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Message-ID: <s5hd1wola54.wl-tiwai@suse.de>
Date: Fri, 09 Oct 2015 10:27:19 +0200
From: Takashi Iwai <tiwai@...e.de>
To: Linus Torvalds <torvalds@...uxfoundation.org>
Cc: Mark Brown <broonie@...nel.org>,
Liam Girdwood <liam.r.girdwood@...ux.intel.com>,
linux-kernel@...r.kernel.org
Subject: [GIT PULL] sound fixes for 4.3-rc5
Linus,
please pull sound fixes for v4.3-rc5 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git tags/sound-4.3-rc5
The topmost commit is 601d62959d08a450d4666c728ddd2f47c5ba1cfe
----------------------------------------------------------------
sound fixes for 4.3-rc5
We see various small fixes, but nothing looks too scary, all are
small gentle bug fixes:
- Most of changes are for ASoC codecs: Realtek, SGTL5000, TAS2552,
TLV320, WM8962
- A couple of dwc and imx-ssi fixes
- Usual oneliner HD-audio quirks
- An old emux synth code fix
----------------------------------------------------------------
Andreas Dannenberg (1):
ASoC: tas2552: fix dBscale-min declaration
Benoît Thébaudeau (1):
ASoC: imx-ssi: Fix DAI hardware signal inversions
Gianluca Renzi (2):
ASoC: sgtl5000: fix error message output for MicBias voltage
ASoC: sgtl5000: fix wrong register MIC_BIAS_VOLTAGE setup on probe
Jiada Wang (1):
ASoC: wm8962: balance pm_runtime_enable
John Flatness (1):
ALSA: hda - Apply SPDIF pin ctl to MacBookPro 12,1
Lars-Peter Clausen (1):
ASoC: db1200: Fix DAI link format for db1300 and db1550
Laura Abbott (1):
ALSA: hda: Add dock support for ThinkPad T550
Mark Brown (1):
MAINTAINERS: Remove wm97xx entry
Oder Chiou (1):
ASoC: rt5645: Correct the naming and setting of ADC Boost Volume Control
Rick Mann (1):
ASoC: tlv320aic3x: Prevent writing reserved registers on tlv320aic3104 CODECs
Takashi Iwai (2):
ALSA: hda - Disable power_save_node for IDT 92HD73xx chips
ALSA: synth: Fix conflicting OSS device registration on AWE32
Yitian Bu (1):
ASoC: dwc: correct irq clear method
yitian (1):
ASoC: dwc: fix dma stop transferring issue
---
MAINTAINERS | 9 ---------
sound/pci/hda/patch_cirrus.c | 1 +
sound/pci/hda/patch_realtek.c | 1 +
sound/pci/hda/patch_sigmatel.c | 6 +++++-
sound/soc/au1x/db1200.c | 4 ++++
sound/soc/codecs/rt5645.c | 6 +++---
sound/soc/codecs/rt5645.h | 16 +++++++++-------
sound/soc/codecs/sgtl5000.c | 6 +++---
sound/soc/codecs/tas2552.c | 2 +-
sound/soc/codecs/tlv320aic3x.c | 19 +++++++++++--------
sound/soc/codecs/wm8962.c | 5 ++++-
sound/soc/dwc/designware_i2s.c | 19 ++++++++++++++-----
sound/soc/fsl/imx-ssi.c | 19 +++++++++----------
sound/synth/emux/emux_oss.c | 3 ++-
14 files changed, 67 insertions(+), 49 deletions(-)
diff --git a/MAINTAINERS b/MAINTAINERS
index 797236befd27..60aacd88bd7f 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -11378,15 +11378,6 @@ W: http://oops.ghostprotocols.net:81/blog
S: Maintained
F: drivers/net/wireless/wl3501*
-WM97XX TOUCHSCREEN DRIVERS
-M: Mark Brown <broonie@...nel.org>
