From bb362e2e4f4874f3fd4cbc2497385b9bceb3a08a Mon Sep 17 00:00:00 2001 From: Zeng Zhaoming Date: Wed, 18 Jan 2012 13:58:07 +0800 Subject: [PATCH 01/36] ASoC: sgtl5000: Fix wrong register name in restore Correct SGTL5000_CHIP_CLK_CTRL to SGTL5000_CHIP_REF_CTRL in sgtl5000_restore_regs(), and add comment to explain the restore order. Reported-by: Julia Lawall Signed-off-by: Zeng Zhaoming Acked-by: Dong Aisheng Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 17 +++++++++++++---- 1 file changed, 13 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index f8863ebb4304..7f4ba819a9f6 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -987,12 +987,12 @@ static int sgtl5000_restore_regs(struct snd_soc_codec *codec) /* restore regular registers */ for (reg = 0; reg <= SGTL5000_CHIP_SHORT_CTRL; reg += 2) { - /* this regs depends on the others */ + /* These regs should restore in particular order */ if (reg == SGTL5000_CHIP_ANA_POWER || reg == SGTL5000_CHIP_CLK_CTRL || reg == SGTL5000_CHIP_LINREG_CTRL || reg == SGTL5000_CHIP_LINE_OUT_CTRL || - reg == SGTL5000_CHIP_CLK_CTRL) + reg == SGTL5000_CHIP_REF_CTRL) continue; snd_soc_write(codec, reg, cache[reg]); @@ -1003,8 +1003,17 @@ static int sgtl5000_restore_regs(struct snd_soc_codec *codec) snd_soc_write(codec, reg, cache[reg]); /* - * restore power and other regs according - * to set_power() and set_clock() + * restore these regs according to the power setting sequence in + * sgtl5000_set_power_regs() and clock setting sequence in + * sgtl5000_set_clock(). + * + * The order of restore is: + * 1. SGTL5000_CHIP_CLK_CTRL MCLK_FREQ bits (1:0) should be restore after + * SGTL5000_CHIP_ANA_POWER PLL bits set + * 2. SGTL5000_CHIP_LINREG_CTRL should be set before + * SGTL5000_CHIP_ANA_POWER LINREG_D restored + * 3. SGTL5000_CHIP_REF_CTRL controls Analog Ground Voltage, + * prefer to resotre it after SGTL5000_CHIP_ANA_POWER restored */ snd_soc_write(codec, SGTL5000_CHIP_LINREG_CTRL, cache[SGTL5000_CHIP_LINREG_CTRL]); From 01b37e94c04bc6dae2c4837a2eb6fff6819ea82a Mon Sep 17 00:00:00 2001 From: Wolfram Sang Date: Wed, 18 Jan 2012 11:48:58 +0100 Subject: [PATCH 02/36] ASoC: tlv320aic32x4: always enable dividers Dividers (such as MDAC) are always needed, independent of the codec being I2S master or slave. Needed on a custom board where the codec has to be slave. Signed-off-by: Wolfram Sang Acked-by: Javier Martin Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic32x4.c | 82 ++++++++++++++------------------ 1 file changed, 36 insertions(+), 46 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index eb401ef021fb..3806cb6d9d4d 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -60,7 +60,6 @@ struct aic32x4_rate_divs { struct aic32x4_priv { u32 sysclk; - s32 master; u8 page_no; void *control_data; u32 power_cfg; @@ -369,7 +368,6 @@ static int aic32x4_set_dai_sysclk(struct snd_soc_dai *codec_dai, static int aic32x4_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; - struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); u8 iface_reg_1; u8 iface_reg_2; u8 iface_reg_3; @@ -384,11 +382,9 @@ static int aic32x4_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: - aic32x4->master = 1; iface_reg_1 |= AIC32X4_BCLKMASTER | AIC32X4_WCLKMASTER; break; case SND_SOC_DAIFMT_CBS_CFS: - aic32x4->master = 0; break; default: printk(KERN_ERR "aic32x4: invalid DAI master/slave interface\n"); @@ -526,64 +522,58 @@ static int aic32x4_mute(struct snd_soc_dai *dai, int mute) static int aic32x4_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); - switch (level) { case SND_SOC_BIAS_ON: - if (aic32x4->master) { - /* Switch on PLL */ - snd_soc_update_bits(codec, AIC32X4_PLLPR, - AIC32X4_PLLEN, AIC32X4_PLLEN); + /* Switch on PLL */ + snd_soc_update_bits(codec, AIC32X4_PLLPR, + AIC32X4_PLLEN, AIC32X4_PLLEN); - /* Switch on NDAC Divider */ - snd_soc_update_bits(codec, AIC32X4_NDAC, - AIC32X4_NDACEN, AIC32X4_NDACEN); + /* Switch on NDAC Divider */ + snd_soc_update_bits(codec, AIC32X4_NDAC, + AIC32X4_NDACEN, AIC32X4_NDACEN); - /* Switch on MDAC Divider */ - snd_soc_update_bits(codec, AIC32X4_MDAC, - AIC32X4_MDACEN, AIC32X4_MDACEN); + /* Switch on MDAC Divider */ + snd_soc_update_bits(codec, AIC32X4_MDAC, + AIC32X4_MDACEN, AIC32X4_MDACEN); - /* Switch on NADC Divider */ - snd_soc_update_bits(codec, AIC32X4_NADC, - AIC32X4_NADCEN, AIC32X4_NADCEN); + /* Switch on NADC Divider */ + snd_soc_update_bits(codec, AIC32X4_NADC, + AIC32X4_NADCEN, AIC32X4_NADCEN); - /* Switch