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authorTakashi Iwai <tiwai@suse.de>2019-08-06 12:46:26 +0200
committerTakashi Iwai <tiwai@suse.de>2019-08-06 12:46:35 +0200
commitdae02c8c64937532466d95fdb6ead6ada8000673 (patch)
tree1aa28cc48c37996a7e75aceb5f2430b7f0305127 /sound
parent74f94e41825549638a5b24410c1f62d5ea57134e (diff)
parenta67060201b746a308b1674f66bf289c9faef6d09 (diff)
Merge branch 'for-linus' into for-next
Pull 5.3 development branch for further fixes of USB-audio stuff. Signed-off-by: Takashi Iwai <tiwai@suse.de>
Diffstat (limited to 'sound')
-rw-r--r--sound/ac97/bus.c13
-rw-r--r--sound/core/compress_offload.c60
-rw-r--r--sound/core/pcm_native.c3
-rw-r--r--sound/hda/hdac_i915.c10
-rw-r--r--sound/pci/hda/hda_intel.c5
-rw-r--r--sound/pci/hda/patch_conexant.c1
-rw-r--r--sound/soc/amd/raven/acp3x-pcm-dma.c20
-rw-r--r--sound/soc/codecs/cs42xx8.c116
-rw-r--r--sound/soc/codecs/max98357a.c25
-rw-r--r--sound/soc/codecs/max98373.c6
-rw-r--r--sound/soc/codecs/max98373.h2
-rw-r--r--sound/soc/codecs/pcm3060-i2c.c4
-rw-r--r--sound/soc/codecs/pcm3060-spi.c4
-rw-r--r--sound/soc/codecs/pcm3060.c4
-rw-r--r--sound/soc/codecs/pcm3060.h2
-rw-r--r--sound/soc/codecs/rt1011.c4
-rw-r--r--[-rwxr-xr-x]sound/soc/codecs/rt1308.c0
-rw-r--r--[-rwxr-xr-x]sound/soc/codecs/rt1308.h0
-rw-r--r--sound/soc/generic/audio-graph-card.c30
-rw-r--r--sound/soc/generic/simple-card-utils.c7
-rw-r--r--sound/soc/generic/simple-card.c26
-rw-r--r--sound/soc/intel/boards/bytcht_es8316.c8
-rw-r--r--sound/soc/intel/common/soc-acpi-intel-bxt-match.c2
-rw-r--r--sound/soc/intel/common/soc-acpi-intel-byt-match.c2
-rw-r--r--sound/soc/intel/common/soc-acpi-intel-cht-match.c2
-rw-r--r--sound/soc/intel/common/soc-acpi-intel-cnl-match.c2
-rw-r--r--sound/soc/intel/common/soc-acpi-intel-glk-match.c2
-rw-r--r--sound/soc/intel/common/soc-acpi-intel-hda-match.c2
-rw-r--r--sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c2
-rw-r--r--sound/soc/intel/common/soc-acpi-intel-icl-match.c2
-rw-r--r--sound/soc/intel/common/soc-acpi-intel-kbl-match.c2
-rw-r--r--sound/soc/intel/common/soc-acpi-intel-skl-match.c2
-rw-r--r--sound/soc/qcom/apq8016_sbc.c16
-rw-r--r--sound/soc/rockchip/rockchip_i2s.c5
-rw-r--r--sound/soc/rockchip/rockchip_max98090.c32
-rw-r--r--sound/soc/samsung/odroid.c8
-rw-r--r--sound/soc/soc-core.c7
-rw-r--r--sound/soc/soc-dapm.c10
-rw-r--r--sound/soc/sof/intel/cnl.c4
-rw-r--r--sound/soc/sof/intel/hda-ipc.c4
-rw-r--r--sound/soc/sunxi/sun4i-i2s.c4
-rw-r--r--sound/soc/ti/davinci-mcasp.c46
-rw-r--r--sound/usb/helper.c2
-rw-r--r--sound/usb/stream.c1
44 files changed, 347 insertions, 162 deletions
diff --git a/sound/ac97/bus.c b/sound/ac97/bus.c
index 7b977b753a03..7985dd8198b6 100644
--- a/sound/ac97/bus.c
+++ b/sound/ac97/bus.c
@@ -122,17 +122,12 @@ static int ac97_codec_add(struct ac97_controller *ac97_ctrl, int idx,
vendor_id);
ret = device_add(&codec->dev);
- if (ret)
- goto err_free_codec;
+ if (ret) {
+ put_device(&codec->dev);
+ return ret;
+ }
return 0;
-err_free_codec:
- of_node_put(codec->dev.of_node);
- put_device(&codec->dev);
- kfree(codec);
- ac97_ctrl->codecs[idx] = NULL;
-
- return ret;
}
unsigned int snd_ac97_bus_scan_one(struct ac97_controller *adrv,
diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c
index 99b882158705..41905afada63 100644
--- a/sound/core/compress_offload.c
+++ b/sound/core/compress_offload.c
@@ -574,10 +574,7 @@ snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg)
stream->metadata_set = false;
stream->next_track = false;
- if (stream->direction == SND_COMPRESS_PLAYBACK)
- stream->runtime->state = SNDRV_PCM_STATE_SETUP;
- else
- stream->runtime->state = SNDRV_PCM_STATE_PREPARED;
+ stream->runtime->state = SNDRV_PCM_STATE_SETUP;
} else {
return -EPERM;
}
@@ -693,8 +690,17 @@ static int snd_compr_start(struct snd_compr_stream *stream)
{
int retval;
- if (stream->runtime->state != SNDRV_PCM_STATE_PREPARED)
+ switch (stream->runtime->state) {
+ case SNDRV_PCM_STATE_SETUP:
+ if (stream->direction != SND_COMPRESS_CAPTURE)
+ return -EPERM;
+ break;
+ case SNDRV_PCM_STATE_PREPARED:
+ break;
+ default:
return -EPERM;
+ }
+
retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_START);
if (!retval)
stream->runtime->state = SNDRV_PCM_STATE_RUNNING;
@@ -705,9 +711,15 @@ static int snd_compr_stop(struct snd_compr_stream *stream)
{
int retval;
- if (stream->runtime->state == SNDRV_PCM_STATE_PREPARED ||
- stream->runtime->state == SNDRV_PCM_STATE_SETUP)
+ switch (stream->runtime->state) {
+ case SNDRV_PCM_STATE_OPEN:
+ case SNDRV_PCM_STATE_SETUP:
+ case SNDRV_PCM_STATE_PREPARED:
return -EPERM;
+ default:
+ break;
+ }
+
retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_STOP);
if (!retval) {
snd_compr_drain_notify(stream);
@@ -795,9 +807,17 @@ static int snd_compr_drain(struct snd_compr_stream *stream)
{
int retval;
- if (stream->runtime->state == SNDRV_PCM_STATE_PREPARED ||
- stream->runtime->state == SNDRV_PCM_STATE_SETUP)
+ switch (stream->runtime->state) {
+ case SNDRV_PCM_STATE_OPEN:
+ case SNDRV_PCM_STATE_SETUP:
+ case SNDRV_PCM_STATE_PREPARED:
+ case SNDRV_PCM_STATE_PAUSED:
return -EPERM;
+ case SNDRV_PCM_STATE_XRUN:
+ return -EPIPE;
+ default:
+ break;