-M: Liam Girdwood <lrg@...mlogic.co.uk>
-L: linux-input@...r.kernel.org
-W: https://github.com/CirrusLogic/linux-drivers/wiki
-S: Supported
-F: drivers/input/touchscreen/*wm97*
-F: include/linux/wm97xx.h
-
WOLFSON MICROELECTRONICS DRIVERS
L: patches@...nsource.wolfsonmicro.com
T: git https://github.com/CirrusLogic/linux-drivers.git
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 584a0343ab0c..85813de26da8 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -633,6 +633,7 @@ static const struct snd_pci_quirk cs4208_mac_fixup_tbl[] = {
SND_PCI_QUIRK(0x106b, 0x5e00, "MacBookPro 11,2", CS4208_MBP11),
SND_PCI_QUIRK(0x106b, 0x7100, "MacBookAir 6,1", CS4208_MBA6),
SND_PCI_QUIRK(0x106b, 0x7200, "MacBookAir 6,2", CS4208_MBA6),
+ SND_PCI_QUIRK(0x106b, 0x7b00, "MacBookPro 12,1", CS4208_MBP11),
{} /* terminator */
};
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index afec6dc9f91f..16b8dcba5c12 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -5306,6 +5306,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad T440", ALC292_FIXUP_TPT440_DOCK),
SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad X240", ALC292_FIXUP_TPT440_DOCK),
SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+ SND_PCI_QUIRK(0x17aa, 0x2223, "ThinkPad T550", ALC292_FIXUP_TPT440_DOCK),
SND_PCI_QUIRK(0x17aa, 0x2226, "ThinkPad X250", ALC292_FIXUP_TPT440_DOCK),
SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC),
SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP),
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 9d947aef2c8b..def5cc8dff02 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4520,7 +4520,11 @@ static int patch_stac92hd73xx(struct hda_codec *codec)
return err;
spec = codec->spec;
- codec->power_save_node = 1;
+ /* enable power_save_node only for new 92HD89xx chips, as it causes
+ * click noises on old 92HD73xx chips.
+ */
+ if ((codec->core.vendor_id & 0xfffffff0) != 0x111d7670)
+ codec->power_save_node = 1;
spec->linear_tone_beep = 0;
spec->gen.mixer_nid = 0x1d;
spec->have_spdif_mux = 1;
diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c
index 58c3164802b8..8c907ebea189 100644
--- a/sound/soc/au1x/db1200.c
+++ b/sound/soc/au1x/db1200.c
@@ -129,6 +129,8 @@ static struct snd_soc_dai_link db1300_i2s_dai = {
.cpu_dai_name = "au1xpsc_i2s.2",
.platform_name = "au1xpsc-pcm.2",
.codec_name = "wm8731.0-001b",
+ .dai_fmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM,
.ops = &db1200_i2s_wm8731_ops,
};
@@ -146,6 +148,8 @@ static struct snd_soc_dai_link db1550_i2s_dai = {
.cpu_dai_name = "au1xpsc_i2s.3",
.platform_name = "au1xpsc-pcm.3",
.codec_name = "wm8731.0-001b",
+ .dai_fmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM,
.ops = &db1200_i2s_wm8731_ops,
};
diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index 268a28bd1df4..5c101af0ac63 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -519,11 +519,11 @@ static const struct snd_kcontrol_new rt5645_snd_controls[] = {
