Update Linux to v5.4.148
Sourced from [1]
[1] https://cdn.kernel.org/pub/linux/kernel/v5.x/linux-5.4.148.tar.gz
Change-Id: Ib3d26c5ba9b022e2e03533005c4fed4d7c30b61b
Signed-off-by: Olivier Deprez <olivier.deprez@arm.com>
diff --git a/sound/ac97/bus.c b/sound/ac97/bus.c
index 7985dd8..99e1728 100644
--- a/sound/ac97/bus.c
+++ b/sound/ac97/bus.c
@@ -520,7 +520,7 @@
struct ac97_codec_driver *adrv = to_ac97_driver(dev->driver);
int ret;
- ret = pm_runtime_get_sync(dev);
+ ret = pm_runtime_resume_and_get(dev);
if (ret < 0)
return ret;
diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c
index f34ce56..1afa06b 100644
--- a/sound/core/compress_offload.c
+++ b/sound/core/compress_offload.c
@@ -722,6 +722,9 @@
retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_STOP);
if (!retval) {
+ /* clear flags and stop any drain wait */
+ stream->partial_drain = false;
+ stream->metadata_set = false;
snd_compr_drain_notify(stream);
stream->runtime->total_bytes_available = 0;
stream->runtime->total_bytes_transferred = 0;
@@ -879,6 +882,7 @@
if (stream->next_track == false)
return -EPERM;
+ stream->partial_drain = true;
retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_PARTIAL_DRAIN);
if (retval) {
pr_debug("Partial drain returned failure\n");
diff --git a/sound/core/control.c b/sound/core/control.c
index 7a4d869..15efa4b 100644
--- a/sound/core/control.c
+++ b/sound/core/control.c
@@ -1350,7 +1350,7 @@
unlock:
up_write(&card->controls_rwsem);
- return 0;
+ return err;
}
static int snd_ctl_elem_add_user(struct snd_ctl_file *file,
@@ -1430,8 +1430,9 @@
if (kctl->tlv.c == NULL)
return -ENXIO;
- /* When locked, this is unavailable. */
- if (vd->owner != NULL && vd->owner != file)
+ /* Write and command operations are not allowed for locked element. */
+ if (op_flag != SNDRV_CTL_TLV_OP_READ &&
+ vd->owner != NULL && vd->owner != file)
return -EPERM;
return kctl->tlv.c(kctl, op_flag, size, buf);
diff --git a/sound/core/hwdep.c b/sound/core/hwdep.c
index 00cb5ae..28bec15 100644
--- a/sound/core/hwdep.c
+++ b/sound/core/hwdep.c
@@ -216,12 +216,12 @@
if (info.index >= 32)
return -EINVAL;
/* check whether the dsp was already loaded */
- if (hw->dsp_loaded & (1 << info.index))
+ if (hw->dsp_loaded & (1u << info.index))
return -EBUSY;
err = hw->ops.dsp_load(hw, &info);
if (err < 0)
return err;
- hw->dsp_loaded |= (1 << info.index);
+ hw->dsp_loaded |= (1u << info.index);
return 0;
}
diff --git a/sound/core/info.c b/sound/core/info.c
index e051a02..f18f4ef 100644
--- a/sound/core/info.c
+++ b/sound/core/info.c
@@ -608,7 +608,9 @@
{
int c = -1;
- if (snd_BUG_ON(!buffer || !buffer->buffer))
+ if (snd_BUG_ON(!buffer))
+ return 1;
+ if (!buffer->buffer)
return 1;
if (len <= 0 || buffer->stop || buffer->error)
return 1;
diff --git a/sound/core/init.c b/sound/core/init.c
index db99b7f..45bbc48 100644
--- a/sound/core/init.c
+++ b/sound/core/init.c
@@ -384,10 +384,8 @@
return 0;
}
card->shutdown = 1;
- spin_unlock(&card->files_lock);
/* replace file->f_op with special dummy operations */
- spin_lock(&card->files_lock);
list_for_each_entry(mfile, &card->files_list, list) {
/* it's critical part, use endless loop */
/* we have no room to fail */
diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c
index 6850d13..fe1ea03 100644
--- a/sound/core/memalloc.c
+++ b/sound/core/memalloc.c
@@ -76,7 +76,8 @@
/* Assign the pool into private_data field */
dmab->private_data = pool;
- dmab->area = gen_pool_dma_alloc(pool, size, &dmab->addr);
+ dmab->area = gen_pool_dma_alloc_align(pool, size, &dmab->addr,
+ PAGE_SIZE);
}
/**
diff --git a/sound/core/oss/linear.c b/sound/core/oss/linear.c
index 2045697..797d838 100644
--- a/sound/core/oss/linear.c
+++ b/sound/core/oss/linear.c
@@ -107,6 +107,8 @@
}
}
#endif
+ if (frames > dst_channels[0].frames)
+ frames = dst_channels[0].frames;
convert(plugin, src_channels, dst_channels, frames);
return frames;
}
diff --git a/sound/core/oss/mulaw.c b/sound/core/oss/mulaw.c
index 7915564..fe27034 100644
--- a/sound/core/oss/mulaw.c
+++ b/sound/core/oss/mulaw.c
@@ -269,6 +269,8 @@
}
}
#endif
+ if (frames > dst_channels[0].frames)
+ frames = dst_channels[0].frames;
data = (struct mulaw_priv *)plugin->extra_data;
data->func(plugin, src_channels, dst_channels, frames);
return frames;
@@ -327,8 +329,8 @@
snd_BUG();
return -EINVAL;
}
- if (snd_BUG_ON(!snd_pcm_format_linear(format->format)))
- return -ENXIO;
+ if (!snd_pcm_format_linear(format->format))
+ return -EINVAL;
err = snd_pcm_plugin_build(plug, "Mu-Law<->linear conversion",
src_format, dst_format,
diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c
index f57c610..0b03777 100644
--- a/sound/core/oss/pcm_oss.c
+++ b/sound/core/oss/pcm_oss.c
@@ -693,6 +693,8 @@
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)
+ return -EINVAL;
oss_buffer_size = rounddown_pow_of_two(oss_buffer_size);
if (atomic_read(&substream->mmap_count)) {
if (oss_buffer_size > runtime->oss.mmap_bytes)
@@ -728,17 +730,21 @@
min_period_size = snd_pcm_plug_client_size(substream,
snd_pcm_hw_param_value_min(slave_params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, NULL));
- min_period_size *= oss_frame_size;
- min_period_size = roundup_pow_of_two(min_period_size);
- if (oss_period_size < min_period_size)
- oss_period_size = min_period_size;
+ if (min_period_size) {
+ min_period_size *= oss_frame_size;
+ min_period_size = roundup_pow_of_two(min_period_size);
+ if (oss_period_size < min_period_size)
+ oss_period_size = min_period_size;
+ }
max_period_size = snd_pcm_plug_client_size(substream,
snd_pcm_hw_param_value_max(slave_params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, NULL));
- max_period_size *= oss_frame_size;
- max_period_size = rounddown_pow_of_two(max_period_size);
- if (oss_period_size > max_period_size)
- oss_period_size = max_period_size;
+ if (max_period_size) {
+ max_period_size *= oss_frame_size;
+ max_period_size = rounddown_pow_of_two(max_period_size);
+ if (oss_period_size > max_period_size)
+ oss_period_size = max_period_size;
+ }
oss_periods = oss_buffer_size / oss_period_size;
@@ -1934,11 +1940,15 @@
static int snd_pcm_oss_set_fragment1(struct snd_pcm_substream *substream, unsigned int val)
{
struct snd_pcm_runtime *runtime;
+ int fragshift;
runtime = substream->runtime;
if (runtime->oss.subdivision || runtime->oss.fragshift)
return -EINVAL;
- runtime->oss.fragshift = val & 0xffff;
+ fragshift = val & 0xffff;
+ if (fragshift >= 31)
+ return -EINVAL;
+ runtime->oss.fragshift = fragshift;
runtime->oss.maxfrags = (val >> 16) & 0xffff;
if (runtime->oss.fragshift < 4) /* < 16 */
runtime->oss.fragshift = 4;
diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c
index 31cb2ac..da400da 100644
--- a/sound/core/oss/pcm_plugin.c
+++ b/sound/core/oss/pcm_plugin.c
@@ -111,7 +111,7 @@
while (plugin->next) {
if (plugin->dst_frames)
frames = plugin->dst_frames(plugin, frames);
- if (snd_BUG_ON((snd_pcm_sframes_t)frames <= 0))
+ if ((snd_pcm_sframes_t)frames <= 0)
return -ENXIO;
plugin = plugin->next;
err = snd_pcm_plugin_alloc(plugin, frames);
@@ -123,7 +123,7 @@
while (plugin->prev) {
if (plugin->src_frames)
frames = plugin->src_frames(plugin, frames);
- if (snd_BUG_ON((snd_pcm_sframes_t)frames <= 0))
+ if ((snd_pcm_sframes_t)frames <= 0)
return -ENXIO;
plugin = plugin->prev;
err = snd_pcm_plugin_alloc(plugin, frames);
@@ -196,7 +196,9 @@
return 0;
}
-snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *plug, snd_pcm_uframes_t drv_frames)
+static snd_pcm_sframes_t plug_client_size(struct snd_pcm_substream *plug,
+ snd_pcm_uframes_t drv_frames,
+ bool check_size)
{
struct snd_pcm_plugin *plugin, *plugin_prev, *plugin_next;
int stream;
@@ -212,12 +214,18 @@
plugin_prev = plugin->prev;
if (plugin->src_frames)
drv_frames = plugin->src_frames(plugin, drv_frames);
+ if (check_size && plugin->buf_frames &&
+ drv_frames > plugin->buf_frames)
+ drv_frames = plugin->buf_frames;
plugin = plugin_prev;
}
} else if (stream == SNDRV_PCM_STREAM_CAPTURE) {
plugin = snd_pcm_plug_first(plug);
while (plugin && drv_frames > 0) {
plugin_next = plugin->next;
+ if (check_size && plugin->buf_frames &&
+ drv_frames > plugin->buf_frames)
+ drv_frames = plugin->buf_frames;
if (plugin->dst_frames)
drv_frames = plugin->dst_frames(plugin, drv_frames);
plugin = plugin_next;
@@ -227,7 +235,9 @@
return drv_frames;
}
-snd_pcm_sframes_t snd_pcm_plug_slave_size(struct snd_pcm_substream *plug, snd_pcm_uframes_t clt_frames)
+static snd_pcm_sframes_t plug_slave_size(struct snd_pcm_substream *plug,
+ snd_pcm_uframes_t clt_frames,
+ bool check_size)
{
struct snd_pcm_plugin *plugin, *plugin_prev, *plugin_next;
snd_pcm_sframes_t frames;
@@ -243,6 +253,9 @@
plugin = snd_pcm_plug_first(plug);
while (plugin && frames > 0) {
plugin_next = plugin->next;
+ if (check_size && plugin->buf_frames &&
+ frames > plugin->buf_frames)
+ frames = plugin->buf_frames;
if (plugin->dst_frames) {
frames = plugin->dst_frames(plugin, frames);
if (frames < 0)
@@ -259,6 +272,9 @@
if (frames < 0)
return frames;
}
+ if (check_size && plugin->buf_frames &&
+ frames > plugin->buf_frames)
+ frames = plugin->buf_frames;
plugin = plugin_prev;
}
} else
@@ -266,6 +282,18 @@
return frames;
}
+snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *plug,
+ snd_pcm_uframes_t drv_frames)
+{
+ return plug_client_size(plug, drv_frames, false);
+}
+
+snd_pcm_sframes_t snd_pcm_plug_slave_size(struct snd_pcm_substream *plug,
+ snd_pcm_uframes_t clt_frames)
+{
+ return plug_slave_size(plug, clt_frames, false);
+}
+
static int snd_pcm_plug_formats(const struct snd_mask *mask,
snd_pcm_format_t format)
{
@@ -622,7 +650,7 @@
src_channels = dst_channels;
plugin = next;
}
- return snd_pcm_plug_client_size(plug, frames);
+ return plug_client_size(plug, frames, true);
}
snd_pcm_sframes_t snd_pcm_plug_read_transfer(struct snd_pcm_substream *plug, struct snd_pcm_plugin_channel *dst_channels_final, snd_pcm_uframes_t size)
@@ -632,7 +660,7 @@
snd_pcm_sframes_t frames = size;
int err;
- frames = snd_pcm_plug_slave_size(plug, frames);
+ frames = plug_slave_size(plug, frames, true);
if (frames < 0)
return frames;
diff --git a/sound/core/oss/route.c b/sound/core/oss/route.c
index c8171f5..72dea04 100644
--- a/sound/core/oss/route.c
+++ b/sound/core/oss/route.c
@@ -57,6 +57,8 @@
return -ENXIO;
if (frames == 0)
return 0;
+ if (frames > dst_channels[0].frames)
+ frames = dst_channels[0].frames;
nsrcs = plugin->src_format.channels;
ndsts = plugin->dst_format.channels;
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 2236b5e..fd300c3 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -423,6 +423,7 @@
no_delta_check:
if (runtime->status->hw_ptr == new_hw_ptr) {
+ runtime->hw_ptr_jiffies = curr_jiffies;
update_audio_tstamp(substream, &curr_tstamp, &audio_tstamp);
return 0;
}
@@ -1735,7 +1736,7 @@
channels = params_channels(params);
frame_size = snd_pcm_format_size(format, channels);
if (frame_size > 0)
- params->fifo_size /= (unsigned)frame_size;
+ params->fifo_size /= frame_size;
}
return 0;
}
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 91c6ad5..0c5b7a5 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -136,6 +136,16 @@
}
EXPORT_SYMBOL_GPL(snd_pcm_stream_lock_irq);
+static void snd_pcm_stream_lock_nested(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_group *group = &substream->self_group;
+
+ if (substream->pcm->nonatomic)
+ mutex_lock_nested(&group->mutex, SINGLE_DEPTH_NESTING);
+ else
+ spin_lock_nested(&group->lock, SINGLE_DEPTH_NESTING);
+}
+
/**
* snd_pcm_stream_unlock_irq - Unlock the PCM stream
* @substream: PCM substream
@@ -222,7 +232,8 @@
return false;
if (substream->ops->mmap ||
- substream->dma_buffer.dev.type != SNDRV_DMA_TYPE_DEV)
+ (substream->dma_buffer.dev.type != SNDRV_DMA_TYPE_DEV &&
+ substream->dma_buffer.dev.type != SNDRV_DMA_TYPE_DEV_UC))
return true;
return dma_can_mmap(substream->dma_buffer.dev.dev);
@@ -705,6 +716,15 @@
while (runtime->boundary * 2 <= LONG_MAX - runtime->buffer_size)
runtime->boundary *= 2;
+ /* clear the buffer for avoiding possible kernel info leaks */
+ if (runtime->dma_area && !substream->ops->copy_user) {
+ size_t size = runtime->dma_bytes;
+
+ if (runtime->info & SNDRV_PCM_INFO_MMAP)
+ size = PAGE_ALIGN(size);
+ memset(runtime->dma_area, 0, size);
+ }
+
snd_pcm_timer_resolution_change(substream);
snd_pcm_set_state(substream, SNDRV_PCM_STATE_SETUP);
@@ -1989,6 +2009,12 @@
}
pcm_file = f.file->private_data;
substream1 = pcm_file->substream;
+
+ if (substream == substream1) {
+ res = -EINVAL;
+ goto _badf;
+ }
+
group = kzalloc(sizeof(*group), GFP_KERNEL);
if (!group) {
res = -ENOMEM;
@@ -2017,7 +2043,7 @@
snd_pcm_stream_unlock_irq(substream);
snd_pcm_group_lock_irq(target_group, nonatomic);
- snd_pcm_stream_lock(substream1);
+ snd_pcm_stream_lock_nested(substream1);
snd_pcm_group_assign(substream1, target_group);
refcount_inc(&target_group->refs);
snd_pcm_stream_unlock(substream1);
@@ -2033,7 +2059,7 @@
static void relink_to_local(struct snd_pcm_substream *substream)
{
- snd_pcm_stream_lock(substream);
+ snd_pcm_stream_lock_nested(substream);
snd_pcm_group_assign(substream, &substream->self_group);
snd_pcm_stream_unlock(substream);
}
@@ -3403,7 +3429,8 @@
#endif /* CONFIG_GENERIC_ALLOCATOR */
#ifndef CONFIG_X86 /* for avoiding warnings arch/x86/mm/pat.c */
if (IS_ENABLED(CONFIG_HAS_DMA) && !substream->ops->page &&
- substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV)
+ (substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV ||
+ substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV_UC))
return dma_mmap_coherent(substream->dma_buffer.dev.dev,
area,
substream->runtime->dma_area,
diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c
index 8a12a75..6a3543b 100644
--- a/sound/core/rawmidi.c
+++ b/sound/core/rawmidi.c
@@ -72,11 +72,21 @@
}
}
-static inline int snd_rawmidi_ready(struct snd_rawmidi_substream *substream)
+static inline bool __snd_rawmidi_ready(struct snd_rawmidi_runtime *runtime)
+{
+ return runtime->avail >= runtime->avail_min;
+}
+
+static bool snd_rawmidi_ready(struct snd_rawmidi_substream *substream)
{
struct snd_rawmidi_runtime *runtime = substream->runtime;
+ unsigned long flags;
+ bool ready;
- return runtime->avail >= runtime->avail_min;
+ spin_lock_irqsave(&runtime->lock, flags);
+ ready = __snd_rawmidi_ready(runtime);
+ spin_unlock_irqrestore(&runtime->lock, flags);
+ return ready;
}
static inline int snd_rawmidi_ready_append(struct snd_rawmidi_substream *substream,
@@ -97,6 +107,17 @@
runtime->event(runtime->substream);
}
+/* buffer refcount management: call with runtime->lock held */
+static inline void snd_rawmidi_buffer_ref(struct snd_rawmidi_runtime *runtime)
+{
+ runtime->buffer_ref++;
+}
+
+static inline void snd_rawmidi_buffer_unref(struct snd_rawmidi_runtime *runtime)
+{
+ runtime->buffer_ref--;
+}
+
static int snd_rawmidi_runtime_create(struct snd_rawmidi_substream *substream)
{
struct snd_rawmidi_runtime *runtime;
@@ -646,6 +667,11 @@
if (!newbuf)
return -ENOMEM;
spin_lock_irq(&runtime->lock);
+ if (runtime->buffer_ref) {
+ spin_unlock_irq(&runtime->lock);
+ kvfree(newbuf);
+ return -EBUSY;
+ }
oldbuf = runtime->buffer;
runtime->buffer = newbuf;
runtime->buffer_size = params->buffer_size;
@@ -929,7 +955,7 @@
if (result > 0) {
if (runtime->event)
schedule_work(&runtime->event_work);
- else if (snd_rawmidi_ready(substream))
+ else if (__snd_rawmidi_ready(runtime))
wake_up(&runtime->sleep);
}
spin_unlock_irqrestore(&runtime->lock, flags);
@@ -945,8 +971,10 @@
long result = 0, count1;
struct snd_rawmidi_runtime *runtime = substream->runtime;
unsigned long appl_ptr;
+ int err = 0;
spin_lock_irqsave(&runtime->lock, flags);
+ snd_rawmidi_buffer_ref(runtime);
while (count > 0 && runtime->avail) {
count1 = runtime->buffer_size - runtime->appl_ptr;
if (count1 > count)
@@ -965,16 +993,19 @@
if (userbuf) {
spin_unlock_irqrestore(&runtime->lock, flags);
if (copy_to_user(userbuf + result,
- runtime->buffer + appl_ptr, count1)) {
- return result > 0 ? result : -EFAULT;
- }
+ runtime->buffer + appl_ptr, count1))
+ err = -EFAULT;
spin_lock_irqsave(&runtime->lock, flags);
+ if (err)
+ goto out;
}
result += count1;
count -= count1;
}
+ out:
+ snd_rawmidi_buffer_unref(runtime);
spin_unlock_irqrestore(&runtime->lock, flags);
- return result;
+ return result > 0 ? result : err;
}
long snd_rawmidi_kernel_read(struct snd_rawmidi_substream *substream,
@@ -1003,7 +1034,7 @@
result = 0;
while (count > 0) {
spin_lock_irq(&runtime->lock);
- while (!snd_rawmidi_ready(substream)) {
+ while (!__snd_rawmidi_ready(runtime)) {
wait_queue_entry_t wait;
if ((file->f_flags & O_NONBLOCK) != 0 || result > 0) {
@@ -1020,9 +1051,11 @@
return -ENODEV;
if (signal_pending(current))
return result > 0 ? result : -ERESTARTSYS;
- if (!runtime->avail)
- return result > 0 ? result : -EIO;
spin_lock_irq(&runtime->lock);
+ if (!runtime->avail) {
+ spin_unlock_irq(&runtime->lock);
+ return result > 0 ? result : -EIO;
+ }
}
spin_unlock_irq(&runtime->lock);
count1 = snd_rawmidi_kernel_read1(substream,
@@ -1160,7 +1193,7 @@
runtime->avail += count;
substream->bytes += count;
if (count > 0) {
- if (runtime->drain || snd_rawmidi_ready(substream))
+ if (runtime->drain || __snd_rawmidi_ready(runtime))
wake_up(&runtime->sleep);
}
return count;
@@ -1268,6 +1301,7 @@
return -EAGAIN;
}
}
+ snd_rawmidi_buffer_ref(runtime);
while (count > 0 && runtime->avail > 0) {
count1 = runtime->buffer_size - runtime->appl_ptr;
if (count1 > count)
@@ -1299,6 +1333,7 @@
}
__end:
count1 = runtime->avail < runtime->buffer_size;
+ snd_rawmidi_buffer_unref(runtime);
spin_unlock_irqrestore(&runtime->lock, flags);
if (count1)
snd_rawmidi_output_trigger(substream, 1);
@@ -1347,9 +1382,11 @@
return -ENODEV;
if (signal_pending(current))
return result > 0 ? result : -ERESTARTSYS;
- if (!runtime->avail && !timeout)
- return result > 0 ? result : -EIO;
spin_lock_irq(&runtime->lock);
+ if (!runtime->avail && !timeout) {
+ spin_unlock_irq(&runtime->lock);
+ return result > 0 ? result : -EIO;
+ }
}
spin_unlock_irq(&runtime->lock);
count1 = snd_rawmidi_kernel_write1(substream, buf, NULL, count);
@@ -1429,6 +1466,7 @@
struct snd_rawmidi *rmidi;
struct snd_rawmidi_substream *substream;
struct snd_rawmidi_runtime *runtime;
+ unsigned long buffer_size, avail, xruns;
rmidi = entry->private_data;
snd_iprintf(buffer, "%s\n\n", rmidi->name);
@@ -1447,13 +1485,16 @@
" Owner PID : %d\n",
pid_vnr(substream->pid));
runtime = substream->runtime;
+ spin_lock_irq(&runtime->lock);
+ buffer_size = runtime->buffer_size;
+ avail = runtime->avail;
+ spin_unlock_irq(&runtime->lock);
snd_iprintf(buffer,
" Mode : %s\n"
" Buffer size : %lu\n"
" Avail : %lu\n",
runtime->oss ? "OSS compatible" : "native",
- (unsigned long) runtime->buffer_size,
- (unsigned long) runtime->avail);
+ buffer_size, avail);
}
}
}
@@ -1471,13 +1512,16 @@
" Owner PID : %d\n",
pid_vnr(substream->pid));
runtime = substream->runtime;
+ spin_lock_irq(&runtime->lock);
+ buffer_size = runtime->buffer_size;
+ avail = runtime->avail;
+ xruns = runtime->xruns;
+ spin_unlock_irq(&runtime->lock);
snd_iprintf(buffer,
" Buffer size : %lu\n"
" Avail : %lu\n"
" Overruns : %lu\n",
- (unsigned long) runtime->buffer_size,
- (unsigned long) runtime->avail,
- (unsigned long) runtime->xruns);
+ buffer_size, avail, xruns);
}
}
}
diff --git a/sound/core/seq/oss/seq_oss.c b/sound/core/seq/oss/seq_oss.c
index 17f9136..250a92b 100644
--- a/sound/core/seq/oss/seq_oss.c
+++ b/sound/core/seq/oss/seq_oss.c
@@ -168,10 +168,19 @@
odev_ioctl(struct file *file, unsigned int cmd, unsigned long arg)
{
struct seq_oss_devinfo *dp;
+ long rc;
+
dp = file->private_data;
if (snd_BUG_ON(!dp))
return -ENXIO;
- return snd_seq_oss_ioctl(dp, cmd, arg);
+
+ if (cmd != SNDCTL_SEQ_SYNC &&
+ mutex_lock_interruptible(®ister_mutex))
+ return -ERESTARTSYS;
+ rc = snd_seq_oss_ioctl(dp, cmd, arg);
+ if (cmd != SNDCTL_SEQ_SYNC)
+ mutex_unlock(®ister_mutex);
+ return rc;
}
#ifdef CONFIG_COMPAT
diff --git a/sound/core/seq/oss/seq_oss_midi.c b/sound/core/seq/oss/seq_oss_midi.c
index a88c235..2ddfe22 100644
--- a/sound/core/seq/oss/seq_oss_midi.c
+++ b/sound/core/seq/oss/seq_oss_midi.c
@@ -602,6 +602,7 @@
len = snd_seq_oss_timer_start(dp->timer);
if (ev->type == SNDRV_SEQ_EVENT_SYSEX) {
snd_seq_oss_readq_sysex(dp->readq, mdev->seq_device, ev);
+ snd_midi_event_reset_decode(mdev->coder);
} else {
len = snd_midi_event_decode(mdev->coder, msg, sizeof(msg), ev);
if (len > 0)
diff --git a/sound/core/seq/oss/seq_oss_synth.c b/sound/core/seq/oss/seq_oss_synth.c
index 11554d0..1b8409e 100644
--- a/sound/core/seq/oss/seq_oss_synth.c
+++ b/sound/core/seq/oss/seq_oss_synth.c
@@ -611,7 +611,8 @@
if (info->is_midi) {
struct midi_info minf;
- snd_seq_oss_midi_make_info(dp, info->midi_mapped, &minf);
+ if (snd_seq_oss_midi_make_info(dp, info->midi_mapped, &minf))
+ return -ENXIO;
inf->synth_type = SYNTH_TYPE_MIDI;
inf->synth_subtype = 0;
inf->nr_voices = 16;
diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c
index 6d9592f..cc93157 100644
--- a/sound/core/seq/seq_clientmgr.c
+++ b/sound/core/seq/seq_clientmgr.c
@@ -580,7 +580,7 @@
event->queue = queue;
event->flags &= ~SNDRV_SEQ_TIME_STAMP_MASK;
if (real_time) {
- event->time.time = snd_seq_timer_get_cur_time(q->timer);
+ event->time.time = snd_seq_timer_get_cur_time(q->timer, true);
event->flags |= SNDRV_SEQ_TIME_STAMP_REAL;
} else {
event->time.tick = snd_seq_timer_get_cur_tick(q->timer);
@@ -1659,7 +1659,7 @@
tmr = queue->timer;
status->events = queue->tickq->cells + queue->timeq->cells;
- status->time = snd_seq_timer_get_cur_time(tmr);
+ status->time = snd_seq_timer_get_cur_time(tmr, true);
status->tick = snd_seq_timer_get_cur_tick(tmr);
status->running = tmr->running;
diff --git a/sound/core/seq/seq_ports.c b/sound/core/seq/seq_ports.c
index 83be6b9..97e8eb3 100644
--- a/sound/core/seq/seq_ports.c
+++ b/sound/core/seq/seq_ports.c
@@ -514,10 +514,11 @@
return err;
}
-static void delete_and_unsubscribe_port(struct snd_seq_client *client,
- struct snd_seq_client_port *port,
- struct snd_seq_subscribers *subs,
- bool is_src, bool ack)
+/* called with grp->list_mutex held */
+static void __delete_and_unsubscribe_port(struct snd_seq_client *client,
+ struct snd_seq_client_port *port,
+ struct snd_seq_subscribers *subs,
+ bool is_src, bool ack)
{
struct snd_seq_port_subs_info *grp;
struct list_head *list;
@@ -525,7 +526,6 @@
grp = is_src ? &port->c_src : &port->c_dest;
list = is_src ? &subs->src_list : &subs->dest_list;
- down_write(&grp->list_mutex);
write_lock_irq(&grp->list_lock);
empty = list_empty(list);
if (!empty)
@@ -535,6 +535,18 @@
if (!empty)
unsubscribe_port(client, port, grp, &subs->info, ack);
+}
+
+static void delete_and_unsubscribe_port(struct snd_seq_client *client,
+ struct snd_seq_client_port *port,
+ struct snd_seq_subscribers *subs,
+ bool is_src, bool ack)
+{
+ struct snd_seq_port_subs_info *grp;
+
+ grp = is_src ? &port->c_src : &port->c_dest;
+ down_write(&grp->list_mutex);
+ __delete_and_unsubscribe_port(client, port, subs, is_src, ack);
up_write(&grp->list_mutex);
}
@@ -590,27 +602,30 @@
struct snd_seq_client_port *dest_port,
struct snd_seq_port_subscribe *info)
{
- struct snd_seq_port_subs_info *src = &src_port->c_src;
+ struct snd_seq_port_subs_info *dest = &dest_port->c_dest;
struct snd_seq_subscribers *subs;
int err = -ENOENT;
- down_write(&src->list_mutex);
+ /* always start from deleting the dest port for avoiding concurrent
+ * deletions
+ */
+ down_write(&dest->list_mutex);
/* look for the connection */
- list_for_each_entry(subs, &src->list_head, src_list) {
+ list_for_each_entry(subs, &dest->list_head, dest_list) {
if (match_subs_info(info, &subs->info)) {
- atomic_dec(&subs->ref_count); /* mark as not ready */
+ __delete_and_unsubscribe_port(dest_client, dest_port,
+ subs, false,
+ connector->number != dest_client->number);
err = 0;
break;
}
}
- up_write(&src->list_mutex);
+ up_write(&dest->list_mutex);
if (err < 0)
return err;
delete_and_unsubscribe_port(src_client, src_port, subs, true,
connector->number != src_client->number);
- delete_and_unsubscribe_port(dest_client, dest_port, subs, false,
- connector->number != dest_client->number);
kfree(subs);
return 0;
}
diff --git a/sound/core/seq/seq_queue.c b/sound/core/seq/seq_queue.c
index caf68bf..71a6ea6 100644
--- a/sound/core/seq/seq_queue.c
+++ b/sound/core/seq/seq_queue.c
@@ -238,6 +238,8 @@
{
unsigned long flags;
struct snd_seq_event_cell *cell;
+ snd_seq_tick_time_t cur_tick;
+ snd_seq_real_time_t cur_time;
if (q == NULL)
return;
@@ -254,17 +256,18 @@
__again:
/* Process tick queue... */
+ cur_tick = snd_seq_timer_get_cur_tick(q->timer);
for (;;) {
- cell = snd_seq_prioq_cell_out(q->tickq,
- &q->timer->tick.cur_tick);
+ cell = snd_seq_prioq_cell_out(q->tickq, &cur_tick);
if (!cell)
break;
snd_seq_dispatch_event(cell, atomic, hop);
}
/* Process time queue... */
+ cur_time = snd_seq_timer_get_cur_time(q->timer, false);
for (;;) {
- cell = snd_seq_prioq_cell_out(q->timeq, &q->timer->cur_time);
+ cell = snd_seq_prioq_cell_out(q->timeq, &cur_time);
if (!cell)
break;
snd_seq_dispatch_event(cell, atomic, hop);
@@ -392,6 +395,7 @@
int snd_seq_queue_set_owner(int queueid, int client, int locked)
{
struct snd_seq_queue *q = queueptr(queueid);
+ unsigned long flags;
if (q == NULL)
return -EINVAL;
@@ -401,8 +405,10 @@
return -EPERM;
}
+ spin_lock_irqsave(&q->owner_lock, flags);
q->locked = locked ? 1 : 0;
q->owner = client;
+ spin_unlock_irqrestore(&q->owner_lock, flags);
queue_access_unlock(q);
queuefree(q);
@@ -539,15 +545,17 @@
unsigned long flags;
int i;
struct snd_seq_queue *q;
+ bool matched;
for (i = 0; i < SNDRV_SEQ_MAX_QUEUES; i++) {
if ((q = queueptr(i)) == NULL)
continue;
spin_lock_irqsave(&q->owner_lock, flags);
- if (q->owner == client)
+ matched = (q->owner == client);
+ if (matched)
q->klocked = 1;
spin_unlock_irqrestore(&q->owner_lock, flags);
- if (q->owner == client) {
+ if (matched) {
if (q->timer->running)
snd_seq_timer_stop(q->timer);
snd_seq_timer_reset(q->timer);
@@ -739,6 +747,8 @@
int i, bpm;
struct snd_seq_queue *q;
struct snd_seq_timer *tmr;
+ bool locked;
+ int owner;
for (i = 0; i < SNDRV_SEQ_MAX_QUEUES; i++) {
if ((q = queueptr(i)) == NULL)
@@ -750,9 +760,14 @@
else
bpm = 0;
+ spin_lock_irq(&q->owner_lock);
+ locked = q->locked;
+ owner = q->owner;
+ spin_unlock_irq(&q->owner_lock);
+
snd_iprintf(buffer, "queue %d: [%s]\n", q->queue, q->name);
- snd_iprintf(buffer, "owned by client : %d\n", q->owner);
- snd_iprintf(buffer, "lock status : %s\n", q->locked ? "Locked" : "Free");
+ snd_iprintf(buffer, "owned by client : %d\n", owner);
+ snd_iprintf(buffer, "lock status : %s\n", locked ? "Locked" : "Free");
snd_iprintf(buffer, "queued time events : %d\n", snd_seq_prioq_avail(q->timeq));
snd_iprintf(buffer, "queued tick events : %d\n", snd_seq_prioq_avail(q->tickq));
snd_iprintf(buffer, "timer state : %s\n", tmr->running ? "Running" : "Stopped");
diff --git a/sound/core/seq/seq_queue.h b/sound/core/seq/seq_queue.h
index 9254c8d..25d2d6b 100644
--- a/sound/core/seq/seq_queue.h
+++ b/sound/core/seq/seq_queue.h
@@ -26,10 +26,10 @@
struct snd_seq_timer *timer; /* time keeper for this queue */
int owner; /* client that 'owns' the timer */
- unsigned int locked:1, /* timer is only accesibble by owner if set */
- klocked:1, /* kernel lock (after START) */
- check_again:1,
- check_blocked:1;
+ bool locked; /* timer is only accesibble by owner if set */
+ bool klocked; /* kernel lock (after START) */
+ bool check_again; /* concurrent access happened during check */
+ bool check_blocked; /* queue being checked */
unsigned int flags; /* status flags */
unsigned int info_flags; /* info for sync */
diff --git a/sound/core/seq/seq_timer.c b/sound/core/seq/seq_timer.c
index 161f317..0b43fc5 100644
--- a/sound/core/seq/seq_timer.c
+++ b/sound/core/seq/seq_timer.c
@@ -422,14 +422,15 @@
}
/* return current 'real' time. use timeofday() to get better granularity. */
-snd_seq_real_time_t snd_seq_timer_get_cur_time(struct snd_seq_timer *tmr)
+snd_seq_real_time_t snd_seq_timer_get_cur_time(struct snd_seq_timer *tmr,
+ bool adjust_ktime)
{
snd_seq_real_time_t cur_time;
unsigned long flags;
spin_lock_irqsave(&tmr->lock, flags);
cur_time = tmr->cur_time;
- if (tmr->running) {
+ if (adjust_ktime && tmr->running) {
struct timespec64 tm;
ktime_get_ts64(&tm);
@@ -446,7 +447,13 @@
high PPQ values) */
snd_seq_tick_time_t snd_seq_timer_get_cur_tick(struct snd_seq_timer *tmr)
{
- return tmr->tick.cur_tick;
+ snd_seq_tick_time_t cur_tick;
+ unsigned long flags;
+
+ spin_lock_irqsave(&tmr->lock, flags);
+ cur_tick = tmr->tick.cur_tick;
+ spin_unlock_irqrestore(&tmr->lock, flags);
+ return cur_tick;
}
@@ -465,15 +472,19 @@
q = queueptr(idx);
if (q == NULL)
continue;
- if ((tmr = q->timer) == NULL ||
- (ti = tmr->timeri) == NULL) {
- queuefree(q);
- continue;
- }
+ mutex_lock(&q->timer_mutex);
+ tmr = q->timer;
+ if (!tmr)
+ goto unlock;
+ ti = tmr->timeri;
+ if (!ti)
+ goto unlock;
snd_iprintf(buffer, "Timer for queue %i : %s\n", q->queue, ti->timer->name);
resolution = snd_timer_resolution(ti) * tmr->ticks;
snd_iprintf(buffer, " Period time : %lu.%09lu\n", resolution / 1000000000, resolution % 1000000000);
snd_iprintf(buffer, " Skew : %u / %u\n", tmr->skew, tmr->skew_base);
+unlock:
+ mutex_unlock(&q->timer_mutex);
queuefree(q);
}
}
diff --git a/sound/core/seq/seq_timer.h b/sound/core/seq/seq_timer.h
index 66c3e34..4bec57d 100644
--- a/sound/core/seq/seq_timer.h
+++ b/sound/core/seq/seq_timer.h
@@ -120,7 +120,8 @@
int snd_seq_timer_set_position_tick(struct snd_seq_timer *tmr, snd_seq_tick_time_t position);
int snd_seq_timer_set_position_time(struct snd_seq_timer *tmr, snd_seq_real_time_t position);
int snd_seq_timer_set_skew(struct snd_seq_timer *tmr, unsigned int skew, unsigned int base);
-snd_seq_real_time_t snd_seq_timer_get_cur_time(struct snd_seq_timer *tmr);
+snd_seq_real_time_t snd_seq_timer_get_cur_time(struct snd_seq_timer *tmr,
+ bool adjust_ktime);
snd_seq_tick_time_t snd_seq_timer_get_cur_tick(struct snd_seq_timer *tmr);
extern int seq_default_timer_class;
diff --git a/sound/core/seq/seq_virmidi.c b/sound/core/seq/seq_virmidi.c
index 626d87c..77d7037 100644
--- a/sound/core/seq/seq_virmidi.c
+++ b/sound/core/seq/seq_virmidi.c
@@ -81,6 +81,7 @@
if ((ev->flags & SNDRV_SEQ_EVENT_LENGTH_MASK) != SNDRV_SEQ_EVENT_LENGTH_VARIABLE)
continue;
snd_seq_dump_var_event(ev, (snd_seq_dump_func_t)snd_rawmidi_receive, vmidi->substream);
+ snd_midi_event_reset_decode(vmidi->parser);
} else {
len = snd_midi_event_decode(vmidi->parser, msg, sizeof(msg), ev);
if (len > 0)
diff --git a/sound/core/timer.c b/sound/core/timer.c
index 59ae21b..b5a0ba7 100644
--- a/sound/core/timer.c
+++ b/sound/core/timer.c
@@ -74,6 +74,9 @@
/* lock for slave active lists */
static DEFINE_SPINLOCK(slave_active_lock);
+#define MAX_SLAVE_INSTANCES 1000
+static int num_slaves;
+
static DEFINE_MUTEX(register_mutex);
static int snd_timer_free(struct snd_timer *timer);
@@ -252,6 +255,10 @@
err = -EINVAL;
goto unlock;
}
+ if (num_slaves >= MAX_SLAVE_INSTANCES) {
+ err = -EBUSY;
+ goto unlock;
+ }
timeri = snd_timer_instance_new(owner, NULL);
if (!timeri) {
err = -ENOMEM;
@@ -261,6 +268,7 @@
timeri->slave_id = tid->device;
timeri->flags |= SNDRV_TIMER_IFLG_SLAVE;
list_add_tail(&timeri->open_list, &snd_timer_slave_list);
+ num_slaves++;
err = snd_timer_check_slave(timeri);
if (err < 0) {
snd_timer_close_locked(timeri, &card_dev_to_put);
@@ -356,6 +364,8 @@
}
list_del(&timeri->open_list);
+ if (timeri->flags & SNDRV_TIMER_IFLG_SLAVE)
+ num_slaves--;
/* force to stop the timer */
snd_timer_stop(timeri);
@@ -481,9 +491,10 @@
return;
if (timer->hw.flags & SNDRV_TIMER_HW_SLAVE)
return;
+ event += 10; /* convert to SNDRV_TIMER_EVENT_MXXX */
list_for_each_entry(ts, &ti->slave_active_head, active_list)
if (ts->ccallback)
- ts->ccallback(ts, event + 100, &tstamp, resolution);
+ ts->ccallback(ts, event, &tstamp, resolution);
}
/* start/continue a master timer */
diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c
index 9ccdad8..452b9ea 100644
--- a/sound/drivers/aloop.c
+++ b/sound/drivers/aloop.c
@@ -1035,6 +1035,14 @@
return -ENOMEM;
kctl->id.device = dev;
kctl->id.subdevice = substr;
+
+ /* Add the control before copying the id so that
+ * the numid field of the id is set in the copy.
+ */
+ err = snd_ctl_add(card, kctl);
+ if (err < 0)
+ return err;
+
switch (idx) {
case ACTIVE_IDX:
setup->active_id = kctl->id;
@@ -1051,9 +1059,6 @@
default:
break;
}
- err = snd_ctl_add(card, kctl);
- if (err < 0)
- return err;
}
}
}
diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c
index aee7c04..b61ba03 100644
--- a/sound/drivers/dummy.c
+++ b/sound/drivers/dummy.c
@@ -915,7 +915,7 @@
{
int i;
- for (i = 0; i < SNDRV_PCM_FORMAT_LAST; i++) {
+ for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) {
if (dummy->pcm_hw.formats & (1ULL << i))
snd_iprintf(buffer, " %s", snd_pcm_format_name(i));
}
diff --git a/sound/drivers/opl3/opl3_synth.c b/sound/drivers/opl3/opl3_synth.c
index e69a4ef..08c10ac 100644
--- a/sound/drivers/opl3/opl3_synth.c
+++ b/sound/drivers/opl3/opl3_synth.c
@@ -91,6 +91,8 @@
{
struct snd_dm_fm_info info;
+ memset(&info, 0, sizeof(info));
+
info.fm_mode = opl3->fm_mode;
info.rhythm = opl3->rhythm;
if (copy_to_user(argp, &info, sizeof(struct snd_dm_fm_info)))
diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig
index b0a904c..e469375 100644
--- a/sound/firewire/Kconfig
+++ b/sound/firewire/Kconfig
@@ -38,7 +38,7 @@
* Mackie(Loud) Onyx 1640i (former model)
* Mackie(Loud) Onyx Satellite
* Mackie(Loud) Tapco Link.Firewire
- * Mackie(Loud) d.2 pro/d.4 pro
+ * Mackie(Loud) d.2 pro/d.4 pro (built-in FireWire card with OXFW971 ASIC)
* Mackie(Loud) U.420/U.420d
* TASCAM FireOne
* Stanton Controllers & Systems 1 Deck/Mixer
@@ -84,7 +84,7 @@
* PreSonus FIREBOX/FIREPOD/FP10/Inspire1394
* BridgeCo RDAudio1/Audio5
* Mackie Onyx 1220/1620/1640 (FireWire I/O Card)
- * Mackie d.2 (FireWire Option)
+ * Mackie d.2 (optional FireWire card with DM1000 ASIC)
* Stanton FinalScratch 2 (ScratchAmp)
* Tascam IF-FW/DM
* Behringer XENIX UFX 1204/1604
@@ -110,6 +110,7 @@
* M-Audio Ozonic/NRV10/ProfireLightBridge
* M-Audio FireWire 1814/ProjectMix IO
* Digidesign Mbox 2 Pro
+ * ToneWeal FW66
To compile this driver as a module, choose M here: the module
will be called snd-bebob.
diff --git a/sound/firewire/amdtp-am824.c b/sound/firewire/amdtp-am824.c
index 67d735e..fea92e1 100644
--- a/sound/firewire/amdtp-am824.c
+++ b/sound/firewire/amdtp-am824.c
@@ -82,7 +82,8 @@
if (err < 0)
return err;
- s->ctx_data.rx.fdf = AMDTP_FDF_AM824 | s->sfc;
+ if (s->direction == AMDTP_OUT_STREAM)
+ s->ctx_data.rx.fdf = AMDTP_FDF_AM824 | s->sfc;
p->pcm_channels = pcm_channels;
p->midi_ports = midi_ports;
diff --git a/sound/firewire/amdtp-stream-trace.h b/sound/firewire/amdtp-stream-trace.h
index 16c7f66..26e7cb5 100644
--- a/sound/firewire/amdtp-stream-trace.h
+++ b/sound/firewire/amdtp-stream-trace.h
@@ -66,8 +66,7 @@
__entry->irq,
__entry->index,
__print_array(__get_dynamic_array(cip_header),
- __get_dynamic_array_len(cip_header),
- sizeof(u8)))
+ __get_dynamic_array_len(cip_header), 1))
);
#endif
diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c
index e50e28f..07e58dc 100644
--- a/sound/firewire/amdtp-stream.c
+++ b/sound/firewire/amdtp-stream.c
@@ -617,18 +617,24 @@
unsigned int *syt, unsigned int index)
{
const __be32 *cip_header;
+ unsigned int cip_header_size;
int err;
*payload_length = be32_to_cpu(ctx_header[0]) >> ISO_DATA_LENGTH_SHIFT;
- if (*payload_length > s->ctx_data.tx.ctx_header_size +
- s->ctx_data.tx.max_ctx_payload_length) {
+
+ if (!(s->flags & CIP_NO_HEADER))
+ cip_header_size = 8;
+ else
+ cip_header_size = 0;
+
+ if (*payload_length > cip_header_size + s->ctx_data.tx.max_ctx_payload_length) {
dev_err(&s->unit->device,
"Detect jumbo payload: %04x %04x\n",
- *payload_length, s->ctx_data.tx.max_ctx_payload_length);
+ *payload_length, cip_header_size + s->ctx_data.tx.max_ctx_payload_length);
return -EIO;
}
- if (!(s->flags & CIP_NO_HEADER)) {
+ if (cip_header_size > 0) {
cip_header = ctx_header + 2;
err = check_cip_header(s, cip_header, *payload_length,
data_blocks, data_block_counter, syt);
@@ -932,23 +938,22 @@
s->ctx_data.rx.last_syt_offset = TICKS_PER_CYCLE;
}
- /* initialize packet buffer */
+ // initialize packet buffer.
+ max_ctx_payload_size = amdtp_stream_get_max_payload(s);
if (s->direction == AMDTP_IN_STREAM) {
dir = DMA_FROM_DEVICE;
type = FW_ISO_CONTEXT_RECEIVE;
- if (!(s->flags & CIP_NO_HEADER))
+ if (!(s->flags & CIP_NO_HEADER)) {
+ max_ctx_payload_size -= 8;
ctx_header_size = IR_CTX_HEADER_SIZE_CIP;
- else
+ } else {
ctx_header_size = IR_CTX_HEADER_SIZE_NO_CIP;
-
- max_ctx_payload_size = amdtp_stream_get_max_payload(s) -
- ctx_header_size;
+ }
} else {
dir = DMA_TO_DEVICE;
type = FW_ISO_CONTEXT_TRANSMIT;
ctx_header_size = 0; // No effect for IT context.
- max_ctx_payload_size = amdtp_stream_get_max_payload(s);
if (!(s->flags & CIP_NO_HEADER))
max_ctx_payload_size -= IT_PKT_HEADER_SIZE_CIP;
}
diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c
index 976d8cb..d58f4fe 100644
--- a/sound/firewire/bebob/bebob.c
+++ b/sound/firewire/bebob/bebob.c
@@ -59,6 +59,7 @@
#define VEN_MAUDIO1 0x00000d6c
#define VEN_MAUDIO2 0x000007f5
#define VEN_DIGIDESIGN 0x00a07e
+#define OUI_SHOUYO 0x002327
#define MODEL_FOCUSRITE_SAFFIRE_BOTH 0x00000000
#define MODEL_MAUDIO_AUDIOPHILE_BOTH 0x00010060
@@ -387,7 +388,7 @@
SND_BEBOB_DEV_ENTRY(VEN_BRIDGECO, 0x00010049, &spec_normal),
/* Mackie, Onyx 1220/1620/1640 (Firewire I/O Card) */
SND_BEBOB_DEV_ENTRY(VEN_MACKIE2, 0x00010065, &spec_normal),
- /* Mackie, d.2 (Firewire Option) */
+ // Mackie, d.2 (optional Firewire card with DM1000).
SND_BEBOB_DEV_ENTRY(VEN_MACKIE1, 0x00010067, &spec_normal),
/* Stanton, ScratchAmp */
SND_BEBOB_DEV_ENTRY(VEN_STANTON, 0x00000001, &spec_normal),
@@ -486,6 +487,8 @@
&maudio_special_spec),
/* Digidesign Mbox 2 Pro */
SND_BEBOB_DEV_ENTRY(VEN_DIGIDESIGN, 0x0000a9, &spec_normal),
+ // Toneweal FW66.
+ SND_BEBOB_DEV_ENTRY(OUI_SHOUYO, 0x020002, &spec_normal),
/* IDs are unknown but able to be supported */
/* Apogee, Mini-ME Firewire */
/* Apogee, Mini-DAC Firewire */
diff --git a/sound/firewire/bebob/bebob_hwdep.c b/sound/firewire/bebob/bebob_hwdep.c
index 45b740f..c362eb3 100644
--- a/sound/firewire/bebob/bebob_hwdep.c
+++ b/sound/firewire/bebob/bebob_hwdep.c
@@ -36,12 +36,11 @@
}
memset(&event, 0, sizeof(event));
+ count = min_t(long, count, sizeof(event.lock_status));
if (bebob->dev_lock_changed) {
event.lock_status.type = SNDRV_FIREWIRE_EVENT_LOCK_STATUS;
event.lock_status.status = (bebob->dev_lock_count > 0);
bebob->dev_lock_changed = false;
-
- count = min_t(long, count, sizeof(event.lock_status));
}
spin_unlock_irq(&bebob->lock);
diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c
index 6c1497d..3935e90 100644
--- a/sound/firewire/bebob/bebob_stream.c
+++ b/sound/firewire/bebob/bebob_stream.c
@@ -415,15 +415,16 @@
return 0;
}
-static void
-break_both_connections(struct snd_bebob *bebob)
+static void break_both_connections(struct snd_bebob *bebob)
{
cmp_connection_break(&bebob->in_conn);
cmp_connection_break(&bebob->out_conn);
- /* These models seems to be in transition state for a longer time. */
- if (bebob->maudio_special_quirk != NULL)
- msleep(200);
+ // These models seem to be in transition state for a longer time. When
+ // accessing in the state, any transactions is corrupted. In the worst
+ // case, the device is going to reboot.
+ if (bebob->version < 2)
+ msleep(600);
}
static int
@@ -533,20 +534,22 @@
static int keep_resources(struct snd_bebob *bebob, struct amdtp_stream *stream,
unsigned int rate, unsigned int index)
{
- struct snd_bebob_stream_formation *formation;
+ unsigned int pcm_channels;
+ unsigned int midi_ports;
struct cmp_connection *conn;
int err;
if (stream == &bebob->tx_stream) {
- formation = bebob->tx_stream_formations + index;
+ pcm_channels = bebob->tx_stream_formations[index].pcm;
+ midi_ports = bebob->midi_input_ports;
conn = &bebob->out_conn;
} else {
- formation = bebob->rx_stream_formations + index;
+ pcm_channels = bebob->rx_stream_formations[index].pcm;
+ midi_ports = bebob->midi_output_ports;
conn = &bebob->in_conn;
}
- err = amdtp_am824_set_parameters(stream, rate, formation->pcm,
- formation->midi, false);
+ err = amdtp_am824_set_parameters(stream, rate, pcm_channels, midi_ports, false);
if (err < 0)
return err;
diff --git a/sound/firewire/dice/dice-alesis.c b/sound/firewire/dice/dice-alesis.c
index f5b3252..39a4ef8 100644
--- a/sound/firewire/dice/dice-alesis.c
+++ b/sound/firewire/dice/dice-alesis.c
@@ -16,7 +16,7 @@
static const unsigned int
alesis_io26_tx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT] = {
{10, 10, 4}, /* Tx0 = Analog + S/PDIF. */
- {16, 8, 0}, /* Tx1 = ADAT1 + ADAT2. */
+ {16, 4, 0}, /* Tx1 = ADAT1 + ADAT2 (available at low rate). */
};
int snd_dice_detect_alesis_formats(struct snd_dice *dice)
diff --git a/sound/firewire/dice/dice-extension.c b/sound/firewire/dice/dice-extension.c
index a63fcbc..02f4a83 100644
--- a/sound/firewire/dice/dice-extension.c
+++ b/sound/firewire/dice/dice-extension.c
@@ -159,8 +159,11 @@
int j;
for (j = i + 1; j < 9; ++j) {
- if (pointers[i * 2] == pointers[j * 2])
+ if (pointers[i * 2] == pointers[j * 2]) {
+ // Fallback to limited functionality.
+ err = -ENXIO;
goto end;
+ }
}
}
diff --git a/sound/firewire/dice/dice-stream.c b/sound/firewire/dice/dice-stream.c
index f6a8627..0aa3c56 100644
--- a/sound/firewire/dice/dice-stream.c
+++ b/sound/firewire/dice/dice-stream.c
@@ -489,11 +489,10 @@
struct reg_params tx_params, rx_params;
if (dice->substreams_counter == 0) {
- if (get_register_params(dice, &tx_params, &rx_params) >= 0) {
- amdtp_domain_stop(&dice->domain);
+ if (get_register_params(dice, &tx_params, &rx_params) >= 0)
finish_session(dice, &tx_params, &rx_params);
- }
+ amdtp_domain_stop(&dice->domain);
release_resources(dice);
}
}
diff --git a/sound/firewire/dice/dice-tcelectronic.c b/sound/firewire/dice/dice-tcelectronic.c
index a8875d2..43a3bcb 100644
--- a/sound/firewire/dice/dice-tcelectronic.c
+++ b/sound/firewire/dice/dice-tcelectronic.c
@@ -38,8 +38,8 @@
};
static const struct dice_tc_spec konnekt_live = {
- .tx_pcm_chs = {{16, 16, 16}, {0, 0, 0} },
- .rx_pcm_chs = {{16, 16, 16}, {0, 0, 0} },
+ .tx_pcm_chs = {{16, 16, 6}, {0, 0, 0} },
+ .rx_pcm_chs = {{16, 16, 6}, {0, 0, 0} },
.has_midi = true,
};
diff --git a/sound/firewire/digi00x/digi00x.c b/sound/firewire/digi00x/digi00x.c
index 1f5fc0e..0e4b0ea 100644
--- a/sound/firewire/digi00x/digi00x.c
+++ b/sound/firewire/digi00x/digi00x.c
@@ -14,6 +14,7 @@
#define VENDOR_DIGIDESIGN 0x00a07e
#define MODEL_CONSOLE 0x000001
#define MODEL_RACK 0x000002
+#define SPEC_VERSION 0x000001
static int name_card(struct snd_dg00x *dg00x)
{
@@ -175,14 +176,18 @@
/* Both of 002/003 use the same ID. */
{
.match_flags = IEEE1394_MATCH_VENDOR_ID |
+ IEEE1394_MATCH_VERSION |
IEEE1394_MATCH_MODEL_ID,
.vendor_id = VENDOR_DIGIDESIGN,
+ .version = SPEC_VERSION,
.model_id = MODEL_CONSOLE,
},
{
.match_flags = IEEE1394_MATCH_VENDOR_ID |
+ IEEE1394_MATCH_VERSION |
IEEE1394_MATCH_MODEL_ID,
.vendor_id = VENDOR_DIGIDESIGN,
+ .version = SPEC_VERSION,
.model_id = MODEL_RACK,
},
{}
diff --git a/sound/firewire/fireface/ff-pcm.c b/sound/firewire/fireface/ff-pcm.c
index 9eab3ad..df6ff2d 100644
--- a/sound/firewire/fireface/ff-pcm.c
+++ b/sound/firewire/fireface/ff-pcm.c
@@ -219,7 +219,7 @@
mutex_unlock(&ff->mutex);
}
- return 0;
+ return err;
}
static int pcm_hw_free(struct snd_pcm_substream *substream)
diff --git a/sound/firewire/fireface/ff-protocol-latter.c b/sound/firewire/fireface/ff-protocol-latter.c
index 0e4c3a9..76ae568 100644
--- a/sound/firewire/fireface/ff-protocol-latter.c
+++ b/sound/firewire/fireface/ff-protocol-latter.c
@@ -107,18 +107,18 @@
int err;
// Set the number of data blocks transferred in a second.
- if (rate % 32000 == 0)
- code = 0x00;
+ if (rate % 48000 == 0)
+ code = 0x04;
else if (rate % 44100 == 0)
code = 0x02;
- else if (rate % 48000 == 0)
- code = 0x04;
+ else if (rate % 32000 == 0)
+ code = 0x00;
else
return -EINVAL;
if (rate >= 64000 && rate < 128000)
code |= 0x08;
- else if (rate >= 128000 && rate < 192000)
+ else if (rate >= 128000)
code |= 0x10;
reg = cpu_to_le32(code);
@@ -140,7 +140,7 @@
if (curr_rate == rate)
break;
}
- if (count == 10)
+ if (count > 10)
return -ETIMEDOUT;
for (i = 0; i < ARRAY_SIZE(amdtp_rate_table); ++i) {
diff --git a/sound/firewire/fireface/ff-transaction.c b/sound/firewire/fireface/ff-transaction.c
index 7f82762..ee7122c 100644
--- a/sound/firewire/fireface/ff-transaction.c
+++ b/sound/firewire/fireface/ff-transaction.c
@@ -88,7 +88,7 @@
/* Set interval to next transaction. */
ff->next_ktime[port] = ktime_add_ns(ktime_get(),
- ff->rx_bytes[port] * 8 * NSEC_PER_SEC / 31250);
+ ff->rx_bytes[port] * 8 * (NSEC_PER_SEC / 31250));
if (quad_count == 1)
tcode = TCODE_WRITE_QUADLET_REQUEST;
diff --git a/sound/firewire/fireworks/fireworks_transaction.c b/sound/firewire/fireworks/fireworks_transaction.c
index 0f533f5..9f8c53b 100644
--- a/sound/firewire/fireworks/fireworks_transaction.c
+++ b/sound/firewire/fireworks/fireworks_transaction.c
@@ -123,7 +123,7 @@
t = (struct snd_efw_transaction *)data;
length = min_t(size_t, be32_to_cpu(t->length) * sizeof(u32), length);
- spin_lock_irq(&efw->lock);
+ spin_lock(&efw->lock);
if (efw->push_ptr < efw->pull_ptr)
capacity = (unsigned int)(efw->pull_ptr - efw->push_ptr);
@@ -190,7 +190,7 @@
copy_resp_to_buf(efw, data, length, rcode);
end:
- spin_unlock_irq(&instances_lock);
+ spin_unlock(&instances_lock);
}
static void
diff --git a/sound/firewire/motu/motu-proc.c b/sound/firewire/motu/motu-proc.c
index ea46fb4..126a7bd 100644
--- a/sound/firewire/motu/motu-proc.c
+++ b/sound/firewire/motu/motu-proc.c
@@ -16,7 +16,7 @@
[SND_MOTU_CLOCK_SOURCE_SPDIF_ON_OPT] = "S/PDIF on optical interface",
[SND_MOTU_CLOCK_SOURCE_SPDIF_ON_OPT_A] = "S/PDIF on optical interface A",
[SND_MOTU_CLOCK_SOURCE_SPDIF_ON_OPT_B] = "S/PDIF on optical interface B",
- [SND_MOTU_CLOCK_SOURCE_SPDIF_ON_COAX] = "S/PCIF on coaxial interface",
+ [SND_MOTU_CLOCK_SOURCE_SPDIF_ON_COAX] = "S/PDIF on coaxial interface",
[SND_MOTU_CLOCK_SOURCE_AESEBU_ON_XLR] = "AESEBU on XLR interface",
[SND_MOTU_CLOCK_SOURCE_WORD_ON_BNC] = "Word clock on BNC interface",
};
diff --git a/sound/firewire/oxfw/oxfw-pcm.c b/sound/firewire/oxfw/oxfw-pcm.c
index 7c6d1c2..78d906a 100644
--- a/sound/firewire/oxfw/oxfw-pcm.c
+++ b/sound/firewire/oxfw/oxfw-pcm.c
@@ -255,7 +255,7 @@
mutex_unlock(&oxfw->mutex);
}
- return 0;
+ return err;
}
static int pcm_capture_hw_free(struct snd_pcm_substream *substream)
diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c
index fb6df3f..bebb2b8 100644
--- a/sound/firewire/oxfw/oxfw.c
+++ b/sound/firewire/oxfw/oxfw.c
@@ -350,8 +350,7 @@
* Onyx-i series (former models): 0x081216
* Mackie Onyx Satellite: 0x00200f
* Tapco LINK.firewire 4x6: 0x000460
- * d.2 pro: Unknown
- * d.4 pro: Unknown
+ * d.2 pro/d.4 pro (built-in card): Unknown
* U.420: Unknown
* U.420d: Unknown
*/
diff --git a/sound/firewire/tascam/amdtp-tascam.c b/sound/firewire/tascam/amdtp-tascam.c
index e80bb84..f823a2a 100644
--- a/sound/firewire/tascam/amdtp-tascam.c
+++ b/sound/firewire/tascam/amdtp-tascam.c
@@ -157,14 +157,15 @@
if ((before ^ after) & mask) {
struct snd_firewire_tascam_change *entry =
&tscm->queue[tscm->push_pos];
+ unsigned long flag;
- spin_lock_irq(&tscm->lock);
+ spin_lock_irqsave(&tscm->lock, flag);
entry->index = index;
entry->before = before;
entry->after = after;
if (++tscm->push_pos >= SND_TSCM_QUEUE_COUNT)
tscm->push_pos = 0;
- spin_unlock_irq(&tscm->lock);
+ spin_unlock_irqrestore(&tscm->lock, flag);
wake_up(&tscm->hwdep_wait);
}
diff --git a/sound/firewire/tascam/tascam-transaction.c b/sound/firewire/tascam/tascam-transaction.c
index 90288b4..a073cec 100644
--- a/sound/firewire/tascam/tascam-transaction.c
+++ b/sound/firewire/tascam/tascam-transaction.c
@@ -209,7 +209,7 @@
/* Set interval to next transaction. */
port->next_ktime = ktime_add_ns(ktime_get(),
- port->consume_bytes * 8 * NSEC_PER_SEC / 31250);
+ port->consume_bytes * 8 * (NSEC_PER_SEC / 31250));
/* Start this transaction. */
port->idling = false;
diff --git a/sound/firewire/tascam/tascam.c b/sound/firewire/tascam/tascam.c
index addc464..0175e3e 100644
--- a/sound/firewire/tascam/tascam.c
+++ b/sound/firewire/tascam/tascam.c
@@ -39,9 +39,6 @@
.midi_capture_ports = 2,
.midi_playback_ports = 4,
},
- // This kernel module doesn't support FE-8 because the most of features
- // can be implemented in userspace without any specific support of this
- // module.
};
static int identify_model(struct snd_tscm *tscm)
@@ -211,11 +208,39 @@
}
static const struct ieee1394_device_id snd_tscm_id_table[] = {
+ // Tascam, FW-1884.
{
.match_flags = IEEE1394_MATCH_VENDOR_ID |
- IEEE1394_MATCH_SPECIFIER_ID,
+ IEEE1394_MATCH_SPECIFIER_ID |
+ IEEE1394_MATCH_VERSION,
.vendor_id = 0x00022e,
.specifier_id = 0x00022e,
+ .version = 0x800000,
+ },
+ // Tascam, FE-8 (.version = 0x800001)
+ // This kernel module doesn't support FE-8 because the most of features
+ // can be implemented in userspace without any specific support of this
+ // module.
+ //
+ // .version = 0x800002 is unknown.
+ //
+ // Tascam, FW-1082.
+ {
+ .match_flags = IEEE1394_MATCH_VENDOR_ID |
+ IEEE1394_MATCH_SPECIFIER_ID |
+ IEEE1394_MATCH_VERSION,
+ .vendor_id = 0x00022e,
+ .specifier_id = 0x00022e,
+ .version = 0x800003,
+ },
+ // Tascam, FW-1804.
+ {
+ .match_flags = IEEE1394_MATCH_VENDOR_ID |
+ IEEE1394_MATCH_SPECIFIER_ID |
+ IEEE1394_MATCH_VERSION,
+ .vendor_id = 0x00022e,
+ .specifier_id = 0x00022e,
+ .version = 0x800004,
},
{}
};
diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c
index cfab60d..c87187f 100644
--- a/sound/hda/ext/hdac_ext_controller.c
+++ b/sound/hda/ext/hdac_ext_controller.c
@@ -148,6 +148,8 @@
return NULL;
if (bus->idx != bus_idx)
return NULL;
+ if (addr < 0 || addr > 31)
+ return NULL;
list_for_each_entry(hlink, &bus->hlink_list, list) {
for (i = 0; i < HDA_MAX_CODECS; i++) {
@@ -254,6 +256,7 @@
int snd_hdac_ext_bus_link_get(struct hdac_bus *bus,
struct hdac_ext_link *link)
{
+ unsigned long codec_mask;
int ret = 0;
mutex_lock(&bus->lock);
@@ -280,9 +283,11 @@
* HDA spec section 4.3 - Codec Discovery
*/
udelay(521);
- bus->codec_mask = snd_hdac_chip_readw(bus, STATESTS);
- dev_dbg(bus->dev, "codec_mask = 0x%lx\n", bus->codec_mask);
- snd_hdac_chip_writew(bus, STATESTS, bus->codec_mask);
+ codec_mask = snd_hdac_chip_readw(bus, STATESTS);
+ dev_dbg(bus->dev, "codec_mask = 0x%lx\n", codec_mask);
+ snd_hdac_chip_writew(bus, STATESTS, codec_mask);
+ if (!bus->codec_mask)
+ bus->codec_mask = codec_mask;
}
mutex_unlock(&bus->lock);
diff --git a/sound/hda/hdac_bus.c b/sound/hda/hdac_bus.c
index 8f19876..53be2ca 100644
--- a/sound/hda/hdac_bus.c
+++ b/sound/hda/hdac_bus.c
@@ -158,6 +158,7 @@
struct hdac_driver *drv;
unsigned int rp, caddr, res;
+ spin_lock_irq(&bus->reg_lock);
while (bus->unsol_rp != bus->unsol_wp) {
rp = (bus->unsol_rp + 1) % HDA_UNSOL_QUEUE_SIZE;
bus->unsol_rp = rp;
@@ -169,10 +170,13 @@
codec = bus->caddr_tbl[caddr & 0x0f];
if (!codec || !codec->dev.driver)
continue;
+ spin_unlock_irq(&bus->reg_lock);
drv = drv_to_hdac_driver(codec->dev.driver);
if (drv->unsol_event)
drv->unsol_event(codec, res);
+ spin_lock_irq(&bus->reg_lock);
}
+ spin_unlock_irq(&bus->reg_lock);
}
/**
diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c
index 9f3e375..b84e12f 100644
--- a/sound/hda/hdac_device.c
+++ b/sound/hda/hdac_device.c
@@ -57,6 +57,7 @@
codec->addr = addr;
codec->type = HDA_DEV_CORE;
mutex_init(&codec->widget_lock);
+ mutex_init(&codec->regmap_lock);
pm_runtime_set_active(&codec->dev);
pm_runtime_get_noresume(&codec->dev);
atomic_set(&codec->in_pm, 0);
@@ -126,6 +127,8 @@
void snd_hdac_device_exit(struct hdac_device *codec)
{
pm_runtime_put_noidle(&codec->dev);
+ /* keep balance of runtime PM child_count in parent device */
+ pm_runtime_set_suspended(&codec->dev);
snd_hdac_bus_remove_device(codec->bus, codec);
kfree(codec->vendor_name);
kfree(codec->chip_name);
diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c
index 286361e..4978039 100644
--- a/sound/hda/hdac_regmap.c
+++ b/sound/hda/hdac_regmap.c
@@ -425,12 +425,29 @@
static int reg_raw_write(struct hdac_device *codec, unsigned int reg,
unsigned int val)
{
+ int err;
+
+ mutex_lock(&codec->regmap_lock);
if (!codec->regmap)
- return hda_reg_write(codec, reg, val);
+ err = hda_reg_write(codec, reg, val);
else
- return regmap_write(codec->regmap, reg, val);
+ err = regmap_write(codec->regmap, reg, val);
+ mutex_unlock(&codec->regmap_lock);
+ return err;
}
+/* a helper macro to call @func_call; retry with power-up if failed */
+#define CALL_RAW_FUNC(codec, func_call) \
+ ({ \
+ int _err = func_call; \
+ if (_err == -EAGAIN) { \
+ _err = snd_hdac_power_up_pm(codec); \
+ if (_err >= 0) \
+ _err = func_call; \
+ snd_hdac_power_down_pm(codec); \
+ } \
+ _err;})
+
/**
* snd_hdac_regmap_write_raw - write a pseudo register with power mgmt
* @codec: the codec object
@@ -442,42 +459,29 @@
int snd_hdac_regmap_write_raw(struct hdac_device *codec, unsigned int reg,
unsigned int val)
{
- int err;
-
- err = reg_raw_write(codec, reg, val);
- if (err == -EAGAIN) {
- err = snd_hdac_power_up_pm(codec);
- if (err >= 0)
- err = reg_raw_write(codec, reg, val);
- snd_hdac_power_down_pm(codec);
- }
- return err;
+ return CALL_RAW_FUNC(codec, reg_raw_write(codec, reg, val));
}
EXPORT_SYMBOL_GPL(snd_hdac_regmap_write_raw);
static int reg_raw_read(struct hdac_device *codec, unsigned int reg,
unsigned int *val, bool uncached)
{
+ int err;
+
+ mutex_lock(&codec->regmap_lock);
if (uncached || !codec->regmap)
- return hda_reg_read(codec, reg, val);
+ err = hda_reg_read(codec, reg, val);
else
- return regmap_read(codec->regmap, reg, val);
+ err = regmap_read(codec->regmap, reg, val);
+ mutex_unlock(&codec->regmap_lock);
+ return err;
}
static int __snd_hdac_regmap_read_raw(struct hdac_device *codec,
unsigned int reg, unsigned int *val,
bool uncached)
{
- int err;
-
- err = reg_raw_read(codec, reg, val, uncached);
- if (err == -EAGAIN) {
- err = snd_hdac_power_up_pm(codec);
- if (err >= 0)
- err = reg_raw_read(codec, reg, val, uncached);
- snd_hdac_power_down_pm(codec);
- }
- return err;
+ return CALL_RAW_FUNC(codec, reg_raw_read(codec, reg, val, uncached));
}
/**
@@ -504,6 +508,35 @@
return __snd_hdac_regmap_read_raw(codec, reg, val, true);
}
+static int reg_raw_update(struct hdac_device *codec, unsigned int reg,
+ unsigned int mask, unsigned int val)
+{
+ unsigned int orig;
+ bool change;
+ int err;
+
+ mutex_lock(&codec->regmap_lock);
+ if (codec->regmap) {
+ err = regmap_update_bits_check(codec->regmap, reg, mask, val,
+ &change);
+ if (!err)
+ err = change ? 1 : 0;
+ } else {
+ err = hda_reg_read(codec, reg, &orig);
+ if (!err) {
+ val &= mask;
+ val |= orig & ~mask;
+ if (val != orig) {
+ err = hda_reg_write(codec, reg, val);
+ if (!err)
+ err = 1;
+ }
+ }
+ }
+ mutex_unlock(&codec->regmap_lock);
+ return err;
+}
+
/**
* snd_hdac_regmap_update_raw - update a pseudo register with power mgmt
* @codec: the codec object
@@ -516,19 +549,57 @@
int snd_hdac_regmap_update_raw(struct hdac_device *codec, unsigned int reg,
unsigned int mask, unsigned int val)
{
+ return CALL_RAW_FUNC(codec, reg_raw_update(codec, reg, mask, val));
+}
+EXPORT_SYMBOL_GPL(snd_hdac_regmap_update_raw);
+
+static int reg_raw_update_once(struct hdac_device *codec, unsigned int reg,
+ unsigned int mask, unsigned int val)
+{
unsigned int orig;
int err;
- val &= mask;
- err = snd_hdac_regmap_read_raw(codec, reg, &orig);
+ if (!codec->regmap)
+ return reg_raw_update(codec, reg, mask, val);
+
+ mutex_lock(&codec->regmap_lock);
+ regcache_cache_only(codec->regmap, true);
+ err = regmap_read(codec->regmap, reg, &orig);
+ regcache_cache_only(codec->regmap, false);
if (err < 0)
- return err;
- val |= orig & ~mask;
- if (val == orig)
- return 0;
- err = snd_hdac_regmap_write_raw(codec, reg, val);
- if (err < 0)
- return err;
- return 1;
+ err = regmap_update_bits(codec->regmap, reg, mask, val);
+ mutex_unlock(&codec->regmap_lock);
+ return err;
}
-EXPORT_SYMBOL_GPL(snd_hdac_regmap_update_raw);
+
+/**
+ * snd_hdac_regmap_update_raw_once - initialize the register value only once
+ * @codec: the codec object
+ * @reg: pseudo register
+ * @mask: bit mask to update
+ * @val: value to update
+ *
+ * Performs the update of the register bits only once when the register
+ * hasn't been initialized yet. Used in HD-audio legacy driver.
+ * Returns zero if successful or a negative error code
+ */
+int snd_hdac_regmap_update_raw_once(struct hdac_device *codec, unsigned int reg,
+ unsigned int mask, unsigned int val)
+{
+ return CALL_RAW_FUNC(codec, reg_raw_update_once(codec, reg, mask, val));
+}
+EXPORT_SYMBOL_GPL(snd_hdac_regmap_update_raw_once);
+
+/**
+ * snd_hdac_regmap_sync - sync out the cached values for PM resume
+ * @codec: the codec object
+ */
+void snd_hdac_regmap_sync(struct hdac_device *codec)
+{
+ if (codec->regmap) {
+ mutex_lock(&codec->regmap_lock);
+ regcache_sync(codec->regmap);
+ mutex_unlock(&codec->regmap_lock);
+ }
+}
+EXPORT_SYMBOL_GPL(snd_hdac_regmap_sync);
diff --git a/sound/hda/hdac_stream.c b/sound/hda/hdac_stream.c
index d8fe7ff..682ed39 100644
--- a/sound/hda/hdac_stream.c
+++ b/sound/hda/hdac_stream.c
@@ -96,12 +96,14 @@
1 << azx_dev->index,
1 << azx_dev->index);
/* set stripe control */
- if (azx_dev->substream)
- stripe_ctl = snd_hdac_get_stream_stripe_ctl(bus, azx_dev->substream);
- else
- stripe_ctl = 0;
- snd_hdac_stream_updateb(azx_dev, SD_CTL_3B, SD_CTL_STRIPE_MASK,
- stripe_ctl);
+ if (azx_dev->stripe) {
+ if (azx_dev->substream)
+ stripe_ctl = snd_hdac_get_stream_stripe_ctl(bus, azx_dev->substream);
+ else
+ stripe_ctl = 0;
+ snd_hdac_stream_updateb(azx_dev, SD_CTL_3B, SD_CTL_STRIPE_MASK,
+ stripe_ctl);
+ }
/* set DMA start and interrupt mask */
snd_hdac_stream_updateb(azx_dev, SD_CTL,
0, SD_CTL_DMA_START | SD_INT_MASK);
@@ -118,7 +120,8 @@
snd_hdac_stream_updateb(azx_dev, SD_CTL,
SD_CTL_DMA_START | SD_INT_MASK, 0);
snd_hdac_stream_writeb(azx_dev, SD_STS, SD_INT_MASK); /* to be sure */
- snd_hdac_stream_updateb(azx_dev, SD_CTL_3B, SD_CTL_STRIPE_MASK, 0);
+ if (azx_dev->stripe)
+ snd_hdac_stream_updateb(azx_dev, SD_CTL_3B, SD_CTL_STRIPE_MASK, 0);
azx_dev->running = false;
}
EXPORT_SYMBOL_GPL(snd_hdac_stream_clear);
diff --git a/sound/hda/hdmi_chmap.c b/sound/hda/hdmi_chmap.c
index 886cb78..2efee79 100644
--- a/sound/hda/hdmi_chmap.c
+++ b/sound/hda/hdmi_chmap.c
@@ -250,7 +250,7 @@
for (i = 0, j = 0; i < ARRAY_SIZE(cea_speaker_allocation_names); i++) {
if (spk_alloc & (1 << i))
- j += snprintf(buf + j, buflen - j, " %s",
+ j += scnprintf(buf + j, buflen - j, " %s",
cea_speaker_allocation_names[i]);
}
buf[j] = '\0'; /* necessary when j == 0 */
diff --git a/sound/hda/intel-nhlt.c b/sound/hda/intel-nhlt.c
index daede96..6ed80a4 100644
--- a/sound/hda/intel-nhlt.c
+++ b/sound/hda/intel-nhlt.c
@@ -64,18 +64,49 @@
struct nhlt_endpoint *epnt;
struct nhlt_dmic_array_config *cfg;
struct nhlt_vendor_dmic_array_config *cfg_vendor;
+ struct nhlt_fmt *fmt_configs;
unsigned int dmic_geo = 0;
- u8 j;
+ u16 max_ch = 0;
+ u8 i, j;
if (!nhlt)
return 0;
- epnt = (struct nhlt_endpoint *)nhlt->desc;
+ if (nhlt->header.length <= sizeof(struct acpi_table_header)) {
+ dev_warn(dev, "Invalid DMIC description table\n");
+ return 0;
+ }
- for (j = 0; j < nhlt->endpoint_count; j++) {
- if (epnt->linktype == NHLT_LINK_DMIC) {
- cfg = (struct nhlt_dmic_array_config *)
- (epnt->config.caps);
+ for (j = 0, epnt = nhlt->desc; j < nhlt->endpoint_count; j++,
+ epnt = (struct nhlt_endpoint *)((u8 *)epnt + epnt->length)) {
+
+ if (epnt->linktype != NHLT_LINK_DMIC)
+ continue;
+
+ cfg = (struct nhlt_dmic_array_config *)(epnt->config.caps);
+ fmt_configs = (struct nhlt_fmt *)(epnt->config.caps + epnt->config.size);
+
+ /* find max number of channels based on format_configuration */
+ if (fmt_configs->fmt_count) {
+ dev_dbg(dev, "%s: found %d format definitions\n",
+ __func__, fmt_configs->fmt_count);
+
+ for (i = 0; i < fmt_configs->fmt_count; i++) {
+ struct wav_fmt_ext *fmt_ext;
+
+ fmt_ext = &fmt_configs->fmt_config[i].fmt_ext;
+
+ if (fmt_ext->fmt.channels > max_ch)
+ max_ch = fmt_ext->fmt.channels;
+ }
+ dev_dbg(dev, "%s: max channels found %d\n", __func__, max_ch);
+ } else {
+ dev_dbg(dev, "%s: No format information found\n", __func__);
+ }
+
+ if (cfg->device_config.config_type != NHLT_CONFIG_TYPE_MIC_ARRAY) {
+ dmic_geo = max_ch;
+ } else {
switch (cfg->array_type) {
case NHLT_MIC_ARRAY_2CH_SMALL:
case NHLT_MIC_ARRAY_2CH_BIG:
@@ -92,13 +123,23 @@
dmic_geo = cfg_vendor->nb_mics;
break;
default:
- dev_warn(dev, "undefined DMIC array_type 0x%0x\n",
- cfg->array_type);
+ dev_warn(dev, "%s: undefined DMIC array_type 0x%0x\n",
+ __func__, cfg->array_type);
+ }
+
+ if (dmic_geo > 0) {
+ dev_dbg(dev, "%s: Array with %d dmics\n", __func__, dmic_geo);
+ }
+ if (max_ch > dmic_geo) {
+ dev_dbg(dev, "%s: max channels %d exceed dmic number %d\n",
+ __func__, max_ch, dmic_geo);
}
}
- epnt = (struct nhlt_endpoint *)((u8 *)epnt + epnt->length);
}
+ dev_dbg(dev, "%s: dmic number %d max_ch %d\n",
+ __func__, dmic_geo, max_ch);
+
return dmic_geo;
}
EXPORT_SYMBOL_GPL(intel_nhlt_get_dmic_geo);
diff --git a/sound/isa/cmi8330.c b/sound/isa/cmi8330.c
index bb7d494..281ecd0 100644
--- a/sound/isa/cmi8330.c
+++ b/sound/isa/cmi8330.c
@@ -549,7 +549,7 @@
}
if (acard->sb->hardware != SB_HW_16) {
snd_printk(KERN_ERR PFX "SB16 not found during probe\n");
- return err;
+ return -ENODEV;
}
snd_wss_out(acard->wss, CS4231_MISC_INFO, 0x40); /* switch on MODE2 */
diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c
index 78dd213..fa3c39c 100644
--- a/sound/isa/cs423x/cs4236.c
+++ b/sound/isa/cs423x/cs4236.c
@@ -278,7 +278,8 @@
} else {
mpu_port[dev] = pnp_port_start(pdev, 0);
if (mpu_irq[dev] >= 0 &&
- pnp_irq_valid(pdev, 0) && pnp_irq(pdev, 0) >= 0) {
+ pnp_irq_valid(pdev, 0) &&
+ pnp_irq(pdev, 0) != (resource_size_t)-1) {
mpu_irq[dev] = pnp_irq(pdev, 0);
} else {
mpu_irq[dev] = -1; /* disable interrupt */
diff --git a/sound/isa/es1688/es1688.c b/sound/isa/es1688/es1688.c
index 9be8937..b4e9b0d 100644
--- a/sound/isa/es1688/es1688.c
+++ b/sound/isa/es1688/es1688.c
@@ -267,8 +267,10 @@
return error;
}
error = snd_es1688_probe(card, dev);
- if (error < 0)
+ if (error < 0) {
+ snd_card_free(card);
return error;
+ }
pnp_set_card_drvdata(pcard, card);
snd_es968_pnp_is_probed = 1;
return 0;
diff --git a/sound/isa/gus/gus_main.c b/sound/isa/gus/gus_main.c
index af6b4d8..39911a6 100644
--- a/sound/isa/gus/gus_main.c
+++ b/sound/isa/gus/gus_main.c
@@ -77,17 +77,8 @@
static void snd_gus_init_control(struct snd_gus_card *gus)
{
- int ret;
-
- if (!gus->ace_flag) {
- ret =
- snd_ctl_add(gus->card,
- snd_ctl_new1(&snd_gus_joystick_control,
- gus));
- if (ret)
- snd_printk(KERN_ERR "gus: snd_ctl_add failed: %d\n",
- ret);
- }
+ if (!gus->ace_flag)
+ snd_ctl_add(gus->card, snd_ctl_new1(&snd_gus_joystick_control, gus));
}
/*
diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c
index 0458934..9ca5c83 100644
--- a/sound/isa/opti9xx/miro.c
+++ b/sound/isa/opti9xx/miro.c
@@ -867,10 +867,13 @@
spin_unlock_irqrestore(&chip->lock, flags);
}
+static inline void snd_miro_write_mask(struct snd_miro *chip,
+ unsigned char reg, unsigned char value, unsigned char mask)
+{
+ unsigned char oldval = snd_miro_read(chip, reg);
-#define snd_miro_write_mask(chip, reg, value, mask) \
- snd_miro_write(chip, reg, \
- (snd_miro_read(chip, reg) & ~(mask)) | ((value) & (mask)))
+ snd_miro_write(chip, reg, (oldval & ~mask) | (value & mask));
+}
/*
* Proc Interface
diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c
index fb36bb5..fb87eed 100644
--- a/sound/isa/opti9xx/opti92x-ad1848.c
+++ b/sound/isa/opti9xx/opti92x-ad1848.c
@@ -317,10 +317,13 @@
}
-#define snd_opti9xx_write_mask(chip, reg, value, mask) \
- snd_opti9xx_write(chip, reg, \
- (snd_opti9xx_read(chip, reg) & ~(mask)) | ((value) & (mask)))
+static inline void snd_opti9xx_write_mask(struct snd_opti9xx *chip,
+ unsigned char reg, unsigned char value, unsigned char mask)
+{
+ unsigned char oldval = snd_opti9xx_read(chip, reg);
+ snd_opti9xx_write(chip, reg, (oldval & ~mask) | (value & mask));
+}
static int snd_opti9xx_configure(struct snd_opti9xx *chip,
long port,
diff --git a/sound/isa/sb/emu8000.c b/sound/isa/sb/emu8000.c
index 433e32e..91c96fa 100644
--- a/sound/isa/sb/emu8000.c
+++ b/sound/isa/sb/emu8000.c
@@ -1029,8 +1029,10 @@
memset(emu->controls, 0, sizeof(emu->controls));
for (i = 0; i < EMU8000_NUM_CONTROLS; i++) {
- if ((err = snd_ctl_add(card, emu->controls[i] = snd_ctl_new1(mixer_defs[i], emu))) < 0)
+ if ((err = snd_ctl_add(card, emu->controls[i] = snd_ctl_new1(mixer_defs[i], emu))) < 0) {
+ emu->controls[i] = NULL;
goto __error;
+ }
}
return 0;
diff --git a/sound/isa/sb/sb16_csp.c b/sound/isa/sb/sb16_csp.c
index 4ad0ff0..30021ab 100644
--- a/sound/isa/sb/sb16_csp.c
+++ b/sound/isa/sb/sb16_csp.c
@@ -814,6 +814,7 @@
mixR = snd_sbmixer_read(p->chip, SB_DSP4_PCM_DEV + 1);
snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV, mixL & 0x7);
snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV + 1, mixR & 0x7);
+ spin_unlock_irqrestore(&p->chip->mixer_lock, flags);
spin_lock(&p->chip->reg_lock);
set_mode_register(p->chip, 0xc0); /* c0 = STOP */
@@ -853,6 +854,7 @@
spin_unlock(&p->chip->reg_lock);
/* restore PCM volume */
+ spin_lock_irqsave(&p->chip->mixer_lock, flags);
snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV, mixL);
snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV + 1, mixR);
spin_unlock_irqrestore(&p->chip->mixer_lock, flags);
@@ -878,6 +880,7 @@
mixR = snd_sbmixer_read(p->chip, SB_DSP4_PCM_DEV + 1);
snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV, mixL & 0x7);
snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV + 1, mixR & 0x7);
+ spin_unlock_irqrestore(&p->chip->mixer_lock, flags);
spin_lock(&p->chip->reg_lock);
if (p->running & SNDRV_SB_CSP_ST_QSOUND) {
@@ -892,6 +895,7 @@
spin_unlock(&p->chip->reg_lock);
/* restore PCM volume */
+ spin_lock_irqsave(&p->chip->mixer_lock, flags);
snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV, mixL);
snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV + 1, mixR);
spin_unlock_irqrestore(&p->chip->mixer_lock, flags);
@@ -1045,10 +1049,14 @@
spin_lock_init(&p->q_lock);
- if ((err = snd_ctl_add(card, p->qsound_switch = snd_ctl_new1(&snd_sb_qsound_switch, p))) < 0)
+ if ((err = snd_ctl_add(card, p->qsound_switch = snd_ctl_new1(&snd_sb_qsound_switch, p))) < 0) {
+ p->qsound_switch = NULL;
goto __error;
- if ((err = snd_ctl_add(card, p->qsound_space = snd_ctl_new1(&snd_sb_qsound_space, p))) < 0)
+ }
+ if ((err = snd_ctl_add(card, p->qsound_space = snd_ctl_new1(&snd_sb_qsound_space, p))) < 0) {
+ p->qsound_space = NULL;
goto __error;
+ }
return 0;
@@ -1068,10 +1076,14 @@
card = p->chip->card;
down_write(&card->controls_rwsem);
- if (p->qsound_switch)
+ if (p->qsound_switch) {
snd_ctl_remove(card, p->qsound_switch);
- if (p->qsound_space)
+ p->qsound_switch = NULL;
+ }
+ if (p->qsound_space) {
snd_ctl_remove(card, p->qsound_space);
+ p->qsound_space = NULL;
+ }
up_write(&card->controls_rwsem);
/* cancel pending transfer of QSound parameters */
diff --git a/sound/isa/sb/sb16_main.c b/sound/isa/sb/sb16_main.c
index 0768bbf..679f9f4 100644
--- a/sound/isa/sb/sb16_main.c
+++ b/sound/isa/sb/sb16_main.c
@@ -864,14 +864,10 @@
snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_sb16_playback_ops);
snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_sb16_capture_ops);
- if (chip->dma16 >= 0 && chip->dma8 != chip->dma16) {
- err = snd_ctl_add(card, snd_ctl_new1(
- &snd_sb16_dma_control, chip));
- if (err)
- return err;
- } else {
+ if (chip->dma16 >= 0 && chip->dma8 != chip->dma16)
+ snd_ctl_add(card, snd_ctl_new1(&snd_sb16_dma_control, chip));
+ else
pcm->info_flags = SNDRV_PCM_INFO_HALF_DUPLEX;
- }
snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
card->dev,
diff --git a/sound/isa/sb/sb8.c b/sound/isa/sb/sb8.c
index d67eae3..6f1fc87 100644
--- a/sound/isa/sb/sb8.c
+++ b/sound/isa/sb/sb8.c
@@ -96,10 +96,6 @@
/* block the 0x388 port to avoid PnP conflicts */
acard->fm_res = request_region(0x388, 4, "SoundBlaster FM");
- if (!acard->fm_res) {
- err = -EBUSY;
- goto _err;
- }
if (port[dev] != SNDRV_AUTO_PORT) {
if ((err = snd_sbdsp_create(card, port[dev], irq[dev],
diff --git a/sound/isa/wavefront/wavefront_synth.c b/sound/isa/wavefront/wavefront_synth.c
index c5b1d59..d6420d2 100644
--- a/sound/isa/wavefront/wavefront_synth.c
+++ b/sound/isa/wavefront/wavefront_synth.c
@@ -1171,7 +1171,10 @@
"alias for %d\n",
header->number,
header->hdr.a.OriginalSample);
-
+
+ if (header->number >= WF_MAX_SAMPLE)
+ return -EINVAL;
+
munge_int32 (header->number, &alias_hdr[0], 2);
munge_int32 (header->hdr.a.OriginalSample, &alias_hdr[2], 2);
munge_int32 (*((unsigned int *)&header->hdr.a.sampleStartOffset),
@@ -1202,6 +1205,9 @@
int num_samples;
unsigned char *msample_hdr;
+ if (header->number >= WF_MAX_SAMPLE)
+ return -EINVAL;
+
msample_hdr = kmalloc(WF_MSAMPLE_BYTES, GFP_KERNEL);
if (! msample_hdr)
return -ENOMEM;
diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c
index 496dcde..9790f51 100644
--- a/sound/pci/asihpi/hpioctl.c
+++ b/sound/pci/asihpi/hpioctl.c
@@ -343,7 +343,7 @@
struct hpi_message hm;
struct hpi_response hr;
struct hpi_adapter adapter;
- struct hpi_pci pci;
+ struct hpi_pci pci = { 0 };
memset(&adapter, 0, sizeof(adapter));
@@ -499,7 +499,7 @@
return 0;
err:
- for (idx = 0; idx < HPI_MAX_ADAPTER_MEM_SPACES; idx++) {
+ while (--idx >= 0) {
if (pci.ap_mem_base[idx]) {
iounmap(pci.ap_mem_base[idx]);
pci.ap_mem_base[idx] = NULL;
diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c
index 478412e..7aedaeb 100644
--- a/sound/pci/ca0106/ca0106_main.c
+++ b/sound/pci/ca0106/ca0106_main.c
@@ -537,7 +537,8 @@
else
/* Power down */
chip->spi_dac_reg[reg] |= bit;
- return snd_ca0106_spi_write(chip, chip->spi_dac_reg[reg]);
+ if (snd_ca0106_spi_write(chip, chip->spi_dac_reg[reg]) != 0)
+ return -ENXIO;
}
return 0;
}
diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c
index 5b888b7..c07a9e7 100644
--- a/sound/pci/cs46xx/cs46xx_lib.c
+++ b/sound/pci/cs46xx/cs46xx_lib.c
@@ -766,7 +766,7 @@
rate = 48000 / 9;
/*
- * We can not capture at at rate greater than the Input Rate (48000).
+ * We can not capture at a rate greater than the Input Rate (48000).
* Return an error if an attempt is made to stray outside that limit.
*/
if (rate > 48000)
diff --git a/sound/pci/cs46xx/dsp_spos_scb_lib.c b/sound/pci/cs46xx/dsp_spos_scb_lib.c
index 715ead5..0bef823 100644
--- a/sound/pci/cs46xx/dsp_spos_scb_lib.c
+++ b/sound/pci/cs46xx/dsp_spos_scb_lib.c
@@ -1716,7 +1716,7 @@
struct dsp_spos_instance * ins = chip->dsp_spos_instance;
if ( ins->spdif_status_out & DSP_SPDIF_STATUS_OUTPUT_ENABLED ) {
- /* remove AsynchFGTxSCB and and PCMSerialInput_II */
+ /* remove AsynchFGTxSCB and PCMSerialInput_II */
cs46xx_dsp_disable_spdif_out (chip);
/* save state */
diff --git a/sound/pci/ctxfi/cthw20k2.c b/sound/pci/ctxfi/cthw20k2.c
index 3cd4b7d..b1cc4cd 100644
--- a/sound/pci/ctxfi/cthw20k2.c
+++ b/sound/pci/ctxfi/cthw20k2.c
@@ -991,7 +991,7 @@
if (idx < 4) {
/* S/PDIF output */
- switch ((conf & 0x7)) {
+ switch ((conf & 0xf)) {
case 1:
set_field(&ctl->txctl[idx], ATXCTL_NUC, 0);
break;
diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c
index ca91257..8596ae4 100644
--- a/sound/pci/echoaudio/echoaudio.c
+++ b/sound/pci/echoaudio/echoaudio.c
@@ -2198,7 +2198,6 @@
if (err < 0) {
kfree(commpage_bak);
dev_err(dev, "resume init_hw err=%d\n", err);
- snd_echo_free(chip);
return err;
}
@@ -2225,7 +2224,6 @@
if (request_irq(pci->irq, snd_echo_interrupt, IRQF_SHARED,
KBUILD_MODNAME, chip)) {
dev_err(chip->card->dev, "cannot grab irq\n");
- snd_echo_free(chip);
return -EBUSY;
}
chip->irq = pci->irq;
diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c
index 2c6d2be..824f4ac 100644
--- a/sound/pci/hda/hda_auto_parser.c
+++ b/sound/pci/hda/hda_auto_parser.c
@@ -72,6 +72,12 @@
if (a->type != b->type)
return (int)(a->type - b->type);
+ /* If has both hs_mic and hp_mic, pick the hs_mic ahead of hp_mic. */
+ if (a->is_headset_mic && b->is_headphone_mic)
+ return -1; /* don't swap */
+ else if (a->is_headphone_mic && b->is_headset_mic)
+ return 1; /* swap */
+
/* In case one has boost and the other one has not,
pick the one with boost first. */
return (int)(b->has_boost_on_pin - a->has_boost_on_pin);
diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c
index b7d9160..c6e1e03 100644
--- a/sound/pci/hda/hda_beep.c
+++ b/sound/pci/hda/hda_beep.c
@@ -290,8 +290,12 @@
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct hda_beep *beep = codec->beep;
+ int chs = get_amp_channels(kcontrol);
+
if (beep && (!beep->enabled || !ctl_has_mute(kcontrol))) {
- ucontrol->value.integer.value[0] =
+ if (chs & 1)
+ ucontrol->value.integer.value[0] = beep->enabled;
+ if (chs & 2)
ucontrol->value.integer.value[1] = beep->enabled;
return 0;
}
diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c
index 8272b50..17a25e4 100644
--- a/sound/pci/hda/hda_bind.c
+++ b/sound/pci/hda/hda_bind.c
@@ -43,6 +43,14 @@
{
struct hda_codec *codec = container_of(dev, struct hda_codec, core);
+ /* ignore unsol events during shutdown */
+ if (codec->bus->shutdown)
+ return;
+
+ /* ignore unsol events during system suspend/resume */
+ if (codec->core.dev.power.power_state.event != PM_EVENT_ON)
+ return;
+
if (codec->patch_ops.unsol_event)
codec->patch_ops.unsol_event(codec, ev);
}
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index a2fb191..326f95c 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -641,8 +641,18 @@
struct hda_codec *codec =
container_of(work, struct hda_codec, jackpoll_work.work);
- snd_hda_jack_set_dirty_all(codec);
- snd_hda_jack_poll_all(codec);
+ /* for non-polling trigger: we need nothing if already powered on */
+ if (!codec->jackpoll_interval && snd_hdac_is_power_on(&codec->core))
+ return;
+
+ /* the power-up/down sequence triggers the runtime resume */
+ snd_hda_power_up_pm(codec);
+ /* update jacks manually if polling is required, too */
+ if (codec->jackpoll_interval) {
+ snd_hda_jack_set_dirty_all(codec);
+ snd_hda_jack_poll_all(codec);
+ }
+ snd_hda_power_down_pm(codec);
if (!codec->jackpoll_interval)
return;
@@ -1267,6 +1277,18 @@
}
EXPORT_SYMBOL_GPL(snd_hda_override_amp_caps);
+static unsigned int encode_amp(struct hda_codec *codec, hda_nid_t nid,
+ int ch, int dir, int idx)
+{
+ unsigned int cmd = snd_hdac_regmap_encode_amp(nid, ch, dir, idx);
+
+ /* enable fake mute if no h/w mute but min=mute */
+ if ((query_amp_caps(codec, nid, dir) &
+ (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE)) == AC_AMPCAP_MIN_MUTE)
+ cmd |= AC_AMP_FAKE_MUTE;
+ return cmd;
+}
+
/**
* snd_hda_codec_amp_update - update the AMP mono value
* @codec: HD-audio codec
@@ -1282,12 +1304,8 @@
int snd_hda_codec_amp_update(struct hda_codec *codec, hda_nid_t nid,
int ch, int dir, int idx, int mask, int val)
{
- unsigned int cmd = snd_hdac_regmap_encode_amp(nid, ch, dir, idx);
+ unsigned int cmd = encode_amp(codec, nid, ch, dir, idx);
- /* enable fake mute if no h/w mute but min=mute */
- if ((query_amp_caps(codec, nid, dir) &
- (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE)) == AC_AMPCAP_MIN_MUTE)
- cmd |= AC_AMP_FAKE_MUTE;
return snd_hdac_regmap_update_raw(&codec->core, cmd, mask, val);
}
EXPORT_SYMBOL_GPL(snd_hda_codec_amp_update);
@@ -1335,16 +1353,11 @@
int snd_hda_codec_amp_init(struct hda_codec *codec, hda_nid_t nid, int ch,
int dir, int idx, int mask, int val)
{
- int orig;
+ unsigned int cmd = encode_amp(codec, nid, ch, dir, idx);
if (!codec->core.regmap)
return -EINVAL;
- regcache_cache_only(codec->core.regmap, true);
- orig = snd_hda_codec_amp_read(codec, nid, ch, dir, idx);
- regcache_cache_only(codec->core.regmap, false);
- if (orig >= 0)
- return 0;
- return snd_hda_codec_amp_update(codec, nid, ch, dir, idx, mask, val);
+ return snd_hdac_regmap_update_raw_once(&codec->core, cmd, mask, val);
}
EXPORT_SYMBOL_GPL(snd_hda_codec_amp_init);
@@ -1785,7 +1798,7 @@
return -EBUSY;
/* OK, let it free */
- snd_hdac_device_unregister(&codec->core);
+ device_release_driver(hda_codec_dev(codec));
/* allow device access again */
snd_hda_unlock_devices(bus);
@@ -2905,8 +2918,7 @@
else {
if (codec->patch_ops.init)
codec->patch_ops.init(codec);
- if (codec->core.regmap)
- regcache_sync(codec->core.regmap);
+ snd_hda_regmap_sync(codec);
}
if (codec->jackpoll_interval)
@@ -2922,6 +2934,10 @@
struct hda_codec *codec = dev_to_hda_codec(dev);
unsigned int state;
+ /* Nothing to do if card registration fails and the component driver never probes */
+ if (!codec->card)
+ return 0;
+
cancel_delayed_work_sync(&codec->jackpoll_work);
state = hda_call_codec_suspend(codec);
if (codec->link_down_at_suspend ||
@@ -2936,6 +2952,10 @@
{
struct hda_codec *codec = dev_to_hda_codec(dev);
+ /* Nothing to do if card registration fails and the component driver never probes */
+ if (!codec->card)
+ return 0;
+
codec_display_power(codec, true);
snd_hdac_codec_link_up(&codec->core);
hda_call_codec_resume(codec);
@@ -2948,18 +2968,14 @@
static int hda_codec_force_resume(struct device *dev)
{
struct hda_codec *codec = dev_to_hda_codec(dev);
- bool forced_resume = !codec->relaxed_resume && codec->jacktbl.used;
int ret;
- /* The get/put pair below enforces the runtime resume even if the
- * device hasn't been used at suspend time. This trick is needed to
- * update the jack state change during the sleep.