+ }
retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_DRAIN);
if (retval) {
@@ -817,6 +837,10 @@ static int snd_compr_next_track(struct snd_compr_stream *stream)
if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING)
return -EPERM;
+ /* next track doesn't have any meaning for capture streams */
+ if (stream->direction == SND_COMPRESS_CAPTURE)
+ return -EPERM;
+
/* you can signal next track if this is intended to be a gapless stream
* and current track metadata is set
*/
@@ -834,9 +858,23 @@ static int snd_compr_next_track(struct snd_compr_stream *stream)
static int snd_compr_partial_drain(struct snd_compr_stream *stream)
{
int retval;
- if (stream->runtime->state == SNDRV_PCM_STATE_PREPARED ||
- stream->runtime->state == SNDRV_PCM_STATE_SETUP)
+
+ switch (stream->runtime->state) {
+ case SNDRV_PCM_STATE_OPEN:
+ case SNDRV_PCM_STATE_SETUP:
+ case SNDRV_PCM_STATE_PREPARED:
+ case SNDRV_PCM_STATE_PAUSED:
+ return -EPERM;
+ case SNDRV_PCM_STATE_XRUN:
+ return -EPIPE;
+ default:
+ break;
+ }
+
+ /* partial drain doesn't have any meaning for capture streams */
+ if (stream->direction == SND_COMPRESS_CAPTURE)
return -EPERM;
+
/* stream can be drained only when next track has been signalled */
if (stream->next_track == false)
return -EPERM;
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 12dd9b318db1..703857aab00f 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -1873,6 +1873,7 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream,
if (!to_check)
break; /* all drained */
init_waitqueue_entry(&wait, current);
+ set_current_state(TASK_INTERRUPTIBLE);
add_wait_queue(&to_check->sleep, &wait);
snd_pcm_stream_unlock_irq(substream);
if (runtime->no_period_wakeup)
@@ -1885,7 +1886,7 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream,
}
tout = msecs_to_jiffies(tout * 1000);
}
- tout = schedule_timeout_interruptible(tout);
+ tout = schedule_timeout(tout);
snd_pcm_stream_lock_irq(substream);
group = snd_pcm_stream_group_ref(substream);
diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c
index 1192c7561d62..3c2db3816029 100644
--- a/sound/hda/hdac_i915.c
+++ b/sound/hda/hdac_i915.c
@@ -136,10 +136,12 @@ int snd_hdac_i915_init(struct hdac_bus *bus)
if (!acomp)
return -ENODEV;
if (!acomp->ops) {
- request_module("i915");
- /* 60s timeout */
- wait_for_completion_timeout(&bind_complete,
- msecs_to_jiffies(60 * 1000));
+ if (!IS_ENABLED(CONFIG_MODULES) ||
+ !request_module("i915")) {
+ /* 60s timeout */
+ wait_for_completion_timeout(&bind_complete,
+ msecs_to_jiffies(60 * 1000));
+ }
}
if (!acomp->ops) {
dev_info(bus->dev, "couldn't bind with audio component\n");
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 151f206e1c02..2af1e0868e55 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -318,11 +318,10 @@ enum {
#define AZX_DCAPS_INTEL_SKYLAKE \
(AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_PM_RUNTIME |\
+ AZX_DCAPS_SYNC_WRITE |\
AZX_DCAPS_SEPARATE_STREAM_TAG | AZX_DCAPS_I915_COMPONENT)
-#define AZX_DCAPS_INTEL_BROXTON \
- (AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_PM_RUNTIME |\
- AZX_DCAPS_SEPARATE_STREAM_TAG | AZX_DCAPS_I915_COMPONENT)
+#define AZX_DCAPS_INTEL_BROXTON AZX_DCAPS_INTEL_SKYLAKE
/* quirks for ATI SB / AMD Hudson */
#define AZX_DCAPS_PRESET_ATI_SB \
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 4f8d0845ee1e..f299f137eaea 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -1083,6 +1083,7 @@ static int patch_conexant_auto(struct hda_codec *codec)
*/
static const struct hda_device_id snd_hda_id_conexant[] = {
+ HDA_CODEC_ENTRY(0x14f11f86, "CX8070", patch_conexant_auto),
HDA_CODEC_ENTRY(0x14f12008, "CX8200", patch_conexant_auto),
HDA_CODEC_ENTRY(0x14f15045, "CX20549 (Venice)", patch_conexant_auto),
HDA_CODEC_ENTRY(0x14f15047, "CX20551 (Waikiki)", patch_conexant_auto),
diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c
index a4ade6bb5beb..bc4dfafdfcd1 100644
--- a/sound/soc/amd/raven/acp3x-pcm-dma.c
+++ b/sound/soc/amd/raven/acp3x-pcm-dma.c
@@ -31,8 +31,8 @@ struct i2s_stream_instance {
u16 num_pages;
u16 channels;
u32 xfer_resolution;
- struct page *pg;
u64 bytescount;
+ dma_addr_t dma_addr;
void __iomem *acp3x_base;
};
@@ -211,9 +211,8 @@ static irqreturn_t i2s_irq_handler(int irq, void *dev_id)
static void config_acp3x_dma(struct i2s_stream_instance *rtd, int direction)
{
u16 page_idx;
- u64 addr;
u32 low, high, val, acp_fifo_addr;
- struct page *pg = rtd->pg;
+ dma_addr_t addr = rtd->dma_addr;
/* 8 scratch registers used to map one 64 bit address */
if (direction == SNDRV_PCM_STREAM_PLAYBACK)
@@ -229,7 +228,6 @@ static void config_acp3x_dma(struct i2s_stream_instance *rtd, int direction)
for (page_idx = 0; page_idx < rtd->num_pages; page_idx++) {
/* Load the low address of page int ACP SRAM through SRBM */
- addr = page_to_phys(pg);
low = lower_32_bits(addr);
high = upper_32_bits(addr);
@@ -239,7 +237,7 @@ static void config_acp3x_dma(struct i2s_stream_instance *rtd, int direction)
+ 4);
/* Move to next physically contiguos page */
val += 8;
- pg++;
+ addr += PAGE_SIZE;
}
if (direction == SNDRV_PCM_STREAM_PLAYBACK) {
@@ -341,7 +339,6 @@ static int acp3x_dma_hw_params(struct snd_pcm_substream *substream,
