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authorMark Brown <broonie@linaro.org>2014-08-04 16:31:05 +0100
committerMark Brown <broonie@linaro.org>2014-08-04 16:31:05 +0100
commitf0d766adbcac4eff4a114844b56d64aef1b8f5cd (patch)
treef8099e682afb9ef30d3770700c28b8cc8a042711 /sound
parent4a226ec97d797f4be486aab93458fc56a7d4de8c (diff)
parentd0ab92d63cd6df4c47d93940bd5e4e7737fa4909 (diff)
Merge tag 'asoc-v3.16-rc5' into asoc-linus
ASoC: Fixes for v3.16 A bigger batch of changes than I would like as I didn't send any for a few weeks without noticing how many had built up. They are almost all driver specific though, larger changes are: - Fixes to the newly added Baytrail/MAX98090 which look like some QA was missed on the microphone detection. - Deletion of some erroniously listed audio formats for Haswell. - Fix debugfs creation in the core so that we don't try to generate multiple directories with the same name, relatively large textually but simple to inspect by eye and test. - A couple of bugfixes for the rcar driver one of which which involves a bit of code motion to move initailisation of some hardware out of common paths into device specific ones. - Ensure both channels are powered up for mono outputs on Arizona devices, involving some simple data tables listing the outputs and a loop over them. - A couple of fixes to save and restore information on suspended and idle Samsung I2S controllers. # gpg: Signature made Tue 22 Jul 2014 00:52:53 BST using RSA key ID 7EA229BD # gpg: Good signature from "Mark Brown <broonie@sirena.org.uk>" # gpg: aka "Mark Brown <broonie@debian.org>" # gpg: aka "Mark Brown <broonie@kernel.org>" # gpg: aka "Mark Brown <broonie@tardis.ed.ac.uk>" # gpg: aka "Mark Brown <broonie@linaro.org>" # gpg: aka "Mark Brown <Mark.Brown@linaro.org>"
Diffstat (limited to 'sound')
-rw-r--r--sound/soc/blackfin/bf5xx-i2s-pcm.c8
-rw-r--r--sound/soc/codecs/adau1701.c6
-rw-r--r--sound/soc/codecs/arizona.c25
-rw-r--r--sound/soc/codecs/arizona.h1
-rw-r--r--sound/soc/codecs/cs42l56.c4
-rw-r--r--sound/soc/codecs/max98090.c2
-rw-r--r--sound/soc/codecs/sgtl5000.c11
-rw-r--r--sound/soc/codecs/tlv320aic3x.c2
-rw-r--r--sound/soc/codecs/wm5110.c1
-rw-r--r--sound/soc/codecs/wm_adsp.c2
-rw-r--r--sound/soc/davinci/Kconfig1
-rw-r--r--sound/soc/davinci/davinci-mcasp.c12
-rw-r--r--sound/soc/fsl/fsl_sai.c9
-rw-r--r--sound/soc/generic/simple-card.c13
-rw-r--r--sound/soc/intel/byt-max98090.c19
-rw-r--r--sound/soc/intel/sst-baytrail-pcm.c2
-rw-r--r--sound/soc/intel/sst-haswell-dsp.c13
-rw-r--r--sound/soc/intel/sst-haswell-pcm.c27
-rw-r--r--sound/soc/s6000/s6000-i2s.c4
-rw-r--r--sound/soc/samsung/i2s.c29
-rw-r--r--sound/soc/sh/rcar/core.c4
-rw-r--r--sound/soc/sh/rcar/gen.c33
-rw-r--r--sound/soc/soc-core.c28
-rw-r--r--sound/soc/soc-pcm.c1
24 files changed, 185 insertions, 72 deletions
diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c
index a3881c4381c9..bcf591373a7a 100644
--- a/sound/soc/blackfin/bf5xx-i2s-pcm.c
+++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c
@@ -290,19 +290,19 @@ 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 d71c59cf7bdd..370b742117ef 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/arizona.c b/sound/soc/codecs/arizona.c