RT5645_L_VOL_SFT + 1, RT5645_R_VOL_SFT + 1, 63, 0, adc_vol_tlv),
/* ADC Boost Volume Control */
- SOC_DOUBLE_TLV("STO1 ADC Boost Gain", RT5645_ADC_BST_VOL1,
+ SOC_DOUBLE_TLV("ADC Boost Capture Volume", RT5645_ADC_BST_VOL1,
RT5645_STO1_ADC_L_BST_SFT, RT5645_STO1_ADC_R_BST_SFT, 3, 0,
adc_bst_tlv),
- SOC_DOUBLE_TLV("STO2 ADC Boost Gain", RT5645_ADC_BST_VOL1,
- RT5645_STO2_ADC_L_BST_SFT, RT5645_STO2_ADC_R_BST_SFT, 3, 0,
+ SOC_DOUBLE_TLV("Mono ADC Boost Capture Volume", RT5645_ADC_BST_VOL2,
+ RT5645_MONO_ADC_L_BST_SFT, RT5645_MONO_ADC_R_BST_SFT, 3, 0,
adc_bst_tlv),
/* I2S2 function select */
diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h
index 0e4cfc6ac649..8c964cfb120d 100644
--- a/sound/soc/codecs/rt5645.h
+++ b/sound/soc/codecs/rt5645.h
@@ -39,8 +39,8 @@
#define RT5645_STO1_ADC_DIG_VOL 0x1c
#define RT5645_MONO_ADC_DIG_VOL 0x1d
#define RT5645_ADC_BST_VOL1 0x1e
-/* Mixer - D-D */
#define RT5645_ADC_BST_VOL2 0x20
+/* Mixer - D-D */
#define RT5645_STO1_ADC_MIXER 0x27
#define RT5645_MONO_ADC_MIXER 0x28
#define RT5645_AD_DA_MIXER 0x29
@@ -315,12 +315,14 @@
#define RT5645_STO1_ADC_R_BST_SFT 12
#define RT5645_STO1_ADC_COMP_MASK (0x3 << 10)
#define RT5645_STO1_ADC_COMP_SFT 10
-#define RT5645_STO2_ADC_L_BST_MASK (0x3 << 8)
-#define RT5645_STO2_ADC_L_BST_SFT 8
-#define RT5645_STO2_ADC_R_BST_MASK (0x3 << 6)
-#define RT5645_STO2_ADC_R_BST_SFT 6
-#define RT5645_STO2_ADC_COMP_MASK (0x3 << 4)
-#define RT5645_STO2_ADC_COMP_SFT 4
+
+/* ADC Boost Volume Control (0x20) */
+#define RT5645_MONO_ADC_L_BST_MASK (0x3 << 14)
+#define RT5645_MONO_ADC_L_BST_SFT 14
+#define RT5645_MONO_ADC_R_BST_MASK (0x3 << 12)
+#define RT5645_MONO_ADC_R_BST_SFT 12
+#define RT5645_MONO_ADC_COMP_MASK (0x3 << 10)
+#define RT5645_MONO_ADC_COMP_SFT 10
/* Stereo2 ADC Mixer Control (0x26) */
#define RT5645_STO2_ADC_SRC_MASK (0x1 << 15)
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index bfda25ef0dd4..f540f82b1f27 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -1376,8 +1376,8 @@ static int sgtl5000_probe(struct snd_soc_codec *codec)
sgtl5000->micbias_resistor << SGTL5000_BIAS_R_SHIFT);
snd_soc_update_bits(codec, SGTL5000_CHIP_MIC_CTRL,
- SGTL5000_BIAS_R_MASK,
- sgtl5000->micbias_voltage << SGTL5000_BIAS_R_SHIFT);
+ SGTL5000_BIAS_VOLT_MASK,
+ sgtl5000->micbias_voltage << SGTL5000_BIAS_VOLT_SHIFT);
/*
* disable DAP
* TODO:
@@ -1549,7 +1549,7 @@ static int sgtl5000_i2c_probe(struct i2c_client *client,
else {
sgtl5000->micbias_voltage = 0;
dev_err(&client->dev,
- "Unsuitable MicBias resistor\n");
+ "Unsuitable MicBias voltage\n");
}
} else {
sgtl5000->micbias_voltage = 0;
diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c
index e3a0bca28bcf..cc1d3981fa4b 100644
--- a/sound/soc/codecs/tas2552.c
+++ b/sound/soc/codecs/tas2552.c
@@ -549,7 +549,7 @@ static struct snd_soc_dai_driver tas2552_dai[] = {
/*
* DAC digital volumes. From -7 to 24 dB in 1 dB steps
