[GIT PULL] sound fixes for 3.19-rc7

From: Takashi Iwai
Date: Wed Jan 28 2015 - 16:35:00 EST

Linus,

please pull sound fixes for v3.19-rc7 from:

git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git tags/sound-3.19-rc7

The topmost commit is 0767e95bb96d7fdddcd590fb809e6975d93aebc5

----------------------------------------------------------------

sound fixes for 3.19-rc7

This batch ended up being larger than wished, but there is nothing to
worry too much there. Most of commits are for ASoC, a compress NULL
dereference fix, a fix for probe error handling, and the rest are
device-specific fixes. In addition, we have a fix for a long-standing
but of seq-dummy driver, which just cuts off the buggy part in the
end.

----------------------------------------------------------------

Andrew Jackson (1):
ASoC: adi: Add missing return statement.

Aurelien BOUIN (1):
ASoC: fsl_esai: Fix incorrect xDC field width of xCCR registers

Bard Liao (1):
ASoC: rt286: set the same format for dac and adc

Bo Shen (1):
ASoC: wm8904: fix runtime warning

Cheng-Yi Chiang (1):
ASoC: ts3a227e: Check and report jack status at probe

Clemens Ladisch (1):
ALSA: seq-dummy: remove deadlock-causing events on close

Fabio Estevam (2):
ASoC: fsl_ssi: Fix irq error check
ASoC: fsl: imx-wm8962: Set the card owner field

Geert Uytterhoeven (1):
ASoC: simple-card: Fix crash in asoc_simple_card_unref()

Jianqun Xu (1):
ASoC: rockchip: i2s: applys rate symmetry for CPU DAI

Jie Yang (2):
ASoC: Intel: Don't change offset of block allocator during fixed allocate
ASoC: Intel: Add NULL checks for the stream pointer

Oder Chiou (1):
ASoC: rt5677: Modify the behavior that updates the PLL parameter.

Peter Rosin (1):
ASoC: pcm512x: Fix DSP program selection

Peter Ujfalusi (1):
ASoC: omap-mcbsp: Correct CBM_CFS dai format configuration

Qais Yousef (1):
ASoC: soc-compress.c: fix NULL dereference

Zidan Wang (1):
ASoC: wm8960: Fix capture sample rate from 11250 to 11025

---
sound/core/seq/seq_dummy.c | 31 -------------------------------
sound/soc/adi/axi-i2s.c | 2 ++
sound/soc/codecs/pcm512x.c | 2 +-
sound/soc/codecs/rt286.c | 6 ++----
sound/soc/codecs/rt5677.c | 18 ++++++++++++++----
sound/soc/codecs/ts3a227e.c | 6 ++++++
sound/soc/codecs/wm8904.c | 23 +++++++++++++++--------
sound/soc/codecs/wm8960.c | 2 +-
sound/soc/fsl/fsl_esai.h | 2 +-
sound/soc/fsl/fsl_ssi.c | 4 ++--
sound/soc/fsl/imx-wm8962.c | 1 +
sound/soc/generic/simple-card.c | 7 +++----
sound/soc/intel/sst-firmware.c | 13 +++++++------
sound/soc/intel/sst-haswell-ipc.c | 30 ++++++++++++++++++++++++++++++
sound/soc/omap/omap-mcbsp.c | 2 +-
sound/soc/rockchip/rockchip_i2s.c | 1 +
sound/soc/soc-compress.c | 9 ++++++---
17 files changed, 93 insertions(+), 66 deletions(-)

diff --git a/sound/core/seq/seq_dummy.c b/sound/core/seq/seq_dummy.c
index ec667f158f19..5d905d90d504 100644
--- a/sound/core/seq/seq_dummy.c
+++ b/sound/core/seq/seq_dummy.c
@@ -82,36 +82,6 @@ struct snd_seq_dummy_port {
static int my_client = -1;

