diff options
Diffstat (limited to 'sound')
77 files changed, 748 insertions, 274 deletions
diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 555df64d46ff..7ae8e24dc1e6 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -529,7 +529,7 @@ static int snd_compress_check_input(struct snd_compr_params *params) { /* first let's check the buffer parameter's */ if (params->buffer.fragment_size == 0 || - params->buffer.fragments > INT_MAX / params->buffer.fragment_size || + params->buffer.fragments > U32_MAX / params->buffer.fragment_size || params->buffer.fragments == 0) return -EINVAL; @@ -575,10 +575,7 @@ snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg) stream->metadata_set = false; stream->next_track = false; - if (stream->direction == SND_COMPRESS_PLAYBACK) - stream->runtime->state = SNDRV_PCM_STATE_SETUP; - else - stream->runtime->state = SNDRV_PCM_STATE_PREPARED; + stream->runtime->state = SNDRV_PCM_STATE_SETUP; } else { return -EPERM; } @@ -694,8 +691,17 @@ static int snd_compr_start(struct snd_compr_stream *stream) { int retval; - if (stream->runtime->state != SNDRV_PCM_STATE_PREPARED) + switch (stream->runtime->state) { + case SNDRV_PCM_STATE_SETUP: + if (stream->direction != SND_COMPRESS_CAPTURE) + return -EPERM; + break; + case SNDRV_PCM_STATE_PREPARED: + break; + default: return -EPERM; + } + retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_START); if (!retval) stream->runtime->state = SNDRV_PCM_STATE_RUNNING; @@ -706,9 +712,15 @@ static int snd_compr_stop(struct snd_compr_stream *stream) { int retval; - if (stream->runtime->state == SNDRV_PCM_STATE_PREPARED || - stream->runtime->state == SNDRV_PCM_STATE_SETUP) + switch (stream->runtime->state) { + case SNDRV_PCM_STATE_OPEN: + case SNDRV_PCM_STATE_SETUP: + case SNDRV_PCM_STATE_PREPARED: return -EPERM; + default: + break; + } + retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_STOP); if (!retval) { snd_compr_drain_notify(stream); @@ -796,9 +808,17 @@ static int snd_compr_drain(struct snd_compr_stream *stream) { int retval; - if (stream->runtime->state == SNDRV_PCM_STATE_PREPARED || - stream->runtime->state == SNDRV_PCM_STATE_SETUP) + switch (stream->runtime->state) { + case SNDRV_PCM_STATE_OPEN: + case SNDRV_PCM_STATE_SETUP: + case SNDRV_PCM_STATE_PREPARED: + case SNDRV_PCM_STATE_PAUSED: return -EPERM; + case SNDRV_PCM_STATE_XRUN: + return -EPIPE; + default: + break; + } retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_DRAIN); if (retval) { @@ -818,6 +838,10 @@ static int snd_compr_next_track(struct snd_compr_stream *stream) if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) return -EPERM; + /* next track doesn't have any meaning for capture streams */ + if (stream->direction == SND_COMPRESS_CAPTURE) + return -EPERM; + /* you can signal next track if this is intended to be a gapless stream * and current track metadata is set */ @@ -835,9 +859,23 @@ static int snd_compr_next_track(struct snd_compr_stream *stream) static int snd_compr_partial_drain(struct snd_compr_stream *stream) { int retval; - if (stream->runtime->state == SNDRV_PCM_STATE_PREPARED || - stream->runtime->state == SNDRV_PCM_STATE_SETUP) + + switch (stream->runtime->state) { + case SNDRV_PCM_STATE_OPEN: + case SNDRV_PCM_STATE_SETUP: + case SNDRV_PCM_STATE_PREPARED: + case SNDRV_PCM_STATE_PAUSED: + return -EPERM; + case SNDRV_PCM_STATE_XRUN: + return -EPIPE; + default: + break; + } + + /* partial drain doesn't have any meaning for capture streams */ + if (stream->direction == SND_COMPRESS_CAPTURE) return -EPERM; + /* stream can be drained only when next track has been signalled */ if (stream->next_track == false) return -EPERM; diff --git a/sound/core/oss/linear.c b/sound/core/oss/linear.c index 2045697f449d..797d838a2f9e 100644 --- a/sound/core/oss/linear.c +++ b/sound/core/oss/linear.c @@ -107,6 +107,8 @@ static snd_pcm_sframes_t linear_transfer(struct snd_pcm_plugin *plugin, } } #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 7915564bd394..3788906421a7 100644 --- a/sound/core/oss/mulaw.c +++ b/sound/core/oss/mulaw.c @@ -269,6 +269,8 @@ static snd_pcm_sframes_t mulaw_transfer(struct snd_pcm_plugin *plugin, } } #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; diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index 617845d4a811..b8ab46b8298d 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -111,7 +111,7 @@ int snd_pcm_plug_alloc(struct snd_pcm_substream *plug, snd_pcm_uframes_t frames) while (plugin->next) { if (plugin->dst_frames) frames = plugin->dst_frames(plugin, frames); - if (snd_BUG_ON(frames <= 0)) + if (snd_BUG_ON((snd_pcm_sframes_t)frames <= 0)) return -ENXIO; plugin = plugin->next; err = snd_pcm_plugin_alloc(plugin, frames); @@ -123,7 +123,7 @@ int snd_pcm_plug_alloc(struct snd_pcm_substream *plug, snd_pcm_uframes_t frames) while (plugin->prev) { if (plugin->src_frames) frames = plugin->src_frames(plugin, frames); - if (snd_BUG_ON(frames <= 0)) + if (snd_BUG_ON((snd_pcm_sframes_t)frames <= 0)) return -ENXIO; plugin = plugin->prev; err = snd_pcm_plugin_alloc(plugin, frames); diff --git a/sound/core/oss/route.c b/sound/core/oss/route.c index c8171f5783c8..72dea04197ef 100644 --- a/sound/core/oss/route.c +++ b/sound/core/oss/route.c @@ -57,6 +57,8 @@ static snd_pcm_sframes_t route_transfer(struct snd_pcm_plugin *plugin, 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 729a85a6211d..80453266a2de 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1803,11 +1803,14 @@ void snd_pcm_period_elapsed(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime; unsigned long flags; - if (PCM_RUNTIME_CHECK(substream)) + if (snd_BUG_ON(!substream)) return; - runtime = substream->runtime; snd_pcm_stream_lock_irqsave(substream, flags); + if (PCM_RUNTIME_CHECK(substream)) + goto _unlock; + runtime = substream->runtime; + if (!snd_pcm_running(substream) || snd_pcm_update_hw_ptr0(substream, 1) < 0) goto _end; @@ -1818,6 +1821,7 @@ void snd_pcm_period_elapsed(struct snd_pcm_substream *substream) #endif _end: kill_fasync(&runtime->fasync, SIGIO, POLL_IN); + _unlock: snd_pcm_stream_unlock_irqrestore(substream, flags); } EXPORT_SYMBOL(snd_pcm_period_elapsed); diff --git a/sound/core/seq/oss/seq_oss_ioctl.c b/sound/core/seq/oss/seq_oss_ioctl.c index 5b8520177b0e..7d72e3d48ad5 100644 --- a/sound/core/seq/oss/seq_oss_ioctl.c +++ b/sound/core/seq/oss/seq_oss_ioctl.c @@ -62,7 +62,7 @@ static int snd_seq_oss_oob_user(struct seq_oss_devinfo *dp, void __user *arg) if (copy_from_user(ev, arg, 8)) return -EFAULT; memset(&tmpev, 0, sizeof(tmpev)); - snd_seq_oss_fill_addr(dp, &tmpev, dp->addr.port, dp->addr.client); + snd_seq_oss_fill_addr(dp, &tmpev, dp->addr.client, dp->addr.port); tmpev.time.tick = 0; if (! snd_seq_oss_process_event(dp, (union evrec *)ev, &tmpev)) { snd_seq_oss_dispatch(dp, &tmpev, 0, 0); diff --git a/sound/core/seq/oss/seq_oss_rw.c b/sound/core/seq/oss/seq_oss_rw.c index 6a7b6aceeca9..499f3e8f4949 100644 --- a/sound/core/seq/oss/seq_oss_rw.c +++ b/sound/core/seq/oss/seq_oss_rw.c @@ -174,7 +174,7 @@ insert_queue(struct seq_oss_devinfo *dp, union evrec *rec, struct file *opt) memset(&event, 0, sizeof(event)); /* set dummy -- to be sure */ event.type = SNDRV_SEQ_EVENT_NOTEOFF; - snd_seq_oss_fill_addr(dp, &event, dp->addr.port, dp->addr.client); + snd_seq_oss_fill_addr(dp, &event, dp->addr.client, dp->addr.port); if (snd_seq_oss_process_event(dp, rec, &event)) return 0; /* invalid event - no need to insert queue */ diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 3bcd7a2f0394..92b0d4523a07 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -1001,7 +1001,7 @@ static ssize_t snd_seq_write(struct file *file, const char __user *buf, { struct snd_seq_client *client = file->private_data; int written = 0, len; - int err; + int err, handled; struct snd_seq_event event; if (!(snd_seq_file_flags(file) & SNDRV_SEQ_LFLG_OUTPUT)) @@ -1014,6 +1014,8 @@ static ssize_t snd_seq_write(struct file *file, const char __user *buf, if (!client->accept_output || client->pool == NULL) return -ENXIO; + repeat: + handled = 0; /* allocate the pool now if the pool is not allocated yet */ mutex_lock(&client->ioctl_mutex); if (client->pool->size > 0 && !snd_seq_write_pool_allocated(client)) { @@ -1073,12 +1075,19 @@ static ssize_t snd_seq_write(struct file *file, const char __user *buf, 0, 0, &client->ioctl_mutex); if (err < 0) break; + handled++; __skip_event: /* Update pointers and counts */ count -= len; buf += len; written += len; + + /* let's have a coffee break if too many events are queued */ + if (++handled >= 200) { + mutex_unlock(&client->ioctl_mutex); + goto repeat; + } } out: @@ -1812,8 +1821,7 @@ static int snd_seq_ioctl_get_client_pool(struct snd_seq_client *client, if (cptr->type == USER_CLIENT) { info->input_pool = cptr->data.user.fifo_pool_size; info->input_free = info->input_pool; - if (cptr->data.user.fifo) - info->input_free = snd_seq_unused_cells(cptr->data.user.fifo->pool); + info->input_free = snd_seq_fifo_unused_cells(cptr->data.user.fifo); } else { info->input_pool = 0; info->input_free = 0; @@ -1904,20 +1912,14 @@ static int snd_seq_ioctl_get_subscription(struct snd_seq_client *client, int result; struct snd_seq_client *sender = NULL; struct snd_seq_client_port *sport = NULL; - struct snd_seq_subscribers *p; result = -EINVAL; if ((sender = snd_seq_client_use_ptr(subs->sender.client)) == NULL) goto __end; if ((sport = snd_seq_port_use_ptr(sender, subs->sender.port)) == NULL) goto __end; - p = snd_seq_port_get_subscription(&sport->c_src, &subs->dest); - if (p) { - result = 0; - *subs = p->info; - } else - result = -ENOENT; - + result = snd_seq_port_get_subscription(&sport->c_src, &subs->dest, + subs); __end: if (sport) snd_seq_port_unlock(sport); diff --git a/sound/core/seq/seq_fifo.c b/sound/core/seq/seq_fifo.c index 72c0302a55d2..6a24732704fc 100644 --- a/sound/core/seq/seq_fifo.c +++ b/sound/core/seq/seq_fifo.c @@ -280,3 +280,20 @@ int snd_seq_fifo_resize(struct snd_seq_fifo *f, int poolsize) return 0; } + +/* get the number of unused cells safely */ +int snd_seq_fifo_unused_cells(struct snd_seq_fifo *f) +{ + unsigned long flags; + int cells; + + if (!f) + return 0; + + snd_use_lock_use(&f->use_lock); + spin_lock_irqsave(&f->lock, flags); + cells = snd_seq_unused_cells(f->pool); + spin_unlock_irqrestore(&f->lock, flags); + snd_use_lock_free(&f->use_lock); + return cells; +} diff --git a/sound/core/seq/seq_fifo.h b/sound/core/seq/seq_fifo.h index 062c446e7867..5d38a0d7f0cd 100644 --- a/sound/core/seq/seq_fifo.h +++ b/sound/core/seq/seq_fifo.h @@ -68,5 +68,7 @@ int snd_seq_fifo_poll_wait(struct snd_seq_fifo *f, struct file *file, poll_table /* resize pool in fifo */ int snd_seq_fifo_resize(struct snd_seq_fifo *f, int poolsize); +/* get the number of unused cells safely */ +int snd_seq_fifo_unused_cells(struct snd_seq_fifo *f); #endif diff --git a/sound/core/seq/seq_ports.c b/sound/core/seq/seq_ports.c index d3fc73ac230b..c8fa4336bccd 100644 --- a/sound/core/seq/seq_ports.c +++ b/sound/core/seq/seq_ports.c @@ -635,20 +635,23 @@ int snd_seq_port_disconnect(struct snd_seq_client *connector, /* get matched subscriber */ -struct snd_seq_subscribers *snd_seq_port_get_subscription(struct snd_seq_port_subs_info *src_grp, - struct snd_seq_addr *dest_addr) +int snd_seq_port_get_subscription(struct snd_seq_port_subs_info *src_grp, + struct snd_seq_addr *dest_addr, + struct snd_seq_port_subscribe *subs) { - struct snd_seq_subscribers *s, *found = NULL; + struct snd_seq_subscribers *s; + int err = -ENOENT; down_read(&src_grp->list_mutex); list_for_each_entry(s, &src_grp->list_head, src_list) { if (addr_match(dest_addr, &s->info.dest)) { - found = s; + *subs = s->info; + err = 0; break; } } up_read(&src_grp->list_mutex); - return found; + return err; } /* diff --git a/sound/core/seq/seq_ports.h b/sound/core/seq/seq_ports.h index 26bd71f36c41..06003b36652e 100644 --- a/sound/core/seq/seq_ports.h +++ b/sound/core/seq/seq_ports.h @@ -135,7 +135,8 @@ int snd_seq_port_subscribe(struct snd_seq_client_port *port, struct snd_seq_port_subscribe *info); /* get matched subscriber */ -struct snd_seq_subscribers *snd_seq_port_get_subscription(struct snd_seq_port_subs_info *src_grp, - struct snd_seq_addr *dest_addr); +int snd_seq_port_get_subscription(struct snd_seq_port_subs_info *src_grp, + struct snd_seq_addr *dest_addr, + struct snd_seq_port_subscribe *subs); #endif diff --git a/sound/core/seq/seq_system.c b/sound/core/seq/seq_system.c index 8ce1d0b40dce..ce1f1e4727ab 100644 --- a/sound/core/seq/seq_system.c +++ b/sound/core/seq/seq_system.c @@ -123,6 +123,7 @@ int __init snd_seq_system_client_init(void) { struct snd_seq_port_callback pcallbacks; struct snd_seq_port_info *port; + int err; port = kzalloc(sizeof(*port), GFP_KERNEL); if (!port) @@ -144,7 +145,10 @@ int __init snd_seq_system_client_init(void) port->flags = SNDRV_SEQ_PORT_FLG_GIVEN_PORT; port->addr.client = sysclient; port->addr.port = SNDRV_SEQ_PORT_SYSTEM_TIMER; - snd_seq_kernel_client_ctl(sysclient, SNDRV_SEQ_IOCTL_CREATE_PORT, port); + err = snd_seq_kernel_client_ctl(sysclient, SNDRV_SEQ_IOCTL_CREATE_PORT, + port); + if (err < 0) + goto error_port; /* register announcement port */ strcpy(port->name, "Announce"); @@ -154,16 +158,24 @@ int __init snd_seq_system_client_init(void) port->flags = SNDRV_SEQ_PORT_FLG_GIVEN_PORT; port->addr.client = sysclient; port->addr.port = SNDRV_SEQ_PORT_SYSTEM_ANNOUNCE; - snd_seq_kernel_client_ctl(sysclient, SNDRV_SEQ_IOCTL_CREATE_PORT, port); + err = snd_seq_kernel_client_ctl(sysclient, SNDRV_SEQ_IOCTL_CREATE_PORT, + port); + if (err < 0) + goto error_port; announce_port = port->addr.port; kfree(port); return 0; + + error_port: + snd_seq_system_client_done(); + kfree(port); + return err; } /* unregister our internal client */ -void __exit snd_seq_system_client_done(void) +void snd_seq_system_client_done(void) { int oldsysclient = sysclient; diff --git a/sound/core/timer.c b/sound/core/timer.c index 2c0f292226d7..c60dfd52e8a6 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -240,7 +240,8 @@ static int snd_timer_check_master(struct snd_timer_instance *master) return 0; } -static int snd_timer_close_locked(struct snd_timer_instance *timeri); +static int snd_timer_close_locked(struct snd_timer_instance *timeri, + struct device **card_devp_to_put); /* * open a timer instance @@ -252,21 +253,23 @@ int snd_timer_open(struct snd_timer_instance **ti, { struct snd_timer *timer; struct snd_timer_instance *timeri = NULL; + struct device *card_dev_to_put = NULL; int err; + mutex_lock(®ister_mutex); if (tid->dev_class == SNDRV_TIMER_CLASS_SLAVE) { /* open a slave instance */ if (tid->dev_sclass <= SNDRV_TIMER_SCLASS_NONE || tid->dev_sclass > SNDRV_TIMER_SCLASS_OSS_SEQUENCER) { pr_debug("ALSA: timer: invalid slave class %i\n", tid->dev_sclass); - return -EINVAL; + err = -EINVAL; + goto unlock; } - mutex_lock(®ister_mutex); timeri = snd_timer_instance_new(owner, NULL); if (!timeri) { - mutex_unlock(®ister_mutex); - return -ENOMEM; + err = -ENOMEM; + goto unlock; } timeri->slave_class = tid->dev_sclass; timeri->slave_id = tid->device; @@ -274,16 +277,13 @@ int snd_timer_open(struct snd_timer_instance **ti, list_add_tail(&timeri->open_list, &snd_timer_slave_list); err = snd_timer_check_slave(timeri); if (err < 0) { - snd_timer_close_locked(timeri); + snd_timer_close_locked(timeri, &card_dev_to_put); timeri = NULL; } - mutex_unlock(®ister_mutex); - *ti = timeri; - return err; + goto unlock; } /* open a master instance */ - mutex_lock(®ister_mutex); timer = snd_timer_find(tid); #ifdef CONFIG_MODULES if (!timer) { @@ -294,25 +294,26 @@ int snd_timer_open(struct snd_timer_instance **ti, } #endif if (!timer) { - mutex_unlock(®ister_mutex); - return -ENODEV; + err = -ENODEV; + goto unlock; } if (!list_empty(&timer->open_list_head)) { - timeri = list_entry(timer->open_list_head.next, + struct snd_timer_instance *t = + list_entry(timer->open_list_head.next, struct snd_timer_instance, open_list); - if (timeri->flags & SNDRV_TIMER_IFLG_EXCLUSIVE) { - mutex_unlock(®ister_mutex); - return -EBUSY; + if (t->flags & SNDRV_TIMER_IFLG_EXCLUSIVE) { + err = -EBUSY; + goto unlock; } } if (timer->num_instances >= timer->max_instances) { - mutex_unlock(®ister_mutex); - return -EBUSY; + err = -EBUSY; + goto unlock; } timeri = snd_timer_instance_new(owner, timer); if (!timeri) { - mutex_unlock(®ister_mutex); - return -ENOMEM; + err = -ENOMEM; + goto unlock; } /* take a card refcount for safe disconnection */ if (timer->card) @@ -321,16 +322,16 @@ int snd_timer_open(struct snd_timer_instance **ti, timeri->slave_id = slave_id; if (list_empty(&timer->open_list_head) && timer->hw.open) { - int err = timer->hw.open(timer); + err = timer->hw.open(timer); if (err) { kfree(timeri->owner); kfree(timeri); + timeri = NULL; if (timer->card) - put_device(&timer->card->card_dev); + card_dev_to_put = &timer->card->card_dev; module_put(timer->module); - mutex_unlock(®ister_mutex); - return err; + goto unlock; } } @@ -338,10 +339,15 @@ int snd_timer_open(struct snd_timer_instance **ti, timer->num_instances++; err = snd_timer_check_master(timeri); if (err < 0) { - snd_timer_close_locked(timeri); + snd_timer_close_locked(timeri, &card_dev_to_put); timeri = NULL; } + + unlock: mutex_unlock(®ister_mutex); + /* put_device() is called after unlock for avoiding deadlock */ + if (card_dev_to_put) + put_device(card_dev_to_put); *ti = timeri; return err; } @@ -351,7 +357,8 @@ EXPORT_SYMBOL(snd_timer_open); * close a timer instance * call this with register_mutex down. */ -static int snd_timer_close_locked(struct snd_timer_instance *timeri) +static int snd_timer_close_locked(struct snd_timer_instance *timeri, + struct device **card_devp_to_put) { struct snd_timer *timer = NULL; struct snd_timer_instance *slave, *tmp; @@ -403,7 +410,7 @@ static int snd_timer_close_locked(struct snd_timer_instance *timeri) timer->hw.close(timer); /* release a card refcount for safe disconnection */ if (timer->card) - put_device(&timer->card->card_dev); + *card_devp_to_put = &timer->card->card_dev; module_put(timer->module); } @@ -415,14 +422,18 @@ static int snd_timer_close_locked(struct snd_timer_instance *timeri) */ int snd_timer_close(struct snd_timer_instance *timeri) { + struct device *card_dev_to_put = NULL; int err; if (snd_BUG_ON(!timeri)) return -ENXIO; mutex_lock(®ister_mutex); - err = snd_timer_close_locked(timeri); + err = snd_timer_close_locked(timeri, &card_dev_to_put); mutex_unlock(®ister_mutex); + /* put_device() is called after unlock for avoiding deadlock */ + if (card_dev_to_put) + put_device(card_dev_to_put); return err; } EXPORT_SYMBOL(snd_timer_close); diff --git a/sound/firewire/amdtp-am824.c b/sound/firewire/amdtp-am824.c index 23ccddb20de1..8fc554d745f6 100644 --- a/sound/firewire/amdtp-am824.c +++ b/sound/firewire/amdtp-am824.c @@ -321,7 +321,7 @@ static void read_midi_messages(struct amdtp_stream *s, u8 *b; for (f = 0; f < frames; f++) { - port = (s->data_block_counter + f) % 8; + port = (8 - s->tx_first_dbc + s->data_block_counter + f) % 8; b = (u8 *)&buffer[p->midi_position]; len = b[0] - 0x80; diff --git a/sound/firewire/bebob/bebob_focusrite.c b/sound/firewire/bebob/bebob_focusrite.c index 52b8b61ecddd..62d989edd129 100644 --- a/sound/firewire/bebob/bebob_focusrite.c +++ b/sound/firewire/bebob/bebob_focusrite.c @@ -28,6 +28,8 @@ #define SAFFIRE_CLOCK_SOURCE_SPDIF 1 /* clock sources as returned from register of Saffire Pro 10 and 26 */ +#define SAFFIREPRO_CLOCK_SOURCE_SELECT_MASK 0x000000ff +#define SAFFIREPRO_CLOCK_SOURCE_DETECT_MASK 0x0000ff00 #define SAFFIREPRO_CLOCK_SOURCE_INTERNAL 0 #define SAFFIREPRO_CLOCK_SOURCE_SKIP 1 /* never used on hardware */ #define SAFFIREPRO_CLOCK_SOURCE_SPDIF 2 @@ -190,6 +192,7 @@ saffirepro_both_clk_src_get(struct snd_bebob *bebob, unsigned int *id) map = saffirepro_clk_maps[1]; /* In a case that this driver cannot handle the value of register. */ + value &= SAFFIREPRO_CLOCK_SOURCE_SELECT_MASK; if (value >= SAFFIREPRO_CLOCK_SOURCE_COUNT || map[value] < 0) { err = -EIO; goto end; diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c index 4d3034a68bdf..be2c056eb62d 100644 --- a/sound/firewire/bebob/bebob_stream.c +++ b/sound/firewire/bebob/bebob_stream.c @@ -253,8 +253,7 @@ end: return err; } -static unsigned int -map_data_channels(struct snd_bebob *bebob, struct amdtp_stream *s) +static int map_data_channels(struct snd_bebob *bebob, struct amdtp_stream *s) { unsigned int sec, sections, ch, channels; unsigned int pcm, midi, location; diff --git a/sound/firewire/isight.c b/sound/firewire/isight.c index 5826aa8362f1..9edb26ab16e9 100644 --- a/sound/firewire/isight.c +++ b/sound/firewire/isight.c @@ -639,7 +639,7 @@ static int isight_probe(struct fw_unit *unit, if (!isight->audio_base) { dev_err(&unit->device, "audio unit base not found\n"); err = -ENXIO; - goto err_unit; + goto error; } fw_iso_resources_init(&isight->resources, unit); @@ -668,12 +668,12 @@ static int isight_probe(struct fw_unit *unit, dev_set_drvdata(&unit->device, isight); return 0; - -err_unit: - fw_unit_put(isight->unit); - mutex_destroy(&isight->mutex); error: snd_card_free(card); + + mutex_destroy(&isight->mutex); + fw_unit_put(isight->unit); + return err; } diff --git a/sound/firewire/motu/motu-stream.c b/sound/firewire/motu/motu-stream.c index 73e7a5e527fc..483a8771d502 100644 --- a/sound/firewire/motu/motu-stream.c +++ b/sound/firewire/motu/motu-stream.c @@ -345,7 +345,7 @@ static void destroy_stream(struct snd_motu *motu, } amdtp_stream_destroy(stream); - fw_iso_resources_free(resources); + fw_iso_resources_destroy(resources); } int snd_motu_stream_init_duplex(struct snd_motu *motu) diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index 1554dc98e092..6f941720141a 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -176,9 +176,6 @@ static int detect_quirks(struct snd_oxfw *oxfw) oxfw->midi_input_ports = 0; oxfw->midi_output_ports = 0; - /* Output stream exists but no data channels are useful. */ - oxfw->has_output = false; - return snd_oxfw_scs1x_add(oxfw); } diff --git a/sound/firewire/packets-buffer.c b/sound/firewire/packets-buffer.c index ea1506679c66..3b09b8ef3a09 100644 --- a/sound/firewire/packets-buffer.c +++ b/sound/firewire/packets-buffer.c @@ -37,7 +37,7 @@ int iso_packets_buffer_init(struct iso_packets_buffer *b, struct fw_unit *unit, packets_per_page = PAGE_SIZE / packet_size; if (WARN_ON(!packets_per_page)) { err = -EINVAL; - goto error; + goto err_packets; } pages = DIV_ROUND_UP(count, packets_per_page); diff --git a/sound/firewire/tascam/tascam-pcm.c b/sound/firewire/tascam/tascam-pcm.c index 6ec8ec634d4d..87da80727c22 100644 --- a/sound/firewire/tascam/tascam-pcm.c +++ b/sound/firewire/tascam/tascam-pcm.c @@ -57,6 +57,9 @@ static int pcm_open(struct snd_pcm_substream *substream) goto err_locked; err = snd_tscm_stream_get_clock(tscm, &clock); + if (err < 0) + goto err_locked; + if (clock != SND_TSCM_CLOCK_INTERNAL || amdtp_stream_pcm_running(&tscm->rx_stream) || amdtp_stream_pcm_running(&tscm->tx_stream)) { diff --git a/sound/firewire/tascam/tascam-stream.c b/sound/firewire/tascam/tascam-stream.c index