From b90bf1de7cb65e7f61798fcfbcf74ae72207b0dc Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 19 Jan 2012 11:42:55 +0100 Subject: ALSA: hda/realtek - Avoid multi-ios conflicting with multi-speakers When a machine has multiple speakers, we don't need to create the controls for multi-ios. Check the number of primary outputs beforehand. Note that this workaround might not work always with new codecs in future; this assumes that both speakers and multi-io jacks share the same mixers/DACs. If they are routed with different mixers, the individual mixer controls should be needed. But, so far, this doesn't happen with the existing ALC codecs. 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 5e82acf..61ccbe8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3233,7 +3233,7 @@ static int alc_auto_create_multi_out_ctls(struct hda_codec *codec, int i, err, noutputs; noutputs = cfg->line_outs; - if (spec->multi_ios > 0) + if (spec->multi_ios > 0 && cfg->line_outs < 3) noutputs += spec->multi_ios; for (i = 0; i < noutputs; i++) { -- cgit v1.1 From f21d78e2698b6380a5387461e3b126bb2dee23aa Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 19 Jan 2012 12:10:29 +0100 Subject: ALSA: hda/realtek - Avoid conflict of unsol-events with static quirks The recently added jack-kctl support sets the unsol event tags dynamically, while static quirks usually set the fixed tags in the init_verbs array. Due to this conflict, the own unsol event handler can't retrieve the tag and handle it properly any more. For fixing this, avoid calling snd_hda_jack_add_kctls() for static quirks, and always let them use own handlers instead of the standard one for the auto-pareser. Reported-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/alc880_quirks.c | 17 +++++++++++----- sound/pci/hda/alc882_quirks.c | 15 ++++++++++----- sound/pci/hda/patch_realtek.c | 45 +++++++++++++++++++++++++++++-------------- 3 files changed, 53 insertions(+), 24 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/alc880_quirks.c b/sound/pci/hda/alc880_quirks.c index 5b68435..501501e 100644 --- a/sound/pci/hda/alc880_quirks.c +++ b/sound/pci/hda/alc880_quirks.c @@ -762,16 +762,22 @@ static void alc880_uniwill_unsol_event(struct hda_codec *codec, /* Looks like the unsol event is incompatible with the standard * definition. 4bit tag is placed at 28 bit! */ - switch (res >> 28) { + res >>= 28; + switch (res) { case ALC_MIC_EVENT: alc88x_simple_mic_automute(codec); break; default: - alc_sku_unsol_event(codec, res); + alc_exec_unsol_event(codec, res); break; } } +static void alc880_unsol_event(struct hda_codec *codec, unsigned int res) +{ + alc_exec_unsol_event(codec, res >> 28); +} + static void alc880_uniwill_p53_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -800,10 +806,11 @@ static void alc880_uniwill_p53_unsol_event(struct hda_codec *codec, /* Looks like the unsol event is incompatible with the standard * definition. 4bit tag is placed at 28 bit! */ - if ((res >> 28) == ALC_DCVOL_EVENT) + res >>= 28; + if (res == ALC_DCVOL_EVENT) alc880_uniwill_p53_dcvol_automute(codec); else - alc_sku_unsol_event(codec, res); + alc_exec_unsol_event(codec, res); } /* @@ -1677,7 +1684,7 @@ static const struct alc_config_preset alc880_presets[] = { .channel_mode = alc880_lg_ch_modes, .need_dac_fix = 1, .input_mux = &alc880_lg_capture_source, - .unsol_event = alc_sku_unsol_event, + .unsol_event = alc880_unsol_event, .setup = alc880_lg_setup, .init_hook = alc_hp_automute, #ifdef CONFIG_SND_HDA_POWER_SAVE diff --git a/sound/pci/hda/alc882_quirks.c b/sound/pci/hda/alc882_quirks.c index bdf0ed4..bb364a5 100644 --- a/sound/pci/hda/alc882_quirks.c +++ b/sound/pci/hda/alc882_quirks.c @@ -730,6 +730,11 @@ static void alc889A_mb31_unsol_event(struct hda_codec *codec, unsigned int res) alc889A_mb31_automute(codec); } +static void alc882_unsol_event(struct hda_codec *codec, unsigned int res) +{ + alc_exec_unsol_event(codec, res >> 26); +} + /* * configuration and preset */ @@ -775,7 +780,7 @@ static const struct alc_config_preset alc882_presets[] = { .channel_mode = alc885_mba21_ch_modes, .num_channel_mode = ARRAY_SIZE(alc885_mba21_ch_modes), .input_mux = &alc882_capture_source, - .unsol_event = alc_sku_unsol_event, + .unsol_event = alc882_unsol_event, .setup = alc885_mba21_setup, .init_hook = alc_hp_automute, }, @@ -791,7 +796,7 @@ static const struct alc_config_preset alc882_presets[] = { .input_mux = &alc882_capture_source, .dig_out_nid = ALC882_DIGOUT_NID, .dig_in_nid = ALC882_DIGIN_NID, - .unsol_event = alc_sku_unsol_event, + .unsol_event = alc882_unsol_event, .setup = alc885_mbp3_setup, .init_hook = alc_hp_automute, }, @@ -806,7 +811,7 @@ static const struct alc_config_preset alc882_presets[] = { .input_mux = &mb5_capture_source, .dig_out_nid = ALC882_DIGOUT_NID, .dig_in_nid = ALC882_DIGIN_NID, - .unsol_event = alc_sku_unsol_event, + .unsol_event = alc882_unsol_event, .setup = alc885_mb5_setup, .init_hook = alc_hp_automute, }, @@ -821,7 +826,7 @@ static const struct alc_config_preset alc882_presets[] = { .input_mux = &macmini3_capture_source, .dig_out_nid = ALC882_DIGOUT_NID, .dig_in_nid = ALC882_DIGIN_NID, - .unsol_event = alc_sku_unsol_event, + .unsol_event = alc882_unsol_event, .setup = alc885_macmini3_setup, .init_hook = alc_hp_automute, }, @@ -836,7 +841,7 @@ static const struct alc_config_preset alc882_presets[] = { .input_mux = &alc889A_imac91_capture_source, .dig_out_nid = ALC882_DIGOUT_NID, .dig_in_nid = ALC882_DIGIN_NID, - .unsol_event = alc_sku_unsol_event, + .unsol_event = alc882_unsol_event, .setup = alc885_imac91_setup, .init_hook = alc_hp_automute, }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 61ccbe8..2326bf3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -621,17 +621,10 @@ static void alc_mic_automute(struct hda_codec *codec) alc_mux_select(codec, 0, spec->int_mic_idx, false); } -/* unsolicited event for HP jack sensing */ -static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res) +/* handle the specified unsol action (ALC_XXX_EVENT) */ +static void alc_exec_unsol_event(struct hda_codec *codec, int action) { - struct alc_spec *spec = codec->spec; - if (codec->vendor_id == 0x10ec0880) - res >>= 28; - else - res >>= 26; - if (spec->use_jack_tbl) - res = snd_hda_jack_get_action(codec, res); - switch (res) { + switch (action) { case ALC_HP_EVENT: alc_hp_automute(codec); break; @@ -645,6 +638,19 @@ static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res) snd_hda_jack_report_sync(codec); } +/* unsolicited event for HP jack sensing */ +static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res) +{ + struct alc_spec *spec = codec->spec; + if (codec->vendor_id == 0x10ec0880) + res >>= 28; + else + res >>= 26; + if (spec->use_jack_tbl) + res = snd_hda_jack_get_action(codec, res); + alc_exec_unsol_event(codec, res); +} + /* call init functions of standard auto-mute helpers */ static void alc_inithook(struct hda_codec *codec) { @@ -1883,7 +1889,7 @@ static const struct snd_kcontrol_new alc_beep_mixer[] = { }; #endif -static int alc_build_controls(struct hda_codec *codec) +static int __alc_build_controls(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; struct snd_kcontrol *kctl = NULL; @@ -2029,11 +2035,16 @@ static int alc_build_controls(struct hda_codec *codec) alc_free_kctls(codec); /* no longer needed */ - err = snd_hda_jack_add_kctls(codec, &spec->autocfg); + return 0; +} + +static int alc_build_controls(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + int err = __alc_build_controls(codec); if (err < 0) return err; - - return 0; + return snd_hda_jack_add_kctls(codec, &spec->autocfg); } @@ -4168,6 +4179,8 @@ static int patch_alc880(struct hda_codec *codec) codec->patch_ops = alc_patch_ops; if (board_config == ALC_MODEL_AUTO) spec->init_hook = alc_auto_init_std; + else + codec->patch_ops.build_controls = __alc_build_controls; #ifdef CONFIG_SND_HDA_POWER_SAVE if (!spec->loopback.amplist) spec->loopback.amplist = alc880_loopbacks; @@ -4297,6 +4310,8 @@ static int patch_alc260(struct hda_codec *codec) codec->patch_ops = alc_patch_ops; if (board_config == ALC_MODEL_AUTO) spec->init_hook = alc_auto_init_std; + else + codec->patch_ops.build_controls = __alc_build_controls; spec->shutup = alc_eapd_shutup; #ifdef CONFIG_SND_HDA_POWER_SAVE if (!spec->loopback.amplist) @@ -4691,6 +4706,8 @@ static int patch_alc882(struct hda_codec *codec) codec->patch_ops = alc_patch_ops; if (board_config == ALC_MODEL_AUTO) spec->init_hook = alc_auto_init_std; + else + codec->patch_ops.build_controls = __alc_build_controls; #ifdef CONFIG_SND_HDA_POWER_SAVE if (!spec->loopback.amplist) -- cgit v1.1 From a7309792c4e313d4e4c30084dc0ecbc834082433 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 19 Jan 2012 15:03:48 +0100 Subject: ALSA: hda/realtek - Remove use_jack_tbl field Now that all quirks have the own unsol handlers, we don't need to check use_jack_tbl flag any more. Let's kill it. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2326bf3..ddbed97 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -185,7 +185,6 @@ struct alc_spec { unsigned int vol_in_capsrc:1; /* use capsrc volume (ADC has no vol) */ unsigned int parse_flags; /* passed to snd_hda_parse_pin_defcfg() */ unsigned int shared_mic_hp:1; /* HP/Mic-in sharing */ - unsigned int use_jack_tbl:1; /* 1 for model=auto */ /* auto-mute control */ int automute_mode; @@ -646,8 +645,7 @@ static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res) res >>= 28; else res >>= 26; - if (spec->use_jack_tbl) - res = snd_hda_jack_get_action(codec, res); + res = snd_hda_jack_get_action(codec, res); alc_exec_unsol_event(codec, res); } @@ -3915,7 +3913,6 @@ static void set_capture_mixer(struct hda_codec *codec) static void alc_auto_init_std(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - spec->use_jack_tbl = 1; alc_auto_init_multi_out(codec); alc_auto_init_extra_out(codec); alc_auto_init_analog_input(codec); -- cgit v1.1 From b9ecc4ee28a5ff5b3997da247cd9df1320c602a9 Mon Sep 17 00:00:00 2001 From: Albert Pool Date: Thu, 19 Jan 2012 22:08:50 +0100 Subject: snd-hda-intel: better Alienware M17x R3 quirk I have been told that this way the rear headphone connector is working as well; with model=alienware only laptop speakers work. The subsystem of both controller and codec is 1028:0490. Signed-off-by: Albert Pool Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 3556408..1a26dbc 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1608,7 +1608,7 @@ static const struct snd_pci_quirk stac92hd73xx_codec_id_cfg_tbl[] = { SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x043a, "Alienware M17x", STAC_ALIENWARE_M17X), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0490, - "Alienware M17x", STAC_ALIENWARE_M17X), + "Alienware M17x R3", STAC_DELL_EQ), {} /* terminator */ }; -- cgit v1.1 From cb0cdebbf8b834110ef67ed9335d5bafed7835df Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 20 Jan 2012 12:14:12 +0100 Subject: ALSA: hda - Fix a unused variable warning MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Just overlooked. sound/pci/hda/patch_realtek.c: In function ‘alc_sku_unsol_event’: sound/pci/hda/patch_realtek.c:643:19: warning: unused variable ‘spec’ [-Wunused-variable] 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 ddbed97..c95c8bd 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -640,7 +640,6 @@ static void alc_exec_unsol_event(struct hda_codec *codec, int action) /* unsolicited event for HP jack sensing */ static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res) { - struct alc_spec *spec = codec->spec; if (codec->vendor_id == 0x10ec0880) res >>= 28; else -- cgit v1.1 From bb362e2e4f4874f3fd4cbc2497385b9bceb3a08a Mon Sep 17 00:00:00 2001 From: Zeng Zhaoming Date: Wed, 18 Jan 2012 13:58:07 +0800 Subject: 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(-) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index f8863eb..7f4ba81 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]); -- cgit v1.1 From 01b37e94c04bc6dae2c4837a2eb6fff6819ea82a Mon Sep 17 00:00:00 2001 From: Wolfram Sang Date: Wed, 18 Jan 2012 11:48:58 +0100 Subject: 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 | 102 ++++++++++++++++++--------------------- 1 file changed, 46 insertions(+), 56 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index eb401ef..3806cb6 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 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 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 BCLK_N Divider */ - snd_soc_update_bits(codec, AIC32X4_BCLKN, - AIC32X4_BCLKEN, AIC32X4_BCLKEN); - } + /* 