index 29e198f57d4c..747c71e59c04 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -243,6 +243,31 @@ int arizona_init_spk(struct snd_soc_codec *codec)
}
EXPORT_SYMBOL_GPL(arizona_init_spk);
+static const struct snd_soc_dapm_route arizona_mono_routes[] = {
+ { "OUT1R", NULL, "OUT1L" },
+ { "OUT2R", NULL, "OUT2L" },
+ { "OUT3R", NULL, "OUT3L" },
+ { "OUT4R", NULL, "OUT4L" },
+ { "OUT5R", NULL, "OUT5L" },
+ { "OUT6R", NULL, "OUT6L" },
+};
+
+int arizona_init_mono(struct snd_soc_codec *codec)
+{
+ struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+ struct arizona *arizona = priv->arizona;
+ int i;
+
+ for (i = 0; i < ARIZONA_MAX_OUTPUT; ++i) {
+ if (arizona->pdata.out_mono[i])
+ snd_soc_dapm_add_routes(&codec->dapm,
+ &arizona_mono_routes[i], 1);
+ }
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(arizona_init_mono);
+
int arizona_init_gpio(struct snd_soc_codec *codec)
{
struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h
index 05ae17f5bca3..942cfb197b6d 100644
--- a/sound/soc/codecs/arizona.h
+++ b/sound/soc/codecs/arizona.h
@@ -249,6 +249,7 @@ extern int arizona_set_fll(struct arizona_fll *fll, int source,
extern int arizona_init_spk(struct snd_soc_codec *codec);
extern int arizona_init_gpio(struct snd_soc_codec *codec);
+extern int arizona_init_mono(struct snd_soc_codec *codec);
extern int arizona_init_dai(struct arizona_priv *priv, int dai);
diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c
index fdc4bd27b0df..8e68ef5de849 100644
--- a/sound/soc/codecs/cs42l56.c
+++ b/sound/soc/codecs/cs42l56.c
@@ -445,9 +445,9 @@ static const struct snd_kcontrol_new cs42l56_snd_controls[] = {
SOC_DOUBLE("ADC Boost Switch", CS42L56_GAIN_BIAS_CTL, 3, 2, 1, 1),
SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L56_HPA_VOLUME,
- CS42L56_HPA_VOLUME, 0, 0x44, 0x55, hl_tlv),
+ CS42L56_HPB_VOLUME, 0, 0x44, 0x55, hl_tlv),
SOC_DOUBLE_R_SX_TLV("LineOut Volume", CS42L56_LOA_VOLUME,
- CS42L56_LOA_VOLUME, 0, 0x44, 0x55, hl_tlv),
+ CS42L56_LOB_VOLUME, 0, 0x44, 0x55, hl_tlv),
SOC_SINGLE_TLV("Bass Shelving Volume", CS42L56_TONE_CTL,
0, 0x00, 1, tone_tlv),
diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c
index f5fccc7a8e89..d97f1ce7ff7d 100644
--- a/sound/soc/codecs/max98090.c
+++ b/sound/soc/codecs/max98090.c
@@ -2284,7 +2284,7 @@ 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) {
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index 3d39f0b5b4a8..8f4c73d17c87 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -1277,7 +1277,7 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec)
return ret;
}
- ret = devm_regulator_bulk_get(codec->dev, ARRAY_SIZE(sgtl5000->supplies),
+ ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(sgtl5000->supplies),
sgtl5000->supplies);
if (ret)
goto err_ldo_remove;
@@ -1285,13 +1285,16 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec)
ret = regulator_bulk_enable(ARRAY_SIZE(sgtl5000->supplies),
sgtl5000->supplies);
if (ret)
- goto err_ldo_remove;
+ goto err_regulator_free;
/* wait for all power rails bring up */
udelay(10);
return 0;
+err_regulator_free:
+ regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies),
+ sgtl5000->supplies);
err_ldo_remove:
if (!external_vddd)
ldo_regulator_remove(codec);
