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authorPaul Mundt <lethal@linux-sh.org>2012-02-24 08:23:23 +0400
committerPaul Mundt <lethal@linux-sh.org>2012-02-24 08:23:23 +0400
commit35eb304b5cd7b49d581bda79218b8134f3b689ea (patch)
tree3d75d9ada70814161d035b2f9166fee05d257dfb /sound/soc
parentca0cc30109241f280eb871794620d7cf198bb582 (diff)
parentbb4c7e9a9908548b458f34afb2fee74dc0d49f90 (diff)
downloadlinux-35eb304b5cd7b49d581bda79218b8134f3b689ea.tar.xz
Merge git://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux into rmobile-fixes-for-linus
Diffstat (limited to 'sound/soc')
-rw-r--r--sound/soc/codecs/ak4642.c31
-rw-r--r--sound/soc/codecs/cs42l73.c2
-rw-r--r--sound/soc/codecs/sgtl5000.c19
-rw-r--r--sound/soc/codecs/tlv320aic32x4.c110
-rw-r--r--sound/soc/codecs/wm2000.c31
-rw-r--r--sound/soc/codecs/wm5100.c15
-rw-r--r--sound/soc/codecs/wm8958-dsp2.c2
-rw-r--r--sound/soc/codecs/wm8962.c10
-rw-r--r--sound/soc/codecs/wm8993.c6
-rw-r--r--sound/soc/codecs/wm8994.c16
-rw-r--r--sound/soc/codecs/wm8996.c9
-rw-r--r--sound/soc/codecs/wm8996.h4
-rw-r--r--sound/soc/codecs/wm_hubs.c18
-rw-r--r--sound/soc/ep93xx/ep93xx-pcm.c4
-rw-r--r--sound/soc/imx/imx-pcm-dma-mx2.c18
-rw-r--r--sound/soc/mxs/mxs-pcm.c2
-rw-r--r--sound/soc/mxs/mxs-saif.c5
-rw-r--r--sound/soc/samsung/dma.c4
-rw-r--r--sound/soc/samsung/neo1973_wm8753.c65
-rw-r--r--sound/soc/sh/fsi.c6
-rw-r--r--sound/soc/sh/siu_pcm.c4
-rw-r--r--sound/soc/soc-core.c15
-rw-r--r--sound/soc/soc-dapm.c2
-rw-r--r--sound/soc/txx9/txx9aclc.c2
24 files changed, 195 insertions, 205 deletions
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 5ef70b5d27e4..278c0a0575f5 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -146,13 +146,10 @@ static const struct snd_kcontrol_new ak4642_snd_controls[] = {
SOC_DOUBLE_R_TLV("Digital Playback Volume", L_DVC, R_DVC,
0, 0xFF, 1, out_tlv),
-
- SOC_SINGLE("Headphone Switch", PW_MGMT2, 6, 1, 0),
};
-static const struct snd_kcontrol_new ak4642_hpout_mixer_controls[] = {
- SOC_DAPM_SINGLE("DACH", MD_CTL4, 0, 1, 0),
-};
+static const struct snd_kcontrol_new ak4642_headphone_control =
+ SOC_DAPM_SINGLE("Switch", PW_MGMT2, 6, 1, 0);
static const struct snd_kcontrol_new ak4642_lout_mixer_controls[] = {
SOC_DAPM_SINGLE("DACL", SG_SL1, 4, 1, 0),
@@ -165,13 +162,12 @@ static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = {
SND_SOC_DAPM_OUTPUT("HPOUTR"),
SND_SOC_DAPM_OUTPUT("LINEOUT"),
- SND_SOC_DAPM_MIXER("HPOUTL Mixer", PW_MGMT2, 5, 0,
- &ak4642_hpout_mixer_controls[0],
- ARRAY_SIZE(ak4642_hpout_mixer_controls)),
+ SND_SOC_DAPM_PGA("HPL Out", PW_MGMT2, 5, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("HPR Out", PW_MGMT2, 4, 0, NULL, 0),
+ SND_SOC_DAPM_SWITCH("Headphone Enable", SND_SOC_NOPM, 0, 0,
+ &ak4642_headphone_control),
- SND_SOC_DAPM_MIXER("HPOUTR Mixer", PW_MGMT2, 4, 0,
- &ak4642_hpout_mixer_controls[0],
- ARRAY_SIZE(ak4642_hpout_mixer_controls)),
+ SND_SOC_DAPM_PGA("DACH", MD_CTL4, 0, 0, NULL, 0),
SND_SOC_DAPM_MIXER("LINEOUT Mixer", PW_MGMT1, 3, 0,
&ak4642_lout_mixer_controls[0],
@@ -184,12 +180,17 @@ static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = {
static const struct snd_soc_dapm_route ak4642_intercon[] = {
/* Outputs */
- {"HPOUTL", NULL, "HPOUTL Mixer"},
- {"HPOUTR", NULL, "HPOUTR Mixer"},
+ {"HPOUTL", NULL, "HPL Out"},
+ {"HPOUTR", NULL, "HPR Out"},
{"LINEOUT", NULL, "LINEOUT Mixer"},
- {"HPOUTL Mixer", "DACH", "DAC"},
- {"HPOUTR Mixer", "DACH", "DAC"},
+ {"HPL Out", NULL, "Headphone Enable"},
+ {"HPR Out", NULL, "Headphone Enable"},
+
+ {"Headphone Enable", "Switch", "DACH"},
+
+ {"DACH", NULL, "DAC"},
+
{"LINEOUT Mixer", "DACL", "DAC"},
};
diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c
index 9d38db8f1919..78979b3e0e95 100644
--- a/sound/soc/codecs/cs42l73.c
+++ b/sound/soc/codecs/cs42l73.c
@@ -1113,7 +1113,7 @@ static int cs42l73_pcm_hw_params(struct snd_pcm_substream *substream,
priv->config[id].mmcc &= 0xC0;
priv->config[id].mmcc |= cs42l73_mclk_coeffs[mclk_coeff].mmcc;
priv->config[id].spc &= 0xFC;
- priv->config[id].spc &= MCK_SCLK_64FS;
+ priv->config[id].spc |= MCK_SCLK_MCLK;
} else {
/* CS42L73 Slave */
priv->config[id].spc &= 0xFC;
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index d7bd91831611..7f4ba819a9f6 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -987,12 +987,12 @@ static int sgtl5000_restore_regs(struct snd_soc_codec *codec)
/* restore regular registers */
for (reg = 0; reg <= SGTL5000_CHIP_SHORT_CTRL; reg += 2) {
- /* this regs depends on the others */
+ /* These regs should restore in particular order */
if (reg == SGTL5000_CHIP_ANA_POWER ||
reg == SGTL5000_CHIP_CLK_CTRL ||
reg == SGTL5000_CHIP_LINREG_CTRL ||
reg == SGTL5000_CHIP_LINE_OUT_CTRL ||
- reg == SGTL5000_CHIP_CLK_CTRL)
+ reg == SGTL5000_CHIP_REF_CTRL)
continue;
snd_soc_write(codec, reg, cache[reg]);
@@ -1003,8 +1003,17 @@ static int sgtl5000_restore_regs(struct snd_soc_codec *codec)
snd_soc_write(codec, reg, cache[reg]);
/*
- * restore power and other regs according
- * to set_power() and set_clock()
+ * restore these regs according to the power setting sequence in
+ * sgtl5000_set_power_regs() and clock setting sequence in
+ * sgtl5000_set_clock().