{
int status;
u64 size;
- struct page *pg;
struct snd_pcm_runtime *runtime = substream->runtime;
struct i2s_stream_instance *rtd = runtime->private_data;
@@ -354,9 +351,8 @@ static int acp3x_dma_hw_params(struct snd_pcm_substream *substream,
return status;
memset(substream->runtime->dma_area, 0, params_buffer_bytes(params));
- pg = virt_to_page(substream->dma_buffer.area);
- if (pg) {
- rtd->pg = pg;
+ if (substream->dma_buffer.area) {
+ rtd->dma_addr = substream->dma_buffer.addr;
rtd->num_pages = (PAGE_ALIGN(size) >> PAGE_SHIFT);
config_acp3x_dma(rtd, substream->stream);
status = 0;
@@ -385,9 +381,11 @@ static snd_pcm_uframes_t acp3x_dma_pointer(struct snd_pcm_substream *substream)
static int acp3x_dma_new(struct snd_soc_pcm_runtime *rtd)
{
+ struct snd_soc_component *component = snd_soc_rtdcom_lookup(rtd,
+ DRV_NAME);
+ struct device *parent = component->dev->parent;
snd_pcm_lib_preallocate_pages_for_all(rtd->pcm, SNDRV_DMA_TYPE_DEV,
- rtd->pcm->card->dev,
- MIN_BUFFER, MAX_BUFFER);
+ parent, MIN_BUFFER, MAX_BUFFER);
return 0;
}
diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c
index 6203f54d9f25..5b049fcdba20 100644
--- a/sound/soc/codecs/cs42xx8.c
+++ b/sound/soc/codecs/cs42xx8.c
@@ -47,6 +47,7 @@ struct cs42xx8_priv {
unsigned long sysclk;
u32 tx_channels;
struct gpio_desc *gpiod_reset;
+ u32 rate[2];
};
/* -127.5dB to 0dB with step of 0.5dB */
@@ -176,21 +177,27 @@ static const struct snd_soc_dapm_route cs42xx8_adc3_dapm_routes[] = {
};
struct cs42xx8_ratios {
- unsigned int ratio;
- unsigned char speed;
- unsigned char mclk;
+ unsigned int mfreq;
+ unsigned int min_mclk;
+ unsigned int max_mclk;
+ unsigned int ratio[3];
};
+/*
+ * According to reference mannual, define the cs42xx8_ratio struct
+ * MFreq2 | MFreq1 | MFreq0 | Description | SSM | DSM | QSM |
+ * 0 | 0 | 0 |1.029MHz to 12.8MHz | 256 | 128 | 64 |
+ * 0 | 0 | 1 |1.536MHz to 19.2MHz | 384 | 192 | 96 |
+ * 0 | 1 | 0 |2.048MHz to 25.6MHz | 512 | 256 | 128 |
+ * 0 | 1 | 1 |3.072MHz to 38.4MHz | 768 | 384 | 192 |
+ * 1 | x | x |4.096MHz to 51.2MHz |1024 | 512 | 256 |
+ */
static const struct cs42xx8_ratios cs42xx8_ratios[] = {
- { 64, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_256(4) },
- { 96, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_384(4) },
- { 128, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_512(4) },
- { 192, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_768(4) },
- { 256, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_256(1) },
- { 384, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_384(1) },
- { 512, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_512(1) },
- { 768, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_768(1) },
- { 1024, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_1024(1) }
+ { 0, 1029000, 12800000, {256, 128, 64} },
+ { 2, 1536000, 19200000, {384, 192, 96} },
+ { 4, 2048000, 25600000, {512, 256, 128} },
+ { 6, 3072000, 38400000, {768, 384, 192} },
+ { 8, 4096000, 51200000, {1024, 512, 256} },
};
static int cs42xx8_set_dai_sysclk(struct snd_soc_dai *codec_dai,
@@ -257,14 +264,68 @@ static int cs42xx8_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_component *component = dai->component;
struct cs42xx8_priv *cs42xx8 = snd_soc_component_get_drvdata(component);
bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
- u32 ratio = cs42xx8->sysclk / params_rate(params);
- u32 i, fm, val, mask;
+ u32 ratio[2];
+ u32 rate[2];
+ u32 fm[2];
+ u32 i, val, mask;
+ bool condition1, condition2;
if (tx)
cs42xx8->tx_channels = params_channels(params);
+ rate[tx] = params_rate(params);
+ rate[!tx] = cs42xx8->rate[!tx];
+
+ ratio[tx] = rate[tx] > 0 ? cs42xx8->sysclk / rate[tx] : 0;
+ ratio[!tx] = rate[!tx] > 0 ? cs42xx8->sysclk / rate[!tx] : 0;
+
+ /* Get functional mode for tx and rx according to rate */
+ for (i = 0; i < 2; i++) {
+ if (cs42xx8->slave_mode) {
+ fm[i] = CS42XX8_FM_AUTO;
+ } else {
+ if (rate[i] < 50000) {
+ fm[i] = CS42XX8_FM_SINGLE;
+ } else if (rate[i] > 50000 && rate[i] < 100000) {
+ fm[i] = CS42XX8_FM_DOUBLE;
+ } else if (rate[i] > 100000 && rate[i] < 200000) {
+ fm[i] = CS42XX8_FM_QUAD;
+ } else {
+ dev_err(component->dev,
+ "unsupported sample rate\n");
+ return -EINVAL;
+ }
+ }
+ }
+
for (i = 0; i < ARRAY_SIZE(cs42xx8_ratios); i++) {
- if (cs42xx8_ratios[i].ratio == ratio)
+ /* Is the ratio[tx] valid ? */
+ condition1 = ((fm[tx] == CS42XX8_FM_AUTO) ?
+ (cs42xx8_ratios[i].ratio[0] == ratio[tx] ||
+ cs42xx8_ratios[i].ratio[1] == ratio[tx] ||
+ cs42xx8_ratios[i].ratio[2] == ratio[tx]) :
+ (cs42xx8_ratios[i].ratio[fm[tx]] == ratio[tx])) &&
+ cs42xx8->sysclk >= cs42xx8_ratios[i].min_mclk &&
+ cs42xx8->sysclk <= cs42xx8_ratios[i].max_mclk;
+
+ if (!ratio[tx])
+ condition1 = true;
+
+ /* Is the ratio[!tx] valid ? */
+ condition2 = ((fm[!tx] == CS42XX8_FM_AUTO) ?
+ (cs42xx8_ratios[i].ratio[0] == ratio[!tx] ||
+ cs42xx8_ratios[i].ratio[1] == ratio[!tx] ||
+ cs42xx8_ratios[i].ratio[2] == ratio[!tx]) :
+ (cs42xx8_ratios[i].ratio[fm[!tx]] == ratio[!tx]));
+
+ if (!ratio[!tx])
+ condition2 = true;
+
+ /*
+ * Both ratio[tx] and ratio[!tx] is valid, then we get
+ * a proper MFreq.