@@ -1361,6 +1364,8 @@ static int sgtl5000_probe(struct snd_soc_codec *codec)
err:
regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies),
sgtl5000->supplies);
+ regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies),
+ sgtl5000->supplies);
ldo_regulator_remove(codec);
return ret;
@@ -1374,6 +1379,8 @@ static int sgtl5000_remove(struct snd_soc_codec *codec)
regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies),
sgtl5000->supplies);
+ regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies),
+ sgtl5000->supplies);
ldo_regulator_remove(codec);
return 0;
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index e12fafbb1e09..5360772bc1ad 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -879,7 +879,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream,
case SNDRV_PCM_FORMAT_S20_3LE:
data |= (0x01 << 4);
break;
- case SNDRV_PCM_FORMAT_S24_LE:
+ case SNDRV_PCM_FORMAT_S24_3LE:
data |= (0x02 << 4);
break;
case SNDRV_PCM_FORMAT_S32_LE:
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index 2e5fcb559e90..62ef54456499 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -1596,6 +1596,7 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec)
arizona_init_spk(codec);
arizona_init_gpio(codec);
+ arizona_init_mono(codec);
ret = snd_soc_add_codec_controls(codec, wm_adsp2_fw_controls, 8);
if (ret != 0)
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index 060027182dcb..2537725dd53f 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -1758,3 +1758,5 @@ 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/Kconfig b/sound/soc/davinci/Kconfig
index 50a098749b9e..fdbb16fffd30 100644
--- a/sound/soc/davinci/Kconfig
+++ b/sound/soc/davinci/Kconfig
@@ -6,6 +6,7 @@ config SND_DAVINCI_SOC_I2S
tristate
config SND_DAVINCI_SOC_MCASP
+ depends on SND_DAVINCI_SOC || SND_OMAP_SOC
tristate
config SND_DAVINCI_SOC_VCIF
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 9afb14629a17..bfcc6c3dc2fd 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -720,6 +720,10 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
case SNDRV_PCM_FORMAT_U24_LE:
case SNDRV_PCM_FORMAT_S24_LE:
+ dma_params->data_type = 4;
+ word_length = 24;
+ break;
+
case SNDRV_PCM_FORMAT_U32_LE:
case SNDRV_PCM_FORMAT_S32_LE:
dma_params->data_type = 4;
@@ -1223,14 +1227,22 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
goto err;
switch (mcasp->version) {
+#if IS_BUILTIN(CONFIG_SND_DAVINCI_SOC) || \
+ (IS_MODULE(CONFIG_SND_DAVINCI_SOC_MCASP) && \
+ IS_MODULE(CONFIG_SND_DAVINCI_SOC))
case MCASP_VERSION_1:
case MCASP_VERSION_2:
case MCASP_VERSION_3:
ret = davinci_soc_platform_register(&pdev->dev);
break;
+#endif
+#if IS_BUILTIN(CONFIG_SND_OMAP_SOC) || \
+ (IS_MODULE(CONFIG_SND_DAVINCI_SOC_MCASP) && \
+ IS_MODULE(CONFIG_SND_OMAP_SOC))
case MCASP_VERSION_4:
ret = omap_pcm_platform_register(&pdev->dev);
break;
+#endif
default:
dev_err(&pdev->dev, "Invalid McASP version: %d\n",
mcasp->version);
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index c5a0e8af8226..1b6ee2ce849f 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -106,7 +106,7 @@ irq_rx:
xcsr &= ~FSL_SAI_CSR_xF_MASK;
if (flags)
- regmap_write(sai->regmap, FSL_SAI_TCSR, flags | xcsr);
+ regmap_write(sai->regmap, FSL_SAI_RCSR, flags | xcsr);
