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-rw-r--r--sound/soc/au1x/psc-ac97.c140
-rw-r--r--sound/soc/au1x/psc-i2s.c100
-rw-r--r--sound/soc/au1x/psc.h22
-rw-r--r--sound/soc/codecs/arizona.c16
-rw-r--r--sound/soc/codecs/cs4265.c18
-rw-r--r--sound/soc/codecs/da732x.h2
-rw-r--r--sound/soc/codecs/pcm512x.c4
-rw-r--r--sound/soc/codecs/rt286.c7
-rw-r--r--sound/soc/codecs/rt5640.c1
-rw-r--r--sound/soc/codecs/rt5677.c8
-rw-r--r--sound/soc/codecs/ssm2602.c2
-rw-r--r--sound/soc/codecs/sta529.c4
-rw-r--r--sound/soc/codecs/tlv320aic31xx.c51
-rw-r--r--sound/soc/davinci/davinci-mcasp.c25
-rw-r--r--sound/soc/dwc/designware_i2s.c4
-rw-r--r--sound/soc/fsl/Kconfig1
-rw-r--r--sound/soc/fsl/fsl_esai.c2
-rw-r--r--sound/soc/fsl/fsl_ssi.c12
-rw-r--r--sound/soc/generic/simple-card.c8
-rw-r--r--sound/soc/intel/sst-acpi.c4
-rw-r--r--sound/soc/intel/sst-baytrail-ipc.c10
-rw-r--r--sound/soc/intel/sst-baytrail-ipc.h1
-rw-r--r--sound/soc/intel/sst-baytrail-pcm.c43
-rw-r--r--sound/soc/omap/omap-twl4030.c2
-rw-r--r--sound/soc/pxa/pxa-ssp.c4
-rw-r--r--sound/soc/rockchip/rockchip_i2s.c13
-rw-r--r--sound/soc/samsung/goni_wm8994.c2
-rw-r--r--sound/soc/samsung/i2s.c5
-rw-r--r--sound/soc/sh/rcar/gen.c2
-rw-r--r--sound/soc/soc-compress.c6
-rw-r--r--sound/soc/soc-core.c4
-rw-r--r--sound/soc/soc-dapm.c12
-rw-r--r--sound/soc/soc-pcm.c6
-rw-r--r--sound/soc/spear/spear_pcm.c4
-rw-r--r--sound/soc/tegra/tegra_asoc_utils.h2
35 files changed, 293 insertions, 254 deletions
diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c
index 986dcec79fa0..84f31e1f9d24 100644
--- a/sound/soc/au1x/psc-ac97.c
+++ b/sound/soc/au1x/psc-ac97.c
@@ -79,28 +79,28 @@ static unsigned short au1xpsc_ac97_read(struct snd_ac97 *ac97,
unsigned short retry, tmo;
unsigned long data;
- au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata));
- au_sync();
+ __raw_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata));
+ wmb(); /* drain writebuffer */
retry = AC97_RW_RETRIES;
do {
mutex_lock(&pscdata->lock);
- au_writel(PSC_AC97CDC_RD | PSC_AC97CDC_INDX(reg),
+ __raw_writel(PSC_AC97CDC_RD | PSC_AC97CDC_INDX(reg),
AC97_CDC(pscdata));
- au_sync();
+ wmb(); /* drain writebuffer */
tmo = 20;
do {
udelay(21);
- if (au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)
+ if (__raw_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)
break;
} while (--tmo);
- data = au_readl(AC97_CDC(pscdata));
+ data = __raw_readl(AC97_CDC(pscdata));
- au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata));
- au_sync();
+ __raw_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata));
+ wmb(); /* drain writebuffer */
mutex_unlock(&pscdata->lock);
@@ -119,26 +119,26 @@ static void au1xpsc_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
struct au1xpsc_audio_data *pscdata = ac97_to_pscdata(ac97);
unsigned int tmo, retry;
- au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata));
- au_sync();
+ __raw_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata));
+ wmb(); /* drain writebuffer */
retry = AC97_RW_RETRIES;
do {
mutex_lock(&pscdata->lock);
- au_writel(PSC_AC97CDC_INDX(reg) | (val & 0xffff),
+ __raw_writel(PSC_AC97CDC_INDX(reg) | (val & 0xffff),
AC97_CDC(pscdata));
- au_sync();
+ wmb(); /* drain writebuffer */
tmo = 20;
do {
udelay(21);
- if (au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)
+ if (__raw_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)
break;
} while (--tmo);
- au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata));
- au_sync();
+ __raw_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata));
+ wmb(); /* drain writebuffer */
mutex_unlock(&pscdata->lock);
} while (--retry && !tmo);
@@ -149,11 +149,11 @@ static void au1xpsc_ac97_warm_reset(struct snd_ac97 *ac97)
{
struct au1xpsc_audio_data *pscdata = ac97_to_pscdata(ac97);
- au_writel(PSC_AC97RST_SNC, AC97_RST(pscdata));
- au_sync();
+ __raw_writel(PSC_AC97RST_SNC, AC97_RST(pscdata));
+ wmb(); /* drain writebuffer */
msleep(10);
- au_writel(0, AC97_RST(pscdata));
- au_sync();
+ __raw_writel(0, AC97_RST(pscdata));
+ wmb(); /* drain writebuffer */
}
static void au1xpsc_ac97_cold_reset(struct snd_ac97 *ac97)
@@ -162,25 +162,25 @@ static void au1xpsc_ac97_cold_reset(struct snd_ac97 *ac97)
int i;
/* disable PSC during cold reset */
- au_writel(0, AC97_CFG(au1xpsc_ac97_workdata));
- au_sync();
- au_writel(PSC_CTRL_DISABLE, PSC_CTRL(pscdata));
- au_sync();
+ __raw_writel(0, AC97_CFG(au1xpsc_ac97_workdata));
+ wmb(); /* drain writebuffer */
+ __raw_writel(PSC_CTRL_DISABLE, PSC_CTRL(pscdata));
+ wmb(); /* drain writebuffer */
/* issue cold reset */
- au_writel(PSC_AC97RST_RST, AC97_RST(pscdata));
- au_sync();
+ __raw_writel(PSC_AC97RST_RST, AC97_RST(pscdata));
+ wmb(); /* drain writebuffer */
msleep(500);
- au_writel(0, AC97_RST(pscdata));
- au_sync();
+ __raw_writel(0, AC97_RST(pscdata));
+ wmb(); /* drain writebuffer */
/* enable PSC */
- au_writel(PSC_CTRL_ENABLE, PSC_CTRL(pscdata));
- au_sync();
+ __raw_writel(PSC_CTRL_ENABLE, PSC_CTRL(pscdata));
+ wmb(); /* drain writebuffer */
/* wait for PSC to indicate it's ready */
i = 1000;
- while (!((au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_SR)) && (--i))
+ while (!((__raw_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_SR)) && (--i))
msleep(1);
if (i == 0) {
@@ -189,12 +189,12 @@ static void au1xpsc_ac97_cold_reset(struct snd_ac97 *ac97)
}
/* enable the ac97 function */
- au_writel(pscdata->cfg | PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata));
- au_sync();
+ __raw_writel(pscdata->cfg | PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata));
+ wmb(); /* drain writebuffer */
/* wait for AC97 core to become ready */
i = 1000;
- while (!((au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR)) && (--i))
+ while (!((__raw_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR)) && (--i))
msleep(1);
if (i == 0)
printk(KERN_ERR "au1xpsc-ac97: AC97 ctrl not ready\n");
@@ -218,8 +218,8 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream,
chans = params_channels(params);
- r = ro = au_readl(AC97_CFG(pscdata));
- stat = au_readl(AC97_STAT(pscdata));
