From f1e10354fc2a12773e5e8efcf841380aa57d4aa5 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sat, 5 Nov 2011 14:47:19 +0800 Subject: ASoC: wm9081: Fix reading wrong register for setting VMID 2*240k VMID Divider Enable and Select is controlled by BIT[2:1] of WM9081_VMID_CONTROL register (04h). Current code reads wrong register (WM9081_BIAS_CONTROL_1) for setting VMID 2*240k. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm9081.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 3cd35a02c28c..fe6561885f39 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -818,7 +818,7 @@ static int wm9081_set_bias_level(struct snd_soc_codec *codec, } /* VMID 2*240k */ - reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); + reg = snd_soc_read(codec, WM9081_VMID_CONTROL); reg &= ~WM9081_VMID_SEL_MASK; reg |= 0x04; snd_soc_write(codec, WM9081_VMID_CONTROL, reg); -- cgit v1.2.3 From adf463626ad8e0a2cdbe17d8bb64c1d9d0ac160d Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sat, 5 Nov 2011 14:49:21 +0800 Subject: ASoC: wm9081: Don't write WM9081_BIAS_ENA bit to WM9081_VMID_CONTROL register WM9081_BIAS_ENA is the bit[1] of WM9081_BIAS_CONTROL_1 register (05h). Current code incorrectly write it to WM9081_VMID_CONTROL(04h) register. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm9081.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index fe6561885f39..4a398c3bfe84 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -807,7 +807,6 @@ static int wm9081_set_bias_level(struct snd_soc_codec *codec, mdelay(100); /* Normal bias enable & soft start off */ - reg |= WM9081_BIAS_ENA; reg &= ~WM9081_VMID_RAMP; snd_soc_write(codec, WM9081_VMID_CONTROL, reg); @@ -830,14 +829,15 @@ static int wm9081_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_OFF: - /* Startup bias source */ + /* Startup bias source and disable bias */ reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); reg |= WM9081_BIAS_SRC; + reg &= ~WM9081_BIAS_ENA; snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg); - /* Disable VMID and biases with soft ramping */ + /* Disable VMID with soft ramping */ reg = snd_soc_read(codec, WM9081_VMID_CONTROL); - reg &= ~(WM9081_VMID_SEL_MASK | WM9081_BIAS_ENA); + reg &= ~WM9081_VMID_SEL_MASK; reg |= WM9081_VMID_RAMP; snd_soc_write(codec, WM9081_VMID_CONTROL, reg); -- cgit v1.2.3 From 54dc6cabe684375b3cf549c7b0545613d694aba8 Mon Sep 17 00:00:00 2001 From: Johannes Stezenbach Date: Mon, 14 Nov 2011 17:23:16 +0100 Subject: ASoC: sta32x: preserve coefficient RAM The coefficient RAM must be saved in a shadow so it can be restored when the codec is powered on using regulator_bulk_enable(). Signed-off-by: Johannes Stezenbach Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/sta32x.c | 63 ++++++++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/sta32x.h | 1 + 2 files changed, 63 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index bb82408ab8e1..d2f37152f940 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -76,6 +76,8 @@ struct sta32x_priv { unsigned int mclk; unsigned int format; + + u32 coef_shadow[STA32X_COEF_COUNT]; }; static const DECLARE_TLV_DB_SCALE(mvol_tlv, -12700, 50, 1); @@ -227,6 +229,7 @@ static int sta32x_coefficient_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); int numcoef = kcontrol->private_value >> 16; int index = kcontrol->private_value & 0xffff; unsigned int cfud; @@ -239,6 +242,11 @@ static int sta32x_coefficient_put(struct snd_kcontrol *kcontrol, snd_soc_write(codec, STA32X_CFUD, cfud); snd_soc_write(codec, STA32X_CFADDR2, index); + for (i = 0; i < numcoef && (index + i < STA32X_COEF_COUNT); i++) + sta32x->coef_shadow[index + i] = + (ucontrol->value.bytes.data[3 * i] << 16) + | (ucontrol->value.bytes.data[3 * i + 1] << 8) + | (ucontrol->value.bytes.data[3 * i + 2]); for (i = 0; i < 3 * numcoef; i++) snd_soc_write(codec, STA32X_B1CF1 + i, ucontrol->value.bytes.data[i]); @@ -252,6 +260,48 @@ static int sta32x_coefficient_put(struct snd_kcontrol *kcontrol, return 0; } +int sta32x_sync_coef_shadow(struct snd_soc_codec *codec) +{ + struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); + unsigned int cfud; + int i; + + /* preserve reserved bits in STA32X_CFUD */ + cfud = snd_soc_read(codec, STA32X_CFUD) & 0xf0; + + for (i = 0; i < STA32X_COEF_COUNT; i++) { + snd_soc_write(codec, STA32X_CFADDR2, i); + snd_soc_write(codec, STA32X_B1CF1, + (sta32x->coef_shadow[i] >> 16) & 0xff); + snd_soc_write(codec, STA32X_B1CF2, + (sta32x->coef_shadow[i] >> 8) & 0xff); + snd_soc_write(codec, STA32X_B1CF3, + (sta32x->coef_shadow[i]) & 0xff); + /* chip documentation does not say if the bits are + * self-clearing, so do it explicitly */ + snd_soc_write(codec, STA32X_CFUD, cfud); + snd_soc_write(codec, STA32X_CFUD, cfud | 0x01); + } + return 0; +} + +int sta32x_cache_sync(struct snd_soc_codec *codec) +{ + unsigned int mute; + int rc; + + if (!codec->cache_sync) + return 0; + + /* mute during register sync */ + mute = snd_soc_read(codec, STA32X_MMUTE); + snd_soc_write(codec, STA32X_MMUTE, mute | STA32X_MMUTE_MMUTE); + sta32x_sync_coef_shadow(codec); + rc = snd_soc_cache_sync(codec); + snd_soc_write(codec, STA32X_MMUTE, mute); + return rc; +} + #define SINGLE_COEF(xname, index) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = sta32x_coefficient_info, \ @@ -661,7 +711,7 @@ static int sta32x_set_bias_level(struct snd_soc_codec *codec, return ret; } - snd_soc_cache_sync(codec); + sta32x_cache_sync(codec); } /* Power up to mute */ @@ -790,6 +840,17 @@ static int sta32x_probe(struct snd_soc_codec *codec) STA32X_CxCFG_OM_MASK, 2 << STA32X_CxCFG_OM_SHIFT); + /* initialize coefficient shadow RAM with reset values */ + for (i = 4; i <= 49; i += 5) + sta32x->coef_shadow[i] = 0x400000; + for (i = 50; i <= 54; i++) + sta32x->coef_shadow[i] = 0x7fffff; + sta32x->coef_shadow[55] = 0x5a9df7; + sta32x->coef_shadow[56] = 0x7fffff; + sta32x->coef_shadow[59] = 0x7fffff; + sta32x->coef_shadow[60] = 0x400000; + sta32x->coef_shadow[61] = 0x400000; + sta32x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Bias level configuration will have done an extra enable */ regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); diff --git a/sound/soc/codecs/sta32x.h b/sound/soc/codecs/sta32x.h index b97ee5a75667..d8e32a6262ee 100644 --- a/sound/soc/codecs/sta32x.h +++ b/sound/soc/codecs/sta32x.h @@ -19,6 +19,7 @@ /* STA326 register addresses */ #define STA32X_REGISTER_COUNT 0x2d +#define STA32X_COEF_COUNT 62 #define STA32X_CONFA 0x00 #define STA32X_CONFB 0x01 -- cgit v1.2.3 From 0f768a7235d3dfb6f4833030a95a06419df089cb Mon Sep 17 00:00:00 2001 From: Timur Tabi Date: Mon, 14 Nov 2011 16:35:26 -0600 Subject: ASoC: fsl_ssi: properly initialize the sysfs attribute object Commit 6992f533 ("sysfs: Use one lockdep class per sysfs attribute") requires 'struct attribute' objects to be initialized with sysfs_attr_init(). Signed-off-by: Timur Tabi Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/fsl/fsl_ssi.