out:
if (irq_none)
@@ -371,10 +371,13 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd,
/* Check if the opposite FRDE is also disabled */
if (!(tx ? rcsr & FSL_SAI_CSR_FRDE : tcsr & FSL_SAI_CSR_FRDE)) {
+ /* Disable both directions and reset their FIFOs */
regmap_update_bits(sai->regmap, FSL_SAI_TCSR,
- FSL_SAI_CSR_TERE, 0);
+ FSL_SAI_CSR_TERE | FSL_SAI_CSR_FR,
+ FSL_SAI_CSR_FR);
regmap_update_bits(sai->regmap, FSL_SAI_RCSR,
- FSL_SAI_CSR_TERE, 0);
+ FSL_SAI_CSR_TERE | FSL_SAI_CSR_FR,
+ FSL_SAI_CSR_FR);
}
break;
default:
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 03a7fdcdf114..159e517fa09a 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -116,6 +116,7 @@ asoc_simple_card_sub_parse_of(struct device_node *np,
{
struct device_node *node;
struct clk *clk;
+ u32 val;
int ret;
/*
@@ -151,10 +152,8 @@ asoc_simple_card_sub_parse_of(struct device_node *np,
}
dai->sysclk = clk_get_rate(clk);
- } else if (of_property_read_bool(np, "system-clock-frequency")) {
- of_property_read_u32(np,
- "system-clock-frequency",
- &dai->sysclk);
+ } else if (!of_property_read_u32(np, "system-clock-frequency", &val)) {
+ dai->sysclk = val;
} else {
clk = of_clk_get(node, 0);
if (!IS_ERR(clk))
@@ -303,6 +302,7 @@ static int asoc_simple_card_parse_of(struct device_node *node,
{
struct snd_soc_dai_link *dai_link = priv->snd_card.dai_link;
struct simple_dai_props *dai_props = priv->dai_props;
+ u32 val;
int ret;
/* parsing the card name from DT */
@@ -325,8 +325,9 @@ static int asoc_simple_card_parse_of(struct device_node *node,
}
/* Factor to mclk, used in hw_params() */
- of_property_read_u32(node, "simple-audio-card,mclk-fs",
- &priv->mclk_fs);
+ ret = of_property_read_u32(node, "simple-audio-card,mclk-fs", &val);
+ if (ret == 0)
+ priv->mclk_fs = val;
dev_dbg(dev, "New simple-card: %s\n", priv->snd_card.name ?
priv->snd_card.name : "");
diff --git a/sound/soc/intel/byt-max98090.c b/sound/soc/intel/byt-max98090.c
index 5fc98c64a3f4..5cfb41ec3fab 100644
--- a/sound/soc/intel/byt-max98090.c
+++ b/sound/soc/intel/byt-max98090.c
@@ -39,8 +39,7 @@ static const struct snd_soc_dapm_widget byt_max98090_widgets[] = {
static const struct snd_soc_dapm_route byt_max98090_audio_map[] = {
{"IN34", NULL, "Headset Mic"},
- {"IN34", NULL, "MICBIAS"},
- {"MICBIAS", NULL, "Headset Mic"},
+ {"Headset Mic", NULL, "MICBIAS"},
{"DMICL", NULL, "Int Mic"},
{"Headphone", NULL, "HPL"},
{"Headphone", NULL, "HPR"},
@@ -84,7 +83,8 @@ static struct snd_soc_jack_gpio hs_jack_gpios[] = {
{
.name = "mic-gpio",
.idx = 1,
- .report = SND_JACK_MICROPHONE | SND_JACK_LINEIN,
+ .invert = 1,
+ .report = SND_JACK_MICROPHONE,
.debounce_time = 200,
},
};
@@ -108,7 +108,8 @@ static int byt_max98090_init(struct snd_soc_pcm_runtime *runtime)
}
/* Enable jack detection */
- ret = snd_soc_jack_new(codec, "Headphone", SND_JACK_HEADPHONE, jack);
+ ret = snd_soc_jack_new(codec, "Headset",
+ SND_JACK_LINEOUT | SND_JACK_HEADSET, jack);
if (ret)
return ret;
@@ -117,13 +118,9 @@ static int byt_max98090_init(struct snd_soc_pcm_runtime *runtime)
if (ret)
return ret;
- ret = snd_soc_jack_add_gpiods(card->dev->parent, jack,
- ARRAY_SIZE(hs_jack_gpios),
- hs_jack_gpios);
- if (ret)
- return ret;
-
