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(-) diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 3cd35a02c28..fe6561885f3 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 v1.2.3-70-g09d2 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(-) diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index fe6561885f3..4a398c3bfe8 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 v1.2.3-70-g09d2 From 98d97019c88bd832da1457729739cf739ece493f Mon Sep 17 00:00:00 2001 From: Jassi Brar Date: Thu, 10 Nov 2011 17:19:07 +0530 Subject: MAINTAINERS: Drop inactive Samsung ASoC maintainer Signed-off-by: Jassi Brar Signed-off-by: Mark Brown --- MAINTAINERS | 1 - 1 file changed, 1 deletion(-) diff --git a/MAINTAINERS b/MAINTAINERS index c802e5fa2d1..fd7e441b5ea 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -5648,7 +5648,6 @@ F: drivers/media/video/*7146* F: include/media/*7146* SAMSUNG AUDIO (ASoC) DRIVERS -M: Jassi Brar M: Sangbeom Kim L: alsa-devel@alsa-project.org (moderated for non-subscribers) S: Supported -- cgit v1.2.3-70-g09d2 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(-) diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index bb82408ab8e..d2f37152f94 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 b97ee5a7566..d8e32a6262e 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 v1.2.3-70-g09d2 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(+) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 0268cf98973..83c4bd5b2dd 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 v1.2.3-70-g09d2 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(+) diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index a9504710bb6..3a629d0d690 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 v1.2.3-70-g09d2 From dbd1b5473ce8ae40fe7385eacc9294355eec0676 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 19 Nov 2011 11:41:30 +0100 Subject: ALSA: hda - Add pin fix for Alienware M17x R3 Reported-by: Albert Pool Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 470f6f286e8..f3658658548 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1641,6 +1641,8 @@ static const struct snd_pci_quirk stac92hd73xx_codec_id_cfg_tbl[] = { "Alienware M17x", STAC_ALIENWARE_M17X), 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), {} /* terminator */ }; -- cgit v1.2.3-70-g09d2 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(-) diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index 1ccf8dd4757..45c63028b40 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 v1.2.3-70-g09d2 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(-) diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index 27a078cbb6e..4646e808b90 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 v1.2.3-70-g09d2 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(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index d15695d1c27..bbcf921166f 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 v1.2.3-70-g09d2 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(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 91d3c6dbeba..53edd9a8c75 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 v1.2.3-70-g09d2 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(-) diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index eec8e143511..d1a142f48b0 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 v1.2.3-70-g09d2 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(-) diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index 2b5252c9e37..f94c06057c6 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 v1.2.3-70-g09d2 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(-) diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 84f33d4ea2c..48e61e91240 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 v1.2.3-70-g09d2 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(-) diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 23d1bd5dadd..69fde1506fe 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 v1.2.3-70-g09d2 From 27533df80e93dc164e39d47281bbbd608f9014a6 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Sun, 20 Nov 2011 23:57:49 +0300 Subject: ALSA: cs5535 - Fix an endianness conversion desc->size is supposed to be a le16 type. On a big endian system the current code will set ->size to zero. We fixed a similar bug on the next line but missed this one. