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authorScott Wood <scottwood@freescale.com>2015-02-13 22:12:06 (GMT)
committerScott Wood <scottwood@freescale.com>2015-02-13 22:19:22 (GMT)
commit6faa2909871d8937cb2f79a10e1b21ffe193fac1 (patch)
treef558a94f1553814cc122ab8d9e04c0ebad5262a5 /sound/soc
parentfcb2fb84301c673ee15ca04e7a2fc965712d49a0 (diff)
downloadlinux-fsl-qoriq-6faa2909871d8937cb2f79a10e1b21ffe193fac1.tar.xz
Reset to 3.12.37
Diffstat (limited to 'sound/soc')
-rw-r--r--sound/soc/blackfin/bf5xx-i2s-pcm.c8
-rw-r--r--sound/soc/codecs/adau1701.c6
-rw-r--r--sound/soc/codecs/cs42l51.c9
-rw-r--r--sound/soc/codecs/cs42l52.c8
-rw-r--r--sound/soc/codecs/cs42l73.c6
-rw-r--r--sound/soc/codecs/max98090.c9
-rw-r--r--sound/soc/codecs/rt5640.c1
-rw-r--r--sound/soc/codecs/sgtl5000.c3
-rw-r--r--sound/soc/codecs/sgtl5000.h2
-rw-r--r--sound/soc/codecs/sigmadsp.c7
-rw-r--r--sound/soc/codecs/tlv320aic3x.c2
-rw-r--r--sound/soc/codecs/wm8962.c15
-rw-r--r--sound/soc/codecs/wm8962.h4
-rw-r--r--sound/soc/codecs/wm8994.c9
-rw-r--r--sound/soc/codecs/wm_adsp.c3
-rw-r--r--sound/soc/davinci/davinci-mcasp.c11
-rw-r--r--sound/soc/dwc/designware_i2s.c14
-rw-r--r--sound/soc/pxa/pxa-ssp.c7
-rw-r--r--sound/soc/samsung/i2s.c16
-rw-r--r--sound/soc/sh/fsi.c3
-rw-r--r--sound/soc/sh/rcar/core.c3
-rw-r--r--sound/soc/soc-dapm.c26
-rw-r--r--sound/soc/soc-pcm.c73
23 files changed, 164 insertions, 81 deletions
diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c
index 9cb4a80..bc9983d 100644
--- a/sound/soc/blackfin/bf5xx-i2s-pcm.c
+++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c
@@ -293,19 +293,19 @@ static int bf5xx_pcm_silence(struct snd_pcm_substream *substream,
unsigned int sample_size = runtime->sample_bits / 8;
void *buf = runtime->dma_area;
struct bf5xx_i2s_pcm_data *dma_data;
- unsigned int offset, size;
+ unsigned int offset, samples;
dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
if (dma_data->tdm_mode) {
offset = pos * 8 * sample_size;
- size = count * 8 * sample_size;
+ samples = count * 8;
} else {
offset = frames_to_bytes(runtime, pos);
- size = frames_to_bytes(runtime, count);
+ samples = count * runtime->channels;
}
- snd_pcm_format_set_silence(runtime->format, buf + offset, size);
+ snd_pcm_format_set_silence(runtime->format, buf + offset, samples);
return 0;
}
diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c
index adee866..56bfc67 100644
--- a/sound/soc/codecs/adau1701.c
+++ b/sound/soc/codecs/adau1701.c
@@ -230,8 +230,10 @@ static int adau1701_reg_read(void *context, unsigned int reg,
*value = 0;
- for (i = 0; i < size; i++)
- *value |= recv_buf[i] << (i * 8);
+ for (i = 0; i < size; i++) {
+ *value <<= 8;
+ *value |= recv_buf[i];
+ }
return 0;
}
diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c
index 1e0fa3b..e1dfebb 100644
--- a/sound/soc/codecs/cs42l51.c
+++ b/sound/soc/codecs/cs42l51.c
