dapm_route lineout2_diff_routes[] = { - { "LINEOUT2 Mixer", "IN2L Switch", "IN2L PGA" }, - { "LINEOUT2 Mixer", "IN2R Switch", "IN2R PGA" }, + { "LINEOUT2 Mixer", "IN1L Switch", "IN1L PGA" }, + { "LINEOUT2 Mixer", "IN1R Switch", "IN1R PGA" }, { "LINEOUT2 Mixer", "Output Switch", "Right Output PGA" }, { "LINEOUT2N Driver", NULL, "LINEOUT2 Mixer" }, -- cgit v1.1 From 05c3b36e539627b7aed67d038381d0d9fa9d61e7 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Tue, 31 Jan 2012 09:04:15 +0100 Subject: ALSA: HDA: Fix jack creation for codecs with front and rear Line In If a codec has both a front and a rear Line In, two controls both named "Line Jack" will be created, which causes parsing to fail. While a long term solution might be to name the jacks differently, this extra check is consistent with what is already being done in many auto-parsers, and will also protect against other cases when two inputs have the same label. BugLink: https://bugs.launchpad.net/bugs/923409 Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_jack.c | 24 +++++++++++++++--------- 1 file changed, 15 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index d8a35da..9d819c4 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -282,7 +282,8 @@ int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, EXPORT_SYMBOL_HDA(snd_hda_jack_add_kctl); static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid, - const struct auto_pin_cfg *cfg) + const struct auto_pin_cfg *cfg, + char *lastname, int *lastidx) { unsigned int def_conf, conn; char name[44]; @@ -298,6 +299,10 @@ static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid, return 0; snd_hda_get_pin_label(codec, nid, cfg, name, sizeof(name), &idx); + if (!strcmp(name, lastname) && idx == *lastidx) + idx++; + strncpy(lastname, name, 44); + *lastidx = idx; err = snd_hda_jack_add_kctl(codec, nid, name, idx); if (err < 0) return err; @@ -311,41 +316,42 @@ int snd_hda_jack_add_kctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { const hda_nid_t *p; - int i, err; + int i, err, lastidx = 0; + char lastname[44] = ""; for (i = 0, p = cfg->line_out_pins; i < cfg->line_outs; i++, p++) { - err = add_jack_kctl(codec, *p, cfg); + err = add_jack_kctl(codec, *p, cfg, lastname, &lastidx); if (err < 0) return err; } for (i = 0, p = cfg->hp_pins; i < cfg->hp_outs; i++, p++) { if (*p == *cfg->line_out_pins) /* might be duplicated */ break; - err = add_jack_kctl(codec, *p, cfg); + err = add_jack_kctl(codec, *p, cfg, lastname, &lastidx); if (err < 0) return err; } for (i = 0, p = cfg->speaker_pins; i < cfg->speaker_outs; i++, p++) { if (*p == *cfg->line_out_pins) /* might be duplicated */ break; - err = add_jack_kctl(codec, *p, cfg); + err = add_jack_kctl(codec, *p, cfg, lastname, &lastidx); if (err < 0) return err; } for (i = 0; i < cfg->num_inputs; i++) { - err = add_jack_kctl(codec, cfg->inputs[i].pin, cfg); + err = add_jack_kctl(codec, cfg->inputs[i].pin, cfg, lastname, &lastidx); if (err < 0) return err; } for (i = 0, p = cfg->dig_out_pins; i < cfg->dig_outs; i++, p++) { - err = add_jack_kctl(codec, *p, cfg); + err = add_jack_kctl(codec, *p, cfg, lastname, &lastidx); if (err < 0) return err; } - err = add_jack_kctl(codec, cfg->dig_in_pin, cfg); + err = add_jack_kctl(codec, cfg->dig_in_pin, cfg, lastname, &lastidx); if (err < 0) return err; - err = add_jack_kctl(codec, cfg->mono_out_pin, cfg); + err = add_jack_kctl(codec, cfg->mono_out_pin, cfg, lastname, &lastidx); if (err < 0) return err; return 0; -- cgit v1.1 From 3422a47041b8cb8f14ac1e3926bcf711121df6dc Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Tue, 31 Jan 2012 10:31:49 +0100 Subject: ALSA: HDA: Remove quirk for Toshiba Qosmio G50 The user reports that model=auto works better than current handling on a 3.2 based kernel (with jack detection patches backported). Since model=auto is what we prefer these days anyway, the quirk should be removed. Alsa-info for the relevant machine: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/923316/+attachment/2702812/+files/alsa-info.txt.Pbfno2x7bp BugLink: https://bugs.launchpad.net/bugs/923316 Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 13 ------------- 1 file changed, 13 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 42b6a01..a8e82be 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4747,7 +4747,6 @@ enum { ALC262_FIXUP_FSC_H270, ALC262_FIXUP_HP_Z200, ALC262_FIXUP_TYAN, - ALC262_FIXUP_TOSHIBA_RX1, ALC262_FIXUP_LENOVO_3000, ALC262_FIXUP_BENQ, ALC262_FIXUP_BENQ_T31, @@ -4777,16 +4776,6 @@ static const struct alc_fixup alc262_fixups[] = { { } } }, - [ALC262_FIXUP_TOSHIBA_RX1] = { - .type = ALC_FIXUP_PINS, - .v.pins = (const struct alc_pincfg[]) { - { 0x14, 0x90170110 }, /* speaker */ - { 0x15, 0x0421101f }, /* HP */ - { 0x1a, 0x40f000f0 }, /* N/A */ - { 0x1b, 0x40f000f0 }, /* N/A */ - { 0x1e, 0x40f000f0 }, /* N/A */ - } - }, [ALC262_FIXUP_LENOVO_3000] = { .type = ALC_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -4819,8 +4808,6 @@ static const struct snd_pci_quirk alc262_fixup_tbl[] = { SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FIXUP_BENQ), SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FIXUP_BENQ), SND_PCI_QUIRK(0x10f1, 0x2915, "Tyan Thunder n6650W", ALC262_FIXUP_TYAN), - SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1", - ALC262_FIXUP_TOSHIBA_RX1), SND_PCI_QUIRK(0x1734, 0x1147, "FSC Celsius H270", ALC262_FIXUP_FSC_H270), SND_PCI_QUIRK(0x17aa, 0x384e, "Lenovo 3000", ALC262_FIXUP_LENOVO_3000), SND_PCI_QUIRK(0x17ff, 0x0560, "Benq ED8", ALC262_FIXUP_BENQ), -- cgit v1.1 From 67f97f5c3edad35c4d37a94f994c76111a177fb6 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 31 Jan 2012 14:51:29 +0000 Subject: ASoC: wm8994: Remove ASoC level register cache sync Now we've switched over to regmap the ASoC level cache sync will be ineffectual and potentially harmful as there is no longer an ASoC level cache. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 5 ----- 1 file changed, 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 93d27b6..8623950 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2753,11 +2753,6 @@ static int wm8994_resume(struct snd_soc_codec *codec) codec->cache_only = 0; } - /* Restore the registers */ - ret = snd_soc_cache_sync(codec); - if (ret != 0) - dev_err(codec->dev, "Failed to sync cache: %d\n", ret); - wm8994_set_bias_level(codec, SND_SOC_BIAS_STANDBY); for (i = 0; i < ARRAY_SIZE(wm8994->fll); i++) { -- cgit v1.1 From 125a25da5729740b7d1dc417a3d5549321baae17 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 31 Jan 2012 15:49:10 +0000 Subject: ASoC: core: Better support for idle_bias_off suspend ignores If an idle_bias_off device is in any state other than off then it is still active for some reason (typically a low power function such as accessory detection). This wasn't an issue when the feature was implemented as we always went to _ON for any active function, subsequent power improvements have changed things. With the modern way of doing things we should overhaul the infrastructure to allow devices to explicitly take references for these functions but that's a much more invasive change and will require driver updates to deploy, this will bring the framework into line with the existing driver set before we do that work. