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-rw-r--r--sound/core/Makefile2
-rw-r--r--sound/core/control_compat.c3
-rw-r--r--sound/core/jack.c7
-rw-r--r--sound/core/oss/pcm_oss.c51
-rw-r--r--sound/core/oss/pcm_plugin.c5
-rw-r--r--sound/core/pcm.c9
-rw-r--r--sound/core/pcm_lib.c5
-rw-r--r--sound/core/pcm_memory.c11
-rw-r--r--sound/core/pcm_misc.c2
-rw-r--r--sound/core/pcm_native.c114
-rw-r--r--sound/core/seq/seq_queue.c14
-rw-r--r--sound/drivers/opl3/opl3_midi.c2
-rw-r--r--sound/firewire/fcp.c4
-rw-r--r--sound/firewire/fireworks/fireworks_hwdep.c1
-rw-r--r--sound/hda/ext/hdac_ext_stream.c46
-rw-r--r--sound/hda/hdac_stream.c4
-rw-r--r--sound/isa/Kconfig2
-rw-r--r--sound/isa/cs423x/cs4236.c8
-rw-r--r--sound/isa/gus/gus_dma.c2
-rw-r--r--sound/pci/Kconfig1
-rw-r--r--sound/pci/ac97/ac97_codec.c4
-rw-r--r--sound/pci/cmipci.c3
-rw-r--r--sound/pci/ctxfi/ctamixer.c14
-rw-r--r--sound/pci/ctxfi/ctdaio.c16
-rw-r--r--sound/pci/ctxfi/ctresource.c7
-rw-r--r--sound/pci/ctxfi/ctresource.h4
-rw-r--r--sound/pci/ctxfi/ctsrc.c7
-rw-r--r--sound/pci/hda/hda_codec.c3
-rw-r--r--sound/pci/hda/hda_intel.c5
-rw-r--r--sound/pci/hda/patch_realtek.c95
-rw-r--r--sound/soc/atmel/atmel_ssc_dai.c5
-rw-r--r--sound/soc/atmel/sam9g20_wm8731.c62
-rw-r--r--sound/soc/codecs/Kconfig5
-rw-r--r--sound/soc/codecs/cpcap.c2
-rw-r--r--sound/soc/codecs/cs4265.c3
-rw-r--r--sound/soc/codecs/da7219.c14
-rw-r--r--sound/soc/codecs/max9759.c3
-rw-r--r--sound/soc/codecs/msm8916-wcd-analog.c16
-rw-r--r--sound/soc/codecs/msm8916-wcd-digital.c14
-rw-r--r--sound/soc/codecs/mt6358.c4
-rw-r--r--sound/soc/codecs/nau8824.c40
-rw-r--r--sound/soc/codecs/rt5663.c14
-rw-r--r--sound/soc/codecs/rt5668.c12
-rw-r--r--sound/soc/codecs/rt5682.c12
-rw-r--r--sound/soc/codecs/wm8350.c28
-rw-r--r--sound/soc/codecs/wm8731.c19
-rw-r--r--sound/soc/codecs/wm8958-dsp2.c8
-rw-r--r--sound/soc/fsl/imx-es8328.c1
-rw-r--r--sound/soc/fsl/pcm030-audio-fabric.c11
-rw-r--r--sound/soc/mediatek/mt8173/mt8173-max98090.c3
-rw-r--r--sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c2
-rw-r--r--sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c2
-rw-r--r--sound/soc/mediatek/mt8173/mt8173-rt5650.c2
-rw-r--r--sound/soc/meson/g12a-tohdmitx.c2
-rw-r--r--sound/soc/mxs/mxs-saif.c5
-rw-r--r--sound/soc/mxs/mxs-sgtl5000.c3
-rw-r--r--sound/soc/qcom/qdsp6/q6routing.c6
-rw-r--r--sound/soc/samsung/idma.c2
-rw-r--r--sound/soc/sh/fsi.c19
-rw-r--r--sound/soc/soc-compress.c5
-rw-r--r--sound/soc/soc-core.c2
-rw-r--r--sound/soc/soc-dapm.c35
-rw-r--r--sound/soc/soc-ops.c58
-rw-r--r--sound/soc/soc-topology.c6
-rw-r--r--sound/soc/sof/intel/hda-dai.c7
-rw-r--r--sound/soc/sof/intel/hda-loader.c9
-rw-r--r--sound/soc/sti/uniperif_player.c6
-rw-r--r--sound/soc/sti/uniperif_reader.c2
-rw-r--r--sound/soc/ti/davinci-i2s.c5
-rw-r--r--sound/soc/uniphier/Kconfig2
-rw-r--r--sound/soc/xilinx/xlnx_formatter_pcm.c27
-rw-r--r--sound/spi/at73c213.c27
-rw-r--r--sound/usb/midi.c1
-rw-r--r--sound/usb/mixer_quirks.c7
-rw-r--r--sound/usb/quirks-table.h10
-rw-r--r--sound/usb/usbaudio.h2
-rw-r--r--sound/x86/intel_hdmi_audio.c2
77 files changed, 695 insertions, 288 deletions
diff --git a/sound/core/Makefile b/sound/core/Makefile
index ee4a4a6b99ba..d123587c0fd8 100644
--- a/sound/core/Makefile
+++ b/sound/core/Makefile
@@ -9,7 +9,9 @@ ifneq ($(CONFIG_SND_PROC_FS),)
snd-y += info.o
snd-$(CONFIG_SND_OSSEMUL) += info_oss.o
endif
+ifneq ($(CONFIG_M68K),y)
snd-$(CONFIG_ISA_DMA_API) += isadma.o
+endif
snd-$(CONFIG_SND_OSSEMUL) += sound_oss.o
snd-$(CONFIG_SND_VMASTER) += vmaster.o
snd-$(CONFIG_SND_JACK) += ctljack.o jack.o
diff --git a/sound/core/control_compat.c b/sound/core/control_compat.c
index d55be1db1a8a..cca3ed9b0629 100644
--- a/sound/core/control_compat.c
+++ b/sound/core/control_compat.c
@@ -266,6 +266,7 @@ static int copy_ctl_value_to_user(void __user *userdata,
struct snd_ctl_elem_value *data,
int type, int count)
{
+ struct snd_ctl_elem_value32 __user *data32 = userdata;
int i, size;
if (type == SNDRV_CTL_ELEM_TYPE_BOOLEAN ||
@@ -282,6 +283,8 @@ static int copy_ctl_value_to_user(void __user *userdata,
if (copy_to_user(valuep, data->value.bytes.data, size))
return -EFAULT;
}
+ if (copy_to_user(&data32->id, &data->id, sizeof(data32->id)))
+ return -EFAULT;
return 0;
}
diff --git a/sound/core/jack.c b/sound/core/jack.c
index fb26196571a7..b00ae6f39f05 100644
--- a/sound/core/jack.c
+++ b/sound/core/jack.c
@@ -54,10 +54,13 @@ static int snd_jack_dev_free(struct snd_device *device)
struct snd_card *card = device->card;
struct snd_jack_kctl *jack_kctl, *tmp_jack_kctl;
+ down_write(&card->controls_rwsem);
list_for_each_entry_safe(jack_kctl, tmp_jack_kctl, &jack->kctl_list, list) {
list_del_init(&jack_kctl->list);
snd_ctl_remove(card, jack_kctl->kctl);
}
+ up_write(&card->controls_rwsem);
+
if (jack->private_free)
jack->private_free(jack);
@@ -220,6 +223,10 @@ int snd_jack_new(struct snd_card *card, const char *id, int type,
return -ENOMEM;
jack->id = kstrdup(id, GFP_KERNEL);
+ if (jack->id == NULL) {
+ kfree(jack);
+ return -ENOMEM;
+ }
/* don't creat input device for phantom jack */
if (!phantom_jack) {
diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c
index 0b03777d0111..ad4e0af2d0d0 100644
--- a/sound/core/oss/pcm_oss.c
+++ b/sound/core/oss/pcm_oss.c
@@ -147,7 +147,7 @@ snd_pcm_hw_param_value_min(const struct snd_pcm_hw_params *params,
*
* Return the maximum value for field PAR.
*/
-static unsigned int
+static int
snd_pcm_hw_param_value_max(const struct snd_pcm_hw_params *params,
snd_pcm_hw_param_t var, int *dir)
{
@@ -682,18 +682,24 @@ static int snd_pcm_oss_period_size(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *oss_params,
struct snd_pcm_hw_params *slave_params)
{
- size_t s;
- size_t oss_buffer_size, oss_period_size, oss_periods;
- size_t min_period_size, max_period_size;
+ ssize_t s;
+ ssize_t oss_buffer_size;
+ ssize_t oss_period_size, oss_periods;
+ ssize_t min_period_size, max_period_size;
struct snd_pcm_runtime *runtime = substream->runtime;
size_t oss_frame_size;
oss_frame_size = snd_pcm_format_physical_width(params_format(oss_params)) *
params_channels(oss_params) / 8;
+ oss_buffer_size = snd_pcm_hw_param_value_max(slave_params,
+ SNDRV_PCM_HW_PARAM_BUFFER_SIZE,
+ NULL);
+ if (oss_buffer_size <= 0)
+ return -EINVAL;
oss_buffer_size = snd_pcm_plug_client_size(substream,
- snd_pcm_hw_param_value_max(slave_params, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, NULL)) * oss_frame_size;
- if (!oss_buffer_size)
+ oss_buffer_size * oss_frame_size);
+ if (oss_buffer_size <= 0)
return -EINVAL;
oss_buffer_size = rounddown_pow_of_two(oss_buffer_size);
if (atomic_read(&substream->mmap_count)) {
@@ -730,7 +736,7 @@ static int snd_pcm_oss_period_size(struct snd_pcm_substream *substream,
min_period_size = snd_pcm_plug_client_size(substream,
snd_pcm_hw_param_value_min(slave_params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, NULL));
- if (min_period_size) {
+ if (min_period_size > 0) {
min_period_size *= oss_frame_size;
min_period_size = roundup_pow_of_two(min_period_size);
if (oss_period_size < min_period_size)
@@ -739,7 +745,7 @@ static int snd_pcm_oss_period_size(struct snd_pcm_substream *substream,
max_period_size = snd_pcm_plug_client_size(substream,
snd_pcm_hw_param_value_max(slave_params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, NULL));
- if (max_period_size) {
+ if (max_period_size > 0) {
max_period_size *= oss_frame_size;
max_period_size = rounddown_pow_of_two(max_period_size);
if (oss_period_size > max_period_size)
@@ -752,7 +758,7 @@ static int snd_pcm_oss_period_size(struct snd_pcm_substream *substream,
oss_periods = substream->oss.setup.periods;
s = snd_pcm_hw_param_value_max(slave_params, SNDRV_PCM_HW_PARAM_PERIODS, NULL);
- if (runtime->oss.maxfrags && s > runtime->oss.maxfrags)
+ if (s > 0 && runtime->oss.maxfrags && s > runtime->oss.maxfrags)
s = runtime->oss.maxfrags;
if (oss_periods > s)
oss_periods = s;
@@ -768,6 +774,11 @@ static int snd_pcm_oss_period_size(struct snd_pcm_substream *substream,
if (oss_period_size < 16)
return -EINVAL;
+
+ /* don't allocate too large period; 1MB period must be enough */
+ if (oss_period_size > 1024 * 1024)
+ return -ENOMEM;
+
runtime->oss.period_bytes = oss_period_size;
runtime->oss.period_frames = 1;
runtime->oss.periods = oss_periods;
@@ -878,8 +889,15 @@ static int snd_pcm_oss_change_params_locked(struct snd_pcm_substream *substream)
err = -EINVAL;
goto failure;
}
- choose_rate(substream, sparams, runtime->oss.rate);
- snd_pcm_hw_param_near(substream, sparams, SNDRV_PCM_HW_PARAM_CHANNELS, runtime->oss.channels, NULL);
+
+ err = choose_rate(substream, sparams, runtime->oss.rate);
+ if (err < 0)
+ goto failure;
+ err = snd_pcm_hw_param_near(substream, sparams,
+ SNDRV_PCM_HW_PARAM_CHANNELS,
+ runtime->oss.channels, NULL);
+ if (err < 0)
+ goto failure;
format = snd_pcm_oss_format_from(runtime->oss.format);
@@ -1032,10 +1050,9 @@ static int snd_pcm_oss_change_params_locked(struct snd_pcm_substream *substream)
goto failure;
}
#endif
- oss_period_size *= oss_frame_size;
-
- oss_buffer_size = oss_period_size * runtime->oss.periods;
- if (oss_buffer_size < 0) {
+ oss_period_size = array_size(oss_period_size, oss_frame_size);
+ oss_buffer_size = array_size(oss_period_size, runtime->oss.periods);
+ if (oss_buffer_size <= 0) {
err = -EINVAL;
goto failure;
}
@@ -1946,7 +1963,7 @@ static int snd_pcm_oss_set_fragment1(struct snd_pcm_substream *substream, unsign
if (runtime->oss.subdivision || runtime->oss.fragshift)
return -EINVAL;
fragshift = val & 0xffff;
- if (fragshift >= 31)
+ if (fragshift >= 25) /* should be large enough */
return -EINVAL;
runtime->oss.fragshift = fragshift;
runtime->oss.maxfrags = (val >> 16) & 0xffff;
@@ -2042,7 +2059,7 @@ static int snd_pcm_oss_set_trigger(struct snd_pcm_oss_file *pcm_oss_file, int tr
int err, cmd;
#ifdef OSS_DEBUG
- pcm_dbg(substream->pcm, "pcm_oss: trigger = 0x%x\n", trigger);
+ pr_debug("pcm_oss: trigger = 0x%x\n", trigger);
#endif
psubstream = pcm_oss_file->streams[SNDRV_PCM_STREAM_PLAYBACK];
diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c
index da400da1fafe..8b7bbabeea24 100644
--- a/sound/core/oss/pcm_plugin.c
+++ b/sound/core/oss/pcm_plugin.c
@@ -61,7 +61,10 @@ static int snd_pcm_plugin_alloc(struct snd_pcm_plugin *plugin, snd_pcm_uframes_t
}
if ((width = snd_pcm_format_physical_width(format->format)) < 0)
return width;
- size = frames * format->channels * width;
+ size = array3_size(frames, format->channels, width);
+ /* check for too large period size once again */
+ if (size > 1024 * 1024)
+ return -ENOMEM;
if (snd_BUG_ON(size % 8))
return -ENXIO;
size /= 8;
diff --git a/sound/core/pcm.c b/sound/core/pcm.c
index 9a72d641743d..3561cdceaadc 100644
--- a/sound/core/pcm.c
+++ b/sound/core/pcm.c
@@ -810,7 +810,11 @@ EXPORT_SYMBOL(snd_pcm_new_internal);
static void free_chmap(struct snd_pcm_str *pstr)
{
if (pstr->chmap_kctl) {
- snd_ctl_remove(pstr->pcm->card, pstr->chmap_kctl);
+ struct snd_card *card = pstr->pcm->card;
+
+ down_write(&card->controls_rwsem);
+ snd_ctl_remove(card, pstr->chmap_kctl);
+ up_write(&card->controls_rwsem);
pstr->chmap_kctl = NULL;
}
}
@@ -965,6 +969,8 @@ int snd_pcm_attach_substream(struct snd_pcm *pcm, int stream,
init_waitqueue_head(&runtime->tsleep);
runtime->status->state = SNDRV_PCM_STATE_OPEN;