/*
- * unuse callback - send ALL_SOUNDS_OFF and RESET_CONTROLLERS events
- * to subscribers.
- * Note: this callback is called only after all subscribers are removed.
- */
-static int
-dummy_unuse(void *private_data, struct snd_seq_port_subscribe *info)
-{
- struct snd_seq_dummy_port *p;
- int i;
- struct snd_seq_event ev;
-
- p = private_data;
- memset(&ev, 0, sizeof(ev));
- if (p->duplex)
- ev.source.port = p->connect;
- else
- ev.source.port = p->port;
- ev.dest.client = SNDRV_SEQ_ADDRESS_SUBSCRIBERS;
- ev.type = SNDRV_SEQ_EVENT_CONTROLLER;
- for (i = 0; i < 16; i++) {
- ev.data.control.channel = i;
- ev.data.control.param = MIDI_CTL_ALL_SOUNDS_OFF;
- snd_seq_kernel_client_dispatch(p->client, &ev, 0, 0);
- ev.data.control.param = MIDI_CTL_RESET_CONTROLLERS;
- snd_seq_kernel_client_dispatch(p->client, &ev, 0, 0);
- }
- return 0;
-}
-
-/*
* event input callback - just redirect events to subscribers
*/
static int
@@ -175,7 +145,6 @@ create_port(int idx, int type)
| SNDRV_SEQ_PORT_TYPE_PORT;
memset(&pcb, 0, sizeof(pcb));
pcb.owner = THIS_MODULE;
- pcb.unuse = dummy_unuse;
pcb.event_input = dummy_input;
pcb.private_free = dummy_free;
pcb.private_data = rec;
diff --git a/sound/soc/adi/axi-i2s.c b/sound/soc/adi/axi-i2s.c
index 7752860f7230..4c23381727a1 100644
--- a/sound/soc/adi/axi-i2s.c
+++ b/sound/soc/adi/axi-i2s.c
@@ -240,6 +240,8 @@ static int axi_i2s_probe(struct platform_device *pdev)
if (ret)
goto err_clk_disable;

+ return 0;
+
err_clk_disable:
clk_disable_unprepare(i2s->clk);
return ret;
diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c
index e5f2fb884bf3..30c673cdc12e 100644
--- a/sound/soc/codecs/pcm512x.c
+++ b/sound/soc/codecs/pcm512x.c
@@ -188,8 +188,8 @@ static const DECLARE_TLV_DB_SCALE(boost_tlv, 0, 80, 0);
static const char * const pcm512x_dsp_program_texts[] = {
"FIR interpolation with de-emphasis",
"Low latency IIR with de-emphasis",
- "Fixed process flow",
"High attenuation with de-emphasis",
+ "Fixed process flow",
"Ringing-less low latency FIR",
};

diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c
index 2cd4fe463102..1d1c7f8a9af2 100644
--- a/sound/soc/codecs/rt286.c
+++ b/sound/soc/codecs/rt286.c
@@ -861,10 +861,8 @@ static int rt286_hw_params(struct snd_pcm_substream *substream,
RT286_I2S_CTRL1, 0x0018, d_len_code << 3);
dev_dbg(codec->dev, "format val = 0x%x\n", val);

- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- snd_soc_update_bits(codec, RT286_DAC_FORMAT, 0x407f, val);
- else
- snd_soc_update_bits(codec, RT286_ADC_FORMAT, 0x407f, val);
+ snd_soc_update_bits(codec, RT286_DAC_FORMAT, 0x407f, val);
+ snd_soc_update_bits(codec, RT286_ADC_FORMAT, 0x407f, val);

return 0;
}
diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c
index c0fbe1881439..918ada9738b0 100644
--- a/sound/soc/codecs/rt5677.c
+++ b/sound/soc/codecs/rt5677.c
@@ -2083,10 +2083,14 @@ static int rt5677_set_pll1_event(struct snd_soc_dapm_widget *w,
struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec);