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 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 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); 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 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 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 BCLK_N Divider */ - snd_soc_update_bits(codec, AIC32X4_BCLKN, - AIC32X4_BCLKEN, 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 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 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); break; case SND_SOC_BIAS_OFF: break; -- cgit v1.1 From 0c93a167a6b3fa510c74e88477852c41defda075 Mon Sep 17 00:00:00 2001 From: Wolfram Sang Date: Wed, 18 Jan 2012 11:48:59 +0100 Subject: 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(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 3806cb6..372b0b8 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; -- cgit v1.1 From e53e417331c57b9b97e3f8be870214a02c99265c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 18 Jan 2012 20:02:38 +0000 Subject: 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(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 8b24323..3fd9cfe 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), -- cgit v1.1 From 495174a8ffbaa0d15153d855cf206cdc46d51cf4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 19 Jan 2012 11:16:37 +0000 Subject: 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(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 3fd9cfe..66f0611 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; -- cgit v1.1 From fed22007113cb857e917913ce016d9b539dc3a80 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 18 Jan 2012 19:17:06 +0000 Subject: 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(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index d8da10f..86f5b6b 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 0fde643..de9ac3e 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 */ -- cgit v1.1 From 6b35f924b80a0e6d71711e66f5b3c16f427f3d2a Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 19 Jan 2012 10:23:22 -0200 Subject: 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(+) (limited to 'sound') diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index dccfb37..f204dba 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; -- cgit v1.1 From a14304edcd5e8323205db34b08f709feb5357e64 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 21 Jan 2012 21:48:53 +0000 Subject: 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(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 86f5b6b..13aa2bd 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), -- cgit v1.1 From 52409aa6a0e96337da137c069856298f4dd825a0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 23 Jan 2012 17:10:24 +0100 Subject: ALSA: hda - Fix buffer-alignment regression with Nvidia HDMI The commit 2ae66c26550cd94b0e2606a9275eb0ab7070ad0e ALSA: hda: option to enable arbitrary buffer/period sizes introduced a regression on machines with Intel controller and Nvidia HDMI. The reason is that the driver modifies the global variable align_buffer_size when an Intel controller is found, and the Nvidia HDMI controller is probed after Intel although Nvidia chips require the aligned buffers. This patch fixes the problem by moving the flag into the local struct so that it's not affected by other controllers. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=42567 Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index fb35474..95dfb68 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -469,6 +469,7 @@ struct azx { unsigned int irq_pending_warned :1; unsigned int probing :1; /* codec probing phase */ unsigned int snoop:1; + unsigned int align_buffer_size:1; /* for debugging */ unsigned int last_cmd[AZX_MAX_CODECS]; @@ -1690,7 +1691,7 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) runtime->hw.rates = hinfo->rates; snd_pcm_limit_hw_rates(runtime); snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); - if (align_buffer_size) + if (chip->align_buffer_size) /* constrain buffer sizes to be multiple of 128 bytes. This is more efficient in terms of memory access