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-rw-r--r--sound/soc/atmel/atmel-pcm-dma.c2
-rw-r--r--sound/soc/blackfin/bf5xx-i2s.c1
-rw-r--r--sound/soc/codecs/88pm860x-codec.c3
-rw-r--r--sound/soc/codecs/ab8500-codec.c7
-rw-r--r--sound/soc/codecs/ak4642.c2
-rw-r--r--sound/soc/codecs/arizona.c4
-rw-r--r--sound/soc/codecs/cs42l52.c2
-rw-r--r--sound/soc/codecs/cs42l52.h2
-rw-r--r--sound/soc/codecs/max98088.c2
-rw-r--r--sound/soc/codecs/max98095.c4
-rw-r--r--sound/soc/codecs/mc13783.c4
-rw-r--r--sound/soc/codecs/sgtl5000.c2
-rw-r--r--sound/soc/codecs/sgtl5000.h2
-rw-r--r--sound/soc/codecs/wm5110.c43
-rw-r--r--sound/soc/codecs/wm8731.c4
-rw-r--r--sound/soc/codecs/wm8960.c6
-rw-r--r--sound/soc/codecs/wm8962.c26
-rw-r--r--sound/soc/codecs/wm8990.c2
-rw-r--r--sound/soc/codecs/wm_hubs.c1
-rw-r--r--sound/soc/fsl/imx-pcm-fiq.c29
-rw-r--r--sound/soc/s6000/s6000-pcm.c2
-rw-r--r--sound/soc/soc-dapm.c9
-rw-r--r--sound/soc/tegra/tegra20_ac97.c6
-rw-r--r--sound/soc/tegra/tegra20_spdif.c4
-rw-r--r--sound/soc/tegra/tegra30_i2s.c2
25 files changed, 123 insertions, 48 deletions
diff --git a/sound/soc/atmel/atmel-pcm-dma.c b/sound/soc/atmel/atmel-pcm-dma.c
index 1d38fd0bc4e..d1282652679 100644
--- a/sound/soc/atmel/atmel-pcm-dma.c
+++ b/sound/soc/atmel/atmel-pcm-dma.c
@@ -81,7 +81,9 @@ static void atmel_pcm_dma_irq(u32 ssc_sr,
/* stop RX and capture: will be enabled again at restart */
ssc_writex(prtd->ssc->regs, SSC_CR, prtd->mask->ssc_disable);
+ snd_pcm_stream_lock(substream);
snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
+ snd_pcm_stream_unlock(substream);
/* now drain RHR and read status to remove xrun condition */
ssc_readx(prtd->ssc->regs, SSC_RHR);
diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c
index dd0c2a4f83a..e0869aaa1e9 100644
--- a/sound/soc/blackfin/bf5xx-i2s.c
+++ b/sound/soc/blackfin/bf5xx-i2s.c
@@ -111,6 +111,7 @@ static int bf5xx_i2s_hw_params(struct snd_pcm_substream *substream,
bf5xx_i2s->tcr2 |= 7;
bf5xx_i2s->rcr2 |= 7;
sport_handle->wdsize = 1;
+ break;
case SNDRV_PCM_FORMAT_S16_LE:
bf5xx_i2s->tcr2 |= 15;
bf5xx_i2s->rcr2 |= 15;
diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c
index 60159c07448..6fd174be3bd 100644
--- a/sound/soc/codecs/88pm860x-codec.c
+++ b/sound/soc/codecs/88pm860x-codec.c
@@ -351,6 +351,9 @@ static int snd_soc_put_volsw_2r_st(struct snd_kcontrol *kcontrol,
val = ucontrol->value.integer.value[0];
val2 = ucontrol->value.integer.value[1];
+ if (val >= ARRAY_SIZE(st_table) || val2 >= ARRAY_SIZE(st_table))
+ return -EINVAL;
+
err = snd_soc_update_bits(codec, reg, 0x3f, st_table[val].m);
if (err < 0)
return err;
diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c
index a153b168129..bce45c197e1 100644
--- a/sound/soc/codecs/ab8500-codec.c
+++ b/sound/soc/codecs/ab8500-codec.c
@@ -1225,13 +1225,18 @@ static int anc_status_control_put(struct snd_kcontrol *kcontrol,
struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(codec->dev);
struct device *dev = codec->dev;
bool apply_fir, apply_iir;
- int req, status;
+ unsigned int req;
+ int status;
dev_dbg(dev, "%s: Enter.\n", __func__);
mutex_lock(&drvdata->anc_lock);
req = ucontrol->value.integer.value[0];
+ if (req >= ARRAY_SIZE(enum_anc_state)) {
+ status = -EINVAL;
+ goto cleanup;
+ }
if (req != ANC_APPLY_FIR_IIR && req != ANC_APPLY_FIR &&
req != ANC_APPLY_IIR) {
dev_err(dev, "%s: ERROR: Unsupported status to set '%s'!\n",
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 2d037870970..687565d08d9 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -257,7 +257,7 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream,
* This operation came from example code of
* "ASAHI KASEI AK4642" (japanese) manual p94.