- */
- if (forced_resume)
- pm_runtime_get_noresume(dev);
ret = pm_runtime_force_resume(dev);
- if (forced_resume)
- pm_runtime_put(dev);
+ /* schedule jackpoll work for jack detection update */
+ if (codec->jackpoll_interval ||
+ (pm_runtime_suspended(dev) && hda_codec_need_resume(codec)))
+ schedule_delayed_work(&codec->jackpoll_work,
+ codec->jackpoll_interval);
return ret;
}
@@ -3410,7 +3426,7 @@
* @nid: NID to check / update
*
* Check whether the given NID is in the amp list. If it's in the list,
- * check the current AMP status, and update the the power-status according
+ * check the current AMP status, and update the power-status according
* to the mute status.
*
* This function is supposed to be set or called from the check_power_status
@@ -4019,7 +4035,7 @@
for (i = 0, j = 0; i < ARRAY_SIZE(bits); i++)
if (pcm & (AC_SUPPCM_BITS_8 << i))
- j += snprintf(buf + j, buflen - j, " %d", bits[i]);
+ j += scnprintf(buf + j, buflen - j, " %d", bits[i]);
buf[j] = '\0'; /* necessary when j == 0 */
}
diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c
index 6387c7e..6a159c6 100644
--- a/sound/pci/hda/hda_controller.c
+++ b/sound/pci/hda/hda_controller.c
@@ -613,13 +613,6 @@
20,
178000000);
- /* by some reason, the playback stream stalls on PulseAudio with
- * tsched=1 when a capture stream triggers. Until we figure out the
- * real cause, disable tsched mode by telling the PCM info flag.
- */
- if (chip->driver_caps & AZX_DCAPS_AMD_WORKAROUND)
- runtime->hw.info |= SNDRV_PCM_INFO_BATCH;
-
if (chip->align_buffer_size)
/* constrain buffer sizes to be multiple of 128
bytes. This is more efficient in terms of memory
@@ -884,7 +877,7 @@
return -EAGAIN; /* give a chance to retry */
}
- dev_WARN(chip->card->dev,
+ dev_err(chip->card->dev,
"azx_get_response timeout, switching to single_cmd mode: last cmd=0x%08x\n",
bus->last_cmd[addr]);
chip->single_cmd = 1;
@@ -1159,16 +1152,23 @@
if (snd_hdac_bus_handle_stream_irq(bus, status, stream_update))
active = true;
- /* clear rirb int */
status = azx_readb(chip, RIRBSTS);
if (status & RIRB_INT_MASK) {
+ /*
+ * Clearing the interrupt status here ensures that no
+ * interrupt gets masked after the RIRB wp is read in
+ * snd_hdac_bus_update_rirb. This avoids a possible
+ * race condition where codec response in RIRB may
+ * remain unserviced by IRQ, eventually falling back
+ * to polling mode in azx_rirb_get_response.
+ */
+ azx_writeb(chip, RIRBSTS, RIRB_INT_MASK);
active = true;
if (status & RIRB_INT_RESPONSE) {
if (chip->driver_caps & AZX_DCAPS_CTX_WORKAROUND)
udelay(80);
snd_hdac_bus_update_rirb(bus);
}
- azx_writeb(chip, RIRBSTS, RIRB_INT_MASK);
}
} while (active && ++repeat < 10);
diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h
index 82e2644..9da7a06 100644
--- a/sound/pci/hda/hda_controller.h
+++ b/sound/pci/hda/hda_controller.h
@@ -143,6 +143,7 @@
unsigned int align_buffer_size:1;
unsigned int region_requested:1;
unsigned int disabled:1; /* disabled by vga_switcheroo */
+ unsigned int pm_prepared:1;
/* GTS present */
unsigned int gts_present:1;
diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c
index d081fb2..82cf1da 100644
--- a/sound/pci/hda/hda_eld.c
+++ b/sound/pci/hda/hda_eld.c
@@ -360,7 +360,7 @@
for (i = 0, j = 0; i < ARRAY_SIZE(alsa_rates); i++)
if (pcm & (1 << i))
- j += snprintf(buf + j, buflen - j, " %d",
+ j += scnprintf(buf + j, buflen - j, " %d",
alsa_rates[i]);
buf[j] = '\0'; /* necessary when j == 0 */
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 10d5023..e92fcb1 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -813,7 +813,7 @@
}
}
-/* sync power of each widget in the the given path */
+/* sync power of each widget in the given path */
static hda_nid_t path_power_update(struct hda_codec *codec,
struct nid_path *path,
bool allow_powerdown)
@@ -1202,11 +1202,17 @@
*index = ch;
return "Headphone";
case AUTO_PIN_LINE_OUT:
- /* This deals with the case where we have two DACs and
- * one LO, one HP and one Speaker */
- if (!ch && cfg->speaker_outs && cfg->hp_outs) {
- bool hp_lo_shared = !path_has_mixer(codec, spec->hp_paths[0], ctl_type);
- bool spk_lo_shared = !path_has_mixer(codec, spec->speaker_paths[0], ctl_type);
+ /* This deals with the case where one HP or one Speaker or
+ * one HP + one Speaker need to share the DAC with LO
+ */
+ if (!ch) {
+ bool hp_lo_shared = false, spk_lo_shared = false;
+
+ if (cfg->speaker_outs)
+ spk_lo_shared = !path_has_mixer(codec,
+ spec->speaker_paths[0], ctl_type);
+ if (cfg->hp_outs)
+ hp_lo_shared = !path_has_mixer(codec, spec->hp_paths[0], ctl_type);
if (hp_lo_shared && spk_lo_shared)
return spec->vmaster_mute.hook ? "PCM" : "Master";
if (hp_lo_shared)
@@ -1364,16 +1370,20 @@
struct nid_path *path;
hda_nid_t pin = pins[i];
- path = snd_hda_get_path_from_idx(codec, path_idx[i]);
- if (path) {
- badness += assign_out_path_ctls(codec, path);
- continue;
+ if (!spec->obey_preferred_dacs) {
+ path = snd_hda_get_path_from_idx(codec, path_idx[i]);
+ if (path) {
+ badness += assign_out_path_ctls(codec, path);
+ continue;
+ }
}
dacs[i] = get_preferred_dac(codec, pin);
if (dacs[i]) {
if (is_dac_already_used(codec, dacs[i]))
badness += bad->shared_primary;
+ } else if (spec->obey_preferred_dacs) {
+ badness += BAD_NO_PRIMARY_DAC;
}
if (!dacs[i])
@@ -3448,7 +3458,7 @@
struct hda_gen_spec *spec = codec->spec;
const struct hda_input_mux *imux;
struct nid_path *path;
- int i, adc_idx, err = 0;
+ int i, adc_idx, ret, err = 0;
imux = &spec->input_mux;
adc_idx = kcontrol->id.index;
@@ -3458,9 +3468,13 @@
if (!path || !path->ctls[type])
continue;
kcontrol->private_value = path->ctls[type];
- err = func(kcontrol, ucontrol);
- if (err < 0)
+ ret = func(kcontrol, ucontrol);
+ if (ret < 0) {
+ err = ret;
break;
+ }
+ if (ret > 0)
+ err = 1;
}
mutex_unlock(&codec->control_mutex);
if (err >= 0 && spec->cap_sync_hook)
@@ -4013,7 +4027,7 @@
spec->micmute_led.led_mode = MICMUTE_LED_FOLLOW_MUTE;
spec->micmute_led.capture = 0;
- spec->micmute_led.led_value = 0;
+ spec->micmute_led.led_value = -1;
spec->micmute_led.old_hook = spec->cap_sync_hook;
spec->micmute_led.update = hook;
spec->cap_sync_hook = update_micmute_led;
@@ -4401,7 +4415,7 @@
*/
/* check each pin in the given array; returns true if any of them is plugged */
-static bool detect_jacks(struct hda_codec *codec, int num_pins, hda_nid_t *pins)
+static bool detect_jacks(struct hda_codec *codec, int num_pins, const hda_nid_t *pins)
{
int i;
bool present = false;
@@ -4420,7 +4434,7 @@
}
/* standard HP/line-out auto-mute helper */
-static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins,
+static void do_automute(struct hda_codec *codec, int num_pins, const hda_nid_t *pins,
int *paths, bool mute)
{
struct hda_gen_spec *spec = codec->spec;
@@ -6027,7 +6041,7 @@
/* call init functions of standard auto-mute helpers */
update_automute_all(codec);
- regcache_sync(codec->core.regmap);
+ snd_hda_regmap_sync(codec);
if (spec->vmaster_mute.sw_kctl && spec->vmaster_mute.hook)
snd_hda_sync_vmaster_hook(&spec->vmaster_mute);
diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h
index fb9f1a9..e728df6 100644
--- a/sound/pci/hda/hda_generic.h
+++ b/sound/pci/hda/hda_generic.h
@@ -236,6 +236,7 @@
unsigned int power_down_unused:1; /* power down unused widgets */
unsigned int dac_min_mute:1; /* minimal = mute for DACs */
unsigned int suppress_vmaster:1; /* don't create vmaster kctls */
+ unsigned int obey_preferred_dacs:1; /* obey preferred_dacs assignment */
/* other internal flags */
unsigned int no_analog:1; /* digital I/O only */
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index c524193..ebb1ee6 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -280,12 +280,13 @@
/* quirks for old Intel chipsets */
#define AZX_DCAPS_INTEL_ICH \
- (AZX_DCAPS_OLD_SSYNC | AZX_DCAPS_NO_ALIGN_BUFSIZE)
+ (AZX_DCAPS_OLD_SSYNC | AZX_DCAPS_NO_ALIGN_BUFSIZE |\
+ AZX_DCAPS_SYNC_WRITE)
/* quirks for Intel PCH */
#define AZX_DCAPS_INTEL_PCH_BASE \
(AZX_DCAPS_NO_ALIGN_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY |\
- AZX_DCAPS_SNOOP_TYPE(SCH))
+ AZX_DCAPS_SNOOP_TYPE(SCH) | AZX_DCAPS_SYNC_WRITE)
/* PCH up to IVB; no runtime PM; bind with i915 gfx */
#define AZX_DCAPS_INTEL_PCH_NOPM \
@@ -300,13 +301,13 @@
#define AZX_DCAPS_INTEL_HASWELL \
(/*AZX_DCAPS_ALIGN_BUFSIZE |*/ AZX_DCAPS_COUNT_LPIB_DELAY |\
AZX_DCAPS_PM_RUNTIME | AZX_DCAPS_I915_COMPONENT |\
- AZX_DCAPS_SNOOP_TYPE(SCH))
+ AZX_DCAPS_SNOOP_TYPE(SCH) | AZX_DCAPS_SYNC_WRITE)
/* Broadwell HDMI can't use position buffer reliably, force to use LPIB */
#define AZX_DCAPS_INTEL_BROADWELL \
(/*AZX_DCAPS_ALIGN_BUFSIZE |*/ AZX_DCAPS_POSFIX_LPIB |\
AZX_DCAPS_PM_RUNTIME | AZX_DCAPS_I915_COMPONENT |\
- AZX_DCAPS_SNOOP_TYPE(SCH))
+ AZX_DCAPS_SNOOP_TYPE(SCH) | AZX_DCAPS_SYNC_WRITE)
#define AZX_DCAPS_INTEL_BAYTRAIL \
(AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_I915_COMPONENT)
@@ -982,7 +983,7 @@
display_power(chip, false);
}
-static void __azx_runtime_resume(struct azx *chip, bool from_rt)
+static void __azx_runtime_resume(struct azx *chip)
{
struct hda_intel *hda = container_of(chip, struct hda_intel, chip);
struct hdac_bus *bus = azx_bus(chip);
@@ -999,11 +1000,15 @@
azx_init_pci(chip);
hda_intel_init_chip(chip, true);
- if (status && from_rt) {
- list_for_each_codec(codec, &chip->bus)
- if (status & (1 << codec->addr))
- schedule_delayed_work(&codec->jackpoll_work,
- codec->jackpoll_interval);
+ /* Avoid codec resume if runtime resume is for system suspend */
+ if (!chip->pm_prepared) {
+ list_for_each_codec(codec, &chip->bus) {
+ if (codec->relaxed_resume)
+ continue;
+
+ if (codec->forced_resume || (status & (1 << codec->addr)))
+ pm_request_resume(hda_codec_dev(codec));
+ }
}
/* power down again for link-controlled chips */
@@ -1012,6 +1017,39 @@
}
#ifdef CONFIG_PM_SLEEP
+static int azx_prepare(struct device *dev)
+{
+ struct snd_card *card = dev_get_drvdata(dev);
+ struct azx *chip;
+
+ if (!azx_is_pm_ready(card))
+ return 0;
+
+ chip = card->private_data;
+ chip->pm_prepared = 1;
+ snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+
+ flush_work(&azx_bus(chip)->unsol_work);
+
+ /* HDA controller always requires different WAKEEN for runtime suspend
+ * and system suspend, so don't use direct-complete here.
+ */
+ return 0;
+}
+
+static void azx_complete(struct device *dev)
+{
+ struct snd_card *card = dev_get_drvdata(dev);
+ struct azx *chip;
+
+ if (!azx_is_pm_ready(card))
+ return;
+
+ chip = card->private_data;
+ snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+ chip->pm_prepared = 0;
+}
+
static int azx_suspend(struct device *dev)
{
struct snd_card *card = dev_get_drvdata(dev);
@@ -1023,7 +1061,6 @@
chip = card->private_data;
bus = azx_bus(chip);
- snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
__azx_runtime_suspend(chip);
if (bus->irq >= 0) {
free_irq(bus->irq, chip);
@@ -1051,8 +1088,8 @@
chip->msi = 0;
if (azx_acquire_irq(chip, 1) < 0)
return -EIO;
- __azx_runtime_resume(chip, false);
- snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+
+ __azx_runtime_resume(chip);
trace_azx_resume(chip);
return 0;
@@ -1067,6 +1104,8 @@
struct azx *chip = card->private_data;
struct pci_dev *pci = to_pci_dev(dev);
+ if (!azx_is_pm_ready(card))
+ return 0;
if (chip->driver_type == AZX_DRIVER_SKL)
pci_set_power_state(pci, PCI_D3hot);
@@ -1079,6 +1118,8 @@
struct azx *chip = card->private_data;
struct pci_dev *pci = to_pci_dev(dev);
+ if (!azx_is_pm_ready(card))
+ return 0;
if (chip->driver_type == AZX_DRIVER_SKL)
pci_set_power_state(pci, PCI_D0);
@@ -1094,12 +1135,9 @@
if (!azx_is_pm_ready(card))
return 0;
chip = card->private_data;
- if (!azx_has_pm_runtime(chip))
- return 0;
/* enable controller wake up event */
- azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) |
- STATESTS_INT_MASK);
+ azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) | STATESTS_INT_MASK);
__azx_runtime_suspend(chip);
trace_azx_runtime_suspend(chip);
@@ -1114,13 +1152,10 @@
if (!azx_is_pm_ready(card))
return 0;
chip = card->private_data;
- if (!azx_has_pm_runtime(chip))
- return 0;
- __azx_runtime_resume(chip, true);
+ __azx_runtime_resume(chip);
/* disable controller Wake Up event*/
- azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) &
- ~STATESTS_INT_MASK);
+ azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) & ~STATESTS_INT_MASK);
trace_azx_runtime_resume(chip);
return 0;
@@ -1154,6 +1189,8 @@
static const struct dev_pm_ops azx_pm = {
SET_SYSTEM_SLEEP_PM_OPS(azx_suspend, azx_resume)
#ifdef CONFIG_PM_SLEEP
+ .prepare = azx_prepare,
+ .complete = azx_complete,
.freeze_noirq = azx_freeze_noirq,
.thaw_noirq = azx_thaw_noirq,
#endif
@@ -1195,10 +1232,8 @@
if (!disabled) {
dev_info(chip->card->dev,
"Start delayed initialization\n");
- if (azx_probe_continue(chip) < 0) {
+ if (azx_probe_continue(chip) < 0)
dev_err(chip->card->dev, "initialization error\n");
- hda->init_failed = true;
- }
}
} else {
dev_info(chip->card->dev, "%s via vga_switcheroo\n",
@@ -1280,11 +1315,17 @@
{
struct hda_intel *hda = container_of(chip, struct hda_intel, chip);
struct pci_dev *p = get_bound_vga(chip->pci);
+ struct pci_dev *parent;
if (p) {
dev_info(chip->card->dev,
"Handle vga_switcheroo audio client\n");
hda->use_vga_switcheroo = 1;
- chip->bus.keep_power = 1; /* cleared in either gpu_bound op or codec probe */
+
+ /* cleared in either gpu_bound op or codec probe, or when its
+ * upstream port has _PR3 (i.e. dGPU).
+ */
+ parent = pci_upstream_bridge(p);
+ chip->bus.keep_power = parent ? !pci_pr3_present(parent) : 1;
chip->driver_caps |= AZX_DCAPS_PM_RUNTIME;
pci_dev_put(p);
}
@@ -1325,12 +1366,15 @@
/*
* destructor
*/
-static int azx_free(struct azx *chip)
+static void azx_free(struct azx *chip)
{
struct pci_dev *pci = chip->pci;
struct hda_intel *hda = container_of(chip, struct hda_intel, chip);
struct hdac_bus *bus = azx_bus(chip);
+ if (hda->freed)
+ return;
+
if (azx_has_pm_runtime(chip) && chip->running)
pm_runtime_get_noresume(&pci->dev);
chip->running = 0;
@@ -1374,22 +1418,25 @@
if (chip->driver_caps & AZX_DCAPS_I915_COMPONENT)
snd_hdac_i915_exit(bus);
- kfree(hda);
- return 0;
+ hda->freed = 1;
}
static int azx_dev_disconnect(struct snd_device *device)
{
struct azx *chip = device->device_data;
+ struct hdac_bus *bus = azx_bus(chip);
chip->bus.shutdown = 1;
+ cancel_work_sync(&bus->unsol_work);
+
return 0;
}
static int azx_dev_free(struct snd_device *device)
{
- return azx_free(device->device_data);
+ azx_free(device->device_data);
+ return 0;
}
#ifdef SUPPORT_VGA_SWITCHEROO
@@ -1703,7 +1750,7 @@
if (err < 0)
return err;
- hda = kzalloc(sizeof(*hda), GFP_KERNEL);
+ hda = devm_kzalloc(&pci->dev, sizeof(*hda), GFP_KERNEL);
if (!hda) {
pci_disable_device(pci);
return -ENOMEM;
@@ -1744,7 +1791,6 @@
err = azx_bus_init(chip, model[dev]);
if (err < 0) {
- kfree(hda);
pci_disable_device(pci);
return err;
}
@@ -1944,7 +1990,7 @@
/* codec detection */
if (!azx_bus(chip)->codec_mask) {
dev_err(card->dev, "no codecs found!\n");
- return -ENODEV;
+ /* keep running the rest for the runtime PM */
}
if (azx_acquire_irq(chip, 0) < 0)
@@ -1966,24 +2012,15 @@
{
struct snd_card *card = context;
struct azx *chip = card->private_data;
- struct pci_dev *pci = chip->pci;
- if (!fw) {
- dev_err(card->dev, "Cannot load firmware, aborting\n");
- goto error;
- }
-
- chip->fw = fw;
+ if (fw)
+ chip->fw = fw;
+ else
+ dev_err(card->dev, "Cannot load firmware, continue without patching\n");
if (!chip->disabled) {
/* continue probing */
- if (azx_probe_continue(chip))
- goto error;
+ azx_probe_continue(chip);
}
- return; /* OK */
-
- error:
- snd_card_free(card);
- pci_set_drvdata(pci, NULL);
}
#endif
@@ -2014,6 +2051,17 @@
#endif
}
+/* Blacklist for skipping the whole probe:
+ * some HD-audio PCI entries are exposed without any codecs, and such devices
+ * should be ignored from the beginning.
+ */
+static const struct pci_device_id driver_blacklist[] = {
+ { PCI_DEVICE_SUB(0x1022, 0x1487, 0x1043, 0x874f) }, /* ASUS ROG Zenith II / Strix */
+ { PCI_DEVICE_SUB(0x1022, 0x1487, 0x1462, 0xcb59) }, /* MSI TRX40 Creator */
+ { PCI_DEVICE_SUB(0x1022, 0x1487, 0x1462, 0xcb60) }, /* MSI TRX40 */
+ {}
+};
+
static const struct hda_controller_ops pci_hda_ops = {
.disable_msi_reset_irq = disable_msi_reset_irq,
.pcm_mmap_prepare = pcm_mmap_prepare,
@@ -2049,6 +2097,11 @@
bool schedule_probe;
int err;
+ if (pci_match_id(driver_blacklist, pci)) {
+ dev_info(&pci->dev, "Skipping the blacklisted device\n");
+ return -ENODEV;
+ }
+
if (dev >= SNDRV_CARDS)
return -ENODEV;
if (!enable[dev]) {
@@ -2143,8 +2196,8 @@
SND_PCI_QUIRK(0x1849, 0x7662, "Asrock H81M-HDS", 0),
/* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */
SND_PCI_QUIRK(0x1043, 0x8733, "Asus Prime X370-Pro", 0),
- /* https://bugzilla.redhat.com/show_bug.cgi?id=1581607 */
- SND_PCI_QUIRK(0x1558, 0x3501, "Clevo W35xSS_370SS", 0),
+ /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */
+ SND_PCI_QUIRK(0x1558, 0x6504, "Clevo W65_67SB", 0),
/* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */
SND_PCI_QUIRK(0x1028, 0x0497, "Dell Precision T3600", 0),
/* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */
@@ -2246,9 +2299,11 @@
#endif
/* create codec instances */
- err = azx_probe_codecs(chip, azx_max_codecs[chip->driver_type]);
- if (err < 0)
- goto out_free;
+ if (bus->codec_mask) {
+ err = azx_probe_codecs(chip, azx_max_codecs[chip->driver_type]);
+ if (err < 0)
+ goto out_free;
+ }
#ifdef CONFIG_SND_HDA_PATCH_LOADER
if (chip->fw) {
@@ -2262,7 +2317,7 @@
#endif
}
#endif
- if ((probe_only[dev] & 1) == 0) {
+ if (bus->codec_mask && !(probe_only[dev] & 1)) {
err = azx_codec_configure(chip);
if (err < 0)
goto out_free;
@@ -2279,17 +2334,23 @@
set_default_power_save(chip);
- if (azx_has_pm_runtime(chip))
+ if (azx_has_pm_runtime(chip)) {
+ pm_runtime_use_autosuspend(&pci->dev);
+ pm_runtime_allow(&pci->dev);
pm_runtime_put_autosuspend(&pci->dev);
+ }
out_free:
- if (err < 0 || !hda->need_i915_power)
+ if (err < 0) {
+ azx_free(chip);
+ return err;
+ }
+
+ if (!hda->need_i915_power)
display_power(chip, false);
- if (err < 0)
- hda->init_failed = 1;
complete_all(&hda->probe_wait);
to_hda_bus(bus)->bus_probing = 0;
- return err;
+ return 0;
}
static void azx_remove(struct pci_dev *pci)
@@ -2396,21 +2457,36 @@
/* CometLake-H */
{ PCI_DEVICE(0x8086, 0x06C8),
.driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
+ { PCI_DEVICE(0x8086, 0xf1c8),
+ .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
/* CometLake-S */
{ PCI_DEVICE(0x8086, 0xa3f0),
.driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
+ /* CometLake-R */
+ { PCI_DEVICE(0x8086, 0xf0c8),
+ .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
/* Icelake */
{ PCI_DEVICE(0x8086, 0x34c8),
.driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
+ /* Icelake-H */
+ { PCI_DEVICE(0x8086, 0x3dc8),
+ .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
/* Jasperlake */
{ PCI_DEVICE(0x8086, 0x38c8),
.driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
+ { PCI_DEVICE(0x8086, 0x4dc8),
+ .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
/* Tigerlake */
{ PCI_DEVICE(0x8086, 0xa0c8),
.driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
+ /* Tigerlake-H */
+ { PCI_DEVICE(0x8086, 0x43c8),
+ .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
/* Elkhart Lake */
{ PCI_DEVICE(0x8086, 0x4b55),
.driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
+ { PCI_DEVICE(0x8086, 0x4b58),
+ .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
/* Broxton-P(Apollolake) */
{ PCI_DEVICE(0x8086, 0x5a98),
.driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_BROXTON },
diff --git a/sound/pci/hda/hda_intel.h b/sound/pci/hda/hda_intel.h
index 2acfff3..3fb119f 100644
--- a/sound/pci/hda/hda_intel.h
+++ b/sound/pci/hda/hda_intel.h
@@ -27,6 +27,7 @@
unsigned int use_vga_switcheroo:1;
unsigned int vga_switcheroo_registered:1;
unsigned int init_failed:1; /* delayed init failed */
+ unsigned int freed:1; /* resources already released */
bool need_i915_power:1; /* the hda controller needs i915 power */
};
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index 3942e1b..3dca65d 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -138,6 +138,8 @@
void snd_hda_codec_register(struct hda_codec *codec);
void snd_hda_codec_cleanup_for_unbind(struct hda_codec *codec);
+#define snd_hda_regmap_sync(codec) snd_hdac_regmap_sync(&(codec)->core)
+
enum {
HDA_VMUTE_OFF,
HDA_VMUTE_ON,
diff --git a/sound/pci/hda/hda_sysfs.c b/sound/pci/hda/hda_sysfs.c
index fcc3441..91b4a29 100644
--- a/sound/pci/hda/hda_sysfs.c
+++ b/sound/pci/hda/hda_sysfs.c
@@ -139,7 +139,7 @@
"The codec is being used, can't reconfigure.\n");
goto error;
}
- err = snd_hda_codec_configure(codec);
+ err = device_reprobe(hda_codec_dev(codec));
if (err < 0)
goto error;
err = snd_card_register(codec->card);
@@ -222,7 +222,7 @@
int i, len = 0;
mutex_lock(&codec->user_mutex);
snd_array_for_each(&codec->init_verbs, i, v) {
- len += snprintf(buf + len, PAGE_SIZE - len,
+ len += scnprintf(buf + len, PAGE_SIZE - len,
"0x%02x 0x%03x 0x%04x\n",
v->nid, v->verb, v->param);
}
@@ -272,7 +272,7 @@
int i, len = 0;
mutex_lock(&codec->user_mutex);
snd_array_for_each(&codec->hints, i, hint) {
- len += snprintf(buf + len, PAGE_SIZE - len,
+ len += scnprintf(buf + len, PAGE_SIZE - len,
"%s = %s\n", hint->key, hint->val);
}
mutex_unlock(&codec->user_mutex);
diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c
index 8350954..2971b34 100644
--- a/sound/pci/hda/hda_tegra.c
+++ b/sound/pci/hda/hda_tegra.c
@@ -169,6 +169,10 @@
struct hdac_bus *bus = azx_bus(chip);
if (chip && chip->running) {
+ /* enable controller wake up event */
+ azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) |
+ STATESTS_INT_MASK);
+
azx_stop_chip(chip);
synchronize_irq(bus->irq);
azx_enter_link_reset(chip);
@@ -191,6 +195,9 @@
if (chip && chip->running) {
hda_tegra_init(hda);
azx_init_chip(chip, 1);
+ /* disable controller wake up event*/
+ azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) &
+ ~STATESTS_INT_MASK);
}
return 0;
@@ -285,6 +292,9 @@
const char *sname, *drv_name = "tegra-hda";
struct device_node *np = pdev->dev.of_node;
+ if (irq_id < 0)
+ return irq_id;
+
err = hda_tegra_init_chip(chip, pdev);
if (err)
return err;
@@ -398,6 +408,7 @@
return err;
chip->bus.needs_damn_long_delay = 1;
+ chip->bus.core.aligned_mmio = 1;
err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
if (err < 0) {
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index bc9dd8e..c64895f 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -389,7 +389,7 @@
{
int err;
struct ad198x_spec *spec;
- static hda_nid_t preferred_pairs[] = {
+ static const hda_nid_t preferred_pairs[] = {
0x1a, 0x03,
0x1b, 0x03,
0x1c, 0x04,
@@ -519,9 +519,9 @@
static int patch_ad1983(struct hda_codec *codec)
{
+ static const hda_nid_t conn_0c[] = { 0x08 };
+ static const hda_nid_t conn_0d[] = { 0x09 };
struct ad198x_spec *spec;
- static hda_nid_t conn_0c[] = { 0x08 };
- static hda_nid_t conn_0d[] = { 0x09 };
int err;
err = alloc_ad_spec(codec);
diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index b7a1abb..9412bdd 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -93,7 +93,7 @@
};
/* Strings for Input Source Enum Control */
-static const char *const in_src_str[3] = {"Rear Mic", "Line", "Front Mic" };
+static const char *const in_src_str[3] = { "Microphone", "Line In", "Front Microphone" };
#define IN_SRC_NUM_OF_INPUTS 3
enum {
REAR_MIC,
@@ -1065,6 +1065,7 @@
QUIRK_R3DI,
QUIRK_R3D,
QUIRK_AE5,
+ QUIRK_AE7,
};
#ifdef CONFIG_PCI
@@ -1146,7 +1147,7 @@
{ 0x0e, 0x01c510f0 }, /* SPDIF In */
{ 0x0f, 0x01017114 }, /* Port A -- Rear L/R. */
{ 0x10, 0x01017012 }, /* Port D -- Center/LFE or FP Hp */
- { 0x11, 0x01a170ff }, /* Port B -- LineMicIn2 / Rear Headphone */
+ { 0x11, 0x012170ff }, /* Port B -- LineMicIn2 / Rear Headphone */
{ 0x12, 0x01a170f0 }, /* Port C -- LineIn1 */
{ 0x13, 0x908700f0 }, /* What U Hear In*/
{ 0x18, 0x50d000f0 }, /* N/A */
@@ -1180,8 +1181,12 @@
SND_PCI_QUIRK(0x1458, 0xA016, "Recon3Di", QUIRK_R3DI),
SND_PCI_QUIRK(0x1458, 0xA026, "Gigabyte G1.Sniper Z97", QUIRK_R3DI),
SND_PCI_QUIRK(0x1458, 0xA036, "Gigabyte GA-Z170X-Gaming 7", QUIRK_R3DI),
+ SND_PCI_QUIRK(0x3842, 0x1038, "EVGA X99 Classified", QUIRK_R3DI),
SND_PCI_QUIRK(0x1102, 0x0013, "Recon3D", QUIRK_R3D),
+ SND_PCI_QUIRK(0x1102, 0x0018, "Recon3D", QUIRK_R3D),
SND_PCI_QUIRK(0x1102, 0x0051, "Sound Blaster AE-5", QUIRK_AE5),
+ SND_PCI_QUIRK(0x1102, 0x0191, "Sound Blaster AE-5 Plus", QUIRK_AE5),
+ SND_PCI_QUIRK(0x1102, 0x0081, "Sound Blaster AE-7", QUIRK_AE7),
{}
};
@@ -1809,13 +1814,14 @@
static void dspio_clear_response_queue(struct hda_codec *codec)
{
+ unsigned long timeout = jiffies + msecs_to_jiffies(1000);
unsigned int dummy = 0;
- int status = -1;
+ int status;
/* clear all from the response queue */
do {
status = dspio_read(codec, &dummy);
- } while (status == 0);
+ } while (status == 0 && time_before(jiffies, timeout));
}
static int dspio_get_response_data(struct hda_codec *codec)
@@ -4668,9 +4674,18 @@
tmp = FLOAT_ONE;
break;
case QUIRK_AE5:
- ca0113_mmio_command_set(codec, 0x48, 0x28, 0x00);
+ ca0113_mmio_command_set(codec, 0x30, 0x28, 0x00);
tmp = FLOAT_THREE;
break;
+ case QUIRK_AE7:
+ ca0113_mmio_command_set(codec, 0x30, 0x28, 0x00);
+ tmp = FLOAT_THREE;
+ chipio_set_conn_rate(codec, MEM_CONNID_MICIN2,
+ SR_96_000);
+ chipio_set_conn_rate(codec, MEM_CONNID_MICOUT2,
+ SR_96_000);
+ dspio_set_uint_param(codec, 0x80, 0x01, FLOAT_ZERO);
+ break;
default:
tmp = FLOAT_ONE;
break;
@@ -4714,7 +4729,15 @@
r3di_gpio_mic_set(codec, R3DI_REAR_MIC);
break;
case QUIRK_AE5:
- ca0113_mmio_command_set(codec, 0x48, 0x28, 0x00);
+ ca0113_mmio_command_set(codec, 0x30, 0x28, 0x00);
+ break;
+ case QUIRK_AE7:
+ ca0113_mmio_command_set(codec, 0x30, 0x28, 0x3f);
+ chipio_set_conn_rate(codec, MEM_CONNID_MICIN2,
+ SR_96_000);
+ chipio_set_conn_rate(codec, MEM_CONNID_MICOUT2,
+ SR_96_000);
+ dspio_set_uint_param(codec, 0x80, 0x01, FLOAT_ZERO);
break;
default:
break;
@@ -4725,7 +4748,10 @@
if (ca0132_quirk(spec) == QUIRK_R3DI)
chipio_set_conn_rate(codec, 0x0F, SR_96_000);
- tmp = FLOAT_ZERO;
+ if (ca0132_quirk(spec) == QUIRK_AE7)
+ tmp = FLOAT_THREE;
+ else
+ tmp = FLOAT_ZERO;
dspio_set_uint_param(codec, 0x80, 0x00, tmp);
switch (ca0132_quirk(spec)) {
@@ -4753,7 +4779,7 @@
tmp = FLOAT_ONE;
break;
case QUIRK_AE5:
- ca0113_mmio_command_set(codec, 0x48, 0x28, 0x3f);
+ ca0113_mmio_command_set(codec, 0x30, 0x28, 0x3f);
tmp = FLOAT_THREE;
break;
default:
@@ -5745,6 +5771,11 @@
return 0;
}
+ if (nid == ZXR_HEADPHONE_GAIN) {
+ *valp = spec->zxr_gain_set;
+ return 0;
+ }
+
return 0;
}
@@ -7588,12 +7619,14 @@
struct ca0132_spec *spec = codec->spec;
codec_dbg(codec, "ca0132_process_dsp_response\n");
+ snd_hda_power_up_pm(codec);
if (spec->wait_scp) {
if (dspio_get_response_data(codec) >= 0)
spec->wait_scp = 0;
}
dspio_clear_response_queue(codec);
+ snd_hda_power_down_pm(codec);
}
static void hp_callback(struct hda_codec *codec, struct hda_jack_callback *cb)
@@ -7604,11 +7637,10 @@
/* Delay enabling the HP amp, to let the mic-detection
* state machine run.
*/
- cancel_delayed_work(&spec->unsol_hp_work);
- schedule_delayed_work(&spec->unsol_hp_work, msecs_to_jiffies(500));
tbl = snd_hda_jack_tbl_get(codec, cb->nid);
if (tbl)
tbl->block_report = 1;
+ schedule_delayed_work(&spec->unsol_hp_work, msecs_to_jiffies(500));
}
static void amic_callback(struct hda_codec *codec, struct hda_jack_callback *cb)
@@ -7800,23 +7832,23 @@
static void sbz_set_pin_ctl_default(struct hda_codec *codec)
{
- hda_nid_t pins[5] = {0x0B, 0x0C, 0x0E, 0x12, 0x13};
+ static const hda_nid_t pins[] = {0x0B, 0x0C, 0x0E, 0x12, 0x13};
unsigned int i;
snd_hda_codec_write(codec, 0x11, 0,
AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40);
- for (i = 0; i < 5; i++)
+ for (i = 0; i < ARRAY_SIZE(pins); i++)
snd_hda_codec_write(codec, pins[i], 0,
AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00);
}
static void ca0132_clear_unsolicited(struct hda_codec *codec)
{
- hda_nid_t pins[7] = {0x0B, 0x0E, 0x0F, 0x10, 0x11, 0x12, 0x13};
+ static const hda_nid_t pins[] = {0x0B, 0x0E, 0x0F, 0x10, 0x11, 0x12, 0x13};
unsigned int i;
- for (i = 0; i < 7; i++) {
+ for (i = 0; i < ARRAY_SIZE(pins); i++) {
snd_hda_codec_write(codec, pins[i], 0,
AC_VERB_SET_UNSOLICITED_ENABLE, 0x00);
}
@@ -7840,10 +7872,10 @@
static void zxr_dbpro_power_state_shutdown(struct hda_codec *codec)
{
- hda_nid_t pins[7] = {0x05, 0x0c, 0x09, 0x0e, 0x08, 0x11, 0x01};
+ static const hda_nid_t pins[] = {0x05, 0x0c, 0x09, 0x0e, 0x08, 0x11, 0x01};
unsigned int i;
- for (i = 0; i < 7; i++)
+ for (i = 0; i < ARRAY_SIZE(pins); i++)
snd_hda_codec_write(codec, pins[i], 0,
AC_VERB_SET_POWER_STATE, 0x03);
}
@@ -8454,12 +8486,25 @@
codec->patch_ops.free(codec);
}
+#ifdef CONFIG_PM
+static int ca0132_suspend(struct hda_codec *codec)
+{
+ struct ca0132_spec *spec = codec->spec;
+
+ cancel_delayed_work_sync(&spec->unsol_hp_work);
+ return 0;
+}
+#endif
+
static const struct hda_codec_ops ca0132_patch_ops = {
.build_controls = ca0132_build_controls,
.build_pcms = ca0132_build_pcms,
.init = ca0132_init,
.free = ca0132_free,
.unsol_event = snd_hda_jack_unsol_event,
+#ifdef CONFIG_PM
+ .suspend = ca0132_suspend,
+#endif
.reboot_notify = ca0132_reboot_notify,
};
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 968d3ca..5e2fadb 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -116,7 +116,7 @@
}
static void cx_auto_turn_eapd(struct hda_codec *codec, int num_pins,
- hda_nid_t *pins, bool on)
+ const hda_nid_t *pins, bool on)
{
int i;
for (i = 0; i < num_pins; i++) {
@@ -898,18 +898,19 @@
SND_PCI_QUIRK(0x103c, 0x8079, "HP EliteBook 840 G3", CXT_FIXUP_HP_DOCK),
SND_PCI_QUIRK(0x103c, 0x807C, "HP EliteBook 820 G3", CXT_FIXUP_HP_DOCK),
SND_PCI_QUIRK(0x103c, 0x80FD, "HP ProBook 640 G2", CXT_FIXUP_HP_DOCK),
+ SND_PCI_QUIRK(0x103c, 0x8115, "HP Z1 Gen3", CXT_FIXUP_HP_GATE_MIC),
+ SND_PCI_QUIRK(0x103c, 0x814f, "HP ZBook 15u G3", CXT_FIXUP_MUTE_LED_GPIO),
+ SND_PCI_QUIRK(0x103c, 0x8174, "HP Spectre x360", CXT_FIXUP_HP_SPECTRE),
+ SND_PCI_QUIRK(0x103c, 0x822e, "HP ProBook 440 G4", CXT_FIXUP_MUTE_LED_GPIO),
SND_PCI_QUIRK(0x103c, 0x828c, "HP EliteBook 840 G4", CXT_FIXUP_HP_DOCK),
+ SND_PCI_QUIRK(0x103c, 0x8299, "HP 800 G3 SFF", CXT_FIXUP_HP_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x103c, 0x829a, "HP 800 G3 DM", CXT_FIXUP_HP_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x103c, 0x836e, "HP ProBook 455 G5", CXT_FIXUP_MUTE_LED_GPIO),
+ SND_PCI_QUIRK(0x103c, 0x837f, "HP ProBook 470 G5", CXT_FIXUP_MUTE_LED_GPIO),
SND_PCI_QUIRK(0x103c, 0x83b2, "HP EliteBook 840 G5", CXT_FIXUP_HP_DOCK),
SND_PCI_QUIRK(0x103c, 0x83b3, "HP EliteBook 830 G5", CXT_FIXUP_HP_DOCK),
SND_PCI_QUIRK(0x103c, 0x83d3, "HP ProBook 640 G4", CXT_FIXUP_HP_DOCK),
- SND_PCI_QUIRK(0x103c, 0x8174, "HP Spectre x360", CXT_FIXUP_HP_SPECTRE),
- SND_PCI_QUIRK(0x103c, 0x8115, "HP Z1 Gen3", CXT_FIXUP_HP_GATE_MIC),
- SND_PCI_QUIRK(0x103c, 0x814f, "HP ZBook 15u G3", CXT_FIXUP_MUTE_LED_GPIO),
- SND_PCI_QUIRK(0x103c, 0x822e, "HP ProBook 440 G4", CXT_FIXUP_MUTE_LED_GPIO),
- SND_PCI_QUIRK(0x103c, 0x836e, "HP ProBook 455 G5", CXT_FIXUP_MUTE_LED_GPIO),
- SND_PCI_QUIRK(0x103c, 0x837f, "HP ProBook 470 G5", CXT_FIXUP_MUTE_LED_GPIO),
- SND_PCI_QUIRK(0x103c, 0x8299, "HP 800 G3 SFF", CXT_FIXUP_HP_MIC_NO_PRESENCE),
- SND_PCI_QUIRK(0x103c, 0x829a, "HP 800 G3 DM", CXT_FIXUP_HP_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x103c, 0x8402, "HP ProBook 645 G4", CXT_FIXUP_MUTE_LED_GPIO),
SND_PCI_QUIRK(0x103c, 0x8455, "HP Z2 G4", CXT_FIXUP_HP_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x103c, 0x8456, "HP Z2 G4 SFF", CXT_FIXUP_HP_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x103c, 0x8457, "HP Z2 G4 mini", CXT_FIXUP_HP_MIC_NO_PRESENCE),
@@ -921,6 +922,7 @@
SND_PCI_QUIRK(0x17aa, 0x215f, "Lenovo T510", CXT_PINCFG_LENOVO_TP410),
SND_PCI_QUIRK(0x17aa, 0x21ce, "Lenovo T420", CXT_PINCFG_LENOVO_TP410),
SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520", CXT_PINCFG_LENOVO_TP410),
+ SND_PCI_QUIRK(0x17aa, 0x21d2, "Lenovo T420s", CXT_PINCFG_LENOVO_TP410),
SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT_PINCFG_LENOVO_TP410),
SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT_PINCFG_LENOVO_TP410),
SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo IdeaPad Z560", CXT_FIXUP_MUTE_LED_EAPD),
@@ -958,10 +960,10 @@
static void add_cx5051_fake_mutes(struct hda_codec *codec)
{
struct conexant_spec *spec = codec->spec;
- static hda_nid_t out_nids[] = {
+ static const hda_nid_t out_nids[] = {
0x10, 0x11, 0
};
- hda_nid_t *p;
+ const hda_nid_t *p;
for (p = out_nids; *p; p++)
snd_hda_override_amp_caps(codec, *p, HDA_OUTPUT,
@@ -1073,6 +1075,7 @@
static const struct hda_device_id snd_hda_id_conexant[] = {
HDA_CODEC_ENTRY(0x14f11f86, "CX8070", patch_conexant_auto),
HDA_CODEC_ENTRY(0x14f12008, "CX8200", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f120d0, "CX11970", patch_conexant_auto),
HDA_CODEC_ENTRY(0x14f15045, "CX20549 (Venice)", patch_conexant_auto),
HDA_CODEC_ENTRY(0x14f15047, "CX20551 (Waikiki)", patch_conexant_auto),
HDA_CODEC_ENTRY(0x14f15051, "CX20561 (Hermosa)", patch_conexant_auto),
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index d14f668..5128a5d 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -32,6 +32,7 @@
#include <sound/hda_codec.h>
#include "hda_local.h"
#include "hda_jack.h"
+#include "hda_controller.h"
static bool static_hdmi_pcm;
module_param(static_hdmi_pcm, bool, 0644);
@@ -56,6 +57,10 @@
#define is_cherryview(codec) ((codec)->core.vendor_id == 0x80862883)
#define is_valleyview_plus(codec) (is_valleyview(codec) || is_cherryview(codec))
+static bool enable_acomp = true;
+module_param(enable_acomp, bool, 0444);
+MODULE_PARM_DESC(enable_acomp, "Enable audio component binding (default=yes)");
+
struct hdmi_spec_per_cvt {
hda_nid_t cvt_nid;
int assigned;
@@ -171,6 +176,7 @@
bool use_jack_detect; /* jack detection enabled */
bool use_acomp_notifier; /* use eld_notify callback for hotplug */
bool acomp_registered; /* audio component registered in this driver */
+ bool force_connect; /* force connectivity */
struct drm_audio_component_audio_ops drm_audio_ops;
int (*port2pin)(struct hda_codec *, int); /* reverse port/pin mapping */
@@ -1240,6 +1246,10 @@
per_pin->cvt_nid = per_cvt->cvt_nid;
hinfo->nid = per_cvt->cvt_nid;
+ /* flip stripe flag for the assigned stream if supported */
+ if (get_wcaps(codec, per_cvt->cvt_nid) & AC_WCAP_STRIPE)
+ azx_stream(get_azx_dev(substream))->stripe = 1;
+
snd_hda_set_dev_select(codec, per_pin->pin_nid, per_pin->dev_id);
snd_hda_codec_write_cache(codec, per_pin->pin_nid, 0,
AC_VERB_SET_CONNECT_SEL,
@@ -1702,7 +1712,8 @@
* all device entries on the same pin
*/
config = snd_hda_codec_get_pincfg(codec, pin_nid);
- if (get_defcfg_connect(config) == AC_JACK_PORT_NONE)
+ if (get_defcfg_connect(config) == AC_JACK_PORT_NONE &&
+ !spec->force_connect)
return 0;
/*
@@ -1806,35 +1817,59 @@
return 0;
}
+static const struct snd_pci_quirk force_connect_list[] = {
+ SND_PCI_QUIRK(0x103c, 0x870f, "HP", 1),
+ SND_PCI_QUIRK(0x103c, 0x871a, "HP", 1),
+ SND_PCI_QUIRK(0x1462, 0xec94, "MS-7C94", 1),
+ {}
+};
+
static int hdmi_parse_codec(struct hda_codec *codec)
{
- hda_nid_t nid;
+ struct hdmi_spec *spec = codec->spec;
+ hda_nid_t start_nid;
+ unsigned int caps;
int i, nodes;
+ const struct snd_pci_quirk *q;
- nodes = snd_hda_get_sub_nodes(codec, codec->core.afg, &nid);
- if (!nid || nodes < 0) {
+ nodes = snd_hda_get_sub_nodes(codec, codec->core.afg, &start_nid);
+ if (!start_nid || nodes < 0) {
codec_warn(codec, "HDMI: failed to get afg sub nodes\n");
return -EINVAL;
}
- for (i = 0; i < nodes; i++, nid++) {
- unsigned int caps;
- unsigned int type;
+ q = snd_pci_quirk_lookup(codec->bus->pci, force_connect_list);
+
+ if (q && q->value)
+ spec->force_connect = true;
+
+ /*
+ * hdmi_add_pin() assumes total amount of converters to
+ * be known, so first discover all converters
+ */
+ for (i = 0; i < nodes; i++) {
+ hda_nid_t nid = start_nid + i;
caps = get_wcaps(codec, nid);
- type = get_wcaps_type(caps);
if (!(caps & AC_WCAP_DIGITAL))
continue;
- switch (type) {
- case AC_WID_AUD_OUT:
+ if (get_wcaps_type(caps) == AC_WID_AUD_OUT)
hdmi_add_cvt(codec, nid);
- break;
- case AC_WID_PIN:
+ }
+
+ /* discover audio pins */
+ for (i = 0; i < nodes; i++) {
+ hda_nid_t nid = start_nid + i;
+
+ caps = get_wcaps(codec, nid);
+
+ if (!(caps & AC_WCAP_DIGITAL))
+ continue;
+
+ if (get_wcaps_type(caps) == AC_WID_PIN)
hdmi_add_pin(codec, nid);
- break;
- }
}
return 0;
@@ -1852,8 +1887,10 @@
/* Add sanity check to pass klockwork check.
* This should never happen.
*/
- if (WARN_ON(spdif == NULL))
+ if (WARN_ON(spdif == NULL)) {
+ mutex_unlock(&codec->spdif_mutex);
return true;
+ }
non_pcm = !!(spdif->status & IEC958_AES0_NONAUDIO);
mutex_unlock(&codec->spdif_mutex);
return non_pcm;
@@ -1965,20 +2002,25 @@
int pinctl;
int err = 0;
+ mutex_lock(&spec->pcm_lock);
if (hinfo->nid) {
pcm_idx = hinfo_to_pcm_index(codec, hinfo);
- if (snd_BUG_ON(pcm_idx < 0))
- return -EINVAL;
+ if (snd_BUG_ON(pcm_idx < 0)) {
+ err = -EINVAL;
+ goto unlock;
+ }
cvt_idx = cvt_nid_to_cvt_index(codec, hinfo->nid);
- if (snd_BUG_ON(cvt_idx < 0))
- return -EINVAL;
+ if (snd_BUG_ON(cvt_idx < 0)) {
+ err = -EINVAL;
+ goto unlock;
+ }
per_cvt = get_cvt(spec, cvt_idx);
-
snd_BUG_ON(!per_cvt->assigned);
per_cvt->assigned = 0;
hinfo->nid = 0;
- mutex_lock(&spec->pcm_lock);
+ azx_stream(get_azx_dev(substream))->stripe = 0;
+
snd_hda_spdif_ctls_unassign(codec, pcm_idx);
clear_bit(pcm_idx, &spec->pcm_in_use);
pin_idx = hinfo_to_pin_index(codec, hinfo);
@@ -2006,10 +2048,11 @@
per_pin->setup = false;
per_pin->channels = 0;
mutex_unlock(&per_pin->lock);
- unlock:
- mutex_unlock(&spec->pcm_lock);
}
+unlock:
+ mutex_unlock(&spec->pcm_lock);
+
return err;
}
@@ -2221,7 +2264,9 @@
for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) {
struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx);
+ struct hdmi_eld *pin_eld = &per_pin->sink_eld;
+ pin_eld->eld_valid = false;
hdmi_present_sense(per_pin, 0);
}
@@ -2338,13 +2383,25 @@
}
#ifdef CONFIG_PM
+static int generic_hdmi_suspend(struct hda_codec *codec)
+{
+ struct hdmi_spec *spec = codec->spec;
+ int pin_idx;
+
+ for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) {
+ struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx);
+ cancel_delayed_work_sync(&per_pin->work);
+ }
+ return 0;
+}
+
static int generic_hdmi_resume(struct hda_codec *codec)
{
struct hdmi_spec *spec = codec->spec;
int pin_idx;
codec->patch_ops.init(codec);
- regcache_sync(codec->core.regmap);
+ snd_hda_regmap_sync(codec);
for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) {
struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx);
@@ -2361,6 +2418,7 @@
.build_controls = generic_hdmi_build_controls,
.unsol_event = hdmi_unsol_event,
#ifdef CONFIG_PM
+ .suspend = generic_hdmi_suspend,
.resume = generic_hdmi_resume,
#endif
};
@@ -2458,6 +2516,7 @@
mutex_lock(&spec->bind_lock);
spec->use_acomp_notifier = use_acomp;
spec->codec->relaxed_resume = use_acomp;
+ spec->codec->bus->keep_power = 0;
/* reprogram each jack detection logic depending on the notifier */
if (spec->use_jack_detect) {
for (i = 0; i < spec->num_pins; i++)
@@ -2509,7 +2568,7 @@
/* skip notification during system suspend (but not in runtime PM);
* the state will be updated at resume
*/
- if (snd_power_get_state(codec->card) != SNDRV_CTL_POWER_D0)
+ if (codec->core.dev.power.power_state.event == PM_EVENT_SUSPEND)
return;
/* ditto during suspend/resume process itself */
if (snd_hdac_is_in_pm(&codec->core))
@@ -2543,12 +2602,16 @@
{
struct hdmi_spec *spec = codec->spec;
+ if (!enable_acomp) {
+ codec_info(codec, "audio component disabled by module option\n");
+ return;
+ }
+
spec->port2pin = port2pin;
setup_drm_audio_ops(codec, ops);
if (!snd_hdac_acomp_init(&codec->bus->core, &spec->drm_audio_ops,
match_bound_vga, 0)) {
spec->acomp_registered = true;
- codec->bus->keep_power = 0;
}
}
@@ -2710,7 +2773,7 @@
/* skip notification during system suspend (but not in runtime PM);
* the state will be updated at resume
*/
- if (snd_power_get_state(codec->card) != SNDRV_CTL_POWER_D0)
+ if (codec->core.dev.power.power_state.event == PM_EVENT_SUSPEND)
return;
/* ditto during suspend/resume process itself */
if (snd_hdac_is_in_pm(&codec->core))
@@ -2753,6 +2816,7 @@
hda_nid_t cvt_nid)
{
if (per_pin) {
+ haswell_verify_D0(codec, per_pin->cvt_nid, per_pin->pin_nid);
snd_hda_set_dev_select(codec, per_pin->pin_nid,
per_pin->dev_id);
intel_verify_pin_cvt_connect(codec, per_pin);
@@ -2787,9 +2851,12 @@
/* parse and post-process for Intel codecs */
static int parse_intel_hdmi(struct hda_codec *codec)
{
- int err;
+ int err, retries = 3;
- err = hdmi_parse_codec(codec);
+ do {
+ err = hdmi_parse_codec(codec);
+ } while (err < 0 && retries--);
+
if (err < 0) {
generic_spec_free(codec);
return err;
@@ -3629,6 +3696,7 @@
static int patch_tegra_hdmi(struct hda_codec *codec)
{
+ struct hdmi_spec *spec;
int err;
err = patch_generic_hdmi(codec);
@@ -3636,6 +3704,10 @@
return err;
codec->patch_ops.build_pcms = tegra_hdmi_build_pcms;
+ spec = codec->spec;
+ spec->chmap.ops.chmap_cea_alloc_validate_get_type =
+ nvhdmi_chmap_cea_alloc_validate_get_type;
+ spec->chmap.ops.chmap_validate = nvhdmi_chmap_validate;
return 0;
}
@@ -4123,6 +4195,11 @@
HDA_CODEC_ENTRY(0x10de0097, "GPU 97 HDMI/DP", patch_nvhdmi),
HDA_CODEC_ENTRY(0x10de0098, "GPU 98 HDMI/DP", patch_nvhdmi),
HDA_CODEC_ENTRY(0x10de0099, "GPU 99 HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de009a, "GPU 9a HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de009d, "GPU 9d HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de009e, "GPU 9e HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de009f, "GPU 9f HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de00a0, "GPU a0 HDMI/DP", patch_nvhdmi),
HDA_CODEC_ENTRY(0x10de8001, "MCP73 HDMI", patch_nvhdmi_2ch),
HDA_CODEC_ENTRY(0x10de8067, "MCP67/68 HDMI", patch_nvhdmi_2ch),
HDA_CODEC_ENTRY(0x11069f80, "VX900 HDMI/DP", patch_via_hdmi),
@@ -4146,6 +4223,8 @@
HDA_CODEC_ENTRY(0x8086280d, "Geminilake HDMI", patch_i915_glk_hdmi),
HDA_CODEC_ENTRY(0x8086280f, "Icelake HDMI", patch_i915_icl_hdmi),
HDA_CODEC_ENTRY(0x80862812, "Tigerlake HDMI", patch_i915_tgl_hdmi),
+HDA_CODEC_ENTRY(0x80862816, "Rocketlake HDMI", patch_i915_tgl_hdmi),
+HDA_CODEC_ENTRY(0x8086281a, "Jasperlake HDMI", patch_i915_icl_hdmi),
HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI", patch_generic_hdmi),
HDA_CODEC_ENTRY(0x80862882, "Valleyview2 HDMI", patch_i915_byt_hdmi),
HDA_CODEC_ENTRY(0x80862883, "Braswell HDMI", patch_i915_byt_hdmi),
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 80f66ba..abe371c 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -81,11 +81,20 @@
/* mute LED for HP laptops, see alc269_fixup_mic_mute_hook() */
int mute_led_polarity;
+ int micmute_led_polarity;
hda_nid_t mute_led_nid;
hda_nid_t cap_mute_led_nid;
unsigned int gpio_mute_led_mask;
unsigned int gpio_mic_led_mask;
+ unsigned int mute_led_coef_idx;
+ unsigned int mute_led_coefbit_mask;
+ unsigned int mute_led_coefbit_on;
+ unsigned int mute_led_coefbit_off;
+ unsigned int mic_led_coef_idx;
+ unsigned int mic_led_coefbit_mask;
+ unsigned int mic_led_coefbit_on;
+ unsigned int mic_led_coefbit_off;
hda_nid_t headset_mic_pin;
hda_nid_t headphone_mic_pin;
@@ -107,6 +116,7 @@
unsigned int done_hp_init:1;
unsigned int no_shutup_pins:1;
unsigned int ultra_low_power:1;
+ unsigned int has_hs_key:1;
/* for PLL fix */
hda_nid_t pll_nid;
@@ -365,9 +375,11 @@
alc_update_coef_idx(codec, 0x67, 0xf000, 0x3000);
/* fallthrough */
case 0x10ec0215:
+ case 0x10ec0230:
case 0x10ec0233:
case 0x10ec0235:
case 0x10ec0236:
+ case 0x10ec0245:
case 0x10ec0255:
case 0x10ec0256:
case 0x10ec0257:
@@ -384,6 +396,10 @@
case 0x10ec0275:
alc_update_coef_idx(codec, 0xe, 0, 1<<0);
break;
+ case 0x10ec0287:
+ alc_update_coef_idx(codec, 0x10, 1<<9, 0);
+ alc_write_coef_idx(codec, 0x8, 0x4ab7);
+ break;
case 0x10ec0293:
alc_update_coef_idx(codec, 0xa, 1<<13, 0);
break;
@@ -409,6 +425,7 @@
case 0x10ec0672:
alc_update_coef_idx(codec, 0xd, 0, 1<<14); /* EAPD Ctrl */
break;
+ case 0x10ec0222:
case 0x10ec0623:
alc_update_coef_idx(codec, 0x19, 1<<13, 0);
break;
@@ -423,10 +440,12 @@
alc_update_coef_idx(codec, 0x7, 1<<5, 0);
break;
case 0x10ec0892:
+ case 0x10ec0897:
alc_update_coef_idx(codec, 0x7, 1<<5, 0);
break;
case 0x10ec0899:
case 0x10ec0900:
+ case 0x10ec0b00:
case 0x10ec1168:
case 0x10ec1220:
alc_update_coef_idx(codec, 0x7, 1<<1, 0);
@@ -461,10 +480,10 @@
static void alc_auto_setup_eapd(struct hda_codec *codec, bool on)
{
/* We currently only handle front, HP */
- static hda_nid_t pins[] = {
+ static const hda_nid_t pins[] = {
0x0f, 0x10, 0x14, 0x15, 0x17, 0
};
- hda_nid_t *p;
+ const hda_nid_t *p;
for (p = pins; *p; p++)
set_eapd(codec, *p, on);
}
@@ -498,6 +517,7 @@
struct alc_spec *spec = codec->spec;
switch (codec->core.vendor_id) {
+ case 0x10ec0283:
case 0x10ec0286:
case 0x10ec0288:
case 0x10ec0298:
@@ -785,9 +805,11 @@
{
if (!alc_subsystem_id(codec, ports)) {
struct alc_spec *spec = codec->spec;
- codec_dbg(codec,
- "realtek: Enable default setup for auto mode as fallback\n");
- spec->init_amp = ALC_INIT_DEFAULT;
+ if (spec->init_amp == ALC_INIT_UNDEFINED) {
+ codec_dbg(codec,
+ "realtek: Enable default setup for auto mode as fallback\n");
+ spec->init_amp = ALC_INIT_DEFAULT;
+ }
}
}
@@ -901,7 +923,7 @@
if (!spec->no_depop_delay)
msleep(150); /* to avoid pop noise */
codec->patch_ops.init(codec);
- regcache_sync(codec->core.regmap);
+ snd_hda_regmap_sync(codec);
hda_call_check_power_status(codec, 0x01);
return 0;
}
@@ -943,7 +965,7 @@
const char *name;
};
-static struct alc_codec_rename_table rename_tbl[] = {
+static const struct alc_codec_rename_table rename_tbl[] = {
{ 0x10ec0221, 0xf00f, 0x1003, "ALC231" },
{ 0x10ec0269, 0xfff0, 0x3010, "ALC277" },
{ 0x10ec0269, 0xf0f0, 0x2010, "ALC259" },
@@ -964,7 +986,7 @@
{ } /* terminator */
};
-static struct alc_codec_rename_pci_table rename_pci_tbl[] = {
+static const struct alc_codec_rename_pci_table rename_pci_tbl[] = {
{ 0x10ec0280, 0x1028, 0, "ALC3220" },
{ 0x10ec0282, 0x1028, 0, "ALC3221" },
{ 0x10ec0283, 0x1028, 0, "ALC3223" },
@@ -1124,6 +1146,7 @@
codec->single_adc_amp = 1;
/* FIXME: do we need this for all Realtek codec models? */
codec->spdif_status_reset = 1;
+ codec->forced_resume = 1;
codec->patch_ops = alc_patch_ops;
err = alc_codec_rename_from_preset(codec);
@@ -1876,6 +1899,7 @@
ALC889_FIXUP_FRONT_HP_NO_PRESENCE,
ALC889_FIXUP_VAIO_TT,
ALC888_FIXUP_EEE1601,
+ ALC886_FIXUP_EAPD,
ALC882_FIXUP_EAPD,
ALC883_FIXUP_EAPD,
ALC883_FIXUP_ACER_EAPD,
@@ -1903,6 +1927,8 @@
ALC1220_FIXUP_CLEVO_P950,
ALC1220_FIXUP_CLEVO_PB51ED,
ALC1220_FIXUP_CLEVO_PB51ED_PINS,
+ ALC887_FIXUP_ASUS_AUDIO,
+ ALC887_FIXUP_ASUS_HMIC,
};
static void alc889_fixup_coef(struct hda_codec *codec,
@@ -1932,19 +1958,19 @@
{
if (action == HDA_FIXUP_ACT_PRE_PROBE) {
/* fake the connections during parsing the tree */
- hda_nid_t conn1[2] = { 0x0c, 0x0d };
- hda_nid_t conn2[2] = { 0x0e, 0x0f };
- snd_hda_override_conn_list(codec, 0x14, 2, conn1);
- snd_hda_override_conn_list(codec, 0x15, 2, conn1);
- snd_hda_override_conn_list(codec, 0x18, 2, conn2);
- snd_hda_override_conn_list(codec, 0x1a, 2, conn2);
+ static const hda_nid_t conn1[] = { 0x0c, 0x0d };
+ static const hda_nid_t conn2[] = { 0x0e, 0x0f };
+ snd_hda_override_conn_list(codec, 0x14, ARRAY_SIZE(conn1), conn1);
+ snd_hda_override_conn_list(codec, 0x15, ARRAY_SIZE(conn1), conn1);
+ snd_hda_override_conn_list(codec, 0x18, ARRAY_SIZE(conn2), conn2);
+ snd_hda_override_conn_list(codec, 0x1a, ARRAY_SIZE(conn2), conn2);
} else if (action == HDA_FIXUP_ACT_PROBE) {
/* restore the connections */
- hda_nid_t conn[5] = { 0x0c, 0x0d, 0x0e, 0x0f, 0x26 };
- snd_hda_override_conn_list(codec, 0x14, 5, conn);
- snd_hda_override_conn_list(codec, 0x15, 5, conn);
- snd_hda_override_conn_list(codec, 0x18, 5, conn);
- snd_hda_override_conn_list(codec, 0x1a, 5, conn);
+ static const hda_nid_t conn[] = { 0x0c, 0x0d, 0x0e, 0x0f, 0x26 };
+ snd_hda_override_conn_list(codec, 0x14, ARRAY_SIZE(conn), conn);
+ snd_hda_override_conn_list(codec, 0x15, ARRAY_SIZE(conn), conn);
+ snd_hda_override_conn_list(codec, 0x18, ARRAY_SIZE(conn), conn);
+ snd_hda_override_conn_list(codec, 0x1a, ARRAY_SIZE(conn), conn);
}
}
@@ -1952,8 +1978,8 @@
static void alc889_fixup_mbp_vref(struct hda_codec *codec,
const struct hda_fixup *fix, int action)
{
+ static const hda_nid_t nids[] = { 0x14, 0x15, 0x19 };
struct alc_spec *spec = codec->spec;
- static hda_nid_t nids[3] = { 0x14, 0x15, 0x19 };
int i;
if (action != HDA_FIXUP_ACT_INIT)
@@ -1989,7 +2015,7 @@
static void alc889_fixup_imac91_vref(struct hda_codec *codec,
const struct hda_fixup *fix, int action)
{
- static hda_nid_t nids[2] = { 0x18, 0x1a };
+ static const hda_nid_t nids[] = { 0x18, 0x1a };
if (action == HDA_FIXUP_ACT_INIT)
alc889_fixup_mac_pins(codec, nids, ARRAY_SIZE(nids));
@@ -1999,7 +2025,7 @@
static void alc889_fixup_mba11_vref(struct hda_codec *codec,
const struct hda_fixup *fix, int action)
{
- static hda_nid_t nids[1] = { 0x18 };
+ static const hda_nid_t nids[] = { 0x18 };
if (action == HDA_FIXUP_ACT_INIT)
alc889_fixup_mac_pins(codec, nids, ARRAY_SIZE(nids));
@@ -2009,7 +2035,7 @@
static void alc889_fixup_mba21_vref(struct hda_codec *codec,
const struct hda_fixup *fix, int action)
{
- static hda_nid_t nids[2] = { 0x18, 0x19 };
+ static const hda_nid_t nids[] = { 0x18, 0x19 };
if (action == HDA_FIXUP_ACT_INIT)
alc889_fixup_mac_pins(codec, nids, ARRAY_SIZE(nids));
@@ -2091,7 +2117,7 @@
const struct hda_fixup *fix,
int action)
{
- hda_nid_t conn1[1] = { 0x0c };
+ static const hda_nid_t conn1[] = { 0x0c };
if (action != HDA_FIXUP_ACT_PRE_PROBE)
return;
@@ -2100,8 +2126,8 @@
/* We therefore want to make sure 0x14 (front headphone) and
* 0x1b (speakers) use the stereo DAC 0x02
*/
- snd_hda_override_conn_list(codec, 0x14, 1, conn1);
- snd_hda_override_conn_list(codec, 0x1b, 1, conn1);
+ snd_hda_override_conn_list(codec, 0x14, ARRAY_SIZE(conn1), conn1);
+ snd_hda_override_conn_list(codec, 0x1b, ARRAY_SIZE(conn1), conn1);
}
static void alc_fixup_headset_mode_no_hp_mic(struct hda_codec *codec,
@@ -2115,6 +2141,31 @@
alc_fixup_headset_mode_no_hp_mic(codec, fix, action);
}
+static void alc887_asus_hp_automute_hook(struct hda_codec *codec,
+ struct hda_jack_callback *jack)
+{
+ struct alc_spec *spec = codec->spec;
+ unsigned int vref;
+
+ snd_hda_gen_hp_automute(codec, jack);
+
+ if (spec->gen.hp_jack_present)
+ vref = AC_PINCTL_VREF_80;
+ else
+ vref = AC_PINCTL_VREF_HIZ;
+ snd_hda_set_pin_ctl(codec, 0x19, PIN_HP | vref);
+}
+
+static void alc887_fixup_asus_jack(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ struct alc_spec *spec = codec->spec;
+ if (action != HDA_FIXUP_ACT_PROBE)
+ return;
+ snd_hda_set_pin_ctl_cache(codec, 0x1b, PIN_HP);
+ spec->gen.hp_automute_hook = alc887_asus_hp_automute_hook;
+}
+
static const struct hda_fixup alc882_fixups[] = {
[ALC882_FIXUP_ABIT_AW9D_MAX] = {
.type = HDA_FIXUP_PINS,
@@ -2182,6 +2233,15 @@
{ }
}
},
+ [ALC886_FIXUP_EAPD] = {
+ .type = HDA_FIXUP_VERBS,
+ .v.verbs = (const struct hda_verb[]) {
+ /* change to EAPD mode */
+ { 0x20, AC_VERB_SET_COEF_INDEX, 0x07 },
+ { 0x20, AC_VERB_SET_PROC_COEF, 0x0068 },
+ { }
+ }
+ },
[ALC882_FIXUP_EAPD] = {
.type = HDA_FIXUP_VERBS,
.v.verbs = (const struct hda_verb[]) {
@@ -2372,6 +2432,20 @@
.chained = true,
.chain_id = ALC1220_FIXUP_CLEVO_PB51ED,
},
+ [ALC887_FIXUP_ASUS_AUDIO] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x15, 0x02a14150 }, /* use as headset mic, without its own jack detect */
+ { 0x19, 0x22219420 },
+ {}
+ },
+ },
+ [ALC887_FIXUP_ASUS_HMIC] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc887_fixup_asus_jack,
+ .chained = true,
+ .chain_id = ALC887_FIXUP_ASUS_AUDIO,
+ },
};
static const struct snd_pci_quirk alc882_fixup_tbl[] = {
@@ -2390,13 +2464,13 @@
ALC882_FIXUP_ACER_ASPIRE_8930G),
SND_PCI_QUIRK(0x1025, 0x0146, "Acer Aspire 6935G",
ALC882_FIXUP_ACER_ASPIRE_8930G),
+ SND_PCI_QUIRK(0x1025, 0x0142, "Acer Aspire 7730G",
+ ALC882_FIXUP_ACER_ASPIRE_4930G),
+ SND_PCI_QUIRK(0x1025, 0x0155, "Packard-Bell M5120", ALC882_FIXUP_PB_M5210),
SND_PCI_QUIRK(0x1025, 0x015e, "Acer Aspire 6930G",
ALC882_FIXUP_ACER_ASPIRE_4930G),
SND_PCI_QUIRK(0x1025, 0x0166, "Acer Aspire 6530G",
ALC882_FIXUP_ACER_ASPIRE_4930G),
- SND_PCI_QUIRK(0x1025, 0x0142, "Acer Aspire 7730G",
- ALC882_FIXUP_ACER_ASPIRE_4930G),
- SND_PCI_QUIRK(0x1025, 0x0155, "Packard-Bell M5120", ALC882_FIXUP_PB_M5210),
SND_PCI_QUIRK(0x1025, 0x021e, "Acer Aspire 5739G",
ALC882_FIXUP_ACER_ASPIRE_4930G),
SND_PCI_QUIRK(0x1025, 0x0259, "Acer Aspire 5935", ALC889_FIXUP_DAC_ROUTE),
@@ -2405,14 +2479,15 @@
SND_PCI_QUIRK(0x1043, 0x13c2, "Asus A7M", ALC882_FIXUP_EAPD),
SND_PCI_QUIRK(0x1043, 0x1873, "ASUS W90V", ALC882_FIXUP_ASUS_W90V),
SND_PCI_QUIRK(0x1043, 0x1971, "Asus W2JC", ALC882_FIXUP_ASUS_W2JC),
+ SND_PCI_QUIRK(0x1043, 0x2390, "Asus D700SA", ALC887_FIXUP_ASUS_HMIC),
SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_FIXUP_EEE1601),
SND_PCI_QUIRK(0x1043, 0x84bc, "ASUS ET2700", ALC887_FIXUP_ASUS_BASS),
SND_PCI_QUIRK(0x1043, 0x8691, "ASUS ROG Ranger VIII", ALC882_FIXUP_GPIO3),
+ SND_PCI_QUIRK(0x104d, 0x9043, "Sony Vaio VGC-LN51JGB", ALC882_FIXUP_NO_PRIMARY_HP),
+ SND_PCI_QUIRK(0x104d, 0x9044, "Sony VAIO AiO", ALC882_FIXUP_NO_PRIMARY_HP),
SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC889_FIXUP_VAIO_TT),
SND_PCI_QUIRK(0x104d, 0x905a, "Sony Vaio Z", ALC882_FIXUP_NO_PRIMARY_HP),
SND_PCI_QUIRK(0x104d, 0x9060, "Sony Vaio VPCL14M1R", ALC882_FIXUP_NO_PRIMARY_HP),
- SND_PCI_QUIRK(0x104d, 0x9043, "Sony Vaio VGC-LN51JGB", ALC882_FIXUP_NO_PRIMARY_HP),
- SND_PCI_QUIRK(0x104d, 0x9044, "Sony VAIO AiO", ALC882_FIXUP_NO_PRIMARY_HP),
/* All Apple entries are in codec SSIDs */
SND_PCI_QUIRK(0x106b, 0x00a0, "MacBookPro 3,1", ALC889_FIXUP_MBP_VREF),
@@ -2439,19 +2514,44 @@
SND_PCI_QUIRK(0x106b, 0x4a00, "Macbook 5,2", ALC889_FIXUP_MBA11_VREF),
SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC882_FIXUP_EAPD),
+ 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(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),
+ SND_PCI_QUIRK(0x1462, 0x1275, "MSI-GL63", ALC1220_FIXUP_CLEVO_P950),
+ SND_PCI_QUIRK(0x1462, 0x1276, "MSI-GL73", ALC1220_FIXUP_CLEVO_P950),
+ SND_PCI_QUIRK(0x1462, 0x1293, "MSI-GP65", ALC1220_FIXUP_CLEVO_P950),
SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD),
+ SND_PCI_QUIRK(0x1462, 0xcc34, "MSI Godlike X570", ALC1220_FIXUP_GB_DUAL_CODECS),
SND_PCI_QUIRK(0x1462, 0xda57, "MSI Z270-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS),
SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3),
SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX),
+ SND_PCI_QUIRK(0x1558, 0x50d3, "Clevo PC50[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS),
+ SND_PCI_QUIRK(0x1558, 0x65d1, "Clevo PB51[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS),
+ SND_PCI_QUIRK(0x1558, 0x65d2, "Clevo PB51R[CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS),
+ 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, 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),
+ SND_PCI_QUIRK(0x1558, 0x70d1, "Clevo PC70[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS),
+ SND_PCI_QUIRK(0x1558, 0x7714, "Clevo X170", ALC1220_FIXUP_CLEVO_PB51ED_PINS),
SND_PCI_QUIRK(0x1558, 0x9501, "Clevo P950HR", ALC1220_FIXUP_CLEVO_P950),
+ SND_PCI_QUIRK(0x1558, 0x9506, "Clevo P955HQ", ALC1220_FIXUP_CLEVO_P950),
+ SND_PCI_QUIRK(0x1558, 0x950a, "Clevo P955H[PR]", ALC1220_FIXUP_CLEVO_P950),
SND_PCI_QUIRK(0x1558, 0x95e1, "Clevo P95xER", ALC1220_FIXUP_CLEVO_P950),
SND_PCI_QUIRK(0x1558, 0x95e2, "Clevo P950ER", ALC1220_FIXUP_CLEVO_P950),
+ SND_PCI_QUIRK(0x1558, 0x95e3, "Clevo P955[ER]T", ALC1220_FIXUP_CLEVO_P950),
+ SND_PCI_QUIRK(0x1558, 0x95e4, "Clevo P955ER", ALC1220_FIXUP_CLEVO_P950),
+ SND_PCI_QUIRK(0x1558, 0x95e5, "Clevo P955EE6", ALC1220_FIXUP_CLEVO_P950),
+ SND_PCI_QUIRK(0x1558, 0x95e6, "Clevo P950R[CDF]", ALC1220_FIXUP_CLEVO_P950),
SND_PCI_QUIRK(0x1558, 0x96e1, "Clevo P960[ER][CDFN]-K", ALC1220_FIXUP_CLEVO_P950),
SND_PCI_QUIRK(0x1558, 0x97e1, "Clevo P970[ER][CDFN]", ALC1220_FIXUP_CLEVO_P950),
- SND_PCI_QUIRK(0x1558, 0x65d1, "Clevo PB51[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS),
- SND_PCI_QUIRK(0x1558, 0x67d1, "Clevo PB71[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS),
+ SND_PCI_QUIRK(0x1558, 0x97e2, "Clevo P970RC-M", ALC1220_FIXUP_CLEVO_P950),
SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC882_FIXUP_EAPD),
SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_FIXUP_EAPD),
SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Y530", ALC882_FIXUP_LENOVO_Y530),
@@ -2494,6 +2594,28 @@
{}
};
+static const struct snd_hda_pin_quirk alc882_pin_fixup_tbl[] = {
+ SND_HDA_PIN_QUIRK(0x10ec1220, 0x1043, "ASUS", ALC1220_FIXUP_CLEVO_P950,
+ {0x14, 0x01014010},
+ {0x15, 0x01011012},
+ {0x16, 0x01016011},
+ {0x18, 0x01a19040},
+ {0x19, 0x02a19050},
+ {0x1a, 0x0181304f},
+ {0x1b, 0x0221401f},
+ {0x1e, 0x01456130}),
+ SND_HDA_PIN_QUIRK(0x10ec1220, 0x1462, "MS-7C35", ALC1220_FIXUP_CLEVO_P950,
+ {0x14, 0x01015010},
+ {0x15, 0x01011012},
+ {0x16, 0x01011011},
+ {0x18, 0x01a11040},
+ {0x19, 0x02a19050},
+ {0x1a, 0x0181104f},
+ {0x1b, 0x0221401f},
+ {0x1e, 0x01451130}),
+ {}
+};
+
/*
* BIOS auto configuration
*/
@@ -2522,6 +2644,7 @@
case 0x10ec0882:
case 0x10ec0885:
case 0x10ec0900:
+ case 0x10ec0b00:
case 0x10ec1220:
break;
default:
@@ -2534,6 +2657,7 @@
snd_hda_pick_fixup(codec, alc882_fixup_models, alc882_fixup_tbl,
alc882_fixups);
+ snd_hda_pick_pin_fixup(codec, alc882_pin_fixup_tbl, alc882_fixups, true);
snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE);
alc_auto_parse_customize_define(codec);
@@ -2971,6 +3095,111 @@
return alc_parse_auto_config(codec, alc269_ignore, ssids);
}
+static const struct hda_jack_keymap alc_headset_btn_keymap[] = {
+ { SND_JACK_BTN_0, KEY_PLAYPAUSE },
+ { SND_JACK_BTN_1, KEY_VOICECOMMAND },
+ { SND_JACK_BTN_2, KEY_VOLUMEUP },
+ { SND_JACK_BTN_3, KEY_VOLUMEDOWN },
+ {}
+};
+
+static void alc_headset_btn_callback(struct hda_codec *codec,
+ struct hda_jack_callback *jack)
+{
+ int report = 0;
+
+ if (jack->unsol_res & (7 << 13))
+ report |= SND_JACK_BTN_0;
+
+ if (jack->unsol_res & (1 << 16 | 3 << 8))
+ report |= SND_JACK_BTN_1;
+
+ /* Volume up key */
+ if (jack->unsol_res & (7 << 23))
+ report |= SND_JACK_BTN_2;
+
+ /* Volume down key */
+ if (jack->unsol_res & (7 << 10))
+ report |= SND_JACK_BTN_3;
+
+ jack->jack->button_state = report;
+}
+
+static void alc_disable_headset_jack_key(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ if (!spec->has_hs_key)
+ return;
+
+ switch (codec->core.vendor_id) {
+ case 0x10ec0215:
+ case 0x10ec0225:
+ case 0x10ec0285:
+ case 0x10ec0287:
+ case 0x10ec0295:
+ case 0x10ec0289:
+ case 0x10ec0299:
+ alc_write_coef_idx(codec, 0x48, 0x0);
+ alc_update_coef_idx(codec, 0x49, 0x0045, 0x0);
+ alc_update_coef_idx(codec, 0x44, 0x0045 << 8, 0x0);
+ break;
+ case 0x10ec0230:
+ case 0x10ec0236:
+ case 0x10ec0256:
+ alc_write_coef_idx(codec, 0x48, 0x0);
+ alc_update_coef_idx(codec, 0x49, 0x0045, 0x0);
+ break;
+ }
+}
+
+static void alc_enable_headset_jack_key(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ if (!spec->has_hs_key)
+ return;
+
+ switch (codec->core.vendor_id) {
+ case 0x10ec0215:
+ case 0x10ec0225:
+ case 0x10ec0285:
+ case 0x10ec0287:
+ case 0x10ec0295:
+ case 0x10ec0289:
+ case 0x10ec0299:
+ alc_write_coef_idx(codec, 0x48, 0xd011);
+ alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045);
+ alc_update_coef_idx(codec, 0x44, 0x007f << 8, 0x0045 << 8);
+ break;
+ case 0x10ec0230:
+ case 0x10ec0236:
+ case 0x10ec0256:
+ alc_write_coef_idx(codec, 0x48, 0xd011);
+ alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045);
+ break;
+ }
+}
+
+static void alc_fixup_headset_jack(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ struct alc_spec *spec = codec->spec;
+
+ switch (action) {
+ case HDA_FIXUP_ACT_PRE_PROBE:
+ spec->has_hs_key = 1;
+ snd_hda_jack_detect_enable_callback(codec, 0x55,
+ alc_headset_btn_callback);
+ snd_hda_jack_add_kctl(codec, 0x55, "Headset Jack", false,
+ SND_JACK_HEADSET, alc_headset_btn_keymap);
+ break;
+ case HDA_FIXUP_ACT_INIT:
+ alc_enable_headset_jack_key(codec);
+ break;
+ }
+}
+
static void alc269vb_toggle_power_output(struct hda_codec *codec, int power_up)
{
alc_update_coef_idx(codec, 0x04, 1 << 11, power_up ? (1 << 11) : 0);
@@ -2989,7 +3218,7 @@
alc_shutup_pins(codec);
}
-static struct coef_fw alc282_coefs[] = {
+static const struct coef_fw alc282_coefs[] = {
WRITE_COEF(0x03, 0x0002), /* Power Down Control */
UPDATE_COEF(0x05, 0xff3f, 0x0700), /* FIFO and filter clock */
WRITE_COEF(0x07, 0x0200), /* DMIC control */
@@ -3101,7 +3330,7 @@
alc_write_coef_idx(codec, 0x78, coef78);
}
-static struct coef_fw alc283_coefs[] = {
+static const struct coef_fw alc283_coefs[] = {
WRITE_COEF(0x03, 0x0002), /* Power Down Control */
UPDATE_COEF(0x05, 0xff3f, 0x0700), /* FIFO and filter clock */
WRITE_COEF(0x07, 0x0200), /* DMIC control */
@@ -3258,7 +3487,13 @@
alc_update_coefex_idx(codec, 0x57, 0x04, 0x0007, 0x4); /* Hight power */
alc_update_coefex_idx(codec, 0x53, 0x02, 0x8000, 1 << 15); /* Clear bit */
alc_update_coefex_idx(codec, 0x53, 0x02, 0x8000, 0 << 15);
- alc_update_coef_idx(codec, 0x36, 1 << 13, 1 << 5); /* Switch pcbeep path to Line in path*/
+ /*
+ * Expose headphone mic (or possibly Line In on some machines) instead
+ * of PC Beep on 1Ah, and disable 1Ah loopback for all outputs. See
+ * Documentation/sound/hd-audio/realtek-pc-beep.rst for details of
+ * this register.