@@ -124,9 +124,8 @@ static int cs42l51_set_chan_mix(struct snd_kcontrol *kcontrol,
static const DECLARE_TLV_DB_SCALE(adc_pcm_tlv, -5150, 50, 0);
static const DECLARE_TLV_DB_SCALE(tone_tlv, -1050, 150, 0);
-/* This is a lie. after -102 db, it stays at -102 */
-/* maybe a range would be better */
-static const DECLARE_TLV_DB_SCALE(aout_tlv, -11550, 50, 0);
+
+static const DECLARE_TLV_DB_SCALE(aout_tlv, -10200, 50, 0);
static const DECLARE_TLV_DB_SCALE(boost_tlv, 1600, 1600, 0);
static const char *chan_mix[] = {
@@ -141,7 +140,7 @@ static const struct soc_enum cs42l51_chan_mix =
static const struct snd_kcontrol_new cs42l51_snd_controls[] = {
SOC_DOUBLE_R_SX_TLV("PCM Playback Volume",
CS42L51_PCMA_VOL, CS42L51_PCMB_VOL,
- 6, 0x19, 0x7F, adc_pcm_tlv),
+ 0, 0x19, 0x7F, adc_pcm_tlv),
SOC_DOUBLE_R("PCM Playback Switch",
CS42L51_PCMA_VOL, CS42L51_PCMB_VOL, 7, 1, 1),
SOC_DOUBLE_R_SX_TLV("Analog Playback Volume",
@@ -149,7 +148,7 @@ static const struct snd_kcontrol_new cs42l51_snd_controls[] = {
0, 0x34, 0xE4, aout_tlv),
SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume",
CS42L51_ADCA_VOL, CS42L51_ADCB_VOL,
- 6, 0x19, 0x7F, adc_pcm_tlv),
+ 0, 0x19, 0x7F, adc_pcm_tlv),
SOC_DOUBLE_R("ADC Mixer Switch",
CS42L51_ADCA_VOL, CS42L51_ADCB_VOL, 7, 1, 1),
SOC_SINGLE("Playback Deemphasis Switch", CS42L51_DAC_CTL, 3, 1, 0),
diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
index be2ba1b..ab3ac7b 100644
--- a/sound/soc/codecs/cs42l52.c
+++ b/sound/soc/codecs/cs42l52.c
@@ -352,7 +352,7 @@ static const char * const right_swap_text[] = {
static const unsigned int swap_values[] = { 0, 1, 3 };
static const struct soc_enum adca_swap_enum =
- SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 2, 1,
+ SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 2, 3,
ARRAY_SIZE(left_swap_text),
left_swap_text,
swap_values);
@@ -361,7 +361,7 @@ static const struct snd_kcontrol_new adca_mixer =
SOC_DAPM_ENUM("Route", adca_swap_enum);
static const struct soc_enum pcma_swap_enum =
- SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 6, 1,
+ SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 6, 3,
ARRAY_SIZE(left_swap_text),
left_swap_text,
swap_values);
@@ -370,7 +370,7 @@ static const struct snd_kcontrol_new pcma_mixer =
SOC_DAPM_ENUM("Route", pcma_swap_enum);
static const struct soc_enum adcb_swap_enum =
- SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 0, 1,
+ SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 0, 3,
ARRAY_SIZE(right_swap_text),
right_swap_text,
swap_values);
@@ -379,7 +379,7 @@ static const struct snd_kcontrol_new adcb_mixer =
SOC_DAPM_ENUM("Route", adcb_swap_enum);
static const struct soc_enum pcmb_swap_enum =
- SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 4, 1,
+ SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 4, 3,
ARRAY_SIZE(right_swap_text),
right_swap_text,
swap_values);
diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c