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 0268cf989736..83c4bd5b2dd7 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -694,6 +694,7 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev) /* Initialize the the device_attribute structure */ dev_attr = &ssi_private->dev_attr; + sysfs_attr_init(&dev_attr->attr); dev_attr->attr.name = "statistics"; dev_attr->attr.mode = S_IRUGO; dev_attr->show = fsl_sysfs_ssi_show; -- cgit v1.2.3 From 2391a0e06789a3f1718dee30b282562f7ed28c87 Mon Sep 17 00:00:00 2001 From: Timo Juhani Lindfors Date: Thu, 17 Nov 2011 02:52:50 +0200 Subject: ASoC: wm8753: Skip noop reconfiguration of DAI mode This patch makes it possible to set DAI mode to its currently applied value even if codec is active. This is necessary to allow aplay -t raw -r 44100 -f S16_LE -c 2 < /dev/urandom & alsactl store -f backup.state alsactl restore -f backup.state to work without returning errors. This patch is based on a patch sent by Klaus Kurzmann . Signed-off-by: Timo Juhani Lindfors Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm8753.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index a9504710bb69..3a629d0d690e 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -190,6 +190,9 @@ static int wm8753_set_dai(struct snd_kcontrol *kcontrol, struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec); u16 ioctl; + if (wm8753->dai_func == ucontrol->value.integer.value[0]) + return 0; + if (codec->active) return -EBUSY; -- cgit v1.2.3 From 0aed4a95ce3b39acfceb38ab7f93c7906b4a27f8 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sun, 20 Nov 2011 15:10:27 +0100 Subject: ASoC: adau1373: fix DB_RANGE size Give the correct number of entries to TLV_DB_RANGE_HEAD to prevent reading more data than actually is in the array. Signed-off-by: Clemens Ladisch --- sound/soc/codecs/adau1373.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index 1ccf8dd47576..45c63028b40d 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -245,7 +245,7 @@ static const char *adau1373_bass_hpf_cutoff_text[] = { }; static const unsigned int adau1373_bass_tlv[] = { - TLV_DB_RANGE_HEAD(4), + TLV_DB_RANGE_HEAD(3), 0, 2, TLV_DB_SCALE_ITEM(-600, 600, 1), 3, 4, TLV_DB_SCALE_ITEM(950, 250, 0), 5, 7, TLV_DB_SCALE_ITEM(1400, 150, 0), -- cgit v1.2.3 From a19ea0b8ec1f6892bf18f461d5023c9299e1417b Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sun, 20 Nov 2011 15:11:54 +0100 Subject: ASoC: rt5631: fix DB_RANGE size Give the correct number of entries to TLV_DB_RANGE_HEAD to prevent the last entry from being omitted. Signed-off-by: Clemens Ladisch --- sound/soc/codecs/rt5631.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index 27a078cbb6eb..4646e808b90a 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -177,7 +177,7 @@ static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -95625, 375, 0); static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); /* {0, +20, +24, +30, +35, +40, +44, +50, +52}dB */ static unsigned int mic_bst_tlv[] = { - TLV_DB_RANGE_HEAD(6), + TLV_DB_RANGE_HEAD(7), 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0), 2, 2, TLV_DB_SCALE_ITEM(2400, 0, 0), -- cgit v1.2.3 From 740fb9d512d91b1d6192ea13c109efa05b101424 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sun, 20 Nov 2011 15:12:26 +0100 Subject: ASoC: sgtl5000: fix DB_RANGE size Give the correct number of entries to TLV_DB_RANGE_HEAD to prevent reading more data than actually is in the array. Signed-off-by: Clemens Ladisch --- sound/soc/codecs/sgtl5000.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index d15695d1c273..bbcf921166f7 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -365,7 +365,7 @@ static const DECLARE_TLV_DB_SCALE(capture_6db_attenuate, -600, 600, 0); /* tlv for mic gain, 0db 20db 30db 40db */ static const unsigned int mic_gain_tlv[] = { - TLV_DB_RANGE_HEAD(4), + TLV_DB_RANGE_HEAD(2), 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), 1, 3, TLV_DB_SCALE_ITEM(2000, 1000, 0), }; -- cgit v1.2.3 From 43e9dc7bce9f21355cd2aa493a99281eae03b156 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sun, 20 Nov 2011 15:13:27 +0100 Subject: ASoC: wm8962: fix DB_RANGE size Give the correct number of entries to TLV_DB_RANGE_HEAD to prevent reading more data than actually is in the arrays. Signed-off-by: Clemens Ladisch --- sound/soc/codecs/wm8962.