+ *
+ * The order of restore is:
+ * 1. SGTL5000_CHIP_CLK_CTRL MCLK_FREQ bits (1:0) should be restore after
+ * SGTL5000_CHIP_ANA_POWER PLL bits set
+ * 2. SGTL5000_CHIP_LINREG_CTRL should be set before
+ * SGTL5000_CHIP_ANA_POWER LINREG_D restored
+ * 3. SGTL5000_CHIP_REF_CTRL controls Analog Ground Voltage,
+ * prefer to resotre it after SGTL5000_CHIP_ANA_POWER restored
*/
snd_soc_write(codec, SGTL5000_CHIP_LINREG_CTRL,
cache[SGTL5000_CHIP_LINREG_CTRL]);
@@ -1457,5 +1466,5 @@ static void __exit sgtl5000_exit(void)
module_exit(sgtl5000_exit);
MODULE_DESCRIPTION("Freescale SGTL5000 ALSA SoC Codec Driver");
-MODULE_AUTHOR("Zeng Zhaoming <zhaoming.zeng@freescale.com>");
+MODULE_AUTHOR("Zeng Zhaoming <zengzm.kernel@gmail.com>");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
index eb401ef021fb..372b0b83bd9f 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -60,7 +60,6 @@ struct aic32x4_rate_divs {
struct aic32x4_priv {
u32 sysclk;
- s32 master;
u8 page_no;
void *control_data;
u32 power_cfg;
@@ -369,7 +368,6 @@ static int aic32x4_set_dai_sysclk(struct snd_soc_dai *codec_dai,
static int aic32x4_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
- struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec);
u8 iface_reg_1;
u8 iface_reg_2;
u8 iface_reg_3;
@@ -384,11 +382,9 @@ static int aic32x4_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
/* set master/slave audio interface */
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBM_CFM:
- aic32x4->master = 1;
iface_reg_1 |= AIC32X4_BCLKMASTER | AIC32X4_WCLKMASTER;
break;
case SND_SOC_DAIFMT_CBS_CFS:
- aic32x4->master = 0;
break;
default:
printk(KERN_ERR "aic32x4: invalid DAI master/slave interface\n");
@@ -526,64 +522,58 @@ static int aic32x4_mute(struct snd_soc_dai *dai, int mute)
static int aic32x4_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
- struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec);
-
switch (level) {
case SND_SOC_BIAS_ON:
- if (aic32x4->master) {
- /* Switch on PLL */
- snd_soc_update_bits(codec, AIC32X4_PLLPR,
- AIC32X4_PLLEN, AIC32X4_PLLEN);
-
- /* Switch on NDAC Divider */
- snd_soc_update_bits(codec, AIC32X4_NDAC,
- AIC32X4_NDACEN, AIC32X4_NDACEN);
-
- /* Switch on MDAC Divider */
- snd_soc_update_bits(codec, AIC32X4_MDAC,
- AIC32X4_MDACEN, AIC32X4_MDACEN);
-
- /* Switch on NADC Divider */
- snd_soc_update_bits(codec, AIC32X4_NADC,
- AIC32X4_NADCEN, AIC32X4_NADCEN);
-
- /* Switch on MADC Divider */
- snd_soc_update_bits(codec, AIC32X4_MADC,
- AIC32X4_MADCEN, AIC32X4_MADCEN);
-
- /* Switch on BCLK_N Divider */
- snd_soc_update_bits(codec, AIC32X4_BCLKN,
- AIC32X4_BCLKEN, AIC32X4_BCLKEN);
- }
+ /* Switch on PLL */
+ snd_soc_update_bits(codec, AIC32X4_PLLPR,
+ AIC32X4_PLLEN, AIC32X4_PLLEN);
+
+ /* Switch on NDAC Divider */
+ snd_soc_update_bits(codec, AIC32X4_NDAC,
+ AIC32X4_NDACEN, AIC32X4_NDACEN);
+
+ /* Switch on MDAC Divider */
+ snd_soc_update_bits(codec, AIC32X4_MDAC,
+ AIC32X4_MDACEN, AIC32X4_MDACEN);
+
+ /* Switch on NADC Divider */
+ snd_soc_update_bits(codec, AIC32X4_NADC,
+ AIC32X4_NADCEN, AIC32X4_NADCEN);
+
+ /* Switch on MADC Divider */
+ snd_soc_update_bits(codec, AIC32X4_MADC,
+ AIC32X4_MADCEN, AIC32X4_MADCEN);
+
+ /* Switch on BCLK_N Divider */
+ snd_soc_update_bits(codec, AIC32X4_BCLKN,
+ AIC32X4_BCLKEN, AIC32X4_BCLKEN);
break;
case SND_SOC_BIAS_PREPARE:
break;
case SND_SOC_BIAS_STANDBY:
- if (aic32x4->master) {
- /* Switch off PLL */
- snd_soc_update_bits(codec, AIC32X4_PLLPR,
- AIC32X4_PLLEN, 0);
-
- /* Switch off NDAC Divider */
- snd_soc_update_bits(codec, AIC32X4_NDAC,
- AIC32X4_NDACEN, 0);
-
- /* Switch off MDAC Divider */
- snd_soc_update_bits(codec, AIC32X4_MDAC,
- AIC32X4_MDACEN, 0);
-
- /* Switch off NADC Divider */
- snd_soc_update_bits(codec, AIC32X4_NADC,
- AIC32X4_NADCEN, 0);
-
- /* Switch off MADC Divider */
- snd_soc_update_bits(codec, AIC32X4_MADC,
- AIC32X4_MADCEN, 0);
-
- /* Switch off BCLK_N Divider */
- snd_soc_update_bits(codec, AIC32X4_BCLKN,
- AIC32X4_BCLKEN, 0);
- }
+ /* Switch off PLL */
+ snd_soc_update_bits(codec, AIC32X4_PLLPR,
+ AIC32X4_PLLEN, 0);
+
+ /* Switch off NDAC Divider */
+ snd_soc_update_bits(codec, AIC32X4_NDAC,
+ AIC32X4_NDACEN, 0);
+
+ /* Switch off MDAC Divider */
+ snd_soc_update_bits(codec, AIC32X4_MDAC,
+ AIC32X4_MDACEN, 0);
+
+ /* Switch off NADC Divider */
+ snd_soc_update_bits(codec, AIC32X4_NADC,
+ AIC32X4_NADCEN, 0);