index 3b20c86..eade6e2 100644
--- a/sound/soc/codecs/cs42l73.c
+++ b/sound/soc/codecs/cs42l73.c
@@ -325,7 +325,7 @@ static const char * const cs42l73_mono_mix_texts[] = {
static const unsigned int cs42l73_mono_mix_values[] = { 0, 1, 2 };
static const struct soc_enum spk_asp_enum =
- SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 6, 1,
+ SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 6, 3,
ARRAY_SIZE(cs42l73_mono_mix_texts),
cs42l73_mono_mix_texts,
cs42l73_mono_mix_values);
@@ -343,7 +343,7 @@ static const struct snd_kcontrol_new spk_xsp_mixer =
SOC_DAPM_ENUM("Route", spk_xsp_enum);
static const struct soc_enum esl_asp_enum =
- SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 2, 5,
+ SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 2, 3,
ARRAY_SIZE(cs42l73_mono_mix_texts),
cs42l73_mono_mix_texts,
cs42l73_mono_mix_values);
@@ -352,7 +352,7 @@ static const struct snd_kcontrol_new esl_asp_mixer =
SOC_DAPM_ENUM("Route", esl_asp_enum);
static const struct soc_enum esl_xsp_enum =
- SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 0, 7,
+ SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 0, 3,
ARRAY_SIZE(cs42l73_mono_mix_texts),
cs42l73_mono_mix_texts,
cs42l73_mono_mix_values);
diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c
index 8bddf3f..9c20ef5 100644
--- a/sound/soc/codecs/max98090.c
+++ b/sound/soc/codecs/max98090.c
@@ -255,6 +255,7 @@ static struct reg_default max98090_reg[] = {
static bool max98090_volatile_register(struct device *dev, unsigned int reg)
{
switch (reg) {
+ case M98090_REG_SOFTWARE_RESET:
case M98090_REG_DEVICE_STATUS:
case M98090_REG_JACK_STATUS:
case M98090_REG_REVISION_ID:
@@ -1377,8 +1378,8 @@ static const struct snd_soc_dapm_route max98090_dapm_routes[] = {
{"STENL Mux", "Sidetone Left", "DMICL"},
{"STENR Mux", "Sidetone Right", "ADCR"},
{"STENR Mux", "Sidetone Right", "DMICR"},
- {"DACL", "NULL", "STENL Mux"},
- {"DACR", "NULL", "STENL Mux"},
+ {"DACL", NULL, "STENL Mux"},
+ {"DACR", NULL, "STENL Mux"},
{"AIFINL", NULL, "SHDN"},
{"AIFINR", NULL, "SHDN"},
@@ -2249,7 +2250,7 @@ static int max98090_probe(struct snd_soc_codec *codec)
/* Register for interrupts */
dev_dbg(codec->dev, "irq = %d\n", max98090->irq);
- ret = request_threaded_irq(max98090->irq, NULL,
+ ret = devm_request_threaded_irq(codec->dev, max98090->irq, NULL,
max98090_interrupt, IRQF_TRIGGER_FALLING | IRQF_ONESHOT,
"max98090_interrupt", codec);
if (ret < 0) {
@@ -2360,6 +2361,8 @@ static int max98090_runtime_resume(struct device *dev)
regcache_cache_only(max98090->regmap, false);
+ max98090_reset(max98090);
+
regcache_sync(max98090->regmap);
return 0;
diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c
index c26a8f8..aa5253a 100644
--- a/sound/soc/codecs/rt5640.c
+++ b/sound/soc/codecs/rt5640.c