+ */
+ if (condition1 && condition2)
break;
}
@@ -273,15 +334,31 @@ static int cs42xx8_hw_params(struct snd_pcm_substream *substream,
return -EINVAL;
}
- mask = CS42XX8_FUNCMOD_MFREQ_MASK;
- val = cs42xx8_ratios[i].mclk;
+ cs42xx8->rate[tx] = params_rate(params);
- fm = cs42xx8->slave_mode ? CS42XX8_FM_AUTO : cs42xx8_ratios[i].speed;
+ mask = CS42XX8_FUNCMOD_MFREQ_MASK;
+ val = cs42xx8_ratios[i].mfreq;
regmap_update_bits(cs42xx8->regmap, CS42XX8_FUNCMOD,
CS42XX8_FUNCMOD_xC_FM_MASK(tx) | mask,
- CS42XX8_FUNCMOD_xC_FM(tx, fm) | val);
+ CS42XX8_FUNCMOD_xC_FM(tx, fm[tx]) | val);
+
+ return 0;
+}
+
+static int cs42xx8_hw_free(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_component *component = dai->component;
+ struct cs42xx8_priv *cs42xx8 = snd_soc_component_get_drvdata(component);
+ bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+ /* Clear stored rate */
+ cs42xx8->rate[tx] = 0;
+
+ regmap_update_bits(cs42xx8->regmap, CS42XX8_FUNCMOD,
+ CS42XX8_FUNCMOD_xC_FM_MASK(tx),
+ CS42XX8_FUNCMOD_xC_FM(tx, CS42XX8_FM_AUTO));
return 0;
}
@@ -302,6 +379,7 @@ static const struct snd_soc_dai_ops cs42xx8_dai_ops = {
.set_fmt = cs42xx8_set_dai_fmt,
.set_sysclk = cs42xx8_set_dai_sysclk,
.hw_params = cs42xx8_hw_params,
+ .hw_free = cs42xx8_hw_free,
.digital_mute = cs42xx8_digital_mute,
};
diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c
index 6f0e28f903bf..16313b973eaa 100644
--- a/sound/soc/codecs/max98357a.c
+++ b/sound/soc/codecs/max98357a.c
@@ -20,20 +20,10 @@
#include <sound/soc-dapm.h>
struct max98357a_priv {
- struct delayed_work enable_sdmode_work;
struct gpio_desc *sdmode;
unsigned int sdmode_delay;
};
-static void max98357a_enable_sdmode_work(struct work_struct *work)
-{
- struct max98357a_priv *max98357a =
- container_of(work, struct max98357a_priv,
- enable_sdmode_work.work);
-
- gpiod_set_value(max98357a->sdmode, 1);
-}
-
static int max98357a_daiops_trigger(struct snd_pcm_substream *substream,
int cmd, struct snd_soc_dai *dai)
{
@@ -46,14 +36,12 @@ static int max98357a_daiops_trigger(struct snd_pcm_substream *substream,
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
- queue_delayed_work(system_power_efficient_wq,
- &max98357a->enable_sdmode_work,
- msecs_to_jiffies(max98357a->sdmode_delay));
+ mdelay(max98357a->sdmode_delay);
+ gpiod_set_value(max98357a->sdmode, 1);
break;
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
- cancel_delayed_work_sync(&max98357a->enable_sdmode_work);
gpiod_set_value(max98357a->sdmode, 0);
break;
}
@@ -112,30 +100,25 @@ static int max98357a_platform_probe(struct platform_device *pdev)
int ret;
max98357a = devm_kzalloc(&pdev->dev, sizeof(*max98357a), GFP_KERNEL);
-
if (!max98357a)
return -ENOMEM;
max98357a->sdmode = devm_gpiod_get_optional(&pdev->dev,
"sdmode", GPIOD_OUT_LOW);
-
if (IS_ERR(max98357a->sdmode))
return PTR_ERR(max98357a->sdmode);
ret = device_property_read_u32(&pdev->dev, "sdmode-delay",
&max98357a->sdmode_delay);
-
if (ret) {
max98357a->sdmode_delay = 0;
dev_dbg(&pdev->dev,
- "no optional property 'sdmode-delay' found, default: no delay\n");
+ "no optional property 'sdmode-delay' found, "
+ "default: no delay\n");
}
dev_set_drvdata(&pdev->dev, max98357a);
- INIT_DELAYED_WORK(&max98357a->enable_sdmode_work,
- max98357a_enable_sdmode_work);
-
return devm_snd_soc_register_component(&pdev->dev,
&max98357a_component_driver,
&max98357a_dai_driver, 1);
diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c
index 528695cd6a1c..8c601a3ebc27 100644
--- a/sound/soc/codecs/max98373.c
+++ b/sound/soc/codecs/max98373.c
@@ -267,6 +267,12 @@ static int max98373_dai_hw_params(struct snd_pcm_substream *substream,
case 48000:
sampling_rate = MAX98373_PCM_SR_SET1_SR_48000;
break;
+ case 88200:
+ sampling_rate = MAX98373_PCM_SR_SET1_SR_88200;
+ break;
+ case 96000:
+ sampling_rate = MAX98373_PCM_SR_SET1_SR_96000;
+ break;
default:
dev_err(component->dev, "rate %d not supported\n",
params_rate(params));
diff --git a/sound/soc/codecs/max98373.h b/sound/soc/codecs/max98373.h
index f6a37aa02f26..a59e51355a84 100644
--- a/sound/soc/codecs/max98373.h
+++ b/sound/soc/codecs/max98373.h
@@ -130,6 +130,8 @@
#define MAX98373_PCM_SR_SET1_SR_32000 (0x6 << 0)
#define MAX98373_PCM_SR_SET1_SR_44100 (0x7 << 0)
#define MAX98373_PCM_SR_SET1_SR_48000 (0x8 << 0)
+#define MAX98373_PCM_SR_SET1_SR_88200 (0x9 << 0)
+#define MAX98373_PCM_SR_SET1_SR_96000 (0xA << 0)
/* MAX98373_R2028_PCM_SR_SETUP_2 */
#define MAX98373_PCM_SR_SET2_SR_MASK (0xF << 4)
diff --git a/sound/soc/codecs/pcm3060-i2c.c b/sound/soc/codecs/pcm3060-i2c.c
index cdc8314882bc..abcdeb922201 100644
--- a/sound/soc/codecs/pcm3060-i2c.c
+++ b/sound/soc/codecs/pcm3060-i2c.c
@@ -2,7 +2,7 @@
//
// PCM3060 I2C driver
//
-// Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.tech>
+// Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.com>
#include <linux/i2c.h>
#include <linux/module.h>
@@ -56,5 +56,5 @@ static struct i2c_driver pcm3060_i2c_driver = {
module_i2c_driver(pcm3060_i2c_driver);
MODULE_DESCRIPTION("PCM3060 I2C driver");
-MODULE_AUTHOR("Kirill Marinushkin <kmarinushkin@birdec.tech>");
+MODULE_AUTHOR("Kirill Marinushkin <kmarinushkin@birdec.com>");
MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/pcm3060-spi.c b/sound/soc/codecs/pcm3060-spi.c
index f6f19fa80932..3b79734b832b 100644
--- a/sound/soc/codecs/pcm3060-spi.c
+++ b/sound/soc/codecs/pcm3060-spi.c
@@ -2,7 +2,7 @@
//
// PCM3060 SPI driver
//
-// Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.tech>
+// Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.com>