on MADC Divider */ - snd_soc_update_bits(codec, AIC32X4_MADC, - AIC32X4_MADCEN, AIC32X4_MADCEN); + /* Switch on MADC Divider */ + snd_soc_update_bits(codec, AIC32X4_MADC, + AIC32X4_MADCEN, AIC32X4_MADCEN); - /* Switch on BCLK_N Divider */ - snd_soc_update_bits(codec, AIC32X4_BCLKN, - AIC32X4_BCLKEN, AIC32X4_BCLKEN); - } + /* Switch on BCLK_N Divider */ + snd_soc_update_bits(codec, AIC32X4_BCLKN, + AIC32X4_BCLKEN, AIC32X4_BCLKEN); break; case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: - if (aic32x4->master) { - /* Switch off PLL */ - snd_soc_update_bits(codec, AIC32X4_PLLPR, - AIC32X4_PLLEN, 0); + /* Switch off PLL */ + snd_soc_update_bits(codec, AIC32X4_PLLPR, + AIC32X4_PLLEN, 0); - /* Switch off NDAC Divider */ - snd_soc_update_bits(codec, AIC32X4_NDAC, - AIC32X4_NDACEN, 0); + /* Switch off NDAC Divider */ + snd_soc_update_bits(codec, AIC32X4_NDAC, + AIC32X4_NDACEN, 0); - /* Switch off MDAC Divider */ - snd_soc_update_bits(codec, AIC32X4_MDAC, - AIC32X4_MDACEN, 0); + /* Switch off MDAC Divider */ + snd_soc_update_bits(codec, AIC32X4_MDAC, + AIC32X4_MDACEN, 0); - /* Switch off NADC Divider */ - snd_soc_update_bits(codec, AIC32X4_NADC, - AIC32X4_NADCEN, 0); + /* Switch off NADC Divider */ + snd_soc_update_bits(codec, AIC32X4_NADC, + AIC32X4_NADCEN, 0); - /* Switch off MADC Divider */ - snd_soc_update_bits(codec, AIC32X4_MADC, - AIC32X4_MADCEN, 0); + /* Switch off MADC Divider */ + snd_soc_update_bits(codec, AIC32X4_MADC, + AIC32X4_MADCEN, 0); - /* Switch off BCLK_N Divider */ - snd_soc_update_bits(codec, AIC32X4_BCLKN, - AIC32X4_BCLKEN, 0); - } + /* Switch off BCLK_N Divider */ + snd_soc_update_bits(codec, AIC32X4_BCLKN, + AIC32X4_BCLKEN, 0); break; case SND_SOC_BIAS_OFF: break; From 0c93a167a6b3fa510c74e88477852c41defda075 Mon Sep 17 00:00:00 2001 From: Wolfram Sang Date: Wed, 18 Jan 2012 11:48:59 +0100 Subject: [PATCH 03/36] ASoC: tlv320aic32x4: always enable analouge block Register LDOCTLEN must always be initialized to clear the analog power control bit, otherwise the analog block will stay deactivated. Signed-off-by: Wolfram Sang Acked-by: Javier Martin Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic32x4.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 3806cb6d9d4d..372b0b83bd9f 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -641,9 +641,11 @@ static int aic32x4_probe(struct snd_soc_codec *codec) if (aic32x4->power_cfg & AIC32X4_PWR_AVDD_DVDD_WEAK_DISABLE) { snd_soc_write(codec, AIC32X4_PWRCFG, AIC32X4_AVDDWEAKDISABLE); } - if (aic32x4->power_cfg & AIC32X4_PWR_AIC32X4_LDO_ENABLE) { - snd_soc_write(codec, AIC32X4_LDOCTL, AIC32X4_LDOCTLEN); - } + + tmp_reg = (aic32x4->power_cfg & AIC32X4_PWR_AIC32X4_LDO_ENABLE) ? + AIC32X4_LDOCTLEN : 0; + snd_soc_write(codec, AIC32X4_LDOCTL, tmp_reg); + tmp_reg = snd_soc_read(codec, AIC32X4_CMMODE); if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_LDOIN_RANGE_18_36) { tmp_reg |= AIC32X4_LDOIN_18_36; From e53e417331c57b9b97e3f8be870214a02c99265c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 18 Jan 2012 20:02:38 +0000 Subject: [PATCH 04/36] ASoC: Mark WM5100 register map cache only when going into BIAS_OFF Writing to the registers won't work if we do actually manage to hit a fully powered off state. Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/wm5100.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 8b24323d6b2c..3fd9cfe6dcd7 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -1402,6 +1402,7 @@ static int wm5100_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_OFF: + regcache_cache_only(wm5100->regmap, true); if (wm5100->pdata.ldo_ena) gpio_set_value_cansleep(wm5100->pdata.ldo_ena, 0); regulator_bulk_disable(ARRAY_SIZE(wm5100->core_supplies), From 495174a8ffbaa0d15153d855cf206cdc46d51cf4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 19 Jan 2012 11:16:37 +0000 Subject: [PATCH 05/36] ASoC: Don't go through cache when applying WM5100 rev A updates These are all to either uncached registers or fixes to register defaults, in the former case the cache won't do anything and in the latter case we're fixing things so the cache sync will do the right thing. Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/wm5100.