+
+ /* Switch off MADC Divider */
+ snd_soc_update_bits(codec, AIC32X4_MADC,
+ AIC32X4_MADCEN, 0);
+
+ /* Switch off BCLK_N Divider */
+ snd_soc_update_bits(codec, AIC32X4_BCLKN,
+ AIC32X4_BCLKEN, 0);
break;
case SND_SOC_BIAS_OFF:
break;
@@ -651,9 +641,11 @@ static int aic32x4_probe(struct snd_soc_codec *codec)
if (aic32x4->power_cfg & AIC32X4_PWR_AVDD_DVDD_WEAK_DISABLE) {
snd_soc_write(codec, AIC32X4_PWRCFG, AIC32X4_AVDDWEAKDISABLE);
}
- if (aic32x4->power_cfg & AIC32X4_PWR_AIC32X4_LDO_ENABLE) {
- snd_soc_write(codec, AIC32X4_LDOCTL, AIC32X4_LDOCTLEN);
- }
+
+ tmp_reg = (aic32x4->power_cfg & AIC32X4_PWR_AIC32X4_LDO_ENABLE) ?
+ AIC32X4_LDOCTLEN : 0;
+ snd_soc_write(codec, AIC32X4_LDOCTL, tmp_reg);
+
tmp_reg = snd_soc_read(codec, AIC32X4_CMMODE);
if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_LDOIN_RANGE_18_36) {
tmp_reg |= AIC32X4_LDOIN_18_36;
diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c
index c2880907fced..a75c3766aede 100644
--- a/sound/soc/codecs/wm2000.c
+++ b/sound/soc/codecs/wm2000.c
@@ -733,8 +733,9 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c,
struct wm2000_priv *wm2000;
struct wm2000_platform_data *pdata;
const char *filename;
- const struct firmware *fw;
- int reg, ret;
+ const struct firmware *fw = NULL;
+ int ret;
+ int reg;
u16 id;
wm2000 = devm_kzalloc(&i2c->dev, sizeof(struct wm2000_priv),
@@ -751,7 +752,7 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c,
ret = PTR_ERR(wm2000->regmap);
dev_err(&i2c->dev, "Failed to allocate register map: %d\n",
ret);
- goto err;
+ goto out;
}
/* Verify that this is a WM2000 */
@@ -763,7 +764,7 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c,
if (id != 0x2000) {
dev_err(&i2c->dev, "Device is not a WM2000 - ID %x\n", id);
ret = -ENODEV;
- goto err_regmap;
+ goto out_regmap_exit;
}
reg = wm2000_read(i2c, WM2000_REG_REVISON);
@@ -782,7 +783,7 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c,
ret = request_firmware(&fw, filename, &i2c->dev);
if (ret != 0) {
dev_err(&i2c->dev, "Failed to acquire ANC data: %d\n", ret);
- goto err_regmap;
+ goto out_regmap_exit;
}
/* Pre-cook the concatenation of the register address onto the image */
@@ -793,15 +794,13 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c,
if (wm2000->anc_download == NULL) {
dev_err(&i2c->dev, "Out of memory\n");
ret = -ENOMEM;
- goto err_fw;
+ goto out_regmap_exit;
}
wm2000->anc_download[0] = 0x80;
wm2000->anc_download[1] = 0x00;
memcpy(wm2000->anc_download + 2, fw->data, fw->size);
- release_firmware(fw);
-
wm2000->anc_eng_ena = 1;
wm2000->anc_active = 1;
wm2000->spk_ena = 1;
@@ -809,18 +808,14 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c,
wm2000_reset(wm2000);
- ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm2000,
- NULL, 0);
- if (ret != 0)
- goto err_fw;
+ ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm2000, NULL, 0);
+ if (!ret)
+ goto out;
- return 0;
-
-err_fw:
- release_firmware(fw);
-err_regmap:
+out_regmap_exit:
regmap_exit(wm2000->regmap);
-err:
+out:
+ release_firmware(fw);
return ret;
}
diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c
index 8b24323d6b2c..89f2af77b1c3 100644
--- a/sound/soc/codecs/wm5100.c
+++ b/sound/soc/codecs/wm5100.c
@@ -1377,6 +1377,7 @@ static int wm5100_set_bias_level(struct snd_soc_codec *codec,
switch (wm5100->rev) {
case 0:
+ regcache_cache_bypass(wm5100->regmap, true);
snd_soc_write(codec, 0x11, 0x3);
snd_soc_write(codec, 0x203, 0xc);
snd_soc_write(codec, 0x206, 0);
@@ -1392,6 +1393,7 @@ static int wm5100_set_bias_level(struct snd_soc_codec *codec,
snd_soc_write(codec,
wm5100_reva_patches[i].reg,
wm5100_reva_patches[i].val);
+ regcache_cache_bypass(wm5100->regmap, false);
break;
default:
break;
@@ -1402,6 +1404,8 @@ static int wm5100_set_bias_level(struct snd_soc_codec *codec,
break;
case SND_SOC_BIAS_OFF:
+ regcache_cache_only(wm5100->regmap, true);
+ regcache_mark_dirty(wm5100->regmap);
if (wm5100->pdata.ldo_ena)
gpio_set_value_cansleep(wm5100->pdata.ldo_ena, 0);
regulator_bulk_disable(ARRAY_SIZE(wm5100->core_supplies),
@@ -2180,6 +2184,7 @@ static void wm5100_micd_irq(struct snd_soc_codec *codec)
if (wm5100->jack_detecting) {
dev_dbg(codec->dev, "Microphone detected\n");
wm5100->jack_mic = true;
+ wm5100->jack_detecting = false;
snd_soc_jack_report(wm5100->jack,
SND_JACK_HEADSET,
SND_JACK_HEADSET | SND_JACK_BTN_0);