*/
-static DECLARE_TLV_DB_SCALE(dac_tlv, -7, 100, 0);
+static DECLARE_TLV_DB_SCALE(dac_tlv, -700, 100, 0);
static const char * const tas2552_din_source_select[] = {
"Muted",
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 1a82b19b2644..8739126a1f6f 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -1509,14 +1509,17 @@ static int aic3x_init(struct snd_soc_codec *codec)
snd_soc_write(codec, PGAL_2_LLOPM_VOL, DEFAULT_VOL);
snd_soc_write(codec, PGAR_2_RLOPM_VOL, DEFAULT_VOL);
- /* Line2 to HP Bypass default volume, disconnect from Output Mixer */
- snd_soc_write(codec, LINE2L_2_HPLOUT_VOL, DEFAULT_VOL);
- snd_soc_write(codec, LINE2R_2_HPROUT_VOL, DEFAULT_VOL);
- snd_soc_write(codec, LINE2L_2_HPLCOM_VOL, DEFAULT_VOL);
- snd_soc_write(codec, LINE2R_2_HPRCOM_VOL, DEFAULT_VOL);
- /* Line2 Line Out default volume, disconnect from Output Mixer */
- snd_soc_write(codec, LINE2L_2_LLOPM_VOL, DEFAULT_VOL);
- snd_soc_write(codec, LINE2R_2_RLOPM_VOL, DEFAULT_VOL);
+ /* On tlv320aic3104, these registers are reserved and must not be written */
+ if (aic3x->model != AIC3X_MODEL_3104) {
+ /* Line2 to HP Bypass default volume, disconnect from Output Mixer */
+ snd_soc_write(codec, LINE2L_2_HPLOUT_VOL, DEFAULT_VOL);
+ snd_soc_write(codec, LINE2R_2_HPROUT_VOL, DEFAULT_VOL);
+ snd_soc_write(codec, LINE2L_2_HPLCOM_VOL, DEFAULT_VOL);
+ snd_soc_write(codec, LINE2R_2_HPRCOM_VOL, DEFAULT_VOL);
+ /* Line2 Line Out default volume, disconnect from Output Mixer */
+ snd_soc_write(codec, LINE2L_2_LLOPM_VOL, DEFAULT_VOL);
+ snd_soc_write(codec, LINE2R_2_RLOPM_VOL, DEFAULT_VOL);
+ }
switch (aic3x->model) {
case AIC3X_MODEL_3X:
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 293e47a6ff59..2fbc6ef8cbdb 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -3760,7 +3760,7 @@ static int wm8962_i2c_probe(struct i2c_client *i2c,
ret = snd_soc_register_codec(&i2c->dev,
&soc_codec_dev_wm8962, &wm8962_dai, 1);
if (ret < 0)
- goto err_enable;
+ goto err_pm_runtime;
regcache_cache_only(wm8962->regmap, true);
@@ -3769,6 +3769,8 @@ static int wm8962_i2c_probe(struct i2c_client *i2c,
return 0;
+err_pm_runtime:
+ pm_runtime_disable(&i2c->dev);
err_enable:
regulator_bulk_disable(ARRAY_SIZE(wm8962->supplies), wm8962->supplies);
err:
@@ -3778,6 +3780,7 @@ err:
static int wm8962_i2c_remove(struct i2c_client *client)
{
snd_soc_unregister_codec(&client->dev);
+ pm_runtime_disable(&client->dev);
return 0;
}
diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c
index a3e97b46b64e..ba34252b7bba 100644
--- a/sound/soc/dwc/designware_i2s.c
+++ b/sound/soc/dwc/designware_i2s.c
@@ -131,23 +131,32 @@ static inline void i2s_clear_irqs(struct dw_i2s_dev *dev, u32 stream)
if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
for (i = 0; i < 4; i++)
- i2s_write_reg(dev->i2s_base, TOR(i), 0);
+ i2s_read_reg(dev->i2s_base, TOR(i));
} else {
for (i = 0; i < 4; i++)
- i2s_write_reg(dev->i2s_base, ROR(i), 0);