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 3fd9cfe6dcd7..66f0611e68b6 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -1377,6 +1377,7 @@ static int wm5100_set_bias_level(struct snd_soc_codec *codec, switch (wm5100->rev) { case 0: + regcache_cache_bypass(wm5100->regmap, true); snd_soc_write(codec, 0x11, 0x3); snd_soc_write(codec, 0x203, 0xc); snd_soc_write(codec, 0x206, 0); @@ -1392,6 +1393,7 @@ static int wm5100_set_bias_level(struct snd_soc_codec *codec, snd_soc_write(codec, wm5100_reva_patches[i].reg, wm5100_reva_patches[i].val); + regcache_cache_bypass(wm5100->regmap, false); break; default: break; From fed22007113cb857e917913ce016d9b539dc3a80 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 18 Jan 2012 19:17:06 +0000 Subject: [PATCH 06/36] ASoC: Disable register synchronisation for low frequency WM8996 SYSCLK With a low frequency SYSCLK and a fast I2C clock register synchronisation may occasionally take too long to take effect, causing I/O issues. Disable synchronisation in order to avoid any issues. Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/wm8996.c | 4 ++++ sound/soc/codecs/wm8996.h | 4 ++++ 2 files changed, 8 insertions(+) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index d8da10fe5b52..86f5b6bd7af2 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2007,6 +2007,7 @@ static int wm8996_set_sysclk(struct snd_soc_dai *dai, struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); int lfclk = 0; int ratediv = 0; + int sync = WM8996_REG_SYNC; int src; int old; @@ -2051,6 +2052,7 @@ static int wm8996_set_sysclk(struct snd_soc_dai *dai, case 32000: case 32768: lfclk = WM8996_LFCLK_ENA; + sync = 0; break; default: dev_warn(codec->dev, "Unsupported clock rate %dHz\n", @@ -2064,6 +2066,8 @@ static int wm8996_set_sysclk(struct snd_soc_dai *dai, WM8996_SYSCLK_SRC_MASK | WM8996_SYSCLK_DIV_MASK, src << WM8996_SYSCLK_SRC_SHIFT | ratediv); snd_soc_update_bits(codec, WM8996_CLOCKING_1, WM8996_LFCLK_ENA, lfclk); + snd_soc_update_bits(codec, WM8996_CONTROL_INTERFACE_1, + WM8996_REG_SYNC, sync); snd_soc_update_bits(codec, WM8996_AIF_CLOCKING_1, WM8996_SYSCLK_ENA, old); diff --git a/sound/soc/codecs/wm8996.h b/sound/soc/codecs/wm8996.h index 0fde643194ce..de9ac3e44aec 100644 --- a/sound/soc/codecs/wm8996.h +++ b/sound/soc/codecs/wm8996.h @@ -1567,6 +1567,10 @@ int wm8996_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, /* * R257 (0x101) - Control Interface (1) */ +#define WM8996_REG_SYNC 0x8000 /* REG_SYNC */ +#define WM8996_REG_SYNC_MASK 0x8000 /* REG_SYNC */ +#define WM8996_REG_SYNC_SHIFT 15 /* REG_SYNC */ +#define WM8996_REG_SYNC_WIDTH 1 /* REG_SYNC */ #define WM8996_AUTO_INC 0x0004 /* AUTO_INC */ #define WM8996_AUTO_INC_MASK 0x0004 /* AUTO_INC */ #define WM8996_AUTO_INC_SHIFT 2 /* AUTO_INC */ From 6b35f924b80a0e6d71711e66f5b3c16f427f3d2a Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 19 Jan 2012 10:23:22 -0200 Subject: [PATCH 07/36] ASoC: mxs: Fix mxs-saif timeout On a mx28evk board the following errors happens on mxs-sgtl5000 probe: [ 0.660000] saif0_clk_set_rate: divider writing timeout [ 0.670000] mxs-sgtl5000: probe of mxs-sgtl5000.0 failed with error -110 [ 0.670000] ALSA device list: [ 0.680000] No soundcards found. This timeout happens because clk_set_rate will result in writing to the DIV bits of register HW_CLKCTRL_SAIF0 with the saif clock gated (CLKGATE bit set to one). MX28 Reference states the following about CLKGATE: "The DIV field can change ONLY when this clock gate bit field is low." So call clk_prepare_enable prior to clk_set_rate to fix this problem. After this change the mxs-saif driver can be correctly probed and audio is functional. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/mxs/mxs-saif.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index dccfb37a9626..f204dbac11d4 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -124,6 +124,8 @@ static int mxs_saif_set_clk(struct mxs_saif *saif, * * If MCLK is not used, we just set saif clk to 512*fs. */ + clk_prepare_enable(master_saif->clk); + if (master_saif->mclk_in_use) { if (mclk % 32 == 0) { scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE; @@ -133,6 +135,7 @@ static int mxs_saif_set_clk(struct mxs_saif *saif, ret = clk_set_rate(master_saif->clk, 384 * rate); } else { /* SAIF MCLK should be either 32x or 48x */ + clk_disable_unprepare(master_saif->clk); return -EINVAL; } } else { @@ -140,6 +143,8 @@ static int mxs_saif_set_clk(struct mxs_saif *saif, scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE; } + clk_disable_unprepare(master_saif->clk); + if (ret) return ret; From a14304edcd5e8323205db34b08f709feb5357e64 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 21 Jan 2012 21:48:53 +0000 Subject: [PATCH 08/36] ASoC: wm8996: Call _POST_PMU callback for CPVDD We should be allowing a 5ms delay after the charge pump is started in order to ensure it has finished ramping. Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/wm8996.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 86f5b6bd7af2..13aa2bdaa7d7 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -1120,7 +1120,8 @@ SND_SOC_DAPM_SUPPLY_S("SYSCLK", 1, WM8996_AIF_CLOCKING_1, 0, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("SYSDSPCLK", 2, WM8996_CLOCKING_1, 1, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("AIFCLK", 2, WM8996_CLOCKING_1, 2, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("Charge Pump", 2, WM8996_CHARGE_PUMP_1, 15, 0, cp_event, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_SUPPLY("Bandgap", SND_SOC_NOPM, 0, 0, bg_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_SUPPLY("LDO2", WM8996_POWER_MANAGEMENT_2, 1, 0, NULL, 0), From b4ead019afc201f71c39cd0dfcaafed4a97b3dd2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 23 Jan 2012 18:23:36 +0100 Subject: [PATCH 09/36] ALSA: hda - Fix silent outputs from docking-station jacks of Dell laptops The recent change of the power-widget handling for IDT codecs caused the silent output from the docking-station line-out jack. This was partially fixed by the commit f2cbba7602383cd9cdd21f0a5d0b8bd1aad47b33 "ALSA: hda - Fix the lost power-setup of seconary pins after PM resume". But the line-out on the docking-station is still silent when booted with the jack plugged even by this fix. The remainig bug is that the power-widget is set off in stac92xx_init() because the pins in cfg->line_out_pins[] aren't checked there properly but only hp_pins[] are checked in is_nid_hp_pin(). This patch fixes the problem by checking both HP and line-out pins and leaving the power-map correctly. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=42637 Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 1a26dbca9483..336cfcd324f9 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4163,13 +4163,15 @@ static int enable_pin_detect(struct hda_codec *codec, hda_nid_t nid, return 1; } -static int is_nid_hp_pin(struct auto_pin_cfg *cfg, hda_nid_t nid) +static int is_nid_out_jack_pin(struct auto_pin_cfg *cfg, hda_nid_t nid) { int i; for (i = 0; i < cfg->hp_outs; i++) if (cfg->hp_pins[i] == nid) return 1; /* nid is a HP-Out */ - + for (i = 0; i < cfg->line_outs; i++) + if (cfg->line_out_pins[i] == nid) + return 1; /* nid is a line-Out */ return 0; /* nid is not a HP-Out */ }; @@ -4375,7 +4377,7 @@ static int stac92xx_init(struct hda_codec *codec) continue; } - if (is_nid_hp_pin(cfg, nid)) + if (is_nid_out_jack_pin(cfg, nid)) continue; /* already has an unsol event */ pinctl = snd_hda_codec_read(codec, nid, 0, From 7edf1a4f27f44588d69cbde955651990090eb25d Mon Sep 17 00:00:00 2001 From: Jesper Juhl Date: Mon, 23 Jan 2012 21:15:48 +0100 Subject: [PATCH 10/36] ASoC: wm8958: Use correct format string in dev_err() call To print a value of type size_t one should use %zd, not %d. Signed-off-by: Jesper Juhl Signed-off-by: Mark Brown --- sound/soc/codecs/wm8958-dsp2.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index 8d4ea43d40a3..40ac888faf3d 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -55,7 +55,7 @@ static int wm8958_dsp2_fw(struct snd_soc_codec *codec, const char *name, return 0; if (fw->size < 32) { - dev_err(codec->dev, "%s: firmware too short (%d bytes)\n", + dev_err(codec->dev, "%s: firmware too short (%zd bytes)\n", name, fw->size); goto err; } From c83f1d7e71625801c72f4013291194e09b6f0a6e Mon Sep 17 00:00:00 2001 From: Jesper Juhl Date: Mon, 23 Jan 2012 22:28:44 +0100 Subject: [PATCH 11/36] ASoC: wm2000: Fix use-after-free - don't release_firmware() twice on error In wm2000_i2c_probe(), if we take the true branch in " ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm2000, NULL, 0); if (ret != 0) goto err_fw; " then we'll release_firmware(fw) at the 'err_fw' label. But we've already done that just a few lines above. That's a use-after-free bug. This patch restructures the code so that we always call release_firmware(fw) before leaving the function, but only ever call it once. This means that we have to initialize 'fw' to NULL since some paths may now end up calling it without having called request_firmware(), but since request_firmware() deals gracefully with NULL pointers, we are fine if we just NULL