+ mutex_init(&runtime->buffer_mutex);
+ atomic_set(&runtime->buffer_accessing, 0);
substream->runtime = runtime;
substream->private_data = pcm->private_data;
@@ -996,6 +1002,7 @@ void snd_pcm_detach_substream(struct snd_pcm_substream *substream)
substream->runtime = NULL;
if (substream->timer)
spin_unlock_irq(&substream->timer->lock);
+ mutex_destroy(&runtime->buffer_mutex);
kfree(runtime);
put_pid(substream->pid);
substream->pid = NULL;
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index fd300c3addde..1bce55533519 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -2211,10 +2211,15 @@ snd_pcm_sframes_t __snd_pcm_lib_xfer(struct snd_pcm_substream *substream,
err = -EINVAL;
goto _end_unlock;
}
+ if (!atomic_inc_unless_negative(&runtime->buffer_accessing)) {
+ err = -EBUSY;
+ goto _end_unlock;
+ }
snd_pcm_stream_unlock_irq(substream);
err = writer(substream, appl_ofs, data, offset, frames,
transfer);
snd_pcm_stream_lock_irq(substream);
+ atomic_dec(&runtime->buffer_accessing);
if (err < 0)
goto _end_unlock;
err = pcm_accessible_state(runtime);
diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c
index 7600dcdf5fd4..9aea1d6fb054 100644
--- a/sound/core/pcm_memory.c
+++ b/sound/core/pcm_memory.c
@@ -133,19 +133,20 @@ static void snd_pcm_lib_preallocate_proc_write(struct snd_info_entry *entry,
size_t size;
struct snd_dma_buffer new_dmab;
+ mutex_lock(&substream->pcm->open_mutex);
if (substream->runtime) {
buffer->error = -EBUSY;
- return;
+ goto unlock;
}
if (!snd_info_get_line(buffer, line, sizeof(line))) {
snd_info_get_str(str, line, sizeof(str));
size = simple_strtoul(str, NULL, 10) * 1024;
if ((size != 0 && size < 8192) || size > substream->dma_max) {
buffer->error = -EINVAL;
- return;
+ goto unlock;
}
if (substream->dma_buffer.bytes == size)
- return;
+ goto unlock;
memset(&new_dmab, 0, sizeof(new_dmab));
new_dmab.dev = substream->dma_buffer.dev;
if (size > 0) {
@@ -153,7 +154,7 @@ static void snd_pcm_lib_preallocate_proc_write(struct snd_info_entry *entry,
substream->dma_buffer.dev.dev,
size, &new_dmab) < 0) {
buffer->error = -ENOMEM;
- return;
+ goto unlock;
}
substream->buffer_bytes_max = size;
} else {
@@ -165,6 +166,8 @@ static void snd_pcm_lib_preallocate_proc_write(struct snd_info_entry *entry,
} else {
buffer->error = -EINVAL;
}
+ unlock:
+ mutex_unlock(&substream->pcm->open_mutex);
}
static inline void preallocate_info_init(struct snd_pcm_substream *substream)
diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c
index c4eb561d2008..0956be39b035 100644
--- a/sound/core/pcm_misc.c
+++ b/sound/core/pcm_misc.c
@@ -423,7 +423,7 @@ int snd_pcm_format_set_silence(snd_pcm_format_t format, void *data, unsigned int
return 0;
width = pcm_formats[(INT)format].phys; /* physical width */
pat = pcm_formats[(INT)format].silence;
- if (! width)
+ if (!width || !pat)
return -EINVAL;
/* signed or 1 byte data */
if (pcm_formats[(INT)format].signd == 1 || width <= 8) {
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 0c5b7a54ca81..57a4991fa0f3 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -630,6 +630,30 @@ static int snd_pcm_hw_params_choose(struct snd_pcm_substream *pcm,
return 0;
}
+/* acquire buffer_mutex; if it's in r/w operation, return -EBUSY, otherwise
+ * block the further r/w operations
+ */
+static int snd_pcm_buffer_access_lock(struct snd_pcm_runtime *runtime)
+{
+ if (!atomic_dec_unless_positive(&runtime->buffer_accessing))
+ return -EBUSY;
+ mutex_lock(&runtime->buffer_mutex);
+ return 0; /* keep buffer_mutex, unlocked by below */
+}
+
+/* release buffer_mutex and clear r/w access flag */
+static void snd_pcm_buffer_access_unlock(struct snd_pcm_runtime *runtime)
+{
+ mutex_unlock(&runtime->buffer_mutex);
+ atomic_inc(&runtime->buffer_accessing);
+}
+
+#if IS_ENABLED(CONFIG_SND_PCM_OSS)
+#define is_oss_stream(substream) ((substream)->oss.oss)
+#else
+#define is_oss_stream(substream) false
+#endif
+
static int snd_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
@@ -641,22 +665,25 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream,
if (PCM_RUNTIME_CHECK(substream))
return -ENXIO;
runtime = substream->runtime;
+ err = snd_pcm_buffer_access_lock(runtime);
+ if (err < 0)
+ return err;
snd_pcm_stream_lock_irq(substream);
switch (runtime->status->state) {
case SNDRV_PCM_STATE_OPEN:
case SNDRV_PCM_STATE_SETUP:
case SNDRV_PCM_STATE_PREPARED:
+ if (!is_oss_stream(substream) &&
+ atomic_read(&substream->mmap_count))
+ err = -EBADFD;
break;
default:
- snd_pcm_stream_unlock_irq(substream);
- return -EBADFD;
+ err = -EBADFD;
+ break;
}
snd_pcm_stream_unlock_irq(substream);
-#if IS_ENABLED(CONFIG_SND_PCM_OSS)
- if (!substream->oss.oss)
-#endif
- if (atomic_read(&substream->mmap_count))
- return -EBADFD;
+ if (err)
+ goto unlock;
params->rmask = ~0U;
err = snd_pcm_hw_refine(substream, params);
@@ -733,14 +760,19 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream,
if ((usecs = period_to_usecs(runtime)) >= 0)
pm_qos_add_request(&substream->latency_pm_qos_req,
PM_QOS_CPU_DMA_LATENCY, usecs);
- return 0;
+ err = 0;
_error:
- /* hardware might be unusable from this time,
- so we force application to retry to set
- the correct hardware parameter settings */
- snd_pcm_set_state(substream, SNDRV_PCM_STATE_OPEN);
- if (substream->ops->hw_free != NULL)
- substream->ops->hw_free(substream);
+ if (err) {
+ /* hardware might be unusable from this time,
+ * so we force application to retry to set
+ * the correct hardware parameter settings
+ */
+ snd_pcm_set_state(substream, SNDRV_PCM_STATE_OPEN);
+ if (substream->ops->hw_free != NULL)
+ substream->ops->hw_free(substream);
+ }
+ unlock:
+ snd_pcm_buffer_access_unlock(runtime);
return err;
}
@@ -773,22 +805,29 @@ static int snd_pcm_hw_free(struct snd_pcm_substream *substream)
if (PCM_RUNTIME_CHECK(substream))
return -ENXIO;
runtime = substream->runtime;
+ result = snd_pcm_buffer_access_lock(runtime);
+ if (result < 0)
+ return result;
snd_pcm_stream_lock_irq(substream);
switch (runtime->status->state) {
case SNDRV_PCM_STATE_SETUP:
case SNDRV_PCM_STATE_PREPARED:
+ if (atomic_read(&substream->mmap_count))
+ result = -EBADFD;
break;
default:
- snd_pcm_stream_unlock_irq(substream);
- return -EBADFD;
+ result = -EBADFD;
+ break;
}
snd_pcm_stream_unlock_irq(substream);
- if (atomic_read(&substream->mmap_count))
- return -EBADFD;
+ if (result)
+ goto unlock;
if (substream->ops->hw_free)
result = substream->ops->hw_free(substream);
snd_pcm_set_state(substream, SNDRV_PCM_STATE_OPEN);
pm_qos_remove_request(&substream->latency_pm_qos_req);
+ unlock:
+ snd_pcm_buffer_access_unlock(runtime);
return result;
}
@@ -1025,15 +1064,17 @@ struct action_ops {
*/
static int snd_pcm_action_group(const struct action_ops *ops,
struct snd_pcm_substream *substream,
- int state, int do_lock)
+ int state, int stream_lock)
{
struct snd_pcm_substream *s = NULL;
struct snd_pcm_substream *s1;
int res = 0, depth = 1;
snd_pcm_group_for_each_entry(s, substream) {
- if (do_lock && s != substream) {
- if (s->pcm->nonatomic)
+ if (s != substream) {
+ if (!stream_lock)
+ mutex_lock_nested(&s->runtime->buffer_mutex, depth);
+ else if (s->pcm->nonatomic)
mutex_lock_nested(&s->self_group.mutex, depth);
else
spin_lock_nested(&s->self_group.lock, depth);
@@ -1061,18 +1102,18 @@ static int snd_pcm_action_group(const struct action_ops *ops,
ops->post_action(s, state);
}
_unlock:
- if (do_lock) {
- /* unlock streams */
- snd_pcm_group_for_each_entry(s1, substream) {
- if (s1 != substream) {
- if (s1->pcm->nonatomic)
- mutex_unlock(&s1->self_group.mutex);
- else
- spin_unlock(&s1->self_group.lock);
- }
- if (s1 == s) /* end */
- break;
+ /* unlock streams */
+ snd_pcm_group_for_each_entry(s1, substream) {
+ if (s1 != substream) {
+ if (!stream_lock)
+ mutex_unlock(&s1->runtime->buffer_mutex);
+ else if (s1->pcm->nonatomic)
+ mutex_unlock(&s1->self_group.mutex);
+ else
+ spin_unlock(&s1->self_group.lock);
}
+ if (s1 == s) /* end */
+ break;
}
return res;
}
@@ -1202,10 +1243,15 @@ static int snd_pcm_action_nonatomic(const struct action_ops *ops,
/* Guarantee the group members won't change during non-atomic action */
down_read(&snd_pcm_link_rwsem);
+ res = snd_pcm_buffer_access_lock(substream->runtime);
+ if (res < 0)
+ goto unlock;
if (snd_pcm_stream_linked(substream))
res = snd_pcm_action_group(ops, substream, state, 0);
else
res = snd_pcm_action_single(ops, substream, state);
+ snd_pcm_buffer_access_unlock(substream->runtime);
+ unlock:
up_read(&snd_pcm_link_rwsem);
return res;
}
@@ -1656,21 +1702,25 @@ static int snd_pcm_do_reset(struct snd_pcm_substream *substream, int state)
int err = substream->ops->ioctl(substream, SNDRV_PCM_IOCTL1_RESET, NULL);
if (err < 0)
return err;
+ snd_pcm_stream_lock_irq(substream);
runtime->hw_ptr_base = 0;
runtime->hw_ptr_interrupt = runtime->status->hw_ptr -
runtime->status->hw_ptr % runtime->period_size;
runtime->silence_start = runtime->status->hw_ptr;
runtime->silence_filled = 0;
+ snd_pcm_stream_unlock_irq(substream);
return 0;
}
static void snd_pcm_post_reset(struct snd_pcm_substream *substream, int state)
{
struct snd_pcm_runtime *runtime = substream->runtime;
+ snd_pcm_stream_lock_irq(substream);
runtime->control->appl_ptr = runtime->status->hw_ptr;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
runtime->silence_size > 0)
snd_pcm_playback_silence(substream, ULONG_MAX);
+ snd_pcm_stream_unlock_irq(substream);
}
static const struct action_ops snd_pcm_action_reset = {
diff --git a/sound/core/seq/seq_queue.c b/sound/core/seq/seq_queue.c
index 71a6ea62c3be..4ff0b927230c 100644
--- a/sound/core/seq/seq_queue.c
+++ b/sound/core/seq/seq_queue.c
@@ -234,12 +234,15 @@ struct snd_seq_queue *snd_seq_queue_find_name(char *name)
/* -------------------------------------------------------- */
+#define MAX_CELL_PROCESSES_IN_QUEUE 1000
+
void snd_seq_check_queue(struct snd_seq_queue *q, int atomic, int hop)
{
unsigned long flags;
struct snd_seq_event_cell *cell;
snd_seq_tick_time_t cur_tick;
snd_seq_real_time_t cur_time;
+ int processed = 0;
if (q == NULL)
return;
@@ -262,6 +265,8 @@ void snd_seq_check_queue(struct snd_seq_queue *q, int atomic, int hop)
if (!cell)
break;
snd_seq_dispatch_event(cell, atomic, hop);
+ if (++processed >= MAX_CELL_PROCESSES_IN_QUEUE)
+ goto out; /* the rest processed at the next batch */
}
/* Process time queue... */
@@ -271,14 +276,19 @@ void snd_seq_check_queue(struct snd_seq_queue *q, int atomic, int hop)
if (!cell)
break;
snd_seq_dispatch_event(cell, atomic, hop);
+ if (++processed >= MAX_CELL_PROCESSES_IN_QUEUE)
+ goto out; /* the rest processed at the next batch */
}
+ out:
/* free lock */
spin_lock_irqsave(&q->check_lock, flags);
if (q->check_again) {
q->check_again = 0;
- spin_unlock_irqrestore(&q->check_lock, flags);
- goto __again;
+ if (processed < MAX_CELL_PROCESSES_IN_QUEUE) {
+ spin_unlock_irqrestore(&q->check_lock, flags);
+ goto __again;
+ }
}
q->check_blocked = 0;
spin_unlock_irqrestore(&q->check_lock, flags);
diff --git a/sound/drivers/opl3/opl3_midi.c b/sound/drivers/opl3/opl3_midi.c
index 280cc79870cf..ce38ec09d408 100644
--- a/sound/drivers/opl3/opl3_midi.c
+++ b/sound/drivers/opl3/opl3_midi.c
@@ -398,7 +398,7 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan)
}
if (instr_4op) {
vp2 = &opl3->voices[voice + 3];
- if (vp->state > 0) {
