diff options
author | Dave Airlie <airlied@redhat.com> | 2015-10-16 10:10:32 +1000 |
---|---|---|
committer | Dave Airlie <airlied@redhat.com> | 2015-10-16 10:25:28 +1000 |
commit | 48f87dd146a480c723774962eca675873a8aa1da (patch) | |
tree | 71461989ebe8a20258ca4b0be341b755594a2b0b /sound | |
parent | 6b62b3e134676687d5d666e6edc3b45f1507b2b7 (diff) | |
parent | 06d1ee32a4d25356a710b49d5e95dbdd68bdf505 (diff) | |
download | op-kernel-dev-48f87dd146a480c723774962eca675873a8aa1da.zip op-kernel-dev-48f87dd146a480c723774962eca675873a8aa1da.tar.gz |
Merge commit '06d1ee32a4d25356a710b49d5e95dbdd68bdf505' of git://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux into drm-next
Backmerge the drm-fixes pull from Linus's tree into drm-next.
This is to fix some conflicts and make future pulls cleaner
Diffstat (limited to 'sound')
-rw-r--r-- | sound/pci/hda/patch_cirrus.c | 1 | ||||
-rw-r--r-- | sound/pci/hda/patch_realtek.c | 1 | ||||
-rw-r--r-- | sound/pci/hda/patch_sigmatel.c | 6 | ||||
-rw-r--r-- | sound/soc/au1x/db1200.c | 4 | ||||
-rw-r--r-- | sound/soc/codecs/rt5645.c | 6 | ||||
-rw-r--r-- | sound/soc/codecs/rt5645.h | 16 | ||||
-rw-r--r-- | sound/soc/codecs/sgtl5000.c | 6 | ||||
-rw-r--r-- | sound/soc/codecs/tas2552.c | 2 | ||||
-rw-r--r-- | sound/soc/codecs/tlv320aic3x.c | 19 | ||||
-rw-r--r-- | sound/soc/codecs/wm8962.c | 5 | ||||
-rw-r--r-- | sound/soc/dwc/designware_i2s.c | 19 | ||||
-rw-r--r-- | sound/soc/fsl/imx-ssi.c | 19 | ||||
-rw-r--r-- | sound/synth/emux/emux_oss.c | 3 |
13 files changed, 67 insertions, 40 deletions
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 584a034..85813de 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 afec6dc..16b8dcb 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 9d947ae..def5cc8 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 58c3164..8c907eb 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 268a28b..5c101af 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 0e4cfc6..8c964cfb 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 bfda25e..f540f82 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 e3a0bca..cc1d398 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 1a82b19..8739126 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 293e47a..2fbc6ef 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 a3e97b4..ba34252 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 48b2d24..b95132e 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 82e350e..ac75816 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; |