@@ -2061,6 +2061,7 @@ 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/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index 1f4093f..b76c6b6 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -1398,8 +1398,7 @@ static int sgtl5000_probe(struct snd_soc_codec *codec)
/* enable small pop, introduce 400ms delay in turning off */
snd_soc_update_bits(codec, SGTL5000_CHIP_REF_CTRL,
- SGTL5000_SMALL_POP,
- SGTL5000_SMALL_POP);
+ SGTL5000_SMALL_POP, 1);
/* disable short cut detector */
snd_soc_write(codec, SGTL5000_CHIP_SHORT_CTRL, 0);
diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h
index 2f8c889..bd7a344 100644
--- a/sound/soc/codecs/sgtl5000.h
+++ b/sound/soc/codecs/sgtl5000.h
@@ -275,7 +275,7 @@
#define SGTL5000_BIAS_CTRL_MASK 0x000e
#define SGTL5000_BIAS_CTRL_SHIFT 1
#define SGTL5000_BIAS_CTRL_WIDTH 3
-#define SGTL5000_SMALL_POP 0x0001
+#define SGTL5000_SMALL_POP 0
/*
* SGTL5000_CHIP_MIC_CTRL
diff --git a/sound/soc/codecs/sigmadsp.c b/sound/soc/codecs/sigmadsp.c
index 4068f24..bb3878c 100644
--- a/sound/soc/codecs/sigmadsp.c
+++ b/sound/soc/codecs/sigmadsp.c
@@ -176,6 +176,13 @@ static int _process_sigma_firmware(struct device *dev,
goto done;
}
+ if (ssfw_head->version != 1) {
+ dev_err(dev,
+ "Failed to load firmware: Invalid version %d. Supported firmware versions: 1\n",
+ ssfw_head->version);
+ goto done;
+ }
+
crc = crc32(0, fw->data + sizeof(*ssfw_head),
fw->size - sizeof(*ssfw_head));
pr_debug("%s: crc=%x\n", __func__, crc);
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 64ad84d..11c8d1f 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -164,7 +164,7 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol,
mask <<= shift;
val <<= shift;
- change = snd_soc_test_bits(codec, val, mask, reg);
+ change = snd_soc_test_bits(codec, reg, mask, val);
if (change) {
update.kcontrol = kcontrol;
update.reg = reg;
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 871f851..ea16dc4 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -154,6 +154,7 @@ static struct reg_default wm8962_reg[] = {
{ 40, 0x0000 }, /* R40 - SPKOUTL volume */
{ 41, 0x0000 }, /* R41 - SPKOUTR volume */
+ { 49, 0x0010 }, /* R49 - Class D Control 1 */
{ 51, 0x0003 }, /* R51 - Class D Control 2 */
{ 56, 0x0506 }, /* R56 - Clocking 4 */
@@ -795,7 +796,6 @@ static bool wm8962_volatile_register(struct device *dev, unsigned int reg)
case WM8962_ALC2:
case WM8962_THERMAL_SHUTDOWN_STATUS:
case WM8962_ADDITIONAL_CONTROL_4:
- case WM8962_CLASS_D_CONTROL_1:
case WM8962_DC_SERVO_6:
case WM8962_INTERRUPT_STATUS_1:
case WM8962_INTERRUPT_STATUS_2:
@@ -2901,13 +2901,22 @@ static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
static int wm8962_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_codec *codec = dai->codec;
- int val;
+ int val, ret;
if (mute)
- val = WM8962_DAC_MUTE;
+ val = WM8962_DAC_MUTE | WM8962_DAC_MUTE_ALT;
else
val = 0;
+ /**
+ * The DAC mute bit is mirrored in two registers, update both to keep
+ * the register cache consistent.