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 91d3c6dbeba3..53edd9a8c758 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -1973,7 +1973,7 @@ static int wm8962_reset(struct snd_soc_codec *codec) static const DECLARE_TLV_DB_SCALE(inpga_tlv, -2325, 75, 0); static const DECLARE_TLV_DB_SCALE(mixin_tlv, -1500, 300, 0); static const unsigned int mixinpga_tlv[] = { - TLV_DB_RANGE_HEAD(7), + TLV_DB_RANGE_HEAD(5), 0, 1, TLV_DB_SCALE_ITEM(0, 600, 0), 2, 2, TLV_DB_SCALE_ITEM(1300, 1300, 0), 3, 4, TLV_DB_SCALE_ITEM(1800, 200, 0), @@ -1988,7 +1988,7 @@ static const DECLARE_TLV_DB_SCALE(bypass_tlv, -1500, 300, 0); static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1); static const DECLARE_TLV_DB_SCALE(hp_tlv, -700, 100, 0); static const unsigned int classd_tlv[] = { - TLV_DB_RANGE_HEAD(7), + TLV_DB_RANGE_HEAD(2), 0, 6, TLV_DB_SCALE_ITEM(0, 150, 0), 7, 7, TLV_DB_SCALE_ITEM(1200, 0, 0), }; -- cgit v1.2.3 From dac678f5c281fac55aadfa5f390c12a8d14bbc67 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sun, 20 Nov 2011 15:14:11 +0100 Subject: ASoC: wm8993: fix DB_RANGE size Give the correct number of entries to TLV_DB_RANGE_HEAD to prevent reading more data than actually is in the array. Signed-off-by: Clemens Ladisch --- sound/soc/codecs/wm8993.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index eec8e1435116..d1a142f48b09 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -512,7 +512,7 @@ static const DECLARE_TLV_DB_SCALE(drc_comp_threash, -4500, 75, 0); static const DECLARE_TLV_DB_SCALE(drc_comp_amp, -2250, 75, 0); static const DECLARE_TLV_DB_SCALE(drc_min_tlv, -1800, 600, 0); static const unsigned int drc_max_tlv[] = { - TLV_DB_RANGE_HEAD(4), + TLV_DB_RANGE_HEAD(2), 0, 2, TLV_DB_SCALE_ITEM(1200, 600, 0), 3, 3, TLV_DB_SCALE_ITEM(3600, 0, 0), }; -- cgit v1.2.3 From a1320fee27352b608a82020a47a59bb15e6e5db8 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sun, 20 Nov 2011 15:14:55 +0100 Subject: ASoC: wm9090: fix DB_RANGE size Give the correct number of entries to TLV_DB_RANGE_HEAD to prevent reading more data than actually is in the arrays. Signed-off-by: Clemens Ladisch --- sound/soc/codecs/wm9090.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index 2b5252c9e377..f94c06057c64 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -177,19 +177,19 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec) } static const unsigned int in_tlv[] = { - TLV_DB_RANGE_HEAD(6), + TLV_DB_RANGE_HEAD(3), 0, 0, TLV_DB_SCALE_ITEM(-600, 0, 0), 1, 3, TLV_DB_SCALE_ITEM(-350, 350, 0), 4, 6, TLV_DB_SCALE_ITEM(600, 600, 0), }; static const unsigned int mix_tlv[] = { - TLV_DB_RANGE_HEAD(4), + TLV_DB_RANGE_HEAD(2), 0, 2, TLV_DB_SCALE_ITEM(-1200, 300, 0), 3, 3, TLV_DB_SCALE_ITEM(0, 0, 0), }; static const DECLARE_TLV_DB_SCALE(out_tlv, -5700, 100, 0); static const unsigned int spkboost_tlv[] = { - TLV_DB_RANGE_HEAD(7), + TLV_DB_RANGE_HEAD(2), 0, 6, TLV_DB_SCALE_ITEM(0, 150, 0), 7, 7, TLV_DB_SCALE_ITEM(1200, 0, 0), }; -- cgit v1.2.3 From 028aa634e180107ac93b790c0fed7376c0402d1a Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sun, 20 Nov 2011 15:15:31 +0100 Subject: ASoC: wm_hubs: fix DB_RANGE size Give the correct number of entries to TLV_DB_RANGE_HEAD to prevent reading more data than actually is in the array. Signed-off-by: Clemens Ladisch --- sound/soc/codecs/wm_hubs.