+ */
+ alc_write_coef_idx(codec, 0x36, 0x5757);
}
static void alc256_shutup(struct hda_codec *codec)
@@ -3283,7 +3518,11 @@
/* 3k pull low control for Headset jack. */
/* NOTE: call this before clearing the pin, otherwise codec stalls */
- alc_update_coef_idx(codec, 0x46, 0, 3 << 12);
+ /* 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 (codec->core.vendor_id != 0x10ec0257)
+ alc_update_coef_idx(codec, 0x46, 0, 3 << 12);
if (!spec->no_shutup_pins)
snd_hda_codec_write(codec, hp_pin, 0,
@@ -3361,6 +3600,8 @@
if (!hp_pin)
hp_pin = 0x21;
+
+ alc_disable_headset_jack_key(codec);
/* 3k pull low control for Headset jack. */
alc_update_coef_idx(codec, 0x4a, 0, 3 << 10);
@@ -3400,6 +3641,9 @@
alc_update_coef_idx(codec, 0x4a, 3<<4, 2<<4);
msleep(30);
}
+
+ alc_update_coef_idx(codec, 0x4a, 3 << 10, 0);
+ alc_enable_headset_jack_key(codec);
}
static void alc_default_init(struct hda_codec *codec)
@@ -3594,8 +3838,8 @@
}
#ifdef HALT_REALTEK_ALC5505
-#define alc5505_dsp_suspend(codec) /* NOP */
-#define alc5505_dsp_resume(codec) /* NOP */
+#define alc5505_dsp_suspend(codec) do { } while (0) /* NOP */
+#define alc5505_dsp_resume(codec) do { } while (0) /* NOP */
#else
#define alc5505_dsp_suspend(codec) alc5505_dsp_halt(codec)
#define alc5505_dsp_resume(codec) alc5505_dsp_back_from_halt(codec)
@@ -3631,7 +3875,7 @@
msleep(200);
}
- regcache_sync(codec->core.regmap);
+ snd_hda_regmap_sync(codec);
hda_call_check_power_status(codec, 0x01);
/* on some machine, the BIOS will clear the codec gpio data when enter
@@ -3704,6 +3948,15 @@
snd_hda_sequence_write(codec, verbs);
}
+/* Fix the speaker amp after resume, etc */
+static void alc269vb_fixup_aspire_e1_coef(struct hda_codec *codec,
+ const struct hda_fixup *fix,
+ int action)
+{
+ if (action == HDA_FIXUP_ACT_INIT)
+ alc_update_coef_idx(codec, 0x0d, 0x6000, 0x6000);
+}
+
static void alc269_fixup_pcm_44k(struct hda_codec *codec,
const struct hda_fixup *fix, int action)
{
@@ -3942,11 +4195,9 @@
/* update LED status via GPIO */
static void alc_update_gpio_led(struct hda_codec *codec, unsigned int mask,
- bool enabled)
+ int polarity, bool enabled)
{
- struct alc_spec *spec = codec->spec;
-
- if (spec->mute_led_polarity)
+ if (polarity)
enabled = !enabled;
alc_update_gpio_data(codec, mask, !enabled); /* muted -> LED on */
}
@@ -3957,7 +4208,8 @@
struct hda_codec *codec = private_data;
struct alc_spec *spec = codec->spec;
- alc_update_gpio_led(codec, spec->gpio_mute_led_mask, enabled);
+ alc_update_gpio_led(codec, spec->gpio_mute_led_mask,
+ spec->mute_led_polarity, enabled);
}
/* turn on/off mic-mute LED via GPIO per capture hook */
@@ -3966,6 +4218,7 @@
struct alc_spec *spec = codec->spec;
alc_update_gpio_led(codec, spec->gpio_mic_led_mask,
+ spec->micmute_led_polarity,
spec->gen.micmute_led.led_value);
}
@@ -3997,6 +4250,12 @@
alc_fixup_hp_gpio_led(codec, action, 0x08, 0x10);
}
+static void alc285_fixup_hp_gpio_led(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ alc_fixup_hp_gpio_led(codec, action, 0x04, 0x00);
+}
+
static void alc286_fixup_hp_gpio_led(struct hda_codec *codec,
const struct hda_fixup *fix, int action)
{
@@ -4051,6 +4310,111 @@
}
}
+/* update mute-LED according to the speaker mute state via COEF bit */
+static void alc_fixup_mute_led_coefbit_hook(void *private_data, int enabled)
+{
+ struct hda_codec *codec = private_data;
+ struct alc_spec *spec = codec->spec;
+
+ if (spec->mute_led_polarity)
+ enabled = !enabled;
+
+ /* temporarily power up/down for setting COEF bit */
+ enabled ? alc_update_coef_idx(codec, spec->mute_led_coef_idx,
+ spec->mute_led_coefbit_mask, spec->mute_led_coefbit_off) :
+ alc_update_coef_idx(codec, spec->mute_led_coef_idx,
+ spec->mute_led_coefbit_mask, spec->mute_led_coefbit_on);
+}
+
+static void alc285_fixup_hp_mute_led_coefbit(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->mute_led_polarity = 0;
+ spec->mute_led_coef_idx = 0x0b;
+ spec->mute_led_coefbit_mask = 1<<3;
+ spec->mute_led_coefbit_on = 1<<3;
+ spec->mute_led_coefbit_off = 0;
+ spec->gen.vmaster_mute.hook = alc_fixup_mute_led_coefbit_hook;
+ spec->gen.vmaster_mute_enum = 1;
+ }
+}
+
+static void alc236_fixup_hp_mute_led_coefbit(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->mute_led_polarity = 0;
+ spec->mute_led_coef_idx = 0x34;
+ spec->mute_led_coefbit_mask = 1<<5;
+ spec->mute_led_coefbit_on = 0;
+ spec->mute_led_coefbit_off = 1<<5;
+ spec->gen.vmaster_mute.hook = alc_fixup_mute_led_coefbit_hook;
+ spec->gen.vmaster_mute_enum = 1;
+ }
+}
+
+/* turn on/off mic-mute LED per capture hook by coef bit */
+static void alc_hp_cap_micmute_update(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+
+ if (spec->gen.micmute_led.led_value)
+ alc_update_coef_idx(codec, spec->mic_led_coef_idx,
+ spec->mic_led_coefbit_mask, spec->mic_led_coefbit_on);
+ else
+ alc_update_coef_idx(codec, spec->mic_led_coef_idx,
+ spec->mic_led_coefbit_mask, spec->mic_led_coefbit_off);
+}
+
+static void alc285_fixup_hp_coef_micmute_led(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->mic_led_coef_idx = 0x19;
+ spec->mic_led_coefbit_mask = 1<<13;
+ spec->mic_led_coefbit_on = 1<<13;
+ spec->mic_led_coefbit_off = 0;
+ snd_hda_gen_add_micmute_led(codec, alc_hp_cap_micmute_update);
+ }
+}
+
+static void alc236_fixup_hp_coef_micmute_led(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->mic_led_coef_idx = 0x35;
+ spec->mic_led_coefbit_mask = 3<<2;
+ spec->mic_led_coefbit_on = 2<<2;
+ spec->mic_led_coefbit_off = 1<<2;
+ snd_hda_gen_add_micmute_led(codec, alc_hp_cap_micmute_update);
+ }
+}
+
+static void alc285_fixup_hp_mute_led(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ alc285_fixup_hp_mute_led_coefbit(codec, fix, action);
+ alc285_fixup_hp_coef_micmute_led(codec, fix, action);
+}
+
+static void alc236_fixup_hp_mute_led(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ alc236_fixup_hp_mute_led_coefbit(codec, fix, action);
+ alc236_fixup_hp_coef_micmute_led(codec, fix, action);
+}
+
#if IS_REACHABLE(CONFIG_INPUT)
static void gpio2_mic_hotkey_event(struct hda_codec *codec,
struct hda_jack_callback *event)
@@ -4140,6 +4504,7 @@
{
struct alc_spec *spec = codec->spec;
+ spec->micmute_led_polarity = 1;
alc_fixup_hp_gpio_led(codec, action, 0, 0x04);
if (action == HDA_FIXUP_ACT_PRE_PROBE) {
spec->init_amp = ALC_INIT_DEFAULT;
@@ -4177,7 +4542,7 @@
}
}
-static struct coef_fw alc225_pre_hsmode[] = {
+static const struct coef_fw alc225_pre_hsmode[] = {
UPDATE_COEF(0x4a, 1<<8, 0),
UPDATE_COEFEX(0x57, 0x05, 1<<14, 0),
UPDATE_COEF(0x63, 3<<14, 3<<14),
@@ -4190,7 +4555,7 @@
static void alc_headset_mode_unplugged(struct hda_codec *codec)
{
- static struct coef_fw coef0255[] = {
+ static const struct coef_fw coef0255[] = {
WRITE_COEF(0x1b, 0x0c0b), /* LDO and MISC control */
WRITE_COEF(0x45, 0xd089), /* UAJ function set to menual mode */
UPDATE_COEFEX(0x57, 0x05, 1<<14, 0), /* Direct Drive HP Amp control(Set to verb control)*/
@@ -4198,7 +4563,7 @@
WRITE_COEFEX(0x57, 0x03, 0x8aa6), /* Direct Drive HP Amp control */
{}
};
- static struct coef_fw coef0256[] = {
+ static const struct coef_fw coef0256[] = {
WRITE_COEF(0x1b, 0x0c4b), /* LDO and MISC control */
WRITE_COEF(0x45, 0xd089), /* UAJ function set to menual mode */
WRITE_COEF(0x06, 0x6104), /* Set MIC2 Vref gate with HP */
@@ -4206,7 +4571,7 @@
UPDATE_COEFEX(0x57, 0x05, 1<<14, 0), /* Direct Drive HP Amp control(Set to verb control)*/
{}
};
- static struct coef_fw coef0233[] = {
+ static const struct coef_fw coef0233[] = {
WRITE_COEF(0x1b, 0x0c0b),
WRITE_COEF(0x45, 0xc429),
UPDATE_COEF(0x35, 0x4000, 0),
@@ -4216,7 +4581,7 @@
WRITE_COEF(0x32, 0x42a3),
{}
};
- static struct coef_fw coef0288[] = {
+ static const struct coef_fw coef0288[] = {
UPDATE_COEF(0x4f, 0xfcc0, 0xc400),
UPDATE_COEF(0x50, 0x2000, 0x2000),
UPDATE_COEF(0x56, 0x0006, 0x0006),
@@ -4224,18 +4589,18 @@
UPDATE_COEF(0x67, 0x2000, 0),
{}
};
- static struct coef_fw coef0298[] = {
+ static const struct coef_fw coef0298[] = {
UPDATE_COEF(0x19, 0x1300, 0x0300),
{}
};
- static struct coef_fw coef0292[] = {
+ static const struct coef_fw coef0292[] = {
WRITE_COEF(0x76, 0x000e),
WRITE_COEF(0x6c, 0x2400),
WRITE_COEF(0x18, 0x7308),
WRITE_COEF(0x6b, 0xc429),
{}
};
- static struct coef_fw coef0293[] = {
+ static const struct coef_fw coef0293[] = {
UPDATE_COEF(0x10, 7<<8, 6<<8), /* SET Line1 JD to 0 */
UPDATE_COEFEX(0x57, 0x05, 1<<15|1<<13, 0x0), /* SET charge pump by verb */
UPDATE_COEFEX(0x57, 0x03, 1<<10, 1<<10), /* SET EN_OSW to 1 */
@@ -4244,16 +4609,16 @@
UPDATE_COEF(0x4a, 0x000f, 0x000e), /* Combo Jack auto detect */
{}
};
- static struct coef_fw coef0668[] = {
+ static const struct coef_fw coef0668[] = {
WRITE_COEF(0x15, 0x0d40),
WRITE_COEF(0xb7, 0x802b),
{}
};
- static struct coef_fw coef0225[] = {
+ static const struct coef_fw coef0225[] = {
UPDATE_COEF(0x63, 3<<14, 0),
{}
};
- static struct coef_fw coef0274[] = {
+ static const struct coef_fw coef0274[] = {
UPDATE_COEF(0x4a, 0x0100, 0),
UPDATE_COEFEX(0x57, 0x05, 0x4000, 0),
UPDATE_COEF(0x6b, 0xf000, 0x5000),
@@ -4268,6 +4633,7 @@
case 0x10ec0255:
alc_process_coef_fw(codec, coef0255);
break;
+ case 0x10ec0230:
case 0x10ec0236:
case 0x10ec0256:
alc_process_coef_fw(codec, coef0256);
@@ -4318,25 +4684,25 @@
static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin,
hda_nid_t mic_pin)
{
- static struct coef_fw coef0255[] = {
+ static const struct coef_fw coef0255[] = {
WRITE_COEFEX(0x57, 0x03, 0x8aa6),
WRITE_COEF(0x06, 0x6100), /* Set MIC2 Vref gate to normal */
{}
};
- static struct coef_fw coef0256[] = {
+ static const struct coef_fw coef0256[] = {
UPDATE_COEFEX(0x57, 0x05, 1<<14, 1<<14), /* Direct Drive HP Amp control(Set to verb control)*/
WRITE_COEFEX(0x57, 0x03, 0x09a3),
WRITE_COEF(0x06, 0x6100), /* Set MIC2 Vref gate to normal */
{}
};
- static struct coef_fw coef0233[] = {
+ static const struct coef_fw coef0233[] = {
UPDATE_COEF(0x35, 0, 1<<14),
WRITE_COEF(0x06, 0x2100),
WRITE_COEF(0x1a, 0x0021),
WRITE_COEF(0x26, 0x008c),
{}
};
- static struct coef_fw coef0288[] = {
+ static const struct coef_fw coef0288[] = {
UPDATE_COEF(0x4f, 0x00c0, 0),
UPDATE_COEF(0x50, 0x2000, 0),
UPDATE_COEF(0x56, 0x0006, 0),
@@ -4345,30 +4711,30 @@
UPDATE_COEF(0x67, 0x2000, 0x2000),
{}
};
- static struct coef_fw coef0292[] = {
+ static const struct coef_fw coef0292[] = {
WRITE_COEF(0x19, 0xa208),
WRITE_COEF(0x2e, 0xacf0),
{}
};
- static struct coef_fw coef0293[] = {
+ static const struct coef_fw coef0293[] = {
UPDATE_COEFEX(0x57, 0x05, 0, 1<<15|1<<13), /* SET charge pump by verb */
UPDATE_COEFEX(0x57, 0x03, 1<<10, 0), /* SET EN_OSW to 0 */
UPDATE_COEF(0x1a, 1<<3, 0), /* Combo JD gating without LINE1-VREFO */
{}
};
- static struct coef_fw coef0688[] = {
+ static const struct coef_fw coef0688[] = {
WRITE_COEF(0xb7, 0x802b),
WRITE_COEF(0xb5, 0x1040),
UPDATE_COEF(0xc3, 0, 1<<12),
{}
};
- static struct coef_fw coef0225[] = {
+ static const struct coef_fw coef0225[] = {
UPDATE_COEFEX(0x57, 0x05, 1<<14, 1<<14),
UPDATE_COEF(0x4a, 3<<4, 2<<4),
UPDATE_COEF(0x63, 3<<14, 0),
{}
};
- static struct coef_fw coef0274[] = {
+ static const struct coef_fw coef0274[] = {
UPDATE_COEFEX(0x57, 0x05, 0x4000, 0x4000),
UPDATE_COEF(0x4a, 0x0010, 0),
UPDATE_COEF(0x6b, 0xf000, 0),
@@ -4382,6 +4748,7 @@
alc_process_coef_fw(codec, coef0255);
snd_hda_set_pin_ctl_cache(codec, mic_pin, PIN_VREF50);
break;
+ case 0x10ec0230:
case 0x10ec0236:
case 0x10ec0256:
alc_write_coef_idx(codec, 0x45, 0xc489);
@@ -4454,7 +4821,7 @@
static void alc_headset_mode_default(struct hda_codec *codec)
{
- static struct coef_fw coef0225[] = {
+ static const struct coef_fw coef0225[] = {
UPDATE_COEF(0x45, 0x3f<<10, 0x30<<10),
UPDATE_COEF(0x45, 0x3f<<10, 0x31<<10),
UPDATE_COEF(0x49, 3<<8, 0<<8),
@@ -4463,14 +4830,14 @@
UPDATE_COEF(0x67, 0xf000, 0x3000),
{}
};
- static struct coef_fw coef0255[] = {
+ static const struct coef_fw coef0255[] = {
WRITE_COEF(0x45, 0xc089),
WRITE_COEF(0x45, 0xc489),
WRITE_COEFEX(0x57, 0x03, 0x8ea6),
WRITE_COEF(0x49, 0x0049),
{}
};
- static struct coef_fw coef0256[] = {
+ static const struct coef_fw coef0256[] = {
WRITE_COEF(0x45, 0xc489),
WRITE_COEFEX(0x57, 0x03, 0x0da3),
WRITE_COEF(0x49, 0x0049),
@@ -4478,12 +4845,12 @@
WRITE_COEF(0x06, 0x6100),
{}
};
- static struct coef_fw coef0233[] = {
+ static const struct coef_fw coef0233[] = {
WRITE_COEF(0x06, 0x2100),
WRITE_COEF(0x32, 0x4ea3),
{}
};
- static struct coef_fw coef0288[] = {
+ static const struct coef_fw coef0288[] = {
UPDATE_COEF(0x4f, 0xfcc0, 0xc400), /* Set to TRS type */
UPDATE_COEF(0x50, 0x2000, 0x2000),
UPDATE_COEF(0x56, 0x0006, 0x0006),
@@ -4491,26 +4858,26 @@
UPDATE_COEF(0x67, 0x2000, 0),
{}
};
- static struct coef_fw coef0292[] = {
+ static const struct coef_fw coef0292[] = {
WRITE_COEF(0x76, 0x000e),
WRITE_COEF(0x6c, 0x2400),
WRITE_COEF(0x6b, 0xc429),
WRITE_COEF(0x18, 0x7308),
{}
};
- static struct coef_fw coef0293[] = {
+ static const struct coef_fw coef0293[] = {
UPDATE_COEF(0x4a, 0x000f, 0x000e), /* Combo Jack auto detect */
WRITE_COEF(0x45, 0xC429), /* Set to TRS type */
UPDATE_COEF(0x1a, 1<<3, 0), /* Combo JD gating without LINE1-VREFO */
{}
};
- static struct coef_fw coef0688[] = {
+ static const struct coef_fw coef0688[] = {
WRITE_COEF(0x11, 0x0041),
WRITE_COEF(0x15, 0x0d40),
WRITE_COEF(0xb7, 0x802b),
{}
};
- static struct coef_fw coef0274[] = {
+ static const struct coef_fw coef0274[] = {
WRITE_COEF(0x45, 0x4289),
UPDATE_COEF(0x4a, 0x0010, 0x0010),
UPDATE_COEF(0x6b, 0x0f00, 0),
@@ -4531,6 +4898,7 @@
case 0x10ec0255:
alc_process_coef_fw(codec, coef0255);
break;
+ case 0x10ec0230:
case 0x10ec0236:
case 0x10ec0256:
alc_write_coef_idx(codec, 0x1b, 0x0e4b);
@@ -4573,53 +4941,53 @@
{
int val;
- static struct coef_fw coef0255[] = {
+ static const struct coef_fw coef0255[] = {
WRITE_COEF(0x45, 0xd489), /* Set to CTIA type */
WRITE_COEF(0x1b, 0x0c2b),
WRITE_COEFEX(0x57, 0x03, 0x8ea6),
{}
};
- static struct coef_fw coef0256[] = {
+ static const struct coef_fw coef0256[] = {
WRITE_COEF(0x45, 0xd489), /* Set to CTIA type */
WRITE_COEF(0x1b, 0x0e6b),
{}
};
- static struct coef_fw coef0233[] = {
+ static const struct coef_fw coef0233[] = {
WRITE_COEF(0x45, 0xd429),
WRITE_COEF(0x1b, 0x0c2b),
WRITE_COEF(0x32, 0x4ea3),
{}
};
- static struct coef_fw coef0288[] = {
+ static const struct coef_fw coef0288[] = {
UPDATE_COEF(0x50, 0x2000, 0x2000),
UPDATE_COEF(0x56, 0x0006, 0x0006),
UPDATE_COEF(0x66, 0x0008, 0),
UPDATE_COEF(0x67, 0x2000, 0),
{}
};
- static struct coef_fw coef0292[] = {
+ static const struct coef_fw coef0292[] = {
WRITE_COEF(0x6b, 0xd429),
WRITE_COEF(0x76, 0x0008),
WRITE_COEF(0x18, 0x7388),
{}
};
- static struct coef_fw coef0293[] = {
+ static const struct coef_fw coef0293[] = {
WRITE_COEF(0x45, 0xd429), /* Set to ctia type */
UPDATE_COEF(0x10, 7<<8, 7<<8), /* SET Line1 JD to 1 */
{}
};
- static struct coef_fw coef0688[] = {
+ static const struct coef_fw coef0688[] = {
WRITE_COEF(0x11, 0x0001),
WRITE_COEF(0x15, 0x0d60),
WRITE_COEF(0xc3, 0x0000),
{}
};
- static struct coef_fw coef0225_1[] = {
+ static const struct coef_fw coef0225_1[] = {
UPDATE_COEF(0x45, 0x3f<<10, 0x35<<10),
UPDATE_COEF(0x63, 3<<14, 2<<14),
{}
};
- static struct coef_fw coef0225_2[] = {
+ static const struct coef_fw coef0225_2[] = {
UPDATE_COEF(0x45, 0x3f<<10, 0x35<<10),
UPDATE_COEF(0x63, 3<<14, 1<<14),
{}
@@ -4629,6 +4997,7 @@
case 0x10ec0255:
alc_process_coef_fw(codec, coef0255);
break;
+ case 0x10ec0230:
case 0x10ec0236:
case 0x10ec0256:
alc_process_coef_fw(codec, coef0256);
@@ -4691,48 +5060,48 @@
/* Nokia type */
static void alc_headset_mode_omtp(struct hda_codec *codec)
{
- static struct coef_fw coef0255[] = {
+ static const struct coef_fw coef0255[] = {
WRITE_COEF(0x45, 0xe489), /* Set to OMTP Type */
WRITE_COEF(0x1b, 0x0c2b),
WRITE_COEFEX(0x57, 0x03, 0x8ea6),
{}
};
- static struct coef_fw coef0256[] = {
+ static const struct coef_fw coef0256[] = {
WRITE_COEF(0x45, 0xe489), /* Set to OMTP Type */
WRITE_COEF(0x1b, 0x0e6b),
{}
};
- static struct coef_fw coef0233[] = {
+ static const struct coef_fw coef0233[] = {
WRITE_COEF(0x45, 0xe429),
WRITE_COEF(0x1b, 0x0c2b),
WRITE_COEF(0x32, 0x4ea3),
{}
};
- static struct coef_fw coef0288[] = {
+ static const struct coef_fw coef0288[] = {
UPDATE_COEF(0x50, 0x2000, 0x2000),
UPDATE_COEF(0x56, 0x0006, 0x0006),
UPDATE_COEF(0x66, 0x0008, 0),
UPDATE_COEF(0x67, 0x2000, 0),
{}
};
- static struct coef_fw coef0292[] = {
+ static const struct coef_fw coef0292[] = {
WRITE_COEF(0x6b, 0xe429),
WRITE_COEF(0x76, 0x0008),
WRITE_COEF(0x18, 0x7388),
{}
};
- static struct coef_fw coef0293[] = {
+ static const struct coef_fw coef0293[] = {
WRITE_COEF(0x45, 0xe429), /* Set to omtp type */
UPDATE_COEF(0x10, 7<<8, 7<<8), /* SET Line1 JD to 1 */
{}
};
- static struct coef_fw coef0688[] = {
+ static const struct coef_fw coef0688[] = {
WRITE_COEF(0x11, 0x0001),
WRITE_COEF(0x15, 0x0d50),
WRITE_COEF(0xc3, 0x0000),
{}
};
- static struct coef_fw coef0225[] = {
+ static const struct coef_fw coef0225[] = {
UPDATE_COEF(0x45, 0x3f<<10, 0x39<<10),
UPDATE_COEF(0x63, 3<<14, 2<<14),
{}
@@ -4742,6 +5111,7 @@
case 0x10ec0255:
alc_process_coef_fw(codec, coef0255);
break;
+ case 0x10ec0230:
case 0x10ec0236:
case 0x10ec0256:
alc_process_coef_fw(codec, coef0256);
@@ -4792,17 +5162,17 @@
int val;
bool is_ctia = false;
struct alc_spec *spec = codec->spec;
- static struct coef_fw coef0255[] = {
+ static const struct coef_fw coef0255[] = {
WRITE_COEF(0x45, 0xd089), /* combo jack auto switch control(Check type)*/
WRITE_COEF(0x49, 0x0149), /* combo jack auto switch control(Vref
conteol) */
{}
};
- static struct coef_fw coef0288[] = {
+ static const struct coef_fw coef0288[] = {
UPDATE_COEF(0x4f, 0xfcc0, 0xd400), /* Check Type */
{}
};
- static struct coef_fw coef0298[] = {
+ static const struct coef_fw coef0298[] = {
UPDATE_COEF(0x50, 0x2000, 0x2000),
UPDATE_COEF(0x56, 0x0006, 0x0006),
UPDATE_COEF(0x66, 0x0008, 0),
@@ -4810,19 +5180,19 @@
UPDATE_COEF(0x19, 0x1300, 0x1300),
{}
};
- static struct coef_fw coef0293[] = {
+ static const struct coef_fw coef0293[] = {
UPDATE_COEF(0x4a, 0x000f, 0x0008), /* Combo Jack auto detect */
WRITE_COEF(0x45, 0xD429), /* Set to ctia type */
{}
};
- static struct coef_fw coef0688[] = {
+ static const struct coef_fw coef0688[] = {
WRITE_COEF(0x11, 0x0001),
WRITE_COEF(0xb7, 0x802b),
WRITE_COEF(0x15, 0x0d60),
WRITE_COEF(0xc3, 0x0c00),
{}
};
- static struct coef_fw coef0274[] = {
+ static const struct coef_fw coef0274[] = {
UPDATE_COEF(0x4a, 0x0010, 0),
UPDATE_COEF(0x4a, 0x8000, 0),
WRITE_COEF(0x45, 0xd289),
@@ -4837,6 +5207,7 @@
val = alc_read_coef_idx(codec, 0x46);
is_ctia = (val & 0x0070) == 0x0070;
break;
+ case 0x10ec0230:
case 0x10ec0236:
case 0x10ec0256:
alc_write_coef_idx(codec, 0x1b, 0x0e4b);
@@ -4867,7 +5238,7 @@
case 0x10ec0274:
case 0x10ec0294:
alc_process_coef_fw(codec, coef0274);
- msleep(80);
+ msleep(850);
val = alc_read_coef_idx(codec, 0x46);
is_ctia = (val & 0x00f0) == 0x00f0;
break;
@@ -5051,6 +5422,7 @@
struct hda_jack_callback *jack)
{
snd_hda_gen_hp_automute(codec, jack);
+ alc_update_headset_mode(codec);
}
static void alc_probe_headset_mode(struct hda_codec *codec)
@@ -5109,7 +5481,7 @@
static void alc255_set_default_jack_type(struct hda_codec *codec)
{
/* Set to iphone type */
- static struct coef_fw alc255fw[] = {
+ static const struct coef_fw alc255fw[] = {
WRITE_COEF(0x1b, 0x880b),
WRITE_COEF(0x45, 0xd089),
WRITE_COEF(0x1b, 0x080b),
@@ -5117,7 +5489,7 @@
WRITE_COEF(0x1b, 0x0c0b),
{}
};
- static struct coef_fw alc256fw[] = {
+ static const struct coef_fw alc256fw[] = {
WRITE_COEF(0x1b, 0x884b),
WRITE_COEF(0x45, 0xd089),
WRITE_COEF(0x1b, 0x084b),
@@ -5129,6 +5501,7 @@
case 0x10ec0255:
alc_process_coef_fw(codec, alc255fw);
break;
+ case 0x10ec0230:
case 0x10ec0236:
case 0x10ec0256:
alc_process_coef_fw(codec, alc256fw);
@@ -5236,18 +5609,9 @@
{ 0x19, 0x21a11010 }, /* dock mic */
{ }
};
- /* Assure the speaker pin to be coupled with DAC NID 0x03; otherwise
- * the speaker output becomes too low by some reason on Thinkpads with
- * ALC298 codec
- */
- static hda_nid_t preferred_pairs[] = {
- 0x14, 0x03, 0x17, 0x02, 0x21, 0x02,
- 0
- };
struct alc_spec *spec = codec->spec;
if (action == HDA_FIXUP_ACT_PRE_PROBE) {
- spec->gen.preferred_dacs = preferred_pairs;
spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP;
snd_hda_apply_pincfgs(codec, pincfgs);
} else if (action == HDA_FIXUP_ACT_INIT) {
@@ -5260,6 +5624,35 @@
}
}
+static void alc_fixup_tpt470_dacs(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ /* Assure the speaker pin to be coupled with DAC NID 0x03; otherwise
+ * the speaker output becomes too low by some reason on Thinkpads with
+ * ALC298 codec
+ */
+ static const hda_nid_t preferred_pairs[] = {
+ 0x14, 0x03, 0x17, 0x02, 0x21, 0x02,
+ 0
+ };
+ struct alc_spec *spec = codec->spec;
+
+ if (action == HDA_FIXUP_ACT_PRE_PROBE)
+ spec->gen.preferred_dacs = preferred_pairs;
+}
+
+static void alc295_fixup_asus_dacs(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ static const hda_nid_t preferred_pairs[] = {
+ 0x17, 0x02, 0x21, 0x03, 0
+ };
+ struct alc_spec *spec = codec->spec;
+
+ if (action == HDA_FIXUP_ACT_PRE_PROBE)
+ spec->gen.preferred_dacs = preferred_pairs;
+}
+
static void alc_shutup_dell_xps13(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
@@ -5364,17 +5757,6 @@
}
}
-static void alc256_fixup_dell_xps_13_headphone_noise2(struct hda_codec *codec,
- const struct hda_fixup *fix,
- int action)
-{
- if (action != HDA_FIXUP_ACT_PRE_PROBE)
- return;
-
- snd_hda_codec_amp_stereo(codec, 0x1a, HDA_INPUT, 0, HDA_AMP_VOLMASK, 1);
- snd_hda_override_wcaps(codec, 0x1a, get_wcaps(codec, 0x1a) & ~AC_WCAP_IN_AMP);
-}
-
static void alc269_fixup_limit_int_mic_boost(struct hda_codec *codec,
const struct hda_fixup *fix,
int action)
@@ -5512,9 +5894,9 @@
/* DAC node 0x03 is giving mono output. We therefore want to
make sure 0x14 (front speaker) and 0x15 (headphones) use the
stereo DAC, while leaving 0x17 (bass speaker) for node 0x03. */
- hda_nid_t conn1[2] = { 0x0c };
- snd_hda_override_conn_list(codec, 0x14, 1, conn1);
- snd_hda_override_conn_list(codec, 0x15, 1, conn1);
+ static const hda_nid_t conn1[] = { 0x0c };
+ snd_hda_override_conn_list(codec, 0x14, ARRAY_SIZE(conn1), conn1);
+ snd_hda_override_conn_list(codec, 0x15, ARRAY_SIZE(conn1), conn1);
}
}
@@ -5529,8 +5911,8 @@
Pin Complex), since Node 0x02 has Amp-out caps, we can adjust
speaker's volume now. */
- hda_nid_t conn1[1] = { 0x0c };
- snd_hda_override_conn_list(codec, 0x17, 1, conn1);
+ static const hda_nid_t conn1[] = { 0x0c };
+ snd_hda_override_conn_list(codec, 0x17, ARRAY_SIZE(conn1), conn1);
}
}
@@ -5539,8 +5921,18 @@
const struct hda_fixup *fix, int action)
{
if (action == HDA_FIXUP_ACT_PRE_PROBE) {
- hda_nid_t conn[2] = { 0x02, 0x03 };
- snd_hda_override_conn_list(codec, 0x17, 2, conn);
+ static const hda_nid_t conn[] = { 0x02, 0x03 };
+ snd_hda_override_conn_list(codec, 0x17, ARRAY_SIZE(conn), conn);
+ }
+}
+
+/* force NID 0x17 (Bass Speaker) to DAC1 to share it with the main speaker */
+static void alc285_fixup_speaker2_to_dac1(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ if (action == HDA_FIXUP_ACT_PRE_PROBE) {
+ static const hda_nid_t conn[] = { 0x02 };
+ snd_hda_override_conn_list(codec, 0x17, ARRAY_SIZE(conn), conn);
}
}
@@ -5552,7 +5944,8 @@
snd_hda_gen_hp_automute(codec, jack);
/* mute_led_polarity is set to 0, so we pass inverted value here */
- alc_update_gpio_led(codec, 0x10, !spec->gen.hp_jack_present);
+ alc_update_gpio_led(codec, 0x10, spec->mute_led_polarity,
+ !spec->gen.hp_jack_present);
}
/* Manage GPIOs for HP EliteBook Folio 9480m.
@@ -5589,6 +5982,39 @@
}
}
+/* Quirk for Thinkpad X1 7th and 8th Gen
+ * The following fixed routing needed
+ * DAC1 (NID 0x02) -> Speaker (NID 0x14); some eq applied secretly
+ * DAC2 (NID 0x03) -> Bass (NID 0x17) & Headphone (NID 0x21); sharing a DAC
+ * DAC3 (NID 0x06) -> Unused, due to the lack of volume amp
+ */
+static void alc285_fixup_thinkpad_x1_gen7(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ static const hda_nid_t conn[] = { 0x02, 0x03 }; /* exclude 0x06 */
+ static const hda_nid_t preferred_pairs[] = {
+ 0x14, 0x02, 0x17, 0x03, 0x21, 0x03, 0
+ };
+ struct alc_spec *spec = codec->spec;
+
+ switch (action) {
+ case HDA_FIXUP_ACT_PRE_PROBE:
+ snd_hda_override_conn_list(codec, 0x17, ARRAY_SIZE(conn), conn);
+ spec->gen.preferred_dacs = preferred_pairs;
+ break;
+ case HDA_FIXUP_ACT_BUILD:
+ /* The generic parser creates somewhat unintuitive volume ctls
+ * with the fixed routing above, and the shared DAC2 may be
+ * confusing for PA.
+ * Rename those to unique names so that PA doesn't touch them
+ * and use only Master volume.
+ */
+ rename_ctl(codec, "Front Playback Volume", "DAC1 Playback Volume");
+ rename_ctl(codec, "Bass Speaker Playback Volume", "DAC2 Playback Volume");
+ break;
+ }
+}
+
static void alc233_alc662_fixup_lenovo_dual_codecs(struct hda_codec *codec,
const struct hda_fixup *fix,
int action)
@@ -5613,12 +6039,21 @@
}
}
+static void alc225_fixup_s3_pop_noise(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ if (action != HDA_FIXUP_ACT_PRE_PROBE)
+ return;
+
+ codec->power_save_node = 1;
+}
+
/* Forcibly assign NID 0x03 to HP/LO while NID 0x02 to SPK for EQ */
static void alc274_fixup_bind_dacs(struct hda_codec *codec,
const struct hda_fixup *fix, int action)
{
struct alc_spec *spec = codec->spec;
- static hda_nid_t preferred_pairs[] = {
+ static const hda_nid_t preferred_pairs[] = {
0x21, 0x03, 0x1b, 0x03, 0x16, 0x02,
0
};
@@ -5631,6 +6066,21 @@
codec->power_save_node = 0;
}
+/* avoid DAC 0x06 for bass speaker 0x17; it has no volume control */
+static void alc289_fixup_asus_ga401(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ static const hda_nid_t preferred_pairs[] = {
+ 0x14, 0x02, 0x17, 0x02, 0x21, 0x03, 0
+ };
+ struct alc_spec *spec = codec->spec;
+
+ if (action == HDA_FIXUP_ACT_PRE_PROBE) {
+ spec->gen.preferred_dacs = preferred_pairs;
+ spec->gen.obey_preferred_dacs = 1;
+ }
+}
+
/* The DAC of NID 0x3 will introduce click/pop noise on headphones, so invalidate it */
static void alc285_fixup_invalidate_dacs(struct hda_codec *codec,
const struct hda_fixup *fix, int action)
@@ -5641,62 +6091,24 @@
snd_hda_override_wcaps(codec, 0x03, 0);
}
-static const struct hda_jack_keymap alc_headset_btn_keymap[] = {
- { SND_JACK_BTN_0, KEY_PLAYPAUSE },
- { SND_JACK_BTN_1, KEY_VOICECOMMAND },
- { SND_JACK_BTN_2, KEY_VOLUMEUP },
- { SND_JACK_BTN_3, KEY_VOLUMEDOWN },
- {}
-};
-
-static void alc_headset_btn_callback(struct hda_codec *codec,
- struct hda_jack_callback *jack)
+static void alc_combo_jack_hp_jd_restart(struct hda_codec *codec)
{
- int report = 0;
-
- if (jack->unsol_res & (7 << 13))
- report |= SND_JACK_BTN_0;
-
- if (jack->unsol_res & (1 << 16 | 3 << 8))
- report |= SND_JACK_BTN_1;
-
- /* Volume up key */
- if (jack->unsol_res & (7 << 23))
- report |= SND_JACK_BTN_2;
-
- /* Volume down key */
- if (jack->unsol_res & (7 << 10))
- report |= SND_JACK_BTN_3;
-
- jack->jack->button_state = report;
-}
-
-static void alc_fixup_headset_jack(struct hda_codec *codec,
- const struct hda_fixup *fix, int action)
-{
-
- switch (action) {
- case HDA_FIXUP_ACT_PRE_PROBE:
- snd_hda_jack_detect_enable_callback(codec, 0x55,
- alc_headset_btn_callback);
- snd_hda_jack_add_kctl(codec, 0x55, "Headset Jack", false,
- SND_JACK_HEADSET, alc_headset_btn_keymap);
+ switch (codec->core.vendor_id) {
+ case 0x10ec0274:
+ case 0x10ec0294:
+ case 0x10ec0225:
+ case 0x10ec0295:
+ case 0x10ec0299:
+ alc_update_coef_idx(codec, 0x4a, 0x8000, 1 << 15); /* Reset HP JD */
+ alc_update_coef_idx(codec, 0x4a, 0x8000, 0 << 15);
break;
- case HDA_FIXUP_ACT_INIT:
- switch (codec->core.vendor_id) {
- case 0x10ec0225:
- case 0x10ec0295:
- case 0x10ec0299:
- alc_write_coef_idx(codec, 0x48, 0xd011);
- alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045);
- alc_update_coef_idx(codec, 0x44, 0x007f << 8, 0x0045 << 8);
- break;
- case 0x10ec0236:
- case 0x10ec0256:
- alc_write_coef_idx(codec, 0x48, 0xd011);
- alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045);
- break;
- }
+ case 0x10ec0230:
+ case 0x10ec0235:
+ case 0x10ec0236:
+ case 0x10ec0255:
+ case 0x10ec0256:
+ alc_update_coef_idx(codec, 0x1b, 0x8000, 1 << 15); /* Reset HP JD */
+ alc_update_coef_idx(codec, 0x1b, 0x8000, 0 << 15);
break;
}
}
@@ -5711,16 +6123,7 @@
spec->ultra_low_power = true;
break;
case HDA_FIXUP_ACT_INIT:
- switch (codec->core.vendor_id) {
- case 0x10ec0295:
- alc_update_coef_idx(codec, 0x4a, 0x8000, 1 << 15); /* Reset HP JD */
- alc_update_coef_idx(codec, 0x4a, 0x8000, 0 << 15);
- break;
- case 0x10ec0236:
- alc_update_coef_idx(codec, 0x1b, 0x8000, 1 << 15); /* Reset HP JD */
- alc_update_coef_idx(codec, 0x1b, 0x8000, 0 << 15);
- break;
- }
+ alc_combo_jack_hp_jd_restart(codec);
break;
}
}
@@ -5732,6 +6135,107 @@
snd_hda_codec_set_pin_target(codec, 0x19, PIN_VREFHIZ);
}
+
+static void alc294_gx502_toggle_output(struct hda_codec *codec,
+ struct hda_jack_callback *cb)
+{
+ /* The Windows driver sets the codec up in a very different way where
+ * it appears to leave 0x10 = 0x8a20 set. For Linux we need to toggle it
+ */
+ if (snd_hda_jack_detect_state(codec, 0x21) == HDA_JACK_PRESENT)
+ alc_write_coef_idx(codec, 0x10, 0x8a20);
+ else
+ alc_write_coef_idx(codec, 0x10, 0x0a20);
+}
+
+static void alc294_fixup_gx502_hp(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ /* Pin 0x21: headphones/headset mic */
+ if (!is_jack_detectable(codec, 0x21))
+ return;
+
+ switch (action) {
+ case HDA_FIXUP_ACT_PRE_PROBE:
+ snd_hda_jack_detect_enable_callback(codec, 0x21,
+ alc294_gx502_toggle_output);
+ break;
+ case HDA_FIXUP_ACT_INIT:
+ /* Make sure to start in a correct state, i.e. if
+ * headphones have been plugged in before powering up the system
+ */
+ alc294_gx502_toggle_output(codec, NULL);
+ break;
+ }
+}
+
+static void alc294_gu502_toggle_output(struct hda_codec *codec,
+ struct hda_jack_callback *cb)
+{
+ /* Windows sets 0x10 to 0x8420 for Node 0x20 which is
+ * responsible from changes between speakers and headphones
+ */
+ if (snd_hda_jack_detect_state(codec, 0x21) == HDA_JACK_PRESENT)
+ alc_write_coef_idx(codec, 0x10, 0x8420);
+ else
+ alc_write_coef_idx(codec, 0x10, 0x0a20);
+}
+
+static void alc294_fixup_gu502_hp(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ if (!is_jack_detectable(codec, 0x21))
+ return;
+
+ switch (action) {
+ case HDA_FIXUP_ACT_PRE_PROBE:
+ snd_hda_jack_detect_enable_callback(codec, 0x21,
+ alc294_gu502_toggle_output);
+ break;
+ case HDA_FIXUP_ACT_INIT:
+ alc294_gu502_toggle_output(codec, NULL);
+ break;
+ }
+}
+
+static void alc285_fixup_hp_gpio_amp_init(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ if (action != HDA_FIXUP_ACT_INIT)
+ return;
+
+ msleep(100);
+ alc_write_coef_idx(codec, 0x65, 0x0);
+}
+
+static void alc274_fixup_hp_headset_mic(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ switch (action) {
+ case HDA_FIXUP_ACT_INIT:
+ alc_combo_jack_hp_jd_restart(codec);
+ break;
+ }
+}
+
+static void alc285_fixup_hp_spectre_x360(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ static const hda_nid_t conn[] = { 0x02 };
+ static const struct hda_pintbl pincfgs[] = {
+ { 0x14, 0x90170110 }, /* rear speaker */
+ { }
+ };
+
+ switch (action) {
+ case HDA_FIXUP_ACT_PRE_PROBE:
+ snd_hda_apply_pincfgs(codec, pincfgs);
+ /* force front speaker to DAC1 */
+ snd_hda_override_conn_list(codec, 0x17, ARRAY_SIZE(conn), conn);
+ break;
+ }
+}
+
/* for hda_fixup_thinkpad_acpi() */
#include "thinkpad_helper.c"
@@ -5746,6 +6250,7 @@
#include "hp_x360_helper.c"
enum {
+ ALC269_FIXUP_GPIO2,
ALC269_FIXUP_SONY_VAIO,
ALC275_FIXUP_SONY_VAIO_GPIO2,
ALC269_FIXUP_DELL_M101Z,
@@ -5777,6 +6282,7 @@
ALC269_FIXUP_HP_LINE1_MIC1_LED,
ALC269_FIXUP_INV_DMIC,
ALC269_FIXUP_LENOVO_DOCK,
+ ALC269_FIXUP_LENOVO_DOCK_LIMIT_BOOST,
ALC269_FIXUP_NO_SHUTUP,
ALC286_FIXUP_SONY_MIC_NO_PRESENCE,
ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT,
@@ -5821,6 +6327,7 @@
ALC283_FIXUP_HEADSET_MIC,
ALC255_FIXUP_MIC_MUTE_LED,
ALC282_FIXUP_ASPIRE_V5_PINS,
+ ALC269VB_FIXUP_ASPIRE_E1_COEF,
ALC280_FIXUP_HP_GPIO4,
ALC286_FIXUP_HP_GPIO_LED,
ALC280_FIXUP_HP_GPIO2_MIC_HOTKEY,
@@ -5831,6 +6338,7 @@
ALC288_FIXUP_DELL1_MIC_NO_PRESENCE,
ALC288_FIXUP_DELL_XPS_13,
ALC288_FIXUP_DISABLE_AAMIX,
+ ALC292_FIXUP_DELL_E7X_AAMIX,
ALC292_FIXUP_DELL_E7X,
ALC292_FIXUP_DISABLE_AAMIX,
ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK,
@@ -5838,18 +6346,18 @@
ALC298_FIXUP_DELL1_MIC_NO_PRESENCE,
ALC298_FIXUP_DELL_AIO_MIC_NO_PRESENCE,
ALC275_FIXUP_DELL_XPS,
- ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE,
- ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2,
ALC293_FIXUP_LENOVO_SPK_NOISE,
ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY,
ALC255_FIXUP_DELL_SPK_NOISE,
ALC225_FIXUP_DISABLE_MIC_VREF,
ALC225_FIXUP_DELL1_MIC_NO_PRESENCE,
ALC295_FIXUP_DISABLE_DAC3,
+ ALC285_FIXUP_SPEAKER2_TO_DAC1,
ALC280_FIXUP_HP_HEADSET_MIC,
ALC221_FIXUP_HP_FRONT_MIC,
ALC292_FIXUP_TPT460,
ALC298_FIXUP_SPK_VOLUME,
+ ALC298_FIXUP_LENOVO_SPK_VOLUME,
ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER,
ALC269_FIXUP_ATIV_BOOK_8,
ALC221_FIXUP_HP_MIC_NO_PRESENCE,
@@ -5862,9 +6370,11 @@
ALC233_FIXUP_ACER_HEADSET_MIC,
ALC294_FIXUP_LENOVO_MIC_LOCATION,
ALC225_FIXUP_DELL_WYSE_MIC_NO_PRESENCE,
+ ALC225_FIXUP_S3_POP_NOISE,
ALC700_FIXUP_INTEL_REFERENCE,
ALC274_FIXUP_DELL_BIND_DACS,
ALC274_FIXUP_DELL_AIO_LINEOUT_VERB,
+ ALC298_FIXUP_TPT470_DOCK_FIX,
ALC298_FIXUP_TPT470_DOCK,
ALC255_FIXUP_DUMMY_LINEOUT_VERB,
ALC255_FIXUP_DELL_HEADSET_MIC,
@@ -5890,11 +6400,57 @@
ALC256_FIXUP_ASUS_HEADSET_MIC,
ALC256_FIXUP_ASUS_MIC_NO_PRESENCE,
ALC299_FIXUP_PREDATOR_SPK,
- ALC294_FIXUP_ASUS_INTSPK_HEADSET_MIC,
ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE,
+ ALC289_FIXUP_DELL_SPK2,
+ ALC289_FIXUP_DUAL_SPK,
+ ALC294_FIXUP_SPK2_TO_DAC1,
+ ALC294_FIXUP_ASUS_DUAL_SPK,
+ ALC285_FIXUP_THINKPAD_X1_GEN7,
+ ALC285_FIXUP_THINKPAD_HEADSET_JACK,
+ ALC294_FIXUP_ASUS_HPE,
+ ALC294_FIXUP_ASUS_COEF_1B,
+ ALC294_FIXUP_ASUS_GX502_HP,
+ ALC294_FIXUP_ASUS_GX502_PINS,
+ ALC294_FIXUP_ASUS_GX502_VERBS,
+ ALC294_FIXUP_ASUS_GU502_HP,
+ ALC294_FIXUP_ASUS_GU502_PINS,
+ ALC294_FIXUP_ASUS_GU502_VERBS,
+ ALC285_FIXUP_HP_GPIO_LED,
+ ALC285_FIXUP_HP_MUTE_LED,
+ ALC236_FIXUP_HP_MUTE_LED,
+ ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET,
+ ALC295_FIXUP_ASUS_MIC_NO_PRESENCE,
+ ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS,
+ ALC269VC_FIXUP_ACER_HEADSET_MIC,
+ ALC269VC_FIXUP_ACER_MIC_NO_PRESENCE,
+ ALC289_FIXUP_ASUS_GA401,
+ ALC289_FIXUP_ASUS_GA502,
+ ALC256_FIXUP_ACER_MIC_NO_PRESENCE,
+ ALC285_FIXUP_HP_GPIO_AMP_INIT,
+ ALC269_FIXUP_CZC_B20,
+ ALC269_FIXUP_CZC_TMI,
+ ALC269_FIXUP_CZC_L101,
+ ALC269_FIXUP_LEMOTE_A1802,
+ ALC269_FIXUP_LEMOTE_A190X,
+ ALC256_FIXUP_INTEL_NUC8_RUGGED,
+ ALC233_FIXUP_INTEL_NUC8_DMIC,
+ ALC233_FIXUP_INTEL_NUC8_BOOST,
+ ALC256_FIXUP_INTEL_NUC10,
+ ALC255_FIXUP_XIAOMI_HEADSET_MIC,
+ ALC274_FIXUP_HP_MIC,
+ ALC274_FIXUP_HP_HEADSET_MIC,
+ ALC256_FIXUP_ASUS_HPE,
+ ALC285_FIXUP_THINKPAD_NO_BASS_SPK_HEADSET_JACK,
+ ALC295_FIXUP_ASUS_DACS,
+ ALC295_FIXUP_HP_OMEN,
+ ALC285_FIXUP_HP_SPECTRE_X360,
};
static const struct hda_fixup alc269_fixups[] = {
+ [ALC269_FIXUP_GPIO2] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc_fixup_gpio2,
+ },
[ALC269_FIXUP_SONY_VAIO] = {
.type = HDA_FIXUP_PINCTLS,
.v.pins = (const struct hda_pintbl[]) {
@@ -6089,6 +6645,12 @@
.chained = true,
.chain_id = ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT
},
+ [ALC269_FIXUP_LENOVO_DOCK_LIMIT_BOOST] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc269_fixup_limit_int_mic_boost,
+ .chained = true,
+ .chain_id = ALC269_FIXUP_LENOVO_DOCK,
+ },
[ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT] = {
.type = HDA_FIXUP_FUNC,
.v.func = alc269_fixup_pincfg_no_hp_to_lineout,
@@ -6444,6 +7006,10 @@
{ },
},
},
+ [ALC269VB_FIXUP_ASPIRE_E1_COEF] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc269vb_fixup_aspire_e1_coef,
+ },
[ALC280_FIXUP_HP_GPIO4] = {
.type = HDA_FIXUP_FUNC,
.v.func = alc280_fixup_hp_gpio4,
@@ -6521,12 +7087,19 @@
.chained = true,
.chain_id = ALC293_FIXUP_DELL1_MIC_NO_PRESENCE
},
- [ALC292_FIXUP_DELL_E7X] = {
+ [ALC292_FIXUP_DELL_E7X_AAMIX] = {
.type = HDA_FIXUP_FUNC,
.v.func = alc_fixup_dell_xps13,
.chained = true,
.chain_id = ALC292_FIXUP_DISABLE_AAMIX
},
+ [ALC292_FIXUP_DELL_E7X] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = snd_hda_gen_fixup_micmute_led,
+ /* micmute fixup must be applied at last */
+ .chained_before = true,
+ .chain_id = ALC292_FIXUP_DELL_E7X_AAMIX,
+ },
[ALC298_FIXUP_ALIENWARE_MIC_NO_PRESENCE] = {
.type = HDA_FIXUP_PINS,
.v.pins = (const struct hda_pintbl[]) {
@@ -6566,23 +7139,6 @@
{}
}
},
- [ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE] = {
- .type = HDA_FIXUP_VERBS,
- .v.verbs = (const struct hda_verb[]) {
- /* Disable pass-through path for FRONT 14h */
- {0x20, AC_VERB_SET_COEF_INDEX, 0x36},
- {0x20, AC_VERB_SET_PROC_COEF, 0x1737},
- {}
- },
- .chained = true,
- .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE
- },
- [ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2] = {
- .type = HDA_FIXUP_FUNC,
- .v.func = alc256_fixup_dell_xps_13_headphone_noise2,
- .chained = true,
- .chain_id = ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE
- },
[ALC293_FIXUP_LENOVO_SPK_NOISE] = {
.type = HDA_FIXUP_FUNC,
.v.func = alc_fixup_disable_aamix,
@@ -6593,6 +7149,16 @@
.type = HDA_FIXUP_FUNC,
.v.func = alc233_fixup_lenovo_line2_mic_hotkey,
},
+ [ALC233_FIXUP_INTEL_NUC8_DMIC] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc_fixup_inv_dmic,
+ .chained = true,
+ .chain_id = ALC233_FIXUP_INTEL_NUC8_BOOST,
+ },
+ [ALC233_FIXUP_INTEL_NUC8_BOOST] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc269_fixup_limit_int_mic_boost
+ },
[ALC255_FIXUP_DELL_SPK_NOISE] = {
.type = HDA_FIXUP_FUNC,
.v.func = alc_fixup_disable_aamix,
@@ -6641,10 +7207,20 @@
.chained = true,
.chain_id = ALC298_FIXUP_DELL_AIO_MIC_NO_PRESENCE,
},
+ [ALC298_FIXUP_LENOVO_SPK_VOLUME] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc298_fixup_speaker_volume,
+ },
[ALC295_FIXUP_DISABLE_DAC3] = {
.type = HDA_FIXUP_FUNC,
.v.func = alc295_fixup_disable_dac3,
},
+ [ALC285_FIXUP_SPEAKER2_TO_DAC1] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc285_fixup_speaker2_to_dac1,
+ .chained = true,
+ .chain_id = ALC269_FIXUP_THINKPAD_ACPI
+ },
[ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER] = {
.type = HDA_FIXUP_PINS,
.v.pins = (const struct hda_pintbl[]) {
@@ -6712,6 +7288,8 @@
[ALC233_FIXUP_LENOVO_MULTI_CODECS] = {
.type = HDA_FIXUP_FUNC,
.v.func = alc233_alc662_fixup_lenovo_dual_codecs,
+ .chained = true,
+ .chain_id = ALC269_FIXUP_GPIO2
},
[ALC233_FIXUP_ACER_HEADSET_MIC] = {
.type = HDA_FIXUP_VERBS,
@@ -6745,6 +7323,12 @@
{ }
},
.chained = true,
+ .chain_id = ALC225_FIXUP_S3_POP_NOISE
+ },
+ [ALC225_FIXUP_S3_POP_NOISE] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc225_fixup_s3_pop_noise,
+ .chained = true,
.chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC
},
[ALC700_FIXUP_INTEL_REFERENCE] = {
@@ -6777,12 +7361,18 @@
.chained = true,
.chain_id = ALC274_FIXUP_DELL_BIND_DACS
},
- [ALC298_FIXUP_TPT470_DOCK] = {
+ [ALC298_FIXUP_TPT470_DOCK_FIX] = {
.type = HDA_FIXUP_FUNC,
.v.func = alc_fixup_tpt470_dock,
.chained = true,
.chain_id = ALC293_FIXUP_LENOVO_SPK_NOISE
},
+ [ALC298_FIXUP_TPT470_DOCK] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc_fixup_tpt470_dacs,
+ .chained = true,
+ .chain_id = ALC298_FIXUP_TPT470_DOCK_FIX
+ },
[ALC255_FIXUP_DUMMY_LINEOUT_VERB] = {
.type = HDA_FIXUP_PINS,
.v.pins = (const struct hda_pintbl[]) {
@@ -6843,7 +7433,7 @@
{ }
},
.chained = true,
- .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC
+ .chain_id = ALC269_FIXUP_HEADSET_MIC
},
[ALC294_FIXUP_ASUS_HEADSET_MIC] = {
.type = HDA_FIXUP_PINS,
@@ -6852,7 +7442,7 @@
{ }
},
.chained = true,
- .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC
+ .chain_id = ALC269_FIXUP_HEADSET_MIC
},
[ALC294_FIXUP_ASUS_SPK] = {
.type = HDA_FIXUP_VERBS,
@@ -6860,6 +7450,8 @@
/* Set EAPD high */
{ 0x20, AC_VERB_SET_COEF_INDEX, 0x40 },
{ 0x20, AC_VERB_SET_PROC_COEF, 0x8800 },
+ { 0x20, AC_VERB_SET_COEF_INDEX, 0x0f },
+ { 0x20, AC_VERB_SET_PROC_COEF, 0x7774 },
{ }
},
.chained = true,
@@ -6962,16 +7554,6 @@
{ }
}
},
- [ALC294_FIXUP_ASUS_INTSPK_HEADSET_MIC] = {
- .type = HDA_FIXUP_PINS,
- .v.pins = (const struct hda_pintbl[]) {
- { 0x14, 0x411111f0 }, /* disable confusing internal speaker */
- { 0x19, 0x04a11150 }, /* use as headset mic, without its own jack detect */
- { }
- },
- .chained = true,
- .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC
- },
[ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE] = {
.type = HDA_FIXUP_PINS,
.v.pins = (const struct hda_pintbl[]) {
@@ -6982,6 +7564,381 @@
.chained = true,
.chain_id = ALC256_FIXUP_ASUS_HEADSET_MODE
},
+ [ALC289_FIXUP_DELL_SPK2] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x17, 0x90170130 }, /* bass spk */
+ { }
+ },
+ .chained = true,
+ .chain_id = ALC269_FIXUP_DELL4_MIC_NO_PRESENCE
+ },
+ [ALC289_FIXUP_DUAL_SPK] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc285_fixup_speaker2_to_dac1,
+ .chained = true,
+ .chain_id = ALC289_FIXUP_DELL_SPK2
+ },
+ [ALC294_FIXUP_SPK2_TO_DAC1] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc285_fixup_speaker2_to_dac1,
+ .chained = true,
+ .chain_id = ALC294_FIXUP_ASUS_HEADSET_MIC
+ },
+ [ALC294_FIXUP_ASUS_DUAL_SPK] = {
+ .type = HDA_FIXUP_FUNC,
+ /* The GPIO must be pulled to initialize the AMP */
+ .v.func = alc_fixup_gpio4,
+ .chained = true,
+ .chain_id = ALC294_FIXUP_SPK2_TO_DAC1
+ },
+ [ALC285_FIXUP_THINKPAD_X1_GEN7] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc285_fixup_thinkpad_x1_gen7,
+ .chained = true,
+ .chain_id = ALC269_FIXUP_THINKPAD_ACPI
+ },
+ [ALC285_FIXUP_THINKPAD_HEADSET_JACK] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc_fixup_headset_jack,
+ .chained = true,
+ .chain_id = ALC285_FIXUP_THINKPAD_X1_GEN7
+ },
+ [ALC294_FIXUP_ASUS_HPE] = {
+ .type = HDA_FIXUP_VERBS,
+ .v.verbs = (const struct hda_verb[]) {
+ /* Set EAPD high */
+ { 0x20, AC_VERB_SET_COEF_INDEX, 0x0f },
+ { 0x20, AC_VERB_SET_PROC_COEF, 0x7774 },
+ { }
+ },
+ .chained = true,
+ .chain_id = ALC294_FIXUP_ASUS_HEADSET_MIC
+ },
+ [ALC294_FIXUP_ASUS_GX502_PINS] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x19, 0x03a11050 }, /* front HP mic */
+ { 0x1a, 0x01a11830 }, /* rear external mic */
+ { 0x21, 0x03211020 }, /* front HP out */
+ { }
+ },
+ .chained = true,
+ .chain_id = ALC294_FIXUP_ASUS_GX502_VERBS
+ },
+ [ALC294_FIXUP_ASUS_GX502_VERBS] = {
+ .type = HDA_FIXUP_VERBS,
+ .v.verbs = (const struct hda_verb[]) {
+ /* set 0x15 to HP-OUT ctrl */
+ { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
+ /* unmute the 0x15 amp */
+ { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000 },
+ { }
+ },
+ .chained = true,
+ .chain_id = ALC294_FIXUP_ASUS_GX502_HP
+ },
+ [ALC294_FIXUP_ASUS_GX502_HP] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc294_fixup_gx502_hp,
+ },
+ [ALC294_FIXUP_ASUS_GU502_PINS] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x19, 0x01a11050 }, /* rear HP mic */
+ { 0x1a, 0x01a11830 }, /* rear external mic */
+ { 0x21, 0x012110f0 }, /* rear HP out */
+ { }
+ },
+ .chained = true,
+ .chain_id = ALC294_FIXUP_ASUS_GU502_VERBS
+ },
+ [ALC294_FIXUP_ASUS_GU502_VERBS] = {
+ .type = HDA_FIXUP_VERBS,
+ .v.verbs = (const struct hda_verb[]) {
+ /* set 0x15 to HP-OUT ctrl */
+ { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
+ /* unmute the 0x15 amp */
+ { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000 },
+ /* set 0x1b to HP-OUT */
+ { 0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+ { }
+ },
+ .chained = true,
+ .chain_id = ALC294_FIXUP_ASUS_GU502_HP
+ },
+ [ALC294_FIXUP_ASUS_GU502_HP] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc294_fixup_gu502_hp,
+ },
+ [ALC294_FIXUP_ASUS_COEF_1B] = {
+ .type = HDA_FIXUP_VERBS,
+ .v.verbs = (const struct hda_verb[]) {
+ /* Set bit 10 to correct noisy output after reboot from
+ * Windows 10 (due to pop noise reduction?)
+ */
+ { 0x20, AC_VERB_SET_COEF_INDEX, 0x1b },
+ { 0x20, AC_VERB_SET_PROC_COEF, 0x4e4b },
+ { }
+ },
+ .chained = true,
+ .chain_id = ALC289_FIXUP_ASUS_GA401,
+ },
+ [ALC285_FIXUP_HP_GPIO_LED] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc285_fixup_hp_gpio_led,
+ },
+ [ALC285_FIXUP_HP_MUTE_LED] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc285_fixup_hp_mute_led,
+ },
+ [ALC236_FIXUP_HP_MUTE_LED] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc236_fixup_hp_mute_led,
+ },
+ [ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET] = {
+ .type = HDA_FIXUP_VERBS,
+ .v.verbs = (const struct hda_verb[]) {
+ { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc5 },
+ { }
+ },
+ },
+ [ALC295_FIXUP_ASUS_MIC_NO_PRESENCE] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x19, 0x01a1913c }, /* use as headset mic, without its own jack detect */
+ { }
+ },
+ .chained = true,
+ .chain_id = ALC269_FIXUP_HEADSET_MODE
+ },
+ [ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x14, 0x90100120 }, /* use as internal speaker */
+ { 0x18, 0x02a111f0 }, /* use as headset mic, without its own jack detect */
+ { 0x1a, 0x01011020 }, /* use as line out */
+ { },
+ },
+ .chained = true,
+ .chain_id = ALC269_FIXUP_HEADSET_MIC
+ },
+ [ALC269VC_FIXUP_ACER_HEADSET_MIC] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x18, 0x02a11030 }, /* use as headset mic */
+ { }
+ },
+ .chained = true,
+ .chain_id = ALC269_FIXUP_HEADSET_MIC
+ },
+ [ALC269VC_FIXUP_ACER_MIC_NO_PRESENCE] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x18, 0x01a11130 }, /* use as headset mic, without its own jack detect */
+ { }
+ },
+ .chained = true,
+ .chain_id = ALC269_FIXUP_HEADSET_MIC
+ },
+ [ALC289_FIXUP_ASUS_GA401] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc289_fixup_asus_ga401,
+ .chained = true,
+ .chain_id = ALC289_FIXUP_ASUS_GA502,
+ },
+ [ALC289_FIXUP_ASUS_GA502] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x19, 0x03a11020 }, /* headset mic with jack detect */
+ { }
+ },
+ },
+ [ALC256_FIXUP_ACER_MIC_NO_PRESENCE] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x19, 0x02a11120 }, /* use as headset mic, without its own jack detect */
+ { }
+ },
+ .chained = true,
+ .chain_id = ALC256_FIXUP_ASUS_HEADSET_MODE
+ },
+ [ALC285_FIXUP_HP_GPIO_AMP_INIT] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc285_fixup_hp_gpio_amp_init,
+ .chained = true,
+ .chain_id = ALC285_FIXUP_HP_GPIO_LED
+ },
+ [ALC269_FIXUP_CZC_B20] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x12, 0x411111f0 },
+ { 0x14, 0x90170110 }, /* speaker */
+ { 0x15, 0x032f1020 }, /* HP out */
+ { 0x17, 0x411111f0 },
+ { 0x18, 0x03ab1040 }, /* mic */
+ { 0x19, 0xb7a7013f },
+ { 0x1a, 0x0181305f },
+ { 0x1b, 0x411111f0 },
+ { 0x1d, 0x411111f0 },
+ { 0x1e, 0x411111f0 },
+ { }
+ },
+ .chain_id = ALC269_FIXUP_DMIC,
+ },
+ [ALC269_FIXUP_CZC_TMI] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x12, 0x4000c000 },
+ { 0x14, 0x90170110 }, /* speaker */
+ { 0x15, 0x0421401f }, /* HP out */
+ { 0x17, 0x411111f0 },
+ { 0x18, 0x04a19020 }, /* mic */
+ { 0x19, 0x411111f0 },
+ { 0x1a, 0x411111f0 },
+ { 0x1b, 0x411111f0 },
+ { 0x1d, 0x40448505 },
+ { 0x1e, 0x411111f0 },
+ { 0x20, 0x8000ffff },
+ { }
+ },
+ .chain_id = ALC269_FIXUP_DMIC,
+ },
+ [ALC269_FIXUP_CZC_L101] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x12, 0x40000000 },
+ { 0x14, 0x01014010 }, /* speaker */
+ { 0x15, 0x411111f0 }, /* HP out */
+ { 0x16, 0x411111f0 },
+ { 0x18, 0x01a19020 }, /* mic */
+ { 0x19, 0x02a19021 },
+ { 0x1a, 0x0181302f },
+ { 0x1b, 0x0221401f },
+ { 0x1c, 0x411111f0 },
+ { 0x1d, 0x4044c601 },
+ { 0x1e, 0x411111f0 },
+ { }
+ },
+ .chain_id = ALC269_FIXUP_DMIC,
+ },
+ [ALC269_FIXUP_LEMOTE_A1802] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x12, 0x40000000 },
+ { 0x14, 0x90170110 }, /* speaker */
+ { 0x17, 0x411111f0 },
+ { 0x18, 0x03a19040 }, /* mic1 */
+ { 0x19, 0x90a70130 }, /* mic2 */
+ { 0x1a, 0x411111f0 },
+ { 0x1b, 0x411111f0 },
+ { 0x1d, 0x40489d2d },
+ { 0x1e, 0x411111f0 },
+ { 0x20, 0x0003ffff },
+ { 0x21, 0x03214020 },
+ { }
+ },
+ .chain_id = ALC269_FIXUP_DMIC,
+ },
+ [ALC269_FIXUP_LEMOTE_A190X] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x14, 0x99130110 }, /* speaker */
+ { 0x15, 0x0121401f }, /* HP out */
+ { 0x18, 0x01a19c20 }, /* rear mic */
+ { 0x19, 0x99a3092f }, /* front mic */
+ { 0x1b, 0x0201401f }, /* front lineout */
+ { }
+ },
+ .chain_id = ALC269_FIXUP_DMIC,
+ },
+ [ALC256_FIXUP_INTEL_NUC8_RUGGED] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x1b, 0x01a1913c }, /* use as headset mic, without its own jack detect */
+ { }
+ },
+ .chained = true,
+ .chain_id = ALC269_FIXUP_HEADSET_MODE
+ },
+ [ALC256_FIXUP_INTEL_NUC10] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x19, 0x01a1913c }, /* use as headset mic, without its own jack detect */
+ { }
+ },
+ .chained = true,
+ .chain_id = ALC269_FIXUP_HEADSET_MODE
+ },
+ [ALC255_FIXUP_XIAOMI_HEADSET_MIC] = {
+ .type = HDA_FIXUP_VERBS,
+ .v.verbs = (const struct hda_verb[]) {
+ { 0x20, AC_VERB_SET_COEF_INDEX, 0x45 },
+ { 0x20, AC_VERB_SET_PROC_COEF, 0x5089 },
+ { }
+ },
+ .chained = true,
+ .chain_id = ALC289_FIXUP_ASUS_GA502
+ },
+ [ALC274_FIXUP_HP_MIC] = {
+ .type = HDA_FIXUP_VERBS,
+ .v.verbs = (const struct hda_verb[]) {
+ { 0x20, AC_VERB_SET_COEF_INDEX, 0x45 },
+ { 0x20, AC_VERB_SET_PROC_COEF, 0x5089 },
+ { }
+ },
+ },
+ [ALC274_FIXUP_HP_HEADSET_MIC] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc274_fixup_hp_headset_mic,
+ .chained = true,
+ .chain_id = ALC274_FIXUP_HP_MIC
+ },
+ [ALC256_FIXUP_ASUS_HPE] = {
+ .type = HDA_FIXUP_VERBS,
+ .v.verbs = (const struct hda_verb[]) {
+ /* Set EAPD high */
+ { 0x20, AC_VERB_SET_COEF_INDEX, 0x0f },
+ { 0x20, AC_VERB_SET_PROC_COEF, 0x7778 },
+ { }
+ },
+ .chained = true,
+ .chain_id = ALC294_FIXUP_ASUS_HEADSET_MIC
+ },
+ [ALC285_FIXUP_THINKPAD_NO_BASS_SPK_HEADSET_JACK] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc_fixup_headset_jack,
+ .chained = true,
+ .chain_id = ALC269_FIXUP_THINKPAD_ACPI
+ },
+ [ALC295_FIXUP_ASUS_DACS] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc295_fixup_asus_dacs,
+ },
+ [ALC295_FIXUP_HP_OMEN] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x12, 0xb7a60130 },
+ { 0x13, 0x40000000 },
+ { 0x14, 0x411111f0 },
+ { 0x16, 0x411111f0 },
+ { 0x17, 0x90170110 },
+ { 0x18, 0x411111f0 },
+ { 0x19, 0x02a11030 },
+ { 0x1a, 0x411111f0 },
+ { 0x1b, 0x04a19030 },
+ { 0x1d, 0x40600001 },
+ { 0x1e, 0x411111f0 },
+ { 0x21, 0x03211020 },
+ {}
+ },
+ .chained = true,
+ .chain_id = ALC269_FIXUP_HP_LINE1_MIC1_LED,
+ },
+ [ALC285_FIXUP_HP_SPECTRE_X360] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc285_fixup_hp_spectre_x360,
+ },
};
static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -6990,23 +7947,32 @@
SND_PCI_QUIRK(0x1025, 0x0349, "Acer AOD260", ALC269_FIXUP_INV_DMIC),
SND_PCI_QUIRK(0x1025, 0x047c, "Acer AC700", ALC269_FIXUP_ACER_AC700),
SND_PCI_QUIRK(0x1025, 0x072d, "Acer Aspire V5-571G", ALC269_FIXUP_ASPIRE_HEADSET_MIC),
- SND_PCI_QUIRK(0x1025, 0x080d, "Acer Aspire V5-122P", ALC269_FIXUP_ASPIRE_HEADSET_MIC),
SND_PCI_QUIRK(0x1025, 0x0740, "Acer AO725", ALC271_FIXUP_HP_GATE_MIC_JACK),
SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK),
SND_PCI_QUIRK(0x1025, 0x0762, "Acer Aspire E1-472", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572),
SND_PCI_QUIRK(0x1025, 0x0775, "Acer Aspire E1-572", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572),
SND_PCI_QUIRK(0x1025, 0x079b, "Acer Aspire V5-573G", ALC282_FIXUP_ASPIRE_V5_PINS),
+ SND_PCI_QUIRK(0x1025, 0x080d, "Acer Aspire V5-122P", ALC269_FIXUP_ASPIRE_HEADSET_MIC),
+ SND_PCI_QUIRK(0x1025, 0x0840, "Acer Aspire E1", ALC269VB_FIXUP_ASPIRE_E1_COEF),
+ SND_PCI_QUIRK(0x1025, 0x101c, "Acer Veriton N2510G", ALC269_FIXUP_LIFEBOOK),
SND_PCI_QUIRK(0x1025, 0x102b, "Acer Aspire C24-860", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1025, 0x1065, "Acer Aspire C20-820", ALC269VC_FIXUP_ACER_HEADSET_MIC),
SND_PCI_QUIRK(0x1025, 0x106d, "Acer Cloudbook 14", ALC283_FIXUP_CHROME_BOOK),
SND_PCI_QUIRK(0x1025, 0x1099, "Acer Aspire E5-523G", ALC255_FIXUP_ACER_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1025, 0x110e, "Acer Aspire ES1-432", ALC255_FIXUP_ACER_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1025, 0x1166, "Acer Veriton N4640G", ALC269_FIXUP_LIFEBOOK),
+ SND_PCI_QUIRK(0x1025, 0x1167, "Acer Veriton N6640G", ALC269_FIXUP_LIFEBOOK),
SND_PCI_QUIRK(0x1025, 0x1246, "Acer Predator Helios 500", ALC299_FIXUP_PREDATOR_SPK),
+ SND_PCI_QUIRK(0x1025, 0x1247, "Acer vCopperbox", ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS),
+ SND_PCI_QUIRK(0x1025, 0x1248, "Acer Veriton N4660G", ALC269VC_FIXUP_ACER_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1025, 0x128f, "Acer Veriton Z6860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC),
SND_PCI_QUIRK(0x1025, 0x1290, "Acer Veriton Z4860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC),
SND_PCI_QUIRK(0x1025, 0x1291, "Acer Veriton Z4660G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC),
SND_PCI_QUIRK(0x1025, 0x1308, "Acer Aspire Z24-890", ALC286_FIXUP_ACER_AIO_HEADSET_MIC),
SND_PCI_QUIRK(0x1025, 0x132a, "Acer TravelMate B114-21", ALC233_FIXUP_ACER_HEADSET_MIC),
SND_PCI_QUIRK(0x1025, 0x1330, "Acer TravelMate X514-51T", ALC255_FIXUP_ACER_HEADSET_MIC),
+ SND_PCI_QUIRK(0x1025, 0x142b, "Acer Swift SF314-42", ALC255_FIXUP_ACER_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1025, 0x1430, "Acer TravelMate B311R-31", ALC256_FIXUP_ACER_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
SND_PCI_QUIRK(0x1028, 0x054b, "Dell XPS one 2710", ALC275_FIXUP_DELL_XPS),
SND_PCI_QUIRK(0x1028, 0x05bd, "Dell Latitude E6440", ALC292_FIXUP_DELL_E7X),
@@ -7035,17 +8001,14 @@
SND_PCI_QUIRK(0x1028, 0x06de, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK),
SND_PCI_QUIRK(0x1028, 0x06df, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK),
SND_PCI_QUIRK(0x1028, 0x06e0, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK),
- SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13 9350", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2),
SND_PCI_QUIRK(0x1028, 0x0706, "Dell Inspiron 7559", ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER),
SND_PCI_QUIRK(0x1028, 0x0725, "Dell Inspiron 3162", ALC255_FIXUP_DELL_SPK_NOISE),
SND_PCI_QUIRK(0x1028, 0x0738, "Dell Precision 5820", ALC269_FIXUP_NO_SHUTUP),
- SND_PCI_QUIRK(0x1028, 0x075b, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2),
SND_PCI_QUIRK(0x1028, 0x075c, "Dell XPS 27 7760", ALC298_FIXUP_SPK_VOLUME),
SND_PCI_QUIRK(0x1028, 0x075d, "Dell AIO", ALC298_FIXUP_SPK_VOLUME),
- SND_PCI_QUIRK(0x1028, 0x07b0, "Dell Precision 7520", ALC295_FIXUP_DISABLE_DAC3),
SND_PCI_QUIRK(0x1028, 0x0798, "Dell Inspiron 17 7000 Gaming", ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER),
+ SND_PCI_QUIRK(0x1028, 0x07b0, "Dell Precision 7520", ALC295_FIXUP_DISABLE_DAC3),
SND_PCI_QUIRK(0x1028, 0x080c, "Dell WYSE", ALC225_FIXUP_DELL_WYSE_MIC_NO_PRESENCE),
- SND_PCI_QUIRK(0x1028, 0x082a, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2),
SND_PCI_QUIRK(0x1028, 0x084b, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB),
SND_PCI_QUIRK(0x1028, 0x084e, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB),
SND_PCI_QUIRK(0x1028, 0x0871, "Dell Precision 3630", ALC255_FIXUP_DELL_HEADSET_MIC),
@@ -7054,40 +8017,27 @@
SND_PCI_QUIRK(0x1028, 0x08ad, "Dell WYSE AIO", ALC225_FIXUP_DELL_WYSE_AIO_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x08ae, "Dell WYSE NB", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x0935, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB),
+ SND_PCI_QUIRK(0x1028, 0x097d, "Dell Precision", ALC289_FIXUP_DUAL_SPK),
+ SND_PCI_QUIRK(0x1028, 0x097e, "Dell Precision", ALC289_FIXUP_DUAL_SPK),
+ SND_PCI_QUIRK(0x1028, 0x098d, "Dell Precision", ALC233_FIXUP_ASUS_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1028, 0x09bf, "Dell Precision", ALC233_FIXUP_ASUS_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2),
SND_PCI_QUIRK(0x103c, 0x18e6, "HP", ALC269_FIXUP_HP_GPIO_LED),
SND_PCI_QUIRK(0x103c, 0x218b, "HP", ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED),
- SND_PCI_QUIRK(0x103c, 0x225f, "HP", ALC280_FIXUP_HP_GPIO2_MIC_HOTKEY),
- /* ALC282 */
SND_PCI_QUIRK(0x103c, 0x21f9, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
SND_PCI_QUIRK(0x103c, 0x2210, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
SND_PCI_QUIRK(0x103c, 0x2214, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
+ SND_PCI_QUIRK(0x103c, 0x221b, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED),
+ SND_PCI_QUIRK(0x103c, 0x221c, "HP EliteBook 755 G2", ALC280_FIXUP_HP_HEADSET_MIC),
+ SND_PCI_QUIRK(0x103c, 0x2221, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED),
+ SND_PCI_QUIRK(0x103c, 0x2225, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED),
SND_PCI_QUIRK(0x103c, 0x2236, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED),
SND_PCI_QUIRK(0x103c, 0x2237, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED),
SND_PCI_QUIRK(0x103c, 0x2238, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED),
SND_PCI_QUIRK(0x103c, 0x2239, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED),
SND_PCI_QUIRK(0x103c, 0x224b, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED),
- SND_PCI_QUIRK(0x103c, 0x2268, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
- SND_PCI_QUIRK(0x103c, 0x226a, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
- SND_PCI_QUIRK(0x103c, 0x226b, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
- SND_PCI_QUIRK(0x103c, 0x226e, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
- SND_PCI_QUIRK(0x103c, 0x2271, "HP", ALC286_FIXUP_HP_GPIO_LED),
- SND_PCI_QUIRK(0x103c, 0x2272, "HP", ALC280_FIXUP_HP_DOCK_PINS),
- SND_PCI_QUIRK(0x103c, 0x2273, "HP", ALC280_FIXUP_HP_DOCK_PINS),
- SND_PCI_QUIRK(0x103c, 0x229e, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
- SND_PCI_QUIRK(0x103c, 0x22b2, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
- SND_PCI_QUIRK(0x103c, 0x22b7, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
- SND_PCI_QUIRK(0x103c, 0x22bf, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
- SND_PCI_QUIRK(0x103c, 0x22cf, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
- SND_PCI_QUIRK(0x103c, 0x22db, "HP", ALC280_FIXUP_HP_9480M),
- SND_PCI_QUIRK(0x103c, 0x22dc, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED),
- SND_PCI_QUIRK(0x103c, 0x22fb, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED),
- /* ALC290 */
- SND_PCI_QUIRK(0x103c, 0x221b, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED),
- SND_PCI_QUIRK(0x103c, 0x2221, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED),
- SND_PCI_QUIRK(0x103c, 0x2225, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED),
SND_PCI_QUIRK(0x103c, 0x2253, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED),
SND_PCI_QUIRK(0x103c, 0x2254, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED),
SND_PCI_QUIRK(0x103c, 0x2255, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED),
@@ -7095,36 +8045,63 @@
SND_PCI_QUIRK(0x103c, 0x2257, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED),
SND_PCI_QUIRK(0x103c, 0x2259, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED),
SND_PCI_QUIRK(0x103c, 0x225a, "HP", ALC269_FIXUP_HP_DOCK_GPIO_MIC1_LED),
+ SND_PCI_QUIRK(0x103c, 0x225f, "HP", ALC280_FIXUP_HP_GPIO2_MIC_HOTKEY),
SND_PCI_QUIRK(0x103c, 0x2260, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
SND_PCI_QUIRK(0x103c, 0x2263, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
SND_PCI_QUIRK(0x103c, 0x2264, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
SND_PCI_QUIRK(0x103c, 0x2265, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
+ SND_PCI_QUIRK(0x103c, 0x2268, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
+ SND_PCI_QUIRK(0x103c, 0x226a, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
+ SND_PCI_QUIRK(0x103c, 0x226b, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
+ SND_PCI_QUIRK(0x103c, 0x226e, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
+ SND_PCI_QUIRK(0x103c, 0x2271, "HP", ALC286_FIXUP_HP_GPIO_LED),
SND_PCI_QUIRK(0x103c, 0x2272, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED),
+ SND_PCI_QUIRK(0x103c, 0x2272, "HP", ALC280_FIXUP_HP_DOCK_PINS),
SND_PCI_QUIRK(0x103c, 0x2273, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED),
+ SND_PCI_QUIRK(0x103c, 0x2273, "HP", ALC280_FIXUP_HP_DOCK_PINS),
SND_PCI_QUIRK(0x103c, 0x2278, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED),
SND_PCI_QUIRK(0x103c, 0x227f, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
SND_PCI_QUIRK(0x103c, 0x2282, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
SND_PCI_QUIRK(0x103c, 0x228b, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
SND_PCI_QUIRK(0x103c, 0x228e, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
+ SND_PCI_QUIRK(0x103c, 0x229e, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
+ SND_PCI_QUIRK(0x103c, 0x22b2, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
+ SND_PCI_QUIRK(0x103c, 0x22b7, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
+ SND_PCI_QUIRK(0x103c, 0x22bf, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
+ SND_PCI_QUIRK(0x103c, 0x22c4, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
SND_PCI_QUIRK(0x103c, 0x22c5, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
SND_PCI_QUIRK(0x103c, 0x22c7, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
SND_PCI_QUIRK(0x103c, 0x22c8, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
- SND_PCI_QUIRK(0x103c, 0x22c4, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
+ SND_PCI_QUIRK(0x103c, 0x22cf, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
+ SND_PCI_QUIRK(0x103c, 0x22db, "HP", ALC280_FIXUP_HP_9480M),
+ SND_PCI_QUIRK(0x103c, 0x22dc, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED),
+ SND_PCI_QUIRK(0x103c, 0x22fb, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED),
SND_PCI_QUIRK(0x103c, 0x2334, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
SND_PCI_QUIRK(0x103c, 0x2335, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
SND_PCI_QUIRK(0x103c, 0x2336, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
SND_PCI_QUIRK(0x103c, 0x2337, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
- SND_PCI_QUIRK(0x103c, 0x221c, "HP EliteBook 755 G2", ALC280_FIXUP_HP_HEADSET_MIC),
SND_PCI_QUIRK(0x103c, 0x802e, "HP Z240 SFF", ALC221_FIXUP_HP_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x103c, 0x802f, "HP Z240", ALC221_FIXUP_HP_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x103c, 0x820d, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3),
SND_PCI_QUIRK(0x103c, 0x8256, "HP", ALC221_FIXUP_HP_FRONT_MIC),
SND_PCI_QUIRK(0x103c, 0x827e, "HP x360", ALC295_FIXUP_HP_X360),
+ SND_PCI_QUIRK(0x103c, 0x827f, "HP x360", ALC269_FIXUP_HP_MUTE_LED_MIC3),
SND_PCI_QUIRK(0x103c, 0x82bf, "HP G3 mini", ALC221_FIXUP_HP_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x103c, 0x82c0, "HP G3 mini premium", ALC221_FIXUP_HP_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x103c, 0x83b9, "HP Spectre x360", ALC269_FIXUP_HP_MUTE_LED_MIC3),
+ SND_PCI_QUIRK(0x103c, 0x841c, "HP Pavilion 15-CK0xx", ALC269_FIXUP_HP_MUTE_LED_MIC3),
SND_PCI_QUIRK(0x103c, 0x8497, "HP Envy x360", ALC269_FIXUP_HP_MUTE_LED_MIC3),
+ 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, 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),
+ SND_PCI_QUIRK(0x103c, 0x8729, "HP", ALC285_FIXUP_HP_GPIO_LED),
+ SND_PCI_QUIRK(0x103c, 0x8736, "HP", ALC285_FIXUP_HP_GPIO_AMP_INIT),
+ SND_PCI_QUIRK(0x103c, 0x8760, "HP", ALC285_FIXUP_HP_MUTE_LED),
+ SND_PCI_QUIRK(0x103c, 0x877a, "HP", ALC285_FIXUP_HP_MUTE_LED),
+ SND_PCI_QUIRK(0x103c, 0x877d, "HP", ALC236_FIXUP_HP_MUTE_LED),
SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC),
SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300),
SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
@@ -7133,61 +8110,143 @@
SND_PCI_QUIRK(0x1043, 0x10d0, "ASUS X540LA/X540LJ", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1043, 0x115d, "Asus 1015E", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
SND_PCI_QUIRK(0x1043, 0x11c0, "ASUS X556UR", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1043, 0x125e, "ASUS Q524UQK", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1043, 0x1271, "ASUS X430UN", ALC256_FIXUP_ASUS_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1043, 0x1290, "ASUS X441SA", ALC233_FIXUP_EAPD_COEF_AND_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1043, 0x12a0, "ASUS X441UV", ALC233_FIXUP_EAPD_COEF_AND_MIC_NO_PRESENCE),
- SND_PCI_QUIRK(0x1043, 0x12f0, "ASUS X541UV", ALC256_FIXUP_ASUS_MIC),
SND_PCI_QUIRK(0x1043, 0x12e0, "ASUS X541SA", ALC256_FIXUP_ASUS_MIC),
+ SND_PCI_QUIRK(0x1043, 0x12f0, "ASUS X541UV", ALC256_FIXUP_ASUS_MIC),
SND_PCI_QUIRK(0x1043, 0x13b0, "ASUS Z550SA", ALC256_FIXUP_ASUS_MIC),
SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_ASUS_ZENBOOK),
SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A),
SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC),
- SND_PCI_QUIRK(0x1043, 0x17d1, "ASUS UX431FL", ALC294_FIXUP_ASUS_INTSPK_HEADSET_MIC),
+ SND_PCI_QUIRK(0x1043, 0x1740, "ASUS UX430UA", ALC295_FIXUP_ASUS_DACS),
+ SND_PCI_QUIRK(0x1043, 0x17d1, "ASUS UX431FL", ALC294_FIXUP_ASUS_DUAL_SPK),
+ SND_PCI_QUIRK(0x1043, 0x1662, "ASUS GV301QH", ALC294_FIXUP_ASUS_DUAL_SPK),
+ SND_PCI_QUIRK(0x1043, 0x1881, "ASUS Zephyrus S/M", ALC294_FIXUP_ASUS_GX502_PINS),
SND_PCI_QUIRK(0x1043, 0x18b1, "Asus MJ401TA", ALC256_FIXUP_ASUS_HEADSET_MIC),
+ SND_PCI_QUIRK(0x1043, 0x18f1, "Asus FX505DT", ALC256_FIXUP_ASUS_HEADSET_MIC),
+ SND_PCI_QUIRK(0x1043, 0x194e, "ASUS UX563FD", ALC294_FIXUP_ASUS_HPE),
+ SND_PCI_QUIRK(0x1043, 0x1982, "ASUS B1400CEPE", ALC256_FIXUP_ASUS_HPE),
+ SND_PCI_QUIRK(0x1043, 0x19ce, "ASUS B9450FA", ALC294_FIXUP_ASUS_HPE),
+ SND_PCI_QUIRK(0x1043, 0x19e1, "ASUS UX581LV", ALC295_FIXUP_ASUS_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW),
SND_PCI_QUIRK(0x1043, 0x1a30, "ASUS X705UD", ALC256_FIXUP_ASUS_MIC),
+ SND_PCI_QUIRK(0x1043, 0x1b11, "ASUS UX431DA", ALC294_FIXUP_ASUS_COEF_1B),
SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC),
SND_PCI_QUIRK(0x1043, 0x1bbd, "ASUS Z550MA", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1043, 0x1c23, "Asus X55U", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
SND_PCI_QUIRK(0x1043, 0x1ccd, "ASUS X555UB", ALC256_FIXUP_ASUS_MIC),
+ SND_PCI_QUIRK(0x1043, 0x1d4e, "ASUS TM420", ALC256_FIXUP_ASUS_HPE),
+ SND_PCI_QUIRK(0x1043, 0x1e11, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA502),
+ 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, 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),
SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC),
SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC),
SND_PCI_QUIRK(0x1043, 0x8516, "ASUS X101CH", ALC269_FIXUP_ASUS_X101),
- SND_PCI_QUIRK(0x104d, 0x90b5, "Sony VAIO Pro 11", ALC286_FIXUP_SONY_MIC_NO_PRESENCE),
- SND_PCI_QUIRK(0x104d, 0x90b6, "Sony VAIO Pro 13", ALC286_FIXUP_SONY_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x104d, 0x9073, "Sony VAIO", ALC275_FIXUP_SONY_VAIO_GPIO2),
SND_PCI_QUIRK(0x104d, 0x907b, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
SND_PCI_QUIRK(0x104d, 0x9099, "Sony VAIO S13", ALC275_FIXUP_SONY_DISABLE_AAMIX),
+ SND_PCI_QUIRK(0x104d, 0x90b5, "Sony VAIO Pro 11", ALC286_FIXUP_SONY_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x104d, 0x90b6, "Sony VAIO Pro 13", ALC286_FIXUP_SONY_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook", ALC269_FIXUP_LIFEBOOK),
SND_PCI_QUIRK(0x10cf, 0x159f, "Lifebook E780", ALC269_FIXUP_LIFEBOOK_NO_HP_TO_LINEOUT),
SND_PCI_QUIRK(0x10cf, 0x15dc, "Lifebook T731", ALC269_FIXUP_LIFEBOOK_HP_PIN),
- SND_PCI_QUIRK(0x10cf, 0x1757, "Lifebook E752", ALC269_FIXUP_LIFEBOOK_HP_PIN),
SND_PCI_QUIRK(0x10cf, 0x1629, "Lifebook U7x7", ALC255_FIXUP_LIFEBOOK_U7x7_HEADSET_MIC),
+ SND_PCI_QUIRK(0x10cf, 0x1757, "Lifebook E752", ALC269_FIXUP_LIFEBOOK_HP_PIN),
SND_PCI_QUIRK(0x10cf, 0x1845, "Lifebook U904", ALC269_FIXUP_LIFEBOOK_EXTMIC),
SND_PCI_QUIRK(0x10ec, 0x10f2, "Intel Reference board", ALC700_FIXUP_INTEL_REFERENCE),
+ SND_PCI_QUIRK(0x10ec, 0x118c, "Medion EE4254 MD62100", ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE),
+ SND_PCI_QUIRK(0x10ec, 0x1230, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK),
+ SND_PCI_QUIRK(0x10ec, 0x1252, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK),
+ SND_PCI_QUIRK(0x10ec, 0x1254, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK),
SND_PCI_QUIRK(0x10f7, 0x8338, "Panasonic CF-SZ6", ALC269_FIXUP_HEADSET_MODE),
SND_PCI_QUIRK(0x144d, 0xc109, "Samsung Ativ book 9 (NP900X3G)", ALC269_FIXUP_INV_DMIC),
+ SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
+ SND_PCI_QUIRK(0x144d, 0xc176, "Samsung Notebook 9 Pro (NP930MBE-K04US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
+ SND_PCI_QUIRK(0x144d, 0xc189, "Samsung Galaxy Flex Book (NT950QCG-X716)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
+ SND_PCI_QUIRK(0x144d, 0xc18a, "Samsung Galaxy Book Ion (NP930XCJ-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET),
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(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),
+ SND_PCI_QUIRK(0x152d, 0x1082, "Quanta NL3", ALC269_FIXUP_LIFEBOOK),
+ SND_PCI_QUIRK(0x1558, 0x1323, "Clevo N130ZU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1558, 0x1325, "System76 Darter Pro (darp5)", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1558, 0x1401, "Clevo L140[CZ]U", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1558, 0x1403, "Clevo N140CU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1558, 0x1404, "Clevo N150CU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1558, 0x14a1, "Clevo L141MU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1558, 0x4018, "Clevo NV40M[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1558, 0x4019, "Clevo NV40MZ", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1558, 0x4020, "Clevo NV40MB", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1558, 0x40a1, "Clevo NL40GU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1558, 0x40c1, "Clevo NL40[CZ]U", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1558, 0x40d1, "Clevo NL41DU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1558, 0x50a3, "Clevo NJ51GU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1558, 0x50b3, "Clevo NK50S[BEZ]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1558, 0x50b6, "Clevo NK50S5", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1558, 0x50b8, "Clevo NK50SZ", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1558, 0x50d5, "Clevo NP50D5", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1558, 0x50f0, "Clevo NH50A[CDF]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1558, 0x50f2, "Clevo NH50E[PR]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1558, 0x50f3, "Clevo NH58DPQ", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1558, 0x50f5, "Clevo NH55EPY", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1558, 0x50f6, "Clevo NH55DPQ", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1558, 0x5101, "Clevo S510WU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1558, 0x5157, "Clevo W517GU1", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1558, 0x51a1, "Clevo NS50MU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1558, 0x70a1, "Clevo NB70T[HJK]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1558, 0x70b3, "Clevo NK70SB", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1558, 0x70f2, "Clevo NH79EPY", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1558, 0x70f3, "Clevo NH77DPQ", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1558, 0x70f4, "Clevo NH77EPY", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1558, 0x70f6, "Clevo NH77DPQ-Y", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1558, 0x8228, "Clevo NR40BU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1558, 0x8520, "Clevo NH50D[CD]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1558, 0x8521, "Clevo NH77D[CD]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1558, 0x8535, "Clevo NH50D[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1558, 0x8536, "Clevo NH79D[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1558, 0x8550, "System76 Gazelle (gaze14)", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1558, 0x8551, "System76 Gazelle (gaze14)", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1558, 0x8560, "System76 Gazelle (gaze14)", ALC269_FIXUP_HEADSET_MIC),
SND_PCI_QUIRK(0x1558, 0x8561, "System76 Gazelle (gaze14)", ALC269_FIXUP_HEADSET_MIC),
+ SND_PCI_QUIRK(0x1558, 0x8562, "Clevo NH[5|7][0-9]RZ[Q]", ALC269_FIXUP_DMIC),
+ SND_PCI_QUIRK(0x1558, 0x8668, "Clevo NP50B[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1558, 0x8680, "Clevo NJ50LU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1558, 0x8686, "Clevo NH50[CZ]U", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1558, 0x8a20, "Clevo NH55DCQ-Y", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1558, 0x8a51, "Clevo NH70RCQ-Y", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1558, 0x8d50, "Clevo NH55RCQ-M", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1558, 0x951d, "Clevo N950T[CDF]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1558, 0x9600, "Clevo N960K[PR]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1558, 0x961d, "Clevo N960S[CDF]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1558, 0x971d, "Clevo N970T[CDF]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1558, 0xa500, "Clevo NL53RU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1558, 0xa600, "Clevo NL5XNU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1558, 0xb018, "Clevo NP50D[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1558, 0xb019, "Clevo NH77D[BE]Q", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1558, 0xb022, "Clevo NH77D[DC][QW]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1558, 0xc018, "Clevo NP50D[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1558, 0xc019, "Clevo NH77D[BE]Q", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1558, 0xc022, "Clevo NH77[DC][QW]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x17aa, 0x1036, "Lenovo P520", ALC233_FIXUP_LENOVO_MULTI_CODECS),
+ SND_PCI_QUIRK(0x17aa, 0x1048, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC),
SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE),
SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE),
SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE),
SND_PCI_QUIRK(0x17aa, 0x21ca, "Thinkpad L412", ALC269_FIXUP_SKU_IGNORE),
SND_PCI_QUIRK(0x17aa, 0x21e9, "Thinkpad Edge 15", ALC269_FIXUP_SKU_IGNORE),
- SND_PCI_QUIRK(0x17aa, 0x21f6, "Thinkpad T530", ALC269_FIXUP_LENOVO_DOCK),
- SND_PCI_QUIRK(0x17aa, 0x21fa, "Thinkpad X230", ALC269_FIXUP_LENOVO_DOCK),
SND_PCI_QUIRK(0x17aa, 0x21f3, "Thinkpad T430", ALC269_FIXUP_LENOVO_DOCK),
+ SND_PCI_QUIRK(0x17aa, 0x21f6, "Thinkpad T530", ALC269_FIXUP_LENOVO_DOCK_LIMIT_BOOST),
+ SND_PCI_QUIRK(0x17aa, 0x21fa, "Thinkpad X230", ALC269_FIXUP_LENOVO_DOCK),
SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK),
SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK),
SND_PCI_QUIRK(0x17aa, 0x2208, "Thinkpad T431s", ALC269_FIXUP_LENOVO_DOCK),
@@ -7213,6 +8272,10 @@
SND_PCI_QUIRK(0x17aa, 0x224c, "Thinkpad", ALC298_FIXUP_TPT470_DOCK),
SND_PCI_QUIRK(0x17aa, 0x224d, "Thinkpad", ALC298_FIXUP_TPT470_DOCK),
SND_PCI_QUIRK(0x17aa, 0x225d, "Thinkpad T480", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+ SND_PCI_QUIRK(0x17aa, 0x2292, "Thinkpad X1 Carbon 7th", ALC285_FIXUP_THINKPAD_HEADSET_JACK),
+ SND_PCI_QUIRK(0x17aa, 0x22be, "Thinkpad X1 Carbon 8th", ALC285_FIXUP_THINKPAD_HEADSET_JACK),
+ SND_PCI_QUIRK(0x17aa, 0x22c1, "Thinkpad P1 Gen 3", ALC285_FIXUP_THINKPAD_NO_BASS_SPK_HEADSET_JACK),
+ SND_PCI_QUIRK(0x17aa, 0x22c2, "Thinkpad X1 Extreme Gen 3", ALC285_FIXUP_THINKPAD_NO_BASS_SPK_HEADSET_JACK),
SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY),
SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY),
SND_PCI_QUIRK(0x17aa, 0x310c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION),
@@ -7223,9 +8286,11 @@
SND_PCI_QUIRK(0x17aa, 0x3151, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC),
SND_PCI_QUIRK(0x17aa, 0x3176, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC),
SND_PCI_QUIRK(0x17aa, 0x3178, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC),
+ SND_PCI_QUIRK(0x17aa, 0x3818, "Lenovo C940", ALC298_FIXUP_LENOVO_SPK_VOLUME),
SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI),
SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC),
SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI),
+ SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K),
SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
SND_PCI_QUIRK(0x17aa, 0x501a, "Thinkpad", ALC283_FIXUP_INT_MIC),
SND_PCI_QUIRK(0x17aa, 0x501e, "Thinkpad L440", ALC292_FIXUP_TPT440_DOCK),
@@ -7244,11 +8309,21 @@
SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
SND_PCI_QUIRK(0x17aa, 0x511e, "Thinkpad", ALC298_FIXUP_TPT470_DOCK),
SND_PCI_QUIRK(0x17aa, 0x511f, "Thinkpad", ALC298_FIXUP_TPT470_DOCK),
- SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K),
SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD),
SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MACH-WX9", ALC256_FIXUP_HUAWEI_MACH_WX9_PINS),
+ SND_PCI_QUIRK(0x1b35, 0x1235, "CZC B20", ALC269_FIXUP_CZC_B20),
+ SND_PCI_QUIRK(0x1b35, 0x1236, "CZC TMI", ALC269_FIXUP_CZC_TMI),
+ SND_PCI_QUIRK(0x1b35, 0x1237, "CZC L101", ALC269_FIXUP_CZC_L101),
SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */
- SND_PCI_QUIRK(0x10ec, 0x118c, "Medion EE4254 MD62100", ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1c06, 0x2013, "Lemote A1802", ALC269_FIXUP_LEMOTE_A1802),
+ SND_PCI_QUIRK(0x1c06, 0x2015, "Lemote A190X", ALC269_FIXUP_LEMOTE_A190X),
+ SND_PCI_QUIRK(0x1d72, 0x1602, "RedmiBook", ALC255_FIXUP_XIAOMI_HEADSET_MIC),
+ SND_PCI_QUIRK(0x1d72, 0x1701, "XiaomiNotebook Pro", ALC298_FIXUP_DELL1_MIC_NO_PRESENCE),
+ SND_PCI_QUIRK(0x1d72, 0x1901, "RedmiBook 14", ALC256_FIXUP_ASUS_HEADSET_MIC),
+ SND_PCI_QUIRK(0x1d72, 0x1947, "RedmiBook Air", ALC255_FIXUP_XIAOMI_HEADSET_MIC),
+ SND_PCI_QUIRK(0x8086, 0x2074, "Intel NUC 8", ALC233_FIXUP_INTEL_NUC8_DMIC),
+ SND_PCI_QUIRK(0x8086, 0x2080, "Intel NUC 8 Rugged", ALC256_FIXUP_INTEL_NUC8_RUGGED),
+ SND_PCI_QUIRK(0x8086, 0x2081, "Intel NUC 10", ALC256_FIXUP_INTEL_NUC10),
#if 0
/* Below is a quirk table taken from the old code.
@@ -7320,6 +8395,7 @@
{.id = ALC269_FIXUP_HEADSET_MODE, .name = "headset-mode"},
{.id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC, .name = "headset-mode-no-hp-mic"},
{.id = ALC269_FIXUP_LENOVO_DOCK, .name = "lenovo-dock"},
+ {.id = ALC269_FIXUP_LENOVO_DOCK_LIMIT_BOOST, .name = "lenovo-dock-limit-boost"},
{.id = ALC269_FIXUP_HP_GPIO_LED, .name = "hp-gpio-led"},
{.id = ALC269_FIXUP_HP_DOCK_GPIO_MIC1_LED, .name = "hp-dock-gpio-mic1-led"},
{.id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "dell-headset-multi"},
@@ -7331,6 +8407,7 @@
{.id = ALC292_FIXUP_TPT440_DOCK, .name = "tpt440-dock"},
{.id = ALC292_FIXUP_TPT440, .name = "tpt440"},
{.id = ALC292_FIXUP_TPT460, .name = "tpt460"},
+ {.id = ALC298_FIXUP_TPT470_DOCK_FIX, .name = "tpt470-dock-fix"},
{.id = ALC298_FIXUP_TPT470_DOCK, .name = "tpt470-dock"},
{.id = ALC233_FIXUP_LENOVO_MULTI_CODECS, .name = "dual-codecs"},
{.id = ALC700_FIXUP_INTEL_REFERENCE, .name = "alc700-ref"},
@@ -7376,6 +8453,7 @@
{.id = ALC283_FIXUP_HEADSET_MIC, .name = "alc283-headset"},
{.id = ALC255_FIXUP_MIC_MUTE_LED, .name = "alc255-dell-mute"},
{.id = ALC282_FIXUP_ASPIRE_V5_PINS, .name = "aspire-v5"},
+ {.id = ALC269VB_FIXUP_ASPIRE_E1_COEF, .name = "aspire-e1-coef"},
{.id = ALC280_FIXUP_HP_GPIO4, .name = "hp-gpio4"},
{.id = ALC286_FIXUP_HP_GPIO_LED, .name = "hp-gpio-led"},
{.id = ALC280_FIXUP_HP_GPIO2_MIC_HOTKEY, .name = "hp-gpio2-hotkey"},
@@ -7390,12 +8468,12 @@
{.id = ALC298_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "alc298-dell1"},
{.id = ALC298_FIXUP_DELL_AIO_MIC_NO_PRESENCE, .name = "alc298-dell-aio"},
{.id = ALC275_FIXUP_DELL_XPS, .name = "alc275-dell-xps"},
- {.id = ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE, .name = "alc256-dell-xps13"},
{.id = ALC293_FIXUP_LENOVO_SPK_NOISE, .name = "lenovo-spk-noise"},
{.id = ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY, .name = "lenovo-hotkey"},
{.id = ALC255_FIXUP_DELL_SPK_NOISE, .name = "dell-spk-noise"},
{.id = ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "alc225-dell1"},
{.id = ALC295_FIXUP_DISABLE_DAC3, .name = "alc295-disable-dac3"},
+ {.id = ALC285_FIXUP_SPEAKER2_TO_DAC1, .name = "alc285-speaker2-to-dac1"},
{.id = ALC280_FIXUP_HP_HEADSET_MIC, .name = "alc280-hp-headset"},
{.id = ALC221_FIXUP_HP_FRONT_MIC, .name = "alc221-hp-mic"},
{.id = ALC298_FIXUP_SPK_VOLUME, .name = "alc298-spk-volume"},
@@ -7418,6 +8496,11 @@
{.id = ALC299_FIXUP_PREDATOR_SPK, .name = "predator-spk"},
{.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 = 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"},
+ {.id = ALC285_FIXUP_HP_SPECTRE_X360, .name = "alc285-hp-spectre-x360"},
{}
};
#define ALC225_STANDARD_PINS \
@@ -7512,20 +8595,6 @@
{0x19, 0x02a11020},
{0x1a, 0x02a11030},
{0x21, 0x0221101f}),
- SND_HDA_PIN_QUIRK(0x10ec0236, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE,
- {0x12, 0x90a60140},
- {0x14, 0x90170110},
- {0x21, 0x02211020}),
- SND_HDA_PIN_QUIRK(0x10ec0236, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE,
- {0x12, 0x90a60140},
- {0x14, 0x90170150},
- {0x21, 0x02211020}),
- SND_HDA_PIN_QUIRK(0x10ec0236, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE,
- {0x21, 0x02211020}),
- SND_HDA_PIN_QUIRK(0x10ec0236, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE,
- {0x12, 0x40000000},
- {0x14, 0x90170110},
- {0x21, 0x02211020}),
SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL2_MIC_NO_PRESENCE,
{0x14, 0x90170110},
{0x21, 0x02211020}),
@@ -7660,11 +8729,10 @@
{0x1a, 0x90a70130},
{0x1b, 0x90170110},
{0x21, 0x03211020}),
- SND_HDA_PIN_QUIRK(0x10ec0274, 0x1028, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB,
- {0x12, 0xb7a60130},
- {0x13, 0xb8a61140},
- {0x16, 0x90170110},
- {0x21, 0x04211020}),
+ SND_HDA_PIN_QUIRK(0x10ec0274, 0x103c, "HP", ALC274_FIXUP_HP_HEADSET_MIC,
+ {0x17, 0x90170110},
+ {0x19, 0x03a11030},
+ {0x21, 0x03211020}),
SND_HDA_PIN_QUIRK(0x10ec0280, 0x103c, "HP", ALC280_FIXUP_HP_GPIO4,
{0x12, 0x90a60130},
{0x14, 0x90170110},
@@ -7720,6 +8788,20 @@
{0x14, 0x90170110},
{0x19, 0x04a11040},
{0x21, 0x04211020}),
+ SND_HDA_PIN_QUIRK(0x10ec0285, 0x17aa, "Lenovo", ALC285_FIXUP_LENOVO_PC_BEEP_IN_NOISE,
+ {0x14, 0x90170110},
+ {0x19, 0x04a11040},
+ {0x1d, 0x40600001},
+ {0x21, 0x04211020}),
+ SND_HDA_PIN_QUIRK(0x10ec0285, 0x17aa, "Lenovo", ALC285_FIXUP_THINKPAD_NO_BASS_SPK_HEADSET_JACK,
+ {0x14, 0x90170110},
+ {0x19, 0x04a11040},
+ {0x21, 0x04211020}),
+ SND_HDA_PIN_QUIRK(0x10ec0287, 0x17aa, "Lenovo", ALC285_FIXUP_THINKPAD_HEADSET_JACK,
+ {0x14, 0x90170110},
+ {0x17, 0x90170111},
+ {0x19, 0x03a11030},
+ {0x21, 0x03211020}),
SND_HDA_PIN_QUIRK(0x10ec0286, 0x1025, "Acer", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE,
{0x12, 0x90a60130},
{0x17, 0x90170110},
@@ -7783,6 +8865,9 @@
SND_HDA_PIN_QUIRK(0x10ec0293, 0x1028, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE,
ALC292_STANDARD_PINS,
{0x13, 0x90a60140}),
+ SND_HDA_PIN_QUIRK(0x10ec0294, 0x1043, "ASUS", ALC294_FIXUP_ASUS_HPE,
+ {0x17, 0x90170110},
+ {0x21, 0x04211020}),
SND_HDA_PIN_QUIRK(0x10ec0294, 0x1043, "ASUS", ALC294_FIXUP_ASUS_MIC,
{0x14, 0x90170110},
{0x1b, 0x90a70130},
@@ -7799,6 +8884,18 @@
{0x12, 0x90a60130},
{0x17, 0x90170110},
{0x21, 0x03211020}),
+ SND_HDA_PIN_QUIRK(0x10ec0295, 0x1043, "ASUS", ALC295_FIXUP_ASUS_MIC_NO_PRESENCE,
+ {0x12, 0x90a60120},
+ {0x17, 0x90170110},
+ {0x21, 0x04211030}),
+ SND_HDA_PIN_QUIRK(0x10ec0295, 0x1043, "ASUS", ALC295_FIXUP_ASUS_MIC_NO_PRESENCE,
+ {0x12, 0x90a60130},
+ {0x17, 0x90170110},
+ {0x21, 0x03211020}),
+ SND_HDA_PIN_QUIRK(0x10ec0295, 0x1043, "ASUS", ALC295_FIXUP_ASUS_MIC_NO_PRESENCE,
+ {0x12, 0x90a60130},
+ {0x17, 0x90170110},
+ {0x21, 0x03211020}),
SND_HDA_PIN_QUIRK(0x10ec0295, 0x1028, "Dell", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE,
{0x14, 0x90170110},
{0x21, 0x04211020}),
@@ -7839,6 +8936,12 @@
ALC225_STANDARD_PINS,
{0x12, 0xb7a60130},
{0x17, 0x90170110}),
+ SND_HDA_PIN_QUIRK(0x10ec0623, 0x17aa, "Lenovo", ALC283_FIXUP_HEADSET_MIC,
+ {0x14, 0x01014010},
+ {0x17, 0x90170120},
+ {0x18, 0x02a11030},
+ {0x19, 0x02a1103f},
+ {0x21, 0x0221101f}),
{}
};
@@ -7852,6 +8955,12 @@
SND_HDA_PIN_QUIRK(0x10ec0289, 0x1028, "Dell", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE,
{0x19, 0x40000000},
{0x1b, 0x40000000}),
+ SND_HDA_PIN_QUIRK(0x10ec0274, 0x1028, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB,
+ {0x19, 0x40000000},
+ {0x1a, 0x40000000}),
+ SND_HDA_PIN_QUIRK(0x10ec0236, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE,
+ {0x19, 0x40000000},
+ {0x1a, 0x40000000}),
{}
};
@@ -7983,6 +9092,7 @@
spec->shutup = alc256_shutup;
spec->init_hook = alc256_init;
break;
+ case 0x10ec0230:
case 0x10ec0236:
case 0x10ec0256:
spec->codec_variant = ALC269_TYPE_ALC256;
@@ -7997,7 +9107,9 @@
spec->gen.mixer_nid = 0;
break;
case 0x10ec0215:
+ case 0x10ec0245:
case 0x10ec0285:
+ case 0x10ec0287:
case 0x10ec0289:
spec->codec_variant = ALC269_TYPE_ALC215;
spec->shutup = alc225_shutup;
@@ -8048,6 +9160,16 @@
snd_hda_pick_fixup(codec, alc269_fixup_models,
alc269_fixup_tbl, alc269_fixups);
+ /* FIXME: both TX300 and ROG Strix G17 have the same SSID, and
+ * the quirk breaks the latter (bko#214101).