+ */
+ ret = snd_soc_update_bits(codec, WM8962_CLASS_D_CONTROL_1,
+ WM8962_DAC_MUTE_ALT, val);
+ if (ret < 0)
+ return ret;
+
return snd_soc_update_bits(codec, WM8962_ADC_DAC_CONTROL_1,
WM8962_DAC_MUTE, val);
}
diff --git a/sound/soc/codecs/wm8962.h b/sound/soc/codecs/wm8962.h
index a1a5d52..910aafd 100644
--- a/sound/soc/codecs/wm8962.h
+++ b/sound/soc/codecs/wm8962.h
@@ -1954,6 +1954,10 @@
#define WM8962_SPKOUTL_ENA_MASK 0x0040 /* SPKOUTL_ENA */
#define WM8962_SPKOUTL_ENA_SHIFT 6 /* SPKOUTL_ENA */
#define WM8962_SPKOUTL_ENA_WIDTH 1 /* SPKOUTL_ENA */
+#define WM8962_DAC_MUTE_ALT 0x0010 /* DAC_MUTE */
+#define WM8962_DAC_MUTE_ALT_MASK 0x0010 /* DAC_MUTE */
+#define WM8962_DAC_MUTE_ALT_SHIFT 4 /* DAC_MUTE */
+#define WM8962_DAC_MUTE_ALT_WIDTH 1 /* DAC_MUTE */
#define WM8962_SPKOUTL_PGA_MUTE 0x0002 /* SPKOUTL_PGA_MUTE */
#define WM8962_SPKOUTL_PGA_MUTE_MASK 0x0002 /* SPKOUTL_PGA_MUTE */
#define WM8962_SPKOUTL_PGA_MUTE_SHIFT 1 /* SPKOUTL_PGA_MUTE */
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 86426a1..c9ce977 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -3492,6 +3492,7 @@ static irqreturn_t wm8994_mic_irq(int irq, void *data)
return IRQ_HANDLED;
}
+/* Should be called with accdet_lock held */
static void wm1811_micd_stop(struct snd_soc_codec *codec)
{
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
@@ -3499,14 +3500,10 @@ static void wm1811_micd_stop(struct snd_soc_codec *codec)
if (!wm8994->jackdet)
return;
- mutex_lock(&wm8994->accdet_lock);
-
snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, WM8958_MICD_ENA, 0);
wm1811_jackdet_set_mode(codec, WM1811_JACKDET_MODE_JACK);
- mutex_unlock(&wm8994->accdet_lock);
-
if (wm8994->wm8994->pdata.jd_ext_cap)
snd_soc_dapm_disable_pin(&codec->dapm,
"MICBIAS2");
@@ -3547,10 +3544,10 @@ static void wm8958_open_circuit_work(struct work_struct *work)
open_circuit_work.work);
struct device *dev = wm8994->wm8994->dev;
- wm1811_micd_stop(wm8994->hubs.codec);
-
mutex_lock(&wm8994->accdet_lock);
+ wm1811_micd_stop(wm8994->hubs.codec);
+
dev_dbg(dev, "Reporting open circuit\n");
wm8994->jack_mic = false;
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index 0d5de60..f0e97fc 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -1341,6 +1341,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp)
file, blocks, pos - firmware->size);
out_fw:
+ regmap_async_complete(regmap);
release_firmware(firmware);
wm_adsp_buf_free(&buf_list);
out:
@@ -1694,3 +1695,5 @@ int wm_adsp2_init(struct wm_adsp *adsp, bool dvfs)
return 0;
}
EXPORT_SYMBOL_GPL(wm_adsp2_init);
+
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 32ddb7f..aab16a7 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -632,8 +632,17 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev,
{
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()
diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c
index 25c31f1..2f63575 100644
--- a/sound/soc/dwc/designware_i2s.c
+++ b/sound/soc/dwc/designware_i2s.c
@@ -263,6 +263,19 @@ static void dw_i2s_shutdown(struct snd_pcm_substream *substream,
snd_soc_dai_set_dma_data(dai, substream, NULL);
}
+static int dw_i2s_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(dai);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ i2s_write_reg(dev->i2s_base, TXFFR, 1);
+ else
+ i2s_write_reg(dev->i2s_base, RXFFR, 1);
+
+ return 0;
+}
+
static int dw_i2s_trigger(struct snd_pcm_substream *substream,
int cmd, struct snd_soc_dai *dai)
{
@@ -294,6 +307,7 @@ static struct snd_soc_dai_ops dw_i2s_dai_ops = {
.startup = dw_i2s_startup,
.shutdown = dw_i2s_shutdown,
.hw_params = dw_i2s_hw_params,
+ .prepare = dw_i2s_prepare,
.trigger = dw_i2s_trigger,
};