+ r = ro = __raw_readl(AC97_CFG(pscdata));
+ stat = __raw_readl(AC97_STAT(pscdata));
/* already active? */
if (stat & (PSC_AC97STAT_TB | PSC_AC97STAT_RB)) {
@@ -252,28 +252,28 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream,
mutex_lock(&pscdata->lock);
/* disable AC97 device controller first... */
- au_writel(r & ~PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata));
- au_sync();
+ __raw_writel(r & ~PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata));
+ wmb(); /* drain writebuffer */
/* ...wait for it... */
t = 100;
- while ((au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR) && --t)
+ while ((__raw_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR) && --t)
msleep(1);
if (!t)
printk(KERN_ERR "PSC-AC97: can't disable!\n");
/* ...write config... */
- au_writel(r, AC97_CFG(pscdata));
- au_sync();
+ __raw_writel(r, AC97_CFG(pscdata));
+ wmb(); /* drain writebuffer */
/* ...enable the AC97 controller again... */
- au_writel(r | PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata));
- au_sync();
+ __raw_writel(r | PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata));
+ wmb(); /* drain writebuffer */
/* ...and wait for ready bit */
t = 100;
- while ((!(au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR)) && --t)
+ while ((!(__raw_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR)) && --t)
msleep(1);
if (!t)
@@ -300,21 +300,21 @@ static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream,
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
- au_writel(AC97PCR_CLRFIFO(stype), AC97_PCR(pscdata));
- au_sync();
- au_writel(AC97PCR_START(stype), AC97_PCR(pscdata));
- au_sync();
+ __raw_writel(AC97PCR_CLRFIFO(stype), AC97_PCR(pscdata));
+ wmb(); /* drain writebuffer */
+ __raw_writel(AC97PCR_START(stype), AC97_PCR(pscdata));
+ wmb(); /* drain writebuffer */
break;
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_SUSPEND:
- au_writel(AC97PCR_STOP(stype), AC97_PCR(pscdata));
- au_sync();
+ __raw_writel(AC97PCR_STOP(stype), AC97_PCR(pscdata));
+ wmb(); /* drain writebuffer */
- while (au_readl(AC97_STAT(pscdata)) & AC97STAT_BUSY(stype))
+ while (__raw_readl(AC97_STAT(pscdata)) & AC97STAT_BUSY(stype))
asm volatile ("nop");
- au_writel(AC97PCR_CLRFIFO(stype), AC97_PCR(pscdata));
- au_sync();
+ __raw_writel(AC97PCR_CLRFIFO(stype), AC97_PCR(pscdata));
+ wmb(); /* drain writebuffer */
break;
default:
@@ -398,13 +398,13 @@ static int au1xpsc_ac97_drvprobe(struct platform_device *pdev)
PSC_AC97CFG_DE_ENABLE;
/* preserve PSC clock source set up by platform */
- sel = au_readl(PSC_SEL(wd)) & PSC_SEL_CLK_MASK;
- au_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd));
- au_sync();
- au_writel(0, PSC_SEL(wd));
- au_sync();
- au_writel(PSC_SEL_PS_AC97MODE | sel, PSC_SEL(wd));
- au_sync();
+ sel = __raw_readl(PSC_SEL(wd)) & PSC_SEL_CLK_MASK;
+ __raw_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd));
+ wmb(); /* drain writebuffer */
+ __raw_writel(0, PSC_SEL(wd));
+ wmb(); /* drain writebuffer */
+ __raw_writel(PSC_SEL_PS_AC97MODE | sel, PSC_SEL(wd));
+ wmb(); /* drain writebuffer */
/* name the DAI like this device instance ("au1xpsc-ac97.PSCINDEX") */
memcpy(&wd->dai_drv, &au1xpsc_ac97_dai_template,
@@ -433,10 +433,10 @@ static int au1xpsc_ac97_drvremove(struct platform_device *pdev)
snd_soc_unregister_component(&pdev->dev);
/* disable PSC completely */
- au_writel(0, AC97_CFG(wd));
- au_sync();
- au_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd));
- au_sync();
+ __raw_writel(0, AC97_CFG(wd));
+ wmb(); /* drain writebuffer */
+ __raw_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd));
+ wmb(); /* drain writebuffer */
au1xpsc_ac97_workdata = NULL; /* MDEV */
@@ -449,12 +449,12 @@ static int au1xpsc_ac97_drvsuspend(struct device *dev)
struct au1xpsc_audio_data *wd = dev_get_drvdata(dev);
/* save interesting registers and disable PSC */
- wd->pm[0] = au_readl(PSC_SEL(wd));
+ wd->pm[0] = __raw_readl(PSC_SEL(wd));
- au_writel(0, AC97_CFG(wd));
- au_sync();
- au_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd));
- au_sync();
+ __raw_writel(0, AC97_CFG(wd));
+ wmb(); /* drain writebuffer */
+ __raw_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd));
+ wmb(); /* drain writebuffer */
return 0;
}
@@ -464,8 +464,8 @@ static int au1xpsc_ac97_drvresume(struct device *dev)
struct au1xpsc_audio_data *wd = dev_get_drvdata(dev);
/* restore PSC clock config */
- au_writel(wd->pm[0] | PSC_SEL_PS_AC97MODE, PSC_SEL(wd));
- au_sync();
+ __raw_writel(wd->pm[0] | PSC_SEL_PS_AC97MODE, PSC_SEL(wd));
+ wmb(); /* drain writebuffer */
/* after this point the ac97 core will cold-reset the codec.
* During cold-reset the PSC is reinitialized and the last
diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c
index fe923a7bdc39..814beffc56f2 100644
--- a/sound/soc/au1x/psc-i2s.c
+++ b/sound/soc/au1x/psc-i2s.c
@@ -120,10 +120,10 @@ static int au1xpsc_i2s_hw_params(struct snd_pcm_substream *substream,
unsigned long stat;
/* check if the PSC is already streaming data */
- stat = au_readl(I2S_STAT(pscdata));
+ stat = __raw_readl(I2S_STAT(pscdata));
if (stat & (PSC_I2SSTAT_TB | PSC_I2SSTAT_RB)) {
/* reject parameters not currently set up in hardware */
- cfgbits = au_readl(I2S_CFG(pscdata));
+ cfgbits = __raw_readl(I2S_CFG(pscdata));
if ((PSC_I2SCFG_GET_LEN(cfgbits) != params->msbits) ||
(params_rate(params) != pscdata->rate))
return -EINVAL;
@@ -149,33 +149,33 @@ static int au1xpsc_i2s_configure(struct au1xpsc_audio_data *pscdata)
unsigned long tmo;
/* bring PSC out of sleep, and configure I2S unit */
- au_writel(PSC_CTRL_ENABLE, PSC_CTRL(pscdata));
- au_sync();
+ __raw_writel(PSC_CTRL_ENABLE, PSC_CTRL(pscdata));
+ wmb(); /* drain writebuffer */
tmo = 1000000;
- while (!(au_readl(I2S_STAT(pscdata)) & PSC_I2SSTAT_SR) && tmo)
+ while (!(__raw_readl(I2S_STAT(pscdata)) & PSC_I2SSTAT_SR) && tmo)
tmo--;
if (!tmo)
goto psc_err;
- au_writel(0, I2S_CFG(pscdata));
- au_sync();
- au_writel(pscdata->cfg | PSC_I2SCFG_DE_ENABLE, I2S_CFG(pscdata));
- au_sync();
+ __raw_writel(0, I2S_CFG(pscdata));