*/
- snd_soc_write(codec, SG_SL1, PMMP | MGAIN0);
+ snd_soc_update_bits(codec, SG_SL1, PMMP | MGAIN0, PMMP | MGAIN0);
snd_soc_write(codec, TIMER, ZTM(0x3) | WTM(0x3));
snd_soc_write(codec, ALC_CTL1, ALC | LMTH0);
snd_soc_update_bits(codec, PW_MGMT1, PMADL, PMADL);
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index 389f2325383..663a2a74862 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -1455,6 +1455,8 @@ static void arizona_enable_fll(struct arizona_fll *fll,
try_wait_for_completion(&fll->ok);
regmap_update_bits(arizona->regmap, fll->base + 1,
+ ARIZONA_FLL1_FREERUN, 0);
+ regmap_update_bits(arizona->regmap, fll->base + 1,
ARIZONA_FLL1_ENA, ARIZONA_FLL1_ENA);
if (fll->ref_src >= 0 && fll->sync_src >= 0 &&
fll->ref_src != fll->sync_src)
@@ -1473,6 +1475,8 @@ static void arizona_disable_fll(struct arizona_fll *fll)
struct arizona *arizona = fll->arizona;
bool change;
+ regmap_update_bits(arizona->regmap, fll->base + 1,
+ ARIZONA_FLL1_FREERUN, ARIZONA_FLL1_FREERUN);
regmap_update_bits_check(arizona->regmap, fll->base + 1,
ARIZONA_FLL1_ENA, 0, &change);
regmap_update_bits(arizona->regmap, fll->base + 0x11,
diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
index 987f728718c..ee25f325d65 100644
--- a/sound/soc/codecs/cs42l52.c
+++ b/sound/soc/codecs/cs42l52.c
@@ -451,7 +451,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = {
SOC_ENUM("Beep Pitch", beep_pitch_enum),
SOC_ENUM("Beep on Time", beep_ontime_enum),
SOC_ENUM("Beep off Time", beep_offtime_enum),
- SOC_SINGLE_TLV("Beep Volume", CS42L52_BEEP_VOL, 0, 0x1f, 0x07, hl_tlv),
+ SOC_SINGLE_SX_TLV("Beep Volume", CS42L52_BEEP_VOL, 0, 0x07, 0x1f, hl_tlv),
SOC_SINGLE("Beep Mixer Switch", CS42L52_BEEP_TONE_CTL, 5, 1, 1),
SOC_ENUM("Beep Treble Corner Freq", beep_treble_enum),
SOC_ENUM("Beep Bass Corner Freq", beep_bass_enum),
diff --git a/sound/soc/codecs/cs42l52.h b/sound/soc/codecs/cs42l52.h
index 4277012c471..a935d7381af 100644
--- a/sound/soc/codecs/cs42l52.h
+++ b/sound/soc/codecs/cs42l52.h
@@ -179,7 +179,7 @@
#define CS42L52_MICB_CTL 0x11
#define CS42L52_MIC_CTL_MIC_SEL_MASK 0xBF
#define CS42L52_MIC_CTL_MIC_SEL_SHIFT 6
-#define CS42L52_MIC_CTL_TYPE_MASK 0xDF
+#define CS42L52_MIC_CTL_TYPE_MASK 0x20
#define CS42L52_MIC_CTL_TYPE_SHIFT 5
diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c
index 3eeada57e87..566a367c94f 100644
--- a/sound/soc/codecs/max98088.c
+++ b/sound/soc/codecs/max98088.c
@@ -1612,7 +1612,7 @@ static int max98088_dai2_digital_mute(struct snd_soc_dai *codec_dai, int mute)
static void max98088_sync_cache(struct snd_soc_codec *codec)
{
- u16 *reg_cache = codec->reg_cache;
+ u8 *reg_cache = codec->reg_cache;
int i;
if (!codec->cache_sync)
diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c
index 41cdd164297..8dbcacd44e6 100644
--- a/sound/soc/codecs/max98095.c
+++ b/sound/soc/codecs/max98095.c
@@ -1863,7 +1863,7 @@ static int max98095_put_eq_enum(struct snd_kcontrol *kcontrol,
struct max98095_pdata *pdata = max98095->pdata;
int channel = max98095_get_eq_channel(kcontrol->id.name);
struct max98095_cdata *cdata;
- int sel = ucontrol->value.integer.value[0];
+ unsigned int sel = ucontrol->value.integer.value[0];
struct max98095_eq_cfg *coef_set;