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index a3119a0..6c6b35e 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -725,7 +725,8 @@ static int pxa_ssp_probe(struct snd_soc_dai *dai)
ssp_handle = of_parse_phandle(dev->of_node, "port", 0);
if (!ssp_handle) {
dev_err(dev, "unable to get 'port' phandle\n");
- return -ENODEV;
+ ret = -ENODEV;
+ goto err_priv;
}
priv->ssp = pxa_ssp_request_of(ssp_handle, "SoC audio");
@@ -766,9 +767,7 @@ 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/samsung/i2s.c b/sound/soc/samsung/i2s.c
index b302f3b..2ac8d88 100644
--- a/sound/soc/samsung/i2s.c
+++ b/sound/soc/samsung/i2s.c
@@ -922,11 +922,9 @@ static int i2s_suspend(struct snd_soc_dai *dai)
{
struct i2s_dai *i2s = to_info(dai);
- if (dai->active) {
- i2s->suspend_i2smod = readl(i2s->addr + I2SMOD);
- i2s->suspend_i2scon = readl(i2s->addr + I2SCON);
- i2s->suspend_i2spsr = readl(i2s->addr + I2SPSR);
- }
+ i2s->suspend_i2smod = readl(i2s->addr + I2SMOD);
+ i2s->suspend_i2scon = readl(i2s->addr + I2SCON);
+ i2s->suspend_i2spsr = readl(i2s->addr + I2SPSR);
return 0;
}
@@ -935,11 +933,9 @@ static int i2s_resume(struct snd_soc_dai *dai)
{
struct i2s_dai *i2s = to_info(dai);
- if (dai->active) {
- writel(i2s->suspend_i2scon, i2s->addr + I2SCON);
- writel(i2s->suspend_i2smod, i2s->addr + I2SMOD);
- writel(i2s->suspend_i2spsr, i2s->addr + I2SPSR);
- }
+ writel(i2s->suspend_i2scon, i2s->addr + I2SCON);
+ writel(i2s->suspend_i2smod, i2s->addr + I2SMOD);
+ writel(i2s->suspend_i2spsr, i2s->addr + I2SPSR);
return 0;
}
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index b33ca7c..5dbf494 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -1775,8 +1775,7 @@ static const struct snd_soc_dai_ops fsi_dai_ops = {
static struct snd_pcm_hardware fsi_pcm_hardware = {
.info = SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_MMAP |
- SNDRV_PCM_INFO_MMAP_VALID |
- SNDRV_PCM_INFO_PAUSE,
+ SNDRV_PCM_INFO_MMAP_VALID,
.formats = FSI_FMTS,
.rates = FSI_RATES,
.rate_min = 8000,
diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index a357060..f6e45b1 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -662,8 +662,7 @@ static void rsnd_dai_remove(struct platform_device *pdev,
static struct snd_pcm_hardware rsnd_pcm_hardware = {
.info = SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_MMAP |
- SNDRV_PCM_INFO_MMAP_VALID |
- SNDRV_PCM_INFO_PAUSE,
+ SNDRV_PCM_INFO_MMAP_VALID,
.formats = RSND_FMTS,
.rates = RSND_RATES,
.rate_min = 8000,
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index b2949ae..d3fa7b7 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -251,7 +251,6 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget,
static void dapm_kcontrol_free(struct snd_kcontrol *kctl)
{
struct dapm_kcontrol_data *data = snd_kcontrol_chip(kctl);
- kfree(data->widget);
kfree(data->wlist);
kfree(data);
}
@@ -676,9 +675,9 @@ static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w,
int shared;
struct snd_kcontrol *kcontrol;
bool wname_in_long_name, kcname_in_long_name;
- char *long_name;
+ char *long_name = NULL;
const char *name;
- int ret;
+ int ret = 0;
if (dapm->codec)
prefix = dapm->codec->name_prefix;
@@ -743,15 +742,17 @@ static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w,
kcontrol = snd_soc_cnew(&w->kcontrol_news[kci], NULL, name,
prefix);
- kfree(long_name);
- if (!kcontrol)
- return -ENOMEM;
+ if (!kcontrol) {
+ ret = -ENOMEM;
+ goto exit_free;
+ }
+
kcontrol->private_free = dapm_kcontrol_free;
ret = dapm_kcontrol_data_alloc(w, kcontrol);
if (ret) {
snd_ctl_free_one(kcontrol);
- return ret;
+ goto exit_free;
}
ret = snd_ctl_add(card, kcontrol);
@@ -759,17 +760,18 @@ static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w,
dev_err(dapm->dev,