+ * Clear the wrong entry.
+ */
+ if (codec->fixup_id == ALC282_FIXUP_ASUS_TX300 &&
+ codec->core.vendor_id == 0x10ec0294) {
+ codec_dbg(codec, "Clear wrong fixup for ASUS ROG Strix G17\n");
+ codec->fixup_id = HDA_FIXUP_ID_NOT_SET;
+ }
+
snd_hda_pick_pin_fixup(codec, alc269_pin_fixup_tbl, alc269_fixups, true);
snd_hda_pick_pin_fixup(codec, alc269_fallback_pin_fixup_tbl, alc269_fixups, false);
snd_hda_pick_fixup(codec, NULL, alc269_fixup_vendor_tbl,
@@ -8165,8 +9287,7 @@
SND_PCI_QUIRK(0x1043, 0x1393, "ASUS A6Rp", ALC861_FIXUP_ASUS_A6RP),
SND_PCI_QUIRK_VENDOR(0x1043, "ASUS laptop", ALC861_FIXUP_AMP_VREF_0F),
SND_PCI_QUIRK(0x1462, 0x7254, "HP DX2200", ALC861_FIXUP_NO_JACK_DETECT),
- SND_PCI_QUIRK(0x1584, 0x2b01, "Haier W18", ALC861_FIXUP_AMP_VREF_0F),
- SND_PCI_QUIRK(0x1584, 0x0000, "Uniwill ECS M31EI", ALC861_FIXUP_AMP_VREF_0F),
+ SND_PCI_QUIRK_VENDOR(0x1584, "Haier/Uniwill", ALC861_FIXUP_AMP_VREF_0F),
SND_PCI_QUIRK(0x1734, 0x10c7, "FSC Amilo Pi1505", ALC861_FIXUP_FSC_AMILO_PI1505),
{}
};
@@ -8455,6 +9576,8 @@
case HDA_FIXUP_ACT_PRE_PROBE:
snd_hda_jack_detect_enable_callback(codec, 0x1b,
alc662_aspire_ethos_mute_speakers);
+ /* subwoofer needs an extra GPIO setting to become audible */
+ alc_setup_gpio(codec, 0x02);
break;
case HDA_FIXUP_ACT_INIT:
/* Make sure to start in a correct state, i.e. if
@@ -8465,7 +9588,30 @@
}
}
-static struct coef_fw alc668_coefs[] = {
+static void alc671_fixup_hp_headset_mic2(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ struct alc_spec *spec = codec->spec;
+
+ static const struct hda_pintbl pincfgs[] = {
+ { 0x19, 0x02a11040 }, /* use as headset mic, with its own jack detect */
+ { 0x1b, 0x0181304f },
+ { }
+ };
+
+ switch (action) {
+ case HDA_FIXUP_ACT_PRE_PROBE:
+ spec->gen.mixer_nid = 0;
+ spec->parse_flags |= HDA_PINCFG_HEADSET_MIC;
+ snd_hda_apply_pincfgs(codec, pincfgs);
+ break;
+ case HDA_FIXUP_ACT_INIT:
+ alc_write_coef_idx(codec, 0x19, 0xa054);
+ break;
+ }
+}
+
+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),
WRITE_COEF(0x08, 0x0031), WRITE_COEF(0x0a, 0x0060), WRITE_COEF(0x0b, 0x0),
@@ -8499,6 +9645,7 @@
ALC662_FIXUP_LED_GPIO1,
ALC662_FIXUP_IDEAPAD,
ALC272_FIXUP_MARIO,
+ ALC662_FIXUP_CZC_ET26,
ALC662_FIXUP_CZC_P10T,
ALC662_FIXUP_SKU_IGNORE,
ALC662_FIXUP_HP_RP5800,
@@ -8537,8 +9684,10 @@
ALC662_FIXUP_USI_HEADSET_MODE,
ALC662_FIXUP_LENOVO_MULTI_CODECS,
ALC669_FIXUP_ACER_ASPIRE_ETHOS,
- ALC669_FIXUP_ACER_ASPIRE_ETHOS_SUBWOOFER,
ALC669_FIXUP_ACER_ASPIRE_ETHOS_HEADSET,
+ ALC671_FIXUP_HP_HEADSET_MIC2,
+ ALC662_FIXUP_ACER_X2660G_HEADSET_MODE,
+ ALC662_FIXUP_ACER_NITRO_HEADSET_MODE,
};
static const struct hda_fixup alc662_fixups[] = {
@@ -8566,6 +9715,25 @@
.type = HDA_FIXUP_FUNC,
.v.func = alc272_fixup_mario,
},
+ [ALC662_FIXUP_CZC_ET26] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ {0x12, 0x403cc000},
+ {0x14, 0x90170110}, /* speaker */
+ {0x15, 0x411111f0},
+ {0x16, 0x411111f0},
+ {0x18, 0x01a19030}, /* mic */
+ {0x19, 0x90a7013f}, /* int-mic */
+ {0x1a, 0x01014020},
+ {0x1b, 0x0121401f},
+ {0x1c, 0x411111f0},
+ {0x1d, 0x411111f0},
+ {0x1e, 0x40478e35},
+ {}
+ },
+ .chained = true,
+ .chain_id = ALC662_FIXUP_SKU_IGNORE
+ },
[ALC662_FIXUP_CZC_P10T] = {
.type = HDA_FIXUP_VERBS,
.v.verbs = (const struct hda_verb[]) {
@@ -8869,18 +10037,6 @@
.type = HDA_FIXUP_FUNC,
.v.func = alc662_fixup_aspire_ethos_hp,
},
- [ALC669_FIXUP_ACER_ASPIRE_ETHOS_SUBWOOFER] = {
- .type = HDA_FIXUP_VERBS,
- /* subwoofer needs an extra GPIO setting to become audible */
- .v.verbs = (const struct hda_verb[]) {
- {0x01, AC_VERB_SET_GPIO_MASK, 0x02},
- {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02},
- {0x01, AC_VERB_SET_GPIO_DATA, 0x00},
- { }
- },
- .chained = true,
- .chain_id = ALC669_FIXUP_ACER_ASPIRE_ETHOS_HEADSET
- },
[ALC669_FIXUP_ACER_ASPIRE_ETHOS] = {
.type = HDA_FIXUP_PINS,
.v.pins = (const struct hda_pintbl[]) {
@@ -8890,7 +10046,30 @@
{ }
},
.chained = true,
- .chain_id = ALC669_FIXUP_ACER_ASPIRE_ETHOS_SUBWOOFER
+ .chain_id = ALC669_FIXUP_ACER_ASPIRE_ETHOS_HEADSET
+ },
+ [ALC671_FIXUP_HP_HEADSET_MIC2] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc671_fixup_hp_headset_mic2,
+ },
+ [ALC662_FIXUP_ACER_X2660G_HEADSET_MODE] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x1a, 0x02a1113c }, /* use as headset mic, without its own jack detect */
+ { }
+ },
+ .chained = true,
+ .chain_id = ALC662_FIXUP_USI_FUNC
+ },
+ [ALC662_FIXUP_ACER_NITRO_HEADSET_MODE] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x1a, 0x01a11140 }, /* use as headset mic, without its own jack detect */
+ { 0x1b, 0x0221144f },
+ { }
+ },
+ .chained = true,
+ .chain_id = ALC662_FIXUP_USI_FUNC
},
};
@@ -8903,6 +10082,9 @@
SND_PCI_QUIRK(0x1025, 0x0349, "eMachines eM250", ALC662_FIXUP_INV_DMIC),
SND_PCI_QUIRK(0x1025, 0x034a, "Gateway LT27", ALC662_FIXUP_INV_DMIC),
SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE),
+ SND_PCI_QUIRK(0x1025, 0x0566, "Acer Aspire Ethos 8951G", ALC669_FIXUP_ACER_ASPIRE_ETHOS),
+ SND_PCI_QUIRK(0x1025, 0x123c, "Acer Nitro N50-600", ALC662_FIXUP_ACER_NITRO_HEADSET_MODE),
+ SND_PCI_QUIRK(0x1025, 0x124e, "Acer 2660G", ALC662_FIXUP_ACER_X2660G_HEADSET_MODE),
SND_PCI_QUIRK(0x1028, 0x05d8, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x05db, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x1028, 0x05fe, "Dell XPS 15", ALC668_FIXUP_DELL_XPS13),
@@ -8914,11 +10096,12 @@
SND_PCI_QUIRK(0x1028, 0x0698, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
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(0x1043, 0x1080, "Asus UX501VW", ALC668_FIXUP_HEADSET_MODE),
SND_PCI_QUIRK(0x1043, 0x11cd, "Asus N550", ALC662_FIXUP_ASUS_Nx50),
- SND_PCI_QUIRK(0x1043, 0x13df, "Asus N550JX", ALC662_FIXUP_BASS_1A),
SND_PCI_QUIRK(0x1043, 0x129d, "Asus N750", ALC662_FIXUP_ASUS_Nx50),
SND_PCI_QUIRK(0x1043, 0x12ff, "ASUS G751", ALC668_FIXUP_ASUS_G751),
+ SND_PCI_QUIRK(0x1043, 0x13df, "Asus N550JX", ALC662_FIXUP_BASS_1A),
SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_BASS_MODE4_CHMAP),
SND_PCI_QUIRK(0x1043, 0x15a7, "ASUS UX51VZH", ALC662_FIXUP_BASS_16),
SND_PCI_QUIRK(0x1043, 0x177d, "ASUS N551", ALC668_FIXUP_ASUS_Nx51),
@@ -8936,8 +10119,8 @@
SND_PCI_QUIRK(0x1849, 0x5892, "ASRock B150M", ALC892_FIXUP_ASROCK_MOBO),
SND_PCI_QUIRK(0x19da, 0xa130, "Zotac Z68", ALC662_FIXUP_ZOTAC_Z68),
SND_PCI_QUIRK(0x1b0a, 0x01b8, "ACER Veriton", ALC662_FIXUP_ACER_VERITON),
+ SND_PCI_QUIRK(0x1b35, 0x1234, "CZC ET26", ALC662_FIXUP_CZC_ET26),
SND_PCI_QUIRK(0x1b35, 0x2206, "CZC P10T", ALC662_FIXUP_CZC_P10T),
- SND_PCI_QUIRK(0x1025, 0x0566, "Acer Aspire Ethos 8951G", ALC669_FIXUP_ACER_ASPIRE_ETHOS),
#if 0
/* Below is a quirk table taken from the old code.
@@ -9074,6 +10257,23 @@
{0x12, 0x90a60130},
{0x14, 0x90170110},
{0x15, 0x0321101f}),
+ SND_HDA_PIN_QUIRK(0x10ec0671, 0x103c, "HP cPC", ALC671_FIXUP_HP_HEADSET_MIC2,
+ {0x14, 0x01014010},
+ {0x17, 0x90170150},
+ {0x19, 0x02a11060},
+ {0x1b, 0x01813030},
+ {0x21, 0x02211020}),
+ SND_HDA_PIN_QUIRK(0x10ec0671, 0x103c, "HP cPC", ALC671_FIXUP_HP_HEADSET_MIC2,
+ {0x14, 0x01014010},
+ {0x18, 0x01a19040},
+ {0x1b, 0x01813030},
+ {0x21, 0x02211020}),
+ SND_HDA_PIN_QUIRK(0x10ec0671, 0x103c, "HP cPC", ALC671_FIXUP_HP_HEADSET_MIC2,
+ {0x14, 0x01014020},
+ {0x17, 0x90170110},
+ {0x18, 0x01a19050},
+ {0x1b, 0x01813040},
+ {0x21, 0x02211030}),
{}
};
@@ -9194,11 +10394,13 @@
HDA_CODEC_ENTRY(0x10ec0221, "ALC221", patch_alc269),
HDA_CODEC_ENTRY(0x10ec0222, "ALC222", patch_alc269),
HDA_CODEC_ENTRY(0x10ec0225, "ALC225", patch_alc269),
+ HDA_CODEC_ENTRY(0x10ec0230, "ALC236", patch_alc269),
HDA_CODEC_ENTRY(0x10ec0231, "ALC231", patch_alc269),
HDA_CODEC_ENTRY(0x10ec0233, "ALC233", patch_alc269),
HDA_CODEC_ENTRY(0x10ec0234, "ALC234", patch_alc269),
HDA_CODEC_ENTRY(0x10ec0235, "ALC233", patch_alc269),
HDA_CODEC_ENTRY(0x10ec0236, "ALC236", patch_alc269),
+ HDA_CODEC_ENTRY(0x10ec0245, "ALC245", patch_alc269),
HDA_CODEC_ENTRY(0x10ec0255, "ALC255", patch_alc269),
HDA_CODEC_ENTRY(0x10ec0256, "ALC256", patch_alc269),
HDA_CODEC_ENTRY(0x10ec0257, "ALC257", patch_alc269),
@@ -9218,6 +10420,7 @@
HDA_CODEC_ENTRY(0x10ec0284, "ALC284", patch_alc269),
HDA_CODEC_ENTRY(0x10ec0285, "ALC285", patch_alc269),
HDA_CODEC_ENTRY(0x10ec0286, "ALC286", patch_alc269),
+ HDA_CODEC_ENTRY(0x10ec0287, "ALC287", patch_alc269),
HDA_CODEC_ENTRY(0x10ec0288, "ALC288", patch_alc269),
HDA_CODEC_ENTRY(0x10ec0289, "ALC289", patch_alc269),
HDA_CODEC_ENTRY(0x10ec0290, "ALC290", patch_alc269),
@@ -9259,8 +10462,10 @@
HDA_CODEC_ENTRY(0x10ec0888, "ALC888", patch_alc882),
HDA_CODEC_ENTRY(0x10ec0889, "ALC889", patch_alc882),
HDA_CODEC_ENTRY(0x10ec0892, "ALC892", patch_alc662),
+ HDA_CODEC_ENTRY(0x10ec0897, "ALC897", patch_alc662),
HDA_CODEC_ENTRY(0x10ec0899, "ALC898", patch_alc882),
HDA_CODEC_ENTRY(0x10ec0900, "ALC1150", patch_alc882),
+ HDA_CODEC_ENTRY(0x10ec0b00, "ALCS1200A", patch_alc882),
HDA_CODEC_ENTRY(0x10ec1168, "ALC1220", patch_alc882),
HDA_CODEC_ENTRY(0x10ec1220, "ALC1220", patch_alc882),
{} /* terminator */
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 894f3f5..bfd3fe5 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -795,7 +795,7 @@
static bool has_builtin_speaker(struct hda_codec *codec)
{
struct sigmatel_spec *spec = codec->spec;
- hda_nid_t *nid_pin;
+ const hda_nid_t *nid_pin;
int nids, i;
if (spec->gen.autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT) {
@@ -832,7 +832,7 @@
static struct snd_kcontrol_new beep_vol_ctl =
HDA_CODEC_VOLUME(NULL, 0, 0, 0);
- /* check for mute support for the the amp */
+ /* check for mute support for the amp */
if ((caps & AC_AMPCAP_MUTE) >> AC_AMPCAP_MUTE_SHIFT) {
const struct snd_kcontrol_new *temp;
if (spec->anabeep_nid == nid)
@@ -2182,7 +2182,7 @@
int action)
{
struct sigmatel_spec *spec = codec->spec;
- static hda_nid_t preferred_pairs[] = {
+ static const hda_nid_t preferred_pairs[] = {
0xd, 0x13,
0
};
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 29dcdb8..a5c1a2c 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -113,6 +113,7 @@
spec->codec_type = VT1708S;
spec->gen.indep_hp = 1;
spec->gen.keep_eapd_on = 1;
+ spec->gen.dac_min_mute = 1;
spec->gen.pcm_playback_hook = via_playback_pcm_hook;
spec->gen.add_stereo_mix_input = HDA_HINT_STEREO_MIX_AUTO;
codec->power_save_node = 1;
@@ -396,7 +397,7 @@
/* some delay here to make jack detection working (bko#98921) */
msleep(10);
codec->patch_ops.init(codec);
- regcache_sync(codec->core.regmap);
+ snd_hda_regmap_sync(codec);
return 0;
}
#endif
@@ -1002,6 +1003,7 @@
enum {
VIA_FIXUP_INTMIC_BOOST,
VIA_FIXUP_ASUS_G75,
+ VIA_FIXUP_POWER_SAVE,
};
static void via_fixup_intmic_boost(struct hda_codec *codec,
@@ -1011,6 +1013,13 @@
override_mic_boost(codec, 0x30, 0, 2, 40);
}
+static void via_fixup_power_save(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ if (action == HDA_FIXUP_ACT_PRE_PROBE)
+ codec->power_save_node = 0;
+}
+
static const struct hda_fixup via_fixups[] = {
[VIA_FIXUP_INTMIC_BOOST] = {
.type = HDA_FIXUP_FUNC,
@@ -1025,11 +1034,16 @@
{ }
}
},
+ [VIA_FIXUP_POWER_SAVE] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = via_fixup_power_save,
+ },
};
static const struct snd_pci_quirk vt2002p_fixups[] = {
SND_PCI_QUIRK(0x1043, 0x1487, "Asus G75", VIA_FIXUP_ASUS_G75),
SND_PCI_QUIRK(0x1043, 0x8532, "Asus X202E", VIA_FIXUP_INTMIC_BOOST),
+ SND_PCI_QUIRK_VENDOR(0x1558, "Clevo", VIA_FIXUP_POWER_SAVE),
{}
};
@@ -1038,8 +1052,8 @@
*/
static void fix_vt1802_connections(struct hda_codec *codec)
{
- static hda_nid_t conn_24[] = { 0x14, 0x1c };
- static hda_nid_t conn_33[] = { 0x1c };
+ static const hda_nid_t conn_24[] = { 0x14, 0x1c };
+ static const hda_nid_t conn_33[] = { 0x1c };
snd_hda_override_conn_list(codec, 0x24, ARRAY_SIZE(conn_24), conn_24);
snd_hda_override_conn_list(codec, 0x33, ARRAY_SIZE(conn_33), conn_33);
diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c
index 4b0dea7..2654eeb 100644
--- a/sound/pci/ice1712/ice1712.c
+++ b/sound/pci/ice1712/ice1712.c
@@ -2360,7 +2360,8 @@
pci_write_config_byte(ice->pci, 0x61, ice->eeprom.data[ICE_EEP1_ACLINK]);
pci_write_config_byte(ice->pci, 0x62, ice->eeprom.data[ICE_EEP1_I2SID]);
pci_write_config_byte(ice->pci, 0x63, ice->eeprom.data[ICE_EEP1_SPDIF]);
- if (ice->eeprom.subvendor != ICE1712_SUBDEVICE_STDSP24) {
+ if (ice->eeprom.subvendor != ICE1712_SUBDEVICE_STDSP24 &&
+ ice->eeprom.subvendor != ICE1712_SUBDEVICE_STAUDIO_ADCIII) {
ice->gpio.write_mask = ice->eeprom.gpiomask;
ice->gpio.direction = ice->eeprom.gpiodir;
snd_ice1712_write(ice, ICE1712_IREG_GPIO_WRITE_MASK,
diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c
index e62c118..f360b33 100644
--- a/sound/pci/ice1712/ice1724.c
+++ b/sound/pci/ice1712/ice1724.c
@@ -647,6 +647,7 @@
unsigned long flags;
unsigned char mclk_change;
unsigned int i, old_rate;
+ bool call_set_rate = false;
if (rate > ice->hw_rates->list[ice->hw_rates->count - 1])
return -EINVAL;
@@ -670,7 +671,7 @@
* setting clock rate for internal clock mode */
old_rate = ice->get_rate(ice);
if (force || (old_rate != rate))
- ice->set_rate(ice, rate);
+ call_set_rate = true;
else if (rate == ice->cur_rate) {
spin_unlock_irqrestore(&ice->reg_lock, flags);
return 0;
@@ -678,12 +679,14 @@
}
ice->cur_rate = rate;
+ spin_unlock_irqrestore(&ice->reg_lock, flags);
+
+ if (call_set_rate)
+ ice->set_rate(ice, rate);
/* setting master clock */
mclk_change = ice->set_mclk(ice, rate);
- spin_unlock_irqrestore(&ice->reg_lock, flags);
-
if (mclk_change && ice->gpio.i2s_mclk_changed)
ice->gpio.i2s_mclk_changed(ice);
if (ice->gpio.set_pro_rate)
diff --git a/sound/pci/ice1712/prodigy192.c b/sound/pci/ice1712/prodigy192.c
index 98f8ac6..243f757 100644
--- a/sound/pci/ice1712/prodigy192.c
+++ b/sound/pci/ice1712/prodigy192.c
@@ -32,7 +32,7 @@
* Experimentally I found out that only a combination of
* OCKS0=1, OCKS1=1 (128fs, 64fs output) and ice1724 -
* VT1724_MT_I2S_MCLK_128X=0 (256fs input) yields correct
- * sampling rate. That means the the FPGA doubles the
+ * sampling rate. That means that the FPGA doubles the
* MCK01 rate.
*
* Copyright (c) 2003 Takashi Iwai <tiwai@suse.de>
diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c
index 9d71e9d..3cf41c1 100644
--- a/sound/pci/ice1712/prodigy_hifi.c
+++ b/sound/pci/ice1712/prodigy_hifi.c
@@ -536,7 +536,7 @@
struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol);
mutex_lock(&ice->gpio_mutex);
- ucontrol->value.integer.value[0] = wm_get(ice, WM_ADC_MUX) & 0x1f;
+ ucontrol->value.enumerated.item[0] = wm_get(ice, WM_ADC_MUX) & 0x1f;
mutex_unlock(&ice->gpio_mutex);
return 0;
}
@@ -550,7 +550,7 @@
mutex_lock(&ice->gpio_mutex);
oval = wm_get(ice, WM_ADC_MUX);
- nval = (oval & 0xe0) | ucontrol->value.integer.value[0];
+ nval = (oval & 0xe0) | ucontrol->value.enumerated.item[0];
if (nval != oval) {
wm_put(ice, WM_ADC_MUX, nval);
change = 1;
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 6ff94d8..5150e8d 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -354,6 +354,7 @@
unsigned int ali_slot; /* ALI DMA slot */
struct ac97_pcm *pcm;
int pcm_open_flag;
+ unsigned int prepared:1;
unsigned int suspended: 1;
};
@@ -714,6 +715,9 @@
int status, civ, i, step;
int ack = 0;
+ if (!(ichdev->prepared || chip->in_measurement) || ichdev->suspended)
+ return;
+
spin_lock_irqsave(&chip->reg_lock, flags);
status = igetbyte(chip, port + ichdev->roff_sr);
civ = igetbyte(chip, port + ICH_REG_OFF_CIV);
@@ -907,6 +911,7 @@
if (ichdev->pcm_open_flag) {
snd_ac97_pcm_close(ichdev->pcm);
ichdev->pcm_open_flag = 0;
+ ichdev->prepared = 0;
}
err = snd_ac97_pcm_open(ichdev->pcm, params_rate(hw_params),
params_channels(hw_params),
@@ -928,6 +933,7 @@
if (ichdev->pcm_open_flag) {
snd_ac97_pcm_close(ichdev->pcm);
ichdev->pcm_open_flag = 0;
+ ichdev->prepared = 0;
}
return snd_pcm_lib_free_pages(substream);
}
@@ -1002,6 +1008,7 @@
ichdev->pos_shift = (runtime->sample_bits > 16) ? 2 : 1;
}
snd_intel8x0_setup_periods(chip, ichdev);
+ ichdev->prepared = 1;
return 0;
}
diff --git a/sound/pci/mixart/mixart_core.c b/sound/pci/mixart/mixart_core.c
index 048a266..363b26e 100644
--- a/sound/pci/mixart/mixart_core.c
+++ b/sound/pci/mixart/mixart_core.c
@@ -70,7 +70,6 @@
unsigned int i;
#endif
- mutex_lock(&mgr->msg_lock);
err = 0;
/* copy message descriptor from miXart to driver */
@@ -119,8 +118,6 @@
writel_be(headptr, MIXART_MEM(mgr, MSG_OUTBOUND_FREE_HEAD));
_clean_exit:
- mutex_unlock(&mgr->msg_lock);
-
return err;
}
@@ -258,7 +255,9 @@
resp.data = resp_data;
resp.size = max_resp_size;
+ mutex_lock(&mgr->msg_lock);
err = get_msg(mgr, &resp, msg_frame);
+ mutex_unlock(&mgr->msg_lock);
if( request->message_id != resp.message_id )
dev_err(&mgr->pci->dev, "RESPONSE ERROR!\n");
diff --git a/sound/pci/oxygen/xonar_dg.c b/sound/pci/oxygen/xonar_dg.c
index c3f8721..b90421a 100644
--- a/sound/pci/oxygen/xonar_dg.c
+++ b/sound/pci/oxygen/xonar_dg.c
@@ -29,7 +29,7 @@
* GPIO 4 <- headphone detect
* GPIO 5 -> enable ADC analog circuit for the left channel
* GPIO 6 -> enable ADC analog circuit for the right channel
- * GPIO 7 -> switch green rear output jack between CS4245 and and the first
+ * GPIO 7 -> switch green rear output jack between CS4245 and the first
* channel of CS4361 (mechanical relay)
* GPIO 8 -> enable output to speakers
*
diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c
index 5cbdc9b..c7b3e76 100644
--- a/sound/pci/rme9652/hdsp.c
+++ b/sound/pci/rme9652/hdsp.c
@@ -5326,7 +5326,8 @@
if (hdsp->port)
pci_release_regions(hdsp->pci);
- pci_disable_device(hdsp->pci);
+ if (pci_is_enabled(hdsp->pci))
+ pci_disable_device(hdsp->pci);
return 0;
}
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index 81a6f4b..e34f07c 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -6889,7 +6889,8 @@
if (hdspm->port)
pci_release_regions(hdspm->pci);
- pci_disable_device(hdspm->pci);
+ if (pci_is_enabled(hdspm->pci))
+ pci_disable_device(hdspm->pci);
return 0;
}
diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c
index 4c851f8..73ad6e7 100644
--- a/sound/pci/rme9652/rme9652.c
+++ b/sound/pci/rme9652/rme9652.c
@@ -1745,7 +1745,8 @@
if (rme9652->port)
pci_release_regions(rme9652->pci);
- pci_disable_device(rme9652->pci);
+ if (pci_is_enabled(rme9652->pci))
+ pci_disable_device(rme9652->pci);
return 0;
}
diff --git a/sound/ppc/powermac.c b/sound/ppc/powermac.c
index 96ef550..b135d11 100644
--- a/sound/ppc/powermac.c
+++ b/sound/ppc/powermac.c
@@ -77,7 +77,11 @@
sprintf(card->shortname, "PowerMac %s", name_ext);
sprintf(card->longname, "%s (Dev %d) Sub-frame %d",
card->shortname, chip->device_id, chip->subframe);
- if ( snd_pmac_tumbler_init(chip) < 0 || snd_pmac_tumbler_post_init() < 0)
+ err = snd_pmac_tumbler_init(chip);
+ if (err < 0)
+ goto __error;
+ err = snd_pmac_tumbler_post_init();
+ if (err < 0)
goto __error;
break;
case PMAC_AWACS:
diff --git a/sound/sh/aica.c b/sound/sh/aica.c
index 52e9cfb..8421b2f 100644
--- a/sound/sh/aica.c
+++ b/sound/sh/aica.c
@@ -101,10 +101,10 @@
}
/* spu_memload - write to SPU address space */
-static void spu_memload(u32 toi, void *from, int length)
+static void spu_memload(u32 toi, const void *from, int length)
{
unsigned long flags;
- u32 *froml = from;
+ const u32 *froml = from;
u32 __iomem *to = (u32 __iomem *) (SPU_MEMORY_BASE + toi);
int i;
u32 val;
diff --git a/sound/sh/sh_dac_audio.c b/sound/sh/sh_dac_audio.c
index ed877a1..7c46494 100644
--- a/sound/sh/sh_dac_audio.c
+++ b/sound/sh/sh_dac_audio.c
@@ -175,7 +175,6 @@
{
/* channel is not used (interleaved data) */
struct snd_sh_dac *chip = snd_pcm_substream_chip(substream);
- struct snd_pcm_runtime *runtime = substream->runtime;
if (copy_from_user_toio(chip->data_buffer + pos, src, count))
return -EFAULT;
@@ -195,7 +194,6 @@
{
/* channel is not used (interleaved data) */
struct snd_sh_dac *chip = snd_pcm_substream_chip(substream);
- struct snd_pcm_runtime *runtime = substream->runtime;
memcpy_toio(chip->data_buffer + pos, src, count);
chip->buffer_end = chip->data_buffer + pos + count;
@@ -214,7 +212,6 @@
{
/* channel is not used (interleaved data) */
struct snd_sh_dac *chip = snd_pcm_substream_chip(substream);
- struct snd_pcm_runtime *runtime = substream->runtime;
memset_io(chip->data_buffer + pos, 0, count);
chip->buffer_end = chip->data_buffer + pos + count;
diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c
index f4ee679..1612ec6 100644
--- a/sound/soc/amd/acp-da7219-max98357a.c
+++ b/sound/soc/amd/acp-da7219-max98357a.c
@@ -73,8 +73,13 @@
return ret;
}
- da7219_dai_wclk = clk_get(component->dev, "da7219-dai-wclk");
- da7219_dai_bclk = clk_get(component->dev, "da7219-dai-bclk");
+ da7219_dai_wclk = devm_clk_get(component->dev, "da7219-dai-wclk");
+ if (IS_ERR(da7219_dai_wclk))
+ return PTR_ERR(da7219_dai_wclk);
+
+ da7219_dai_bclk = devm_clk_get(component->dev, "da7219-dai-bclk");
+ if (IS_ERR(da7219_dai_bclk))
+ return PTR_ERR(da7219_dai_bclk);
ret = snd_soc_card_jack_new(card, "Headset Jack",
SND_JACK_HEADSET | SND_JACK_LINEOUT |
diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig
index f118c22..51e75b7 100644
--- a/sound/soc/atmel/Kconfig
+++ b/sound/soc/atmel/Kconfig
@@ -10,15 +10,16 @@
if SND_ATMEL_SOC
config SND_ATMEL_SOC_PDC
- tristate
- depends on HAS_DMA
+ bool
config SND_ATMEL_SOC_DMA
- tristate
+ bool
select SND_SOC_GENERIC_DMAENGINE_PCM
config SND_ATMEL_SOC_SSC
tristate
+ select SND_ATMEL_SOC_DMA
+ select SND_ATMEL_SOC_PDC
config SND_ATMEL_SOC_SSC_PDC
tristate "SoC PCM DAI support for AT91 SSC controller using PDC"
diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile
index 1f6890e..c7d2989 100644
--- a/sound/soc/atmel/Makefile
+++ b/sound/soc/atmel/Makefile
@@ -6,8 +6,14 @@
snd-soc-atmel-i2s-objs := atmel-i2s.o
snd-soc-mchp-i2s-mcc-objs := mchp-i2s-mcc.o
-obj-$(CONFIG_SND_ATMEL_SOC_PDC) += snd-soc-atmel-pcm-pdc.o
-obj-$(CONFIG_SND_ATMEL_SOC_DMA) += snd-soc-atmel-pcm-dma.o
+# pdc and dma need to both be built-in if any user of
+# ssc is built-in.
+ifdef CONFIG_SND_ATMEL_SOC_PDC
+obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel-pcm-pdc.o
+endif
+ifdef CONFIG_SND_ATMEL_SOC_DMA
+obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel-pcm-dma.o
+endif
obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel_ssc_dai.o
obj-$(CONFIG_SND_ATMEL_SOC_I2S) += snd-soc-atmel-i2s.o
obj-$(CONFIG_SND_MCHP_SOC_I2S_MCC) += snd-soc-mchp-i2s-mcc.o
diff --git a/sound/soc/atmel/atmel-i2s.c b/sound/soc/atmel/atmel-i2s.c
index bbe2b63..d870f56 100644
--- a/sound/soc/atmel/atmel-i2s.c
+++ b/sound/soc/atmel/atmel-i2s.c
@@ -200,6 +200,7 @@
unsigned int fmt;
const struct atmel_i2s_gck_param *gck_param;
const struct atmel_i2s_caps *caps;
+ int clk_use_no;
};
static irqreturn_t atmel_i2s_interrupt(int irq, void *dev_id)
@@ -321,9 +322,16 @@
{
struct atmel_i2s_dev *dev = snd_soc_dai_get_drvdata(dai);
bool is_playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
- unsigned int mr = 0;
+ unsigned int mr = 0, mr_mask;
int ret;
+ mr_mask = ATMEL_I2SC_MR_FORMAT_MASK | ATMEL_I2SC_MR_MODE_MASK |
+ ATMEL_I2SC_MR_DATALENGTH_MASK;
+ if (is_playback)
+ mr_mask |= ATMEL_I2SC_MR_TXMONO;
+ else
+ mr_mask |= ATMEL_I2SC_MR_RXMONO;
+
switch (dev->fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
mr |= ATMEL_I2SC_MR_FORMAT_I2S;
@@ -402,7 +410,7 @@
return -EINVAL;
}
- return regmap_write(dev->regmap, ATMEL_I2SC_MR, mr);
+ return regmap_update_bits(dev->regmap, ATMEL_I2SC_MR, mr_mask, mr);
}
static int atmel_i2s_switch_mck_generator(struct atmel_i2s_dev *dev,
@@ -495,18 +503,28 @@
is_master = (mr & ATMEL_I2SC_MR_MODE_MASK) == ATMEL_I2SC_MR_MODE_MASTER;
/* If master starts, enable the audio clock. */
- if (is_master && mck_enabled)
- err = atmel_i2s_switch_mck_generator(dev, true);
- if (err)
- return err;
+ if (is_master && mck_enabled) {
+ if (!dev->clk_use_no) {
+ err = atmel_i2s_switch_mck_generator(dev, true);
+ if (err)
+ return err;
+ }
+ dev->clk_use_no++;
+ }
err = regmap_write(dev->regmap, ATMEL_I2SC_CR, cr);
if (err)
return err;
/* If master stops, disable the audio clock. */
- if (is_master && !mck_enabled)
- err = atmel_i2s_switch_mck_generator(dev, false);
+ if (is_master && !mck_enabled) {
+ if (dev->clk_use_no == 1) {
+ err = atmel_i2s_switch_mck_generator(dev, false);
+ if (err)
+ return err;
+ }
+ dev->clk_use_no--;
+ }
return err;
}
diff --git a/sound/soc/codecs/ak4458.c b/sound/soc/codecs/ak4458.c
index 7156215..eca5fc5 100644
--- a/sound/soc/codecs/ak4458.c
+++ b/sound/soc/codecs/ak4458.c
@@ -523,18 +523,10 @@
.ops = &ak4458_dai_ops,
};
-static void ak4458_power_off(struct ak4458_priv *ak4458)
+static void ak4458_reset(struct ak4458_priv *ak4458, bool active)
{
if (ak4458->reset_gpiod) {
- gpiod_set_value_cansleep(ak4458->reset_gpiod, 0);
- usleep_range(1000, 2000);
- }
-}
-
-static void ak4458_power_on(struct ak4458_priv *ak4458)
-{
- if (ak4458->reset_gpiod) {
- gpiod_set_value_cansleep(ak4458->reset_gpiod, 1);
+ gpiod_set_value_cansleep(ak4458->reset_gpiod, active);
usleep_range(1000, 2000);
}
}
@@ -548,7 +540,7 @@
if (ak4458->mute_gpiod)
gpiod_set_value_cansleep(ak4458->mute_gpiod, 1);
- ak4458_power_on(ak4458);
+ ak4458_reset(ak4458, false);
ret = snd_soc_component_update_bits(component, AK4458_00_CONTROL1,
0x80, 0x80); /* ACKS bit = 1; 10000000 */
@@ -571,7 +563,7 @@
{
struct ak4458_priv *ak4458 = snd_soc_component_get_drvdata(component);
- ak4458_power_off(ak4458);
+ ak4458_reset(ak4458, true);
}
#ifdef CONFIG_PM
@@ -581,7 +573,7 @@
regcache_cache_only(ak4458->regmap, true);
- ak4458_power_off(ak4458);
+ ak4458_reset(ak4458, true);
if (ak4458->mute_gpiod)
gpiod_set_value_cansleep(ak4458->mute_gpiod, 0);
@@ -596,8 +588,8 @@
if (ak4458->mute_gpiod)
gpiod_set_value_cansleep(ak4458->mute_gpiod, 1);
- ak4458_power_off(ak4458);
- ak4458_power_on(ak4458);
+ ak4458_reset(ak4458, true);
+ ak4458_reset(ak4458, false);
regcache_cache_only(ak4458->regmap, false);
regcache_mark_dirty(ak4458->regmap);
@@ -715,6 +707,7 @@
{ .compatible = "asahi-kasei,ak4497", .data = &ak4497_drvdata},
{ },
};
+MODULE_DEVICE_TABLE(of, ak4458_of_match);
static struct i2c_driver ak4458_i2c_driver = {
.driver = {
diff --git a/sound/soc/codecs/ak5558.c b/sound/soc/codecs/ak5558.c
index 8179512..e98312a 100644
--- a/sound/soc/codecs/ak5558.c
+++ b/sound/soc/codecs/ak5558.c
@@ -264,7 +264,7 @@
if (!ak5558->reset_gpiod)
return;
- gpiod_set_value_cansleep(ak5558->reset_gpiod, 0);
+ gpiod_set_value_cansleep(ak5558->reset_gpiod, 1);
usleep_range(1000, 2000);
}
@@ -273,7 +273,7 @@
if (!ak5558->reset_gpiod)
return;
- gpiod_set_value_cansleep(ak5558->reset_gpiod, 1);
+ gpiod_set_value_cansleep(ak5558->reset_gpiod, 0);
usleep_range(1000, 2000);
}
@@ -389,6 +389,7 @@
{ .compatible = "asahi-kasei,ak5558"},
{ }
};
+MODULE_DEVICE_TABLE(of, ak5558_i2c_dt_ids);
static struct i2c_driver ak5558_i2c_driver = {
.driver = {
diff --git a/sound/soc/codecs/cpcap.c b/sound/soc/codecs/cpcap.c
index d7f05b3..1902689 100644
--- a/sound/soc/codecs/cpcap.c
+++ b/sound/soc/codecs/cpcap.c
@@ -1263,12 +1263,12 @@
if (direction == SNDRV_PCM_STREAM_CAPTURE) {
mask = 0x0000;
- mask |= CPCAP_BIT_MIC1_RX_TIMESLOT0;
- mask |= CPCAP_BIT_MIC1_RX_TIMESLOT1;
- mask |= CPCAP_BIT_MIC1_RX_TIMESLOT2;
- mask |= CPCAP_BIT_MIC2_TIMESLOT0;
- mask |= CPCAP_BIT_MIC2_TIMESLOT1;
- mask |= CPCAP_BIT_MIC2_TIMESLOT2;
+ mask |= BIT(CPCAP_BIT_MIC1_RX_TIMESLOT0);
+ mask |= BIT(CPCAP_BIT_MIC1_RX_TIMESLOT1);
+ mask |= BIT(CPCAP_BIT_MIC1_RX_TIMESLOT2);
+ mask |= BIT(CPCAP_BIT_MIC2_TIMESLOT0);
+ mask |= BIT(CPCAP_BIT_MIC2_TIMESLOT1);
+ mask |= BIT(CPCAP_BIT_MIC2_TIMESLOT2);
val = 0x0000;
if (channels >= 2)
val = BIT(CPCAP_BIT_MIC1_RX_TIMESLOT0);
diff --git a/sound/soc/codecs/cs35l33.c b/sound/soc/codecs/cs35l33.c
index 6042194..8894369 100644
--- a/sound/soc/codecs/cs35l33.c
+++ b/sound/soc/codecs/cs35l33.c
@@ -1201,6 +1201,7 @@
dev_err(&i2c_client->dev,
"CS35L33 Device ID (%X). Expected ID %X\n",
devid, CS35L33_CHIP_ID);
+ ret = -EINVAL;
goto err_enable;
}
diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c
index 5125bb9..6825e87 100644
--- a/sound/soc/codecs/cs42l42.c
+++ b/sound/soc/codecs/cs42l42.c
@@ -398,10 +398,13 @@
.reg_defaults = cs42l42_reg_defaults,
.num_reg_defaults = ARRAY_SIZE(cs42l42_reg_defaults),
.cache_type = REGCACHE_RBTREE,
+
+ .use_single_read = true,
+ .use_single_write = true,
};
-static DECLARE_TLV_DB_SCALE(adc_tlv, -9600, 100, false);
-static DECLARE_TLV_DB_SCALE(mixer_tlv, -6200, 100, false);
+static DECLARE_TLV_DB_SCALE(adc_tlv, -9700, 100, true);
+static DECLARE_TLV_DB_SCALE(mixer_tlv, -6300, 100, true);
static const char * const cs42l42_hpf_freq_text[] = {
"1.86Hz", "120Hz", "235Hz", "466Hz"
@@ -420,34 +423,23 @@
CS42L42_ADC_WNF_CF_SHIFT,
cs42l42_wnf3_freq_text);
-static const char * const cs42l42_wnf05_freq_text[] = {
- "280Hz", "315Hz", "350Hz", "385Hz",
- "420Hz", "455Hz", "490Hz", "525Hz"
-};
-
-static SOC_ENUM_SINGLE_DECL(cs42l42_wnf05_freq_enum, CS42L42_ADC_WNF_HPF_CTL,
- CS42L42_ADC_WNF_CF_SHIFT,
- cs42l42_wnf05_freq_text);
-
static const struct snd_kcontrol_new cs42l42_snd_controls[] = {
/* ADC Volume and Filter Controls */
SOC_SINGLE("ADC Notch Switch", CS42L42_ADC_CTL,
- CS42L42_ADC_NOTCH_DIS_SHIFT, true, false),
+ CS42L42_ADC_NOTCH_DIS_SHIFT, true, true),
SOC_SINGLE("ADC Weak Force Switch", CS42L42_ADC_CTL,
CS42L42_ADC_FORCE_WEAK_VCM_SHIFT, true, false),
SOC_SINGLE("ADC Invert Switch", CS42L42_ADC_CTL,
CS42L42_ADC_INV_SHIFT, true, false),
SOC_SINGLE("ADC Boost Switch", CS42L42_ADC_CTL,
CS42L42_ADC_DIG_BOOST_SHIFT, true, false),
- SOC_SINGLE_SX_TLV("ADC Volume", CS42L42_ADC_VOLUME,
- CS42L42_ADC_VOL_SHIFT, 0xA0, 0x6C, adc_tlv),
+ SOC_SINGLE_S8_TLV("ADC Volume", CS42L42_ADC_VOLUME, -97, 12, adc_tlv),
SOC_SINGLE("ADC WNF Switch", CS42L42_ADC_WNF_HPF_CTL,
CS42L42_ADC_WNF_EN_SHIFT, true, false),
SOC_SINGLE("ADC HPF Switch", CS42L42_ADC_WNF_HPF_CTL,
CS42L42_ADC_HPF_EN_SHIFT, true, false),
SOC_ENUM("HPF Corner Freq", cs42l42_hpf_freq_enum),
SOC_ENUM("WNF 3dB Freq", cs42l42_wnf3_freq_enum),
- SOC_ENUM("WNF 05dB Freq", cs42l42_wnf05_freq_enum),
/* DAC Volume and Filter Controls */
SOC_SINGLE("DACA Invert Switch", CS42L42_DAC_CTL1,
@@ -458,7 +450,7 @@
CS42L42_DAC_HPF_EN_SHIFT, true, false),
SOC_DOUBLE_R_TLV("Mixer Volume", CS42L42_MIXER_CHA_VOL,
CS42L42_MIXER_CHB_VOL, CS42L42_MIXER_CH_VOL_SHIFT,
- 0x3e, 1, mixer_tlv)
+ 0x3f, 1, mixer_tlv)
};
static int cs42l42_hpdrv_evt(struct snd_soc_dapm_widget *w,
@@ -666,15 +658,6 @@
CS42L42_FSYNC_PULSE_WIDTH_MASK,
CS42L42_FRAC1_VAL(fsync - 1) <<
CS42L42_FSYNC_PULSE_WIDTH_SHIFT);
- snd_soc_component_update_bits(component,
- CS42L42_ASP_FRM_CFG,
- CS42L42_ASP_5050_MASK,
- CS42L42_ASP_5050_MASK);
- /* Set the frame delay to 1.0 SCLK clocks */
- snd_soc_component_update_bits(component, CS42L42_ASP_FRM_CFG,
- CS42L42_ASP_FSD_MASK,
- CS42L42_ASP_FSD_1_0 <<
- CS42L42_ASP_FSD_SHIFT);
/* Set the sample rates (96k or lower) */
snd_soc_component_update_bits(component, CS42L42_FS_RATE_EN,
CS42L42_FS_EN_MASK,
@@ -691,24 +674,6 @@
CS42L42_CLK_OASRC_SEL_MASK,
CS42L42_CLK_OASRC_SEL_12 <<
CS42L42_CLK_OASRC_SEL_SHIFT);
- /* channel 1 on low LRCLK, 32 bit */
- snd_soc_component_update_bits(component,
- CS42L42_ASP_RX_DAI0_CH1_AP_RES,
- CS42L42_ASP_RX_CH_AP_MASK |
- CS42L42_ASP_RX_CH_RES_MASK,
- (CS42L42_ASP_RX_CH_AP_LOW <<
- CS42L42_ASP_RX_CH_AP_SHIFT) |
- (CS42L42_ASP_RX_CH_RES_32 <<
- CS42L42_ASP_RX_CH_RES_SHIFT));
- /* Channel 2 on high LRCLK, 32 bit */
- snd_soc_component_update_bits(component,
- CS42L42_ASP_RX_DAI0_CH2_AP_RES,
- CS42L42_ASP_RX_CH_AP_MASK |
- CS42L42_ASP_RX_CH_RES_MASK,
- (CS42L42_ASP_RX_CH_AP_HI <<
- CS42L42_ASP_RX_CH_AP_SHIFT) |
- (CS42L42_ASP_RX_CH_RES_32 <<
- CS42L42_ASP_RX_CH_RES_SHIFT));
if (pll_ratio_table[i].mclk_src_sel == 0) {
/* Pass the clock straight through */
snd_soc_component_update_bits(component,
@@ -788,7 +753,18 @@
/* interface format */
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
- case SND_SOC_DAIFMT_LEFT_J:
+ /*
+ * 5050 mode, frame starts on falling edge of LRCLK,
+ * frame delayed by 1.0 SCLKs
+ */
+ snd_soc_component_update_bits(component,
+ CS42L42_ASP_FRM_CFG,
+ CS42L42_ASP_STP_MASK |
+ CS42L42_ASP_5050_MASK |
+ CS42L42_ASP_FSD_MASK,
+ CS42L42_ASP_5050_MASK |
+ (CS42L42_ASP_FSD_1_0 <<
+ CS42L42_ASP_FSD_SHIFT));
break;
default:
return -EINVAL;
@@ -797,27 +773,23 @@
/* Bitclock/frame inversion */
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
case SND_SOC_DAIFMT_NB_NF:
+ asp_cfg_val |= CS42L42_ASP_SCPOL_NOR << CS42L42_ASP_SCPOL_SHIFT;
break;
case SND_SOC_DAIFMT_NB_IF:
- asp_cfg_val |= CS42L42_ASP_POL_INV <<
- CS42L42_ASP_LCPOL_IN_SHIFT;
+ asp_cfg_val |= CS42L42_ASP_SCPOL_NOR << CS42L42_ASP_SCPOL_SHIFT;
+ asp_cfg_val |= CS42L42_ASP_LCPOL_INV << CS42L42_ASP_LCPOL_SHIFT;
break;
case SND_SOC_DAIFMT_IB_NF:
- asp_cfg_val |= CS42L42_ASP_POL_INV <<
- CS42L42_ASP_SCPOL_IN_DAC_SHIFT;
break;
case SND_SOC_DAIFMT_IB_IF:
- asp_cfg_val |= CS42L42_ASP_POL_INV <<
- CS42L42_ASP_LCPOL_IN_SHIFT;
- asp_cfg_val |= CS42L42_ASP_POL_INV <<
- CS42L42_ASP_SCPOL_IN_DAC_SHIFT;
+ asp_cfg_val |= CS42L42_ASP_LCPOL_INV << CS42L42_ASP_LCPOL_SHIFT;
break;
}
- snd_soc_component_update_bits(component, CS42L42_ASP_CLK_CFG,
- CS42L42_ASP_MODE_MASK |
- CS42L42_ASP_SCPOL_IN_DAC_MASK |
- CS42L42_ASP_LCPOL_IN_MASK, asp_cfg_val);
+ snd_soc_component_update_bits(component, CS42L42_ASP_CLK_CFG, CS42L42_ASP_MODE_MASK |
+ CS42L42_ASP_SCPOL_MASK |
+ CS42L42_ASP_LCPOL_MASK,
+ asp_cfg_val);
return 0;
}
@@ -828,14 +800,29 @@
{
struct snd_soc_component *component = dai->component;
struct cs42l42_private *cs42l42 = snd_soc_component_get_drvdata(component);
- int retval;
+ unsigned int width = (params_width(params) / 8) - 1;
+ unsigned int val = 0;
cs42l42->srate = params_rate(params);
- cs42l42->swidth = params_width(params);
- retval = cs42l42_pll_config(component);
+ switch(substream->stream) {
+ case SNDRV_PCM_STREAM_PLAYBACK:
+ val |= width << CS42L42_ASP_RX_CH_RES_SHIFT;
+ /* channel 1 on low LRCLK */
+ snd_soc_component_update_bits(component, CS42L42_ASP_RX_DAI0_CH1_AP_RES,
+ CS42L42_ASP_RX_CH_AP_MASK |
+ CS42L42_ASP_RX_CH_RES_MASK, val);
+ /* Channel 2 on high LRCLK */
+ val |= CS42L42_ASP_RX_CH_AP_HI << CS42L42_ASP_RX_CH_AP_SHIFT;
+ snd_soc_component_update_bits(component, CS42L42_ASP_RX_DAI0_CH2_AP_RES,
+ CS42L42_ASP_RX_CH_AP_MASK |
+ CS42L42_ASP_RX_CH_RES_MASK, val);
+ break;
+ default:
+ break;
+ }
- return retval;
+ return cs42l42_pll_config(component);
}
static int cs42l42_set_sysclk(struct snd_soc_dai *dai,
@@ -900,9 +887,9 @@
return 0;
}
-#define CS42L42_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S18_3LE | \
- SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE | \
- SNDRV_PCM_FMTBIT_S32_LE)
+#define CS42L42_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\
+ SNDRV_PCM_FMTBIT_S24_LE |\
+ SNDRV_PCM_FMTBIT_S32_LE )
static const struct snd_soc_dai_ops cs42l42_ops = {
@@ -1803,7 +1790,7 @@
dev_dbg(&i2c_client->dev, "Found reset GPIO\n");
gpiod_set_value_cansleep(cs42l42->reset_gpio, 1);
}
- mdelay(3);
+ usleep_range(CS42L42_BOOT_TIME_US, CS42L42_BOOT_TIME_US * 2);
/* Request IRQ */
ret = devm_request_threaded_irq(&i2c_client->dev,
@@ -1928,6 +1915,7 @@
}
gpiod_set_value_cansleep(cs42l42->reset_gpio, 1);
+ usleep_range(CS42L42_BOOT_TIME_US, CS42L42_BOOT_TIME_US * 2);
regcache_cache_only(cs42l42->regmap, false);
regcache_sync(cs42l42->regmap);
diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h
index 9e3cc52..ca20197 100644
--- a/sound/soc/codecs/cs42l42.h
+++ b/sound/soc/codecs/cs42l42.h
@@ -77,7 +77,7 @@
#define CS42L42_HP_PDN_SHIFT 3
#define CS42L42_HP_PDN_MASK (1 << CS42L42_HP_PDN_SHIFT)
#define CS42L42_ADC_PDN_SHIFT 2
-#define CS42L42_ADC_PDN_MASK (1 << CS42L42_HP_PDN_SHIFT)
+#define CS42L42_ADC_PDN_MASK (1 << CS42L42_ADC_PDN_SHIFT)
#define CS42L42_PDN_ALL_SHIFT 0
#define CS42L42_PDN_ALL_MASK (1 << CS42L42_PDN_ALL_SHIFT)
@@ -258,11 +258,12 @@
#define CS42L42_ASP_SLAVE_MODE 0x00
#define CS42L42_ASP_MODE_SHIFT 4
#define CS42L42_ASP_MODE_MASK (1 << CS42L42_ASP_MODE_SHIFT)
-#define CS42L42_ASP_SCPOL_IN_DAC_SHIFT 2
-#define CS42L42_ASP_SCPOL_IN_DAC_MASK (1 << CS42L42_ASP_SCPOL_IN_DAC_SHIFT)
-#define CS42L42_ASP_LCPOL_IN_SHIFT 0
-#define CS42L42_ASP_LCPOL_IN_MASK (1 << CS42L42_ASP_LCPOL_IN_SHIFT)
-#define CS42L42_ASP_POL_INV 1
+#define CS42L42_ASP_SCPOL_SHIFT 2
+#define CS42L42_ASP_SCPOL_MASK (3 << CS42L42_ASP_SCPOL_SHIFT)
+#define CS42L42_ASP_SCPOL_NOR 3
+#define CS42L42_ASP_LCPOL_SHIFT 0
+#define CS42L42_ASP_LCPOL_MASK (3 << CS42L42_ASP_LCPOL_SHIFT)
+#define CS42L42_ASP_LCPOL_INV 3
#define CS42L42_ASP_FRM_CFG (CS42L42_PAGE_12 + 0x08)
#define CS42L42_ASP_STP_SHIFT 4
@@ -739,6 +740,7 @@
#define CS42L42_FRAC2_VAL(val) (((val) & 0xff0000) >> 16)
#define CS42L42_NUM_SUPPLIES 5
+#define CS42L42_BOOT_TIME_US 3000
static const char *const cs42l42_supply_names[CS42L42_NUM_SUPPLIES] = {
"VA",
@@ -756,7 +758,6 @@
struct completion pdn_done;
u32 sclk;
u32 srate;
- u32 swidth;
u8 plug_state;
u8 hs_type;
u8 ts_inv;
diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c
index 55408c8..cdd7ae9 100644
--- a/sound/soc/codecs/cs42l51.c
+++ b/sound/soc/codecs/cs42l51.c
@@ -247,8 +247,28 @@
&cs42l51_adcr_mux_controls),
};
+static int mclk_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm);
+ struct cs42l51_private *cs42l51 = snd_soc_component_get_drvdata(comp);
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ return clk_prepare_enable(cs42l51->mclk_handle);
+ case SND_SOC_DAPM_POST_PMD:
+ /* Delay mclk shutdown to fulfill power-down sequence requirements */
+ msleep(20);
+ clk_disable_unprepare(cs42l51->mclk_handle);
+ break;
+ }
+
+ return 0;
+}
+
static const struct snd_soc_dapm_widget cs42l51_dapm_mclk_widgets[] = {
- SND_SOC_DAPM_CLOCK_SUPPLY("MCLK")
+ SND_SOC_DAPM_SUPPLY("MCLK", SND_SOC_NOPM, 0, 0, mclk_event,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
};
static const struct snd_soc_dapm_route cs42l51_routes[] = {
diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c
index ac569ab..51d7a87 100644
--- a/sound/soc/codecs/cs42l56.c
+++ b/sound/soc/codecs/cs42l56.c
@@ -1248,6 +1248,7 @@
dev_err(&i2c_client->dev,
"CS42L56 Device ID (%X). Expected %X\n",
devid, CS42L56_DEVID);
+ ret = -EINVAL;
goto err_enable;
}
alpha_rev = reg & CS42L56_AREV_MASK;
@@ -1305,7 +1306,7 @@
ret = devm_snd_soc_register_component(&i2c_client->dev,
&soc_component_dev_cs42l56, &cs42l56_dai, 1);
if (ret < 0)
- return ret;
+ goto err_enable;
return 0;
diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c
index 7fb3442..8f70dee 100644
--- a/sound/soc/codecs/cs43130.c
+++ b/sound/soc/codecs/cs43130.c
@@ -1735,6 +1735,14 @@
static DEVICE_ATTR(hpload_ac_l, 0444, cs43130_show_ac_l, NULL);
static DEVICE_ATTR(hpload_ac_r, 0444, cs43130_show_ac_r, NULL);
+static struct attribute *hpload_attrs[] = {
+ &dev_attr_hpload_dc_l.attr,
+ &dev_attr_hpload_dc_r.attr,
+ &dev_attr_hpload_ac_l.attr,
+ &dev_attr_hpload_ac_r.attr,
+};
+ATTRIBUTE_GROUPS(hpload);
+
static struct reg_sequence hp_en_cal_seq[] = {
{CS43130_INT_MASK_4, CS43130_INT_MASK_ALL},
{CS43130_HP_MEAS_LOAD_1, 0},
@@ -2302,25 +2310,15 @@
cs43130->hpload_done = false;
if (cs43130->dc_meas) {
- ret = device_create_file(component->dev, &dev_attr_hpload_dc_l);
- if (ret < 0)
- return ret;
-
- ret = device_create_file(component->dev, &dev_attr_hpload_dc_r);
- if (ret < 0)
- return ret;
-
- ret = device_create_file(component->dev, &dev_attr_hpload_ac_l);
- if (ret < 0)
- return ret;
-
- ret = device_create_file(component->dev, &dev_attr_hpload_ac_r);
- if (ret < 0)
+ ret = sysfs_create_groups(&component->dev->kobj, hpload_groups);
+ if (ret)
return ret;
cs43130->wq = create_singlethread_workqueue("cs43130_hp");
- if (!cs43130->wq)
+ if (!cs43130->wq) {
+ sysfs_remove_groups(&component->dev->kobj, hpload_groups);
return -ENOMEM;
+ }
INIT_WORK(&cs43130->work, cs43130_imp_meas);
}
diff --git a/sound/soc/codecs/cx2072x.c b/sound/soc/codecs/cx2072x.c
index 1c1ba7b..8ee4b2e 100644
--- a/sound/soc/codecs/cx2072x.c
+++ b/sound/soc/codecs/cx2072x.c
@@ -1579,7 +1579,7 @@
.id = CX2072X_DAI_DSP,
.probe = cx2072x_dsp_dai_probe,
.playback = {
- .stream_name = "Playback",
+ .stream_name = "DSP Playback",
.channels_min = 2,
.channels_max = 2,
.rates = CX2072X_RATES_DSP,
@@ -1591,7 +1591,7 @@
.name = "cx2072x-aec",
.id = 3,
.capture = {
- .stream_name = "Capture",
+ .stream_name = "AEC Capture",
.channels_min = 2,
.channels_max = 2,
.rates = CX2072X_RATES_DSP,
diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c
index 36eef1f..b781b28 100644
--- a/sound/soc/codecs/es8316.c
+++ b/sound/soc/codecs/es8316.c
@@ -63,13 +63,8 @@
1, 1, TLV_DB_SCALE_ITEM(0, 0, 0),
2, 2, TLV_DB_SCALE_ITEM(250, 0, 0),
3, 3, TLV_DB_SCALE_ITEM(450, 0, 0),
- 4, 4, TLV_DB_SCALE_ITEM(700, 0, 0),
- 5, 5, TLV_DB_SCALE_ITEM(1000, 0, 0),
- 6, 6, TLV_DB_SCALE_ITEM(1300, 0, 0),
- 7, 7, TLV_DB_SCALE_ITEM(1600, 0, 0),
- 8, 8, TLV_DB_SCALE_ITEM(1800, 0, 0),
- 9, 9, TLV_DB_SCALE_ITEM(2100, 0, 0),
- 10, 10, TLV_DB_SCALE_ITEM(2400, 0, 0),
+ 4, 7, TLV_DB_SCALE_ITEM(700, 300, 0),
+ 8, 10, TLV_DB_SCALE_ITEM(1800, 300, 0),
);
static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(hpout_vol_tlv,
diff --git a/sound/soc/codecs/hdac_hda.c b/sound/soc/codecs/hdac_hda.c
index 4570f66..d78f4d8 100644
--- a/sound/soc/codecs/hdac_hda.c
+++ b/sound/soc/codecs/hdac_hda.c
@@ -498,7 +498,9 @@
struct hdac_hda_priv *hda_pvt;
hda_pvt = dev_get_drvdata(&hdev->dev);
- cancel_delayed_work_sync(&hda_pvt->codec.jackpoll_work);
+ if (hda_pvt && hda_pvt->codec.registered)
+ cancel_delayed_work_sync(&hda_pvt->codec.jackpoll_work);
+
return 0;
}
diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c
index 18c173e..78d5b4d 100644
--- a/sound/soc/codecs/hdac_hdmi.c
+++ b/sound/soc/codecs/hdac_hdmi.c
@@ -150,14 +150,14 @@
hdac_hdmi_get_pcm_from_cvt(struct hdac_hdmi_priv *hdmi,
struct hdac_hdmi_cvt *cvt)
{
- struct hdac_hdmi_pcm *pcm = NULL;
+ struct hdac_hdmi_pcm *pcm;
list_for_each_entry(pcm, &hdmi->pcm_list, head) {
if (pcm->cvt == cvt)
- break;
+ return pcm;
}
- return pcm;
+ return NULL;
}
static void hdac_hdmi_jack_report(struct hdac_hdmi_pcm *pcm,
diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c
index f031d2c..fa4cdbf 100644
--- a/sound/soc/codecs/max98088.c
+++ b/sound/soc/codecs/max98088.c
@@ -41,6 +41,7 @@
enum max98088_type devtype;
struct max98088_pdata *pdata;
struct clk *mclk;
+ unsigned char mclk_prescaler;
unsigned int sysclk;
struct max98088_cdata dai[2];
int eq_textcnt;
@@ -998,13 +999,16 @@
/* Configure NI when operating as master */
if (snd_soc_component_read32(component, M98088_REG_14_DAI1_FORMAT)
& M98088_DAI_MAS) {
+ unsigned long pclk;
+
if (max98088->sysclk == 0) {
dev_err(component->dev, "Invalid system clock frequency\n");
return -EINVAL;
}
ni = 65536ULL * (rate < 50000 ? 96ULL : 48ULL)
* (unsigned long long int)rate;
- do_div(ni, (unsigned long long int)max98088->sysclk);
+ pclk = DIV_ROUND_CLOSEST(max98088->sysclk, max98088->mclk_prescaler);
+ ni = DIV_ROUND_CLOSEST_ULL(ni, pclk);
snd_soc_component_write(component, M98088_REG_12_DAI1_CLKCFG_HI,
(ni >> 8) & 0x7F);
snd_soc_component_write(component, M98088_REG_13_DAI1_CLKCFG_LO,
@@ -1065,13 +1069,16 @@
/* Configure NI when operating as master */
if (snd_soc_component_read32(component, M98088_REG_1C_DAI2_FORMAT)
& M98088_DAI_MAS) {
+ unsigned long pclk;
+
if (max98088->sysclk == 0) {
dev_err(component->dev, "Invalid system clock frequency\n");
return -EINVAL;
}
ni = 65536ULL * (rate < 50000 ? 96ULL : 48ULL)
* (unsigned long long int)rate;
- do_div(ni, (unsigned long long int)max98088->sysclk);
+ pclk = DIV_ROUND_CLOSEST(max98088->sysclk, max98088->mclk_prescaler);
+ ni = DIV_ROUND_CLOSEST_ULL(ni, pclk);
snd_soc_component_write(component, M98088_REG_1A_DAI2_CLKCFG_HI,
(ni >> 8) & 0x7F);
snd_soc_component_write(component, M98088_REG_1B_DAI2_CLKCFG_LO,
@@ -1113,8 +1120,10 @@
*/
if ((freq >= 10000000) && (freq < 20000000)) {
snd_soc_component_write(component, M98088_REG_10_SYS_CLK, 0x10);
+ max98088->mclk_prescaler = 1;
} else if ((freq >= 20000000) && (freq < 30000000)) {
snd_soc_component_write(component, M98088_REG_10_SYS_CLK, 0x20);
+ max98088->mclk_prescaler = 2;
} else {
dev_err(component->dev, "Invalid master clock frequency\n");
return -EINVAL;
diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c
index f6bf4cf..6b9d326 100644
--- a/sound/soc/codecs/max98090.c
+++ b/sound/soc/codecs/max98090.c
@@ -2103,10 +2103,8 @@
M98090_IULK_MASK, 0);
}
-static void max98090_pll_work(struct work_struct *work)
+static void max98090_pll_work(struct max98090_priv *max98090)
{
- struct max98090_priv *max98090 =
- container_of(work, struct max98090_priv, pll_work);
struct snd_soc_component *component = max98090->component;
if (!snd_soc_component_is_active(component))
@@ -2114,10 +2112,16 @@
dev_info_ratelimited(component->dev, "PLL unlocked\n");
+ /*
+ * As the datasheet suggested, the maximum PLL lock time should be
+ * 7 msec. The workaround resets the codec softly by toggling SHDN
+ * off and on if PLL failed to lock for 10 msec. Notably, there is
+ * no suggested hold time for SHDN off.
+ */
+
/* Toggle shutdown OFF then ON */
snd_soc_component_update_bits(component, M98090_REG_DEVICE_SHUTDOWN,
M98090_SHDNN_MASK, 0);
- msleep(10);
snd_soc_component_update_bits(component, M98090_REG_DEVICE_SHUTDOWN,
M98090_SHDNN_MASK, M98090_SHDNN_MASK);
@@ -2259,7 +2263,7 @@
if (active & M98090_ULK_MASK) {
dev_dbg(component->dev, "M98090_ULK_MASK\n");
- schedule_work(&max98090->pll_work);
+ max98090_pll_work(max98090);
}
if (active & M98090_JDET_MASK) {
@@ -2422,7 +2426,6 @@
max98090_pll_det_enable_work);
INIT_WORK(&max98090->pll_det_disable_work,
max98090_pll_det_disable_work);
- INIT_WORK(&max98090->pll_work, max98090_pll_work);
/* Enable jack detection */
snd_soc_component_write(component, M98090_REG_JACK_DETECT,
@@ -2475,7 +2478,6 @@
cancel_delayed_work_sync(&max98090->jack_work);
cancel_delayed_work_sync(&max98090->pll_det_enable_work);
cancel_work_sync(&max98090->pll_det_disable_work);
- cancel_work_sync(&max98090->pll_work);
max98090->component = NULL;
}
diff --git a/sound/soc/codecs/max98090.h b/sound/soc/codecs/max98090.h
index 57965cd..a197114 100644
--- a/sound/soc/codecs/max98090.h
+++ b/sound/soc/codecs/max98090.h
@@ -1530,7 +1530,6 @@
struct delayed_work jack_work;
struct delayed_work pll_det_enable_work;
struct work_struct pll_det_disable_work;
- struct work_struct pll_work;
struct snd_soc_jack *jack;
unsigned int dai_fmt;
int tdm_slots;
diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c
index cae1def..16fbc9f 100644
--- a/sound/soc/codecs/max98373.c
+++ b/sound/soc/codecs/max98373.c
@@ -410,11 +410,13 @@
regmap_update_bits(max98373->regmap,
MAX98373_R20FF_GLOBAL_SHDN,
MAX98373_GLOBAL_EN_MASK, 1);
+ usleep_range(30000, 31000);
break;
case SND_SOC_DAPM_POST_PMD:
regmap_update_bits(max98373->regmap,
MAX98373_R20FF_GLOBAL_SHDN,
MAX98373_GLOBAL_EN_MASK, 0);
+ usleep_range(30000, 31000);
max98373->tdm_mode = false;
break;
default:
@@ -850,8 +852,8 @@
{
struct max98373_priv *max98373 = dev_get_drvdata(dev);
- max98373_reset(max98373, dev);
regcache_cache_only(max98373->regmap, false);
+ max98373_reset(max98373, dev);
regcache_sync(max98373->regmap);
return 0;
}
diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c
index 8600c54..2e4aa23 100644
--- a/sound/soc/codecs/max9867.c
+++ b/sound/soc/codecs/max9867.c
@@ -46,13 +46,13 @@
static const struct snd_kcontrol_new max9867_snd_controls[] = {
SOC_DOUBLE_R_TLV("Master Playback Volume", MAX9867_LEFTVOL,
- MAX9867_RIGHTVOL, 0, 41, 1, max9867_master_tlv),
+ MAX9867_RIGHTVOL, 0, 40, 1, max9867_master_tlv),
SOC_DOUBLE_R_TLV("Line Capture Volume", MAX9867_LEFTLINELVL,
MAX9867_RIGHTLINELVL, 0, 15, 1, max9867_line_tlv),
SOC_DOUBLE_R_TLV("Mic Capture Volume", MAX9867_LEFTMICGAIN,
MAX9867_RIGHTMICGAIN, 0, 20, 1, max9867_mic_tlv),
SOC_DOUBLE_R_TLV("Mic Boost Capture Volume", MAX9867_LEFTMICGAIN,
- MAX9867_RIGHTMICGAIN, 5, 4, 0, max9867_micboost_tlv),
+ MAX9867_RIGHTMICGAIN, 5, 3, 0, max9867_micboost_tlv),
SOC_SINGLE("Digital Sidetone Volume", MAX9867_SIDETONE, 0, 31, 1),
SOC_SINGLE_TLV("Digital Playback Volume", MAX9867_DACLEVEL, 0, 15, 1,
max9867_dac_tlv),
diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c
index e3d311f..337bddb 100644
--- a/sound/soc/codecs/msm8916-wcd-analog.c
+++ b/sound/soc/codecs/msm8916-wcd-analog.c
@@ -19,8 +19,8 @@
#define CDC_D_REVISION1 (0xf000)
#define CDC_D_PERPH_SUBTYPE (0xf005)
-#define CDC_D_INT_EN_SET (0x015)
-#define CDC_D_INT_EN_CLR (0x016)
+#define CDC_D_INT_EN_SET (0xf015)
+#define CDC_D_INT_EN_CLR (0xf016)
#define MBHC_SWITCH_INT BIT(7)
#define MBHC_MIC_ELECTRICAL_INS_REM_DET BIT(6)
#define MBHC_BUTTON_PRESS_DET BIT(5)
@@ -391,9 +391,6 @@
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
- snd_soc_component_update_bits(component, CDC_A_MICB_1_INT_RBIAS,
- MICB_1_INT_TX2_INT_RBIAS_EN_MASK,
- MICB_1_INT_TX2_INT_RBIAS_EN_ENABLE);
snd_soc_component_update_bits(component, reg, MICB_1_EN_PULL_DOWN_EN_MASK, 0);
snd_soc_component_update_bits(component, CDC_A_MICB_1_EN,
MICB_1_EN_OPA_STG2_TAIL_CURR_MASK,
@@ -443,6 +440,14 @@
struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm);
struct pm8916_wcd_analog_priv *wcd = snd_soc_component_get_drvdata(component);
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ snd_soc_component_update_bits(component, CDC_A_MICB_1_INT_RBIAS,
+ MICB_1_INT_TX1_INT_RBIAS_EN_MASK,
+ MICB_1_INT_TX1_INT_RBIAS_EN_ENABLE);
+ break;
+ }
+
return pm8916_wcd_analog_enable_micbias_int(component, event, w->reg,
wcd->micbias1_cap_mode);
}
@@ -553,6 +558,11 @@
struct pm8916_wcd_analog_priv *wcd = snd_soc_component_get_drvdata(component);
switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ snd_soc_component_update_bits(component, CDC_A_MICB_1_INT_RBIAS,
+ MICB_1_INT_TX2_INT_RBIAS_EN_MASK,
+ MICB_1_INT_TX2_INT_RBIAS_EN_ENABLE);
+ break;
case SND_SOC_DAPM_POST_PMU:
pm8916_mbhc_configure_bias(wcd, true);
break;
@@ -888,10 +898,10 @@
SND_SOC_DAPM_SUPPLY("MIC BIAS External1", CDC_A_MICB_1_EN, 7, 0,
pm8916_wcd_analog_enable_micbias_ext1,
- SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_SUPPLY("MIC BIAS External2", CDC_A_MICB_2_EN, 7, 0,
pm8916_wcd_analog_enable_micbias_ext2,
- SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_ADC_E("ADC1", NULL, CDC_A_TX_1_EN, 7, 0,
pm8916_wcd_analog_enable_adc,
diff --git a/sound/soc/codecs/msm8916-wcd-digital.c b/sound/soc/codecs/msm8916-wcd-digital.c
index 58b2468..09fccac 100644
--- a/sound/soc/codecs/msm8916-wcd-digital.c
+++ b/sound/soc/codecs/msm8916-wcd-digital.c
@@ -586,6 +586,12 @@
snd_soc_component_write(component, rx_gain_reg[w->shift],
snd_soc_component_read32(component, rx_gain_reg[w->shift]));
break;
+ case SND_SOC_DAPM_POST_PMD:
+ snd_soc_component_update_bits(component, LPASS_CDC_CLK_RX_RESET_CTL,
+ 1 << w->shift, 1 << w->shift);
+ snd_soc_component_update_bits(component, LPASS_CDC_CLK_RX_RESET_CTL,
+ 1 << w->shift, 0x0);
+ break;
}
return 0;
}
diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c
index 88b7569..b37e5fb 100644
--- a/sound/soc/codecs/pcm3168a.c
+++ b/sound/soc/codecs/pcm3168a.c
@@ -302,6 +302,13 @@
struct pcm3168a_priv *pcm3168a = snd_soc_component_get_drvdata(dai->component);
int ret;
+ /*
+ * Some sound card sets 0 Hz as reset,
+ * but it is impossible to set. Ignore it here
+ */
+ if (freq == 0)
+ return 0;
+
if (freq > PCM3168A_MAX_SYSCLK)
return -EINVAL;
diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c
index 861210f..4cbef9a 100644
--- a/sound/soc/codecs/pcm512x.c
+++ b/sound/soc/codecs/pcm512x.c
@@ -1564,13 +1564,15 @@
}
pcm512x->sclk = devm_clk_get(dev, NULL);
- if (PTR_ERR(pcm512x->sclk) == -EPROBE_DEFER)
- return -EPROBE_DEFER;
+ if (PTR_ERR(pcm512x->sclk) == -EPROBE_DEFER) {
+ ret = -EPROBE_DEFER;
+ goto err;
+ }
if (!IS_ERR(pcm512x->sclk)) {
ret = clk_prepare_enable(pcm512x->sclk);
if (ret != 0) {
dev_err(dev, "Failed to enable SCLK: %d\n", ret);
- return ret;
+ goto err;
}
}
diff --git a/sound/soc/codecs/rk3328_codec.c b/sound/soc/codecs/rk3328_codec.c
index 287c962..514ebe1 100644
--- a/sound/soc/codecs/rk3328_codec.c
+++ b/sound/soc/codecs/rk3328_codec.c
@@ -472,7 +472,8 @@
rk3328->pclk = devm_clk_get(&pdev->dev, "pclk");
if (IS_ERR(rk3328->pclk)) {
dev_err(&pdev->dev, "can't get acodec pclk\n");
- return PTR_ERR(rk3328->pclk);
+ ret = PTR_ERR(rk3328->pclk);
+ goto err_unprepare_mclk;
}
ret = clk_prepare_enable(rk3328->pclk);
@@ -482,19 +483,34 @@
}
base = devm_platform_ioremap_resource(pdev, 0);
- if (IS_ERR(base))
- return PTR_ERR(base);
+ if (IS_ERR(base)) {
+ ret = PTR_ERR(base);
+ goto err_unprepare_pclk;
+ }
rk3328->regmap = devm_regmap_init_mmio(&pdev->dev, base,
&rk3328_codec_regmap_config);
- if (IS_ERR(rk3328->regmap))
- return PTR_ERR(rk3328->regmap);
+ if (IS_ERR(rk3328->regmap)) {
+ ret = PTR_ERR(rk3328->regmap);
+ goto err_unprepare_pclk;
+ }
platform_set_drvdata(pdev, rk3328);
- return devm_snd_soc_register_component(&pdev->dev, &soc_codec_rk3328,
+ ret = devm_snd_soc_register_component(&pdev->dev, &soc_codec_rk3328,
rk3328_dai,
ARRAY_SIZE(rk3328_dai));
+ if (ret)
+ goto err_unprepare_pclk;
+
+ return 0;
+
+err_unprepare_pclk:
+ clk_disable_unprepare(rk3328->pclk);
+
+err_unprepare_mclk:
+ clk_disable_unprepare(rk3328->mclk);
+ return ret;
}
static const struct of_device_id rk3328_codec_of_match[] = {
diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c
index 9593a9a..d8ab8af 100644
--- a/sound/soc/codecs/rt286.c
+++ b/sound/soc/codecs/rt286.c
@@ -171,6 +171,9 @@
case RT286_PROC_COEF:
case RT286_SET_AMP_GAIN_ADC_IN1:
case RT286_SET_AMP_GAIN_ADC_IN2:
+ case RT286_SET_GPIO_MASK:
+ case RT286_SET_GPIO_DIRECTION:
+ case RT286_SET_GPIO_DATA:
case RT286_SET_POWER(RT286_DAC_OUT1):
case RT286_SET_POWER(RT286_DAC_OUT2):
case RT286_SET_POWER(RT286_ADC_IN1):
@@ -1115,12 +1118,11 @@
{ }
};
-static const struct dmi_system_id dmi_dell_dino[] = {
+static const struct dmi_system_id dmi_dell[] = {
{
- .ident = "Dell Dino",
+ .ident = "Dell",
.matches = {
DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc."),
- DMI_MATCH(DMI_PRODUCT_NAME, "XPS 13 9343")
}
},
{ }
@@ -1131,7 +1133,7 @@
{
struct rt286_platform_data *pdata = dev_get_platdata(&i2c->dev);
struct rt286_priv *rt286;
- int i, ret, val;
+ int i, ret, vendor_id;
rt286 = devm_kzalloc(&i2c->dev, sizeof(*rt286),
GFP_KERNEL);
@@ -1147,14 +1149,15 @@
}
ret = regmap_read(rt286->regmap,
- RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &val);
+ RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &vendor_id);
if (ret != 0) {
dev_err(&i2c->dev, "I2C error %d\n", ret);
return ret;
}
- if (val != RT286_VENDOR_ID && val != RT288_VENDOR_ID) {
+ if (vendor_id != RT286_VENDOR_ID && vendor_id != RT288_VENDOR_ID) {
dev_err(&i2c->dev,
- "Device with ID register %#x is not rt286\n", val);
+ "Device with ID register %#x is not rt286\n",
+ vendor_id);
return -ENODEV;
}
@@ -1178,8 +1181,8 @@
if (pdata)
rt286->pdata = *pdata;
- if (dmi_check_system(force_combo_jack_table) ||
- dmi_check_system(dmi_dell_dino))
+ if ((vendor_id == RT288_VENDOR_ID && dmi_check_system(dmi_dell)) ||
+ dmi_check_system(force_combo_jack_table))
rt286->pdata.cbj_en = true;
regmap_write(rt286->regmap, RT286_SET_AUDIO_POWER, AC_PWRST_D3);
@@ -1218,7 +1221,7 @@
regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL3, 0xf777, 0x4737);
regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL4, 0x00ff, 0x003f);
- if (dmi_check_system(dmi_dell_dino)) {
+ if (vendor_id == RT288_VENDOR_ID && dmi_check_system(dmi_dell)) {
regmap_update_bits(rt286->regmap,
RT286_SET_GPIO_MASK, 0x40, 0x40);
regmap_update_bits(rt286->regmap,
diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c
index f70b9f7..281957a 100644
--- a/sound/soc/codecs/rt5631.c
+++ b/sound/soc/codecs/rt5631.c
@@ -1691,6 +1691,8 @@
.reg_defaults = rt5631_reg,
.num_reg_defaults = ARRAY_SIZE(rt5631_reg),
.cache_type = REGCACHE_RBTREE,
+ .use_single_read = true,
+ .use_single_write = true,
};
static int rt5631_i2c_probe(struct i2c_client *i2c,
diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c
index adbae1f..3bc63fb 100644
--- a/sound/soc/codecs/rt5640.c
+++ b/sound/soc/codecs/rt5640.c
@@ -339,9 +339,9 @@
}
static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0);
-static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0);
+static const DECLARE_TLV_DB_MINMAX(dac_vol_tlv, -6562, 0);
static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0);
-static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0);
+static const DECLARE_TLV_DB_MINMAX(adc_vol_tlv, -1762, 3000);
static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0);
/* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */
@@ -2432,6 +2432,13 @@
{
struct rt5640_priv *rt5640 = snd_soc_component_get_drvdata(component);
+ /*
+ * soc_remove_component() force-disables jack and thus rt5640->jack
+ * could be NULL at the time of driver's module unloading.
+ */
+ if (!rt5640->jack)
+ return;
+
disable_irq(rt5640->irq);
rt5640_cancel_work(rt5640);
diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index 1c06b3b..c83f7f5 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -3270,6 +3270,9 @@
snd_soc_jack_report(rt5645->mic_jack,
report, SND_JACK_MICROPHONE);
return;
+ case 4:
+ val = snd_soc_component_read32(rt5645->component, RT5645_A_JD_CTRL1) & 0x0020;
+ break;
default: /* read rt5645 jd1_1 status */
val = snd_soc_component_read32(rt5645->component, RT5645_INT_IRQ_ST) & 0x1000;
break;
@@ -3603,7 +3606,7 @@
static const struct rt5645_platform_data buddy_platform_data = {
.dmic1_data_pin = RT5645_DMIC_DATA_GPIO5,
.dmic2_data_pin = RT5645_DMIC_DATA_IN2P,
- .jd_mode = 3,
+ .jd_mode = 4,
.level_trigger_irq = true,
};
@@ -3622,6 +3625,12 @@
.inv_jd1_1 = true,
};
+static const struct rt5645_platform_data asus_t101ha_platform_data = {
+ .dmic1_data_pin = RT5645_DMIC_DATA_IN2N,
+ .dmic2_data_pin = RT5645_DMIC2_DISABLE,
+ .jd_mode = 3,
+};
+
static const struct rt5645_platform_data lenovo_ideapad_miix_310_pdata = {
.jd_mode = 3,
.in2_diff = true,
@@ -3700,6 +3709,14 @@
.driver_data = (void *)&asus_t100ha_platform_data,
},
{
+ .ident = "ASUS T101HA",
+ .matches = {
+ DMI_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."),
+ DMI_MATCH(DMI_PRODUCT_NAME, "T101HA"),
+ },
+ .driver_data = (void *)&asus_t101ha_platform_data,
+ },
+ {
.ident = "MINIX Z83-4",
.matches = {
DMI_EXACT_MATCH(DMI_SYS_VENDOR, "MINIX"),
@@ -3999,6 +4016,7 @@
RT5645_JD1_MODE_1);
break;
case 3:
+ case 4:
regmap_update_bits(rt5645->regmap, RT5645_A_JD_CTRL1,
RT5645_JD1_MODE_MASK,
RT5645_JD1_MODE_2);
diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c
index c506c93..829cf55 100644
--- a/sound/soc/codecs/rt5651.c
+++ b/sound/soc/codecs/rt5651.c
@@ -285,9 +285,9 @@
}
static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0);
-static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0);
+static const DECLARE_TLV_DB_MINMAX(dac_vol_tlv, -6562, 0);
static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0);
-static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0);
+static const DECLARE_TLV_DB_MINMAX(adc_vol_tlv, -1762, 3000);
static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0);
/* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */
diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c
index e66d083..a28afb4 100644
--- a/sound/soc/codecs/rt5659.c
+++ b/sound/soc/codecs/rt5659.c
@@ -2470,13 +2470,18 @@
return 0;
}
-static const struct snd_soc_dapm_widget rt5659_dapm_widgets[] = {
+static const struct snd_soc_dapm_widget rt5659_particular_dapm_widgets[] = {
SND_SOC_DAPM_SUPPLY("LDO2", RT5659_PWR_ANLG_3, RT5659_PWR_LDO2_BIT, 0,
NULL, 0),
- SND_SOC_DAPM_SUPPLY("PLL", RT5659_PWR_ANLG_3, RT5659_PWR_PLL_BIT, 0,
- NULL, 0),
+ SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5659_PWR_ANLG_2, RT5659_PWR_MB1_BIT,
+ 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("Mic Det Power", RT5659_PWR_VOL,
RT5659_PWR_MIC_DET_BIT, 0, NULL, 0),
+};
+
+static const struct snd_soc_dapm_widget rt5659_dapm_widgets[] = {
+ SND_SOC_DAPM_SUPPLY("PLL", RT5659_PWR_ANLG_3, RT5659_PWR_PLL_BIT, 0,
+ NULL, 0),
SND_SOC_DAPM_SUPPLY("Mono Vref", RT5659_PWR_ANLG_1,
RT5659_PWR_VREF3_BIT, 0, NULL, 0),
@@ -2501,8 +2506,6 @@
RT5659_ADC_MONO_R_ASRC_SFT, 0, NULL, 0),
/* Input Side */
- SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5659_PWR_ANLG_2, RT5659_PWR_MB1_BIT,
- 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("MICBIAS2", RT5659_PWR_ANLG_2, RT5659_PWR_MB2_BIT,
0, NULL, 0),
SND_SOC_DAPM_SUPPLY("MICBIAS3", RT5659_PWR_ANLG_2, RT5659_PWR_MB3_BIT,
@@ -3463,12 +3466,17 @@
{
struct rt5659_priv *rt5659 = snd_soc_component_get_drvdata(component);
unsigned int reg_val = 0;
+ int ret;
if (freq == rt5659->sysclk && clk_id == rt5659->sysclk_src)
return 0;
switch (clk_id) {
case RT5659_SCLK_S_MCLK:
+ ret = clk_set_rate(rt5659->mclk, freq);
+ if (ret)
+ return ret;
+
reg_val |= RT5659_SCLK_SRC_MCLK;
break;
case RT5659_SCLK_S_PLL1:
@@ -3692,10 +3700,23 @@
static int rt5659_probe(struct snd_soc_component *component)
{
+ struct snd_soc_dapm_context *dapm =
+ snd_soc_component_get_dapm(component);
struct rt5659_priv *rt5659 = snd_soc_component_get_drvdata(component);
rt5659->component = component;
+ switch (rt5659->pdata.jd_src) {
+ case RT5659_JD_HDA_HEADER:
+ break;
+
+ default:
+ snd_soc_dapm_new_controls(dapm,
+ rt5659_particular_dapm_widgets,
+ ARRAY_SIZE(rt5659_particular_dapm_widgets));
+ break;
+ }
+
return 0;
}
diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c
index 70fee68..f211817 100644
--- a/sound/soc/codecs/rt5670.c
+++ b/sound/soc/codecs/rt5670.c
@@ -31,18 +31,19 @@
#include "rt5670.h"
#include "rt5670-dsp.h"
-#define RT5670_DEV_GPIO BIT(0)
-#define RT5670_IN2_DIFF BIT(1)
-#define RT5670_DMIC_EN BIT(2)
-#define RT5670_DMIC1_IN2P BIT(3)
-#define RT5670_DMIC1_GPIO6 BIT(4)
-#define RT5670_DMIC1_GPIO7 BIT(5)
-#define RT5670_DMIC2_INR BIT(6)
-#define RT5670_DMIC2_GPIO8 BIT(7)
-#define RT5670_DMIC3_GPIO5 BIT(8)
-#define RT5670_JD_MODE1 BIT(9)
-#define RT5670_JD_MODE2 BIT(10)
-#define RT5670_JD_MODE3 BIT(11)
+#define RT5670_DEV_GPIO BIT(0)
+#define RT5670_IN2_DIFF BIT(1)
+#define RT5670_DMIC_EN BIT(2)
+#define RT5670_DMIC1_IN2P BIT(3)
+#define RT5670_DMIC1_GPIO6 BIT(4)
+#define RT5670_DMIC1_GPIO7 BIT(5)
+#define RT5670_DMIC2_INR BIT(6)
+#define RT5670_DMIC2_GPIO8 BIT(7)
+#define RT5670_DMIC3_GPIO5 BIT(8)
+#define RT5670_JD_MODE1 BIT(9)
+#define RT5670_JD_MODE2 BIT(10)
+#define RT5670_JD_MODE3 BIT(11)
+#define RT5670_GPIO1_IS_EXT_SPK_EN BIT(12)
static unsigned long rt5670_quirk;
static unsigned int quirk_override;
@@ -1447,6 +1448,33 @@
return 0;
}
+static int rt5670_spk_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm);
+ struct rt5670_priv *rt5670 = snd_soc_component_get_drvdata(component);
+
+ if (!rt5670->pdata.gpio1_is_ext_spk_en)
+ return 0;
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL2,
+ RT5670_GP1_OUT_MASK, RT5670_GP1_OUT_HI);
+ break;
+
+ case SND_SOC_DAPM_PRE_PMD:
+ regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL2,
+ RT5670_GP1_OUT_MASK, RT5670_GP1_OUT_LO);
+ break;
+
+ default:
+ return 0;
+ }
+
+ return 0;
+}
+
static int rt5670_bst1_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
@@ -1860,7 +1888,9 @@
};
static const struct snd_soc_dapm_widget rt5672_specific_dapm_widgets[] = {
- SND_SOC_DAPM_PGA("SPO Amp", SND_SOC_NOPM, 0, 0, NULL, 0),
+ SND_SOC_DAPM_PGA_E("SPO Amp", SND_SOC_NOPM, 0, 0, NULL, 0,
+ rt5670_spk_event, SND_SOC_DAPM_PRE_PMD |
+ SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_OUTPUT("SPOLP"),
SND_SOC_DAPM_OUTPUT("SPOLN"),
SND_SOC_DAPM_OUTPUT("SPORP"),
@@ -2857,14 +2887,14 @@
},
{
.callback = rt5670_quirk_cb,
- .ident = "Lenovo Thinkpad Tablet 10",
+ .ident = "Lenovo Miix 2 10",
.matches = {
DMI_MATCH(DMI_SYS_VENDOR, "LENOVO"),
DMI_MATCH(DMI_PRODUCT_VERSION, "Lenovo Miix 2 10"),
},
.driver_data = (unsigned long *)(RT5670_DMIC_EN |
RT5670_DMIC1_IN2P |
- RT5670_DEV_GPIO |
+ RT5670_GPIO1_IS_EXT_SPK_EN |
RT5670_JD_MODE2),
},
{
@@ -2924,6 +2954,10 @@
rt5670->pdata.dev_gpio = true;
dev_info(&i2c->dev, "quirk dev_gpio\n");
}
+ if (rt5670_quirk & RT5670_GPIO1_IS_EXT_SPK_EN) {
+ rt5670->pdata.gpio1_is_ext_spk_en = true;
+ dev_info(&i2c->dev, "quirk GPIO1 is external speaker enable\n");
+ }
if (rt5670_quirk & RT5670_IN2_DIFF) {
rt5670->pdata.in2_diff = true;
dev_info(&i2c->dev, "quirk IN2_DIFF\n");
@@ -3023,6 +3057,13 @@
RT5670_GP1_PF_MASK, RT5670_GP1_PF_OUT);
}
+ if (rt5670->pdata.gpio1_is_ext_spk_en) {
+ regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL1,
+ RT5670_GP1_PIN_MASK, RT5670_GP1_PIN_GPIO1);
+ regmap_update_bits(rt5670->regmap, RT5670_GPIO_CTRL2,
+ RT5670_GP1_PF_MASK, RT5670_GP1_PF_OUT);
+ }
+
if (rt5670->pdata.jd_mode) {
regmap_update_bits(rt5670->regmap, RT5670_GLB_CLK,
RT5670_SCLK_SRC_MASK, RT5670_SCLK_SRC_RCCLK);
diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h
index a8c3e44..de02033 100644
--- a/sound/soc/codecs/rt5670.h
+++ b/sound/soc/codecs/rt5670.h
@@ -757,7 +757,7 @@
#define RT5670_PWR_VREF2_BIT 4
#define RT5670_PWR_FV2 (0x1 << 3)
#define RT5670_PWR_FV2_BIT 3
-#define RT5670_LDO_SEL_MASK (0x3)
+#define RT5670_LDO_SEL_MASK (0x7)
#define RT5670_LDO_SEL_SFT 0
/* Power Management for Analog 2 (0x64) */
diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c
index 315a3d3..8bc9450 100644
--- a/sound/soc/codecs/rt5677.c
+++ b/sound/soc/codecs/rt5677.c
@@ -298,6 +298,7 @@
case RT5677_I2C_MASTER_CTRL7:
case RT5677_I2C_MASTER_CTRL8:
case RT5677_HAP_GENE_CTRL2:
+ case RT5677_PWR_ANLG2: /* Modified by DSP firmware */
case RT5677_PWR_DSP_ST:
case RT5677_PRIV_DATA:
case RT5677_ASRC_22:
diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c
index c50b75c..05e883a 100644
--- a/sound/soc/codecs/rt5682.c
+++ b/sound/soc/codecs/rt5682.c
@@ -72,6 +72,7 @@
static const struct reg_sequence patch_list[] = {
{RT5682_HP_IMP_SENS_CTRL_19, 0x1000},
{RT5682_DAC_ADC_DIG_VOL1, 0xa020},
+ {RT5682_I2C_CTRL, 0x000f},
};
static const struct reg_default rt5682_reg[] = {
@@ -2481,6 +2482,7 @@
mutex_lock(&rt5682->calibrate_mutex);
rt5682_reset(rt5682->regmap);
+ regmap_write(rt5682->regmap, RT5682_I2C_CTRL, 0x000f);
regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xa2af);
usleep_range(15000, 20000);
regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xf2af);
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index aa1f963..8a1e485 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -71,7 +71,7 @@
{ SGTL5000_DAP_EQ_BASS_BAND4, 0x002f },
{ SGTL5000_DAP_MAIN_CHAN, 0x8000 },
{ SGTL5000_DAP_MIX_CHAN, 0x0000 },
- { SGTL5000_DAP_AVC_CTRL, 0x0510 },
+ { SGTL5000_DAP_AVC_CTRL, 0x5100 },
{ SGTL5000_DAP_AVC_THRESHOLD, 0x1473 },
{ SGTL5000_DAP_AVC_ATTACK, 0x0028 },
{ SGTL5000_DAP_AVC_DECAY, 0x0050 },
@@ -1344,7 +1344,8 @@
* if vddio == vdda the source of charge pump should be
* assigned manually to VDDIO
*/
- if (vddio == vdda) {
+ if (regulator_is_equal(sgtl5000->supplies[VDDA].consumer,
+ sgtl5000->supplies[VDDIO].consumer)) {
lreg_ctrl |= SGTL5000_VDDC_ASSN_OVRD;
lreg_ctrl |= SGTL5000_VDDC_MAN_ASSN_VDDIO <<
SGTL5000_VDDC_MAN_ASSN_SHIFT;
@@ -1644,6 +1645,40 @@
dev_err(&client->dev,
"Error %d initializing CHIP_CLK_CTRL\n", ret);
+ /* Mute everything to avoid pop from the following power-up */
+ ret = regmap_write(sgtl5000->regmap, SGTL5000_CHIP_ANA_CTRL,
+ SGTL5000_CHIP_ANA_CTRL_DEFAULT);
+ if (ret) {
+ dev_err(&client->dev,
+ "Error %d muting outputs via CHIP_ANA_CTRL\n", ret);
+ goto disable_clk;
+ }
+
+ /*
+ * If VAG is powered-on (e.g. from previous boot), it would be disabled
+ * by the write to ANA_POWER in later steps of the probe code. This
+ * may create a loud pop even with all outputs muted. The proper way
+ * to circumvent this is disabling the bit first and waiting the proper
+ * cool-down time.