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-core.c | 11 +++++++++++ 1 file changed, 11 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b5ecf6d..92cee24 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -567,6 +567,17 @@ int snd_soc_suspend(struct device *dev) if (!codec->suspended && codec->driver->suspend) { switch (codec->dapm.bias_level) { case SND_SOC_BIAS_STANDBY: + /* + * If the CODEC is capable of idle + * bias off then being in STANDBY + * means it's doing something, + * otherwise fall through. + */ + if (codec->dapm.idle_bias_off) { + dev_dbg(codec->dev, + "idle_bias_off CODEC on over suspend\n"); + break; + } case SND_SOC_BIAS_OFF: codec->driver->suspend(codec); codec->suspended = 1; -- cgit v1.1 From f70eecde3bca92630d3886496e73316ff353f185 Mon Sep 17 00:00:00 2001 From: Dylan Reid Date: Tue, 31 Jan 2012 13:04:41 -0800 Subject: ALSA: hda - Fix calling cs_automic twice for Cirrus codecs. If cs_automic is called twice (like it is during init) while the mic is present, it will over-write the last_input with the new one, causing it to switch back to the automic input when the mic is unplugged. This leaves the driver in a state (cur_input, last_input, and automix_idx the same) where the internal mic can not be selected until it is rebooted without the mic attached. Check that the mic hasn't already been switched to before setting last_input. Signed-off-by: Dylan Reid Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cirrus.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 0e99357..bc5a993 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -988,8 +988,10 @@ static void cs_automic(struct hda_codec *codec) change_cur_input(codec, !spec->automic_idx, 0); } else { if (present) { - spec->last_input = spec->cur_input; - spec->cur_input = spec->automic_idx; + if (spec->cur_input != spec->automic_idx) { + spec->last_input = spec->cur_input; + spec->cur_input = spec->automic_idx; + } } else { spec->cur_input = spec->last_input; } -- cgit v1.1 From 2b6712b19531e22455e7fa18371c5ba9eec76699 Mon Sep 17 00:00:00 2001 From: Susan Gao Date: Mon, 30 Jan 2012 13:57:04 -0800 Subject: ASoC: wm8962: Fix word length configuration Signed-off-by: Susan Gao Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm8962.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index bda3da8..29c4b02 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3159,13 +3159,13 @@ static int wm8962_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S16_LE: break; case SNDRV_PCM_FORMAT_S20_3LE: - aif0 |= 0x40; + aif0 |= 0x4; break; case SNDRV_PCM_FORMAT_S24_LE: - aif0 |= 0x80; + aif0 |= 0x8; break; case SNDRV_PCM_FORMAT_S32_LE: - aif0 |= 0xc0; + aif0 |= 0xc; break; default: return -EINVAL; -- cgit v1.1 From 44bed4838dc191988fd1d03deccc3a845705d2de Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 31 Jan 2012 09:49:04 +0800 Subject: ASoC: cs42l73: Fix Output [X|A|V]SP_SCLK Sourcing Mode setting for master mode For master mode, set Output [X|A|V]SP_SCLK Sourcing Mode to MCLK Mode. Signed-off-by: Axel Lin Acked-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l73.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 9d38db8..78979b3 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1113,7 +1113,7 @@ static int cs42l73_pcm_hw_params(struct snd_pcm_substream *substream, priv->config[id].mmcc &= 0xC0; priv->config[id].mmcc |= cs42l73_mclk_coeffs[mclk_coeff].mmcc; priv->config[id].spc &= 0xFC; - priv->config[id].spc &= MCK_SCLK_64FS; + priv->config[id].spc |= MCK_SCLK_MCLK; } else { /* CS42L73 Slave */ priv->config[id].spc &= 0xFC; -- cgit v1.1 From 54c2a89f60fd71b924d0f848ac892442951401a6 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Wed, 1 Feb 2012 12:05:41 +0100 Subject: ALSA: HDA: Fix duplicated output to more than one codec This typo caused the wrong codec's nid to be checked for wcaps type. As a result, sometimes speakers would duplicate the output sent to HDMI output. Cc: stable@kernel.org BugLink: https://bugs.launchpad.net/bugs/924320 Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 4df72c0..c2c65f6 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1447,7 +1447,7 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, for (i = 0; i < c->cvt_setups.used; i++) { p = snd_array_elem(&c->cvt_setups, i); if (!p->active && p->stream_tag == stream_tag && - get_wcaps_type(get_wcaps(codec, p->nid)) == type) + get_wcaps_type(get_wcaps(c, p->nid)) == type) p->dirty = 1; } } -- cgit v1.1 From 43b6cec27e1e50a1de3eff47e66e502f3fe7e66e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 1 Feb 2012 23:46:58 +0000 Subject: ASoC: wm_hubs: Correct line input to line output 2 paths The second line output mixer has the controls for the line input bypasses in the opposite order. Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm_hubs.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index c1a3f8c..8a68cea 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -586,8 +586,8 @@ SOC_DAPM_SINGLE("Left Output Switch", WM8993_LINE_MIXER1, 0, 1, 0), }; static const struct snd_kcontrol_new line2_mix[] = { -SOC_DAPM_SINGLE("IN1R Switch", WM8993_LINE_MIXER2, 2, 1, 0), -SOC_DAPM_SINGLE("IN1L Switch", WM8993_LINE_MIXER2, 1, 1, 0), +SOC_DAPM_SINGLE("IN1L Switch", WM8993_LINE_MIXER2, 2, 1, 0), +SOC_DAPM_SINGLE("IN1R Switch", WM8993_LINE_MIXER2, 1, 1, 0), SOC_DAPM_SINGLE("Output Switch", WM8993_LINE_MIXER2, 0, 1, 0), }; -- cgit v1.1 From 054d867e032daf55c3343fc6d36c5c5f1e7954db Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 24 Jan 2012 12:25:50 +0100 Subject: ALSA: hda - Check power-state before changing in patch_via.c Instead of always writing AC_VERB_SET_POWER_STATE, check the current power-state and don't write again if the value is already set. This may reduce the click noise upon the dynamic power-state change (e.g. in analog-input mixer). Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 256 +++++++++++++++++++--------------------------- 1 file changed, 107 insertions(+), 149 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 03e63fe..fb1f0ff 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -687,6 +687,15 @@ static void via_auto_init_analog_input(struct hda_codec *codec) } } +static void update_power_state(struct hda_codec *codec, hda_nid_t nid, + unsigned int parm) +{ + if (snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_POWER_STATE, 0) == parm) + return; + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, parm); +} + static void set_pin_power_state(struct hda_codec *codec, hda_nid_t nid, unsigned int *affected_parm) { @@ -709,7 +718,7 @@ static void set_pin_power_state(struct hda_codec *codec, hda_nid_t nid, } else parm = AC_PWRST_D3; - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, nid, parm); } static int via_pin_power_ctl_info(struct snd_kcontrol *kcontrol, @@ -2295,10 +2304,7 @@ static int via_mux_enum_put(struct snd_kcontrol *kcontrol, if (mux) { /* switch to D0 beofre change index */ - if (snd_hda_codec_read(codec, mux, 0, - AC_VERB_GET_POWER_STATE, 0x00) != AC_PWRST_D0) - snd_hda_codec_write(codec, mux, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + update_power_state(codec, mux, AC_PWRST_D0); snd_hda_codec_write(codec, mux, 0, AC_VERB_SET_CONNECT_SEL, spec->inputs[cur].mux_idx); @@ -2922,9 +2928,9 @@ static void set_widgets_power_state_vt1708B(struct hda_codec *codec) if (imux_is_smixer) parm = AC_PWRST_D0; /* SW0 (17h), AIW 0/1 (13h/14h) */ - snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x17, parm); + update_power_state(codec, 0x13, parm); + update_power_state(codec, 0x14, parm); /* outputs */ /* PW0 (19h), SW1 (18h), AOW1 (11h) */ @@ -2932,8 +2938,8 @@ static void set_widgets_power_state_vt1708B(struct hda_codec *codec) set_pin_power_state(codec, 0x19, &parm); if (spec->smart51_enabled) set_pin_power_state(codec, 0x1b, &parm); - snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x18, parm); + update_power_state(codec, 0x11, parm); /* PW6 (22h), SW2 (26h), AOW2 (24h) */ if (is_8ch) { @@ -2941,20 +2947,16 @@ static void set_widgets_power_state_vt1708B(struct hda_codec *codec) set_pin_power_state(codec, 0x22, &parm); if (spec->smart51_enabled) set_pin_power_state(codec, 0x1a, &parm); - snd_hda_codec_write(codec, 0x26, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x24, 0, - AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x26, parm); + update_power_state(codec, 0x24, parm); } else if (codec->vendor_id == 0x11064397) { /* PW7(23h), SW2(27h), AOW2(25h) */ parm = AC_PWRST_D3; set_pin_power_state(codec, 0x23, &parm); if (spec->smart51_enabled) set_pin_power_state(codec, 0x1a, &parm); - snd_hda_codec_write(codec, 