initialize it. Signed-off-by: Jesper Juhl Signed-off-by: Mark Brown --- sound/soc/codecs/wm2000.c | 31 +++++++++++++------------------ 1 file changed, 13 insertions(+), 18 deletions(-) diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index c2880907fced..a75c3766aede 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -733,8 +733,9 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c, struct wm2000_priv *wm2000; struct wm2000_platform_data *pdata; const char *filename; - const struct firmware *fw; - int reg, ret; + const struct firmware *fw = NULL; + int ret; + int reg; u16 id; wm2000 = devm_kzalloc(&i2c->dev, sizeof(struct wm2000_priv), @@ -751,7 +752,7 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c, ret = PTR_ERR(wm2000->regmap); dev_err(&i2c->dev, "Failed to allocate register map: %d\n", ret); - goto err; + goto out; } /* Verify that this is a WM2000 */ @@ -763,7 +764,7 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c, if (id != 0x2000) { dev_err(&i2c->dev, "Device is not a WM2000 - ID %x\n", id); ret = -ENODEV; - goto err_regmap; + goto out_regmap_exit; } reg = wm2000_read(i2c, WM2000_REG_REVISON); @@ -782,7 +783,7 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c, ret = request_firmware(&fw, filename, &i2c->dev); if (ret != 0) { dev_err(&i2c->dev, "Failed to acquire ANC data: %d\n", ret); - goto err_regmap; + goto out_regmap_exit; } /* Pre-cook the concatenation of the register address onto the image */ @@ -793,15 +794,13 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c, if (wm2000->anc_download == NULL) { dev_err(&i2c->dev, "Out of memory\n"); ret = -ENOMEM; - goto err_fw; + goto out_regmap_exit; } wm2000->anc_download[0] = 0x80; wm2000->anc_download[1] = 0x00; memcpy(wm2000->anc_download + 2, fw->data, fw->size); - release_firmware(fw); - wm2000->anc_eng_ena = 1; wm2000->anc_active = 1; wm2000->spk_ena = 1; @@ -809,18 +808,14 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c, wm2000_reset(wm2000); - ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm2000, - NULL, 0); - if (ret != 0) - goto err_fw; + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm2000, NULL, 0); + if (!ret) + goto out; - return 0; - -err_fw: - release_firmware(fw); -err_regmap: +out_regmap_exit: regmap_exit(wm2000->regmap); -err: +out: + release_firmware(fw); return ret; } From 4d20bb1d5fe1afbdbff951c06cd3d3654fa5ceed Mon Sep 17 00:00:00 2001 From: Raymond Yau Date: Tue, 17 Jan 2012 11:41:47 +0800 Subject: [PATCH 12/36] ALSA: ymfpci - Don't create invalid PCM & mixers when AC97 doesn't support - check SDAC bit of AC97 primary codec when create "rear" device 3, "4ch" device 2 and "4ch Duplication" switch as the card need a four channels AC97 codec to support surround40. Signed-off-by: Raymond Yau Signed-off-by: Takashi Iwai --- sound/pci/ymfpci/ymfpci.c | 21 +++++++++++++-------- sound/pci/ymfpci/ymfpci_main.c | 21 ++++++++++++++------- 2 files changed, 27 insertions(+), 15 deletions(-) diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c index e57b89e8aa89..94ab728f5ca8 100644 --- a/sound/pci/ymfpci/ymfpci.c +++ b/sound/pci/ymfpci/ymfpci.c @@ -286,17 +286,22 @@ static int __devinit snd_card_ymfpci_probe(struct pci_dev *pci, snd_card_free(card); return err; } - if ((err = snd_ymfpci_pcm_4ch(chip, 2, NULL)) < 0) { + err = snd_ymfpci_mixer(chip, rear_switch[dev]); + if (err < 0) { snd_card_free(card); return err; } - if ((err = snd_ymfpci_pcm2(chip, 3, NULL)) < 0) { - snd_card_free(card); - return err; - } - if ((err = snd_ymfpci_mixer(chip, rear_switch[dev])) < 0) { - snd_card_free(card); - return err; + if (chip->ac97->ext_id & AC97_EI_SDAC) { + err = snd_ymfpci_pcm_4ch(chip, 2, NULL); + if (err < 0) { + snd_card_free(card); + return err; + } + err = snd_ymfpci_pcm2(chip, 3, NULL); + if (err < 0) { + snd_card_free(card); + return err; + } } if ((err = snd_ymfpci_timer(chip, 0)) < 0) { snd_card_free(card); diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c index 03ee4e365311..12a9a2b03387 100644 --- a/sound/pci/ymfpci/ymfpci_main.c +++ b/sound/pci/ymfpci/ymfpci_main.c @@ -1614,6 +1614,14 @@ static int snd_ymfpci_put_dup4ch(struct snd_kcontrol *kcontrol, struct snd_ctl_e return change; } +static struct snd_kcontrol_new snd_ymfpci_dup4ch __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "4ch Duplication", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_ymfpci_info_dup4ch, + .get = snd_ymfpci_get_dup4ch, + .put = snd_ymfpci_put_dup4ch, +}; static