Merge tag 'sound-3.19-rc7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"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"
* tag 'sound-3.19-rc7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: seq-dummy: remove deadlock-causing events on close
ASoC: omap-mcbsp: Correct CBM_CFS dai format configuration
ASoC: soc-compress.c: fix NULL dereference
ASoC: rt286: set the same format for dac and adc
ASoC: wm8904: fix runtime warning
ASoC: simple-card: Fix crash in asoc_simple_card_unref()
ASoC: fsl: imx-wm8962: Set the card owner field
ASoC: pcm512x: Fix DSP program selection
ASoC: rt5677: Modify the behavior that updates the PLL parameter.
ASoC: fsl_ssi: Fix irq error check
ASoC: rockchip: i2s: applys rate symmetry for CPU DAI
ASoC: Intel: Add NULL checks for the stream pointer
ASoC: wm8960: Fix capture sample rate from 11250 to 11025
ASoC: adi: Add missing return statement.
ASoC: Intel: Don't change offset of block allocator during fixed allocate
ASoC: ts3a227e: Check and report jack status at probe
ASoC: fsl_esai: Fix incorrect xDC field width of xCCR registers
diff --git a/sound/core/seq/seq_dummy.c b/sound/core/seq/seq_dummy.c
index ec667f1..5d905d9 100644
--- a/sound/core/seq/seq_dummy.c
+++ b/sound/core/seq/seq_dummy.c
@@ -82,36 +82,6 @@
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 @@
| 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 7752860..4c23381 100644
--- a/sound/soc/adi/axi-i2s.c
+++ b/sound/soc/adi/axi-i2s.c
@@ -240,6 +240,8 @@
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 e5f2fb8..30c673c 100644
--- a/sound/soc/codecs/pcm512x.c
+++ b/sound/soc/codecs/pcm512x.c
@@ -188,8 +188,8 @@
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 2cd4fe4..1d1c7f8 100644
--- a/sound/soc/codecs/rt286.c
+++ b/sound/soc/codecs/rt286.c
@@ -861,10 +861,8 @@
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 c0fbe18..918ada9 100644
--- a/sound/soc/codecs/rt5677.c
+++ b/sound/soc/codecs/rt5677.c
@@ -2083,10 +2083,14 @@
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 @@
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 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 1d12057..9f2dced 100644
--- a/sound/soc/codecs/ts3a227e.c
+++ b/sound/soc/codecs/ts3a227e.c
@@ -254,6 +254,7 @@
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 @@
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 4d2d2b1..75b87c5 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -1076,10 +1076,13 @@
{ "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 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 031a1ae..a96eb49 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -556,7 +556,7 @@
{ 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 91a550f..5e793bb 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 a65f17d..059496e 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -1362,9 +1362,9 @@
}
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 4caacb0..cd146d4 100644
--- a/sound/soc/fsl/imx-wm8962.c
+++ b/sound/soc/fsl/imx-wm8962.c
@@ -257,6 +257,7 @@
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 fb9240f..7fe3009 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -452,9 +452,8 @@
}
/* 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 @@
return ret;
err:
- asoc_simple_card_unref(pdev);
+ asoc_simple_card_unref(&priv->snd_card);
return ret;
}
@@ -572,7 +571,7 @@
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 ef2e8b5..b3f9489 100644
--- a/sound/soc/intel/sst-firmware.c
+++ b/sound/soc/intel/sst-firmware.c
@@ -706,6 +706,7 @@
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 @@
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 @@
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 3f8c482..5bf1404 100644
--- a/sound/soc/intel/sst-haswell-ipc.c
+++ b/sound/soc/intel/sst-haswell-ipc.c
@@ -1228,6 +1228,11 @@
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 @@
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 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 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 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 8b79cafa..c7eb9dd 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -434,7 +434,7 @@
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 13d8507..dcc26ed 100644
--- a/sound/soc/rockchip/rockchip_i2s.c
+++ b/sound/soc/rockchip/rockchip_i2s.c
@@ -335,6 +335,7 @@
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 590a82f..025c38f 100644
--- a/sound/soc/soc-compress.c
+++ b/sound/soc/soc-compress.c
@@ -659,7 +659,8 @@
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 @@
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));