0x27, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x25, 0, - AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x27, parm); + update_power_state(codec, 0x25, parm); } /* PW 3/4/7 (1ch/1dh/23h) */ @@ -2966,17 +2968,13 @@ static void set_widgets_power_state_vt1708B(struct hda_codec *codec) set_pin_power_state(codec, 0x23, &parm); /* MW0 (16h), Sw3 (27h), AOW 0/3 (10h/25h) */ - snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_POWER_STATE, - imux_is_smixer ? AC_PWRST_D0 : parm); - snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x16, imux_is_smixer ? AC_PWRST_D0 : parm); + update_power_state(codec, 0x10, parm); if (is_8ch) { - snd_hda_codec_write(codec, 0x25, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x27, 0, - AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x25, parm); + update_power_state(codec, 0x27, parm); } else if (codec->vendor_id == 0x11064397 && spec->hp_independent_mode) - snd_hda_codec_write(codec, 0x25, 0, - AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x25, parm); } static int patch_vt1708S(struct hda_codec *codec); @@ -3149,10 +3147,10 @@ static void set_widgets_power_state_vt1702(struct hda_codec *codec) if (imux_is_smixer) parm = AC_PWRST_D0; /* SW0 (13h) = stereo mixer (idx 3) */ /* SW0 (13h), AIW 0/1/2 (12h/1fh/20h) */ - snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x12, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x13, parm); + update_power_state(codec, 0x12, parm); + update_power_state(codec, 0x1f, parm); + update_power_state(codec, 0x20, parm); /* outputs */ /* PW 3/4 (16h/17h) */ @@ -3160,10 +3158,9 @@ static void set_widgets_power_state_vt1702(struct hda_codec *codec) set_pin_power_state(codec, 0x17, &parm); set_pin_power_state(codec, 0x16, &parm); /* MW0 (1ah), AOW 0/1 (10h/1dh) */ - snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_POWER_STATE, - imux_is_smixer ? AC_PWRST_D0 : parm); - snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x1d, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x1a, imux_is_smixer ? AC_PWRST_D0 : parm); + update_power_state(codec, 0x10, parm); + update_power_state(codec, 0x1d, parm); } static int patch_vt1702(struct hda_codec *codec) @@ -3228,52 +3225,48 @@ static void set_widgets_power_state_vt1718S(struct hda_codec *codec) if (imux_is_smixer) parm = AC_PWRST_D0; /* MUX6/7 (1eh/1fh), AIW 0/1 (10h/11h) */ - snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x1e, parm); + update_power_state(codec, 0x1f, parm); + update_power_state(codec, 0x10, parm); + update_power_state(codec, 0x11, parm); /* outputs */ /* PW3 (27h), MW2 (1ah), AOW3 (bh) */ parm = AC_PWRST_D3; set_pin_power_state(codec, 0x27, &parm); - snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0xb, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x1a, parm); + update_power_state(codec, 0xb, parm); /* PW2 (26h), AOW2 (ah) */ parm = AC_PWRST_D3; set_pin_power_state(codec, 0x26, &parm); if (spec->smart51_enabled) set_pin_power_state(codec, 0x2b, &parm); - snd_hda_codec_write(codec, 0xa, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0xa, parm); /* PW0 (24h), AOW0 (8h) */ parm = AC_PWRST_D3; set_pin_power_state(codec, 0x24, &parm); if (!spec->hp_independent_mode) /* check for redirected HP */ set_pin_power_state(codec, 0x28, &parm); - snd_hda_codec_write(codec, 0x8, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x8, parm); /* MW9 (21h), Mw2 (1ah), AOW0 (8h) */ - snd_hda_codec_write(codec, 0x21, 0, AC_VERB_SET_POWER_STATE, - imux_is_smixer ? AC_PWRST_D0 : parm); + update_power_state(codec, 0x21, imux_is_smixer ? AC_PWRST_D0 : parm); /* PW1 (25h), AOW1 (9h) */ parm = AC_PWRST_D3; set_pin_power_state(codec, 0x25, &parm); if (spec->smart51_enabled) set_pin_power_state(codec, 0x2a, &parm); - snd_hda_codec_write(codec, 0x9, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x9, parm); if (spec->hp_independent_mode) { /* PW4 (28h), MW3 (1bh), MUX1(34h), AOW4 (ch) */ parm = AC_PWRST_D3; set_pin_power_state(codec, 0x28, &parm); - snd_hda_codec_write(codec, 0x1b, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x34, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0xc, 0, - AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x1b, parm); + update_power_state(codec, 0x34, parm); + update_power_state(codec, 0xc, parm); } } @@ -3433,8 +3426,8 @@ static void set_widgets_power_state_vt1716S(struct hda_codec *codec) if (imux_is_smixer) parm = AC_PWRST_D0; /* SW0 (17h), AIW0(13h) */ - snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x17, parm); + update_power_state(codec, 0x13, parm); parm = AC_PWRST_D3; set_pin_power_state(codec, 0x1e, &parm); @@ -3442,12 +3435,11 @@ static void set_widgets_power_state_vt1716S(struct hda_codec *codec) if (spec->dmic_enabled) set_pin_power_state(codec, 0x22, &parm); else - snd_hda_codec_write(codec, 0x22, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + update_power_state(codec, 0x22, AC_PWRST_D3); /* SW2(26h), AIW1(14h) */ - snd_hda_codec_write(codec, 0x26, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x26, parm); + update_power_state(codec, 0x14, parm); /* outputs */ /* PW0 (19h), SW1 (18h), AOW1 (11h) */ @@ -3456,8 +3448,8 @@ static void set_widgets_power_state_vt1716S(struct hda_codec *codec) /* Smart 5.1 PW2(1bh) */ if (spec->smart51_enabled) set_pin_power_state(codec, 0x1b, &parm); - snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x18, parm); + update_power_state(codec, 0x11, parm); /* PW7 (23h), SW3 (27h), AOW3 (25h) */ parm = AC_PWRST_D3; @@ -3465,12 +3457,12 @@ static void set_widgets_power_state_vt1716S(struct hda_codec *codec) /* Smart 5.1 PW1(1ah) */ if (spec->smart51_enabled) set_pin_power_state(codec, 0x1a, &parm); - snd_hda_codec_write(codec, 0x27, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x27, parm); /* Smart 5.1 PW5(1eh) */ if (spec->smart51_enabled) set_pin_power_state(codec, 0x1e, &parm); - snd_hda_codec_write(codec, 0x25, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x25, parm); /* Mono out */ /* SW4(28h)->MW1(29h)-> PW12 (2ah)*/ @@ -3486,9 +3478,9 @@ static void set_widgets_power_state_vt1716S(struct hda_codec *codec) mono_out = 1; } parm = mono_out ? AC_PWRST_D0 : AC_PWRST_D3; - snd_hda_codec_write(codec, 0x28, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x29, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x2a, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x28, parm); + update_power_state(codec, 0x29, parm); + update_power_state(codec, 0x2a, parm); /* PW 3/4 (1ch/1dh) */ parm = AC_PWRST_D3; @@ -3496,15 +3488,12 @@ static void set_widgets_power_state_vt1716S(struct hda_codec *codec) set_pin_power_state(codec, 0x1d, &parm); /* HP Independent Mode, power on AOW3 */ if (spec->hp_independent_mode) - snd_hda_codec_write(codec, 0x25, 0, - AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x25, parm); /* force to D0 for internal Speaker */ /* MW0 (16h), AOW0 (10h) */ - snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_POWER_STATE, - imux_is_smixer ? AC_PWRST_D0 : parm); - snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, - mono_out ? AC_PWRST_D0 : parm); + update_power_state(codec, 0x16, imux_is_smixer ? AC_PWRST_D0 : parm); + update_power_state(codec, 0x10, mono_out ? AC_PWRST_D0 : parm); } static int patch_vt1716S(struct hda_codec *codec) @@ -3580,54 +3569,45 @@ static void set_widgets_power_state_vt2002P(struct hda_codec *codec) set_pin_power_state(codec, 0x2b, &parm); parm = AC_PWRST_D0; /* MUX9/10 (1eh/1fh), AIW 0/1 (10h/11h) */ - snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x1e, parm); + update_power_state(codec, 0x1f, parm); + update_power_state(codec, 0x10, parm); + update_power_state(codec, 0x11, parm); /* outputs */ /* AOW0 (8h)*/ - snd_hda_codec_write(codec, 0x8, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x8, parm); if (spec->codec_type == VT1802) { /* PW4 (28h), MW4 (18h), MUX4(38h) */ parm = AC_PWRST_D3; set_pin_power_state(codec, 