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 84f33d4ea2cd..48e61e912400 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -40,7 +40,7 @@ static const DECLARE_TLV_DB_SCALE(outmix_tlv, -2100, 300, 0); static const DECLARE_TLV_DB_SCALE(spkmixout_tlv, -1800, 600, 1); static const DECLARE_TLV_DB_SCALE(outpga_tlv, -5700, 100, 0); static const unsigned int spkboost_tlv[] = { - TLV_DB_RANGE_HEAD(7), + TLV_DB_RANGE_HEAD(2), 0, 6, TLV_DB_SCALE_ITEM(0, 150, 0), 7, 7, TLV_DB_SCALE_ITEM(1200, 0, 0), }; -- cgit v1.2.3 From ef0cd47093a6c4b8a1f17d7be02a966f7805ff41 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sat, 19 Nov 2011 14:41:07 +0800 Subject: ASoC: cs4271: Fix wrong mask parameter in some snd_soc_update_bits calls Signed-off-by: Axel Lin Acked-by: Alexander Sverdlin Signed-off-by: Mark Brown --- sound/soc/codecs/cs4271.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 23d1bd5dadda..69fde1506fe1 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -434,7 +434,8 @@ static int cs4271_soc_suspend(struct snd_soc_codec *codec, pm_message_t mesg) { int ret; /* Set power-down bit */ - ret = snd_soc_update_bits(codec, CS4271_MODE2, 0, CS4271_MODE2_PDN); + ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, + CS4271_MODE2_PDN); if (ret < 0) return ret; return 0; @@ -501,8 +502,9 @@ static int cs4271_probe(struct snd_soc_codec *codec) return ret; } - ret = snd_soc_update_bits(codec, CS4271_MODE2, 0, - CS4271_MODE2_PDN | CS4271_MODE2_CPEN); + ret = snd_soc_update_bits(codec, CS4271_MODE2, + CS4271_MODE2_PDN | CS4271_MODE2_CPEN, + CS4271_MODE2_PDN | CS4271_MODE2_CPEN); if (ret < 0) return ret; ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0); -- cgit v1.2.3 From ed3e80c4c991a52f9fce3421536a78e331ae0949 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 21 Nov 2011 11:55:41 +0000 Subject: ASoC: Ensure WM8731 register cache is synced when resuming from disabled Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm8731.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 7e5ec03f6f8d..a7c9ae17fc7e 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -453,6 +453,7 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, snd_soc_write(codec, WM8731_PWR, 0xffff); regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); + codec->cache_sync = 1; break; } codec->dapm.bias_level = level; -- cgit v1.2.3 From 92bb43e6aae3dbdb199feba93da5f2d05d7716d0 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Thu, 24 Nov 2011 14:48:24 +0300 Subject: ALSA: hda - cut and paste typo in cs420x_models[] The CS420X_IMAC27 was copied from the line before but CS420X_APPLE was clearly intented. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cirrus.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 7bd2a52f2bac..70a7abda7e22 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -1278,7 +1278,7 @@ static const char * const cs420x_models[CS420X_MODELS] = { [CS420X_MBP53] = "mbp53", [CS420X_MBP55] = "mbp55", [CS420X_IMAC27] = "imac27", - [CS420X_IMAC27] = "apple", + [CS420X_APPLE] = "apple", [CS420X_AUTO] = "auto", }; -- cgit v1.2.3 From 187d333edc0a8e1bb507900ce89853ffe3bd2c84 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 24 Nov 2011 16:33:09 +0100 Subject: ALSA: hda - Fix jack-detection control of VT1708 VT1708 has no support for unsolicited events per jack-plug, the driver implements the workq for polling the jack-detection. The mixer element "Jack Detect" was supposed to control this behavior on/off, but this doesn't work properly as is now. The workq is always started and the HP automute is always enabled. This patch fixes the jack-detect control behavior by triggering / stopping the work appropriately at the state change. Also the work checks the internal state to continue scheduling or not. Cc: [v3.1] Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 76 +++++++++++++++++++++++++++-------------------- 1 file changed, 43 insertions(+), 33 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 