f1657a4e0621..a1308f12a65b 100644 --- a/sound/firewire/tascam/tascam-stream.c +++ b/sound/firewire/tascam/tascam-stream.c @@ -9,20 +9,37 @@ #include <linux/delay.h> #include "tascam.h" +#define CLOCK_STATUS_MASK 0xffff0000 +#define CLOCK_CONFIG_MASK 0x0000ffff + #define CALLBACK_TIMEOUT 500 static int get_clock(struct snd_tscm *tscm, u32 *data) { + int trial = 0; __be32 reg; int err; - err = snd_fw_transaction(tscm->unit, TCODE_READ_QUADLET_REQUEST, - TSCM_ADDR_BASE + TSCM_OFFSET_CLOCK_STATUS, - ®, sizeof(reg), 0); - if (err >= 0) + while (trial++ < 5) { + err = snd_fw_transaction(tscm->unit, TCODE_READ_QUADLET_REQUEST, + TSCM_ADDR_BASE + TSCM_OFFSET_CLOCK_STATUS, + ®, sizeof(reg), 0); + if (err < 0) + return err; + *data = be32_to_cpu(reg); + if (*data & CLOCK_STATUS_MASK) + break; - return err; + // In intermediate state after changing clock status. + msleep(50); + } + + // Still in the intermediate state. + if (trial >= 5) + return -EAGAIN; + + return 0; } static int set_clock(struct snd_tscm *tscm, unsigned int rate, @@ -35,7 +52,7 @@ static int set_clock(struct snd_tscm *tscm, unsigned int rate, err = get_clock(tscm, &data); if (err < 0) return err; - data &= 0x0000ffff; + data &= CLOCK_CONFIG_MASK; if (rate > 0) { data &= 0x000000ff; @@ -80,17 +97,14 @@ static int set_clock(struct snd_tscm *tscm, unsigned int rate, int snd_tscm_stream_get_rate(struct snd_tscm *tscm, unsigned int *rate) { - u32 data = 0x0; - unsigned int trials = 0; + u32 data; int err; - while (data == 0x0 || trials++ < 5) { - err = get_clock(tscm, &data); - if (err < 0) - return err; + err = get_clock(tscm, &data); + if (err < 0) + return err; - data = (data & 0xff000000) >> 24; - } + data = (data & 0xff000000) >> 24; /* Check base rate. */ if ((data & 0x0f) == 0x01) diff --git a/sound/i2c/cs8427.c b/sound/i2c/cs8427.c index 7e21621e492a..7fd1b4000883 100644 --- a/sound/i2c/cs8427.c +++ b/sound/i2c/cs8427.c @@ -118,7 +118,7 @@ static int snd_cs8427_send_corudata(struct snd_i2c_device *device, struct cs8427 *chip = device->private_data; char *hw_data = udata ? chip->playback.hw_udata : chip->playback.hw_status; - char data[32]; + unsigned char data[32]; int err, idx; if (!memcmp(hw_data, ndata, count)) diff --git a/sound/i2c/other/ak4xxx-adda.c b/sound/i2c/other/ak4xxx-adda.c index bf377dc192aa..d33e02c31712 100644 --- a/sound/i2c/other/ak4xxx-adda.c +++ b/sound/i2c/other/ak4xxx-adda.c @@ -789,11 +789,12 @@ static int build_adc_controls(struct snd_akm4xxx *ak) return err; memset(&knew, 0, sizeof(knew)); - knew.name = ak->adc_info[mixer_ch].selector_name; - if (!knew.name) { + if (!ak->adc_info || + !ak->adc_info[mixer_ch].selector_name) { knew.name = "Capture Channel"; knew.index = mixer_ch + ak->idx_offset * 2; - } + } else + knew.name = ak->adc_info[mixer_ch].selector_name; knew.iface = SNDRV_CTL_ELEM_IFACE_MIXER; knew.info = ak4xxx_capture_source_info; diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index d3ea73171a3d..8b1cf237b96e 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -828,6 +828,8 @@ static void apply_fixup(struct hda_codec *codec, int id, int action, int depth) while (id >= 0) { const struct hda_fixup *fix = codec->fixup_list + id; + if (++depth > 10) + break; if (fix->chained_before) apply_fixup(codec, fix->chain_id, action, depth + 1); @@ -867,8 +869,6 @@ static void apply_fixup(struct hda_codec *codec, int id, int action, int depth) } if (!fix->chained || fix->chained_before) break; - if (++depth > 10) - break; id = fix->chain_id; } } diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index 8db1890605f6..c175b2cf63f7 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -42,6 +42,10 @@ static void hda_codec_unsol_event(struct hdac_device *dev, unsigned int ev) { struct hda_codec *codec = container_of(dev, struct hda_codec, core); + /* ignore unsol events during shutdown */ + if (codec->bus->shutdown) + return; + if (codec->patch_ops.unsol_event) codec->patch_ops.unsol_event(codec, ev); } diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index a12e594d4e3b..8fcb421193e0 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -609,11 +609,9 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) } runtime->private_data = azx_dev; - if (chip->gts_present) - azx_pcm_hw.info = azx_pcm_hw.info | - SNDRV_PCM_INFO_HAS_LINK_SYNCHRONIZED_ATIME; - runtime->hw = azx_pcm_hw; + if (chip->gts_present) + runtime->hw.info |= SNDRV_PCM_INFO_HAS_LINK_SYNCHRONIZED_ATIME; runtime->hw.channels_min = hinfo->channels_min; runtime->hw.channels_max = hinfo->channels_max; runtime->hw.formats = hinfo->formats; @@ -626,6 +624,13 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) 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 @@ -872,10 +877,13 @@ static int azx_rirb_get_response(struct hdac_bus *bus, unsigned int addr, */ if (hbus->allow_bus_reset && !hbus->response_reset && !hbus->in_reset) { hbus->response_reset = 1; + dev_err(chip->card->dev, + "No response from codec, resetting bus: last cmd=0x%08x\n", + bus->last_cmd[addr]); return -EAGAIN; /* give a chance to retry */ } - dev_err(chip->card->dev, + dev_WARN(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; diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index 53c3cd28bc99..8a9dd4767b1e 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -40,7 +40,7 @@ /* 14 unused */ #define AZX_DCAPS_CTX_WORKAROUND (1 << 15) /* X-Fi workaround */ #define AZX_DCAPS_POSFIX_LPIB (1 << 16) /* Use LPIB as default */ -/* 17 unused */ +#define AZX_DCAPS_AMD_WORKAROUND (1 << 17) /* AMD-specific workaround */ #define AZX_DCAPS_NO_64BIT (1 << 18) /* No 64bit address */ #define AZX_DCAPS_SYNC_WRITE (1 << 19) /* sync each cmd write */ #define AZX_DCAPS_OLD_SSYNC (1 << 20) /* Old SSYNC reg for ICH */ diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 28e265a88383..28ef409a9e6a 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -5854,7 +5854,8 @@ int snd_hda_gen_init(struct hda_codec *codec) if (spec->init_hook) spec->init_hook(codec); - snd_hda_apply_verbs(codec); + if (!spec->skip_verbs) + snd_hda_apply_verbs(codec); init_multi_out(codec); init_extra_out(codec); @@ -5896,6 +5897,24 @@ void snd_hda_gen_free(struct hda_codec *codec) } EXPORT_SYMBOL_GPL(snd_hda_gen_free); +/** + * snd_hda_gen_reboot_notify - Make codec enter D3 before rebooting + * @codec: the HDA codec + * + * This can be put as patch_ops reboot_notify function. + */ +void snd_hda_gen_reboot_notify(struct hda_codec *codec) +{ + /* Make the codec enter D3 to avoid spurious noises from the internal + * speaker during (and after) reboot + */ + snd_hda_codec_set_power_to_all(codec, codec->core.afg, AC_PWRST_D3); + snd_hda_codec_write(codec, codec->core.afg, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + msleep(10); +} +EXPORT_SYMBOL_GPL(snd_hda_gen_reboot_notify); + #ifdef CONFIG_PM /** * snd_hda_gen_check_power_status - check the loopback power save state @@ -5923,6 +5942,7 @@ static const struct hda_codec_ops generic_patch_ops = { .init = snd_hda_gen_init, .free = snd_hda_gen_free, .unsol_event = snd_hda_jack_unsol_event, + .reboot_notify = snd_hda_gen_reboot_notify, #ifdef CONFIG_PM .check_power_status = snd_hda_gen_check_power_status, #endif @@ -5945,7 +5965,7 @@ static int snd_hda_parse_generic_codec(struct hda_codec *codec) err = snd_hda_parse_pin_defcfg(codec, &spec->autocfg, NULL, 0); if (err < 0) - return err; + goto error; err = snd_hda_gen_parse_auto_config(codec, &spec->autocfg); if (err < 0) diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index 61772317de46..17a6bff8e94e 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -237,6 +237,7 @@ struct hda_gen_spec { unsigned int indep_hp_enabled:1; /* independent HP enabled */ unsigned int have_aamix_ctl:1; unsigned int hp_mic_jack_modes:1; + unsigned int skip_verbs:1; /* don't apply verbs at snd_hda_gen_init() */ /* additional mute flags (only effective with auto_mute_via_amp=1) */ u64 mute_bits; @@ -323,6 +324,7 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec, struct auto_pin_cfg *cfg); int snd_hda_gen_build_controls(struct hda_codec *codec); int snd_hda_gen_build_pcms(struct hda_codec *codec); +void snd_hda_gen_reboot_notify(struct hda_codec *codec); /* standard jack event callbacks */ void snd_hda_gen_hp_automute(struct hda_codec *codec, diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 65fb1e7edb9c..890793ad85ca 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -78,6 +78,7 @@ enum { POS_FIX_VIACOMBO, POS_FIX_COMBO, POS_FIX_SKL, + POS_FIX_FIFO, }; /* Defines for ATI HD Audio support in SB450 south bridge */ @@ -149,7 +150,7 @@ module_param_array(model, charp, NULL, 0444); MODULE_PARM_DESC(model, "Use the given board model."); module_param_array(position_fix, int, NULL, 0444); MODULE_PARM_DESC(position_fix, "DMA pointer read method." - "(-1 = system default, 0 = auto, 1 = LPIB, 2 = POSBUF, 3 = VIACOMBO, 4 = COMBO, 5 = SKL+)."); + "(-1 = system default, 0 = auto, 1 = LPIB, 2 = POSBUF, 3 = VIACOMBO, 4 = COMBO, 5 = SKL+, 6 = FIFO)."); module_param_array(bdl_pos_adj, int, NULL, 0644); MODULE_PARM_DESC(bdl_pos_adj, "BDL position adjustment offset."); module_param_array(probe_mask, int, NULL, 0444); @@ -350,6 +351,11 @@ enum { #define AZX_DCAPS_PRESET_ATI_HDMI_NS \ (AZX_DCAPS_PRESET_ATI_HDMI | AZX_DCAPS_SNOOP_OFF) +/* quirks for AMD SB */ +#define AZX_DCAPS_PRESET_AMD_SB \ + (AZX_DCAPS_NO_TCSEL | AZX_DCAPS_SYNC_WRITE | AZX_DCAPS_AMD_WORKAROUND |\ + AZX_DCAPS_SNOOP_TYPE(ATI) | AZX_DCAPS_PM_RUNTIME) + /* quirks for Nvidia */ #define AZX_DCAPS_PRESET_NVIDIA \ (AZX_DCAPS_NO_MSI | AZX_DCAPS_CORBRP_SELF_CLEAR |\ @@ -376,6 +382,7 @@ enum { #define IS_BXT(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x5a98) #define IS_CFL(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0xa348) +#define IS_CNL(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x9dc8) static char *driver_short_names[] = { [AZX_DRIVER_ICH] = "HDA Intel", @@ -916,6 +923,49 @@ static unsigned int azx_via_get_position(struct azx *chip, return bound_pos + mod_dma_pos; } +#define AMD_FIFO_SIZE 32 + +/* get the current DMA position with FIFO size correction */ +static unsigned int azx_get_pos_fifo(struct azx *chip, struct azx_dev *azx_dev) +{ + struct snd_pcm_substream *substream = azx_dev->core.substream; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned int pos, delay; + + pos = snd_hdac_stream_get_pos_lpib(azx_stream(azx_dev)); + if (!runtime) + return pos; + + runtime->delay = AMD_FIFO_SIZE; + delay = frames_to_bytes(runtime, AMD_FIFO_SIZE); + if (azx_dev->insufficient) { + if (pos < delay) { + delay = pos; + runtime->delay = bytes_to_frames(runtime, pos); + } else { + azx_dev->insufficient = 0; + } + } + + /* correct the DMA position for capture stream */ + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (pos < delay) + pos += azx_dev->core.bufsize; + pos -= delay; + } + + return pos; +} + +static int azx_get_delay_from_fifo(struct azx *chip, struct azx_dev *azx_dev, + unsigned int pos) +{ + struct snd_pcm_substream *substream = azx_dev->core.substream; + + /* just read back the calculated value in the above */ + return substream->runtime->delay; +} + static unsigned int azx_skl_get_dpib_pos(struct azx *chip, struct azx_dev *azx_dev) { @@ -1400,8 +1450,11 @@ static int azx_free(struct azx *chip) 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; } @@ -1483,6 +1536,7 @@ static int check_position_fix(struct azx *chip, int fix) case POS_FIX_VIACOMBO: case POS_FIX_COMBO: case POS_FIX_SKL: + case POS_FIX_FIFO: return fix; } @@ -1499,6 +1553,10 @@ static int check_position_fix(struct azx *chip, int fix) dev_dbg(chip->card->dev, "Using VIACOMBO position fix\n"); return POS_FIX_VIACOMBO; } + if (chip->driver_caps & AZX_DCAPS_AMD_WORKAROUND) { + dev_dbg(chip->card->dev, "Using FIFO position fix\n"); + return POS_FIX_FIFO; + } if (chip->driver_caps & AZX_DCAPS_POSFIX_LPIB) { dev_dbg(chip->card->dev, "Using LPIB position fix\n"); return POS_FIX_LPIB; @@ -1519,6 +1577,7 @@ static void assign_position_fix(struct azx *chip, int fix) [POS_FIX_VIACOMBO] = azx_via_get_position, [POS_FIX_COMBO] = azx_get_pos_lpib, [POS_FIX_SKL] = azx_get_pos_skl, + [POS_FIX_FIFO] = azx_get_pos_fifo, }; chip->get_position[0] = chip->get_position[1] = callbacks[fix]; @@ -1533,6 +1592,9 @@ static void assign_position_fix(struct azx *chip, int fix) azx_get_delay_from_lpib; } + if (fix == POS_FIX_FIFO) + chip->get_delay[0] = chip->get_delay[1] = + azx_get_delay_from_fifo; } /* @@ -1751,8 +1813,8 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci, else chip->bdl_pos_adj = bdl_pos_adj[dev]; - /* Workaround for a communication error on CFL (bko#199007) */ - if (IS_CFL(pci)) + /* Workaround for a communication error on CFL (bko#199007) and CNL */ + if (IS_CFL(pci) || IS_CNL(pci)) chip->polling_mode = 1; err = azx_bus_init(chip, model[dev], &pci_hda_io_ops); @@ -2515,14 +2577,19 @@ static const struct pci_device_id azx_ids[] = { /* AMD Hudson */ { PCI_DEVICE(0x1022, 0x780d), .