int fs, best, best_val, i;
int regmask, regsave;
@@ -2016,7 +2016,7 @@ static int max98095_put_bq_enum(struct snd_kcontrol *kcontrol,
struct max98095_pdata *pdata = max98095->pdata;
int channel = max98095_get_bq_channel(codec, kcontrol->id.name);
struct max98095_cdata *cdata;
- int sel = ucontrol->value.integer.value[0];
+ unsigned int sel = ucontrol->value.integer.value[0];
struct max98095_biquad_cfg *coef_set;
int fs, best, best_val, i;
int regmask, regsave;
diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c
index 5402dfbbb71..8a8d9364e87 100644
--- a/sound/soc/codecs/mc13783.c
+++ b/sound/soc/codecs/mc13783.c
@@ -126,6 +126,10 @@ static int mc13783_write(struct snd_soc_codec *codec,
ret = mc13xxx_reg_write(priv->mc13xxx, reg, value);
+ /* include errata fix for spi audio problems */
+ if (reg == MC13783_AUDIO_CODEC || reg == MC13783_AUDIO_DAC)
+ ret = mc13xxx_reg_write(priv->mc13xxx, reg, value);
+
mc13xxx_unlock(priv->mc13xxx);
return ret;
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index 92bbfec9b10..ea479388fb5 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -37,7 +37,7 @@
static const u16 sgtl5000_regs[SGTL5000_MAX_REG_OFFSET] = {
[SGTL5000_CHIP_CLK_CTRL] = 0x0008,
[SGTL5000_CHIP_I2S_CTRL] = 0x0010,
- [SGTL5000_CHIP_SSS_CTRL] = 0x0008,
+ [SGTL5000_CHIP_SSS_CTRL] = 0x0010,
[SGTL5000_CHIP_DAC_VOL] = 0x3c3c,
[SGTL5000_CHIP_PAD_STRENGTH] = 0x015f,
[SGTL5000_CHIP_ANA_HP_CTRL] = 0x1818,
diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h
index 8a9f43534b7..d3a68bbfea0 100644
--- a/sound/soc/codecs/sgtl5000.h
+++ b/sound/soc/codecs/sgtl5000.h
@@ -347,7 +347,7 @@
#define SGTL5000_PLL_INT_DIV_MASK 0xf800
#define SGTL5000_PLL_INT_DIV_SHIFT 11
#define SGTL5000_PLL_INT_DIV_WIDTH 5
-#define SGTL5000_PLL_FRAC_DIV_MASK 0x0700
+#define SGTL5000_PLL_FRAC_DIV_MASK 0x07ff
#define SGTL5000_PLL_FRAC_DIV_SHIFT 0
#define SGTL5000_PLL_FRAC_DIV_WIDTH 11
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index 88ad7db52dd..cd6ed88409d 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -37,6 +37,47 @@ struct wm5110_priv {
struct arizona_fll fll[2];
};
+static const struct reg_default wm5110_sysclk_revd_patch[] = {
+ { 0x3093, 0x1001 },
+ { 0x30E3, 0x1301 },
+ { 0x3133, 0x1201 },
+ { 0x3183, 0x1501 },
+ { 0x31D3, 0x1401 },
+};
+
+static int wm5110_sysclk_ev(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ struct arizona *arizona = dev_get_drvdata(codec->dev->parent);
+ struct regmap *regmap = codec->control_data;
+ const struct reg_default *patch = NULL;
+ int i, patch_size;
+
+ switch (arizona->rev) {
+ case 3:
+ patch = wm5110_sysclk_revd_patch;
+ patch_size = ARRAY_SIZE(wm5110_sysclk_revd_patch);
+ break;
+ default:
+ return 0;
+ }
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ if (patch)
+ for (i = 0; i < patch_size; i++)
+ regmap_write(regmap, patch[i].reg,
+ patch[i].def);
+ break;
+
+ default:
+ break;
+ }
+
+ return 0;
+}
+
static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0);
static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0);
static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0);
@@ -386,7 +427,7 @@ static const struct snd_kcontrol_new wm5110_aec_loopback_mux =