#include <linux/module.h>
#include <linux/spi/spi.h>
@@ -55,5 +55,5 @@ static struct spi_driver pcm3060_spi_driver = {
module_spi_driver(pcm3060_spi_driver);
MODULE_DESCRIPTION("PCM3060 SPI driver");
-MODULE_AUTHOR("Kirill Marinushkin <kmarinushkin@birdec.tech>");
+MODULE_AUTHOR("Kirill Marinushkin <kmarinushkin@birdec.com>");
MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/pcm3060.c b/sound/soc/codecs/pcm3060.c
index 32b26f1c2282..b2358069cf9b 100644
--- a/sound/soc/codecs/pcm3060.c
+++ b/sound/soc/codecs/pcm3060.c
@@ -2,7 +2,7 @@
//
// PCM3060 codec driver
//
-// Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.tech>
+// Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.com>
#include <linux/module.h>
#include <sound/pcm_params.h>
@@ -342,5 +342,5 @@ int pcm3060_probe(struct device *dev)
EXPORT_SYMBOL(pcm3060_probe);
MODULE_DESCRIPTION("PCM3060 codec driver");
-MODULE_AUTHOR("Kirill Marinushkin <kmarinushkin@birdec.tech>");
+MODULE_AUTHOR("Kirill Marinushkin <kmarinushkin@birdec.com>");
MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/pcm3060.h b/sound/soc/codecs/pcm3060.h
index 75931c9a9d85..18d51e5dac2c 100644
--- a/sound/soc/codecs/pcm3060.h
+++ b/sound/soc/codecs/pcm3060.h
@@ -2,7 +2,7 @@
/*
* PCM3060 codec driver
*
- * Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.tech>
+ * Copyright (C) 2018 Kirill Marinushkin <kmarinushkin@birdec.com>
*/
#ifndef _SND_SOC_PCM3060_H
diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c
index 5605b660f4bf..0a6ff13d76e1 100644
--- a/sound/soc/codecs/rt1011.c
+++ b/sound/soc/codecs/rt1011.c
@@ -39,7 +39,7 @@ static const struct reg_sequence init_list[] = {
{ RT1011_POWER_9, 0xa840 },
{ RT1011_ADC_SET_5, 0x0a20 },
- { RT1011_DAC_SET_2, 0xa232 },
+ { RT1011_DAC_SET_2, 0xa032 },
{ RT1011_ADC_SET_1, 0x2925 },
{ RT1011_SPK_PRO_DC_DET_1, 0xb00c },
@@ -1917,7 +1917,7 @@ static int rt1011_set_bias_level(struct snd_soc_component *component,
snd_soc_component_write(component,
RT1011_SYSTEM_RESET_2, 0x0000);
snd_soc_component_write(component,
- RT1011_SYSTEM_RESET_3, 0x0000);
+ RT1011_SYSTEM_RESET_3, 0x0001);
snd_soc_component_write(component,
RT1011_SYSTEM_RESET_1, 0x003f);
snd_soc_component_write(component,
diff --git a/sound/soc/codecs/rt1308.c b/sound/soc/codecs/rt1308.c
index d673506c7c39..d673506c7c39 100755..100644
--- a/sound/soc/codecs/rt1308.c
+++ b/sound/soc/codecs/rt1308.c
diff --git a/sound/soc/codecs/rt1308.h b/sound/soc/codecs/rt1308.h
index c330aae1d527..c330aae1d527 100755..100644
--- a/sound/soc/codecs/rt1308.h
+++ b/sound/soc/codecs/rt1308.h
diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c
index 30a4e8399ec3..288df245b2f0 100644
--- a/sound/soc/generic/audio-graph-card.c
+++ b/sound/soc/generic/audio-graph-card.c
@@ -63,6 +63,7 @@ static int graph_get_dai_id(struct device_node *ep)
struct device_node *endpoint;
struct of_endpoint info;
int i, id;
+ const u32 *reg;
int ret;
/* use driver specified DAI ID if exist */
@@ -83,8 +84,9 @@ static int graph_get_dai_id(struct device_node *ep)
return info.id;
node = of_get_parent(ep);
+ reg = of_get_property(node, "reg", NULL);
of_node_put(node);
- if (of_get_property(node, "reg", NULL))
+ if (reg)
return info.port;
}
node = of_graph_get_port_parent(ep);
@@ -208,10 +210,6 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv,
dev_dbg(dev, "link_of DPCM (%pOF)\n", ep);
- of_node_put(ports);
- of_node_put(port);
- of_node_put(node);
-
if (li->cpu) {
int is_single_links = 0;
@@ -229,17 +227,17 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv,
ret = asoc_simple_parse_cpu(ep, dai_link, &is_single_links);
if (ret)
- return ret;
+ goto out_put_node;
ret = asoc_simple_parse_clk_cpu(dev, ep, dai_link, dai);
if (ret < 0)
- return ret;
+ goto out_put_node;
ret = asoc_simple_set_dailink_name(dev, dai_link,
"fe.%s",
cpus->dai_name);
if (ret < 0)
- return ret;
+ goto out_put_node;
/* card->num_links includes Codec */
asoc_simple_canonicalize_cpu(dai_link, is_single_links);
@@ -263,17 +261,17 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv,
ret = asoc_simple_parse_codec(ep, dai_link);
if (ret < 0)
- return ret;
+ goto out_put_node;
ret = asoc_simple_parse_clk_codec(dev, ep, dai_link, dai);
if (ret < 0)
- return ret;
+ goto out_put_node;
ret = asoc_simple_set_dailink_name(dev, dai_link,
"be.%s",
codecs->dai_name);
if (ret < 0)
- return ret;
+ goto out_put_node;
/* check "prefix" from top node */
snd_soc_of_parse_node_prefix(top, cconf, codecs->of_node,
@@ -293,19 +291,23 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv,
ret = asoc_simple_parse_tdm(ep, dai);
if (ret)
- return ret;
+ goto out_put_node;
ret = asoc_simple_parse_daifmt(dev, cpu_ep, codec_ep,
NULL, &dai_link->dai_fmt);
if (ret < 0)
- return ret;
+ goto out_put_node;
dai_link->dpcm_playback = 1;
dai_link->dpcm_capture = 1;
dai_link->ops = &graph_ops;
dai_link->init = asoc_simple_dai_init;
- return 0;
+out_put_node:
+ of_node_put(ports);
+ of_node_put(port);
+ of_node_put(node);
+ return ret;
}
static int graph_dai_link_of(struct asoc_simple_priv *priv,
diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c
index ac8678fe55ff..556b1a789629 100644
--- a/sound/soc/generic/simple-card-utils.c
+++ b/sound/soc/generic/simple-card-utils.c
@@ -349,6 +349,13 @@ void asoc_simple_canonicalize_platform(struct snd_soc_dai_link *dai_link)
/* Assumes platform == cpu */
if (!dai_link->platforms->of_node)
dai_link->platforms->of_node = dai_link->cpus->of_node;
+
+ /*
+ * DPCM BE can be no platform.
+ * Alloced memory will be waste, but not leak.