+ */
+ ret = regmap_read(sgtl5000->regmap, SGTL5000_CHIP_ANA_POWER, &value);
+ if (ret) {
+ dev_err(&client->dev, "Failed to read ANA_POWER: %d\n", ret);
+ goto disable_clk;
+ }
+ if (value & SGTL5000_VAG_POWERUP) {
+ ret = regmap_update_bits(sgtl5000->regmap,
+ SGTL5000_CHIP_ANA_POWER,
+ SGTL5000_VAG_POWERUP,
+ 0);
+ if (ret) {
+ dev_err(&client->dev, "Error %d disabling VAG\n", ret);
+ goto disable_clk;
+ }
+
+ msleep(SGTL5000_VAG_POWERDOWN_DELAY);
+ }
+
/* Follow section 2.2.1.1 of AN3663 */
ana_pwr = SGTL5000_ANA_POWER_DEFAULT;
if (sgtl5000->num_supplies <= VDDD) {
diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h
index a4bf4bc..56ec586 100644
--- a/sound/soc/codecs/sgtl5000.h
+++ b/sound/soc/codecs/sgtl5000.h
@@ -233,6 +233,7 @@
/*
* SGTL5000_CHIP_ANA_CTRL
*/
+#define SGTL5000_CHIP_ANA_CTRL_DEFAULT 0x0133
#define SGTL5000_LINE_OUT_MUTE 0x0100
#define SGTL5000_HP_SEL_MASK 0x0040
#define SGTL5000_HP_SEL_SHIFT 6
diff --git a/sound/soc/codecs/sti-sas.c b/sound/soc/codecs/sti-sas.c
index ec9933b..423daac 100644
--- a/sound/soc/codecs/sti-sas.c
+++ b/sound/soc/codecs/sti-sas.c
@@ -411,6 +411,7 @@
},
{},
};
+MODULE_DEVICE_TABLE(of, sti_sas_dev_match);
static int sti_sas_driver_probe(struct platform_device *pdev)
{
diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c
index 1554631..5b7f9fc 100644
--- a/sound/soc/codecs/tas571x.c
+++ b/sound/soc/codecs/tas571x.c
@@ -820,8 +820,10 @@
priv->regmap = devm_regmap_init(dev, NULL, client,
priv->chip->regmap_config);
- if (IS_ERR(priv->regmap))
- return PTR_ERR(priv->regmap);
+ if (IS_ERR(priv->regmap)) {
+ ret = PTR_ERR(priv->regmap);
+ goto disable_regs;
+ }
priv->pdn_gpio = devm_gpiod_get_optional(dev, "pdn", GPIOD_OUT_LOW);
if (IS_ERR(priv->pdn_gpio)) {
@@ -845,7 +847,7 @@
ret = regmap_write(priv->regmap, TAS571X_OSC_TRIM_REG, 0);
if (ret)
- return ret;
+ goto disable_regs;
usleep_range(50000, 60000);
@@ -861,12 +863,20 @@
*/
ret = regmap_update_bits(priv->regmap, TAS571X_MVOL_REG, 1, 0);
if (ret)
- return ret;
+ goto disable_regs;
}
- return devm_snd_soc_register_component(&client->dev,
+ ret = devm_snd_soc_register_component(&client->dev,
&priv->component_driver,
&tas571x_dai, 1);
+ if (ret)
+ goto disable_regs;
+
+ return ret;
+
+disable_regs:
+ regulator_bulk_disable(priv->chip->num_supply_names, priv->supplies);
+ return ret;
}
static int tas571x_i2c_remove(struct i2c_client *client)
diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h
index cb02495..73c5f6c 100644
--- a/sound/soc/codecs/tlv320aic31xx.h
+++ b/sound/soc/codecs/tlv320aic31xx.h
@@ -151,8 +151,8 @@
#define AIC31XX_WORD_LEN_24BITS 0x02
#define AIC31XX_WORD_LEN_32BITS 0x03
#define AIC31XX_IFACE1_MASTER_MASK GENMASK(3, 2)
-#define AIC31XX_BCLK_MASTER BIT(2)
-#define AIC31XX_WCLK_MASTER BIT(3)
+#define AIC31XX_BCLK_MASTER BIT(3)
+#define AIC31XX_WCLK_MASTER BIT(2)
/* AIC31XX_DATA_OFFSET */
#define AIC31XX_DATA_OFFSET_MASK GENMASK(7, 0)
diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
index 68165de..7a1ffba 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -662,7 +662,7 @@
}
static int aic32x4_setup_clocks(struct snd_soc_component *component,
- unsigned int sample_rate)
+ unsigned int sample_rate, unsigned int channels)
{
u8 aosr;
u16 dosr;
@@ -750,7 +750,9 @@
dosr);
clk_set_rate(clocks[5].clk,
- sample_rate * 32);
+ sample_rate * 32 *
+ channels);
+
return 0;
}
}
@@ -772,7 +774,8 @@
u8 iface1_reg = 0;
u8 dacsetup_reg = 0;
- aic32x4_setup_clocks(component, params_rate(params));
+ aic32x4_setup_clocks(component, params_rate(params),
+ params_channels(params));
switch (params_width(params)) {
case 16:
diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c
index f318403..016aff9 100644
--- a/sound/soc/codecs/wcd9335.c
+++ b/sound/soc/codecs/wcd9335.c
@@ -618,7 +618,7 @@
"ZERO", "RX_MIX_TX8", "DEC8", "DEC8_192"
};
-static const DECLARE_TLV_DB_SCALE(digital_gain, 0, 1, 0);
+static const DECLARE_TLV_DB_SCALE(digital_gain, -8400, 100, -8400);
static const DECLARE_TLV_DB_SCALE(line_gain, 0, 7, 1);
static const DECLARE_TLV_DB_SCALE(analog_gain, 0, 25, 1);
static const DECLARE_TLV_DB_SCALE(ear_pa_gain, 0, 150, 0);
@@ -4076,6 +4076,16 @@
return ret;
}
+static void wcd9335_teardown_irqs(struct wcd9335_codec *wcd)
+{
+ int i;
+
+ /* disable interrupts on all slave ports */
+ for (i = 0; i < WCD9335_SLIM_NUM_PORT_REG; i++)
+ regmap_write(wcd->if_regmap, WCD9335_SLIM_PGD_PORT_INT_EN0 + i,
+ 0x00);
+}
+
static void wcd9335_cdc_sido_ccl_enable(struct wcd9335_codec *wcd,
bool ccl_flag)
{
@@ -4844,6 +4854,7 @@
static int wcd9335_codec_probe(struct snd_soc_component *component)
{
struct wcd9335_codec *wcd = dev_get_drvdata(component->dev);
+ int ret;
int i;
snd_soc_component_init_regmap(component, wcd->regmap);
@@ -4861,7 +4872,15 @@
for (i = 0; i < NUM_CODEC_DAIS; i++)
INIT_LIST_HEAD(&wcd->dai[i].slim_ch_list);
- return wcd9335_setup_irqs(wcd);
+ ret = wcd9335_setup_irqs(wcd);
+ if (ret)
+ goto free_clsh_ctrl;
+
+ return 0;
+
+free_clsh_ctrl:
+ wcd_clsh_ctrl_free(wcd->clsh_ctrl);
+ return ret;
}
static void wcd9335_codec_remove(struct snd_soc_component *comp)
@@ -4869,7 +4888,7 @@
struct wcd9335_codec *wcd = dev_get_drvdata(comp->dev);
wcd_clsh_ctrl_free(wcd->clsh_ctrl);
- free_irq(regmap_irq_get_virq(wcd->irq_data, WCD9335_IRQ_SLIMBUS), wcd);
+ wcd9335_teardown_irqs(wcd);
}
static int wcd9335_codec_set_sysclk(struct snd_soc_component *comp,
diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c
index cf64e10..7b087d9 100644
--- a/sound/soc/codecs/wm2200.c
+++ b/sound/soc/codecs/wm2200.c
@@ -2410,6 +2410,8 @@
err_pm_runtime:
pm_runtime_disable(&i2c->dev);
+ if (i2c->irq)
+ free_irq(i2c->irq, wm2200);
err_reset:
if (wm2200->pdata.reset)
gpio_set_value_cansleep(wm2200->pdata.reset, 0);
@@ -2426,12 +2428,15 @@
{
struct wm2200_priv *wm2200 = i2c_get_clientdata(i2c);
+ pm_runtime_disable(&i2c->dev);
if (i2c->irq)
free_irq(i2c->irq, wm2200);
if (wm2200->pdata.reset)
gpio_set_value_cansleep(wm2200->pdata.reset, 0);
if (wm2200->pdata.ldo_ena)
gpio_set_value_cansleep(wm2200->pdata.ldo_ena, 0);
+ regulator_bulk_disable(ARRAY_SIZE(wm2200->core_supplies),
+ wm2200->core_supplies);
return 0;
}
diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c
index 4af0e51..91cc63c 100644
--- a/sound/soc/codecs/wm5100.c
+++ b/sound/soc/codecs/wm5100.c
@@ -2617,6 +2617,7 @@
return ret;
err_reset:
+ pm_runtime_disable(&i2c->dev);
if (i2c->irq)
free_irq(i2c->irq, wm5100);
wm5100_free_gpio(i2c);
@@ -2640,6 +2641,7 @@
{
struct wm5100_priv *wm5100 = i2c_get_clientdata(i2c);
+ pm_runtime_disable(&i2c->dev);
if (i2c->irq)
free_irq(i2c->irq, wm5100);
wm5100_free_gpio(i2c);
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index bcb3c9d..9e8c564 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -1917,6 +1917,7 @@
snd_soc_component_update_bits(component, WM8904_BIAS_CONTROL_0,
WM8904_BIAS_ENA, 0);
+ snd_soc_component_write(component, WM8904_SW_RESET_AND_ID, 0);
regcache_cache_only(wm8904->regmap, true);
regcache_mark_dirty(wm8904->regmap);
diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c
index 18535b3..04f2347 100644
--- a/sound/soc/codecs/wm8958-dsp2.c
+++ b/sound/soc/codecs/wm8958-dsp2.c
@@ -416,8 +416,12 @@
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm);
+ struct wm8994 *control = dev_get_drvdata(component->dev->parent);
int i;
+ if (control->type != WM8958)
+ return 0;
+
switch (event) {
case SND_SOC_DAPM_POST_PMU:
case SND_SOC_DAPM_PRE_PMU:
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index 55112c1..708fc4e 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -707,7 +707,13 @@
best_freq_out = -EINVAL;
*sysclk_idx = *dac_idx = *bclk_idx = -1;
- for (i = 0; i < ARRAY_SIZE(sysclk_divs); ++i) {
+ /*
+ * From Datasheet, the PLL performs best when f2 is between
+ * 90MHz and 100MHz, the desired sysclk output is 11.2896MHz
+ * or 12.288MHz, then sysclkdiv = 2 is the best choice.
+ * So search sysclk_divs from 2 to 1 other than from 1 to 2.
+ */
+ for (i = ARRAY_SIZE(sysclk_divs) - 1; i >= 0; --i) {
if (sysclk_divs[i] == -1)
continue;
for (j = 0; j < ARRAY_SIZE(dac_divs); ++j) {
@@ -860,8 +866,7 @@
wm8960->is_stream_in_use[tx] = true;
- if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_ON &&
- !wm8960->is_stream_in_use[!tx])
+ if (!wm8960->is_stream_in_use[!tx])
return wm8960_configure_clocking(component);
return 0;
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 3e5c69f..d9d59f4 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -2788,7 +2788,7 @@
if (target % Fref == 0) {
fll_div->theta = 0;
- fll_div->lambda = 0;
+ fll_div->lambda = 1;
} else {
gcd_fll = gcd(target, fratio * Fref);
@@ -2858,7 +2858,7 @@
return -EINVAL;
}
- if (fll_div.theta || fll_div.lambda)
+ if (fll_div.theta)
fll1 |= WM8962_FLL_FRAC;
/* Stop the FLL while we reconfigure */
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index d5fb7f5..6dbab3f 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -3372,6 +3372,8 @@
return -EINVAL;
}
+ pm_runtime_get_sync(component->dev);
+
switch (micbias) {
case 1:
micdet = &wm8994->micdet[0];
@@ -3419,6 +3421,8 @@
snd_soc_dapm_sync(dapm);
+ pm_runtime_put(component->dev);
+
return 0;
}
EXPORT_SYMBOL_GPL(wm8994_mic_detect);
@@ -3786,6 +3790,8 @@
return -EINVAL;
}
+ pm_runtime_get_sync(component->dev);
+
if (jack) {
snd_soc_dapm_force_enable_pin(dapm, "CLK_SYS");
snd_soc_dapm_sync(dapm);
@@ -3854,6 +3860,8 @@
snd_soc_dapm_sync(dapm);
}
+ pm_runtime_put(component->dev);
+
return 0;
}
EXPORT_SYMBOL_GPL(wm8958_mic_detect);
@@ -4047,11 +4055,13 @@
wm8994->hubs.dcs_readback_mode = 2;
break;
}
+ wm8994->hubs.micd_scthr = true;
break;
case WM8958:
wm8994->hubs.dcs_readback_mode = 1;
wm8994->hubs.hp_startup_mode = 1;
+ wm8994->hubs.micd_scthr = true;
switch (control->revision) {
case 0:
diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c
index 37e4bb3..229f298 100644
--- a/sound/soc/codecs/wm8997.c
+++ b/sound/soc/codecs/wm8997.c
@@ -1177,6 +1177,8 @@
goto err_spk_irqs;
}
+ return ret;
+
err_spk_irqs:
arizona_free_spk_irqs(arizona);
diff --git a/sound/soc/codecs/wm8998.c b/sound/soc/codecs/wm8998.c
index 7c18992..817ccdd 100644
--- a/sound/soc/codecs/wm8998.c
+++ b/sound/soc/codecs/wm8998.c
@@ -1375,7 +1375,7 @@
ret = arizona_init_spk_irqs(arizona);
if (ret < 0)
- return ret;
+ goto err_pm_disable;
ret = devm_snd_soc_register_component(&pdev->dev,
&soc_component_dev_wm8998,
@@ -1390,6 +1390,8 @@
err_spk_irqs:
arizona_free_spk_irqs(arizona);
+err_pm_disable:
+ pm_runtime_disable(&pdev->dev);
return ret;
}
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index 9b8bb7b..1367292 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -1496,7 +1496,7 @@
ctl_work = kzalloc(sizeof(*ctl_work), GFP_KERNEL);
if (!ctl_work) {
ret = -ENOMEM;
- goto err_ctl_cache;
+ goto err_list_del;
}
ctl_work->dsp = dsp;
@@ -1506,7 +1506,8 @@
return 0;
-err_ctl_cache:
+err_list_del:
+ list_del(&ctl->list);
kfree(ctl->cache);
err_ctl_name:
kfree(ctl->name);
@@ -1912,6 +1913,7 @@
mem = wm_adsp_find_region(dsp, type);
if (!mem) {
adsp_err(dsp, "No region of type: %x\n", type);
+ ret = -EINVAL;
goto out_fw;
}
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index e93af7e..dd421e2 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -1223,6 +1223,9 @@
snd_soc_component_update_bits(component, WM8993_ADDITIONAL_CONTROL,
WM8993_LINEOUT2_FB, WM8993_LINEOUT2_FB);
+ if (!hubs->micd_scthr)
+ return 0;
+
snd_soc_component_update_bits(component, WM8993_MICBIAS,
WM8993_JD_SCTHR_MASK | WM8993_JD_THR_MASK |
WM8993_MICB1_LVL | WM8993_MICB2_LVL,
diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h
index 4b8e5f0..988b29e 100644
--- a/sound/soc/codecs/wm_hubs.h
+++ b/sound/soc/codecs/wm_hubs.h
@@ -27,6 +27,7 @@
int hp_startup_mode;
int series_startup;
int no_series_update;
+ bool micd_scthr;
bool no_cache_dac_hp_direct;
struct list_head dcs_cache;
diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c
index 01052a0..5aee6b8 100644
--- a/sound/soc/fsl/fsl_asrc_dma.c
+++ b/sound/soc/fsl/fsl_asrc_dma.c
@@ -241,6 +241,7 @@
ret = dmaengine_slave_config(pair->dma_chan[dir], &config_be);
if (ret) {
dev_err(dev, "failed to config DMA channel for Back-End\n");
+ dma_release_channel(pair->dma_chan[dir]);
return ret;
}
diff --git a/sound/soc/fsl/fsl_audmix.c b/sound/soc/fsl/fsl_audmix.c
index c7e4e97..5faecbe 100644
--- a/sound/soc/fsl/fsl_audmix.c
+++ b/sound/soc/fsl/fsl_audmix.c
@@ -286,6 +286,7 @@
struct snd_soc_dai *dai)
{
struct fsl_audmix *priv = snd_soc_dai_get_drvdata(dai);
+ unsigned long lock_flags;
/* Capture stream shall not be handled */
if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
@@ -295,12 +296,16 @@
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ spin_lock_irqsave(&priv->lock, lock_flags);
priv->tdms |= BIT(dai->driver->id);
+ spin_unlock_irqrestore(&priv->lock, lock_flags);
break;
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ spin_lock_irqsave(&priv->lock, lock_flags);
priv->tdms &= ~BIT(dai->driver->id);
+ spin_unlock_irqrestore(&priv->lock, lock_flags);
break;
default:
return -EINVAL;
@@ -491,6 +496,7 @@
return PTR_ERR(priv->ipg_clk);
}
+ spin_lock_init(&priv->lock);
platform_set_drvdata(pdev, priv);
pm_runtime_enable(dev);
@@ -499,15 +505,20 @@
ARRAY_SIZE(fsl_audmix_dai));
if (ret) {
dev_err(dev, "failed to register ASoC DAI\n");
- return ret;
+ goto err_disable_pm;
}
priv->pdev = platform_device_register_data(dev, mdrv, 0, NULL, 0);
if (IS_ERR(priv->pdev)) {
ret = PTR_ERR(priv->pdev);
dev_err(dev, "failed to register platform %s: %d\n", mdrv, ret);
+ goto err_disable_pm;
}
+ return 0;
+
+err_disable_pm:
+ pm_runtime_disable(dev);
return ret;
}
@@ -515,6 +526,8 @@
{
struct fsl_audmix *priv = dev_get_drvdata(&pdev->dev);
+ pm_runtime_disable(&pdev->dev);
+
if (priv->pdev)
platform_device_unregister(priv->pdev);
diff --git a/sound/soc/fsl/fsl_audmix.h b/sound/soc/fsl/fsl_audmix.h
index 7812ffe..479f056 100644
--- a/sound/soc/fsl/fsl_audmix.h
+++ b/sound/soc/fsl/fsl_audmix.h
@@ -96,6 +96,7 @@
struct platform_device *pdev;
struct regmap *regmap;
struct clk *ipg_clk;
+ spinlock_t lock; /* Protect tdms */
u8 tdms;
};
diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index a78e4ab..33ade79 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -33,6 +33,7 @@
* @fsysclk: system clock source to derive HCK, SCK and FS
* @spbaclk: SPBA clock (optional, depending on SoC design)
* @task: tasklet to handle the reset operation
+ * @lock: spin lock between hw_reset() and trigger()
* @fifo_depth: depth of tx/rx FIFO
* @slot_width: width of each DAI slot
* @slots: number of slots
@@ -56,6 +57,7 @@
struct clk *fsysclk;
struct clk *spbaclk;
struct tasklet_struct task;
+ spinlock_t lock; /* Protect hw_reset and trigger */
u32 fifo_depth;
u32 slot_width;
u32 slots;
@@ -85,6 +87,10 @@
if ((saisr & (ESAI_SAISR_TUE | ESAI_SAISR_ROE)) &&
esai_priv->reset_at_xrun) {
dev_dbg(&pdev->dev, "reset module for xrun\n");
+ regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR,
+ ESAI_xCR_xEIE_MASK, 0);
+ regmap_update_bits(esai_priv->regmap, REG_ESAI_RCR,
+ ESAI_xCR_xEIE_MASK, 0);
tasklet_schedule(&esai_priv->task);
}
@@ -488,11 +494,13 @@
ESAI_SAICR_SYNC, esai_priv->synchronous ?
ESAI_SAICR_SYNC : 0);
- /* Set a default slot number -- 2 */
+ /* Set slots count */
regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR,
- ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(2));
+ ESAI_xCCR_xDC_MASK,
+ ESAI_xCCR_xDC(esai_priv->slots));
regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR,
- ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(2));
+ ESAI_xCCR_xDC_MASK,
+ ESAI_xCCR_xDC(esai_priv->slots));
}
return 0;
@@ -676,8 +684,10 @@
{
struct fsl_esai *esai_priv = (struct fsl_esai *)arg;
bool tx = true, rx = false, enabled[2];
+ unsigned long lock_flags;
u32 tfcr, rfcr;
+ spin_lock_irqsave(&esai_priv->lock, lock_flags);
/* Save the registers */
regmap_read(esai_priv->regmap, REG_ESAI_TFCR, &tfcr);
regmap_read(esai_priv->regmap, REG_ESAI_RFCR, &rfcr);
@@ -715,6 +725,8 @@
fsl_esai_trigger_start(esai_priv, tx);
if (enabled[rx])
fsl_esai_trigger_start(esai_priv, rx);
+
+ spin_unlock_irqrestore(&esai_priv->lock, lock_flags);
}
static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd,
@@ -722,6 +734,7 @@
{
struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai);
bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+ unsigned long lock_flags;
esai_priv->channels[tx] = substream->runtime->channels;
@@ -729,12 +742,16 @@
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ spin_lock_irqsave(&esai_priv->lock, lock_flags);
fsl_esai_trigger_start(esai_priv, tx);
+ spin_unlock_irqrestore(&esai_priv->lock, lock_flags);
break;
case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ spin_lock_irqsave(&esai_priv->lock, lock_flags);
fsl_esai_trigger_stop(esai_priv, tx);
+ spin_unlock_irqrestore(&esai_priv->lock, lock_flags);
break;
default:
return -EINVAL;
@@ -1002,6 +1019,7 @@
dev_set_drvdata(&pdev->dev, esai_priv);
+ spin_lock_init(&esai_priv->lock);
ret = fsl_esai_hw_init(esai_priv);
if (ret)
return ret;
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index b517e4b..0272596 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -680,10 +680,11 @@
regmap_write(sai->regmap, FSL_SAI_RCSR(ofs), 0);
regmap_update_bits(sai->regmap, FSL_SAI_TCR1(ofs),
- FSL_SAI_CR1_RFW_MASK,
+ FSL_SAI_CR1_RFW_MASK(sai->soc_data->fifo_depth),
sai->soc_data->fifo_depth - FSL_SAI_MAXBURST_TX);
regmap_update_bits(sai->regmap, FSL_SAI_RCR1(ofs),
- FSL_SAI_CR1_RFW_MASK, FSL_SAI_MAXBURST_RX - 1);
+ FSL_SAI_CR1_RFW_MASK(sai->soc_data->fifo_depth),
+ FSL_SAI_MAXBURST_RX - 1);
snd_soc_dai_init_dma_data(cpu_dai, &sai->dma_params_tx,
&sai->dma_params_rx);
@@ -693,7 +694,7 @@
return 0;
}
-static struct snd_soc_dai_driver fsl_sai_dai = {
+static struct snd_soc_dai_driver fsl_sai_dai_template = {
.probe = fsl_sai_dai_probe,
.playback = {
.stream_name = "CPU-Playback",
@@ -964,12 +965,15 @@
return ret;
}
+ memcpy(&sai->cpu_dai_drv, &fsl_sai_dai_template,
+ sizeof(fsl_sai_dai_template));
+
/* Sync Tx with Rx as default by following old DT binding */
sai->synchronous[RX] = true;
sai->synchronous[TX] = false;
- fsl_sai_dai.symmetric_rates = 1;
- fsl_sai_dai.symmetric_channels = 1;
- fsl_sai_dai.symmetric_samplebits = 1;
+ sai->cpu_dai_drv.symmetric_rates = 1;
+ sai->cpu_dai_drv.symmetric_channels = 1;
+ sai->cpu_dai_drv.symmetric_samplebits = 1;
if (of_find_property(np, "fsl,sai-synchronous-rx", NULL) &&
of_find_property(np, "fsl,sai-asynchronous", NULL)) {
@@ -986,9 +990,9 @@
/* Discard all settings for asynchronous mode */
sai->synchronous[RX] = false;
sai->synchronous[TX] = false;
- fsl_sai_dai.symmetric_rates = 0;
- fsl_sai_dai.symmetric_channels = 0;
- fsl_sai_dai.symmetric_samplebits = 0;
+ sai->cpu_dai_drv.symmetric_rates = 0;
+ sai->cpu_dai_drv.symmetric_channels = 0;
+ sai->cpu_dai_drv.symmetric_samplebits = 0;
}
if (of_find_property(np, "fsl,sai-mclk-direction-output", NULL) &&
@@ -1017,14 +1021,26 @@
pm_runtime_enable(&pdev->dev);
ret = devm_snd_soc_register_component(&pdev->dev, &fsl_component,
- &fsl_sai_dai, 1);
+ &sai->cpu_dai_drv, 1);
if (ret)
- return ret;
+ goto err_pm_disable;
- if (sai->soc_data->use_imx_pcm)
- return imx_pcm_dma_init(pdev, IMX_SAI_DMABUF_SIZE);
- else
- return devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0);
+ if (sai->soc_data->use_imx_pcm) {
+ ret = imx_pcm_dma_init(pdev, IMX_SAI_DMABUF_SIZE);
+ if (ret)
+ goto err_pm_disable;
+ } else {
+ ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0);
+ if (ret)
+ goto err_pm_disable;
+ }
+
+ return ret;
+
+err_pm_disable:
+ pm_runtime_disable(&pdev->dev);
+
+ return ret;
}
static int fsl_sai_remove(struct platform_device *pdev)
diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h
index 76b15de..677ecfc 100644
--- a/sound/soc/fsl/fsl_sai.h
+++ b/sound/soc/fsl/fsl_sai.h
@@ -94,7 +94,7 @@
#define FSL_SAI_CSR_FRDE BIT(0)
/* SAI Transmit and Receive Configuration 1 Register */
-#define FSL_SAI_CR1_RFW_MASK 0x1f
+#define FSL_SAI_CR1_RFW_MASK(x) ((x) - 1)
/* SAI Transmit and Receive Configuration 2 Register */
#define FSL_SAI_CR2_SYNC BIT(30)
@@ -180,6 +180,7 @@
unsigned int bclk_ratio;
const struct fsl_sai_soc_data *soc_data;
+ struct snd_soc_dai_driver cpu_dai_drv;
struct snd_dmaengine_dai_dma_data dma_params_rx;
struct snd_dmaengine_dai_dma_data dma_params_tx;
};
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 537dc69..ed18bc69 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -678,8 +678,9 @@
struct regmap *regs = ssi->regs;
u32 pm = 999, div2, psr, stccr, mask, afreq, factor, i;
unsigned long clkrate, baudrate, tmprate;
- unsigned int slots = params_channels(hw_params);
- unsigned int slot_width = 32;
+ unsigned int channels = params_channels(hw_params);
+ unsigned int slot_width = params_width(hw_params);
+ unsigned int slots = 2;
u64 sub, savesub = 100000;
unsigned int freq;
bool baudclk_is_used;
@@ -688,10 +689,14 @@
/* Override slots and slot_width if being specifically set... */
if (ssi->slots)
slots = ssi->slots;
- /* ...but keep 32 bits if slots is 2 -- I2S Master mode */
- if (ssi->slot_width && slots != 2)
+ if (ssi->slot_width)
slot_width = ssi->slot_width;
+ /* ...but force 32 bits for stereo audio using I2S Master Mode */
+ if (channels == 2 &&
+ (ssi->i2s_net & SSI_SCR_I2S_MODE_MASK) == SSI_SCR_I2S_MODE_MASTER)
+ slot_width = 32;
+
/* Generate bit clock based on the slot number and slot width */
freq = slots * slot_width * params_rate(hw_params);
@@ -868,6 +873,7 @@
static int _fsl_ssi_set_dai_fmt(struct fsl_ssi *ssi, unsigned int fmt)
{
u32 strcr = 0, scr = 0, stcr, srcr, mask;
+ unsigned int slots;
ssi->dai_fmt = fmt;
@@ -899,10 +905,11 @@
return -EINVAL;
}
+ slots = ssi->slots ? : 2;
regmap_update_bits(ssi->regs, REG_SSI_STCCR,
- SSI_SxCCR_DC_MASK, SSI_SxCCR_DC(2));
+ SSI_SxCCR_DC_MASK, SSI_SxCCR_DC(slots));
regmap_update_bits(ssi->regs, REG_SSI_SRCCR,
- SSI_SxCCR_DC_MASK, SSI_SxCCR_DC(2));
+ SSI_SxCCR_DC_MASK, SSI_SxCCR_DC(slots));
/* Data on rising edge of bclk, frame low, 1clk before data */
strcr |= SSI_STCR_TFSI | SSI_STCR_TSCKP | SSI_STCR_TEFS;
diff --git a/sound/soc/fsl/imx-es8328.c b/sound/soc/fsl/imx-es8328.c
index 15a27a2..fad1eb6 100644
--- a/sound/soc/fsl/imx-es8328.c
+++ b/sound/soc/fsl/imx-es8328.c
@@ -145,13 +145,13 @@
data = devm_kzalloc(dev, sizeof(*data), GFP_KERNEL);
if (!data) {
ret = -ENOMEM;
- goto fail;
+ goto put_device;
}
comp = devm_kzalloc(dev, 3 * sizeof(*comp), GFP_KERNEL);
if (!comp) {
ret = -ENOMEM;
- goto fail;
+ goto put_device;
}
data->dev = dev;
@@ -182,12 +182,12 @@
ret = snd_soc_of_parse_card_name(&data->card, "model");
if (ret) {
dev_err(dev, "Unable to parse card name\n");
- goto fail;
+ goto put_device;
}
ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing");
if (ret) {
dev_err(dev, "Unable to parse routing: %d\n", ret);
- goto fail;
+ goto put_device;
}
data->card.num_links = 1;
data->card.owner = THIS_MODULE;
@@ -196,10 +196,12 @@
ret = snd_soc_register_card(&data->card);
if (ret) {
dev_err(dev, "Unable to register: %d\n", ret);
- goto fail;
+ goto put_device;
}
platform_set_drvdata(pdev, data);
+put_device:
+ put_device(&ssi_pdev->dev);
fail:
of_node_put(ssi_np);
of_node_put(codec_np);
diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c
index 6007e63..1bc4981 100644
--- a/sound/soc/generic/audio-graph-card.c
+++ b/sound/soc/generic/audio-graph-card.c
@@ -340,7 +340,7 @@
struct device_node *top = dev->of_node;
struct asoc_simple_dai *cpu_dai;
struct asoc_simple_dai *codec_dai;
- int ret, single_cpu;
+ int ret, single_cpu = 0;
/* Do it only CPU turn */
if (!li->cpu)
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index fc9c753..4484c40 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -258,7 +258,7 @@
struct device_node *plat = NULL;
char prop[128];
char *prefix = "";
- int ret, single_cpu;
+ int ret, single_cpu = 0;
/*
* |CPU |Codec : turn
diff --git a/sound/soc/hisilicon/hi6210-i2s.c b/sound/soc/hisilicon/hi6210-i2s.c
index ab3b76d..03470e8 100644
--- a/sound/soc/hisilicon/hi6210-i2s.c
+++ b/sound/soc/hisilicon/hi6210-i2s.c
@@ -102,18 +102,15 @@
for (n = 0; n < i2s->clocks; n++) {
ret = clk_prepare_enable(i2s->clk[n]);
- if (ret) {
- while (n--)
- clk_disable_unprepare(i2s->clk[n]);
- return ret;
- }
+ if (ret)
+ goto err_unprepare_clk;
}
ret = clk_set_rate(i2s->clk[CLK_I2S_BASE], 49152000);
if (ret) {
dev_err(i2s->dev, "%s: setting 49.152MHz base rate failed %d\n",
__func__, ret);
- return ret;
+ goto err_unprepare_clk;
}
/* enable clock before frequency division */
@@ -165,6 +162,11 @@
hi6210_write_reg(i2s, HII2S_SW_RST_N, val);
return 0;
+
+err_unprepare_clk:
+ while (n--)
+ clk_disable_unprepare(i2s->clk[n]);
+ return ret;
}
static void hi6210_i2s_shutdown(struct snd_pcm_substream *substream,
diff --git a/sound/soc/img/img-i2s-in.c b/sound/soc/img/img-i2s-in.c
index fdd2c73..243f916 100644
--- a/sound/soc/img/img-i2s-in.c
+++ b/sound/soc/img/img-i2s-in.c
@@ -343,8 +343,10 @@
chan_control_mask = IMG_I2S_IN_CH_CTL_CLK_TRANS_MASK;
ret = pm_runtime_get_sync(i2s->dev);
- if (ret < 0)
+ if (ret < 0) {
+ pm_runtime_put_noidle(i2s->dev);
return ret;
+ }
for (i = 0; i < i2s->active_channels; i++)
img_i2s_in_ch_disable(i2s, i);
@@ -462,7 +464,7 @@
if (ret)
goto err_pm_disable;
}
- ret = pm_runtime_get_sync(&pdev->dev);
+ ret = pm_runtime_resume_and_get(&pdev->dev);
if (ret < 0)
goto err_suspend;
@@ -482,6 +484,7 @@
if (IS_ERR(rst)) {
if (PTR_ERR(rst) == -EPROBE_DEFER) {
ret = -EPROBE_DEFER;
+ pm_runtime_put(&pdev->dev);
goto err_suspend;
}
diff --git a/sound/soc/img/img-i2s-out.c b/sound/soc/img/img-i2s-out.c
index 4b18534..9c4212f 100644
--- a/sound/soc/img/img-i2s-out.c
+++ b/sound/soc/img/img-i2s-out.c
@@ -347,8 +347,10 @@
chan_control_mask = IMG_I2S_OUT_CHAN_CTL_CLKT_MASK;
ret = pm_runtime_get_sync(i2s->dev);
- if (ret < 0)
+ if (ret < 0) {
+ pm_runtime_put_noidle(i2s->dev);
return ret;
+ }
img_i2s_out_disable(i2s);
@@ -488,8 +490,10 @@
goto err_pm_disable;
}
ret = pm_runtime_get_sync(&pdev->dev);
- if (ret < 0)
+ if (ret < 0) {
+ pm_runtime_put_noidle(&pdev->dev);
goto err_suspend;
+ }
reg = IMG_I2S_OUT_CTL_FRM_SIZE_MASK;
img_i2s_out_writel(i2s, reg, IMG_I2S_OUT_CTL);
diff --git a/sound/soc/img/img-parallel-out.c b/sound/soc/img/img-parallel-out.c
index 5ddbe3a..4da49a4 100644
--- a/sound/soc/img/img-parallel-out.c
+++ b/sound/soc/img/img-parallel-out.c
@@ -163,8 +163,10 @@
}
ret = pm_runtime_get_sync(prl->dev);
- if (ret < 0)
+ if (ret < 0) {
+ pm_runtime_put_noidle(prl->dev);
return ret;
+ }
reg = img_prl_out_readl(prl, IMG_PRL_OUT_CTL);
reg = (reg & ~IMG_PRL_OUT_CTL_EDGE_MASK) | control_set;
diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig
index 01c9975..ef493ca 100644
--- a/sound/soc/intel/Kconfig
+++ b/sound/soc/intel/Kconfig
@@ -59,6 +59,9 @@
If you have a Intel Haswell or Broadwell platform connected to
an I2S codec, then enable this option by saying Y or m. This is
typically used for Chromebooks. This is a recommended option.
+ This option is mutually exclusive with the SOF support on
+ Broadwell. If you want to enable SOF on Broadwell, you need to
+ deselect this option first.
config SND_SOC_INTEL_BAYTRAIL
tristate "Baytrail (legacy) Platforms"
diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c
index baef461..df8f799 100644
--- a/sound/soc/intel/atom/sst-atom-controls.c
+++ b/sound/soc/intel/atom/sst-atom-controls.c
@@ -966,7 +966,9 @@
dev_dbg(c->dev, "Enter: widget=%s\n", w->name);
if (SND_SOC_DAPM_EVENT_ON(event)) {
+ mutex_lock(&drv->lock);
ret = sst_send_slot_map(drv);
+ mutex_unlock(&drv->lock);
if (ret)
return ret;
ret = sst_send_pipe_module_params(w, k);
@@ -1333,7 +1335,7 @@
dai->capture_widget->name);
w = dai->capture_widget;
snd_soc_dapm_widget_for_each_source_path(w, p) {
- if (p->connected && !p->connected(w, p->sink))
+ if (p->connected && !p->connected(w, p->source))
continue;
if (p->connect && p->source->power &&
diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
index 8cc3cc3..7d59846 100644
--- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c
+++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
@@ -127,7 +127,7 @@
snd_pcm_uframes_t period_size;
ssize_t periodbytes;
ssize_t buffer_bytes = snd_pcm_lib_buffer_bytes(substream);
- u32 buffer_addr = virt_to_phys(substream->dma_buffer.area);
+ u32 buffer_addr = virt_to_phys(substream->runtime->dma_area);
channels = substream->runtime->channels;
period_size = substream->runtime->period_size;
@@ -233,7 +233,6 @@
/* set codec params and inform SST driver the same */
sst_fill_pcm_params(substream, ¶m);
sst_fill_alloc_params(substream, &alloc_params);
- substream->runtime->dma_area = substream->dma_buffer.area;
str_params.sparams = param;
str_params.aparams = alloc_params;
str_params.codec = SST_CODEC_TYPE_PCM;
@@ -331,7 +330,7 @@
ret_val = power_up_sst(stream);
if (ret_val < 0)
- return ret_val;
+ goto out_power_up;
/* Make sure, that the period size is always even */
snd_pcm_hw_constraint_step(substream->runtime, 0,
@@ -340,8 +339,9 @@
return snd_pcm_hw_constraint_integer(runtime,
SNDRV_PCM_HW_PARAM_PERIODS);
out_ops:
- kfree(stream);
mutex_unlock(&sst_lock);
+out_power_up:
+ kfree(stream);
return ret_val;
}
@@ -499,14 +499,14 @@
.channels_min = SST_STEREO,
.channels_max = SST_STEREO,
.rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000,
- .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
},
.capture = {
.stream_name = "Headset Capture",
.channels_min = 1,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000,
- .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
},
},
{
@@ -517,7 +517,7 @@
.channels_min = SST_STEREO,
.channels_max = SST_STEREO,
.rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000,
- .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
},
},
{
diff --git a/sound/soc/intel/atom/sst/sst_pci.c b/sound/soc/intel/atom/sst/sst_pci.c
index d952719..5862fe9 100644
--- a/sound/soc/intel/atom/sst/sst_pci.c
+++ b/sound/soc/intel/atom/sst/sst_pci.c
@@ -99,7 +99,7 @@
dev_dbg(ctx->dev, "DRAM Ptr %p\n", ctx->dram);
do_release_regions:
pci_release_regions(pci);
- return 0;
+ return ret;
}
/*
diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c
index adf416a..60fb874 100644
--- a/sound/soc/intel/boards/bxt_rt298.c
+++ b/sound/soc/intel/boards/bxt_rt298.c
@@ -556,6 +556,7 @@
/* broxton audio machine driver for SPT + RT298S */
static struct snd_soc_card broxton_rt298 = {
.name = "broxton-rt298",
+ .owner = THIS_MODULE,
.dai_link = broxton_rt298_dais,
.num_links = ARRAY_SIZE(broxton_rt298_dais),
.controls = broxton_controls,
@@ -571,6 +572,7 @@
static struct snd_soc_card geminilake_rt298 = {
.name = "geminilake-rt298",
+ .owner = THIS_MODULE,
.dai_link = broxton_rt298_dais,
.num_links = ARRAY_SIZE(broxton_rt298_dais),
.controls = broxton_controls,
diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c
index 4661233..ed33217 100644
--- a/sound/soc/intel/boards/bytcht_es8316.c
+++ b/sound/soc/intel/boards/bytcht_es8316.c
@@ -442,7 +442,8 @@
DMI_MATCH(DMI_SYS_VENDOR, "IRBIS"),
DMI_MATCH(DMI_PRODUCT_NAME, "NB41"),
},
- .driver_data = (void *)(BYT_CHT_ES8316_INTMIC_IN2_MAP
+ .driver_data = (void *)(BYT_CHT_ES8316_SSP0
+ | BYT_CHT_ES8316_INTMIC_IN2_MAP
| BYT_CHT_ES8316_JD_INVERTED),
},
{ /* Teclast X98 Plus II */
@@ -547,8 +548,10 @@
if (cnt) {
ret = device_add_properties(codec_dev, props);
- if (ret)
+ if (ret) {
+ put_device(codec_dev);
return ret;
+ }
}
devm_acpi_dev_add_driver_gpios(codec_dev, byt_cht_es8316_gpios);
diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c
index 9c1aa4e..7830d01 100644
--- a/sound/soc/intel/boards/bytcr_rt5640.c
+++ b/sound/soc/intel/boards/bytcr_rt5640.c
@@ -284,9 +284,6 @@
static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = {
{"Headphone", NULL, "Platform Clock"},
{"Headset Mic", NULL, "Platform Clock"},
- {"Internal Mic", NULL, "Platform Clock"},
- {"Speaker", NULL, "Platform Clock"},
-
{"Headset Mic", NULL, "MICBIAS1"},
{"IN2P", NULL, "Headset Mic"},
{"Headphone", NULL, "HPOL"},
@@ -294,19 +291,23 @@
};
static const struct snd_soc_dapm_route byt_rt5640_intmic_dmic1_map[] = {
+ {"Internal Mic", NULL, "Platform Clock"},
{"DMIC1", NULL, "Internal Mic"},
};
static const struct snd_soc_dapm_route byt_rt5640_intmic_dmic2_map[] = {
+ {"Internal Mic", NULL, "Platform Clock"},
{"DMIC2", NULL, "Internal Mic"},
};
static const struct snd_soc_dapm_route byt_rt5640_intmic_in1_map[] = {
+ {"Internal Mic", NULL, "Platform Clock"},
{"Internal Mic", NULL, "MICBIAS1"},
{"IN1P", NULL, "Internal Mic"},
};
static const struct snd_soc_dapm_route byt_rt5640_intmic_in3_map[] = {
+ {"Internal Mic", NULL, "Platform Clock"},
{"Internal Mic", NULL, "MICBIAS1"},
{"IN3P", NULL, "Internal Mic"},
};
@@ -348,6 +349,7 @@
};
static const struct snd_soc_dapm_route byt_rt5640_stereo_spk_map[] = {
+ {"Speaker", NULL, "Platform Clock"},
{"Speaker", NULL, "SPOLP"},
{"Speaker", NULL, "SPOLN"},
{"Speaker", NULL, "SPORP"},
@@ -355,6 +357,7 @@
};
static const struct snd_soc_dapm_route byt_rt5640_mono_spk_map[] = {
+ {"Speaker", NULL, "Platform Clock"},
{"Speaker", NULL, "SPOLP"},
{"Speaker", NULL, "SPOLN"},
};
@@ -400,15 +403,30 @@
BYT_RT5640_SSP0_AIF1 |
BYT_RT5640_MCLK_EN),
},
+ { /* Acer One 10 S1002 */
+ .matches = {
+ DMI_MATCH(DMI_SYS_VENDOR, "Acer"),
+ DMI_MATCH(DMI_PRODUCT_NAME, "One S1002"),
+ },
+ .driver_data = (void *)(BYT_RT5640_IN1_MAP |
+ BYT_RT5640_JD_SRC_JD2_IN4N |
+ BYT_RT5640_OVCD_TH_2000UA |
+ BYT_RT5640_OVCD_SF_0P75 |
+ BYT_RT5640_DIFF_MIC |
+ BYT_RT5640_SSP0_AIF2 |
+ BYT_RT5640_MCLK_EN),
+ },
{
.matches = {
DMI_MATCH(DMI_SYS_VENDOR, "Acer"),
DMI_MATCH(DMI_PRODUCT_NAME, "Aspire SW5-012"),
},
- .driver_data = (void *)(BYT_RT5640_IN1_MAP |
- BYT_RT5640_MCLK_EN |
- BYT_RT5640_SSP0_AIF1),
-
+ .driver_data = (void *)(BYT_RT5640_DMIC1_MAP |
+ BYT_RT5640_JD_SRC_JD2_IN4N |
+ BYT_RT5640_OVCD_TH_2000UA |
+ BYT_RT5640_OVCD_SF_0P75 |
+ BYT_RT5640_SSP0_AIF1 |
+ BYT_RT5640_MCLK_EN),
},
{
.matches = {
@@ -422,6 +440,18 @@
},
{
.matches = {
+ DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ARCHOS"),
+ DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "ARCHOS 140 CESIUM"),
+ },
+ .driver_data = (void *)(BYT_RT5640_IN1_MAP |
+ BYT_RT5640_JD_SRC_JD2_IN4N |
+ BYT_RT5640_OVCD_TH_2000UA |
+ BYT_RT5640_OVCD_SF_0P75 |
+ BYT_RT5640_SSP0_AIF1 |
+ BYT_RT5640_MCLK_EN),
+ },
+ {
+ .matches = {
DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."),
DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "ME176C"),
},
@@ -449,6 +479,9 @@
DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "T100TAF"),
},
.driver_data = (void *)(BYT_RT5640_IN1_MAP |
+ BYT_RT5640_JD_SRC_JD2_IN4N |
+ BYT_RT5640_OVCD_TH_2000UA |
+ BYT_RT5640_OVCD_SF_0P75 |
BYT_RT5640_MONO_SPEAKER |
BYT_RT5640_DIFF_MIC |
BYT_RT5640_SSP0_AIF2 |
@@ -483,6 +516,23 @@
BYT_RT5640_MCLK_EN),
},
{
+ /* Chuwi Hi8 (CWI509) */
+ .matches = {
+ DMI_MATCH(DMI_BOARD_VENDOR, "Hampoo"),
+ DMI_MATCH(DMI_BOARD_NAME, "BYT-PA03C"),
+ DMI_MATCH(DMI_SYS_VENDOR, "ilife"),
+ DMI_MATCH(DMI_PRODUCT_NAME, "S806"),
+ },
+ .driver_data = (void *)(BYT_RT5640_IN1_MAP |
+ BYT_RT5640_JD_SRC_JD2_IN4N |
+ BYT_RT5640_OVCD_TH_2000UA |
+ BYT_RT5640_OVCD_SF_0P75 |
+ BYT_RT5640_MONO_SPEAKER |
+ BYT_RT5640_DIFF_MIC |
+ BYT_RT5640_SSP0_AIF1 |
+ BYT_RT5640_MCLK_EN),
+ },
+ {
.matches = {
DMI_MATCH(DMI_SYS_VENDOR, "Circuitco"),
DMI_MATCH(DMI_PRODUCT_NAME, "Minnowboard Max B3 PLATFORM"),
@@ -511,6 +561,27 @@
BYT_RT5640_MONO_SPEAKER |
BYT_RT5640_MCLK_EN),
},
+ { /* Estar Beauty HD MID 7316R */
+ .matches = {
+ DMI_MATCH(DMI_SYS_VENDOR, "Estar"),
+ DMI_MATCH(DMI_PRODUCT_NAME, "eSTAR BEAUTY HD Intel Quad core"),
+ },
+ .driver_data = (void *)(BYTCR_INPUT_DEFAULTS |
+ BYT_RT5640_MONO_SPEAKER |
+ BYT_RT5640_SSP0_AIF1 |
+ BYT_RT5640_MCLK_EN),
+ },
+ { /* Glavey TM800A550L */
+ .matches = {
+ DMI_MATCH(DMI_BOARD_VENDOR, "AMI Corporation"),
+ DMI_MATCH(DMI_BOARD_NAME, "Aptio CRB"),
+ /* Above strings are too generic, also match on BIOS version */
+ DMI_MATCH(DMI_BIOS_VERSION, "ZY-8-BI-PX4S70VTR400-X423B-005-D"),
+ },
+ .driver_data = (void *)(BYTCR_INPUT_DEFAULTS |
+ BYT_RT5640_SSP0_AIF1 |
+ BYT_RT5640_MCLK_EN),
+ },
{
.matches = {
DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Hewlett-Packard"),
@@ -578,6 +649,20 @@
BYT_RT5640_MONO_SPEAKER |
BYT_RT5640_MCLK_EN),
},
+ { /* Lenovo Miix 3-830 */
+ .matches = {
+ DMI_EXACT_MATCH(DMI_SYS_VENDOR, "LENOVO"),
+ DMI_EXACT_MATCH(DMI_PRODUCT_VERSION, "Lenovo MIIX 3-830"),
+ },
+ .driver_data = (void *)(BYT_RT5640_IN1_MAP |
+ BYT_RT5640_JD_SRC_JD2_IN4N |
+ BYT_RT5640_OVCD_TH_2000UA |
+ BYT_RT5640_OVCD_SF_0P75 |
+ BYT_RT5640_MONO_SPEAKER |
+ BYT_RT5640_DIFF_MIC |
+ BYT_RT5640_SSP0_AIF1 |
+ BYT_RT5640_MCLK_EN),
+ },
{ /* Linx Linx7 tablet */
.matches = {
DMI_EXACT_MATCH(DMI_SYS_VENDOR, "LINX"),
@@ -589,6 +674,27 @@
BYT_RT5640_SSP0_AIF1 |
BYT_RT5640_MCLK_EN),
},
+ { /* MPMAN Converter 9, similar hw as the I.T.Works TW891 2-in-1 */
+ .matches = {
+ DMI_MATCH(DMI_SYS_VENDOR, "MPMAN"),
+ DMI_MATCH(DMI_PRODUCT_NAME, "Converter9"),
+ },
+ .driver_data = (void *)(BYTCR_INPUT_DEFAULTS |
+ BYT_RT5640_MONO_SPEAKER |
+ BYT_RT5640_SSP0_AIF1 |
+ BYT_RT5640_MCLK_EN),
+ },
+ {
+ /* MPMAN MPWIN895CL */
+ .matches = {
+ DMI_EXACT_MATCH(DMI_SYS_VENDOR, "MPMAN"),
+ DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "MPWIN8900CL"),
+ },
+ .driver_data = (void *)(BYTCR_INPUT_DEFAULTS |
+ BYT_RT5640_MONO_SPEAKER |
+ BYT_RT5640_SSP0_AIF1 |
+ BYT_RT5640_MCLK_EN),
+ },
{ /* MSI S100 tablet */
.matches = {
DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Micro-Star International Co., Ltd."),
@@ -705,13 +811,17 @@
BYT_RT5640_MCLK_EN),
},
{
+ /* Teclast X89 */
.matches = {
DMI_MATCH(DMI_BOARD_VENDOR, "TECLAST"),
DMI_MATCH(DMI_BOARD_NAME, "tPAD"),
},
.driver_data = (void *)(BYT_RT5640_IN3_MAP |
- BYT_RT5640_MCLK_EN |
- BYT_RT5640_SSP0_AIF1),
+ BYT_RT5640_JD_SRC_JD1_IN4P |
+ BYT_RT5640_OVCD_TH_2000UA |
+ BYT_RT5640_OVCD_SF_1P0 |
+ BYT_RT5640_SSP0_AIF1 |
+ BYT_RT5640_MCLK_EN),
},
{ /* Toshiba Satellite Click Mini L9W-B */
.matches = {
@@ -725,6 +835,44 @@
BYT_RT5640_SSP0_AIF1 |
BYT_RT5640_MCLK_EN),
},
+ { /* Toshiba Encore WT8-A */
+ .matches = {
+ DMI_EXACT_MATCH(DMI_SYS_VENDOR, "TOSHIBA"),
+ DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "TOSHIBA WT8-A"),
+ },
+ .driver_data = (void *)(BYT_RT5640_DMIC1_MAP |
+ BYT_RT5640_JD_SRC_JD2_IN4N |
+ BYT_RT5640_OVCD_TH_2000UA |
+ BYT_RT5640_OVCD_SF_0P75 |
+ BYT_RT5640_JD_NOT_INV |
+ BYT_RT5640_MCLK_EN),
+ },
+ { /* Toshiba Encore WT10-A */
+ .matches = {
+ DMI_EXACT_MATCH(DMI_SYS_VENDOR, "TOSHIBA"),
+ DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "TOSHIBA WT10-A-103"),
+ },
+ .driver_data = (void *)(BYT_RT5640_DMIC1_MAP |
+ BYT_RT5640_JD_SRC_JD1_IN4P |
+ BYT_RT5640_OVCD_TH_2000UA |
+ BYT_RT5640_OVCD_SF_0P75 |
+ BYT_RT5640_SSP0_AIF2 |
+ BYT_RT5640_MCLK_EN),
+ },
+ { /* Voyo Winpad A15 */
+ .matches = {
+ DMI_MATCH(DMI_BOARD_VENDOR, "AMI Corporation"),
+ DMI_MATCH(DMI_BOARD_NAME, "Aptio CRB"),
+ /* Above strings are too generic, also match on BIOS date */
+ DMI_MATCH(DMI_BIOS_DATE, "11/20/2014"),
+ },
+ .driver_data = (void *)(BYT_RT5640_IN1_MAP |
+ BYT_RT5640_JD_SRC_JD2_IN4N |
+ BYT_RT5640_OVCD_TH_2000UA |
+ BYT_RT5640_OVCD_SF_0P75 |
+ BYT_RT5640_DIFF_MIC |
+ BYT_RT5640_MCLK_EN),
+ },
{ /* Catch-all for generic Insyde tablets, must be last */
.matches = {
DMI_MATCH(DMI_SYS_VENDOR, "Insyde"),
diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c
index 4606f6f..921c09c 100644
--- a/sound/soc/intel/boards/bytcr_rt5651.c
+++ b/sound/soc/intel/boards/bytcr_rt5651.c
@@ -436,6 +436,19 @@
BYT_RT5651_MONO_SPEAKER),
},
{
+ /* Jumper EZpad 7 */
+ .callback = byt_rt5651_quirk_cb,
+ .matches = {
+ DMI_MATCH(DMI_SYS_VENDOR, "Jumper"),
+ DMI_MATCH(DMI_PRODUCT_NAME, "EZpad"),
+ /* Jumper12x.WJ2012.bsBKRCP05 with the version dropped */
+ DMI_MATCH(DMI_BIOS_VERSION, "Jumper12x.WJ2012.bsBKRCP"),
+ },
+ .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS |
+ BYT_RT5651_IN2_MAP |
+ BYT_RT5651_JD_NOT_INV),
+ },
+ {
/* KIANO SlimNote 14.2 */
.callback = byt_rt5651_quirk_cb,
.matches = {
diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c
index 8879c3b..c68a5b8 100644
--- a/sound/soc/intel/boards/cht_bsw_rt5645.c
+++ b/sound/soc/intel/boards/cht_bsw_rt5645.c
@@ -48,6 +48,7 @@
#define CHT_RT5645_SSP2_AIF2 BIT(16) /* default is using AIF1 */
#define CHT_RT5645_SSP0_AIF1 BIT(17)
#define CHT_RT5645_SSP0_AIF2 BIT(18)
+#define CHT_RT5645_PMC_PLT_CLK_0 BIT(19)
static unsigned long cht_rt5645_quirk = 0;
@@ -59,6 +60,8 @@
dev_info(dev, "quirk SSP0_AIF1 enabled");
if (cht_rt5645_quirk & CHT_RT5645_SSP0_AIF2)
dev_info(dev, "quirk SSP0_AIF2 enabled");
+ if (cht_rt5645_quirk & CHT_RT5645_PMC_PLT_CLK_0)
+ dev_info(dev, "quirk PMC_PLT_CLK_0 enabled");
}
static int platform_clock_control(struct snd_soc_dapm_widget *w,
@@ -226,16 +229,22 @@
return 0;
}
-/* uncomment when we have a real quirk
static int cht_rt5645_quirk_cb(const struct dmi_system_id *id)
{
cht_rt5645_quirk = (unsigned long)id->driver_data;
return 1;
}
-*/
static const struct dmi_system_id cht_rt5645_quirk_table[] = {
{
+ /* Strago family Chromebooks */
+ .callback = cht_rt5645_quirk_cb,
+ .matches = {
+ DMI_MATCH(DMI_PRODUCT_FAMILY, "Intel_Strago"),
+ },
+ .driver_data = (void *)CHT_RT5645_PMC_PLT_CLK_0,
+ },
+ {
},
};
@@ -526,6 +535,7 @@
int dai_index = 0;
int ret_val = 0;
int i;
+ const char *mclk_name;
drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_KERNEL);
if (!drv)
@@ -662,11 +672,15 @@
if (ret_val)
return ret_val;
- drv->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3");
+ if (cht_rt5645_quirk & CHT_RT5645_PMC_PLT_CLK_0)
+ mclk_name = "pmc_plt_clk_0";
+ else
+ mclk_name = "pmc_plt_clk_3";
+
+ drv->mclk = devm_clk_get(&pdev->dev, mclk_name);
if (IS_ERR(drv->mclk)) {
- dev_err(&pdev->dev,
- "Failed to get MCLK from pmc_plt_clk_3: %ld\n",
- PTR_ERR(drv->mclk));
+ dev_err(&pdev->dev, "Failed to get MCLK from %s: %ld\n",
+ mclk_name, PTR_ERR(drv->mclk));
return PTR_ERR(drv->mclk);
}
diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c
index 3dadf9b..cf47fd9 100644
--- a/sound/soc/intel/boards/haswell.c
+++ b/sound/soc/intel/boards/haswell.c
@@ -206,6 +206,7 @@
.probe = haswell_audio_probe,
.driver = {
.name = "haswell-audio",
+ .pm = &snd_soc_pm_ops,
},
};
diff --git a/sound/soc/intel/boards/kbl_da7219_max98357a.c b/sound/soc/intel/boards/kbl_da7219_max98357a.c
index 537a889..69362ea 100644
--- a/sound/soc/intel/boards/kbl_da7219_max98357a.c
+++ b/sound/soc/intel/boards/kbl_da7219_max98357a.c
@@ -607,7 +607,7 @@
static const struct platform_device_id kbl_board_ids[] = {
{
- .name = "kbl_da7219_max98357a",
+ .name = "kbl_da7219_mx98357a",
.driver_data =
(kernel_ulong_t)&kabylake_audio_card_da7219_m98357a,
},
@@ -629,4 +629,4 @@
MODULE_DESCRIPTION("Audio Machine driver-DA7219 & MAX98357A in I2S mode");
MODULE_AUTHOR("Naveen Manohar <naveen.m@intel.com>");
MODULE_LICENSE("GPL v2");
-MODULE_ALIAS("platform:kbl_da7219_max98357a");
+MODULE_ALIAS("platform:kbl_da7219_mx98357a");
diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c
index 829f95f..16996f6 100644
--- a/sound/soc/intel/boards/kbl_da7219_max98927.c
+++ b/sound/soc/intel/boards/kbl_da7219_max98927.c
@@ -282,12 +282,34 @@
struct snd_interval *channels = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_CHANNELS);
struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
- struct snd_soc_dpcm *dpcm = container_of(
- params, struct snd_soc_dpcm, hw_params);
- struct snd_soc_dai_link *fe_dai_link = dpcm->fe->dai_link;
- struct snd_soc_dai_link *be_dai_link = dpcm->be->dai_link;
+ struct snd_soc_dpcm *dpcm, *rtd_dpcm = NULL;
/*
+ * The following loop will be called only for playback stream
+ * In this platform, there is only one playback device on every SSP
+ */
+ for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_PLAYBACK, dpcm) {
+ rtd_dpcm = dpcm;
+ break;
+ }
+
+ /*
+ * This following loop will be called only for capture stream
+ * In this platform, there is only one capture device on every SSP
+ */
+ for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_CAPTURE, dpcm) {
+ rtd_dpcm = dpcm;
+ break;
+ }
+
+ if (!rtd_dpcm)
+ return -EINVAL;
+
+ /*
+ * The above 2 loops are mutually exclusive based on the stream direction,
+ * thus rtd_dpcm variable will never be overwritten
+ */
+ /*
* Topology for kblda7219m98373 & kblmax98373 supports only S24_LE,
* where as kblda7219m98927 & kblmax98927 supports S16_LE by default.
* Skipping the port wise FE and BE configuration for kblda7219m98373 &
@@ -309,9 +331,9 @@
/*
* The ADSP will convert the FE rate to 48k, stereo, 24 bit
*/
- if (!strcmp(fe_dai_link->name, "Kbl Audio Port") ||
- !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") ||
- !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) {
+ if (!strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Port") ||
+ !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Headset Playback") ||
+ !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Capture Port")) {
rate->min = rate->max = 48000;
channels->min = channels->max = 2;
snd_mask_none(fmt);
@@ -322,7 +344,7 @@
* The speaker on the SSP0 supports S16_LE and not S24_LE.
* thus changing the mask here
*/
- if (!strcmp(be_dai_link->name, "SSP0-Codec"))
+ if (!strcmp(rtd_dpcm->be->dai_link->name, "SSP0-Codec"))
snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE);
return 0;
diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c
index 7cefda3..a540a2d 100644
--- a/sound/soc/intel/boards/kbl_rt5663_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c
@@ -401,17 +401,40 @@
struct snd_interval *channels = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_CHANNELS);
struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
- struct snd_soc_dpcm *dpcm = container_of(
- params, struct snd_soc_dpcm, hw_params);
- struct snd_soc_dai_link *fe_dai_link = dpcm->fe->dai_link;
- struct snd_soc_dai_link *be_dai_link = dpcm->be->dai_link;
+ struct snd_soc_dpcm *dpcm, *rtd_dpcm = NULL;
+
+ /*
+ * The following loop will be called only for playback stream
+ * In this platform, there is only one playback device on every SSP
+ */
+ for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_PLAYBACK, dpcm) {
+ rtd_dpcm = dpcm;
+ break;
+ }
+
+ /*
+ * This following loop will be called only for capture stream
+ * In this platform, there is only one capture device on every SSP
+ */
+ for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_CAPTURE, dpcm) {
+ rtd_dpcm = dpcm;
+ break;
+ }
+
+ if (!rtd_dpcm)
+ return -EINVAL;
+
+ /*
+ * The above 2 loops are mutually exclusive based on the stream direction,
+ * thus rtd_dpcm variable will never be overwritten
+ */
/*
* The ADSP will convert the FE rate to 48k, stereo, 24 bit
*/
- if (!strcmp(fe_dai_link->name, "Kbl Audio Port") ||
- !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") ||
- !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) {
+ if (!strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Port") ||
+ !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Headset Playback") ||
+ !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Capture Port")) {
rate->min = rate->max = 48000;
channels->min = channels->max = 2;
snd_mask_none(fmt);
@@ -421,7 +444,7 @@
* The speaker on the SSP0 supports S16_LE and not S24_LE.
* thus changing the mask here
*/
- if (!strcmp(be_dai_link->name, "SSP0-Codec"))
+ if (!strcmp(rtd_dpcm->be->dai_link->name, "SSP0-Codec"))
snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE);
return 0;
diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
index 74dda87..8ad31c9 100644
--- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
@@ -400,6 +400,9 @@
snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
dmic_constraints);
+ runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE;
+ snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16);
+
return snd_pcm_hw_constraint_list(substream->runtime, 0,
SNDRV_PCM_HW_PARAM_RATE, &constraints_rates);
}
@@ -623,7 +626,7 @@
* kabylake audio machine driver for MAX98927 + RT5514 + RT5663
*/
static struct snd_soc_card kabylake_audio_card = {
- .name = "kbl_r5514_5663_max",
+ .name = "kbl-r5514-5663-max",
.owner = THIS_MODULE,
.dai_link = kabylake_dais,
.num_links = ARRAY_SIZE(kabylake_dais),
diff --git a/sound/soc/intel/boards/skl_hda_dsp_common.c b/sound/soc/intel/boards/skl_hda_dsp_common.c
index 58409b6..e3d405e 100644
--- a/sound/soc/intel/boards/skl_hda_dsp_common.c
+++ b/sound/soc/intel/boards/skl_hda_dsp_common.c
@@ -38,16 +38,19 @@
return 0;
}
-SND_SOC_DAILINK_DEFS(idisp1,
- DAILINK_COMP_ARRAY(COMP_CPU("iDisp1 Pin")),
+SND_SOC_DAILINK_DEF(idisp1_cpu,
+ DAILINK_COMP_ARRAY(COMP_CPU("iDisp1 Pin")));
+SND_SOC_DAILINK_DEF(idisp1_codec,
DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi1")));
-SND_SOC_DAILINK_DEFS(idisp2,
- DAILINK_COMP_ARRAY(COMP_CPU("iDisp2 Pin")),
+SND_SOC_DAILINK_DEF(idisp2_cpu,
+ DAILINK_COMP_ARRAY(COMP_CPU("iDisp2 Pin")));
+SND_SOC_DAILINK_DEF(idisp2_codec,
DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi2")));
-SND_SOC_DAILINK_DEFS(idisp3,
- DAILINK_COMP_ARRAY(COMP_CPU("iDisp3 Pin")),
+SND_SOC_DAILINK_DEF(idisp3_cpu,
+ DAILINK_COMP_ARRAY(COMP_CPU("iDisp3 Pin")));
+SND_SOC_DAILINK_DEF(idisp3_codec,
DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi3")));
SND_SOC_DAILINK_DEF(analog_cpu,
@@ -80,21 +83,21 @@
.id = 1,
.dpcm_playback = 1,
.no_pcm = 1,
- SND_SOC_DAILINK_REG(idisp1),
+ SND_SOC_DAILINK_REG(idisp1_cpu, idisp1_codec, platform),
},
{
.name = "iDisp2",
.id = 2,
.dpcm_playback = 1,
.no_pcm = 1,
- SND_SOC_DAILINK_REG(idisp2),
+ SND_SOC_DAILINK_REG(idisp2_cpu, idisp2_codec, platform),
},
{
.name = "iDisp3",
.id = 3,
.dpcm_playback = 1,
.no_pcm = 1,
- SND_SOC_DAILINK_REG(idisp3),
+ SND_SOC_DAILINK_REG(idisp3_cpu, idisp3_codec, platform),
},
{
.name = "Analog Playback and Capture",
diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c
index 1778acd..e8d676c 100644
--- a/sound/soc/intel/boards/skl_hda_dsp_generic.c
+++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c
@@ -90,7 +90,7 @@
}
static struct snd_soc_card hda_soc_card = {
- .name = "skl_hda_card",
+ .name = "hda-dsp",
.owner = THIS_MODULE,
.dai_link = skl_hda_be_dai_links,
.dapm_widgets = skl_hda_widgets,
diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c
index 4f6e58c..302ca19 100644
--- a/sound/soc/intel/boards/sof_rt5682.c
+++ b/sound/soc/intel/boards/sof_rt5682.c
@@ -34,6 +34,10 @@
#define SOF_RT5682_SSP_AMP(quirk) \
(((quirk) << SOF_RT5682_SSP_AMP_SHIFT) & SOF_RT5682_SSP_AMP_MASK)
#define SOF_RT5682_MCLK_BYTCHT_EN BIT(9)
+#define SOF_RT5682_NUM_HDMIDEV_SHIFT 10
+#define SOF_RT5682_NUM_HDMIDEV_MASK (GENMASK(12, 10))
+#define SOF_RT5682_NUM_HDMIDEV(quirk) \
+ ((quirk << SOF_RT5682_NUM_HDMIDEV_SHIFT) & SOF_RT5682_NUM_HDMIDEV_MASK)
/* Default: MCLK on, MCLK 19.2M, SSP0 */
static unsigned long sof_rt5682_quirk = SOF_RT5682_MCLK_EN |
@@ -370,7 +374,7 @@
/* sof audio machine driver for rt5682 codec */
static struct snd_soc_card sof_audio_card_rt5682 = {
- .name = "sof_rt5682",
+ .name = "rt5682", /* the sof- prefix is added by the core */
.owner = THIS_MODULE,
.controls = sof_controls,
.num_controls = ARRAY_SIZE(sof_controls),
@@ -585,6 +589,19 @@
if (!ctx)
return -ENOMEM;
+ if (pdev->id_entry && pdev->id_entry->driver_data)
+ sof_rt5682_quirk = (unsigned long)pdev->id_entry->driver_data;
+
+ dmi_check_system(sof_rt5682_quirk_table);
+
+ mach = (&pdev->dev)->platform_data;
+
+ /* A speaker amp might not be present when the quirk claims one is.
+ * Detect this via whether the machine driver match includes quirk_data.
+ */
+ if ((sof_rt5682_quirk & SOF_SPEAKER_AMP_PRESENT) && !mach->quirk_data)
+ sof_rt5682_quirk &= ~SOF_SPEAKER_AMP_PRESENT;
+
if (soc_intel_is_byt() || soc_intel_is_cht()) {
is_legacy_cpu = 1;
dmic_be_num = 0;
@@ -595,11 +612,13 @@
SOF_RT5682_SSP_CODEC(2);
} else {
dmic_be_num = 2;
- hdmi_num = 3;
+ hdmi_num = (sof_rt5682_quirk & SOF_RT5682_NUM_HDMIDEV_MASK) >>
+ SOF_RT5682_NUM_HDMIDEV_SHIFT;
+ /* default number of HDMI DAI's */
+ if (!hdmi_num)
+ hdmi_num = 3;
}
- dmi_check_system(sof_rt5682_quirk_table);
-
/* need to get main clock from pmc */
if (sof_rt5682_quirk & SOF_RT5682_MCLK_BYTCHT_EN) {
ctx->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3");
@@ -643,7 +662,6 @@
INIT_LIST_HEAD(&ctx->hdmi_pcm_list);
sof_audio_card_rt5682.dev = &pdev->dev;
- mach = (&pdev->dev)->platform_data;
/* set platform name for each dailink */
ret = snd_soc_fixup_dai_links_platform_name(&sof_audio_card_rt5682,
@@ -672,6 +690,21 @@
return 0;
}
+static const struct platform_device_id board_ids[] = {
+ {
+ .name = "sof_rt5682",
+ },
+ {
+ .name = "tgl_max98357a_rt5682",
+ .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN |
+ SOF_RT5682_SSP_CODEC(0) |
+ SOF_SPEAKER_AMP_PRESENT |
+ SOF_RT5682_SSP_AMP(1) |
+ SOF_RT5682_NUM_HDMIDEV(4)),
+ },
+ { }
+};
+
static struct platform_driver sof_audio = {
.probe = sof_audio_probe,
.remove = sof_rt5682_remove,
@@ -679,6 +712,7 @@
.name = "sof_rt5682",
.pm = &snd_soc_pm_ops,
},
+ .id_table = board_ids,
};
module_platform_driver(sof_audio)
@@ -688,3 +722,4 @@
MODULE_AUTHOR("Sathya Prakash M R <sathya.prakash.m.r@intel.com>");
MODULE_LICENSE("GPL v2");
MODULE_ALIAS("platform:sof_rt5682");
+MODULE_ALIAS("platform:tgl_max98357a_rt5682");
diff --git a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c
index e200baa..df7f82e 100644
--- a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c
@@ -113,7 +113,7 @@
},
{
.id = "DLGS7219",
- .drv_name = "kbl_da7219_max98373",
+ .drv_name = "kbl_da7219_mx98373",
.fw_filename = "intel/dsp_fw_kbl.bin",
.machine_quirk = snd_soc_acpi_codec_list,
.quirk_data = &kbl_7219_98373_codecs,
diff --git a/sound/soc/intel/common/soc-intel-quirks.h b/sound/soc/intel/common/soc-intel-quirks.h
index 863a477..645baf0 100644
--- a/sound/soc/intel/common/soc-intel-quirks.h
+++ b/sound/soc/intel/common/soc-intel-quirks.h
@@ -11,6 +11,7 @@
#if IS_ENABLED(CONFIG_X86)
+#include <linux/dmi.h>
#include <asm/cpu_device_id.h>
#include <asm/intel-family.h>
#include <asm/iosf_mbi.h>
@@ -40,12 +41,36 @@
static inline bool soc_intel_is_byt_cr(struct platform_device *pdev)
{
+ /*
+ * List of systems which:
+ * 1. Use a non CR version of the Bay Trail SoC
+ * 2. Contain at least 6 interrupt resources so that the
+ * platform_get_resource(pdev, IORESOURCE_IRQ, 5) check below
+ * succeeds
+ * 3. Despite 1. and 2. still have their IPC IRQ at index 0 rather then 5
+ *
+ * This needs to be here so that it can be shared between the SST and
+ * SOF drivers. We rely on the compiler to optimize this out in files
+ * where soc_intel_is_byt_cr is not used.
+ */
+ static const struct dmi_system_id force_bytcr_table[] = {
+ { /* Lenovo Yoga Tablet 2 series */
+ .matches = {
+ DMI_MATCH(DMI_SYS_VENDOR, "LENOVO"),
+ DMI_MATCH(DMI_PRODUCT_FAMILY, "YOGATablet2"),
+ },
+ },
+ {}
+ };
struct device *dev = &pdev->dev;
int status = 0;
if (!soc_intel_is_byt())
return false;
+ if (dmi_check_system(force_bytcr_table))
+ return true;
+
if (iosf_mbi_available()) {
u32 bios_status;
diff --git a/sound/soc/intel/skylake/bxt-sst.c b/sound/soc/intel/skylake/bxt-sst.c
index 92a82e6..38b9d74 100644
--- a/sound/soc/intel/skylake/bxt-sst.c
+++ b/sound/soc/intel/skylake/bxt-sst.c
@@ -17,7 +17,6 @@
#include "skl.h"
#define BXT_BASEFW_TIMEOUT 3000
-#define BXT_INIT_TIMEOUT 300
#define BXT_ROM_INIT_TIMEOUT 70
#define BXT_IPC_PURGE_FW 0x01004000
@@ -38,8 +37,6 @@
/* Delay before scheduling D0i3 entry */
#define BXT_D0I3_DELAY 5000
-#define BXT_FW_ROM_INIT_RETRY 3
-
static unsigned int bxt_get_errorcode(struct sst_dsp *ctx)
{
return sst_dsp_shim_read(ctx, BXT_ADSP_ERROR_CODE);
diff --git a/sound/soc/intel/skylake/cnl-sst.c b/sound/soc/intel/skylake/cnl-sst.c
index 4f64f09..e808f62 100644
--- a/sound/soc/intel/skylake/cnl-sst.c
+++ b/sound/soc/intel/skylake/cnl-sst.c
@@ -57,18 +57,34 @@
ctx->dsp_ops.stream_tag = stream_tag;
memcpy(ctx->dmab.area, fwdata, fwsize);
+ ret = skl_dsp_core_power_up(ctx, SKL_DSP_CORE0_MASK);
+ if (ret < 0) {
+ dev_err(ctx->dev, "dsp core0 power up failed\n");
+ ret = -EIO;
+ goto base_fw_load_failed;
+ }
+
/* purge FW request */
sst_dsp_shim_write(ctx, CNL_ADSP_REG_HIPCIDR,
CNL_ADSP_REG_HIPCIDR_BUSY | (CNL_IPC_PURGE |
((stream_tag - 1) << CNL_ROM_CTRL_DMA_ID)));
- ret = cnl_dsp_enable_core(ctx, SKL_DSP_CORE0_MASK);
+ ret = skl_dsp_start_core(ctx, SKL_DSP_CORE0_MASK);
if (ret < 0) {
- dev_err(ctx->dev, "dsp boot core failed ret: %d\n", ret);
+ dev_err(ctx->dev, "Start dsp core failed ret: %d\n", ret);
ret = -EIO;
goto base_fw_load_failed;
}
+ ret = sst_dsp_register_poll(ctx, CNL_ADSP_REG_HIPCIDA,
+ CNL_ADSP_REG_HIPCIDA_DONE,
+ CNL_ADSP_REG_HIPCIDA_DONE,
+ BXT_INIT_TIMEOUT, "HIPCIDA Done");
+ if (ret < 0) {
+ dev_err(ctx->dev, "timeout for purge request: %d\n", ret);
+ goto base_fw_load_failed;
+ }
+
/* enable interrupt */
cnl_ipc_int_enable(ctx);
cnl_ipc_op_int_enable(ctx);
@@ -109,7 +125,7 @@
{
struct firmware stripped_fw;
struct skl_dev *cnl = ctx->thread_context;
- int ret;
+ int ret, i;
if (!ctx->fw) {
ret = request_firmware(&ctx->fw, ctx->fw_name, ctx->dev);
@@ -131,12 +147,16 @@
stripped_fw.size = ctx->fw->size;
skl_dsp_strip_extended_manifest(&stripped_fw);
- ret = cnl_prepare_fw(ctx, stripped_fw.data, stripped_fw.size);
- if (ret < 0) {
- dev_err(ctx->dev, "prepare firmware failed: %d\n", ret);
- goto cnl_load_base_firmware_failed;
+ for (i = 0; i < BXT_FW_ROM_INIT_RETRY; i++) {
+ ret = cnl_prepare_fw(ctx, stripped_fw.data, stripped_fw.size);
+ if (!ret)
+ break;
+ dev_dbg(ctx->dev, "prepare firmware failed: %d\n", ret);
}
+ if (ret < 0)
+ goto cnl_load_base_firmware_failed;
+
ret = sst_transfer_fw_host_dma(ctx);
if (ret < 0) {
dev_err(ctx->dev, "transfer firmware failed: %d\n", ret);
@@ -158,6 +178,7 @@
return 0;
cnl_load_base_firmware_failed:
+ dev_err(ctx->dev, "firmware load failed: %d\n", ret);
release_firmware(ctx->fw);
ctx->fw = NULL;
@@ -203,6 +224,7 @@
"dsp boot timeout, status=%#x error=%#x\n",
sst_dsp_shim_read(ctx, CNL_ADSP_FW_STATUS),
sst_dsp_shim_read(ctx, CNL_ADSP_ERROR_CODE));
+ ret = -ETIMEDOUT;
goto err;
}
} else {
diff --git a/sound/soc/intel/skylake/skl-debug.c b/sound/soc/intel/skylake/skl-debug.c
index 3466675..a15aa2f 100644
--- a/sound/soc/intel/skylake/skl-debug.c
+++ b/sound/soc/intel/skylake/skl-debug.c
@@ -34,8 +34,8 @@
int i;
ssize_t ret = 0;
- for (i = 0; i < max_pin; i++)
- ret += snprintf(buf + size, MOD_BUF - size,
+ for (i = 0; i < max_pin; i++) {
+ ret += scnprintf(buf + size, MOD_BUF - size,
"%s %d\n\tModule %d\n\tInstance %d\n\t"
"In-used %s\n\tType %s\n"
"\tState %d\n\tIndex %d\n",
@@ -45,13 +45,15 @@
m_pin[i].in_use ? "Used" : "Unused",
m_pin[i].is_dynamic ? "Dynamic" : "Static",
m_pin[i].pin_state, i);
+ size += ret;
+ }
return ret;
}
static ssize_t skl_print_fmt(struct skl_module_fmt *fmt, char *buf,
ssize_t size, bool direction)
{
- return snprintf(buf + size, MOD_BUF - size,
+ return scnprintf(buf + size, MOD_BUF - size,
"%s\n\tCh %d\n\tFreq %d\n\tBit depth %d\n\t"
"Valid bit depth %d\n\tCh config %#x\n\tInterleaving %d\n\t"
"Sample Type %d\n\tCh Map %#x\n",
@@ -75,16 +77,16 @@
if (!buf)
return -ENOMEM;
- ret = snprintf(buf, MOD_BUF, "Module:\n\tUUID %pUL\n\tModule id %d\n"
+ ret = scnprintf(buf, MOD_BUF, "Module:\n\tUUID %pUL\n\tModule id %d\n"
"\tInstance id %d\n\tPvt_id %d\n", mconfig->guid,
mconfig->id.module_id, mconfig->id.instance_id,
mconfig->id.pvt_id);
- ret += snprintf(buf + ret, MOD_BUF - ret,
+ ret += scnprintf(buf + ret, MOD_BUF - ret,
"Resources:\n\tCPC %#x\n\tIBS %#x\n\tOBS %#x\t\n",
res->cpc, res->ibs, res->obs);
- ret += snprintf(buf + ret, MOD_BUF - ret,
+ ret += scnprintf(buf + ret, MOD_BUF - ret,
"Module data:\n\tCore %d\n\tIn queue %d\n\t"
"Out queue %d\n\tType %s\n",
mconfig->core_id, mconfig->max_in_queue,
@@ -94,38 +96,38 @@
ret += skl_print_fmt(mconfig->in_fmt, buf, ret, true);
ret += skl_print_fmt(mconfig->out_fmt, buf, ret, false);
- ret += snprintf(buf + ret, MOD_BUF - ret,
+ ret += scnprintf(buf + ret, MOD_BUF - ret,
"Fixup:\n\tParams %#x\n\tConverter %#x\n",
mconfig->params_fixup, mconfig->converter);
- ret += snprintf(buf + ret, MOD_BUF - ret,
+ ret += scnprintf(buf + ret, MOD_BUF - ret,
"Module Gateway:\n\tType %#x\n\tVbus %#x\n\tHW conn %#x\n\tSlot %#x\n",
mconfig->dev_type, mconfig->vbus_id,
mconfig->hw_conn_type, mconfig->time_slot);
- ret += snprintf(buf + ret, MOD_BUF - ret,
+ ret += scnprintf(buf + ret, MOD_BUF - ret,
"Pipeline:\n\tID %d\n\tPriority %d\n\tConn Type %d\n\t"
"Pages %#x\n", mconfig->pipe->ppl_id,
mconfig->pipe->pipe_priority, mconfig->pipe->conn_type,
mconfig->pipe->memory_pages);
- ret += snprintf(buf + ret, MOD_BUF - ret,
+ ret += scnprintf(buf + ret, MOD_BUF - ret,
"\tParams:\n\t\tHost DMA %d\n\t\tLink DMA %d\n",
mconfig->pipe->p_params->host_dma_id,
mconfig->pipe->p_params->link_dma_id);
- ret += snprintf(buf + ret, MOD_BUF - ret,
+ ret += scnprintf(buf + ret, MOD_BUF - ret,
"\tPCM params:\n\t\tCh %d\n\t\tFreq %d\n\t\tFormat %d\n",
mconfig->pipe->p_params->ch,
mconfig->pipe->p_params->s_freq,
mconfig->pipe->p_params->s_fmt);
- ret += snprintf(buf + ret, MOD_BUF - ret,
+ ret += scnprintf(buf + ret, MOD_BUF - ret,
"\tLink %#x\n\tStream %#x\n",
mconfig->pipe->p_params->linktype,
mconfig->pipe->p_params->stream);
- ret += snprintf(buf + ret, MOD_BUF - ret,
+ ret += scnprintf(buf + ret, MOD_BUF - ret,
"\tState %d\n\tPassthru %s\n",
mconfig->pipe->state,
mconfig->pipe->passthru ? "true" : "false");
@@ -135,7 +137,7 @@
ret += skl_print_pins(mconfig->m_out_pin, buf,
mconfig->max_out_queue, ret, false);
- ret += snprintf(buf + ret, MOD_BUF - ret,
+ ret += scnprintf(buf + ret, MOD_BUF - ret,
"Other:\n\tDomain %d\n\tHomogeneous Input %s\n\t"
"Homogeneous Output %s\n\tIn Queue Mask %d\n\t"
"Out Queue Mask %d\n\tDMA ID %d\n\tMem Pages %d\n\t"
@@ -191,7 +193,7 @@
__ioread32_copy(d->fw_read_buff, fw_reg_addr, w0_stat_sz >> 2);
for (offset = 0; offset < FW_REG_SIZE; offset += 16) {
- ret += snprintf(tmp + ret, FW_REG_BUF - ret, "%#.4x: ", offset);
+ ret += scnprintf(tmp + ret, FW_REG_BUF - ret, "%#.4x: ", offset);
hex_dump_to_buffer(d->fw_read_buff + offset, 16, 16, 4,
tmp + ret, FW_REG_BUF - ret, 0);
ret += strlen(tmp + ret);
diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c
index 476ef18..79c6cf2 100644
--- a/sound/soc/intel/skylake/skl-messages.c
+++ b/sound/soc/intel/skylake/skl-messages.c
@@ -802,9 +802,12 @@
case SKL_MODULE_TYPE_BASE_OUTFMT:
case SKL_MODULE_TYPE_MIC_SELECT:
- case SKL_MODULE_TYPE_KPB:
return sizeof(struct skl_base_outfmt_cfg);
+ case SKL_MODULE_TYPE_MIXER:
+ case SKL_MODULE_TYPE_KPB:
+ return sizeof(struct skl_base_cfg);
+
default:
/*
* return only base cfg when no specific module type is
@@ -857,10 +860,14 @@
case SKL_MODULE_TYPE_BASE_OUTFMT:
case SKL_MODULE_TYPE_MIC_SELECT:
- case SKL_MODULE_TYPE_KPB:
skl_set_base_outfmt_format(skl, module_config, *param_data);
break;
+ case SKL_MODULE_TYPE_MIXER:
+ case SKL_MODULE_TYPE_KPB:
+ skl_set_base_module_format(skl, module_config, *param_data);
+ break;
+
default:
skl_set_base_module_format(skl, module_config, *param_data);
break;
diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c
index 19f328d..d9c8f5c 100644
--- a/sound/soc/intel/skylake/skl-nhlt.c
+++ b/sound/soc/intel/skylake/skl-nhlt.c
@@ -182,7 +182,8 @@
{
struct device *dev = &skl->pci->dev;
- sysfs_remove_file(&dev->kobj, &dev_attr_platform_id.attr);
+ if (skl->nhlt)
+ sysfs_remove_file(&dev->kobj, &dev_attr_platform_id.attr);
}
/*
diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c
index 7f28742..439dd4b 100644
--- a/sound/soc/intel/skylake/skl-pcm.c
+++ b/sound/soc/intel/skylake/skl-pcm.c
@@ -1333,21 +1333,6 @@
return -EIO;
}
- list_for_each_entry(module, &skl->uuid_list, list) {
- if (guid_equal(uuid_mod, &module->uuid)) {
- mconfig->id.module_id = module->id;
- if (mconfig->module)
- mconfig->module->loadable = module->is_loadable;
- ret = 0;
- break;
- }
- }
-
- if (ret)
- return ret;
-
- uuid_mod = &module->uuid;
- ret = -EIO;
for (i = 0; i < skl->nr_modules; i++) {
skl_module = skl->modules[i];
uuid_tplg = &skl_module->uuid;
@@ -1357,10 +1342,18 @@
break;
}
}
+
if (skl->nr_modules && ret)
return ret;
+ ret = -EIO;
list_for_each_entry(module, &skl->uuid_list, list) {
+ if (guid_equal(uuid_mod, &module->uuid)) {
+ mconfig->id.module_id = module->id;
+ mconfig->module->loadable = module->is_loadable;
+ ret = 0;
+ }
+
for (i = 0; i < MAX_IN_QUEUE; i++) {
pin_id = &mconfig->m_in_pin[i].id;
if (guid_equal(&pin_id->mod_uuid, &module->uuid))
@@ -1374,7 +1367,7 @@
}
}
- return 0;
+ return ret;
}
static int skl_populate_modules(struct skl_dev *skl)
diff --git a/sound/soc/intel/skylake/skl-ssp-clk.c b/sound/soc/intel/skylake/skl-ssp-clk.c
index 1c0e522..bd43885 100644
--- a/sound/soc/intel/skylake/skl-ssp-clk.c
+++ b/sound/soc/intel/skylake/skl-ssp-clk.c
@@ -384,9 +384,11 @@
&clks[i], clk_pdata, i);
if (IS_ERR(data->clk[data->avail_clk_cnt])) {
- ret = PTR_ERR(data->clk[data->avail_clk_cnt++]);
+ ret = PTR_ERR(data->clk[data->avail_clk_cnt]);
goto err_unreg_skl_clk;
}
+
+ data->avail_clk_cnt++;
}
platform_set_drvdata(pdev, data);
diff --git a/sound/soc/intel/skylake/skl-sst-dsp.h b/sound/soc/intel/skylake/skl-sst-dsp.h
index cdfec0f..1df9ef4 100644
--- a/sound/soc/intel/skylake/skl-sst-dsp.h
+++ b/sound/soc/intel/skylake/skl-sst-dsp.h
@@ -67,6 +67,8 @@
#define SKL_FW_INIT 0x1
#define SKL_FW_RFW_START 0xf
+#define BXT_FW_ROM_INIT_RETRY 3
+#define BXT_INIT_TIMEOUT 300
#define SKL_ADSPIC_IPC 1
#define SKL_ADSPIS_IPC 1
diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c
index 69cd7a8..254b796 100644
--- a/sound/soc/intel/skylake/skl-topology.c
+++ b/sound/soc/intel/skylake/skl-topology.c
@@ -14,6 +14,7 @@
#include <linux/uuid.h>
#include <sound/intel-nhlt.h>
#include <sound/soc.h>
+#include <sound/soc-acpi.h>
#include <sound/soc-topology.h>
#include <uapi/sound/snd_sst_tokens.h>
#include <uapi/sound/skl-tplg-interface.h>
@@ -112,7 +113,7 @@
static void skl_dump_mconfig(struct skl_dev *skl, struct skl_module_cfg *mcfg)
{
- struct skl_module_iface *iface = &mcfg->module->formats[0];
+ struct skl_module_iface *iface = &mcfg->module->formats[mcfg->fmt_idx];
dev_dbg(skl->dev, "Dumping config\n");
dev_dbg(skl->dev, "Input Format:\n");
@@ -194,8 +195,8 @@
struct skl_module_fmt *in_fmt, *out_fmt;
/* Fixups will be applied to pin 0 only */
- in_fmt = &m_cfg->module->formats[0].inputs[0].fmt;
- out_fmt = &m_cfg->module->formats[0].outputs[0].fmt;
+ in_fmt = &m_cfg->module->formats[m_cfg->fmt_idx].inputs[0].fmt;
+ out_fmt = &m_cfg->module->formats[m_cfg->fmt_idx].outputs[0].fmt;
if (params->stream == SNDRV_PCM_STREAM_PLAYBACK) {
if (is_fe) {
@@ -238,9 +239,9 @@
/* Since fixups is applied to pin 0 only, ibs, obs needs
* change for pin 0 only
*/
- res = &mcfg->module->resources[0];
- in_fmt = &mcfg->module->formats[0].inputs[0].fmt;
- out_fmt = &mcfg->module->formats[0].outputs[0].fmt;
+ res = &mcfg->module->resources[mcfg->res_idx];
+ in_fmt = &mcfg->module->formats[mcfg->fmt_idx].inputs[0].fmt;
+ out_fmt = &mcfg->module->formats[mcfg->fmt_idx].outputs[0].fmt;
if (mcfg->m_type == SKL_MODULE_TYPE_SRCINT)
multiplier = 5;
@@ -578,6 +579,38 @@
return ret;
}
+static bool skl_tplg_is_multi_fmt(struct skl_dev *skl, struct skl_pipe *pipe)
+{
+ struct skl_pipe_fmt *cur_fmt;
+ struct skl_pipe_fmt *next_fmt;
+ int i;
+
+ if (pipe->nr_cfgs <= 1)
+ return false;
+
+ if (pipe->conn_type != SKL_PIPE_CONN_TYPE_FE)
+ return true;
+
+ for (i = 0; i < pipe->nr_cfgs - 1; i++) {
+ if (pipe->direction == SNDRV_PCM_STREAM_PLAYBACK) {
+ cur_fmt = &pipe->configs[i].out_fmt;
+ next_fmt = &pipe->configs[i + 1].out_fmt;
+ } else {
+ cur_fmt = &pipe->configs[i].in_fmt;
+ next_fmt = &pipe->configs[i + 1].in_fmt;
+ }
+
+ if (!CHECK_HW_PARAMS(cur_fmt->channels, cur_fmt->freq,
+ cur_fmt->bps,
+ next_fmt->channels,
+ next_fmt->freq,
+ next_fmt->bps))
+ return true;
+ }
+
+ return false;
+}
+
/*
* Here, we select pipe format based on the pipe type and pipe
* direction to determine the current config index for the pipeline.