static const struct snd_soc_dapm_widget wm5110_dapm_widgets[] = {
SND_SOC_DAPM_SUPPLY("SYSCLK", ARIZONA_SYSTEM_CLOCK_1, ARIZONA_SYSCLK_ENA_SHIFT,
- 0, NULL, 0),
+ 0, wm5110_sysclk_ev, SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_SUPPLY("ASYNCCLK", ARIZONA_ASYNC_CLOCK_1,
ARIZONA_ASYNC_CLK_ENA_SHIFT, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("OPCLK", ARIZONA_OUTPUT_SYSTEM_CLOCK,
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 5276062d6c7..10d492b6a5b 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -407,10 +407,10 @@ static int wm8731_set_dai_fmt(struct snd_soc_dai *codec_dai,
iface |= 0x0001;
break;
case SND_SOC_DAIFMT_DSP_A:
- iface |= 0x0003;
+ iface |= 0x0013;
break;
case SND_SOC_DAIFMT_DSP_B:
- iface |= 0x0013;
+ iface |= 0x0003;
break;
default:
return -EINVAL;
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index 0a4ffdd1d2a..5e5af898f7f 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -857,9 +857,9 @@ static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
if (pll_div.k) {
reg |= 0x20;
- snd_soc_write(codec, WM8960_PLL2, (pll_div.k >> 18) & 0x3f);
- snd_soc_write(codec, WM8960_PLL3, (pll_div.k >> 9) & 0x1ff);
- snd_soc_write(codec, WM8960_PLL4, pll_div.k & 0x1ff);
+ snd_soc_write(codec, WM8960_PLL2, (pll_div.k >> 16) & 0xff);
+ snd_soc_write(codec, WM8960_PLL3, (pll_div.k >> 8) & 0xff);
+ snd_soc_write(codec, WM8960_PLL4, pll_div.k & 0xff);
}
snd_soc_write(codec, WM8960_PLL1, reg);
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index e9710280e5e..e3cd86514ce 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -1600,7 +1600,6 @@ static int wm8962_put_hp_sw(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
- u16 *reg_cache = codec->reg_cache;
int ret;
/* Apply the update (if any) */
@@ -1609,16 +1608,19 @@ static int wm8962_put_hp_sw(struct snd_kcontrol *kcontrol,
return 0;
/* If the left PGA is enabled hit that VU bit... */
- if (snd_soc_read(codec, WM8962_PWR_MGMT_2) & WM8962_HPOUTL_PGA_ENA)
- return snd_soc_write(codec, WM8962_HPOUTL_VOLUME,
- reg_cache[WM8962_HPOUTL_VOLUME]);
+ ret = snd_soc_read(codec, WM8962_PWR_MGMT_2);
+ if (ret & WM8962_HPOUTL_PGA_ENA) {
+ snd_soc_write(codec, WM8962_HPOUTL_VOLUME,
+ snd_soc_read(codec, WM8962_HPOUTL_VOLUME));
+ return 1;
+ }
/* ...otherwise the right. The VU is stereo. */
- if (snd_soc_read(codec, WM8962_PWR_MGMT_2) & WM8962_HPOUTR_PGA_ENA)
- return snd_soc_write(codec, WM8962_HPOUTR_VOLUME,
- reg_cache[WM8962_HPOUTR_VOLUME]);
+ if (ret & WM8962_HPOUTR_PGA_ENA)
+ snd_soc_write(codec, WM8962_HPOUTR_VOLUME,
+ snd_soc_read(codec, WM8962_HPOUTR_VOLUME));
- return 0;
+ return 1;
}
/* The VU bits for the speakers are in a different register to the mute
@@ -3374,7 +3376,6 @@ static int wm8962_probe(struct snd_soc_codec *codec)
int ret;
struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec);
struct wm8962_pdata *pdata = dev_get_platdata(codec->dev);
- u16 *reg_cache = codec->reg_cache;
int i, trigger, irq_pol;
bool dmicclk, dmicdat;
@@ -3432,8 +3433,9 @@ static int wm8962_probe(struct snd_soc_codec *codec)
/* Put the speakers into mono mode? */
if (pdata->spk_mono)
- reg_cache[WM8962_CLASS_D_CONTROL_2]
- |= WM8962_SPK_MONO;