- return max98090_mic_detect(codec, jack);
+ return snd_soc_jack_add_gpiods(card->dev->parent, jack,
+ ARRAY_SIZE(hs_jack_gpios),
+ hs_jack_gpios);
}
static struct snd_soc_dai_link byt_max98090_dais[] = {
diff --git a/sound/soc/intel/sst-baytrail-pcm.c b/sound/soc/intel/sst-baytrail-pcm.c
index 8eab97368ea7..599401c0c655 100644
--- a/sound/soc/intel/sst-baytrail-pcm.c
+++ b/sound/soc/intel/sst-baytrail-pcm.c
@@ -32,7 +32,7 @@ static const struct snd_pcm_hardware sst_byt_pcm_hardware = {
SNDRV_PCM_INFO_PAUSE |
SNDRV_PCM_INFO_RESUME,
.formats = SNDRV_PCM_FMTBIT_S16_LE |
- SNDRV_PCM_FORMAT_S24_LE,
+ SNDRV_PCM_FMTBIT_S24_LE,
.period_bytes_min = 384,
.period_bytes_max = 48000,
.periods_min = 2,
diff --git a/sound/soc/intel/sst-haswell-dsp.c b/sound/soc/intel/sst-haswell-dsp.c
index 535f517629fd..a33b931181dc 100644
--- a/sound/soc/intel/sst-haswell-dsp.c
+++ b/sound/soc/intel/sst-haswell-dsp.c
@@ -359,6 +359,17 @@ static u32 hsw_block_get_bit(struct sst_mem_block *block)
return bit;
}
+/*dummy read a SRAM block.*/
+static void sst_mem_block_dummy_read(struct sst_mem_block *block)
+{
+ u32 size;
+ u8 tmp_buf[4];
+ struct sst_dsp *sst = block->dsp;
+
+ size = block->size > 4 ? 4 : block->size;
+ memcpy_fromio(tmp_buf, sst->addr.lpe + block->offset, size);
+}
+
/* enable 32kB memory block - locks held by caller */
static int hsw_block_enable(struct sst_mem_block *block)
{
@@ -378,6 +389,8 @@ static int hsw_block_enable(struct sst_mem_block *block)
/* wait 18 DSP clock ticks */
udelay(10);
+ /*add a dummy read before the SRAM block is written, otherwise the writing may miss bytes sometimes.*/
+ sst_mem_block_dummy_read(block);
return 0;
}
diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c
index 058efb17c568..61bf6da4bb02 100644
--- a/sound/soc/intel/sst-haswell-pcm.c
+++ b/sound/soc/intel/sst-haswell-pcm.c
@@ -80,7 +80,7 @@ static const struct snd_pcm_hardware hsw_pcm_hardware = {
SNDRV_PCM_INFO_PAUSE |
SNDRV_PCM_INFO_RESUME |
SNDRV_PCM_INFO_NO_PERIOD_WAKEUP,
- .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE |
+ .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |
SNDRV_PCM_FMTBIT_S32_LE,
.period_bytes_min = PAGE_SIZE,
.period_bytes_max = (HSW_PCM_PERIODS_MAX / HSW_PCM_PERIODS_MIN) * PAGE_SIZE,
@@ -400,7 +400,15 @@ static int hsw_pcm_hw_params(struct snd_pcm_substream *substream,
sst_hsw_stream_set_valid(hsw, pcm_data->stream, 16);
break;
case SNDRV_PCM_FORMAT_S24_LE:
- bits = SST_HSW_DEPTH_24BIT;
+ bits = SST_HSW_DEPTH_32BIT;
+ sst_hsw_stream_set_valid(hsw, pcm_data->stream, 24);
+ break;
+ case SNDRV_PCM_FORMAT_S8:
+ bits = SST_HSW_DEPTH_8BIT;
+ sst_hsw_stream_set_valid(hsw, pcm_data->stream, 8);
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ bits = SST_HSW_DEPTH_32BIT;
sst_hsw_stream_set_valid(hsw, pcm_data->stream, 32);
break;
default:
@@ -685,8 +693,9 @@ static int hsw_pcm_new(struct snd_soc_pcm_runtime *rtd)
}
#define HSW_FORMATS \
- (SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S16_LE |\
- SNDRV_PCM_FMTBIT_S32_LE)
+ (SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S24_LE | \
+ SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S16_LE |\
+ SNDRV_PCM_FMTBIT_S8)
static struct snd_soc_dai_driver hsw_dais[] = {
{