"ASoC: failed to add widget %s dapm kcontrol %s: %d\n",
w->name, name, ret);
- return ret;
+ goto exit_free;
}
}
ret = dapm_kcontrol_add_widget(kcontrol, w);
- if (ret)
- return ret;
+ if (ret == 0)
+ w->kcontrols[kci] = kcontrol;
- w->kcontrols[kci] = kcontrol;
+exit_free:
+ kfree(long_name);
- return 0;
+ return ret;
}
/* create new dapm mixer control */
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 330c9a6..8457ebb 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -1129,13 +1129,36 @@ static void dpcm_set_fe_runtime(struct snd_pcm_substream *substream)
}
}
+static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd);
+
+/* Set FE's runtime_update state; the state is protected via PCM stream lock
+ * for avoiding the race with trigger callback.
+ * If the state is unset and a trigger is pending while the previous operation,
+ * process the pending trigger action here.
+ */
+static void dpcm_set_fe_update_state(struct snd_soc_pcm_runtime *fe,
+ int stream, enum snd_soc_dpcm_update state)
+{
+ struct snd_pcm_substream *substream =
+ snd_soc_dpcm_get_substream(fe, stream);
+
+ snd_pcm_stream_lock_irq(substream);
+ if (state == SND_SOC_DPCM_UPDATE_NO && fe->dpcm[stream].trigger_pending) {
+ dpcm_fe_dai_do_trigger(substream,
+ fe->dpcm[stream].trigger_pending - 1);
+ fe->dpcm[stream].trigger_pending = 0;
+ }
+ fe->dpcm[stream].runtime_update = state;
+ snd_pcm_stream_unlock_irq(substream);
+}
+
static int dpcm_fe_dai_startup(struct snd_pcm_substream *fe_substream)
{
struct snd_soc_pcm_runtime *fe = fe_substream->private_data;
struct snd_pcm_runtime *runtime = fe_substream->runtime;
int stream = fe_substream->stream, ret = 0;
- fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE;
+ dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_FE);
ret = dpcm_be_dai_startup(fe, fe_substream->stream);
if (ret < 0) {
@@ -1157,13 +1180,13 @@ static int dpcm_fe_dai_startup(struct snd_pcm_substream *fe_substream)
dpcm_set_fe_runtime(fe_substream);
snd_pcm_limit_hw_rates(runtime);
- fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO;
+ dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO);
return 0;
unwind:
dpcm_be_dai_startup_unwind(fe, fe_substream->stream);
be_err:
- fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO;
+ dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO);
return ret;
}
@@ -1210,7 +1233,7 @@ static int dpcm_fe_dai_shutdown(struct snd_pcm_substream *substream)
struct snd_soc_pcm_runtime *fe = substream->private_data;
int stream = substream->stream;
- fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE;
+ dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_FE);
/* shutdown the BEs */
dpcm_be_dai_shutdown(fe, substream->stream);
@@ -1224,7 +1247,7 @@ static int dpcm_fe_dai_shutdown(struct snd_pcm_substream *substream)
dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_STOP);
fe->dpcm[stream].state = SND_SOC_DPCM_STATE_CLOSE;
- fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO;
+ dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO);
return 0;
}
@@ -1272,7 +1295,7 @@ static int dpcm_fe_dai_hw_free(struct snd_pcm_substream *substream)
int err, stream = substream->stream;
mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME);
- fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE;
+ dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_FE);
dev_dbg(fe->dev, "ASoC: hw_free FE %s\n", fe->dai_link->name);
@@ -1287,7 +1310,7 @@ static int dpcm_fe_dai_hw_free(struct snd_pcm_substream *substream)
err = dpcm_be_dai_hw_free(fe, stream);
fe->dpcm[stream].state = SND_SOC_DPCM_STATE_HW_FREE;
- fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO;
+ dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO);