0x28, &parm); - snd_hda_codec_write(codec, 0x18, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x38, 0, - AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x18, parm); + update_power_state(codec, 0x38, parm); } else { /* PW4 (26h), MW4 (1ch), MUX4(37h) */ parm = AC_PWRST_D3; set_pin_power_state(codec, 0x26, &parm); - snd_hda_codec_write(codec, 0x1c, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x37, 0, - AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x1c, parm); + update_power_state(codec, 0x37, parm); } if (spec->codec_type == VT1802) { /* PW1 (25h), MW1 (15h), MUX1(35h), AOW1 (9h) */ parm = AC_PWRST_D3; set_pin_power_state(codec, 0x25, &parm); - snd_hda_codec_write(codec, 0x15, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x35, 0, - AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x15, parm); + update_power_state(codec, 0x35, parm); } else { /* PW1 (25h), MW1 (19h), MUX1(35h), AOW1 (9h) */ parm = AC_PWRST_D3; set_pin_power_state(codec, 0x25, &parm); - snd_hda_codec_write(codec, 0x19, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x35, 0, - AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x19, parm); + update_power_state(codec, 0x35, parm); } if (spec->hp_independent_mode) - snd_hda_codec_write(codec, 0x9, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + update_power_state(codec, 0x9, AC_PWRST_D0); /* Class-D */ /* PW0 (24h), MW0(18h/14h), MUX0(34h) */ @@ -3637,12 +3617,10 @@ static void set_widgets_power_state_vt2002P(struct hda_codec *codec) set_pin_power_state(codec, 0x24, &parm); parm = present ? AC_PWRST_D3 : AC_PWRST_D0; if (spec->codec_type == VT1802) - snd_hda_codec_write(codec, 0x14, 0, - AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x14, parm); else - snd_hda_codec_write(codec, 0x18, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x34, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x18, parm); + update_power_state(codec, 0x34, parm); /* Mono Out */ present = snd_hda_jack_detect(codec, 0x26); @@ -3650,28 +3628,20 @@ static void set_widgets_power_state_vt2002P(struct hda_codec *codec) parm = present ? AC_PWRST_D3 : AC_PWRST_D0; if (spec->codec_type == VT1802) { /* PW15 (33h), MW8(1ch), MUX8(3ch) */ - snd_hda_codec_write(codec, 0x33, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x1c, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x3c, 0, - AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x33, parm); + update_power_state(codec, 0x1c, parm); + update_power_state(codec, 0x3c, parm); } else { /* PW15 (31h), MW8(17h), MUX8(3bh) */ - snd_hda_codec_write(codec, 0x31, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x17, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x3b, 0, - AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x31, parm); + update_power_state(codec, 0x17, parm); + update_power_state(codec, 0x3b, parm); } /* MW9 (21h) */ if (imux_is_smixer || !is_aa_path_mute(codec)) - snd_hda_codec_write(codec, 0x21, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + update_power_state(codec, 0x21, AC_PWRST_D0); else - snd_hda_codec_write(codec, 0x21, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + update_power_state(codec, 0x21, AC_PWRST_D3); } /* patch for vt2002P */ @@ -3731,30 +3701,28 @@ static void set_widgets_power_state_vt1812(struct hda_codec *codec) set_pin_power_state(codec, 0x2b, &parm); parm = AC_PWRST_D0; /* MUX10/11 (1eh/1fh), AIW 0/1 (10h/11h) */ - snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x1e, parm); + update_power_state(codec, 0x1f, parm); + update_power_state(codec, 0x10, parm); + update_power_state(codec, 0x11, parm); /* outputs */ /* AOW0 (8h)*/ - snd_hda_codec_write(codec, 0x8, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + update_power_state(codec, 0x8, AC_PWRST_D0); /* PW4 (28h), MW4 (18h), MUX4(38h) */ parm = AC_PWRST_D3; set_pin_power_state(codec, 0x28, &parm); - snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x38, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x18, parm); + update_power_state(codec, 0x38, parm); /* PW1 (25h), MW1 (15h), MUX1(35h), AOW1 (9h) */ parm = AC_PWRST_D3; set_pin_power_state(codec, 0x25, &parm); - snd_hda_codec_write(codec, 0x15, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x35, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x15, parm); + update_power_state(codec, 0x35, parm); if (spec->hp_independent_mode) - snd_hda_codec_write(codec, 0x9, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + update_power_state(codec, 0x9, AC_PWRST_D0); /* Internal Speaker */ /* PW0 (24h), MW0(14h), MUX0(34h) */ @@ -3763,15 +3731,11 @@ static void set_widgets_power_state_vt1812(struct hda_codec *codec) parm = AC_PWRST_D3; set_pin_power_state(codec, 0x24, &parm); if (present) { - snd_hda_codec_write(codec, 0x14, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - snd_hda_codec_write(codec, 0x34, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + update_power_state(codec, 0x14, AC_PWRST_D3); + update_power_state(codec, 0x34, AC_PWRST_D3); } else { - snd_hda_codec_write(codec, 0x14, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - snd_hda_codec_write(codec, 0x34, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + update_power_state(codec, 0x14, AC_PWRST_D0); + update_power_state(codec, 0x34, AC_PWRST_D0); } @@ -3782,26 +3746,20 @@ static void set_widgets_power_state_vt1812(struct hda_codec *codec) parm = AC_PWRST_D3; set_pin_power_state(codec, 0x31, &parm); if (present) { - snd_hda_codec_write(codec, 0x1c, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - snd_hda_codec_write(codec, 0x3c, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - snd_hda_codec_write(codec, 0x3e, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + update_power_state(codec, 0x1c, AC_PWRST_D3); + update_power_state(codec, 0x3c, AC_PWRST_D3); + update_power_state(codec, 0x3e, AC_PWRST_D3); } else { - snd_hda_codec_write(codec, 0x1c, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - snd_hda_codec_write(codec, 0x3c, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - snd_hda_codec_write(codec, 0x3e, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + update_power_state(codec, 0x1c, AC_PWRST_D0); + update_power_state(codec, 0x3c, AC_PWRST_D0); + update_power_state(codec, 0x3e, AC_PWRST_D0); } /* PW15 (33h), MW15 (1dh), MUX15(3dh) */ parm = AC_PWRST_D3; set_pin_power_state(codec, 0x33, &parm); - snd_hda_codec_write(codec, 0x1d, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x3d, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x1d, parm); + update_power_state(codec, 0x3d, parm); } -- cgit v1.1 From 924339239fd5ba3e505f9420d41f0939196f3530 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 24 Jan 2012 13:58:36 +0100 Subject: ALSA: hda - Fix the logic to detect VIA analog low-current mode The analog low-current mode must be enabled when the no stream is running but the current detection checks it in a wrong way. Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=741128 Cc: [v3.1+] Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index fb1f0ff..de43cd9 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1051,7 +1051,7 @@ static void analog_low_current_mode(struct hda_codec *codec) bool enable; unsigned int verb, parm; - enable = is_aa_path_mute(codec) && (spec->opened_streams != 0); + enable = is_aa_path_mute(codec) && !spec->opened_streams; /* decide low current mode's verb & parameter */ switch (spec->codec_type) { -- cgit v1.1 From e9d010c2e8f03952e67a6fd8aed0f0dc92084ccc Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 1 Feb 2012 10:33:23 +0100 Subject: ALSA: hda - Allow analog low-current mode when dynamic power-control is on VIA codecs have several different power-saving features, and one of them is the analog low-current mode. But it turned out that the ALC mode causes pop-noises at each on/off time on some machines. As a quick workaround, disable the ALC when another power-saving feature, the dynamic pin power-control, is turned off, too, since the dynamic power-control is already exposed as a mixer enum element so that user can turn it on/off freely. Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=741128 Cc: [v3.1+] Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 27 +++++++++++++++++++++------ 1 file changed, 21 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index de43cd9..79166fb 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -199,6 +199,9 @@ struct via_spec { unsigned int no_pin_power_ctl; enum VIA_HDA_CODEC codec_type; + /* analog low-power control */ + bool alc_mode; + /* smart51 setup */ unsigned int smart51_nums; hda_nid_t smart51_pins[2]; @@ -758,6 +761,7 @@ static int via_pin_power_ctl_put(struct snd_kcontrol *kcontrol, return 0; spec->no_pin_power_ctl = val; set_widgets_power_state(codec); + analog_low_current_mode(codec); return 1; } @@ -1045,13 +1049,19 @@ static bool is_aa_path_mute(struct hda_codec *codec) } /* enter/exit analog low-current mode */ -static void analog_low_current_mode(struct hda_codec *codec) +static void __analog_low_current_mode(struct hda_codec *codec, bool force) { struct via_spec *spec = codec->spec; bool enable; unsigned int verb, parm; - enable = is_aa_path_mute(codec) && !spec->opened_streams; + if (spec->no_pin_power_ctl) + enable = false; + else + enable = is_aa_path_mute(codec) && !spec->opened_streams; + if (enable == spec->alc_mode && !force) + return; + spec->alc_mode = enable; /* decide low current mode's verb & parameter */ switch (spec->codec_type) { @@ -1083,6 +1093,11 @@ static void analog_low_current_mode(struct hda_codec *codec) snd_hda_codec_write(codec, codec->afg, 0, verb, parm); } +static void analog_low_current_mode(struct hda_codec *codec) +{ + return __analog_low_current_mode(codec, false); +} + /* * generic initialization of ADC, input mixers and output mixers */ @@ -1508,10 +1523,6 @@ static int via_build_controls(struct hda_codec *codec) return err; } - /* init power states */ - set_widgets_power_state(codec); - analog_low_current_mode(codec); - via_free_kctls(codec); /* no longer needed */ err = snd_hda_jack_add_kctls(codec, &spec->autocfg); @@ -2782,6 +2793,10 @@ static int via_init(struct hda_codec *codec) for (i = 0; i < spec->num_iverbs; i++) snd_hda_sequence_write(codec, spec->init_verbs[i]); + /* init power states */ + set_widgets_power_state(codec); + __analog_low_current_mode(codec, true); + via_auto_init_multi_out(codec); via_auto_init_hp_out(codec); via_auto_init_speaker_out(codec); -- cgit v1.1 From b5bcc189401c815988b7dd37611fc56f40c9139d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 2 Feb 2012 10:30:17 +0100 Subject: ALSA: hda - Disable dynamic-power control for VIA as default Since the dynamic pin power-control and the analog low-current mode may lead to pop-noise, it's safer to set it off as default. Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=741128 Cc: [v3.1+] Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 79166fb..284e311 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1470,6 +1470,7 @@ static int via_build_controls(struct hda_codec *codec) struct snd_kcontrol *kctl; int err, i; + spec->no_pin_power_ctl = 1; if (spec->set_widgets_power_state) if (!via_clone_control(spec, &via_pin_power_ctl_enum)) return -ENOMEM; -- cgit v1.1 From b544d1e0e233f83a2e6d20ee96b54ea272d5d5ba Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 3 Feb 2012 11:56:35 +0100 Subject: ALSA: hda/realtek - Add missing Bass and CLFE as vmaster slaves The recent changes in Realtek auto-parser added the new "Bass Speaker" and "CLFE" mixer elements which should be tracked as vmaster slaves, too. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=42720 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a8e82be..33b6077 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1855,6 +1855,8 @@ static const char * const alc_slave_vols[] = { "Speaker Playback Volume", "Mono Playback Volume", "Line-Out Playback Volume", + "CLFE Playback Volume", + "Bass Speaker Playback Volume", "PCM Playback Volume", NULL, }; @@ -1870,6 +1872,8 @@ static const char * const alc_slave_sws[] = { "Mono Playback Switch", "IEC958 Playback Switch", "Line-Out Playback Switch", + "CLFE Playback Switch", + "Bass Speaker Playback Switch", "PCM Playback Switch", NULL, }; -- cgit v1.1 From 226e01ef0da0b1a4c2c3922fb83ff3f9e4dfb508 Mon Sep 17 00:00:00 2001 From: Jesper Juhl Date: Sun, 5 Feb 2012 01:27:44 +0100 Subject: ALSA: emu8000: Remove duplicate linux/moduleparam.h include from emu8000_patch.c The header 'linux/moduleparam.h' is included twice in 'sound/isa/sb/emu8000_patch.c'. Once is enough. Signed-off-by: Jesper Juhl Signed-off-by: Takashi Iwai --- sound/isa/sb/emu8000_patch.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/isa/sb/emu8000_patch.c b/sound/isa/sb/emu8000_patch.c index e09f144..c99c607 100644 --- a/sound/isa/sb/emu8000_patch.c +++ b/sound/isa/sb/emu8000_patch.c @@ -22,7 +22,6 @@ #include "emu8000_local.h" #include #include -#include static int emu8000_reset_addr; module_param(emu8000_reset_addr, int, 0444); -- cgit v1.1 From eedec3d3854a390fc14008f265930f8c22b0373f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 6 Feb 2012 10:24:04 +0100 Subject: ALSA: hda/realtek - Fix a wrong condition sparse complains that "spec->multiout.dac_nids" is a pointer. sound/pci/hda/patch_realtek.c:2321:37: error: incompatible types for operation (>) sound/pci/hda/patch_realtek.c:2321:37: left side has type unsigned short const [usertype] *dac_nids sound/pci/hda/patch_realtek.c:2321:37: right side has type int It was meant to be num_dacs instead of dac_nids. Although the current code still works as expected (when num_dacs is zero, dac_nids should be NULL, too), better to fix now, of course. Reported-by: Dan Carpenter Cc: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 33b6077..485a837 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2322,7 +2322,7 @@ static int alc_build_pcms(struct hda_codec *codec) "%s Analog", codec->chip_name); info->name = spec->stream_name_analog; - if (spec->multiout.dac_nids > 0) { + if (spec->multiout.num_dacs > 0) { p = spec->stream_analog_playback; if (!p) p = &alc_pcm_analog_playback; -- cgit v1.1 From db966f8abb9ba74f7d5a7230f51572f52c31c4e5 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 6 Feb 2012 12:07:08 +0000 Subject: ASoC: wm8994: Enabling VMID should take a runtime PM reference We can enable VMID independently of the bias in some use cases so we need to ensure that the core device is powered up. Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm8994.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 8623950..81795eb 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -770,6 +770,8 @@ static void vmid_reference(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + pm_runtime_get_sync(codec->dev); + wm8994->vmid_refcount++; dev_dbg(codec->dev, "Referencing VMID, refcount is now %d\n", @@ -837,6 +839,8 @@ static void vmid_dereference(struct snd_soc_codec *codec) WM8994_VMID_BUF_ENA | WM8994_VMID_RAMP_MASK, 0); } + + pm_runtime_put(codec->dev); } static int vmid_event(struct snd_soc_dapm_widget *w, -- cgit v1.1 From b97f6bfdd1af95681de5a9f652da644a6525e376 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 Feb 2012 11:00:53 +0100 Subject: ALSA: hda - Fix error handling in patch_ca0132.c In patch_ca0132.c, the error returned from chipio_write() isn't checked always. Also, the power-up/down sequence isn't tracked properly in some error paths. Reported-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0132.c | 33 +++++++++++++++++++-------------- 1 file changed, 19 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 35abe3c..21d91d5 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -728,18 +728,19 @@ static int ca0132_hp_switch_put(struct snd_kcontrol *kcontrol, err = chipio_read(codec, REG_CODEC_MUTE, &data); if (err < 0) - return err; + goto exit; /* *valp 0 is mute, 1 is unmute */ data = (data & 0x7f) | (*valp ? 