@@ -696,7 +705,7 @@ static struct snd_soc_dai_driver hsw_dais[] = {
.channels_min = 2,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_48000,
- .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .formats = SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE,
},
},
{
@@ -727,8 +736,8 @@ static struct snd_soc_dai_driver hsw_dais[] = {
.stream_name = "Loopback Capture",
.channels_min = 2,
.channels_max = 2,
- .rates = SNDRV_PCM_RATE_8000_192000,
- .formats = HSW_FORMATS,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE,
},
},
{
@@ -737,8 +746,8 @@ static struct snd_soc_dai_driver hsw_dais[] = {
.stream_name = "Analog Capture",
.channels_min = 2,
.channels_max = 2,
- .rates = SNDRV_PCM_RATE_8000_192000,
- .formats = HSW_FORMATS,
+ .rates = SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE,
},
},
};
diff --git a/sound/soc/s6000/s6000-i2s.c b/sound/soc/s6000/s6000-i2s.c
index 7eba7979b9af..1c8d01166e5b 100644
--- a/sound/soc/s6000/s6000-i2s.c
+++ b/sound/soc/s6000/s6000-i2s.c
@@ -570,7 +570,7 @@ err_release_none:
return ret;
}
-static void s6000_i2s_remove(struct platform_device *pdev)
+static int s6000_i2s_remove(struct platform_device *pdev)
{
struct s6000_i2s_dev *dev = dev_get_drvdata(&pdev->dev);
struct resource *region;
@@ -597,6 +597,8 @@ static void s6000_i2s_remove(struct platform_device *pdev)
iounmap(mmio);
region = platform_get_resource(pdev, IORESOURCE_IO, 0);
release_mem_region(region->start, resource_size(region));
+
+ return 0;
}
static struct platform_driver s6000_i2s_driver = {
diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c
index 2ac76fa3e742..d2533dbc8399 100644
--- a/sound/soc/samsung/i2s.c
+++ b/sound/soc/samsung/i2s.c
@@ -68,6 +68,8 @@ struct i2s_dai {
#define DAI_OPENED (1 << 0) /* Dai is opened */
#define DAI_MANAGER (1 << 1) /* Dai is the manager */
unsigned mode;
+ /* CDCLK pin direction: 0 - input, 1 - output */
+ unsigned int cdclk_out:1;
/* Driver for this DAI */
struct snd_soc_dai_driver i2s_dai_drv;
/* DMA parameters */
@@ -737,6 +739,9 @@ static int i2s_startup(struct snd_pcm_substream *substream,
spin_unlock_irqrestore(&lock, flags);
+ if (!is_opened(other) && i2s->cdclk_out)
+ i2s_set_sysclk(dai, SAMSUNG_I2S_CDCLK,
+ 0, SND_SOC_CLOCK_OUT);
return 0;
}
@@ -752,9 +757,13 @@ static void i2s_shutdown(struct snd_pcm_substream *substream,
i2s->mode &= ~DAI_OPENED;
i2s->mode &= ~DAI_MANAGER;
- if (is_opened(other))
+ if (is_opened(other)) {
other->mode |= DAI_MANAGER;
-
+ } else {
+ u32 mod = readl(i2s->addr + I2SMOD);
+ i2s->cdclk_out = !(mod & MOD_CDCLKCON);
+ other->cdclk_out = i2s->cdclk_out;
+ }
/* Reset any constraint on RFS and BFS */
i2s->rfs = 0;
i2s->bfs = 0;
@@ -920,11 +929,9 @@ 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;
}
@@ -933,11 +940,9 @@ 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/rcar/core.c b/sound/soc/sh/rcar/core.c
index 4e86265f625c..ed76901f8202 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -297,7 +297,6 @@ static void rsnd_dma_of_name(struct rsnd_dma *dma,
for (i = 1; i < MOD_MAX; i++) {
if (!src) {
mod[i] = ssi;
- break;
} else if (!dvc) {
mod[i] = src;
src = NULL;
@@ -308,6 +307,9 @@ static void rsnd_dma_of_name(struct rsnd_dma *dma,
if (mod[i] == this)
index = i;
+
+ if (mod[i] == ssi)