+ if (vp2->state > 0) {
opl3_reg = reg_side | (OPL3_REG_KEYON_BLOCK +
voice_offset + 3);
reg_val = vp->keyon_reg & ~OPL3_KEYON_BIT;
diff --git a/sound/firewire/fcp.c b/sound/firewire/fcp.c
index bbfbebf4affb..df44dd5dc4b2 100644
--- a/sound/firewire/fcp.c
+++ b/sound/firewire/fcp.c
@@ -240,9 +240,7 @@ int fcp_avc_transaction(struct fw_unit *unit,
t.response_match_bytes = response_match_bytes;
t.state = STATE_PENDING;
init_waitqueue_head(&t.wait);
-
- if (*(const u8 *)command == 0x00 || *(const u8 *)command == 0x03)
- t.deferrable = true;
+ t.deferrable = (*(const u8 *)command == 0x00 || *(const u8 *)command == 0x03);
spin_lock_irq(&transactions_lock);
list_add_tail(&t.list, &transactions);
diff --git a/sound/firewire/fireworks/fireworks_hwdep.c b/sound/firewire/fireworks/fireworks_hwdep.c
index e93eb4616c5f..c739173c668f 100644
--- a/sound/firewire/fireworks/fireworks_hwdep.c
+++ b/sound/firewire/fireworks/fireworks_hwdep.c
@@ -34,6 +34,7 @@ hwdep_read_resp_buf(struct snd_efw *efw, char __user *buf, long remained,
type = SNDRV_FIREWIRE_EVENT_EFW_RESPONSE;
if (copy_to_user(buf, &type, sizeof(type)))
return -EFAULT;
+ count += sizeof(type);
remained -= sizeof(type);
buf += sizeof(type);
diff --git a/sound/hda/ext/hdac_ext_stream.c b/sound/hda/ext/hdac_ext_stream.c
index 6b1b4b834bae..04f4070fbf36 100644
--- a/sound/hda/ext/hdac_ext_stream.c
+++ b/sound/hda/ext/hdac_ext_stream.c
@@ -106,20 +106,14 @@ void snd_hdac_stream_free_all(struct hdac_bus *bus)
}
EXPORT_SYMBOL_GPL(snd_hdac_stream_free_all);
-/**
- * snd_hdac_ext_stream_decouple - decouple the hdac stream
- * @bus: HD-audio core bus
- * @stream: HD-audio ext core stream object to initialize
- * @decouple: flag to decouple
- */
-void snd_hdac_ext_stream_decouple(struct hdac_bus *bus,
- struct hdac_ext_stream *stream, bool decouple)
+void snd_hdac_ext_stream_decouple_locked(struct hdac_bus *bus,
+ struct hdac_ext_stream *stream,
+ bool decouple)
{
struct hdac_stream *hstream = &stream->hstream;
u32 val;
int mask = AZX_PPCTL_PROCEN(hstream->index);
- spin_lock_irq(&bus->reg_lock);
val = readw(bus->ppcap + AZX_REG_PP_PPCTL) & mask;
if (decouple && !val)
@@ -128,6 +122,20 @@ void snd_hdac_ext_stream_decouple(struct hdac_bus *bus,
snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, mask, 0);
stream->decoupled = decouple;
+}
+EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_decouple_locked);
+
+/**
+ * snd_hdac_ext_stream_decouple - decouple the hdac stream
+ * @bus: HD-audio core bus
+ * @stream: HD-audio ext core stream object to initialize
+ * @decouple: flag to decouple
+ */
+void snd_hdac_ext_stream_decouple(struct hdac_bus *bus,
+ struct hdac_ext_stream *stream, bool decouple)
+{
+ spin_lock_irq(&bus->reg_lock);
+ snd_hdac_ext_stream_decouple_locked(bus, stream, decouple);
spin_unlock_irq(&bus->reg_lock);
}
EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_decouple);
@@ -252,6 +260,7 @@ hdac_ext_link_stream_assign(struct hdac_bus *bus,
return NULL;
}
+ spin_lock_irq(&bus->reg_lock);
list_for_each_entry(stream, &bus->stream_list, list) {
struct hdac_ext_stream *hstream = container_of(stream,
struct hdac_ext_stream,
@@ -266,17 +275,16 @@ hdac_ext_link_stream_assign(struct hdac_bus *bus,
}
if (!hstream->link_locked) {
- snd_hdac_ext_stream_decouple(bus, hstream, true);
+ snd_hdac_ext_stream_decouple_locked(bus, hstream, true);
res = hstream;
break;
}
}
if (res) {
- spin_lock_irq(&bus->reg_lock);
res->link_locked = 1;
res->link_substream = substream;
- spin_unlock_irq(&bus->reg_lock);
}
+ spin_unlock_irq(&bus->reg_lock);
return res;
}
@@ -292,6 +300,7 @@ hdac_ext_host_stream_assign(struct hdac_bus *bus,
return NULL;
}
+ spin_lock_irq(&bus->reg_lock);
list_for_each_entry(stream, &bus->stream_list, list) {
struct hdac_ext_stream *hstream = container_of(stream,
struct hdac_ext_stream,
@@ -301,18 +310,17 @@ hdac_ext_host_stream_assign(struct hdac_bus *bus,
if (!stream->opened) {
if (!hstream->decoupled)
- snd_hdac_ext_stream_decouple(bus, hstream, true);
+ snd_hdac_ext_stream_decouple_locked(bus, hstream, true);
res = hstream;
break;
}
}
if (res) {
- spin_lock_irq(&bus->reg_lock);
res->hstream.opened = 1;
res->hstream.running = 0;
res->hstream.substream = substream;
- spin_unlock_irq(&bus->reg_lock);
}
+ spin_unlock_irq(&bus->reg_lock);
return res;
}
@@ -378,15 +386,17 @@ void snd_hdac_ext_stream_release(struct hdac_ext_stream *stream, int type)
break;
case HDAC_EXT_STREAM_TYPE_HOST:
+ spin_lock_irq(&bus->reg_lock);
if (stream->decoupled && !stream->link_locked)
- snd_hdac_ext_stream_decouple(bus, stream, false);
+ snd_hdac_ext_stream_decouple_locked(bus, stream, false);
+ spin_unlock_irq(&bus->reg_lock);
snd_hdac_stream_release(&stream->hstream);
break;
case HDAC_EXT_STREAM_TYPE_LINK:
- if (stream->decoupled && !stream->hstream.opened)
- snd_hdac_ext_stream_decouple(bus, stream, false);
spin_lock_irq(&bus->reg_lock);
+ if (stream->decoupled && !stream->hstream.opened)
+ snd_hdac_ext_stream_decouple_locked(bus, stream, false);
stream->link_locked = 0;
stream->link_substream = NULL;
spin_unlock_irq(&bus->reg_lock);
diff --git a/sound/hda/hdac_stream.c b/sound/hda/hdac_stream.c
index 682ed39f79b0..b299b8b7f871 100644
--- a/sound/hda/hdac_stream.c
+++ b/sound/hda/hdac_stream.c
@@ -289,6 +289,7 @@ struct hdac_stream *snd_hdac_stream_assign(struct hdac_bus *bus,
int key = (substream->pcm->device << 16) | (substream->number << 2) |
(substream->stream + 1);
+ spin_lock_irq(&bus->reg_lock);
list_for_each_entry(azx_dev, &bus->stream_list, list) {
if (azx_dev->direction != substream->stream)
continue;
@@ -302,13 +303,12 @@ struct hdac_stream *snd_hdac_stream_assign(struct hdac_bus *bus,
res = azx_dev;
}
if (res) {
- spin_lock_irq(&bus->reg_lock);
res->opened = 1;
res->running = 0;
res->assigned_key = key;
res->substream = substream;
- spin_unlock_irq(&bus->reg_lock);
}
+ spin_unlock_irq(&bus->reg_lock);
return res;
}
EXPORT_SYMBOL_GPL(snd_hdac_stream_assign);
diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig
index b690ed937cbe..df2e45c8814e 100644
--- a/sound/isa/Kconfig
+++ b/sound/isa/Kconfig
@@ -22,7 +22,7 @@ config SND_SB16_DSP
menuconfig SND_ISA
bool "ISA sound devices"
depends on ISA || COMPILE_TEST
- depends on ISA_DMA_API
+ depends on ISA_DMA_API && !M68K
default y
help
Support for sound devices connected via the ISA bus.
diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c
index fa3c39cff5f8..9ee3a312c679 100644
--- a/sound/isa/cs423x/cs4236.c
+++ b/sound/isa/cs423x/cs4236.c
@@ -544,7 +544,7 @@ static int snd_cs423x_pnpbios_detect(struct pnp_dev *pdev,
static int dev;
int err;
struct snd_card *card;
- struct pnp_dev *cdev;
+ struct pnp_dev *cdev, *iter;
char cid[PNP_ID_LEN];
if (pnp_device_is_isapnp(pdev))
@@ -560,9 +560,11 @@ static int snd_cs423x_pnpbios_detect(struct pnp_dev *pdev,
strcpy(cid, pdev->id[0].id);
cid[5] = '1';
cdev = NULL;
- list_for_each_entry(cdev, &(pdev->protocol->devices), protocol_list) {
- if (!strcmp(cdev->id[0].id, cid))
+ list_for_each_entry(iter, &(pdev->protocol->devices), protocol_list) {
+ if (!strcmp(iter->id[0].id, cid)) {
+ cdev = iter;
break;
+ }
}
err = snd_cs423x_card_new(&pdev->dev, dev, &card);
if (err < 0)
diff --git a/sound/isa/gus/gus_dma.c b/sound/isa/gus/gus_dma.c
index a1c770d826dd..6d664dd8dde0 100644
--- a/sound/isa/gus/gus_dma.c
+++ b/sound/isa/gus/gus_dma.c
@@ -126,6 +126,8 @@ static void snd_gf1_dma_interrupt(struct snd_gus_card * gus)
}
block = snd_gf1_dma_next_block(gus);
spin_unlock(&gus->dma_lock);
+ if (!block)
+ return;
snd_gf1_dma_program(gus, block->addr, block->buf_addr, block->count, (unsigned short) block->cmd);
kfree(block);
#if 0
diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
index 7630f808d087..6edde2f14502 100644
--- a/sound/pci/Kconfig
+++ b/sound/pci/Kconfig
@@ -279,6 +279,7 @@ config SND_CS46XX_NEW_DSP
config SND_CS5530
tristate "CS5530 Audio"
depends on ISA_DMA_API && (X86_32 || COMPILE_TEST)
+ depends on !M68K
select SND_SB16_DSP
help
Say Y here to include support for audio on Cyrix/NatSemi CS5530 chips.
diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c
index 66f6c3bf08e3..6fb192a94762 100644
--- a/sound/pci/ac97/ac97_codec.c
+++ b/sound/pci/ac97/ac97_codec.c
@@ -938,8 +938,8 @@ static int snd_ac97_ad18xx_pcm_get_volume(struct snd_kcontrol *kcontrol, struct
int codec = kcontrol->private_value & 3;
mutex_lock(&ac97->page_mutex);
- ucontrol->value.integer.value[0] = 31 - ((ac97->spec.ad18xx.pcmreg[codec] >> 0) & 31);
- ucontrol->value.integer.value[1] = 31 - ((ac97->spec.ad18xx.pcmreg[codec] >> 8) & 31);
+ ucontrol->value.integer.value[0] = 31 - ((ac97->spec.ad18xx.pcmreg[codec] >> 8) & 31);
+ ucontrol->value.integer.value[1] = 31 - ((ac97->spec.ad18xx.pcmreg[codec] >> 0) & 31);
mutex_unlock(&ac97->page_mutex);
return 0;
}
diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c
index df720881eb99..db9d89ba3658 100644
--- a/sound/pci/cmipci.c
+++ b/sound/pci/cmipci.c
@@ -302,7 +302,6 @@ MODULE_PARM_DESC(joystick_port, "Joystick port address.");
#define CM_MICGAINZ 0x01 /* mic boost */
#define CM_MICGAINZ_SHIFT 0
-#define CM_REG_MIXER3 0x24
#define CM_REG_AUX_VOL 0x26
#define CM_VAUXL_MASK 0xf0
#define CM_VAUXR_MASK 0x0f
@@ -3310,7 +3309,7 @@ static void snd_cmipci_remove(struct pci_dev *pci)
*/
static unsigned char saved_regs[] = {
CM_REG_FUNCTRL1, CM_REG_CHFORMAT, CM_REG_LEGACY_CTRL, CM_REG_MISC_CTRL,
- CM_REG_MIXER0, CM_REG_MIXER1, CM_REG_MIXER2, CM_REG_MIXER3, CM_REG_PLL,
+ CM_REG_MIXER0, CM_REG_MIXER1, CM_REG_MIXER2, CM_REG_AUX_VOL, CM_REG_PLL,
CM_REG_CH0_FRAME1, CM_REG_CH0_FRAME2,
CM_REG_CH1_FRAME1, CM_REG_CH1_FRAME2, CM_REG_EXT_MISC,
CM_REG_INT_STATUS, CM_REG_INT_HLDCLR, CM_REG_FUNCTRL0,
diff --git a/sound/pci/ctxfi/ctamixer.c b/sound/pci/ctxfi/ctamixer.c
index d4ff377eb3a3..6d636bdcaa5a 100644
--- a/sound/pci/ctxfi/ctamixer.c
+++ b/sound/pci/ctxfi/ctamixer.c
@@ -23,16 +23,15 @@
#define BLANK_SLOT 4094
-static int amixer_master(struct rsc *rsc)
+static void amixer_master(struct rsc *rsc)
{
rsc->conj = 0;
- return rsc->idx = container_of(rsc, struct amixer, rsc)->idx[0];
+ rsc->idx = container_of(rsc, struct amixer, rsc)->idx[0];
}
-static int amixer_next_conj(struct rsc *rsc)
+static void amixer_next_conj(struct rsc *rsc)
{
rsc->conj++;
- return container_of(rsc, struct amixer, rsc)->idx[rsc->conj];
}
static int amixer_index(const struct rsc *rsc)
@@ -331,16 +330,15 @@ int amixer_mgr_destroy(struct amixer_mgr *amixer_mgr)
/* SUM resource management */
-static int sum_master(struct rsc *rsc)
+static void sum_master(struct rsc *rsc)
{
rsc->conj = 0;
- return rsc->idx = container_of(rsc, struct sum, rsc)->idx[0];
+ rsc->idx = container_of(rsc, struct sum, rsc)->idx[0];
}
-static int sum_next_conj(struct rsc *rsc)
+static void sum_next_conj(struct rsc *rsc)
{
rsc->conj++;
- return container_of(rsc, struct sum, rsc)->idx[rsc->conj];
}
static int sum_index(const struct rsc *rsc)
diff --git a/sound/pci/ctxfi/ctdaio.c b/sound/pci/ctxfi/ctdaio.c
index 27441d498968..b5e1296af09e 100644
--- a/sound/pci/ctxfi/ctdaio.c
+++ b/sound/pci/ctxfi/ctdaio.c
@@ -51,12 +51,12 @@ static struct daio_rsc_idx idx_20k2[NUM_DAIOTYP] = {
[SPDIFIO] = {.left = 0x05, .right = 0x85},
};
-static int daio_master(struct rsc *rsc)
+static void daio_master(struct rsc *rsc)
{
/* Actually, this is not the resource index of DAIO.