0 : 0x80); - chipio_write(codec, REG_CODEC_MUTE, data); + err = chipio_write(codec, REG_CODEC_MUTE, data); if (err < 0) - return err; + goto exit; spec->curr_hp_switch = *valp; + exit: snd_hda_power_down(codec); - return 1; + return err < 0 ? err : 1; } static int ca0132_speaker_switch_get(struct snd_kcontrol *kcontrol, @@ -770,18 +771,19 @@ static int ca0132_speaker_switch_put(struct snd_kcontrol *kcontrol, err = chipio_read(codec, REG_CODEC_MUTE, &data); if (err < 0) - return err; + goto exit; /* *valp 0 is mute, 1 is unmute */ data = (data & 0xef) | (*valp ? 0 : 0x10); - chipio_write(codec, REG_CODEC_MUTE, data); + err = chipio_write(codec, REG_CODEC_MUTE, data); if (err < 0) - return err; + goto exit; spec->curr_speaker_switch = *valp; + exit: snd_hda_power_down(codec); - return 1; + return err < 0 ? err : 1; } static int ca0132_hp_volume_get(struct snd_kcontrol *kcontrol, @@ -819,25 +821,26 @@ static int ca0132_hp_volume_put(struct snd_kcontrol *kcontrol, err = chipio_read(codec, REG_CODEC_HP_VOL_L, &data); if (err < 0) - return err; + goto exit; val = 31 - left_vol; data = (data & 0xe0) | val; - chipio_write(codec, REG_CODEC_HP_VOL_L, data); + err = chipio_write(codec, REG_CODEC_HP_VOL_L, data); if (err < 0) - return err; + goto exit; val = 31 - right_vol; data = (data & 0xe0) | val; - chipio_write(codec, REG_CODEC_HP_VOL_R, data); + err = chipio_write(codec, REG_CODEC_HP_VOL_R, data); if (err < 0) - return err; + goto exit; spec->curr_hp_volume[0] = left_vol; spec->curr_hp_volume[1] = right_vol; + exit: snd_hda_power_down(codec); - return 1; + return err < 0 ? err : 1; } static int add_hp_switch(struct hda_codec *codec, hda_nid_t nid) @@ -936,6 +939,8 @@ static int ca0132_build_controls(struct hda_codec *codec) if (err < 0) return err; err = add_in_volume(codec, spec->dig_in, "IEC958"); + if (err < 0) + return err; } return 0; } -- cgit v1.1 From 416846d2b31fc740ed9d5a5ec116964fb43c4358 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 7 Feb 2012 14:18:14 +0100 Subject: ALSA: hda - add support for Uniwill ECS M31EI notebook This hardware requires same fixup for the node 0x0f like Asus A6Rp. More information: https://bugzilla.redhat.com/show_bug.cgi?id=785417 Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 485a837..9350f3c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5627,6 +5627,7 @@ static const struct alc_fixup alc861_fixups[] = { static const struct snd_pci_quirk alc861_fixup_tbl[] = { SND_PCI_QUIRK_VENDOR(0x1043, "ASUS laptop", PINFIX_ASUS_A6RP), + SND_PCI_QUIRK(0x1584, 0x0000, "Uniwill ECS M31EI", PINFIX_ASUS_A6RP), SND_PCI_QUIRK(0x1584, 0x2b01, "Haier W18", PINFIX_ASUS_A6RP), SND_PCI_QUIRK(0x1734, 0x10c7, "FSC Amilo Pi1505", PINFIX_FSC_AMILO_PI1505), {} -- cgit v1.1 From 927c9423dd5f2d1c0b93d5e694ab84b4a5559713 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sat, 4 Feb 2012 20:51:43 +0100 Subject: ALSA: usb-audio: add Edirol UM-3G support Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/usb/quirks-table.h | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 8edc503..d89ab4c 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1618,6 +1618,14 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { + /* Edirol UM-3G */ + USB_DEVICE_VENDOR_SPEC(0x0582, 0x0108), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .ifnum = 0, + .type = QUIRK_MIDI_STANDARD_INTERFACE + } +}, +{ /* Boss JS-8 Jam Station */ USB_DEVICE(0x0582, 0x0109), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { -- cgit v1.1 From 2492250e4412c6411324c14ab289629360640b0a Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sat, 4 Feb 2012 20:56:47 +0100 Subject: ALSA: oxygen, virtuoso: fix exchanged L/R volumes of aux and CD inputs The driver accidentally exchanged the left/right fields for stereo AC'97 mixer registers. This affected only the aux and CD inputs because the line input bypasses the AC'97 codec and the mic input is mono; cards without AC'97 (Xonar DS/DG/HDAV Slim, HG2PCI, HiFier) were not affected. Reported-and-tested-by: Abby Cedar Signed-off-by: Clemens Ladisch Cc: 2.6.31+ Signed-off-by: Takashi Iwai --- sound/pci/oxygen/oxygen_mixer.c | 25 ++++++++++++++----------- 1 file changed, 14 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index 26c7e8b..c0dbb52 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -618,9 +618,12 @@ static int ac97_volume_get(struct snd_kcontrol *ctl, mutex_lock(&chip->mutex); reg = oxygen_read_ac97(chip, codec, index); mutex_unlock(&chip->mutex); - value->value.integer.value[0] = 31 - (reg & 0x1f); - if (stereo) - value->value.integer.value[1] = 31 - ((reg >> 8) & 0x1f); + if (!stereo) { + value->value.integer.value[0] = 31 - (reg & 0x1f); + } else { + value->value.integer.value[0] = 31 - ((reg >> 8) & 0x1f); + value->value.integer.value[1] = 31 - (reg & 0x1f); + } return 0; } @@ -636,14 +639,14 @@ static int ac97_volume_put(struct snd_kcontrol *ctl, mutex_lock(&chip->mutex); oldreg = oxygen_read_ac97(chip, codec, index); - newreg = oldreg; - newreg = (newreg & ~0x1f) | - (31 - (value->value.integer.value[0] & 0x1f)); - if (stereo) - newreg = (newreg & ~0x1f00) | - ((31 - (value->value.integer.value[1] & 0x1f)) << 8); - else - newreg = (newreg & ~0x1f00) | ((newreg & 0x1f) << 8); + if (!stereo) { + newreg = oldreg & ~0x1f; + newreg |= 31 - (value->value.integer.value[0] & 0x1f); + } else { + newreg = oldreg & ~0x1f1f; + newreg |= (31 - (value->value.integer.value[0] & 0x1f)) << 8; + newreg |= 31 - (value->value.integer.value[1] & 0x1f); + } change = newreg != oldreg; if (change) oxygen_write_ac97(chip, codec, index, newreg); -- cgit v1.1 From f647e1526fd6c7c8ab720781c40d11e11f930e93 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 7 Feb 2012 17:24:19 +0000 Subject: ASoC: wm8994: Fix typo in VMID ramp setting The VMID ramp rate is supposed to be 0x3, not 11b. Fix that. Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm8994.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 81795eb..b75a652 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -785,7 +785,7 @@ static void vmid_reference(struct snd_soc_codec *codec) WM8994_VMID_RAMP_MASK, WM8994_STARTUP_BIAS_ENA | WM8994_VMID_BUF_ENA | - (0x11 << WM8994_VMID_RAMP_SHIFT)); + (0x3 << WM8994_VMID_RAMP_SHIFT)); /* Main bias enable, VMID=2x40k */ snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1, -- cgit v1.1 From a7c4183be2d6a7da8c97a9b671b5f3aed321127e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 7 Feb 2012 14:18:29 +0000 Subject: ASoC: wm8994: Disable line output discharge prior to ramping VMID Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index b75a652..ec69a6c 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -787,6 +787,11 @@ static void vmid_reference(struct snd_soc_codec *codec) WM8994_VMID_BUF_ENA | (0x3 << WM8994_VMID_RAMP_SHIFT)); + /* Remove discharge for line out */ + snd_soc_update_bits(codec, WM8994_ANTIPOP_1, + WM8994_LINEOUT1_DISCH | + WM8994_LINEOUT2_DISCH, 0); + /* Main bias enable, VMID=2x40k */ snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1, WM8994_BIAS_ENA | -- cgit v1.1 From a1e0c3cf7fb07227fe1f26161d969101dba78287 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 9 Feb 2012 09:32:19 +0100 Subject: ALSA: hda - Fix mute-LED VREF value for new HP laptops The new HP laptops turns off the mute LED with VREF50 or VREF80, but not in HIZ unlike the previous models. Since VREF50 (also 80) works with the previous models, let's use VREF50 for all. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 948f0be..6345df1 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -5078,9 +5078,9 @@ static int stac92xx_update_led_status(struct hda_codec *codec) spec->gpio_dir, spec->gpio_data); } else { notmtd_lvl = spec->gpio_led_polarity ? - AC_PINCTL_VREF_HIZ : AC_PINCTL_VREF_GRD; + AC_PINCTL_VREF_50 : AC_PINCTL_VREF_GRD; muted_lvl = spec->gpio_led_polarity ? - AC_PINCTL_VREF_GRD : AC_PINCTL_VREF_HIZ; + AC_PINCTL_VREF_GRD : AC_PINCTL_VREF_50; spec->vref_led = muted ? muted_lvl : notmtd_lvl; stac_vrefout_set(codec, spec->vref_mute_led_nid, spec->vref_led); -- cgit v1.1 From 1987877d869027ab63dc9df515e11f19279a8091 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 8 Feb 2012 16:57:29 -0800 Subject: ASoC: fsi: fixup fsi_pointer() calculation method current fsi_pointer() calculation was not correct for FSI driver. This patch fix it up. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index db6c89a..ea4a82d0 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1152,12 +1152,8 @@ static snd_pcm_uframes_t fsi_pointer(struct snd_pcm_substream *substream) { struct fsi_priv *fsi = fsi_get_priv(substream); struct fsi_stream *io = fsi_get_stream(fsi, fsi_is_play(substream)); - int samples_pos = io->buff_sample_pos - 1; - if (samples_pos < 0) - samples_pos = 0; - - return fsi_sample2frame(fsi, samples_pos); + return fsi_sample2frame(fsi, io->buff_sample_pos); } static struct snd_pcm_ops fsi_pcm_ops = { -- cgit v1.1 From fc1156c0b0f7ad45ec03d919866349eeca2bf18c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 13 Feb 2012 15:04:06 +0100 Subject: ALSA: hda - Fix initialization of secondary capture source on VT1705 VT1705 codec has two ADCs where the secondary ADC has no MUX but only a fixed connection to the mic pin. This confused the driver and it tries always overriding the input-source selection by assumption of the existing MUX for the secondary ADC, resulted in resetting the input-source at each time PM (including power-saving) occurs. The fix is simply to check the existence of MUX for secondary ADCs in the initialization code. Tested-by: Anisse Astier Cc: [v3.1+] Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 284e311..dff9a00 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -666,6 +666,9 @@ static void via_auto_init_analog_input(struct hda_codec *codec) /* init input-src */ for (i = 0; i < spec->num_adc_nids; i++) { int adc_idx = spec->inputs[spec->cur_mux[i]].adc_idx; + /* secondary ADCs must have the unique MUX */ + if (i > 0 && !spec->mux_nids[i]) + break; if (spec->mux_nids[adc_idx]) { int mux_idx = spec->inputs[spec->cur_mux[i]].mux_idx; snd_hda_codec_write(codec, spec->mux_nids[adc_idx], 0, -- cgit v1.1 From 02a237b24d57e2e2d5402c92549e9e792aa24359 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 13 Feb 2012 15:25:07 +0100 Subject: ALSA: hda - Fix silent speaker output on Acer Aspire 6935 Since 3.2 kernel, the driver starts trying to assign the multi-io DACs before the speaker, thus it assigns DAC2/3 for multi-io and DAC4 for the speaker for a standard laptop setup like a HP, a speaker, a mic-in and a line-in. However, on Acer Aspire 6935, it seems that the speaker pin 0x14 must be connected with either DAC1 or 2; otherwise it results in silence by some reason, although the codec itself allows the routing to DAC3/4. As a workaround, the connection list of each pin is reduced to be mapped to either only DAC1/2 or DAC3/4, so that the compatible assignment as in kernel 3.1 is achieved. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=42740 Cc: [v3.2+] Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 23 +++++++++++++++++++++++ 1 file changed, 23 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1d07e8f..c4bde71 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4201,8 +4201,26 @@ enum { PINFIX_PB_M5210, PINFIX_ACER_ASPIRE_7736, PINFIX_ASUS_W90V, + ALC889_FIXUP_DAC_ROUTE, }; +/* Fix the connection of some pins for ALC889: + * At least, Acer Aspire 5935 shows the connections to DAC3/4 don't + * work correctly (bko#42740) + */ +static void alc889_fixup_dac_route(struct hda_codec *codec, + const struct alc_fixup *fix, int action) +{ + if (action == ALC_FIXUP_ACT_PRE_PROBE) { + hda_nid_t conn1[2] = { 0x0c, 0x0d }; + hda_nid_t conn2[2] = { 0x0e, 0x0f }; + snd_hda_override_conn_list(codec, 0x14, 2, conn1); + snd_hda_override_conn_list(codec, 0x15, 2, conn1); + snd_hda_override_conn_list(codec, 0x18, 2, conn2); + snd_hda_override_conn_list(codec, 0x1a, 2, conn2); + } +} + static const struct alc_fixup alc882_fixups[] = { [PINFIX_ABIT_AW9D_MAX] = { .type = ALC_FIXUP_PINS, @@ -4239,10 +4257,15 @@ static const struct alc_fixup alc882_fixups[] = { { } } }, + [ALC889_FIXUP_DAC_ROUTE] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc889_fixup_dac_route, + }, }; static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0155, "Packard-Bell M5120", PINFIX_PB_M5210), + SND_PCI_QUIRK(0x1025, 0x0259, "Acer Aspire 5935", ALC889_FIXUP_DAC_ROUTE), SND_PCI_QUIRK(0x1043, 0x1873, "ASUS W90V", PINFIX_ASUS_W90V), SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Y530", PINFIX_LENOVO_Y530), SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", PINFIX_ABIT_AW9D_MAX), -- cgit v1.1 From 27c3afe6e1cf129faac90405121203962da08ff4 Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Mon, 13 Feb 2012 23:44:22 -0500 Subject: ALSA: intel8x0: Fix default inaudible sound on Gateway M520 BugLink: https://bugs.launchpad.net/bugs/930842 The reporter states that audio is inaudible by default without muting 'External Amplifier'. Add a quirk to handle his SSID so that changing the control is not necessary. Reported-and-tested-by: Benjamin Carlson Cc: Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/intel8x0.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 9f3b01b..e0a4263 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -2102,6 +2102,12 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = { }, { .subvendor = 0x161f, + .subdevice = 0x202f, + .name = "Gateway M520", + .type = AC97_TUNE_INV_EAPD + }, + { + .subvendor = 0x161f, .subdevice = 0x203a, .name = "Gateway 4525GZ", /* AD1981B */ .type = AC97_TUNE_INV_EAPD -- cgit v1.1 From 31794bc37bf2db84f085da52b72bfba65739b2d2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 13 Feb 2012 22:00:47 -0800 Subject: ASoC: wm8962: Fix sidetone enumeration texts The sidetone enumeration texts have left and right swapped. Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm8962.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 29c4b02..0ac228b 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2564,7 +2564,7 @@ static int dsp2_event(struct snd_soc_dapm_widget *w, return 0; } -static const char *st_text[] = { "None", "Right", "Left" }; +static const char *st_text[] = { "None", "Left", "Right" }; static const struct soc_enum str_enum = SOC_ENUM_SINGLE(WM8962_DAC_DSP_MIXING_1, 2, 3, st_text); -- cgit v1.1 From 8866f405efd4171f9d9c91901d2dd02f01bacb60 Mon Sep 17 00:00:00 2001 From: Xi Wang Date: Tue, 14 Feb 2012 05:18:48 -0500 Subject: ALSA: usb-audio: avoid integer overflow in create_fixed_stream_quirk() A malicious USB device could feed in a large nr_rates value. This would cause the subsequent call to kmemdup() to allocate a smaller buffer than expected, leading to out-of-bounds access. This patch validates the nr_rates value and reuses the limit introduced in commit 4fa0e81b ("ALSA: usb-audio: fix possible hang and overflow in parse_uac2_sample_rate_range()"). Signed-off-by: Xi Wang Signed-off-by: Takashi Iwai --- sound/usb/card.h | 1 + sound/usb/format.c | 4 +--- sound/usb/quirks.c | 6 +++++- 3 files changed, 7 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/usb/card.h b/sound/usb/card.h index a39edcc..da5fa1a 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -1,6 +1,7 @@ #ifndef __USBAUDIO_CARD_H #define __USBAUDIO_CARD_H +#define MAX_NR_RATES 1024 #define MAX_PACKS 20 #define MAX_PACKS_HS (MAX_PACKS * 8) /* in high speed mode */ #define MAX_URBS 8 diff --git a/sound/usb/format.c b/sound/usb/format.c index e09aba1..ddfef57 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -209,8 +209,6 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof return 0; } -#define MAX_UAC2_NR_RATES 1024 - /* * Helper function to walk the array of sample rate triplets reported by * the device. The problem is that we need