diff options
Diffstat (limited to 'sound')
163 files changed, 1413 insertions, 559 deletions
diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 7ae8e24dc1e6..81624f6e3f33 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -723,6 +723,9 @@ static int snd_compr_stop(struct snd_compr_stream *stream) retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_STOP); if (!retval) { + /* clear flags and stop any drain wait */ + stream->partial_drain = false; + stream->metadata_set = false; snd_compr_drain_notify(stream); stream->runtime->total_bytes_available = 0; stream->runtime->total_bytes_transferred = 0; @@ -880,6 +883,7 @@ static int snd_compr_partial_drain(struct snd_compr_stream *stream) if (stream->next_track == false) return -EPERM; + stream->partial_drain = true; retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_PARTIAL_DRAIN); if (retval) { pr_debug("Partial drain returned failure\n"); diff --git a/sound/core/control.c b/sound/core/control.c index a0ce22164957..29012534ffb2 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1387,7 +1387,7 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, unlock: up_write(&card->controls_rwsem); - return 0; + return err; } static int snd_ctl_elem_add_user(struct snd_ctl_file *file, diff --git a/sound/core/control_compat.c b/sound/core/control_compat.c index 507fd5210c1c..3fc216644e0e 100644 --- a/sound/core/control_compat.c +++ b/sound/core/control_compat.c @@ -279,6 +279,7 @@ static int copy_ctl_value_to_user(void __user *userdata, struct snd_ctl_elem_value *data, int type, int count) { + struct snd_ctl_elem_value32 __user *data32 = userdata; int i, size; if (type == SNDRV_CTL_ELEM_TYPE_BOOLEAN || @@ -295,6 +296,8 @@ static int copy_ctl_value_to_user(void __user *userdata, if (copy_to_user(valuep, data->value.bytes.data, size)) return -EFAULT; } + if (copy_to_user(&data32->id, &data->id, sizeof(data32->id))) + return -EFAULT; return 0; } diff --git a/sound/core/hwdep.c b/sound/core/hwdep.c index a73baa1242be..727219f40201 100644 --- a/sound/core/hwdep.c +++ b/sound/core/hwdep.c @@ -229,14 +229,14 @@ static int snd_hwdep_dsp_load(struct snd_hwdep *hw, if (copy_from_user(&info, _info, sizeof(info))) return -EFAULT; /* check whether the dsp was already loaded */ - if (hw->dsp_loaded & (1 << info.index)) + if (hw->dsp_loaded & (1u << info.index)) return -EBUSY; if (!access_ok(VERIFY_READ, info.image, info.length)) return -EFAULT; err = hw->ops.dsp_load(hw, &info); if (err < 0) return err; - hw->dsp_loaded |= (1 << info.index); + hw->dsp_loaded |= (1u << info.index); return 0; } diff --git a/sound/core/info.c b/sound/core/info.c index 5fb00437507b..f15569cd124d 100644 --- a/sound/core/info.c +++ b/sound/core/info.c @@ -634,7 +634,9 @@ int snd_info_get_line(struct snd_info_buffer *buffer, char *line, int len) { int c = -1; - if (snd_BUG_ON(!buffer || !buffer->buffer)) + if (snd_BUG_ON(!buffer)) + return 1; + if (!buffer->buffer) return 1; if (len <= 0 || buffer->stop || buffer->error) return 1; diff --git a/sound/core/init.c b/sound/core/init.c index dcb9199f5e4f..7fdeae4dc820 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -404,10 +404,8 @@ int snd_card_disconnect(struct snd_card *card) return 0; } card->shutdown = 1; - spin_unlock(&card->files_lock); /* replace file->f_op with special dummy operations */ - spin_lock(&card->files_lock); list_for_each_entry(mfile, &card->files_list, list) { /* it's critical part, use endless loop */ /* we have no room to fail */ diff --git a/sound/core/jack.c b/sound/core/jack.c index f652e90efd7e..5ddf81f091fa 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -234,6 +234,10 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, return -ENOMEM; jack->id = kstrdup(id, GFP_KERNEL); + if (jack->id == NULL) { + kfree(jack); + return -ENOMEM; + } /* don't creat input device for phantom jack */ if (!phantom_jack) { diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index 379bf486ccc7..c91048fd2822 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -145,11 +145,13 @@ static int snd_mixer_oss_devmask(struct snd_mixer_oss_file *fmixer) if (mixer == NULL) return -EIO; + mutex_lock(&mixer->reg_mutex); for (chn = 0; chn < 31; chn++) { pslot = &mixer->slots[chn]; if (pslot->put_volume || pslot->put_recsrc) result |= 1 << chn; } + mutex_unlock(&mixer->reg_mutex); return result; } @@ -161,11 +163,13 @@ static int snd_mixer_oss_stereodevs(struct snd_mixer_oss_file *fmixer) if (mixer == NULL) return -EIO; + mutex_lock(&mixer->reg_mutex); for (chn = 0; chn < 31; chn++) { pslot = &mixer->slots[chn]; if (pslot->put_volume && pslot->stereo) result |= 1 << chn; } + mutex_unlock(&mixer->reg_mutex); return result; } @@ -176,6 +180,7 @@ static int snd_mixer_oss_recmask(struct snd_mixer_oss_file *fmixer) if (mixer == NULL) return -EIO; + mutex_lock(&mixer->reg_mutex); if (mixer->put_recsrc && mixer->get_recsrc) { /* exclusive */ result = mixer->mask_recsrc; } else { @@ -187,6 +192,7 @@ static int snd_mixer_oss_recmask(struct snd_mixer_oss_file *fmixer) result |= 1 << chn; } } + mutex_unlock(&mixer->reg_mutex); return result; } @@ -197,11 +203,12 @@ static int snd_mixer_oss_get_recsrc(struct snd_mixer_oss_file *fmixer) if (mixer == NULL) return -EIO; + mutex_lock(&mixer->reg_mutex); if (mixer->put_recsrc && mixer->get_recsrc) { /* exclusive */ - int err; unsigned int index; - if ((err = mixer->get_recsrc(fmixer, &index)) < 0) - return err; + result = mixer->get_recsrc(fmixer, &index); + if (result < 0) + goto unlock; result = 1 << index; } else { struct snd_mixer_oss_slot *pslot; @@ -216,7 +223,10 @@ static int snd_mixer_oss_get_recsrc(struct snd_mixer_oss_file *fmixer) } } } - return mixer->oss_recsrc = result; + mixer->oss_recsrc = result; + unlock: + mutex_unlock(&mixer->reg_mutex); + return result; } static int snd_mixer_oss_set_recsrc(struct snd_mixer_oss_file *fmixer, int recsrc) @@ -229,6 +239,7 @@ static int snd_mixer_oss_set_recsrc(struct snd_mixer_oss_file *fmixer, int recsr if (mixer == NULL) return -EIO; + mutex_lock(&mixer->reg_mutex); if (mixer->get_recsrc && mixer->put_recsrc) { /* exclusive input */ if (recsrc & ~mixer->oss_recsrc) recsrc &= ~mixer->oss_recsrc; @@ -254,6 +265,7 @@ static int snd_mixer_oss_set_recsrc(struct snd_mixer_oss_file *fmixer, int recsr } } } + mutex_unlock(&mixer->reg_mutex); return result; } @@ -265,6 +277,7 @@ static int snd_mixer_oss_get_volume(struct snd_mixer_oss_file *fmixer, int slot) if (mixer == NULL || slot > 30) return -EIO; + mutex_lock(&mixer->reg_mutex); pslot = &mixer->slots[slot]; left = pslot->volume[0]; right = pslot->volume[1]; @@ -272,15 +285,21 @@ static int snd_mixer_oss_get_volume(struct snd_mixer_oss_file *fmixer, int slot) result = pslot->get_volume(fmixer, pslot, &left, &right); if (!pslot->stereo) right = left; - if (snd_BUG_ON(left < 0 || left > 100)) - return -EIO; - if (snd_BUG_ON(right < 0 || right > 100)) - return -EIO; + if (snd_BUG_ON(left < 0 || left > 100)) { + result = -EIO; + goto unlock; + } + if (snd_BUG_ON(right < 0 || right > 100)) { + result = -EIO; + goto unlock; + } if (result >= 0) { pslot->volume[0] = left; pslot->volume[1] = right; result = (left & 0xff) | ((right & 0xff) << 8); } + unlock: + mutex_unlock(&mixer->reg_mutex); return result; } @@ -293,6 +312,7 @@ static int snd_mixer_oss_set_volume(struct snd_mixer_oss_file *fmixer, if (mixer == NULL || slot > 30) return -EIO; + mutex_lock(&mixer->reg_mutex); pslot = &mixer->slots[slot]; if (left > 100) left = 100; @@ -303,10 +323,13 @@ static int snd_mixer_oss_set_volume(struct snd_mixer_oss_file *fmixer, if (pslot->put_volume) result = pslot->put_volume(fmixer, pslot, left, right); if (result < 0) - return result; + goto unlock; pslot->volume[0] = left; pslot->volume[1] = right; - return (left & 0xff) | ((right & 0xff) << 8); + result = (left & 0xff) | ((right & 0xff) << 8); + unlock: + mutex_unlock(&mixer->reg_mutex); + return result; } static int snd_mixer_oss_ioctl1(struct snd_mixer_oss_file *fmixer, unsigned int cmd, unsigned long arg) diff --git a/sound/core/oss/mulaw.c b/sound/core/oss/mulaw.c index 3788906421a7..fe27034f2846 100644 --- a/sound/core/oss/mulaw.c +++ b/sound/core/oss/mulaw.c @@ -329,8 +329,8 @@ int snd_pcm_plugin_build_mulaw(struct snd_pcm_substream *plug, snd_BUG(); return -EINVAL; } - if (snd_BUG_ON(!snd_pcm_format_linear(format->format))) - return -ENXIO; + if (!snd_pcm_format_linear(format->format)) + return -EINVAL; err = snd_pcm_plugin_build(plug, "Mu-Law<->linear conversion", src_format, dst_format, diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index bb0ab0f6ce9d..b092f257c1c6 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -162,7 +162,7 @@ snd_pcm_hw_param_value_min(const struct snd_pcm_hw_params *params, * * Return the maximum value for field PAR. */ -static unsigned int +static int snd_pcm_hw_param_value_max(const struct snd_pcm_hw_params *params, snd_pcm_hw_param_t var, int *dir) { @@ -697,17 +697,25 @@ static int snd_pcm_oss_period_size(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *oss_params, struct snd_pcm_hw_params *slave_params) { - size_t s; - size_t oss_buffer_size, oss_period_size, oss_periods; - size_t min_period_size, max_period_size; + ssize_t s; + ssize_t oss_buffer_size; + ssize_t oss_period_size, oss_periods; + ssize_t min_period_size, max_period_size; struct snd_pcm_runtime *runtime = substream->runtime; size_t oss_frame_size; oss_frame_size = snd_pcm_format_physical_width(params_format(oss_params)) * params_channels(oss_params) / 8; + oss_buffer_size = snd_pcm_hw_param_value_max(slave_params, + SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + NULL); + if (oss_buffer_size <= 0) + return -EINVAL; oss_buffer_size = snd_pcm_plug_client_size(substream, - snd_pcm_hw_param_value_max(slave_params, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, NULL)) * oss_frame_size; + oss_buffer_size * oss_frame_size); + if (oss_buffer_size <= 0) + return -EINVAL; oss_buffer_size = rounddown_pow_of_two(oss_buffer_size); if (atomic_read(&substream->mmap_count)) { if (oss_buffer_size > runtime->oss.mmap_bytes) @@ -743,17 +751,21 @@ static int snd_pcm_oss_period_size(struct snd_pcm_substream *substream, min_period_size = snd_pcm_plug_client_size(substream, snd_pcm_hw_param_value_min(slave_params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, NULL)); - min_period_size *= oss_frame_size; - min_period_size = roundup_pow_of_two(min_period_size); - if (oss_period_size < min_period_size) - oss_period_size = min_period_size; + if (min_period_size > 0) { + min_period_size *= oss_frame_size; + min_period_size = roundup_pow_of_two(min_period_size); + if (oss_period_size < min_period_size) + oss_period_size = min_period_size; + } max_period_size = snd_pcm_plug_client_size(substream, snd_pcm_hw_param_value_max(slave_params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, NULL)); - max_period_size *= oss_frame_size; - max_period_size = rounddown_pow_of_two(max_period_size); - if (oss_period_size > max_period_size) - oss_period_size = max_period_size; + if (max_period_size > 0) { + max_period_size *= oss_frame_size; + max_period_size = rounddown_pow_of_two(max_period_size); + if (oss_period_size > max_period_size) + oss_period_size = max_period_size; + } oss_periods = oss_buffer_size / oss_period_size; @@ -761,7 +773,7 @@ static int snd_pcm_oss_period_size(struct snd_pcm_substream *substream, oss_periods = substream->oss.setup.periods; s = snd_pcm_hw_param_value_max(slave_params, SNDRV_PCM_HW_PARAM_PERIODS, NULL); - if (runtime->oss.maxfrags && s > runtime->oss.maxfrags) + if (s > 0 && runtime->oss.maxfrags && s > runtime->oss.maxfrags) s = runtime->oss.maxfrags; if (oss_periods > s) oss_periods = s; @@ -887,8 +899,15 @@ static int snd_pcm_oss_change_params_locked(struct snd_pcm_substream *substream) err = -EINVAL; goto failure; } - choose_rate(substream, sparams, runtime->oss.rate); - snd_pcm_hw_param_near(substream, sparams, SNDRV_PCM_HW_PARAM_CHANNELS, runtime->oss.channels, NULL); + + err = choose_rate(substream, sparams, runtime->oss.rate); + if (err < 0) + goto failure; + err = snd_pcm_hw_param_near(substream, sparams, + SNDRV_PCM_HW_PARAM_CHANNELS, + runtime->oss.channels, NULL); + if (err < 0) + goto failure; format = snd_pcm_oss_format_from(runtime->oss.format); @@ -1949,11 +1968,15 @@ static int snd_pcm_oss_set_subdivide(struct snd_pcm_oss_file *pcm_oss_file, int static int snd_pcm_oss_set_fragment1(struct snd_pcm_substream *substream, unsigned int val) { struct snd_pcm_runtime *runtime; + int fragshift; runtime = substream->runtime; if (runtime->oss.subdivision || runtime->oss.fragshift) return -EINVAL; - runtime->oss.fragshift = val & 0xffff; + fragshift = val & 0xffff; + if (fragshift >= 25) /* should be large enough */ + return -EINVAL; + runtime->oss.fragshift = fragshift; runtime->oss.maxfrags = (val >> 16) & 0xffff; if (runtime->oss.fragshift < 4) /* < 16 */ runtime->oss.fragshift = 4; diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index 40d2d39151bf..c1315ce98b54 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -196,7 +196,9 @@ int snd_pcm_plugin_free(struct snd_pcm_plugin *plugin) return 0; } -snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *plug, snd_pcm_uframes_t drv_frames) +static snd_pcm_sframes_t plug_client_size(struct snd_pcm_substream *plug, + snd_pcm_uframes_t drv_frames, + bool check_size) { struct snd_pcm_plugin *plugin, *plugin_prev, *plugin_next; int stream; @@ -209,21 +211,23 @@ snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *plug, snd_p if (stream == SNDRV_PCM_STREAM_PLAYBACK) { plugin = snd_pcm_plug_last(plug); while (plugin && drv_frames > 0) { - if (drv_frames > plugin->buf_frames) - drv_frames = plugin->buf_frames; plugin_prev = plugin->prev; if (plugin->src_frames) drv_frames = plugin->src_frames(plugin, drv_frames); + if (check_size && plugin->buf_frames && + drv_frames > plugin->buf_frames) + drv_frames = plugin->buf_frames; plugin = plugin_prev; } } else if (stream == SNDRV_PCM_STREAM_CAPTURE) { plugin = snd_pcm_plug_first(plug); while (plugin && drv_frames > 0) { plugin_next = plugin->next; + if (check_size && plugin->buf_frames && + drv_frames > plugin->buf_frames) + drv_frames = plugin->buf_frames; if (plugin->dst_frames) drv_frames = plugin->dst_frames(plugin, drv_frames); - if (drv_frames > plugin->buf_frames) - drv_frames = plugin->buf_frames; plugin = plugin_next; } } else @@ -231,7 +235,9 @@ snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *plug, snd_p return drv_frames; } -snd_pcm_sframes_t snd_pcm_plug_slave_size(struct snd_pcm_substream *plug, snd_pcm_uframes_t clt_frames) +static snd_pcm_sframes_t plug_slave_size(struct snd_pcm_substream *plug, + snd_pcm_uframes_t clt_frames, + bool check_size) { struct snd_pcm_plugin *plugin, *plugin_prev, *plugin_next; snd_pcm_sframes_t frames; @@ -247,26 +253,28 @@ snd_pcm_sframes_t snd_pcm_plug_slave_size(struct snd_pcm_substream *plug, snd_pc plugin = snd_pcm_plug_first(plug); while (plugin && frames > 0) { plugin_next = plugin->next; + if (check_size && plugin->buf_frames && + frames > plugin->buf_frames) + frames = plugin->buf_frames; if (plugin->dst_frames) { frames = plugin->dst_frames(plugin, frames); if (frames < 0) return frames; } - if (frames > plugin->buf_frames) - frames = plugin->buf_frames; plugin = plugin_next; } } else if (stream == SNDRV_PCM_STREAM_CAPTURE) { plugin = snd_pcm_plug_last(plug); while (plugin) { - if (frames > plugin->buf_frames) - frames = plugin->buf_frames; plugin_prev = plugin->prev; if (plugin->src_frames) { frames = plugin->src_frames(plugin, frames); if (frames < 0) return frames; } + if (check_size && plugin->buf_frames && + frames > plugin->buf_frames) + frames = plugin->buf_frames; plugin = plugin_prev; } } else @@ -274,6 +282,18 @@ snd_pcm_sframes_t snd_pcm_plug_slave_size(struct snd_pcm_substream *plug, snd_pc return frames; } +snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *plug, + snd_pcm_uframes_t drv_frames) +{ + return plug_client_size(plug, drv_frames, false); +} + +snd_pcm_sframes_t snd_pcm_plug_slave_size(struct snd_pcm_substream *plug, + snd_pcm_uframes_t clt_frames) +{ + return plug_slave_size(plug, clt_frames, false); +} + static int snd_pcm_plug_formats(const struct snd_mask *mask, snd_pcm_format_t format) { @@ -629,7 +649,7 @@ snd_pcm_sframes_t snd_pcm_plug_write_transfer(struct snd_pcm_substream *plug, st src_channels = dst_channels; plugin = next; } - return snd_pcm_plug_client_size(plug, frames); + return plug_client_size(plug, frames, true); } snd_pcm_sframes_t snd_pcm_plug_read_transfer(struct snd_pcm_substream *plug, struct snd_pcm_plugin_channel *dst_channels_final, snd_pcm_uframes_t size) @@ -639,7 +659,7 @@ snd_pcm_sframes_t snd_pcm_plug_read_transfer(struct snd_pcm_substream *plug, str snd_pcm_sframes_t frames = size; int err; - frames = snd_pcm_plug_slave_size(plug, frames); + frames = plug_slave_size(plug, frames, true); if (frames < 0) return frames; diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 80453266a2de..82a7387ba9d2 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -440,6 +440,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, no_delta_check: if (runtime->status->hw_ptr == new_hw_ptr) { + runtime->hw_ptr_jiffies = curr_jiffies; update_audio_tstamp(substream, &curr_tstamp, &audio_tstamp); return 0; } @@ -1756,7 +1757,7 @@ static int snd_pcm_lib_ioctl_fifo_size(struct snd_pcm_substream *substream, channels = params_channels(params); frame_size = snd_pcm_format_size(format, channels); if (frame_size > 0) - params->fifo_size /= (unsigned)frame_size; + params->fifo_size /= frame_size; } return 0; } diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 14b1ee29509d..c78db361cbba 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -721,8 +721,13 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, runtime->boundary *= 2; /* clear the buffer for avoiding possible kernel info leaks */ - if (runtime->dma_area && !substream->ops->copy_user) - memset(runtime->dma_area, 0, runtime->dma_bytes); + if (runtime->dma_area && !substream->ops->copy_user) { + size_t size = runtime->dma_bytes; + + if (runtime->info & SNDRV_PCM_INFO_MMAP) + size = PAGE_ALIGN(size); + memset(runtime->dma_area, 0, size); + } snd_pcm_timer_resolution_change(substream); snd_pcm_set_state(substream, SNDRV_PCM_STATE_SETUP); @@ -1950,6 +1955,11 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd) } pcm_file = f.file->private_data; substream1 = pcm_file->substream; + if (substream == substream1) { + res = -EINVAL; + goto _badf; + } + group = kmalloc(sizeof(*group), GFP_KERNEL); if (!group) { res = -ENOMEM; diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index d22472ba211e..0120624c4120 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -108,6 +108,17 @@ static void snd_rawmidi_input_event_work(struct work_struct *work) runtime->event(runtime->substream); } +/* buffer refcount management: call with runtime->lock held */ +static inline void snd_rawmidi_buffer_ref(struct snd_rawmidi_runtime *runtime) +{ + runtime->buffer_ref++; +} + +static inline void snd_rawmidi_buffer_unref(struct snd_rawmidi_runtime *runtime) +{ + runtime->buffer_ref--; +} + static int snd_rawmidi_runtime_create(struct snd_rawmidi_substream *substream) { struct snd_rawmidi_runtime *runtime; @@ -125,7 +136,7 @@ static int snd_rawmidi_runtime_create(struct snd_rawmidi_substream *substream) runtime->avail = 0; else runtime->avail = runtime->buffer_size; - if ((runtime->buffer = kmalloc(runtime->buffer_size, GFP_KERNEL)) == NULL) { + if ((runtime->buffer = kzalloc(runtime->buffer_size, GFP_KERNEL)) == NULL) { kfree(runtime); return -ENOMEM; } @@ -650,10 +661,15 @@ int snd_rawmidi_output_params(struct snd_rawmidi_substream *substream, return -EINVAL; } if (params->buffer_size != runtime->buffer_size) { - newbuf = kmalloc(params->buffer_size, GFP_KERNEL); + newbuf = kzalloc(params->buffer_size, GFP_KERNEL); if (!newbuf) return -ENOMEM; spin_lock_irq(&runtime->lock); + if (runtime->buffer_ref) { + spin_unlock_irq(&runtime->lock); + kfree(newbuf); + return -EBUSY; + } oldbuf = runtime->buffer; runtime->buffer = newbuf; runtime->buffer_size = params->buffer_size; @@ -962,8 +978,10 @@ static long snd_rawmidi_kernel_read1(struct snd_rawmidi_substream *substream, long result = 0, count1; struct snd_rawmidi_runtime *runtime = substream->runtime; unsigned long appl_ptr; + int err = 0; spin_lock_irqsave(&runtime->lock, flags); + snd_rawmidi_buffer_ref(runtime); while (count > 0 && runtime->avail) { count1 = runtime->buffer_size - runtime->appl_ptr; if (count1 > count) @@ -982,16 +1000,19 @@ static long snd_rawmidi_kernel_read1(struct snd_rawmidi_substream *substream, if (userbuf) { spin_unlock_irqrestore(&runtime->lock, flags); if (copy_to_user(userbuf + result, - runtime->buffer + appl_ptr, count1)) { - return result > 0 ? result : -EFAULT; - } + runtime->buffer + appl_ptr, count1)) + err = -EFAULT; spin_lock_irqsave(&runtime->lock, flags); + if (err) + goto out; } result += count1; count -= count1; } + out: + snd_rawmidi_buffer_unref(runtime); spin_unlock_irqrestore(&runtime->lock, flags); - return result; + return result > 0 ? result : err; } long snd_rawmidi_kernel_read(struct snd_rawmidi_substream *substream, @@ -1262,6 +1283,7 @@ static long snd_rawmidi_kernel_write1(struct snd_rawmidi_substream *substream, return -EAGAIN; } } + snd_rawmidi_buffer_ref(runtime); while (count > 0 && runtime->avail > 0) { count1 = runtime->buffer_size - runtime->appl_ptr; if (count1 > count) @@ -1293,6 +1315,7 @@ static long snd_rawmidi_kernel_write1(struct snd_rawmidi_substream *substream, } __end: count1 = runtime->avail < runtime->buffer_size; + snd_rawmidi_buffer_unref(runtime); spin_unlock_irqrestore(&runtime->lock, flags); if (count1) snd_rawmidi_output_trigger(substream, 1); diff --git a/sound/core/seq/oss/seq_oss.c b/sound/core/seq/oss/seq_oss.c index 8cdf489df80e..ade880fe24a4 100644 --- a/sound/core/seq/oss/seq_oss.c +++ b/sound/core/seq/oss/seq_oss.c @@ -181,10 +181,19 @@ static long odev_ioctl(struct file *file, unsigned int cmd, unsigned long arg) { struct seq_oss_devinfo *dp; + long rc; + dp = file->private_data; if (snd_BUG_ON(!dp)) return -ENXIO; - return snd_seq_oss_ioctl(dp, cmd, arg); + + if (cmd != SNDCTL_SEQ_SYNC && + mutex_lock_interruptible(®ister_mutex)) + return -ERESTARTSYS; + rc = snd_seq_oss_ioctl(dp, cmd, arg); + if (cmd != SNDCTL_SEQ_SYNC) + mutex_unlock(®ister_mutex); + return rc; } #ifdef CONFIG_COMPAT diff --git a/sound/core/seq/oss/seq_oss_synth.c b/sound/core/seq/oss/seq_oss_synth.c index c93945917235..247b68790a52 100644 --- a/sound/core/seq/oss/seq_oss_synth.c +++ b/sound/core/seq/oss/seq_oss_synth.c @@ -624,7 +624,8 @@ snd_seq_oss_synth_make_info(struct seq_oss_devinfo *dp, int dev, struct synth_in if (info->is_midi) { struct midi_info minf; - snd_seq_oss_midi_make_info(dp, info->midi_mapped, &minf); + if (snd_seq_oss_midi_make_info(dp, info->midi_mapped, &minf)) + return -ENXIO; inf->synth_type = SYNTH_TYPE_MIDI; inf->synth_subtype = 0; inf->nr_voices = 16; diff --git a/sound/core/seq/seq_ports.c b/sound/core/seq/seq_ports.c index c8fa4336bccd..86fb5eea9e4d 100644 --- a/sound/core/seq/seq_ports.c +++ b/sound/core/seq/seq_ports.c @@ -532,10 +532,11 @@ static int check_and_subscribe_port(struct snd_seq_client *client, return err; } -static void delete_and_unsubscribe_port(struct snd_seq_client *client, - struct snd_seq_client_port *port, - struct snd_seq_subscribers *subs, - bool is_src, bool ack) +/* called with grp->list_mutex held */ +static void __delete_and_unsubscribe_port(struct snd_seq_client *client, + struct snd_seq_client_port *port, + struct snd_seq_subscribers *subs, + bool is_src, bool ack) { struct snd_seq_port_subs_info *grp; struct list_head *list; @@ -543,7 +544,6 @@ static void delete_and_unsubscribe_port(struct snd_seq_client *client, grp = is_src ? &port->c_src : &port->c_dest; list = is_src ? &subs->src_list : &subs->dest_list; - down_write(&grp->list_mutex); write_lock_irq(&grp->list_lock); empty = list_empty(list); if (!empty) @@ -553,6 +553,18 @@ static void delete_and_unsubscribe_port(struct snd_seq_client *client, if (!empty) unsubscribe_port(client, port, grp, &subs->info, ack); +} + +static void delete_and_unsubscribe_port(struct snd_seq_client *client, + struct snd_seq_client_port *port, + struct snd_seq_subscribers *subs, + bool is_src, bool ack) +{ + struct snd_seq_port_subs_info *grp; + + grp = is_src ? &port->c_src : &port->c_dest; + down_write(&grp->list_mutex); + __delete_and_unsubscribe_port(client, port, subs, is_src, ack); up_write(&grp->list_mutex); } @@ -608,27 +620,30 @@ int snd_seq_port_disconnect(struct snd_seq_client *connector, struct snd_seq_client_port *dest_port, struct snd_seq_port_subscribe *info) { - struct snd_seq_port_subs_info *src = &src_port->c_src; + struct snd_seq_port_subs_info *dest = &dest_port->c_dest; struct snd_seq_subscribers *subs; int err = -ENOENT; - down_write(&src->list_mutex); + /* always start from deleting the dest port for avoiding concurrent + * deletions + */ + down_write(&dest->list_mutex); /* look for the connection */ - list_for_each_entry(subs, &src->list_head, src_list) { + list_for_each_entry(subs, &dest->list_head, dest_list) { if (match_subs_info(info, &subs->info)) { - atomic_dec(&subs->ref_count); /* mark as not ready */ + __delete_and_unsubscribe_port(dest_client, dest_port, + subs, false, + connector->number != dest_client->number); err = 0; break; } } - up_write(&src->list_mutex); + up_write(&dest->list_mutex); if (err < 0) return err; delete_and_unsubscribe_port(src_client, src_port, subs, true, connector->number != src_client->number); - delete_and_unsubscribe_port(dest_client, dest_port, subs, false, - connector->number != dest_client->number); kfree(subs); return 0; } diff --git a/sound/core/seq/seq_queue.h b/sound/core/seq/seq_queue.h index 719093489a2c..7909cf6040e3 100644 --- a/sound/core/seq/seq_queue.h +++ b/sound/core/seq/seq_queue.h @@ -40,10 +40,10 @@ struct snd_seq_queue { struct snd_seq_timer *timer; /* time keeper for this queue */ int owner; /* client that 'owns' the timer */ - unsigned int locked:1, /* timer is only accesibble by owner if set */ - klocked:1, /* kernel lock (after START) */ - check_again:1, - check_blocked:1; + bool locked; /* timer is only accesibble by owner if set */ + bool klocked; /* kernel lock (after START) */ + bool check_again; /* concurrent access happened during check */ + bool check_blocked; /* queue being checked */ unsigned int flags; /* status flags */ unsigned int info_flags; /* info for sync */ diff --git a/sound/core/seq_device.c b/sound/core/seq_device.c index e40a2cba5002..5d16b2079119 100644 --- a/sound/core/seq_device.c +++ b/sound/core/seq_device.c @@ -162,6 +162,8 @@ static int snd_seq_device_dev_free(struct snd_device *device) struct snd_seq_device *dev = device->device_data; cancel_autoload_drivers(); + if (dev->private_free) + dev->private_free(dev); put_device(&dev->dev); return 0; } @@ -189,11 +191,7 @@ static int snd_seq_device_dev_disconnect(struct snd_device *device) static void snd_seq_dev_release(struct device *dev) { - struct snd_seq_device *sdev = to_seq_dev(dev); - - if (sdev->private_free) - sdev->private_free(sdev); - kfree(sdev); + kfree(to_seq_dev(dev)); } /* diff --git a/sound/core/timer.c b/sound/core/timer.c index 22589a073423..c333ceb80d5f 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -490,9 +490,10 @@ static void snd_timer_notify1(struct snd_timer_instance *ti, int event) return; if (timer->hw.flags & SNDRV_TIMER_HW_SLAVE) return; + event += 10; /* convert to SNDRV_TIMER_EVENT_MXXX */ list_for_each_entry(ts, &ti->slave_active_head, active_list) if (ts->ccallback) - ts->ccallback(ts, event + 100, &tstamp, resolution); + ts->ccallback(ts, event, &tstamp, resolution); } /* start/continue a master timer */ @@ -582,13 +583,13 @@ static int snd_timer_stop1(struct snd_timer_instance *timeri, bool stop) if (!timer) return -EINVAL; spin_lock_irqsave(&timer->lock, flags); + list_del_init(&timeri->ack_list); + list_del_init(&timeri->active_list); if (!(timeri->flags & (SNDRV_TIMER_IFLG_RUNNING | SNDRV_TIMER_IFLG_START))) { result = -EBUSY; goto unlock; } - list_del_init(&timeri->ack_list); - list_del_init(&timeri->active_list); if (timer->card && timer->card->shutdown) goto unlock; if (stop) { @@ -623,23 +624,22 @@ static int snd_timer_stop1(struct snd_timer_instance *timeri, bool stop) static int snd_timer_stop_slave(struct snd_timer_instance *timeri, bool stop) { unsigned long flags; + bool running; spin_lock_irqsave(&slave_active_lock, flags); - if (!(timeri->flags & SNDRV_TIMER_IFLG_RUNNING)) { - spin_unlock_irqrestore(&slave_active_lock, flags); - return -EBUSY; - } + running = timeri->flags & SNDRV_TIMER_IFLG_RUNNING; timeri->flags &= ~SNDRV_TIMER_IFLG_RUNNING; if (timeri->timer) { spin_lock(&timeri->timer->lock); list_del_init(&timeri->ack_list); list_del_init(&timeri->active_list); - snd_timer_notify1(timeri, stop ? SNDRV_TIMER_EVENT_STOP : - SNDRV_TIMER_EVENT_PAUSE); + if (running) + snd_timer_notify1(timeri, stop ? SNDRV_TIMER_EVENT_STOP : + SNDRV_TIMER_EVENT_PAUSE); spin_unlock(&timeri->timer->lock); } spin_unlock_irqrestore(&slave_active_lock, flags); - return 0; + return running ? 