* For DAO, it is the input mapper index. And, for DAI,
* it is the output time-slot index. */
- return rsc->conj = rsc->idx;
+ rsc->conj = rsc->idx;
}
static int daio_index(const struct rsc *rsc)
@@ -64,19 +64,19 @@ static int daio_index(const struct rsc *rsc)
return rsc->conj;
}
-static int daio_out_next_conj(struct rsc *rsc)
+static void daio_out_next_conj(struct rsc *rsc)
{
- return rsc->conj += 2;
+ rsc->conj += 2;
}
-static int daio_in_next_conj_20k1(struct rsc *rsc)
+static void daio_in_next_conj_20k1(struct rsc *rsc)
{
- return rsc->conj += 0x200;
+ rsc->conj += 0x200;
}
-static int daio_in_next_conj_20k2(struct rsc *rsc)
+static void daio_in_next_conj_20k2(struct rsc *rsc)
{
- return rsc->conj += 0x100;
+ rsc->conj += 0x100;
}
static const struct rsc_ops daio_out_rsc_ops = {
diff --git a/sound/pci/ctxfi/ctresource.c b/sound/pci/ctxfi/ctresource.c
index 0bb5696e44b3..ec5f597b580a 100644
--- a/sound/pci/ctxfi/ctresource.c
+++ b/sound/pci/ctxfi/ctresource.c
@@ -109,18 +109,17 @@ static int audio_ring_slot(const struct rsc *rsc)
return (rsc->conj << 4) + offset_in_audio_slot_block[rsc->type];
}
-static int rsc_next_conj(struct rsc *rsc)
+static void rsc_next_conj(struct rsc *rsc)
{
unsigned int i;
for (i = 0; (i < 8) && (!(rsc->msr & (0x1 << i))); )
i++;
rsc->conj += (AUDIO_SLOT_BLOCK_NUM >> i);
- return rsc->conj;
}
-static int rsc_master(struct rsc *rsc)
+static void rsc_master(struct rsc *rsc)
{
- return rsc->conj = rsc->idx;
+ rsc->conj = rsc->idx;
}
static const struct rsc_ops rsc_generic_ops = {
diff --git a/sound/pci/ctxfi/ctresource.h b/sound/pci/ctxfi/ctresource.h
index 93e47488a1c1..92146054af58 100644
--- a/sound/pci/ctxfi/ctresource.h
+++ b/sound/pci/ctxfi/ctresource.h
@@ -39,8 +39,8 @@ struct rsc {
};
struct rsc_ops {
- int (*master)(struct rsc *rsc); /* Move to master resource */
- int (*next_conj)(struct rsc *rsc); /* Move to next conjugate resource */
+ void (*master)(struct rsc *rsc); /* Move to master resource */
+ void (*next_conj)(struct rsc *rsc); /* Move to next conjugate resource */
int (*index)(const struct rsc *rsc); /* Return the index of resource */
/* Return the output slot number */
int (*output_slot)(const struct rsc *rsc);
diff --git a/sound/pci/ctxfi/ctsrc.c b/sound/pci/ctxfi/ctsrc.c
index 37c18ce84974..7d2bda0c3d3d 100644
--- a/sound/pci/ctxfi/ctsrc.c
+++ b/sound/pci/ctxfi/ctsrc.c
@@ -590,16 +590,15 @@ int src_mgr_destroy(struct src_mgr *src_mgr)
/* SRCIMP resource manager operations */
-static int srcimp_master(struct rsc *rsc)
+static void srcimp_master(struct rsc *rsc)
{
rsc->conj = 0;
- return rsc->idx = container_of(rsc, struct srcimp, rsc)->idx[0];
+ rsc->idx = container_of(rsc, struct srcimp, rsc)->idx[0];
}
-static int srcimp_next_conj(struct rsc *rsc)
+static void srcimp_next_conj(struct rsc *rsc)
{
rsc->conj++;
- return container_of(rsc, struct srcimp, rsc)->idx[rsc->conj];
}
static int srcimp_index(const struct rsc *rsc)
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 326f95ce5ceb..c8847de8388f 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -1721,8 +1721,11 @@ void snd_hda_ctls_clear(struct hda_codec *codec)
{
int i;
struct hda_nid_item *items = codec->mixers.list;
+
+ down_write(&codec->card->controls_rwsem);
for (i = 0; i < codec->mixers.used; i++)
snd_ctl_remove(codec->card, items[i].kctl);
+ up_write(&codec->card->controls_rwsem);
snd_array_free(&codec->mixers);
snd_array_free(&codec->nids);
}
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 95d472d433e7..b8fe0ec5d624 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1608,6 +1608,7 @@ static struct snd_pci_quirk probe_mask_list[] = {
/* forced codec slots */
SND_PCI_QUIRK(0x1043, 0x1262, "ASUS W5Fm", 0x103),
SND_PCI_QUIRK(0x1046, 0x1262, "ASUS W5F", 0x103),
+ SND_PCI_QUIRK(0x1558, 0x0351, "Schenker Dock 15", 0x105),
/* WinFast VP200 H (Teradici) user reported broken communication */
SND_PCI_QUIRK(0x3a21, 0x040d, "WinFast VP200 H", 0x101),
{}
@@ -1793,8 +1794,6 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci,
assign_position_fix(chip, check_position_fix(chip, position_fix[dev]));
- check_probe_mask(chip, dev);
-
if (single_cmd < 0) /* allow fallback to single_cmd at errors */
chip->fallback_to_single_cmd = 1;
else /* explicitly set to single_cmd or not */
@@ -1826,6 +1825,8 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci,
chip->bus.needs_damn_long_delay = 1;
}
+ check_probe_mask(chip, dev);
+
err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
if (err < 0) {
dev_err(card->dev, "Error creating device [card]!\n");
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 94fc17b28e9c..851ea79da31c 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1926,6 +1926,7 @@ enum {
ALC887_FIXUP_ASUS_BASS,
ALC887_FIXUP_BASS_CHMAP,
ALC1220_FIXUP_GB_DUAL_CODECS,
+ ALC1220_FIXUP_GB_X570,
ALC1220_FIXUP_CLEVO_P950,
ALC1220_FIXUP_CLEVO_PB51ED,
ALC1220_FIXUP_CLEVO_PB51ED_PINS,
@@ -2115,6 +2116,30 @@ static void alc1220_fixup_gb_dual_codecs(struct hda_codec *codec,
}
}
+static void alc1220_fixup_gb_x570(struct hda_codec *codec,
+ const struct hda_fixup *fix,
+ int action)
+{
+ static const hda_nid_t conn1[] = { 0x0c };
+ static const struct coef_fw gb_x570_coefs[] = {
+ WRITE_COEF(0x07, 0x03c0),
+ WRITE_COEF(0x1a, 0x01c1),
+ WRITE_COEF(0x1b, 0x0202),
+ WRITE_COEF(0x43, 0x3005),
+ {}
+ };
+
+ switch (action) {
+ case HDA_FIXUP_ACT_PRE_PROBE:
+ snd_hda_override_conn_list(codec, 0x14, ARRAY_SIZE(conn1), conn1);
+ snd_hda_override_conn_list(codec, 0x1b, ARRAY_SIZE(conn1), conn1);
+ break;
+ case HDA_FIXUP_ACT_INIT:
+ alc_process_coef_fw(codec, gb_x570_coefs);
+ break;
+ }
+}
+
static void alc1220_fixup_clevo_p950(struct hda_codec *codec,
const struct hda_fixup *fix,
int action)
@@ -2417,6 +2442,10 @@ static const struct hda_fixup alc882_fixups[] = {
.type = HDA_FIXUP_FUNC,
.v.func = alc1220_fixup_gb_dual_codecs,
},
+ [ALC1220_FIXUP_GB_X570] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc1220_fixup_gb_x570,
+ },
[ALC1220_FIXUP_CLEVO_P950] = {
.type = HDA_FIXUP_FUNC,
.v.func = alc1220_fixup_clevo_p950,
@@ -2519,8 +2548,9 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
SND_PCI_QUIRK(0x13fe, 0x1009, "Advantech MIT-W101", ALC886_FIXUP_EAPD),
SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte EP45-DS3/Z87X-UD3H", ALC889_FIXUP_FRONT_HP_NO_PRESENCE),
SND_PCI_QUIRK(0x1458, 0xa0b8, "Gigabyte AZ370-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS),
- SND_PCI_QUIRK(0x1458, 0xa0cd, "Gigabyte X570 Aorus Master", ALC1220_FIXUP_CLEVO_P950),
- SND_PCI_QUIRK(0x1458, 0xa0ce, "Gigabyte X570 Aorus Xtreme", ALC1220_FIXUP_CLEVO_P950),
+ SND_PCI_QUIRK(0x1458, 0xa0cd, "Gigabyte X570 Aorus Master", ALC1220_FIXUP_GB_X570),
+ SND_PCI_QUIRK(0x1458, 0xa0ce, "Gigabyte X570 Aorus Xtreme", ALC1220_FIXUP_GB_X570),
+ SND_PCI_QUIRK(0x1458, 0xa0d5, "Gigabyte X570S Aorus Master", ALC1220_FIXUP_GB_X570),
SND_PCI_QUIRK(0x1462, 0x11f7, "MSI-GE63", ALC1220_FIXUP_CLEVO_P950),
SND_PCI_QUIRK(0x1462, 0x1228, "MSI-GP63", ALC1220_FIXUP_CLEVO_P950),
SND_PCI_QUIRK(0x1462, 0x1229, "MSI-GP73", ALC1220_FIXUP_CLEVO_P950),
@@ -2538,6 +2568,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
SND_PCI_QUIRK(0x1558, 0x65e1, "Clevo PB51[ED][DF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS),
SND_PCI_QUIRK(0x1558, 0x65e5, "Clevo PC50D[PRS](?:-D|-G)?", ALC1220_FIXUP_CLEVO_PB51ED_PINS),
SND_PCI_QUIRK(0x1558, 0x65f1, "Clevo PC50HS", ALC1220_FIXUP_CLEVO_PB51ED_PINS),
+ SND_PCI_QUIRK(0x1558, 0x65f5, "Clevo PD50PN[NRT]", ALC1220_FIXUP_CLEVO_PB51ED_PINS),
SND_PCI_QUIRK(0x1558, 0x67d1, "Clevo PB71[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS),
SND_PCI_QUIRK(0x1558, 0x67e1, "Clevo PB71[DE][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS),
SND_PCI_QUIRK(0x1558, 0x67e5, "Clevo PC70D[PRS](?:-D|-G)?", ALC1220_FIXUP_CLEVO_PB51ED_PINS),
@@ -2595,6 +2626,7 @@ static const struct hda_model_fixup alc882_fixup_models[] = {
{.id = ALC882_FIXUP_NO_PRIMARY_HP, .name = "no-primary-hp"},
{.id = ALC887_FIXUP_ASUS_BASS, .name = "asus-bass"},
{.id = ALC1220_FIXUP_GB_DUAL_CODECS, .name = "dual-codecs"},
+ {.id = ALC1220_FIXUP_GB_X570, .name = "gb-x570"},
{.id = ALC1220_FIXUP_CLEVO_P950, .name = "clevo-p950"},
{}
};
@@ -3526,8 +3558,8 @@ static void alc256_shutup(struct hda_codec *codec)
/* If disable 3k pulldown control for alc257, the Mic detection will not work correctly
* when booting with headset plugged. So skip setting it for the codec alc257
*/
- if (spec->codec_variant != ALC269_TYPE_ALC257 &&
- spec->codec_variant != ALC269_TYPE_ALC256)
+ if (codec->core.vendor_id != 0x10ec0236 &&
+ codec->core.vendor_id != 0x10ec0257)
alc_update_coef_idx(codec, 0x46, 0, 3 << 12);
if (!spec->no_shutup_pins)
@@ -6425,6 +6457,7 @@ enum {
ALC285_FIXUP_HP_MUTE_LED,
ALC236_FIXUP_HP_MUTE_LED,
ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET,
+ ALC256_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET,
ALC295_FIXUP_ASUS_MIC_NO_PRESENCE,
ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS,
ALC269VC_FIXUP_ACER_HEADSET_MIC,
@@ -7709,6 +7742,14 @@ static const struct hda_fixup alc269_fixups[] = {
{ }
},
},
+ [ALC256_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET] = {
+ .type = HDA_FIXUP_VERBS,
+ .v.verbs = (const struct hda_verb[]) {
+ { 0x20, AC_VERB_SET_COEF_INDEX, 0x08},
+ { 0x20, AC_VERB_SET_PROC_COEF, 0x2fcf},
+ { }
+ },
+ },
[ALC295_FIXUP_ASUS_MIC_NO_PRESENCE] = {
.type = HDA_FIXUP_PINS,
.v.pins = (const struct hda_pintbl[]) {
@@ -8101,6 +8142,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x103c, 0x84da, "HP OMEN dc0019-ur", ALC295_FIXUP_HP_OMEN),
SND_PCI_QUIRK(0x103c, 0x84e7, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3),
SND_PCI_QUIRK(0x103c, 0x8519, "HP Spectre x360 15-df0xxx", ALC285_FIXUP_HP_SPECTRE_X360),
+ SND_PCI_QUIRK(0x103c, 0x860f, "HP ZBook 15 G6", ALC285_FIXUP_HP_GPIO_AMP_INIT),
SND_PCI_QUIRK(0x103c, 0x861f, "HP Elite Dragonfly G1", ALC285_FIXUP_HP_GPIO_AMP_INIT),
SND_PCI_QUIRK(0x103c, 0x869d, "HP", ALC236_FIXUP_HP_MUTE_LED),
SND_PCI_QUIRK(0x103c, 0x8724, "HP EliteBook 850 G7", ALC285_FIXUP_HP_GPIO_LED),
@@ -8151,6 +8193,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1043, 0x1e51, "ASUS Zephyrus M15", ALC294_FIXUP_ASUS_GU502_PINS),
SND_PCI_QUIRK(0x1043, 0x1e8e, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA401),
SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_GA401),
+ SND_PCI_QUIRK(0x1043, 0x1d42, "ASUS Zephyrus G14 2022", ALC289_FIXUP_ASUS_GA401),