mutex_unlock(&fe->card->mutex);
return 0;
@@ -1380,7 +1403,7 @@ static int dpcm_fe_dai_hw_params(struct snd_pcm_substream *substream,
int ret, stream = substream->stream;
mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME);
- fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE;
+ dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_FE);
memcpy(&fe->dpcm[substream->stream].hw_params, params,
sizeof(struct snd_pcm_hw_params));
@@ -1403,7 +1426,7 @@ static int dpcm_fe_dai_hw_params(struct snd_pcm_substream *substream,
fe->dpcm[stream].state = SND_SOC_DPCM_STATE_HW_PARAMS;
out:
- fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO;
+ dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO);
mutex_unlock(&fe->card->mutex);
return ret;
}
@@ -1517,7 +1540,7 @@ static int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream,
}
EXPORT_SYMBOL_GPL(dpcm_be_dai_trigger);
-static int dpcm_fe_dai_trigger(struct snd_pcm_substream *substream, int cmd)
+static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd)
{
struct snd_soc_pcm_runtime *fe = substream->private_data;
int stream = substream->stream, ret;
@@ -1591,6 +1614,23 @@ out:
return ret;
}
+static int dpcm_fe_dai_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_soc_pcm_runtime *fe = substream->private_data;
+ int stream = substream->stream;
+
+ /* if FE's runtime_update is already set, we're in race;
+ * process this trigger later at exit
+ */
+ if (fe->dpcm[stream].runtime_update != SND_SOC_DPCM_UPDATE_NO) {
+ fe->dpcm[stream].trigger_pending = cmd + 1;
+ return 0; /* delayed, assuming it's successful */
+ }
+
+ /* we're alone, let's trigger */
+ return dpcm_fe_dai_do_trigger(substream, cmd);
+}
+
static int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream)
{
struct snd_soc_dpcm *dpcm;
@@ -1634,7 +1674,7 @@ static int dpcm_fe_dai_prepare(struct snd_pcm_substream *substream)
dev_dbg(fe->dev, "ASoC: prepare FE %s\n", fe->dai_link->name);
- fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE;
+ dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_FE);
/* there is no point preparing this FE if there are no BEs */
if (list_empty(&fe->dpcm[stream].be_clients)) {
@@ -1661,7 +1701,7 @@ static int dpcm_fe_dai_prepare(struct snd_pcm_substream *substream)
fe->dpcm[stream].state = SND_SOC_DPCM_STATE_PREPARE;
out:
- fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO;
+ dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO);
mutex_unlock(&fe->card->mutex);
return ret;
@@ -1808,11 +1848,11 @@ static int dpcm_run_new_update(struct snd_soc_pcm_runtime *fe, int stream)
{
int ret;
- fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_BE;
+ dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_BE);
ret = dpcm_run_update_startup(fe, stream);
if (ret < 0)
dev_err(fe->dev, "ASoC: failed to startup some BEs\n");
- fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO;
+ dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO);
return ret;
}
@@ -1821,11 +1861,11 @@ static int dpcm_run_old_update(struct snd_soc_pcm_runtime *fe, int stream)
{
int ret;
- fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_BE;
+ dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_BE);
ret = dpcm_run_update_shutdown(fe, stream);
if (ret < 0)
dev_err(fe->dev, "ASoC: failed to shutdown some BEs\n");
- fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO;
+ dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO);
return ret;
}
@@ -1882,6 +1922,7 @@ int soc_dpcm_runtime_update(struct snd_soc_card *card)
dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK);
}
+ dpcm_path_put(&list);
capture:
/* skip if FE doesn't have capture capability */
if (!fe->cpu_dai->driver->capture.channels_min)