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB }, + /* AMD, X370 & co */ + { PCI_DEVICE(0x1022, 0x1457), + .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_AMD_SB }, + /* AMD, X570 & co */ + { PCI_DEVICE(0x1022, 0x1487), + .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_AMD_SB }, /* AMD Stoney */ { PCI_DEVICE(0x1022, 0x157a), .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB | AZX_DCAPS_PM_RUNTIME }, /* AMD Raven */ { PCI_DEVICE(0x1022, 0x15e3), - .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB | - AZX_DCAPS_PM_RUNTIME }, + .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_AMD_SB }, /* ATI HDMI */ { PCI_DEVICE(0x1002, 0x0002), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 757857313426..87eff3c39611 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -370,6 +370,7 @@ static const struct hda_fixup ad1986a_fixups[] = { static const struct snd_pci_quirk ad1986a_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x30af, "HP B2800", AD1986A_FIXUP_LAPTOP_IMIC), + SND_PCI_QUIRK(0x1043, 0x1153, "ASUS M9V", AD1986A_FIXUP_LAPTOP_IMIC), SND_PCI_QUIRK(0x1043, 0x1443, "ASUS Z99He", AD1986A_FIXUP_EAPD), SND_PCI_QUIRK(0x1043, 0x1447, "ASUS A8JN", AD1986A_FIXUP_EAPD), SND_PCI_QUIRK_MASK(0x1043, 0xff00, 0x8100, "ASUS P5", AD1986A_FIXUP_3STACK), diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 119f3b504765..9876d8dc2ede 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -4440,7 +4440,7 @@ static void hp_callback(struct hda_codec *codec, struct hda_jack_callback *cb) /* Delay enabling the HP amp, to let the mic-detection * state machine run. */ - cancel_delayed_work_sync(&spec->unsol_hp_work); + 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) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index d14516f31679..382b6d2ed803 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -210,23 +210,10 @@ static void cx_auto_reboot_notify(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; - switch (codec->core.vendor_id) { - case 0x14f12008: /* CX8200 */ - case 0x14f150f2: /* CX20722 */ - case 0x14f150f4: /* CX20724 */ - break; - default: - return; - } - /* Turn the problematic codec into D3 to avoid spurious noises from the internal speaker during (and after) reboot */ cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, false); - - snd_hda_codec_set_power_to_all(codec, codec->core.afg, AC_PWRST_D3); - snd_hda_codec_write(codec, codec->core.afg, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - msleep(10); + snd_hda_gen_reboot_notify(codec); } static void cx_auto_free(struct hda_codec *codec) @@ -973,6 +960,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { 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(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN), SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT_FIXUP_OLPC_XO), @@ -1127,6 +1115,7 @@ static int patch_conexant_auto(struct hda_codec *codec) */ static const struct hda_device_id snd_hda_id_conexant[] = { + HDA_CODEC_ENTRY(0x14f11f86, "CX8070", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f12008, "CX8200", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f15045, "CX20549 (Venice)", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f15047, "CX20551 (Waikiki)", patch_conexant_auto), diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index f5803f9bba9b..f21405597215 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2555,13 +2555,20 @@ static void i915_pin_cvt_fixup(struct hda_codec *codec, /* precondition and allocation for Intel codecs */ static int alloc_intel_hdmi(struct hda_codec *codec) { + int err; + /* requires i915 binding */ if (!codec->bus->core.audio_component) { codec_info(codec, "No i915 binding for Intel HDMI/DP codec\n"); return -ENODEV; } - return alloc_generic_hdmi(codec); + err = alloc_generic_hdmi(codec); + if (err < 0) + return err; + /* no need to handle unsol events */ + codec->patch_ops.unsol_event = NULL; + return 0; } /* parse and post-process for Intel codecs */ diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a9e3ee233833..41e3c77d5fb7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -333,9 +333,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0215: case 0x10ec0233: case 0x10ec0235: - case 0x10ec0236: case 0x10ec0255: - case 0x10ec0256: case 0x10ec0257: case 0x10ec0282: case 0x10ec0283: @@ -347,6 +345,11 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0300: alc_update_coef_idx(codec, 0x10, 1<<9, 0); break; + case 0x10ec0236: + case 0x10ec0256: + alc_write_coef_idx(codec, 0x36, 0x5757); + alc_update_coef_idx(codec, 0x10, 1<<9, 0); + break; case 0x10ec0275: alc_update_coef_idx(codec, 0xe, 0, 1<<0); break; @@ -359,6 +362,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0700: case 0x10ec0701: case 0x10ec0703: + case 0x10ec0711: alc_update_coef_idx(codec, 0x10, 1<<15, 0); break; case 0x10ec0662: @@ -374,6 +378,9 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0672: alc_update_coef_idx(codec, 0xd, 0, 1<<14); /* EAPD Ctrl */ break; + case 0x10ec0623: + alc_update_coef_idx(codec, 0x19, 1<<13, 0); + break; case 0x10ec0668: alc_update_coef_idx(codec, 0x7, 3<<13, 0); break; @@ -781,9 +788,11 @@ static int alc_init(struct hda_codec *codec) if (spec->init_hook) spec->init_hook(codec); + spec->gen.skip_verbs = 1; /* applied in below */ snd_hda_gen_init(codec); alc_fix_pll(codec); alc_auto_init_amp(codec, spec->init_amp); + snd_hda_apply_verbs(codec); /* apply verbs here after own init */ snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_INIT); @@ -810,15 +819,6 @@ static void alc_reboot_notify(struct hda_codec *codec) alc_shutup(codec); } -/* power down codec to D3 at reboot/shutdown; set as reboot_notify ops */ -static void alc_d3_at_reboot(struct hda_codec *codec) -{ - snd_hda_codec_set_power_to_all(codec, codec->core.afg, AC_PWRST_D3); - snd_hda_codec_write(codec, codec->core.afg, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - msleep(10); -} - #define alc_free snd_hda_gen_free #ifdef CONFIG_PM @@ -984,6 +984,9 @@ static const struct snd_pci_quirk beep_white_list[] = { SND_PCI_QUIRK(0x1043, 0x834a, "EeePC", 1), SND_PCI_QUIRK(0x1458, 0xa002, "GA-MA790X", 1), SND_PCI_QUIRK(0x8086, 0xd613, "Intel", 1), + /* blacklist -- no beep available */ + SND_PCI_QUIRK(0x17aa, 0x309e, "Lenovo ThinkCentre M73", 0), + SND_PCI_QUIRK(0x17aa, 0x30a3, "Lenovo ThinkCentre M93", 0), {} }; @@ -2760,6 +2763,7 @@ enum { ALC269_TYPE_ALC225, ALC269_TYPE_ALC294, ALC269_TYPE_ALC300, + ALC269_TYPE_ALC623, ALC269_TYPE_ALC700, }; @@ -2795,6 +2799,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec) case ALC269_TYPE_ALC225: case ALC269_TYPE_ALC294: case ALC269_TYPE_ALC300: + case ALC269_TYPE_ALC623: case ALC269_TYPE_ALC700: ssids = alc269_ssids; break; @@ -3114,6 +3119,7 @@ static void alc256_init(struct hda_codec *codec) 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*/ } static void alc256_shutup(struct hda_codec *codec) @@ -3248,7 +3254,9 @@ static void alc294_init(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - if (!spec->done_hp_init) { + /* required only at boot or S4 resume time */ + if (!spec->done_hp_init || + codec->core.dev.power.power_state.event == PM_EVENT_RESTORE) { alc294_hp_init(codec); spec->done_hp_init = true; } @@ -3614,6 +3622,19 @@ static void alc269_fixup_hp_mute_led_mic2(struct hda_codec *codec, } } +static void alc269_fixup_hp_mute_led_mic3(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_nid = 0x1b; + spec->gen.vmaster_mute.hook = alc269_fixup_mic_mute_hook; + spec->gen.vmaster_mute_enum = 1; + codec->power_filter = led_power_filter; + } +} + /* update LED status via GPIO */ static void alc_update_gpio_led(struct hda_codec *codec, unsigned int mask, bool enabled) @@ -3936,18 +3957,19 @@ static struct coef_fw alc225_pre_hsmode[] = { static void alc_headset_mode_unplugged(struct hda_codec *codec) { static 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)*/ WRITE_COEF(0x06, 0x6104), /* Set MIC2 Vref gate with HP */ WRITE_COEFEX(0x57, 0x03, 0x8aa6), /* Direct Drive HP Amp control */ {} }; - static struct coef_fw coef0255_1[] = { - WRITE_COEF(0x1b, 0x0c0b), /* LDO and MISC control */ - {} - }; static 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 */ + WRITE_COEFEX(0x57, 0x03, 0x09a3), /* Direct Drive HP Amp control */ + UPDATE_COEFEX(0x57, 0x05, 1<<14, 0), /* Direct Drive HP Amp control(Set to verb control)*/ {} }; static struct coef_fw coef0233[] = { @@ -4010,13 +4032,11 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) switch (codec->core.vendor_id) { case 0x10ec0255: - alc_process_coef_fw(codec, coef0255_1); alc_process_coef_fw(codec, coef0255); break; case 0x10ec0236: case 0x10ec0256: alc_process_coef_fw(codec, coef0256); - alc_process_coef_fw(codec, coef0255); break; case 0x10ec0234: case 0x10ec0274: @@ -4066,6 +4086,12 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, WRITE_COEF(0x06, 0x6100), /* Set MIC2 Vref gate to normal */ {} }; + static 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[] = { UPDATE_COEF(0x35, 0, 1<<14), WRITE_COEF(0x06, 0x2100), @@ -4113,14 +4139,19 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, }; switch (codec->core.vendor_id) { - case 0x10ec0236: case 0x10ec0255: - case 0x10ec0256: alc_write_coef_idx(codec, 0x45, 0xc489); snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); alc_process_coef_fw(codec, coef0255); snd_hda_set_pin_ctl_cache(codec, mic_pin, PIN_VREF50); break; + case 0x10ec0236: + case 0x10ec0256: + alc_write_coef_idx(codec, 0x45, 0xc489); + snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); + alc_process_coef_fw(codec, coef0256); + snd_hda_set_pin_ctl_cache(codec, mic_pin, PIN_VREF50); + break; case 0x10ec0234: case 0x10ec0274: case 0x10ec0294: @@ -4199,6 +4230,14 @@ static void alc_headset_mode_default(struct hda_codec *codec) WRITE_COEF(0x49, 0x0049), {} }; + static struct coef_fw coef0256[] = { + WRITE_COEF(0x45, 0xc489), + WRITE_COEFEX(0x57, 0x03, 0x0da3), + WRITE_COEF(0x49, 0x0049), + UPDATE_COEFEX(0x57, 0x05, 1<<14, 0), /* Direct Drive HP Amp control(Set to verb control)*/ + WRITE_COEF(0x06, 0x6100), + {} + }; static struct coef_fw