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/pci/cs5535audio/cs5535audio_pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/cs5535audio/cs5535audio_pcm.c b/sound/pci/cs5535audio/cs5535audio_pcm.c index e083122ca55..dbf94b189e7 100644 --- a/sound/pci/cs5535audio/cs5535audio_pcm.c +++ b/sound/pci/cs5535audio/cs5535audio_pcm.c @@ -148,7 +148,7 @@ static int cs5535audio_build_dma_packets(struct cs5535audio *cs5535au, struct cs5535audio_dma_desc *desc = &((struct cs5535audio_dma_desc *) dma->desc_buf.area)[i]; desc->addr = cpu_to_le32(addr); - desc->size = cpu_to_le32(period_bytes); + desc->size = cpu_to_le16(period_bytes); desc->ctlreserved = cpu_to_le16(PRD_EOP); desc_addr += sizeof(struct cs5535audio_dma_desc); addr += period_bytes; -- cgit v1.2.3-70-g09d2 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(+) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 7e5ec03f6f8..a7c9ae17fc7 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 v1.2.3-70-g09d2 From 05c7cc9ccab7d9229fdae68d7d6231edd2c93741 Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Mon, 21 Nov 2011 16:15:36 +0100 Subject: ALSA: hdspm - Fix PCI ID for PCIe RME MADI cards Commit c09403dcc5698abf214329fbbf3cf8dbb5558bea has introduced a regression: PCIe versions of RME MADI were no longer detected, because the MADIface ID (0xd5) was used instead of the correct 0xd2. This commit fixes the problem. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index e760adad952..19ee2203cbb 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -6518,7 +6518,7 @@ static int __devinit snd_hdspm_create(struct snd_card *card, hdspm->io_type = AES32; hdspm->card_name = "RME AES32"; hdspm->midiPorts = 2; - } else if ((hdspm->firmware_rev == 0xd5) || + } else if ((hdspm->firmware_rev == 0xd2) || ((hdspm->firmware_rev >= 0xc8) && (hdspm->firmware_rev <= 0xcf))) { hdspm->io_type = MADI; -- cgit v1.2.3-70-g09d2 From c6e8453e7511001e453f8b20b9ceefd231946867 Mon Sep 17 00:00:00 2001 From: Wu Fengguang Date: Fri, 18 Nov 2011 16:59:32 -0600 Subject: ALSA: hda - repoll ELD content for multiple times Improve the one-shot ELD repoll to up to 6 retries. Up to now the 300ms looks sufficient for the test boxes. However I'm a bit worried about how well it can fit the wider user base. Signed-off-by: Wu Fengguang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 16 ++++++++++------ 1 file changed, 10 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 9850c5b481e..c505fd5d338 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -69,6 +69,7 @@ struct hdmi_spec_per_pin { struct hda_codec *codec; struct hdmi_eld sink_eld; struct delayed_work work; + int repoll_count; }; struct hdmi_spec { @@ -748,7 +749,7 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, int pin_idx, * Unsolicited events */ -static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, bool retry); +static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll); static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) { @@ -766,7 +767,7 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) if (pin_idx < 0) return; - hdmi_present_sense(&spec->pins[pin_idx], true); + hdmi_present_sense(&spec->pins[pin_idx], 1); } static void hdmi_non_intrinsic_event(struct hda_codec *codec, unsigned int res) @@ -960,7 +961,7 @@ static int hdmi_read_pin_conn(struct hda_codec *codec, int pin_idx) return 0; } -static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, bool retry) +static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) { struct hda_codec *codec = per_pin->codec; struct hdmi_eld *eld = &per_pin->sink_eld; @@ -989,7 +990,7 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, bool retry) if (eld_valid) { if (!snd_hdmi_get_eld(eld, codec, pin_nid)) snd_hdmi_show_eld(eld); - else if (retry) { + else if (repoll) { queue_delayed_work(codec->bus->workq, &per_pin->work, msecs_to_jiffies(300)); @@ -1004,7 +1005,10 @@ static void hdmi_repoll_eld(struct work_struct *work) struct hdmi_spec_per_pin *per_pin = container_of(to_delayed_work(work), struct hdmi_spec_per_pin, work); - hdmi_present_sense(per_pin, false); + if (per_pin->repoll_count++ > 6) + per_pin->repoll_count = 0; + + hdmi_present_sense(per_pin, per_pin->repoll_count); } static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) @@ -1235,7 +1239,7 @@ static int generic_hdmi_build_jack(struct hda_codec *codec, int pin_idx) if (err < 0) return err; - hdmi_present_sense(per_pin, false); + hdmi_present_sense(per_pin, 0); return 0; } -- cgit v1.2.3-70-g09d2 From afd00d7235c1989d06d75cf8ac3d7722fcf2f394 Mon Sep 17 00:00:00 2001 From: Tim Blechmann Date: Tue, 22 Nov 2011 11:15:44 +0100 Subject: ALSA: lx6464es - command buffer API cleanup the command buffer is only accessed from one file, so we can declare the specific functions as static in that file Signed-off-by: Tim Blechmann Signed-off-by: Takashi Iwai --- sound/pci/lx6464es/lx_core.c | 7 ++++--- sound/pci/lx6464es/lx_core.h | 3 --- 2 files changed, 4 insertions(+), 6 deletions(-) diff --git a/sound/pci/lx6464es/lx_core.c b/sound/pci/lx6464es/lx_core.c index 5c8717e29ee..ad52f4187e4 100644 --- a/sound/pci/lx6464es/lx_core.c +++ b/sound/pci/lx6464es/lx_core.c @@ -78,7 +78,8 @@ unsigned long lx_dsp_reg_read(struct lx6464es *chip, int port) return ioread32(address); } -void lx_dsp_reg_readbuf(struct lx6464es *chip, int port, u32 *data, u32 len) +static void lx_dsp_reg_readbuf(struct lx6464es *chip, int port, u32 *data, + u32 len) { void __iomem *address = lx_dsp_register(chip, port); memcpy_fromio(data, address, len*sizeof(u32)); @@ -91,8 +92,8 @@ void lx_dsp_reg_write(struct lx6464es *chip, int port, unsigned data) iowrite32(data, address); } -void lx_dsp_reg_writebuf(struct lx6464es *chip, int port, const u32 *data, - u32 len) +static void lx_dsp_reg_writebuf(struct lx6464es *chip, int port, + const u32 *data, u32 len) { void __iomem *address = lx_dsp_register(chip, port); memcpy_toio(address, data, len*sizeof(u32)); diff --git a/sound/pci/lx6464es/lx_core.h b/sound/pci/lx6464es/lx_core.h index 1dd562980b6..4d7ff797a64 100644 --- a/sound/pci/lx6464es/lx_core.h +++ b/sound/pci/lx6464es/lx_core.h @@ -72,10 +72,7 @@ enum { }; unsigned long lx_dsp_reg_read(struct lx6464es *chip, int port); -void lx_dsp_reg_readbuf(struct lx6464es *chip, int port, u32 *data, u32 len); void lx_dsp_reg_write(struct lx6464es *chip, int port, unsigned data); -void lx_dsp_reg_writebuf(struct lx6464es *chip, int port, const u32 *data, - u32 len); /* plx register access */ enum { -- cgit v1.2.3-70-g09d2 From a29878553a9a7b4c06f93c7e383527cf014d4ceb Mon Sep 17 00:00:00 2001 From: Tim Blechmann Date: Tue, 22 Nov 2011 11:15:45 +0100 Subject: ALSA: lx6464es - fix device communication via command bus commit 6175ddf06b6172046a329e3abfd9c901a43efd2e optimized the mem*io functions that have been used to send commands to the device. these optimizations somehow corrupted the communication with the lx6464es, that resulted the device to be unusable with kernels after 2.6.33. this patch emulates the memcpy_*_io functions via a loop to avoid these problems. Signed-off-by: Tim Blechmann LKML-Reference: <4ECB5257.4040600@ladisch.de> Cc: Signed-off-by: Takashi Iwai --- sound/pci/lx6464es/lx_core.c | 16 ++++++++++++---- 1 file changed, 12 insertions(+), 4 deletions(-) diff --git a/sound/pci/lx6464es/lx_core.c b/sound/pci/lx6464es/lx_core.c index ad52f4187e4..8c3e7fcefd9 100644 --- a/sound/pci/lx6464es/lx_core.c +++ b/sound/pci/lx6464es/lx_core.c @@ -81,8 +81,12 @@ unsigned long lx_dsp_reg_read(struct lx6464es *chip, int port) static void lx_dsp_reg_readbuf(struct lx6464es *chip, int port, u32 *data, u32 len) { - void __iomem *address = lx_dsp_register(chip, port); - memcpy_fromio(data, address, len*sizeof(u32)); + u32 __iomem *address = lx_dsp_register(chip, port); + int i; + + /* we cannot use memcpy_fromio */ + for (i = 0; i != len; ++i) + data[i] = ioread32(address + i); } @@ -95,8 +99,12 @@ void lx_dsp_reg_write(struct lx6464es *chip, int port, unsigned data) static void lx_dsp_reg_writebuf(struct lx6464es *chip, int port, const u32 *data, u32 len) { - void __iomem *address = lx_dsp_register(chip, port); - memcpy_toio(address, data, len*sizeof(u32)); + u32 __iomem *address = lx_dsp_register(chip, port); + int i; + + /* we cannot use memcpy_to */ + for (i = 0; i != len; ++i) + iowrite32(data[i], address + i); } -- cgit v1.2.3-70-g09d2 From a370fc62b9ad3f73abe2a721de6c03cdcce95b54 Mon Sep 17 00:00:00 2001 From: Wu Fengguang Date: Tue, 22 Nov 2011 16:46:23 +0800 Subject: ALSA: hda - fail ELD reading early With the ELD repoll mechanism, we can (and should) fail the ELD reading immediately when find something obviously wrong and let the caller retry after some delay. Signed-off-by: Wu Fengguang Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_eld.c | 28 +++++++++++++++++++--------- 1 file changed, 19 insertions(+), 9 deletions(-) diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index 7ae7578bdcc..c1da422e085 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -347,18 +347,28 @@ int snd_hdmi_get_eld(struct hdmi_eld *eld, for (i = 0; i < size; i++) { unsigned int val = hdmi_get_eld_data(codec, nid, i); + /* + * Graphics driver might be writing to ELD buffer right now. + * Just abort. The caller will repoll after a while. + */ if (!(val & AC_ELDD_ELD_VALID)) { - if (!i) { - snd_printd(KERN_INFO - "HDMI: invalid ELD data\n"); - ret = -EINVAL; - goto error; - } snd_printd(KERN_INFO "HDMI: invalid ELD data byte %d\n", i); - val = 0; - } else - val &= AC_ELDD_ELD_DATA; + ret = -EINVAL; + goto error; + } + val &= AC_ELDD_ELD_DATA; + /* + * The first byte cannot be zero. This can happen on some DVI + * connections. Some Intel chips may also need some 250ms delay + * to return non-zero ELD data, even when the graphics driver + * correctly writes ELD content before setting ELD_valid bit. + */ + if (!val && !i) { + snd_printdd(KERN_INFO "HDMI: 0 ELD data\n"); + ret = -EINVAL; + goto error; + } buf[i] = val; } -- cgit v1.2.3-70-g09d2 From e2301a4de22c438f5a962c7cefc3e9cba736991c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 22 Nov 2011 19:58:56 +0100 Subject: ALSA: hda - Check subdevice mask in snd_hda_check_board_codec_sid_config() In snd_hda_check_board_codec_sid_config(), not only comparing with the exact value but allow the bit-mask comparison for vendor-only, etc. Tested-by: Linus Torvalds Tested-by: Dirk Hohndel Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index e44b107fdc7..4562e9de6a1 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4046,9 +4046,9 @@ int snd_hda_check_board_codec_sid_config(struct hda_codec *codec, /* Search for codec ID */ for (q = tbl; q->subvendor; q++) { - unsigned long vendorid = (q->subdevice) | (q->subvendor << 16); - - if (vendorid == codec->subsystem_id) + unsigned int mask = 0xffff0000 | q->subdevice_mask; + unsigned int id = (q->subdevice | (q->subvendor << 16)) & mask; + if ((codec->subsystem_id & mask) == id) break; } -- cgit v1.2.3-70-g09d2 From 6dfeb703e386369d9f1585d29482efe7b2b4401d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 22 Nov 2011 20:00:31 +0100 Subject: ALSA: hda - Fix invalid pin and GPIO for Apple laptops with CS codecs The PCI SSID 8086:7270 is commonly used for multiple Apple machines, thus we can't use it as identifier for a unique model. Because of this conflict, some machines show weird behavior. For example, MacBook Air shows Front and Surround