+ snd_soc_update_bits(codec, WM8962_CLASS_D_CONTROL_2,
+ WM8962_SPK_MONO_MASK, WM8962_SPK_MONO);
+
/* Micbias setup, detection enable and detection
* threasholds. */
@@ -3684,6 +3686,8 @@ static int wm8962_i2c_probe(struct i2c_client *i2c,
if (ret < 0)
goto err_enable;
+ regcache_cache_only(wm8962->regmap, true);
+
/* The drivers should power up as needed */
regulator_bulk_disable(ARRAY_SIZE(wm8962->supplies), wm8962->supplies);
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 837978e16e9..ded9ed854a1 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -1264,6 +1264,8 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec,
/* disable POBCTRL, SOFT_ST and BUFDCOPEN */
snd_soc_write(codec, WM8990_ANTIPOP2, 0x0);
+
+ codec->cache_sync = 1;
break;
}
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index f5d81b94875..7a0466eb7ed 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -530,6 +530,7 @@ static int hp_supply_event(struct snd_soc_dapm_widget *w,
hubs->hp_startup_mode);
break;
}
+ break;
case SND_SOC_DAPM_PRE_PMD:
snd_soc_update_bits(codec, WM8993_CHARGE_PUMP_1,
diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c
index 670b96b0ce2..dcfd0fae0b3 100644
--- a/sound/soc/fsl/imx-pcm-fiq.c
+++ b/sound/soc/fsl/imx-pcm-fiq.c
@@ -42,7 +42,8 @@ struct imx_pcm_runtime_data {
struct hrtimer hrt;
int poll_time_ns;
struct snd_pcm_substream *substream;
- atomic_t running;
+ atomic_t playing;
+ atomic_t capturing;
};
static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt)
@@ -54,7 +55,7 @@ static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt)
struct pt_regs regs;
unsigned long delta;
- if (!atomic_read(&iprtd->running))
+ if (!atomic_read(&iprtd->playing) && !atomic_read(&iprtd->capturing))
return HRTIMER_NORESTART;
get_fiq_regs(&regs);
@@ -122,7 +123,6 @@ static int snd_imx_pcm_prepare(struct snd_pcm_substream *substream)
return 0;
}
-static int fiq_enable;
static int imx_pcm_fiq;
static int snd_imx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
@@ -134,23 +134,27 @@ static int snd_imx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
- atomic_set(&iprtd->running, 1);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ atomic_set(&iprtd->playing, 1);
+ else
+ atomic_set(&iprtd->capturing, 1);
hrtimer_start(&iprtd->hrt, ns_to_ktime(iprtd->poll_time_ns),
HRTIMER_MODE_REL);
- if (++fiq_enable == 1)
- enable_fiq(imx_pcm_fiq);
-
+ enable_fiq(imx_pcm_fiq);
break;
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
- atomic_set(&iprtd->running, 0);
-
- if (--fiq_enable == 0)
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ atomic_set(&iprtd->playing, 0);
+ else
+ atomic_set(&iprtd->capturing, 0);
+ if (!atomic_read(&iprtd->playing) &&
+ !atomic_read(&iprtd->capturing))
disable_fiq(imx_pcm_fiq);
-
break;
+
default:
return -EINVAL;
}
@@ -198,7 +202,8 @@ static int snd_imx_open(struct snd_pcm_substream *substream)
iprtd->substream = substream;
- atomic_set(&iprtd->running, 0);
+ atomic_set(&iprtd->playing, 0);
+ atomic_set(&iprtd->capturing, 0);
hrtimer_init(&iprtd->hrt, CLOCK_MONOTONIC, HRTIMER_MODE_REL);
iprtd->hrt.function = snd_hrtimer_callback;
diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c