+ wmb(); /* drain writebuffer */
+ __raw_writel(pscdata->cfg | PSC_I2SCFG_DE_ENABLE, I2S_CFG(pscdata));
+ wmb(); /* drain writebuffer */
/* wait for I2S controller to become ready */
tmo = 1000000;
- while (!(au_readl(I2S_STAT(pscdata)) & PSC_I2SSTAT_DR) && tmo)
+ while (!(__raw_readl(I2S_STAT(pscdata)) & PSC_I2SSTAT_DR) && tmo)
tmo--;
if (tmo)
return 0;
psc_err:
- au_writel(0, I2S_CFG(pscdata));
- au_writel(PSC_CTRL_SUSPEND, PSC_CTRL(pscdata));
- au_sync();
+ __raw_writel(0, I2S_CFG(pscdata));
+ __raw_writel(PSC_CTRL_SUSPEND, PSC_CTRL(pscdata));
+ wmb(); /* drain writebuffer */
return -ETIMEDOUT;
}
@@ -187,26 +187,26 @@ static int au1xpsc_i2s_start(struct au1xpsc_audio_data *pscdata, int stype)
ret = 0;
/* if both TX and RX are idle, configure the PSC */
- stat = au_readl(I2S_STAT(pscdata));
+ stat = __raw_readl(I2S_STAT(pscdata));
if (!(stat & (PSC_I2SSTAT_TB | PSC_I2SSTAT_RB))) {
ret = au1xpsc_i2s_configure(pscdata);
if (ret)
goto out;
}
- au_writel(I2SPCR_CLRFIFO(stype), I2S_PCR(pscdata));
- au_sync();
- au_writel(I2SPCR_START(stype), I2S_PCR(pscdata));
- au_sync();
+ __raw_writel(I2SPCR_CLRFIFO(stype), I2S_PCR(pscdata));
+ wmb(); /* drain writebuffer */
+ __raw_writel(I2SPCR_START(stype), I2S_PCR(pscdata));
+ wmb(); /* drain writebuffer */
/* wait for start confirmation */
tmo = 1000000;
- while (!(au_readl(I2S_STAT(pscdata)) & I2SSTAT_BUSY(stype)) && tmo)
+ while (!(__raw_readl(I2S_STAT(pscdata)) & I2SSTAT_BUSY(stype)) && tmo)
tmo--;
if (!tmo) {
- au_writel(I2SPCR_STOP(stype), I2S_PCR(pscdata));
- au_sync();
+ __raw_writel(I2SPCR_STOP(stype), I2S_PCR(pscdata));
+ wmb(); /* drain writebuffer */
ret = -ETIMEDOUT;
}
out:
@@ -217,21 +217,21 @@ static int au1xpsc_i2s_stop(struct au1xpsc_audio_data *pscdata, int stype)
{
unsigned long tmo, stat;
- au_writel(I2SPCR_STOP(stype), I2S_PCR(pscdata));
- au_sync();
+ __raw_writel(I2SPCR_STOP(stype), I2S_PCR(pscdata));
+ wmb(); /* drain writebuffer */
/* wait for stop confirmation */
tmo = 1000000;
- while ((au_readl(I2S_STAT(pscdata)) & I2SSTAT_BUSY(stype)) && tmo)
+ while ((__raw_readl(I2S_STAT(pscdata)) & I2SSTAT_BUSY(stype)) && tmo)
tmo--;
/* if both TX and RX are idle, disable PSC */
- stat = au_readl(I2S_STAT(pscdata));
+ stat = __raw_readl(I2S_STAT(pscdata));
if (!(stat & (PSC_I2SSTAT_TB | PSC_I2SSTAT_RB))) {
- au_writel(0, I2S_CFG(pscdata));
- au_sync();
- au_writel(PSC_CTRL_SUSPEND, PSC_CTRL(pscdata));
- au_sync();
+ __raw_writel(0, I2S_CFG(pscdata));
+ wmb(); /* drain writebuffer */
+ __raw_writel(PSC_CTRL_SUSPEND, PSC_CTRL(pscdata));
+ wmb(); /* drain writebuffer */
}
return 0;
}
@@ -332,12 +332,12 @@ static int au1xpsc_i2s_drvprobe(struct platform_device *pdev)
/* preserve PSC clock source set up by platform (dev.platform_data
* is already occupied by soc layer)
*/
- sel = au_readl(PSC_SEL(wd)) & PSC_SEL_CLK_MASK;
- au_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd));
- au_sync();
- au_writel(PSC_SEL_PS_I2SMODE | sel, PSC_SEL(wd));
- au_writel(0, I2S_CFG(wd));
- au_sync();
+ sel = __raw_readl(PSC_SEL(wd)) & PSC_SEL_CLK_MASK;
+ __raw_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd));
+ wmb(); /* drain writebuffer */
+ __raw_writel(PSC_SEL_PS_I2SMODE | sel, PSC_SEL(wd));
+ __raw_writel(0, I2S_CFG(wd));
+ wmb(); /* drain writebuffer */
/* preconfigure: set max rx/tx fifo depths */
wd->cfg |= PSC_I2SCFG_RT_FIFO8 | PSC_I2SCFG_TT_FIFO8;
@@ -364,10 +364,10 @@ static int au1xpsc_i2s_drvremove(struct platform_device *pdev)
snd_soc_unregister_component(&pdev->dev);
- au_writel(0, I2S_CFG(wd));
- au_sync();
- au_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd));
- au_sync();
+ __raw_writel(0, I2S_CFG(wd));
+ wmb(); /* drain writebuffer */
+ __raw_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd));
+ wmb(); /* drain writebuffer */
return 0;
}
@@ -378,12 +378,12 @@ static int au1xpsc_i2s_drvsuspend(struct device *dev)
struct au1xpsc_audio_data *wd = dev_get_drvdata(dev);
/* save interesting register and disable PSC */
- wd->pm[0] = au_readl(PSC_SEL(wd));
+ wd->pm[0] = __raw_readl(PSC_SEL(wd));
- au_writel(0, I2S_CFG(wd));
- au_sync();
- au_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd));
- au_sync();
+ __raw_writel(0, I2S_CFG(wd));
+ wmb(); /* drain writebuffer */
+ __raw_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd));
+ wmb(); /* drain writebuffer */
return 0;
}
@@ -393,12 +393,12 @@ static int au1xpsc_i2s_drvresume(struct device *dev)
struct au1xpsc_audio_data *wd = dev_get_drvdata(dev);
/* select I2S mode and PSC clock */
- au_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd));
- au_sync();
- au_writel(0, PSC_SEL(wd));
- au_sync();
- au_writel(wd->pm[0], PSC_SEL(wd));
- au_sync();
+ __raw_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd));
+ wmb(); /* drain writebuffer */
+ __raw_writel(0, PSC_SEL(wd));
+ wmb(); /* drain writebuffer */
+ __raw_writel(wd->pm[0], PSC_SEL(wd));
+ wmb(); /* drain writebuffer */
return 0;
}
diff --git a/sound/soc/au1x/psc.h b/sound/soc/au1x/psc.h
index b16b2e02e0c9..74dffeb641fa 100644
--- a/sound/soc/au1x/psc.h
+++ b/sound/soc/au1x/psc.h
@@ -27,16 +27,16 @@ struct au1xpsc_audio_data {
};
/* easy access macros */
-#define PSC_CTRL(x) ((unsigned long)((x)->mmio) + PSC_CTRL_OFFSET)
-#define PSC_SEL(x) ((unsigned long)((x)->mmio) + PSC_SEL_OFFSET)
-#define I2S_STAT(x) ((unsigned long)((x)->mmio) + PSC_I2SSTAT_OFFSET)
-#define I2S_CFG(x) ((unsigned long)((x)->mmio) + PSC_I2SCFG_OFFSET)
-#define I2S_PCR(x) ((unsigned long)((x)->mmio) + PSC_I2SPCR_OFFSET)
-#define AC97_CFG(x) ((unsigned long)((x)->mmio) + PSC_AC97CFG_OFFSET)
-#define AC97_CDC(x) ((unsigned long)((x)->mmio) + PSC_AC97CDC_OFFSET)
-#define AC97_EVNT(x) ((unsigned long)((x)->mmio) + PSC_AC97EVNT_OFFSET)
-#define AC97_PCR(x) ((unsigned long)((x)->mmio) + PSC_AC97PCR_OFFSET)
-#define AC97_RST(x) ((unsigned long)((x)->mmio) + PSC_AC97RST_OFFSET)
-#define AC97_STAT(x) ((unsigned long)((x)->mmio) + PSC_AC97STAT_OFFSET)
+#define PSC_CTRL(x) ((x)->mmio + PSC_CTRL_OFFSET)
+#define PSC_SEL(x) ((x)->mmio + PSC_SEL_OFFSET)
+#define I2S_STAT(x) ((x)->mmio + PSC_I2SSTAT_OFFSET)