@@ -600,6 +633,14 @@
return 0;
}
+ if (skl_tplg_is_multi_fmt(skl, pipe)) {
+ pipe->cur_config_idx = pipe->pipe_config_idx;
+ pipe->memory_pages = pconfig->mem_pages;
+ dev_dbg(skl->dev, "found pipe config idx:%d\n",
+ pipe->cur_config_idx);
+ return 0;
+ }
+
if (pipe->conn_type == SKL_PIPE_CONN_TYPE_NONE) {
dev_dbg(skl->dev, "No conn_type detected, take 0th config\n");
pipe->cur_config_idx = 0;
@@ -1314,6 +1355,68 @@
return 0;
}
+static int skl_tplg_multi_config_set_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol,
+ bool is_set)
+{
+ struct snd_soc_component *component =
+ snd_soc_kcontrol_component(kcontrol);
+ struct hdac_bus *bus = snd_soc_component_get_drvdata(component);
+ struct skl_dev *skl = bus_to_skl(bus);
+ struct skl_pipeline *ppl;
+ struct skl_pipe *pipe = NULL;
+ struct soc_enum *ec = (struct soc_enum *)kcontrol->private_value;
+ u32 *pipe_id;
+
+ if (!ec)
+ return -EINVAL;
+
+ if (is_set && ucontrol->value.enumerated.item[0] > ec->items)
+ return -EINVAL;
+
+ pipe_id = ec->dobj.private;
+
+ list_for_each_entry(ppl, &skl->ppl_list, node) {
+ if (ppl->pipe->ppl_id == *pipe_id) {
+ pipe = ppl->pipe;
+ break;
+ }
+ }
+ if (!pipe)
+ return -EIO;
+
+ if (is_set)
+ pipe->pipe_config_idx = ucontrol->value.enumerated.item[0];
+ else
+ ucontrol->value.enumerated.item[0] = pipe->pipe_config_idx;
+
+ return 0;
+}
+
+static int skl_tplg_multi_config_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ return skl_tplg_multi_config_set_get(kcontrol, ucontrol, false);
+}
+
+static int skl_tplg_multi_config_set(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ return skl_tplg_multi_config_set_get(kcontrol, ucontrol, true);
+}
+
+static int skl_tplg_multi_config_get_dmic(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ return skl_tplg_multi_config_set_get(kcontrol, ucontrol, false);
+}
+
+static int skl_tplg_multi_config_set_dmic(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ return skl_tplg_multi_config_set_get(kcontrol, ucontrol, true);
+}
+
static int skl_tplg_tlv_control_get(struct snd_kcontrol *kcontrol,
unsigned int __user *data, unsigned int size)
{
@@ -1360,12 +1463,6 @@
struct skl_dev *skl = get_skl_ctx(w->dapm->dev);
if (ac->params) {
- /*
- * Widget data is expected to be stripped of T and L
- */
- size -= 2 * sizeof(unsigned int);
- data += 2;
-
if (size > ac->max)
return -EINVAL;
ac->size = size;
@@ -1534,11 +1631,12 @@
struct skl_module_cfg *mconfig,
struct skl_pipe_params *params)
{
- struct skl_module_res *res = &mconfig->module->resources[0];
+ struct skl_module_res *res;
struct skl_dev *skl = get_skl_ctx(dev);
struct skl_module_fmt *format = NULL;
u8 cfg_idx = mconfig->pipe->cur_config_idx;
+ res = &mconfig->module->resources[mconfig->res_idx];
skl_tplg_fill_dma_id(mconfig, params);
mconfig->fmt_idx = mconfig->mod_cfg[cfg_idx].fmt_idx;
mconfig->res_idx = mconfig->mod_cfg[cfg_idx].res_idx;
@@ -1547,9 +1645,9 @@
return 0;
if (params->stream == SNDRV_PCM_STREAM_PLAYBACK)
- format = &mconfig->module->formats[0].inputs[0].fmt;
+ format = &mconfig->module->formats[mconfig->fmt_idx].inputs[0].fmt;
else
- format = &mconfig->module->formats[0].outputs[0].fmt;
+ format = &mconfig->module->formats[mconfig->fmt_idx].outputs[0].fmt;
/* set the hw_params */
format->s_freq = params->s_freq;
@@ -1853,6 +1951,16 @@
.get = skl_tplg_mic_control_get,
.put = skl_tplg_mic_control_set,
},
+ {
+ .id = SKL_CONTROL_TYPE_MULTI_IO_SELECT,
+ .get = skl_tplg_multi_config_get,
+ .put = skl_tplg_multi_config_set,
+ },
+ {
+ .id = SKL_CONTROL_TYPE_MULTI_IO_SELECT_DMIC,
+ .get = skl_tplg_multi_config_get_dmic,
+ .put = skl_tplg_multi_config_set_dmic,
+ }
};
static int skl_tplg_fill_pipe_cfg(struct device *dev,
@@ -3013,12 +3121,21 @@
case SND_SOC_TPLG_CTL_ENUM:
tplg_ec = container_of(hdr,
struct snd_soc_tplg_enum_control, hdr);
- if (kctl->access & SNDRV_CTL_ELEM_ACCESS_READWRITE) {
+ if (kctl->access & SNDRV_CTL_ELEM_ACCESS_READ) {
se = (struct soc_enum *)kctl->private_value;
if (tplg_ec->priv.size)
- return skl_init_enum_data(bus->dev, se,
- tplg_ec);
+ skl_init_enum_data(bus->dev, se, tplg_ec);
}
+
+ /*
+ * now that the control initializations are done, remove
+ * write permission for the DMIC configuration enums to
+ * avoid conflicts between NHLT settings and user interaction
+ */
+
+ if (hdr->ops.get == SKL_CONTROL_TYPE_MULTI_IO_SELECT_DMIC)
+ kctl->access = SNDRV_CTL_ELEM_ACCESS_READ;
+
break;
default:
@@ -3488,6 +3605,38 @@
return 0;
}
+static void skl_tplg_complete(struct snd_soc_component *component)
+{
+ struct snd_soc_dobj *dobj;
+ struct snd_soc_acpi_mach *mach =
+ dev_get_platdata(component->card->dev);
+ int i;
+
+ list_for_each_entry(dobj, &component->dobj_list, list) {
+ struct snd_kcontrol *kcontrol = dobj->control.kcontrol;
+ struct soc_enum *se;
+ char **texts;
+ char chan_text[4];
+
+ if (dobj->type != SND_SOC_DOBJ_ENUM || !kcontrol ||
+ kcontrol->put != skl_tplg_multi_config_set_dmic)
+ continue;
+
+ se = (struct soc_enum *)kcontrol->private_value;
+ texts = dobj->control.dtexts;
+ sprintf(chan_text, "c%d", mach->mach_params.dmic_num);
+
+ for (i = 0; i < se->items; i++) {
+ struct snd_ctl_elem_value val = {};
+
+ if (strstr(texts[i], chan_text)) {
+ val.value.enumerated.item[0] = i;
+ kcontrol->put(kcontrol, &val);
+ }
+ }
+ }
+}
+
static struct snd_soc_tplg_ops skl_tplg_ops = {
.widget_load = skl_tplg_widget_load,
.control_load = skl_tplg_control_load,
@@ -3497,6 +3646,7 @@
.io_ops_count = ARRAY_SIZE(skl_tplg_kcontrol_ops),
.manifest = skl_manifest_load,
.dai_load = skl_dai_load,
+ .complete = skl_tplg_complete,
};
/*
@@ -3565,8 +3715,20 @@
ret = request_firmware(&fw, skl->tplg_name, bus->dev);
if (ret < 0) {
- dev_info(bus->dev, "tplg fw %s load failed with %d, falling back to dfw_sst.bin",
- skl->tplg_name, ret);
+ char alt_tplg_name[64];
+
+ snprintf(alt_tplg_name, sizeof(alt_tplg_name), "%s-tplg.bin",
+ skl->mach->drv_name);
+ dev_info(bus->dev, "tplg fw %s load failed with %d, trying alternative tplg name %s",
+ skl->tplg_name, ret, alt_tplg_name);
+
+ ret = request_firmware(&fw, alt_tplg_name, bus->dev);
+ if (!ret)
+ goto component_load;
+
+ dev_info(bus->dev, "tplg %s failed with %d, falling back to dfw_sst.bin",
+ alt_tplg_name, ret);
+
ret = request_firmware(&fw, "dfw_sst.bin", bus->dev);
if (ret < 0) {
dev_err(bus->dev, "Fallback tplg fw %s load failed with %d\n",
@@ -3575,6 +3737,8 @@
}
}
+component_load:
+
/*
* The complete tplg for SKL is loaded as index 0, we don't use
* any other index
diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h
index e967800..0657614 100644
--- a/sound/soc/intel/skylake/skl-topology.h
+++ b/sound/soc/intel/skylake/skl-topology.h
@@ -306,6 +306,7 @@
struct skl_path_config configs[SKL_MAX_PATH_CONFIGS];
struct list_head w_list;
bool passthru;
+ u32 pipe_config_idx;
};
enum skl_module_state {
diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c
index 141dbbf..2e5fbd2 100644
--- a/sound/soc/intel/skylake/skl.c
+++ b/sound/soc/intel/skylake/skl.c
@@ -129,6 +129,7 @@
struct hdac_ext_link *hlink;
int ret;
+ snd_hdac_set_codec_wakeup(bus, true);
skl_enable_miscbdcge(bus->dev, false);
ret = snd_hdac_bus_init_chip(bus, full_reset);
@@ -137,6 +138,7 @@
writel(0, hlink->ml_addr + AZX_REG_ML_LOSIDV);
skl_enable_miscbdcge(bus->dev, true);
+ snd_hdac_set_codec_wakeup(bus, false);
return ret;
}
@@ -480,13 +482,8 @@
static struct snd_soc_acpi_mach *skl_find_hda_machine(struct skl_dev *skl,
struct snd_soc_acpi_mach *machines)
{
- struct hdac_bus *bus = skl_to_bus(skl);
struct snd_soc_acpi_mach *mach;
- /* check if we have any codecs detected on bus */
- if (bus->codec_mask == 0)
- return NULL;
-
/* point to common table */
mach = snd_soc_acpi_intel_hda_machines;
@@ -635,6 +632,9 @@
struct platform_device_info pdevinfo = {NULL};
struct skl_clk_pdata *clk_pdata;
+ if (!skl->nhlt)
+ return 0;
+
clk_pdata = devm_kzalloc(&skl->pci->dev, sizeof(*clk_pdata),
GFP_KERNEL);
if (!clk_pdata)
@@ -807,6 +807,9 @@
return;
}
+ skl_init_pci(skl);
+ skl_dum_set(bus);
+
err = skl_init_chip(bus, true);
if (err < 0) {
dev_err(bus->dev, "Init chip failed with err: %d\n", err);
@@ -922,8 +925,6 @@
return -ENXIO;
}
- snd_hdac_bus_reset_link(bus, true);
-
snd_hdac_bus_parse_capabilities(bus);
/* check if PPCAP exists */
@@ -971,11 +972,7 @@
if (err < 0)
return err;
- /* initialize chip */
- skl_init_pci(skl);
- skl_dum_set(bus);
-
- return skl_init_chip(bus, true);
+ return 0;
}
static int skl_probe(struct pci_dev *pci,
@@ -1080,8 +1077,6 @@
if (bus->mlcap)
snd_hdac_ext_bus_get_ml_capabilities(bus);
- snd_hdac_bus_stop_chip(bus);
-
/* create device for soc dmic */
err = skl_dmic_device_register(skl);
if (err < 0) {
@@ -1098,7 +1093,8 @@
out_clk_free:
skl_clock_device_unregister(skl);
out_nhlt_free:
- intel_nhlt_free(skl->nhlt);
+ if (skl->nhlt)
+ intel_nhlt_free(skl->nhlt);
out_free:
skl_free(bus);
@@ -1147,7 +1143,8 @@
skl_dmic_device_unregister(skl);
skl_clock_device_unregister(skl);
skl_nhlt_remove_sysfs(skl);
- intel_nhlt_free(skl->nhlt);
+ if (skl->nhlt)
+ intel_nhlt_free(skl->nhlt);
skl_free(bus);
dev_set_drvdata(&pci->dev, NULL);
}
diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c
index 13408de..9bfd2aa 100644
--- a/sound/soc/jz4740/jz4740-i2s.c
+++ b/sound/soc/jz4740/jz4740-i2s.c
@@ -83,7 +83,7 @@
#define JZ_AIC_I2S_STATUS_BUSY BIT(2)
#define JZ_AIC_CLK_DIV_MASK 0xf
-#define I2SDIV_DV_SHIFT 8
+#define I2SDIV_DV_SHIFT 0
#define I2SDIV_DV_MASK (0xf << I2SDIV_DV_SHIFT)
#define I2SDIV_IDV_SHIFT 8
#define I2SDIV_IDV_MASK (0xf << I2SDIV_IDV_SHIFT)
@@ -309,10 +309,14 @@
switch (clk_id) {
case JZ4740_I2S_CLKSRC_EXT:
parent = clk_get(NULL, "ext");
+ if (IS_ERR(parent))
+ return PTR_ERR(parent);
clk_set_parent(i2s->clk_i2s, parent);
break;
case JZ4740_I2S_CLKSRC_PLL:
parent = clk_get(NULL, "pll half");
+ if (IS_ERR(parent))
+ return PTR_ERR(parent);
clk_set_parent(i2s->clk_i2s, parent);
ret = clk_set_rate(i2s->clk_i2s, freq);
break;
diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c
index 6f69f31..d2d5c25 100644
--- a/sound/soc/kirkwood/kirkwood-dma.c
+++ b/sound/soc/kirkwood/kirkwood-dma.c
@@ -132,7 +132,7 @@
err = request_irq(priv->irq, kirkwood_dma_irq, IRQF_SHARED,
"kirkwood-i2s", priv);
if (err)
- return -EBUSY;
+ return err;
/*
* Enable Error interrupts. We're only ack'ing them but
diff --git a/sound/soc/mediatek/common/mtk-btcvsd.c b/sound/soc/mediatek/common/mtk-btcvsd.c
index d00608c..b66f7de 100644
--- a/sound/soc/mediatek/common/mtk-btcvsd.c
+++ b/sound/soc/mediatek/common/mtk-btcvsd.c
@@ -1302,7 +1302,7 @@
static int mtk_btcvsd_snd_probe(struct platform_device *pdev)
{
- int ret = 0;
+ int ret;
int irq_id;
u32 offset[5] = {0, 0, 0, 0, 0};
struct mtk_btcvsd_snd *btcvsd;
@@ -1358,7 +1358,8 @@
btcvsd->bt_sram_bank2_base = of_iomap(dev->of_node, 1);
if (!btcvsd->bt_sram_bank2_base) {
dev_err(dev, "iomap bt_sram_bank2_base fail\n");
- return -EIO;
+ ret = -EIO;
+ goto unmap_pkv_err;
}
btcvsd->infra = syscon_regmap_lookup_by_phandle(dev->of_node,
@@ -1366,7 +1367,8 @@
if (IS_ERR(btcvsd->infra)) {
dev_err(dev, "cannot find infra controller: %ld\n",
PTR_ERR(btcvsd->infra));
- return PTR_ERR(btcvsd->infra);
+ ret = PTR_ERR(btcvsd->infra);
+ goto unmap_bank2_err;
}
/* get offset */
@@ -1375,7 +1377,7 @@
ARRAY_SIZE(offset));
if (ret) {
dev_warn(dev, "%s(), get offset fail, ret %d\n", __func__, ret);
- return ret;
+ goto unmap_bank2_err;
}
btcvsd->infra_misc_offset = offset[0];
btcvsd->conn_bt_cvsd_mask = offset[1];
@@ -1394,8 +1396,18 @@
mtk_btcvsd_snd_set_state(btcvsd, btcvsd->tx, BT_SCO_STATE_IDLE);
mtk_btcvsd_snd_set_state(btcvsd, btcvsd->rx, BT_SCO_STATE_IDLE);
- return devm_snd_soc_register_component(dev, &mtk_btcvsd_snd_platform,
- NULL, 0);
+ ret = devm_snd_soc_register_component(dev, &mtk_btcvsd_snd_platform,
+ NULL, 0);
+ if (ret)
+ goto unmap_bank2_err;
+
+ return 0;
+
+unmap_bank2_err:
+ iounmap(btcvsd->bt_sram_bank2_base);
+unmap_pkv_err:
+ iounmap(btcvsd->bt_pkv_base);
+ return ret;
}
static int mtk_btcvsd_snd_remove(struct platform_device *pdev)
diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig
index 2e36761..e0d2459 100644
--- a/sound/soc/meson/Kconfig
+++ b/sound/soc/meson/Kconfig
@@ -1,6 +1,6 @@
# SPDX-License-Identifier: GPL-2.0-only
menu "ASoC support for Amlogic platforms"
- depends on ARCH_MESON || COMPILE_TEST
+ depends on ARCH_MESON || (COMPILE_TEST && COMMON_CLK)
config SND_MESON_AXG_FIFO
tristate
diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c
index 1f698ad..7126344 100644
--- a/sound/soc/meson/axg-card.c
+++ b/sound/soc/meson/axg-card.c
@@ -266,7 +266,7 @@
lb = &card->dai_link[*index + 1];
- lb->name = kasprintf(GFP_KERNEL, "%s-lb", pad->name);
+ lb->name = devm_kasprintf(card->dev, GFP_KERNEL, "%s-lb", pad->name);
if (!lb->name)
return -ENOMEM;
diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c
index 5a37499..898ef1d 100644
--- a/sound/soc/meson/axg-fifo.c
+++ b/sound/soc/meson/axg-fifo.c
@@ -108,10 +108,12 @@
{
struct snd_pcm_runtime *runtime = ss->runtime;
struct axg_fifo *fifo = axg_fifo_data(ss);
+ unsigned int burst_num, period, threshold;
dma_addr_t end_ptr;
- unsigned int burst_num;
int ret;
+ period = params_period_bytes(params);
+
ret = snd_pcm_lib_malloc_pages(ss, params_buffer_bytes(params));
if (ret < 0)
return ret;
@@ -122,9 +124,25 @@
regmap_write(fifo->map, FIFO_FINISH_ADDR, end_ptr);
/* Setup interrupt periodicity */
- burst_num = params_period_bytes(params) / AXG_FIFO_BURST;
+ burst_num = period / AXG_FIFO_BURST;
regmap_write(fifo->map, FIFO_INT_ADDR, burst_num);
+ /*
+ * Start the fifo request on the smallest of the following:
+ * - Half the fifo size
+ * - Half the period size
+ */
+ threshold = min(period / 2,
+ (unsigned int)AXG_FIFO_MIN_DEPTH / 2);
+
+ /*
+ * With the threshold in bytes, register value is:
+ * V = (threshold / burst) - 1
+ */
+ threshold /= AXG_FIFO_BURST;
+ regmap_field_write(fifo->field_threshold,
+ threshold ? threshold - 1 : 0);
+
/* Enable block count irq */
regmap_update_bits(fifo->map, FIFO_CTRL0,
CTRL0_INT_EN(FIFO_INT_COUNT_REPEAT),
@@ -226,7 +244,7 @@
/* Enable pclk to access registers and clock the fifo ip */
ret = clk_prepare_enable(fifo->pclk);
if (ret)
- return ret;
+ goto free_irq;
/* Setup status2 so it reports the memory pointer */
regmap_update_bits(fifo->map, FIFO_CTRL1,
@@ -246,8 +264,14 @@
/* Take memory arbitror out of reset */
ret = reset_control_deassert(fifo->arb);
if (ret)
- clk_disable_unprepare(fifo->pclk);
+ goto free_clk;
+ return 0;
+
+free_clk:
+ clk_disable_unprepare(fifo->pclk);
+free_irq:
+ free_irq(fifo->irq, ss);
return ret;
}
@@ -360,6 +384,11 @@
return fifo->irq;
}
+ fifo->field_threshold =
+ devm_regmap_field_alloc(dev, fifo->map, data->field_threshold);
+ if (IS_ERR(fifo->field_threshold))
+ return PTR_ERR(fifo->field_threshold);
+
return devm_snd_soc_register_component(dev, data->component_drv,
data->dai_drv, 1);
}
diff --git a/sound/soc/meson/axg-fifo.h b/sound/soc/meson/axg-fifo.h
index bb1e2ce..ab546a3 100644
--- a/sound/soc/meson/axg-fifo.h
+++ b/sound/soc/meson/axg-fifo.h
@@ -9,7 +9,9 @@
struct clk;
struct platform_device;
+struct reg_field;
struct regmap;
+struct regmap_field;
struct reset_control;
struct snd_soc_component_driver;
@@ -50,8 +52,6 @@
#define CTRL1_STATUS2_SEL_MASK GENMASK(11, 8)
#define CTRL1_STATUS2_SEL(x) ((x) << 8)
#define STATUS2_SEL_DDR_READ 0
-#define CTRL1_THRESHOLD_MASK GENMASK(23, 16)
-#define CTRL1_THRESHOLD(x) ((x) << 16)
#define CTRL1_FRDDR_DEPTH_MASK GENMASK(31, 24)
#define CTRL1_FRDDR_DEPTH(x) ((x) << 24)
#define FIFO_START_ADDR 0x08
@@ -67,12 +67,14 @@
struct regmap *map;
struct clk *pclk;
struct reset_control *arb;
+ struct regmap_field *field_threshold;
int irq;
};
struct axg_fifo_match_data {
const struct snd_soc_component_driver *component_drv;
struct snd_soc_dai_driver *dai_drv;
+ struct reg_field field_threshold;
};
extern const struct snd_pcm_ops axg_fifo_pcm_ops;
diff --git a/sound/soc/meson/axg-frddr.c b/sound/soc/meson/axg-frddr.c
index 6ab111c..09773a9 100644
--- a/sound/soc/meson/axg-frddr.c
+++ b/sound/soc/meson/axg-frddr.c
@@ -50,7 +50,7 @@
struct snd_soc_dai *dai)
{
struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai);
- unsigned int fifo_depth, fifo_threshold;
+ unsigned int fifo_depth;
int ret;
/* Enable pclk to access registers and clock the fifo ip */
@@ -68,11 +68,8 @@
* Depth and threshold are zero based.
*/
fifo_depth = AXG_FIFO_MIN_CNT - 1;
- fifo_threshold = (AXG_FIFO_MIN_CNT / 2) - 1;
- regmap_update_bits(fifo->map, FIFO_CTRL1,
- CTRL1_FRDDR_DEPTH_MASK | CTRL1_THRESHOLD_MASK,
- CTRL1_FRDDR_DEPTH(fifo_depth) |
- CTRL1_THRESHOLD(fifo_threshold));
+ regmap_update_bits(fifo->map, FIFO_CTRL1, CTRL1_FRDDR_DEPTH_MASK,
+ CTRL1_FRDDR_DEPTH(fifo_depth));
return 0;
}
@@ -153,8 +150,9 @@
};
static const struct axg_fifo_match_data axg_frddr_match_data = {
- .component_drv = &axg_frddr_component_drv,
- .dai_drv = &axg_frddr_dai_drv
+ .field_threshold = REG_FIELD(FIFO_CTRL1, 16, 23),
+ .component_drv = &axg_frddr_component_drv,
+ .dai_drv = &axg_frddr_dai_drv
};
static const struct snd_soc_dai_ops g12a_frddr_ops = {
@@ -271,8 +269,9 @@
};
static const struct axg_fifo_match_data g12a_frddr_match_data = {
- .component_drv = &g12a_frddr_component_drv,
- .dai_drv = &g12a_frddr_dai_drv
+ .field_threshold = REG_FIELD(FIFO_CTRL1, 16, 23),
+ .component_drv = &g12a_frddr_component_drv,
+ .dai_drv = &g12a_frddr_dai_drv
};
/* On SM1, the output selection in on CTRL2 */
@@ -335,8 +334,9 @@
};
static const struct axg_fifo_match_data sm1_frddr_match_data = {
- .component_drv = &sm1_frddr_component_drv,
- .dai_drv = &g12a_frddr_dai_drv
+ .field_threshold = REG_FIELD(FIFO_CTRL1, 16, 23),
+ .component_drv = &sm1_frddr_component_drv,
+ .dai_drv = &g12a_frddr_dai_drv
};
static const struct of_device_id axg_frddr_of_match[] = {
diff --git a/sound/soc/meson/axg-tdm-formatter.c b/sound/soc/meson/axg-tdm-formatter.c
index 358c8c0..f7e8e9d 100644
--- a/sound/soc/meson/axg-tdm-formatter.c
+++ b/sound/soc/meson/axg-tdm-formatter.c
@@ -70,7 +70,7 @@
static int axg_tdm_formatter_enable(struct axg_tdm_formatter *formatter)
{
struct axg_tdm_stream *ts = formatter->stream;
- bool invert = formatter->drv->quirks->invert_sclk;
+ bool invert;
int ret;
/* Do nothing if the formatter is already enabled */
@@ -96,11 +96,12 @@
return ret;
/*
- * If sclk is inverted, invert it back and provide the inversion
- * required by the formatter
+ * If sclk is inverted, it means the bit should latched on the
+ * rising edge which is what our HW expects. If not, we need to
+ * invert it before the formatter.
*/
- invert ^= axg_tdm_sclk_invert(ts->iface->fmt);
- ret = clk_set_phase(formatter->sclk, invert ? 180 : 0);
+ invert = axg_tdm_sclk_invert(ts->iface->fmt);
+ ret = clk_set_phase(formatter->sclk, invert ? 0 : 180);
if (ret)
return ret;
diff --git a/sound/soc/meson/axg-tdm-formatter.h b/sound/soc/meson/axg-tdm-formatter.h
index 9ef98e9..a1f0dcc 100644
--- a/sound/soc/meson/axg-tdm-formatter.h
+++ b/sound/soc/meson/axg-tdm-formatter.h
@@ -16,7 +16,6 @@
struct axg_tdm_formatter_hw {
unsigned int skew_offset;
- bool invert_sclk;
};
struct axg_tdm_formatter_ops {
diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c
index d51f334..f5a431b 100644
--- a/sound/soc/meson/axg-tdm-interface.c
+++ b/sound/soc/meson/axg-tdm-interface.c
@@ -119,16 +119,23 @@
{
struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai);
- /* These modes are not supported */
- if (fmt & (SND_SOC_DAIFMT_CBS_CFM | SND_SOC_DAIFMT_CBM_CFS)) {
- dev_err(dai->dev, "only CBS_CFS and CBM_CFM are supported\n");
- return -EINVAL;
- }
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ if (!iface->mclk) {
+ dev_err(dai->dev, "cpu clock master: mclk missing\n");
+ return -ENODEV;
+ }
+ break;
- /* If the TDM interface is the clock master, it requires mclk */
- if (!iface->mclk && (fmt & SND_SOC_DAIFMT_CBS_CFS)) {
- dev_err(dai->dev, "cpu clock master: mclk missing\n");
- return -ENODEV;
+ case SND_SOC_DAIFMT_CBM_CFM:
+ break;
+
+ case SND_SOC_DAIFMT_CBS_CFM:
+ case SND_SOC_DAIFMT_CBM_CFS:
+ dev_err(dai->dev, "only CBS_CFS and CBM_CFM are supported\n");
+ /* Fall-through */
+ default:
+ return -EINVAL;
}
iface->fmt = fmt;
@@ -319,7 +326,8 @@
if (ret)
return ret;
- if (iface->fmt & SND_SOC_DAIFMT_CBS_CFS) {
+ if ((iface->fmt & SND_SOC_DAIFMT_MASTER_MASK) ==
+ SND_SOC_DAIFMT_CBS_CFS) {
ret = axg_tdm_iface_set_sclk(dai, params);
if (ret)
return ret;
@@ -459,8 +467,20 @@
return ret;
}
+static const struct snd_soc_dapm_widget axg_tdm_iface_dapm_widgets[] = {
+ SND_SOC_DAPM_SIGGEN("Playback Signal"),
+};
+
+static const struct snd_soc_dapm_route axg_tdm_iface_dapm_routes[] = {
+ { "Loopback", NULL, "Playback Signal" },
+};
+
static const struct snd_soc_component_driver axg_tdm_iface_component_drv = {
- .set_bias_level = axg_tdm_iface_set_bias_level,
+ .dapm_widgets = axg_tdm_iface_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(axg_tdm_iface_dapm_widgets),
+ .dapm_routes = axg_tdm_iface_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(axg_tdm_iface_dapm_routes),
+ .set_bias_level = axg_tdm_iface_set_bias_level,
};
static const struct of_device_id axg_tdm_iface_of_match[] = {
diff --git a/sound/soc/meson/axg-tdmin.c b/sound/soc/meson/axg-tdmin.c
index 973d4c0..b4faf9d 100644
--- a/sound/soc/meson/axg-tdmin.c
+++ b/sound/soc/meson/axg-tdmin.c
@@ -228,8 +228,7 @@
.regmap_cfg = &axg_tdmin_regmap_cfg,
.ops = &axg_tdmin_ops,
.quirks = &(const struct axg_tdm_formatter_hw) {
- .invert_sclk = false,
- .skew_offset = 2,
+ .skew_offset = 3,
},
};
@@ -237,6 +236,12 @@
{
.compatible = "amlogic,axg-tdmin",
.data = &axg_tdmin_drv,
+ }, {
+ .compatible = "amlogic,g12a-tdmin",
+ .data = &axg_tdmin_drv,
+ }, {
+ .compatible = "amlogic,sm1-tdmin",
+ .data = &axg_tdmin_drv,
}, {}
};
MODULE_DEVICE_TABLE(of, axg_tdmin_of_match);
diff --git a/sound/soc/meson/axg-tdmout.c b/sound/soc/meson/axg-tdmout.c
index 418ec31..3ceabdd 100644
--- a/sound/soc/meson/axg-tdmout.c
+++ b/sound/soc/meson/axg-tdmout.c
@@ -238,7 +238,6 @@
.regmap_cfg = &axg_tdmout_regmap_cfg,
.ops = &axg_tdmout_ops,
.quirks = &(const struct axg_tdm_formatter_hw) {
- .invert_sclk = true,
.skew_offset = 1,
},
};
@@ -248,7 +247,6 @@
.regmap_cfg = &axg_tdmout_regmap_cfg,
.ops = &axg_tdmout_ops,
.quirks = &(const struct axg_tdm_formatter_hw) {
- .invert_sclk = true,
.skew_offset = 2,
},
};
@@ -309,7 +307,6 @@
.regmap_cfg = &axg_tdmout_regmap_cfg,
.ops = &axg_tdmout_ops,
.quirks = &(const struct axg_tdm_formatter_hw) {
- .invert_sclk = true,
.skew_offset = 2,
},
};
diff --git a/sound/soc/meson/axg-toddr.c b/sound/soc/meson/axg-toddr.c
index c8ea214..32b9fd5 100644
--- a/sound/soc/meson/axg-toddr.c
+++ b/sound/soc/meson/axg-toddr.c
@@ -18,6 +18,7 @@
#define CTRL0_TODDR_SEL_RESAMPLE BIT(30)
#define CTRL0_TODDR_EXT_SIGNED BIT(29)
#define CTRL0_TODDR_PP_MODE BIT(28)
+#define CTRL0_TODDR_SYNC_CH BIT(27)
#define CTRL0_TODDR_TYPE_MASK GENMASK(15, 13)
#define CTRL0_TODDR_TYPE(x) ((x) << 13)
#define CTRL0_TODDR_MSB_POS_MASK GENMASK(12, 8)
@@ -89,7 +90,6 @@
struct snd_soc_dai *dai)
{
struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai);
- unsigned int fifo_threshold;
int ret;
/* Enable pclk to access registers and clock the fifo ip */
@@ -107,11 +107,6 @@
/* Apply single buffer mode to the interface */
regmap_update_bits(fifo->map, FIFO_CTRL0, CTRL0_TODDR_PP_MODE, 0);
- /* TODDR does not have a configurable fifo depth */
- fifo_threshold = AXG_FIFO_MIN_CNT - 1;
- regmap_update_bits(fifo->map, FIFO_CTRL1, CTRL1_THRESHOLD_MASK,
- CTRL1_THRESHOLD(fifo_threshold));
-
return 0;
}
@@ -185,14 +180,36 @@
};
static const struct axg_fifo_match_data axg_toddr_match_data = {
- .component_drv = &axg_toddr_component_drv,
- .dai_drv = &axg_toddr_dai_drv
+ .field_threshold = REG_FIELD(FIFO_CTRL1, 16, 23),
+ .component_drv = &axg_toddr_component_drv,
+ .dai_drv = &axg_toddr_dai_drv
};
+static int g12a_toddr_dai_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai);
+ int ret;
+
+ ret = axg_toddr_dai_startup(substream, dai);
+ if (ret)
+ return ret;
+
+ /*
+ * Make sure the first channel ends up in the at beginning of the output
+ * As weird as it looks, without this the first channel may be misplaced
+ * in memory, with a random shift of 2 channels.
+ */
+ regmap_update_bits(fifo->map, FIFO_CTRL0, CTRL0_TODDR_SYNC_CH,
+ CTRL0_TODDR_SYNC_CH);
+
+ return 0;
+}
+
static const struct snd_soc_dai_ops g12a_toddr_ops = {
.prepare = g12a_toddr_dai_prepare,
.hw_params = axg_toddr_dai_hw_params,
- .startup = axg_toddr_dai_startup,
+ .startup = g12a_toddr_dai_startup,
.shutdown = axg_toddr_dai_shutdown,
};
@@ -218,8 +235,9 @@
};
static const struct axg_fifo_match_data g12a_toddr_match_data = {
- .component_drv = &g12a_toddr_component_drv,
- .dai_drv = &g12a_toddr_dai_drv
+ .field_threshold = REG_FIELD(FIFO_CTRL1, 16, 23),
+ .component_drv = &g12a_toddr_component_drv,
+ .dai_drv = &g12a_toddr_dai_drv
};
static const char * const sm1_toddr_sel_texts[] = {
@@ -282,8 +300,9 @@
};
static const struct axg_fifo_match_data sm1_toddr_match_data = {
- .component_drv = &sm1_toddr_component_drv,
- .dai_drv = &g12a_toddr_dai_drv
+ .field_threshold = REG_FIELD(FIFO_CTRL1, 12, 23),
+ .component_drv = &sm1_toddr_component_drv,
+ .dai_drv = &g12a_toddr_dai_drv
};
static const struct of_device_id axg_toddr_of_match[] = {
diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig
index 6008685..b9d8fe9 100644
--- a/sound/soc/qcom/Kconfig
+++ b/sound/soc/qcom/Kconfig
@@ -72,7 +72,7 @@
config SND_SOC_QDSP6
tristate "SoC ALSA audio driver for QDSP6"
- depends on QCOM_APR && HAS_DMA
+ depends on QCOM_APR
select SND_SOC_QDSP6_COMMON
select SND_SOC_QDSP6_CORE
select SND_SOC_QDSP6_AFE
diff --git a/sound/soc/qcom/apq8016_sbc.c b/sound/soc/qcom/apq8016_sbc.c
index ac75838..15a8802 100644
--- a/sound/soc/qcom/apq8016_sbc.c
+++ b/sound/soc/qcom/apq8016_sbc.c
@@ -235,6 +235,7 @@
return -ENOMEM;
card->dev = dev;
+ card->owner = THIS_MODULE;
card->dapm_widgets = apq8016_sbc_dapm_widgets;
card->num_dapm_widgets = ARRAY_SIZE(apq8016_sbc_dapm_widgets);
data = apq8016_sbc_parse_of(card);
diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c
index 94363fd..c10c5f2 100644
--- a/sound/soc/qcom/apq8096.c
+++ b/sound/soc/qcom/apq8096.c
@@ -114,6 +114,7 @@
return -ENOMEM;
card->dev = dev;
+ card->owner = THIS_MODULE;
dev_set_drvdata(dev, card);
ret = qcom_snd_parse_of(card);
if (ret) {
diff --git a/sound/soc/qcom/common.c b/sound/soc/qcom/common.c
index 6c20bdd..1032269 100644
--- a/sound/soc/qcom/common.c
+++ b/sound/soc/qcom/common.c
@@ -4,6 +4,7 @@
#include <linux/module.h>
#include "common.h"
+#include "qdsp6/q6afe.h"
int qcom_snd_parse_of(struct snd_soc_card *card)
{
@@ -44,8 +45,10 @@
for_each_child_of_node(dev->of_node, np) {
dlc = devm_kzalloc(dev, 2 * sizeof(*dlc), GFP_KERNEL);
- if (!dlc)
- return -ENOMEM;
+ if (!dlc) {
+ ret = -ENOMEM;
+ goto err;
+ }
link->cpus = &dlc[0];
link->platforms = &dlc[1];
@@ -101,6 +104,15 @@
}
link->no_pcm = 1;
link->ignore_pmdown_time = 1;
+
+ if (q6afe_is_rx_port(link->id)) {
+ link->dpcm_playback = 1;
+ link->dpcm_capture = 0;
+ } else {
+ link->dpcm_playback = 0;
+ link->dpcm_capture = 1;
+ }
+
} else {
dlc = devm_kzalloc(dev, sizeof(*dlc), GFP_KERNEL);
if (!dlc)
@@ -113,12 +125,12 @@
link->codecs->dai_name = "snd-soc-dummy-dai";
link->codecs->name = "snd-soc-dummy";
link->dynamic = 1;
+ link->dpcm_playback = 1;
+ link->dpcm_capture = 1;
}
link->ignore_suspend = 1;
link->nonatomic = 1;
- link->dpcm_playback = 1;
- link->dpcm_capture = 1;
link->stream_name = link->name;
link++;
diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c
index dbce7e9..c5d6952 100644
--- a/sound/soc/qcom/lpass-cpu.c
+++ b/sound/soc/qcom/lpass-cpu.c
@@ -174,21 +174,6 @@
return 0;
}
-static int lpass_cpu_daiops_hw_free(struct snd_pcm_substream *substream,
- struct snd_soc_dai *dai)
-{
- struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai);
- int ret;
-
- ret = regmap_write(drvdata->lpaif_map,
- LPAIF_I2SCTL_REG(drvdata->variant, dai->driver->id),
- 0);
- if (ret)
- dev_err(dai->dev, "error writing to i2sctl reg: %d\n", ret);
-
- return ret;
-}
-
static int lpass_cpu_daiops_prepare(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
@@ -269,7 +254,6 @@
.startup = lpass_cpu_daiops_startup,
.shutdown = lpass_cpu_daiops_shutdown,
.hw_params = lpass_cpu_daiops_hw_params,
- .hw_free = lpass_cpu_daiops_hw_free,
.prepare = lpass_cpu_daiops_prepare,
.trigger = lpass_cpu_daiops_trigger,
};
diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c
index 4c745ba..b1981d8 100644
--- a/sound/soc/qcom/lpass-platform.c
+++ b/sound/soc/qcom/lpass-platform.c
@@ -61,7 +61,7 @@
int ret, dma_ch, dir = substream->stream;
struct lpass_pcm_data *data;
- data = devm_kzalloc(soc_runtime->dev, sizeof(*data), GFP_KERNEL);
+ data = kzalloc(sizeof(*data), GFP_KERNEL);
if (!data)
return -ENOMEM;
@@ -73,8 +73,10 @@
else
dma_ch = 0;
- if (dma_ch < 0)
+ if (dma_ch < 0) {
+ kfree(data);
return dma_ch;
+ }
drvdata->substream[dma_ch] = substream;
@@ -95,6 +97,7 @@
ret = snd_pcm_hw_constraint_integer(runtime,
SNDRV_PCM_HW_PARAM_PERIODS);
if (ret < 0) {
+ kfree(data);
dev_err(soc_runtime->dev, "setting constraints failed: %d\n",
ret);
return -EINVAL;
@@ -119,6 +122,7 @@
if (v->free_dma_channel)
v->free_dma_channel(drvdata, data->dma_ch);
+ kfree(data);
return 0;
}
diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c
index c1a7624..0168af8 100644
--- a/sound/soc/qcom/qdsp6/q6afe-dai.c
+++ b/sound/soc/qcom/qdsp6/q6afe-dai.c
@@ -902,6 +902,8 @@
SNDRV_PCM_RATE_16000,
.formats = SNDRV_PCM_FMTBIT_S16_LE |
SNDRV_PCM_FMTBIT_S24_LE,
+ .channels_min = 1,
+ .channels_max = 8,
.rate_min = 8000,
.rate_max = 48000,
},
@@ -917,6 +919,8 @@
SNDRV_PCM_RATE_16000,
.formats = SNDRV_PCM_FMTBIT_S16_LE |
SNDRV_PCM_FMTBIT_S24_LE,
+ .channels_min = 1,
+ .channels_max = 8,
.rate_min = 8000,
.rate_max = 48000,
},
@@ -931,6 +935,8 @@
.rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
SNDRV_PCM_RATE_16000,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 8,
.rate_min = 8000,
.rate_max = 48000,
},
@@ -946,6 +952,8 @@
SNDRV_PCM_RATE_16000,
.formats = SNDRV_PCM_FMTBIT_S16_LE |
SNDRV_PCM_FMTBIT_S24_LE,
+ .channels_min = 1,
+ .channels_max = 8,
.rate_min = 8000,
.rate_max = 48000,
},
@@ -960,6 +968,8 @@
.rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
SNDRV_PCM_RATE_16000,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 8,
.rate_min = 8000,
.rate_max = 48000,
},
@@ -975,6 +985,8 @@
SNDRV_PCM_RATE_16000,
.formats = SNDRV_PCM_FMTBIT_S16_LE |
SNDRV_PCM_FMTBIT_S24_LE,
+ .channels_min = 1,
+ .channels_max = 8,
.rate_min = 8000,
.rate_max = 48000,
},
@@ -989,6 +1001,8 @@
.rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
SNDRV_PCM_RATE_16000,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels_min = 1,
+ .channels_max = 8,
.rate_min = 8000,
.rate_max = 48000,
},
@@ -1004,6 +1018,8 @@
SNDRV_PCM_RATE_16000,
.formats = SNDRV_PCM_FMTBIT_S16_LE |
SNDRV_PCM_FMTBIT_S24_LE,
+ .channels_min = 1,
+ .channels_max = 8,
.rate_min = 8000,
.rate_max = 48000,
},
@@ -1134,206 +1150,206 @@
}
static const struct snd_soc_dapm_widget q6afe_dai_widgets[] = {
- SND_SOC_DAPM_AIF_IN("HDMI_RX", NULL, 0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("SLIMBUS_0_RX", NULL, 0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("SLIMBUS_1_RX", NULL, 0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("SLIMBUS_2_RX", NULL, 0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("SLIMBUS_3_RX", NULL, 0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("SLIMBUS_4_RX", NULL, 0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("SLIMBUS_5_RX", NULL, 0, 0, 0, 0),
- SND_SOC_DAPM_AIF_IN("SLIMBUS_6_RX", NULL, 0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("SLIMBUS_0_TX", NULL, 0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("SLIMBUS_1_TX", NULL, 0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("SLIMBUS_2_TX", NULL, 0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("SLIMBUS_3_TX", NULL, 0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("SLIMBUS_4_TX", NULL, 0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("SLIMBUS_5_TX", NULL, 0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("SLIMBUS_6_TX", NULL, 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("HDMI_RX", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("SLIMBUS_0_RX", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("SLIMBUS_1_RX", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("SLIMBUS_2_RX", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("SLIMBUS_3_RX", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("SLIMBUS_4_RX", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("SLIMBUS_5_RX", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("SLIMBUS_6_RX", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("SLIMBUS_0_TX", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("SLIMBUS_1_TX", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("SLIMBUS_2_TX", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("SLIMBUS_3_TX", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("SLIMBUS_4_TX", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("SLIMBUS_5_TX", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("SLIMBUS_6_TX", NULL, 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("QUAT_MI2S_RX", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("QUAT_MI2S_TX", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("TERT_MI2S_RX", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("TERT_MI2S_TX", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("SEC_MI2S_RX", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("SEC_MI2S_TX", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("SEC_MI2S_RX_SD1",
"Secondary MI2S Playback SD1",
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("PRI_MI2S_RX", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("PRI_MI2S_TX", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_0", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_1", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_2", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_3", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_4", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_5", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_6", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_7", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_0", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_1", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_2", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_3", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_4", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_5", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_6", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_7", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_0", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_1", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_2", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_3", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_4", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_5", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_6", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_7", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_0", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_1", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_2", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_3", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_4", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_5", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_6", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_7", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_0", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_1", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_2", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_3", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_4", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_5", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_6", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_7", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_0", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_1", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_2", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_3", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_4", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_5", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_6", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_7", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_0", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_1", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_2", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_3", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_4", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_5", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_6", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_7", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_0", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_1", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_2", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_3", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_4", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_5", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_6", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_7", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_0", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_1", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_2", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_3", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_4", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_5", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_6", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_7", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_0", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_1", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_2", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_3", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_4", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_5", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_6", NULL,
- 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_7", NULL,
- 0, 0, 0, 0),
- SND_SOC_DAPM_AIF_OUT("DISPLAY_PORT_RX", "NULL", 0, 0, 0, 0),
+ 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("DISPLAY_PORT_RX", "NULL", 0, SND_SOC_NOPM, 0, 0),
};
static const struct snd_soc_component_driver q6afe_dai_component = {
diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c
index e0945f7..0ce4eb6 100644
--- a/sound/soc/qcom/qdsp6/q6afe.c
+++ b/sound/soc/qcom/qdsp6/q6afe.c
@@ -800,6 +800,14 @@
}
EXPORT_SYMBOL_GPL(q6afe_get_port_id);
+int q6afe_is_rx_port(int index)
+{
+ if (index < 0 || index >= AFE_PORT_MAX)
+ return -EINVAL;
+
+ return port_maps[index].is_rx;
+}
+EXPORT_SYMBOL_GPL(q6afe_is_rx_port);
static int afe_apr_send_pkt(struct q6afe *afe, struct apr_pkt *pkt,
struct q6afe_port *port)
{
diff --git a/sound/soc/qcom/qdsp6/q6afe.h b/sound/soc/qcom/qdsp6/q6afe.h
index c7ed542..1a0f80a 100644
--- a/sound/soc/qcom/qdsp6/q6afe.h
+++ b/sound/soc/qcom/qdsp6/q6afe.h
@@ -198,6 +198,7 @@
int q6afe_port_stop(struct q6afe_port *port);
void q6afe_port_put(struct q6afe_port *port);
int q6afe_get_port_id(int index);
+int q6afe_is_rx_port(int index);
void q6afe_hdmi_port_prepare(struct q6afe_port *port,
struct q6afe_hdmi_cfg *cfg);
void q6afe_slim_port_prepare(struct q6afe_port *port,
diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
index 548eb4f..9f0ffdc 100644
--- a/sound/soc/qcom/qdsp6/q6asm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
@@ -171,7 +171,7 @@
};
static void event_handler(uint32_t opcode, uint32_t token,
- uint32_t *payload, void *priv)
+ void *payload, void *priv)
{
struct q6asm_dai_rtd *prtd = priv;
struct snd_pcm_substream *substream = prtd->substream;
@@ -494,7 +494,7 @@
};
static void compress_event_handler(uint32_t opcode, uint32_t token,
- uint32_t *payload, void *priv)
+ void *payload, void *priv)
{
struct q6asm_dai_rtd *prtd = priv;
struct snd_compr_stream *substream = prtd->cstream;
diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
index e8141a3..835ac98 100644
--- a/sound/soc/qcom/qdsp6/q6asm.c
+++ b/sound/soc/qcom/qdsp6/q6asm.c
@@ -25,6 +25,7 @@
#define ASM_STREAM_CMD_FLUSH 0x00010BCE
#define ASM_SESSION_CMD_PAUSE 0x00010BD3
#define ASM_DATA_CMD_EOS 0x00010BDB
+#define ASM_DATA_EVENT_RENDERED_EOS 0x00010C1C
#define ASM_NULL_POPP_TOPOLOGY 0x00010C68
#define ASM_STREAM_CMD_FLUSH_READBUFS 0x00010C09
#define ASM_STREAM_CMD_SET_ENCDEC_PARAM 0x00010C10
@@ -546,9 +547,6 @@
case ASM_SESSION_CMD_SUSPEND:
client_event = ASM_CLIENT_EVENT_CMD_SUSPEND_DONE;
break;
- case ASM_DATA_CMD_EOS:
- client_event = ASM_CLIENT_EVENT_CMD_EOS_DONE;
- break;
case ASM_STREAM_CMD_FLUSH:
client_event = ASM_CLIENT_EVENT_CMD_FLUSH_DONE;
break;
@@ -652,6 +650,9 @@
}
break;
+ case ASM_DATA_EVENT_RENDERED_EOS:
+ client_event = ASM_CLIENT_EVENT_CMD_EOS_DONE;
+ break;
}
if (ac->cb)
diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c
index ddcd997..745cc9d 100644
--- a/sound/soc/qcom/qdsp6/q6routing.c
+++ b/sound/soc/qcom/qdsp6/q6routing.c
@@ -996,6 +996,20 @@
return 0;
}
+static unsigned int q6routing_reg_read(struct snd_soc_component *component,
+ unsigned int reg)
+{
+ /* default value */
+ return 0;
+}
+
+static int q6routing_reg_write(struct snd_soc_component *component,
+ unsigned int reg, unsigned int val)
+{
+ /* dummy */
+ return 0;
+}
+
static const struct snd_soc_component_driver msm_soc_routing_component = {
.ops = &q6pcm_routing_ops,
.probe = msm_routing_probe,
@@ -1004,6 +1018,8 @@
.num_dapm_widgets = ARRAY_SIZE(msm_qdsp6_widgets),
.dapm_routes = intercon,
.num_dapm_routes = ARRAY_SIZE(intercon),
+ .read = q6routing_reg_read,
+ .write = q6routing_reg_write,
};
static int q6pcm_routing_probe(struct platform_device *pdev)
diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c
index 28f3cef..23e1de6 100644
--- a/sound/soc/qcom/sdm845.c
+++ b/sound/soc/qcom/sdm845.c
@@ -16,6 +16,7 @@
#include "qdsp6/q6afe.h"
#include "../codecs/rt5663.h"
+#define DRIVER_NAME "sdm845"
#define DEFAULT_SAMPLE_RATE_48K 48000
#define DEFAULT_MCLK_RATE 24576000
#define TDM_BCLK_RATE 6144000
@@ -407,9 +408,11 @@
goto data_alloc_fail;
}
+ card->driver_name = DRIVER_NAME;
card->dapm_widgets = sdm845_snd_widgets;
card->num_dapm_widgets = ARRAY_SIZE(sdm845_snd_widgets);
card->dev = dev;
+ card->owner = THIS_MODULE;
dev_set_drvdata(dev, card);
ret = qcom_snd_parse_of(card);
if (ret) {
diff --git a/sound/soc/qcom/storm.c b/sound/soc/qcom/storm.c
index e6666e5..2367591 100644
--- a/sound/soc/qcom/storm.c
+++ b/sound/soc/qcom/storm.c
@@ -96,6 +96,7 @@
return -ENOMEM;
card->dev = &pdev->dev;
+ card->owner = THIS_MODULE;
ret = snd_soc_of_parse_card_name(card, "qcom,model");
if (ret) {
diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c
index 61c984f..086c90e 100644
--- a/sound/soc/rockchip/rockchip_i2s.c
+++ b/sound/soc/rockchip/rockchip_i2s.c
@@ -186,7 +186,9 @@
{
struct rk_i2s_dev *i2s = to_info(cpu_dai);
unsigned int mask = 0, val = 0;
+ int ret = 0;
+ pm_runtime_get_sync(cpu_dai->dev);
mask = I2S_CKR_MSS_MASK;
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBS_CFS:
@@ -199,7 +201,8 @@
i2s->is_master_mode = false;
break;
default:
- return -EINVAL;
+ ret = -EINVAL;
+ goto err_pm_put;
}
regmap_update_bits(i2s->regmap, I2S_CKR, mask, val);
@@ -213,7 +216,8 @@
val = I2S_CKR_CKP_POS;
break;
default:
- return -EINVAL;
+ ret = -EINVAL;
+ goto err_pm_put;
}
regmap_update_bits(i2s->regmap, I2S_CKR, mask, val);
@@ -229,14 +233,15 @@
case SND_SOC_DAIFMT_I2S:
val = I2S_TXCR_IBM_NORMAL;
break;
- case SND_SOC_DAIFMT_DSP_A: /* PCM no delay mode */
- val = I2S_TXCR_TFS_PCM;
- break;
- case SND_SOC_DAIFMT_DSP_B: /* PCM delay 1 mode */
+ case SND_SOC_DAIFMT_DSP_A: /* PCM delay 1 bit mode */
val = I2S_TXCR_TFS_PCM | I2S_TXCR_PBM_MODE(1);
break;
+ case SND_SOC_DAIFMT_DSP_B: /* PCM no delay mode */
+ val = I2S_TXCR_TFS_PCM;
+ break;
default:
- return -EINVAL;
+ ret = -EINVAL;
+ goto err_pm_put;
}
regmap_update_bits(i2s->regmap, I2S_TXCR, mask, val);
@@ -252,19 +257,23 @@
case SND_SOC_DAIFMT_I2S:
val = I2S_RXCR_IBM_NORMAL;
break;
- case SND_SOC_DAIFMT_DSP_A: /* PCM no delay mode */
- val = I2S_RXCR_TFS_PCM;
- break;
- case SND_SOC_DAIFMT_DSP_B: /* PCM delay 1 mode */
+ case SND_SOC_DAIFMT_DSP_A: /* PCM delay 1 bit mode */
val = I2S_RXCR_TFS_PCM | I2S_RXCR_PBM_MODE(1);
break;
+ case SND_SOC_DAIFMT_DSP_B: /* PCM no delay mode */
+ val = I2S_RXCR_TFS_PCM;
+ break;
default:
- return -EINVAL;
+ ret = -EINVAL;
+ goto err_pm_put;
}
regmap_update_bits(i2s->regmap, I2S_RXCR, mask, val);
- return 0;
+err_pm_put:
+ pm_runtime_put(cpu_dai->dev);
+
+ return ret;
}
static int rockchip_i2s_hw_params(struct snd_pcm_substream *substream,
diff --git a/sound/soc/rockchip/rockchip_pdm.c b/sound/soc/rockchip/rockchip_pdm.c
index 7cd42fc..1707414 100644
--- a/sound/soc/rockchip/rockchip_pdm.c
+++ b/sound/soc/rockchip/rockchip_pdm.c
@@ -590,8 +590,10 @@
int ret;
ret = pm_runtime_get_sync(dev);
- if (ret < 0)
+ if (ret < 0) {
+ pm_runtime_put(dev);
return ret;
+ }
ret = regcache_sync(pdm->regmap);
diff --git a/sound/soc/samsung/tm2_wm5110.c b/sound/soc/samsung/tm2_wm5110.c
index bb9910d..0588470 100644
--- a/sound/soc/samsung/tm2_wm5110.c
+++ b/sound/soc/samsung/tm2_wm5110.c
@@ -553,7 +553,7 @@
ret = of_parse_phandle_with_args(dev->of_node, "i2s-controller",
cells_name, i, &args);
- if (!args.np) {
+ if (ret) {
dev_err(dev, "i2s-controller property parse error: %d\n", i);
ret = -EINVAL;
goto dai_node_put;
diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c
index b9aacf3..7532ab2 100644
--- a/sound/soc/sh/rcar/adg.c
+++ b/sound/soc/sh/rcar/adg.c
@@ -289,7 +289,6 @@
int rsnd_adg_clk_query(struct rsnd_priv *priv, unsigned int rate)
{
struct rsnd_adg *adg = rsnd_priv_to_adg(priv);
- struct clk *clk;
int i;
int sel_table[] = {
[CLKA] = 0x1,
@@ -302,10 +301,9 @@
* find suitable clock from
* AUDIO_CLKA/AUDIO_CLKB/AUDIO_CLKC/AUDIO_CLKI.
*/
- for_each_rsnd_clk(clk, adg, i) {
+ for (i = 0; i < CLKMAX; i++)
if (rate == adg->clk_rate[i])
return sel_table[i];
- }
/*
* find divided clock from BRGA/BRGB
diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c
index e9596c2..df8d7b5 100644
--- a/sound/soc/sh/rcar/core.c
+++ b/sound/soc/sh/rcar/core.c
@@ -376,6 +376,17 @@
*/
u32 rsnd_get_dalign(struct rsnd_mod *mod, struct rsnd_dai_stream *io)
{
+ static const u32 dalign_values[8][2] = {
+ {0x76543210, 0x67452301},
+ {0x00000032, 0x00000023},
+ {0x00007654, 0x00006745},
+ {0x00000076, 0x00000067},
+ {0xfedcba98, 0xefcdab89},
+ {0x000000ba, 0x000000ab},
+ {0x0000fedc, 0x0000efcd},
+ {0x000000fe, 0x000000ef},
+ };
+ int id = 0, inv;
struct rsnd_mod *ssiu = rsnd_io_to_mod_ssiu(io);
struct rsnd_mod *target;
struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io);
@@ -411,13 +422,18 @@
target = cmd ? cmd : ssiu;
}
+ if (mod == ssiu)
+ id = rsnd_mod_id_sub(mod);
+
/* Non target mod or non 16bit needs normal DALIGN */
if ((snd_pcm_format_width(runtime->format) != 16) ||
(mod != target))
- return 0x76543210;
+ inv = 0;
/* Target mod needs inverted DALIGN when 16bit */
else
- return 0x67452301;
+ inv = 1;
+
+ return dalign_values[id][inv];
}
u32 rsnd_get_busif_shift(struct rsnd_dai_stream *io, struct rsnd_mod *mod)
@@ -1410,8 +1426,75 @@
}
if (io->converted_chan)
dev_dbg(dev, "convert channels = %d\n", io->converted_chan);
- if (io->converted_rate)
+ if (io->converted_rate) {
+ /*
+ * SRC supports convert rates from params_rate(hw_params)/k_down
+ * to params_rate(hw_params)*k_up, where k_up is always 6, and
+ * k_down depends on number of channels and SRC unit.
+ * So all SRC units can upsample audio up to 6 times regardless
+ * its number of channels. And all SRC units can downsample
+ * 2 channel audio up to 6 times too.
+ */
+ int k_up = 6;
+ int k_down = 6;
+ int channel;
+ struct rsnd_mod *src_mod = rsnd_io_to_mod_src(io);
+
dev_dbg(dev, "convert rate = %d\n", io->converted_rate);
+
+ channel = io->converted_chan ? io->converted_chan :
+ params_channels(hw_params);
+
+ switch (rsnd_mod_id(src_mod)) {
+ /*
+ * SRC0 can downsample 4, 6 and 8 channel audio up to 4 times.
+ * SRC1, SRC3 and SRC4 can downsample 4 channel audio
+ * up to 4 times.
+ * SRC1, SRC3 and SRC4 can downsample 6 and 8 channel audio
+ * no more than twice.
+ */
+ case 1:
+ case 3:
+ case 4:
+ if (channel > 4) {
+ k_down = 2;
+ break;
+ }
+ fallthrough;
+ case 0:
+ if (channel > 2)
+ k_down = 4;
+ break;
+
+ /* Other SRC units do not support more than 2 channels */
+ default:
+ if (channel > 2)
+ return -EINVAL;
+ }
+
+ if (params_rate(hw_params) > io->converted_rate * k_down) {
+ hw_param_interval(hw_params, SNDRV_PCM_HW_PARAM_RATE)->min =
+ io->converted_rate * k_down;
+ hw_param_interval(hw_params, SNDRV_PCM_HW_PARAM_RATE)->max =
+ io->converted_rate * k_down;
+ hw_params->cmask |= SNDRV_PCM_HW_PARAM_RATE;
+ } else if (params_rate(hw_params) * k_up < io->converted_rate) {
+ hw_param_interval(hw_params, SNDRV_PCM_HW_PARAM_RATE)->min =
+ (io->converted_rate + k_up - 1) / k_up;
+ hw_param_interval(hw_params, SNDRV_PCM_HW_PARAM_RATE)->max =
+ (io->converted_rate + k_up - 1) / k_up;
+ hw_params->cmask |= SNDRV_PCM_HW_PARAM_RATE;
+ }
+
+ /*
+ * TBD: Max SRC input and output rates also depend on number
+ * of channels and SRC unit:
+ * SRC1, SRC3 and SRC4 do not support more than 128kHz
+ * for 6 channel and 96kHz for 8 channel audio.
+ * Perhaps this function should return EINVAL if the input or
+ * the output rate exceeds the limitation.
+ */
+ }
}
ret = rsnd_dai_call(hw_params, io, substream, hw_params);
diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c
index af19010..8bd49c8 100644
--- a/sound/soc/sh/rcar/gen.c
+++ b/sound/soc/sh/rcar/gen.c
@@ -224,6 +224,14 @@
RSND_GEN_S_REG(SSI_SYS_STATUS5, 0x884),
RSND_GEN_S_REG(SSI_SYS_STATUS6, 0x888),
RSND_GEN_S_REG(SSI_SYS_STATUS7, 0x88c),
+ RSND_GEN_S_REG(SSI_SYS_INT_ENABLE0, 0x850),
+ RSND_GEN_S_REG(SSI_SYS_INT_ENABLE1, 0x854),
+ RSND_GEN_S_REG(SSI_SYS_INT_ENABLE2, 0x858),
+ RSND_GEN_S_REG(SSI_SYS_INT_ENABLE3, 0x85c),
+ RSND_GEN_S_REG(SSI_SYS_INT_ENABLE4, 0x890),
+ RSND_GEN_S_REG(SSI_SYS_INT_ENABLE5, 0x894),
+ RSND_GEN_S_REG(SSI_SYS_INT_ENABLE6, 0x898),
+ RSND_GEN_S_REG(SSI_SYS_INT_ENABLE7, 0x89c),
RSND_GEN_S_REG(HDMI0_SEL, 0x9e0),
RSND_GEN_S_REG(HDMI1_SEL, 0x9e4),
diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h
index ea6cbaa..d47608f 100644
--- a/sound/soc/sh/rcar/rsnd.h
+++ b/sound/soc/sh/rcar/rsnd.h
@@ -189,6 +189,14 @@
SSI_SYS_STATUS5,
SSI_SYS_STATUS6,
SSI_SYS_STATUS7,
+ SSI_SYS_INT_ENABLE0,
+ SSI_SYS_INT_ENABLE1,
+ SSI_SYS_INT_ENABLE2,
+ SSI_SYS_INT_ENABLE3,
+ SSI_SYS_INT_ENABLE4,
+ SSI_SYS_INT_ENABLE5,
+ SSI_SYS_INT_ENABLE6,
+ SSI_SYS_INT_ENABLE7,
HDMI0_SEL,
HDMI1_SEL,
SSI9_BUSIF0_MODE,
@@ -237,6 +245,7 @@
#define SSI9_BUSIF_ADINR(i) (SSI9_BUSIF0_ADINR + (i))
#define SSI9_BUSIF_DALIGN(i) (SSI9_BUSIF0_DALIGN + (i))
#define SSI_SYS_STATUS(i) (SSI_SYS_STATUS0 + (i))
+#define SSI_SYS_INT_ENABLE(i) (SSI_SYS_INT_ENABLE0 + (i))
struct rsnd_priv;
diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index fc5d089..09af402 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -372,6 +372,9 @@
u32 wsr = ssi->wsr;
int width;
int is_tdm, is_tdm_split;
+ int id = rsnd_mod_id(mod);
+ int i;
+ u32 sys_int_enable = 0;
is_tdm = rsnd_runtime_is_tdm(io);
is_tdm_split = rsnd_runtime_is_tdm_split(io);
@@ -447,6 +450,38 @@
cr_mode = DIEN; /* PIO : enable Data interrupt */
}
+ /* enable busif buffer over/under run interrupt. */
+ if (is_tdm || is_tdm_split) {
+ switch (id) {
+ case 0:
+ case 1:
+ case 2:
+ case 3:
+ case 4:
+ for (i = 0; i < 4; i++) {
+ sys_int_enable = rsnd_mod_read(mod,
+ SSI_SYS_INT_ENABLE(i * 2));
+ sys_int_enable |= 0xf << (id * 4);
+ rsnd_mod_write(mod,
+ SSI_SYS_INT_ENABLE(i * 2),
+ sys_int_enable);
+ }
+
+ break;
+ case 9:
+ for (i = 0; i < 4; i++) {
+ sys_int_enable = rsnd_mod_read(mod,
+ SSI_SYS_INT_ENABLE((i * 2) + 1));
+ sys_int_enable |= 0xf << 4;
+ rsnd_mod_write(mod,
+ SSI_SYS_INT_ENABLE((i * 2) + 1),
+ sys_int_enable);
+ }
+
+ break;
+ }
+ }
+
init_end:
ssi->cr_own = cr_own;
ssi->cr_mode = cr_mode;
@@ -472,10 +507,15 @@
struct rsnd_priv *priv)
{
struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
+ int ret;
if (!rsnd_ssi_is_run_mods(mod, io))
return 0;
+ ret = rsnd_ssi_master_clk_start(mod, io);
+ if (ret < 0)
+ return ret;
+
ssi->usrcnt++;
rsnd_mod_power_on(mod);
@@ -496,6 +536,13 @@
{
struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod);
struct device *dev = rsnd_priv_to_dev(priv);
+ int is_tdm, is_tdm_split;
+ int id = rsnd_mod_id(mod);
+ int i;
+ u32 sys_int_enable = 0;
+
+ is_tdm = rsnd_runtime_is_tdm(io);
+ is_tdm_split = rsnd_runtime_is_tdm_split(io);
if (!rsnd_ssi_is_run_mods(mod, io))
return 0;
@@ -517,6 +564,38 @@
ssi->wsr = 0;
}
+ /* disable busif buffer over/under run interrupt. */
+ if (is_tdm || is_tdm_split) {
+ switch (id) {
+ case 0:
+ case 1:
+ case 2:
+ case 3:
+ case 4:
+ for (i = 0; i < 4; i++) {
+ sys_int_enable = rsnd_mod_read(mod,
+ SSI_SYS_INT_ENABLE(i * 2));
+ sys_int_enable &= ~(0xf << (id * 4));
+ rsnd_mod_write(mod,
+ SSI_SYS_INT_ENABLE(i * 2),
+ sys_int_enable);
+ }
+
+ break;
+ case 9:
+ for (i = 0; i < 4; i++) {
+ sys_int_enable = rsnd_mod_read(mod,
+ SSI_SYS_INT_ENABLE((i * 2) + 1));
+ sys_int_enable &= ~(0xf << 4);
+ rsnd_mod_write(mod,
+ SSI_SYS_INT_ENABLE((i * 2) + 1),
+ sys_int_enable);
+ }
+
+ break;
+ }
+ }
+
return 0;
}
@@ -594,10 +673,16 @@
* Capture: It might not receave data. Do nothing
*/
if (rsnd_io_is_play(io)) {
- rsnd_mod_write(mod, SSICR, cr | EN);
+ rsnd_mod_write(mod, SSICR, cr | ssi->cr_en);
rsnd_ssi_status_check(mod, DIRQ);
}
+ /* In multi-SSI mode, stop is performed by setting ssi0129 in
+ * SSI_CONTROL to 0 (in rsnd_ssio_stop_gen2). Do nothing here.
+ */
+ if (rsnd_ssi_multi_slaves_runtime(io))
+ return 0;
+
/*
* disable SSI,
* and, wait idle state
@@ -616,6 +701,11 @@
int enable)
{
u32 val = 0;
+ int is_tdm, is_tdm_split;
+ int id = rsnd_mod_id(mod);
+
+ is_tdm = rsnd_runtime_is_tdm(io);
+ is_tdm_split = rsnd_runtime_is_tdm_split(io);
if (rsnd_is_gen1(priv))
return 0;
@@ -629,6 +719,19 @@
if (enable)
val = rsnd_ssi_is_dma_mode(mod) ? 0x0e000000 : 0x0f000000;
+ if (is_tdm || is_tdm_split) {
+ switch (id) {
+ case 0:
+ case 1:
+ case 2:
+ case 3:
+ case 4:
+ case 9:
+ val |= 0x0000ff00;
+ break;
+ }
+ }
+
rsnd_mod_write(mod, SSI_INT_ENABLE, val);
return 0;
@@ -645,6 +748,12 @@
u32 status;
bool elapsed = false;
bool stop = false;
+ int id = rsnd_mod_id(mod);
+ int i;
+ int is_tdm, is_tdm_split;
+
+ is_tdm = rsnd_runtime_is_tdm(io);
+ is_tdm_split = rsnd_runtime_is_tdm_split(io);
spin_lock(&priv->lock);
@@ -666,6 +775,51 @@
stop = true;
}
+ status = 0;
+
+ if (is_tdm || is_tdm_split) {
+ switch (id) {
+ case 0:
+ case 1:
+ case 2:
+ case 3:
+ case 4:
+ for (i = 0; i < 4; i++) {
+ status = rsnd_mod_read(mod,
+ SSI_SYS_STATUS(i * 2));
+ status &= 0xf << (id * 4);
+
+ if (status) {
+ rsnd_dbg_irq_status(dev,
+ "%s err status : 0x%08x\n",
+ rsnd_mod_name(mod), status);
+ rsnd_mod_write(mod,
+ SSI_SYS_STATUS(i * 2),
+ 0xf << (id * 4));
+ stop = true;
+ }
+ }
+ break;
+ case 9:
+ for (i = 0; i < 4; i++) {
+ status = rsnd_mod_read(mod,
+ SSI_SYS_STATUS((i * 2) + 1));
+ status &= 0xf << 4;
+
+ if (status) {
+ rsnd_dbg_irq_status(dev,
+ "%s err status : 0x%08x\n",
+ rsnd_mod_name(mod), status);
+ rsnd_mod_write(mod,
+ SSI_SYS_STATUS((i * 2) + 1),
+ 0xf << 4);
+ stop = true;
+ }
+ }
+ break;
+ }
+ }
+
rsnd_ssi_status_clear(mod);
rsnd_ssi_interrupt_out:
spin_unlock(&priv->lock);
@@ -737,6 +891,9 @@
if (!rsnd_rdai_is_clk_master(rdai))
return;
+ if (rsnd_ssi_is_multi_slave(mod, io))
+ return;
+
switch (rsnd_mod_id(mod)) {
case 1:
case 2:
@@ -906,13 +1063,6 @@
return 0;
}
-static int rsnd_ssi_prepare(struct rsnd_mod *mod,
- struct rsnd_dai_stream *io,
- struct rsnd_priv *priv)
-{
- return rsnd_ssi_master_clk_start(mod, io);
-}
-
static struct rsnd_mod_ops rsnd_ssi_pio_ops = {
.name = SSI_NAME,
.probe = rsnd_ssi_common_probe,
@@ -925,7 +1075,6 @@
.pointer = rsnd_ssi_pio_pointer,
.pcm_new = rsnd_ssi_pcm_new,
.hw_params = rsnd_ssi_hw_params,
- .prepare = rsnd_ssi_prepare,
.get_status = rsnd_ssi_get_status,
};
@@ -1012,7 +1161,6 @@
.pcm_new = rsnd_ssi_pcm_new,
.fallback = rsnd_ssi_fallback,
.hw_params = rsnd_ssi_hw_params,
- .prepare = rsnd_ssi_prepare,
.get_status = rsnd_ssi_get_status,
};
diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c
index f35d882..9c7c3e7 100644
--- a/sound/soc/sh/rcar/ssiu.c
+++ b/sound/soc/sh/rcar/ssiu.c
@@ -221,7 +221,7 @@
i;
for_each_rsnd_mod_array(i, pos, io, rsnd_ssi_array) {
- shift = (i * 4) + 16;
+ shift = (i * 4) + 20;
val = (val & ~(0xF << shift)) |
rsnd_mod_id(pos) << shift;
}
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 88978a3..c0e03cc 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1076,8 +1076,18 @@
ret = snd_soc_dapm_add_routes(dapm,
component->driver->dapm_routes,
component->driver->num_dapm_routes);
- if (ret < 0)
- goto err_probe;
+ if (ret < 0) {
+ if (card->disable_route_checks) {
+ dev_info(card->dev,
+ "%s: disable_route_checks set, ignoring errors on add_routes\n",
+ __func__);
+ } else {
+ dev_err(card->dev,
+ "%s: snd_soc_dapm_add_routes failed: %d\n",
+ __func__, ret);
+ goto err_probe;
+ }
+ }
/* see for_each_card_components */
list_add(&component->card_list, &card->component_dev_list);
@@ -1885,7 +1895,25 @@
dai_link->platforms->name = component->name;
/* convert non BE into BE */
- dai_link->no_pcm = 1;
+ if (!dai_link->no_pcm) {
+ dai_link->no_pcm = 1;
+
+ if (dai_link->dpcm_playback)
+ dev_warn(card->dev,
+ "invalid configuration, dailink %s has flags no_pcm=0 and dpcm_playback=1\n",
+ dai_link->name);
+ if (dai_link->dpcm_capture)
+ dev_warn(card->dev,
+ "invalid configuration, dailink %s has flags no_pcm=0 and dpcm_capture=1\n",
+ dai_link->name);
+
+ /* convert normal link into DPCM one */
+ if (!(dai_link->dpcm_playback ||
+ dai_link->dpcm_capture)) {
+ dai_link->dpcm_playback = !dai_link->capture_only;
+ dai_link->dpcm_capture = !dai_link->playback_only;
+ }
+ }
/* override any BE fixups */
dai_link->be_hw_params_fixup =
@@ -2065,8 +2093,18 @@
ret = snd_soc_dapm_add_routes(&card->dapm, card->dapm_routes,
card->num_dapm_routes);
- if (ret < 0)
- goto probe_end;
+ if (ret < 0) {
+ if (card->disable_route_checks) {
+ dev_info(card->dev,
+ "%s: disable_route_checks set, ignoring errors on add_routes\n",
+ __func__);
+ } else {
+ dev_err(card->dev,
+ "%s: snd_soc_dapm_add_routes failed: %d\n",
+ __func__, ret);
+ goto probe_end;
+ }
+ }
ret = snd_soc_dapm_add_routes(&card->dapm, card->of_dapm_routes,
card->num_of_dapm_routes);
@@ -3140,7 +3178,7 @@
if (!routes) {
dev_err(card->dev,
"ASoC: Could not allocate DAPM route table\n");
- return -EINVAL;
+ return -ENOMEM;
}
for (i = 0; i < num_routes; i++) {
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index b6378f0..7c4d596 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -423,7 +423,7 @@
memset(&template, 0, sizeof(template));
template.reg = e->reg;
- template.mask = e->mask << e->shift_l;
+ template.mask = e->mask;
template.shift = e->shift_l;
template.off_val = snd_soc_enum_item_to_val(e, 0);
template.on_val = template.off_val;
@@ -546,8 +546,22 @@
if (data->value == value)
return false;
- if (data->widget)
- data->widget->on_val = value;
+ if (data->widget) {
+ switch (dapm_kcontrol_get_wlist(kcontrol)->widgets[0]->id) {
+ case snd_soc_dapm_switch:
+ case snd_soc_dapm_mixer:
+ case snd_soc_dapm_mixer_named_ctl:
+ data->widget->on_val = value & data->widget->mask;
+ break;
+ case snd_soc_dapm_demux:
+ case snd_soc_dapm_mux:
+ data->widget->on_val = value >> data->widget->shift;
+ break;
+ default:
+ data->widget->on_val = value;
+ break;
+ }
+ }
data->value = value;
@@ -802,7 +816,13 @@
val = max - val;
p->connect = !!val;
} else {
- p->connect = 0;
+ /* since a virtual mixer has no backing registers to
+ * decide which path to connect, it will try to match
+ * with initial state. This is to ensure
+ * that the default mixer choice will be
+ * correctly powered up during initialization.
+ */
+ p->connect = invert;
}
}
@@ -2464,6 +2484,7 @@
enum snd_soc_dapm_direction dir;
list_del(&w->list);
+ list_del(&w->dirty);
/*
* remove source and sink paths associated to this widget.
* While removing the path, remove reference to it from both
@@ -3888,9 +3909,6 @@
runtime->rate = params_rate(params);
out:
- if (ret < 0)
- kfree(runtime);
-
kfree(params);
return ret;
}
@@ -4752,7 +4770,7 @@
continue;
if (w->power) {
dapm_seq_insert(w, &down_list, false);
- w->power = 0;
+ w->new_power = 0;
powerdown = 1;
}
}
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index a71d234..b5748dc 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -82,10 +82,9 @@
unsigned int sync = 0;
int enable;
- trace_snd_soc_jack_report(jack, mask, status);
-
if (!jack)
return;
+ trace_snd_soc_jack_report(jack, mask, status);
dapm = &jack->card->dapm;
diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c
index f4dc3d4..95fc245 100644
--- a/sound/soc/soc-ops.c
+++ b/sound/soc/soc-ops.c
@@ -832,7 +832,7 @@
unsigned int regbase = mc->regbase;
unsigned int regcount = mc->regcount;
unsigned int regwshift = component->val_bytes * BITS_PER_BYTE;
- unsigned int regwmask = (1<<regwshift)-1;
+ unsigned int regwmask = (1UL<<regwshift)-1;
unsigned int invert = mc->invert;
unsigned long mask = (1UL<<mc->nbits)-1;
long min = mc->min;
@@ -881,7 +881,7 @@
unsigned int regbase = mc->regbase;
unsigned int regcount = mc->regcount;
unsigned int regwshift = component->val_bytes * BITS_PER_BYTE;
- unsigned int regwmask = (1<<regwshift)-1;
+ unsigned int regwmask = (1UL<<regwshift)-1;
unsigned int invert = mc->invert;
unsigned long mask = (1UL<<mc->nbits)-1;
long max = mc->max;
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index b600d3e..1196167 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -877,6 +877,11 @@
int i, ret = 0;
mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass);
+
+ ret = soc_pcm_params_symmetry(substream, params);
+ if (ret)
+ goto out;
+
if (rtd->dai_link->ops->hw_params) {
ret = rtd->dai_link->ops->hw_params(substream, params);
if (ret < 0) {
@@ -958,9 +963,6 @@
}
component = NULL;
- ret = soc_pcm_params_symmetry(substream, params);
- if (ret)
- goto component_err;
out:
mutex_unlock(&rtd->card->pcm_mutex);
return ret;
@@ -1146,7 +1148,9 @@
{
struct snd_soc_dpcm *dpcm;
unsigned long flags;
+#ifdef CONFIG_DEBUG_FS
char *name;
+#endif
/* only add new dpcms */
for_each_dpcm_be(fe, stream, dpcm) {
@@ -1385,6 +1389,7 @@
struct snd_soc_dapm_widget *widget;
struct snd_soc_dai *dai;
int prune = 0;
+ int do_prune;
/* Destroy any old FE <--> BE connections */
for_each_dpcm_be(fe, stream, dpcm) {
@@ -1398,13 +1403,16 @@
continue;
/* is there a valid CODEC DAI widget for this BE */
+ do_prune = 1;
for_each_rtd_codec_dai(dpcm->be, i, dai) {
widget = dai_get_widget(dai, stream);
/* prune the BE if it's no longer in our active list */
if (widget && widget_in_list(list, widget))
- continue;
+ do_prune = 0;
}
+ if (!do_prune)
+ continue;
dev_dbg(fe->dev, "ASoC: pruning %s BE %s for %s\n",
stream ? "capture" : "playback",
@@ -2214,7 +2222,8 @@
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) &&
- (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP))
+ (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) &&
+ (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED))
continue;
ret = dpcm_do_trigger(dpcm, be_substream, cmd);
@@ -2244,7 +2253,8 @@
be->dpcm[stream].state = SND_SOC_DPCM_STATE_START;
break;
case SNDRV_PCM_TRIGGER_STOP:
- if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START)
+ if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) &&
+ (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED))
continue;
if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream))
@@ -2289,42 +2299,83 @@
}
EXPORT_SYMBOL_GPL(dpcm_be_dai_trigger);
+static int dpcm_dai_trigger_fe_be(struct snd_pcm_substream *substream,
+ int cmd, bool fe_first)
+{
+ struct snd_soc_pcm_runtime *fe = substream->private_data;
+ int ret;
+
+ /* call trigger on the frontend before the backend. */
+ if (fe_first) {
+ dev_dbg(fe->dev, "ASoC: pre trigger FE %s cmd %d\n",
+ fe->dai_link->name, cmd);
+
+ ret = soc_pcm_trigger(substream, cmd);
+ if (ret < 0)
+ return ret;
+
+ ret = dpcm_be_dai_trigger(fe, substream->stream, cmd);
+ return ret;
+ }
+
+ /* call trigger on the frontend after the backend. */
+ ret = dpcm_be_dai_trigger(fe, substream->stream, cmd);
+ if (ret < 0)
+ return ret;
+
+ dev_dbg(fe->dev, "ASoC: post trigger FE %s cmd %d\n",
+ fe->dai_link->name, cmd);
+
+ ret = soc_pcm_trigger(substream, cmd);
+
+ return ret;
+}
+
static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd)
{
struct snd_soc_pcm_runtime *fe = substream->private_data;
- int stream = substream->stream, ret;
+ int stream = substream->stream;
+ int ret = 0;
enum snd_soc_dpcm_trigger trigger = fe->dai_link->trigger[stream];
fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE;
switch (trigger) {
case SND_SOC_DPCM_TRIGGER_PRE:
- /* call trigger on the frontend before the backend. */
-
- dev_dbg(fe->dev, "ASoC: pre trigger FE %s cmd %d\n",
- fe->dai_link->name, cmd);
-
- ret = soc_pcm_trigger(substream, cmd);
- if (ret < 0) {
- dev_err(fe->dev,"ASoC: trigger FE failed %d\n", ret);
- goto out;
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ case SNDRV_PCM_TRIGGER_DRAIN:
+ ret = dpcm_dai_trigger_fe_be(substream, cmd, true);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ ret = dpcm_dai_trigger_fe_be(substream, cmd, false);
+ break;
+ default:
+ ret = -EINVAL;
+ break;
}
-
- ret = dpcm_be_dai_trigger(fe, substream->stream, cmd);
break;
case SND_SOC_DPCM_TRIGGER_POST:
- /* call trigger on the frontend after the backend. */
-
- ret = dpcm_be_dai_trigger(fe, substream->stream, cmd);
- if (ret < 0) {
- dev_err(fe->dev,"ASoC: trigger FE failed %d\n", ret);
- goto out;
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ case SNDRV_PCM_TRIGGER_DRAIN:
+ ret = dpcm_dai_trigger_fe_be(substream, cmd, false);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ ret = dpcm_dai_trigger_fe_be(substream, cmd, true);
+ break;
+ default:
+ ret = -EINVAL;
+ break;
}
-
- dev_dbg(fe->dev, "ASoC: post trigger FE %s cmd %d\n",
- fe->dai_link->name, cmd);
-
- ret = soc_pcm_trigger(substream, cmd);
break;
case SND_SOC_DPCM_TRIGGER_BESPOKE:
/* bespoke trigger() - handles both FE and BEs */
@@ -2333,10 +2384,6 @@
fe->dai_link->name, cmd);
ret = soc_pcm_bespoke_trigger(substream, cmd);
- if (ret < 0) {
- dev_err(fe->dev,"ASoC: trigger FE failed %d\n", ret);
- goto out;
- }
break;
default:
dev_err(fe->dev, "ASoC: invalid trigger cmd %d for %s\n", cmd,
@@ -2345,6 +2392,12 @@
goto out;
}
+ if (ret < 0) {
+ dev_err(fe->dev, "ASoC: trigger FE cmd: %d failed: %d\n",
+ cmd, ret);
+ goto out;
+ }
+
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
@@ -3120,16 +3173,16 @@
unsigned long flags;
/* FE state */
- offset += snprintf(buf + offset, size - offset,
+ offset += scnprintf(buf + offset, size - offset,
"[%s - %s]\n", fe->dai_link->name,
stream ? "Capture" : "Playback");
- offset += snprintf(buf + offset, size - offset, "State: %s\n",
+ offset += scnprintf(buf + offset, size - offset, "State: %s\n",
dpcm_state_string(fe->dpcm[stream].state));
if ((fe->dpcm[stream].state >= SND_SOC_DPCM_STATE_HW_PARAMS) &&
(fe->dpcm[stream].state <= SND_SOC_DPCM_STATE_STOP))
- offset += snprintf(buf + offset, size - offset,
+ offset += scnprintf(buf + offset, size - offset,
"Hardware Params: "
"Format = %s, Channels = %d, Rate = %d\n",
snd_pcm_format_name(params_format(params)),
@@ -3137,10 +3190,10 @@
params_rate(params));
/* BEs state */
- offset += snprintf(buf + offset, size - offset, "Backends:\n");
+ offset += scnprintf(buf + offset, size - offset, "Backends:\n");
if (list_empty(&fe->dpcm[stream].be_clients)) {
- offset += snprintf(buf + offset, size - offset,
+ offset += scnprintf(buf + offset, size - offset,
" No active DSP links\n");
goto out;
}
@@ -3150,16 +3203,16 @@
struct snd_soc_pcm_runtime *be = dpcm->be;
params = &dpcm->hw_params;
- offset += snprintf(buf + offset, size - offset,
+ offset += scnprintf(buf + offset, size - offset,
"- %s\n", be->dai_link->name);
- offset += snprintf(buf + offset, size - offset,
+ offset += scnprintf(buf + offset, size - offset,
" State: %s\n",
dpcm_state_string(be->dpcm[stream].state));
if ((be->dpcm[stream].state >= SND_SOC_DPCM_STATE_HW_PARAMS) &&
(be->dpcm[stream].state <= SND_SOC_DPCM_STATE_STOP))
- offset += snprintf(buf + offset, size - offset,
+ offset += scnprintf(buf + offset, size - offset,
" Hardware Params: "
"Format = %s, Channels = %d, Rate = %d\n",
snd_pcm_format_name(params_format(params)),
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index 0fd0329..c367609 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -362,7 +362,7 @@
struct snd_soc_component *comp = tplg->comp;
return soc_tplg_add_dcontrol(comp->card->snd_card,
- comp->dev, k, NULL, comp, kcontrol);
+ comp->dev, k, comp->name_prefix, comp, kcontrol);
}
/* remove a mixer kcontrol */
@@ -548,12 +548,12 @@
if (dobj->ops && dobj->ops->link_unload)
dobj->ops->link_unload(comp, dobj);
+ list_del(&dobj->list);
+ snd_soc_remove_dai_link(comp->card, link);
+
kfree(link->name);
kfree(link->stream_name);
kfree(link->cpus->dai_name);
-
- list_del(&dobj->list);
- snd_soc_remove_dai_link(comp->card, link);
kfree(link);
}
@@ -604,9 +604,11 @@
ext_ops = tplg->bytes_ext_ops;
num_ops = tplg->bytes_ext_ops_count;
for (i = 0; i < num_ops; i++) {
- if (!sbe->put && ext_ops[i].id == be->ext_ops.put)
+ if (!sbe->put &&
+ ext_ops[i].id == le32_to_cpu(be->ext_ops.put))
sbe->put = ext_ops[i].put;
- if (!sbe->get && ext_ops[i].id == be->ext_ops.get)
+ if (!sbe->get &&
+ ext_ops[i].id == le32_to_cpu(be->ext_ops.get))
sbe->get = ext_ops[i].get;
}
@@ -621,11 +623,11 @@
num_ops = tplg->io_ops_count;
for (i = 0; i < num_ops; i++) {
- if (k->put == NULL && ops[i].id == hdr->ops.put)
+ if (k->put == NULL && ops[i].id == le32_to_cpu(hdr->ops.put))
k->put = ops[i].put;
- if (k->get == NULL && ops[i].id == hdr->ops.get)
+ if (k->get == NULL && ops[i].id == le32_to_cpu(hdr->ops.get))
k->get = ops[i].get;
- if (k->info == NULL && ops[i].id == hdr->ops.info)
+ if (k->info == NULL && ops[i].id == le32_to_cpu(hdr->ops.info))
k->info = ops[i].info;
}
@@ -638,11 +640,11 @@
num_ops = ARRAY_SIZE(io_ops);
for (i = 0; i < num_ops; i++) {
- if (k->put == NULL && ops[i].id == hdr->ops.put)
+ if (k->put == NULL && ops[i].id == le32_to_cpu(hdr->ops.put))
k->put = ops[i].put;
- if (k->get == NULL && ops[i].id == hdr->ops.get)
+ if (k->get == NULL && ops[i].id == le32_to_cpu(hdr->ops.get))
k->get = ops[i].get;
- if (k->info == NULL && ops[i].id == hdr->ops.info)
+ if (k->info == NULL && ops[i].id == le32_to_cpu(hdr->ops.info))
k->info = ops[i].info;
}
@@ -891,7 +893,13 @@
}
/* create any TLV data */
- soc_tplg_create_tlv(tplg, &kc, &mc->hdr);
+ err = soc_tplg_create_tlv(tplg, &kc, &mc->hdr);
+ if (err < 0) {
+ dev_err(tplg->dev, "ASoC: failed to create TLV %s\n",
+ mc->hdr.name);
+ kfree(sm);
+ continue;
+ }
/* pass control to driver for optional further init */
err = soc_tplg_init_kcontrol(tplg, &kc,
@@ -931,7 +939,7 @@
if (se->dobj.control.dtexts == NULL)
return -ENOMEM;
- for (i = 0; i < ec->items; i++) {
+ for (i = 0; i < le32_to_cpu(ec->items); i++) {
if (strnlen(ec->texts[i], SNDRV_CTL_ELEM_ID_NAME_MAXLEN) ==
SNDRV_CTL_ELEM_ID_NAME_MAXLEN) {
@@ -974,7 +982,7 @@
return -EINVAL;
se->dobj.control.dvalues = kzalloc(le32_to_cpu(ec->items) *
- sizeof(u32),
+ sizeof(*se->dobj.control.dvalues),
GFP_KERNEL);
if (!se->dobj.control.dvalues)
return -ENOMEM;
@@ -1115,6 +1123,7 @@
struct snd_soc_tplg_hdr *hdr)
{
struct snd_soc_tplg_ctl_hdr *control_hdr;
+ int ret;
int i;
if (tplg->pass != SOC_TPLG_PASS_MIXER) {
@@ -1143,25 +1152,30 @@
case SND_SOC_TPLG_CTL_RANGE:
case SND_SOC_TPLG_DAPM_CTL_VOLSW:
case SND_SOC_TPLG_DAPM_CTL_PIN:
- soc_tplg_dmixer_create(tplg, 1,
- le32_to_cpu(hdr->payload_size));
+ ret = soc_tplg_dmixer_create(tplg, 1,
+ le32_to_cpu(hdr->payload_size));
break;
case SND_SOC_TPLG_CTL_ENUM:
case SND_SOC_TPLG_CTL_ENUM_VALUE:
case SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE:
case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT:
case SND_SOC_TPLG_DAPM_CTL_ENUM_VALUE:
- soc_tplg_denum_create(tplg, 1,
- le32_to_cpu(hdr->payload_size));
+ ret = soc_tplg_denum_create(tplg, 1,
+ le32_to_cpu(hdr->payload_size));
break;
case SND_SOC_TPLG_CTL_BYTES:
- soc_tplg_dbytes_create(tplg, 1,
- le32_to_cpu(hdr->payload_size));
+ ret = soc_tplg_dbytes_create(tplg, 1,
+ le32_to_cpu(hdr->payload_size));
break;
default:
soc_bind_err(tplg, control_hdr, i);
return -EINVAL;
}
+ if (ret < 0) {
+ dev_err(tplg->dev, "ASoC: invalid control\n");
+ return ret;
+ }
+
}
return 0;
@@ -1269,16 +1283,30 @@
routes[i]->dobj.index = tplg->index;
list_add(&routes[i]->dobj.list, &tplg->comp->dobj_list);
- soc_tplg_add_route(tplg, routes[i]);
+ ret = soc_tplg_add_route(tplg, routes[i]);
+ if (ret < 0) {
+ /*
+ * this route was added to the list, it will
+ * be freed in remove_route() so increment the
+ * counter to skip it in the error handling
+ * below.
+ */
+ i++;
+ break;
+ }
/* add route, but keep going if some fail */
snd_soc_dapm_add_routes(dapm, routes[i], 1);
}
- /* free memory allocated for all dapm routes in case of error */
- if (ret < 0)
- for (i = 0; i < count ; i++)
- kfree(routes[i]);
+ /*
+ * free memory allocated for all dapm routes not added to the
+ * list in case of error
+ */
+ if (ret < 0) {
+ while (i < count)
+ kfree(routes[i++]);
+ }
/*
* free pointer to array of dapm routes as this is no longer needed.
@@ -1325,7 +1353,7 @@
if (kc[i].name == NULL)
goto err_sm;
kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER;
- kc[i].access = mc->hdr.access;
+ kc[i].access = le32_to_cpu(mc->hdr.access);
/* we only support FL/FR channel mapping atm */
sm->reg = tplc_chan_get_reg(tplg, mc->channel,
@@ -1337,10 +1365,10 @@
sm->rshift = tplc_chan_get_shift(tplg, mc->channel,
SNDRV_CHMAP_FR);
- sm->max = mc->max;
- sm->min = mc->min;
- sm->invert = mc->invert;
- sm->platform_max = mc->platform_max;
+ sm->max = le32_to_cpu(mc->max);
+ sm->min = le32_to_cpu(mc->min);
+ sm->invert = le32_to_cpu(mc->invert);
+ sm->platform_max = le32_to_cpu(mc->platform_max);
sm->dobj.index = tplg->index;
INIT_LIST_HEAD(&sm->dobj.list);
@@ -1352,7 +1380,13 @@
}
/* create any TLV data */
- soc_tplg_create_tlv(tplg, &kc[i], &mc->hdr);
+ err = soc_tplg_create_tlv(tplg, &kc[i], &mc->hdr);
+ if (err < 0) {
+ dev_err(tplg->dev, "ASoC: failed to create TLV %s\n",
+ mc->hdr.name);
+ kfree(sm);
+ continue;
+ }
/* pass control to driver for optional further init */
err = soc_tplg_init_kcontrol(tplg, &kc[i],
@@ -1360,7 +1394,6 @@
if (err < 0) {
dev_err(tplg->dev, "ASoC: failed to init %s\n",
mc->hdr.name);
- soc_tplg_free_tlv(tplg, &kc[i]);
goto err_sm;
}
}
@@ -1368,6 +1401,7 @@
err_sm:
for (; i >= 0; i--) {
+ soc_tplg_free_tlv(tplg, &kc[i]);
sm = (struct soc_mixer_control *)kc[i].private_value;
kfree(sm);
kfree(kc[i].name);
@@ -1401,7 +1435,7 @@
goto err_se;
tplg->pos += (sizeof(struct snd_soc_tplg_enum_control) +
- ec->priv.size);
+ le32_to_cpu(ec->priv.size));
dev_dbg(tplg->dev, " adding DAPM widget enum control %s\n",
ec->hdr.name);
@@ -1411,7 +1445,7 @@
if (kc[i].name == NULL)
goto err_se;
kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER;
- kc[i].access = ec->hdr.access;
+ kc[i].access = le32_to_cpu(ec->hdr.access);
/* we only support FL/FR channel mapping atm */
se->reg = tplc_chan_get_reg(tplg, ec->channel, SNDRV_CHMAP_FL);
@@ -1420,8 +1454,8 @@
se->shift_r = tplc_chan_get_shift(tplg, ec->channel,
SNDRV_CHMAP_FR);
- se->items = ec->items;
- se->mask = ec->mask;
+ se->items = le32_to_cpu(ec->items);
+ se->mask = le32_to_cpu(ec->mask);
se->dobj.index = tplg->index;
switch (le32_to_cpu(ec->hdr.ops.info)) {
@@ -1523,9 +1557,9 @@
if (kc[i].name == NULL)
goto err_sbe;
kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER;
- kc[i].access = be->hdr.access;
+ kc[i].access = le32_to_cpu(be->hdr.access);
- sbe->max = be->max;
+ sbe->max = le32_to_cpu(be->max);
INIT_LIST_HEAD(&sbe->dobj.list);
/* map standard io handlers and check for external handlers */
@@ -1891,6 +1925,10 @@
link->num_codecs = 1;
link->num_platforms = 1;
+ link->dobj.index = tplg->index;
+ link->dobj.ops = tplg->ops;
+ link->dobj.type = SND_SOC_DOBJ_DAI_LINK;
+
if (strlen(pcm->pcm_name)) {
link->name = kstrdup(pcm->pcm_name, GFP_KERNEL);
link->stream_name = kstrdup(pcm->pcm_name, GFP_KERNEL);
@@ -1918,20 +1956,24 @@
ret = soc_tplg_dai_link_load(tplg, link, NULL);
if (ret < 0) {
dev_err(tplg->comp->dev, "ASoC: FE link loading failed\n");
- kfree(link->name);
- kfree(link->stream_name);
- kfree(link->cpus->dai_name);
- kfree(link);
- return ret;
+ goto err;
}
- link->dobj.index = tplg->index;
- link->dobj.ops = tplg->ops;
- link->dobj.type = SND_SOC_DOBJ_DAI_LINK;
+ ret = snd_soc_add_dai_link(tplg->comp->card, link);
+ if (ret < 0) {
+ dev_err(tplg->comp->dev, "ASoC: adding FE link failed\n");
+ goto err;
+ }
+
list_add(&link->dobj.list, &tplg->comp->dobj_list);
- snd_soc_add_dai_link(tplg->comp->card, link);
return 0;
+err:
+ kfree(link->name);
+ kfree(link->stream_name);
+ kfree(link->cpus->dai_name);
+ kfree(link);
+ return ret;
}
/* create a FE DAI and DAI link from the PCM object */
@@ -2024,6 +2066,7 @@
int size;
int i;
bool abi_match;
+ int ret;
count = le32_to_cpu(hdr->count);
@@ -2061,11 +2104,18 @@
_pcm = pcm;
} else {
abi_match = false;
- pcm_new_ver(tplg, pcm, &_pcm);
+ ret = pcm_new_ver(tplg, pcm, &_pcm);
+ if (ret < 0)
+ return ret;
}
/* create the FE DAIs and DAI links */
- soc_tplg_pcm_create(tplg, _pcm);
+ ret = soc_tplg_pcm_create(tplg, _pcm);
+ if (ret < 0) {
+ if (!abi_match)
+ kfree(_pcm);
+ return ret;
+ }
/* offset by version-specific struct size and
* real priv data size
@@ -2304,8 +2354,11 @@
}
ret = soc_tplg_link_config(tplg, _link);
- if (ret < 0)
+ if (ret < 0) {
+ if (!abi_match)
+ kfree(_link);
return ret;
+ }
/* offset by version-specific struct size and
* real priv data size
@@ -2390,7 +2443,7 @@
{
struct snd_soc_tplg_dai *dai;
int count;
- int i;
+ int i, ret;
count = le32_to_cpu(hdr->count);
@@ -2405,7 +2458,12 @@
return -EINVAL;
}
- soc_tplg_dai_config(tplg, dai);
+ ret = soc_tplg_dai_config(tplg, dai);
+ if (ret < 0) {
+ dev_err(tplg->dev, "ASoC: failed to configure DAI\n");
+ return ret;
+ }
+
tplg->pos += (sizeof(*dai) + le32_to_cpu(dai->priv.size));
}
@@ -2469,7 +2527,7 @@
{
struct snd_soc_tplg_manifest *manifest, *_manifest;
bool abi_match;
- int err;
+ int ret = 0;
if (tplg->pass != SOC_TPLG_PASS_MANIFEST)
return 0;
@@ -2482,19 +2540,19 @@
_manifest = manifest;
} else {
abi_match = false;
- err = manifest_new_ver(tplg, manifest, &_manifest);
- if (err < 0)
- return err;
+ ret = manifest_new_ver(tplg, manifest, &_manifest);
+ if (ret < 0)
+ return ret;
}
/* pass control to component driver for optional further init */
if (tplg->comp && tplg->ops && tplg->ops->manifest)
- return tplg->ops->manifest(tplg->comp, tplg->index, _manifest);
+ ret = tplg->ops->manifest(tplg->comp, tplg->index, _manifest);
if (!abi_match) /* free the duplicated one */
kfree(_manifest);
- return 0;
+ return ret;
}
/* validate header magic, size and type */
@@ -2513,7 +2571,7 @@
}
/* big endian firmware objects not supported atm */
- if (hdr->magic == SOC_TPLG_MAGIC_BIG_ENDIAN) {
+ if (le32_to_cpu(hdr->magic) == SOC_TPLG_MAGIC_BIG_ENDIAN) {
dev_err(tplg->dev,
"ASoC: pass %d big endian not supported header got %x at offset 0x%lx size 0x%zx.\n",
tplg->pass, hdr->magic,
diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c
index 81f28f7..2a6b84d 100644
--- a/sound/soc/sof/core.c
+++ b/sound/soc/sof/core.c
@@ -288,6 +288,46 @@
#endif
}
+/*
+ * FW Boot State Transition Diagram
+ *
+ * +-----------------------------------------------------------------------+
+ * | |
+ * ------------------ ------------------ |
+ * | | | | |
+ * | BOOT_FAILED | | READY_FAILED |-------------------------+ |
+ * | | | | | |
+ * ------------------ ------------------ | |
+ * ^ ^ | |
+ * | | | |
+ * (FW Boot Timeout) (FW_READY FAIL) | |
+ * | | | |
+ * | | | |
+ * ------------------ | ------------------ | |
+ * | | | | | | |
+ * | IN_PROGRESS |---------------+------------->| COMPLETE | | |
+ * | | (FW Boot OK) (FW_READY OK) | | | |
+ * ------------------ ------------------ | |
+ * ^ | | |
+ * | | | |
+ * (FW Loading OK) (System Suspend/Runtime Suspend)
+ * | | | |
+ * | | | |
+ * ------------------ ------------------ | | |
+ * | | | |<-----+ | |
+ * | PREPARE | | NOT_STARTED |<---------------------+ |
+ * | | | |<---------------------------+
+ * ------------------ ------------------
+ * | ^ | ^
+ * | | | |
+ * | +-----------------------+ |
+ * | (DSP Probe OK) |
+ * | |
+ * | |
+ * +------------------------------------+
+ * (System Suspend/Runtime Suspend)
+ */
+
static int sof_probe_continue(struct snd_sof_dev *sdev)
{
struct snd_sof_pdata *plat_data = sdev->pdata;
@@ -303,6 +343,8 @@
return ret;
}
+ sdev->fw_state = SOF_FW_BOOT_PREPARE;
+
/* check machine info */
ret = sof_machine_check(sdev);
if (ret < 0) {
@@ -330,6 +372,7 @@
/* init the IPC */
sdev->ipc = snd_sof_ipc_init(sdev);
if (!sdev->ipc) {
+ ret = -ENOMEM;
dev_err(sdev->dev, "error: failed to init DSP IPC %d\n", ret);
goto ipc_err;
}
@@ -342,7 +385,12 @@
goto fw_load_err;
}
- /* boot the firmware */
+ sdev->fw_state = SOF_FW_BOOT_IN_PROGRESS;
+
+ /*
+ * Boot the firmware. The FW boot status will be modified
+ * in snd_sof_run_firmware() depending on the outcome.
+ */
ret = snd_sof_run_firmware(sdev);
if (ret < 0) {
dev_err(sdev->dev, "error: failed to boot DSP firmware %d\n",
@@ -368,7 +416,7 @@
if (ret < 0) {
dev_err(sdev->dev,
"error: failed to register DSP DAI driver %d\n", ret);
- goto fw_run_err;
+ goto fw_trace_err;
}
drv_name = plat_data->machine->drv_name;
@@ -382,7 +430,7 @@
if (IS_ERR(plat_data->pdev_mach)) {
ret = PTR_ERR(plat_data->pdev_mach);
- goto fw_run_err;
+ goto fw_trace_err;
}
dev_dbg(sdev->dev, "created machine %s\n",
@@ -393,7 +441,8 @@
return 0;
-#if !IS_ENABLED(CONFIG_SND_SOC_SOF_PROBE_WORK_QUEUE)
+fw_trace_err:
+ snd_sof_free_trace(sdev);
fw_run_err:
snd_sof_fw_unload(sdev);
fw_load_err:
@@ -402,21 +451,10 @@
snd_sof_free_debug(sdev);
dbg_err:
snd_sof_remove(sdev);
-#else
- /*
- * when the probe_continue is handled in a work queue, the
- * probe does not fail so we don't release resources here.
- * They will be released with an explicit call to
- * snd_sof_device_remove() when the PCI/ACPI device is removed
- */
-
-fw_run_err:
-fw_load_err:
-ipc_err:
-dbg_err:
-
-#endif
+ /* all resources freed, update state to match */
+ sdev->fw_state = SOF_FW_BOOT_NOT_STARTED;
+ sdev->first_boot = true;
return ret;
}
@@ -447,6 +485,7 @@
sdev->pdata = plat_data;
sdev->first_boot = true;
+ sdev->fw_state = SOF_FW_BOOT_NOT_STARTED;
dev_set_drvdata(dev, sdev);
/* check all mandatory ops */
@@ -494,10 +533,12 @@
if (IS_ENABLED(CONFIG_SND_SOC_SOF_PROBE_WORK_QUEUE))
cancel_work_sync(&sdev->probe_work);
- snd_sof_fw_unload(sdev);
- snd_sof_ipc_free(sdev);
- snd_sof_free_debug(sdev);
- snd_sof_free_trace(sdev);
+ if (sdev->fw_state > SOF_FW_BOOT_NOT_STARTED) {
+ snd_sof_fw_unload(sdev);
+ snd_sof_ipc_free(sdev);
+ snd_sof_free_debug(sdev);
+ snd_sof_free_trace(sdev);
+ }
/*
* Unregister machine driver. This will unbind the snd_card which
@@ -513,7 +554,8 @@
* scheduled on, when they are unloaded. Therefore, the DSP must be
* removed only after the topology has been unloaded.