+ SND_PCI_QUIRK(0x1043, 0x16b2, "ASUS GU603", ALC289_FIXUP_ASUS_GA401),
SND_PCI_QUIRK(0x1043, 0x3030, "ASUS ZN270IE", ALC256_FIXUP_ASUS_AIO_GPIO2),
SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC),
SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC),
@@ -8183,6 +8227,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x144d, 0xc740, "Samsung Ativ book 8 (NP870Z5G)", ALC269_FIXUP_ATIV_BOOK_8),
SND_PCI_QUIRK(0x144d, 0xc812, "Samsung Notebook Pen S (NT950SBE-X58)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
SND_PCI_QUIRK(0x144d, 0xc830, "Samsung Galaxy Book Ion (NT950XCJ-X716A)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
+ SND_PCI_QUIRK(0x144d, 0xc832, "Samsung Galaxy Book Flex Alpha (NP730QCJ)", ALC256_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC),
SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC),
SND_PCI_QUIRK(0x1462, 0xb171, "Cubi N 8GL (MS-B171)", ALC283_FIXUP_HEADSET_MIC),
@@ -8506,6 +8551,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = {
{.id = ALC298_FIXUP_HUAWEI_MBX_STEREO, .name = "huawei-mbx-stereo"},
{.id = ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE, .name = "alc256-medion-headset"},
{.id = ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, .name = "alc298-samsung-headphone"},
+ {.id = ALC256_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, .name = "alc256-samsung-headphone"},
{.id = ALC255_FIXUP_XIAOMI_HEADSET_MIC, .name = "alc255-xiaomi-headset"},
{.id = ALC274_FIXUP_HP_MIC, .name = "alc274-hp-mic-detect"},
{.id = ALC295_FIXUP_HP_OMEN, .name = "alc295-hp-omen"},
@@ -9620,6 +9666,27 @@ static void alc671_fixup_hp_headset_mic2(struct hda_codec *codec,
}
}
+static void alc897_hp_automute_hook(struct hda_codec *codec,
+ struct hda_jack_callback *jack)
+{
+ struct alc_spec *spec = codec->spec;
+ int vref;
+
+ snd_hda_gen_hp_automute(codec, jack);
+ vref = spec->gen.hp_jack_present ? (PIN_HP | AC_PINCTL_VREF_100) : PIN_HP;
+ snd_hda_codec_write(codec, 0x1b, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+ vref);
+}
+
+static void alc897_fixup_lenovo_headset_mic(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ struct alc_spec *spec = codec->spec;
+ if (action == HDA_FIXUP_ACT_PRE_PROBE) {
+ spec->gen.hp_automute_hook = alc897_hp_automute_hook;
+ }
+}
+
static const struct coef_fw alc668_coefs[] = {
WRITE_COEF(0x01, 0xbebe), WRITE_COEF(0x02, 0xaaaa), WRITE_COEF(0x03, 0x0),
WRITE_COEF(0x04, 0x0180), WRITE_COEF(0x06, 0x0), WRITE_COEF(0x07, 0x0f80),
@@ -9700,6 +9767,8 @@ enum {
ALC668_FIXUP_ASUS_NO_HEADSET_MIC,
ALC668_FIXUP_HEADSET_MIC,
ALC668_FIXUP_MIC_DET_COEF,
+ ALC897_FIXUP_LENOVO_HEADSET_MIC,
+ ALC897_FIXUP_HEADSET_MIC_PIN,
};
static const struct hda_fixup alc662_fixups[] = {
@@ -10106,6 +10175,19 @@ static const struct hda_fixup alc662_fixups[] = {
{}
},
},
+ [ALC897_FIXUP_LENOVO_HEADSET_MIC] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc897_fixup_lenovo_headset_mic,
+ },
+ [ALC897_FIXUP_HEADSET_MIC_PIN] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x1a, 0x03a11050 },
+ { }
+ },
+ .chained = true,
+ .chain_id = ALC897_FIXUP_LENOVO_HEADSET_MIC
+ },
};
static const struct snd_pci_quirk alc662_fixup_tbl[] = {
@@ -10132,6 +10214,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = {
SND_PCI_QUIRK(0x1028, 0x069f, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800),
SND_PCI_QUIRK(0x103c, 0x873e, "HP", ALC671_FIXUP_HP_HEADSET_MIC2),
+ SND_PCI_QUIRK(0x103c, 0x885f, "HP 288 Pro G8", ALC671_FIXUP_HP_HEADSET_MIC2),
SND_PCI_QUIRK(0x1043, 0x1080, "Asus UX501VW", ALC668_FIXUP_HEADSET_MODE),
SND_PCI_QUIRK(0x1043, 0x11cd, "Asus N550", ALC662_FIXUP_ASUS_Nx50),
SND_PCI_QUIRK(0x1043, 0x129d, "Asus N750", ALC662_FIXUP_ASUS_Nx50),
@@ -10150,6 +10233,10 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = {
SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD),
SND_PCI_QUIRK(0x14cd, 0x5003, "USI", ALC662_FIXUP_USI_HEADSET_MODE),
SND_PCI_QUIRK(0x17aa, 0x1036, "Lenovo P520", ALC662_FIXUP_LENOVO_MULTI_CODECS),
+ SND_PCI_QUIRK(0x17aa, 0x32ca, "Lenovo ThinkCentre M80", ALC897_FIXUP_HEADSET_MIC_PIN),
+ SND_PCI_QUIRK(0x17aa, 0x32cb, "Lenovo ThinkCentre M70", ALC897_FIXUP_HEADSET_MIC_PIN),
+ SND_PCI_QUIRK(0x17aa, 0x32cf, "Lenovo ThinkCentre M950", ALC897_FIXUP_HEADSET_MIC_PIN),
+ SND_PCI_QUIRK(0x17aa, 0x32f7, "Lenovo ThinkCentre M90", ALC897_FIXUP_HEADSET_MIC_PIN),
SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo Ideapad Y550P", ALC662_FIXUP_IDEAPAD),
SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Ideapad Y550", ALC662_FIXUP_IDEAPAD),
SND_PCI_QUIRK(0x1849, 0x5892, "ASRock B150M", ALC892_FIXUP_ASROCK_MOBO),
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index ca603397651c..1e0973322cd0 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -280,7 +280,10 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream,
/* Enable PMC peripheral clock for this SSC */
pr_debug("atmel_ssc_dai: Starting clock\n");
- clk_enable(ssc_p->ssc->clk);
+ ret = clk_enable(ssc_p->ssc->clk);
+ if (ret)
+ return ret;
+
ssc_p->mck_rate = clk_get_rate(ssc_p->ssc->clk);
/* Reset the SSC unless initialized to keep it in a clean state */
diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c
index b1bef2bf142d..d1579896f3a1 100644
--- a/sound/soc/atmel/sam9g20_wm8731.c
+++ b/sound/soc/atmel/sam9g20_wm8731.c
@@ -46,35 +46,6 @@
*/
#undef ENABLE_MIC_INPUT
-static struct clk *mclk;
-
-static int at91sam9g20ek_set_bias_level(struct snd_soc_card *card,
- struct snd_soc_dapm_context *dapm,
- enum snd_soc_bias_level level)
-{
- static int mclk_on;
- int ret = 0;
-
- switch (level) {
- case SND_SOC_BIAS_ON:
- case SND_SOC_BIAS_PREPARE:
- if (!mclk_on)
- ret = clk_enable(mclk);
- if (ret == 0)
- mclk_on = 1;
- break;
-
- case SND_SOC_BIAS_OFF:
- case SND_SOC_BIAS_STANDBY:
- if (mclk_on)
- clk_disable(mclk);
- mclk_on = 0;
- break;
- }
-
- return ret;
-}
-
static const struct snd_soc_dapm_widget at91sam9g20ek_dapm_widgets[] = {
SND_SOC_DAPM_MIC("Int Mic", NULL),
SND_SOC_DAPM_SPK("Ext Spk", NULL),
@@ -135,7 +106,6 @@ static struct snd_soc_card snd_soc_at91sam9g20ek = {
.owner = THIS_MODULE,
.dai_link = &at91sam9g20ek_dai,
.num_links = 1,
- .set_bias_level = at91sam9g20ek_set_bias_level,
.dapm_widgets = at91sam9g20ek_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(at91sam9g20ek_dapm_widgets),
@@ -148,7 +118,6 @@ static int at91sam9g20ek_audio_probe(struct platform_device *pdev)
{
struct device_node *np = pdev->dev.of_node;
struct device_node *codec_np, *cpu_np;
- struct clk *pllb;
struct snd_soc_card *card = &snd_soc_at91sam9g20ek;
int ret;
@@ -162,31 +131,6 @@ static int at91sam9g20ek_audio_probe(struct platform_device *pdev)
return -EINVAL;
}
- /*
- * Codec MCLK is supplied by PCK0 - set it up.
- */
- mclk = clk_get(NULL, "pck0");
- if (IS_ERR(mclk)) {
- dev_err(&pdev->dev, "Failed to get MCLK\n");
- ret = PTR_ERR(mclk);
- goto err;
- }
-
- pllb = clk_get(NULL, "pllb");
- if (IS_ERR(pllb)) {
- dev_err(&pdev->dev, "Failed to get PLLB\n");
- ret = PTR_ERR(pllb);
- goto err_mclk;
- }
- ret = clk_set_parent(mclk, pllb);
- clk_put(pllb);
- if (ret != 0) {
- dev_err(&pdev->dev, "Failed to set MCLK parent\n");
- goto err_mclk;
- }
-
- clk_set_rate(mclk, MCLK_RATE);
-
card->dev = &pdev->dev;
/* Parse device node info */
@@ -214,6 +158,7 @@ static int at91sam9g20ek_audio_probe(struct platform_device *pdev)
cpu_np = of_parse_phandle(np, "atmel,ssc-controller", 0);
if (!cpu_np) {
dev_err(&pdev->dev, "dai and pcm info missing\n");
+ of_node_put(codec_np);
return -EINVAL;
}
at91sam9g20ek_dai.cpus->of_node = cpu_np;
@@ -229,9 +174,6 @@ static int at91sam9g20ek_audio_probe(struct platform_device *pdev)
return ret;
-err_mclk:
- clk_put(mclk);
- mclk = NULL;
err:
atmel_ssc_put_audio(0);
return ret;
@@ -241,8 +183,6 @@ static int at91sam9g20ek_audio_remove(struct platform_device *pdev)
{
struct snd_soc_card *card = platform_get_drvdata(pdev);
- clk_disable(mclk);
- mclk = NULL;
snd_soc_unregister_card(card);
atmel_ssc_put_audio(0);
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index b889cb54461d..64c5d64cf4eb 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -590,21 +590,26 @@ config SND_SOC_CS4349
config SND_SOC_CS47L15
tristate
+ depends on MFD_CS47L15
config SND_SOC_CS47L24
tristate
config SND_SOC_CS47L35
tristate
+ depends on MFD_CS47L35
config SND_SOC_CS47L85
tristate
+ depends on MFD_CS47L85
config SND_SOC_CS47L90
tristate
+ depends on MFD_CS47L90
config SND_SOC_CS47L92
tristate
+ depends on MFD_CS47L92
# Cirrus Logic Quad-Channel ADC
config SND_SOC_CS53L30
diff --git a/sound/soc/codecs/cpcap.c b/sound/soc/codecs/cpcap.c
index 1902689c5ea2..acd88fe38cd4 100644
--- a/sound/soc/codecs/cpcap.c
+++ b/sound/soc/codecs/cpcap.c
@@ -1541,6 +1541,8 @@ static int cpcap_codec_probe(struct platform_device *pdev)
{
struct device_node *codec_node =
of_get_child_by_name(pdev->dev.parent->of_node, "audio-codec");
+ if (!codec_node)
+ return -ENODEV;
pdev->dev.of_node = codec_node;
diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c
index 2fb65f246b0c..77af5b67b9bb 100644
--- a/sound/soc/codecs/cs4265.c
+++ b/sound/soc/codecs/cs4265.c
@@ -150,7 +150,6 @@ static const struct snd_kcontrol_new cs4265_snd_controls[] = {
SOC_SINGLE("E to F Buffer Disable Switch", CS4265_SPDIF_CTL1,
6, 1, 0),
SOC_ENUM("C Data Access", cam_mode_enum),
- SOC_SINGLE("SPDIF Switch", CS4265_SPDIF_CTL2, 5, 1, 1),
SOC_SINGLE("Validity Bit Control Switch", CS4265_SPDIF_CTL2,
3, 1, 0),
SOC_ENUM("SPDIF Mono/Stereo", spdif_mono_stereo_enum),
@@ -186,7 +185,7 @@ static const struct snd_soc_dapm_widget cs4265_dapm_widgets[] = {
SND_SOC_DAPM_SWITCH("Loopback", SND_SOC_NOPM, 0, 0,
&loopback_ctl),
- SND_SOC_DAPM_SWITCH("SPDIF", SND_SOC_NOPM, 0, 0,
+ SND_SOC_DAPM_SWITCH("SPDIF", CS4265_SPDIF_CTL2, 5, 1,
&spdif_switch),
SND_SOC_DAPM_SWITCH("DAC", CS4265_PWRCTL, 1, 1,
&dac_switch),
diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c
index f83a6eaba12c..ef8bd9e04637 100644
--- a/sound/soc/codecs/da7219.c
+++ b/sound/soc/codecs/da7219.c
@@ -446,7 +446,7 @@ static int da7219_tonegen_freq_put(struct snd_kcontrol *kcontrol,
struct soc_mixer_control *mixer_ctrl =
(struct soc_mixer_control *) kcontrol->private_value;
unsigned int reg = mixer_ctrl->reg;
- __le16 val;
+ __le16 val_new, val_old;
int ret;
/*
@@ -454,13 +454,19 @@ static int da7219_tonegen_freq_put(struct snd_kcontrol *kcontrol,
* Therefore we need to convert to little endian here to align with
* HW registers.