+#define I2S_CFG(x) ((x)->mmio + PSC_I2SCFG_OFFSET)
+#define I2S_PCR(x) ((x)->mmio + PSC_I2SPCR_OFFSET)
+#define AC97_CFG(x) ((x)->mmio + PSC_AC97CFG_OFFSET)
+#define AC97_CDC(x) ((x)->mmio + PSC_AC97CDC_OFFSET)
+#define AC97_EVNT(x) ((x)->mmio + PSC_AC97EVNT_OFFSET)
+#define AC97_PCR(x) ((x)->mmio + PSC_AC97PCR_OFFSET)
+#define AC97_RST(x) ((x)->mmio + PSC_AC97RST_OFFSET)
+#define AC97_STAT(x) ((x)->mmio + PSC_AC97STAT_OFFSET)
#endif
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index 2f2e91ac690f..2c71f16bd661 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -107,7 +107,7 @@ static int arizona_spk_ev(struct snd_soc_dapm_widget *w,
break;
case SND_SOC_DAPM_POST_PMU:
val = snd_soc_read(codec, ARIZONA_INTERRUPT_RAW_STATUS_3);
- if (val & ARIZONA_SPK_SHUTDOWN_STS) {
+ if (val & ARIZONA_SPK_OVERHEAT_STS) {
dev_crit(arizona->dev,
"Speaker not enabled due to temperature\n");
return -EBUSY;
@@ -159,7 +159,7 @@ static irqreturn_t arizona_thermal_warn(int irq, void *data)
if (ret != 0) {
dev_err(arizona->dev, "Failed to read thermal status: %d\n",
ret);
- } else if (val & ARIZONA_SPK_SHUTDOWN_WARN_STS) {
+ } else if (val & ARIZONA_SPK_OVERHEAT_WARN_STS) {
dev_crit(arizona->dev, "Thermal warning\n");
}
@@ -177,7 +177,7 @@ static irqreturn_t arizona_thermal_shutdown(int irq, void *data)
if (ret != 0) {
dev_err(arizona->dev, "Failed to read thermal status: %d\n",
ret);
- } else if (val & ARIZONA_SPK_SHUTDOWN_STS) {
+ } else if (val & ARIZONA_SPK_OVERHEAT_STS) {
dev_crit(arizona->dev, "Thermal shutdown\n");
ret = regmap_update_bits(arizona->regmap,
ARIZONA_OUTPUT_ENABLES_1,
@@ -223,7 +223,7 @@ int arizona_init_spk(struct snd_soc_codec *codec)
break;
}
- ret = arizona_request_irq(arizona, ARIZONA_IRQ_SPK_SHUTDOWN_WARN,
+ ret = arizona_request_irq(arizona, ARIZONA_IRQ_SPK_OVERHEAT_WARN,
"Thermal warning", arizona_thermal_warn,
arizona);
if (ret != 0)
@@ -231,7 +231,7 @@ int arizona_init_spk(struct snd_soc_codec *codec)
"Failed to get thermal warning IRQ: %d\n",
ret);
- ret = arizona_request_irq(arizona, ARIZONA_IRQ_SPK_SHUTDOWN,
+ ret = arizona_request_irq(arizona, ARIZONA_IRQ_SPK_OVERHEAT,
"Thermal shutdown", arizona_thermal_shutdown,
arizona);
if (ret != 0)
@@ -1278,6 +1278,8 @@ static int arizona_hw_params(struct snd_pcm_substream *substream,
else
rates = &arizona_48k_bclk_rates[0];
+ wl = snd_pcm_format_width(params_format(params));
+
if (tdm_slots) {
arizona_aif_dbg(dai, "Configuring for %d %d bit TDM slots\n",
tdm_slots, tdm_width);
@@ -1285,6 +1287,7 @@ static int arizona_hw_params(struct snd_pcm_substream *substream,
channels = tdm_slots;
} else {
bclk_target = snd_soc_params_to_bclk(params);
+ tdm_width = wl;
}
if (chan_limit && chan_limit < channels) {
@@ -1319,8 +1322,7 @@ static int arizona_hw_params(struct snd_pcm_substream *substream,
arizona_aif_dbg(dai, "BCLK %dHz LRCLK %dHz\n",
rates[bclk], rates[bclk] / lrclk);
- wl = snd_pcm_format_width(params_format(params));
- frame = wl << ARIZONA_AIF1TX_WL_SHIFT | wl;
+ frame = wl << ARIZONA_AIF1TX_WL_SHIFT | tdm_width;
reconfig = arizona_aif_cfg_changed(codec, base, bclk, lrclk, frame);
diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c
index a20b30ca52c0..69a85164357c 100644
--- a/sound/soc/codecs/cs4265.c
+++ b/sound/soc/codecs/cs4265.c
@@ -282,10 +282,10 @@ static const struct cs4265_clk_para clk_map_table[] = {
/*64k*/
{8192000, 64000, 1, 0},
- {1228800, 64000, 1, 1},
- {1693440, 64000, 1, 2},
- {2457600, 64000, 1, 3},
- {3276800, 64000, 1, 4},
+ {12288000, 64000, 1, 1},
+ {16934400, 64000, 1, 2},
+ {24576000, 64000, 1, 3},
+ {32768000, 64000, 1, 4},
/* 88.2k */
{11289600, 88200, 1, 0},
@@ -435,10 +435,10 @@ static int cs4265_pcm_hw_params(struct snd_pcm_substream *substream,
index = cs4265_get_clk_index(cs4265->sysclk, params_rate(params));
if (index >= 0) {
snd_soc_update_bits(codec, CS4265_ADC_CTL,
- CS4265_ADC_FM, clk_map_table[index].fm_mode);
+ CS4265_ADC_FM, clk_map_table[index].fm_mode << 6);
snd_soc_update_bits(codec, CS4265_MCLK_FREQ,
CS4265_MCLK_FREQ_MASK,
- clk_map_table[index].mclkdiv);
+ clk_map_table[index].mclkdiv << 4);
} else {
dev_err(codec->dev, "can't get correct mclk\n");
@@ -458,12 +458,12 @@ static int cs4265_pcm_hw_params(struct snd_pcm_substream *substream,
if (params_width(params) == 16) {
snd_soc_update_bits(codec, CS4265_DAC_CTL,
CS4265_DAC_CTL_DIF, (1 << 5));
- snd_soc_update_bits(codec, CS4265_ADC_CTL,
+ snd_soc_update_bits(codec, CS4265_SPDIF_CTL2,
CS4265_SPDIF_CTL2_DIF, (1 << 7));
} else {
snd_soc_update_bits(codec, CS4265_DAC_CTL,
CS4265_DAC_CTL_DIF, (3 << 5));
- snd_soc_update_bits(codec, CS4265_ADC_CTL,
+ snd_soc_update_bits(codec, CS4265_SPDIF_CTL2,
CS4265_SPDIF_CTL2_DIF, (1 << 7));
}
break;
@@ -472,7 +472,7 @@ static int cs4265_pcm_hw_params(struct snd_pcm_substream *substream,
CS4265_DAC_CTL_DIF, 0);
snd_soc_update_bits(codec, CS4265_ADC_CTL,
CS4265_ADC_DIF, 0);
- snd_soc_update_bits(codec, CS4265_ADC_CTL,
+ snd_soc_update_bits(codec, CS4265_SPDIF_CTL2,
CS4265_SPDIF_CTL2_DIF, (1 << 6));
break;
diff --git a/sound/soc/codecs/da732x.h b/sound/soc/codecs/da732x.h
index 1dceafeec415..f586cbd30b77 100644
--- a/sound/soc/codecs/da732x.h
+++ b/sound/soc/codecs/da732x.h
@@ -11,7 +11,7 @@
*/
#ifndef __DA732X_H_
-#define __DA732X_H
+#define __DA732X_H_
#include <sound/soc.h>
diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c
index 163ec3855fd4..0c8aefab404c 100644
--- a/sound/soc/codecs/pcm512x.c
+++ b/sound/soc/codecs/pcm512x.c
@@ -259,13 +259,13 @@ static const struct soc_enum pcm512x_veds =
pcm512x_ramp_step_text);
static const struct snd_kcontrol_new pcm512x_controls[] = {
-SOC_DOUBLE_R_TLV("Playback Digital Volume", PCM512x_DIGITAL_VOLUME_2,
+SOC_DOUBLE_R_TLV("Digital Playback Volume", PCM512x_DIGITAL_VOLUME_2,
PCM512x_DIGITAL_VOLUME_3, 0, 255, 1, digital_tlv),
SOC_DOUBLE_TLV("Playback Volume", PCM512x_ANALOG_GAIN_CTRL,
PCM512x_LAGN_SHIFT, PCM512x_RAGN_SHIFT, 1, 1, analog_tlv),
SOC_DOUBLE_TLV("Playback Boost Volume", PCM512x_ANALOG_GAIN_BOOST,
PCM512x_AGBL_SHIFT, PCM512x_AGBR_SHIFT, 1, 0, boost_tlv),