*/
- snd_sof_remove(sdev);
+ if (sdev->fw_state > SOF_FW_BOOT_NOT_STARTED)
+ snd_sof_remove(sdev);
/* release firmware */
release_firmware(pdata->fw);
diff --git a/sound/soc/sof/debug.c b/sound/soc/sof/debug.c
index 5529e8e..0872603 100644
--- a/sound/soc/sof/debug.c
+++ b/sound/soc/sof/debug.c
@@ -135,7 +135,7 @@
char *string;
int ret;
- string = kzalloc(count, GFP_KERNEL);
+ string = kzalloc(count+1, GFP_KERNEL);
if (!string)
return -ENOMEM;
diff --git a/sound/soc/sof/imx/Kconfig b/sound/soc/sof/imx/Kconfig
index 5acae75..b4f0426 100644
--- a/sound/soc/sof/imx/Kconfig
+++ b/sound/soc/sof/imx/Kconfig
@@ -11,13 +11,17 @@
if SND_SOC_SOF_IMX_TOPLEVEL
-config SND_SOC_SOF_IMX8
- tristate "SOF support for i.MX8"
+config SND_SOC_SOF_IMX8_SUPPORT
+ bool "SOF support for i.MX8"
depends on IMX_SCU
- depends on IMX_DSP
+ select IMX_DSP
help
This adds support for Sound Open Firmware for NXP i.MX8 platforms
Say Y if you have such a device.
If unsure select "N".
+config SND_SOC_SOF_IMX8
+ def_tristate SND_SOC_SOF_OF
+ depends on SND_SOC_SOF_IMX8_SUPPORT
+
endif ## SND_SOC_SOF_IMX_IMX_TOPLEVEL
diff --git a/sound/soc/sof/imx/imx8.c b/sound/soc/sof/imx/imx8.c
index 2a22b18..69785f6 100644
--- a/sound/soc/sof/imx/imx8.c
+++ b/sound/soc/sof/imx/imx8.c
@@ -209,7 +209,7 @@
priv->pd_dev = devm_kmalloc_array(&pdev->dev, priv->num_domains,
sizeof(*priv->pd_dev), GFP_KERNEL);
- if (!priv)
+ if (!priv->pd_dev)
return -ENOMEM;
priv->link = devm_kmalloc_array(&pdev->dev, priv->num_domains,
@@ -304,6 +304,9 @@
}
sdev->mailbox_bar = SOF_FW_BLK_TYPE_SRAM;
+ /* set default mailbox offset for FW ready message */
+ sdev->dsp_box.offset = MBOX_OFFSET;
+
return 0;
exit_pdev_unregister:
diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig
index d62f51d..8421b97 100644
--- a/sound/soc/sof/intel/Kconfig
+++ b/sound/soc/sof/intel/Kconfig
@@ -76,10 +76,18 @@
config SND_SOC_SOF_BROADWELL_SUPPORT
bool "SOF support for Broadwell"
+ depends on SND_SOC_INTEL_HASWELL=n
help
This adds support for Sound Open Firmware for Intel(R) platforms
using the Broadwell processors.
- Say Y if you have such a device.
+ This option is mutually exclusive with the Haswell/Broadwell legacy
+ driver. If you want to enable SOF on Broadwell you need to deselect
+ the legacy driver first.
+ SOF does fully support Broadwell yet, so this option is not
+ recommended for distros. At some point all legacy drivers will be
+ deprecated but not before all userspace firmware/topology/UCM files
+ are made available to downstream distros.
+ Say Y if you want to enable SOF on Broadwell
If unsure select "N".
config SND_SOC_SOF_BROADWELL
diff --git a/sound/soc/sof/intel/byt.c b/sound/soc/sof/intel/byt.c
index a1e514f..41008c9 100644
--- a/sound/soc/sof/intel/byt.c
+++ b/sound/soc/sof/intel/byt.c
@@ -24,7 +24,8 @@
#define DRAM_OFFSET 0x100000
#define DRAM_SIZE (160 * 1024)
#define SHIM_OFFSET 0x140000
-#define SHIM_SIZE 0x100
+#define SHIM_SIZE_BYT 0x100
+#define SHIM_SIZE_CHT 0x118
#define MBOX_OFFSET 0x144000
#define MBOX_SIZE 0x1000
#define EXCEPT_OFFSET 0x800
@@ -75,7 +76,7 @@
SOF_DEBUGFS_ACCESS_D0_ONLY},
{"dram", BYT_DSP_BAR, DRAM_OFFSET, DRAM_SIZE,
SOF_DEBUGFS_ACCESS_D0_ONLY},
- {"shim", BYT_DSP_BAR, SHIM_OFFSET, SHIM_SIZE,
+ {"shim", BYT_DSP_BAR, SHIM_OFFSET, SHIM_SIZE_BYT,
SOF_DEBUGFS_ACCESS_ALWAYS},
};
@@ -102,7 +103,7 @@
SOF_DEBUGFS_ACCESS_D0_ONLY},
{"dram", BYT_DSP_BAR, DRAM_OFFSET, DRAM_SIZE,
SOF_DEBUGFS_ACCESS_D0_ONLY},
- {"shim", BYT_DSP_BAR, SHIM_OFFSET, SHIM_SIZE,
+ {"shim", BYT_DSP_BAR, SHIM_OFFSET, SHIM_SIZE_CHT,
SOF_DEBUGFS_ACCESS_ALWAYS},
};
diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c
index 3ca6795..df38616 100644
--- a/sound/soc/sof/intel/hda-codec.c
+++ b/sound/soc/sof/intel/hda-codec.c
@@ -24,19 +24,18 @@
#define IDISP_VID_INTEL 0x80860000
/* load the legacy HDA codec driver */
-#ifdef MODULE
-static void hda_codec_load_module(struct hda_codec *codec)
+static int hda_codec_load_module(struct hda_codec *codec)
{
+#ifdef MODULE
char alias[MODULE_NAME_LEN];
const char *module = alias;
snd_hdac_codec_modalias(&codec->core, alias, sizeof(alias));
dev_dbg(&codec->core.dev, "loading codec module: %s\n", module);
request_module(module);
-}
-#else
-static void hda_codec_load_module(struct hda_codec *codec) {}
#endif
+ return device_attach(hda_codec_dev(codec));
+}
/* enable controller wake up event for all codecs with jack connectors */
void hda_codec_jack_wake_enable(struct snd_sof_dev *sdev)
@@ -69,8 +68,7 @@
* has been recorded in STATESTS
*/
if (codec->jacktbl.used)
- schedule_delayed_work(&codec->jackpoll_work,
- codec->jackpoll_interval);
+ pm_request_resume(&codec->core.dev);
}
#else
void hda_codec_jack_wake_enable(struct snd_sof_dev *sdev) {}
@@ -116,10 +114,16 @@
/* use legacy bus only for HDA codecs, idisp uses ext bus */
if ((resp & 0xFFFF0000) != IDISP_VID_INTEL) {
hdev->type = HDA_DEV_LEGACY;
- hda_codec_load_module(&hda_priv->codec);
+ ret = hda_codec_load_module(&hda_priv->codec);
+ /*
+ * handle ret==0 (no driver bound) as an error, but pass
+ * other return codes without modification
+ */
+ if (ret == 0)
+ ret = -ENOENT;
}
- return 0;
+ return ret;
#else
hdev = devm_kzalloc(sdev->dev, sizeof(*hdev), GFP_KERNEL);
if (!hdev)
diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c
index 8796f38..3f64520 100644
--- a/sound/soc/sof/intel/hda-dai.c
+++ b/sound/soc/sof/intel/hda-dai.c
@@ -216,6 +216,8 @@
link_dev = hda_link_stream_assign(bus, substream);
if (!link_dev)
return -EBUSY;
+
+ snd_soc_dai_set_dma_data(dai, substream, (void *)link_dev);
}
stream_tag = hdac_stream(link_dev)->stream_tag;
@@ -228,8 +230,6 @@
if (ret < 0)
return ret;
- snd_soc_dai_set_dma_data(dai, substream, (void *)link_dev);
-
link = snd_hdac_ext_bus_get_link(bus, codec_dai->component->name);
if (!link)
return -EINVAL;
@@ -261,14 +261,11 @@
{
struct hdac_ext_stream *link_dev =
snd_soc_dai_get_dma_data(dai, substream);
- struct sof_intel_hda_stream *hda_stream;
struct snd_sof_dev *sdev =
snd_soc_component_get_drvdata(dai->component);
struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
int stream = substream->stream;
- hda_stream = hstream_to_sof_hda_stream(link_dev);
-
if (link_dev->link_prepared)
return 0;
@@ -361,6 +358,13 @@
bus = hstream->bus;
rtd = snd_pcm_substream_chip(substream);
link_dev = snd_soc_dai_get_dma_data(dai, substream);
+
+ if (!link_dev) {
+ dev_dbg(dai->dev,
+ "%s: link_dev is not assigned\n", __func__);
+ return -EINVAL;
+ }
+
hda_stream = hstream_to_sof_hda_stream(link_dev);
/* free the link DMA channel in the FW */
@@ -440,6 +444,10 @@
.ops = &hda_link_dai_ops,
},
{
+ .name = "iDisp4 Pin",
+ .ops = &hda_link_dai_ops,
+},
+{
.name = "Analog CPU DAI",
.ops = &hda_link_dai_ops,
},
diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c
index fb55a3c..06715b3 100644
--- a/sound/soc/sof/intel/hda-dsp.c
+++ b/sound/soc/sof/intel/hda-dsp.c
@@ -179,7 +179,7 @@
return snd_sof_dsp_read_poll_timeout(sdev, HDA_DSP_BAR,
HDA_DSP_REG_ADSPCS, adspcs,
- !(adspcs & HDA_DSP_ADSPCS_SPA_MASK(core_mask)),
+ !(adspcs & HDA_DSP_ADSPCS_CPA_MASK(core_mask)),
HDA_DSP_REG_POLL_INTERVAL_US,
HDA_DSP_PD_TIMEOUT * USEC_PER_MSEC);
}
@@ -192,10 +192,17 @@
val = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_ADSPCS);
- is_enable = ((val & HDA_DSP_ADSPCS_CPA_MASK(core_mask)) &&
- (val & HDA_DSP_ADSPCS_SPA_MASK(core_mask)) &&
- !(val & HDA_DSP_ADSPCS_CRST_MASK(core_mask)) &&
- !(val & HDA_DSP_ADSPCS_CSTALL_MASK(core_mask)));
+#define MASK_IS_EQUAL(v, m, field) ({ \
+ u32 _m = field(m); \
+ ((v) & _m) == _m; \
+})
+
+ is_enable = MASK_IS_EQUAL(val, core_mask, HDA_DSP_ADSPCS_CPA_MASK) &&
+ MASK_IS_EQUAL(val, core_mask, HDA_DSP_ADSPCS_SPA_MASK) &&
+ !(val & HDA_DSP_ADSPCS_CRST_MASK(core_mask)) &&
+ !(val & HDA_DSP_ADSPCS_CSTALL_MASK(core_mask));
+
+#undef MASK_IS_EQUAL
dev_dbg(sdev->dev, "DSP core(s) enabled? %d : core_mask %x\n",
is_enable, core_mask);
diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c
index 65c2af3..356bb13 100644
--- a/sound/soc/sof/intel/hda-loader.c
+++ b/sound/soc/sof/intel/hda-loader.c
@@ -278,7 +278,6 @@
/* init for booting wait */
init_waitqueue_head(&sdev->boot_wait);
- sdev->boot_complete = false;
/* prepare DMA for code loader stream */
tag = cl_stream_prepare(sdev, 0x40, stripped_firmware.size,
diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c
index 06e8467..a101667 100644
--- a/sound/soc/sof/intel/hda.c
+++ b/sound/soc/sof/intel/hda.c
@@ -166,7 +166,7 @@
panic = snd_sof_dsp_read(sdev, HDA_DSP_BAR,
HDA_ADSP_ERROR_CODE_SKL + 0x4);
- if (sdev->boot_complete) {
+ if (sdev->fw_state == SOF_FW_BOOT_COMPLETE) {
hda_dsp_get_registers(sdev, &xoops, &panic_info, stack,
HDA_DSP_STACK_DUMP_SIZE);
snd_sof_get_status(sdev, status, panic, &xoops, &panic_info,
@@ -193,7 +193,7 @@
HDA_DSP_SRAM_REG_FW_STATUS);
panic = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_SRAM_REG_FW_TRACEP);
- if (sdev->boot_complete) {
+ if (sdev->fw_state == SOF_FW_BOOT_COMPLETE) {
hda_dsp_get_registers(sdev, &xoops, &panic_info, stack,
HDA_DSP_STACK_DUMP_SIZE);
snd_sof_get_status(sdev, status, panic, &xoops, &panic_info,
@@ -268,6 +268,7 @@
bus->use_posbuf = 1;
bus->bdl_pos_adj = 0;
+ bus->sync_write = 1;
mutex_init(&hbus->prepare_mutex);
hbus->pci = pci;
@@ -671,6 +672,7 @@
/* dsp_unmap: not currently used */
iounmap(sdev->bar[HDA_DSP_BAR]);
hdac_bus_unmap:
+ platform_device_unregister(hdev->dmic_dev);
iounmap(bus->remap_addr);
err:
return ret;
diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h
index 23e430d..4be53ef 100644
--- a/sound/soc/sof/intel/hda.h
+++ b/sound/soc/sof/intel/hda.h
@@ -336,7 +336,7 @@
/* Number of DAIs */
#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA)
-#define SOF_SKL_NUM_DAIS 14
+#define SOF_SKL_NUM_DAIS 15
#else
#define SOF_SKL_NUM_DAIS 8
#endif
diff --git a/sound/soc/sof/ipc.c b/sound/soc/sof/ipc.c
index 086eeea..f38f651 100644
--- a/sound/soc/sof/ipc.c
+++ b/sound/soc/sof/ipc.c
@@ -215,15 +215,17 @@
snd_sof_trace_notify_for_error(ipc->sdev);
ret = -ETIMEDOUT;
} else {
- /* copy the data returned from DSP */
ret = msg->reply_error;
- if (msg->reply_size)
- memcpy(reply_data, msg->reply_data, msg->reply_size);
- if (ret < 0)
+ if (ret < 0) {
dev_err(sdev->dev, "error: ipc error for 0x%x size %zu\n",
hdr->cmd, msg->reply_size);
- else
+ } else {
ipc_log_header(sdev->dev, "ipc tx succeeded", hdr->cmd);
+ if (msg->reply_size)
+ /* copy the data returned from DSP */
+ memcpy(reply_data, msg->reply_data,
+ msg->reply_size);
+ }
}
return ret;
@@ -348,19 +350,12 @@
break;
case SOF_IPC_FW_READY:
/* check for FW boot completion */
- if (!sdev->boot_complete) {
+ if (sdev->fw_state == SOF_FW_BOOT_IN_PROGRESS) {
err = sof_ops(sdev)->fw_ready(sdev, cmd);
- if (err < 0) {
- /*
- * this indicates a mismatch in ABI
- * between the driver and fw
- */
- dev_err(sdev->dev, "error: ABI mismatch %d\n",
- err);
- } else {
- /* firmware boot completed OK */
- sdev->boot_complete = true;
- }
+ if (err < 0)
+ sdev->fw_state = SOF_FW_BOOT_READY_FAILED;
+ else
+ sdev->fw_state = SOF_FW_BOOT_COMPLETE;
/* wake up firmware loader */
wake_up(&sdev->boot_wait);
@@ -504,7 +499,7 @@
/* send IPC to the DSP */
err = sof_ipc_tx_message(sdev->ipc,
- stream.hdr.cmd, &stream, sizeof(stream), &posn,
+ stream.hdr.cmd, &stream, sizeof(stream), posn,
sizeof(*posn));
if (err < 0) {
dev_err(sdev->dev, "error: failed to get stream %d position\n",
@@ -834,6 +829,9 @@
{
struct snd_sof_ipc *ipc = sdev->ipc;
+ if (!ipc)
+ return;
+
/* disable sending of ipc's */
mutex_lock(&ipc->tx_mutex);
ipc->disable_ipc_tx = true;
diff --git a/sound/soc/sof/loader.c b/sound/soc/sof/loader.c
index 9a9a381..ce114df 100644
--- a/sound/soc/sof/loader.c
+++ b/sound/soc/sof/loader.c
@@ -66,6 +66,8 @@
ret = get_ext_windows(sdev, ext_hdr);
break;
default:
+ dev_warn(sdev->dev, "warning: unknown ext header type %d size 0x%x\n",
+ ext_hdr->type, ext_hdr->hdr.size);
break;
}
@@ -509,7 +511,6 @@
int init_core_mask;
init_waitqueue_head(&sdev->boot_wait);
- sdev->boot_complete = false;
/* create read-only fw_version debugfs to store boot version info */
if (sdev->first_boot) {
@@ -541,19 +542,27 @@
init_core_mask = ret;
- /* now wait for the DSP to boot */
- ret = wait_event_timeout(sdev->boot_wait, sdev->boot_complete,
+ /*
+ * now wait for the DSP to boot. There are 3 possible outcomes:
+ * 1. Boot wait times out indicating FW boot failure.
+ * 2. FW boots successfully and fw_ready op succeeds.
+ * 3. FW boots but fw_ready op fails.
+ */
+ ret = wait_event_timeout(sdev->boot_wait,
+ sdev->fw_state > SOF_FW_BOOT_IN_PROGRESS,
msecs_to_jiffies(sdev->boot_timeout));
if (ret == 0) {
dev_err(sdev->dev, "error: firmware boot failure\n");
snd_sof_dsp_dbg_dump(sdev, SOF_DBG_REGS | SOF_DBG_MBOX |
SOF_DBG_TEXT | SOF_DBG_PCI);
- /* after this point FW_READY msg should be ignored */
- sdev->boot_complete = true;
+ sdev->fw_state = SOF_FW_BOOT_FAILED;
return -EIO;
}
- dev_info(sdev->dev, "firmware boot complete\n");
+ if (sdev->fw_state == SOF_FW_BOOT_COMPLETE)
+ dev_info(sdev->dev, "firmware boot complete\n");
+ else
+ return -EIO; /* FW boots but fw_ready op failed */
/* perform post fw run operations */
ret = snd_sof_dsp_post_fw_run(sdev);
diff --git a/sound/soc/sof/nocodec.c b/sound/soc/sof/nocodec.c
index 3d128e5..7141011 100644
--- a/sound/soc/sof/nocodec.c
+++ b/sound/soc/sof/nocodec.c
@@ -14,6 +14,7 @@
static struct snd_soc_card sof_nocodec_card = {
.name = "nocodec", /* the sof- prefix is added by the core */
+ .owner = THIS_MODULE
};
static int sof_nocodec_bes_setup(struct device *dev,
@@ -52,8 +53,10 @@
links[i].platforms->name = dev_name(dev);
links[i].codecs->dai_name = "snd-soc-dummy-dai";
links[i].codecs->name = "snd-soc-dummy";
- links[i].dpcm_playback = 1;
- links[i].dpcm_capture = 1;
+ if (ops->drv[i].playback.channels_min)
+ links[i].dpcm_playback = 1;
+ if (ops->drv[i].capture.channels_min)
+ links[i].dpcm_capture = 1;
}
card->dai_link = links;
diff --git a/sound/soc/sof/pm.c b/sound/soc/sof/pm.c
index e23beae..128680b 100644
--- a/sound/soc/sof/pm.c
+++ b/sound/soc/sof/pm.c
@@ -266,7 +266,14 @@
int ret;
/* do nothing if dsp resume callbacks are not set */
- if (!sof_ops(sdev)->resume || !sof_ops(sdev)->runtime_resume)
+ if (!runtime_resume && !sof_ops(sdev)->resume)
+ return 0;
+
+ if (runtime_resume && !sof_ops(sdev)->runtime_resume)
+ return 0;
+
+ /* DSP was never successfully started, nothing to resume */
+ if (sdev->first_boot)
return 0;
/*
@@ -283,6 +290,8 @@
return ret;
}
+ sdev->fw_state = SOF_FW_BOOT_PREPARE;
+
/* load the firmware */
ret = snd_sof_load_firmware(sdev);
if (ret < 0) {
@@ -292,7 +301,12 @@
return ret;
}
- /* boot the firmware */
+ sdev->fw_state = SOF_FW_BOOT_IN_PROGRESS;
+
+ /*
+ * Boot the firmware. The FW boot status will be modified
+ * in snd_sof_run_firmware() depending on the outcome.
+ */
ret = snd_sof_run_firmware(sdev);
if (ret < 0) {
dev_err(sdev->dev,
@@ -335,9 +349,15 @@
int ret;
/* do nothing if dsp suspend callback is not set */
- if (!sof_ops(sdev)->suspend)
+ if (!runtime_suspend && !sof_ops(sdev)->suspend)
return 0;
+ if (runtime_suspend && !sof_ops(sdev)->runtime_suspend)
+ return 0;
+
+ if (sdev->fw_state != SOF_FW_BOOT_COMPLETE)
+ goto power_down;
+
/* release trace */
snd_sof_release_trace(sdev);
@@ -375,6 +395,12 @@
ret);
}
+power_down:
+
+ /* return if the DSP was not probed successfully */
+ if (sdev->fw_state == SOF_FW_BOOT_NOT_STARTED)
+ return 0;
+
/* power down all DSP cores */
if (runtime_suspend)
ret = snd_sof_dsp_runtime_suspend(sdev);
@@ -385,6 +411,9 @@
"error: failed to power down DSP during suspend %d\n",
ret);
+ /* reset FW state */
+ sdev->fw_state = SOF_FW_BOOT_NOT_STARTED;
+
return ret;
}
diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c
index d66412a..3f79cd0 100644
--- a/sound/soc/sof/sof-pci-dev.c
+++ b/sound/soc/sof/sof-pci-dev.c
@@ -420,6 +420,8 @@
#if IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_H)
{ PCI_DEVICE(0x8086, 0x06c8),
.driver_data = (unsigned long)&cml_desc},
+ { PCI_DEVICE(0x8086, 0xa3f0), /* CML-S */
+ .driver_data = (unsigned long)&cml_desc},
#endif
#if IS_ENABLED(CONFIG_SND_SOC_SOF_TIGERLAKE)
{ PCI_DEVICE(0x8086, 0xa0c8),
diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h
index 730f325..7b329bd 100644
--- a/sound/soc/sof/sof-priv.h
+++ b/sound/soc/sof/sof-priv.h
@@ -356,6 +356,15 @@
struct list_head list; /* list in sdev dai list */
};
+enum snd_sof_fw_state {
+ SOF_FW_BOOT_NOT_STARTED = 0,
+ SOF_FW_BOOT_PREPARE,
+ SOF_FW_BOOT_IN_PROGRESS,
+ SOF_FW_BOOT_FAILED,
+ SOF_FW_BOOT_READY_FAILED, /* firmware booted but fw_ready op failed */
+ SOF_FW_BOOT_COMPLETE,
+};
+
/*
* SOF Device Level.
*/
@@ -372,7 +381,7 @@
/* DSP firmware boot */
wait_queue_head_t boot_wait;
- u32 boot_complete;
+ enum snd_sof_fw_state fw_state;
u32 first_boot;
/* work queue in case the probe is implemented in two steps */
diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c
index 4452594..fa299e0 100644
--- a/sound/soc/sof/topology.c
+++ b/sound/soc/sof/topology.c
@@ -2828,6 +2828,10 @@
if (!link->no_pcm) {
link->nonatomic = true;
+ /* set trigger order */
+ link->trigger[0] = SND_SOC_DPCM_TRIGGER_POST;
+ link->trigger[1] = SND_SOC_DPCM_TRIGGER_POST;
+
/* nothing more to do for FE dai links */
return 0;
}
diff --git a/sound/soc/sof/trace.c b/sound/soc/sof/trace.c
index 4c3cff0..fd6f591 100644
--- a/sound/soc/sof/trace.c
+++ b/sound/soc/sof/trace.c
@@ -328,7 +328,10 @@
{
snd_sof_release_trace(sdev);
- snd_dma_free_pages(&sdev->dmatb);
- snd_dma_free_pages(&sdev->dmatp);
+ if (sdev->dma_trace_pages) {
+ snd_dma_free_pages(&sdev->dmatb);
+ snd_dma_free_pages(&sdev->dmatp);
+ sdev->dma_trace_pages = 0;
+ }
}
EXPORT_SYMBOL(snd_sof_free_trace);
diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c
index 48ea915..2ed92c9 100644
--- a/sound/soc/sti/uniperif_player.c
+++ b/sound/soc/sti/uniperif_player.c
@@ -226,7 +226,6 @@
* sampling frequency. If no sample rate is already specified, then
* set one.
*/
- mutex_lock(&player->ctrl_lock);
if (runtime) {
switch (runtime->rate) {
case 22050:
@@ -303,7 +302,6 @@
player->stream_settings.iec958.status[3 + (n * 4)] << 24;
SET_UNIPERIF_CHANNEL_STA_REGN(player, n, status);
}
- mutex_unlock(&player->ctrl_lock);
/* Update the channel status */
if (player->ver < SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0)
@@ -365,8 +363,10 @@
SET_UNIPERIF_CTRL_ZERO_STUFF_HW(player);
+ mutex_lock(&player->ctrl_lock);
/* Update the channel status */
uni_player_set_channel_status(player, runtime);
+ mutex_unlock(&player->ctrl_lock);
/* Clear the user validity user bits */
SET_UNIPERIF_USER_VALIDITY_VALIDITY_LR(player, 0);
@@ -598,7 +598,6 @@
iec958->status[1] = ucontrol->value.iec958.status[1];
iec958->status[2] = ucontrol->value.iec958.status[2];
iec958->status[3] = ucontrol->value.iec958.status[3];
- mutex_unlock(&player->ctrl_lock);
spin_lock_irqsave(&player->irq_lock, flags);
if (player->substream && player->substream->runtime)
@@ -608,6 +607,8 @@
uni_player_set_channel_status(player, NULL);
spin_unlock_irqrestore(&player->irq_lock, flags);
+ mutex_unlock(&player->ctrl_lock);
+
return 0;
}
diff --git a/sound/soc/stm/stm32_adfsdm.c b/sound/soc/stm/stm32_adfsdm.c
index 3c9a9de..4ecea49 100644
--- a/sound/soc/stm/stm32_adfsdm.c
+++ b/sound/soc/stm/stm32_adfsdm.c
@@ -153,13 +153,13 @@
.name = "stm32_dfsdm_audio",
};
-static void memcpy_32to16(void *dest, const void *src, size_t n)
+static void stm32_memcpy_32to16(void *dest, const void *src, size_t n)
{
unsigned int i = 0;
u16 *d = (u16 *)dest, *s = (u16 *)src;
s++;
- for (i = n; i > 0; i--) {
+ for (i = n >> 1; i > 0; i--) {
*d++ = *s++;
s++;
}
@@ -186,8 +186,8 @@
if ((priv->pos + src_size) > buff_size) {
if (format == SNDRV_PCM_FORMAT_S16_LE)
- memcpy_32to16(&pcm_buff[priv->pos], src_buff,
- buff_size - priv->pos);
+ stm32_memcpy_32to16(&pcm_buff[priv->pos], src_buff,
+ buff_size - priv->pos);
else
memcpy(&pcm_buff[priv->pos], src_buff,
buff_size - priv->pos);
@@ -196,8 +196,8 @@
}
if (format == SNDRV_PCM_FORMAT_S16_LE)
- memcpy_32to16(&pcm_buff[priv->pos],
- &src_buff[src_size - cur_size], cur_size);
+ stm32_memcpy_32to16(&pcm_buff[priv->pos],
+ &src_buff[src_size - cur_size], cur_size);
else
memcpy(&pcm_buff[priv->pos], &src_buff[src_size - cur_size],
cur_size);
diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c
index 48e629a..7e96584 100644
--- a/sound/soc/stm/stm32_sai_sub.c
+++ b/sound/soc/stm/stm32_sai_sub.c
@@ -184,6 +184,56 @@
}
}
+static int stm32_sai_sub_reg_up(struct stm32_sai_sub_data *sai,
+ unsigned int reg, unsigned int mask,
+ unsigned int val)
+{
+ int ret;
+
+ ret = clk_enable(sai->pdata->pclk);
+ if (ret < 0)
+ return ret;
+
+ ret = regmap_update_bits(sai->regmap, reg, mask, val);
+
+ clk_disable(sai->pdata->pclk);
+
+ return ret;
+}
+
+static int stm32_sai_sub_reg_wr(struct stm32_sai_sub_data *sai,
+ unsigned int reg, unsigned int mask,
+ unsigned int val)
+{
+ int ret;
+
+ ret = clk_enable(sai->pdata->pclk);
+ if (ret < 0)
+ return ret;
+
+ ret = regmap_write_bits(sai->regmap, reg, mask, val);
+
+ clk_disable(sai->pdata->pclk);
+
+ return ret;
+}
+
+static int stm32_sai_sub_reg_rd(struct stm32_sai_sub_data *sai,
+ unsigned int reg, unsigned int *val)
+{
+ int ret;
+
+ ret = clk_enable(sai->pdata->pclk);
+ if (ret < 0)
+ return ret;
+
+ ret = regmap_read(sai->regmap, reg, val);
+
+ clk_disable(sai->pdata->pclk);
+
+ return ret;
+}
+
static const struct regmap_config stm32_sai_sub_regmap_config_f4 = {
.reg_bits = 32,
.reg_stride = 4,
@@ -295,7 +345,7 @@
mask = SAI_XCR1_MCKDIV_MASK(SAI_XCR1_MCKDIV_WIDTH(version));
cr1 = SAI_XCR1_MCKDIV_SET(div);
- ret = regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, mask, cr1);
+ ret = stm32_sai_sub_reg_up(sai, STM_SAI_CR1_REGX, mask, cr1);
if (ret < 0)
dev_err(&sai->pdev->dev, "Failed to update CR1 register\n");
@@ -372,8 +422,8 @@
dev_dbg(&sai->pdev->dev, "Enable master clock\n");
- return regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX,
- SAI_XCR1_MCKEN, SAI_XCR1_MCKEN);
+ return stm32_sai_sub_reg_up(sai, STM_SAI_CR1_REGX,
+ SAI_XCR1_MCKEN, SAI_XCR1_MCKEN);
}
static void stm32_sai_mclk_disable(struct clk_hw *hw)
@@ -383,7 +433,7 @@
dev_dbg(&sai->pdev->dev, "Disable master clock\n");
- regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, SAI_XCR1_MCKEN, 0);
+ stm32_sai_sub_reg_up(sai, STM_SAI_CR1_REGX, SAI_XCR1_MCKEN, 0);
}
static const struct clk_ops mclk_ops = {
@@ -446,15 +496,15 @@
unsigned int sr, imr, flags;
snd_pcm_state_t status = SNDRV_PCM_STATE_RUNNING;
- regmap_read(sai->regmap, STM_SAI_IMR_REGX, &imr);
- regmap_read(sai->regmap, STM_SAI_SR_REGX, &sr);
+ stm32_sai_sub_reg_rd(sai, STM_SAI_IMR_REGX, &imr);
+ stm32_sai_sub_reg_rd(sai, STM_SAI_SR_REGX, &sr);
flags = sr & imr;
if (!flags)
return IRQ_NONE;
- regmap_write_bits(sai->regmap, STM_SAI_CLRFR_REGX, SAI_XCLRFR_MASK,
- SAI_XCLRFR_MASK);
+ stm32_sai_sub_reg_wr(sai, STM_SAI_CLRFR_REGX, SAI_XCLRFR_MASK,
+ SAI_XCLRFR_MASK);
if (!sai->substream) {
dev_err(&pdev->dev, "Device stopped. Spurious IRQ 0x%x\n", sr);
@@ -503,8 +553,8 @@
int ret;
if (dir == SND_SOC_CLOCK_OUT && sai->sai_mclk) {
- ret = regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX,
- SAI_XCR1_NODIV,
+ ret = stm32_sai_sub_reg_up(sai, STM_SAI_CR1_REGX,
+ SAI_XCR1_NODIV,
freq ? 0 : SAI_XCR1_NODIV);
if (ret < 0)
return ret;
@@ -583,7 +633,7 @@
slotr_mask |= SAI_XSLOTR_SLOTEN_MASK;
- regmap_update_bits(sai->regmap, STM_SAI_SLOTR_REGX, slotr_mask, slotr);
+ stm32_sai_sub_reg_up(sai, STM_SAI_SLOTR_REGX, slotr_mask, slotr);
sai->slot_width = slot_width;
sai->slots = slots;
@@ -665,7 +715,7 @@
cr1_mask |= SAI_XCR1_CKSTR;
frcr_mask |= SAI_XFRCR_FSPOL;
- regmap_update_bits(sai->regmap, STM_SAI_FRCR_REGX, frcr_mask, frcr);
+ stm32_sai_sub_reg_up(sai, STM_SAI_FRCR_REGX, frcr_mask, frcr);
/* DAI clock master masks */
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
@@ -693,7 +743,7 @@
cr1_mask |= SAI_XCR1_SLAVE;
conf_update:
- ret = regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, cr1_mask, cr1);
+ ret = stm32_sai_sub_reg_up(sai, STM_SAI_CR1_REGX, cr1_mask, cr1);
if (ret < 0) {
dev_err(cpu_dai->dev, "Failed to update CR1 register\n");
return ret;
@@ -730,12 +780,12 @@
}
/* Enable ITs */
- regmap_write_bits(sai->regmap, STM_SAI_CLRFR_REGX,
- SAI_XCLRFR_MASK, SAI_XCLRFR_MASK);
+ stm32_sai_sub_reg_wr(sai, STM_SAI_CLRFR_REGX,
+ SAI_XCLRFR_MASK, SAI_XCLRFR_MASK);
imr = SAI_XIMR_OVRUDRIE;
if (STM_SAI_IS_CAPTURE(sai)) {
- regmap_read(sai->regmap, STM_SAI_CR2_REGX, &cr2);
+ stm32_sai_sub_reg_rd(sai, STM_SAI_CR2_REGX, &cr2);
if (cr2 & SAI_XCR2_MUTECNT_MASK)
imr |= SAI_XIMR_MUTEDETIE;
}
@@ -745,8 +795,8 @@
else
imr |= SAI_XIMR_AFSDETIE | SAI_XIMR_LFSDETIE;
- regmap_update_bits(sai->regmap, STM_SAI_IMR_REGX,
- SAI_XIMR_MASK, imr);
+ stm32_sai_sub_reg_up(sai, STM_SAI_IMR_REGX,
+ SAI_XIMR_MASK, imr);
return 0;
}
@@ -763,10 +813,10 @@
* SAI fifo threshold is set to half fifo, to keep enough space
* for DMA incoming bursts.
*/
- regmap_write_bits(sai->regmap, STM_SAI_CR2_REGX,
- SAI_XCR2_FFLUSH | SAI_XCR2_FTH_MASK,
- SAI_XCR2_FFLUSH |
- SAI_XCR2_FTH_SET(STM_SAI_FIFO_TH_HALF));
+ stm32_sai_sub_reg_wr(sai, STM_SAI_CR2_REGX,
+ SAI_XCR2_FFLUSH | SAI_XCR2_FTH_MASK,
+ SAI_XCR2_FFLUSH |
+ SAI_XCR2_FTH_SET(STM_SAI_FIFO_TH_HALF));
/* DS bits in CR1 not set for SPDIF (size forced to 24 bits).*/
if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) {
@@ -795,7 +845,7 @@
if ((sai->slots == 2) && (params_channels(params) == 1))
cr1 |= SAI_XCR1_MONO;
- ret = regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, cr1_mask, cr1);
+ ret = stm32_sai_sub_reg_up(sai, STM_SAI_CR1_REGX, cr1_mask, cr1);
if (ret < 0) {
dev_err(cpu_dai->dev, "Failed to update CR1 register\n");
return ret;
@@ -809,7 +859,7 @@
struct stm32_sai_sub_data *sai = snd_soc_dai_get_drvdata(cpu_dai);
int slotr, slot_sz;
- regmap_read(sai->regmap, STM_SAI_SLOTR_REGX, &slotr);
+ stm32_sai_sub_reg_rd(sai, STM_SAI_SLOTR_REGX, &slotr);
/*
* If SLOTSZ is set to auto in SLOTR, align slot width on data size
@@ -831,16 +881,16 @@
sai->slots = 2;
/* The number of slots in the audio frame is equal to NBSLOT[3:0] + 1*/
- regmap_update_bits(sai->regmap, STM_SAI_SLOTR_REGX,
- SAI_XSLOTR_NBSLOT_MASK,
- SAI_XSLOTR_NBSLOT_SET((sai->slots - 1)));
+ stm32_sai_sub_reg_up(sai, STM_SAI_SLOTR_REGX,
+ SAI_XSLOTR_NBSLOT_MASK,
+ SAI_XSLOTR_NBSLOT_SET((sai->slots - 1)));
/* Set default slots mask if not already set from DT */
if (!(slotr & SAI_XSLOTR_SLOTEN_MASK)) {
sai->slot_mask = (1 << sai->slots) - 1;
- regmap_update_bits(sai->regmap,
- STM_SAI_SLOTR_REGX, SAI_XSLOTR_SLOTEN_MASK,
- SAI_XSLOTR_SLOTEN_SET(sai->slot_mask));
+ stm32_sai_sub_reg_up(sai,
+ STM_SAI_SLOTR_REGX, SAI_XSLOTR_SLOTEN_MASK,
+ SAI_XSLOTR_SLOTEN_SET(sai->slot_mask));
}
dev_dbg(cpu_dai->dev, "Slots %d, slot width %d\n",
@@ -870,14 +920,14 @@
dev_dbg(cpu_dai->dev, "Frame length %d, frame active %d\n",
sai->fs_length, fs_active);
- regmap_update_bits(sai->regmap, STM_SAI_FRCR_REGX, frcr_mask, frcr);
+ stm32_sai_sub_reg_up(sai, STM_SAI_FRCR_REGX, frcr_mask, frcr);
if ((sai->fmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_LSB) {
offset = sai->slot_width - sai->data_size;
- regmap_update_bits(sai->regmap, STM_SAI_SLOTR_REGX,
- SAI_XSLOTR_FBOFF_MASK,
- SAI_XSLOTR_FBOFF_SET(offset));
+ stm32_sai_sub_reg_up(sai, STM_SAI_SLOTR_REGX,
+ SAI_XSLOTR_FBOFF_MASK,
+ SAI_XSLOTR_FBOFF_SET(offset));
}
}
@@ -994,9 +1044,9 @@
return -EINVAL;
}
- regmap_update_bits(sai->regmap,
- STM_SAI_CR1_REGX,
- SAI_XCR1_OSR, cr1);
+ stm32_sai_sub_reg_up(sai,
+ STM_SAI_CR1_REGX,
+ SAI_XCR1_OSR, cr1);
div = stm32_sai_get_clk_div(sai, sai_clk_rate,
sai->mclk_rate);
@@ -1058,12 +1108,12 @@
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
dev_dbg(cpu_dai->dev, "Enable DMA and SAI\n");
- regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX,
- SAI_XCR1_DMAEN, SAI_XCR1_DMAEN);
+ stm32_sai_sub_reg_up(sai, STM_SAI_CR1_REGX,
+ SAI_XCR1_DMAEN, SAI_XCR1_DMAEN);
/* Enable SAI */
- ret = regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX,
- SAI_XCR1_SAIEN, SAI_XCR1_SAIEN);
+ ret = stm32_sai_sub_reg_up(sai, STM_SAI_CR1_REGX,
+ SAI_XCR1_SAIEN, SAI_XCR1_SAIEN);
if (ret < 0)
dev_err(cpu_dai->dev, "Failed to update CR1 register\n");
break;
@@ -1072,16 +1122,16 @@
case SNDRV_PCM_TRIGGER_STOP:
dev_dbg(cpu_dai->dev, "Disable DMA and SAI\n");
- regmap_update_bits(sai->regmap, STM_SAI_IMR_REGX,
- SAI_XIMR_MASK, 0);
+ stm32_sai_sub_reg_up(sai, STM_SAI_IMR_REGX,
+ SAI_XIMR_MASK, 0);
- regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX,
- SAI_XCR1_SAIEN,
- (unsigned int)~SAI_XCR1_SAIEN);
+ stm32_sai_sub_reg_up(sai, STM_SAI_CR1_REGX,
+ SAI_XCR1_SAIEN,
+ (unsigned int)~SAI_XCR1_SAIEN);
- ret = regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX,
- SAI_XCR1_DMAEN,
- (unsigned int)~SAI_XCR1_DMAEN);
+ ret = stm32_sai_sub_reg_up(sai, STM_SAI_CR1_REGX,
+ SAI_XCR1_DMAEN,
+ (unsigned int)~SAI_XCR1_DMAEN);
if (ret < 0)
dev_err(cpu_dai->dev, "Failed to update CR1 register\n");
@@ -1101,7 +1151,7 @@
struct stm32_sai_sub_data *sai = snd_soc_dai_get_drvdata(cpu_dai);
unsigned long flags;
- regmap_update_bits(sai->regmap, STM_SAI_IMR_REGX, SAI_XIMR_MASK, 0);
+ stm32_sai_sub_reg_up(sai, STM_SAI_IMR_REGX, SAI_XIMR_MASK, 0);
clk_disable_unprepare(sai->sai_ck);
@@ -1169,7 +1219,7 @@
cr1_mask |= SAI_XCR1_SYNCEN_MASK;
cr1 |= SAI_XCR1_SYNCEN_SET(sai->sync);
- return regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, cr1_mask, cr1);
+ return stm32_sai_sub_reg_up(sai, STM_SAI_CR1_REGX, cr1_mask, cr1);
}
static const struct snd_soc_dai_ops stm32_sai_pcm_dai_ops = {
@@ -1322,8 +1372,13 @@
if (STM_SAI_HAS_PDM(sai) && STM_SAI_IS_SUB_A(sai))
sai->regmap_config = &stm32_sai_sub_regmap_config_h7;
- sai->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "sai_ck",
- base, sai->regmap_config);
+ /*
+ * Do not manage peripheral clock through regmap framework as this
+ * can lead to circular locking issue with sai master clock provider.
+ * Manage peripheral clock directly in driver instead.
+ */
+ sai->regmap = devm_regmap_init_mmio(&pdev->dev, base,
+ sai->regmap_config);
if (IS_ERR(sai->regmap)) {
dev_err(&pdev->dev, "Failed to initialize MMIO\n");
return PTR_ERR(sai->regmap);
@@ -1420,6 +1475,10 @@
return PTR_ERR(sai->sai_ck);
}
+ ret = clk_prepare(sai->pdata->pclk);
+ if (ret < 0)
+ return ret;
+
if (STM_SAI_IS_F4(sai->pdata))
return 0;
@@ -1484,20 +1543,31 @@
return ret;
}
- ret = devm_snd_soc_register_component(&pdev->dev, &stm32_component,
- &sai->cpu_dai_drv, 1);
- if (ret)
- return ret;
-
if (STM_SAI_PROTOCOL_IS_SPDIF(sai))
conf = &stm32_sai_pcm_config_spdif;
- ret = devm_snd_dmaengine_pcm_register(&pdev->dev, conf, 0);
+ ret = snd_dmaengine_pcm_register(&pdev->dev, conf, 0);
if (ret) {
dev_err(&pdev->dev, "Could not register pcm dma\n");
return ret;
}
+ ret = snd_soc_register_component(&pdev->dev, &stm32_component,
+ &sai->cpu_dai_drv, 1);
+ if (ret)
+ snd_dmaengine_pcm_unregister(&pdev->dev);
+
+ return ret;
+}
+
+static int stm32_sai_sub_remove(struct platform_device *pdev)
+{
+ struct stm32_sai_sub_data *sai = dev_get_drvdata(&pdev->dev);
+
+ clk_unprepare(sai->pdata->pclk);
+ snd_dmaengine_pcm_unregister(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
+
return 0;
}
@@ -1505,18 +1575,35 @@
static int stm32_sai_sub_suspend(struct device *dev)
{
struct stm32_sai_sub_data *sai = dev_get_drvdata(dev);
+ int ret;
+
+ ret = clk_enable(sai->pdata->pclk);
+ if (ret < 0)
+ return ret;
regcache_cache_only(sai->regmap, true);
regcache_mark_dirty(sai->regmap);
+
+ clk_disable(sai->pdata->pclk);
+
return 0;
}
static int stm32_sai_sub_resume(struct device *dev)
{
struct stm32_sai_sub_data *sai = dev_get_drvdata(dev);
+ int ret;
+
+ ret = clk_enable(sai->pdata->pclk);
+ if (ret < 0)
+ return ret;
regcache_cache_only(sai->regmap, false);
- return regcache_sync(sai->regmap);
+ ret = regcache_sync(sai->regmap);
+
+ clk_disable(sai->pdata->pclk);
+
+ return ret;
}
#endif /* CONFIG_PM_SLEEP */
@@ -1531,6 +1618,7 @@
.pm = &stm32_sai_sub_pm_ops,
},
.probe = stm32_sai_sub_probe,
+ .remove = stm32_sai_sub_remove,
};
module_platform_driver(stm32_sai_sub_driver);
diff --git a/sound/soc/stm/stm32_spdifrx.c b/sound/soc/stm/stm32_spdifrx.c
index cd4b235..9fc2a17 100644
--- a/sound/soc/stm/stm32_spdifrx.c
+++ b/sound/soc/stm/stm32_spdifrx.c
@@ -12,7 +12,6 @@
#include <linux/delay.h>
#include <linux/module.h>
#include <linux/of_platform.h>
-#include <linux/pinctrl/consumer.h>
#include <linux/regmap.h>
#include <linux/reset.h>
@@ -220,6 +219,7 @@
* @slave_config: dma slave channel runtime config pointer
* @phys_addr: SPDIFRX registers physical base address
* @lock: synchronization enabling lock
+ * @irq_lock: prevent race condition with IRQ on stream state
* @cs: channel status buffer
* @ub: user data buffer
* @irq: SPDIFRX interrupt line
@@ -240,6 +240,7 @@
struct dma_slave_config slave_config;
dma_addr_t phys_addr;
spinlock_t lock; /* Sync enabling lock */
+ spinlock_t irq_lock; /* Prevent race condition on stream state */
unsigned char cs[SPDIFRX_CS_BYTES_NB];
unsigned char ub[SPDIFRX_UB_BYTES_NB];
int irq;
@@ -320,6 +321,7 @@
static int stm32_spdifrx_start_sync(struct stm32_spdifrx_data *spdifrx)
{
int cr, cr_mask, imr, ret;
+ unsigned long flags;
/* Enable IRQs */
imr = SPDIFRX_IMR_IFEIE | SPDIFRX_IMR_SYNCDIE | SPDIFRX_IMR_PERRIE;
@@ -327,7 +329,7 @@
if (ret)
return ret;
- spin_lock(&spdifrx->lock);
+ spin_lock_irqsave(&spdifrx->lock, flags);
spdifrx->refcount++;
@@ -360,7 +362,7 @@
"Failed to start synchronization\n");
}
- spin_unlock(&spdifrx->lock);
+ spin_unlock_irqrestore(&spdifrx->lock, flags);
return ret;
}
@@ -368,11 +370,12 @@
static void stm32_spdifrx_stop(struct stm32_spdifrx_data *spdifrx)
{
int cr, cr_mask, reg;
+ unsigned long flags;
- spin_lock(&spdifrx->lock);
+ spin_lock_irqsave(&spdifrx->lock, flags);
if (--spdifrx->refcount) {
- spin_unlock(&spdifrx->lock);
+ spin_unlock_irqrestore(&spdifrx->lock, flags);
return;
}
@@ -391,7 +394,7 @@
regmap_read(spdifrx->regmap, STM32_SPDIFRX_DR, ®);
regmap_read(spdifrx->regmap, STM32_SPDIFRX_CSR, ®);
- spin_unlock(&spdifrx->lock);
+ spin_unlock_irqrestore(&spdifrx->lock, flags);
}
static int stm32_spdifrx_dma_ctrl_register(struct device *dev,
@@ -478,8 +481,6 @@
memset(spdifrx->cs, 0, SPDIFRX_CS_BYTES_NB);
memset(spdifrx->ub, 0, SPDIFRX_UB_BYTES_NB);
- pinctrl_pm_select_default_state(&spdifrx->pdev->dev);
-
ret = stm32_spdifrx_dma_ctrl_start(spdifrx);
if (ret < 0)
return ret;
@@ -511,7 +512,6 @@
end:
clk_disable_unprepare(spdifrx->kclk);
- pinctrl_pm_select_sleep_state(&spdifrx->pdev->dev);
return ret;
}
@@ -663,7 +663,6 @@
static irqreturn_t stm32_spdifrx_isr(int irq, void *devid)
{
struct stm32_spdifrx_data *spdifrx = (struct stm32_spdifrx_data *)devid;
- struct snd_pcm_substream *substream = spdifrx->substream;
struct platform_device *pdev = spdifrx->pdev;
unsigned int cr, mask, sr, imr;
unsigned int flags;
@@ -731,14 +730,19 @@
regmap_update_bits(spdifrx->regmap, STM32_SPDIFRX_CR,
SPDIFRX_CR_SPDIFEN_MASK, cr);
- if (substream)
- snd_pcm_stop(substream, SNDRV_PCM_STATE_DISCONNECTED);
+ spin_lock(&spdifrx->irq_lock);
+ if (spdifrx->substream)
+ snd_pcm_stop(spdifrx->substream,
+ SNDRV_PCM_STATE_DISCONNECTED);
+ spin_unlock(&spdifrx->irq_lock);
return IRQ_HANDLED;
}
- if (err_xrun && substream)
- snd_pcm_stop_xrun(substream);
+ spin_lock(&spdifrx->irq_lock);
+ if (err_xrun && spdifrx->substream)
+ snd_pcm_stop_xrun(spdifrx->substream);
+ spin_unlock(&spdifrx->irq_lock);
return IRQ_HANDLED;
}
@@ -747,9 +751,12 @@
struct snd_soc_dai *cpu_dai)
{
struct stm32_spdifrx_data *spdifrx = snd_soc_dai_get_drvdata(cpu_dai);
+ unsigned long flags;
int ret;
+ spin_lock_irqsave(&spdifrx->irq_lock, flags);
spdifrx->substream = substream;
+ spin_unlock_irqrestore(&spdifrx->irq_lock, flags);
ret = clk_prepare_enable(spdifrx->kclk);
if (ret)
@@ -825,8 +832,12 @@
struct snd_soc_dai *cpu_dai)
{
struct stm32_spdifrx_data *spdifrx = snd_soc_dai_get_drvdata(cpu_dai);
+ unsigned long flags;
+ spin_lock_irqsave(&spdifrx->irq_lock, flags);
spdifrx->substream = NULL;
+ spin_unlock_irqrestore(&spdifrx->irq_lock, flags);
+
clk_disable_unprepare(spdifrx->kclk);
}
@@ -930,6 +941,7 @@
spdifrx->pdev = pdev;
init_completion(&spdifrx->cs_completion);
spin_lock_init(&spdifrx->lock);
+ spin_lock_init(&spdifrx->irq_lock);
platform_set_drvdata(pdev, spdifrx);
@@ -983,6 +995,8 @@
if (idr == SPDIFRX_IPIDR_NUMBER) {
ret = regmap_read(spdifrx->regmap, STM32_SPDIFRX_VERR, &ver);
+ if (ret)
+ goto error;
dev_dbg(&pdev->dev, "SPDIFRX version: %lu.%lu registered\n",
FIELD_GET(SPDIFRX_VERR_MAJ_MASK, ver),
diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c
index ee448d5..c4021d6 100644
--- a/sound/soc/sunxi/sun4i-codec.c
+++ b/sound/soc/sunxi/sun4i-codec.c
@@ -1364,6 +1364,7 @@
return ERR_PTR(-ENOMEM);
card->dev = dev;
+ card->owner = THIS_MODULE;
card->name = "sun4i-codec";
card->dapm_widgets = sun4i_codec_card_dapm_widgets;
card->num_dapm_widgets = ARRAY_SIZE(sun4i_codec_card_dapm_widgets);
@@ -1396,6 +1397,7 @@
return ERR_PTR(-ENOMEM);
card->dev = dev;
+ card->owner = THIS_MODULE;
card->name = "A31 Audio Codec";
card->dapm_widgets = sun6i_codec_card_dapm_widgets;
card->num_dapm_widgets = ARRAY_SIZE(sun6i_codec_card_dapm_widgets);
@@ -1449,6 +1451,7 @@
return ERR_PTR(-ENOMEM);
card->dev = dev;
+ card->owner = THIS_MODULE;
card->name = "A23 Audio Codec";
card->dapm_widgets = sun6i_codec_card_dapm_widgets;
card->num_dapm_widgets = ARRAY_SIZE(sun6i_codec_card_dapm_widgets);
@@ -1487,6 +1490,7 @@
return ERR_PTR(-ENOMEM);
card->dev = dev;
+ card->owner = THIS_MODULE;
card->name = "H3 Audio Codec";
card->dapm_widgets = sun6i_codec_card_dapm_widgets;
card->num_dapm_widgets = ARRAY_SIZE(sun6i_codec_card_dapm_widgets);
@@ -1525,6 +1529,7 @@
return ERR_PTR(-ENOMEM);
card->dev = dev;
+ card->owner = THIS_MODULE;
card->name = "V3s Audio Codec";
card->dapm_widgets = sun6i_codec_card_dapm_widgets;
card->num_dapm_widgets = ARRAY_SIZE(sun6i_codec_card_dapm_widgets);
diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c
index d0a8d58..9655dec 100644
--- a/sound/soc/sunxi/sun4i-i2s.c
+++ b/sound/soc/sunxi/sun4i-i2s.c
@@ -442,11 +442,11 @@
switch (i2s->format & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_DSP_A:
case SND_SOC_DAIFMT_DSP_B:
- case SND_SOC_DAIFMT_LEFT_J:
- case SND_SOC_DAIFMT_RIGHT_J:
lrck_period = params_physical_width(params) * slots;
break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ case SND_SOC_DAIFMT_RIGHT_J:
case SND_SOC_DAIFMT_I2S:
lrck_period = params_physical_width(params);
break;
diff --git a/sound/soc/sunxi/sun8i-codec.c b/sound/soc/sunxi/sun8i-codec.c
index 55798bc..686561d 100644
--- a/sound/soc/sunxi/sun8i-codec.c
+++ b/sound/soc/sunxi/sun8i-codec.c
@@ -80,6 +80,7 @@
#define SUN8I_SYS_SR_CTRL_AIF1_FS_MASK GENMASK(15, 12)
#define SUN8I_SYS_SR_CTRL_AIF2_FS_MASK GENMASK(11, 8)
+#define SUN8I_AIF1CLK_CTRL_AIF1_DATA_FMT_MASK GENMASK(3, 2)
#define SUN8I_AIF1CLK_CTRL_AIF1_WORD_SIZ_MASK GENMASK(5, 4)
#define SUN8I_AIF1CLK_CTRL_AIF1_LRCK_DIV_MASK GENMASK(8, 6)
#define SUN8I_AIF1CLK_CTRL_AIF1_BCLK_DIV_MASK GENMASK(12, 9)
@@ -241,7 +242,7 @@
return -EINVAL;
}
regmap_update_bits(scodec->regmap, SUN8I_AIF1CLK_CTRL,
- BIT(SUN8I_AIF1CLK_CTRL_AIF1_DATA_FMT),
+ SUN8I_AIF1CLK_CTRL_AIF1_DATA_FMT_MASK,
value << SUN8I_AIF1CLK_CTRL_AIF1_DATA_FMT);
return 0;
diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c
index 635eacb..156e3b9 100644
--- a/sound/soc/tegra/tegra30_ahub.c
+++ b/sound/soc/tegra/tegra30_ahub.c
@@ -643,8 +643,10 @@
int ret;
ret = pm_runtime_get_sync(dev);
- if (ret < 0)
+ if (ret < 0) {
+ pm_runtime_put(dev);
return ret;
+ }
ret = regcache_sync(ahub->regmap_ahub);
ret |= regcache_sync(ahub->regmap_apbif);
pm_runtime_put(dev);
diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c
index e6d548f..8894b7c 100644
--- a/sound/soc/tegra/tegra30_i2s.c
+++ b/sound/soc/tegra/tegra30_i2s.c
@@ -538,8 +538,10 @@
int ret;
ret = pm_runtime_get_sync(dev);
- if (ret < 0)
+ if (ret < 0) {
+ pm_runtime_put(dev);
return ret;
+ }
ret = regcache_sync(i2s->regmap);
pm_runtime_put(dev);
diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c
index 9e8b149..a281ceb 100644
--- a/sound/soc/tegra/tegra_alc5632.c
+++ b/sound/soc/tegra/tegra_alc5632.c
@@ -139,6 +139,7 @@
static struct snd_soc_card snd_soc_tegra_alc5632 = {
.name = "tegra-alc5632",
+ .driver_name = "tegra",
.owner = THIS_MODULE,
.dai_link = &tegra_alc5632_dai,
.num_links = 1,
diff --git a/sound/soc/tegra/tegra_max98090.c b/sound/soc/tegra/tegra_max98090.c
index 4954a33..30edd70 100644
--- a/sound/soc/tegra/tegra_max98090.c
+++ b/sound/soc/tegra/tegra_max98090.c
@@ -182,6 +182,7 @@
static struct snd_soc_card snd_soc_tegra_max98090 = {
.name = "tegra-max98090",
+ .driver_name = "tegra",
.owner = THIS_MODULE,
.dai_link = &tegra_max98090_dai,
.num_links = 1,
diff --git a/sound/soc/tegra/tegra_rt5640.c b/sound/soc/tegra/tegra_rt5640.c
index d46915a..3d68a41 100644
--- a/sound/soc/tegra/tegra_rt5640.c
+++ b/sound/soc/tegra/tegra_rt5640.c
@@ -132,6 +132,7 @@
static struct snd_soc_card snd_soc_tegra_rt5640 = {
.name = "tegra-rt5640",
+ .driver_name = "tegra",
.owner = THIS_MODULE,
.dai_link = &tegra_rt5640_dai,
.num_links = 1,
diff --git a/sound/soc/tegra/tegra_rt5677.c b/sound/soc/tegra/tegra_rt5677.c
index 81cb6cc..ae150ad 100644
--- a/sound/soc/tegra/tegra_rt5677.c
+++ b/sound/soc/tegra/tegra_rt5677.c
@@ -175,6 +175,7 @@
static struct snd_soc_card snd_soc_tegra_rt5677 = {
.name = "tegra-rt5677",
+ .driver_name = "tegra",
.owner = THIS_MODULE,
.dai_link = &tegra_rt5677_dai,
.num_links = 1,
diff --git a/sound/soc/tegra/tegra_sgtl5000.c b/sound/soc/tegra/tegra_sgtl5000.c
index e13b81d..fe21d9e 100644
--- a/sound/soc/tegra/tegra_sgtl5000.c
+++ b/sound/soc/tegra/tegra_sgtl5000.c
@@ -97,6 +97,7 @@
static struct snd_soc_card snd_soc_tegra_sgtl5000 = {
.name = "tegra-sgtl5000",
+ .driver_name = "tegra",
.owner = THIS_MODULE,
.dai_link = &tegra_sgtl5000_dai,
.num_links = 1,
diff --git a/sound/soc/tegra/tegra_wm8753.c b/sound/soc/tegra/tegra_wm8753.c
index f6dd790..a2362a2 100644
--- a/sound/soc/tegra/tegra_wm8753.c
+++ b/sound/soc/tegra/tegra_wm8753.c
@@ -101,6 +101,7 @@
static struct snd_soc_card snd_soc_tegra_wm8753 = {
.name = "tegra-wm8753",
+ .driver_name = "tegra",
.owner = THIS_MODULE,
.dai_link = &tegra_wm8753_dai,
.num_links = 1,
diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c
index 6211dfd..08bcc94 100644
--- a/sound/soc/tegra/tegra_wm8903.c
+++ b/sound/soc/tegra/tegra_wm8903.c
@@ -159,6 +159,7 @@
struct snd_soc_component *component = codec_dai->component;
struct snd_soc_card *card = rtd->card;
struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card);
+ int shrt = 0;
if (gpio_is_valid(machine->gpio_hp_det)) {
tegra_wm8903_hp_jack_gpio.gpio = machine->gpio_hp_det;
@@ -171,12 +172,15 @@
&tegra_wm8903_hp_jack_gpio);
}
+ if (of_property_read_bool(card->dev->of_node, "nvidia,headset"))
+ shrt = SND_JACK_MICROPHONE;
+
snd_soc_card_jack_new(rtd->card, "Mic Jack", SND_JACK_MICROPHONE,
&tegra_wm8903_mic_jack,
tegra_wm8903_mic_jack_pins,
ARRAY_SIZE(tegra_wm8903_mic_jack_pins));
wm8903_mic_detect(component, &tegra_wm8903_mic_jack, SND_JACK_MICROPHONE,
- 0);
+ shrt);
snd_soc_dapm_force_enable_pin(&card->dapm, "MICBIAS");
@@ -213,6 +217,7 @@
static struct snd_soc_card snd_soc_tegra_wm8903 = {
.name = "tegra-wm8903",
+ .driver_name = "tegra",
.owner = THIS_MODULE,
.dai_link = &tegra_wm8903_dai,
.num_links = 1,
diff --git a/sound/soc/tegra/tegra_wm9712.c b/sound/soc/tegra/tegra_wm9712.c
index b85bd9f..232eac5 100644
--- a/sound/soc/tegra/tegra_wm9712.c
+++ b/sound/soc/tegra/tegra_wm9712.c
@@ -54,6 +54,7 @@
static struct snd_soc_card snd_soc_tegra_wm9712 = {
.name = "tegra-wm9712",
+ .driver_name = "tegra",
.owner = THIS_MODULE,
.dai_link = &tegra_wm9712_dai,
.num_links = 1,
diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c
index 3f67ddd..5086bc2 100644
--- a/sound/soc/tegra/trimslice.c
+++ b/sound/soc/tegra/trimslice.c
@@ -94,6 +94,7 @@
static struct snd_soc_card snd_soc_trimslice = {
.name = "tegra-trimslice",
+ .driver_name = "tegra",
.owner = THIS_MODULE,
.dai_link = &trimslice_tlv320aic23_dai,
.num_links = 1,
diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c
index 7aa3c32..0541071 100644
--- a/sound/soc/ti/davinci-mcasp.c
+++ b/sound/soc/ti/davinci-mcasp.c
@@ -1875,8 +1875,10 @@
PTR_ERR(chan));
return PTR_ERR(chan);
}
- if (WARN_ON(!chan->device || !chan->device->dev))
+ if (WARN_ON(!chan->device || !chan->device->dev)) {
+ dma_release_channel(chan);
return -EINVAL;
+ }
if (chan->device->dev->of_node)
ret = of_property_read_string(chan->device->dev->of_node,
diff --git a/sound/soc/ti/omap-mcbsp.c b/sound/soc/ti/omap-mcbsp.c
index 26b503b..3273b31 100644
--- a/sound/soc/ti/omap-mcbsp.c
+++ b/sound/soc/ti/omap-mcbsp.c
@@ -686,7 +686,7 @@
mcbsp->dma_data[1].addr = omap_mcbsp_dma_reg_params(mcbsp,
SNDRV_PCM_STREAM_CAPTURE);
- mcbsp->fclk = clk_get(&pdev->dev, "fck");
+ mcbsp->fclk = devm_clk_get(&pdev->dev, "fck");
if (IS_ERR(mcbsp->fclk)) {
ret = PTR_ERR(mcbsp->fclk);
dev_err(mcbsp->dev, "unable to get fck: %d\n", ret);
@@ -711,7 +711,7 @@
if (ret) {
dev_err(mcbsp->dev,
"Unable to create additional controls\n");
- goto err_thres;
+ return ret;
}
}
@@ -724,8 +724,6 @@
err_st:
if (mcbsp->pdata->buffer_size)
sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group);
-err_thres:
- clk_put(mcbsp->fclk);
return ret;
}
@@ -1442,8 +1440,6 @@
omap_mcbsp_st_cleanup(pdev);
- clk_put(mcbsp->fclk);
-
return 0;
}
diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c
index 2873e8e..cdae119 100644
--- a/sound/soc/ux500/mop500.c
+++ b/sound/soc/ux500/mop500.c
@@ -63,10 +63,11 @@
{
int i;
- for (i = 0; i < 2; i++) {
+ for (i = 0; i < 2; i++)
of_node_put(mop500_dai_links[i].cpus->of_node);
- of_node_put(mop500_dai_links[i].codecs->of_node);
- }
+
+ /* Both links use the same codec, which is refcounted only once */
+ of_node_put(mop500_dai_links[0].codecs->of_node);
}
static int mop500_of_probe(struct platform_device *pdev,
@@ -81,7 +82,9 @@
if (!(msp_np[0] && msp_np[1] && codec_np)) {
dev_err(&pdev->dev, "Phandle missing or invalid\n");
- mop500_of_node_put();
+ for (i = 0; i < 2; i++)
+ of_node_put(msp_np[i]);
+ of_node_put(codec_np);
return -EINVAL;
}
diff --git a/sound/soc/xilinx/xlnx_formatter_pcm.c b/sound/soc/xilinx/xlnx_formatter_pcm.c
index 48970ef..1f15c11 100644
--- a/sound/soc/xilinx/xlnx_formatter_pcm.c
+++ b/sound/soc/xilinx/xlnx_formatter_pcm.c
@@ -461,8 +461,8 @@
stream_data->buffer_size = size;
- low = lower_32_bits(substream->dma_buffer.addr);
- high = upper_32_bits(substream->dma_buffer.addr);
+ low = lower_32_bits(runtime->dma_addr);
+ high = upper_32_bits(runtime->dma_addr);
writel(low, stream_data->mmio + XLNX_AUD_BUFF_ADDR_LSB);
writel(high, stream_data->mmio + XLNX_AUD_BUFF_ADDR_MSB);
diff --git a/sound/usb/card.c b/sound/usb/card.c
index db91dc7..c2dd18c 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -176,9 +176,8 @@
ctrlif, interface);
return -EINVAL;
}
- usb_driver_claim_interface(&usb_audio_driver, iface, (void *)-1L);
-
- return 0;
+ return usb_driver_claim_interface(&usb_audio_driver, iface,
+ USB_AUDIO_IFACE_UNUSED);
}
if ((altsd->bInterfaceClass != USB_CLASS_AUDIO &&
@@ -198,7 +197,8 @@
if (! snd_usb_parse_audio_interface(chip, interface)) {
usb_set_interface(dev, interface, 0); /* reset the current interface */
- usb_driver_claim_interface(&usb_audio_driver, iface, (void *)-1L);
+ return usb_driver_claim_interface(&usb_audio_driver, iface,
+ USB_AUDIO_IFACE_UNUSED);
}
return 0;
@@ -597,6 +597,10 @@
}
}
if (! chip) {
+ err = snd_usb_apply_boot_quirk_once(dev, intf, quirk, id);
+ if (err < 0)
+ goto __error;
+
/* it's a fresh one.
* now look for an empty slot and create a new card instance
*/
@@ -655,10 +659,14 @@
goto __error;
}
- /* we are allowed to call snd_card_register() many times */
- err = snd_card_register(chip->card);
- if (err < 0)
- goto __error;
+ /* we are allowed to call snd_card_register() many times, but first
+ * check to see if a device needs to skip it or do anything special
+ */
+ if (!snd_usb_registration_quirk(chip, ifnum)) {
+ err = snd_card_register(chip->card);
+ if (err < 0)
+ goto __error;
+ }
if (quirk && quirk->shares_media_device) {
/* don't want to fail when snd_media_device_create() fails */
@@ -695,7 +703,7 @@
struct snd_card *card;
struct list_head *p;
- if (chip == (void *)-1L)
+ if (chip == USB_AUDIO_IFACE_UNUSED)
return;
card = chip->card;
@@ -803,12 +811,9 @@
struct usb_mixer_interface *mixer;
struct list_head *p;
- if (chip == (void *)-1L)
+ if (chip == USB_AUDIO_IFACE_UNUSED)
return 0;
- chip->autosuspended = !!PMSG_IS_AUTO(message);
- if (!chip->autosuspended)
- snd_power_change_state(chip->card, SNDRV_CTL_POWER_D3hot);
if (!chip->num_suspended_intf++) {
list_for_each_entry(as, &chip->pcm_list, list) {
snd_usb_pcm_suspend(as);
@@ -821,6 +826,11 @@
snd_usb_mixer_suspend(mixer);
}
+ if (!PMSG_IS_AUTO(message) && !chip->system_suspend) {
+ snd_power_change_state(chip->card, SNDRV_CTL_POWER_D3hot);
+ chip->system_suspend = chip->num_suspended_intf;
+ }
+
return 0;
}
@@ -832,12 +842,12 @@
struct list_head *p;
int err = 0;
- if (chip == (void *)-1L)
- return 0;
- if (--chip->num_suspended_intf)
+ if (chip == USB_AUDIO_IFACE_UNUSED)
return 0;
atomic_inc(&chip->active); /* avoid autopm */
+ if (chip->num_suspended_intf > 1)
+ goto out;
list_for_each_entry(as, &chip->pcm_list, list) {
err = snd_usb_pcm_resume(as);
@@ -859,9 +869,12 @@
snd_usbmidi_resume(p);
}
- if (!chip->autosuspended)
+ out:
+ if (chip->num_suspended_intf == chip->system_suspend) {
snd_power_change_state(chip->card, SNDRV_CTL_POWER_D0);
- chip->autosuspended = 0;
+ chip->system_suspend = 0;
+ }
+ chip->num_suspended_intf--;
err_out:
atomic_dec(&chip->active); /* allow autopm after this point */
diff --git a/sound/usb/card.h b/sound/usb/card.h
index 2991b99..d8ec5ca 100644
--- a/sound/usb/card.h
+++ b/sound/usb/card.h
@@ -104,6 +104,7 @@
int iface, altsetting;
int skip_packets; /* quirks for devices to ignore the first n packets
in a stream */
+ bool is_implicit_feedback; /* This endpoint is used as implicit feedback */
spinlock_t lock;
struct list_head list;
@@ -132,6 +133,7 @@
unsigned int tx_length_quirk:1; /* add length specifier to transfers */
unsigned int fmt_type; /* USB audio format type (1-3) */
unsigned int pkt_offset_adj; /* Bytes to drop from beginning of packets (for non-compliant devices) */
+ unsigned int stream_offset_adj; /* Bytes to drop from beginning of stream (for non-compliant devices) */
unsigned int running: 1; /* running status */
@@ -145,6 +147,7 @@
struct snd_usb_endpoint *sync_endpoint;
unsigned long flags;
bool need_setup_ep; /* (re)configure EP at prepare? */
+ bool need_setup_fmt; /* (re)configure fmt after resume? */
unsigned int speed; /* USB_SPEED_XXX */
u64 formats; /* format bitmasks (all or'ed) */
diff --git a/sound/usb/clock.c b/sound/usb/clock.c
index 6b8c14f..3d1c0ec 100644
--- a/sound/usb/clock.c
+++ b/sound/usb/clock.c
@@ -151,8 +151,71 @@
return ret;
}
+static bool uac_clock_source_is_valid_quirk(struct snd_usb_audio *chip,
+ struct audioformat *fmt,
+ int source_id)
+{
+ bool ret = false;
+ int count;
+ unsigned char data;
+ struct usb_device *dev = chip->dev;
+
+ if (fmt->protocol == UAC_VERSION_2) {
+ struct uac_clock_source_descriptor *cs_desc =
+ snd_usb_find_clock_source(chip->ctrl_intf, source_id);
+
+ if (!cs_desc)
+ return false;
+
+ /*
+ * Assume the clock is valid if clock source supports only one
+ * single sample rate, the terminal is connected directly to it
+ * (there is no clock selector) and clock type is internal.
+ * This is to deal with some Denon DJ controllers that always
+ * reports that clock is invalid.
+ */
+ if (fmt->nr_rates == 1 &&
+ (fmt->clock & 0xff) == cs_desc->bClockID &&
+ (cs_desc->bmAttributes & 0x3) !=
+ UAC_CLOCK_SOURCE_TYPE_EXT)
+ return true;
+ }
+
+ /*
+ * MOTU MicroBook IIc
+ * Sample rate changes takes more than 2 seconds for this device. Clock
+ * validity request returns false during that period.
+ */
+ if (chip->usb_id == USB_ID(0x07fd, 0x0004)) {
+ count = 0;
+
+ while ((!ret) && (count < 50)) {
+ int err;
+
+ msleep(100);
+
+ err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR,
+ USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
+ UAC2_CS_CONTROL_CLOCK_VALID << 8,
+ snd_usb_ctrl_intf(chip) | (source_id << 8),
+ &data, sizeof(data));
+ if (err < 0) {
+ dev_warn(&dev->dev,
+ "%s(): cannot get clock validity for id %d\n",
+ __func__, source_id);
+ return false;
+ }
+
+ ret = !!data;
+ count++;
+ }
+ }
+
+ return ret;
+}
+
static bool uac_clock_source_is_valid(struct snd_usb_audio *chip,
- int protocol,
+ struct audioformat *fmt,
int source_id)
{
int err;
@@ -160,26 +223,26 @@
struct usb_device *dev = chip->dev;
u32 bmControls;
- if (protocol == UAC_VERSION_3) {
+ if (fmt->protocol == UAC_VERSION_3) {
struct uac3_clock_source_descriptor *cs_desc =
snd_usb_find_clock_source_v3(chip->ctrl_intf, source_id);
if (!cs_desc)
- return 0;
+ return false;
bmControls = le32_to_cpu(cs_desc->bmControls);
} else { /* UAC_VERSION_1/2 */
struct uac_clock_source_descriptor *cs_desc =
snd_usb_find_clock_source(chip->ctrl_intf, source_id);
if (!cs_desc)
- return 0;
+ return false;
bmControls = cs_desc->bmControls;
}
/* If a clock source can't tell us whether it's valid, we assume it is */
if (!uac_v2v3_control_is_readable(bmControls,
UAC2_CS_CONTROL_CLOCK_VALID))
- return 1;
+ return true;
err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR,
USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
@@ -191,13 +254,17 @@
dev_warn(&dev->dev,
"%s(): cannot get clock validity for id %d\n",
__func__, source_id);
- return 0;
+ return false;
}
- return !!data;
+ if (data)
+ return true;
+ else
+ return uac_clock_source_is_valid_quirk(chip, fmt, source_id);
}
-static int __uac_clock_find_source(struct snd_usb_audio *chip, int entity_id,
+static int __uac_clock_find_source(struct snd_usb_audio *chip,
+ struct audioformat *fmt, int entity_id,
unsigned long *visited, bool validate)
{
struct uac_clock_source_descriptor *source;
@@ -217,7 +284,7 @@
source = snd_usb_find_clock_source(chip->ctrl_intf, entity_id);
if (source) {
entity_id = source->bClockID;
- if (validate && !uac_clock_source_is_valid(chip, UAC_VERSION_2,
+ if (validate && !uac_clock_source_is_valid(chip, fmt,
entity_id)) {
usb_audio_err(chip,
"clock source %d is not valid, cannot use\n",
@@ -229,7 +296,7 @@
selector = snd_usb_find_clock_selector(chip->ctrl_intf, entity_id);
if (selector) {
- int ret, i, cur;
+ int ret, i, cur, err;
/* the entity ID we are looking for is a selector.
* find out what it currently selects */
@@ -248,20 +315,32 @@
}
cur = ret;
- ret = __uac_clock_find_source(chip, selector->baCSourceID[ret - 1],
- visited, validate);
+ ret = __uac_clock_find_source(chip, fmt,
+ selector->baCSourceID[ret - 1],
+ visited, validate);
+ if (ret > 0) {
+ /*
+ * For Samsung USBC Headset (AKG), setting clock selector again
+ * will result in incorrect default clock setting problems
+ */
+ if (chip->usb_id == USB_ID(0x04e8, 0xa051))
+ return ret;
+ err = uac_clock_selector_set_val(chip, entity_id, cur);
+ if (err < 0)
+ return err;
+ }
+
if (!validate || ret > 0 || !chip->autoclock)
return ret;
/* The current clock source is invalid, try others. */
for (i = 1; i <= selector->bNrInPins; i++) {
- int err;
-
if (i == cur)
continue;
- ret = __uac_clock_find_source(chip, selector->baCSourceID[i - 1],
- visited, true);
+ ret = __uac_clock_find_source(chip, fmt,
+ selector->baCSourceID[i - 1],
+ visited, true);
if (ret < 0)
continue;
@@ -281,14 +360,16 @@
/* FIXME: multipliers only act as pass-thru element for now */
multiplier = snd_usb_find_clock_multiplier(chip->ctrl_intf, entity_id);
if (multiplier)
- return __uac_clock_find_source(chip, multiplier->bCSourceID,
- visited, validate);
+ return __uac_clock_find_source(chip, fmt,
+ multiplier->bCSourceID,
+ visited, validate);
return -EINVAL;
}
-static int __uac3_clock_find_source(struct snd_usb_audio *chip, int entity_id,
- unsigned long *visited, bool validate)
+static int __uac3_clock_find_source(struct snd_usb_audio *chip,
+ struct audioformat *fmt, int entity_id,
+ unsigned long *visited, bool validate)
{
struct uac3_clock_source_descriptor *source;
struct uac3_clock_selector_descriptor *selector;
@@ -307,7 +388,7 @@
source = snd_usb_find_clock_source_v3(chip->ctrl_intf, entity_id);
if (source) {
entity_id = source->bClockID;
- if (validate && !uac_clock_source_is_valid(chip, UAC_VERSION_3,
+ if (validate && !uac_clock_source_is_valid(chip, fmt,
entity_id)) {
usb_audio_err(chip,
"clock source %d is not valid, cannot use\n",
@@ -319,7 +400,7 @@
selector = snd_usb_find_clock_selector_v3(chip->ctrl_intf, entity_id);
if (selector) {
- int ret, i, cur;
+ int ret, i, cur, err;
/* the entity ID we are looking for is a selector.
* find out what it currently selects */
@@ -338,8 +419,15 @@
}
cur = ret;
- ret = __uac3_clock_find_source(chip, selector->baCSourceID[ret - 1],
+ ret = __uac3_clock_find_source(chip, fmt,
+ selector->baCSourceID[ret - 1],
visited, validate);
+ if (ret > 0) {
+ err = uac_clock_selector_set_val(chip, entity_id, cur);
+ if (err < 0)
+ return err;
+ }
+
if (!validate || ret > 0 || !chip->autoclock)
return ret;
@@ -350,8 +438,9 @@
if (i == cur)
continue;
- ret = __uac3_clock_find_source(chip, selector->baCSourceID[i - 1],
- visited, true);
+ ret = __uac3_clock_find_source(chip, fmt,
+ selector->baCSourceID[i - 1],
+ visited, true);
if (ret < 0)
continue;
@@ -372,7 +461,8 @@
multiplier = snd_usb_find_clock_multiplier_v3(chip->ctrl_intf,
entity_id);
if (multiplier)
- return __uac3_clock_find_source(chip, multiplier->bCSourceID,
+ return __uac3_clock_find_source(chip, fmt,
+ multiplier->bCSourceID,
visited, validate);
return -EINVAL;
@@ -389,18 +479,18 @@
*
* Returns the clock source UnitID (>=0) on success, or an error.
*/
-int snd_usb_clock_find_source(struct snd_usb_audio *chip, int protocol,
- int entity_id, bool validate)
+int snd_usb_clock_find_source(struct snd_usb_audio *chip,
+ struct audioformat *fmt, bool validate)
{
DECLARE_BITMAP(visited, 256);
memset(visited, 0, sizeof(visited));
- switch (protocol) {
+ switch (fmt->protocol) {
case UAC_VERSION_2:
- return __uac_clock_find_source(chip, entity_id, visited,
+ return __uac_clock_find_source(chip, fmt, fmt->clock, visited,
validate);
case UAC_VERSION_3:
- return __uac3_clock_find_source(chip, entity_id, visited,
+ return __uac3_clock_find_source(chip, fmt, fmt->clock, visited,
validate);
default:
return -EINVAL;
@@ -457,6 +547,12 @@
}
crate = data[0] | (data[1] << 8) | (data[2] << 16);
+ if (!crate) {
+ dev_info(&dev->dev, "failed to read current rate; disabling the check\n");
+ chip->sample_rate_read_error = 3; /* three strikes, see above */
+ return 0;
+ }
+
if (crate != rate) {
dev_warn(&dev->dev, "current rate %d is different from the runtime rate %d\n", crate, rate);
// runtime->rate = crate;
@@ -501,8 +597,7 @@
* automatic clock selection if the current clock is not
* valid.
*/
- clock = snd_usb_clock_find_source(chip, fmt->protocol,
- fmt->clock, true);
+ clock = snd_usb_clock_find_source(chip, fmt, true);
if (clock < 0) {
/* We did not find a valid clock, but that might be
* because the current sample rate does not match an
@@ -510,8 +605,7 @@
* and we will do another validation after setting the
* rate.
*/
- clock = snd_usb_clock_find_source(chip, fmt->protocol,
- fmt->clock, false);
+ clock = snd_usb_clock_find_source(chip, fmt, false);
if (clock < 0)
return clock;
}
@@ -577,7 +671,7 @@
validation:
/* validate clock after rate change */
- if (!uac_clock_source_is_valid(chip, fmt->protocol, clock))
+ if (!uac_clock_source_is_valid(chip, fmt, clock))
return -ENXIO;
return 0;
}
diff --git a/sound/usb/clock.h b/sound/usb/clock.h
index 076e31b..68df0fb 100644
--- a/sound/usb/clock.h
+++ b/sound/usb/clock.h
@@ -6,7 +6,7 @@
struct usb_host_interface *alts,
struct audioformat *fmt, int rate);
-int snd_usb_clock_find_source(struct snd_usb_audio *chip, int protocol,
- int entity_id, bool validate);
+int snd_usb_clock_find_source(struct snd_usb_audio *chip,
+ struct audioformat *fmt, bool validate);
#endif /* __USBAUDIO_CLOCK_H */
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index 4a9a2f6..87cc249 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -321,17 +321,17 @@
ep->next_packet_read_pos %= MAX_URBS;
/* take URB out of FIFO */
- if (!list_empty(&ep->ready_playback_urbs))
+ if (!list_empty(&ep->ready_playback_urbs)) {
ctx = list_first_entry(&ep->ready_playback_urbs,
struct snd_urb_ctx, ready_list);
+ list_del_init(&ctx->ready_list);
+ }
}
spin_unlock_irqrestore(&ep->lock, flags);
if (ctx == NULL)
return;
- list_del_init(&ctx->ready_list);
-
/* copy over the length information */
for (i = 0; i < packet->packets; i++)
ctx->packet_size[i] = packet->packet_size[i];
@@ -497,6 +497,8 @@
list_add_tail(&ep->list, &chip->ep_list);
+ ep->is_implicit_feedback = 0;
+
__exit_unlock:
mutex_unlock(&chip->mutex);
@@ -597,6 +599,178 @@
}
/*
+ * Check data endpoint for format differences
+ */
+static bool check_ep_params(struct snd_usb_endpoint *ep,
+ snd_pcm_format_t pcm_format,
+ unsigned int channels,
+ unsigned int period_bytes,
+ unsigned int frames_per_period,
+ unsigned int periods_per_buffer,
+ struct audioformat *fmt,
+ struct snd_usb_endpoint *sync_ep)
+{
+ unsigned int maxsize, minsize, packs_per_ms, max_packs_per_urb;
+ unsigned int max_packs_per_period, urbs_per_period, urb_packs;
+ unsigned int max_urbs;
+ int frame_bits = snd_pcm_format_physical_width(pcm_format) * channels;
+ int tx_length_quirk = (ep->chip->tx_length_quirk &&
+ usb_pipeout(ep->pipe));
+ bool ret = 1;
+
+ if (pcm_format == SNDRV_PCM_FORMAT_DSD_U16_LE && fmt->dsd_dop) {
+ /*
+ * When operating in DSD DOP mode, the size of a sample frame
+ * in hardware differs from the actual physical format width
+ * because we need to make room for the DOP markers.
+ */
+ frame_bits += channels << 3;
+ }
+
+ ret = ret && (ep->datainterval == fmt->datainterval);
+ ret = ret && (ep->stride == frame_bits >> 3);
+
+ switch (pcm_format) {
+ case SNDRV_PCM_FORMAT_U8:
+ ret = ret && (ep->silence_value == 0x80);
+ break;
+ case SNDRV_PCM_FORMAT_DSD_U8:
+ case SNDRV_PCM_FORMAT_DSD_U16_LE:
+ case SNDRV_PCM_FORMAT_DSD_U32_LE:
+ case SNDRV_PCM_FORMAT_DSD_U16_BE:
+ case SNDRV_PCM_FORMAT_DSD_U32_BE:
+ ret = ret && (ep->silence_value == 0x69);
+ break;
+ default:
+ ret = ret && (ep->silence_value == 0);
+ }
+
+ /* assume max. frequency is 50% higher than nominal */
+ ret = ret && (ep->freqmax == ep->freqn + (ep->freqn >> 1));
+ /* Round up freqmax to nearest integer in order to calculate maximum
+ * packet size, which must represent a whole number of frames.
+ * This is accomplished by adding 0x0.ffff before converting the
+ * Q16.16 format into integer.
+ * In order to accurately calculate the maximum packet size when
+ * the data interval is more than 1 (i.e. ep->datainterval > 0),
+ * multiply by the data interval prior to rounding. For instance,
+ * a freqmax of 41 kHz will result in a max packet size of 6 (5.125)
+ * frames with a data interval of 1, but 11 (10.25) frames with a
+ * data interval of 2.
+ * (ep->freqmax << ep->datainterval overflows at 8.192 MHz for the
+ * maximum datainterval value of 3, at USB full speed, higher for
+ * USB high speed, noting that ep->freqmax is in units of
+ * frames per packet in Q16.16 format.)
+ */
+ maxsize = (((ep->freqmax << ep->datainterval) + 0xffff) >> 16) *
+ (frame_bits >> 3);
+ if (tx_length_quirk)
+ maxsize += sizeof(__le32); /* Space for length descriptor */
+ /* but wMaxPacketSize might reduce this */
+ if (ep->maxpacksize && ep->maxpacksize < maxsize) {
+ /* whatever fits into a max. size packet */
+ unsigned int data_maxsize = maxsize = ep->maxpacksize;
+
+ if (tx_length_quirk)
+ /* Need to remove the length descriptor to calc freq */
+ data_maxsize -= sizeof(__le32);
+ ret = ret && (ep->freqmax == (data_maxsize / (frame_bits >> 3))
+ << (16 - ep->datainterval));
+ }
+
+ if (ep->fill_max)
+ ret = ret && (ep->curpacksize == ep->maxpacksize);
+ else
+ ret = ret && (ep->curpacksize == maxsize);
+
+ if (snd_usb_get_speed(ep->chip->dev) != USB_SPEED_FULL) {
+ packs_per_ms = 8 >> ep->datainterval;
+ max_packs_per_urb = MAX_PACKS_HS;
+ } else {
+ packs_per_ms = 1;
+ max_packs_per_urb = MAX_PACKS;
+ }
+ if (sync_ep && !snd_usb_endpoint_implicit_feedback_sink(ep))
+ max_packs_per_urb = min(max_packs_per_urb,
+ 1U << sync_ep->syncinterval);
+ max_packs_per_urb = max(1u, max_packs_per_urb >> ep->datainterval);
+
+ /*
+ * Capture endpoints need to use small URBs because there's no way
+ * to tell in advance where the next period will end, and we don't
+ * want the next URB to complete much after the period ends.
+ *
+ * Playback endpoints with implicit sync much use the same parameters
+ * as their corresponding capture endpoint.
+ */
+ if (usb_pipein(ep->pipe) ||
+ snd_usb_endpoint_implicit_feedback_sink(ep)) {
+
+ urb_packs = packs_per_ms;
+ /*
+ * Wireless devices can poll at a max rate of once per 4ms.
+ * For dataintervals less than 5, increase the packet count to
+ * allow the host controller to use bursting to fill in the
+ * gaps.
+ */
+ if (snd_usb_get_speed(ep->chip->dev) == USB_SPEED_WIRELESS) {
+ int interval = ep->datainterval;
+
+ while (interval < 5) {
+ urb_packs <<= 1;
+ ++interval;
+ }
+ }
+ /* make capture URBs <= 1 ms and smaller than a period */
+ urb_packs = min(max_packs_per_urb, urb_packs);
+ while (urb_packs > 1 && urb_packs * maxsize >= period_bytes)
+ urb_packs >>= 1;
+ ret = ret && (ep->nurbs == MAX_URBS);
+
+ /*
+ * Playback endpoints without implicit sync are adjusted so that
+ * a period fits as evenly as possible in the smallest number of
+ * URBs. The total number of URBs is adjusted to the size of the
+ * ALSA buffer, subject to the MAX_URBS and MAX_QUEUE limits.
+ */
+ } else {
+ /* determine how small a packet can be */
+ minsize = (ep->freqn >> (16 - ep->datainterval)) *
+ (frame_bits >> 3);
+ /* with sync from device, assume it can be 12% lower */
+ if (sync_ep)
+ minsize -= minsize >> 3;
+ minsize = max(minsize, 1u);
+
+ /* how many packets will contain an entire ALSA period? */
+ max_packs_per_period = DIV_ROUND_UP(period_bytes, minsize);
+
+ /* how many URBs will contain a period? */
+ urbs_per_period = DIV_ROUND_UP(max_packs_per_period,
+ max_packs_per_urb);
+ /* how many packets are needed in each URB? */
+ urb_packs = DIV_ROUND_UP(max_packs_per_period, urbs_per_period);
+
+ /* limit the number of frames in a single URB */
+ ret = ret && (ep->max_urb_frames ==
+ DIV_ROUND_UP(frames_per_period, urbs_per_period));
+
+ /* try to use enough URBs to contain an entire ALSA buffer */
+ max_urbs = min((unsigned) MAX_URBS,
+ MAX_QUEUE * packs_per_ms / urb_packs);
+ ret = ret && (ep->nurbs == min(max_urbs,
+ urbs_per_period * periods_per_buffer));
+ }
+
+ ret = ret && (ep->datainterval == fmt->datainterval);
+ ret = ret && (ep->maxpacksize == fmt->maxpacksize);
+ ret = ret &&
+ (ep->fill_max == !!(fmt->attributes & UAC_EP_CS_ATTR_FILL_MAX));
+
+ return ret;
+}
+
+/*
* configure a data endpoint
*/
static int data_ep_set_params(struct snd_usb_endpoint *ep,
@@ -861,10 +1035,23 @@
int err;
if (ep->use_count != 0) {
- usb_audio_warn(ep->chip,
- "Unable to change format on ep #%x: already in use\n",
- ep->ep_num);
- return -EBUSY;
+ bool check = ep->is_implicit_feedback &&
+ check_ep_params(ep, pcm_format,
+ channels, period_bytes,
+ period_frames, buffer_periods,
+ fmt, sync_ep);
+
+ if (!check) {
+ usb_audio_warn(ep->chip,
+ "Unable to change format on ep #%x: already in use\n",
+ ep->ep_num);
+ return -EBUSY;
+ }
+
+ usb_audio_dbg(ep->chip,
+ "Ep #%x already in use as implicit feedback but format not changed\n",
+ ep->ep_num);
+ return 0;
}
/* release old buffers, if any */
diff --git a/sound/usb/format.c b/sound/usb/format.c
index d79db71..84b66f7 100644
--- a/sound/usb/format.c
+++ b/sound/usb/format.c
@@ -40,6 +40,8 @@
case UAC_VERSION_1:
default: {
struct uac_format_type_i_discrete_descriptor *fmt = _fmt;
+ if (format >= 64)
+ return 0; /* invalid format */
sample_width = fmt->bBitResolution;
sample_bytes = fmt->bSubframeSize;
format = 1ULL << format;
@@ -193,9 +195,11 @@
continue;
/* C-Media CM6501 mislabels its 96 kHz altsetting */
/* Terratec Aureon 7.1 USB C-Media 6206, too */
+ /* Ozone Z90 USB C-Media, too */
if (rate == 48000 && nr_rates == 1 &&
(chip->usb_id == USB_ID(0x0d8c, 0x0201) ||
chip->usb_id == USB_ID(0x0d8c, 0x0102) ||
+ chip->usb_id == USB_ID(0x0d8c, 0x0078) ||
chip->usb_id == USB_ID(0x0ccd, 0x00b1)) &&
fp->altsetting == 5 && fp->maxpacksize == 392)
rate = 96000;
@@ -227,6 +231,52 @@
}
/*
+ * Many Focusrite devices supports a limited set of sampling rates per
+ * altsetting. Maximum rate is exposed in the last 4 bytes of Format Type
+ * descriptor which has a non-standard bLength = 10.
+ */
+static bool focusrite_valid_sample_rate(struct snd_usb_audio *chip,
+ struct audioformat *fp,
+ unsigned int rate)
+{
+ struct usb_interface *iface;
+ struct usb_host_interface *alts;
+ unsigned char *fmt;
+ unsigned int max_rate;
+
+ iface = usb_ifnum_to_if(chip->dev, fp->iface);
+ if (!iface)
+ return true;
+
+ alts = &iface->altsetting[fp->altset_idx];
+ fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen,
+ NULL, UAC_FORMAT_TYPE);
+ if (!fmt)
+ return true;
+
+ if (fmt[0] == 10) { /* bLength */
+ max_rate = combine_quad(&fmt[6]);
+
+ /* Validate max rate */
+ if (max_rate != 48000 &&
+ max_rate != 96000 &&
+ max_rate != 192000 &&
+ max_rate != 384000) {
+
+ usb_audio_info(chip,
+ "%u:%d : unexpected max rate: %u\n",
+ fp->iface, fp->altsetting, max_rate);
+
+ return true;
+ }
+
+ return rate <= max_rate;
+ }
+
+ return true;
+}
+
+/*
* Helper function to walk the array of sample rate triplets reported by
* the device. The problem is that we need to parse whole array first to
* get to know how many sample rates we have to expect.
@@ -262,6 +312,11 @@
}
for (rate = min; rate <= max; rate += res) {
+ /* Filter out invalid rates on Focusrite devices */
+ if (USB_ID_VENDOR(chip->usb_id) == 0x1235 &&
+ !focusrite_valid_sample_rate(chip, fp, rate))
+ goto skip_rate;
+
if (fp->rate_table)
fp->rate_table[nr_rates] = rate;
if (!fp->rate_min || rate < fp->rate_min)
@@ -276,6 +331,7 @@
break;
}
+skip_rate:
/* avoid endless loop */
if (res == 0)
break;
@@ -296,6 +352,9 @@
case USB_ID(0x0E41, 0x4242): /* Line6 Helix Rack */
case USB_ID(0x0E41, 0x4244): /* Line6 Helix LT */
case USB_ID(0x0E41, 0x4246): /* Line6 HX-Stomp */
+ case USB_ID(0x0E41, 0x4248): /* Line6 Helix >= fw 2.82 */
+ case USB_ID(0x0E41, 0x4249): /* Line6 Helix Rack >= fw 2.82 */
+ case USB_ID(0x0E41, 0x424a): /* Line6 Helix LT >= fw 2.82 */
/* supported rates: 48Khz */
kfree(fp->rate_table);
fp->rate_table = kmalloc(sizeof(int), GFP_KERNEL);
@@ -322,8 +381,7 @@
struct usb_device *dev = chip->dev;
unsigned char tmp[2], *data;
int nr_triplets, data_size, ret = 0, ret_l6;
- int clock = snd_usb_clock_find_source(chip, fp->protocol,
- fp->clock, false);
+ int clock = snd_usb_clock_find_source(chip, fp, false);
if (clock < 0) {
dev_err(&dev->dev,
diff --git a/sound/usb/line6/capture.c b/sound/usb/line6/capture.c
index 82abef3..4b6e99e 100644
--- a/sound/usb/line6/capture.c
+++ b/sound/usb/line6/capture.c
@@ -287,6 +287,8 @@
urb->interval = LINE6_ISO_INTERVAL;
urb->error_count = 0;
urb->complete = audio_in_callback;
+ if (usb_urb_ep_type_check(urb))
+ return -EINVAL;
}
return 0;
diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c
index b5a3f75..1e38cdd 100644
--- a/sound/usb/line6/driver.c
+++ b/sound/usb/line6/driver.c
@@ -305,7 +305,7 @@
line6_midibuf_read(mb, line6->buffer_message,
LINE6_MIDI_MESSAGE_MAXLEN);
- if (done == 0)
+ if (done <= 0)
break;
line6->message_length = done;
@@ -690,6 +690,10 @@
line6->buffer_message = kmalloc(LINE6_MIDI_MESSAGE_MAXLEN, GFP_KERNEL);
if (!line6->buffer_message)
return -ENOMEM;
+
+ ret = line6_init_midi(line6);
+ if (ret < 0)
+ return ret;
} else {
ret = line6_hwdep_init(line6);
if (ret < 0)
@@ -820,7 +824,7 @@
if (WARN_ON(usbdev != line6->usbdev))
return;
- cancel_delayed_work(&line6->startup_work);
+ cancel_delayed_work_sync(&line6->startup_work);
if (line6->urb_listen != NULL)
line6_stop_listen(line6);
diff --git a/sound/usb/line6/midibuf.c b/sound/usb/line6/midibuf.c
index 8d6eefa..6a70463 100644
--- a/sound/usb/line6/midibuf.c
+++ b/sound/usb/line6/midibuf.c
@@ -159,7 +159,7 @@
int midi_length_prev =
midibuf_message_length(this->command_prev);
- if (midi_length_prev > 0) {
+ if (midi_length_prev > 1) {
midi_length = midi_length_prev - 1;
repeat = 1;
} else
diff --git a/sound/usb/line6/playback.c b/sound/usb/line6/playback.c
index 2e8ead3..797ced3 100644
--- a/sound/usb/line6/playback.c
+++ b/sound/usb/line6/playback.c
@@ -432,6 +432,8 @@
urb->interval = LINE6_ISO_INTERVAL;
urb->error_count = 0;
urb->complete = audio_out_callback;
+ if (usb_urb_ep_type_check(urb))
+ return -EINVAL;
}
return 0;
diff --git a/sound/usb/line6/pod.c b/sound/usb/line6/pod.c
index ee4c9d2..b667bb8 100644
--- a/sound/usb/line6/pod.c
+++ b/sound/usb/line6/pod.c
@@ -376,11 +376,6 @@
if (err < 0)
return err;
- /* initialize MIDI subsystem: */
- err = line6_init_midi(line6);
- if (err < 0)
- return err;
-
/* initialize PCM subsystem: */
err = line6_init_pcm(line6, &pod_pcm_properties);
if (err < 0)
diff --git a/sound/usb/line6/podhd.c b/sound/usb/line6/podhd.c
index 27bf61c..5d9954a 100644
--- a/sound/usb/line6/podhd.c
+++ b/sound/usb/line6/podhd.c
@@ -21,8 +21,7 @@
enum {
LINE6_PODHD300,
LINE6_PODHD400,
- LINE6_PODHD500_0,
- LINE6_PODHD500_1,
+ LINE6_PODHD500,
LINE6_PODX3,
LINE6_PODX3LIVE,
LINE6_PODHD500X,
@@ -318,8 +317,7 @@
/* TODO: no need to alloc data interfaces when only audio is used */
{ LINE6_DEVICE(0x5057), .driver_info = LINE6_PODHD300 },
{ LINE6_DEVICE(0x5058), .driver_info = LINE6_PODHD400 },
- { LINE6_IF_NUM(0x414D, 0), .driver_info = LINE6_PODHD500_0 },
- { LINE6_IF_NUM(0x414D, 1), .driver_info = LINE6_PODHD500_1 },
+ { LINE6_IF_NUM(0x414D, 0), .driver_info = LINE6_PODHD500 },
{ LINE6_IF_NUM(0x414A, 0), .driver_info = LINE6_PODX3 },
{ LINE6_IF_NUM(0x414B, 0), .driver_info = LINE6_PODX3LIVE },
{ LINE6_IF_NUM(0x4159, 0), .driver_info = LINE6_PODHD500X },
@@ -352,23 +350,13 @@
.ep_audio_r = 0x82,
.ep_audio_w = 0x01,
},
- [LINE6_PODHD500_0] = {
+ [LINE6_PODHD500] = {
.id = "PODHD500",
.name = "POD HD500",
- .capabilities = LINE6_CAP_PCM
+ .capabilities = LINE6_CAP_PCM | LINE6_CAP_CONTROL
| LINE6_CAP_HWMON,
.altsetting = 1,
- .ep_ctrl_r = 0x81,
- .ep_ctrl_w = 0x01,
- .ep_audio_r = 0x86,
- .ep_audio_w = 0x02,
- },
- [LINE6_PODHD500_1] = {
- .id = "PODHD500",
- .name = "POD HD500",
- .capabilities = LINE6_CAP_PCM
- | LINE6_CAP_HWMON,
- .altsetting = 0,
+ .ctrl_if = 1,
.ep_ctrl_r = 0x81,
.ep_ctrl_w = 0x01,
.ep_audio_r = 0x86,
diff --git a/sound/usb/line6/variax.c b/sound/usb/line6/variax.c
index ed158f0..c2245aa 100644
--- a/sound/usb/line6/variax.c
+++ b/sound/usb/line6/variax.c
@@ -159,7 +159,6 @@
const struct usb_device_id *id)
{
struct usb_line6_variax *variax = line6_to_variax(line6);
- int err;
line6->process_message = line6_variax_process_message;
line6->disconnect = line6_variax_disconnect;
@@ -172,11 +171,6 @@
if (variax->buffer_activate == NULL)
return -ENOMEM;
- /* initialize MIDI subsystem: */
- err = line6_init_midi(&variax->line6);
- if (err < 0)
- return err;
-
/* initiate startup procedure: */
schedule_delayed_work(&line6->startup_work,
msecs_to_jiffies(VARIAX_STARTUP_DELAY1));
diff --git a/sound/usb/midi.c b/sound/usb/midi.c
index b737f0e..33e9a7f 100644
--- a/sound/usb/midi.c
+++ b/sound/usb/midi.c
@@ -1332,7 +1332,7 @@
error:
snd_usbmidi_in_endpoint_delete(ep);
- return -ENOMEM;
+ return err;
}
/*
@@ -1499,6 +1499,8 @@
spin_unlock_irq(&umidi->disc_lock);
up_write(&umidi->disc_rwsem);
+ del_timer_sync(&umidi->error_timer);
+
for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i) {
struct snd_usb_midi_endpoint *ep = &umidi->endpoints[i];
if (ep->out)
@@ -1525,7 +1527,6 @@
ep->in = NULL;
}
}
- del_timer_sync(&umidi->error_timer);
}
EXPORT_SYMBOL(snd_usbmidi_disconnect);
@@ -1826,6 +1827,28 @@
return 0;
}
+static struct usb_ms_endpoint_descriptor *find_usb_ms_endpoint_descriptor(
+ struct usb_host_endpoint *hostep)
+{
+ unsigned char *extra = hostep->extra;
+ int extralen = hostep->extralen;
+
+ while (extralen > 3) {
+ struct usb_ms_endpoint_descriptor *ms_ep =
+ (struct usb_ms_endpoint_descriptor *)extra;
+
+ if (ms_ep->bLength > 3 &&
+ ms_ep->bDescriptorType == USB_DT_CS_ENDPOINT &&
+ ms_ep->bDescriptorSubtype == UAC_MS_GENERAL)
+ return ms_ep;
+ if (!extra[0])
+ break;
+ extralen -= extra[0];
+ extra += extra[0];
+ }
+ return NULL;
+}
+
/*
* Returns MIDIStreaming device capabilities.