*/
- val = cpu_to_le16(ucontrol->value.integer.value[0]);
+ val_new = cpu_to_le16(ucontrol->value.integer.value[0]);
mutex_lock(&da7219->ctrl_lock);
- ret = regmap_raw_write(da7219->regmap, reg, &val, sizeof(val));
+ ret = regmap_raw_read(da7219->regmap, reg, &val_old, sizeof(val_old));
+ if (ret == 0 && (val_old != val_new))
+ ret = regmap_raw_write(da7219->regmap, reg,
+ &val_new, sizeof(val_new));
mutex_unlock(&da7219->ctrl_lock);
- return ret;
+ if (ret < 0)
+ return ret;
+
+ return val_old != val_new;
}
diff --git a/sound/soc/codecs/max9759.c b/sound/soc/codecs/max9759.c
index 00e9d4fd1651..0c261335c8a1 100644
--- a/sound/soc/codecs/max9759.c
+++ b/sound/soc/codecs/max9759.c
@@ -64,7 +64,8 @@ static int speaker_gain_control_put(struct snd_kcontrol *kcontrol,
struct snd_soc_component *c = snd_soc_kcontrol_component(kcontrol);
struct max9759 *priv = snd_soc_component_get_drvdata(c);
- if (ucontrol->value.integer.value[0] > 3)
+ if (ucontrol->value.integer.value[0] < 0 ||
+ ucontrol->value.integer.value[0] > 3)
return -EINVAL;
priv->gain = ucontrol->value.integer.value[0];
diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c
index cf6516693e4e..5a8eedea6be0 100644
--- a/sound/soc/codecs/msm8916-wcd-analog.c
+++ b/sound/soc/codecs/msm8916-wcd-analog.c
@@ -1196,8 +1196,8 @@ static int pm8916_wcd_analog_spmi_probe(struct platform_device *pdev)
irq = platform_get_irq_byname(pdev, "mbhc_switch_int");
if (irq < 0) {
- dev_err(dev, "failed to get mbhc switch irq\n");
- return irq;
+ ret = irq;
+ goto err_disable_clk;
}
ret = devm_request_threaded_irq(dev, irq, NULL,
@@ -1211,8 +1211,8 @@ static int pm8916_wcd_analog_spmi_probe(struct platform_device *pdev)
if (priv->mbhc_btn_enabled) {
irq = platform_get_irq_byname(pdev, "mbhc_but_press_det");
if (irq < 0) {
- dev_err(dev, "failed to get button press irq\n");
- return irq;
+ ret = irq;
+ goto err_disable_clk;
}
ret = devm_request_threaded_irq(dev, irq, NULL,
@@ -1225,8 +1225,8 @@ static int pm8916_wcd_analog_spmi_probe(struct platform_device *pdev)
irq = platform_get_irq_byname(pdev, "mbhc_but_rel_det");
if (irq < 0) {
- dev_err(dev, "failed to get button release irq\n");
- return irq;
+ ret = irq;
+ goto err_disable_clk;
}
ret = devm_request_threaded_irq(dev, irq, NULL,
@@ -1244,6 +1244,10 @@ static int pm8916_wcd_analog_spmi_probe(struct platform_device *pdev)
return devm_snd_soc_register_component(dev, &pm8916_wcd_analog,
pm8916_wcd_analog_dai,
ARRAY_SIZE(pm8916_wcd_analog_dai));
+
+err_disable_clk:
+ clk_disable_unprepare(priv->mclk);
+ return ret;
}
static int pm8916_wcd_analog_spmi_remove(struct platform_device *pdev)
diff --git a/sound/soc/codecs/msm8916-wcd-digital.c b/sound/soc/codecs/msm8916-wcd-digital.c
index 09fccacadd6b..e4cde214b7b2 100644
--- a/sound/soc/codecs/msm8916-wcd-digital.c
+++ b/sound/soc/codecs/msm8916-wcd-digital.c
@@ -1201,14 +1201,24 @@ static int msm8916_wcd_digital_probe(struct platform_device *pdev)
ret = clk_prepare_enable(priv->mclk);
if (ret < 0) {
dev_err(dev, "failed to enable mclk %d\n", ret);
- return ret;
+ goto err_clk;
}
dev_set_drvdata(dev, priv);
- return devm_snd_soc_register_component(dev, &msm8916_wcd_digital,
+ ret = devm_snd_soc_register_component(dev, &msm8916_wcd_digital,
msm8916_wcd_digital_dai,
ARRAY_SIZE(msm8916_wcd_digital_dai));
+ if (ret)
+ goto err_mclk;
+
+ return 0;
+
+err_mclk:
+ clk_disable_unprepare(priv->mclk);
+err_clk:
+ clk_disable_unprepare(priv->ahbclk);
+ return ret;
}
static int msm8916_wcd_digital_remove(struct platform_device *pdev)
diff --git a/sound/soc/codecs/mt6358.c b/sound/soc/codecs/mt6358.c
index bb737fd678cc..494ba0eeb433 100644
--- a/sound/soc/codecs/mt6358.c
+++ b/sound/soc/codecs/mt6358.c
@@ -103,6 +103,7 @@ int mt6358_set_mtkaif_protocol(struct snd_soc_component *cmpnt,
priv->mtkaif_protocol = mtkaif_protocol;
return 0;
}
+EXPORT_SYMBOL_GPL(mt6358_set_mtkaif_protocol);
static void playback_gpio_set(struct mt6358_priv *priv)
{
@@ -269,6 +270,7 @@ int mt6358_mtkaif_calibration_enable(struct snd_soc_component *cmpnt)
1 << RG_AUD_PAD_TOP_DAT_MISO_LOOPBACK_SFT);
return 0;
}
+EXPORT_SYMBOL_GPL(mt6358_mtkaif_calibration_enable);
int mt6358_mtkaif_calibration_disable(struct snd_soc_component *cmpnt)
{
@@ -292,6 +294,7 @@ int mt6358_mtkaif_calibration_disable(struct snd_soc_component *cmpnt)
capture_gpio_reset(priv);
return 0;
}
+EXPORT_SYMBOL_GPL(mt6358_mtkaif_calibration_disable);
int mt6358_set_mtkaif_calibration_phase(struct snd_soc_component *cmpnt,
int phase_1, int phase_2)
@@ -306,6 +309,7 @@ int mt6358_set_mtkaif_calibration_phase(struct snd_soc_component *cmpnt,
phase_2 << RG_AUD_PAD_TOP_PHASE_MODE2_SFT);
return 0;
}
+EXPORT_SYMBOL_GPL(mt6358_set_mtkaif_calibration_phase);
/* dl pga gain */
enum {
diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c
index 15bd8335f667..c8ccfa2fff84 100644
--- a/sound/soc/codecs/nau8824.c
+++ b/sound/soc/codecs/nau8824.c
@@ -8,6 +8,7 @@
#include <linux/module.h>
#include <linux/delay.h>
+#include <linux/dmi.h>
#include <linux/init.h>
#include <linux/i2c.h>
#include <linux/regmap.h>
@@ -27,6 +28,12 @@
#include "nau8824.h"
+#define NAU8824_JD_ACTIVE_HIGH BIT(0)
+
+static int nau8824_quirk;
+static int quirk_override = -1;
+module_param_named(quirk, quirk_override, uint, 0444);
+MODULE_PARM_DESC(quirk, "Board-specific quirk override");
static int nau8824_config_sysclk(struct nau8824 *nau8824,
int clk_id, unsigned int freq);
@@ -1875,6 +1882,34 @@ static int nau8824_read_device_properties(struct device *dev,
return 0;
}
+/* Please keep this list alphabetically sorted */
+static const struct dmi_system_id nau8824_quirk_table[] = {
+ {
+ /* Cyberbook T116 rugged tablet */
+ .matches = {
+ DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "Default string"),
+ DMI_EXACT_MATCH(DMI_BOARD_NAME, "Cherry Trail CR"),
+ DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "20170531"),
+ },
+ .driver_data = (void *)(NAU8824_JD_ACTIVE_HIGH),
+ },
+ {}
+};
+
+static void nau8824_check_quirks(void)
+{
+ const struct dmi_system_id *dmi_id;
+
+ if (quirk_override != -1) {
+ nau8824_quirk = quirk_override;
+ return;
+ }
+
+ dmi_id = dmi_first_match(nau8824_quirk_table);
+ if (dmi_id)
+ nau8824_quirk = (unsigned long)dmi_id->driver_data;
+}
+
static int nau8824_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
@@ -1899,6 +1934,11 @@ static int nau8824_i2c_probe(struct i2c_client *i2c,
nau8824->irq = i2c->irq;
sema_init(&nau8824->jd_sem, 1);
+ nau8824_check_quirks();
+
+ if (nau8824_quirk & NAU8824_JD_ACTIVE_HIGH)
+ nau8824->jkdet_polarity = 0;
+
nau8824_print_device_properties(nau8824);
ret = regmap_read(nau8824->regmap, NAU8824_REG_I2C_DEVICE_ID, &value);
diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c
index 2943692f66ed..19e2f622718d 100644
--- a/sound/soc/codecs/rt5663.c
+++ b/sound/soc/codecs/rt5663.c
@@ -3461,6 +3461,7 @@ static void rt5663_calibrate(struct rt5663_priv *rt5663)
static int rt5663_parse_dp(struct rt5663_priv *rt5663, struct device *dev)
{
int table_size;
+ int ret;
device_property_read_u32(dev, "realtek,dc_offset_l_manual",
&rt5663->pdata.dc_offset_l_manual);
@@ -3477,9 +3478,13 @@ static int rt5663_parse_dp(struct rt5663_priv *rt5663, struct device *dev)
table_size = sizeof(struct impedance_mapping_table) *
rt5663->pdata.impedance_sensing_num;
rt5663->imp_table = devm_kzalloc(dev, table_size, GFP_KERNEL);
- device_property_read_u32_array(dev,
+ if (!rt5663->imp_table)
+ return -ENOMEM;
+ ret = device_property_read_u32_array(dev,
"realtek,impedance_sensing_table",
(u32 *)rt5663->imp_table, table_size);
+ if (ret)
+ return ret;
}
return 0;
@@ -3504,8 +3509,11 @@ static int rt5663_i2c_probe(struct i2c_client *i2c,
if (pdata)
rt5663->pdata = *pdata;
- else
- rt5663_parse_dp(rt5663, &i2c->dev);
+ else {
+ ret = rt5663_parse_dp(rt5663, &i2c->dev);
+ if (ret)
+ return ret;
+ }
for (i = 0; i < ARRAY_SIZE(rt5663->supplies); i++)
rt5663->supplies[i].supply = rt5663_supply_names[i];
diff --git a/sound/soc/codecs/rt5668.c b/sound/soc/codecs/rt5668.c
index 5716cede99cb..acc2b34ca334 100644
--- a/sound/soc/codecs/rt5668.c
+++ b/sound/soc/codecs/rt5668.c
@@ -1022,11 +1022,13 @@ static void rt5668_jack_detect_handler(struct work_struct *work)
container_of(work, struct rt5668_priv, jack_detect_work.work);
int val, btn_type;
- while (!rt5668->component)
- usleep_range(10000, 15000);
-
- while (!rt5668->component->card->instantiated)
- usleep_range(10000, 15000);
+ if (!rt5668->component || !rt5668->component->card ||
+ !rt5668->component->card->instantiated) {
+ /* card not yet ready, try later */
+ mod_delayed_work(system_power_efficient_wq,
+ &rt5668->jack_detect_work, msecs_to_jiffies(15));
+ return;
+ }
mutex_lock(&rt5668->calibrate_mutex);
diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c
index 05e883a65d7a..a8cf4c745130 100644
--- a/sound/soc/codecs/rt5682.c
+++ b/sound/soc/codecs/rt5682.c
@@ -1052,11 +1052,13 @@ static void rt5682_jack_detect_handler(struct work_struct *work)
container_of(work, struct rt5682_priv, jack_detect_work.work);
int val, btn_type;
- while (!rt5682->component)
- usleep_range(10000, 15000);
-
- while (!rt5682->component->card->instantiated)
- usleep_range(10000, 15000);
+ if (!rt5682->component || !rt5682->component->card ||
+ !rt5682->component->card->instantiated) {
+ /* card not yet ready, try later */
+ mod_delayed_work(system_power_efficient_wq,
+ &rt5682->jack_detect_work, msecs_to_jiffies(15));
+ return;
+ }
mutex_lock(&rt5682->calibrate_mutex);
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index fe99584c917f..9cd91bb0a902 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -1535,18 +1535,38 @@ static int wm8350_component_probe(struct snd_soc_component *component)
wm8350_clear_bits(wm8350, WM8350_JACK_DETECT,
WM8350_JDL_ENA | WM8350_JDR_ENA);
- wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L,
+ ret = wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L,
wm8350_hpl_jack_handler, 0, "Left jack detect",
priv);
- wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R,
+ if (ret != 0)
+ goto err;
+
+ ret = wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R,
wm8350_hpr_jack_handler, 0, "Right jack detect",
priv);
- wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_MICSCD,
+ if (ret != 0)
+ goto free_jck_det_l;
+
+ ret = wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_MICSCD,
wm8350_mic_handler, 0, "Microphone short", priv);
- wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_MICD,
+ if (ret != 0)
+ goto free_jck_det_r;
+
+ ret = wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_MICD,
wm8350_mic_handler, 0, "Microphone detect", priv);
+ if (ret != 0)
+ goto free_micscd;
return 0;
+
+free_micscd:
+ wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_MICSCD, priv);
+free_jck_det_r:
+ wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R, priv);
+free_jck_det_l:
+ wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L, priv);
+err:
+ return ret;
}
static void wm8350_component_remove(struct snd_soc_component *component)
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 6fd1bef848ed..fa55d79b39b6 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -601,7 +601,7 @@ static int wm8731_hw_init(struct device *dev, struct wm8731_priv *wm8731)
ret = wm8731_reset(wm8731->regmap);
if (ret < 0) {
dev_err(dev, "Failed to issue reset: %d\n", ret);
- goto err_regulator_enable;
+ goto err;
}
/* Clear POWEROFF, keep everything else disabled */
@@ -618,10 +618,7 @@ static int wm8731_hw_init(struct device *dev, struct wm8731_priv *wm8731)
regcache_mark_dirty(wm8731->regmap);
-err_regulator_enable:
- /* Regulators will be enabled by bias management */
- regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies);
-
+err:
return ret;
}
@@ -765,21 +762,27 @@ static int wm8731_i2c_probe(struct i2c_client *i2c,
ret = PTR_ERR(wm8731->regmap);
dev_err(&i2c->dev, "Failed to allocate register map: %d\n",
ret);
- return ret;
+ goto err_regulator_enable;
}
ret = wm8731_hw_init(&i2c->dev, wm8731);
if (ret != 0)
- return ret;
+ goto err_regulator_enable;
ret = devm_snd_soc_register_component(&i2c->dev,
&soc_component_dev_wm8731, &wm8731_dai, 1);
if (ret != 0) {
dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret);
- return ret;
+ goto err_regulator_enable;
}
return 0;
+
+err_regulator_enable:
+ /* Regulators will be enabled by bias management */
+ regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies);
+
+ return ret;
}
static int wm8731_i2c_remove(struct i2c_client *client)
diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c
index 04f23477039a..c677c068b05e 100644
--- a/sound/soc/codecs/wm8958-dsp2.c
+++ b/sound/soc/codecs/wm8958-dsp2.c
@@ -534,7 +534,7 @@ static int wm8958_mbc_put(struct snd_kcontrol *kcontrol,
wm8958_dsp_apply(component, mbc, wm8994->mbc_ena[mbc]);
- return 0;
+ return 1;
}
#define WM8958_MBC_SWITCH(xname, xval) {\
@@ -660,7 +660,7 @@ static int wm8958_vss_put(struct snd_kcontrol *kcontrol,
wm8958_dsp_apply(component, vss, wm8994->vss_ena[vss]);
- return 0;
+ return 1;
}
@@ -734,7 +734,7 @@ static int wm8958_hpf_put(struct snd_kcontrol *kcontrol,
wm8958_dsp_apply(component, hpf % 3, ucontrol->value.integer.value[0]);
- return 0;
+ return 1;
}
#define WM8958_HPF_SWITCH(xname, xval) {\
@@ -828,7 +828,7 @@ static int wm8958_enh_eq_put(struct snd_kcontrol *kcontrol,
wm8958_dsp_apply(component, eq, ucontrol->value.integer.value[0]);
- return 0;
+ return 1;
}
#define WM8958_ENH_EQ_SWITCH(xname, xval) {\
diff --git a/sound/soc/fsl/imx-es8328.c b/sound/soc/fsl/imx-es8328.c
index fad1eb6253d5..9e602c345619 100644
--- a/sound/soc/fsl/imx-es8328.c
+++ b/sound/soc/fsl/imx-es8328.c
@@ -87,6 +87,7 @@ static int imx_es8328_probe(struct platform_device *pdev)
if (int_port > MUX_PORT_MAX || int_port == 0) {
dev_err(dev, "mux-int-port: hardware only has %d mux ports\n",
MUX_PORT_MAX);
+ ret = -EINVAL;
goto fail;
}
diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c
index af3c3b90c0ac..83b4a22bf15a 100644
--- a/sound/soc/fsl/pcm030-audio-fabric.c
+++ b/sound/soc/fsl/pcm030-audio-fabric.c
@@ -93,16 +93,21 @@ static int pcm030_fabric_probe(struct platform_device *op)
dev_err(&op->dev, "platform_device_alloc() failed\n");
ret = platform_device_add(pdata->codec_device);
- if (ret)
+ if (ret) {
dev_err(&op->dev, "platform_device_add() failed: %d\n", ret);
+ platform_device_put(pdata->codec_device);
+ }
ret = snd_soc_register_card(card);
- if (ret)
+ if (ret) {
dev_err(&op->dev, "snd_soc_register_card() failed: %d\n", ret);
+ platform_device_del(pdata->codec_device);
+ platform_device_put(pdata->codec_device);
+ }
platform_set_drvdata(op, pdata);
-
return ret;
+
}
static int pcm030_fabric_remove(struct platform_device *op)
diff --git a/sound/soc/mediatek/mt8173/mt8173-max98090.c b/sound/soc/mediatek/mt8173/mt8173-max98090.c
index 22c00600c999..de1410c2c446 100644
--- a/sound/soc/mediatek/mt8173/mt8173-max98090.c
+++ b/sound/soc/mediatek/mt8173/mt8173-max98090.c
@@ -180,6 +180,9 @@ static int mt8173_max98090_dev_probe(struct platform_device *pdev)
if (ret)
dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n",
__func__, ret);
+
+ of_node_put(codec_node);
+ of_node_put(platform_node);
return ret;
}
diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c
index 8717e87bfe26..6f8542329bab 100644
--- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c
+++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c
@@ -218,6 +218,8 @@ static int mt8173_rt5650_rt5514_dev_probe(struct platform_device *pdev)
if (ret)
dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n",
__func__, ret);
+
+ of_node_put(platform_node);
return ret;
}
diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c
index 9d4dd9721154..727ff0f7f20b 100644
--- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c
+++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c
@@ -285,6 +285,8 @@ static int mt8173_rt5650_rt5676_dev_probe(struct platform_device *pdev)
if (ret)
dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n",
__func__, ret);
+
+ of_node_put(platform_node);
return ret;
}
diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650.c b/sound/soc/mediatek/mt8173/mt8173-rt5650.c
index ef6f23675286..21e7d4d3ded5 100644
--- a/sound/soc/mediatek/mt8173/mt8173-rt5650.c
+++ b/sound/soc/mediatek/mt8173/mt8173-rt5650.c
@@ -309,6 +309,8 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev)
if (ret)
dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n",
__func__, ret);
+
+ of_node_put(platform_node);
return ret;
}
diff --git a/sound/soc/meson/g12a-tohdmitx.c b/sound/soc/meson/g12a-tohdmitx.c
index 9cfbd343a00c..cbe47e0cae42 100644
--- a/sound/soc/meson/g12a-tohdmitx.c
+++ b/sound/soc/meson/g12a-tohdmitx.c
@@ -127,7 +127,7 @@ static int g12a_tohdmitx_i2s_mux_put_enum(struct snd_kcontrol *kcontrol,
snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL);
- return 0;
+ return 1;
}
static const struct snd_kcontrol_new g12a_tohdmitx_i2s_mux =
diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c
index a2c79426513b..d7d272bbebb2 100644
--- a/sound/soc/mxs/mxs-saif.c
+++ b/sound/soc/mxs/mxs-saif.c
@@ -455,7 +455,10 @@ static int mxs_saif_hw_params(struct snd_pcm_substream *substream,
* basic clock which should be fast enough for the internal
* logic.