switch (event) {
- case SND_SOC_DAPM_POST_PMU:
+ case SND_SOC_DAPM_PRE_PMU:
regmap_update_bits(rt5677->regmap, RT5677_PLL1_CTRL2, 0x2, 0x2);
+ break;
+
+ case SND_SOC_DAPM_POST_PMU:
regmap_update_bits(rt5677->regmap, RT5677_PLL1_CTRL2, 0x2, 0x0);
break;
+
default:
return 0;
}
@@ -2101,10 +2105,14 @@ static int rt5677_set_pll2_event(struct snd_soc_dapm_widget *w,
struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec);

switch (event) {
- case SND_SOC_DAPM_POST_PMU:
+ case SND_SOC_DAPM_PRE_PMU:
regmap_update_bits(rt5677->regmap, RT5677_PLL2_CTRL2, 0x2, 0x2);
+ break;
+
+ case SND_SOC_DAPM_POST_PMU:
regmap_update_bits(rt5677->regmap, RT5677_PLL2_CTRL2, 0x2, 0x0);
break;
+
default:
return 0;
}
@@ -2212,9 +2220,11 @@ static int rt5677_vref_event(struct snd_soc_dapm_widget *w,

static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = {
SND_SOC_DAPM_SUPPLY("PLL1", RT5677_PWR_ANLG2, RT5677_PWR_PLL1_BIT,
- 0, rt5677_set_pll1_event, SND_SOC_DAPM_POST_PMU),
+ 0, rt5677_set_pll1_event, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_SUPPLY("PLL2", RT5677_PWR_ANLG2, RT5677_PWR_PLL2_BIT,
- 0, rt5677_set_pll2_event, SND_SOC_DAPM_POST_PMU),
+ 0, rt5677_set_pll2_event, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMU),

/* Input Side */
/* micbias */
diff --git a/sound/soc/codecs/ts3a227e.c b/sound/soc/codecs/ts3a227e.c
index 1d1205702d23..9f2dced046de 100644
--- a/sound/soc/codecs/ts3a227e.c
+++ b/sound/soc/codecs/ts3a227e.c
@@ -254,6 +254,7 @@ static int ts3a227e_i2c_probe(struct i2c_client *i2c,
struct ts3a227e *ts3a227e;
struct device *dev = &i2c->dev;
int ret;
+ unsigned int acc_reg;

ts3a227e = devm_kzalloc(&i2c->dev, sizeof(*ts3a227e), GFP_KERNEL);
if (ts3a227e == NULL)
@@ -283,6 +284,11 @@ static int ts3a227e_i2c_probe(struct i2c_client *i2c,
INTB_DISABLE | ADC_COMPLETE_INT_DISABLE,
ADC_COMPLETE_INT_DISABLE);

+ /* Read jack status because chip might not trigger interrupt at boot. */
+ regmap_read(ts3a227e->regmap, TS3A227E_REG_ACCESSORY_STATUS, &acc_reg);
+ ts3a227e_new_jack_state(ts3a227e, acc_reg);
+ ts3a227e_jack_report(ts3a227e);
+
return 0;
}

diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 4d2d2b1380d5..75b87c5c0f04 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -1076,10 +1076,13 @@ static const struct snd_soc_dapm_route adc_intercon[] = {
{ "Right Capture PGA", NULL, "Right Capture Mux" },
{ "Right Capture PGA", NULL, "Right Capture Inverting Mux" },

- { "AIFOUTL", "Left", "ADCL" },
- { "AIFOUTL", "Right", "ADCR" },
- { "AIFOUTR", "Left", "ADCL" },
- { "AIFOUTR", "Right", "ADCR" },
+ { "AIFOUTL Mux", "Left", "ADCL" },
+ { "AIFOUTL Mux", "Right", "ADCR" },
+ { "AIFOUTR Mux", "Left", "ADCL" },
+ { "AIFOUTR Mux", "Right", "ADCR" },
+
+ { "AIFOUTL", NULL, "AIFOUTL Mux" },
+ { "AIFOUTR", NULL, "AIFOUTR Mux" },