@@ -2218,6 +2223,7 @@ static void wm5100_micd_irq(struct snd_soc_codec *codec)
SND_JACK_BTN_0);
} else if (wm5100->jack_detecting) {
dev_dbg(codec->dev, "Headphone detected\n");
+ wm5100->jack_detecting = false;
snd_soc_jack_report(wm5100->jack, SND_JACK_HEADPHONE,
SND_JACK_HEADPHONE);
@@ -2607,6 +2613,13 @@ static const struct regmap_config wm5100_regmap = {
.cache_type = REGCACHE_RBTREE,
};
+static const unsigned int wm5100_mic_ctrl_reg[] = {
+ WM5100_IN1L_CONTROL,
+ WM5100_IN2L_CONTROL,
+ WM5100_IN3L_CONTROL,
+ WM5100_IN4L_CONTROL,
+};
+
static __devinit int wm5100_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
@@ -2739,7 +2752,7 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c,
}
for (i = 0; i < ARRAY_SIZE(wm5100->pdata.in_mode); i++) {
- regmap_update_bits(wm5100->regmap, WM5100_IN1L_CONTROL,
+ regmap_update_bits(wm5100->regmap, wm5100_mic_ctrl_reg[i],
WM5100_IN1_MODE_MASK |
WM5100_IN1_DMIC_SUP_MASK,
(wm5100->pdata.in_mode[i] <<
diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c
index 8d4ea43d40a3..40ac888faf3d 100644
--- a/sound/soc/codecs/wm8958-dsp2.c
+++ b/sound/soc/codecs/wm8958-dsp2.c
@@ -55,7 +55,7 @@ static int wm8958_dsp2_fw(struct snd_soc_codec *codec, const char *name,
return 0;
if (fw->size < 32) {
- dev_err(codec->dev, "%s: firmware too short (%d bytes)\n",
+ dev_err(codec->dev, "%s: firmware too short (%zd bytes)\n",
name, fw->size);
goto err;
}
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 296de4e30d26..0ac228b7dc04 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -96,7 +96,7 @@ static int wm8962_regulator_event_##n(struct notifier_block *nb, \
struct wm8962_priv *wm8962 = container_of(nb, struct wm8962_priv, \
disable_nb[n]); \
if (event & REGULATOR_EVENT_DISABLE) { \
- regcache_cache_only(wm8962->regmap, true); \
+ regcache_mark_dirty(wm8962->regmap); \
} \
return 0; \
}
@@ -2564,7 +2564,7 @@ static int dsp2_event(struct snd_soc_dapm_widget *w,
return 0;
}
-static const char *st_text[] = { "None", "Right", "Left" };
+static const char *st_text[] = { "None", "Left", "Right" };
static const struct soc_enum str_enum =
SOC_ENUM_SINGLE(WM8962_DAC_DSP_MIXING_1, 2, 3, st_text);
@@ -3159,13 +3159,13 @@ static int wm8962_hw_params(struct snd_pcm_substream *substream,
case SNDRV_PCM_FORMAT_S16_LE:
break;
case SNDRV_PCM_FORMAT_S20_3LE:
- aif0 |= 0x40;
+ aif0 |= 0x4;
break;
case SNDRV_PCM_FORMAT_S24_LE:
- aif0 |= 0x80;
+ aif0 |= 0x8;
break;
case SNDRV_PCM_FORMAT_S32_LE:
- aif0 |= 0xc0;
+ aif0 |= 0xc;
break;
default:
return -EINVAL;
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index 2b40c93601ed..7c7fd925db8d 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -444,6 +444,12 @@ static int _wm8993_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
/* Enable the FLL */
snd_soc_write(codec, WM8993_FLL_CONTROL_1, reg1 | WM8993_FLL_ENA);
+ /* Both overestimates */
+ if (Fref < 1000000)
+ msleep(3);
+ else
+ msleep(1);
+
dev_dbg(codec->dev, "FLL enabled at %dHz->%dHz\n", Fref, Fout);
wm8993->fll_fref = Fref;
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 93d27b660257..ec69a6c152fe 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -770,6 +770,8 @@ static void vmid_reference(struct snd_soc_codec *codec)
{
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
+ pm_runtime_get_sync(codec->dev);
+
wm8994->vmid_refcount++;
dev_dbg(codec->dev, "Referencing VMID, refcount is now %d\n",
@@ -783,7 +785,12 @@ static void vmid_reference(struct snd_soc_codec *codec)
WM8994_VMID_RAMP_MASK,
WM8994_STARTUP_BIAS_ENA |
WM8994_VMID_BUF_ENA |
- (0x11 << WM8994_VMID_RAMP_SHIFT));
+ (0x3 << WM8994_VMID_RAMP_SHIFT));
+
+ /* Remove discharge for line out */
+ snd_soc_update_bits(codec, WM8994_ANTIPOP_1,
+ WM8994_LINEOUT1_DISCH |
+ WM8994_LINEOUT2_DISCH, 0);
/* Main bias enable, VMID=2x40k */
snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1,
@@ -837,6 +844,8 @@ static void vmid_dereference(struct snd_soc_codec *codec)
WM8994_VMID_BUF_ENA |
WM8994_VMID_RAMP_MASK, 0);
}
+
+ pm_runtime_put(codec->dev);
}
static int vmid_event(struct snd_soc_dapm_widget *w,
@@ -2753,11 +2762,6 @@ static int wm8994_resume(struct snd_soc_codec *codec)
codec->cache_only = 0;
}
- /* Restore the registers */
- ret = snd_soc_cache_sync(codec);
- if (ret != 0)
- dev_err(codec->dev, "Failed to sync cache: %d\n", ret);
-
wm8994_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