*/
- clk_enable(saif->clk);
+ ret = clk_enable(saif->clk);
+ if (ret)
+ return ret;
+
ret = clk_set_rate(saif->clk, 24000000);
clk_disable(saif->clk);
if (ret)
diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c
index 9841e1da9782..8282fe6d00dd 100644
--- a/sound/soc/mxs/mxs-sgtl5000.c
+++ b/sound/soc/mxs/mxs-sgtl5000.c
@@ -118,6 +118,9 @@ static int mxs_sgtl5000_probe(struct platform_device *pdev)
codec_np = of_parse_phandle(np, "audio-codec", 0);
if (!saif_np[0] || !saif_np[1] || !codec_np) {
dev_err(&pdev->dev, "phandle missing or invalid\n");
+ of_node_put(codec_np);
+ of_node_put(saif_np[0]);
+ of_node_put(saif_np[1]);
return -EINVAL;
}
diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c
index 745cc9dd14f3..bc65009be875 100644
--- a/sound/soc/qcom/qdsp6/q6routing.c
+++ b/sound/soc/qcom/qdsp6/q6routing.c
@@ -440,9 +440,15 @@ static int msm_routing_put_audio_mixer(struct snd_kcontrol *kcontrol,
struct session_data *session = &data->sessions[session_id];
if (ucontrol->value.integer.value[0]) {
+ if (session->port_id == be_id)
+ return 0;
+
session->port_id = be_id;
snd_soc_dapm_mixer_update_power(dapm, kcontrol, 1, update);
} else {
+ if (session->port_id == -1 || session->port_id != be_id)
+ return 0;
+
session->port_id = -1;
snd_soc_dapm_mixer_update_power(dapm, kcontrol, 0, update);
}
diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c
index 65497cd477a5..47f6f5d70853 100644
--- a/sound/soc/samsung/idma.c
+++ b/sound/soc/samsung/idma.c
@@ -363,6 +363,8 @@ static int preallocate_idma_buffer(struct snd_pcm *pcm, int stream)
buf->addr = idma.lp_tx_addr;
buf->bytes = idma_hardware.buffer_bytes_max;
buf->area = (unsigned char * __force)ioremap(buf->addr, buf->bytes);
+ if (!buf->area)
+ return -ENOMEM;
return 0;
}
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 3447dbdba1f1..6ac7df30a289 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -816,14 +816,27 @@ static int fsi_clk_enable(struct device *dev,
return ret;
}
- clk_enable(clock->xck);
- clk_enable(clock->ick);
- clk_enable(clock->div);
+ ret = clk_enable(clock->xck);
+ if (ret)
+ goto err;
+ ret = clk_enable(clock->ick);
+ if (ret)
+ goto disable_xck;
+ ret = clk_enable(clock->div);
+ if (ret)
+ goto disable_ick;
clock->count++;
}
return ret;
+
+disable_ick:
+ clk_disable(clock->ick);
+disable_xck:
+ clk_disable(clock->xck);
+err:
+ return ret;
}
static int fsi_clk_disable(struct device *dev,
diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c
index 9e54d8ae6d2c..da6e40aef7b6 100644
--- a/sound/soc/soc-compress.c
+++ b/sound/soc/soc-compress.c
@@ -871,6 +871,11 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num)
return -EINVAL;
}
+ if (!codec_dai) {
+ dev_err(rtd->card->dev, "Missing codec\n");
+ return -EINVAL;
+ }
+
/* check client and interface hw capabilities */
if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_PLAYBACK) &&
snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_PLAYBACK))
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 7c10b284555c..66a99d6f9434 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -3364,7 +3364,7 @@ int snd_soc_get_dai_name(struct of_phandle_args *args,
for_each_component(pos) {
component_of_node = soc_component_to_node(pos);
- if (component_of_node != args->np)
+ if (component_of_node != args->np || !pos->num_dai)
continue;
ret = snd_soc_component_of_xlate_dai_name(pos, args, dai_name);
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 66f6b698a543..1c09dfb0c0f0 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -1676,8 +1676,7 @@ static void dapm_seq_run(struct snd_soc_card *card,
switch (w->id) {
case snd_soc_dapm_pre:
if (!w->event)
- list_for_each_entry_safe_continue(w, n, list,
- power_list);
+ continue;
if (event == SND_SOC_DAPM_STREAM_START)
ret = w->event(w,
@@ -1689,8 +1688,7 @@ static void dapm_seq_run(struct snd_soc_card *card,
case snd_soc_dapm_post:
if (!w->event)
- list_for_each_entry_safe_continue(w, n, list,
- power_list);
+ continue;
if (event == SND_SOC_DAPM_STREAM_START)
ret = w->event(w,
@@ -2542,8 +2540,13 @@ static struct snd_soc_dapm_widget *dapm_find_widget(
return NULL;
}
-static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm,
- const char *pin, int status)
+/*
+ * set the DAPM pin status:
+ * returns 1 when the value has been updated, 0 when unchanged, or a negative
+ * error code; called from kcontrol put callback
+ */
+static int __snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm,
+ const char *pin, int status)
{
struct snd_soc_dapm_widget *w = dapm_find_widget(dapm, pin, true);
int ret = 0;
@@ -2569,6 +2572,18 @@ static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm,
return ret;
}
+/*
+ * similar as __snd_soc_dapm_set_pin(), but returns 0 when successful;
+ * called from several API functions below
+ */
+static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm,
+ const char *pin, int status)
+{
+ int ret = __snd_soc_dapm_set_pin(dapm, pin, status);
+
+ return ret < 0 ? ret : 0;
+}
+
/**
* snd_soc_dapm_sync_unlocked - scan and power dapm paths
* @dapm: DAPM context
@@ -3584,10 +3599,10 @@ int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol,
const char *pin = (const char *)kcontrol->private_value;
int ret;
- if (ucontrol->value.integer.value[0])
- ret = snd_soc_dapm_enable_pin(&card->dapm, pin);
- else
- ret = snd_soc_dapm_disable_pin(&card->dapm, pin);
+ mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
+ ret = __snd_soc_dapm_set_pin(&card->dapm, pin,
+ !!ucontrol->value.integer.value[0]);
+ mutex_unlock(&card->dapm_mutex);
snd_soc_dapm_sync(&card->dapm);
return ret;
diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c
index 95fc24580f85..c88bc6bb41cf 100644
--- a/sound/soc/soc-ops.c
+++ b/sound/soc/soc-ops.c
@@ -314,7 +314,7 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol,
unsigned int sign_bit = mc->sign_bit;
unsigned int mask = (1 << fls(max)) - 1;
unsigned int invert = mc->invert;
- int err;
+ int err, ret;
bool type_2r = false;
unsigned int val2 = 0;
unsigned int val, val_mask;
@@ -322,13 +322,27 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol,
if (sign_bit)
mask = BIT(sign_bit + 1) - 1;
- val = ((ucontrol->value.integer.value[0] + min) & mask);
+ val = ucontrol->value.integer.value[0];
+ if (mc->platform_max && ((int)val + min) > mc->platform_max)
+ return -EINVAL;
+ if (val > max - min)
+ return -EINVAL;
+ if (val < 0)
+ return -EINVAL;
+ val = (val + min) & mask;
if (invert)
val = max - val;
val_mask = mask << shift;
val = val << shift;
if (snd_soc_volsw_is_stereo(mc)) {
- val2 = ((ucontrol->value.integer.value[1] + min) & mask);
+ val2 = ucontrol->value.integer.value[1];
+ if (mc->platform_max && ((int)val2 + min) > mc->platform_max)
+ return -EINVAL;
+ if (val2 > max - min)
+ return -EINVAL;
+ if (val2 < 0)
+ return -EINVAL;
+ val2 = (val2 + min) & mask;
if (invert)
val2 = max - val2;
if (reg == reg2) {
@@ -342,12 +356,18 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol,
err = snd_soc_component_update_bits(component, reg, val_mask, val);
if (err < 0)
return err;
+ ret = err;
- if (type_2r)
+ if (type_2r) {
err = snd_soc_component_update_bits(component, reg2, val_mask,
- val2);
+ val2);
+ /* Don't discard any error code or drop change flag */
+ if (ret == 0 || err < 0) {
+ ret = err;
+ }
+ }
- return err;
+ return ret;
}
EXPORT_SYMBOL_GPL(snd_soc_put_volsw);
@@ -422,8 +442,15 @@ int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol,
int err = 0;
unsigned int val, val_mask, val2 = 0;
+ val = ucontrol->value.integer.value[0];
+ if (mc->platform_max && val > mc->platform_max)
+ return -EINVAL;
+ if (val > max - min)
+ return -EINVAL;
+ if (val < 0)
+ return -EINVAL;
val_mask = mask << shift;
- val = (ucontrol->value.integer.value[0] + min) & mask;
+ val = (val + min) & mask;
val = val << shift;
err = snd_soc_component_update_bits(component, reg, val_mask, val);
@@ -496,7 +523,7 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol,
unsigned int mask = (1 << fls(max)) - 1;
unsigned int invert = mc->invert;
unsigned int val, val_mask;
- int ret;
+ int err, ret;
if (invert)
val = (max - ucontrol->value.integer.value[0]) & mask;
@@ -505,9 +532,10 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol,
val_mask = mask << shift;
val = val << shift;
- ret = snd_soc_component_update_bits(component, reg, val_mask, val);
- if (ret < 0)
- return ret;
+ err = snd_soc_component_update_bits(component, reg, val_mask, val);
+ if (err < 0)
+ return err;
+ ret = err;
if (snd_soc_volsw_is_stereo(mc)) {
if (invert)
@@ -517,8 +545,12 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol,
val_mask = mask << shift;
val = val << shift;
- ret = snd_soc_component_update_bits(component, rreg, val_mask,
+ err = snd_soc_component_update_bits(component, rreg, val_mask,
val);
+ /* Don't discard any error code or drop change flag */
+ if (ret == 0 || err < 0) {
+ ret = err;
+ }
}
return ret;
@@ -889,6 +921,8 @@ int snd_soc_put_xr_sx(struct snd_kcontrol *kcontrol,
unsigned int i, regval, regmask;
int err;
+ if (val < mc->min || val > mc->max)
+ return -EINVAL;
if (invert)
val = max - val;
val &= mask;
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index c367609433bf..870b00229353 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -587,7 +587,8 @@ static int soc_tplg_kcontrol_bind_io(struct snd_soc_tplg_ctl_hdr *hdr,
if (le32_to_cpu(hdr->ops.info) == SND_SOC_TPLG_CTL_BYTES
&& k->iface & SNDRV_CTL_ELEM_IFACE_MIXER
- && k->access & SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE
+ && (k->access & SNDRV_CTL_ELEM_ACCESS_TLV_READ
+ || k->access & SNDRV_CTL_ELEM_ACCESS_TLV_WRITE)
&& k->access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK) {
struct soc_bytes_ext *sbe;
struct snd_soc_tplg_bytes_control *be;
@@ -2777,6 +2778,7 @@ EXPORT_SYMBOL_GPL(snd_soc_tplg_widget_remove_all);
/* remove dynamic controls from the component driver */
int snd_soc_tplg_component_remove(struct snd_soc_component *comp, u32 index)
{
+ struct snd_card *card = comp->card->snd_card;
struct snd_soc_dobj *dobj, *next_dobj;
int pass = SOC_TPLG_PASS_END;
@@ -2784,6 +2786,7 @@ int snd_soc_tplg_component_remove(struct snd_soc_component *comp, u32 index)
while (pass >= SOC_TPLG_PASS_START) {
/* remove mixer controls */
+ down_write(&card->controls_rwsem);
list_for_each_entry_safe(dobj, next_dobj, &comp->dobj_list,
list) {
@@ -2827,6 +2830,7 @@ int snd_soc_tplg_component_remove(struct snd_soc_component *comp, u32 index)
break;
}
}
+ up_write(&card->controls_rwsem);
pass--;
}
diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c
index 3f645200d3a5..b3cdd10c83ae 100644
--- a/sound/soc/sof/intel/hda-dai.c
+++ b/sound/soc/sof/intel/hda-dai.c
@@ -67,6 +67,7 @@ static struct hdac_ext_stream *
return NULL;
}
+ spin_lock_irq(&bus->reg_lock);
list_for_each_entry(stream, &bus->stream_list, list) {
struct hdac_ext_stream *hstream =
stream_to_hdac_ext_stream(stream);
@@ -106,12 +107,12 @@ static struct hdac_ext_stream *
* is updated in snd_hdac_ext_stream_decouple().