coef0233[] = { WRITE_COEF(0x06, 0x2100), WRITE_COEF(0x32, 0x4ea3), @@ -4246,11 +4285,16 @@ static void alc_headset_mode_default(struct hda_codec *codec) alc_process_coef_fw(codec, alc225_pre_hsmode); alc_process_coef_fw(codec, coef0225); break; - case 0x10ec0236: case 0x10ec0255: - case 0x10ec0256: alc_process_coef_fw(codec, coef0255); break; + case 0x10ec0236: + case 0x10ec0256: + alc_write_coef_idx(codec, 0x1b, 0x0e4b); + alc_write_coef_idx(codec, 0x45, 0xc089); + msleep(50); + alc_process_coef_fw(codec, coef0256); + break; case 0x10ec0234: case 0x10ec0274: case 0x10ec0294: @@ -4294,8 +4338,7 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) }; static struct coef_fw coef0256[] = { WRITE_COEF(0x45, 0xd489), /* Set to CTIA type */ - WRITE_COEF(0x1b, 0x0c6b), - WRITE_COEFEX(0x57, 0x03, 0x8ea6), + WRITE_COEF(0x1b, 0x0e6b), {} }; static struct coef_fw coef0233[] = { @@ -4410,8 +4453,7 @@ static void alc_headset_mode_omtp(struct hda_codec *codec) }; static struct coef_fw coef0256[] = { WRITE_COEF(0x45, 0xe489), /* Set to OMTP Type */ - WRITE_COEF(0x1b, 0x0c6b), - WRITE_COEFEX(0x57, 0x03, 0x8ea6), + WRITE_COEF(0x1b, 0x0e6b), {} }; static struct coef_fw coef0233[] = { @@ -4540,13 +4582,37 @@ static void alc_determine_headset_type(struct hda_codec *codec) }; switch (codec->core.vendor_id) { - case 0x10ec0236: case 0x10ec0255: + alc_process_coef_fw(codec, coef0255); + msleep(300); + val = alc_read_coef_idx(codec, 0x46); + is_ctia = (val & 0x0070) == 0x0070; + break; + case 0x10ec0236: case 0x10ec0256: + alc_write_coef_idx(codec, 0x1b, 0x0e4b); + alc_write_coef_idx(codec, 0x06, 0x6104); + alc_write_coefex_idx(codec, 0x57, 0x3, 0x09a3); + + snd_hda_codec_write(codec, 0x21, 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); + msleep(80); + snd_hda_codec_write(codec, 0x21, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); + alc_process_coef_fw(codec, coef0255); msleep(300); val = alc_read_coef_idx(codec, 0x46); is_ctia = (val & 0x0070) == 0x0070; + + alc_write_coefex_idx(codec, 0x57, 0x3, 0x0da3); + alc_update_coefex_idx(codec, 0x57, 0x5, 1<<14, 0); + + snd_hda_codec_write(codec, 0x21, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); + msleep(80); + snd_hda_codec_write(codec, 0x21, 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); break; case 0x10ec0234: case 0x10ec0274: @@ -4891,7 +4957,7 @@ static void alc_fixup_tpt440_dock(struct hda_codec *codec, struct alc_spec *spec = codec->spec; if (action == HDA_FIXUP_ACT_PRE_PROBE) { - spec->reboot_notify = alc_d3_at_reboot; /* reduce noise */ + spec->reboot_notify = snd_hda_gen_reboot_notify; /* reduce noise */ spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP; codec->power_save_node = 0; /* avoid click noises */ snd_hda_apply_pincfgs(codec, pincfgs); @@ -5343,6 +5409,7 @@ enum { ALC269_FIXUP_HP_MUTE_LED, ALC269_FIXUP_HP_MUTE_LED_MIC1, ALC269_FIXUP_HP_MUTE_LED_MIC2, + ALC269_FIXUP_HP_MUTE_LED_MIC3, ALC269_FIXUP_HP_GPIO_LED, ALC269_FIXUP_HP_GPIO_MIC1_LED, ALC269_FIXUP_HP_LINE1_MIC1_LED, @@ -5606,6 +5673,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_hp_mute_led_mic2, }, + [ALC269_FIXUP_HP_MUTE_LED_MIC3] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_hp_mute_led_mic3, + }, [ALC269_FIXUP_HP_GPIO_LED] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_hp_gpio_led, @@ -6460,6 +6531,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { 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, 0x8256, "HP", ALC221_FIXUP_HP_FRONT_MIC), + SND_PCI_QUIRK(0x103c, 0x827e, "HP x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x82bf, "HP", ALC221_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x82c0, "HP", ALC221_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), @@ -6545,9 +6617,13 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { 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), + SND_PCI_QUIRK(0x17aa, 0x3111, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), SND_PCI_QUIRK(0x17aa, 0x312a, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), SND_PCI_QUIRK(0x17aa, 0x312f, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), SND_PCI_QUIRK(0x17aa, 0x313c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), + 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, 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), @@ -7172,7 +7248,6 @@ static int patch_alc269(struct hda_codec *codec) spec->shutup = alc256_shutup; spec->init_hook = alc256_init; spec->gen.mixer_nid = 0; /* ALC256 does not have any loopback mixer path */ - alc_update_coef_idx(codec, 0x36, 1 << 13, 1 << 5); /* Switch pcbeep path to Line in path*/ break; case 0x10ec0257: spec->codec_variant = ALC269_TYPE_ALC257; @@ -7207,9 +7282,13 @@ static int patch_alc269(struct hda_codec *codec) spec->codec_variant = ALC269_TYPE_ALC300; spec->gen.mixer_nid = 0; /* no loopback on ALC300 */ break; + case 0x10ec0623: + spec->codec_variant = ALC269_TYPE_ALC623; + break; case 0x10ec0700: case 0x10ec0701: case 0x10ec0703: + case 0x10ec0711: spec->codec_variant = ALC269_TYPE_ALC700; spec->gen.mixer_nid = 0; /* ALC700 does not have any loopback mixer path */ alc_update_coef_idx(codec, 0x4a, 1 << 15, 0); /* Combo jack auto trigger control */ @@ -8103,6 +8182,11 @@ static const struct snd_hda_pin_quirk alc662_pin_fixup_tbl[] = { {0x18, 0x01a19030}, {0x1a, 0x01813040}, {0x21, 0x01014020}), + SND_HDA_PIN_QUIRK(0x10ec0867, 0x1028, "Dell", ALC891_FIXUP_DELL_MIC_NO_PRESENCE, + {0x16, 0x01813030}, + {0x17, 0x02211010}, + {0x18, 0x01a19040}, + {0x21, 0x01014020}), SND_HDA_PIN_QUIRK(0x10ec0662, 0x1028, "Dell", ALC662_FIXUP_DELL_MIC_NO_PRESENCE, {0x14, 0x01014010}, {0x18, 0x01a19020}, @@ -8245,6 +8329,7 @@ static int patch_alc680(struct hda_codec *codec) static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0215, "ALC215", patch_alc269), HDA_CODEC_ENTRY(0x10ec0221, "ALC221", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0222, "ALC222", patch_alc269), HDA_CODEC_ENTRY(0x10ec0225, "ALC225", patch_alc269), HDA_CODEC_ENTRY(0x10ec0231, "ALC231", patch_alc269), HDA_CODEC_ENTRY(0x10ec0233, "ALC233", patch_alc269), @@ -8280,6 +8365,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0298, "ALC298", patch_alc269), HDA_CODEC_ENTRY(0x10ec0299, "ALC299", patch_alc269), HDA_CODEC_ENTRY(0x10ec0300, "ALC300", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0623, "ALC623", patch_alc269), HDA_CODEC_REV_ENTRY(0x10ec0861, 0x100340, "ALC660", patch_alc861), HDA_CODEC_ENTRY(0x10ec0660, "ALC660-VD", patch_alc861vd), HDA_CODEC_ENTRY(0x10ec0861, "ALC861", patch_alc861), @@ -8297,6 +8383,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0700, "ALC700", patch_alc269), HDA_CODEC_ENTRY(0x10ec0701, "ALC701", patch_alc269), HDA_CODEC_ENTRY(0x10ec0703, "ALC703", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0711, "ALC711", patch_alc269), HDA_CODEC_ENTRY(0x10ec0867, "ALC891", patch_alc662), HDA_CODEC_ENTRY(0x10ec0880, "ALC880", patch_alc880), HDA_CODEC_ENTRY(0x10ec0882, "ALC882", patch_alc882), diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 63d15b545b33..7cd147411b22 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -77,6 +77,7 @@ enum { STAC_DELL_M6_BOTH, STAC_DELL_EQ, STAC_ALIENWARE_M17X, + STAC_ELO_VUPOINT_15MX, STAC_92HD89XX_HP_FRONT_JACK, STAC_92HD89XX_HP_Z1_G2_RIGHT_MIC_JACK, STAC_92HD73XX_ASUS_MOBO, @@ -1897,6 +1898,18 @@ static void stac92hd73xx_fixup_no_jd(struct hda_codec *codec, codec->no_jack_detect = 1; } + +static void stac92hd73xx_disable_automute(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct sigmatel_spec *spec = codec->spec; + + if (action != HDA_FIXUP_ACT_PRE_PROBE) + return; + + spec->gen.suppress_auto_mute = 1; +} + static const struct hda_fixup stac92hd73xx_fixups[] = { [STAC_92HD73XX_REF] = { .type = HDA_FIXUP_FUNC, @@ -1922,6 +1935,10 @@ static const struct hda_fixup stac92hd73xx_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = stac92hd73xx_fixup_alienware_m17x, }, + [STAC_ELO_VUPOINT_15MX] = { + .type = HDA_FIXUP_FUNC, + .v.func = stac92hd73xx_disable_automute, + }, [STAC_92HD73XX_INTEL] = { .type = HDA_FIXUP_PINS, .v.pins = intel_dg45id_pin_configs, @@ -1960,6 +1977,7 @@ static const struct hda_model_fixup stac92hd73xx_models[] = { { .id = STAC_DELL_M6_BOTH, .name = "dell-m6" }, { .id = STAC_DELL_EQ, .name = "dell-eq" }, { .id = STAC_ALIENWARE_M17X, .name = "alienware" }, + { .id = STAC_ELO_VUPOINT_15MX, .name = "elo-vupoint-15mx" }, { .id = STAC_92HD73XX_ASUS_MOBO, .name = "asus-mobo" }, {} }; @@ -2009,6 +2027,8 @@ static const struct snd_pci_quirk stac92hd73xx_fixup_tbl[] = { "Alienware M17x", STAC_ALIENWARE_M17X), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0490, "Alienware M17x R3", STAC_DELL_EQ), + SND_PCI_QUIRK(0x1059, 0x1011, + "ELO VuPoint 15MX", STAC_ELO_VUPOINT_15MX), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1927, "HP Z1 G2", STAC_92HD89XX_HP_Z1_G2_RIGHT_MIC_JACK), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x2b17, diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index 3a4769a97d29..a626ee18628e 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -1171,16 +1171,6 @@ static int snd_intel8x0m_create(struct snd_card *card, } port_inited: - if (request_irq(pci->irq, snd_intel8x0m_interrupt, IRQF_SHARED, - KBUILD_MODNAME, chip)) { - dev_err(card->dev, "unable to grab IRQ %d\n", pci->irq); - snd_intel8x0m_free(chip); - return -EBUSY; - } - chip->irq = pci->irq; - pci_set_master(pci); - synchronize_irq(chip->irq); - /* initialize offsets */ chip->bdbars_count = 2; tbl = intel_regs; @@ -1224,11 +1214,21 @@ static int snd_intel8x0m_create(struct snd_card *card, chip->int_sta_reg = ICH_REG_GLOB_STA; chip->int_sta_mask = int_sta_masks; + pci_set_master(pci); + if ((err = snd_intel8x0m_chip_init(chip, 1)) < 0) { snd_intel8x0m_free(chip); return err; } + if (request_irq(pci->irq, snd_intel8x0m_interrupt, IRQF_SHARED, + KBUILD_MODNAME, chip)) { + dev_err(card->dev, "unable to grab IRQ %d\n", pci->irq); + snd_intel8x0m_free(chip); + return -EBUSY; + } + chip->irq = pci->irq; + if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) { snd_intel8x0m_free(chip); return err; diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index 6e8eb1f5a041..bed64723e5d9 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -60,7 +60,7 @@ static const struct reg_default cs4265_reg_defaults[] = { static bool cs4265_readable_register(struct device *dev, unsigned int reg) { switch (reg) { - case CS4265_CHIP_ID ... CS4265_SPDIF_CTL2: + case CS4265_CHIP_ID ... CS4265_MAX_REGISTER: return true; default: return false; diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c index 2e772427b48a..cedddee67199 100644 --- a/sound/soc/codecs/cs42xx8.c +++ b/sound/soc/codecs/cs42xx8.c @@ -668,6 +668,7 @@ static int cs42xx8_runtime_resume(struct device *dev) CS42XX8_PWRCTL_PDN_MASK, 0); regcache_cache_only(cs42xx8->regmap, false); + regcache_mark_dirty(cs42xx8->regmap); ret = regcache_sync(cs42xx8->regmap); if (ret) { diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index da2d353af5ba..949dbdc0445e 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -46,7 +46,10 @@ static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_vol_tlv, -9600, 50, 1); static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(alc_max_gain_tlv, -650, 150, 0); static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(alc_min_gain_tlv, -1200, 150, 0); static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(alc_target_tlv, -1650, 150, 0); -static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(hpmixer_gain_tlv, -1200, 150, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(hpmixer_gain_tlv, + 0, 4, TLV_DB_SCALE_ITEM(-1200, 150, 0), + 8, 11, TLV_DB_SCALE_ITEM(-450, 150, 0), +); static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(adc_pga_gain_tlv, 0, 0, TLV_DB_SCALE_ITEM(-350, 0, 0), @@ -84,7 +87,7 @@ static const struct snd_kcontrol_new es8316_snd_controls[] = { SOC_DOUBLE_TLV("Headphone Playback Volume", ES8316_CPHP_ICAL_VOL, 4, 0, 3, 1, hpout_vol_tlv), SOC_DOUBLE_TLV("Headphone Mixer Volume", ES8316_HPMIX_VOL, - 0, 4, 7, 0, hpmixer_gain_tlv), + 0, 4, 11, 0, hpmixer_gain_tlv), SOC_ENUM("Playback Polarity", dacpol), SOC_DOUBLE_R_TLV("DAC Playback Volume", ES8316_DAC_VOLL, diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index e824d47cc22b..1c3626347e12 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1408,6 +1408,12 @@ static int hdac_hdmi_create_dais(struct hdac_device *hdac, if (ret) return ret; + /* Filter out 44.1, 88.2 and 176.4Khz */ + rates &= ~(SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_176400); + if (!rates) + return -EINVAL; + sprintf(dai_name, "intel-hdmi-hifi%d", i+1); hdmi_dais[i].name = devm_kstrdup(&hdac->dev, dai_name, GFP_KERNEL); diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index cc66ea5cc776..3fe09828745a 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -1924,6 +1924,21 @@ static int max98090_configure_dmic(struct max98090_priv *max98090, return 0; } +static int max98090_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct max98090_priv *max98090 = snd_soc_component_get_drvdata(component); + unsigned int fmt = max98090->dai_fmt; + + /* Remove 24-bit format support if it is not in right justified mode. */ + if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != SND_SOC_DAIFMT_RIGHT_J) { + substream->runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; + snd_pcm_hw_constraint_msbits(substream->runtime, 0, 16, 16); + } + return 0; +} + static int max98090_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -2331,6 +2346,7 @@ EXPORT_SYMBOL_GPL(max98090_mic_detect); #define MAX98090_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) static const struct snd_soc_dai_ops max98090_dai_ops = { + .startup = max98090_dai_startup, .set_sysclk = max98090_dai_set_sysclk, .set_fmt = max98090_dai_set_fmt, .set_tdm_slot = max98090_set_tdm_slot, diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index 0b9b014b4bb6..969283737787 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -303,7 +303,7 @@ struct pm8916_wcd_analog_priv { }; static const char *const adc2_mux_text[] = { "ZERO", "INP2", "INP3" }; -static const char *const rdac2_mux_text[] = { "ZERO", "RX2", "RX1" }; +static const char *const rdac2_mux_text[] = { "RX1", "RX2" }; static const char *const hph_text[] = { "ZERO", "Switch", }; static const struct soc_enum hph_enum = SOC_ENUM_SINGLE_VIRT( @@ -318,7 +318,7 @@ static const struct soc_enum adc2_enum = SOC_ENUM_SINGLE_VIRT( /* RDAC2 MUX */ static const struct soc_enum rdac2_mux_enum = SOC_ENUM_SINGLE( - CDC_D_CDC_CONN_HPHR_DAC_CTL, 0, 3, rdac2_mux_text); + CDC_D_CDC_CONN_HPHR_DAC_CTL, 0, 2, rdac2_mux_text); static const struct snd_kcontrol_new spkr_switch[] = { SOC_DAPM_SINGLE("Switch", CDC_A_SPKR_DAC_CTL, 7, 1, 0) diff --git a/sound/soc/codecs/nau8540.c b/sound/soc/codecs/nau8540.c index f9c9933acffb..c0c64f90a61b 100644 --- a/sound/soc/codecs/nau8540.c +++ b/sound/soc/codecs/nau8540.c @@ -548,7 +548,7 @@ static int nau8540_calc_fll_param(unsigned int fll_in, fvco_max = 0; fvco_sel = ARRAY_SIZE(mclk_src_scaling); for (i = 0; i < ARRAY_SIZE(mclk_src_scaling); i++) { - fvco = 256 * fs * 2 * mclk_src_scaling[i].param; + fvco = 256ULL * fs * 2 * mclk_src_scaling[i].param; if (fvco > NAU_FVCO_MIN && fvco < NAU_FVCO_MAX && fvco_max < fvco) { fvco_max = fvco; diff --git a/sound/soc/codecs/rt274.c b/sound/soc/codecs/rt274.c index cd048df76232..43086ac9ffec 100644 --- a/sound/soc/codecs/rt274.c +++ b/sound/soc/codecs/rt274.c @@ -398,6 +398,8 @@ static int rt274_mic_detect(struct snd_soc_codec *codec, { struct rt274_priv *rt274 = snd_soc_codec_get_drvdata(codec); + rt274->jack = jack; + if (jack == NULL) { /* Disable jack detection */ regmap_update_bits(rt274->regmap, RT274_EAPD_GPIO_IRQ_CTRL, @@ -405,7 +407,6 @@ static int rt274_mic_detect(struct snd_soc_codec *codec, return 0; } - rt274->jack = jack; regmap_update_bits(rt274->regmap, RT274_EAPD_GPIO_IRQ_CTRL, RT274_IRQ_EN, RT274_IRQ_EN); diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index c13faf07fa19..5d54a4828b42 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -162,7 +162,7 @@ static inline int hp_sel_input(struct snd_soc_component *component) static inline u16 mute_output(struct snd_soc_component *component, u16 mute_mask) { - unsigned int mute_reg; + unsigned int mute_reg = 0; snd_soc_component_read(component, SGTL5000_CHIP_ANA_CTRL, &mute_reg); @@ -180,7 +180,7 @@ static inline void restore_output(struct snd_soc_component *component, static void vag_power_on(struct snd_soc_component *component, u32 source) { - unsigned int ana_reg; + unsigned int ana_reg = 0; snd_soc_component_read(component, SGTL5000_CHIP_ANA_POWER, &ana_reg); @@ -228,7 +228,7 @@ static int vag_power_consumers(struct snd_soc_component *component, static void vag_power_off(struct snd_soc_component *component, u32 source) { - unsigned int ana_pwr; + unsigned int ana_pwr = SGTL5000_VAG_POWERUP; snd_soc_component_read(component, SGTL5000_CHIP_ANA_POWER, &ana_pwr); @@ -301,7 +301,7 @@ static int vag_and_mute_control(struct snd_soc_component *component, SGTL5000_HP_MUTE, /* * Masks for DAC_POWER_EVENT/ADC_POWER_EVENT. - * Muxing DAC or ADC block have to wrapped with mute/unmute + * Muxing DAC or ADC block have to be wrapped with mute/unmute * both headphones and line-out. */ SGTL5000_OUTPUTS_MUTE, diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 54a87a905eb6..cc95c15ceceb 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -924,23 +924,31 @@ static int aic31xx_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } + /* signal polarity */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + iface_reg2 |= AIC31XX_BCLKINV_MASK; + break; + default: + dev_err(codec->dev, "Invalid DAI clock signal polarity\n"); + return -EINVAL; + } + /* interface format */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: break; case SND_SOC_DAIFMT_DSP_A: - dsp_a_val = 0x1; + dsp_a_val = 0x1; /* fall through */ case SND_SOC_DAIFMT_DSP_B: - /* NOTE: BCLKINV bit value 1 equas NB and 0 equals IB */ - switch (fmt & SND_SOC_DAIFMT_INV_MASK) { - case SND_SOC_DAIFMT_NB_NF: - iface_reg2 |= AIC31XX_BCLKINV_MASK; - break; - case SND_SOC_DAIFMT_IB_NF: - break; - default: - return -EINVAL; - } + /* + * NOTE: This CODEC samples on the falling edge of BCLK in + * DSP mode, this is inverted compared to what most DAIs + * expect, so we invert for this mode + */ + iface_reg2 ^= AIC31XX_BCLKINV_MASK; iface_reg1 |= (AIC31XX_DSP_MODE << AIC31XX_IFACE1_DATATYPE_SHIFT); break; diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index d632a0511d62..158ce68bc9bf 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1169,8 +1169,7 @@ static unsigned int wmfw_convert_flags(unsigned int in, unsigned int len) } if (in) { - if (in & WMFW_CTL_FLAG_READABLE) - out |= rd; + out |= rd; if (in & WMFW_CTL_FLAG_WRITEABLE) out |= wr; if (in & WMFW_CTL_FLAG_VOLATILE) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 9aa741d27279..07bac9ea65c4 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1158,6 +1158,28 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream, return ret; } +static int davinci_mcasp_hw_rule_slot_width(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct davinci_mcasp_ruledata *rd = rule->private; + struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + struct snd_mask nfmt; + int i, slot_width; + + snd_mask_none(&nfmt); + slot_width = rd->mcasp->slot_width; + + for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) { + if (snd_mask_test(fmt, i)) { + if (snd_pcm_format_width(i) <= slot_width) { + snd_mask_set(&nfmt, i); + } + } + } + + return snd_mask_refine(fmt, &nfmt); +} + static const unsigned int davinci_mcasp_dai_rates[] = { 8000, 11025, 16000, 22050, 32000, 44100, 48000, 64000, 88200, 96000, 176400, 192000, @@ -1251,7 +1273,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, struct davinci_mcasp_ruledata *ruledata = &mcasp->ruledata[substream->stream]; u32 max_channels = 0; - int i, dir; + int i, dir, ret; int tdm_slots = mcasp->tdm_slots; /* Do not allow more then one stream per direction */ @@ -1280,6 +1302,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, max_channels++; } ruledata->serializers = max_channels; + ruledata->mcasp = mcasp; max_channels *= tdm_slots; /* * If the already active stream has less channels than the calculated @@ -1305,20 +1328,22 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, 0, SNDRV_PCM_HW_PARAM_CHANNELS, &mcasp->chconstr[substream->stream]); - if (mcasp->slot_width) - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_SAMPLE_BITS, - 8, mcasp->slot_width); + if (mcasp->slot_width) { + /* Only allow formats require <= slot_width bits on the bus */ + ret = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_FORMAT, + davinci_mcasp_hw_rule_slot_width, + ruledata, + SNDRV_PCM_HW_PARAM_FORMAT, -1); + if (ret) + return ret; + } /* * If we rely on implicit BCLK divider setting we should * set constraints based on what we can provide. */ if (mcasp->bclk_master && mcasp->bclk_div == 0 && mcasp->sysclk_freq) { - int ret; - - ruledata->mcasp = mcasp; - ret = snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, davinci_mcasp_hw_rule_rate, @@ -1723,7 +1748,8 @@ static int davinci_mcasp_get_dma_type(struct davinci_mcasp *mcasp) PTR_ERR(chan)); return PTR_ERR(chan); } - BUG_ON(!chan->device || !chan->device->dev); + if (WARN_ON(!chan->device || !chan->device->dev)) + return -EINVAL; if (chan->device->dev->of_node) ret = of_property_read_string(chan->device->dev->of_node, @@ -1869,6 +1895,10 @@ static int davinci_mcasp_probe(struct platform_device *pdev) if (irq >= 0) { irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_common", dev_name(&pdev->dev)); + if (!irq_name) { + ret = -ENOMEM; + goto err; + } ret = devm_request_threaded_irq(&pdev->dev, irq, NULL, davinci_mcasp_common_irq_handler, IRQF_ONESHOT | IRQF_SHARED, @@ -1886,6 +1916,10 @@ static int davinci_mcasp_probe(struct platform_device *pdev) if (irq >= 0) { irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_rx", dev_name(&pdev->dev)); + if (!irq_name) { + ret = -ENOMEM; + goto err; + } ret = devm_request_threaded_irq(&pdev->dev, irq, NULL, davinci_mcasp_rx_irq_handler, IRQF_ONESHOT, irq_name, mcasp); @@ -1901,6 +1935,10 @@ static int davinci_mcasp_probe(struct platform_device *pdev) if (irq >= 0) { irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_tx", dev_name(&pdev->dev)); + if (!irq_name) { + ret = -ENOMEM; + goto err; + } ret = devm_request_threaded_irq(&pdev->dev, irq, NULL, davinci_mcasp_tx_irq_handler, IRQF_ONESHOT, irq_name, mcasp); @@ -1984,8 +2022,10 @@ static int davinci_mcasp_probe(struct platform_device *pdev) GFP_KERNEL); if (!mcasp->chconstr[SNDRV_PCM_STREAM_PLAYBACK].list || - !mcasp->chconstr[SNDRV_PCM_STREAM_CAPTURE].list) - return -ENOMEM; + !mcasp->chconstr[SNDRV_PCM_STREAM_CAPTURE].list) { + ret = -ENOMEM; + goto err; + } ret = davinci_mcasp_set_ch_constraints(mcasp); if (ret) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index 04e61aa6fc9b..f0d75177cde3 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -353,8 +353,8 