speakers although only Surround works due to the wrongly overridden pin-configuration for imac27. This patch fixes two things: - Stop the wrong pin-config override of imac27 by removing PCI SSID entry for avoiding the wrong mappings, - Add the generic GPIO setup for Apple machines by checking the codec SSID vendor bits Tested-by: Linus Torvalds Tested-by: Dirk Hohndel Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cirrus.c | 32 +++++++++++++++++++++++--------- 1 file changed, 23 insertions(+), 9 deletions(-) diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 2fbab8e2957..7bd2a52f2ba 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -58,6 +58,8 @@ struct cs_spec { unsigned int gpio_mask; unsigned int gpio_dir; unsigned int gpio_data; + unsigned int gpio_eapd_hp; /* EAPD GPIO bit for headphones */ + unsigned int gpio_eapd_speaker; /* EAPD GPIO bit for speakers */ struct hda_pcm pcm_rec[2]; /* PCM information */ @@ -76,6 +78,7 @@ enum { CS420X_MBP53, CS420X_MBP55, CS420X_IMAC27, + CS420X_APPLE, CS420X_AUTO, CS420X_MODELS }; @@ -928,10 +931,9 @@ static void cs_automute(struct hda_codec *codec) spdif_present ? 0 : PIN_OUT); } } - if (spec->board_config == CS420X_MBP53 || - spec->board_config == CS420X_MBP55 || - spec->board_config == CS420X_IMAC27) { - unsigned int gpio = hp_present ? 0x02 : 0x08; + if (spec->gpio_eapd_hp) { + unsigned int gpio = hp_present ? + spec->gpio_eapd_hp : spec->gpio_eapd_speaker; snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, gpio); } @@ -1276,6 +1278,7 @@ static const char * const cs420x_models[CS420X_MODELS] = { [CS420X_MBP53] = "mbp53", [CS420X_MBP55] = "mbp55", [CS420X_IMAC27] = "imac27", + [CS420X_IMAC27] = "apple", [CS420X_AUTO] = "auto", }; @@ -1285,7 +1288,13 @@ static const struct snd_pci_quirk cs420x_cfg_tbl[] = { SND_PCI_QUIRK(0x10de, 0x0d94, "MacBookAir 3,1(2)", CS420X_MBP55), SND_PCI_QUIRK(0x10de, 0xcb79, "MacBookPro 5,5", CS420X_MBP55), SND_PCI_QUIRK(0x10de, 0xcb89, "MacBookPro 7,1", CS420X_MBP55), - SND_PCI_QUIRK(0x8086, 0x7270, "IMac 27 Inch", CS420X_IMAC27), + /* this conflicts with too many other models */ + /*SND_PCI_QUIRK(0x8086, 0x7270, "IMac 27 Inch", CS420X_IMAC27),*/ + {} /* terminator */ +}; + +static const struct snd_pci_quirk cs420x_codec_cfg_tbl[] = { + SND_PCI_QUIRK_VENDOR(0x106b, "Apple", CS420X_APPLE), {} /* terminator */ }; @@ -1367,6 +1376,10 @@ static int patch_cs420x(struct hda_codec *codec) spec->board_config = snd_hda_check_board_config(codec, CS420X_MODELS, cs420x_models, cs420x_cfg_tbl); + if (spec->board_config < 0) + spec->board_config = + snd_hda_check_board_codec_sid_config(codec, + CS420X_MODELS, NULL, cs420x_codec_cfg_tbl); if (spec->board_config >= 0) fix_pincfg(codec, spec->board_config, cs_pincfgs); @@ -1374,10 +1387,11 @@ static int patch_cs420x(struct hda_codec *codec) case CS420X_IMAC27: case CS420X_MBP53: case CS420X_MBP55: - /* GPIO1 = headphones */ - /* GPIO3 = speakers */ - spec->gpio_mask = 0x0a; - spec->gpio_dir = 0x0a; + case CS420X_APPLE: + spec->gpio_eapd_hp = 2; /* GPIO1 = headphones */ + spec->gpio_eapd_speaker = 8; /* GPIO3 = speakers */ + spec->gpio_mask = spec->gpio_dir = + spec->gpio_eapd_hp | spec->gpio_eapd_speaker; break; } -- cgit v1.2.3-70-g09d2 From 6759dc323826c2c806c998cd93945c5476688dd2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 23 Nov 2011 07:38:59 +0100 Subject: ALSA: hda/realtek - Fix missing inits of item indices for auto-mic When the imux entries are rebuilt in alc_rebuild_imux_for_auto_mic(), the initialization of index field is missing. It may work without it casually when the original imux was created by the auto-parser, but it's definitely broken in the case of static configs where no imux was parsed beforehand. Because of this, the auto-mic switching doesn't work properly on some model options. This patch adds the missing initialization of index field. Reported-by: Dmitry Nezhevenko Cc: [v3.1] Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 