*/
if (!res->decoupled)
- snd_hdac_ext_stream_decouple(bus, res, true);
- spin_lock_irq(&bus->reg_lock);
+ snd_hdac_ext_stream_decouple_locked(bus, res, true);
+
res->link_locked = 1;
res->link_substream = substream;
- spin_unlock_irq(&bus->reg_lock);
}
+ spin_unlock_irq(&bus->reg_lock);
return res;
}
diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c
index 356bb134ae93..7573f3f9f0f2 100644
--- a/sound/soc/sof/intel/hda-loader.c
+++ b/sound/soc/sof/intel/hda-loader.c
@@ -50,7 +50,7 @@ static int cl_stream_prepare(struct snd_sof_dev *sdev, unsigned int format,
ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV_SG, &pci->dev, size, dmab);
if (ret < 0) {
dev_err(sdev->dev, "error: memory alloc failed: %x\n", ret);
- goto error;
+ goto out_put;
}
hstream->period_bytes = 0;/* initialize period_bytes */
@@ -60,16 +60,17 @@ static int cl_stream_prepare(struct snd_sof_dev *sdev, unsigned int format,
ret = hda_dsp_stream_hw_params(sdev, dsp_stream, dmab, NULL);
if (ret < 0) {
dev_err(sdev->dev, "error: hdac prepare failed: %x\n", ret);
- goto error;
+ goto out_free;
}
hda_dsp_stream_spib_config(sdev, dsp_stream, HDA_DSP_SPIB_ENABLE, size);
return hstream->stream_tag;
-error:
- hda_dsp_stream_put(sdev, direction, hstream->stream_tag);
+out_free:
snd_dma_free_pages(dmab);
+out_put:
+ hda_dsp_stream_put(sdev, direction, hstream->stream_tag);
return ret;
}
diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c
index 2ed92c990b97..dd9013c47664 100644
--- a/sound/soc/sti/uniperif_player.c
+++ b/sound/soc/sti/uniperif_player.c
@@ -91,7 +91,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id)
SET_UNIPERIF_ITM_BCLR_FIFO_ERROR(player);
/* Stop the player */
- snd_pcm_stop_xrun(player->substream);
+ snd_pcm_stop(player->substream, SNDRV_PCM_STATE_XRUN);
}
ret = IRQ_HANDLED;
@@ -105,7 +105,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id)
SET_UNIPERIF_ITM_BCLR_DMA_ERROR(player);
/* Stop the player */
- snd_pcm_stop_xrun(player->substream);
+ snd_pcm_stop(player->substream, SNDRV_PCM_STATE_XRUN);
ret = IRQ_HANDLED;
}
@@ -138,7 +138,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id)
dev_err(player->dev, "Underflow recovery failed\n");
/* Stop the player */
- snd_pcm_stop_xrun(player->substream);
+ snd_pcm_stop(player->substream, SNDRV_PCM_STATE_XRUN);
ret = IRQ_HANDLED;
}
diff --git a/sound/soc/sti/uniperif_reader.c b/sound/soc/sti/uniperif_reader.c
index 136059331211..065c5f0d1f5f 100644
--- a/sound/soc/sti/uniperif_reader.c
+++ b/sound/soc/sti/uniperif_reader.c
@@ -65,7 +65,7 @@ static irqreturn_t uni_reader_irq_handler(int irq, void *dev_id)
if (unlikely(status & UNIPERIF_ITS_FIFO_ERROR_MASK(reader))) {
dev_err(reader->dev, "FIFO error detected\n");
- snd_pcm_stop_xrun(reader->substream);
+ snd_pcm_stop(reader->substream, SNDRV_PCM_STATE_XRUN);
ret = IRQ_HANDLED;
}
diff --git a/sound/soc/ti/davinci-i2s.c b/sound/soc/ti/davinci-i2s.c
index d89b5c928c4d..b2b2dcdb05d4 100644
--- a/sound/soc/ti/davinci-i2s.c
+++ b/sound/soc/ti/davinci-i2s.c
@@ -708,7 +708,9 @@ static int davinci_i2s_probe(struct platform_device *pdev)
dev->clk = clk_get(&pdev->dev, NULL);
if (IS_ERR(dev->clk))
return -ENODEV;
- clk_enable(dev->clk);
+ ret = clk_enable(dev->clk);
+ if (ret)
+ goto err_put_clk;
dev->dev = &pdev->dev;
dev_set_drvdata(&pdev->dev, dev);
@@ -730,6 +732,7 @@ err_unregister_component:
snd_soc_unregister_component(&pdev->dev);
err_release_clk:
clk_disable(dev->clk);
+err_put_clk:
clk_put(dev->clk);
return ret;
}
diff --git a/sound/soc/uniphier/Kconfig b/sound/soc/uniphier/Kconfig
index aa3592ee1358..ddfa6424c656 100644
--- a/sound/soc/uniphier/Kconfig
+++ b/sound/soc/uniphier/Kconfig
@@ -23,7 +23,6 @@ config SND_SOC_UNIPHIER_LD11
tristate "UniPhier LD11/LD20 Device Driver"
depends on SND_SOC_UNIPHIER
select SND_SOC_UNIPHIER_AIO
- select SND_SOC_UNIPHIER_AIO_DMA
help
This adds ASoC driver for Socionext UniPhier LD11/LD20
input and output that can be used with other codecs.
@@ -34,7 +33,6 @@ config SND_SOC_UNIPHIER_PXS2
tristate "UniPhier PXs2 Device Driver"
depends on SND_SOC_UNIPHIER
select SND_SOC_UNIPHIER_AIO
- select SND_SOC_UNIPHIER_AIO_DMA
help
This adds ASoC driver for Socionext UniPhier PXs2
input and output that can be used with other codecs.
diff --git a/sound/soc/xilinx/xlnx_formatter_pcm.c b/sound/soc/xilinx/xlnx_formatter_pcm.c
index d0e22dafadeb..f6b3a5bdbcea 100644
--- a/sound/soc/xilinx/xlnx_formatter_pcm.c
+++ b/sound/soc/xilinx/xlnx_formatter_pcm.c
@@ -37,6 +37,7 @@
#define XLNX_AUD_XFER_COUNT 0x28
#define XLNX_AUD_CH_STS_START 0x2C
#define XLNX_BYTES_PER_CH 0x44
+#define XLNX_AUD_ALIGN_BYTES 64
#define AUD_STS_IOC_IRQ_MASK BIT(31)
#define AUD_STS_CH_STS_MASK BIT(29)
@@ -370,12 +371,32 @@ static int xlnx_formatter_pcm_open(struct snd_pcm_substream *substream)
snd_soc_set_runtime_hwparams(substream, &xlnx_pcm_hardware);
runtime->private_data = stream_data;
- /* Resize the period size divisible by 64 */
+ /* Resize the period bytes as divisible by 64 */
err = snd_pcm_hw_constraint_step(runtime, 0,
- SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64);
+ SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
+ XLNX_AUD_ALIGN_BYTES);
if (err) {
dev_err(component->dev,
- "unable to set constraint on period bytes\n");
+ "Unable to set constraint on period bytes\n");
+ return err;
+ }
+
+ /* Resize the buffer bytes as divisible by 64 */
+ err = snd_pcm_hw_constraint_step(runtime, 0,
+ SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
+ XLNX_AUD_ALIGN_BYTES);
+ if (err) {
+ dev_err(component->dev,
+ "Unable to set constraint on buffer bytes\n");
+ return err;
+ }
+
+ /* Set periods as integer multiple */
+ err = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ if (err < 0) {
+ dev_err(component->dev,
+ "Unable to set constraint on periods to be integer\n");
return err;
}
diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c
index 4de1ba9a418d..6e5d315bab59 100644
--- a/sound/spi/at73c213.c
+++ b/sound/spi/at73c213.c
@@ -218,7 +218,9 @@ static int snd_at73c213_pcm_open(struct snd_pcm_substream *substream)
runtime->hw = snd_at73c213_playback_hw;
chip->substream = substream;
- clk_enable(chip->ssc->clk);
+ err = clk_enable(chip->ssc->clk);
+ if (err)
+ return err;
return 0;
}
@@ -784,7 +786,9 @@ static int snd_at73c213_chip_init(struct snd_at73c213 *chip)
goto out;
/* Enable DAC master clock. */
- clk_enable(chip->board->dac_clk);
+ retval = clk_enable(chip->board->dac_clk);
+ if (retval)
+ goto out;
/* Initialize at73c213 on SPI bus. */
retval = snd_at73c213_write_reg(chip, DAC_RST, 0x04);
@@ -897,7 +901,9 @@ static int snd_at73c213_dev_init(struct snd_card *card,
chip->card = card;
chip->irq = -1;
- clk_enable(chip->ssc->clk);
+ retval = clk_enable(chip->ssc->clk);
+ if (retval)
+ return retval;
retval = request_irq(irq, snd_at73c213_interrupt, 0, "at73c213", chip);
if (retval) {
@@ -1016,7 +1022,9 @@ static int snd_at73c213_remove(struct spi_device *spi)
int retval;
/* Stop playback. */
- clk_enable(chip->ssc->clk);
+ retval = clk_enable(chip->ssc->clk);
+ if (retval)
+ goto out;
ssc_writel(chip->ssc->regs, CR, SSC_BIT(CR_TXDIS));
clk_disable(chip->ssc->clk);
@@ -1096,9 +1104,16 @@ static int snd_at73c213_resume(struct device *dev)
{
struct snd_card *card = dev_get_drvdata(dev);
struct snd_at73c213 *chip = card->private_data;
+ int retval;
- clk_enable(chip->board->dac_clk);
- clk_enable(chip->ssc->clk);
+ retval = clk_enable(chip->board->dac_clk);
+ if (retval)
+ return retval;
+ retval = clk_enable(chip->ssc->clk);
+ if (retval) {
+ clk_disable(chip->board->dac_clk);
+ return retval;
+ }
ssc_writel(chip->ssc->regs, CR, SSC_BIT(CR_TXEN));
return 0;
diff --git a/sound/usb/midi.c b/sound/usb/midi.c
index 33e9a7f6246f..ce501200e592 100644
--- a/sound/usb/midi.c
+++ b/sound/usb/midi.c
@@ -1210,6 +1210,7 @@ static void snd_usbmidi_output_drain(struct snd_rawmidi_substream *substream)
} while (drain_urbs && timeout);
finish_wait(&ep->drain_wait, &wait);
}
+ port->active = 0;
spin_unlock_irq(&ep->buffer_lock);
}
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index d926869c031b..1f7c80541d03 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -2370,9 +2370,10 @@ void snd_usb_mixer_fu_apply_quirk(struct usb_mixer_interface *mixer,
if (unitid == 7 && cval->control == UAC_FU_VOLUME)
snd_dragonfly_quirk_db_scale(mixer, cval, kctl);
break;
- /* lowest playback value is muted on C-Media devices */
- case USB_ID(0x0d8c, 0x000c):
- case USB_ID(0x0d8c, 0x0014):
+ /* lowest playback value is muted on some devices */
+ case USB_ID(0x0d8c, 0x000c): /* C-Media */
+ case USB_ID(0x0d8c, 0x0014): /* C-Media */
+ case USB_ID(0x19f7, 0x0003): /* RODE NT-USB */
if (strstr(kctl->id.name, "Playback"))
cval->min_mute = 1;
break;
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 01dee2074ab3..c29ccdf9e8bc 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -25,6 +25,16 @@
.idProduct = prod, \
.bInterfaceClass = USB_CLASS_VENDOR_SPEC
+/* A standard entry matching with vid/pid and the audio class/subclass */
+#define USB_AUDIO_DEVICE(vend, prod) \
+ .match_flags = USB_DEVICE_ID_MATCH_DEVICE | \
+ USB_DEVICE_ID_MATCH_INT_CLASS | \
+ USB_DEVICE_ID_MATCH_INT_SUBCLASS, \
+ .idVendor = vend, \
+ .idProduct = prod, \
+ .bInterfaceClass = USB_CLASS_AUDIO, \
+ .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL
+
/* HP Thunderbolt Dock Audio Headset */
{
USB_DEVICE(0x03f0, 0x0269),
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index ff97fdcf63bd..b1959e04cbb1 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -8,7 +8,7 @@
*/
/* handling of USB vendor/product ID pairs as 32-bit numbers */
-#define USB_ID(vendor, product) (((vendor) << 16) | (product))
+#define USB_ID(vendor, product) (((unsigned int)(vendor) << 16) | (product))
#define USB_ID_VENDOR(id) ((id) >> 16)
#define USB_ID_PRODUCT(id) ((u16)(id))
diff --git a/sound/x86/intel_hdmi_audio.c b/sound/x86/intel_hdmi_audio.c
index 5fd4e32247a6..a314f13e3292 100644
--- a/sound/x86/intel_hdmi_audio.c
+++ b/sound/x86/intel_hdmi_audio.c
@@ -1279,7 +1279,7 @@ static int had_pcm_mmap(struct snd_pcm_substream *substream,
{
vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot);
return remap_pfn_range(vma, vma->vm_start,
- substream->dma_buffer.addr >> PAGE_SHIFT,
+ substream->runtime->dma_addr >> PAGE_SHIFT,
vma->vm_end - vma->vm_start, vma->vm_page_prot);
}