431c0d417eeb..b5137629f8e9 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -208,6 +208,7 @@ struct via_spec { /* work to check hp jack state */ struct hda_codec *codec; struct delayed_work vt1708_hp_work; + int hp_work_active; int vt1708_jack_detect; int vt1708_hp_present; @@ -305,27 +306,35 @@ enum { static void analog_low_current_mode(struct hda_codec *codec); static bool is_aa_path_mute(struct hda_codec *codec); -static void vt1708_start_hp_work(struct via_spec *spec) +#define hp_detect_with_aa(codec) \ + (snd_hda_get_bool_hint(codec, "analog_loopback_hp_detect") == 1 && \ + !is_aa_path_mute(codec)) + +static void vt1708_stop_hp_work(struct via_spec *spec) { if (spec->codec_type != VT1708 || spec->autocfg.hp_pins[0] == 0) return; - snd_hda_codec_write(spec->codec, 0x1, 0, 0xf81, - !spec->vt1708_jack_detect); - if (!delayed_work_pending(&spec->vt1708_hp_work)) - schedule_delayed_work(&spec->vt1708_hp_work, - msecs_to_jiffies(100)); + if (spec->hp_work_active) { + snd_hda_codec_write(spec->codec, 0x1, 0, 0xf81, 1); + cancel_delayed_work_sync(&spec->vt1708_hp_work); + spec->hp_work_active = 0; + } } -static void vt1708_stop_hp_work(struct via_spec *spec) +static void vt1708_update_hp_work(struct via_spec *spec) { if (spec->codec_type != VT1708 || spec->autocfg.hp_pins[0] == 0) return; - if (snd_hda_get_bool_hint(spec->codec, "analog_loopback_hp_detect") == 1 - && !is_aa_path_mute(spec->codec)) - return; - snd_hda_codec_write(spec->codec, 0x1, 0, 0xf81, - !spec->vt1708_jack_detect); - cancel_delayed_work_sync(&spec->vt1708_hp_work); + if (spec->vt1708_jack_detect && + (spec->active_streams || hp_detect_with_aa(spec->codec))) { + if (!spec->hp_work_active) { + snd_hda_codec_write(spec->codec, 0x1, 0, 0xf81, 0); + schedule_delayed_work(&spec->vt1708_hp_work, + msecs_to_jiffies(100)); + spec->hp_work_active = 1; + } + } else if (!hp_detect_with_aa(spec->codec)) + vt1708_stop_hp_work(spec); } static void set_widgets_power_state(struct hda_codec *codec) @@ -343,12 +352,7 @@ static int analog_input_switch_put(struct snd_kcontrol *kcontrol, set_widgets_power_state(codec); analog_low_current_mode(snd_kcontrol_chip(kcontrol)); - if (snd_hda_get_bool_hint(codec, "analog_loopback_hp_detect") == 1) { - if (is_aa_path_mute(codec)) - vt1708_start_hp_work(codec->spec); - else - vt1708_stop_hp_work(codec->spec); - } + vt1708_update_hp_work(codec->spec); return change; } @@ -1154,7 +1158,7 @@ static int via_playback_multi_pcm_prepare(struct hda_pcm_stream *hinfo, spec->cur_dac_stream_tag = stream_tag; spec->cur_dac_format = format; mutex_unlock(&spec->config_mutex); - vt1708_start_hp_work(spec); + vt1708_update_hp_work(spec); return 0; } @@ -1174,7 +1178,7 @@ static int via_playback_hp_pcm_prepare(struct hda_pcm_stream *hinfo, spec->cur_hp_stream_tag = stream_tag; spec->cur_hp_format = format; mutex_unlock(&spec->config_mutex); - vt1708_start_hp_work(spec); + vt1708_update_hp_work(spec); return 0; } @@ -1188,7 +1192,7 @@ static int via_playback_multi_pcm_cleanup(struct hda_pcm_stream *hinfo, snd_hda_multi_out_analog_cleanup(codec, &spec->multiout); spec->active_streams &= ~STREAM_MULTI_OUT; mutex_unlock(&spec->config_mutex); - vt1708_stop_hp_work(spec); + vt1708_update_hp_work(spec); return 0; } @@ -1203,7 +1207,7 @@ static int via_playback_hp_pcm_cleanup(struct hda_pcm_stream *hinfo, snd_hda_codec_setup_stream(codec, spec->hp_dac_nid, 0, 0, 0); spec->active_streams &= ~STREAM_INDEP_HP; mutex_unlock(&spec->config_mutex); - vt1708_stop_hp_work(spec); + vt1708_update_hp_work(spec); return 0; } @@ -1645,7 +1649,8 @@ static void via_hp_automute(struct hda_codec *codec) int nums; struct via_spec *spec = codec->spec; - if (!spec->hp_independent_mode && spec->autocfg.hp_pins[0]) + if (!spec->hp_independent_mode && spec->autocfg.hp_pins[0] && + (spec->codec_type != VT1708 || spec->vt1708_jack_detect)) present = snd_hda_jack_detect(codec, spec->autocfg.hp_pins[0]); if (spec->smart51_enabled) @@ -2612,8 +2617,6 @@ static int vt1708_jack_detect_get(struct snd_kcontrol *kcontrol, if (spec->codec_type != VT1708) return 0; - spec->vt1708_jack_detect = - !