+ i2s_read_reg(dev->i2s_base, ROR(i));
}
}
static void i2s_start(struct dw_i2s_dev *dev,
struct snd_pcm_substream *substream)
{
-
+ u32 i, irq;
i2s_write_reg(dev->i2s_base, IER, 1);
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ for (i = 0; i < 4; i++) {
+ irq = i2s_read_reg(dev->i2s_base, IMR(i));
+ i2s_write_reg(dev->i2s_base, IMR(i), irq & ~0x30);
+ }
i2s_write_reg(dev->i2s_base, ITER, 1);
- else
+ } else {
+ for (i = 0; i < 4; i++) {
+ irq = i2s_read_reg(dev->i2s_base, IMR(i));
+ i2s_write_reg(dev->i2s_base, IMR(i), irq & ~0x03);
+ }
i2s_write_reg(dev->i2s_base, IRER, 1);
+ }
i2s_write_reg(dev->i2s_base, CER, 1);
}
diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c
index 48b2d24dd1f0..b95132e2f9dc 100644
--- a/sound/soc/fsl/imx-ssi.c
+++ b/sound/soc/fsl/imx-ssi.c
@@ -95,7 +95,8 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
/* data on rising edge of bclk, frame low 1clk before data */
- strcr |= SSI_STCR_TFSI | SSI_STCR_TEFS | SSI_STCR_TXBIT0;
+ strcr |= SSI_STCR_TXBIT0 | SSI_STCR_TSCKP | SSI_STCR_TFSI |
+ SSI_STCR_TEFS;
scr |= SSI_SCR_NET;
if (ssi->flags & IMX_SSI_USE_I2S_SLAVE) {
scr &= ~SSI_I2S_MODE_MASK;
@@ -104,33 +105,31 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
break;
case SND_SOC_DAIFMT_LEFT_J:
/* data on rising edge of bclk, frame high with data */
- strcr |= SSI_STCR_TXBIT0;
+ strcr |= SSI_STCR_TXBIT0 | SSI_STCR_TSCKP;
break;
case SND_SOC_DAIFMT_DSP_B:
/* data on rising edge of bclk, frame high with data */
- strcr |= SSI_STCR_TFSL | SSI_STCR_TXBIT0;
+ strcr |= SSI_STCR_TXBIT0 | SSI_STCR_TSCKP | SSI_STCR_TFSL;
break;
case SND_SOC_DAIFMT_DSP_A:
/* data on rising edge of bclk, frame high 1clk before data */
- strcr |= SSI_STCR_TFSL | SSI_STCR_TXBIT0 | SSI_STCR_TEFS;
+ strcr |= SSI_STCR_TXBIT0 | SSI_STCR_TSCKP | SSI_STCR_TFSL |
+ SSI_STCR_TEFS;
break;
}
/* DAI clock inversion */
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
case SND_SOC_DAIFMT_IB_IF:
- strcr |= SSI_STCR_TFSI;
- strcr &= ~SSI_STCR_TSCKP;
+ strcr ^= SSI_STCR_TSCKP | SSI_STCR_TFSI;
break;
case SND_SOC_DAIFMT_IB_NF:
- strcr &= ~(SSI_STCR_TSCKP | SSI_STCR_TFSI);
+ strcr ^= SSI_STCR_TSCKP;
break;
case SND_SOC_DAIFMT_NB_IF:
- strcr |= SSI_STCR_TFSI | SSI_STCR_TSCKP;
+ strcr ^= SSI_STCR_TFSI;
break;
case SND_SOC_DAIFMT_NB_NF:
- strcr &= ~SSI_STCR_TFSI;
- strcr |= SSI_STCR_TSCKP;
break;
}
diff --git a/sound/synth/emux/emux_oss.c b/sound/synth/emux/emux_oss.c
index 82e350e9501c..ac75816ada7c 100644
--- a/sound/synth/emux/emux_oss.c
+++ b/sound/synth/emux/emux_oss.c
@@ -69,7 +69,8 @@ snd_emux_init_seq_oss(struct snd_emux *emu)
struct snd_seq_oss_reg *arg;
struct snd_seq_device *dev;
- if (snd_seq_device_new(emu->card, 0, SNDRV_SEQ_DEV_ID_OSS,
+ /* using device#1 here for avoiding conflicts with OPL3 */
+ if (snd_seq_device_new(emu->card, 1, SNDRV_SEQ_DEV_ID_OSS,
sizeof(struct snd_seq_oss_reg), &dev) < 0)
return;
--
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