struct snd_kcontrol_new snd_ymfpci_controls[] __devinitdata = { { @@ -1642,13 +1650,6 @@ YMFPCI_DOUBLE(SNDRV_CTL_NAME_IEC958("",CAPTURE,VOLUME), 1, YDSXGR_SPDIFLOOPVOL), YMFPCI_SINGLE(SNDRV_CTL_NAME_IEC958("",PLAYBACK,SWITCH), 0, YDSXGR_SPDIFOUTCTRL, 0), YMFPCI_SINGLE(SNDRV_CTL_NAME_IEC958("",CAPTURE,SWITCH), 0, YDSXGR_SPDIFINCTRL, 0), YMFPCI_SINGLE(SNDRV_CTL_NAME_IEC958("Loop",NONE,NONE), 0, YDSXGR_SPDIFINCTRL, 4), -{ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "4ch Duplication", - .info = snd_ymfpci_info_dup4ch, - .get = snd_ymfpci_get_dup4ch, - .put = snd_ymfpci_put_dup4ch, -}, }; @@ -1838,6 +1839,12 @@ int __devinit snd_ymfpci_mixer(struct snd_ymfpci *chip, int rear_switch) if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_ymfpci_controls[idx], chip))) < 0) return err; } + if (chip->ac97->ext_id & AC97_EI_SDAC) { + kctl = snd_ctl_new1(&snd_ymfpci_dup4ch, chip); + err = snd_ctl_add(chip->card, kctl); + if (err < 0) + return err; + } /* add S/PDIF control */ if (snd_BUG_ON(!chip->pcm_spdif)) From 769fab2a41da4bd3c59eee38f47d6d5405738fe0 Mon Sep 17 00:00:00 2001 From: Jesper Juhl Date: Mon, 23 Jan 2012 21:02:57 +0100 Subject: [PATCH 13/36] ALSA: Fix memory leak on error in snd_compr_set_params() If copy_from_user() does not return 0 we'll leak the memory we allocated for 'params' when that variable goes out of scope. Also a small CodingStyle cleanup: Use braces on both branches of if/else when one branch needs it. Signed-off-by: Jesper Juhl Acked-by: Vinod Koul Signed-off-by: Takashi Iwai --- sound/core/compress_offload.c | 13 ++++++++----- 1 file changed, 8 insertions(+), 5 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index dac3633507c9..a68aed7fce02 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -441,19 +441,22 @@ snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg) params = kmalloc(sizeof(*params), GFP_KERNEL); if (!params) return -ENOMEM; - if (copy_from_user(params, (void __user *)arg, sizeof(*params))) - return -EFAULT; + if (copy_from_user(params, (void __user *)arg, sizeof(*params))) { + retval = -EFAULT; + goto out; + } retval = snd_compr_allocate_buffer(stream, params); if (retval) { - kfree(params); - return -ENOMEM; + retval = -ENOMEM; + goto out; } retval = stream->ops->set_params(stream, params); if (retval) goto out; stream->runtime->state = SNDRV_PCM_STATE_SETUP; - } else + } else { return -EPERM; + } out: kfree(params); return retval; From 3b25eb690e8c7424eecffe1458c02b87b32aa001 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jan 2012 09:55:46 +0100 Subject: [PATCH 14/36] ALSA: hda - Fix silent output on ASUS A6Rp The refactoring of Realtek codec driver in 3.2 kernel caused a regression for ASUS A6Rp laptop; it doesn't give any output. The reason was that this machine has a secret master mute (or EAPD) control via NID 0x0f VREF. Setting VREF50 on this node makes the sound working again. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=42588 Cc: [v3.2+] Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c95c8bde12d0..a23479926f89 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5586,6 +5586,7 @@ static const struct hda_amp_list alc861_loopbacks[] = { /* Pin config fixes */ enum { PINFIX_FSC_AMILO_PI1505, + PINFIX_ASUS_A6RP, }; static const struct alc_fixup alc861_fixups[] = { @@ -5597,9 +5598,18 @@ 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 }, + { } + }, + }, }; static const struct snd_pci_quirk alc861_fixup_tbl[] = { + SND_PCI_QUIRK(0x1043, 0x1393, "ASUS A6Rp", PINFIX_ASUS_A6RP), SND_PCI_QUIRK(0x1734, 0x10c7, "FSC Amilo Pi1505", PINFIX_FSC_AMILO_PI1505), {} }; From a6a600d10aaddf1da38053c4c6b64f50f56176e6 Mon Sep 17 00:00:00 2001 From: Gustavo Maciel Dias Vieira Date: Tue, 24 Jan 2012 13:27:56 -0200 Subject: [PATCH 15/36] ALSA: hda: set mute led polarity for laptops with buggy BIOS based on SSID HP laptop models with buggy BIOS are apparently frequent, including machines with different codecs. Set the polarity of the mute led based on the SSID and include an entry for the HP Mini 110-3100. Signed-off-by: Gustavo Maciel Dias Vieira Tested-by: Predrag Ivanovic Cc: [v3.2+] Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 336cfcd324f9..948f0be2f4f3 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4870,7 +4870,14 @@ static int find_mute_led_cfg(struct hda_codec *codec, int default_polarity) /* BIOS bug: unfilled OEM string */ if (strstr(dev->name, "HP_Mute_LED_P_G")) { set_hp_led_gpio(codec); - spec->gpio_led_polarity = 1; + switch (codec->subsystem_id) { + case 0x103c148a: + spec->gpio_led_polarity = 0; + break; + default: + spec->gpio_led_polarity = 1; + break; + } return 1; } } From 9fb83526a898f14adbd3f6f52fa7126f528f15ac Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 25 Jan 2012 15:19:20 +0000 Subject: [PATCH 16/36] 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(+) diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 66f0611e68b6..3f8fd3ca9454 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); From a188fcba73837f83a78dc90a44998a978f50ac83 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 25 Jan 2012 17:57:16 +0000 Subject: [PATCH 17/36] 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(-) diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 3f8fd3ca9454..fb757af19363 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] << From 1b76d2ee4012f325ae14e0e71dad1a0835195906 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 25 Jan 2012 21:10:07 +0000 Subject: [PATCH 18/36] 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(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 13aa2bdaa7d7..61f7daa4d0e6 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; \ } From 5539a102882d5ddd1bb95ea9f6f43130a789cb7f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 25 Jan 2012 21:10:21 +0000 Subject: [PATCH 19/36] 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(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 296de4e30d26..bda3da887d7e 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; \ } From 5c1b136b7bf702e550039cb0039ec9c790c48f99 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 25 Jan 2012 21:10:33 +0000 Subject: [PATCH 20/36] 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(+) diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index fb757af19363..89f2af77b1c3 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), From b3a81520bd37a28f77cb0f7002086fb14061824d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 26 Jan 2012 15:56:16 +0100 Subject: [PATCH 21/36] ALSA: hda - Fix silent output on Haier W18 laptop The very same problem is seen on Haier W18 laptop with ALC861 as seen on ASUS A6Rp, which was fixed by the commit 3b25eb69. Now we just need to add a new SSID entry pointing to the same fixup. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=42656 Cc: [v3.2+] Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a23479926f89..0db1dc49382b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5610,6 +5610,7 @@ static const struct alc_fixup alc861_fixups[] = { static const struct snd_pci_quirk alc861_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1393, "ASUS A6Rp", 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), {} }; From 77231abe55433aa17eca712718745275853fa66d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 20 Jan 2012 12:19:43 +0000 Subject: [PATCH 22/36] 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(+) diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 2a61094075f8..9ccc416d8cbc 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" }, From a389d67cf9849aff1722ed73186a584e2196a873 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Fri, 27 Jan 2012 14:31:19 +0100 Subject: [PATCH 23/36] 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(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0db1dc49382b..a7f17becbd7c 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), From 114395c61ad2eb5a7a5cd163fcadb2414e48245a Mon Sep 17 00:00:00 2001 From: UK KIM Date: Sat, 28 Jan 2012 01:52:22 +0900 Subject: [PATCH 24/36] 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(-) diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 9ccc416d8cbc..ea2672455d07 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[] = { From 8422fa110334cea79ab16c474902edb21a8b3168 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Mon, 30 Jan 2012 17:10:58 +0800 Subject: [PATCH 25/36] ALSA: Add #ifdef CONFIG_PCI guard for snd_pci_quirk_* functions This fixes below build warning when CONFIG_PCI is not set. CC sound/sound_core.o In file included from sound/sound_core.c:15: include/sound/core.h:454: warning: 'struct pci_dev' declared inside parameter list include/sound/core.h:454: warning: its scope is only this definition or declaration, which is probably not what you want Signed-off-by: Axel Lin Signed-off-by: Takashi Iwai --- include/sound/core.h | 2 ++ 1 file changed, 2 