for (i = 0; i < ARRAY_SIZE(wm8994->fll); i++) {
diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c
index d8da10fe5b52..61f7daa4d0e6 100644
--- a/sound/soc/codecs/wm8996.c
+++ b/sound/soc/codecs/wm8996.c
@@ -108,7 +108,7 @@ static int wm8996_regulator_event_##n(struct notifier_block *nb, \
struct wm8996_priv *wm8996 = container_of(nb, struct wm8996_priv, \
disable_nb[n]); \
if (event & REGULATOR_EVENT_DISABLE) { \
- regcache_cache_only(wm8996->regmap, true); \
+ regcache_mark_dirty(wm8996->regmap); \
} \
return 0; \
}
@@ -1120,7 +1120,8 @@ SND_SOC_DAPM_SUPPLY_S("SYSCLK", 1, WM8996_AIF_CLOCKING_1, 0, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY_S("SYSDSPCLK", 2, WM8996_CLOCKING_1, 1, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY_S("AIFCLK", 2, WM8996_CLOCKING_1, 2, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY_S("Charge Pump", 2, WM8996_CHARGE_PUMP_1, 15, 0, cp_event,
- SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+ SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_SUPPLY("Bandgap", SND_SOC_NOPM, 0, 0, bg_event,
SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_SUPPLY("LDO2", WM8996_POWER_MANAGEMENT_2, 1, 0, NULL, 0),
@@ -2007,6 +2008,7 @@ static int wm8996_set_sysclk(struct snd_soc_dai *dai,
struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec);
int lfclk = 0;
int ratediv = 0;
+ int sync = WM8996_REG_SYNC;
int src;
int old;
@@ -2051,6 +2053,7 @@ static int wm8996_set_sysclk(struct snd_soc_dai *dai,
case 32000:
case 32768:
lfclk = WM8996_LFCLK_ENA;
+ sync = 0;
break;
default:
dev_warn(codec->dev, "Unsupported clock rate %dHz\n",
@@ -2064,6 +2067,8 @@ static int wm8996_set_sysclk(struct snd_soc_dai *dai,
WM8996_SYSCLK_SRC_MASK | WM8996_SYSCLK_DIV_MASK,
src << WM8996_SYSCLK_SRC_SHIFT | ratediv);
snd_soc_update_bits(codec, WM8996_CLOCKING_1, WM8996_LFCLK_ENA, lfclk);
+ snd_soc_update_bits(codec, WM8996_CONTROL_INTERFACE_1,
+ WM8996_REG_SYNC, sync);
snd_soc_update_bits(codec, WM8996_AIF_CLOCKING_1,
WM8996_SYSCLK_ENA, old);
diff --git a/sound/soc/codecs/wm8996.h b/sound/soc/codecs/wm8996.h
index 0fde643194ce..de9ac3e44aec 100644
--- a/sound/soc/codecs/wm8996.h
+++ b/sound/soc/codecs/wm8996.h
@@ -1567,6 +1567,10 @@ int wm8996_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack,
/*
* R257 (0x101) - Control Interface (1)
*/
+#define WM8996_REG_SYNC 0x8000 /* REG_SYNC */
+#define WM8996_REG_SYNC_MASK 0x8000 /* REG_SYNC */
+#define WM8996_REG_SYNC_SHIFT 15 /* REG_SYNC */
+#define WM8996_REG_SYNC_WIDTH 1 /* REG_SYNC */
#define WM8996_AUTO_INC 0x0004 /* AUTO_INC */
#define WM8996_AUTO_INC_MASK 0x0004 /* AUTO_INC */
#define WM8996_AUTO_INC_SHIFT 2 /* AUTO_INC */
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index 2a61094075f8..8a68cea4a3ee 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -586,14 +586,14 @@ SOC_DAPM_SINGLE("Left Output Switch", WM8993_LINE_MIXER1, 0, 1, 0),
};
static const struct snd_kcontrol_new line2_mix[] = {
-SOC_DAPM_SINGLE("IN2R Switch", WM8993_LINE_MIXER2, 2, 1, 0),
-SOC_DAPM_SINGLE("IN2L Switch", WM8993_LINE_MIXER2, 1, 1, 0),
+SOC_DAPM_SINGLE("IN1L Switch", WM8993_LINE_MIXER2, 2, 1, 0),
+SOC_DAPM_SINGLE("IN1R Switch", WM8993_LINE_MIXER2, 1, 1, 0),
SOC_DAPM_SINGLE("Output Switch", WM8993_LINE_MIXER2, 0, 1, 0),
};
static const struct snd_kcontrol_new line2n_mix[] = {
-SOC_DAPM_SINGLE("Left Output Switch", WM8993_LINE_MIXER2, 6, 1, 0),
-SOC_DAPM_SINGLE("Right Output Switch", WM8993_LINE_MIXER2, 5, 1, 0),
+SOC_DAPM_SINGLE("Left Output Switch", WM8993_LINE_MIXER2, 5, 1, 0),
+SOC_DAPM_SINGLE("Right Output Switch", WM8993_LINE_MIXER2, 6, 1, 0),
};
static const struct snd_kcontrol_new line2p_mix[] = {
@@ -613,6 +613,8 @@ SND_SOC_DAPM_INPUT("IN2RP:VXRP"),
SND_SOC_DAPM_SUPPLY("MICBIAS2", WM8993_POWER_MANAGEMENT_1, 5, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("MICBIAS1", WM8993_POWER_MANAGEMENT_1, 4, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("LINEOUT_VMID_BUF", WM8993_ANTIPOP1, 7, 0, NULL, 0),
+
SND_SOC_DAPM_MIXER("IN1L PGA", WM8993_POWER_MANAGEMENT_2, 6, 0,
in1l_pga, ARRAY_SIZE(in1l_pga)),
SND_SOC_DAPM_MIXER("IN1R PGA", WM8993_POWER_MANAGEMENT_2, 4, 0,
@@ -834,9 +836,11 @@ static const struct snd_soc_dapm_route lineout1_diff_routes[] = {
};
static const struct snd_soc_dapm_route lineout1_se_routes[] = {
+ { "LINEOUT1N Mixer", NULL, "LINEOUT_VMID_BUF" },
{ "LINEOUT1N Mixer", "Left Output Switch", "Left Output PGA" },