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair, bool p2p_in, bool p2 return -EINVAL; } - if ((outrate > 8000 && outrate < 30000) && - (outrate/inrate > 24 || inrate/outrate > 8)) { + if ((outrate >= 8000 && outrate <= 30000) && + (outrate > 24 * inrate || inrate > 8 * outrate)) { pair_err("exceed supported ratio range [1/24, 8] for \ inrate/outrate: %d/%d\n", inrate, outrate); return -EINVAL; diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 45e9de81cea9..1245db8451a1 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1459,6 +1459,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) struct fsl_ssi_private *ssi_private; int ret = 0; struct device_node *np = pdev->dev.of_node; + struct device_node *root; const struct of_device_id *of_id; const char *p, *sprop; const uint32_t *iprop; @@ -1648,7 +1649,9 @@ static int fsl_ssi_probe(struct platform_device *pdev) * device tree. We also pass the address of the CPU DAI driver * structure. */ - sprop = of_get_property(of_find_node_by_path("/"), "compatible", NULL); + root = of_find_node_by_path("/"); + sprop = of_get_property(root, "compatible", NULL); + of_node_put(root); /* Sometimes the compatible name has a "fsl," prefix, so we strip it. */ p = strrchr(sprop, ','); if (p) diff --git a/sound/soc/intel/common/sst-ipc.c b/sound/soc/intel/common/sst-ipc.c index 62f3a8e0ec87..fedce78675e8 100644 --- a/sound/soc/intel/common/sst-ipc.c +++ b/sound/soc/intel/common/sst-ipc.c @@ -231,6 +231,8 @@ struct ipc_message *sst_ipc_reply_find_msg(struct sst_generic_ipc *ipc, if (ipc->ops.reply_msg_match != NULL) header = ipc->ops.reply_msg_match(header, &mask); + else + mask = (u64)-1; if (list_empty(&ipc->rx_list)) { dev_err(ipc->dev, "error: rx list empty but received 0x%llx\n", diff --git a/sound/soc/intel/skylake/skl-debug.c b/sound/soc/intel/skylake/skl-debug.c index dc20d91f62e6..1987f78ea91e 100644 --- a/sound/soc/intel/skylake/skl-debug.c +++ b/sound/soc/intel/skylake/skl-debug.c @@ -196,7 +196,7 @@ static ssize_t fw_softreg_read(struct file *file, char __user *user_buf, memset(d->fw_read_buff, 0, FW_REG_BUF); if (w0_stat_sz > 0) - __iowrite32_copy(d->fw_read_buff, fw_reg_addr, w0_stat_sz >> 2); + __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); diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c index 55859c5b456f..1b0129478a7f 100644 --- a/sound/soc/intel/skylake/skl-nhlt.c +++ b/sound/soc/intel/skylake/skl-nhlt.c @@ -215,7 +215,7 @@ int skl_nhlt_update_topology_bin(struct skl *skl) struct hdac_bus *bus = ebus_to_hbus(&skl->ebus); struct device *dev = bus->dev; - dev_dbg(dev, "oem_id %.6s, oem_table_id %8s oem_revision %d\n", + dev_dbg(dev, "oem_id %.6s, oem_table_id %.8s oem_revision %d\n", nhlt->header.oem_id, nhlt->header.oem_table_id, nhlt->header.oem_revision); diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 105a73cc5158..149b7cba10fb 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -569,10 +569,6 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) return PTR_ERR(priv->clk); } - err = clk_prepare_enable(priv->clk); - if (err < 0) - return err; - priv->extclk = devm_clk_get(&pdev->dev, "extclk"); if (IS_ERR(priv->extclk)) { if (PTR_ERR(priv->extclk) == -EPROBE_DEFER) @@ -588,6 +584,10 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) } } + err = clk_prepare_enable(priv->clk); + if (err < 0) + return err; + /* Some sensible defaults - this reflects the powerup values */ priv->ctl_play = KIRKWOOD_PLAYCTL_SIZE_24; priv->ctl_rec = KIRKWOOD_RECCTL_SIZE_24; diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 66fc13a2396a..0e07e3dea7de 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -676,7 +676,7 @@ static int rockchip_i2s_probe(struct platform_device *pdev) ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); if (ret) { dev_err(&pdev->dev, "Could not register PCM\n"); - return ret; + goto err_suspend; } return 0; diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index eb7879bcc6a7..686401bcd1f5 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -33,6 +33,7 @@ struct rsnd_adg { struct clk *clkout[CLKOUTMAX]; struct clk_onecell_data onecell; struct rsnd_mod mod; + int clk_rate[CLKMAX]; u32 flags; u32 ckr; u32 rbga; @@ -110,9 +111,9 @@ static void __rsnd_adg_get_timesel_ratio(struct rsnd_priv *priv, unsigned int val, en; unsigned int min, diff; unsigned int sel_rate[] = { - clk_get_rate(adg->clk[CLKA]), /* 0000: CLKA */ - clk_get_rate(adg->clk[CLKB]), /* 0001: CLKB */ - clk_get_rate(adg->clk[CLKC]), /* 0010: CLKC */ + adg->clk_rate[CLKA], /* 0000: CLKA */ + adg->clk_rate[CLKB], /* 0001: CLKB */ + adg->clk_rate[CLKC], /* 0010: CLKC */ adg->rbga_rate_for_441khz, /* 0011: RBGA */ adg->rbgb_rate_for_48khz, /* 0100: RBGB */ }; @@ -328,7 +329,7 @@ int rsnd_adg_clk_query(struct rsnd_priv *priv, unsigned int rate) * AUDIO_CLKA/AUDIO_CLKB/AUDIO_CLKC/AUDIO_CLKI. */ for_each_rsnd_clk(clk, adg, i) { - if (rate == clk_get_rate(clk)) + if (rate == adg->clk_rate[i]) return sel_table[i]; } @@ -394,10 +395,18 @@ void rsnd_adg_clk_control(struct rsnd_priv *priv, int enable) for_each_rsnd_clk(clk, adg, i) { ret = 0; - if (enable) + if (enable) { ret = clk_prepare_enable(clk); - else + + /* + * We shouldn't use clk_get_rate() under + * atomic context. Let's keep it when + * rsnd_adg_clk_enable() was called + */ + adg->clk_rate[i] = clk_get_rate(adg->clk[i]); + } else { clk_disable_unprepare(clk); + } if (ret < 0) dev_warn(dev, "can't use clk %d\n", i); diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 710c01cd2ad2..f203c0878e69 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -676,6 +676,7 @@ static int rsnd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) } /* set format */ + rdai->bit_clk_inv = 0; switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: rdai->sys_delay = 0; @@ -1277,6 +1278,18 @@ int rsnd_kctrl_new(struct rsnd_mod *mod, }; int ret; + /* + * 1) Avoid duplicate register for DVC with MIX case + * 2) Allow duplicate register for MIX + * 3) re-register if card was rebinded + */ + list_for_each_entry(kctrl, &card->controls, list) { + struct rsnd_kctrl_cfg *c = kctrl->private_data; + + if (c == cfg) + return 0; + } + if (size > RSND_MAX_CHANNELS) return -EINVAL; diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 1768a0ae469d..c68b31483c7b 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -432,6 +432,7 @@ struct rsnd_dai_stream { char name[RSND_DAI_NAME_SIZE]; struct snd_pcm_substream *substream; struct rsnd_mod *mod[RSND_MOD_MAX]; + struct rsnd_mod *dma; struct rsnd_dai *rdai; u32 parent_ssi_status; }; diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 60cc550c5a4c..cae9ed6a0cdb 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -66,7 +66,6 @@ struct rsnd_ssi { struct rsnd_mod mod; - struct rsnd_mod *dma; u32 flags; u32 cr_own; @@ -868,7 +867,6 @@ static int rsnd_ssi_dma_probe(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct rsnd_priv *priv) { - struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); int ret; /* @@ -883,7 +881,7 @@ static int rsnd_ssi_dma_probe(struct rsnd_mod *mod, return ret; /* SSI probe might be called many times in MUX multi path */ - ret = rsnd_dma_attach(io, mod, &ssi->dma); + ret = rsnd_dma_attach(io, mod, &io->dma); return ret; } diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index e9f7c6287376..104d5f487c7d 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1152,8 +1152,8 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget, list_add_tail(&widget->work_list, list); if (custom_stop_condition && custom_stop_condition(widget, dir)) { - widget->endpoints[dir] = 1; - return widget->endpoints[dir]; + list = NULL; + custom_stop_condition = NULL; } if ((widget->is_ep & SND_SOC_DAPM_DIR_TO_EP(dir)) && widget->connected) { @@ -1190,8 +1190,8 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget, * * Optionally, can be supplied with a function acting as a stopping condition. * This function takes the dapm widget currently being examined and the walk - * direction as an arguments, it should return true if the walk should be - * stopped and false otherwise. + * direction as an arguments, it should return true if widgets from that point + * in the graph onwards should not be added to the widget list. */ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, struct list_head *list, @@ -2120,23 +2120,25 @@ void snd_soc_dapm_debugfs_init(struct snd_soc_dapm_context *dapm, { struct dentry *d; - if (!parent) + if (!parent || IS_ERR(parent)) return; dapm->debugfs_dapm = debugfs_create_dir("dapm", parent); - if (!dapm->debugfs_dapm) { + if (IS_ERR(dapm->debugfs_dapm)) { dev_warn(dapm->dev, - "ASoC: Failed to create DAPM debugfs directory\n"); + "ASoC: Failed to create DAPM debugfs directory %ld\n", + PTR_ERR(dapm->debugfs_dapm)); return; } d = debugfs_create_file("bias_level", 0444, dapm->debugfs_dapm, dapm, &dapm_bias_fops); - if (!d) + if (IS_ERR(d)) dev_warn(dapm->dev, - "ASoC: Failed to create bias level debugfs file\n"); + "ASoC: Failed to create bias level debugfs file: %ld\n", + PTR_ERR(d)); } static void dapm_debugfs_add_widget(struct snd_soc_dapm_widget *w) @@ -2150,10 +2152,10 @@ static void dapm_debugfs_add_widget(struct snd_soc_dapm_widget *w) d = debugfs_create_file(w->name, 0444, dapm->debugfs_dapm, w, &dapm_widget_power_fops); - if (!d) + if (IS_ERR(d)) dev_warn(w->dapm->dev, - "ASoC: Failed to create %s debugfs file\n", - w->name); + "ASoC: Failed to create %s debugfs file: %ld\n", + w->name, PTR_ERR(d)); } static void dapm_debugfs_cleanup(struct snd_soc_dapm_context *dapm) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index d53786498b61..052778c6afad 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -311,6 +311,12 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) if (!dmaengine_pcm_can_report_residue(dev, pcm->chan[i])) pcm->flags |= SND_DMAENGINE_PCM_FLAG_NO_RESIDUE; + + if (rtd->pcm->streams[i].pcm->name[0] == '\0') { + strncpy(rtd->pcm->streams[i].pcm->name, + rtd->pcm->streams[i].pcm->id, + sizeof(rtd->pcm->streams[i].pcm->name)); + } } return 0; diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 99902ae1a2d9..b04ecc633da3 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -127,10 +127,9 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) 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-pcm.c b/sound/soc/soc-pcm.c index 6b290773250b..1bb3d0406c96 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1578,7 +1578,7 @@ static void dpcm_init_runtime_hw(struct snd_pcm_runtime *runtime, u64 formats) { runtime->hw.rate_min = stream->rate_min; - runtime->hw.rate_max = stream->rate_max; + runtime->hw.rate_max = min_not_zero(stream->rate_max, UINT_MAX); runtime->hw.channels_min = stream->channels_min; runtime->hw.channels_max = stream->channels_max; if (runtime->hw.formats) @@ -2276,7 +2276,8 @@ int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream) if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) && - (be->dpcm[stream].state != SND_SOC_DPCM_STATE_SUSPEND)) + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_SUSPEND) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; dev_dbg(be->dev, "ASoC: prepare BE %s\n", diff --git a/sound/soc/stm/stm32_i2s.c b/sound/soc/stm/stm32_i2s.c index 6d0bf78d114d..aa2b1196171a 100644 --- a/sound/soc/stm/stm32_i2s.c +++ b/sound/soc/stm/stm32_i2s.c @@ -246,8 +246,8 @@ static irqreturn_t stm32_i2s_isr(int irq, void *devid) return IRQ_NONE; } - regmap_update_bits(i2s->regmap, STM32_I2S_IFCR_REG, - I2S_IFCR_MASK, flags); + regmap_write_bits(i2s->regmap, STM32_I2S_IFCR_REG, + I2S_IFCR_MASK, flags); if (flags & I2S_SR_OVR) { dev_dbg(&pdev->dev, "Overrun\n"); @@ -276,7 +276,6 @@ static bool stm32_i2s_readable_reg(struct device *dev, unsigned int reg) case STM32_I2S_CFG2_REG: case STM32_I2S_IER_REG: case STM32_I2S_SR_REG: - case STM32_I2S_IFCR_REG: case STM32_I2S_TXDR_REG: case STM32_I2S_RXDR_REG: case STM32_I2S_CGFR_REG: @@ -488,7 +487,7 @@ static int stm32_i2s_configure(struct snd_soc_dai *cpu_dai, { struct stm32_i2s_data *i2s = snd_soc_dai_get_drvdata(cpu_dai); int format = params_width(params); - u32 cfgr, cfgr_mask, cfg1, cfg1_mask; + u32 cfgr, cfgr_mask, cfg1; unsigned int fthlv; int ret; @@ -501,7 +500,7 @@ static int stm32_i2s_configure(struct snd_soc_dai *cpu_dai, switch (format) { case 16: cfgr = I2S_CGFR_DATLEN_SET(I2S_I2SMOD_DATLEN_16); - cfgr_mask = I2S_CGFR_DATLEN_MASK; + cfgr_mask = I2S_CGFR_DATLEN_MASK | I2S_CGFR_CHLEN; break; case 32: cfgr = I2S_CGFR_DATLEN_SET(I2S_I2SMOD_DATLEN_32) | @@ -529,15 +528,11 @@ static int stm32_i2s_configure(struct snd_soc_dai *cpu_dai, if (ret < 0) return ret; - cfg1 = I2S_CFG1_RXDMAEN | I2S_CFG1_TXDMAEN; - cfg1_mask = cfg1; - fthlv = STM32_I2S_FIFO_SIZE * I2S_FIFO_TH_ONE_QUARTER / 4; - cfg1 |= I2S_CFG1_FTHVL_SET(fthlv - 1); - cfg1_mask |= I2S_CFG1_FTHVL_MASK; + cfg1 = I2S_CFG1_FTHVL_SET(fthlv - 1); return regmap_update_bits(i2s->regmap, STM32_I2S_CFG1_REG, - cfg1_mask, cfg1); + I2S_CFG1_FTHVL_MASK, cfg1); } static int stm32_i2s_startup(struct snd_pcm_substream *substream, @@ -551,8 +546,8 @@ static int stm32_i2s_startup(struct snd_pcm_substream *substream, i2s->refcount++; spin_unlock(&i2s->lock_fd); - return regmap_update_bits(i2s->regmap, STM32_I2S_IFCR_REG, - I2S_IFCR_MASK, I2S_IFCR_MASK); + return regmap_write_bits(i2s->regmap, STM32_I2S_IFCR_REG, + I2S_IFCR_MASK, I2S_IFCR_MASK); } static int stm32_i2s_hw_params(struct snd_pcm_substream *substream, @@ -589,6 +584,10 @@ static int stm32_i2s_trigger(struct snd_pcm_substream *substream, int cmd, /* Enable i2s */ dev_dbg(cpu_dai->dev, "start I2S\n"); + cfg1_mask = I2S_CFG1_RXDMAEN | I2S_CFG1_TXDMAEN; + regmap_update_bits(i2s->regmap, STM32_I2S_CFG1_REG, + cfg1_mask, cfg1_mask); + ret = regmap_update_bits(i2s->regmap, STM32_I2S_CR1_REG, I2S_CR1_SPE, I2S_CR1_SPE); if (ret < 0) { @@ -603,8 +602,8 @@ static int stm32_i2s_trigger(struct snd_pcm_substream *substream, int cmd, return ret; } - regmap_update_bits(i2s->regmap, STM32_I2S_IFCR_REG, - I2S_IFCR_MASK, I2S_IFCR_MASK); + regmap_write_bits(i2s->regmap, STM32_I2S_IFCR_REG, + I2S_IFCR_MASK, I2S_IFCR_MASK); if (playback_flg) { ier = I2S_IER_UDRIE; diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index b4af5ce78ecb..da0a2083e12a 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -110,7 +110,7 @@ #define SUN8I_I2S_TX_CHAN_MAP_REG 0x44 #define SUN8I_I2S_TX_CHAN_SEL_REG 0x34 -#define SUN8I_I2S_TX_CHAN_OFFSET_MASK GENMASK(13, 11) +#define SUN8I_I2S_TX_CHAN_OFFSET_MASK GENMASK(13, 12) #define SUN8I_I2S_TX_CHAN_OFFSET(offset) (offset << 12) #define SUN8I_I2S_TX_CHAN_EN_MASK GENMASK(11, 4) #define SUN8I_I2S_TX_CHAN_EN(num_chan) (((1 << num_chan) - 1) << 4) @@ -442,6 +442,10 @@ static int sun4i_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) regmap_update_bits(i2s->regmap, SUN8I_I2S_TX_CHAN_SEL_REG, SUN8I_I2S_TX_CHAN_OFFSET_MASK, SUN8I_I2S_TX_CHAN_OFFSET(offset)); + + regmap_update_bits(i2s->regmap, SUN8I_I2S_RX_CHAN_SEL_REG, + SUN8I_I2S_TX_CHAN_OFFSET_MASK, + SUN8I_I2S_TX_CHAN_OFFSET(offset)); } regmap_field_write(i2s->field_fmt_mode, val); diff --git a/sound/soc/tegra/tegra_sgtl5000.c b/sound/soc/tegra/tegra_sgtl5000.c index 45a4aa9d2a47..901457da25ec 100644 --- a/sound/soc/tegra/tegra_sgtl5000.c +++ b/sound/soc/tegra/tegra_sgtl5000.c @@ -149,14 +149,14 @@ static int tegra_sgtl5000_driver_probe(struct platform_device *pdev) dev_err(&pdev->dev, "Property 'nvidia,i2s-controller' missing/invalid\n"); ret = -EINVAL; - goto err; + goto err_put_codec_of_node; } tegra_sgtl5000_dai.platform_of_node = tegra_sgtl5000_dai.cpu_of_node; ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev); if (ret) - goto err; + goto err_put_cpu_of_node; ret = snd_soc_register_card(card); if (ret) { @@ -169,6 +169,13 @@ static int tegra_sgtl5000_driver_probe(struct platform_device *pdev) err_fini_utils: tegra_asoc_utils_fini(&machine->util_data); +err_put_cpu_of_node: + of_node_put(tegra_sgtl5000_dai.cpu_of_node); + tegra_sgtl5000_dai.cpu_of_node = NULL; + tegra_sgtl5000_dai.platform_of_node = NULL; +err_put_codec_of_node: + of_node_put(tegra_sgtl5000_dai.codec_of_node); + tegra_sgtl5000_dai.codec_of_node = NULL; err: return ret; } @@ -183,6 +190,12 @@ static int tegra_sgtl5000_driver_remove(struct platform_device *pdev) tegra_asoc_utils_fini(&machine->util_data); + of_node_put(tegra_sgtl5000_dai.cpu_of_node); + tegra_sgtl5000_dai.cpu_of_node = NULL; + tegra_sgtl5000_dai.platform_of_node = NULL; + of_node_put(tegra_sgtl5000_dai.codec_of_node); + tegra_sgtl5000_dai.codec_of_node = NULL; + return ret; } diff --git a/sound/sound_core.c b/sound/sound_core.c index 99b73c675743..20d4e2e1bacf 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -287,7 +287,8 @@ retry: goto retry; } spin_unlock(&sound_loader_lock); - return -EBUSY; + r = -EBUSY; + goto fail; } } diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index c90607ebe155..8caf0b57f9c6 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -403,6 +403,9 @@ static void snd_complete_urb(struct urb *urb) } prepare_outbound_urb(ep, ctx); + /* can be stopped during prepare callback */ + if (unlikely(!test_bit(EP_FLAG_RUNNING, &ep->flags))) + goto exit_clear; } else { retire_inbound_urb(ep, ctx); /* can be stopped during retire callback */ diff --git a/sound/usb/line6/pcm.c b/sound/usb/line6/pcm.c index b3854f8c0c67..896add5ffee3 100644 --- a/sound/usb/line6/pcm.c +++ b/sound/usb/line6/pcm.c @@ -552,13 +552,6 @@ int line6_init_pcm(struct usb_line6 *line6, line6pcm->volume_monitor = 255; line6pcm->line6 = line6; - line6pcm->max_packet_size_in = - usb_maxpacket(line6->usbdev, - usb_rcvisocpipe(line6->usbdev, ep_read), 0); - line6pcm->max_packet_size_out = - usb_maxpacket(line6->usbdev, - usb_sndisocpipe(line6->usbdev, ep_write), 1); - spin_lock_init(&line6pcm->out.lock); spin_lock_init(&line6pcm->in.lock); line6pcm->impulse_period = LINE6_IMPULSE_DEFAULT_PERIOD; @@ -568,6 +561,18 @@ int line6_init_pcm(struct usb_line6 *line6, pcm->private_data = line6pcm; pcm->private_free = line6_cleanup_pcm; + line6pcm->max_packet_size_in = + usb_maxpacket(line6->usbdev, + usb_rcvisocpipe(line6->usbdev, ep_read), 0); + line6pcm->max_packet_size_out = + usb_maxpacket(line6->usbdev, + usb_sndisocpipe(line6->usbdev, ep_write), 1); + if (!line6pcm->max_packet_size_in || !line6pcm->max_packet_size_out) { + dev_err(line6pcm->line6->ifcdev, + "cannot get proper max packet size\n"); + return -EINVAL; + } + err = line6_create_audio_out_urbs(line6pcm); if (err < 0) return err; diff --git a/sound/usb/line6/podhd.c b/sound/usb/line6/podhd.c index ee6a68c6e1c1..1513fbaf70c2 100644 --- a/sound/usb/line6/podhd.c +++ b/sound/usb/line6/podhd.c @@ -415,7 +415,7 @@ static const struct line6_properties podhd_properties_table[] = { .name = "POD HD500", .capabilities = LINE6_CAP_PCM | LINE6_CAP_HWMON, - .altsetting = 1, + .altsetting = 0, .ep_ctrl_r = 0x81, .ep_ctrl_w = 0x01, .ep_audio_r = 0x86, diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 3d93e33b3485..044193b2364d 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -82,6 +82,7 @@ struct mixer_build { unsigned char *buffer; unsigned int buflen; DECLARE_BITMAP(unitbitmap, MAX_ID_ELEMS); + DECLARE_BITMAP(termbitmap, MAX_ID_ELEMS); struct usb_audio_term oterm; const struct usbmix_name_map *map; const struct usbmix_selector_map *selector_map; @@ -716,15 +717,24 @@ static int get_term_name(struct mixer_build *state, struct usb_audio_term *iterm * parse the source unit recursively until it reaches to a terminal * or a branched unit. */ -static int check_input_term(struct mixer_build *state, int id, +static int __check_input_term(struct mixer_build *state, int id, struct usb_audio_term *term) { int err; void *p1; + unsigned char *hdr; memset(term, 0, sizeof(*term)); - while ((p1 = find_audio_control_unit(state, id)) != NULL) { - unsigned char *hdr = p1; + for (;;) { + /* a loop in the terminal chain? */ + if (test_and_set_bit(id, state->termbitmap)) + return -EINVAL; + + p1 = find_audio_control_unit(state, id); + if (!p1) + break; + + hdr = p1; term->id = id; switch (hdr[2]) { case UAC_INPUT_TERMINAL: @@ -739,7 +749,7 @@ static int check_input_term(struct mixer_build *state, int id, /* call recursively to verify that the * referenced clock entity is valid */ - err = check_input_term(state, d->bCSourceID, term); + err = __check_input_term(state, d->bCSourceID, term); if (err < 0) return err; @@ -771,7 +781,7 @@ static int check_input_term(struct mixer_build *state, int id, case UAC2_CLOCK_SELECTOR: { struct uac_selector_unit_descriptor *d = p1; /* call recursively to retrieve the channel info */ - err = check_input_term(state, d->baSourceID[0], term); + err = __check_input_term(state, d->baSourceID[0], term); if (err < 0) return err; term->type = d->bDescriptorSubtype << 16; /* virtual type */ @@ -818,6 +828,15 @@ static int check_input_term(struct mixer_build *state, int id, return -ENODEV; } + +static int check_input_term(struct mixer_build *state, int id, + struct usb_audio_term *term) +{ + memset(term, 0, sizeof(*term)); + memset(state->termbitmap, 0, sizeof(state->termbitmap)); + return __check_input_term(state, id, term); +} + /* * Feature Unit */ @@ -1033,7 +1052,8 @@ static int get_min_max_with_quirks(struct usb_mixer_elem_info *cval, if (cval->min + cval->res < cval->max) { int last_valid_res = cval->res; int saved, test, check; - get_cur_mix_raw(cval, minchn, &saved); + if (get_cur_mix_raw(cval, minchn, &saved) < 0) + goto no_res_check; for (;;) { test = saved; if (test < cval->max) @@ -1053,6 +1073,7 @@ static int get_min_max_with_quirks(struct usb_mixer_elem_info *cval, snd_usb_set_cur_mix_value(cval, minchn, 0, saved); } +no_res_check: cval->initialized = 1; } @@ -1700,6 +1721,7 @@ static int parse_audio_mixer_unit(struct mixer_build *state, int unitid, int pin, ich, err; if (desc->bLength < 11 || !(input_pins = desc->bNrInPins) || + desc->bLength < sizeof(*desc) + desc->bNrInPins || !(num_outs = uac_mixer_unit_bNrChannels(desc))) { usb_audio_err(state->chip, "invalid MIXER UNIT descriptor %d\n", diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index e1e7ce9ab217..b54f7dab8372 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -754,7 +754,7 @@ static int snd_ni_control_init_val(struct usb_mixer_interface *mixer, return err; } - kctl->private_value |= (value << 24); + kctl->private_value |= ((unsigned int)value << 24); return 0; } @@ -915,7 +915,7 @@ static int snd_ftu_eff_switch_init(struct usb_mixer_interface *mixer, if (err < 0) return err; - kctl->private_value |= value[0] << 24; + kctl->private_value |= (unsigned int)value[0] << 24; return 0; } diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index b1a1eb1f65aa..ff38fca1781b 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -470,6 +470,7 @@ static int set_sync_endpoint(struct snd_usb_substream *subs, } ep = get_endpoint(alts, 1)->bEndpointAddress; if (get_endpoint(alts, 0)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE && + get_endpoint(alts, 0)->bSynchAddress != 0 && ((is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress | USB_DIR_IN)) || (!is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress & ~USB_DIR_IN)))) { dev_err(&dev->dev, |