*/
@@ -1863,11 +1886,14 @@
ep = get_ep_desc(hostep);
if (!usb_endpoint_xfer_bulk(ep) && !usb_endpoint_xfer_int(ep))
continue;
- ms_ep = (struct usb_ms_endpoint_descriptor *)hostep->extra;
- if (hostep->extralen < 4 ||
- ms_ep->bLength < 4 ||
- ms_ep->bDescriptorType != USB_DT_CS_ENDPOINT ||
- ms_ep->bDescriptorSubtype != UAC_MS_GENERAL)
+ ms_ep = find_usb_ms_endpoint_descriptor(hostep);
+ if (!ms_ep)
+ continue;
+ if (ms_ep->bLength <= sizeof(*ms_ep))
+ continue;
+ if (ms_ep->bNumEmbMIDIJack > 0x10)
+ continue;
+ if (ms_ep->bLength < sizeof(*ms_ep) + ms_ep->bNumEmbMIDIJack)
continue;
if (usb_endpoint_dir_out(ep)) {
if (endpoints[epidx].out_ep) {
@@ -2121,6 +2147,8 @@
cs_desc[1] == USB_DT_CS_INTERFACE &&
cs_desc[2] == 0xf1 &&
cs_desc[3] == 0x02) {
+ if (cs_desc[4] > 0x10 || cs_desc[5] > 0x10)
+ continue;
endpoint->in_cables = (1 << cs_desc[4]) - 1;
endpoint->out_cables = (1 << cs_desc[5]) - 1;
return snd_usbmidi_detect_endpoints(umidi, endpoint, 1);
@@ -2282,16 +2310,22 @@
}
EXPORT_SYMBOL(snd_usbmidi_input_stop);
-static void snd_usbmidi_input_start_ep(struct snd_usb_midi_in_endpoint *ep)
+static void snd_usbmidi_input_start_ep(struct snd_usb_midi *umidi,
+ struct snd_usb_midi_in_endpoint *ep)
{
unsigned int i;
+ unsigned long flags;
if (!ep)
return;
for (i = 0; i < INPUT_URBS; ++i) {
struct urb *urb = ep->urbs[i];
- urb->dev = ep->umidi->dev;
- snd_usbmidi_submit_urb(urb, GFP_KERNEL);
+ spin_lock_irqsave(&umidi->disc_lock, flags);
+ if (!atomic_read(&urb->use_count)) {
+ urb->dev = ep->umidi->dev;
+ snd_usbmidi_submit_urb(urb, GFP_ATOMIC);
+ }
+ spin_unlock_irqrestore(&umidi->disc_lock, flags);
}
}
@@ -2307,7 +2341,7 @@
if (umidi->input_running || !umidi->opened[1])
return;
for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i)
- snd_usbmidi_input_start_ep(umidi->endpoints[i].in);
+ snd_usbmidi_input_start_ep(umidi, umidi->endpoints[i].in);
umidi->input_running = 1;
}
EXPORT_SYMBOL(snd_usbmidi_input_start);
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index 6cd4ff0..67eb129 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -576,8 +576,9 @@
* if failed, give up and free the control instance.
*/
-int snd_usb_mixer_add_control(struct usb_mixer_elem_list *list,
- struct snd_kcontrol *kctl)
+int snd_usb_mixer_add_list(struct usb_mixer_elem_list *list,
+ struct snd_kcontrol *kctl,
+ bool is_std_info)
{
struct usb_mixer_interface *mixer = list->mixer;
int err;
@@ -591,6 +592,7 @@
return err;
}
list->kctl = kctl;
+ list->is_std_info = is_std_info;
list->next_id_elem = mixer->id_elems[list->id];
mixer->id_elems[list->id] = list;
return 0;
@@ -897,6 +899,15 @@
return 0;
}
+static int parse_term_effect_unit(struct mixer_build *state,
+ struct usb_audio_term *term,
+ void *p1, int id)
+{
+ term->type = UAC3_EFFECT_UNIT << 16; /* virtual type */
+ term->id = id;
+ return 0;
+}
+
static int parse_term_uac2_clock_source(struct mixer_build *state,
struct usb_audio_term *term,
void *p1, int id)
@@ -981,8 +992,7 @@
UAC3_PROCESSING_UNIT);
case PTYPE(UAC_VERSION_2, UAC2_EFFECT_UNIT):
case PTYPE(UAC_VERSION_3, UAC3_EFFECT_UNIT):
- return parse_term_proc_unit(state, term, p1, id,
- UAC3_EFFECT_UNIT);
+ return parse_term_effect_unit(state, term, p1, id);
case PTYPE(UAC_VERSION_1, UAC1_EXTENSION_UNIT):
case PTYPE(UAC_VERSION_2, UAC2_EXTENSION_UNIT_V2):
case PTYPE(UAC_VERSION_3, UAC3_EXTENSION_UNIT):
@@ -1020,7 +1030,7 @@
int type_uac2; /* data type for uac2 if different from uac1, else -1 */
};
-static struct usb_feature_control_info audio_feature_info[] = {
+static const struct usb_feature_control_info audio_feature_info[] = {
{ UAC_FU_MUTE, "Mute", USB_MIXER_INV_BOOLEAN, -1 },
{ UAC_FU_VOLUME, "Volume", USB_MIXER_S16, -1 },
{ UAC_FU_BASS, "Tone Control - Bass", USB_MIXER_S8, -1 },
@@ -1163,6 +1173,14 @@
cval->res = 384;
}
break;
+ case USB_ID(0x0495, 0x3042): /* ESS Technology Asus USB DAC */
+ if ((strstr(kctl->id.name, "Playback Volume") != NULL) ||
+ strstr(kctl->id.name, "Capture Volume") != NULL) {
+ cval->min >>= 8;
+ cval->max = 0;
+ cval->res = 1;
+ }
+ break;
}
}
@@ -1438,7 +1456,7 @@
usb_audio_err(chip,
"cannot get connectors status: req = %#x, wValue = %#x, wIndex = %#x, type = %d\n",
UAC_GET_CUR, validx, idx, cval->val_type);
- return ret;
+ return filter_error(cval, ret);
}
ucontrol->value.integer.value[0] = val;
@@ -1526,7 +1544,7 @@
strlcpy(kctl->id.name, "Headphone", sizeof(kctl->id.name));
}
-static struct usb_feature_control_info *get_feature_control_info(int control)
+static const struct usb_feature_control_info *get_feature_control_info(int control)
{
int i;
@@ -1544,7 +1562,7 @@
struct usb_audio_term *oterm,
int unitid, int nameid, int readonly_mask)
{
- struct usb_feature_control_info *ctl_info;
+ const struct usb_feature_control_info *ctl_info;
unsigned int len = 0;
int mapped_name = 0;
struct snd_kcontrol *kctl;
@@ -1666,6 +1684,16 @@
/* get min/max values */
get_min_max_with_quirks(cval, 0, kctl);
+ /* skip a bogus volume range */
+ if (cval->max <= cval->min) {
+ usb_audio_dbg(mixer->chip,
+ "[%d] FU [%s] skipped due to invalid volume\n",
+ cval->head.id, kctl->id.name);
+ snd_ctl_free_one(kctl);
+ return;
+ }
+
+
if (control == UAC_FU_VOLUME) {
check_mapped_dB(map, cval);
if (cval->dBmin < cval->dBmax || !cval->initialized) {
@@ -1742,10 +1770,16 @@
/* Build a mixer control for a UAC connector control (jack-detect) */
static void build_connector_control(struct usb_mixer_interface *mixer,
+ const struct usbmix_name_map *imap,
struct usb_audio_term *term, bool is_input)
{
struct snd_kcontrol *kctl;
struct usb_mixer_elem_info *cval;
+ const struct usbmix_name_map *map;
+
+ map = find_map(imap, term->id, 0);
+ if (check_ignored_ctl(map))
+ return;
cval = kzalloc(sizeof(*cval), GFP_KERNEL);
if (!cval)
@@ -1776,8 +1810,12 @@
usb_mixer_elem_info_free(cval);
return;
}
- get_connector_control_name(mixer, term, is_input, kctl->id.name,
- sizeof(kctl->id.name));
+
+ if (check_mapped_name(map, kctl->id.name, sizeof(kctl->id.name)))
+ strlcat(kctl->id.name, " Jack", sizeof(kctl->id.name));
+ else
+ get_connector_control_name(mixer, term, is_input, kctl->id.name,
+ sizeof(kctl->id.name));
kctl->private_free = snd_usb_mixer_elem_free;
snd_usb_mixer_add_control(&cval->head, kctl);
}
@@ -2080,8 +2118,9 @@
check_input_term(state, term_id, &iterm);
/* Check for jack detection. */
- if (uac_v2v3_control_is_readable(bmctls, control))
- build_connector_control(state->mixer, &iterm, true);
+ if ((iterm.type & 0xff00) != 0x0100 &&
+ uac_v2v3_control_is_readable(bmctls, control))
+ build_connector_control(state->mixer, state->map, &iterm, true);
return 0;
}
@@ -2198,7 +2237,7 @@
*/
struct procunit_value_info {
int control;
- char *suffix;
+ const char *suffix;
int val_type;
int min_value;
};
@@ -2206,44 +2245,44 @@
struct procunit_info {
int type;
char *name;
- struct procunit_value_info *values;
+ const struct procunit_value_info *values;
};
-static struct procunit_value_info undefined_proc_info[] = {
+static const struct procunit_value_info undefined_proc_info[] = {
{ 0x00, "Control Undefined", 0 },
{ 0 }
};
-static struct procunit_value_info updown_proc_info[] = {
+static const struct procunit_value_info updown_proc_info[] = {
{ UAC_UD_ENABLE, "Switch", USB_MIXER_BOOLEAN },
{ UAC_UD_MODE_SELECT, "Mode Select", USB_MIXER_U8, 1 },
{ 0 }
};
-static struct procunit_value_info prologic_proc_info[] = {
+static const struct procunit_value_info prologic_proc_info[] = {
{ UAC_DP_ENABLE, "Switch", USB_MIXER_BOOLEAN },
{ UAC_DP_MODE_SELECT, "Mode Select", USB_MIXER_U8, 1 },
{ 0 }
};
-static struct procunit_value_info threed_enh_proc_info[] = {
+static const struct procunit_value_info threed_enh_proc_info[] = {
{ UAC_3D_ENABLE, "Switch", USB_MIXER_BOOLEAN },
{ UAC_3D_SPACE, "Spaciousness", USB_MIXER_U8 },
{ 0 }
};
-static struct procunit_value_info reverb_proc_info[] = {
+static const struct procunit_value_info reverb_proc_info[] = {
{ UAC_REVERB_ENABLE, "Switch", USB_MIXER_BOOLEAN },
{ UAC_REVERB_LEVEL, "Level", USB_MIXER_U8 },
{ UAC_REVERB_TIME, "Time", USB_MIXER_U16 },
{ UAC_REVERB_FEEDBACK, "Feedback", USB_MIXER_U8 },
{ 0 }
};
-static struct procunit_value_info chorus_proc_info[] = {
+static const struct procunit_value_info chorus_proc_info[] = {
{ UAC_CHORUS_ENABLE, "Switch", USB_MIXER_BOOLEAN },
{ UAC_CHORUS_LEVEL, "Level", USB_MIXER_U8 },
{ UAC_CHORUS_RATE, "Rate", USB_MIXER_U16 },
{ UAC_CHORUS_DEPTH, "Depth", USB_MIXER_U16 },
{ 0 }
};
-static struct procunit_value_info dcr_proc_info[] = {
+static const struct procunit_value_info dcr_proc_info[] = {
{ UAC_DCR_ENABLE, "Switch", USB_MIXER_BOOLEAN },
{ UAC_DCR_RATE, "Ratio", USB_MIXER_U16 },
{ UAC_DCR_MAXAMPL, "Max Amp", USB_MIXER_S16 },
@@ -2253,7 +2292,7 @@
{ 0 }
};
-static struct procunit_info procunits[] = {
+static const struct procunit_info procunits[] = {
{ UAC_PROCESS_UP_DOWNMIX, "Up Down", updown_proc_info },
{ UAC_PROCESS_DOLBY_PROLOGIC, "Dolby Prologic", prologic_proc_info },
{ UAC_PROCESS_STEREO_EXTENDER, "3D Stereo Extender", threed_enh_proc_info },
@@ -2263,16 +2302,16 @@
{ 0 },
};
-static struct procunit_value_info uac3_updown_proc_info[] = {
+static const struct procunit_value_info uac3_updown_proc_info[] = {
{ UAC3_UD_MODE_SELECT, "Mode Select", USB_MIXER_U8, 1 },
{ 0 }
};
-static struct procunit_value_info uac3_stereo_ext_proc_info[] = {
+static const struct procunit_value_info uac3_stereo_ext_proc_info[] = {
{ UAC3_EXT_WIDTH_CONTROL, "Width Control", USB_MIXER_U8 },
{ 0 }
};
-static struct procunit_info uac3_procunits[] = {
+static const struct procunit_info uac3_procunits[] = {
{ UAC3_PROCESS_UP_DOWNMIX, "Up Down", uac3_updown_proc_info },
{ UAC3_PROCESS_STEREO_EXTENDER, "3D Stereo Extender", uac3_stereo_ext_proc_info },
{ UAC3_PROCESS_MULTI_FUNCTION, "Multi-Function", undefined_proc_info },
@@ -2282,23 +2321,23 @@
/*
* predefined data for extension units
*/
-static struct procunit_value_info clock_rate_xu_info[] = {
+static const struct procunit_value_info clock_rate_xu_info[] = {
{ USB_XU_CLOCK_RATE_SELECTOR, "Selector", USB_MIXER_U8, 0 },
{ 0 }
};
-static struct procunit_value_info clock_source_xu_info[] = {
+static const struct procunit_value_info clock_source_xu_info[] = {
{ USB_XU_CLOCK_SOURCE_SELECTOR, "External", USB_MIXER_BOOLEAN },
{ 0 }
};
-static struct procunit_value_info spdif_format_xu_info[] = {
+static const struct procunit_value_info spdif_format_xu_info[] = {
{ USB_XU_DIGITAL_FORMAT_SELECTOR, "SPDIF/AC3", USB_MIXER_BOOLEAN },
{ 0 }
};
-static struct procunit_value_info soft_limit_xu_info[] = {
+static const struct procunit_value_info soft_limit_xu_info[] = {
{ USB_XU_SOFT_LIMIT_SELECTOR, " ", USB_MIXER_BOOLEAN },
{ 0 }
};
-static struct procunit_info extunits[] = {
+static const struct procunit_info extunits[] = {
{ USB_XU_CLOCK_RATE, "Clock rate", clock_rate_xu_info },
{ USB_XU_CLOCK_SOURCE, "DigitalIn CLK source", clock_source_xu_info },
{ USB_XU_DIGITAL_IO_STATUS, "DigitalOut format:", spdif_format_xu_info },
@@ -2310,7 +2349,7 @@
* build a processing/extension unit
*/
static int build_audio_procunit(struct mixer_build *state, int unitid,
- void *raw_desc, struct procunit_info *list,
+ void *raw_desc, const struct procunit_info *list,
bool extension_unit)
{
struct uac_processing_unit_descriptor *desc = raw_desc;
@@ -2318,14 +2357,14 @@
struct usb_mixer_elem_info *cval;
struct snd_kcontrol *kctl;
int i, err, nameid, type, len;
- struct procunit_info *info;
- struct procunit_value_info *valinfo;
+ const struct procunit_info *info;
+ const struct procunit_value_info *valinfo;
const struct usbmix_name_map *map;
- static struct procunit_value_info default_value_info[] = {
+ static const struct procunit_value_info default_value_info[] = {
{ 0x01, "Switch", USB_MIXER_BOOLEAN },
{ 0 }
};
- static struct procunit_info default_info = {
+ static const struct procunit_info default_info = {
0, NULL, default_value_info
};
const char *name = extension_unit ?
@@ -2803,7 +2842,7 @@
int st_chmask; /* side tone mixing channel mask */
};
-static struct uac3_badd_profile uac3_badd_profiles[] = {
+static const struct uac3_badd_profile uac3_badd_profiles[] = {
{
/*
* BAIF, BAOF or combination of both
@@ -2864,7 +2903,7 @@
};
static bool uac3_badd_func_has_valid_channels(struct usb_mixer_interface *mixer,
- struct uac3_badd_profile *f,
+ const struct uac3_badd_profile *f,
int c_chmask, int p_chmask)
{
/*
@@ -2908,7 +2947,7 @@
struct usb_device *dev = mixer->chip->dev;
struct usb_interface_assoc_descriptor *assoc;
int badd_profile = mixer->chip->badd_profile;
- struct uac3_badd_profile *f;
+ const struct uac3_badd_profile *f;
const struct usbmix_ctl_map *map;
int p_chmask = 0, c_chmask = 0, st_chmask = 0;
int i;
@@ -3042,13 +3081,13 @@
memset(&iterm, 0, sizeof(iterm));
iterm.id = UAC3_BADD_IT_ID4;
iterm.type = UAC_BIDIR_TERMINAL_HEADSET;
- build_connector_control(mixer, &iterm, true);
+ build_connector_control(mixer, map->map, &iterm, true);
/* Output Term - Insertion control */
memset(&oterm, 0, sizeof(oterm));
oterm.id = UAC3_BADD_OT_ID3;
oterm.type = UAC_BIDIR_TERMINAL_HEADSET;
- build_connector_control(mixer, &oterm, false);
+ build_connector_control(mixer, map->map, &oterm, false);
}
return 0;
@@ -3077,7 +3116,8 @@
if (map->id == state.chip->usb_id) {
state.map = map->map;
state.selector_map = map->selector_map;
- mixer->ignore_ctl_error = map->ignore_ctl_error;
+ mixer->connector_map = map->connector_map;
+ mixer->ignore_ctl_error |= map->ignore_ctl_error;
break;
}
}
@@ -3120,10 +3160,11 @@
if (err < 0 && err != -EINVAL)
return err;
- if (uac_v2v3_control_is_readable(le16_to_cpu(desc->bmControls),
+ if ((state.oterm.type & 0xff00) != 0x0100 &&
+ uac_v2v3_control_is_readable(le16_to_cpu(desc->bmControls),
UAC2_TE_CONNECTOR)) {
- build_connector_control(state.mixer, &state.oterm,
- false);
+ build_connector_control(state.mixer, state.map,
+ &state.oterm, false);
}
} else { /* UAC_VERSION_3 */
struct uac3_output_terminal_descriptor *desc = p;
@@ -3145,10 +3186,11 @@
if (err < 0 && err != -EINVAL)
return err;
- if (uac_v2v3_control_is_readable(le32_to_cpu(desc->bmControls),
+ if ((state.oterm.type & 0xff00) != 0x0100 &&
+ uac_v2v3_control_is_readable(le32_to_cpu(desc->bmControls),
UAC3_TE_INSERTION)) {
- build_connector_control(state.mixer, &state.oterm,
- false);
+ build_connector_control(state.mixer, state.map,
+ &state.oterm, false);
}
}
}
@@ -3156,13 +3198,38 @@
return 0;
}
+static int delegate_notify(struct usb_mixer_interface *mixer, int unitid,
+ u8 *control, u8 *channel)
+{
+ const struct usbmix_connector_map *map = mixer->connector_map;
+
+ if (!map)
+ return unitid;
+
+ for (; map->id; map++) {
+ if (map->id == unitid) {
+ if (control && map->control)
+ *control = map->control;
+ if (channel && map->channel)
+ *channel = map->channel;
+ return map->delegated_id;
+ }
+ }
+ return unitid;
+}
+
void snd_usb_mixer_notify_id(struct usb_mixer_interface *mixer, int unitid)
{
struct usb_mixer_elem_list *list;
+ unitid = delegate_notify(mixer, unitid, NULL, NULL);
+
for_each_mixer_elem(list, mixer, unitid) {
- struct usb_mixer_elem_info *info =
- mixer_elem_list_to_info(list);
+ struct usb_mixer_elem_info *info;
+
+ if (!list->is_std_info)
+ continue;
+ info = mixer_elem_list_to_info(list);
/* invalidate cache, so the value is read from the device */
info->cached = 0;
snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
@@ -3174,8 +3241,17 @@
struct usb_mixer_elem_list *list)
{
struct usb_mixer_elem_info *cval = mixer_elem_list_to_info(list);
- static char *val_types[] = {"BOOLEAN", "INV_BOOLEAN",
- "S8", "U8", "S16", "U16"};
+ static const char * const val_types[] = {
+ [USB_MIXER_BOOLEAN] = "BOOLEAN",
+ [USB_MIXER_INV_BOOLEAN] = "INV_BOOLEAN",
+ [USB_MIXER_S8] = "S8",
+ [USB_MIXER_U8] = "U8",
+ [USB_MIXER_S16] = "S16",
+ [USB_MIXER_U16] = "U16",
+ [USB_MIXER_S32] = "S32",
+ [USB_MIXER_U32] = "U32",
+ [USB_MIXER_BESPOKEN] = "BESPOKEN",
+ };
snd_iprintf(buffer, " Info: id=%i, control=%i, cmask=0x%x, "
"channels=%i, type=\"%s\"\n", cval->head.id,
cval->control, cval->cmask, cval->channels,
@@ -3229,6 +3305,8 @@
return;
}
+ unitid = delegate_notify(mixer, unitid, &control, &channel);
+
for_each_mixer_elem(list, mixer, unitid)
count++;
@@ -3240,6 +3318,8 @@
if (!list->kctl)
continue;
+ if (!list->is_std_info)
+ continue;
info = mixer_elem_list_to_info(list);
if (count > 1 && info->control != control)
@@ -3527,6 +3607,9 @@
struct usb_mixer_elem_info *cval = mixer_elem_list_to_info(list);
int c, err, idx;
+ if (cval->val_type == USB_MIXER_BESPOKEN)
+ return 0;
+
if (cval->cmask) {
idx = 0;
for (c = 0; c < MAX_CHANNELS; c++) {
diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h
index 37e1b23..0e813cd 100644
--- a/sound/usb/mixer.h
+++ b/sound/usb/mixer.h
@@ -6,6 +6,13 @@
struct media_mixer_ctl;
+struct usbmix_connector_map {
+ u8 id;
+ u8 delegated_id;
+ u8 control;
+ u8 channel;
+};
+
struct usb_mixer_interface {
struct snd_usb_audio *chip;
struct usb_host_interface *hostif;
@@ -18,6 +25,9 @@
/* the usb audio specification version this interface complies to */
int protocol;
+ /* optional connector delegation map */
+ const struct usbmix_connector_map *connector_map;
+
/* Sound Blaster remote control stuff */
const struct rc_config *rc_cfg;
u32 rc_code;
@@ -45,6 +55,7 @@
USB_MIXER_U16,
USB_MIXER_S32,
USB_MIXER_U32,
+ USB_MIXER_BESPOKEN, /* non-standard type */
};
typedef void (*usb_mixer_elem_dump_func_t)(struct snd_info_buffer *buffer,
@@ -56,6 +67,7 @@
struct usb_mixer_elem_list *next_id_elem; /* list of controls with same id */
struct snd_kcontrol *kctl;
unsigned int id;
+ bool is_std_info;
usb_mixer_elem_dump_func_t dump;
usb_mixer_elem_resume_func_t resume;
};
@@ -93,8 +105,12 @@
int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval,
int request, int validx, int value_set);
-int snd_usb_mixer_add_control(struct usb_mixer_elem_list *list,
- struct snd_kcontrol *kctl);
+int snd_usb_mixer_add_list(struct usb_mixer_elem_list *list,
+ struct snd_kcontrol *kctl,
+ bool is_std_info);
+
+#define snd_usb_mixer_add_control(list, kctl) \
+ snd_usb_mixer_add_list(list, kctl, true)
void snd_usb_mixer_elem_init_std(struct usb_mixer_elem_list *list,
struct usb_mixer_interface *mixer,
diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c
index 73baf39..dda1362 100644
--- a/sound/usb/mixer_maps.c
+++ b/sound/usb/mixer_maps.c
@@ -14,7 +14,7 @@
int id;
const char *name;
int control;
- struct usbmix_dB_map *dB;
+ const struct usbmix_dB_map *dB;
};
struct usbmix_selector_map {
@@ -27,6 +27,7 @@
u32 id;
const struct usbmix_name_map *map;
const struct usbmix_selector_map *selector_map;
+ const struct usbmix_connector_map *connector_map;
int ignore_ctl_error;
};
@@ -52,7 +53,7 @@
++--+->SU[11]-->FU[12] --------------------------------------------------------------------------------------> USB_OUT[13]
*/
-static struct usbmix_name_map extigy_map[] = {
+static const struct usbmix_name_map extigy_map[] = {
/* 1: IT pcm */
{ 2, "PCM Playback" }, /* FU */
/* 3: IT pcm */
@@ -93,12 +94,12 @@
* e.g. no Master and fake PCM volume
* Pavel Mihaylov <bin@bash.info>
*/
-static struct usbmix_dB_map mp3plus_dB_1 = {.min = -4781, .max = 0};
+static const struct usbmix_dB_map mp3plus_dB_1 = {.min = -4781, .max = 0};
/* just guess */
-static struct usbmix_dB_map mp3plus_dB_2 = {.min = -1781, .max = 618};
+static const struct usbmix_dB_map mp3plus_dB_2 = {.min = -1781, .max = 618};
/* just guess */
-static struct usbmix_name_map mp3plus_map[] = {
+static const struct usbmix_name_map mp3plus_map[] = {
/* 1: IT pcm */
/* 2: IT mic */
/* 3: IT line */
@@ -139,7 +140,7 @@
| ^
+->FU[13]--------------------------------------+
*/
-static struct usbmix_name_map audigy2nx_map[] = {
+static const struct usbmix_name_map audigy2nx_map[] = {
/* 1: IT pcm playback */
/* 4: IT digital in */
{ 6, "Digital In Playback" }, /* FU */
@@ -167,12 +168,12 @@
{ 0 } /* terminator */
};
-static struct usbmix_name_map mbox1_map[] = {
+static const struct usbmix_name_map mbox1_map[] = {
{ 1, "Clock" },
{ 0 } /* terminator */
};
-static struct usbmix_selector_map c400_selectors[] = {
+static const struct usbmix_selector_map c400_selectors[] = {
{
.id = 0x80,
.count = 2,
@@ -181,7 +182,7 @@
{ 0 } /* terminator */
};
-static struct usbmix_selector_map audigy2nx_selectors[] = {
+static const struct usbmix_selector_map audigy2nx_selectors[] = {
{
.id = 14, /* Capture Source */
.count = 3,
@@ -201,21 +202,21 @@
};
/* Creative SoundBlaster Live! 24-bit External */
-static struct usbmix_name_map live24ext_map[] = {
+static const struct usbmix_name_map live24ext_map[] = {
/* 2: PCM Playback Volume */
{ 5, "Mic Capture" }, /* FU, default PCM Capture Volume */
{ 0 } /* terminator */
};
/* LineX FM Transmitter entry - needed to bypass controls bug */
-static struct usbmix_name_map linex_map[] = {
+static const struct usbmix_name_map linex_map[] = {
/* 1: IT pcm */
/* 2: OT Speaker */
{ 3, "Master" }, /* FU: master volume - left / right / mute */
{ 0 } /* terminator */
};
-static struct usbmix_name_map maya44_map[] = {
+static const struct usbmix_name_map maya44_map[] = {
/* 1: IT line */
{ 2, "Line Playback" }, /* FU */
/* 3: IT line */
@@ -238,7 +239,7 @@
* so this map removes all unwanted sliders from alsamixer
*/
-static struct usbmix_name_map justlink_map[] = {
+static const struct usbmix_name_map justlink_map[] = {
/* 1: IT pcm playback */
/* 2: Not present */
{ 3, NULL}, /* IT mic (No mic input on device) */
@@ -255,7 +256,7 @@
};
/* TerraTec Aureon 5.1 MkII USB */
-static struct usbmix_name_map aureon_51_2_map[] = {
+static const struct usbmix_name_map aureon_51_2_map[] = {
/* 1: IT USB */
/* 2: IT Mic */
/* 3: IT Line */
@@ -274,7 +275,7 @@
{} /* terminator */
};
-static struct usbmix_name_map scratch_live_map[] = {
+static const struct usbmix_name_map scratch_live_map[] = {
/* 1: IT Line 1 (USB streaming) */
/* 2: OT Line 1 (Speaker) */
/* 3: IT Line 1 (Line connector) */
@@ -290,7 +291,7 @@
{ 0 } /* terminator */
};
-static struct usbmix_name_map ebox44_map[] = {
+static const struct usbmix_name_map ebox44_map[] = {
{ 4, NULL }, /* FU */
{ 6, NULL }, /* MU */
{ 7, NULL }, /* FU */
@@ -305,7 +306,7 @@
* FIXME: or mp3plus_map should use "Capture Source" too,
* so this maps can be merget
*/
-static struct usbmix_name_map hercules_usb51_map[] = {
+static const struct usbmix_name_map hercules_usb51_map[] = {
{ 8, "Capture Source" }, /* SU, default "PCM Capture Source" */
{ 9, "Master Playback" }, /* FU, default "Speaker Playback" */
{ 10, "Mic Boost", 7 }, /* FU, default "Auto Gain Input" */
@@ -316,7 +317,7 @@
};
/* Plantronics Gamecom 780 has a broken volume control, better to disable it */
-static struct usbmix_name_map gamecom780_map[] = {
+static const struct usbmix_name_map gamecom780_map[] = {
{ 9, NULL }, /* FU, speaker out */
{}
};
@@ -330,12 +331,19 @@
};
/* Bose companion 5, the dB conversion factor is 16 instead of 256 */
-static struct usbmix_dB_map bose_companion5_dB = {-5006, -6};
-static struct usbmix_name_map bose_companion5_map[] = {
+static const struct usbmix_dB_map bose_companion5_dB = {-5006, -6};
+static const struct usbmix_name_map bose_companion5_map[] = {
{ 3, NULL, .dB = &bose_companion5_dB },
{ 0 } /* terminator */
};
+/* Sennheiser Communications Headset [PC 8], the dB value is reported as -6 negative maximum */
+static const struct usbmix_dB_map sennheiser_pc8_dB = {-9500, 0};
+static const struct usbmix_name_map sennheiser_pc8_map[] = {
+ { 9, NULL, .dB = &sennheiser_pc8_dB },
+ { 0 } /* terminator */
+};
+
/*
* Dell usb dock with ALC4020 codec had a firmware problem where it got
* screwed up when zero volume is passed; just skip it as a workaround
@@ -349,11 +357,63 @@
{ 0 }
};
+/* Some mobos shipped with a dummy HD-audio show the invalid GET_MIN/GET_MAX
+ * response for Input Gain Pad (id=19, control=12) and the connector status
+ * for SPDIF terminal (id=18). Skip them.
+ */
+static const struct usbmix_name_map asus_rog_map[] = {
+ { 18, NULL }, /* OT, connector control */
+ { 19, NULL, 12 }, /* FU, Input Gain Pad */
+ {}
+};
+
+/* TRX40 mobos with Realtek ALC1220-VB */
+static const struct usbmix_name_map trx40_mobo_map[] = {
+ { 18, NULL }, /* OT, IEC958 - broken response, disabled */
+ { 19, NULL, 12 }, /* FU, Input Gain Pad - broken response, disabled */
+ { 16, "Speaker" }, /* OT */
+ { 22, "Speaker Playback" }, /* FU */
+ { 7, "Line" }, /* IT */
+ { 19, "Line Capture" }, /* FU */
+ { 17, "Front Headphone" }, /* OT */
+ { 23, "Front Headphone Playback" }, /* FU */
+ { 8, "Mic" }, /* IT */
+ { 20, "Mic Capture" }, /* FU */
+ { 9, "Front Mic" }, /* IT */
+ { 21, "Front Mic Capture" }, /* FU */
+ { 24, "IEC958 Playback" }, /* FU */
+ {}
+};
+
+static const struct usbmix_connector_map trx40_mobo_connector_map[] = {
+ { 10, 16 }, /* (Back) Speaker */
+ { 11, 17 }, /* Front Headphone */
+ { 13, 7 }, /* Line */
+ { 14, 8 }, /* Mic */
+ { 15, 9 }, /* Front Mic */
+ {}
+};
+
+/* Rear panel + front mic on Gigabyte TRX40 Aorus Master with ALC1220-VB */
+static const struct usbmix_name_map aorus_master_alc1220vb_map[] = {
+ { 17, NULL }, /* OT, IEC958?, disabled */
+ { 19, NULL, 12 }, /* FU, Input Gain Pad - broken response, disabled */
+ { 16, "Line Out" }, /* OT */
+ { 22, "Line Out Playback" }, /* FU */
+ { 7, "Line" }, /* IT */
+ { 19, "Line Capture" }, /* FU */
+ { 8, "Mic" }, /* IT */
+ { 20, "Mic Capture" }, /* FU */
+ { 9, "Front Mic" }, /* IT */
+ { 21, "Front Mic Capture" }, /* FU */
+ {}
+};
+
/*
* Control map entries
*/
-static struct usbmix_ctl_map usbmix_ctl_maps[] = {
+static const struct usbmix_ctl_map usbmix_ctl_maps[] = {
{
.id = USB_ID(0x041e, 0x3000),
.map = extigy_map,
@@ -468,6 +528,38 @@
.id = USB_ID(0x05a7, 0x1020),
.map = bose_companion5_map,
},
+ { /* Gigabyte TRX40 Aorus Master (rear panel + front mic) */
+ .id = USB_ID(0x0414, 0xa001),
+ .map = aorus_master_alc1220vb_map,
+ },
+ { /* Gigabyte TRX40 Aorus Pro WiFi */
+ .id = USB_ID(0x0414, 0xa002),
+ .map = trx40_mobo_map,
+ .connector_map = trx40_mobo_connector_map,
+ },
+ { /* ASUS ROG Zenith II */
+ .id = USB_ID(0x0b05, 0x1916),
+ .map = asus_rog_map,
+ },
+ { /* ASUS ROG Strix */
+ .id = USB_ID(0x0b05, 0x1917),
+ .map = asus_rog_map,
+ },
+ { /* MSI TRX40 Creator */
+ .id = USB_ID(0x0db0, 0x0d64),
+ .map = trx40_mobo_map,
+ .connector_map = trx40_mobo_connector_map,
+ },
+ { /* MSI TRX40 */
+ .id = USB_ID(0x0db0, 0x543d),
+ .map = trx40_mobo_map,
+ .connector_map = trx40_mobo_connector_map,
+ },
+ { /* Asrock TRX40 Creator */
+ .id = USB_ID(0x26ce, 0x0a01),
+ .map = trx40_mobo_map,
+ .connector_map = trx40_mobo_connector_map,
+ },
{ 0 } /* terminator */
};
@@ -475,37 +567,37 @@
* Control map entries for UAC3 BADD profiles
*/
-static struct usbmix_name_map uac3_badd_generic_io_map[] = {
+static const struct usbmix_name_map uac3_badd_generic_io_map[] = {
{ UAC3_BADD_FU_ID2, "Generic Out Playback" },
{ UAC3_BADD_FU_ID5, "Generic In Capture" },
{ 0 } /* terminator */
};
-static struct usbmix_name_map uac3_badd_headphone_map[] = {
+static const struct usbmix_name_map uac3_badd_headphone_map[] = {
{ UAC3_BADD_FU_ID2, "Headphone Playback" },
{ 0 } /* terminator */
};
-static struct usbmix_name_map uac3_badd_speaker_map[] = {
+static const struct usbmix_name_map uac3_badd_speaker_map[] = {
{ UAC3_BADD_FU_ID2, "Speaker Playback" },
{ 0 } /* terminator */
};
-static struct usbmix_name_map uac3_badd_microphone_map[] = {
+static const struct usbmix_name_map uac3_badd_microphone_map[] = {
{ UAC3_BADD_FU_ID5, "Mic Capture" },
{ 0 } /* terminator */
};
/* Covers also 'headset adapter' profile */
-static struct usbmix_name_map uac3_badd_headset_map[] = {
+static const struct usbmix_name_map uac3_badd_headset_map[] = {
{ UAC3_BADD_FU_ID2, "Headset Playback" },
{ UAC3_BADD_FU_ID5, "Headset Capture" },
{ UAC3_BADD_FU_ID7, "Sidetone Mixing" },
{ 0 } /* terminator */
};
-static struct usbmix_name_map uac3_badd_speakerphone_map[] = {
+static const struct usbmix_name_map uac3_badd_speakerphone_map[] = {
{ UAC3_BADD_FU_ID2, "Speaker Playback" },
{ UAC3_BADD_FU_ID5, "Mic Capture" },
{ 0 } /* terminator */
};
-static struct usbmix_ctl_map uac3_badd_usbmix_ctl_maps[] = {
+static const struct usbmix_ctl_map uac3_badd_usbmix_ctl_maps[] = {
{
.id = UAC3_FUNCTION_SUBCLASS_GENERIC_IO,
.map = uac3_badd_generic_io_map,
@@ -534,5 +626,10 @@
.id = UAC3_FUNCTION_SUBCLASS_SPEAKERPHONE,
.map = uac3_badd_speakerphone_map,
},
+ {
+ /* Sennheiser Communications Headset [PC 8] */
+ .id = USB_ID(0x1395, 0x0025),
+ .map = sennheiser_pc8_map,
+ },
{ 0 } /* terminator */
};
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index 39e27ae..d926869 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -119,7 +119,7 @@
* Create a set of standard UAC controls from a table
*/
static int snd_create_std_mono_table(struct usb_mixer_interface *mixer,
- struct std_mono_table *t)
+ const struct std_mono_table *t)
{
int err;
@@ -157,7 +157,8 @@
return -ENOMEM;
}
kctl->private_free = snd_usb_mixer_elem_free;
- return snd_usb_mixer_add_control(list, kctl);
+ /* don't use snd_usb_mixer_add_control() here, this is a special list element */
+ return snd_usb_mixer_add_list(list, kctl, false);
}
/*
@@ -183,6 +184,7 @@
{ USB_ID(0x041e, 0x3042), 0, 1, 1, 1, 1, 0x000d }, /* Usb X-Fi S51 */
{ USB_ID(0x041e, 0x30df), 0, 1, 1, 1, 1, 0x000d }, /* Usb X-Fi S51 Pro */
{ USB_ID(0x041e, 0x3237), 0, 1, 1, 1, 1, 0x000d }, /* Usb X-Fi S51 Pro */
+ { USB_ID(0x041e, 0x3263), 0, 1, 1, 1, 1, 0x000d }, /* Usb X-Fi S51 Pro */
{ USB_ID(0x041e, 0x3048), 2, 2, 6, 6, 2, 0x6e91 }, /* Toshiba SB0500 */
};
@@ -1386,7 +1388,7 @@
* are valid they presents mono controls as L and R channels of
* stereo. So we provide a good mixer here.
*/
-static struct std_mono_table ebox44_table[] = {
+static const struct std_mono_table ebox44_table[] = {
{
.unitid = 4,
.control = 1,
@@ -1508,11 +1510,15 @@
/* use known values for that card: interface#1 altsetting#1 */
iface = usb_ifnum_to_if(chip->dev, 1);
- if (!iface || iface->num_altsetting < 2)
- return -EINVAL;
+ if (!iface || iface->num_altsetting < 2) {
+ err = -EINVAL;
+ goto end;
+ }
alts = &iface->altsetting[1];
- if (get_iface_desc(alts)->bNumEndpoints < 1)
- return -EINVAL;
+ if (get_iface_desc(alts)->bNumEndpoints < 1) {
+ err = -EINVAL;
+ goto end;
+ }
ep = get_endpoint(alts, 0)->bEndpointAddress;
err = snd_usb_ctl_msg(chip->dev,
@@ -1691,7 +1697,7 @@
static int snd_microii_controls_create(struct usb_mixer_interface *mixer)
{
int err, i;
- static usb_mixer_elem_resume_func_t resume_funcs[] = {
+ static const usb_mixer_elem_resume_func_t resume_funcs[] = {
snd_microii_spdif_default_update,
NULL,
snd_microii_spdif_switch_update
@@ -2262,7 +2268,7 @@
case USB_ID(0x1235, 0x8203): /* Focusrite Scarlett 6i6 2nd Gen */
case USB_ID(0x1235, 0x8204): /* Focusrite Scarlett 18i8 2nd Gen */
case USB_ID(0x1235, 0x8201): /* Focusrite Scarlett 18i20 2nd Gen */
- err = snd_scarlett_gen2_controls_create(mixer);
+ err = snd_scarlett_gen2_init(mixer);
break;
case USB_ID(0x041e, 0x323b): /* Creative Sound Blaster E1 */
diff --git a/sound/usb/mixer_scarlett.c b/sound/usb/mixer_scarlett.c
index 83715fd..6217424 100644
--- a/sound/usb/mixer_scarlett.c
+++ b/sound/usb/mixer_scarlett.c
@@ -623,7 +623,7 @@
/********************** device-specific config *************************/
/* untested... */
-static struct scarlett_device_info s6i6_info = {
+static const struct scarlett_device_info s6i6_info = {
.matrix_in = 18,
.matrix_out = 8,
.input_len = 6,
@@ -665,7 +665,7 @@
};
/* untested... */
-static struct scarlett_device_info s8i6_info = {
+static const struct scarlett_device_info s8i6_info = {
.matrix_in = 18,
.matrix_out = 6,
.input_len = 8,
@@ -704,7 +704,7 @@
}
};
-static struct scarlett_device_info s18i6_info = {
+static const struct scarlett_device_info s18i6_info = {
.matrix_in = 18,
.matrix_out = 6,
.input_len = 18,
@@ -741,7 +741,7 @@
}
};
-static struct scarlett_device_info s18i8_info = {
+static const struct scarlett_device_info s18i8_info = {
.matrix_in = 18,
.matrix_out = 8,
.input_len = 18,
@@ -783,7 +783,7 @@
}
};
-static struct scarlett_device_info s18i20_info = {
+static const struct scarlett_device_info s18i20_info = {
.matrix_in = 18,
.matrix_out = 8,
.input_len = 18,
@@ -833,7 +833,7 @@
static int scarlett_controls_create_generic(struct usb_mixer_interface *mixer,
- struct scarlett_device_info *info)
+ const struct scarlett_device_info *info)
{
int i, err;
char mx[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
@@ -896,7 +896,7 @@
{
int err, i, o;
char mx[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
- struct scarlett_device_info *info;
+ const struct scarlett_device_info *info;
struct usb_mixer_elem_info *elem;
static char sample_rate_buffer[4] = { '\x80', '\xbb', '\x00', '\x00' };
diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c
index 94b903d..ab7abe3 100644
--- a/sound/usb/mixer_scarlett_gen2.c
+++ b/sound/usb/mixer_scarlett_gen2.c
@@ -254,10 +254,10 @@
.pad_input_count = 2,
.line_out_descrs = {
- "Monitor L",
- "Monitor R",
- "Headphones L",
- "Headphones R",
+ "Headphones 1 L",
+ "Headphones 1 R",
+ "Headphones 2 L",
+ "Headphones 2 R",
},
.ports = {
@@ -356,7 +356,7 @@
},
[SCARLETT2_PORT_TYPE_PCM] = {
.id = 0x600,
- .num = { 20, 18, 18, 14, 10 },
+ .num = { 8, 18, 18, 14, 10 },
.src_descr = "PCM %d",
.src_num_offset = 1,
.dst_descr = "PCM %02d Capture"
@@ -558,11 +558,11 @@
/* proprietary request/response format */
struct scarlett2_usb_packet {
- u32 cmd;
- u16 size;
- u16 seq;
- u32 error;
- u32 pad;
+ __le32 cmd;
+ __le16 size;
+ __le16 seq;
+ __le32 error;
+ __le32 pad;
u8 data[];
};
@@ -635,7 +635,7 @@
/* send a second message to get the response */
err = snd_usb_ctl_msg(mixer->chip->dev,
- usb_sndctrlpipe(mixer->chip->dev, 0),
+ usb_rcvctrlpipe(mixer->chip->dev, 0),
SCARLETT2_USB_VENDOR_SPECIFIC_CMD_RESP,
USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN,
0,
@@ -664,11 +664,11 @@
"Scarlett Gen 2 USB invalid response; "
"cmd tx/rx %d/%d seq %d/%d size %d/%d "
"error %d pad %d\n",
- le16_to_cpu(req->cmd), le16_to_cpu(resp->cmd),
+ le32_to_cpu(req->cmd), le32_to_cpu(resp->cmd),
le16_to_cpu(req->seq), le16_to_cpu(resp->seq),
resp_size, le16_to_cpu(resp->size),
- le16_to_cpu(resp->error),
- le16_to_cpu(resp->pad));
+ le32_to_cpu(resp->error),
+ le32_to_cpu(resp->pad));
err = -EINVAL;
goto unlock;
}
@@ -687,7 +687,7 @@
/* Send SCARLETT2_USB_DATA_CMD SCARLETT2_USB_CONFIG_SAVE */
static void scarlett2_config_save(struct usb_mixer_interface *mixer)
{
- u32 req = cpu_to_le32(SCARLETT2_USB_CONFIG_SAVE);
+ __le32 req = cpu_to_le32(SCARLETT2_USB_CONFIG_SAVE);
scarlett2_usb(mixer, SCARLETT2_USB_DATA_CMD,
&req, sizeof(u32),
@@ -713,11 +713,11 @@
const struct scarlett2_config config_item =
scarlett2_config_items[config_item_num];
struct {
- u32 offset;
- u32 bytes;
- s32 value;
+ __le32 offset;
+ __le32 bytes;
+ __le32 value;
} __packed req;
- u32 req2;
+ __le32 req2;
int err;
struct scarlett2_mixer_data *private = mixer->private_data;
@@ -753,8 +753,8 @@
int offset, void *buf, int size)
{
struct {
- u32 offset;
- u32 size;
+ __le32 offset;
+ __le32 size;
} __packed req;
req.offset = cpu_to_le32(offset);
@@ -794,8 +794,8 @@
const struct scarlett2_device_info *info = private->info;
struct {
- u16 mix_num;
- u16 data[SCARLETT2_INPUT_MIX_MAX];
+ __le16 mix_num;
+ __le16 data[SCARLETT2_INPUT_MIX_MAX];
} __packed req;
int i, j;
@@ -850,9 +850,9 @@
};
struct {
- u16 pad;
- u16 num;
- u32 data[SCARLETT2_MUX_MAX];
+ __le16 pad;
+ __le16 num;
+ __le32 data[SCARLETT2_MUX_MAX];
} __packed req;
req.pad = 0;
@@ -911,9 +911,9 @@
u16 *levels)
{
struct {
- u16 pad;
- u16 num_meters;
- u32 magic;
+ __le16 pad;
+ __le16 num_meters;
+ __le32 magic;
} __packed req;
u32 resp[SCARLETT2_NUM_METERS];
int i, err;
@@ -949,10 +949,15 @@
if (!elem)
return -ENOMEM;
+ /* We set USB_MIXER_BESPOKEN type, so that the core USB mixer code
+ * ignores them for resume and other operations.
+ * Also, the head.id field is set to 0, as we don't use this field.
+ */
elem->head.mixer = mixer;
elem->control = index;
- elem->head.id = index;
+ elem->head.id = 0;
elem->channels = channels;
+ elem->val_type = USB_MIXER_BESPOKEN;
kctl = snd_ctl_new1(ncontrol, elem);
if (!kctl) {
@@ -1028,11 +1033,10 @@
struct usb_mixer_interface *mixer = elem->head.mixer;
struct scarlett2_mixer_data *private = mixer->private_data;
- if (private->vol_updated) {
- mutex_lock(&private->data_mutex);
+ mutex_lock(&private->data_mutex);
+ if (private->vol_updated)
scarlett2_update_volumes(mixer);
- mutex_unlock(&private->data_mutex);
- }
+ mutex_unlock(&private->data_mutex);
ucontrol->value.integer.value[0] = private->master_vol;
return 0;
@@ -1046,11 +1050,10 @@
struct scarlett2_mixer_data *private = mixer->private_data;
int index = elem->control;
- if (private->vol_updated) {
- mutex_lock(&private->data_mutex);
+ mutex_lock(&private->data_mutex);
+ if (private->vol_updated)
scarlett2_update_volumes(mixer);
- mutex_unlock(&private->data_mutex);
- }
+ mutex_unlock(&private->data_mutex);
ucontrol->value.integer.value[0] = private->vol[index];
return 0;
@@ -1181,6 +1184,8 @@
/* Send SW/HW switch change to the device */
err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_SW_HW_SWITCH,
index, val);
+ if (err == 0)
+ err = 1;
unlock:
mutex_unlock(&private->data_mutex);
@@ -1241,6 +1246,8 @@
/* Send switch change to the device */
err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_LEVEL_SWITCH,
index, val);
+ if (err == 0)
+ err = 1;
unlock:
mutex_unlock(&private->data_mutex);
@@ -1291,6 +1298,8 @@
/* Send switch change to the device */
err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_PAD_SWITCH,
index, val);
+ if (err == 0)
+ err = 1;
unlock:
mutex_unlock(&private->data_mutex);
@@ -1314,11 +1323,10 @@
struct usb_mixer_interface *mixer = elem->head.mixer;
struct scarlett2_mixer_data *private = mixer->private_data;
- if (private->vol_updated) {
- mutex_lock(&private->data_mutex);
+ mutex_lock(&private->data_mutex);
+ if (private->vol_updated)
scarlett2_update_volumes(mixer);
- mutex_unlock(&private->data_mutex);
- }
+ mutex_unlock(&private->data_mutex);
ucontrol->value.enumerated.item[0] = private->buttons[elem->control];
return 0;
@@ -1347,6 +1355,8 @@
/* Send switch change to the device */
err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_BUTTONS,
index, val);
+ if (err == 0)
+ err = 1;
unlock:
mutex_unlock(&private->data_mutex);
@@ -1997,38 +2007,11 @@
return usb_submit_urb(mixer->urb, GFP_KERNEL);
}
-/* Entry point */
-int snd_scarlett_gen2_controls_create(struct usb_mixer_interface *mixer)
+static int snd_scarlett_gen2_controls_create(struct usb_mixer_interface *mixer,
+ const struct scarlett2_device_info *info)
{
- const struct scarlett2_device_info *info;
int err;
- /* only use UAC_VERSION_2 */
- if (!mixer->protocol)
- return 0;
-
- switch (mixer->chip->usb_id) {
- case USB_ID(0x1235, 0x8203):
- info = &s6i6_gen2_info;
- break;
- case USB_ID(0x1235, 0x8204):
- info = &s18i8_gen2_info;
- break;
- case USB_ID(0x1235, 0x8201):
- info = &s18i20_gen2_info;
- break;
- default: /* device not (yet) supported */
- return -EINVAL;
- }
-
- if (!(mixer->chip->setup & SCARLETT2_ENABLE)) {
- usb_audio_err(mixer->chip,
- "Focusrite Scarlett Gen 2 Mixer Driver disabled; "
- "use options snd_usb_audio device_setup=1 "
- "to enable and report any issues to g@b4.vu");
- return 0;
- }
-
/* Initialise private data, routing, sequence number */
err = scarlett2_init_private(mixer, info);
if (err < 0)
@@ -2073,3 +2056,51 @@
return 0;
}
+
+int snd_scarlett_gen2_init(struct usb_mixer_interface *mixer)
+{
+ struct snd_usb_audio *chip = mixer->chip;
+ const struct scarlett2_device_info *info;
+ int err;
+
+ /* only use UAC_VERSION_2 */
+ if (!mixer->protocol)
+ return 0;
+
+ switch (chip->usb_id) {
+ case USB_ID(0x1235, 0x8203):
+ info = &s6i6_gen2_info;
+ break;
+ case USB_ID(0x1235, 0x8204):
+ info = &s18i8_gen2_info;
+ break;
+ case USB_ID(0x1235, 0x8201):
+ info = &s18i20_gen2_info;
+ break;
+ default: /* device not (yet) supported */
+ return -EINVAL;
+ }
+
+ if (!(chip->setup & SCARLETT2_ENABLE)) {
+ usb_audio_info(chip,
+ "Focusrite Scarlett Gen 2 Mixer Driver disabled; "
+ "use options snd_usb_audio vid=0x%04x pid=0x%04x "
+ "device_setup=1 to enable and report any issues "
+ "to g@b4.vu",
+ USB_ID_VENDOR(chip->usb_id),
+ USB_ID_PRODUCT(chip->usb_id));
+ return 0;
+ }
+
+ usb_audio_info(chip,
+ "Focusrite Scarlett Gen 2 Mixer Driver enabled pid=0x%04x",
+ USB_ID_PRODUCT(chip->usb_id));
+
+ err = snd_scarlett_gen2_controls_create(mixer, info);
+ if (err < 0)
+ usb_audio_err(mixer->chip,
+ "Error initialising Scarlett Mixer Driver: %d",
+ err);
+
+ return err;
+}
diff --git a/sound/usb/mixer_scarlett_gen2.h b/sound/usb/mixer_scarlett_gen2.h
index 52e1dad..668c6b0 100644
--- a/sound/usb/mixer_scarlett_gen2.h
+++ b/sound/usb/mixer_scarlett_gen2.h
@@ -2,6 +2,6 @@
#ifndef __USB_MIXER_SCARLETT_GEN2_H
#define __USB_MIXER_SCARLETT_GEN2_H
-int snd_scarlett_gen2_controls_create(struct usb_mixer_interface *mixer);
+int snd_scarlett_gen2_init(struct usb_mixer_interface *mixer);
#endif /* __USB_MIXER_SCARLETT_GEN2_H */
diff --git a/sound/usb/mixer_us16x08.c b/sound/usb/mixer_us16x08.c
index f0e8e15..c6c834a 100644
--- a/sound/usb/mixer_us16x08.c
+++ b/sound/usb/mixer_us16x08.c
@@ -607,7 +607,7 @@
static int snd_us16x08_meter_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
- uinfo->count = 1;
+ uinfo->count = 34;
uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
uinfo->value.integer.max = 0x7FFF;
uinfo->value.integer.min = 0;
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index ff5ab24..7521853 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -323,6 +323,7 @@
switch (subs->stream->chip->usb_id) {
case USB_ID(0x0763, 0x2030): /* M-Audio Fast Track C400 */
case USB_ID(0x0763, 0x2031): /* M-Audio Fast Track C600 */
+ case USB_ID(0x22f0, 0x0006): /* Allen&Heath Qu-16 */
ep = 0x81;
ifnum = 3;
goto add_sync_ep_from_ifnum;
@@ -332,6 +333,7 @@
ifnum = 2;
goto add_sync_ep_from_ifnum;
case USB_ID(0x2466, 0x8003): /* Fractal Audio Axe-Fx II */
+ case USB_ID(0x0499, 0x172a): /* Yamaha MODX */
ep = 0x86;
ifnum = 2;
goto add_sync_ep_from_ifnum;
@@ -339,15 +341,32 @@
ep = 0x81;
ifnum = 2;
goto add_sync_ep_from_ifnum;
+ case USB_ID(0x1686, 0xf029): /* Zoom UAC-2 */
+ ep = 0x82;
+ ifnum = 2;
+ goto add_sync_ep_from_ifnum;
case USB_ID(0x1397, 0x0001): /* Behringer UFX1604 */
case USB_ID(0x1397, 0x0002): /* Behringer UFX1204 */
ep = 0x81;
ifnum = 1;
goto add_sync_ep_from_ifnum;
- case USB_ID(0x07fd, 0x0004): /* MOTU MicroBook II */
+ case USB_ID(0x07fd, 0x0004): /* MOTU MicroBook II/IIc */
+ /* MicroBook IIc */
+ if (altsd->bInterfaceClass == USB_CLASS_AUDIO)
+ return 0;
+
+ /* MicroBook II */
ep = 0x84;
ifnum = 0;
goto add_sync_ep_from_ifnum;
+ case USB_ID(0x07fd, 0x0008): /* MOTU M Series */
+ case USB_ID(0x31e9, 0x0001): /* Solid State Logic SSL2 */
+ case USB_ID(0x31e9, 0x0002): /* Solid State Logic SSL2+ */
+ case USB_ID(0x0499, 0x172f): /* Steinberg UR22C */
+ case USB_ID(0x0d9a, 0x00df): /* RTX6001 */
+ ep = 0x81;
+ ifnum = 2;
+ goto add_sync_ep_from_ifnum;
case USB_ID(0x0582, 0x01d8): /* BOSS Katana */
/* BOSS Katana amplifiers do not need quirks */
return 0;
@@ -370,7 +389,7 @@
add_sync_ep_from_ifnum:
iface = usb_ifnum_to_if(dev, ifnum);
- if (!iface || iface->num_altsetting == 0)
+ if (!iface || iface->num_altsetting < 2)
return -EINVAL;
alts = &iface->altsetting[1];
@@ -382,6 +401,8 @@
if (!subs->sync_endpoint)
return -EINVAL;
+ subs->sync_endpoint->is_implicit_feedback = 1;
+
subs->data_endpoint->sync_master = subs->sync_endpoint;
return 1;
@@ -480,12 +501,15 @@
implicit_fb ?
SND_USB_ENDPOINT_TYPE_DATA :
SND_USB_ENDPOINT_TYPE_SYNC);
+
if (!subs->sync_endpoint) {
if (is_playback && attr == USB_ENDPOINT_SYNC_NONE)
return 0;
return -EINVAL;
}
+ subs->sync_endpoint->is_implicit_feedback = implicit_fb;
+
subs->data_endpoint->sync_master = subs->sync_endpoint;
return 0;
@@ -506,15 +530,15 @@
if (WARN_ON(!iface))
return -EINVAL;
alts = usb_altnum_to_altsetting(iface, fmt->altsetting);
- altsd = get_iface_desc(alts);
- if (WARN_ON(altsd->bAlternateSetting != fmt->altsetting))
+ if (WARN_ON(!alts))
return -EINVAL;
+ altsd = get_iface_desc(alts);
- if (fmt == subs->cur_audiofmt)
+ if (fmt == subs->cur_audiofmt && !subs->need_setup_fmt)
return 0;
/* close the old interface */
- if (subs->interface >= 0 && subs->interface != fmt->iface) {
+ if (subs->interface >= 0 && (subs->interface != fmt->iface || subs->need_setup_fmt)) {
if (!subs->stream->chip->keep_iface) {
err = usb_set_interface(subs->dev, subs->interface, 0);
if (err < 0) {
@@ -528,6 +552,9 @@
subs->altset_idx = 0;
}
+ if (subs->need_setup_fmt)
+ subs->need_setup_fmt = false;
+
/* set interface */
if (iface->cur_altsetting != alts) {
err = snd_usb_select_mode_quirk(subs, fmt);
@@ -1397,6 +1424,12 @@
// continue;
}
bytes = urb->iso_frame_desc[i].actual_length;
+ if (subs->stream_offset_adj > 0) {
+ unsigned int adj = min(subs->stream_offset_adj, bytes);
+ cp += adj;
+ bytes -= adj;
+ subs->stream_offset_adj -= adj;
+ }
frames = bytes / stride;
if (!subs->txfr_quirk)
bytes = frames * stride;
@@ -1735,6 +1768,13 @@
subs->data_endpoint->retire_data_urb = retire_playback_urb;
subs->running = 0;
return 0;
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ if (subs->stream->chip->setup_fmt_after_resume_quirk) {
+ stop_endpoints(subs, true);
+ subs->need_setup_fmt = true;
+ return 0;
+ }
+ break;
}
return -EINVAL;
@@ -1757,6 +1797,7 @@
return 0;
case SNDRV_PCM_TRIGGER_STOP:
stop_endpoints(subs, false);
+ subs->data_endpoint->retire_data_urb = NULL;
subs->running = 0;
return 0;
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
@@ -1767,6 +1808,13 @@
subs->data_endpoint->retire_data_urb = retire_capture_urb;
subs->running = 1;
return 0;
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ if (subs->stream->chip->setup_fmt_after_resume_quirk) {
+ stop_endpoints(subs, true);
+ subs->need_setup_fmt = true;
+ return 0;
+ }
+ break;
}
return -EINVAL;
@@ -1837,7 +1885,7 @@
{
struct snd_pcm *pcm = subs->stream->pcm;
struct snd_pcm_substream *s = pcm->streams[subs->direction].substream;
- struct device *dev = subs->dev->bus->controller;
+ struct device *dev = subs->dev->bus->sysdev;
if (!snd_usb_use_vmalloc)
snd_pcm_lib_preallocate_pages(s, SNDRV_DMA_TYPE_DEV_SG,
diff --git a/sound/usb/proc.c b/sound/usb/proc.c
index 49e3f17..ffbf4bd 100644
--- a/sound/usb/proc.c
+++ b/sound/usb/proc.c
@@ -60,7 +60,7 @@
static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct snd_info_buffer *buffer)
{
struct audioformat *fp;
- static char *sync_types[4] = {
+ static const char * const sync_types[4] = {
"NONE", "ASYNC", "ADAPTIVE", "SYNC"
};
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 70c338f..441335a 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -25,6 +25,26 @@
.idProduct = prod, \
.bInterfaceClass = USB_CLASS_VENDOR_SPEC
+/* HP Thunderbolt Dock Audio Headset */
+{
+ USB_DEVICE(0x03f0, 0x0269),
+ .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
+ .vendor_name = "HP",
+ .product_name = "Thunderbolt Dock Audio Headset",
+ .profile_name = "HP-Thunderbolt-Dock-Audio-Headset",
+ .ifnum = QUIRK_NO_INTERFACE
+ }
+},
+/* HP Thunderbolt Dock Audio Module */
+{
+ USB_DEVICE(0x03f0, 0x0567),
+ .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
+ .vendor_name = "HP",
+ .product_name = "Thunderbolt Dock Audio Module",
+ .profile_name = "HP-Thunderbolt-Dock-Audio-Module",
+ .ifnum = QUIRK_NO_INTERFACE
+ }
+},
/* FTDI devices */
{
USB_DEVICE(0x0403, 0xb8d8),
@@ -2465,6 +2485,16 @@
}
},
+{
+ USB_DEVICE_VENDOR_SPEC(0x0944, 0x0204),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "KORG, Inc.",
+ /* .product_name = "ToneLab EX", */
+ .ifnum = 3,
+ .type = QUIRK_MIDI_STANDARD_INTERFACE,
+ }
+},
+
/* AKAI devices */
{
USB_DEVICE(0x09e8, 0x0062),
@@ -2677,6 +2707,10 @@
.data = (const struct snd_usb_audio_quirk[]) {
{
.ifnum = 0,
+ .type = QUIRK_AUDIO_STANDARD_MIXER,
+ },
+ {
+ .ifnum = 0,
.type = QUIRK_AUDIO_FIXED_ENDPOINT,
.data = &(const struct audioformat) {
.formats = SNDRV_PCM_FMTBIT_S24_3LE,
@@ -2687,6 +2721,32 @@
.attributes = UAC_EP_CS_ATTR_SAMPLE_RATE,
.endpoint = 0x01,
.ep_attr = USB_ENDPOINT_XFER_ISOC,
+ .datainterval = 1,
+ .maxpacksize = 0x024c,
+ .rates = SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000,
+ .rate_min = 44100,
+ .rate_max = 48000,
+ .nr_rates = 2,
+ .rate_table = (unsigned int[]) {
+ 44100, 48000
+ }
+ }
+ },
+ {
+ .ifnum = 0,
+ .type = QUIRK_AUDIO_FIXED_ENDPOINT,
+ .data = &(const struct audioformat) {
+ .formats = SNDRV_PCM_FMTBIT_S24_3LE,
+ .channels = 2,
+ .iface = 0,
+ .altsetting = 1,
+ .altset_idx = 1,
+ .attributes = 0,
+ .endpoint = 0x82,
+ .ep_attr = USB_ENDPOINT_XFER_ISOC,
+ .datainterval = 1,
+ .maxpacksize = 0x0126,
.rates = SNDRV_PCM_RATE_44100 |
SNDRV_PCM_RATE_48000,
.rate_min = 44100,
@@ -2756,90 +2816,6 @@
.type = QUIRK_MIDI_NOVATION
}
},
-{
- /*
- * Focusrite Scarlett Solo 2nd generation
- * Reports that playback should use Synch: Synchronous
- * while still providing a feedback endpoint. Synchronous causes
- * snapping on some sample rates.
- * Force it to use Synch: Asynchronous.
- */
- USB_DEVICE(0x1235, 0x8205),
- .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
- .ifnum = QUIRK_ANY_INTERFACE,
- .type = QUIRK_COMPOSITE,
- .data = (const struct snd_usb_audio_quirk[]) {
- {
- .ifnum = 1,
- .type = QUIRK_AUDIO_FIXED_ENDPOINT,
- .data = & (const struct audioformat) {
- .formats = SNDRV_PCM_FMTBIT_S32_LE,
- .channels = 2,
- .iface = 1,
- .altsetting = 1,
- .altset_idx = 1,
- .attributes = 0,
- .endpoint = 0x01,
- .ep_attr = USB_ENDPOINT_XFER_ISOC |
- USB_ENDPOINT_SYNC_ASYNC,
- .protocol = UAC_VERSION_2,
- .rates = SNDRV_PCM_RATE_44100 |
- SNDRV_PCM_RATE_48000 |
- SNDRV_PCM_RATE_88200 |
- SNDRV_PCM_RATE_96000 |
- SNDRV_PCM_RATE_176400 |
- SNDRV_PCM_RATE_192000,
- .rate_min = 44100,
- .rate_max = 192000,
- .nr_rates = 6,
- .rate_table = (unsigned int[]) {
- 44100, 48000, 88200,
- 96000, 176400, 192000
- },
- .clock = 41
- }
- },
- {
- .ifnum = 2,
- .type = QUIRK_AUDIO_FIXED_ENDPOINT,
- .data = & (const struct audioformat) {
- .formats = SNDRV_PCM_FMTBIT_S32_LE,
- .channels = 2,
- .iface = 2,
- .altsetting = 1,
- .altset_idx = 1,
- .attributes = 0,
- .endpoint = 0x82,
- .ep_attr = USB_ENDPOINT_XFER_ISOC |
- USB_ENDPOINT_SYNC_ASYNC |
- USB_ENDPOINT_USAGE_IMPLICIT_FB,
- .protocol = UAC_VERSION_2,
- .rates = SNDRV_PCM_RATE_44100 |
- SNDRV_PCM_RATE_48000 |
- SNDRV_PCM_RATE_88200 |
- SNDRV_PCM_RATE_96000 |
- SNDRV_PCM_RATE_176400 |
- SNDRV_PCM_RATE_192000,
- .rate_min = 44100,
- .rate_max = 192000,
- .nr_rates = 6,
- .rate_table = (unsigned int[]) {
- 44100, 48000, 88200,
- 96000, 176400, 192000
- },
- .clock = 41
- }
- },
- {
- .ifnum = 3,
- .type = QUIRK_IGNORE_INTERFACE
- },
- {
- .ifnum = -1
- }
- }
- }
-},
/* Access Music devices */
{
@@ -3466,12 +3442,13 @@
.vendor_name = "Dell",
.product_name = "WD19 Dock",
.profile_name = "Dell-WD15-Dock",
- .ifnum = QUIRK_NO_INTERFACE
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_SETUP_FMT_AFTER_RESUME
}
},
/* MOTU Microbook II */
{
- USB_DEVICE(0x07fd, 0x0004),
+ USB_DEVICE_VENDOR_SPEC(0x07fd, 0x0004),
.driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
.vendor_name = "MOTU",
.product_name = "MicroBookII",
@@ -3591,5 +3568,201 @@
}
}
},
+{
+ /*
+ * Pioneer DJ DJM-250MK2
+ * PCM is 8 channels out @ 48 fixed (endpoints 0x01).
+ * The output from computer to the mixer is usable.
+ *
+ * The input (phono or line to computer) is not working.
+ * It should be at endpoint 0x82 and probably also 8 channels,
+ * but it seems that it works only with Pioneer proprietary software.
+ * Even on officially supported OS, the Audacity was unable to record
+ * and Mixxx to recognize the control vinyls.
+ */
+ USB_DEVICE_VENDOR_SPEC(0x2b73, 0x0017),
+ .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = (const struct snd_usb_audio_quirk[]) {
+ {
+ .ifnum = 0,
+ .type = QUIRK_AUDIO_FIXED_ENDPOINT,
+ .data = &(const struct audioformat) {
+ .formats = SNDRV_PCM_FMTBIT_S24_3LE,
+ .channels = 8, // outputs
+ .iface = 0,
+ .altsetting = 1,
+ .altset_idx = 1,
+ .endpoint = 0x01,
+ .ep_attr = USB_ENDPOINT_XFER_ISOC|
+ USB_ENDPOINT_SYNC_ASYNC,
+ .rates = SNDRV_PCM_RATE_48000,
+ .rate_min = 48000,
+ .rate_max = 48000,
+ .nr_rates = 1,
+ .rate_table = (unsigned int[]) { 48000 }
+ }
+ },
+ {
+ .ifnum = -1
+ }
+ }
+ }
+},
+{
+ /*
+ * PIONEER DJ DDJ-RB
+ * PCM is 4 channels out, 2 dummy channels in @ 44.1 fixed
+ * The feedback for the output is the dummy input.
+ */
+ USB_DEVICE_VENDOR_SPEC(0x2b73, 0x000e),
+ .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = (const struct snd_usb_audio_quirk[]) {
+ {
+ .ifnum = 0,
+ .type = QUIRK_AUDIO_FIXED_ENDPOINT,
+ .data = &(const struct audioformat) {
+ .formats = SNDRV_PCM_FMTBIT_S24_3LE,
+ .channels = 4,
+ .iface = 0,
+ .altsetting = 1,
+ .altset_idx = 1,
+ .endpoint = 0x01,
+ .ep_attr = USB_ENDPOINT_XFER_ISOC|
+ USB_ENDPOINT_SYNC_ASYNC,
+ .rates = SNDRV_PCM_RATE_44100,
+ .rate_min = 44100,
+ .rate_max = 44100,
+ .nr_rates = 1,
+ .rate_table = (unsigned int[]) { 44100 }
+ }
+ },
+ {
+ .ifnum = 0,
+ .type = QUIRK_AUDIO_FIXED_ENDPOINT,
+ .data = &(const struct audioformat) {
+ .formats = SNDRV_PCM_FMTBIT_S24_3LE,
+ .channels = 2,
+ .iface = 0,
+ .altsetting = 1,
+ .altset_idx = 1,
+ .endpoint = 0x82,
+ .ep_attr = USB_ENDPOINT_XFER_ISOC|
+ USB_ENDPOINT_SYNC_ASYNC|
+ USB_ENDPOINT_USAGE_IMPLICIT_FB,
+ .rates = SNDRV_PCM_RATE_44100,
+ .rate_min = 44100,
+ .rate_max = 44100,
+ .nr_rates = 1,
+ .rate_table = (unsigned int[]) { 44100 }
+ }
+ },
+ {
+ .ifnum = -1
+ }
+ }
+ }
+},
+
+#define ALC1220_VB_DESKTOP(vend, prod) { \
+ USB_DEVICE(vend, prod), \
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { \
+ .vendor_name = "Realtek", \
+ .product_name = "ALC1220-VB-DT", \
+ .profile_name = "Realtek-ALC1220-VB-Desktop", \
+ .ifnum = QUIRK_NO_INTERFACE \
+ } \
+}
+ALC1220_VB_DESKTOP(0x0414, 0xa002), /* Gigabyte TRX40 Aorus Pro WiFi */
+ALC1220_VB_DESKTOP(0x0db0, 0x0d64), /* MSI TRX40 Creator */
+ALC1220_VB_DESKTOP(0x0db0, 0x543d), /* MSI TRX40 */
+ALC1220_VB_DESKTOP(0x26ce, 0x0a01), /* Asrock TRX40 Creator */
+#undef ALC1220_VB_DESKTOP
+
+/* Two entries for Gigabyte TRX40 Aorus Master:
+ * TRX40 Aorus Master has two USB-audio devices, one for the front headphone
+ * with ESS SABRE9218 DAC chip, while another for the rest I/O (the rear
+ * panel and the front mic) with Realtek ALC1220-VB.
+ * Here we provide two distinct names for making UCM profiles easier.
+ */
+{
+ USB_DEVICE(0x0414, 0xa000),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "Gigabyte",
+ .product_name = "Aorus Master Front Headphone",
+ .profile_name = "Gigabyte-Aorus-Master-Front-Headphone",
+ .ifnum = QUIRK_NO_INTERFACE
+ }
+},
+{
+ USB_DEVICE(0x0414, 0xa001),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "Gigabyte",
+ .product_name = "Aorus Master Main Audio",
+ .profile_name = "Gigabyte-Aorus-Master-Main-Audio",
+ .ifnum = QUIRK_NO_INTERFACE
+ }
+},
+
+/*
+ * MacroSilicon MS2109 based HDMI capture cards
+ *
+ * These claim 96kHz 1ch in the descriptors, but are actually 48kHz 2ch.
+ * They also need QUIRK_AUDIO_ALIGN_TRANSFER, which makes one wonder if
+ * they pretend to be 96kHz mono as a workaround for stereo being broken
+ * by that...
+ *
+ * They also have an issue with initial stream alignment that causes the
+ * channels to be swapped and out of phase, which is dealt with in quirks.c.
+ */
+{
+ .match_flags = USB_DEVICE_ID_MATCH_DEVICE |
+ USB_DEVICE_ID_MATCH_INT_CLASS |
+ USB_DEVICE_ID_MATCH_INT_SUBCLASS,
+ .idVendor = 0x534d,
+ .idProduct = 0x2109,
+ .bInterfaceClass = USB_CLASS_AUDIO,
+ .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL,
+ .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
+ .vendor_name = "MacroSilicon",
+ .product_name = "MS2109",
+ .ifnum = QUIRK_ANY_INTERFACE,
+ .type = QUIRK_COMPOSITE,
+ .data = &(const struct snd_usb_audio_quirk[]) {
+ {
+ .ifnum = 2,
+ .type = QUIRK_AUDIO_ALIGN_TRANSFER,
+ },
+ {
+ .ifnum = 2,
+ .type = QUIRK_AUDIO_STANDARD_MIXER,
+ },
+ {
+ .ifnum = 3,
+ .type = QUIRK_AUDIO_FIXED_ENDPOINT,
+ .data = &(const struct audioformat) {
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .channels = 2,
+ .iface = 3,
+ .altsetting = 1,
+ .altset_idx = 1,
+ .attributes = 0,
+ .endpoint = 0x82,
+ .ep_attr = USB_ENDPOINT_XFER_ISOC |
+ USB_ENDPOINT_SYNC_ASYNC,
+ .rates = SNDRV_PCM_RATE_CONTINUOUS,
+ .rate_min = 48000,
+ .rate_max = 48000,
+ }
+ },
+ {
+ .ifnum = -1
+ }
+ }
+ }
+},
#undef USB_DEVICE_VENDOR_SPEC
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 349e1e5..d5d8288 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -55,8 +55,12 @@
if (!iface)
continue;
if (quirk->ifnum != probed_ifnum &&
- !usb_interface_claimed(iface))
- usb_driver_claim_interface(driver, iface, (void *)-1L);
+ !usb_interface_claimed(iface)) {
+ err = usb_driver_claim_interface(driver, iface,
+ USB_AUDIO_IFACE_UNUSED);
+ if (err < 0)
+ return err;
+ }
}
return 0;
@@ -390,8 +394,12 @@
continue;
err = create_autodetect_quirk(chip, iface, driver);
- if (err >= 0)
- usb_driver_claim_interface(driver, iface, (void *)-1L);
+ if (err >= 0) {
+ err = usb_driver_claim_interface(driver, iface,
+ USB_AUDIO_IFACE_UNUSED);
+ if (err < 0)
+ return err;
+ }
}
return 0;
@@ -508,6 +516,16 @@
return snd_usb_create_mixer(chip, quirk->ifnum, 0);
}
+
+static int setup_fmt_after_resume_quirk(struct snd_usb_audio *chip,
+ struct usb_interface *iface,
+ struct usb_driver *driver,
+ const struct snd_usb_audio_quirk *quirk)
+{
+ chip->setup_fmt_after_resume_quirk = 1;
+ return 1; /* Continue with creating streams and mixer */
+}
+
/*
* audio-interface quirks
*
@@ -546,6 +564,7 @@
[QUIRK_AUDIO_EDIROL_UAXX] = create_uaxx_quirk,
[QUIRK_AUDIO_ALIGN_TRANSFER] = create_align_transfer_quirk,
[QUIRK_AUDIO_STANDARD_MIXER] = create_standard_mixer_quirk,
+ [QUIRK_SETUP_FMT_AFTER_RESUME] = setup_fmt_after_resume_quirk,
};
if (quirk->type < QUIRK_TYPE_COUNT) {
@@ -1102,6 +1121,31 @@
return err;
}
+static int snd_usb_motu_m_series_boot_quirk(struct usb_device *dev)
+{
+ int ret;
+
+ if (snd_usb_pipe_sanity_check(dev, usb_sndctrlpipe(dev, 0)))
+ return -EINVAL;
+ ret = usb_control_msg(dev, usb_sndctrlpipe(dev, 0),
+ 1, USB_TYPE_VENDOR | USB_RECIP_DEVICE,
+ 0x0, 0, NULL, 0, 1000);
+
+ if (ret < 0)
+ return ret;
+
+ msleep(2000);
+
+ ret = usb_control_msg(dev, usb_sndctrlpipe(dev, 0),
+ 1, USB_TYPE_VENDOR | USB_RECIP_DEVICE,
+ 0x20, 0, NULL, 0, 1000);
+
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
/*
* Setup quirks
*/
@@ -1280,7 +1324,28 @@
case USB_ID(0x2466, 0x8010): /* Fractal Audio Axe-Fx 3 */
return snd_usb_axefx3_boot_quirk(dev);
case USB_ID(0x07fd, 0x0004): /* MOTU MicroBook II */
- return snd_usb_motu_microbookii_boot_quirk(dev);
+ /*
+ * For some reason interface 3 with vendor-spec class is
+ * detected on MicroBook IIc.
+ */
+ if (get_iface_desc(intf->altsetting)->bInterfaceClass ==
+ USB_CLASS_VENDOR_SPEC &&
+ get_iface_desc(intf->altsetting)->bInterfaceNumber < 3)
+ return snd_usb_motu_microbookii_boot_quirk(dev);
+ break;
+ }
+
+ return 0;
+}
+
+int snd_usb_apply_boot_quirk_once(struct usb_device *dev,
+ struct usb_interface *intf,
+ const struct snd_usb_audio_quirk *quirk,
+ unsigned int id)
+{
+ switch (id) {
+ case USB_ID(0x07fd, 0x0008): /* MOTU M Series */
+ return snd_usb_motu_m_series_boot_quirk(dev);
}
return 0;
@@ -1375,6 +1440,9 @@
case USB_ID(0x041e, 0x3f19): /* E-Mu 0204 USB */
set_format_emu_quirk(subs, fmt);
break;
+ case USB_ID(0x534d, 0x2109): /* MacroSilicon MS2109 */
+ subs->stream_offset_adj = 2;
+ break;
}
}
@@ -1386,10 +1454,14 @@
case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */
case USB_ID(0x0556, 0x0014): /* Phoenix Audio TMX320VC */
case USB_ID(0x05A3, 0x9420): /* ELP HD USB Camera */
+ case USB_ID(0x05a7, 0x1020): /* Bose Companion 5 */
case USB_ID(0x074D, 0x3553): /* Outlaw RR2150 (Micronas UAC3553B) */
case USB_ID(0x1395, 0x740a): /* Sennheiser DECT */
case USB_ID(0x1901, 0x0191): /* GE B850V3 CP2114 audio interface */
case USB_ID(0x21B4, 0x0081): /* AudioQuest DragonFly */
+ case USB_ID(0x2912, 0x30c8): /* Audioengine D1 */
+ case USB_ID(0x413c, 0xa506): /* Dell AE515 sound bar */
+ case USB_ID(0x046d, 0x084c): /* Logitech ConferenceCam Connect */
return true;
}
@@ -1410,6 +1482,7 @@
static bool is_itf_usb_dsd_dac(unsigned int id)
{
switch (id) {
+ case USB_ID(0x154e, 0x1002): /* Denon DCD-1500RE */
case USB_ID(0x154e, 0x1003): /* Denon DA-300USB */
case USB_ID(0x154e, 0x3005): /* Marantz HD-DAC1 */
case USB_ID(0x154e, 0x3006): /* Marantz SA-14S1 */
@@ -1541,15 +1614,33 @@
&& (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS)
msleep(20);
- /* Zoom R16/24, Logitech H650e, Jabra 550a needs a tiny delay here,
- * otherwise requests like get/set frequency return as failed despite
- * actually succeeding.
+ /*
+ * Plantronics headsets (C320, C320-M, etc) need a delay to avoid
+ * random microhpone failures.
+ */
+ if (USB_ID_VENDOR(chip->usb_id) == 0x047f &&
+ (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS)
+ msleep(20);
+
+ /* Zoom R16/24, many Logitech(at least H650e/H570e/BCC950),
+ * Jabra 550a, Kingston HyperX needs a tiny delay here,
+ * otherwise requests like get/set frequency return
+ * as failed despite actually succeeding.
*/
if ((chip->usb_id == USB_ID(0x1686, 0x00dd) ||
- chip->usb_id == USB_ID(0x046d, 0x0a46) ||
- chip->usb_id == USB_ID(0x0b0e, 0x0349)) &&
+ USB_ID_VENDOR(chip->usb_id) == 0x046d || /* Logitech */
+ chip->usb_id == USB_ID(0x0b0e, 0x0349) ||
+ chip->usb_id == USB_ID(0x0951, 0x16ad)) &&
(requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS)
usleep_range(1000, 2000);
+
+ /*
+ * Samsung USBC Headset (AKG) need a tiny delay after each
+ * class compliant request. (Model number: AAM625R or AAM627R)
+ */
+ if (chip->usb_id == USB_ID(0x04e8, 0xa051) &&
+ (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS)
+ usleep_range(5000, 6000);
}
/*
@@ -1592,7 +1683,7 @@
case USB_ID(0x0d8c, 0x0316): /* Hegel HD12 DSD */
case USB_ID(0x10cb, 0x0103): /* The Bit Opus #3; with fp->dsd_raw */
- case USB_ID(0x16b0, 0x06b2): /* NuPrime DAC-10 */
+ case USB_ID(0x16d0, 0x06b2): /* NuPrime DAC-10 */
case USB_ID(0x16d0, 0x09dd): /* Encore mDSD */
case USB_ID(0x16d0, 0x0733): /* Furutech ADL Stratos */
case USB_ID(0x16d0, 0x09db): /* NuPrime Audio DAC-9 */
@@ -1658,7 +1749,9 @@
case 0x25ce: /* Mytek devices */
case 0x278b: /* Rotel? */
case 0x292b: /* Gustard/Ess based devices */
+ case 0x2972: /* FiiO devices */
case 0x2ab6: /* T+A devices */
+ case 0x3353: /* Khadas devices */
case 0x3842: /* EVGA */
case 0xc502: /* HiBy devices */
if (fp->dsd_raw)
@@ -1703,5 +1796,67 @@
else
fp->ep_attr |= USB_ENDPOINT_SYNC_SYNC;
break;
+ case USB_ID(0x07fd, 0x0004): /* MOTU MicroBook IIc */
+ /*
+ * MaxPacketsOnly attribute is erroneously set in endpoint
+ * descriptors. As a result this card produces noise with
+ * all sample rates other than 96 KHz.
+ */
+ fp->attributes &= ~UAC_EP_CS_ATTR_FILL_MAX;
+ break;
+ case USB_ID(0x1235, 0x8202): /* Focusrite Scarlett 2i2 2nd gen */
+ case USB_ID(0x1235, 0x8205): /* Focusrite Scarlett Solo 2nd gen */
+ /*
+ * Reports that playback should use Synch: Synchronous
+ * while still providing a feedback endpoint.
+ * Synchronous causes snapping on some sample rates.
+ * Force it to use Synch: Asynchronous.
+ */
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ fp->ep_attr &= ~USB_ENDPOINT_SYNCTYPE;
+ fp->ep_attr |= USB_ENDPOINT_SYNC_ASYNC;
+ }
+ break;
}
}
+
+/*
+ * registration quirk:
+ * the registration is skipped if a device matches with the given ID,
+ * unless the interface reaches to the defined one. This is for delaying
+ * the registration until the last known interface, so that the card and
+ * devices appear at the same time.
+ */
+
+struct registration_quirk {
+ unsigned int usb_id; /* composed via USB_ID() */
+ unsigned int interface; /* the interface to trigger register */
+};
+
+#define REG_QUIRK_ENTRY(vendor, product, iface) \
+ { .usb_id = USB_ID(vendor, product), .interface = (iface) }
+
+static const struct registration_quirk registration_quirks[] = {
+ REG_QUIRK_ENTRY(0x0951, 0x16d8, 2), /* Kingston HyperX AMP */
+ REG_QUIRK_ENTRY(0x0951, 0x16ed, 2), /* Kingston HyperX Cloud Alpha S */
+ REG_QUIRK_ENTRY(0x0951, 0x16ea, 2), /* Kingston HyperX Cloud Flight S */
+ REG_QUIRK_ENTRY(0x0ecb, 0x1f46, 2), /* JBL Quantum 600 */
+ REG_QUIRK_ENTRY(0x0ecb, 0x1f47, 2), /* JBL Quantum 800 */
+ REG_QUIRK_ENTRY(0x0ecb, 0x2039, 2), /* JBL Quantum 400 */
+ REG_QUIRK_ENTRY(0x0ecb, 0x203c, 2), /* JBL Quantum 600 */
+ REG_QUIRK_ENTRY(0x0ecb, 0x203e, 2), /* JBL Quantum 800 */
+ { 0 } /* terminator */
+};
+
+/* return true if skipping registration */
+bool snd_usb_registration_quirk(struct snd_usb_audio *chip, int iface)
+{
+ const struct registration_quirk *q;
+
+ for (q = registration_quirks; q->usb_id; q++)
+ if (chip->usb_id == q->usb_id)
+ return iface != q->interface;
+
+ /* Register as normal */
+ return false;
+}
diff --git a/sound/usb/quirks.h b/sound/usb/quirks.h
index a80e0dd..c76cf24 100644
--- a/sound/usb/quirks.h
+++ b/sound/usb/quirks.h
@@ -20,6 +20,11 @@
const struct snd_usb_audio_quirk *quirk,
unsigned int usb_id);
+int snd_usb_apply_boot_quirk_once(struct usb_device *dev,
+ struct usb_interface *intf,
+ const struct snd_usb_audio_quirk *quirk,
+ unsigned int usb_id);
+
void snd_usb_set_format_quirk(struct snd_usb_substream *subs,
struct audioformat *fmt);
@@ -46,4 +51,6 @@
struct audioformat *fp,
int stream);
+bool snd_usb_registration_quirk(struct snd_usb_audio *chip, int iface);
+
#endif /* __USBAUDIO_QUIRKS_H */
diff --git a/sound/usb/stream.c b/sound/usb/stream.c
index 11785f9..eff1ac1 100644
--- a/sound/usb/stream.c
+++ b/sound/usb/stream.c
@@ -94,6 +94,7 @@
subs->tx_length_quirk = as->chip->tx_length_quirk;
subs->speed = snd_usb_get_speed(subs->dev);
subs->pkt_offset_adj = 0;
+ subs->stream_offset_adj = 0;
snd_usb_set_pcm_ops(as->pcm, stream);
@@ -192,16 +193,16 @@
struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol);
struct snd_usb_substream *subs = info->private_data;
struct snd_pcm_chmap_elem *chmap = NULL;
- int i;
+ int i = 0;
- memset(ucontrol->value.integer.value, 0,
- sizeof(ucontrol->value.integer.value));
if (subs->cur_audiofmt)
chmap = subs->cur_audiofmt->chmap;
if (chmap) {
for (i = 0; i < chmap->channels; i++)
ucontrol->value.integer.value[i] = chmap->map[i];
}
+ for (; i < subs->channels_max; i++)
+ ucontrol->value.integer.value[i] = 0;
return 0;
}
@@ -239,7 +240,7 @@
static struct snd_pcm_chmap_elem *convert_chmap(int channels, unsigned int bits,
int protocol)
{
- static unsigned int uac1_maps[] = {
+ static const unsigned int uac1_maps[] = {
SNDRV_CHMAP_FL, /* left front */
SNDRV_CHMAP_FR, /* right front */
SNDRV_CHMAP_FC, /* center front */
@@ -254,7 +255,7 @@
SNDRV_CHMAP_TC, /* top */
0 /* terminator */
};
- static unsigned int uac2_maps[] = {
+ static const unsigned int uac2_maps[] = {
SNDRV_CHMAP_FL, /* front left */
SNDRV_CHMAP_FR, /* front right */
SNDRV_CHMAP_FC, /* front center */
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index feb30f9..ff97fdc 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -26,14 +26,14 @@
struct usb_interface *pm_intf;
u32 usb_id;
struct mutex mutex;
- unsigned int autosuspended:1;
+ unsigned int system_suspend;
atomic_t active;
atomic_t shutdown;
atomic_t usage_count;
wait_queue_head_t shutdown_wait;
unsigned int txfr_quirk:1; /* Subframe boundaries on transfers */
unsigned int tx_length_quirk:1; /* Put length specifier in transfers */
-
+ unsigned int setup_fmt_after_resume_quirk:1; /* setup the format to interface after resume */
int num_interfaces;
int num_suspended_intf;
int sample_rate_read_error;
@@ -59,6 +59,8 @@
struct media_intf_devnode *ctl_intf_media_devnode;
};
+#define USB_AUDIO_IFACE_UNUSED ((void *)-1L)
+
#define usb_audio_err(chip, fmt, args...) \
dev_err(&(chip)->dev->dev, fmt, ##args)
#define usb_audio_warn(chip, fmt, args...) \
@@ -98,6 +100,7 @@
QUIRK_AUDIO_EDIROL_UAXX,
QUIRK_AUDIO_ALIGN_TRANSFER,
QUIRK_AUDIO_STANDARD_MIXER,
+ QUIRK_SETUP_FMT_AFTER_RESUME,
QUIRK_TYPE_COUNT
};
diff --git a/sound/usb/usx2y/usX2Yhwdep.c b/sound/usb/usx2y/usX2Yhwdep.c
index d1caa8e..9985fc1 100644
--- a/sound/usb/usx2y/usX2Yhwdep.c
+++ b/sound/usb/usx2y/usX2Yhwdep.c
@@ -119,7 +119,7 @@
info->num_dsps = 2; // 0: Prepad Data, 1: FPGA Code
if (us428->chip_status & USX2Y_STAT_CHIP_INIT)
info->chip_ready = 1;
- info->version = USX2Y_DRIVER_VERSION;
+ info->version = USX2Y_DRIVER_VERSION;
return 0;
}
diff --git a/sound/usb/usx2y/usb_stream.c b/sound/usb/usx2y/usb_stream.c
index 091c071..cff6849 100644
--- a/sound/usb/usx2y/usb_stream.c
+++ b/sound/usb/usx2y/usb_stream.c
@@ -142,8 +142,11 @@
if (!s)
return;
- free_pages_exact(sk->write_page, s->write_size);
- sk->write_page = NULL;
+ if (sk->write_page) {
+ free_pages_exact(sk->write_page, s->write_size);
+ sk->write_page = NULL;
+ }
+
free_pages_exact(s, s->read_size);
sk->s = NULL;
}
diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c
index 89fa287..e0bace4 100644
--- a/sound/usb/usx2y/usbusx2yaudio.c
+++ b/sound/usb/usx2y/usbusx2yaudio.c
@@ -681,6 +681,8 @@
us->submitted = 2*NOOF_SETRATE_URBS;
for (i = 0; i < NOOF_SETRATE_URBS; ++i) {
struct urb *urb = us->urb[i];
+ if (!urb)
+ continue;
if (urb->status) {
if (!err)
err = -ENODEV;
diff --git a/sound/usb/validate.c b/sound/usb/validate.c
index 389e865..89a48d7 100644
--- a/sound/usb/validate.c
+++ b/sound/usb/validate.c
@@ -110,7 +110,7 @@
default:
if (v->type == UAC1_EXTENSION_UNIT)
return true; /* OK */
- switch (d->wProcessType) {
+ switch (le16_to_cpu(d->wProcessType)) {
case UAC_PROCESS_UP_DOWNMIX:
case UAC_PROCESS_DOLBY_PROLOGIC:
if (d->bLength < len + 1) /* bNrModes */
@@ -125,7 +125,7 @@
case UAC_VERSION_2:
if (v->type == UAC2_EXTENSION_UNIT_V2)
return true; /* OK */
- switch (d->wProcessType) {
+ switch (le16_to_cpu(d->wProcessType)) {
case UAC2_PROCESS_UP_DOWNMIX:
case UAC2_PROCESS_DOLBY_PROLOCIC: /* SiC! */
if (d->bLength < len + 1) /* bNrModes */
@@ -142,7 +142,7 @@
len += 2; /* wClusterDescrID */
break;
}
- switch (d->wProcessType) {
+ switch (le16_to_cpu(d->wProcessType)) {
case UAC3_PROCESS_UP_DOWNMIX:
if (d->bLength < len + 1) /* bNrModes */
return false;
@@ -233,7 +233,7 @@
#define FIXED(p, t, s) { .protocol = (p), .type = (t), .size = sizeof(s) }
#define FUNC(p, t, f) { .protocol = (p), .type = (t), .func = (f) }
-static struct usb_desc_validator audio_validators[] = {
+static const struct usb_desc_validator audio_validators[] = {
/* UAC1 */
FUNC(UAC_VERSION_1, UAC_HEADER, validate_uac1_header),
FIXED(UAC_VERSION_1, UAC_INPUT_TERMINAL,
@@ -288,7 +288,7 @@
{ } /* terminator */
};
-static struct usb_desc_validator midi_validators[] = {
+static const struct usb_desc_validator midi_validators[] = {
FIXED(UAC_VERSION_ALL, USB_MS_HEADER,
struct usb_ms_header_descriptor),
FIXED(UAC_VERSION_ALL, USB_MS_MIDI_IN_JACK,