+ */
+ if (!dai_link->platforms->of_node)
+ dai_link->num_platforms = 0;
}
EXPORT_SYMBOL_GPL(asoc_simple_canonicalize_platform);
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index e5cde0d5e63c..ef849151ba56 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -124,8 +124,6 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv,
li->link++;
- of_node_put(node);
-
/* For single DAI link & old style of DT node */
if (is_top)
prefix = PREFIX;
@@ -147,17 +145,17 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv,
ret = asoc_simple_parse_cpu(np, dai_link, &is_single_links);
if (ret)
- return ret;
+ goto out_put_node;
ret = asoc_simple_parse_clk_cpu(dev, np, dai_link, dai);
if (ret < 0)
- return ret;
+ goto out_put_node;
ret = asoc_simple_set_dailink_name(dev, dai_link,
"fe.%s",
cpus->dai_name);
if (ret < 0)
- return ret;
+ goto out_put_node;
asoc_simple_canonicalize_cpu(dai_link, is_single_links);
} else {
@@ -180,17 +178,17 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv,
ret = asoc_simple_parse_codec(np, dai_link);
if (ret < 0)
- return ret;
+ goto out_put_node;
ret = asoc_simple_parse_clk_codec(dev, np, dai_link, dai);
if (ret < 0)
- return ret;
+ goto out_put_node;
ret = asoc_simple_set_dailink_name(dev, dai_link,
"be.%s",
codecs->dai_name);
if (ret < 0)
- return ret;
+ goto out_put_node;
/* check "prefix" from top node */
snd_soc_of_parse_node_prefix(top, cconf, codecs->of_node,
@@ -208,19 +206,21 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv,
ret = asoc_simple_parse_tdm(np, dai);
if (ret)
- return ret;
+ goto out_put_node;
ret = asoc_simple_parse_daifmt(dev, node, codec,
prefix, &dai_link->dai_fmt);
if (ret < 0)
- return ret;
+ goto out_put_node;
dai_link->dpcm_playback = 1;
dai_link->dpcm_capture = 1;
dai_link->ops = &simple_ops;
dai_link->init = asoc_simple_dai_init;
- return 0;
+out_put_node:
+ of_node_put(node);
+ return ret;
}
static int simple_dai_link_of(struct asoc_simple_priv *priv,
@@ -364,8 +364,6 @@ static int simple_for_each_link(struct asoc_simple_priv *priv,
goto error;
}
- of_node_put(codec);
-
/* get convert-xxx property */
memset(&adata, 0, sizeof(adata));
for_each_child_of_node(node, np)
@@ -387,11 +385,13 @@ static int simple_for_each_link(struct asoc_simple_priv *priv,
ret = func_noml(priv, np, codec, li, is_top);
if (ret < 0) {
+ of_node_put(codec);
of_node_put(np);
goto error;
}
}
+ of_node_put(codec);
node = of_get_next_child(top, node);
} while (!is_top && node);
diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c
index fac09be3cade..46612331f5ea 100644
--- a/sound/soc/intel/boards/bytcht_es8316.c
+++ b/sound/soc/intel/boards/bytcht_es8316.c
@@ -437,6 +437,14 @@ static const struct acpi_gpio_mapping byt_cht_es8316_gpios[] = {
/* Please keep this list alphabetically sorted */
static const struct dmi_system_id byt_cht_es8316_quirk_table[] = {
+ { /* Irbis NB41 */
+ .matches = {
+ DMI_MATCH(DMI_SYS_VENDOR, "IRBIS"),
+ DMI_MATCH(DMI_PRODUCT_NAME, "NB41"),
+ },
+ .driver_data = (void *)(BYT_CHT_ES8316_INTMIC_IN2_MAP
+ | BYT_CHT_ES8316_JD_INVERTED),
+ },
{ /* Teclast X98 Plus II */
.matches = {
DMI_MATCH(DMI_SYS_VENDOR, "TECLAST"),
diff --git a/sound/soc/intel/common/soc-acpi-intel-bxt-match.c b/sound/soc/intel/common/soc-acpi-intel-bxt-match.c
index 229e39586868..4a5adae1d785 100644
--- a/sound/soc/intel/common/soc-acpi-intel-bxt-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-bxt-match.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0
/*
- * soc-apci-intel-bxt-match.c - tables and support for BXT ACPI enumeration.
+ * soc-acpi-intel-bxt-match.c - tables and support for BXT ACPI enumeration.
*
* Copyright (c) 2018, Intel Corporation.
*
diff --git a/sound/soc/intel/common/soc-acpi-intel-byt-match.c b/sound/soc/intel/common/soc-acpi-intel-byt-match.c
index b94b482ac34f..1cc801ba92eb 100644
--- a/sound/soc/intel/common/soc-acpi-intel-byt-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-byt-match.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
/*
- * soc-apci-intel-byt-match.c - tables and support for BYT ACPI enumeration.
+ * soc-acpi-intel-byt-match.c - tables and support for BYT ACPI enumeration.
*
* Copyright (c) 2017, Intel Corporation.
*/
diff --git a/sound/soc/intel/common/soc-acpi-intel-cht-match.c b/sound/soc/intel/common/soc-acpi-intel-cht-match.c
index b7f11f6be1cf..d0fb43c2b9f6 100644
--- a/sound/soc/intel/common/soc-acpi-intel-cht-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-cht-match.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
/*
- * soc-apci-intel-cht-match.c - tables and support for CHT ACPI enumeration.
+ * soc-acpi-intel-cht-match.c - tables and support for CHT ACPI enumeration.
*
* Copyright (c) 2017, Intel Corporation.
*/
diff --git a/sound/soc/intel/common/soc-acpi-intel-cnl-match.c b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c
index c36c0aa4f683..771b0ef21051 100644
--- a/sound/soc/intel/common/soc-acpi-intel-cnl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0
/*
- * soc-apci-intel-cnl-match.c - tables and support for CNL ACPI enumeration.
+ * soc-acpi-intel-cnl-match.c - tables and support for CNL ACPI enumeration.
*
* Copyright (c) 2018, Intel Corporation.
*
diff --git a/sound/soc/intel/common/soc-acpi-intel-glk-match.c b/sound/soc/intel/common/soc-acpi-intel-glk-match.c
index 616eb09e78a0..60dea358fa04 100644
--- a/sound/soc/intel/common/soc-acpi-intel-glk-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-glk-match.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0
/*
- * soc-apci-intel-glk-match.c - tables and support for GLK ACPI enumeration.
+ * soc-acpi-intel-glk-match.c - tables and support for GLK ACPI enumeration.
*
* Copyright (c) 2018, Intel Corporation.
*
diff --git a/sound/soc/intel/common/soc-acpi-intel-hda-match.c b/sound/soc/intel/common/soc-acpi-intel-hda-match.c
index 68ae43f7b4b2..cc972d2ac691 100644
--- a/sound/soc/intel/common/soc-acpi-intel-hda-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-hda-match.c
@@ -2,7 +2,7 @@
// Copyright (c) 2018, Intel Corporation.
/*
- * soc-apci-intel-hda-match.c - tables and support for HDA+ACPI enumeration.
+ * soc-acpi-intel-hda-match.c - tables and support for HDA+ACPI enumeration.
*
*/
diff --git a/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c b/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c
index d27853e7a369..34eb0baaa951 100644
--- a/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0-only
/*
- * soc-apci-intel-hsw-bdw-match.c - tables and support for ACPI enumeration.
+ * soc-acpi-intel-hsw-bdw-match.c - tables and support for ACPI enumeration.
*
* Copyright (c) 2017, Intel Corporation.
*/
diff --git a/sound/soc/intel/common/soc-acpi-intel-icl-match.c b/sound/soc/intel/common/soc-acpi-intel-icl-match.c
index 0b430b9b3673..38977669b576 100644
--- a/sound/soc/intel/common/soc-acpi-intel-icl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-icl-match.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0
/*
- * soc-apci-intel-icl-match.c - tables and support for ICL ACPI enumeration.
+ * soc-acpi-intel-icl-match.c - tables and support for ICL ACPI enumeration.
*
* Copyright (c) 2018, Intel Corporation.
*
diff --git a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c
index 4b331058e807..e200baa11011 100644
--- a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0
/*
- * soc-apci-intel-kbl-match.c - tables and support for KBL ACPI enumeration.
+ * soc-acpi-intel-kbl-match.c - tables and support for KBL ACPI enumeration.
*
* Copyright (c) 2018, Intel Corporation.
*
diff --git a/sound/soc/intel/common/soc-acpi-intel-skl-match.c b/sound/soc/intel/common/soc-acpi-intel-skl-match.c
index 0c9c0edd35b3..42fa40a8d932 100644
--- a/sound/soc/intel/common/soc-acpi-intel-skl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-skl-match.c
@@ -1,6 +1,6 @@
// SPDX-License-Identifier: GPL-2.0
/*
- * soc-apci-intel-skl-match.c - tables and support for SKL ACPI enumeration.