{ "LINEOUT1N Mixer", "Right Output Switch", "Right Output PGA" },
+ { "LINEOUT1P Mixer", NULL, "LINEOUT_VMID_BUF" },
{ "LINEOUT1P Mixer", "Left Output Switch", "Left Output PGA" },
{ "LINEOUT1N Driver", NULL, "LINEOUT1N Mixer" },
@@ -844,8 +848,8 @@ static const struct snd_soc_dapm_route lineout1_se_routes[] = {
};
static const struct snd_soc_dapm_route lineout2_diff_routes[] = {
- { "LINEOUT2 Mixer", "IN2L Switch", "IN2L PGA" },
- { "LINEOUT2 Mixer", "IN2R Switch", "IN2R PGA" },
+ { "LINEOUT2 Mixer", "IN1L Switch", "IN1L PGA" },
+ { "LINEOUT2 Mixer", "IN1R Switch", "IN1R PGA" },
{ "LINEOUT2 Mixer", "Output Switch", "Right Output PGA" },
{ "LINEOUT2N Driver", NULL, "LINEOUT2 Mixer" },
@@ -853,9 +857,11 @@ static const struct snd_soc_dapm_route lineout2_diff_routes[] = {
};
static const struct snd_soc_dapm_route lineout2_se_routes[] = {
+ { "LINEOUT2N Mixer", NULL, "LINEOUT_VMID_BUF" },
{ "LINEOUT2N Mixer", "Left Output Switch", "Left Output PGA" },
{ "LINEOUT2N Mixer", "Right Output Switch", "Right Output PGA" },
+ { "LINEOUT2P Mixer", NULL, "LINEOUT_VMID_BUF" },
{ "LINEOUT2P Mixer", "Right Output Switch", "Right Output PGA" },
{ "LINEOUT2N Driver", NULL, "LINEOUT2N Mixer" },
diff --git a/sound/soc/ep93xx/ep93xx-pcm.c b/sound/soc/ep93xx/ep93xx-pcm.c
index 3fc96130d1a6..de8390449873 100644
--- a/sound/soc/ep93xx/ep93xx-pcm.c
+++ b/sound/soc/ep93xx/ep93xx-pcm.c
@@ -113,9 +113,9 @@ static int ep93xx_pcm_open(struct snd_pcm_substream *substream)
rtd->dma_data.name = dma_params->name;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- rtd->dma_data.direction = DMA_TO_DEVICE;
+ rtd->dma_data.direction = DMA_MEM_TO_DEV;
else
- rtd->dma_data.direction = DMA_FROM_DEVICE;
+ rtd->dma_data.direction = DMA_DEV_TO_MEM;
rtd->dma_chan = dma_request_channel(mask, ep93xx_pcm_dma_filter,
&rtd->dma_data);
diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c
index 1cf2fe889f6a..5780c9b9d569 100644
--- a/sound/soc/imx/imx-pcm-dma-mx2.c
+++ b/sound/soc/imx/imx-pcm-dma-mx2.c
@@ -88,11 +88,13 @@ static int imx_ssi_dma_alloc(struct snd_pcm_substream *substream,
iprtd->dma_data.dma_request = dma_params->dma;
/* Try to grab a DMA channel */
- dma_cap_zero(mask);
- dma_cap_set(DMA_SLAVE, mask);
- iprtd->dma_chan = dma_request_channel(mask, filter, iprtd);
- if (!iprtd->dma_chan)
- return -EINVAL;
+ if (!iprtd->dma_chan) {
+ dma_cap_zero(mask);
+ dma_cap_set(DMA_SLAVE, mask);
+ iprtd->dma_chan = dma_request_channel(mask, filter, iprtd);
+ if (!iprtd->dma_chan)
+ return -EINVAL;
+ }
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
@@ -107,12 +109,12 @@ static int imx_ssi_dma_alloc(struct snd_pcm_substream *substream,
}
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- slave_config.direction = DMA_TO_DEVICE;
+ slave_config.direction = DMA_MEM_TO_DEV;
slave_config.dst_addr = dma_params->dma_addr;
slave_config.dst_addr_width = buswidth;
slave_config.dst_maxburst = dma_params->burstsize;
} else {
- slave_config.direction = DMA_FROM_DEVICE;
+ slave_config.direction = DMA_DEV_TO_MEM;
slave_config.src_addr = dma_params->dma_addr;
slave_config.src_addr_width = buswidth;
slave_config.src_maxburst = dma_params->burstsize;
@@ -159,7 +161,7 @@ static int snd_imx_pcm_hw_params(struct snd_pcm_substream *substream,
iprtd->period_bytes * iprtd->periods,
iprtd->period_bytes,
substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
- DMA_TO_DEVICE : DMA_FROM_DEVICE);
+ DMA_MEM_TO_DEV : DMA_DEV_TO_MEM);
if (!iprtd->desc) {
dev_err(&chan->dev->device, "cannot prepare slave dma\n");
return -EINVAL;
diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c
index 0e12f4e0a76d..105f42a394df 100644
--- a/sound/soc/mxs/mxs-pcm.c
+++ b/sound/soc/mxs/mxs-pcm.c
@@ -136,7 +136,7 @@ static int snd_mxs_pcm_hw_params(struct snd_pcm_substream *substream,
iprtd->period_bytes * iprtd->periods,
iprtd->period_bytes,
substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
- DMA_TO_DEVICE : DMA_FROM_DEVICE);
+ DMA_MEM_TO_DEV : DMA_DEV_TO_MEM);
if (!iprtd->desc) {
dev_err(&chan->dev->device, "cannot prepare slave dma\n");
return -EINVAL;
diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c
index dccfb37a9626..f204dbac11d4 100644
--- a/sound/soc/mxs/mxs-saif.c
+++ b/sound/soc/mxs/mxs-saif.c
@@ -124,6 +124,8 @@ static int mxs_saif_set_clk(struct mxs_saif *saif,
*
* If MCLK is not used, we just set saif clk to 512*fs.