-SOC_DOUBLE("Playback Digital Switch", PCM512x_MUTE, PCM512x_RQML_SHIFT,
+SOC_DOUBLE("Digital Playback Switch", PCM512x_MUTE, PCM512x_RQML_SHIFT,
PCM512x_RQMR_SHIFT, 1, 1),
SOC_SINGLE("Deemphasis Switch", PCM512x_DSP, PCM512x_DEMP_SHIFT, 1, 1),
diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c
index e4f6102efc1a..b86b426f159d 100644
--- a/sound/soc/codecs/rt286.c
+++ b/sound/soc/codecs/rt286.c
@@ -51,7 +51,7 @@ static struct reg_default rt286_index_def[] = {
{ 0x04, 0xaf01 },
{ 0x08, 0x000d },
{ 0x09, 0xd810 },
- { 0x0a, 0x0060 },
+ { 0x0a, 0x0120 },
{ 0x0b, 0x0000 },
{ 0x0d, 0x2800 },
{ 0x0f, 0x0000 },
@@ -60,7 +60,7 @@ static struct reg_default rt286_index_def[] = {
{ 0x33, 0x0208 },
{ 0x49, 0x0004 },
{ 0x4f, 0x50e9 },
- { 0x50, 0x2c00 },
+ { 0x50, 0x2000 },
{ 0x63, 0x2902 },
{ 0x67, 0x1111 },
{ 0x68, 0x1016 },
@@ -104,7 +104,6 @@ static const struct reg_default rt286_reg[] = {
{ 0x02170700, 0x00000000 },
{ 0x02270100, 0x00000000 },
{ 0x02370100, 0x00000000 },
- { 0x02040000, 0x00004002 },
{ 0x01870700, 0x00000020 },
{ 0x00830000, 0x000000c3 },
{ 0x00930000, 0x000000c3 },
@@ -192,7 +191,6 @@ static int rt286_hw_write(void *context, unsigned int reg, unsigned int value)
/*handle index registers*/
if (reg <= 0xff) {
rt286_hw_write(client, RT286_COEF_INDEX, reg);
- reg = RT286_PROC_COEF;
for (i = 0; i < INDEX_CACHE_SIZE; i++) {
if (reg == rt286->index_cache[i].reg) {
rt286->index_cache[i].def = value;
@@ -200,6 +198,7 @@ static int rt286_hw_write(void *context, unsigned int reg, unsigned int value)
}
}
+ reg = RT286_PROC_COEF;
}
data[0] = (reg >> 24) & 0xff;
diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c
index 6bc6efdec550..f1ec6e6bd08a 100644
--- a/sound/soc/codecs/rt5640.c
+++ b/sound/soc/codecs/rt5640.c
@@ -2059,6 +2059,7 @@ static struct snd_soc_codec_driver soc_codec_dev_rt5640 = {
static const struct regmap_config rt5640_regmap = {
.reg_bits = 8,
.val_bits = 16,
+ .use_single_rw = true,
.max_register = RT5640_VENDOR_ID2 + 1 + (ARRAY_SIZE(rt5640_ranges) *
RT5640_PR_SPACING),
diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c
index 67f14556462f..5337c448b5e3 100644
--- a/sound/soc/codecs/rt5677.c
+++ b/sound/soc/codecs/rt5677.c
@@ -2135,10 +2135,10 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = {
{ "BST2", NULL, "IN2P" },
{ "BST2", NULL, "IN2N" },
- { "IN1P", NULL, "micbias1" },
- { "IN1N", NULL, "micbias1" },
- { "IN2P", NULL, "micbias1" },
- { "IN2N", NULL, "micbias1" },
+ { "IN1P", NULL, "MICBIAS1" },
+ { "IN1N", NULL, "MICBIAS1" },
+ { "IN2P", NULL, "MICBIAS1" },
+ { "IN2N", NULL, "MICBIAS1" },
{ "ADC 1", NULL, "BST1" },
{ "ADC 1", NULL, "ADC 1 power" },
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index 484b3bbe8624..4021cd435740 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -647,7 +647,7 @@ int ssm2602_probe(struct device *dev, enum ssm2602_type type,
return -ENOMEM;
dev_set_drvdata(dev, ssm2602);
- ssm2602->type = SSM2602;
+ ssm2602->type = type;
ssm2602->regmap = regmap;
return snd_soc_register_codec(dev, &soc_codec_dev_ssm2602,
diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c
index 9aa1323fb2ab..89c748dd3d6e 100644
--- a/sound/soc/codecs/sta529.c
+++ b/sound/soc/codecs/sta529.c
@@ -4,7 +4,7 @@
* sound/soc/codecs/sta529.c -- spear ALSA Soc codec driver
*
* Copyright (C) 2012 ST Microelectronics
- * Rajeev Kumar <rajeev-dlh.kumar@st.com>
+ * Rajeev Kumar <rajeevkumar.linux@gmail.com>
*
* This file is licensed under the terms of the GNU General Public
* License version 2. This program is licensed "as is" without any
@@ -426,5 +426,5 @@ static struct i2c_driver sta529_i2c_driver = {
module_i2c_driver(sta529_i2c_driver);
MODULE_DESCRIPTION("ASoC STA529 codec driver");
-MODULE_AUTHOR("Rajeev Kumar <rajeev-dlh.kumar@st.com>");
+MODULE_AUTHOR("Rajeev Kumar <rajeevkumar.linux@gmail.com>");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c
index 0f64c7890eed..aea9e1ff9126 100644
--- a/sound/soc/codecs/tlv320aic31xx.c
+++ b/sound/soc/codecs/tlv320aic31xx.c
@@ -189,46 +189,57 @@ static const struct aic31xx_rate_divs aic31xx_divs[] = {
/* mclk rate pll: p j d dosr ndac mdac aors nadc madc */
/* 8k rate */
{12000000, 8000, 1, 8, 1920, 128, 48, 2, 128, 48, 2},
+ {12000000, 8000, 1, 8, 1920, 128, 32, 3, 128, 32, 3},
{24000000, 8000, 2, 8, 1920, 128, 48, 2, 128, 48, 2},
{25000000, 8000, 2, 7, 8643, 128, 48, 2, 128, 48, 2},
/* 11.025k rate */
{12000000, 11025, 1, 7, 5264, 128, 32, 2, 128, 32, 2},
+ {12000000, 11025, 1, 8, 4672, 128, 24, 3, 128, 24, 3},
{24000000, 11025, 2, 7, 5264, 128, 32, 2, 128, 32, 2},
{25000000, 11025, 2, 7, 2253, 128, 32, 2, 128, 32, 2},
/* 16k rate */
{12000000, 16000, 1, 8, 1920, 128, 24, 2, 128, 24, 2},
+ {12000000, 16000, 1, 8, 1920, 128, 16, 3, 128, 16, 3},
{24000000, 16000, 2, 8, 1920, 128, 24, 2, 128, 24, 2},
{25000000, 16000, 2, 7, 8643, 128, 24, 2, 128, 24, 2},
/* 22.05k rate */
{12000000, 22050, 1, 7, 5264, 128, 16, 2, 128, 16, 2},
+ {12000000, 22050, 1, 8, 4672, 128, 12, 3, 128, 12, 3},
{24000000, 22050, 2, 7, 5264, 128, 16, 2, 128, 16, 2},
{25000000, 22050, 2, 7, 2253, 128, 16, 2, 128, 16, 2},
/* 32k rate */
{12000000, 32000, 1, 8, 1920, 128, 12, 2, 128, 12, 2},
+ {12000000, 32000, 1, 8, 1920, 128, 8, 3, 128, 8, 3},
{24000000, 32000, 2, 8, 1920, 128, 12, 2, 128, 12, 2},
{25000000, 32000, 2, 7, 8643, 128, 12, 2, 128, 12, 2},
/* 44.1k rate */
{12000000, 44100, 1, 7, 5264, 128, 8, 2, 128, 8, 2},
+ {12000000, 44100, 1, 8, 4672, 128, 6, 3, 128, 6, 3},
{24000000, 44100, 2, 7, 5264, 128, 8, 2, 128, 8, 2},
{25000000, 44100, 2, 7, 2253, 128, 8, 2, 128, 8, 2},
/* 48k rate */
{12000000, 48000, 1, 8, 1920, 128, 8, 2, 128, 8, 2},
+ {12000000, 48000, 1, 7, 6800, 96, 5, 4, 96, 5, 4},
{24000000, 48000, 2, 8, 1920, 128, 8, 2, 128, 8, 2},
{25000000, 48000, 2, 7, 8643, 128, 8, 2, 128, 8, 2},
/* 88.2k rate */
{12000000, 88200, 1, 7, 5264, 64, 8, 2, 64, 8, 2},
+ {12000000, 88200, 1, 8, 4672, 64, 6, 3, 64, 6, 3},
{24000000, 88200, 2, 7, 5264, 64, 8, 2, 64, 8, 2},
{25000000, 88200, 2, 7, 2253, 64, 8, 2, 64, 8, 2},
/* 96k rate */
{12000000, 96000, 1, 8, 1920, 64, 8, 2, 64, 8, 2},