{ "ADCL", NULL, "CLK_DSP" },
{ "ADCL", NULL, "Left Capture PGA" },
@@ -1089,12 +1092,16 @@ static const struct snd_soc_dapm_route adc_intercon[] = {
};

static const struct snd_soc_dapm_route dac_intercon[] = {
- { "DACL", "Right", "AIFINR" },
- { "DACL", "Left", "AIFINL" },
+ { "DACL Mux", "Left", "AIFINL" },
+ { "DACL Mux", "Right", "AIFINR" },
+
+ { "DACR Mux", "Left", "AIFINL" },
+ { "DACR Mux", "Right", "AIFINR" },
+
+ { "DACL", NULL, "DACL Mux" },
{ "DACL", NULL, "CLK_DSP" },

- { "DACR", "Right", "AIFINR" },
- { "DACR", "Left", "AIFINL" },
+ { "DACR", NULL, "DACR Mux" },
{ "DACR", NULL, "CLK_DSP" },

{ "Charge pump", NULL, "SYSCLK" },
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index 031a1ae71d94..a96eb497a379 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -556,7 +556,7 @@ static struct {
{ 22050, 2 },
{ 24000, 2 },
{ 16000, 3 },
- { 11250, 4 },
+ { 11025, 4 },
{ 12000, 4 },
{ 8000, 5 },
};
diff --git a/sound/soc/fsl/fsl_esai.h b/sound/soc/fsl/fsl_esai.h
index 91a550f4a10d..5e793bbb6b02 100644
--- a/sound/soc/fsl/fsl_esai.h
+++ b/sound/soc/fsl/fsl_esai.h
@@ -302,7 +302,7 @@
#define ESAI_xCCR_xFP_MASK (((1 << ESAI_xCCR_xFP_WIDTH) - 1) << ESAI_xCCR_xFP_SHIFT)
#define ESAI_xCCR_xFP(v) ((((v) - 1) << ESAI_xCCR_xFP_SHIFT) & ESAI_xCCR_xFP_MASK)
#define ESAI_xCCR_xDC_SHIFT 9
-#define ESAI_xCCR_xDC_WIDTH 4
+#define ESAI_xCCR_xDC_WIDTH 5
#define ESAI_xCCR_xDC_MASK (((1 << ESAI_xCCR_xDC_WIDTH) - 1) << ESAI_xCCR_xDC_SHIFT)
#define ESAI_xCCR_xDC(v) ((((v) - 1) << ESAI_xCCR_xDC_SHIFT) & ESAI_xCCR_xDC_MASK)
#define ESAI_xCCR_xPSR_SHIFT 8
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index a65f17d57ffb..059496ed9ad7 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -1362,9 +1362,9 @@ static int fsl_ssi_probe(struct platform_device *pdev)
}

ssi_private->irq = platform_get_irq(pdev, 0);
- if (!ssi_private->irq) {
+ if (ssi_private->irq < 0) {
dev_err(&pdev->dev, "no irq for node %s\n", np->full_name);
- return -ENXIO;
+ return ssi_private->irq;
}

/* Are the RX and the TX clocks locked? */
diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c
index 4caacb05a623..cd146d4fa805 100644
--- a/sound/soc/fsl/imx-wm8962.c
+++ b/sound/soc/fsl/imx-wm8962.c
@@ -257,6 +257,7 @@ static int imx_wm8962_probe(struct platform_device *pdev)
if (ret)
goto clk_fail;
data->card.num_links = 1;
+ data->card.owner = THIS_MODULE;
data->card.dai_link = &data->dai;
data->card.dapm_widgets = imx_wm8962_dapm_widgets;
data->card.num_dapm_widgets = ARRAY_SIZE(imx_wm8962_dapm_widgets);
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index fb9240fdc9b7..7fe3009b1c43 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -452,9 +452,8 @@ static int asoc_simple_card_parse_of(struct device_node *node,
}