((snd_hda_codec_read(codec, 0x1, 0, 0xf84, 0) >> 8) & 0x1); ucontrol->value.integer.value[0] = spec->vt1708_jack_detect; return 0; } @@ -2623,18 +2626,22 @@ static int vt1708_jack_detect_put(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct via_spec *spec = codec->spec; - int change; + int val; if (spec->codec_type != VT1708) return 0; - spec->vt1708_jack_detect = ucontrol->value.integer.value[0]; - change = (0x1 & (snd_hda_codec_read(codec, 0x1, 0, 0xf84, 0) >> 8)) - == !spec->vt1708_jack_detect; - if (spec->vt1708_jack_detect) { + val = !!ucontrol->value.integer.value[0]; + if (spec->vt1708_jack_detect == val) + return 0; + spec->vt1708_jack_detect = val; + if (spec->vt1708_jack_detect && + snd_hda_get_bool_hint(codec, "analog_loopback_hp_detect") != 1) { mute_aa_path(codec, 1); notify_aa_path_ctls(codec); } - return change; + via_hp_automute(codec); + vt1708_update_hp_work(spec); + return 1; } static const struct snd_kcontrol_new vt1708_jack_detect_ctl = { @@ -2771,6 +2778,7 @@ static int via_init(struct hda_codec *codec) via_auto_init_unsol_event(codec); via_hp_automute(codec); + vt1708_update_hp_work(spec); return 0; } @@ -2787,7 +2795,9 @@ static void vt1708_update_hp_jack_state(struct work_struct *work) spec->vt1708_hp_present ^= 1; via_hp_automute(spec->codec); } - vt1708_start_hp_work(spec); + if (spec->vt1708_jack_detect) + schedule_delayed_work(&spec->vt1708_hp_work, + msecs_to_jiffies(100)); } static int get_mux_nids(struct hda_codec *codec) -- cgit v1.2.3 From ae7cc709f2ec11b49fc31b20cd8c943794ae9576 Mon Sep 17 00:00:00 2001 From: John F Leach Date: Mon, 28 Nov 2011 19:41:27 -0500 Subject: ALSA: usb-audio - Support for Roland GAIA SH-01 Synthesizer Added table quirks entry for Roland GAIA SH-01 Synthesizer based upon Roland SH-201 table entry as template. USB MIDI and audio was tested with Muse and Audacity. Signed-off-by: John F Leach Signed-off-by: Takashi Iwai --- sound/usb/quirks-table.h | 31 +++++++++++++++++++++++++++++++ 1 file changed, 31 insertions(+) (limited to 'sound') diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index b61945f3af9e..32d2a21f2e3b 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1632,6 +1632,37 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + /* Roland GAIA SH-01 */ + USB_DEVICE(0x0582, 0x0111), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Roland", + .product_name = "GAIA", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = &(const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0003, + .in_cables = 0x0003 + } + }, + { + .ifnum = -1 + } + } + } +}, { USB_DEVICE(0x0582, 0x0113), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { -- cgit v1.2.3 From 542c9a0a2fa351149c4a3467589a54cafcf0a1dd Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 29 Nov 2011 13:01:30 +0100 Subject: ALSA: hda - Avoid touching mute-VREF pin for IDT codecs Some HP laptops use a pin VREF for controlling the mute LED, and such a pin shouldn't be powered off. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index f3658658548e..f4f4ebeed9ea 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4441,7 +4441,9 @@ static int stac92xx_init(struct hda_codec *codec) int pinctl, def_conf; /* power on when no jack detection is available */ - if (!spec->hp_detect) { + /* or when the VREF is used for controlling LED */ + if (!spec->hp_detect || + (spec->gpio_led > 8 && spec->gpio_led == nid)) { stac_toggle_power_map(codec, nid, 1); continue; } -- cgit v1.2.3 From 4f8b6c7dc80ac9619db033c7f6fc355eab9514f5 Mon Sep 17 00:00:00 2001 From: Marc Vertes Date: Tue, 29 Nov 2011 12:21:17 +0100 Subject: ALSA: hda_intel - revert a quirk that affect VIA chipsets This quirk sould be reverted. It has the following probems: 1) The quirk was intended to "ASUS MV2-MX SE" motherboards only, but the ID used matches a much broader range, potentially all boards containing a VIA chipset model in the family of vendor VIA 0x1106 and audio device ID 0x3288, which encompasses VIA-VT82xx, VIA-VT1xx and VIA-VT20xx chipsets. 2) VIA chipsets rely on azx_via_get_position() to handle correctly dma transfers during capture. Using POS_FIX_LPIB instead of POS_FIX_VIACOMBO leads to partially corrupted input buffers during capture. The effects of this bug are not immediately visible, it took strong DSP expertise, some expensive signal generator and a spectrum analyzer to identify it and verify correct behaviour using original default. 3) It's almost certain that the quirk did not fix the real problem, if there was one. Refer to original submission: http://mailman.alsa-project.org/pipermail/alsa-devel/2010-February/025109.html Signed-of-by: Marc Vertes Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 096507d2ca9a..7d98240def0b 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2508,7 +2508,6 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB), SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB), - SND_PCI_QUIRK(0x1106, 0x3288, "ASUS M2V-MX SE", POS_FIX_LPIB), SND_PCI_QUIRK(0x1297, 0x3166, "Shuttle", POS_FIX_LPIB), SND_PCI_QUIRK(0x1458, 0xa022, "ga-ma770-ud3", POS_FIX_LPIB), SND_PCI_QUIRK(0x1462, 0x1002, "MSI Wind U115", POS_FIX_LPIB), -- cgit v1.2.3 From 88d686027bb43f585914c77dd363f6e817b42c2a Mon Sep 17 00:00:00 2001 From: Charles Chin Date: Thu, 1 Dec 2011 11:21:00 +0100 Subject: ALSA: hda - Fix S3/S4 problem on machines with VREF-pin mute-LED The verb command in stac92xx_post_suspend caused the audio to stop working after resuming from S3 mode on HP laptops with the VREF-pin mute-LED control. Removing relevant post_suspend registering. Although removing D3 on AFG is no optimal solution, the impact should be small in comparison with the broken S3/S4. Signed-off-by: Charles Chin Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 18 ------------------ 1 file changed, 18 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index f4f4ebeed9ea..d8d2f9dccd9b 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -5057,20 +5057,6 @@ static int stac92xx_pre_resume(struct hda_codec *codec) return 0; } -static int stac92xx_post_suspend(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - if (spec->gpio_led > 8) { - /* with vref-out pin used for mute led control - * codec AFG is prevented from D3 state, but on - * system suspend it can (and should) be used - */ - snd_hda_codec_read(codec, codec->afg, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - } - return 0; -} - static void stac92xx_set_power_state(struct hda_codec *codec, hda_nid_t fg, unsigned int power_state) { @@ -5670,8 +5656,6 @@ again: } else { codec->patch_ops.set_power_state = stac92xx_set_power_state; - codec->patch_ops.post_suspend = - stac92xx_post_suspend; } codec->patch_ops.pre_resume = stac92xx_pre_resume; codec->patch_ops.check_power_status = @@ -5985,8 +5969,6 @@ again: } else { codec->patch_ops.set_power_state = stac92xx_set_power_state; - codec->patch_ops.post_suspend = - stac92xx_post_suspend; } codec->patch_ops.pre_resume = stac92xx_pre_resume; codec->patch_ops.check_power_status = -- cgit v1.2.3