+ break;
}
if (is_play) {
diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c
index 1dd2b7d38c2c..0280a11c0899 100644
--- a/sound/soc/sh/rcar/gen.c
+++ b/sound/soc/sh/rcar/gen.c
@@ -184,7 +184,7 @@ static int rsnd_gen_regmap_init(struct rsnd_priv *priv,
#define RDMA_CMD_O_N(addr, i) (addr ##_reg - 0x004f8000 + (0x400 * i))
#define RDMA_CMD_O_P(addr, i) (addr ##_reg - 0x001f8000 + (0x400 * i))
-void rsnd_gen_dma_addr(struct rsnd_priv *priv,
+static void rsnd_gen2_dma_addr(struct rsnd_priv *priv,
struct rsnd_dma *dma,
struct dma_slave_config *cfg,
int is_play, int slave_id)
@@ -226,17 +226,6 @@ void rsnd_gen_dma_addr(struct rsnd_priv *priv,
}
};
- cfg->slave_id = slave_id;
- cfg->src_addr = 0;
- cfg->dst_addr = 0;
- cfg->direction = is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM;
-
- /*
- * gen1 uses default DMA addr
- */
- if (rsnd_is_gen1(priv))
- return;
-
/* it shouldn't happen */
if (use_dvc & !use_src) {
dev_err(dev, "DVC is selected without SRC\n");
@@ -250,6 +239,26 @@ void rsnd_gen_dma_addr(struct rsnd_priv *priv,
id, cfg->src_addr, cfg->dst_addr);
}
+void rsnd_gen_dma_addr(struct rsnd_priv *priv,
+ struct rsnd_dma *dma,
+ struct dma_slave_config *cfg,
+ int is_play, int slave_id)
+{
+ cfg->slave_id = slave_id;
+ cfg->src_addr = 0;
+ cfg->dst_addr = 0;
+ cfg->direction = is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM;
+
+ /*
+ * gen1 uses default DMA addr
+ */
+ if (rsnd_is_gen1(priv))
+ return;
+
+ rsnd_gen2_dma_addr(priv, dma, cfg, is_play, slave_id);
+}
+
+
/*
* Gen2
*/
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index b87d7d882e6d..91120b8e283e 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -270,12 +270,32 @@ static const struct file_operations codec_reg_fops = {
.llseek = default_llseek,
};
+static struct dentry *soc_debugfs_create_dir(struct dentry *parent,
+ const char *fmt, ...)
+{
+ struct dentry *de;
+ va_list ap;
+ char *s;
+
+ va_start(ap, fmt);
+ s = kvasprintf(GFP_KERNEL, fmt, ap);
+ va_end(ap);
+
+ if (!s)
+ return NULL;
+
+ de = debugfs_create_dir(s, parent);
+ kfree(s);
+
+ return de;
+}
+
static void soc_init_codec_debugfs(struct snd_soc_codec *codec)
{
struct dentry *debugfs_card_root = codec->card->debugfs_card_root;
- codec->debugfs_codec_root = debugfs_create_dir(codec->name,
- debugfs_card_root);
+ codec->debugfs_codec_root = soc_debugfs_create_dir(debugfs_card_root,
+ "codec:%s", codec->name);
if (!codec->debugfs_codec_root) {
dev_warn(codec->dev,
"ASoC: Failed to create codec debugfs directory\n");
@@ -306,8 +326,8 @@ static void soc_init_platform_debugfs(struct snd_soc_platform *platform)
{
struct dentry *debugfs_card_root = platform->card->debugfs_card_root;
- platform->debugfs_platform_root = debugfs_create_dir(platform->name,
- debugfs_card_root);
+ platform->debugfs_platform_root = soc_debugfs_create_dir(debugfs_card_root,
+ "platform:%s", platform->name);
if (!platform->debugfs_platform_root) {
dev_warn(platform->dev,
"ASoC: Failed to create platform debugfs directory\n");
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 54d18f22a33e..4ea656770d65 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2069,6 +2069,7 @@ 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)