/* Decrease the reference count of the device nodes */
-static int asoc_simple_card_unref(struct platform_device *pdev)
+static int asoc_simple_card_unref(struct snd_soc_card *card)
{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
struct snd_soc_dai_link *dai_link;
int num_links;

@@ -556,7 +555,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev)
return ret;

err:
- asoc_simple_card_unref(pdev);
+ asoc_simple_card_unref(&priv->snd_card);
return ret;
}

@@ -572,7 +571,7 @@ static int asoc_simple_card_remove(struct platform_device *pdev)
snd_soc_jack_free_gpios(&simple_card_mic_jack, 1,
&simple_card_mic_jack_gpio);

- return asoc_simple_card_unref(pdev);
+ return asoc_simple_card_unref(card);
}

static const struct of_device_id asoc_simple_of_match[] = {
diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c
index ef2e8b5766a1..b3f9489794a6 100644
--- a/sound/soc/intel/sst-firmware.c
+++ b/sound/soc/intel/sst-firmware.c
@@ -706,6 +706,7 @@ static int block_alloc_fixed(struct sst_dsp *dsp, struct sst_block_allocator *ba
struct list_head *block_list)
{
struct sst_mem_block *block, *tmp;
+ struct sst_block_allocator ba_tmp = *ba;
u32 end = ba->offset + ba->size, block_end;
int err;

@@ -730,9 +731,9 @@ static int block_alloc_fixed(struct sst_dsp *dsp, struct sst_block_allocator *ba
if (ba->offset >= block->offset && ba->offset < block_end) {

/* align ba to block boundary */
- ba->size -= block_end - ba->offset;
- ba->offset = block_end;
- err = block_alloc_contiguous(dsp, ba, block_list);
+ ba_tmp.size -= block_end - ba->offset;
+ ba_tmp.offset = block_end;
+ err = block_alloc_contiguous(dsp, &ba_tmp, block_list);
if (err < 0)
return -ENOMEM;

@@ -767,10 +768,10 @@ static int block_alloc_fixed(struct sst_dsp *dsp, struct sst_block_allocator *ba
list_move(&block->list, &dsp->used_block_list);
list_add(&block->module_list, block_list);
/* align ba to block boundary */
- ba->size -= block_end - ba->offset;
- ba->offset = block_end;
+ ba_tmp.size -= block_end - ba->offset;
+ ba_tmp.offset = block_end;

- err = block_alloc_contiguous(dsp, ba, block_list);
+ err = block_alloc_contiguous(dsp, &ba_tmp, block_list);
if (err < 0)
return -ENOMEM;

diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c
index 3f8c48231364..5bf14040c24a 100644
--- a/sound/soc/intel/sst-haswell-ipc.c
+++ b/sound/soc/intel/sst-haswell-ipc.c
@@ -1228,6 +1228,11 @@ int sst_hsw_stream_free(struct sst_hsw *hsw, struct sst_hsw_stream *stream)
struct sst_dsp *sst = hsw->dsp;
unsigned long flags;

+ if (!stream) {
+ dev_warn(hsw->dev, "warning: stream is NULL, no stream to free, ignore it.\n");
+ return 0;
+ }
+
/* dont free DSP streams that are not commited */
if (!stream->commited)
goto out;
@@ -1415,6 +1420,16 @@ int sst_hsw_stream_commit(struct sst_hsw *hsw, struct sst_hsw_stream *stream)
u32 header;
int ret;

+ if (!stream) {
+ dev_warn(hsw->dev, "warning: stream is NULL, no stream to commit, ignore it.\n");
+ return 0;
+ }
+
+ if (stream->commited) {
+ dev_warn(hsw->dev, "warning: stream is already committed, ignore it.\n");
+ return 0;
+ }
+
trace_ipc_request("stream alloc", stream->host_id);

header = IPC_GLB_TYPE(IPC_GLB_ALLOCATE_STREAM);
@@ -1519,6 +1534,11 @@ int sst_hsw_stream_pause(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
{
int ret;