*/
+ clk_prepare_enable(master_saif->clk);
+
if (master_saif->mclk_in_use) {
if (mclk % 32 == 0) {
scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE;
@@ -133,6 +135,7 @@ static int mxs_saif_set_clk(struct mxs_saif *saif,
ret = clk_set_rate(master_saif->clk, 384 * rate);
} else {
/* SAIF MCLK should be either 32x or 48x */
+ clk_disable_unprepare(master_saif->clk);
return -EINVAL;
}
} else {
@@ -140,6 +143,8 @@ static int mxs_saif_set_clk(struct mxs_saif *saif,
scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE;
}
+ clk_disable_unprepare(master_saif->clk);
+
if (ret)
return ret;
diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c
index 427ae0d9817b..e4ba17ce6b32 100644
--- a/sound/soc/samsung/dma.c
+++ b/sound/soc/samsung/dma.c
@@ -86,7 +86,7 @@ static void dma_enqueue(struct snd_pcm_substream *substream)
dma_info.cap = (samsung_dma_has_circular() ? DMA_CYCLIC : DMA_SLAVE);
dma_info.direction =
(substream->stream == SNDRV_PCM_STREAM_PLAYBACK
- ? DMA_TO_DEVICE : DMA_FROM_DEVICE);
+ ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM);
dma_info.fp = audio_buffdone;
dma_info.fp_param = substream;
dma_info.period = prtd->dma_period;
@@ -171,7 +171,7 @@ static int dma_hw_params(struct snd_pcm_substream *substream,
dma_info.client = prtd->params->client;
dma_info.direction =
(substream->stream == SNDRV_PCM_STREAM_PLAYBACK
- ? DMA_TO_DEVICE : DMA_FROM_DEVICE);
+ ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM);
dma_info.width = prtd->params->dma_size;
dma_info.fifo = prtd->params->dma_addr;
prtd->params->ch = prtd->params->ops->request(
diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c
index 7ac0ba2025c3..c6012ff5bd3e 100644
--- a/sound/soc/samsung/neo1973_wm8753.c
+++ b/sound/soc/samsung/neo1973_wm8753.c
@@ -230,8 +230,6 @@ static const struct snd_kcontrol_new neo1973_wm8753_controls[] = {
/* GTA02 specific routes and controls */
-#ifdef CONFIG_MACH_NEO1973_GTA02
-
static int gta02_speaker_enabled;
static int lm4853_set_spk(struct snd_kcontrol *kcontrol,
@@ -311,10 +309,6 @@ static int neo1973_gta02_wm8753_init(struct snd_soc_codec *codec)
return 0;
}
-#else
-static int neo1973_gta02_wm8753_init(struct snd_soc_code *codec) { return 0; }
-#endif
-
static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
@@ -322,10 +316,6 @@ static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd)
int ret;
/* set up NC codec pins */
- if (machine_is_neo1973_gta01()) {
- snd_soc_dapm_nc_pin(dapm, "LOUT2");
- snd_soc_dapm_nc_pin(dapm, "ROUT2");
- }
snd_soc_dapm_nc_pin(dapm, "OUT3");
snd_soc_dapm_nc_pin(dapm, "OUT4");
snd_soc_dapm_nc_pin(dapm, "LINE1");
@@ -370,50 +360,6 @@ static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd)
return 0;
}
-/* GTA01 specific controls */
-
-#ifdef CONFIG_MACH_NEO1973_GTA01
-
-static const struct snd_soc_dapm_route neo1973_lm4857_routes[] = {
- {"Amp IN", NULL, "ROUT1"},
- {"Amp IN", NULL, "LOUT1"},
-
- {"Handset Spk", NULL, "Amp EP"},
- {"Stereo Out", NULL, "Amp LS"},
- {"Headphone", NULL, "Amp HP"},
-};
-
-static const struct snd_soc_dapm_widget neo1973_lm4857_dapm_widgets[] = {
- SND_SOC_DAPM_SPK("Handset Spk", NULL),
- SND_SOC_DAPM_SPK("Stereo Out", NULL),
- SND_SOC_DAPM_HP("Headphone", NULL),
-};
-
-static int neo1973_lm4857_init(struct snd_soc_dapm_context *dapm)
-{
- int ret;
-
- ret = snd_soc_dapm_new_controls(dapm, neo1973_lm4857_dapm_widgets,
- ARRAY_SIZE(neo1973_lm4857_dapm_widgets));
- if (ret)
- return ret;
-
- ret = snd_soc_dapm_add_routes(dapm, neo1973_lm4857_routes,
- ARRAY_SIZE(neo1973_lm4857_routes));
- if (ret)
- return ret;
-
- snd_soc_dapm_ignore_suspend(dapm, "Stereo Out");
- snd_soc_dapm_ignore_suspend(dapm, "Handset Spk");
- snd_soc_dapm_ignore_suspend(dapm, "Headphone");
-
- return 0;
-}
-
-#else
-static int neo1973_lm4857_init(struct snd_soc_dapm_context *dapm) { return 0; };
-#endif
-
static struct snd_soc_dai_link neo1973_dai[] = {
{ /* Hifi Playback - for similatious use with voice below */
.name = "WM8753",
@@ -440,11 +386,6 @@ static struct snd_soc_aux_dev neo1973_aux_devs[] = {
.name = "dfbmcs320",
.codec_name = "dfbmcs320.0",
},
- {
- .name = "lm4857",
- .codec_name = "lm4857.0-007c",
- .init = neo1973_lm4857_init,
- },
};
