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 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(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 470f6f286e81..f3658658548e 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 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 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(-) (limited to 'sound') diff --git a/sound/pci/cs5535audio/cs5535audio_pcm.c b/sound/pci/cs5535audio/cs5535audio_pcm.c index e083122ca55a..dbf94b189e75 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 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 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(-) (limited to 'sound') diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index e760adad9523..19ee2203cbb5 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 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(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 9850c5b481ea..c505fd5d338c 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 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(-) (limited to 'sound') diff --git a/sound/pci/lx6464es/lx_core.c b/sound/pci/lx6464es/lx_core.c index 5c8717e29eeb..ad52f4187e40 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 1dd562980b6c..4d7ff797a646 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 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(-) (limited to 'sound') diff --git a/sound/pci/lx6464es/lx_core.c b/sound/pci/lx6464es/lx_core.c index ad52f4187e40..8c3e7fcefd99 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 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(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index 7ae7578bdcc0..c1da422e085a 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 From 72531c9434fa884d20cb3c36fcec83752f32fdf4 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 22 Nov 2011 09:46:51 +0800 Subject: ASoC: Fix wrong define for AD1836_ADC_WORD_OFFSET According to the datasheet: The BIT[5:4] of ADC Control Register 2 is to control the word width. 00 = 25 Bits 01 = 20 Bits 10 = 16 Bits 11 = Invalid Thus, the AD1836_ADC_WORD_OFFSET should be defined as 4. Signed-off-by: Axel Lin Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/ad1836.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/ad1836.h b/sound/soc/codecs/ad1836.h index 444747f0db26..dd7be0dbbc58 100644 --- a/sound/soc/codecs/ad1836.h +++ b/sound/soc/codecs/ad1836.h @@ -34,7 +34,7 @@ #define AD1836_ADC_CTRL2 13 #define AD1836_ADC_WORD_LEN_MASK 0x30 -#define AD1836_ADC_WORD_OFFSET 5 +#define AD1836_ADC_WORD_OFFSET 4 #define AD1836_ADC_SERFMT_MASK (7 << 6) #define AD1836_ADC_SERFMT_PCK256 (0x4 << 6) #define AD1836_ADC_SERFMT_PCK128 (0x5 << 6) -- cgit v1.2.3 From 5c4b2aa3fd1dc30af098de5dec766a817621ace2 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 22 Nov 2011 14:47:44 +0800 Subject: ASoC: max9877: Update register if either val or val2 is changed In the case of ((max9877_regs[reg] >> shift) & mask) != val but ((max9877_regs[reg2] >> shift) & mask) == val2, current code does not update the registers. Fix the logic to update registers if either val or val2 is changed. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/max9877.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/max9877.c b/sound/soc/codecs/max9877.c index 9e7e964a5fa3..dcf6f2a1600a 100644 --- a/sound/soc/codecs/max9877.c +++ b/sound/soc/codecs/max9877.c @@ -106,13 +106,13 @@ static int max9877_set_2reg(struct snd_kcontrol *kcontrol, unsigned int mask = mc->max; unsigned int val = (ucontrol->value.integer.value[0] & mask); unsigned int val2 = (ucontrol->value.integer.value[1] & mask); - unsigned int change = 1; + unsigned int change = 0; - if (((max9877_regs[reg] >> shift) & mask) == val) - change = 0; + if (((max9877_regs[reg] >> shift) & mask) != val) + change = 1; - if (((max9877_regs[reg2] >> shift) & mask) == val2) - change = 0; + if (((max9877_regs[reg2] >> shift) & mask) != val2) + change = 1; if (change) { max9877_regs[reg] &= ~(mask << shift); -- cgit v1.2.3 From 3d94a2a53a3979c30620e3adea10f20bef8267b3 Mon Sep 17 00:00:00 2001 From: Boojin Kim Date: Tue, 22 Nov 2011 11:03:22 +0900 Subject: ASoC: SAMSUNG: Fix build error This patch adds to fix following build errors. sound/soc/codecs/wm8994.c: In function 'wm8994_readable': sound/soc/codecs/wm8994.c:58: warning: unused variable 'wm8994' sound/soc/samsung/smdk_wm8994.c:176: error: expected declaration specifiers or '...' before string constant sound/soc/samsung/smdk_wm8994.c:176: warning: data definition has no type or storage class sound/soc/samsung/smdk_wm8994.c:176: warning: type defaults to 'int' in declaration of 'MODULE_DESCRIPTION' sound/soc/samsung/smdk_wm8994.c:176: warning: function declaration isn't a prototype sound/soc/samsung/smdk_wm8994.c:177: error: expected declaration specifiers or '...' before string constant Signed-off-by: Boojin Kim Signed-off-by: Mark Brown --- sound/soc/samsung/smdk_wm8994.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c index f75e43997d5b..ad9ac42522e2 100644 --- a/sound/soc/samsung/smdk_wm8994.c +++ b/sound/soc/samsung/smdk_wm8994.c @@ -9,6 +9,7 @@ #include "../codecs/wm8994.h" #include +#include /* * Default CFG switch settings to use this driver: -- cgit v1.2.3 From d66b8537b30fbaf79e0f467fa6b7e1a2191cba83 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Tue, 22 Nov 2011 17:17:23 +0100 Subject: ASoC: cs4720: use snd_soc_cache_sync() Replace the manual register restore mechanism in cs4270.c and call the generic snd_soc_cache_sync() handler instead. This factors code out in favour of core facilities and also fixes a bus confusion that is most probably caused by intermixing i2c-regmap functions and i2c_smbus_* accessors. Signed-off-by: Daniel Mack Reported-and-tested-by: Sven Neumann Acked-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/codecs/cs4270.c | 10 +--------- 1 file changed, 1 insertion(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index f1f237ecec2a..73f46eb459f1 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -601,7 +601,6 @@ static int cs4270_soc_suspend(struct snd_soc_codec *codec, pm_message_t mesg) static int cs4270_soc_resume(struct snd_soc_codec *codec) { struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec); - struct i2c_client *i2c_client = to_i2c_client(codec->dev); int reg; regulator_bulk_enable(ARRAY_SIZE(cs4270->supplies), @@ -612,14 +611,7 @@ static int cs4270_soc_resume(struct snd_soc_codec *codec) ndelay(500); /* first restore the entire register cache ... */ - for (reg = CS4270_FIRSTREG; reg <= CS4270_LASTREG; reg++) { - u8 val = snd_soc_read(codec, reg); - - if (i2c_smbus_write_byte_data(i2c_client, reg, val)) { - dev_err(codec->dev, "i2c write failed\n"); - return -EIO; - } - } + snd_soc_cache_sync(codec); /* ... then disable the power-down bits */ reg = snd_soc_read(codec, CS4270_PWRCTL); -- cgit v1.2.3 From 380c88303812951f6c838241366a66a03fb5c897 Mon Sep 17 00:00:00 2001 From: Timur Tabi Date: Tue, 22 Nov 2011 14:38:59 -0600 Subject: ASoC: mpc8610: tell the CS4270 codec that it's the master Commit ac601555 ("ASoC: Return early with -EINVAL if invalid dai format is detected") requires the machine driver to tell the CS4270 codec driver whether the CS4270 should be configured for master or slave operation. Signed-off-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/fsl/mpc8610_hpcd.c | 24 ++++++++++++++++-------- 1 file changed, 16 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 31af405bda84..ae49f1c78c6d 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -392,7 +392,8 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) } if (strcasecmp(sprop, "i2s-slave") == 0) { - machine_data->dai_format = SND_SOC_DAIFMT_I2S; + machine_data->dai_format = + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM; machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; @@ -409,31 +410,38 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) } machine_data->clk_frequency = be32_to_cpup(iprop); } else if (strcasecmp(sprop, "i2s-master") == 0) { - machine_data->dai_format = SND_SOC_DAIFMT_I2S; + machine_data->dai_format = + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS; machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; } else if (strcasecmp(sprop, "lj-slave") == 0) { - machine_data->dai_format = SND_SOC_DAIFMT_LEFT_J; + machine_data->dai_format = + SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBM_CFM; machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; } else if (strcasecmp(sprop, "lj-master") == 0) { - machine_data->dai_format = SND_SOC_DAIFMT_LEFT_J; + machine_data->dai_format = + SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBS_CFS; machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; } else if (strcasecmp(sprop, "rj-slave") == 0) { - machine_data->dai_format = SND_SOC_DAIFMT_RIGHT_J; + machine_data->dai_format = + SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_CBM_CFM; machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; } else if (strcasecmp(sprop, "rj-master") == 0) { - machine_data->dai_format = SND_SOC_DAIFMT_RIGHT_J; + machine_data->dai_format = + SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_CBS_CFS; machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; } else if (strcasecmp(sprop, "ac97-slave") == 0) { - machine_data->dai_format = SND_SOC_DAIFMT_AC97; + machine_data->dai_format = + SND_SOC_DAIFMT_AC97 | SND_SOC_DAIFMT_CBM_CFM; machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; } else if (strcasecmp(sprop, "ac97-master") == 0) { - machine_data->dai_format = SND_SOC_DAIFMT_AC97; + machine_data->dai_format = + SND_SOC_DAIFMT_AC97 | SND_SOC_DAIFMT_CBS_CFS; machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; } else { -- cgit v1.2.3 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(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index e44b107fdc75..4562e9de6a1a 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 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(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 2fbab8e29576..7bd2a52f2bac 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 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(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 336d14eb72af..06c0c12d4fec 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 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(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 06c0c12d4fec..cbde019d3d52 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 From 4ca8af579c9748376db537575f7a811c179fe50a Mon Sep 17 00:00:00 2001 From: Paul Bolle Date: Wed, 23 Nov 2011 10:39:10 +0100 Subject: ASoC: drop support for PlayPaq with WM8510 SoC Audio support for PlayPaq with WM8510 got added in commit 9aaca9683b ("[ALSA] Revised AT32 ASoC Patch"). That support depends on BOARD_PLAYPAQ. That Kconfig symbol didn't exist when that support got added in v2.6.27. It still doesn't. It has never been possible to even build this driver. Drop it. Signed-off-by: Paul Bolle Signed-off-by: Mark Brown --- sound/soc/atmel/Kconfig | 21 +- sound/soc/atmel/Makefile | 4 - sound/soc/atmel/playpaq_wm8510.c | 473 --------------------------------------- 3 files changed, 1 insertion(+), 497 deletions(-) delete mode 100644 sound/soc/atmel/playpaq_wm8510.c (limited to 'sound') diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index bee3c94f58b0..d1fcc816ce97 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -1,6 +1,6 @@ config SND_ATMEL_SOC tristate "SoC Audio for the Atmel System-on-Chip" - depends on ARCH_AT91 || AVR32 + depends on ARCH_AT91 help Say Y or M if you want to add support for codecs attached to the ATMEL SSC interface. You will also need @@ -24,25 +24,6 @@ config SND_AT91_SOC_SAM9G20_WM8731 Say Y if you want to add support for SoC audio on WM8731-based AT91sam9g20 evaluation board. -config SND_AT32_SOC_PLAYPAQ - tristate "SoC Audio support for PlayPaq with WM8510" - depends on SND_ATMEL_SOC && BOARD_PLAYPAQ && AT91_PROGRAMMABLE_CLOCKS - select SND_ATMEL_SOC_SSC - select SND_SOC_WM8510 - help - Say Y or M here if you want to add support for SoC audio - on the LRS PlayPaq. - -config SND_AT32_SOC_PLAYPAQ_SLAVE - bool "Run CODEC on PlayPaq in slave mode" - depends on SND_AT32_SOC_PLAYPAQ - default n - help - Say Y if you want to run with the AT32 SSC generating the BCLK - and FRAME signals on the PlayPaq. Unless you want to play - with the AT32 as the SSC master, you probably want to say N here, - as this will give you better sound quality. - config SND_AT91_SOC_AFEB9260 tristate "SoC Audio support for AFEB9260 board" depends on ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile index e7ea56bd5f82..a5c0bf19da78 100644 --- a/sound/soc/atmel/Makefile +++ b/sound/soc/atmel/Makefile @@ -8,9 +8,5 @@ obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel_ssc_dai.o # AT91 Machine Support snd-soc-sam9g20-wm8731-objs := sam9g20_wm8731.o -# AT32 Machine Support -snd-soc-playpaq-objs := playpaq_wm8510.o - obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o -obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o obj-$(CONFIG_SND_AT91_SOC_AFEB9260) += snd-soc-afeb9260.o diff --git a/sound/soc/atmel/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c deleted file mode 100644 index 73ae99ad4578..000000000000 --- a/sound/soc/atmel/playpaq_wm8510.c +++ /dev/null @@ -1,473 +0,0 @@ -/* sound/soc/at32/playpaq_wm8510.c - * ASoC machine driver for PlayPaq using WM8510 codec - * - * Copyright (C) 2008 Long Range Systems - * Geoffrey Wossum - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - * - * This code is largely inspired by sound/soc/at91/eti_b1_wm8731.c - * - * NOTE: If you don't have the AT32 enhanced portmux configured (which - * isn't currently in the mainline or Atmel patched kernel), you will - * need to set the MCLK pin (PA30) to peripheral A in your board initialization - * code. Something like: - * at32_select_periph(GPIO_PIN_PA(30), GPIO_PERIPH_A, 0); - * - */ - -/* #define DEBUG */ - -#include -#include -#include -#include -#include -#include -#include -#include - -#include -#include -#include -#include - -#include -#include - -#include "../codecs/wm8510.h" -#include "atmel-pcm.h" -#include "atmel_ssc_dai.h" - - -/*-------------------------------------------------------------------------*\ - * constants -\*-------------------------------------------------------------------------*/ -#define MCLK_PIN GPIO_PIN_PA(30) -#define MCLK_PERIPH GPIO_PERIPH_A - - -/*-------------------------------------------------------------------------*\ - * data types -\*-------------------------------------------------------------------------*/ -/* SSC clocking data */ -struct ssc_clock_data { - /* CMR div */ - unsigned int cmr_div; - - /* Frame period (as needed by xCMR.PERIOD) */ - unsigned int period; - - /* The SSC clock rate these settings where calculated for */ - unsigned long ssc_rate; -}; - - -/*-------------------------------------------------------------------------*\ - * module data -\*-------------------------------------------------------------------------*/ -static struct clk *_gclk0; -static struct clk *_pll0; - -#define CODEC_CLK (_gclk0) - - -/*-------------------------------------------------------------------------*\ - * Sound SOC operations -\*-------------------------------------------------------------------------*/ -#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE -static struct ssc_clock_data playpaq_wm8510_calc_ssc_clock( - struct snd_pcm_hw_params *params, - struct snd_soc_dai *cpu_dai) -{ - struct at32_ssc_info *ssc_p = snd_soc_dai_get_drvdata(cpu_dai); - struct ssc_device *ssc = ssc_p->ssc; - struct ssc_clock_data cd; - unsigned int rate, width_bits, channels; - unsigned int bitrate, ssc_div; - unsigned actual_rate; - - - /* - * Figure out required bitrate - */ - rate = params_rate(params); - channels = params_channels(params); - width_bits = snd_pcm_format_physical_width(params_format(params)); - bitrate = rate * width_bits * channels; - - - /* - * Figure out required SSC divider and period for required bitrate - */ - cd.ssc_rate = clk_get_rate(ssc->clk); - ssc_div = cd.ssc_rate / bitrate; - cd.cmr_div = ssc_div / 2; - if (ssc_div & 1) { - /* round cmr_div up */ - cd.cmr_div++; - } - cd.period = width_bits - 1; - - - /* - * Find actual rate, compare to requested rate - */ - actual_rate = (cd.ssc_rate / (cd.cmr_div * 2)) / (2 * (cd.period + 1)); - pr_debug("playpaq_wm8510: Request rate = %u, actual rate = %u\n", - rate, actual_rate); - - - return cd; -} -#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ - - - -static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct at32_ssc_info *ssc_p = snd_soc_dai_get_drvdata(cpu_dai); - struct ssc_device *ssc = ssc_p->ssc; - unsigned int pll_out = 0, bclk = 0, mclk_div = 0; - int ret; - - - /* Due to difficulties with getting the correct clocks from the AT32's - * PLL0, we're going to let the CODEC be in charge of all the clocks - */ -#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE - const unsigned int fmt = (SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); -#else - struct ssc_clock_data cd; - const unsigned int fmt = (SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS); -#endif - - if (ssc == NULL) { - pr_warning("playpaq_wm8510_hw_params: ssc is NULL!\n"); - return -EINVAL; - } - - - /* - * Figure out PLL and BCLK dividers for WM8510 - */ - switch (params_rate(params)) { - case 48000: - pll_out = 24576000; - mclk_div = WM8510_MCLKDIV_2; - bclk = WM8510_BCLKDIV_8; - break; - - case 44100: - pll_out = 22579200; - mclk_div = WM8510_MCLKDIV_2; - bclk = WM8510_BCLKDIV_8; - break; - - case 22050: - pll_out = 22579200; - mclk_div = WM8510_MCLKDIV_4; - bclk = WM8510_BCLKDIV_8; - break; - - case 16000: - pll_out = 24576000; - mclk_div = WM8510_MCLKDIV_6; - bclk = WM8510_BCLKDIV_8; - break; - - case 11025: - pll_out = 22579200; - mclk_div = WM8510_MCLKDIV_8; - bclk = WM8510_BCLKDIV_8; - break; - - case 8000: - pll_out = 24576000; - mclk_div = WM8510_MCLKDIV_12; - bclk = WM8510_BCLKDIV_8; - break; - - default: - pr_warning("playpaq_wm8510: Unsupported sample rate %d\n", - params_rate(params)); - return -EINVAL; - } - - - /* - * set CPU and CODEC DAI configuration - */ - ret = snd_soc_dai_set_fmt(codec_dai, fmt); - if (ret < 0) { - pr_warning("playpaq_wm8510: " - "Failed to set CODEC DAI format (%d)\n", - ret); - return ret; - } - ret = snd_soc_dai_set_fmt(cpu_dai, fmt); - if (ret < 0) { - pr_warning("playpaq_wm8510: " - "Failed to set CPU DAI format (%d)\n", - ret); - return ret; - } - - - /* - * Set CPU clock configuration - */ -#if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE - cd = playpaq_wm8510_calc_ssc_clock(params, cpu_dai); - pr_debug("playpaq_wm8510: cmr_div = %d, period = %d\n", - cd.cmr_div, cd.period); - ret = snd_soc_dai_set_clkdiv(cpu_dai, AT32_SSC_CMR_DIV, cd.cmr_div); - if (ret < 0) { - pr_warning("playpaq_wm8510: Failed to set CPU CMR_DIV (%d)\n", - ret); - return ret; - } - ret = snd_soc_dai_set_clkdiv(cpu_dai, AT32_SSC_TCMR_PERIOD, - cd.period); - if (ret < 0) { - pr_warning("playpaq_wm8510: " - "Failed to set CPU transmit period (%d)\n", - ret); - return ret; - } -#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ - - - /* - * Set CODEC clock configuration - */ - pr_debug("playpaq_wm8510: " - "pll_in = %ld, pll_out = %u, bclk = %x, mclk = %x\n", - clk_get_rate(CODEC_CLK), pll_out, bclk, mclk_div); - - -#if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8510_BCLKDIV, bclk); - if (ret < 0) { - pr_warning - ("playpaq_wm8510: Failed to set CODEC DAI BCLKDIV (%d)\n", - ret); - return ret; - } -#endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ - - - ret = snd_soc_dai_set_pll(codec_dai, 0, 0, - clk_get_rate(CODEC_CLK), pll_out); - if (ret < 0) { - pr_warning("playpaq_wm8510: Failed to set CODEC DAI PLL (%d)\n", - ret); - return ret; - } - - - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8510_MCLKDIV, mclk_div); - if (ret < 0) { - pr_warning("playpaq_wm8510: Failed to set CODEC MCLKDIV (%d)\n", - ret); - return ret; - } - - - return 0; -} - - - -static struct snd_soc_ops playpaq_wm8510_ops = { - .hw_params = playpaq_wm8510_hw_params, -}; - - - -static const struct snd_soc_dapm_widget playpaq_dapm_widgets[] = { - SND_SOC_DAPM_MIC("Int Mic", NULL), - SND_SOC_DAPM_SPK("Ext Spk", NULL), -}; - - - -static const struct snd_soc_dapm_route intercon[] = { - /* speaker connected to SPKOUT */ - {"Ext Spk", NULL, "SPKOUTP"}, - {"Ext Spk", NULL, "SPKOUTN"}, - - {"Mic Bias", NULL, "Int Mic"}, - {"MICN", NULL, "Mic Bias"}, - {"MICP", NULL, "Mic Bias"}, -}; - - - -static int playpaq_wm8510_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - int i; - - /* - * Add DAPM widgets - */ - for (i = 0; i < ARRAY_SIZE(playpaq_dapm_widgets); i++) - snd_soc_dapm_new_control(dapm, &playpaq_dapm_widgets[i]); - - - - /* - * Setup audio path interconnects - */ - snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); - - - - /* always connected pins */ - snd_soc_dapm_enable_pin(dapm, "Int Mic"); - snd_soc_dapm_enable_pin(dapm, "Ext Spk"); - - - - /* Make CSB show PLL rate */ - snd_soc_dai_set_clkdiv(rtd->codec_dai, WM8510_OPCLKDIV, - WM8510_OPCLKDIV_1 | 4); - - return 0; -} - - - -static struct snd_soc_dai_link playpaq_wm8510_dai = { - .name = "WM8510", - .stream_name = "WM8510 PCM", - .cpu_dai_name= "atmel-ssc-dai.0", - .platform_name = "atmel-pcm-audio", - .codec_name = "wm8510-codec.0-0x1a", - .codec_dai_name = "wm8510-hifi", - .init = playpaq_wm8510_init, - .ops = &playpaq_wm8510_ops, -}; - - - -static struct snd_soc_card snd_soc_playpaq = { - .name = "LRS_PlayPaq_WM8510", - .dai_link = &playpaq_wm8510_dai, - .num_links = 1, -}; - -static struct platform_device *playpaq_snd_device; - - -static int __init playpaq_asoc_init(void) -{ - int ret = 0; - - /* - * Configure MCLK for WM8510 - */ - _gclk0 = clk_get(NULL, "gclk0"); - if (IS_ERR(_gclk0)) { - _gclk0 = NULL; - ret = PTR_ERR(_gclk0); - goto err_gclk0; - } - _pll0 = clk_get(NULL, "pll0"); - if (IS_ERR(_pll0)) { - _pll0 = NULL; - ret = PTR_ERR(_pll0); - goto err_pll0; - } - ret = clk_set_parent(_gclk0, _pll0); - if (ret) { - pr_warning("snd-soc-playpaq: " - "Failed to set PLL0 as parent for DAC clock\n"); - goto err_set_clk; - } - clk_set_rate(CODEC_CLK, 12000000); - clk_enable(CODEC_CLK); - -#if defined CONFIG_AT32_ENHANCED_PORTMUX - at32_select_periph(MCLK_PIN, MCLK_PERIPH, 0); -#endif - - - /* - * Create and register platform device - */ - playpaq_snd_device = platform_device_alloc("soc-audio", 0); - if (playpaq_snd_device == NULL) { - ret = -ENOMEM; - goto err_device_alloc; - } - - platform_set_drvdata(playpaq_snd_device, &snd_soc_playpaq); - - ret = platform_device_add(playpaq_snd_device); - if (ret) { - pr_warning("playpaq_wm8510: platform_device_add failed (%d)\n", - ret); - goto err_device_add; - } - - return 0; - - -err_device_add: - if (playpaq_snd_device != NULL) { - platform_device_put(playpaq_snd_device); - playpaq_snd_device = NULL; - } -err_device_alloc: -err_set_clk: - if (_pll0 != NULL) { - clk_put(_pll0); - _pll0 = NULL; - } -err_pll0: - if (_gclk0 != NULL) { - clk_put(_gclk0); - _gclk0 = NULL; - } - return ret; -} - - -static void __exit playpaq_asoc_exit(void) -{ - if (_gclk0 != NULL) { - clk_put(_gclk0); - _gclk0 = NULL; - } - if (_pll0 != NULL) { - clk_put(_pll0); - _pll0 = NULL; - } - -#if defined CONFIG_AT32_ENHANCED_PORTMUX - at32_free_pin(MCLK_PIN); -#endif - - platform_device_unregister(playpaq_snd_device); - playpaq_snd_device = NULL; -} - -module_init(playpaq_asoc_init); -module_exit(playpaq_asoc_exit); - -MODULE_AUTHOR("Geoffrey Wossum "); -MODULE_DESCRIPTION("ASoC machine driver for LRS PlayPaq"); -MODULE_LICENSE("GPL"); -- cgit v1.2.3 From b284362b6b45150d171ff5bed92bc416b040aead Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 23 Nov 2011 12:46:11 +0800 Subject: ASoC: cs42l51: Fix off-by-one for reg_cache_size Just checking the code in cs42l51_fill_cache(): The cache pointer points to codec->reg_cache + 1. I think it is because CS42L51_FIRSTREG is 0x01, so codec->reg_cache[0] is not used here. Then we read CS42L51_NUMREGS bytes to cache. So we need reg_cache_size to be CS42L51_NUMREGS + 1. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l51.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 8c3c8205d19e..1ee66361f61b 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -555,7 +555,7 @@ static int cs42l51_probe(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_device_cs42l51 = { .probe = cs42l51_probe, - .reg_cache_size = CS42L51_NUMREGS, + .reg_cache_size = CS42L51_NUMREGS + 1, .reg_word_size = sizeof(u8), }; -- cgit v1.2.3 From 5ff1ddf22b2584d00d7e0ba5a8eab07b5338bd84 Mon Sep 17 00:00:00 2001 From: Eric Miao Date: Wed, 23 Nov 2011 22:37:00 +0800 Subject: ASoC: skip resume of soc-audio devices without codecs There are cases where there is no working codec on the soc-audio devices, and snd_soc_suspend() will skip such device when suspending. Yet its counterpart snd_soc_resume() does not check this, causing complaints about spinlock lockup: [ 176.726087] BUG: spinlock lockup on CPU#0, kworker/0:2/1067, d8ab82a8 [ 176.732539] [<80014a14>] (unwind_backtrace+0x0/0xec) from [<805b3fc8>] (dump_stack+0x20/0x24) [ 176.741082] [<805b3fc8>] (dump_stack+0x20/0x24) from [<80322208>] (do_raw_spin_lock+0x118/0x158) [ 176.749882] [<80322208>] (do_raw_spin_lock+0x118/0x158) from [<805b7874>] (_raw_spin_lock_irqsave+0x5c/0x68) [ 176.759723] [<805b7874>] (_raw_spin_lock_irqsave+0x5c/0x68) from [<8002a020>] (__wake_up+0x2c/0x5c) [ 176.768781] [<8002a020>] (__wake_up+0x2c/0x5c) from [<804a6de8>] (soc_resume_deferred+0x3c/0x2b0) [ 176.777666] [<804a6de8>] (soc_resume_deferred+0x3c/0x2b0) from [<8004ee20>] (process_one_work+0x2e8/0x50c) [ 176.787334] [<8004ee20>] (process_one_work+0x2e8/0x50c) from [<8004fd08>] (worker_thread+0x1c8/0x2e0) [ 176.796566] [<8004fd08>] (worker_thread+0x1c8/0x2e0) from [<80053ec8>] (kthread+0xa4/0xb0) [ 176.804843] [<80053ec8>] (kthread+0xa4/0xb0) from [<8000ea70>] (kernel_thread_exit+0x0/0x8) Signed-off-by: Eric Miao Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a5d3685a5d38..a25fa63ce9a2 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -709,6 +709,12 @@ int snd_soc_resume(struct device *dev) struct snd_soc_card *card = dev_get_drvdata(dev); int i, ac97_control = 0; + /* If the initialization of this soc device failed, there is no codec + * associated with it. Just bail out in this case. + */ + if (list_empty(&card->codec_dev_list)) + return 0; + /* AC97 devices might have other drivers hanging off them so * need to resume immediately. Other drivers don't have that * problem and may take a substantial amount of time to resume -- cgit v1.2.3 From 97371fa99c1900a84a5220639edd726b35d73931 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 25 Nov 2011 00:23:28 +0100 Subject: ARM: 7175/1: add subname parameter to mfp_set_groupg callers commit 798681bf "ARM: 7158/1: add new MFP implement for NUC900" adds subname parameter for mfp_set_groupg. Thus add subname parameter to the callers. Signed-off-by: Axel Lin Acked-by: Mark Brown Acked-by: Wan Zongshun Signed-off-by: Russell King --- drivers/i2c/busses/i2c-nuc900.c | 2 +- drivers/spi/spi-nuc900.c | 2 +- sound/soc/nuc900/nuc900-ac97.c | 3 ++- 3 files changed, 4 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/drivers/i2c/busses/i2c-nuc900.c b/drivers/i2c/busses/i2c-nuc900.c index 835e47b39bc2..03b615778887 100644 --- a/drivers/i2c/busses/i2c-nuc900.c +++ b/drivers/i2c/busses/i2c-nuc900.c @@ -593,7 +593,7 @@ static int __devinit nuc900_i2c_probe(struct platform_device *pdev) i2c->adap.algo_data = i2c; i2c->adap.dev.parent = &pdev->dev; - mfp_set_groupg(&pdev->dev); + mfp_set_groupg(&pdev->dev, NULL); clk_get_rate(i2c->clk); diff --git a/drivers/spi/spi-nuc900.c b/drivers/spi/spi-nuc900.c index e763254741c2..21c70b2b8311 100644 --- a/drivers/spi/spi-nuc900.c +++ b/drivers/spi/spi-nuc900.c @@ -426,7 +426,7 @@ static int __devinit nuc900_spi_probe(struct platform_device *pdev) goto err_clk; } - mfp_set_groupg(&pdev->dev); + mfp_set_groupg(&pdev->dev, NULL); nuc900_init_spi(hw); err = spi_bitbang_start(&hw->bitbang); diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c index 9c0edad90d8b..a4e3237956e2 100644 --- a/sound/soc/nuc900/nuc900-ac97.c +++ b/sound/soc/nuc900/nuc900-ac97.c @@ -365,7 +365,8 @@ static int __devinit nuc900_ac97_drvprobe(struct platform_device *pdev) if (ret) goto out3; - mfp_set_groupg(nuc900_audio->dev); /* enbale ac97 multifunction pin*/ + /* enbale ac97 multifunction pin */ + mfp_set_groupg(nuc900_audio->dev, "nuc900-audio"); return 0; -- cgit v1.2.3 From 5b895eec118ab5fec7b69102d73c1b04a86140b9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 27 Nov 2011 15:54:49 +0000 Subject: ASoC: Correct name of Speyside Main Speaker widget Signed-off-by: Mark Brown --- sound/soc/samsung/speyside.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index 85bf541a771d..4b8e35410eb1 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -191,7 +191,7 @@ static int speyside_late_probe(struct snd_soc_card *card) snd_soc_dapm_ignore_suspend(&card->dapm, "Headset Mic"); snd_soc_dapm_ignore_suspend(&card->dapm, "Main AMIC"); snd_soc_dapm_ignore_suspend(&card->dapm, "Main DMIC"); - snd_soc_dapm_ignore_suspend(&card->dapm, "Speaker"); + snd_soc_dapm_ignore_suspend(&card->dapm, "Main Speaker"); snd_soc_dapm_ignore_suspend(&card->dapm, "WM1250 Output"); snd_soc_dapm_ignore_suspend(&card->dapm, "WM1250 Input"); -- 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 fc07ecd851bd082265b52838eff12f50b88f6114 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 28 Nov 2011 21:16:56 +0000 Subject: ASoC: Error out if we can't generate a LRCLK at all for WM8994 Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 9c982e47eb99..36ba1edfff80 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2357,6 +2357,11 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, bclk |= best << WM8994_AIF1_BCLK_DIV_SHIFT; lrclk = bclk_rate / params_rate(params); + if (!lrclk) { + dev_err(dai->dev, "Unable to generate LRCLK from %dHz BCLK\n", + bclk_rate); + return -EINVAL; + } dev_dbg(dai->dev, "Using LRCLK rate %d for actual LRCLK %dHz\n", lrclk, bclk_rate / lrclk); -- cgit v1.2.3 From fc8e6e8668e74fbf8e00d6e143d7f43b20f73f32 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 28 Nov 2011 18:48:46 +0000 Subject: ASoC: Supply dcs_codes for newer WM1811 revisions Based on initial data. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 36ba1edfff80..6c2988549003 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3183,6 +3183,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) switch (wm8994->revision) { case 0: case 1: + case 2: + case 3: wm8994->hubs.dcs_codes_l = -9; wm8994->hubs.dcs_codes_r = -5; break; -- 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 From fc084e0b930d546872ab23667052499f7daf0fed Mon Sep 17 00:00:00 2001 From: David Dillow Date: Thu, 1 Dec 2011 23:26:53 -0500 Subject: ALSA: sis7019 - give slow codecs more time to reset There are some AC97 codec and board combinations that have been observed to take a very long time to respond after the cold reset has completed. In one case, more than 350 ms was required. To allow users to have sound on those platforms, we'll wait up to 500ms for the codec to become ready. As a board may have multiple codecs, with some faster than others to reset, we add a module parameter to inform the driver which codecs should be present. Reported-by: KotCzarny Signed-off-by: David Dillow Cc: Signed-off-by: Takashi Iwai --- sound/pci/sis7019.c | 64 ++++++++++++++++++++++++++++++++++++++++++++--------- 1 file changed, 53 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c index a391e622a192..28dfafb56dd1 100644 --- a/sound/pci/sis7019.c +++ b/sound/pci/sis7019.c @@ -41,6 +41,7 @@ MODULE_SUPPORTED_DEVICE("{{SiS,SiS7019 Audio Accelerator}}"); static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */ static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ static int enable = 1; +static int codecs = 1; module_param(index, int, 0444); MODULE_PARM_DESC(index, "Index value for SiS7019 Audio Accelerator."); @@ -48,6 +49,8 @@ module_param(id, charp, 0444); MODULE_PARM_DESC(id, "ID string for SiS7019 Audio Accelerator."); module_param(enable, bool, 0444); MODULE_PARM_DESC(enable, "Enable SiS7019 Audio Accelerator."); +module_param(codecs, int, 0444); +MODULE_PARM_DESC(codecs, "Set bit to indicate that codec number is expected to be present (default 1)"); static DEFINE_PCI_DEVICE_TABLE(snd_sis7019_ids) = { { PCI_DEVICE(PCI_VENDOR_ID_SI, 0x7019) }, @@ -140,6 +143,9 @@ struct sis7019 { dma_addr_t silence_dma_addr; }; +/* These values are also used by the module param 'codecs' to indicate + * which codecs should be present. + */ #define SIS_PRIMARY_CODEC_PRESENT 0x0001 #define SIS_SECONDARY_CODEC_PRESENT 0x0002 #define SIS_TERTIARY_CODEC_PRESENT 0x0004 @@ -1078,6 +1084,7 @@ static int sis_chip_init(struct sis7019 *sis) { unsigned long io = sis->ioport; void __iomem *ioaddr = sis->ioaddr; + unsigned long timeout; u16 status; int count; int i; @@ -1104,21 +1111,45 @@ static int sis_chip_init(struct sis7019 *sis) while ((inw(io + SIS_AC97_STATUS) & SIS_AC97_STATUS_BUSY) && --count) udelay(1); + /* Command complete, we can let go of the semaphore now. + */ + outl(SIS_AC97_SEMA_RELEASE, io + SIS_AC97_SEMA); + if (!count) + return -EIO; + /* Now that we've finished the reset, find out what's attached. + * There are some codec/board combinations that take an extremely + * long time to come up. 350+ ms has been observed in the field, + * so we'll give them up to 500ms. */ - status = inl(io + SIS_AC97_STATUS); - if (status & SIS_AC97_STATUS_CODEC_READY) - sis->codecs_present |= SIS_PRIMARY_CODEC_PRESENT; - if (status & SIS_AC97_STATUS_CODEC2_READY) - sis->codecs_present |= SIS_SECONDARY_CODEC_PRESENT; - if (status & SIS_AC97_STATUS_CODEC3_READY) - sis->codecs_present |= SIS_TERTIARY_CODEC_PRESENT; - - /* All done, let go of the semaphore, and check for errors + sis->codecs_present = 0; + timeout = msecs_to_jiffies(500) + jiffies; + while (time_before_eq(jiffies, timeout)) { + status = inl(io + SIS_AC97_STATUS); + if (status & SIS_AC97_STATUS_CODEC_READY) + sis->codecs_present |= SIS_PRIMARY_CODEC_PRESENT; + if (status & SIS_AC97_STATUS_CODEC2_READY) + sis->codecs_present |= SIS_SECONDARY_CODEC_PRESENT; + if (status & SIS_AC97_STATUS_CODEC3_READY) + sis->codecs_present |= SIS_TERTIARY_CODEC_PRESENT; + + if (sis->codecs_present == codecs) + break; + + msleep(1); + } + + /* All done, check for errors. */ - outl(SIS_AC97_SEMA_RELEASE, io + SIS_AC97_SEMA); - if (!sis->codecs_present || !count) + if (!sis->codecs_present) { + printk(KERN_ERR "sis7019: could not find any codecs\n"); return -EIO; + } + + if (sis->codecs_present != codecs) { + printk(KERN_WARNING "sis7019: missing codecs, found %0x, expected %0x\n", + sis->codecs_present, codecs); + } /* Let the hardware know that the audio driver is alive, * and enable PCM slots on the AC-link for L/R playback (3 & 4) and @@ -1390,6 +1421,17 @@ static int __devinit snd_sis7019_probe(struct pci_dev *pci, if (!enable) goto error_out; + /* The user can specify which codecs should be present so that we + * can wait for them to show up if they are slow to recover from + * the AC97 cold reset. We default to a single codec, the primary. + * + * We assume that SIS_PRIMARY_*_PRESENT matches bits 0-2. + */ + codecs &= SIS_PRIMARY_CODEC_PRESENT | SIS_SECONDARY_CODEC_PRESENT | + SIS_TERTIARY_CODEC_PRESENT; + if (!codecs) + codecs = SIS_PRIMARY_CODEC_PRESENT; + rc = snd_card_create(index, id, THIS_MODULE, sizeof(*sis), &card); if (rc < 0) goto error_out; -- cgit v1.2.3 From cce4aa378a049f4275416ee6302dd24f37b289df Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 2 Dec 2011 15:29:12 +0100 Subject: ALSA: hda/realtek - Fix Oops in alc_mux_select() When no imux is available (e.g. a single capture source), alc_auto_init_input_src() may trigger an Oops due to the access to -1. Add a proper zero-check to avoid it. Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index cbde019d3d52..a7d1bc4e0d09 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -297,6 +297,8 @@ static int alc_mux_select(struct hda_codec *codec, unsigned int adc_idx, imux = &spec->input_mux[mux_idx]; if (!imux->num_items && mux_idx > 0) imux = &spec->input_mux[0]; + if (!imux->num_items) + return 0; if (idx >= imux->num_items) idx = imux->num_items - 1; -- cgit v1.2.3 From 87b86ade8bb07473596e2551de7bb64c1f44bbe4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 14 Aug 2011 13:39:20 +0900 Subject: ASoC: Mark WM8994 ADC muxes as virtual Since they don't actually have power bits but do have events associated with them it's important that we bootstrap their state properly which making them virtual does. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 6c2988549003..d0c545b73d78 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1325,15 +1325,15 @@ SND_SOC_DAPM_DAC("DAC1R", NULL, WM8994_POWER_MANAGEMENT_5, 0, 0), }; static const struct snd_soc_dapm_widget wm8994_adc_revd_widgets[] = { -SND_SOC_DAPM_MUX_E("ADCL Mux", WM8994_POWER_MANAGEMENT_4, 1, 0, &adcl_mux, - adc_mux_ev, SND_SOC_DAPM_PRE_PMU), -SND_SOC_DAPM_MUX_E("ADCR Mux", WM8994_POWER_MANAGEMENT_4, 0, 0, &adcr_mux, - adc_mux_ev, SND_SOC_DAPM_PRE_PMU), +SND_SOC_DAPM_VIRT_MUX_E("ADCL Mux", WM8994_POWER_MANAGEMENT_4, 1, 0, &adcl_mux, + adc_mux_ev, SND_SOC_DAPM_PRE_PMU), +SND_SOC_DAPM_VIRT_MUX_E("ADCR Mux", WM8994_POWER_MANAGEMENT_4, 0, 0, &adcr_mux, + adc_mux_ev, SND_SOC_DAPM_PRE_PMU), }; static const struct snd_soc_dapm_widget wm8994_adc_widgets[] = { -SND_SOC_DAPM_MUX("ADCL Mux", WM8994_POWER_MANAGEMENT_4, 1, 0, &adcl_mux), -SND_SOC_DAPM_MUX("ADCR Mux", WM8994_POWER_MANAGEMENT_4, 0, 0, &adcr_mux), +SND_SOC_DAPM_VIRT_MUX("ADCL Mux", WM8994_POWER_MANAGEMENT_4, 1, 0, &adcl_mux), +SND_SOC_DAPM_VIRT_MUX("ADCR Mux", WM8994_POWER_MANAGEMENT_4, 0, 0, &adcr_mux), }; static const struct snd_soc_dapm_widget wm8994_dapm_widgets[] = { -- cgit v1.2.3 From 36d54dc0c893b143748bcf13a1e3b7a00696115d Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sat, 3 Dec 2011 18:38:25 +0800 Subject: ASoC: kirkwood: Make SND_KIRKWOOD_SOC_OPENRD and SND_KIRKWOOD_SOC_T5325 depend on I2C SND_KIRKWOOD_SOC_T5325 selects SND_SOC_ALC5623, but SND_SOC_ALC5623 needs CONFIG_I2C. So we need to make SND_KIRKWOOD_SOC_T5325 depend on I2C, otherwise I got below build error if CONFIG_I2C is not selected. CC sound/soc/codecs/alc5623.o sound/soc/codecs/alc5623.c: In function 'alc5623_i2c_probe': sound/soc/codecs/alc5623.c:1002: error: implicit declaration of function 'i2c_smbus_read_word_data' sound/soc/codecs/alc5623.c:1009: error: implicit declaration of function 'i2c_smbus_read_byte_data' sound/soc/codecs/alc5623.c: In function 'alc5623_modinit': sound/soc/codecs/alc5623.c:1096: error: implicit declaration of function 'i2c_add_driver' sound/soc/codecs/alc5623.c: In function 'alc5623_modexit': sound/soc/codecs/alc5623.c:1108: error: implicit declaration of function 'i2c_del_driver' make[3]: *** [sound/soc/codecs/alc5623.o] Error 1 make[2]: *** [sound/soc/codecs] Error 2 make[1]: *** [sound/soc] Error 2 make: *** [sound] Error 2 Also fix the same issue for SND_KIRKWOOD_SOC_OPENRD. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/kirkwood/Kconfig | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig index 8f49e165f4d1..c62d715235e2 100644 --- a/sound/soc/kirkwood/Kconfig +++ b/sound/soc/kirkwood/Kconfig @@ -12,6 +12,7 @@ config SND_KIRKWOOD_SOC_I2S config SND_KIRKWOOD_SOC_OPENRD tristate "SoC Audio support for Kirkwood Openrd Client" depends on SND_KIRKWOOD_SOC && (MACH_OPENRD_CLIENT || MACH_OPENRD_ULTIMATE) + depends on I2C select SND_KIRKWOOD_SOC_I2S select SND_SOC_CS42L51 help @@ -20,7 +21,7 @@ config SND_KIRKWOOD_SOC_OPENRD config SND_KIRKWOOD_SOC_T5325 tristate "SoC Audio support for HP t5325" - depends on SND_KIRKWOOD_SOC && MACH_T5325 + depends on SND_KIRKWOOD_SOC && MACH_T5325 && I2C select SND_KIRKWOOD_SOC_I2S select SND_SOC_ALC5623 help -- cgit v1.2.3 From ef1497707c69f3805bbca97c5d10c2913b6d2fa1 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sun, 4 Dec 2011 19:35:20 +0800 Subject: ASoC: uda1380: Return proper error in uda1380_modinit failure path Return proper error for uda1380_modinit if i2c_add_driver() fails. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/uda1380.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index c5ca8cfea60f..0441893e270e 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -863,13 +863,13 @@ static struct i2c_driver uda1380_i2c_driver = { static int __init uda1380_modinit(void) { - int ret; + int ret = 0; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) ret = i2c_add_driver(&uda1380_i2c_driver); if (ret != 0) pr_err("Failed to register UDA1380 I2C driver: %d\n", ret); #endif - return 0; + return ret; } module_init(uda1380_modinit); -- cgit v1.2.3 From b971c370a4d265848d9df0793b2f7fcc5b16b378 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sun, 4 Dec 2011 16:30:18 +0800 Subject: ASoC: Fix dependency for SND_SOC_RAUMFELD and SND_PXA2XX_SOC_HX4700 SND_SOC_RAUMFELD selects SND_SOC_CS4270 which needs CONFIG_I2C, and also selects SND_SOC_AK4104 which needs SPI_MASTER. Thus make SND_SOC_RAUMFELD depend on I2C && SPI_MASTER. Add depend on SPI_MASTER to fix below build error if CONFIG_SPI_MASTER is not selected. LD .tmp_vmlinux1 sound/built-in.o: In function `ak4104_spi_write': last.c:(.text+0x290cc): undefined reference to `spi_sync' sound/built-in.o: In function `ak4104_probe': last.c:(.text+0x292a0): undefined reference to `spi_write_then_read' sound/built-in.o: In function `ak4104_spi_probe': last.c:(.text+0x29398): undefined reference to `spi_setup' sound/built-in.o: In function `ak4104_init': last.c:(.init.text+0x4ec): undefined reference to `spi_register_driver' make: *** [.tmp_vmlinux1] Error 1 Add depend on I2C to fix below build error if CONFIG_I2C is not selected: CC sound/soc/codecs/cs4270.o sound/soc/codecs/cs4270.c: In function 'cs4270_i2c_probe': sound/soc/codecs/cs4270.c:657: error: implicit declaration of function 'i2c_smbus_read_byte_data' sound/soc/codecs/cs4270.c: In function 'cs4270_init': sound/soc/codecs/cs4270.c:730: error: implicit declaration of function 'i2c_add_driver' sound/soc/codecs/cs4270.c: In function 'cs4270_exit': sound/soc/codecs/cs4270.c:736: error: implicit declaration of function 'i2c_del_driver' make[3]: *** [sound/soc/codecs/cs4270.o] Error 1 make[2]: *** [sound/soc/codecs] Error 2 make[1]: *** [sound/soc] Error 2 make: *** [sound] Error 2 SND_PXA2XX_SOC_HX4700 selects SND_SOC_AK4641 which needs CONFIG_I2C. Thus make SND_PXA2XX_SOC_HX4700 depend on I2C. Add depend on I2C to fix below build error if CONFIG_I2C is not selected: CC sound/soc/codecs/ak4641.o sound/soc/codecs/ak4641.c: In function 'ak4641_modinit': sound/soc/codecs/ak4641.c:646: error: implicit declaration of function 'i2c_add_driver' sound/soc/codecs/ak4641.c: In function 'ak4641_exit': sound/soc/codecs/ak4641.c:656: error: implicit declaration of function 'i2c_del_driver' make[3]: *** [sound/soc/codecs/ak4641.o] Error 1 make[2]: *** [sound/soc/codecs] Error 2 make[1]: *** [sound/soc] Error 2 make: *** [sound] Error 2 Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/pxa/Kconfig | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index ffd2242e305f..a0f7d3cfa470 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -151,6 +151,7 @@ config SND_SOC_ZYLONITE config SND_SOC_RAUMFELD tristate "SoC Audio support Raumfeld audio adapter" depends on SND_PXA2XX_SOC && (MACH_RAUMFELD_SPEAKER || MACH_RAUMFELD_CONNECTOR) + depends on I2C && SPI_MASTER select SND_PXA_SOC_SSP select SND_SOC_CS4270 select SND_SOC_AK4104 @@ -159,7 +160,7 @@ config SND_SOC_RAUMFELD config SND_PXA2XX_SOC_HX4700 tristate "SoC Audio support for HP iPAQ hx4700" - depends on SND_PXA2XX_SOC && MACH_H4700 + depends on SND_PXA2XX_SOC && MACH_H4700 && I2C select SND_PXA2XX_SOC_I2S select SND_SOC_AK4641 help -- cgit v1.2.3 From 570a2429e964f8d63338b3450ba801a263b29b3a Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sun, 4 Dec 2011 16:11:16 +0800 Subject: ASoC: Make SND_SOC_MX27VIS_AIC32X4 depend on I2C SND_SOC_MX27VIS_AIC32X4 selects SND_SOC_TLV320AIC32X4, but SND_SOC_TLV320AIC32X4 needs CONFIG_I2C. So we need to make SND_SOC_MX27VIS_AIC32X4 depend on I2C. otherwise I got below build error if CONFIG_I2C is not selected. CC sound/soc/codecs/tlv320aic32x4.o sound/soc/codecs/tlv320aic32x4.c: In function 'aic32x4_read': sound/soc/codecs/tlv320aic32x4.c:323: error: implicit declaration of function 'i2c_smbus_read_byte_data' sound/soc/codecs/tlv320aic32x4.c: In function 'aic32x4_probe': sound/soc/codecs/tlv320aic32x4.c:641: error: 'i2c_master_send' undeclared (first use in this function) sound/soc/codecs/tlv320aic32x4.c:641: error: (Each undeclared identifier is reported only once sound/soc/codecs/tlv320aic32x4.c:641: error: for each function it appears in.) sound/soc/codecs/tlv320aic32x4.c: In function 'aic32x4_modinit': sound/soc/codecs/tlv320aic32x4.c:763: error: implicit declaration of function 'i2c_add_driver' sound/soc/codecs/tlv320aic32x4.c: In function 'aic32x4_exit': sound/soc/codecs/tlv320aic32x4.c:774: error: implicit declaration of function 'i2c_del_driver' make[3]: *** [sound/soc/codecs/tlv320aic32x4.o] Error 1 make[2]: *** [sound/soc/codecs] Error 2 make[1]: *** [sound/soc] Error 2 make: *** [sound] Error 2 Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/imx/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index b133bfcc5848..738391757f2c 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -28,7 +28,7 @@ config SND_MXC_SOC_WM1133_EV1 config SND_SOC_MX27VIS_AIC32X4 tristate "SoC audio support for Visstrim M10 boards" - depends on MACH_IMX27_VISSTRIM_M10 + depends on MACH_IMX27_VISSTRIM_M10 && I2C select SND_SOC_TLV320AIC32X4 select SND_MXC_SOC_MX2 help -- cgit v1.2.3 From cefcc03ffc9527dde56807339edb1719c8dbae5f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 5 Dec 2011 20:50:45 +0000 Subject: ASoC: Provide a more complete DMA driver stub Allow userspace applications to do more parameter setting by providing a more complete stub DMA driver specifying a wildcard set of formats and channels and essentially random values for the DMA parameters. This is required for useful runtime operation of the dummy DMA driver until we are able to figure out how to power up links and do hw_params() from DAPM. Sending to stable as without this the dummy driver is not terribly useful. Reported-by: Kyung-Kwee Ryu Tested-by: Kyung-Kwee Ryu Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/soc-utils.c | 31 ++++++++++++++++++++++++++++++- 1 file changed, 30 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 0c12b98484bd..4220bb0f2730 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -58,7 +58,36 @@ int snd_soc_params_to_bclk(struct snd_pcm_hw_params *params) } EXPORT_SYMBOL_GPL(snd_soc_params_to_bclk); -static struct snd_soc_platform_driver dummy_platform; +static const struct snd_pcm_hardware dummy_dma_hardware = { + .formats = 0xffffffff, + .channels_min = 1, + .channels_max = UINT_MAX, + + /* Random values to keep userspace happy when checking constraints */ + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER, + .buffer_bytes_max = 128*1024, + .period_bytes_min = PAGE_SIZE, + .period_bytes_max = PAGE_SIZE*2, + .periods_min = 2, + .periods_max = 128, +}; + +static int dummy_dma_open(struct snd_pcm_substream *substream) +{ + snd_soc_set_runtime_hwparams(substream, &dummy_dma_hardware); + + return 0; +} + +static struct snd_pcm_ops dummy_dma_ops = { + .open = dummy_dma_open, + .ioctl = snd_pcm_lib_ioctl, +}; + +static struct snd_soc_platform_driver dummy_platform = { + .ops = &dummy_dma_ops, +}; static __devinit int snd_soc_dummy_probe(struct platform_device *pdev) { -- cgit v1.2.3 From f1a73746c6664442082e3d53e1804f46e1910436 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 4 Dec 2011 13:44:06 +0100 Subject: ALSA: hda - Fix GPIO LED setup for IDT 92HD75 codecs Some HP laptops with IDT 92HD75 codecs may use a GPIO > 4 for the mute LED, but currently the driver doesn't check this properly, and confuses the mute LED behavior. This ended up with the silent output on some HP laptops due to having another GPIO used as external amp control. This patch fixes the problem by checking the max GPIO count and comparing with the given value from DMI entry instead of magic fixed value 4 and 8, and adding a new field to indicate the VREF mute-LED behavior. Reported-and-tested-by: Vitaliy Kulikov Cc: [v3.1] Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 45 ++++++++++++++++++++++-------------------- 1 file changed, 24 insertions(+), 21 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index f3658658548e..e035cf6de278 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -215,6 +215,7 @@ struct sigmatel_spec { unsigned int gpio_mute; unsigned int gpio_led; unsigned int gpio_led_polarity; + unsigned int vref_mute_led_nid; /* pin NID for mute-LED vref control */ unsigned int vref_led; /* stream */ @@ -4318,12 +4319,10 @@ static void stac_store_hints(struct hda_codec *codec) spec->eapd_switch = val; get_int_hint(codec, "gpio_led_polarity", &spec->gpio_led_polarity); if (get_int_hint(codec, "gpio_led", &spec->gpio_led)) { - if (spec->gpio_led <= 8) { - spec->gpio_mask |= spec->gpio_led; - spec->gpio_dir |= spec->gpio_led; - if (spec->gpio_led_polarity) - spec->gpio_data |= spec->gpio_led; - } + spec->gpio_mask |= spec->gpio_led; + spec->gpio_dir |= spec->gpio_led; + if (spec->gpio_led_polarity) + spec->gpio_data |= spec->gpio_led; } } @@ -4913,8 +4912,14 @@ static int find_mute_led_gpio(struct hda_codec *codec, int default_polarity) if (sscanf(dev->name, "HP_Mute_LED_%d_%x", &spec->gpio_led_polarity, &spec->gpio_led) == 2) { - if (spec->gpio_led < 4) + unsigned int max_gpio; + max_gpio = snd_hda_param_read(codec, codec->afg, + AC_PAR_GPIO_CAP); + max_gpio &= AC_GPIO_IO_COUNT; + if (spec->gpio_led < max_gpio) spec->gpio_led = 1 << spec->gpio_led; + else + spec->vref_mute_led_nid = spec->gpio_led; return 1; } if (sscanf(dev->name, "HP_Mute_LED_%d", @@ -5043,15 +5048,12 @@ static int stac92xx_pre_resume(struct hda_codec *codec) struct sigmatel_spec *spec = codec->spec; /* sync mute LED */ - if (spec->gpio_led) { - if (spec->gpio_led <= 8) { - stac_gpio_set(codec, spec->gpio_mask, - spec->gpio_dir, spec->gpio_data); - } else { - stac_vrefout_set(codec, - spec->gpio_led, spec->vref_led); - } - } + if (spec->vref_mute_led_nid) + stac_vrefout_set(codec, spec->vref_mute_led_nid, + spec->vref_led); + else if (spec->gpio_led) + stac_gpio_set(codec, spec->gpio_mask, + spec->gpio_dir, spec->gpio_data); return 0; } @@ -5076,7 +5078,7 @@ static void stac92xx_set_power_state(struct hda_codec *codec, hda_nid_t fg, struct sigmatel_spec *spec = codec->spec; if (power_state == AC_PWRST_D3) { - if (spec->gpio_led > 8) { + if (spec->vref_mute_led_nid) { /* with vref-out pin used for mute led control * codec AFG is prevented from D3 state */ @@ -5129,7 +5131,7 @@ static int stac92xx_update_led_status(struct hda_codec *codec) } } /*polarity defines *not* muted state level*/ - if (spec->gpio_led <= 8) { + if (!spec->vref_mute_led_nid) { if (muted) spec->gpio_data &= ~spec->gpio_led; /* orange */ else @@ -5147,7 +5149,8 @@ static int stac92xx_update_led_status(struct hda_codec *codec) muted_lvl = spec->gpio_led_polarity ? AC_PINCTL_VREF_GRD : AC_PINCTL_VREF_HIZ; spec->vref_led = muted ? muted_lvl : notmtd_lvl; - stac_vrefout_set(codec, spec->gpio_led, spec->vref_led); + stac_vrefout_set(codec, spec->vref_mute_led_nid, + spec->vref_led); } return 0; } @@ -5661,7 +5664,7 @@ again: #ifdef CONFIG_SND_HDA_POWER_SAVE if (spec->gpio_led) { - if (spec->gpio_led <= 8) { + if (!spec->vref_mute_led_nid) { spec->gpio_mask |= spec->gpio_led; spec->gpio_dir |= spec->gpio_led; spec->gpio_data |= spec->gpio_led; @@ -5976,7 +5979,7 @@ again: #ifdef CONFIG_SND_HDA_POWER_SAVE if (spec->gpio_led) { - if (spec->gpio_led <= 8) { + if (!spec->vref_mute_led_nid) { spec->gpio_mask |= spec->gpio_led; spec->gpio_dir |= spec->gpio_led; spec->gpio_data |= spec->gpio_led; -- cgit v1.2.3 From a020428364485751b607105c8f5a608f9b1fd38b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 6 Dec 2011 13:17:23 +0100 Subject: ALSA: hda - Fix remaining VREF mute-LED NID check in post-3.1 changes 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 ea1f157ca38b..eeb25d529e30 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4442,7 +4442,7 @@ static int stac92xx_init(struct hda_codec *codec) /* power on when no jack detection is available */ /* or when the VREF is used for controlling LED */ if (!spec->hp_detect || - (spec->gpio_led > 8 && spec->gpio_led == nid)) { + spec->vref_mute_led_nid == nid) { stac_toggle_power_map(codec, nid, 1); continue; } -- cgit v1.2.3 From 766ddee68bf1e4cc2cdcf73b34d243cf6e6f8ab0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Dec 2011 16:55:19 +0100 Subject: ALSA: hda/realtek - Don't create extra controls with channel suffix The multiple headphone or speaker pins are usually provided to output the same stream unlike line-out jacks (which are supposed to be multi-channel surrounds). Thus giving a mixer name like "Headphone Surround" is rather confusing. Instead, when multiple headphone volumes are available, use index with the same "Headphone" name. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 24 +++++++++++++++--------- 1 file changed, 15 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a7d1bc4e0d09..96f5da9db5cf 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3173,7 +3173,8 @@ static int alc_auto_create_multi_out_ctls(struct hda_codec *codec, } static int alc_auto_create_extra_out(struct hda_codec *codec, hda_nid_t pin, - hda_nid_t dac, const char *pfx) + hda_nid_t dac, const char *pfx, + int cidx) { struct alc_spec *spec = codec->spec; hda_nid_t sw, vol; @@ -3189,15 +3190,15 @@ static int alc_auto_create_extra_out(struct hda_codec *codec, hda_nid_t pin, if (is_ctl_used(spec->sw_ctls, val)) return 0; /* already created */ mark_ctl_usage(spec->sw_ctls, val); - return add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, pfx, val); + return __add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, pfx, cidx, val); } sw = alc_look_for_out_mute_nid(codec, pin, dac); vol = alc_look_for_out_vol_nid(codec, pin, dac); - err = alc_auto_add_stereo_vol(codec, pfx, 0, vol); + err = alc_auto_add_stereo_vol(codec, pfx, cidx, vol); if (err < 0) return err; - err = alc_auto_add_stereo_sw(codec, pfx, 0, sw); + err = alc_auto_add_stereo_sw(codec, pfx, cidx, sw); if (err < 0) return err; return 0; @@ -3238,16 +3239,21 @@ static int alc_auto_create_extra_outs(struct hda_codec *codec, int num_pins, hda_nid_t dac = *dacs; if (!dac) dac = spec->multiout.dac_nids[0]; - return alc_auto_create_extra_out(codec, *pins, dac, pfx); + return alc_auto_create_extra_out(codec, *pins, dac, pfx, 0); } if (dacs[num_pins - 1]) { /* OK, we have a multi-output system with individual volumes */ for (i = 0; i < num_pins; i++) { - snprintf(name, sizeof(name), "%s %s", - pfx, channel_name[i]); - err = alc_auto_create_extra_out(codec, pins[i], dacs[i], - name); + if (num_pins >= 3) { + snprintf(name, sizeof(name), "%s %s", + pfx, channel_name[i]); + err = alc_auto_create_extra_out(codec, pins[i], dacs[i], + name, 0); + } else { + err = alc_auto_create_extra_out(codec, pins[i], dacs[i], + pfx, i); + } if (err < 0) return err; } -- cgit v1.2.3 From fbabc24619f7298626265a3f973786b471b8c29d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Dec 2011 17:14:20 +0100 Subject: ALSA: hda/realtek - Create "Bass Speaker" for two speaker pins On systems with two speaker pins, the secondary speaker pin is mostly assigned to a bass speaker instead of a surround. Thus it makes more sense to rename the control properly. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 96f5da9db5cf..214d4f91d6af 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2631,6 +2631,8 @@ static const char *alc_get_line_out_pfx(struct alc_spec *spec, int ch, case AUTO_PIN_SPEAKER_OUT: if (cfg->line_outs == 1) return "Speaker"; + if (cfg->line_outs == 2) + return ch ? "Bass Speaker" : "Speaker"; break; case AUTO_PIN_HP_OUT: /* for multi-io case, only the primary out */ -- cgit v1.2.3 From 0a34b42b6245ccc710be6759cded5b7fe41ea1f9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Dec 2011 17:20:30 +0100 Subject: ALSA: hda/realtek - Fix lost speaker volume controls When there are the same or more number of HP pins are available, HP pins are used as the primary outputs instead of the speaker pins. But, in some cases (especially with ALC663 & co), some DACs are available only with a later pin and it's assigned to a speaker, and since the driver parses the pins from the lower NID, such a DAC was skipped eventually without assignments. This resulted in a regression, the missing speaker volume control in the new parser. As a workaround for this, now the driver retries the pin->DAC mapping again after restoring the speaker-pins as primary. This is still an ad hoc fix, but it works so far for most of Realtek codecs. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 37 ++++++++++++++++++++++++++++++------- 1 file changed, 30 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 214d4f91d6af..1d07e8fa2433 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2906,7 +2906,7 @@ static hda_nid_t alc_auto_look_for_dac(struct hda_codec *codec, hda_nid_t pin) if (!nid) continue; if (found_in_nid_list(nid, spec->multiout.dac_nids, - spec->multiout.num_dacs)) + ARRAY_SIZE(spec->private_dac_nids))) continue; if (found_in_nid_list(nid, spec->multiout.hp_out_nid, ARRAY_SIZE(spec->multiout.hp_out_nid))) @@ -2927,6 +2927,7 @@ static hda_nid_t get_dac_if_single(struct hda_codec *codec, hda_nid_t pin) return 0; } +/* return 0 if no possible DAC is found, 1 if one or more found */ static int alc_auto_fill_extra_dacs(struct hda_codec *codec, int num_outs, const hda_nid_t *pins, hda_nid_t *dacs) { @@ -2944,7 +2945,7 @@ static int alc_auto_fill_extra_dacs(struct hda_codec *codec, int num_outs, if (!dacs[i]) dacs[i] = alc_auto_look_for_dac(codec, pins[i]); } - return 0; + return 1; } static int alc_auto_fill_multi_ios(struct hda_codec *codec, @@ -2954,7 +2955,7 @@ static int alc_auto_fill_multi_ios(struct hda_codec *codec, static int alc_auto_fill_dac_nids(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - const struct auto_pin_cfg *cfg = &spec->autocfg; + struct auto_pin_cfg *cfg = &spec->autocfg; bool redone = false; int i; @@ -2965,6 +2966,7 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec) spec->multiout.extra_out_nid[0] = 0; memset(spec->private_dac_nids, 0, sizeof(spec->private_dac_nids)); spec->multiout.dac_nids = spec->private_dac_nids; + spec->multi_ios = 0; /* fill hard-wired DACs first */ if (!redone) { @@ -2998,10 +3000,12 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec) for (i = 0; i < cfg->line_outs; i++) { if (spec->private_dac_nids[i]) spec->multiout.num_dacs++; - else + else { memmove(spec->private_dac_nids + i, spec->private_dac_nids + i + 1, sizeof(hda_nid_t) * (cfg->line_outs - i - 1)); + spec->private_dac_nids[cfg->line_outs - 1] = 0; + } } if (cfg->line_outs == 1 && cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) { @@ -3023,9 +3027,28 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec) if (cfg->line_out_type != AUTO_PIN_HP_OUT) alc_auto_fill_extra_dacs(codec, cfg->hp_outs, cfg->hp_pins, spec->multiout.hp_out_nid); - if (cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) - alc_auto_fill_extra_dacs(codec, cfg->speaker_outs, cfg->speaker_pins, - spec->multiout.extra_out_nid); + if (cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) { + int err = alc_auto_fill_extra_dacs(codec, cfg->speaker_outs, + cfg->speaker_pins, + spec->multiout.extra_out_nid); + /* if no speaker volume is assigned, try again as the primary + * output + */ + if (!err && cfg->speaker_outs > 0 && + cfg->line_out_type == AUTO_PIN_HP_OUT) { + cfg->hp_outs = cfg->line_outs; + memcpy(cfg->hp_pins, cfg->line_out_pins, + sizeof(cfg->hp_pins)); + cfg->line_outs = cfg->speaker_outs; + memcpy(cfg->line_out_pins, cfg->speaker_pins, + sizeof(cfg->speaker_pins)); + cfg->speaker_outs = 0; + memset(cfg->speaker_pins, 0, sizeof(cfg->speaker_pins)); + cfg->line_out_type = AUTO_PIN_SPEAKER_OUT; + redone = false; + goto again; + } + } return 0; } -- cgit v1.2.3