+ if (!stream) {
+ dev_warn(hsw->dev, "warning: stream is NULL, no stream to pause, ignore it.\n");
+ return 0;
+ }
+
trace_ipc_request("stream pause", stream->reply.stream_hw_id);

ret = sst_hsw_stream_operations(hsw, IPC_STR_PAUSE,
@@ -1535,6 +1555,11 @@ int sst_hsw_stream_resume(struct sst_hsw *hsw, struct sst_hsw_stream *stream,
{
int ret;

+ if (!stream) {
+ dev_warn(hsw->dev, "warning: stream is NULL, no stream to resume, ignore it.\n");
+ return 0;
+ }
+
trace_ipc_request("stream resume", stream->reply.stream_hw_id);

ret = sst_hsw_stream_operations(hsw, IPC_STR_RESUME,
@@ -1550,6 +1575,11 @@ int sst_hsw_stream_reset(struct sst_hsw *hsw, struct sst_hsw_stream *stream)
{
int ret, tries = 10;

+ if (!stream) {
+ dev_warn(hsw->dev, "warning: stream is NULL, no stream to reset, ignore it.\n");
+ return 0;
+ }
+
/* dont reset streams that are not commited */
if (!stream->commited)
return 0;
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 8b79cafab1e2..c7eb9dd67f60 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -434,7 +434,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
case SND_SOC_DAIFMT_CBM_CFS:
/* McBSP slave. FS clock as output */
regs->srgr2 |= FSGM;
- regs->pcr0 |= FSXM;
+ regs->pcr0 |= FSXM | FSRM;
break;
case SND_SOC_DAIFMT_CBM_CFM:
/* McBSP slave */
diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c
index 13d8507333b8..dcc26eda0539 100644
--- a/sound/soc/rockchip/rockchip_i2s.c
+++ b/sound/soc/rockchip/rockchip_i2s.c
@@ -335,6 +335,7 @@ static struct snd_soc_dai_driver rockchip_i2s_dai = {
SNDRV_PCM_FMTBIT_S24_LE),
},
.ops = &rockchip_i2s_dai_ops,
+ .symmetric_rates = 1,
};

static const struct snd_soc_component_driver rockchip_i2s_component = {
diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c
index 590a82f01d0b..025c38fbe3c0 100644
--- a/sound/soc/soc-compress.c
+++ b/sound/soc/soc-compress.c
@@ -659,7 +659,8 @@ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num)
rtd->dai_link->stream_name);

ret = snd_pcm_new_internal(rtd->card->snd_card, new_name, num,
- 1, 0, &be_pcm);
+ rtd->dai_link->dpcm_playback,
+ rtd->dai_link->dpcm_capture, &be_pcm);
if (ret < 0) {
dev_err(rtd->card->dev, "ASoC: can't create compressed for %s\n",
rtd->dai_link->name);
@@ -668,8 +669,10 @@ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num)

rtd->pcm = be_pcm;
rtd->fe_compr = 1;
- be_pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->private_data = rtd;
- be_pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream->private_data = rtd;
+ if (rtd->dai_link->dpcm_playback)
+ be_pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->private_data = rtd;
+ else if (rtd->dai_link->dpcm_capture)
+ be_pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream->private_data = rtd;
memcpy(compr->ops, &soc_compr_dyn_ops, sizeof(soc_compr_dyn_ops));
} else
memcpy(compr->ops, &soc_compr_ops, sizeof(soc_compr_ops));
--
To unsubscribe from this list: send the line "unsubscribe linux-kernel" in
the body of a message to majordomo@xxxxxxxxxxxxxxx
More majordomo info at http://vger.kernel.org/majordomo-info.html
Please read the FAQ at http://www.tux.org/lkml/