to parse whole array first to @@ -255,7 +253,7 @@ static int parse_uac2_sample_rate_range(struct audioformat *fp, int nr_triplets, fp->rates |= snd_pcm_rate_to_rate_bit(rate); nr_rates++; - if (nr_rates >= MAX_UAC2_NR_RATES) { + if (nr_rates >= MAX_NR_RATES) { snd_printk(KERN_ERR "invalid uac2 rates\n"); break; } diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index a3ddac0..2781726 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -132,10 +132,14 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip, unsigned *rate_table = NULL; fp = kmemdup(quirk->data, sizeof(*fp), GFP_KERNEL); - if (! fp) { + if (!fp) { snd_printk(KERN_ERR "cannot memdup\n"); return -ENOMEM; } + if (fp->nr_rates > MAX_NR_RATES) { + kfree(fp); + return -EINVAL; + } if (fp->nr_rates > 0) { rate_table = kmemdup(fp->rate_table, sizeof(int) * fp->nr_rates, GFP_KERNEL); -- cgit v1.1 From c14c95f62ecb8710af14ae0d48e01991b70bb6f4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 16 Feb 2012 16:38:07 +0100 Subject: ALSA: hda/realtek - Fix overflow of vol/sw check bitmap The bitmap introduced in the commit [527e4d73: ALSA: hda/realtek - Fix missing volume controls with ALC260] is too narrow for some codecs, which may have more NIDs than 0x20, thus it may overflow the bitmap array on them. Just double the number to cover all and also add a sanity-check code to be safer. Cc: [v3.2+] Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1358987..389a28a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -80,6 +80,8 @@ enum { ALC_AUTOMUTE_MIXER, /* mute/unmute mixer widget AMP */ }; +#define MAX_VOL_NIDS 0x40 + struct alc_spec { /* codec parameterization */ const struct snd_kcontrol_new *mixers[5]; /* mixer arrays */ @@ -118,8 +120,8 @@ struct alc_spec { const hda_nid_t *capsrc_nids; hda_nid_t dig_in_nid; /* digital-in NID; optional */ hda_nid_t mixer_nid; /* analog-mixer NID */ - DECLARE_BITMAP(vol_ctls, 0x20 << 1); - DECLARE_BITMAP(sw_ctls, 0x20 << 1); + DECLARE_BITMAP(vol_ctls, MAX_VOL_NIDS << 1); + DECLARE_BITMAP(sw_ctls, MAX_VOL_NIDS << 1); /* capture setup for dynamic dual-adc switch */ hda_nid_t cur_adc; @@ -3149,7 +3151,10 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec) static inline unsigned int get_ctl_pos(unsigned int data) { hda_nid_t nid = get_amp_nid_(data); - unsigned int dir = get_amp_direction_(data); + unsigned int dir; + if (snd_BUG_ON(nid >= MAX_VOL_NIDS)) + return 0; + dir = get_amp_direction_(data); return (nid << 1) | dir; } -- cgit v1.1 From ef8d60fb79614a86a82720dc2402631dbcafb315 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 17 Feb 2012 10:12:38 +0100 Subject: ALSA: hda/realtek - Fix surround output regression on Acer Aspire 5935 The previous fix for the speaker on Acer Aspire 59135 introduced another problem for surround outputs. It changed the connections on the line-in/mic pins for limiting the routes, but it left the modified connections. Thus wrong connection indices were written when set to 4ch or 6ch mode. This patch fixes it by restoring the right connections just after parsing the tree but before the initialization. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=42740 Cc: [v3.2+] Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 389a28a..3647baa 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4441,12 +4441,20 @@ static void alc889_fixup_dac_route(struct hda_codec *codec, const struct alc_fixup *fix, int action) { if (action == ALC_FIXUP_ACT_PRE_PROBE) { + /* fake the connections during parsing the tree */ hda_nid_t conn1[2] = { 0x0c, 0x0d }; hda_nid_t conn2[2] = { 0x0e, 0x0f }; snd_hda_override_conn_list(codec, 0x14, 2, conn1); snd_hda_override_conn_list(codec, 0x15, 2, conn1); snd_hda_override_conn_list(codec, 0x18, 2, conn2); snd_hda_override_conn_list(codec, 0x1a, 2, conn2); + } else if (action == ALC_FIXUP_ACT_PROBE) { + /* restore the connections */ + hda_nid_t conn[5] = { 0x0c, 0x0d, 0x0e, 0x0f, 0x26 }; + snd_hda_override_conn_list(codec, 0x14, 5, conn); + snd_hda_override_conn_list(codec, 0x15, 5, conn); + snd_hda_override_conn_list(codec, 0x18, 5, conn); + snd_hda_override_conn_list(codec, 0x1a, 5, conn); } } -- cgit v1.1 From e555cf363167f09efae96d32a363e24c4de16b7b Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 20 Feb 2012 20:14:16 -0800 Subject: ASoC: ak4642: fixup HeadPhone L/R dapm settings Current ak4642 driver had wrong dapm settings for headphone L/R. If you select headphone L, and select R after that, headphone L setting was removed by R settings. This patch fixes it up. It provides just "Headphone Enable" to user side Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/codecs/ak4642.c | 31 ++++++++++++++++--------------- 1 file changed, 16 insertions(+), 15 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 5ef70b5..278c0a0 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -146,13 +146,10 @@ static const struct snd_kcontrol_new ak4642_snd_controls[] = { SOC_DOUBLE_R_TLV("Digital Playback Volume", L_DVC, R_DVC, 0, 0xFF, 1, out_tlv), - - SOC_SINGLE("Headphone Switch", PW_MGMT2, 6, 1, 0), }; -static const struct snd_kcontrol_new ak4642_hpout_mixer_controls[] = { - SOC_DAPM_SINGLE("DACH", MD_CTL4, 0, 1, 0), -}; +static const struct snd_kcontrol_new ak4642_headphone_control = + SOC_DAPM_SINGLE("Switch", PW_MGMT2, 6, 1, 0); static const struct snd_kcontrol_new ak4642_lout_mixer_controls[] = { SOC_DAPM_SINGLE("DACL", SG_SL1, 4, 1, 0), @@ -165,13 +162,12 @@ static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("HPOUTR"), SND_SOC_DAPM_OUTPUT("LINEOUT"), - SND_SOC_DAPM_MIXER("HPOUTL Mixer", PW_MGMT2, 5, 0, - &ak4642_hpout_mixer_controls[0], - ARRAY_SIZE(ak4642_hpout_mixer_controls)), + SND_SOC_DAPM_PGA("HPL Out", PW_MGMT2, 5, 0, NULL, 0), + SND_SOC_DAPM_PGA("HPR Out", PW_MGMT2, 4, 0, NULL, 0), + SND_SOC_DAPM_SWITCH("Headphone Enable", SND_SOC_NOPM, 0, 0, + &ak4642_headphone_control), - SND_SOC_DAPM_MIXER("HPOUTR Mixer", PW_MGMT2, 4, 0, - &ak4642_hpout_mixer_controls[0], - ARRAY_SIZE(ak4642_hpout_mixer_controls)), + SND_SOC_DAPM_PGA("DACH", MD_CTL4, 0, 0, NULL, 0), SND_SOC_DAPM_MIXER("LINEOUT Mixer", PW_MGMT1, 3, 0, &ak4642_lout_mixer_controls[0], @@ -184,12 +180,17 @@ static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = { static const struct snd_soc_dapm_route ak4642_intercon[] = { /* Outputs */ - {"HPOUTL", NULL, "HPOUTL Mixer"}, - {"HPOUTR", NULL, "HPOUTR Mixer"}, + {"HPOUTL", NULL, "HPL Out"}, + {"HPOUTR", NULL, "HPR Out"}, {"LINEOUT", NULL, "LINEOUT Mixer"}, - {"HPOUTL Mixer", "DACH", "DAC"}, - {"HPOUTR Mixer", "DACH", "DAC"}, + {"HPL Out", NULL, "Headphone Enable"}, + {"HPR Out", NULL, "Headphone Enable"}, + + {"Headphone Enable", "Switch", "DACH"}, + + {"DACH", NULL, "DAC"}, + {"LINEOUT Mixer", "DACL", "DAC"}, }; -- cgit v1.1 From cb74eb15ac88d6aacf7e58db1d8f8dadee710fd9 Mon Sep 17 00:00:00 2001 From: Mark Hills Date: Tue, 21 Feb 2012 21:26:31 +0000 Subject: ALSA: snd-usb-caiaq: Fix the return of XRUN Commit 3702b08 added a lock, but did not account for the case of SNDRV_PCM_POS_XRUN, which would get immediately overwritten. This could be bundled into one if-else-if statement, but the goto helps to clarify the 'exceptional' case. Thanks to Andreas Pape for spotting this. Signed-off-by: Mark Hills Acked-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/caiaq/audio.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index 2cf87f5..fde9a7a 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -311,8 +311,10 @@ snd_usb_caiaq_pcm_pointer(struct snd_pcm_substream *sub) spin_lock(&dev->spinlock); - if (dev->input_panic || dev->output_panic) + if (dev->input_panic || dev->output_panic) { ptr = SNDRV_PCM_POS_XRUN; + goto unlock; + } if (sub->stream == SNDRV_PCM_STREAM_PLAYBACK) ptr = bytes_to_frames(sub->runtime, @@ -321,6 +323,7 @@ snd_usb_caiaq_pcm_pointer(struct snd_pcm_substream *sub) ptr = bytes_to_frames(sub->runtime, dev->audio_in_buf_pos[index]); +unlock: spin_unlock(&dev->spinlock); return ptr; } -- cgit v1.1