+ {12000000, 96000, 1, 7, 6800, 48, 5, 4, 48, 5, 4},
{24000000, 96000, 2, 8, 1920, 64, 8, 2, 64, 8, 2},
{25000000, 96000, 2, 7, 8643, 64, 8, 2, 64, 8, 2},
/* 176.4k rate */
{12000000, 176400, 1, 7, 5264, 32, 8, 2, 32, 8, 2},
+ {12000000, 176400, 1, 8, 4672, 32, 6, 3, 32, 6, 3},
{24000000, 176400, 2, 7, 5264, 32, 8, 2, 32, 8, 2},
{25000000, 176400, 2, 7, 2253, 32, 8, 2, 32, 8, 2},
/* 192k rate */
{12000000, 192000, 1, 8, 1920, 32, 8, 2, 32, 8, 2},
+ {12000000, 192000, 1, 7, 6800, 24, 5, 4, 24, 5, 4},
{24000000, 192000, 2, 8, 1920, 32, 8, 2, 32, 8, 2},
{25000000, 192000, 2, 7, 8643, 32, 8, 2, 32, 8, 2},
};
@@ -680,7 +691,9 @@ static int aic31xx_setup_pll(struct snd_soc_codec *codec,
struct snd_pcm_hw_params *params)
{
struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
+ int bclk_score = snd_soc_params_to_frame_size(params);
int bclk_n = 0;
+ int match = -1;
int i;
/* Use PLL as CODEC_CLKIN and DAC_CLK as BDIV_CLKIN */
@@ -691,15 +704,37 @@ static int aic31xx_setup_pll(struct snd_soc_codec *codec,
for (i = 0; i < ARRAY_SIZE(aic31xx_divs); i++) {
if (aic31xx_divs[i].rate == params_rate(params) &&
- aic31xx_divs[i].mclk == aic31xx->sysclk)
- break;
+ aic31xx_divs[i].mclk == aic31xx->sysclk) {
+ int s = (aic31xx_divs[i].dosr * aic31xx_divs[i].mdac) %
+ snd_soc_params_to_frame_size(params);
+ int bn = (aic31xx_divs[i].dosr * aic31xx_divs[i].mdac) /
+ snd_soc_params_to_frame_size(params);
+ if (s < bclk_score && bn > 0) {
+ match = i;
+ bclk_n = bn;
+ bclk_score = s;
+ }
+ }
}
- if (i == ARRAY_SIZE(aic31xx_divs)) {
- dev_err(codec->dev, "%s: Sampling rate %u not supported\n",
+ if (match == -1) {
+ dev_err(codec->dev,
+ "%s: Sample rate (%u) and format not supported\n",
__func__, params_rate(params));
+ /* See bellow for details how fix this. */
return -EINVAL;
}
+ if (bclk_score != 0) {
+ dev_warn(codec->dev, "Can not produce exact bitclock");
+ /* This is fine if using dsp format, but if using i2s
+ there may be trouble. To fix the issue edit the
+ aic31xx_divs table for your mclk and sample
+ rate. Details can be found from:
+ http://www.ti.com/lit/ds/symlink/tlv320aic3100.pdf
+ Section: 5.6 CLOCK Generation and PLL
+ */
+ }
+ i = match;
/* PLL configuration */
snd_soc_update_bits(codec, AIC31XX_PLLPR, AIC31XX_PLL_MASK,
@@ -729,14 +764,6 @@ static int aic31xx_setup_pll(struct snd_soc_codec *codec,
snd_soc_write(codec, AIC31XX_AOSR, aic31xx_divs[i].aosr);
/* Bit clock divider configuration. */
- bclk_n = (aic31xx_divs[i].dosr * aic31xx_divs[i].mdac)
- / snd_soc_params_to_frame_size(params);
- if (bclk_n == 0) {
- dev_err(codec->dev, "%s: Not enough BLCK bandwidth\n",
- __func__);
- return -EINVAL;
- }
-
snd_soc_update_bits(codec, AIC31XX_BCLKN,
AIC31XX_PLL_MASK, bclk_n);
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index c28508da34cf..68347b55f6e1 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -403,7 +403,8 @@ out:
return ret;
}
-static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div)
+static int __davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id,
+ int div, bool explicit)
{
struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai);
@@ -420,7 +421,8 @@ static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div
ACLKXDIV(div - 1), ACLKXDIV_MASK);
mcasp_mod_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG,
ACLKRDIV(div - 1), ACLKRDIV_MASK);
- mcasp->bclk_div = div;
+ if (explicit)
+ mcasp->bclk_div = div;
break;
case 2: /* BCLK/LRCLK ratio */
@@ -434,6 +436,12 @@ static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div
return 0;
}
+static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id,
+ int div)
+{
+ return __davinci_mcasp_set_clkdiv(dai, div_id, div, 1);
+}
+
static int davinci_mcasp_set_sysclk(struct snd_soc_dai *dai, int clk_id,
unsigned int freq, int dir)
{
@@ -459,8 +467,17 @@ static int davinci_config_channel_size(struct davinci_mcasp *mcasp,
{
u32 fmt;
u32 tx_rotate = (word_length / 4) & 0x7;
- u32 rx_rotate = (32 - word_length) / 4;
u32 mask = (1ULL << word_length) - 1;
+ /*
+ * For captured data we should not rotate, inversion and masking is
+ * enoguh to get the data to the right position:
+ * Format data from bus after reverse (XRBUF)
+ * S16_LE: |LSB|MSB|xxx|xxx| |xxx|xxx|MSB|LSB|
+ * S24_3LE: |LSB|DAT|MSB|xxx| |xxx|MSB|DAT|LSB|
+ * S24_LE: |LSB|DAT|MSB|xxx| |xxx|MSB|DAT|LSB|
+ * S32_LE: |LSB|DAT|DAT|MSB| |MSB|DAT|DAT|LSB|
+ */
+ u32 rx_rotate = 0;
/*
* if s BCLK-to-LRCLK ratio has been configured via the set_clkdiv()
@@ -738,7 +755,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
"Inaccurate BCLK: %u Hz / %u != %u Hz\n",
mcasp->sysclk_freq, div, bclk_freq);
}
- davinci_mcasp_set_clkdiv(cpu_dai, 1, div);
+ __davinci_mcasp_set_clkdiv(cpu_dai, 1, div, 0);
}
ret = mcasp_common_hw_param(mcasp, substream->stream,
diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c
index 25c31f1655f6..e961388e6e9c 100644
--- a/sound/soc/dwc/designware_i2s.c
+++ b/sound/soc/dwc/designware_i2s.c
@@ -4,7 +4,7 @@
* sound/soc/dwc/designware_i2s.c
*
* Copyright (C) 2010 ST Microelectronics
- * Rajeev Kumar <rajeev-dlh.kumar@st.com>
+ * Rajeev Kumar <rajeevkumar.linux@gmail.com>
*
* This file is licensed under the terms of the GNU General Public
* License version 2. This program is licensed "as is" without any
@@ -455,7 +455,7 @@ static struct platform_driver dw_i2s_driver = {
module_platform_driver(dw_i2s_driver);
-MODULE_AUTHOR("Rajeev Kumar <rajeev-dlh.kumar@st.com>");
+MODULE_AUTHOR("Rajeev Kumar <rajeevkumar.linux@gmail.com>");
MODULE_DESCRIPTION("DESIGNWARE I2S SoC Interface");
MODULE_LICENSE("GPL");
MODULE_ALIAS("platform:designware_i2s");
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index f54a8fc99291..f3012b645b51 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -49,7 +49,6 @@ config SND_SOC_FSL_ESAI
tristate "Enhanced Serial Audio Interface (ESAI) module support"
select REGMAP_MMIO
select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n
- select SND_SOC_FSL_UTILS
help
Say Y if you want to add Enhanced Synchronous Audio Interface
(ESAI) support for the Freescale CPUs.
diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index 72d154e7dd03..a3b29ed84963 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -18,7 +18,6 @@
#include "fsl_esai.h"
#include "imx-pcm.h"
-#include "fsl_utils.h"
#define FSL_ESAI_RATES SNDRV_PCM_RATE_8000_192000
#define FSL_ESAI_FORMATS (SNDRV_PCM_FMTBIT_S8 | \
@@ -607,7 +606,6 @@ static struct snd_soc_dai_ops fsl_esai_dai_ops = {
.hw_params = fsl_esai_hw_params,
.set_sysclk = fsl_esai_set_dai_sysclk,
.set_fmt = fsl_esai_set_dai_fmt,
- .xlate_tdm_slot_mask = fsl_asoc_xlate_tdm_slot_mask,
.set_tdm_slot = fsl_esai_set_dai_tdm_slot,
};
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 87eb5776a39b..de6ab06f58a5 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -748,8 +748,9 @@ static int fsl_ssi_hw_free(struct snd_pcm_substream *substream,
return 0;
}
-static int _fsl_ssi_set_dai_fmt(struct fsl_ssi_private *ssi_private,
- unsigned int fmt)
+static int _fsl_ssi_set_dai_fmt(struct device *dev,
+ struct fsl_ssi_private *ssi_private,
+ unsigned int fmt)
{
struct regmap *regs = ssi_private->regs;
u32 strcr = 0, stcr, srcr, scr, mask;
@@ -758,7 +759,7 @@ static int _fsl_ssi_set_dai_fmt(struct fsl_ssi_private *ssi_private,
ssi_private->dai_fmt = fmt;
if (fsl_ssi_is_i2s_master(ssi_private) && IS_ERR(ssi_private->baudclk)) {
- dev_err(&ssi_private->pdev->dev, "baudclk is missing which is necessary for master mode\n");
+ dev_err(dev, "baudclk is missing which is necessary for master mode\n");
return -EINVAL;
}
@@ -913,7 +914,7 @@ static int fsl_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
{
struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai);
- return _fsl_ssi_set_dai_fmt(ssi_private, fmt);
+ return _fsl_ssi_set_dai_fmt(cpu_dai->dev, ssi_private, fmt);
}
/**
@@ -1387,7 +1388,8 @@ static int fsl_ssi_probe(struct platform_device *pdev)
done:
if (ssi_private->dai_fmt)
- _fsl_ssi_set_dai_fmt(ssi_private, ssi_private->dai_fmt);
+ _fsl_ssi_set_dai_fmt(&pdev->dev, ssi_private,
+ ssi_private->dai_fmt);
return 0;
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 159e517fa09a..cef7776b712c 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -481,12 +481,19 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
snd_soc_card_set_drvdata(&priv->snd_card, priv);
ret = devm_snd_soc_register_card(&pdev->dev, &priv->snd_card);
+ if (ret >= 0)
+ return ret;
err:
asoc_simple_card_unref(pdev);
return ret;
}
+static int asoc_simple_card_remove(struct platform_device *pdev)
+{
+ return asoc_simple_card_unref(pdev);
+}
+
static const struct of_device_id asoc_simple_of_match[] = {
{ .compatible = "simple-audio-card", },
{},
@@ -500,6 +507,7 @@ static struct platform_driver asoc_simple_card = {
.of_match_table = asoc_simple_of_match,
},
.probe = asoc_simple_card_probe,
+ .remove = asoc_simple_card_remove,
};
module_platform_driver(asoc_simple_card);
diff --git a/sound/soc/intel/sst-acpi.c b/sound/soc/intel/sst-acpi.c
index 42edc6f4fc4a..03d0a166b635 100644
--- a/sound/soc/intel/sst-acpi.c
+++ b/sound/soc/intel/sst-acpi.c
@@ -246,8 +246,8 @@ static struct sst_acpi_desc sst_acpi_broadwell_desc = {
};
static struct sst_acpi_mach baytrail_machines[] = {
- { "10EC5640", "byt-rt5640", "intel/fw_sst_0f28.bin-i2s_master" },
- { "193C9890", "byt-max98090", "intel/fw_sst_0f28.bin-i2s_master" },
+ { "10EC5640", "byt-rt5640", "intel/fw_sst_0f28.bin-48kHz_i2s_master" },
+ { "193C9890", "byt-max98090", "intel/fw_sst_0f28.bin-48kHz_i2s_master" },
{}
};
diff --git a/sound/soc/intel/sst-baytrail-ipc.c b/sound/soc/intel/sst-baytrail-ipc.c
index 67673a2c0f41..b4ad98c43e5c 100644
--- a/sound/soc/intel/sst-baytrail-ipc.c
+++ b/sound/soc/intel/sst-baytrail-ipc.c
@@ -817,7 +817,7 @@ static struct sst_dsp_device byt_dev = {
.ops = &sst_byt_ops,
};
-int sst_byt_dsp_suspend_noirq(struct device *dev, struct sst_pdata *pdata)
+int sst_byt_dsp_suspend_late(struct device *dev, struct sst_pdata *pdata)
{
struct sst_byt *byt = pdata->dsp;
@@ -826,14 +826,6 @@ int sst_byt_dsp_suspend_noirq(struct device *dev, struct sst_pdata *pdata)
sst_byt_drop_all(byt);
dev_dbg(byt->dev, "dsp in reset\n");
- return 0;
-}
-EXPORT_SYMBOL_GPL(sst_byt_dsp_suspend_noirq);
-
-int sst_byt_dsp_suspend_late(struct device *dev, struct sst_pdata *pdata)
-{
- struct sst_byt *byt = pdata->dsp;
-
dev_dbg(byt->dev, "free all blocks and unload fw\n");
sst_fw_unload(byt->fw);
diff --git a/sound/soc/intel/sst-baytrail-ipc.h b/sound/soc/intel/sst-baytrail-ipc.h
index 06a4d202689b..8faff6dcf25d 100644
--- a/sound/soc/intel/sst-baytrail-ipc.h
+++ b/sound/soc/intel/sst-baytrail-ipc.h
@@ -66,7 +66,6 @@ int sst_byt_get_dsp_position(struct sst_byt *byt,
int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata);
void sst_byt_dsp_free(struct device *dev, struct sst_pdata *pdata);
struct sst_dsp *sst_byt_get_dsp(struct sst_byt *byt);
-int sst_byt_dsp_suspend_noirq(struct device *dev, struct sst_pdata *pdata);
int sst_byt_dsp_suspend_late(struct device *dev, struct sst_pdata *pdata);
int sst_byt_dsp_boot(struct device *dev, struct sst_pdata *pdata);
int sst_byt_dsp_wait_for_ready(struct device *dev, struct sst_pdata *pdata);
diff --git a/sound/soc/intel/sst-baytrail-pcm.c b/sound/soc/intel/sst-baytrail-pcm.c
index 599401c0c655..eab1c7d85187 100644
--- a/sound/soc/intel/sst-baytrail-pcm.c
+++ b/sound/soc/intel/sst-baytrail-pcm.c
@@ -59,6 +59,9 @@ struct sst_byt_priv_data {
/* DAI data */
struct sst_byt_pcm_data pcm[BYT_PCM_COUNT];
+
+ /* flag indicating is stream context restore needed after suspend */
+ bool restore_stream;
};
/* this may get called several times by oss emulation */
@@ -184,7 +187,10 @@ static int sst_byt_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
sst_byt_stream_start(byt, pcm_data->stream, 0);
break;
case SNDRV_PCM_TRIGGER_RESUME:
- schedule_work(&pcm_data->work);
+ if (pdata->restore_stream == true)
+ schedule_work(&pcm_data->work);
+ else
+ sst_byt_stream_resume(byt, pcm_data->stream);
break;
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
sst_byt_stream_resume(byt, pcm_data->stream);
@@ -193,6 +199,7 @@ static int sst_byt_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
sst_byt_stream_stop(byt, pcm_data->stream);
break;
case SNDRV_PCM_TRIGGER_SUSPEND:
+ pdata->restore_stream = false;
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
sst_byt_stream_pause(byt, pcm_data->stream);
break;
@@ -404,26 +411,10 @@ static const struct snd_soc_component_driver byt_dai_component = {
};
#ifdef CONFIG_PM
-static int sst_byt_pcm_dev_suspend_noirq(struct device *dev)
-{
- struct sst_pdata *sst_pdata = dev_get_platdata(dev);
- int ret;
-
- dev_dbg(dev, "suspending noirq\n");
-
- /* at this point all streams will be stopped and context saved */
- ret = sst_byt_dsp_suspend_noirq(dev, sst_pdata);
- if (ret < 0) {
- dev_err(dev, "failed to suspend %d\n", ret);
- return ret;
- }
-
- return ret;
-}
-
static int sst_byt_pcm_dev_suspend_late(struct device *dev)
{
struct sst_pdata *sst_pdata = dev_get_platdata(dev);
+ struct sst_byt_priv_data *priv_data = dev_get_drvdata(dev);
int ret;
dev_dbg(dev, "suspending late\n");
@@ -434,34 +425,30 @@ static int sst_byt_pcm_dev_suspend_late(struct device *dev)
return ret;
}
+ priv_data->restore_stream = true;
+
return ret;
}
static int sst_byt_pcm_dev_resume_early(struct device *dev)
{
struct sst_pdata *sst_pdata = dev_get_platdata(dev);
+ int ret;
dev_dbg(dev, "resume early\n");
/* load fw and boot DSP */
- return sst_byt_dsp_boot(dev, sst_pdata);
-}
-
-static int sst_byt_pcm_dev_resume(struct device *dev)
-{
- struct sst_pdata *sst_pdata = dev_get_platdata(dev);
-
- dev_dbg(dev, "resume\n");
+ ret = sst_byt_dsp_boot(dev, sst_pdata);
+ if (ret)
+ return ret;
/* wait for FW to finish booting */
return sst_byt_dsp_wait_for_ready(dev, sst_pdata);
}
static const struct dev_pm_ops sst_byt_pm_ops = {
- .suspend_noirq = sst_byt_pcm_dev_suspend_noirq,
.suspend_late = sst_byt_pcm_dev_suspend_late,
.resume_early = sst_byt_pcm_dev_resume_early,
- .resume = sst_byt_pcm_dev_resume,
};
#define SST_BYT_PM_OPS (&sst_byt_pm_ops)
diff --git a/sound/soc/omap/omap-twl4030.c b/sound/soc/omap/omap-twl4030.c