14 +++++++++++++- 1 file changed, 13 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 336d14eb72a..06c0c12d4fe 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1054,8 +1054,20 @@ static bool alc_rebuild_imux_for_auto_mic(struct hda_codec *codec) spec->imux_pins[2] = spec->dock_mic_pin; for (i = 0; i < 3; i++) { strcpy(imux->items[i].label, texts[i]); - if (spec->imux_pins[i]) + if (spec->imux_pins[i]) { + hda_nid_t pin = spec->imux_pins[i]; + int c; + for (c = 0; c < spec->num_adc_nids; c++) { + hda_nid_t cap = spec->capsrc_nids ? + spec->capsrc_nids[c] : spec->adc_nids[c]; + int idx = get_connection_index(codec, cap, pin); + if (idx >= 0) { + imux->items[i].index = idx; + break; + } + } imux->num_items = i + 1; + } } spec->num_mux_defs = 1; spec->input_mux = imux; -- cgit v1.2.3-70-g09d2 From 61071594f64ed12328046f94716d1d744bddc0a1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 23 Nov 2011 07:52:15 +0100 Subject: ALSA: hda/realtek - Minor cleanup Use an inline function for the common pattern for assigning a capsrc. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 24 ++++++++++++------------ 1 file changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 06c0c12d4fe..cbde019d3d5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -277,6 +277,12 @@ static bool alc_dyn_adc_pcm_resetup(struct hda_codec *codec, int cur) return false; } +static inline hda_nid_t get_capsrc(struct alc_spec *spec, int idx) +{ + return spec->capsrc_nids ? + spec->capsrc_nids[idx] : spec->adc_nids[idx]; +} + /* select the given imux item; either unmute exclusively or select the route */ static int alc_mux_select(struct hda_codec *codec, unsigned int adc_idx, unsigned int idx, bool force) @@ -303,8 +309,7 @@ static int alc_mux_select(struct hda_codec *codec, unsigned int adc_idx, adc_idx = spec->dyn_adc_idx[idx]; } - nid = spec->capsrc_nids ? - spec->capsrc_nids[adc_idx] : spec->adc_nids[adc_idx]; + nid = get_capsrc(spec, adc_idx); /* no selection? */ num_conns = snd_hda_get_conn_list(codec, nid, NULL); @@ -1058,8 +1063,7 @@ static bool alc_rebuild_imux_for_auto_mic(struct hda_codec *codec) hda_nid_t pin = spec->imux_pins[i]; int c; for (c = 0; c < spec->num_adc_nids; c++) { - hda_nid_t cap = spec->capsrc_nids ? - spec->capsrc_nids[c] : spec->adc_nids[c]; + hda_nid_t cap = get_capsrc(spec, c); int idx = get_connection_index(codec, cap, pin); if (idx >= 0) { imux->items[i].index = idx; @@ -1969,10 +1973,8 @@ static int alc_build_controls(struct hda_codec *codec) if (!kctl) kctl = snd_hda_find_mixer_ctl(codec, "Input Source"); for (i = 0; kctl && i < kctl->count; i++) { - const hda_nid_t *nids = spec->capsrc_nids; - if (!nids) - nids = spec->adc_nids; - err = snd_hda_add_nid(codec, kctl, i, nids[i]); + err = snd_hda_add_nid(codec, kctl, i, + get_capsrc(spec, i)); if (err < 0) return err; } @@ -2759,8 +2761,7 @@ static int alc_auto_create_input_ctls(struct hda_codec *codec) } for (c = 0; c < num_adcs; c++) { - hda_nid_t cap = spec->capsrc_nids ? - spec->capsrc_nids[c] : spec->adc_nids[c]; + hda_nid_t cap = get_capsrc(spec, c); idx = get_connection_index(codec, cap, pin); if (idx >= 0) { spec->imux_pins[imux->num_items] = pin; @@ -3706,8 +3707,7 @@ static int init_capsrc_for_pin(struct hda_codec *codec, hda_nid_t pin) if (!pin) return 0; for (i = 0; i < spec->num_adc_nids; i++) { - hda_nid_t cap = spec->capsrc_nids ? - spec->capsrc_nids[i] : spec->adc_nids[i]; + hda_nid_t cap = get_capsrc(spec, i); int idx; idx = get_connection_index(codec, cap, pin); -- cgit v1.2.3-70-g09d2 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(-) diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 7bd2a52f2ba..70a7abda7e2 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 v1.2.3-70-g09d2 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(-) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 431c0d417ee..b5137629f8e 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 v1.2.3-70-g09d2