index 1358c7de252..d0740a76296 100644
--- a/sound/soc/s6000/s6000-pcm.c
+++ b/sound/soc/s6000/s6000-pcm.c
@@ -128,7 +128,9 @@ static irqreturn_t s6000_pcm_irq(int irq, void *data)
substream->runtime &&
snd_pcm_running(substream)) {
dev_dbg(pcm->dev, "xrun\n");
+ snd_pcm_stream_lock(substream);
snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
+ snd_pcm_stream_unlock(substream);
ret = IRQ_HANDLED;
}
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index c7051c457b7..c2ecb4e0159 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -682,13 +682,14 @@ static int dapm_new_mux(struct snd_soc_dapm_widget *w)
return -EINVAL;
}
- path = list_first_entry(&w->sources, struct snd_soc_dapm_path,
- list_sink);
- if (!path) {
+ if (list_empty(&w->sources)) {
dev_err(dapm->dev, "ASoC: mux %s has no paths\n", w->name);
return -EINVAL;
}
+ path = list_first_entry(&w->sources, struct snd_soc_dapm_path,
+ list_sink);
+
ret = dapm_create_or_share_mixmux_kcontrol(w, 0, path);
if (ret < 0)
return ret;
@@ -1796,7 +1797,7 @@ static ssize_t dapm_widget_power_read_file(struct file *file,
w->active ? "active" : "inactive");
list_for_each_entry(p, &w->sources, list_sink) {
- if (p->connected && !p->connected(w, p->sink))
+ if (p->connected && !p->connected(w, p->source))
continue;
if (p->connect)
diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c
index 2f70ea7f661..05676c022a1 100644
--- a/sound/soc/tegra/tegra20_ac97.c
+++ b/sound/soc/tegra/tegra20_ac97.c
@@ -399,9 +399,9 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev)
ac97->capture_dma_data.slave_id = of_dma[1];
ac97->playback_dma_data.addr = mem->start + TEGRA20_AC97_FIFO_TX1;
- ac97->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES;
- ac97->capture_dma_data.maxburst = 4;
- ac97->capture_dma_data.slave_id = of_dma[0];
+ ac97->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES;
+ ac97->playback_dma_data.maxburst = 4;
+ ac97->playback_dma_data.slave_id = of_dma[1];
ret = snd_soc_register_component(&pdev->dev, &tegra20_ac97_component,
&tegra20_ac97_dai, 1);
diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c
index 5eaa12cdc6e..551b3c93ce9 100644
--- a/sound/soc/tegra/tegra20_spdif.c
+++ b/sound/soc/tegra/tegra20_spdif.c
@@ -323,8 +323,8 @@ static int tegra20_spdif_platform_probe(struct platform_device *pdev)
}
spdif->playback_dma_data.addr = mem->start + TEGRA20_SPDIF_DATA_OUT;
- spdif->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES;
- spdif->capture_dma_data.maxburst = 4;
+ spdif->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES;
+ spdif->playback_dma_data.maxburst = 4;
spdif->playback_dma_data.slave_id = dmareq->start;
pm_runtime_enable(&pdev->dev);
diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c
index 31d092d83c7..a5432b10eac 100644
--- a/sound/soc/tegra/tegra30_i2s.c
+++ b/sound/soc/tegra/tegra30_i2s.c
@@ -228,7 +228,7 @@ static int tegra30_i2s_hw_params(struct snd_pcm_substream *substream,
reg = TEGRA30_I2S_CIF_RX_CTRL;
} else {
val |= TEGRA30_AUDIOCIF_CTRL_DIRECTION_TX;
- reg = TEGRA30_I2S_CIF_RX_CTRL;
+ reg = TEGRA30_I2S_CIF_TX_CTRL;
}
regmap_write(i2s->regmap, reg, val);
generated by cgit v1.2.3 (git 2.25.1) at 2024-06-07 09:25:09 +0000