but isn't required by the HDA spec and @@ -2773,8 +2774,9 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, } /* disable buffer size rounding to 128-byte multiples if supported */ + chip->align_buffer_size = align_buffer_size; if (chip->driver_caps & AZX_DCAPS_BUFSIZE) - align_buffer_size = 0; + chip->align_buffer_size = 0; /* allow 64bit DMA address if supported by H/W */ if ((gcap & ICH6_GCAP_64OK) && !pci_set_dma_mask(pci, DMA_BIT_MASK(64))) -- cgit v1.1 From 29c5fbbcfefba5225a6783683c46c39e10877703 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Mon, 23 Jan 2012 16:39:55 +0100 Subject: ALSA: HDA: Use model=auto for Thinkpad T510 The user reports that model=auto works fine for him. Using model=auto bring in new features such as jack detection notification to userspace. Alsa info is available at http://paste.ubuntu.com/805351/ Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 8a32a69..a7a5733 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3027,7 +3027,7 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21c6, "Thinkpad Edge 13", CXT5066_ASUS), - SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD), + SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T510", CXT5066_AUTO), SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520 & W520", CXT5066_AUTO), SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT5066_THINKPAD), -- cgit v1.1 From b4ead019afc201f71c39cd0dfcaafed4a97b3dd2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 23 Jan 2012 18:23:36 +0100 Subject: 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(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 1a26dbc..336cfcd 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, -- cgit v1.1 From 7edf1a4f27f44588d69cbde955651990090eb25d Mon Sep 17 00:00:00 2001 From: Jesper Juhl Date: Mon, 23 Jan 2012 21:15:48 +0100 Subject: 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(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index 8d4ea43..40ac888 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; } -- cgit v1.1 From c83f1d7e71625801c72f4013291194e09b6f0a6e Mon Sep 17 00:00:00 2001 From: Jesper Juhl Date: Mon, 23 Jan 2012 22:28:44 +0100 Subject: 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(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index c288090..a75c376 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; } -- cgit v1.1 From 4d20bb1d5fe1afbdbff951c06cd3d3654fa5ceed Mon Sep 17 00:00:00 2001 From: Raymond Yau Date: Tue, 17 Jan 2012 11:41:47 +0800 Subject: 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(-) (limited to 'sound') diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c index e57b89e8..94ab728 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 03ee4e3..12a9a2b 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)) -- cgit v1.1 From 769fab2a41da4bd3c59eee38f47d6d5405738fe0 Mon Sep 17 00:00:00 2001 From: Jesper Juhl Date: Mon, 23 Jan 2012 21:02:57 +0100 Subject: 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(-) (limited to 'sound') diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index dac3633..a68aed7 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; -- cgit v1.1 From 3b25eb690e8c7424eecffe1458c02b87b32aa001 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jan 2012 09:55:46 +0100 Subject: 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(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c95c8bd..a234799 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), {} }; -- cgit v1.1 From a6a600d10aaddf1da38053c4c6b64f50f56176e6 Mon Sep 17 00:00:00 2001 From: Gustavo Maciel Dias Vieira Date: Tue, 24 Jan 2012 13:27:56 -0200 Subject: 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(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 336cfcd..948f0be 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; } } -- cgit v1.1 From b3a81520bd37a28f77cb0f7002086fb14061824d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 26 Jan 2012 15:56:16 +0100 Subject: 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(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a234799..0db1dc4 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), {} }; -- cgit v1.1