index f8a6adc2d81c..4336d1831485 100644
--- a/sound/soc/omap/omap-twl4030.c
+++ b/sound/soc/omap/omap-twl4030.c
@@ -260,7 +260,7 @@ static struct snd_soc_dai_link omap_twl4030_dai_links[] = {
.stream_name = "TWL4030 Voice",
.cpu_dai_name = "omap-mcbsp.3",
.codec_dai_name = "twl4030-voice",
- .platform_name = "omap-mcbsp.2",
+ .platform_name = "omap-mcbsp.3",
.codec_name = "twl4030-codec",
.dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF |
SND_SOC_DAIFMT_CBM_CFM,
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 0109f6c2334e..a8e097433074 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -765,9 +765,7 @@ static int pxa_ssp_remove(struct snd_soc_dai *dai)
SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_64000 | \
SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
-#define PXA_SSP_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\
- SNDRV_PCM_FMTBIT_S24_LE | \
- SNDRV_PCM_FMTBIT_S32_LE)
+#define PXA_SSP_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE)
static const struct snd_soc_dai_ops pxa_ssp_dai_ops = {
.startup = pxa_ssp_startup,
diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c
index 8d8e4b59049f..fb9e05c9f471 100644
--- a/sound/soc/rockchip/rockchip_i2s.c
+++ b/sound/soc/rockchip/rockchip_i2s.c
@@ -165,13 +165,14 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
struct rk_i2s_dev *i2s = to_info(cpu_dai);
unsigned int mask = 0, val = 0;
- mask = I2S_CKR_MSS_SLAVE;
+ mask = I2S_CKR_MSS_MASK;
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBS_CFS:
- val = I2S_CKR_MSS_SLAVE;
+ /* Set source clock in Master mode */
+ val = I2S_CKR_MSS_MASTER;
break;
case SND_SOC_DAIFMT_CBM_CFM:
- val = I2S_CKR_MSS_MASTER;
+ val = I2S_CKR_MSS_SLAVE;
break;
default:
return -EINVAL;
@@ -361,6 +362,8 @@ static bool rockchip_i2s_rd_reg(struct device *dev, unsigned int reg)
case I2S_XFER:
case I2S_CLR:
case I2S_RXDR:
+ case I2S_FIFOLR:
+ case I2S_INTSR:
return true;
default:
return false;
@@ -370,8 +373,8 @@ static bool rockchip_i2s_rd_reg(struct device *dev, unsigned int reg)
static bool rockchip_i2s_volatile_reg(struct device *dev, unsigned int reg)
{
switch (reg) {
- case I2S_FIFOLR:
case I2S_INTSR:
+ case I2S_CLR:
return true;
default:
return false;
@@ -381,8 +384,6 @@ static bool rockchip_i2s_volatile_reg(struct device *dev, unsigned int reg)
static bool rockchip_i2s_precious_reg(struct device *dev, unsigned int reg)
{
switch (reg) {
- case I2S_FIFOLR:
- return true;
default:
return false;
}
diff --git a/sound/soc/samsung/goni_wm8994.c b/sound/soc/samsung/goni_wm8994.c
index 9506d7617223..3b527dcfc0aa 100644
--- a/sound/soc/samsung/goni_wm8994.c
+++ b/sound/soc/samsung/goni_wm8994.c
@@ -16,7 +16,7 @@
#include <sound/jack.h>
#include <asm/mach-types.h>
-#include <mach/gpio.h>
+#include <mach/gpio-samsung.h>
#include "../codecs/wm8994.h"
diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c
index 03eec22f0f46..9d513473b300 100644
--- a/sound/soc/samsung/i2s.c
+++ b/sound/soc/samsung/i2s.c
@@ -462,7 +462,7 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai,
if (dir == SND_SOC_CLOCK_IN)
rfs = 0;
- if ((rfs && other->rfs && (other->rfs != rfs)) ||
+ if ((rfs && other && other->rfs && (other->rfs != rfs)) ||
(any_active(i2s) &&
(((dir == SND_SOC_CLOCK_IN)
&& !(mod & MOD_CDCLKCON)) ||
@@ -762,7 +762,8 @@ static void i2s_shutdown(struct snd_pcm_substream *substream,
} else {
u32 mod = readl(i2s->addr + I2SMOD);
i2s->cdclk_out = !(mod & MOD_CDCLKCON);
- other->cdclk_out = i2s->cdclk_out;
+ if (other)
+ other->cdclk_out = i2s->cdclk_out;
}
/* Reset any constraint on RFS and BFS */
i2s->rfs = 0;
diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c
index 3fdf3be7b99a..f95e7ab135e8 100644
--- a/sound/soc/sh/rcar/gen.c
+++ b/sound/soc/sh/rcar/gen.c
@@ -247,7 +247,7 @@ rsnd_gen2_dma_addr(struct rsnd_priv *priv,
};
/* it shouldn't happen */
- if (use_dvc & !use_src)
+ if (use_dvc && !use_src)
dev_err(dev, "DVC is selected without SRC\n");
/* use SSIU or SSI ? */
diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c
index 27c06acce205..cecfab3cc948 100644
--- a/sound/soc/soc-compress.c
+++ b/sound/soc/soc-compress.c
@@ -101,10 +101,12 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream)
fe->dpcm[stream].runtime = fe_substream->runtime;
- if (dpcm_path_get(fe, stream, &list) <= 0) {
+ ret = dpcm_path_get(fe, stream, &list);
+ if (ret < 0)
+ goto fe_err;
+ else if (ret == 0)
dev_dbg(fe->dev, "ASoC: %s no valid %s route\n",
fe->dai_link->name, stream ? "capture" : "playback");
- }
/* calculate valid and active FE <-> BE dpcms */
dpcm_process_paths(fe, stream, &list, 1);
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index d4bfd4a9076f..d074aa91b023 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1325,7 +1325,7 @@ static int soc_post_component_init(struct snd_soc_pcm_runtime *rtd,
device_initialize(rtd->dev);
rtd->dev->parent = rtd->card->dev;
rtd->dev->release = rtd_release;
- rtd->dev->init_name = name;
+ dev_set_name(rtd->dev, "%s", name);
dev_set_drvdata(rtd->dev, rtd);
mutex_init(&rtd->pcm_mutex);
INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].be_clients);
@@ -3203,7 +3203,7 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol,
unsigned int val, mask;
void *data;
- if (!component->regmap)
+ if (!component->regmap || !params->num_regs)
return -EINVAL;
len = params->num_regs * component->val_bytes;
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 8348352dc2c6..177bd8639ef9 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -2860,12 +2860,14 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol,
struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol);
struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
unsigned int reg_val, val;
- int ret = 0;
- if (e->reg != SND_SOC_NOPM)
- ret = soc_dapm_read(dapm, e->reg, &reg_val);
- else
+ if (e->reg != SND_SOC_NOPM) {
+ int ret = soc_dapm_read(dapm, e->reg, &reg_val);
+ if (ret)
+ return ret;
+ } else {
reg_val = dapm_kcontrol_get_value(kcontrol);
+ }
val = (reg_val >> e->shift_l) & e->mask;
ucontrol->value.enumerated.item[0] = snd_soc_enum_val_to_item(e, val);
@@ -2875,7 +2877,7 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol,
ucontrol->value.enumerated.item[1] = val;
}
- return ret;
+ return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_double);
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 731fdb5b5f9b..642c86240752 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2352,7 +2352,11 @@ static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream)
mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME);
fe->dpcm[stream].runtime = fe_substream->runtime;
- if (dpcm_path_get(fe, stream, &list) <= 0) {
+ ret = dpcm_path_get(fe, stream, &list);
+ if (ret < 0) {
+ mutex_unlock(&fe->card->mutex);
+ return ret;
+ } else if (ret == 0) {
dev_dbg(fe->dev, "ASoC: %s no valid %s route\n",
fe->dai_link->name, stream ? "capture" : "playback");
}
diff --git a/sound/soc/spear/spear_pcm.c b/sound/soc/spear/spear_pcm.c
index 0e5a8f35d0ad..a7dc3c56f44d 100644
--- a/sound/soc/spear/spear_pcm.c
+++ b/sound/soc/spear/spear_pcm.c
@@ -4,7 +4,7 @@
* sound/soc/spear/spear_pcm.c
*
* Copyright (C) 2012 ST Microelectronics
- * Rajeev Kumar<rajeev-dlh.kumar@st.com>
+ * Rajeev Kumar<rajeevkumar.linux@gmail.com>
*
* This file is licensed under the terms of the GNU General Public
* License version 2. This program is licensed "as is" without any
@@ -50,6 +50,6 @@ int devm_spear_pcm_platform_register(struct device *dev,
}
EXPORT_SYMBOL_GPL(devm_spear_pcm_platform_register);
-MODULE_AUTHOR("Rajeev Kumar <rajeev-dlh.kumar@st.com>");
+MODULE_AUTHOR("Rajeev Kumar <rajeevkumar.linux@gmail.com>");
MODULE_DESCRIPTION("SPEAr PCM DMA module");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/tegra/tegra_asoc_utils.h b/sound/soc/tegra/tegra_asoc_utils.h
index 9577121ce971..ca8037634100 100644
--- a/sound/soc/tegra/tegra_asoc_utils.h
+++ b/sound/soc/tegra/tegra_asoc_utils.h
@@ -21,7 +21,7 @@
*/
#ifndef __TEGRA_ASOC_UTILS_H__
-#define __TEGRA_ASOC_UTILS_H_
+#define __TEGRA_ASOC_UTILS_H__
struct clk;
struct device;