static struct snd_soc_codec_conf neo1973_codec_conf[] = {
@@ -454,14 +395,10 @@ static struct snd_soc_codec_conf neo1973_codec_conf[] = {
},
};
-#ifdef CONFIG_MACH_NEO1973_GTA02
static const struct gpio neo1973_gta02_gpios[] = {
{ GTA02_GPIO_HP_IN, GPIOF_OUT_INIT_HIGH, "GTA02_HP_IN" },
{ GTA02_GPIO_AMP_SHUT, GPIOF_OUT_INIT_HIGH, "GTA02_AMP_SHUT" },
};
-#else
-static const struct gpio neo1973_gta02_gpios[] = {};
-#endif
static struct snd_soc_card neo1973 = {
.name = "neo1973",
@@ -480,7 +417,7 @@ static int __init neo1973_init(void)
{
int ret;
- if (!machine_is_neo1973_gta01() && !machine_is_neo1973_gta02())
+ if (!machine_is_neo1973_gta02())
return -ENODEV;
if (machine_is_neo1973_gta02()) {
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index db6c89a28bda..ea4a82d01160 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -1152,12 +1152,8 @@ static snd_pcm_uframes_t fsi_pointer(struct snd_pcm_substream *substream)
{
struct fsi_priv *fsi = fsi_get_priv(substream);
struct fsi_stream *io = fsi_get_stream(fsi, fsi_is_play(substream));
- int samples_pos = io->buff_sample_pos - 1;
- if (samples_pos < 0)
- samples_pos = 0;
-
- return fsi_sample2frame(fsi, samples_pos);
+ return fsi_sample2frame(fsi, io->buff_sample_pos);
}
static struct snd_pcm_ops fsi_pcm_ops = {
diff --git a/sound/soc/sh/siu_pcm.c b/sound/soc/sh/siu_pcm.c
index f8f681690a71..0193e595d415 100644
--- a/sound/soc/sh/siu_pcm.c
+++ b/sound/soc/sh/siu_pcm.c
@@ -131,7 +131,7 @@ static int siu_pcm_wr_set(struct siu_port *port_info,
sg_dma_address(&sg) = buff;
desc = siu_stream->chan->device->device_prep_slave_sg(siu_stream->chan,
- &sg, 1, DMA_TO_DEVICE, DMA_PREP_INTERRUPT | DMA_CTRL_ACK);
+ &sg, 1, DMA_MEM_TO_DEV, DMA_PREP_INTERRUPT | DMA_CTRL_ACK);
if (!desc) {
dev_err(dev, "Failed to allocate a dma descriptor\n");
return -ENOMEM;
@@ -181,7 +181,7 @@ static int siu_pcm_rd_set(struct siu_port *port_info,
sg_dma_address(&sg) = buff;
desc = siu_stream->chan->device->device_prep_slave_sg(siu_stream->chan,
- &sg, 1, DMA_FROM_DEVICE, DMA_PREP_INTERRUPT | DMA_CTRL_ACK);
+ &sg, 1, DMA_DEV_TO_MEM, DMA_PREP_INTERRUPT | DMA_CTRL_ACK);
if (!desc) {
dev_err(dev, "Failed to allocate dma descriptor\n");
return -ENOMEM;
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 3986520b4677..92cee24ed2dc 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -567,6 +567,17 @@ int snd_soc_suspend(struct device *dev)
if (!codec->suspended && codec->driver->suspend) {
switch (codec->dapm.bias_level) {
case SND_SOC_BIAS_STANDBY:
+ /*
+ * If the CODEC is capable of idle
+ * bias off then being in STANDBY
+ * means it's doing something,
+ * otherwise fall through.
+ */
+ if (codec->dapm.idle_bias_off) {
+ dev_dbg(codec->dev,
+ "idle_bias_off CODEC on over suspend\n");
+ break;
+ }
case SND_SOC_BIAS_OFF:
codec->driver->suspend(codec);
codec->suspended = 1;
@@ -907,6 +918,10 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num, int order)
if (err < 0)
printk(KERN_ERR "asoc: failed to remove %s\n", platform->name);
}
+
+ /* Make sure all DAPM widgets are freed */
+ snd_soc_dapm_free(&platform->dapm);
+
platform->probed = 0;
list_del(&platform->card_list);
module_put(platform->dev->driver->owner);
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 3ad1f59b8028..1f55ded4047f 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -1426,7 +1426,7 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event)
dapm->target_bias_level = SND_SOC_BIAS_ON;
break;
case SND_SOC_DAPM_STREAM_STOP:
- if (dapm->codec->active)
+ if (dapm->codec && dapm->codec->active)
dapm->target_bias_level = SND_SOC_BIAS_ON;
else
dapm->target_bias_level = SND_SOC_BIAS_STANDBY;
diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c
index 93931def0dce..21554611557c 100644
--- a/sound/soc/txx9/txx9aclc.c
+++ b/sound/soc/txx9/txx9aclc.c
@@ -134,7 +134,7 @@ txx9aclc_dma_submit(struct txx9aclc_dmadata *dmadata, dma_addr_t buf_dma_addr)
sg_dma_address(&sg) = buf_dma_addr;
desc = chan->device->device_prep_slave_sg(chan, &sg, 1,
dmadata->substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
- DMA_TO_DEVICE : DMA_FROM_DEVICE,
+ DMA_MEM_TO_DEV : DMA_DEV_TO_MEM,
DMA_PREP_INTERRUPT | DMA_CTRL_ACK);
if (!desc) {
dev_err(&chan->dev->device, "cannot prepare slave dma\n");