+ * soc-acpi-intel-skl-match.c - tables and support for SKL ACPI enumeration.
*
* Copyright (c) 2018, Intel Corporation.
*
diff --git a/sound/soc/qcom/apq8016_sbc.c b/sound/soc/qcom/apq8016_sbc.c
index f60a71990f66..ac75838bbfab 100644
--- a/sound/soc/qcom/apq8016_sbc.c
+++ b/sound/soc/qcom/apq8016_sbc.c
@@ -150,17 +150,17 @@ static struct apq8016_sbc_data *apq8016_sbc_parse_of(struct snd_soc_card *card)
link = data->dai_link;
- dlc = devm_kzalloc(dev, 2 * sizeof(*dlc), GFP_KERNEL);
- if (!dlc)
- return ERR_PTR(-ENOMEM);
+ for_each_child_of_node(node, np) {
+ dlc = devm_kzalloc(dev, 2 * sizeof(*dlc), GFP_KERNEL);
+ if (!dlc)
+ return ERR_PTR(-ENOMEM);
- link->cpus = &dlc[0];
- link->platforms = &dlc[1];
+ link->cpus = &dlc[0];
+ link->platforms = &dlc[1];
- link->num_cpus = 1;
- link->num_platforms = 1;
+ link->num_cpus = 1;
+ link->num_platforms = 1;
- for_each_child_of_node(node, np) {
cpu = of_get_child_by_name(np, "cpu");
codec = of_get_child_by_name(np, "codec");
diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c
index 0a34d0eb8dba..88ebaf6e1880 100644
--- a/sound/soc/rockchip/rockchip_i2s.c
+++ b/sound/soc/rockchip/rockchip_i2s.c
@@ -326,7 +326,6 @@ static int rockchip_i2s_hw_params(struct snd_pcm_substream *substream,
val |= I2S_CHN_4;
break;
case 2:
- case 1:
val |= I2S_CHN_2;
break;
default:
@@ -459,7 +458,7 @@ static struct snd_soc_dai_driver rockchip_i2s_dai = {
},
.capture = {
.stream_name = "Capture",
- .channels_min = 1,
+ .channels_min = 2,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_192000,
.formats = (SNDRV_PCM_FMTBIT_S8 |
@@ -659,7 +658,7 @@ static int rockchip_i2s_probe(struct platform_device *pdev)
}
if (!of_property_read_u32(node, "rockchip,capture-channels", &val)) {
- if (val >= 1 && val <= 8)
+ if (val >= 2 && val <= 8)
soc_dai->capture.channels_max = val;
}
diff --git a/sound/soc/rockchip/rockchip_max98090.c b/sound/soc/rockchip/rockchip_max98090.c
index c5fc24675a33..782e534d4c0d 100644
--- a/sound/soc/rockchip/rockchip_max98090.c
+++ b/sound/soc/rockchip/rockchip_max98090.c
@@ -61,6 +61,37 @@ static const struct snd_kcontrol_new rk_mc_controls[] = {
SOC_DAPM_PIN_SWITCH("Speaker"),
};
+static int rk_jack_event(struct notifier_block *nb, unsigned long event,
+ void *data)
+{
+ struct snd_soc_jack *jack = (struct snd_soc_jack *)data;
+ struct snd_soc_dapm_context *dapm = &jack->card->dapm;
+
+ if (event & SND_JACK_MICROPHONE)
+ snd_soc_dapm_force_enable_pin(dapm, "MICBIAS");
+ else
+ snd_soc_dapm_disable_pin(dapm, "MICBIAS");
+
+ snd_soc_dapm_sync(dapm);
+
+ return 0;
+}
+
+static struct notifier_block rk_jack_nb = {
+ .notifier_call = rk_jack_event,
+};
+
+static int rk_init(struct snd_soc_pcm_runtime *runtime)
+{
+ /*
+ * The jack has already been created in the rk_98090_headset_init()
+ * function.
+ */
+ snd_soc_jack_notifier_register(&headset_jack, &rk_jack_nb);
+
+ return 0;
+}
+
static int rk_aif1_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
@@ -119,6 +150,7 @@ SND_SOC_DAILINK_DEFS(hifi,
static struct snd_soc_dai_link rk_dailink = {
.name = "max98090",
.stream_name = "Audio",
+ .init = rk_init,
.ops = &rk_aif1_ops,
/* set max98090 as slave */
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
diff --git a/sound/soc/samsung/odroid.c b/sound/soc/samsung/odroid.c
index dfb6e460e7eb..f0f5fa9c27d3 100644
--- a/sound/soc/samsung/odroid.c
+++ b/sound/soc/samsung/odroid.c
@@ -284,9 +284,8 @@ static int odroid_audio_probe(struct platform_device *pdev)
}
of_node_put(cpu);
- of_node_put(codec);
if (ret < 0)
- return ret;
+ goto err_put_node;
ret = snd_soc_of_get_dai_link_codecs(dev, codec, codec_link);
if (ret < 0)
@@ -309,7 +308,6 @@ static int odroid_audio_probe(struct platform_device *pdev)
ret = PTR_ERR(priv->clk_i2s_bus);
goto err_put_sclk;
}
- of_node_put(cpu_dai);
ret = devm_snd_soc_register_card(dev, card);
if (ret < 0) {
@@ -317,6 +315,8 @@ static int odroid_audio_probe(struct platform_device *pdev)
goto err_put_clk_i2s;
}
+ of_node_put(cpu_dai);
+ of_node_put(codec);
return 0;
err_put_clk_i2s:
@@ -326,6 +326,8 @@ err_put_sclk:
err_put_cpu_dai:
of_node_put(cpu_dai);
snd_soc_of_put_dai_link_codecs(codec_link);
+err_put_node:
+ of_node_put(codec);
return ret;
}
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index fd6eaae6c0ed..44f899b970c2 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1515,8 +1515,11 @@ static int soc_probe_link_dais(struct snd_soc_card *card,
}
}
- if (dai_link->dai_fmt)
- snd_soc_runtime_set_dai_fmt(rtd, dai_link->dai_fmt);
+ if (dai_link->dai_fmt) {
+ ret = snd_soc_runtime_set_dai_fmt(rtd, dai_link->dai_fmt);
+ if (ret)
+ return ret;
+ }
ret = soc_post_component_init(rtd, dai_link->name);
if (ret)
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index f013b24c050a..2790c00735f3 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -1157,8 +1157,8 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget,
list_add_tail(&widget->work_list, list);
if (custom_stop_condition && custom_stop_condition(widget, dir)) {
- widget->endpoints[dir] = 1;
- return widget->endpoints[dir];
+ list = NULL;
+ custom_stop_condition = NULL;
}
if ((widget->is_ep & SND_SOC_DAPM_DIR_TO_EP(dir)) && widget->connected) {
@@ -1195,8 +1195,8 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget,
*
* Optionally, can be supplied with a function acting as a stopping condition.
* This function takes the dapm widget currently being examined and the walk
- * direction as an arguments, it should return true if the walk should be
- * stopped and false otherwise.
+ * direction as an arguments, it should return true if widgets from that point
+ * in the graph onwards should not be added to the widget list.