insertions(+) diff --git a/include/sound/core.h b/include/sound/core.h index 5ab255f196cc..cea1b5426dfa 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -417,6 +417,7 @@ static inline int __snd_bug_on(int cond) #define gameport_get_port_data(gp) (gp)->port_data #endif +#ifdef CONFIG_PCI /* PCI quirk list helper */ struct snd_pci_quirk { unsigned short subvendor; /* PCI subvendor ID */ @@ -456,5 +457,6 @@ snd_pci_quirk_lookup(struct pci_dev *pci, const struct snd_pci_quirk *list); const struct snd_pci_quirk * snd_pci_quirk_lookup_id(u16 vendor, u16 device, const struct snd_pci_quirk *list); +#endif #endif /* __SOUND_CORE_H */ From 1ae5cbc52e7c6619a3f44b87809fd25370df31bb Mon Sep 17 00:00:00 2001 From: Denis 'GNUtoo' Carikli Date: Mon, 30 Jan 2012 00:31:47 +0100 Subject: [PATCH 26/36] 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(-) diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index 7ac0ba2025c3..c6012ff5bd3e 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()) { From 31150f2327cbb66363f38e13ca1be973d2f9203a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 30 Jan 2012 10:54:08 +0100 Subject: [PATCH 27/36] 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(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a7f17becbd7c..42b6a01e17db 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), {} From 05c3b36e539627b7aed67d038381d0d9fa9d61e7 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Tue, 31 Jan 2012 09:04:15 +0100 Subject: [PATCH 28/36] 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(-) diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index d8a35da0803f..9d819c4b4923 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; From 3422a47041b8cb8f14ac1e3926bcf711121df6dc Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Tue, 31 Jan 2012 10:31:49 +0100 Subject: [PATCH 29/36] 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(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 42b6a01e17db..a8e82be3d2fc 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), From f70eecde3bca92630d3886496e73316ff353f185 Mon Sep 17 00:00:00 2001 From: Dylan Reid Date: Tue, 31 Jan 2012 13:04:41 -0800 Subject: [PATCH 30/36] 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(-) diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 0e99357e822c..bc5a993d1146 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; } From 54c2a89f60fd71b924d0f848ac892442951401a6 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Wed, 1 Feb 2012 12:05:41 +0100 Subject: [PATCH 31/36] 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(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 4df72c0e8c37..c2c65f63bf06 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; } } From 054d867e032daf55c3343fc6d36c5c5f1e7954db Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 24 Jan 2012 12:25:50 +0100 Subject: [PATCH 32/36] 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(-) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 03e63fed9caf..fb1f0ffc556b 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); } From 924339239fd5ba3e505f9420d41f0939196f3530 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 24 Jan 2012 13:58:36 +0100 Subject: [PATCH 33/36] 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(-) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index fb1f0ffc556b..de43cd92b0a5 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) { From e9d010c2e8f03952e67a6fd8aed0f0dc92084ccc Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 1 Feb 2012 10:33:23 +0100 Subject: [PATCH 34/36] 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(-) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index de43cd92b0a5..79166fb8b074 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); From b5bcc189401c815988b7dd37611fc56f40c9139d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 2 Feb 2012 10:30:17 +0100 Subject: [PATCH 35/36] 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(+) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 79166fb8b074..284e311040fe 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; From b544d1e0e233f83a2e6d20ee96b54ea272d5d5ba Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 3 Feb 2012 11:56:35 +0100 Subject: [PATCH 36/36] 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(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a8e82be3d2fc..33b6077fcdb8 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, };