*/
static int is_connected_output_ep(struct snd_soc_dapm_widget *widget,
struct list_head *list,
@@ -3706,6 +3706,8 @@ request_failed:
dev_err(dapm->dev, "ASoC: Failed to request %s: %d\n",
w->name, ret);
+ kfree_const(w->sname);
+ kfree(w);
return ERR_PTR(ret);
}
diff --git a/sound/soc/sof/intel/cnl.c b/sound/soc/sof/intel/cnl.c
index f2b392998f20..ffd8d4394537 100644
--- a/sound/soc/sof/intel/cnl.c
+++ b/sound/soc/sof/intel/cnl.c
@@ -101,8 +101,8 @@ static irqreturn_t cnl_ipc_irq_thread(int irq, void *context)
/*
* This interrupt is not shared so no need to return IRQ_NONE.
*/
- dev_err_ratelimited(sdev->dev,
- "error: nothing to do in IRQ thread\n");
+ dev_dbg_ratelimited(sdev->dev,
+ "nothing to do in IPC IRQ thread\n");
}
/* re-enable IPC interrupt */
diff --git a/sound/soc/sof/intel/hda-ipc.c b/sound/soc/sof/intel/hda-ipc.c
index 50244b82600c..2ecba91f5219 100644
--- a/sound/soc/sof/intel/hda-ipc.c
+++ b/sound/soc/sof/intel/hda-ipc.c
@@ -224,8 +224,8 @@ irqreturn_t hda_dsp_ipc_irq_thread(int irq, void *context)
/*
* This interrupt is not shared so no need to return IRQ_NONE.
*/
- dev_err_ratelimited(sdev->dev,
- "error: nothing to do in IRQ thread\n");
+ dev_dbg_ratelimited(sdev->dev,
+ "nothing to do in IPC IRQ thread\n");
}
/* re-enable IPC interrupt */
diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c
index 9b2232908b65..7fa5c61169db 100644
--- a/sound/soc/sunxi/sun4i-i2s.c
+++ b/sound/soc/sunxi/sun4i-i2s.c
@@ -1002,8 +1002,8 @@ static const struct sun4i_i2s_quirks sun50i_a64_codec_i2s_quirks = {
.field_rxchanmap = REG_FIELD(SUN4I_I2S_RX_CHAN_MAP_REG, 0, 31),
.field_txchansel = REG_FIELD(SUN4I_I2S_TX_CHAN_SEL_REG, 0, 2),
.field_rxchansel = REG_FIELD(SUN4I_I2S_RX_CHAN_SEL_REG, 0, 2),
- .get_sr = sun8i_i2s_get_sr_wss,
- .get_wss = sun8i_i2s_get_sr_wss,
+ .get_sr = sun4i_i2s_get_sr,
+ .get_wss = sun4i_i2s_get_wss,
};
static int sun4i_i2s_init_regmap_fields(struct device *dev,
diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c
index ac59b509ead5..bc7bf15ed7a4 100644
--- a/sound/soc/ti/davinci-mcasp.c
+++ b/sound/soc/ti/davinci-mcasp.c
@@ -195,7 +195,7 @@ static inline void mcasp_set_axr_pdir(struct davinci_mcasp *mcasp, bool enable)
{
u32 bit;
- for_each_set_bit(bit, &mcasp->pdir, PIN_BIT_AFSR) {
+ for_each_set_bit(bit, &mcasp->pdir, PIN_BIT_AMUTE) {
if (enable)
mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, BIT(bit));
else
@@ -223,6 +223,7 @@ static void mcasp_start_rx(struct davinci_mcasp *mcasp)
if (mcasp_is_synchronous(mcasp)) {
mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXHCLKRST);
mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXCLKRST);
+ mcasp_set_clk_pdir(mcasp, true);
}
/* Activate serializer(s) */
@@ -1256,6 +1257,28 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream,
return ret;
}
+static int davinci_mcasp_hw_rule_slot_width(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct davinci_mcasp_ruledata *rd = rule->private;
+ struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+ struct snd_mask nfmt;
+ int i, slot_width;
+
+ snd_mask_none(&nfmt);
+ slot_width = rd->mcasp->slot_width;
+
+ for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) {
+ if (snd_mask_test(fmt, i)) {
+ if (snd_pcm_format_width(i) <= slot_width) {
+ snd_mask_set(&nfmt, i);
+ }
+ }
+ }
+
+ return snd_mask_refine(fmt, &nfmt);
+}
+
static const unsigned int davinci_mcasp_dai_rates[] = {
8000, 11025, 16000, 22050, 32000, 44100, 48000, 64000,
88200, 96000, 176400, 192000,
@@ -1377,7 +1400,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
struct davinci_mcasp_ruledata *ruledata =
&mcasp->ruledata[substream->stream];
u32 max_channels = 0;
- int i, dir;
+ int i, dir, ret;
int tdm_slots = mcasp->tdm_slots;
/* Do not allow more then one stream per direction */
@@ -1406,6 +1429,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
max_channels++;
}
ruledata->serializers = max_channels;
+ ruledata->mcasp = mcasp;
max_channels *= tdm_slots;
/*
* If the already active stream has less channels than the calculated
@@ -1431,20 +1455,22 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
0, SNDRV_PCM_HW_PARAM_CHANNELS,
&mcasp->chconstr[substream->stream]);
- if (mcasp->slot_width)
- snd_pcm_hw_constraint_minmax(substream->runtime,
- SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
- 8, mcasp->slot_width);
+ if (mcasp->slot_width) {
+ /* Only allow formats require <= slot_width bits on the bus */
+ ret = snd_pcm_hw_rule_add(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_FORMAT,
+ davinci_mcasp_hw_rule_slot_width,
+ ruledata,
+ SNDRV_PCM_HW_PARAM_FORMAT, -1);
+ if (ret)
+ return ret;
+ }
/*
* If we rely on implicit BCLK divider setting we should
* set constraints based on what we can provide.
*/
if (mcasp->bclk_master && mcasp->bclk_div == 0 && mcasp->sysclk_freq) {
- int ret;
-
- ruledata->mcasp = mcasp;
-
ret = snd_pcm_hw_rule_add(substream->runtime, 0,
SNDRV_PCM_HW_PARAM_RATE,
davinci_mcasp_hw_rule_rate,
diff --git a/sound/usb/helper.c b/sound/usb/helper.c
index 71d5f540334a..4c12cc5b53fd 100644
--- a/sound/usb/helper.c
+++ b/sound/usb/helper.c
@@ -72,7 +72,7 @@ int snd_usb_pipe_sanity_check(struct usb_device *dev, unsigned int pipe)
struct usb_host_endpoint *ep;
ep = usb_pipe_endpoint(dev, pipe);
- if (usb_pipetype(pipe) != pipetypes[usb_endpoint_type(&ep->desc)])
+ if (!ep || usb_pipetype(pipe) != pipetypes[usb_endpoint_type(&ep->desc)])
return -EINVAL;
return 0;
}
diff --git a/sound/usb/stream.c b/sound/usb/stream.c
index 7ee9d17d0143..e852c7fd6109 100644
--- a/sound/usb/stream.c
+++ b/sound/usb/stream.c
@@ -1043,6 +1043,7 @@ found_clock:
pd = kzalloc(sizeof(*pd), GFP_KERNEL);
if (!pd) {
+ kfree(fp->chmap);
kfree(fp->rate_table);
kfree(fp);
return NULL;