0 : -EBUSY; } /* diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index dfd30a80ece8..8a32a276bd70 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -1062,6 +1062,14 @@ static int loopback_mixer_new(struct loopback *loopback, int notify) return -ENOMEM; kctl->id.device = dev; kctl->id.subdevice = substr; + + /* Add the control before copying the id so that + * the numid field of the id is set in the copy. + */ + err = snd_ctl_add(card, kctl); + if (err < 0) + return err; + switch (idx) { case ACTIVE_IDX: setup->active_id = kctl->id; @@ -1078,9 +1086,6 @@ static int loopback_mixer_new(struct loopback *loopback, int notify) default: break; } - err = snd_ctl_add(card, kctl); - if (err < 0) - return err; } } } diff --git a/sound/drivers/opl3/opl3_midi.c b/sound/drivers/opl3/opl3_midi.c index 13c0a7e1bc2b..5f934b2f1486 100644 --- a/sound/drivers/opl3/opl3_midi.c +++ b/sound/drivers/opl3/opl3_midi.c @@ -415,7 +415,7 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan) } if (instr_4op) { vp2 = &opl3->voices[voice + 3]; - if (vp->state > 0) { + if (vp2->state > 0) { opl3_reg = reg_side | (OPL3_REG_KEYON_BLOCK + voice_offset + 3); reg_val = vp->keyon_reg & ~OPL3_KEYON_BIT; diff --git a/sound/drivers/opl3/opl3_synth.c b/sound/drivers/opl3/opl3_synth.c index 42920a243328..3f94746d587a 100644 --- a/sound/drivers/opl3/opl3_synth.c +++ b/sound/drivers/opl3/opl3_synth.c @@ -104,6 +104,8 @@ int snd_opl3_ioctl(struct snd_hwdep * hw, struct file *file, { struct snd_dm_fm_info info; + memset(&info, 0, sizeof(info)); + info.fm_mode = opl3->fm_mode; info.rhythm = opl3->rhythm; if (copy_to_user(argp, &info, sizeof(struct snd_dm_fm_info))) diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig index 0cb65d0864cc..f7b26b1d7084 100644 --- a/sound/firewire/Kconfig +++ b/sound/firewire/Kconfig @@ -37,7 +37,7 @@ config SND_OXFW * Mackie(Loud) Onyx 1640i (former model) * Mackie(Loud) Onyx Satellite * Mackie(Loud) Tapco Link.Firewire - * Mackie(Loud) d.2 pro/d.4 pro + * Mackie(Loud) d.2 pro/d.4 pro (built-in FireWire card with OXFW971 ASIC) * Mackie(Loud) U.420/U.420d * TASCAM FireOne * Stanton Controllers & Systems 1 Deck/Mixer @@ -83,7 +83,7 @@ config SND_BEBOB * PreSonus FIREBOX/FIREPOD/FP10/Inspire1394 * BridgeCo RDAudio1/Audio5 * Mackie Onyx 1220/1620/1640 (FireWire I/O Card) - * Mackie d.2 (FireWire Option) + * Mackie d.2 (optional FireWire card with DM1000 ASIC) * Stanton FinalScratch 2 (ScratchAmp) * Tascam IF-FW/DM * Behringer XENIX UFX 1204/1604 @@ -109,6 +109,7 @@ config SND_BEBOB * M-Audio Ozonic/NRV10/ProfireLightBridge * M-Audio FireWire 1814/ProjectMix IO * Digidesign Mbox 2 Pro + * ToneWeal FW66 To compile this driver as a module, choose M here: the module will be called snd-bebob. diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c index 5636e89ce5c7..eac3ff24e55d 100644 --- a/sound/firewire/bebob/bebob.c +++ b/sound/firewire/bebob/bebob.c @@ -60,6 +60,7 @@ static DECLARE_BITMAP(devices_used, SNDRV_CARDS); #define VEN_MAUDIO1 0x00000d6c #define VEN_MAUDIO2 0x000007f5 #define VEN_DIGIDESIGN 0x00a07e +#define OUI_SHOUYO 0x002327 #define MODEL_FOCUSRITE_SAFFIRE_BOTH 0x00000000 #define MODEL_MAUDIO_AUDIOPHILE_BOTH 0x00010060 @@ -414,7 +415,7 @@ static const struct ieee1394_device_id bebob_id_table[] = { SND_BEBOB_DEV_ENTRY(VEN_BRIDGECO, 0x00010049, &spec_normal), /* Mackie, Onyx 1220/1620/1640 (Firewire I/O Card) */ SND_BEBOB_DEV_ENTRY(VEN_MACKIE2, 0x00010065, &spec_normal), - /* Mackie, d.2 (Firewire Option) */ + // Mackie, d.2 (optional Firewire card with DM1000). SND_BEBOB_DEV_ENTRY(VEN_MACKIE1, 0x00010067, &spec_normal), /* Stanton, ScratchAmp */ SND_BEBOB_DEV_ENTRY(VEN_STANTON, 0x00000001, &spec_normal), @@ -513,6 +514,8 @@ static const struct ieee1394_device_id bebob_id_table[] = { &maudio_special_spec), /* Digidesign Mbox 2 Pro */ SND_BEBOB_DEV_ENTRY(VEN_DIGIDESIGN, 0x0000a9, &spec_normal), + // Toneweal FW66. + SND_BEBOB_DEV_ENTRY(OUI_SHOUYO, 0x020002, &spec_normal), /* IDs are unknown but able to be supported */ /* Apogee, Mini-ME Firewire */ /* Apogee, Mini-DAC Firewire */ diff --git a/sound/firewire/bebob/bebob_hwdep.c b/sound/firewire/bebob/bebob_hwdep.c index 2b367c21b80c..9bea8d6d5e06 100644 --- a/sound/firewire/bebob/bebob_hwdep.c +++ b/sound/firewire/bebob/bebob_hwdep.c @@ -37,12 +37,11 @@ hwdep_read(struct snd_hwdep *hwdep, char __user *buf, long count, } memset(&event, 0, sizeof(event)); + count = min_t(long, count, sizeof(event.lock_status)); if (bebob->dev_lock_changed) { event.lock_status.type = SNDRV_FIREWIRE_EVENT_LOCK_STATUS; event.lock_status.status = (bebob->dev_lock_count > 0); bebob->dev_lock_changed = false; - - count = min_t(long, count, sizeof(event.lock_status)); } spin_unlock_irq(&bebob->lock); diff --git a/sound/firewire/digi00x/digi00x.c b/sound/firewire/digi00x/digi00x.c index ef689997d6a5..bf53e342788e 100644 --- a/sound/firewire/digi00x/digi00x.c +++ b/sound/firewire/digi00x/digi00x.c @@ -15,6 +15,7 @@ MODULE_LICENSE("GPL v2"); #define VENDOR_DIGIDESIGN 0x00a07e #define MODEL_CONSOLE 0x000001 #define MODEL_RACK 0x000002 +#define SPEC_VERSION 0x000001 static int name_card(struct snd_dg00x *dg00x) { @@ -185,14 +186,18 @@ static const struct ieee1394_device_id snd_dg00x_id_table[] = { /* Both of 002/003 use the same ID. */ { .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_VERSION | IEEE1394_MATCH_MODEL_ID, .vendor_id = VENDOR_DIGIDESIGN, + .version = SPEC_VERSION, .model_id = MODEL_CONSOLE, }, { .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_VERSION | IEEE1394_MATCH_MODEL_ID, .vendor_id = VENDOR_DIGIDESIGN, + .version = SPEC_VERSION, .model_id = MODEL_RACK, }, {} diff --git a/sound/firewire/fireface/ff-transaction.c b/sound/firewire/fireface/ff-transaction.c index dd6c8e839647..dd6dac25e8a3 100644 --- a/sound/firewire/fireface/ff-transaction.c +++ b/sound/firewire/fireface/ff-transaction.c @@ -99,7 +99,7 @@ static void transmit_midi_msg(struct snd_ff *ff, unsigned int port) /* Set interval to next transaction. */ ff->next_ktime[port] = ktime_add_ns(ktime_get(), - len * 8 * NSEC_PER_SEC / 31250); + len * 8 * (NSEC_PER_SEC / 31250)); ff->rx_bytes[port] = len; /* diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index 6f941720141a..74d588bea6a4 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -406,8 +406,7 @@ static const struct ieee1394_device_id oxfw_id_table[] = { * Onyx-i series (former models): 0x081216 * Mackie Onyx Satellite: 0x00200f * Tapco LINK.firewire 4x6: 0x000460 - * d.2 pro: Unknown - * d.4 pro: Unknown + * d.2 pro/d.4 pro (built-in card): Unknown * U.420: Unknown * U.420d: Unknown */ diff --git a/sound/firewire/tascam/tascam-transaction.c b/sound/firewire/tascam/tascam-transaction.c index 8967c52f5032..8653cb4fb982 100644 --- a/sound/firewire/tascam/tascam-transaction.c +++ b/sound/firewire/tascam/tascam-transaction.c @@ -210,7 +210,7 @@ static void midi_port_work(struct work_struct *work) /* Set interval to next transaction. */ port->next_ktime = ktime_add_ns(ktime_get(), - port->consume_bytes * 8 * NSEC_PER_SEC / 31250); + port->consume_bytes * 8 * (NSEC_PER_SEC / 31250)); /* Start this transaction. */ port->idling = false; diff --git a/sound/firewire/tascam/tascam.c b/sound/firewire/tascam/tascam.c index d3fdc463a884..1e61cdce2895 100644 --- a/sound/firewire/tascam/tascam.c +++ b/sound/firewire/tascam/tascam.c @@ -225,11 +225,39 @@ static void snd_tscm_remove(struct fw_unit *unit) } static const struct ieee1394_device_id snd_tscm_id_table[] = { + // Tascam, FW-1884. { .match_flags = IEEE1394_MATCH_VENDOR_ID | - IEEE1394_MATCH_SPECIFIER_ID, + IEEE1394_MATCH_SPECIFIER_ID | + IEEE1394_MATCH_VERSION, .vendor_id = 0x00022e, .specifier_id = 0x00022e, + .version = 0x800000, + }, + // Tascam, FE-8 (.version = 0x800001) + // This kernel module doesn't support FE-8 because the most of features + // can be implemented in userspace without any specific support of this + // module. + // + // .version = 0x800002 is unknown. + // + // Tascam, FW-1082. + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_SPECIFIER_ID | + IEEE1394_MATCH_VERSION, + .vendor_id = 0x00022e, + .specifier_id = 0x00022e, + .version = 0x800003, + }, + // Tascam, FW-1804. + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_SPECIFIER_ID | + IEEE1394_MATCH_VERSION, + .vendor_id = 0x00022e, + .specifier_id = 0x00022e, + .version = 0x800004, }, /* FE-08 requires reverse-engineering because it just has faders. */ {} diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c index 84f3b8168716..b679d5f37e4d 100644 --- a/sound/hda/ext/hdac_ext_controller.c +++ b/sound/hda/ext/hdac_ext_controller.c @@ -155,6 +155,8 @@ struct hdac_ext_link *snd_hdac_ext_bus_get_link(struct hdac_ext_bus *ebus, return NULL; if (ebus->idx != bus_idx) return NULL; + if (addr < 0 || addr > 31) + return NULL; list_for_each_entry(hlink, &ebus->hlink_list, list) { for (i = 0; i < HDA_MAX_CODECS; i++) { diff --git a/sound/hda/hdac_bus.c b/sound/hda/hdac_bus.c index 714a51721a31..ab9236e4c157 100644 --- a/sound/hda/hdac_bus.c +++ b/sound/hda/hdac_bus.c @@ -155,6 +155,7 @@ static void process_unsol_events(struct work_struct *work) struct hdac_driver *drv; unsigned int rp, caddr, res; + spin_lock_irq(&bus->reg_lock); while (bus->unsol_rp != bus->unsol_wp) { rp = (bus->unsol_rp + 1) % HDA_UNSOL_QUEUE_SIZE; bus->unsol_rp = rp; @@ -166,10 +167,13 @@ static void process_unsol_events(struct work_struct *work) codec = bus->caddr_tbl[caddr & 0x0f]; if (!codec || !codec->dev.driver) continue; + spin_unlock_irq(&bus->reg_lock); drv = drv_to_hdac_driver(codec->dev.driver); if (drv->unsol_event) drv->unsol_event(codec, res); + spin_lock_irq(&bus->reg_lock); } + spin_unlock_irq(&bus->reg_lock); } /** diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index 778b42ba90b8..5ae72134159a 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -389,8 +389,9 @@ int snd_hdac_bus_reset_link(struct hdac_bus *bus, bool full_reset) if (!full_reset) goto skip_reset; - /* clear STATESTS */ - snd_hdac_chip_writew(bus, STATESTS, STATESTS_INT_MASK); + /* clear STATESTS if not in reset */ + if (snd_hdac_chip_readb(bus, GCTL) & AZX_GCTL_RESET) + snd_hdac_chip_writew(bus, STATESTS, STATESTS_INT_MASK); /* reset controller */ snd_hdac_bus_enter_link_reset(bus); diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 19deb306facb..4a843eb7cc94 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -123,6 +123,8 @@ EXPORT_SYMBOL_GPL(snd_hdac_device_init); void snd_hdac_device_exit(struct hdac_device *codec) { pm_runtime_put_noidle(&codec->dev); + /* keep balance of runtime PM child_count in parent device */ + pm_runtime_set_suspended(&codec->dev); snd_hdac_bus_remove_device(codec->bus, codec); kfree(codec->vendor_name); kfree(codec->chip_name); diff --git a/sound/isa/cmi8330.c b/sound/isa/cmi8330.c index 6b8c46942efb..75b3d76eb852 100644 --- a/sound/isa/cmi8330.c +++ b/sound/isa/cmi8330.c @@ -564,7 +564,7 @@ static int snd_cmi8330_probe(struct snd_card *card, int dev) } if (acard->sb->hardware != SB_HW_16) { snd_printk(KERN_ERR PFX "SB16 not found during probe\n"); - return err; + return -ENODEV; } snd_wss_out(acard->wss, CS4231_MISC_INFO, 0x40); /* switch on MODE2 */ diff --git a/sound/isa/es1688/es1688.c b/sound/isa/es1688/es1688.c index a826c138e7f5..8a58ed168756 100644 --- a/sound/isa/es1688/es1688.c +++ b/sound/isa/es1688/es1688.c @@ -284,8 +284,10 @@ static int snd_es968_pnp_detect(struct pnp_card_link *pcard, return error; } error = snd_es1688_probe(card, dev); - if (error < 0) + if (error < 0) { + snd_card_free(card); return error; + } pnp_set_card_drvdata(pcard, card); snd_es968_pnp_is_probed = 1; return 0; diff --git a/sound/isa/gus/gus_dma.c b/sound/isa/gus/gus_dma.c index 36c27c832360..2e27cd3427c8 100644 --- a/sound/isa/gus/gus_dma.c +++ b/sound/isa/gus/gus_dma.c @@ -141,6 +141,8 @@ static void snd_gf1_dma_interrupt(struct snd_gus_card * gus) } block = snd_gf1_dma_next_block(gus); spin_unlock(&gus->dma_lock); + if (!block) + return; snd_gf1_dma_program(gus, block->addr, block->buf_addr, block->count, (unsigned short) block->cmd); kfree(block); #if 0 diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index 8894c7c18ad6..d92c3c6b6051 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -875,10 +875,13 @@ static void snd_miro_write(struct snd_miro *chip, unsigned char reg, spin_unlock_irqrestore(&chip->lock, flags); } +static inline void snd_miro_write_mask(struct snd_miro *chip, + unsigned char reg, unsigned char value, unsigned char mask) +{ + unsigned char oldval = snd_miro_read(chip, reg); -#define snd_miro_write_mask(chip, reg, value, mask) \ - snd_miro_write(chip, reg, \ - (snd_miro_read(chip, reg) & ~(mask)) | ((value) & (mask))) + snd_miro_write(chip, reg, (oldval & ~mask) | (value & mask)); +} /* * Proc Interface diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index 505cd81e19fa..4ef3caaf4354 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -327,10 +327,13 @@ static void snd_opti9xx_write(struct snd_opti9xx *chip, unsigned char reg, } -#define snd_opti9xx_write_mask(chip, reg, value, mask) \ - snd_opti9xx_write(chip, reg, \ - (snd_opti9xx_read(chip, reg) & ~(mask)) | ((value) & (mask))) +static inline void snd_opti9xx_write_mask(struct snd_opti9xx *chip, + unsigned char reg, unsigned char value, unsigned char mask) +{ + unsigned char oldval = snd_opti9xx_read(chip, reg); + snd_opti9xx_write(chip, reg, (oldval & ~mask) | (value & mask)); +} static int snd_opti9xx_configure(struct snd_opti9xx *chip, long port, diff --git a/sound/isa/sb/emu8000.c b/sound/isa/sb/emu8000.c index d56973b770c7..24b91cb32839 100644 --- a/sound/isa/sb/emu8000.c +++ b/sound/isa/sb/emu8000.c @@ -1042,8 +1042,10 @@ snd_emu8000_create_mixer(struct snd_card *card, struct snd_emu8000 *emu) memset(emu->controls, 0, sizeof(emu->controls)); for (i = 0; i < EMU8000_NUM_CONTROLS; i++) { - if ((err = snd_ctl_add(card, emu->controls[i] = snd_ctl_new1(mixer_defs[i], emu))) < 0) + if ((err = snd_ctl_add(card, emu->controls[i] = snd_ctl_new1(mixer_defs[i], emu))) < 0) { + emu->controls[i] = NULL; goto __error; + } } return 0; diff --git a/sound/isa/sb/sb16_csp.c b/sound/isa/sb/sb16_csp.c index fa5780bb0c68..00d059412c8a 100644 --- a/sound/isa/sb/sb16_csp.c +++ b/sound/isa/sb/sb16_csp.c @@ -828,6 +828,7 @@ static int snd_sb_csp_start(struct snd_sb_csp * p, int sample_width, int channel mixR = snd_sbmixer_read(p->chip, SB_DSP4_PCM_DEV + 1); snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV, mixL & 0x7); snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV + 1, mixR & 0x7); + spin_unlock_irqrestore(&p->chip->mixer_lock, flags); spin_lock(&p->chip->reg_lock); set_mode_register(p->chip, 0xc0); /* c0 = STOP */ @@ -867,6 +868,7 @@ static int snd_sb_csp_start(struct snd_sb_csp * p, int sample_width, int channel spin_unlock(&p->chip->reg_lock); /* restore PCM volume */ + spin_lock_irqsave(&p->chip->mixer_lock, flags); snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV, mixL); snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV + 1, mixR); spin_unlock_irqrestore(&p->chip->mixer_lock, flags); @@ -892,6 +894,7 @@ static int snd_sb_csp_stop(struct snd_sb_csp * p) mixR = snd_sbmixer_read(p->chip, SB_DSP4_PCM_DEV + 1); snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV, mixL & 0x7); snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV + 1, mixR & 0x7); + spin_unlock_irqrestore(&p->chip->mixer_lock, flags); spin_lock(&p->chip->reg_lock); if (p->running & SNDRV_SB_CSP_ST_QSOUND) { @@ -906,6 +909,7 @@ static int snd_sb_csp_stop(struct snd_sb_csp * p) spin_unlock(&p->chip->reg_lock); /* restore PCM volume */ + spin_lock_irqsave(&p->chip->mixer_lock, flags); snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV, mixL); snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV + 1, mixR); spin_unlock_irqrestore(&p->chip->mixer_lock, flags); @@ -1059,10 +1063,14 @@ static int snd_sb_qsound_build(struct snd_sb_csp * p) spin_lock_init(&p->q_lock); - if ((err = snd_ctl_add(card, p->qsound_switch = snd_ctl_new1(&snd_sb_qsound_switch, p))) < 0) + if ((err = snd_ctl_add(card, p->qsound_switch = snd_ctl_new1(&snd_sb_qsound_switch, p))) < 0) { + p->qsound_switch = NULL; goto __error; - if ((err = snd_ctl_add(card, p->qsound_space = snd_ctl_new1(&snd_sb_qsound_space, p))) < 0) + } + if ((err = snd_ctl_add(card, p->qsound_space = snd_ctl_new1(&snd_sb_qsound_space, p))) < 0) { + p->qsound_space = NULL; goto __error; + } return 0; @@ -1082,10 +1090,14 @@ static void snd_sb_qsound_destroy(struct snd_sb_csp * p) card = p->chip->card; down_write(&card->controls_rwsem); - if (p->qsound_switch) + if (p->qsound_switch) { snd_ctl_remove(card, p->qsound_switch); - if (p->qsound_space) + p->qsound_switch = NULL; + } + if (p->qsound_space) { snd_ctl_remove(card, p->qsound_space); + p->qsound_space = NULL; + } up_write(&card->controls_rwsem); /* cancel pending transfer of QSound parameters */ diff --git a/sound/isa/sb/sb8.c b/sound/isa/sb/sb8.c index 1eb8b61a185b..d77dcba276b5 100644 --- a/sound/isa/sb/sb8.c +++ b/sound/isa/sb/sb8.c @@ -111,10 +111,6 @@ static int snd_sb8_probe(struct device *pdev, unsigned int dev) /* block the 0x388 port to avoid PnP conflicts */ acard->fm_res = request_region(0x388, 4, "SoundBlaster FM"); - if (!acard->fm_res) { - err = -EBUSY; - goto _err; - } if (port[dev] != SNDRV_AUTO_PORT) { if ((err = snd_sbdsp_create(card, port[dev], irq[dev], diff --git a/sound/isa/wavefront/wavefront_synth.c b/sound/isa/wavefront/wavefront_synth.c index 0b1e4b34b299..13c8e6542a2f 100644 --- a/sound/isa/wavefront/wavefront_synth.c +++ b/sound/isa/wavefront/wavefront_synth.c @@ -1175,7 +1175,10 @@ wavefront_send_alias (snd_wavefront_t *dev, wavefront_patch_info *header) "alias for %d\n", header->number, header->hdr.a.OriginalSample); - + + if (header->number >= WF_MAX_SAMPLE) + return -EINVAL; + munge_int32 (header->number, &alias_hdr[0], 2); munge_int32 (header->hdr.a.OriginalSample, &alias_hdr[2], 2); munge_int32 (*((unsigned int *)&header->hdr.a.sampleStartOffset), @@ -1206,6 +1209,9 @@ wavefront_send_multisample (snd_wavefront_t *dev, wavefront_patch_info *header) int num_samples; unsigned char *msample_hdr; + if (header->number >= WF_MAX_SAMPLE) + return -EINVAL; + msample_hdr = kmalloc(WF_MSAMPLE_BYTES, GFP_KERNEL); if (! msample_hdr) return -ENOMEM; diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c index b1a2a7ea4172..b4ccd9f92400 100644 --- a/sound/pci/asihpi/hpioctl.c +++ b/sound/pci/asihpi/hpioctl.c @@ -350,7 +350,7 @@ int asihpi_adapter_probe(struct pci_dev *pci_dev, struct hpi_message hm; struct hpi_response hr; struct hpi_adapter adapter; - struct hpi_pci pci; + struct hpi_pci pci = { 0 }; memset(&adapter, 0, sizeof(adapter)); @@ -506,7 +506,7 @@ int asihpi_adapter_probe(struct pci_dev *pci_dev, return 0; err: - for (idx = 0; idx < HPI_MAX_ADAPTER_MEM_SPACES; idx++) { + while (--idx >= 0) { if (pci.ap_mem_base[idx]) { iounmap(pci.ap_mem_base[idx]); pci.ap_mem_base[idx] = NULL; diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index cd27b5536654..675b812e96d6 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -551,7 +551,8 @@ static int snd_ca0106_pcm_power_dac(struct snd_ca0106 *chip, int channel_id, else /* Power down */ chip->spi_dac_reg[reg] |= bit; - return snd_ca0106_spi_write(chip, chip->spi_dac_reg[reg]); + if (snd_ca0106_spi_write(chip, chip->spi_dac_reg[reg]) != 0) + return -ENXIO; } return 0; } diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index 0020fd0efc46..09c547f4cc18 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -780,7 +780,7 @@ static void snd_cs46xx_set_capture_sample_rate(struct snd_cs46xx *chip, unsigned rate = 48000 / 9; /* - * We can not capture at at rate greater than the Input Rate (48000). + * We can not capture at a rate greater than the Input Rate (48000). * Return an error if an attempt is made to stray outside that limit. */ if (rate > 48000) diff --git a/sound/pci/cs46xx/dsp_spos_scb_lib.c b/sound/pci/cs46xx/dsp_spos_scb_lib.c index 7488e1b7a770..4e726d39b05d 100644 --- a/sound/pci/cs46xx/dsp_spos_scb_lib.c +++ b/sound/pci/cs46xx/dsp_spos_scb_lib.c @@ -1742,7 +1742,7 @@ int cs46xx_iec958_pre_open (struct snd_cs46xx *chip) struct dsp_spos_instance * ins = chip->dsp_spos_instance; if ( ins->spdif_status_out & DSP_SPDIF_STATUS_OUTPUT_ENABLED ) { - /* remove AsynchFGTxSCB and and PCMSerialInput_II */ + /* remove AsynchFGTxSCB and PCMSerialInput_II */ cs46xx_dsp_disable_spdif_out (chip); /* save state */ diff --git a/sound/pci/ctxfi/ctamixer.c b/sound/pci/ctxfi/ctamixer.c index 5fcbb065d870..d32685ce6c05 100644 --- a/sound/pci/ctxfi/ctamixer.c +++ b/sound/pci/ctxfi/ctamixer.c @@ -27,16 +27,15 @@ #define BLANK_SLOT 4094 -static int amixer_master(struct rsc *rsc) +static void amixer_master(struct rsc *rsc) { rsc->conj = 0; - return rsc->idx = container_of(rsc, struct amixer, rsc)->idx[0]; + rsc->idx = container_of(rsc, struct amixer, rsc)->idx[0]; } -static int amixer_next_conj(struct rsc *rsc) +static void amixer_next_conj(struct rsc *rsc) { rsc->conj++; - return container_of(rsc, struct amixer, rsc)->idx[rsc->conj]; } static int amixer_index(const struct rsc *rsc) @@ -335,16 +334,15 @@ int amixer_mgr_destroy(struct amixer_mgr *amixer_mgr) /* SUM resource management */ -static int sum_master(struct rsc *rsc) +static void sum_master(struct rsc *rsc) { rsc->conj = 0; - return rsc->idx = container_of(rsc, struct sum, rsc)->idx[0]; + rsc->idx = container_of(rsc, struct sum, rsc)->idx[0]; } -static int sum_next_conj(struct rsc *rsc) +static void sum_next_conj(struct rsc *rsc) { rsc->conj++; - return container_of(rsc, struct sum, rsc)->idx[rsc->conj]; } static int sum_index(const struct rsc *rsc) diff --git a/sound/pci/ctxfi/ctdaio.c b/sound/pci/ctxfi/ctdaio.c index 7f089cb433e1..df326b7663a2 100644 --- a/sound/pci/ctxfi/ctdaio.c +++ b/sound/pci/ctxfi/ctdaio.c @@ -55,12 +55,12 @@ static struct daio_rsc_idx idx_20k2[NUM_DAIOTYP] = { [SPDIFIO] = {.left = 0x05, .right = 0x85}, }; -static int daio_master(struct rsc *rsc) +static void daio_master(struct rsc *rsc) { /* Actually, this is not the resource index of DAIO. * For DAO, it is the input mapper index. And, for DAI, * it is the output time-slot index. */ - return rsc->conj = rsc->idx; + rsc->conj = rsc->idx; } static int daio_index(const struct rsc *rsc) @@ -68,19 +68,19 @@ static int daio_index(const struct rsc *rsc) return rsc->conj; } -static int daio_out_next_conj(struct rsc *rsc) +static void daio_out_next_conj(struct rsc *rsc) { - return rsc->conj += 2; + rsc->conj += 2; } -static int daio_in_next_conj_20k1(struct rsc *rsc) +static void daio_in_next_conj_20k1(struct rsc *rsc) { - return rsc->conj += 0x200; + rsc->conj += 0x200; } -static int daio_in_next_conj_20k2(struct rsc *rsc) +static void daio_in_next_conj_20k2(struct rsc *rsc) { - return rsc->conj += 0x100; + rsc->conj += 0x100; } static const struct rsc_ops daio_out_rsc_ops = { diff --git a/sound/pci/ctxfi/cthw20k2.c b/sound/pci/ctxfi/cthw20k2.c index b866d6b2c923..e603db4d5ef3 100644 --- a/sound/pci/ctxfi/cthw20k2.c +++ b/sound/pci/ctxfi/cthw20k2.c @@ -995,7 +995,7 @@ static int daio_mgr_dao_init(void *blk, unsigned int idx, unsigned int conf) if (idx < 4) { /* S/PDIF output */ - switch ((conf & 0x7)) { + switch ((conf & 0xf)) { case 1: set_field(&ctl->txctl[idx], ATXCTL_NUC, 0); break; diff --git a/sound/pci/ctxfi/ctresource.c b/sound/pci/ctxfi/ctresource.c index 80c4d84f9667..f05a09ed42b8 100644 --- a/sound/pci/ctxfi/ctresource.c +++ b/sound/pci/ctxfi/ctresource.c @@ -113,18 +113,17 @@ static int audio_ring_slot(const struct rsc *rsc) return (rsc->conj << 4) + offset_in_audio_slot_block[rsc->type]; } -static int rsc_next_conj(struct rsc *rsc) +static void rsc_next_conj(struct rsc *rsc) { unsigned int i; for (i = 0; (i < 8) && (!(rsc->msr & (0x1 << i))); ) i++; rsc->conj += (AUDIO_SLOT_BLOCK_NUM >> i); - return rsc->conj; } -static int rsc_master(struct rsc *rsc) +static void rsc_master(struct rsc *rsc) { - return rsc->conj = rsc->idx; + rsc->conj = rsc->idx; } static const struct rsc_ops rsc_generic_ops = { diff --git a/sound/pci/ctxfi/ctresource.h b/sound/pci/ctxfi/ctresource.h index 736d9f7e9e16..29b6fe6de659 100644 --- a/sound/pci/ctxfi/ctresource.h +++ b/sound/pci/ctxfi/ctresource.h @@ -43,8 +43,8 @@ struct rsc { }; struct rsc_ops { - int (*master)(struct rsc *rsc); /* Move to master resource */ - int (*next_conj)(struct rsc *rsc); /* Move to next conjugate resource */ + void (*master)(struct rsc *rsc); /* Move to master resource */ + void (*next_conj)(struct rsc *rsc); /* Move to next conjugate resource */ int (*index)(const struct rsc *rsc); /* Return the index of resource */ /* Return the output slot number */ int (*output_slot)(const struct rsc *rsc); diff --git a/sound/pci/ctxfi/ctsrc.c b/sound/pci/ctxfi/ctsrc.c index bb4c9c3c89ae..93d660276c82 100644 --- a/sound/pci/ctxfi/ctsrc.c +++ b/sound/pci/ctxfi/ctsrc.c @@ -594,16 +594,15 @@ int src_mgr_destroy(struct src_mgr *src_mgr) /* SRCIMP resource manager operations */ -static int srcimp_master(struct rsc *rsc) +static void srcimp_master(struct rsc *rsc) { rsc->conj = 0; - return rsc->idx = container_of(rsc, struct srcimp, rsc)->idx[0]; + rsc->idx = container_of(rsc, struct srcimp, rsc)->idx[0]; } -static int srcimp_next_conj(struct rsc *rsc) +static void srcimp_next_conj(struct rsc *rsc) { rsc->conj++; - return container_of(rsc, struct srcimp, rsc)->idx[rsc->conj]; } static int srcimp_index(const struct rsc *rsc) diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index e1f0bcd45c37..b58a098a7270 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -2215,7 +2215,6 @@ static int snd_echo_resume(struct device *dev) if (err < 0) { kfree(commpage_bak); dev_err(dev, "resume init_hw err=%d\n", err); - snd_echo_free(chip); return err; } @@ -2242,7 +2241,6 @@ static int snd_echo_resume(struct device *dev) if (request_irq(pci->irq, snd_echo_interrupt, IRQF_SHARED, KBUILD_MODNAME, chip)) { dev_err(chip->card->dev, "cannot grab irq\n"); - snd_echo_free(chip); return -EBUSY; } chip->irq = pci->irq; diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 8b1cf237b96e..c5dc8587d2ac 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -76,6 +76,12 @@ static int compare_input_type(const void *ap, const void *bp) if (a->type != b->type) return (int)(a->type - b->type); + /* If has both hs_mic and hp_mic, pick the hs_mic ahead of hp_mic. */ + if (a->is_headset_mic && b->is_headphone_mic) + return -1; /* don't swap */ + else if (a->is_headphone_mic && b->is_headset_mic) + return 1; /* swap */ + /* In case one has boost and the other one has not, pick the one with boost first. */ return (int)(b->has_boost_on_pin - a->has_boost_on_pin); diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index c397e7da0eac..7ccfb09535e1 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -310,8 +310,12 @@ int snd_hda_mixer_amp_switch_get_beep(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct hda_beep *beep = codec->beep; + int chs = get_amp_channels(kcontrol); + if (beep && (!beep->enabled || !ctl_has_mute(kcontrol))) { - ucontrol->value.integer.value[0] = + if (chs & 1) + ucontrol->value.integer.value[0] = beep->enabled; + if (chs & 2) ucontrol->value.integer.value[1] = beep->enabled; return 0; } diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index c175b2cf63f7..66010d0774b4 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -46,6 +46,10 @@ static void hda_codec_unsol_event(struct hdac_device *dev, unsigned int ev) if (codec->bus->shutdown) return; + /* ignore unsol events during system suspend/resume */ + if (codec->core.dev.power.power_state.event != PM_EVENT_ON) + return; + if (codec->patch_ops.unsol_event) codec->patch_ops.unsol_event(codec, ev); } diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index a6f7561e7bb9..a56f018d586f 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -942,6 +942,7 @@ int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, /* power-up all before initialization */ hda_set_power_state(codec, AC_PWRST_D0); + codec->core.dev.power.power_state = PMSG_ON; snd_hda_codec_proc_new(codec); @@ -3393,7 +3394,7 @@ EXPORT_SYMBOL_GPL(snd_hda_set_power_save); * @nid: NID to check / update * * Check whether the given NID is in the amp list. If it's in the list, - * check the current AMP status, and update the the power-status according + * check the current AMP status, and update the power-status according * to the mute status. * * This function is supposed to be set or called from the check_power_status diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index fa261b27d858..0c5d41e5d146 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -624,13 +624,6 @@ 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 @@ -1169,16 +1162,23 @@ irqreturn_t azx_interrupt(int irq, void *dev_id) if (snd_hdac_bus_handle_stream_irq(bus, status, stream_update)) active = true; - /* clear rirb int */ status = azx_readb(chip, RIRBSTS); if (status & RIRB_INT_MASK) { + /* + * Clearing the interrupt status here ensures that no + * interrupt gets masked after the RIRB wp is read in + * snd_hdac_bus_update_rirb. This avoids a possible + * race condition where codec response in RIRB may + * remain unserviced by IRQ, eventually falling back + * to polling mode in azx_rirb_get_response. + */ + azx_writeb(chip, RIRBSTS, RIRB_INT_MASK); active = true; if (status & RIRB_INT_RESPONSE) { if (chip->driver_caps & AZX_DCAPS_CTX_WORKAROUND) udelay(80); snd_hdac_bus_update_rirb(bus); } - azx_writeb(chip, RIRBSTS, RIRB_INT_MASK); } } while (active && ++repeat < 10); diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 28ef409a9e6a..cf406f22f406 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -823,7 +823,7 @@ static void activate_amp_in(struct hda_codec *codec, struct nid_path *path, } } -/* sync power of each widget in the the given path */ +/* sync power of each widget in the given path */ static hda_nid_t path_power_update(struct hda_codec *codec, struct nid_path *path, bool allow_powerdown) @@ -1212,11 +1212,17 @@ static const char *get_line_out_pfx(struct hda_codec *codec, int ch, *index = ch; return "Headphone"; case AUTO_PIN_LINE_OUT: - /* This deals with the case where we have two DACs and - * one LO, one HP and one Speaker */ - if (!ch && cfg->speaker_outs && cfg->hp_outs) { - bool hp_lo_shared = !path_has_mixer(codec, spec->hp_paths[0], ctl_type); - bool spk_lo_shared = !path_has_mixer(codec, spec->speaker_paths[0], ctl_type); + /* This deals with the case where one HP or one Speaker or + * one HP + one Speaker need to share the DAC with LO + */ + if (!ch) { + bool hp_lo_shared = false, spk_lo_shared = false; + + if (cfg->speaker_outs) + spk_lo_shared = !path_has_mixer(codec, + spec->speaker_paths[0], ctl_type); + if (cfg->hp_outs) + hp_lo_shared = !path_has_mixer(codec, spec->hp_paths[0], ctl_type); if (hp_lo_shared && spk_lo_shared) return spec->vmaster_mute.hook ? "PCM" : "Master"; if (hp_lo_shared) @@ -1374,16 +1380,20 @@ static int try_assign_dacs(struct hda_codec *codec, int num_outs, struct nid_path *path; hda_nid_t pin = pins[i]; - path = snd_hda_get_path_from_idx(codec, path_idx[i]); - if (path) { - badness += assign_out_path_ctls(codec, path); - continue; + if (!spec->obey_preferred_dacs) { + path = snd_hda_get_path_from_idx(codec, path_idx[i]); + if (path) { + badness += assign_out_path_ctls(codec, path); + continue; + } } dacs[i] = get_preferred_dac(codec, pin); if (dacs[i]) { if (is_dac_already_used(codec, dacs[i])) badness += bad->shared_primary; + } else if (spec->obey_preferred_dacs) { + badness += BAD_NO_PRIMARY_DAC; } if (!dacs[i]) @@ -3458,7 +3468,7 @@ static int cap_put_caller(struct snd_kcontrol *kcontrol, struct hda_gen_spec *spec = codec->spec; const struct hda_input_mux *imux; struct nid_path *path; - int i, adc_idx, err = 0; + int i, adc_idx, ret, err = 0; imux = &spec->input_mux; adc_idx = kcontrol->id.index; @@ -3468,9 +3478,13 @@ static int cap_put_caller(struct snd_kcontrol *kcontrol, if (!path || !path->ctls[type]) continue; kcontrol->private_value = path->ctls[type]; - err = func(kcontrol, ucontrol); - if (err < 0) + ret = func(kcontrol, ucontrol); + if (ret < 0) { + err = ret; break; + } + if (ret > 0) + err = 1; } mutex_unlock(&codec->control_mutex); if (err >= 0 && spec->cap_sync_hook) diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index 17a6bff8e94e..b94e69c60e38 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -230,6 +230,7 @@ struct hda_gen_spec { unsigned int power_down_unused:1; /* power down unused widgets */ unsigned int dac_min_mute:1; /* minimal = mute for DACs */ unsigned int suppress_vmaster:1; /* don't create vmaster kctls */ + unsigned int obey_preferred_dacs:1; /* obey preferred_dacs assignment */ /* other internal flags */ unsigned int no_analog:1; /* digital I/O only */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 890793ad85ca..de090a3d2b38 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -742,13 +742,17 @@ static int azx_intel_link_power(struct azx *chip, bool enable) * the update-IRQ timing. The IRQ is issued before actually the * data is processed. So, we need to process it afterwords in a * workqueue. + * + * Returns 1 if OK to proceed, 0 for delay handling, -1 for skipping update */ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev) { struct snd_pcm_substream *substream = azx_dev->core.substream; + struct snd_pcm_runtime *runtime = substream->runtime; int stream = substream->stream; u32 wallclk; unsigned int pos; + snd_pcm_uframes_t hwptr, target; wallclk = azx_readl(chip, WALLCLK) - azx_dev->core.start_wallclk; if (wallclk < (azx_dev->core.period_wallclk * 2) / 3) @@ -785,6 +789,24 @@ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev) /* NG - it's below the first next period boundary */ return chip->bdl_pos_adj ? 0 : -1; azx_dev->core.start_wallclk += wallclk; + + if (azx_dev->core.no_period_wakeup) + return 1; /* OK, no need to check period boundary */ + + if (runtime->hw_ptr_base != runtime->hw_ptr_interrupt) + return 1; /* OK, already in hwptr updating process */ + + /* check whether the period gets really elapsed */ + pos = bytes_to_frames(runtime, pos); + hwptr = runtime->hw_ptr_base + pos; + if (hwptr < runtime->status->hw_ptr) + hwptr += runtime->buffer_size; + target = runtime->hw_ptr_interrupt + runtime->period_size; + if (hwptr < target) { + /* too early wakeup, process it later */ + return chip->bdl_pos_adj ? 0 : -1; + } + return 1; /* OK, it's fine */ } @@ -982,11 +1004,7 @@ static unsigned int azx_get_pos_skl(struct azx *chip, struct azx_dev *azx_dev) if (azx_dev->core.substream->stream == SNDRV_PCM_STREAM_PLAYBACK) return azx_skl_get_dpib_pos(chip, azx_dev); - /* For capture, we need to read posbuf, but it requires a delay - * for the possible boundary overlap; the read of DPIB fetches the - * actual posbuf - */ - udelay(20); + /* read of DPIB fetches the actual posbuf */ azx_skl_get_dpib_pos(chip, azx_dev); return azx_get_pos_posbuf(chip, azx_dev); } @@ -1282,6 +1300,7 @@ static void azx_vs_set_state(struct pci_dev *pci, struct snd_card *card = pci_get_drvdata(pci); struct azx *chip = card->private_data; struct hda_intel *hda = container_of(chip, struct hda_intel, chip); + struct hda_codec *codec; bool disabled; wait_for_completion(&hda->probe_wait); @@ -1306,8 +1325,12 @@ static void azx_vs_set_state(struct pci_dev *pci, dev_info(chip->card->dev, "%s via vga_switcheroo\n", disabled ? "Disabling" : "Enabling"); if (disabled) { - pm_runtime_put_sync_suspend(card->dev); - azx_suspend(card->dev); + list_for_each_codec(codec, &chip->bus) { + pm_runtime_suspend(hda_codec_dev(codec)); + pm_runtime_disable(hda_codec_dev(codec)); + } + pm_runtime_suspend(card->dev); + pm_runtime_disable(card->dev); /* when we get suspended by vga_switcheroo we end up in D3cold, * however we have no ACPI handle, so pci/acpi can't put us there, * put ourselves there */ @@ -1318,9 +1341,12 @@ static void azx_vs_set_state(struct pci_dev *pci, "Cannot lock devices!\n"); } else { snd_hda_unlock_devices(&chip->bus); - pm_runtime_get_noresume(card->dev); chip->disabled = false; - azx_resume(card->dev); + pm_runtime_enable(card->dev); + list_for_each_codec(codec, &chip->bus) { + pm_runtime_enable(hda_codec_dev(codec)); + pm_runtime_resume(hda_codec_dev(codec)); + } } } } @@ -1350,6 +1376,7 @@ static void init_vga_switcheroo(struct azx *chip) dev_info(chip->card->dev, "Handle vga_switcheroo audio client\n"); hda->use_vga_switcheroo = 1; + chip->driver_caps |= AZX_DCAPS_PM_RUNTIME; pci_dev_put(p); } } @@ -1375,9 +1402,6 @@ static int register_vga_switcheroo(struct azx *chip) return err; hda->vga_switcheroo_registered = 1; - /* register as an optimus hdmi audio power domain */ - vga_switcheroo_init_domain_pm_optimus_hdmi_audio(chip->card->dev, - &hda->hdmi_pm_domain); return 0; } #else @@ -1406,10 +1430,8 @@ static int azx_free(struct azx *chip) if (use_vga_switcheroo(hda)) { if (chip->disabled && hda->probe_continued) snd_hda_unlock_devices(&chip->bus); - if (hda->vga_switcheroo_registered) { + if (hda->vga_switcheroo_registered) vga_switcheroo_unregister_client(chip->pci); - vga_switcheroo_fini_domain_pm_ops(chip->card->dev); - } } if (bus->chip_init) { @@ -2034,24 +2056,15 @@ static void azx_firmware_cb(const struct firmware *fw, void *context) { struct snd_card *card = context; struct azx *chip = card->private_data; - struct pci_dev *pci = chip->pci; - - if (!fw) { - dev_err(card->dev, "Cannot load firmware, aborting\n"); - goto error; - } - chip->fw = fw; + if (fw) + chip->fw = fw; + else + dev_err(card->dev, "Cannot load firmware, continue without patching\n"); if (!chip->disabled) { /* continue probing */ - if (azx_probe_continue(chip)) - goto error; + azx_probe_continue(chip); } - return; /* OK */ - - error: - snd_card_free(card); - pci_set_drvdata(pci, NULL); } #endif @@ -2177,6 +2190,17 @@ static const struct hdac_io_ops pci_hda_io_ops = { .dma_free_pages = dma_free_pages, }; +/* Blacklist for skipping the whole probe: + * some HD-audio PCI entries are exposed without any codecs, and such devices + * should be ignored from the beginning. + */ +static const struct pci_device_id driver_blacklist[] = { + { PCI_DEVICE_SUB(0x1022, 0x1487, 0x1043, 0x874f) }, /* ASUS ROG Zenith II / Strix */ + { PCI_DEVICE_SUB(0x1022, 0x1487, 0x1462, 0xcb59) }, /* MSI TRX40 Creator */ + { PCI_DEVICE_SUB(0x1022, 0x1487, 0x1462, 0xcb60) }, /* MSI TRX40 */ + {} +}; + static const struct hda_controller_ops pci_hda_ops = { .disable_msi_reset_irq = disable_msi_reset_irq, .substream_alloc_pages = substream_alloc_pages, @@ -2196,6 +2220,11 @@ static int azx_probe(struct pci_dev *pci, bool schedule_probe; int err; + if (pci_match_id(driver_blacklist, pci)) { + dev_info(&pci->dev, "Skipping the blacklisted device\n"); + return -ENODEV; + } + if (dev >= SNDRV_CARDS) return -ENODEV; if (!enable[dev]) { @@ -2294,6 +2323,7 @@ static int azx_probe_continue(struct azx *chip) struct hda_intel *hda = container_of(chip, struct hda_intel, chip); struct hdac_bus *bus = azx_bus(chip); struct pci_dev *pci = chip->pci; + struct hda_codec *codec; int dev = chip->dev_index; int val; int err; @@ -2378,6 +2408,14 @@ static int azx_probe_continue(struct azx *chip) chip->running = 1; azx_add_card_list(chip); + /* + * The discrete GPU cannot power down unless the HDA controller runtime + * suspends, so activate runtime PM on codecs even if power_save == 0. + */ + if (use_vga_switcheroo(hda)) + list_for_each_codec(codec, &chip->bus) + codec->auto_runtime_pm = 1; + val = power_save; #ifdef CONFIG_PM if (pm_blacklist) { @@ -2392,7 +2430,7 @@ static int azx_probe_continue(struct azx *chip) } #endif /* CONFIG_PM */ snd_hda_set_power_save(&chip->bus, val * 1000); - if (azx_has_pm_runtime(chip) || hda->use_vga_switcheroo) + if (azx_has_pm_runtime(chip)) pm_runtime_put_autosuspend(&pci->dev); out_free: diff --git a/sound/pci/hda/hda_intel.h b/sound/pci/hda/hda_intel.h index ff0c4d617bc1..e3a3d318d2e5 100644 --- a/sound/pci/hda/hda_intel.h +++ b/sound/pci/hda/hda_intel.h @@ -40,9 +40,6 @@ struct hda_intel { unsigned int vga_switcheroo_registered:1; unsigned int init_failed:1; /* delayed init failed */ - /* secondary power domain for hdmi audio under vga device */ - struct dev_pm_domain hdmi_pm_domain; - bool need_i915_power:1; /* the hda controller needs i915 power */ }; diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index e85fb04ec7be..b567c4bdae00 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -363,6 +363,9 @@ static int hda_tegra_first_init(struct azx *chip, struct platform_device *pdev) unsigned short gcap; int irq_id = platform_get_irq(pdev, 0); + if (irq_id < 0) + return irq_id; + err = hda_tegra_init_chip(chip, pdev); if (err) return err; diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 92f5f452bee2..369f812d7072 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -4443,11 +4443,10 @@ 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(&spec->unsol_hp_work); - schedule_delayed_work(&spec->unsol_hp_work, msecs_to_jiffies(500)); tbl = snd_hda_jack_tbl_get(codec, cb->nid); if (tbl) tbl->block_report = 1; + schedule_delayed_work(&spec->unsol_hp_work, msecs_to_jiffies(500)); } static void amic_callback(struct hda_codec *codec, struct hda_jack_callback *cb) @@ -4625,12 +4624,25 @@ static void ca0132_free(struct hda_codec *codec) kfree(codec->spec); } +#ifdef CONFIG_PM +static int ca0132_suspend(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + + cancel_delayed_work_sync(&spec->unsol_hp_work); + return 0; +} +#endif + static const struct hda_codec_ops ca0132_patch_ops = { .build_controls = ca0132_build_controls, .build_pcms = ca0132_build_pcms, .init = ca0132_init, .free = ca0132_free, .unsol_event = snd_hda_jack_unsol_event, +#ifdef CONFIG_PM + .suspend = ca0132_suspend, +#endif }; static void ca0132_config(struct hda_codec *codec) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 9cc9304ff21a..d790c8604a9c 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -948,18 +948,18 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x103c, 0x8079, "HP EliteBook 840 G3", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x807C, "HP EliteBook 820 G3", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x80FD, "HP ProBook 640 G2", CXT_FIXUP_HP_DOCK), - SND_PCI_QUIRK(0x103c, 0x828c, "HP EliteBook 840 G4", CXT_FIXUP_HP_DOCK), - SND_PCI_QUIRK(0x103c, 0x83b2, "HP EliteBook 840 G5", CXT_FIXUP_HP_DOCK), - SND_PCI_QUIRK(0x103c, 0x83b3, "HP EliteBook 830 G5", CXT_FIXUP_HP_DOCK), - SND_PCI_QUIRK(0x103c, 0x83d3, "HP ProBook 640 G4", CXT_FIXUP_HP_DOCK), - SND_PCI_QUIRK(0x103c, 0x8174, "HP Spectre x360", CXT_FIXUP_HP_SPECTRE), SND_PCI_QUIRK(0x103c, 0x8115, "HP Z1 Gen3", CXT_FIXUP_HP_GATE_MIC), SND_PCI_QUIRK(0x103c, 0x814f, "HP ZBook 15u G3", CXT_FIXUP_MUTE_LED_GPIO), + SND_PCI_QUIRK(0x103c, 0x8174, "HP Spectre x360", CXT_FIXUP_HP_SPECTRE), SND_PCI_QUIRK(0x103c, 0x822e, "HP ProBook 440 G4", CXT_FIXUP_MUTE_LED_GPIO), - SND_PCI_QUIRK(0x103c, 0x836e, "HP ProBook 455 G5", CXT_FIXUP_MUTE_LED_GPIO), - SND_PCI_QUIRK(0x103c, 0x837f, "HP ProBook 470 G5", CXT_FIXUP_MUTE_LED_GPIO), + SND_PCI_QUIRK(0x103c, 0x828c, "HP EliteBook 840 G4", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x8299, "HP 800 G3 SFF", CXT_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x829a, "HP 800 G3 DM", CXT_FIXUP_HP_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x103c, 0x836e, "HP ProBook 455 G5", CXT_FIXUP_MUTE_LED_GPIO), + SND_PCI_QUIRK(0x103c, 0x837f, "HP ProBook 470 G5", CXT_FIXUP_MUTE_LED_GPIO), + SND_PCI_QUIRK(0x103c, 0x83b2, "HP EliteBook 840 G5", CXT_FIXUP_HP_DOCK), + SND_PCI_QUIRK(0x103c, 0x83b3, "HP EliteBook 830 G5", CXT_FIXUP_HP_DOCK), + SND_PCI_QUIRK(0x103c, 0x83d3, "HP ProBook 640 G4", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 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), @@ -1118,6 +1118,7 @@ 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(0x14f120d0, "CX11970", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f15045, "CX20549 (Venice)", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f15047, "CX20551 (Waikiki)", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f15051, "CX20561 (Hermosa)", patch_conexant_auto), diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 12913368c231..f7b5f980455a 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -339,13 +339,13 @@ static int hdmi_eld_ctl_info(struct snd_kcontrol *kcontrol, if (!per_pin) { /* no pin is bound to the pcm */ uinfo->count = 0; - mutex_unlock(&spec->pcm_lock); - return 0; + goto unlock; } eld = &per_pin->sink_eld; uinfo->count = eld->eld_valid ? eld->eld_size : 0; - mutex_unlock(&spec->pcm_lock); + unlock: + mutex_unlock(&spec->pcm_lock); return 0; } @@ -357,6 +357,7 @@ static int hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol, struct hdmi_spec_per_pin *per_pin; struct hdmi_eld *eld; int pcm_idx; + int err = 0; pcm_idx = kcontrol->private_value; mutex_lock(&spec->pcm_lock); @@ -365,16 +366,15 @@ static int hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol, /* no pin is bound to the pcm */ memset(ucontrol->value.bytes.data, 0, ARRAY_SIZE(ucontrol->value.bytes.data)); - mutex_unlock(&spec->pcm_lock); - return 0; + goto unlock; } - eld = &per_pin->sink_eld; + eld = &per_pin->sink_eld; if (eld->eld_size > ARRAY_SIZE(ucontrol->value.bytes.data) || eld->eld_size > ELD_MAX_SIZE) { - mutex_unlock(&spec->pcm_lock); snd_BUG(); - return -EINVAL; + err = -EINVAL; + goto unlock; } memset(ucontrol->value.bytes.data, 0, @@ -382,9 +382,10 @@ static int hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol, if (eld->eld_valid) memcpy(ucontrol->value.bytes.data, eld->eld_buffer, eld->eld_size); - mutex_unlock(&spec->pcm_lock); - return 0; + unlock: + mutex_unlock(&spec->pcm_lock); + return err; } static const struct snd_kcontrol_new eld_bytes_ctl = { @@ -1209,8 +1210,8 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, pin_idx = hinfo_to_pin_index(codec, hinfo); if (!spec->dyn_pcm_assign) { if (snd_BUG_ON(pin_idx < 0)) { - mutex_unlock(&spec->pcm_lock); - return -EINVAL; + err = -EINVAL; + goto unlock; } } else { /* no pin is assigned to the PCM @@ -1218,16 +1219,13 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, */ if (pin_idx < 0) { err = hdmi_pcm_open_no_pin(hinfo, codec, substream); - mutex_unlock(&spec->pcm_lock); - return err; + goto unlock; } } err = hdmi_choose_cvt(codec, pin_idx, &cvt_idx); - if (err < 0) { - mutex_unlock(&spec->pcm_lock); - return err; - } + if (err < 0) + goto unlock; per_cvt = get_cvt(spec, cvt_idx); /* Claim converter */ @@ -1264,12 +1262,11 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, per_cvt->assigned = 0; hinfo->nid = 0; snd_hda_spdif_ctls_unassign(codec, pcm_idx); - mutex_unlock(&spec->pcm_lock); - return -ENODEV; + err = -ENODEV; + goto unlock; } } - mutex_unlock(&spec->pcm_lock); /* Store the updated parameters */ runtime->hw.channels_min = hinfo->channels_min; runtime->hw.channels_max = hinfo->channels_max; @@ -1278,7 +1275,9 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, snd_pcm_hw_constraint_step(substream->runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, 2); - return 0; + unlock: + mutex_unlock(&spec->pcm_lock); + return err; } /* @@ -1849,8 +1848,10 @@ static bool check_non_pcm_per_cvt(struct hda_codec *codec, hda_nid_t cvt_nid) /* Add sanity check to pass klockwork check. * This should never happen. */ - if (WARN_ON(spdif == NULL)) + if (WARN_ON(spdif == NULL)) { + mutex_unlock(&codec->spdif_mutex); return true; + } non_pcm = !!(spdif->status & IEC958_AES0_NONAUDIO); mutex_unlock(&codec->spdif_mutex); return non_pcm; @@ -1874,7 +1875,7 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct snd_pcm_runtime *runtime = substream->runtime; bool non_pcm; int pinctl; - int err; + int err = 0; mutex_lock(&spec->pcm_lock); pin_idx = hinfo_to_pin_index(codec, hinfo); @@ -1886,13 +1887,12 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, pin_cvt_fixup(codec, NULL, cvt_nid); snd_hda_codec_setup_stream(codec, cvt_nid, stream_tag, 0, format); - mutex_unlock(&spec->pcm_lock); - return 0; + goto unlock; } if (snd_BUG_ON(pin_idx < 0)) { - mutex_unlock(&spec->pcm_lock); - return -EINVAL; + err = -EINVAL; + goto unlock; } per_pin = get_pin(spec, pin_idx); pin_nid = per_pin->pin_nid; @@ -1931,6 +1931,7 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, /* snd_hda_set_dev_select() has been called before */ err = spec->ops.setup_stream(codec, cvt_nid, pin_nid, stream_tag, format); + unlock: mutex_unlock(&spec->pcm_lock); return err; } @@ -1952,32 +1953,34 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo, struct hdmi_spec_per_cvt *per_cvt; struct hdmi_spec_per_pin *per_pin; int pinctl; + int err = 0; + mutex_lock(&spec->pcm_lock); if (hinfo->nid) { pcm_idx = hinfo_to_pcm_index(codec, hinfo); - if (snd_BUG_ON(pcm_idx < 0)) - return -EINVAL; + if (snd_BUG_ON(pcm_idx < 0)) { + err = -EINVAL; + goto unlock; + } cvt_idx = cvt_nid_to_cvt_index(codec, hinfo->nid); - if (snd_BUG_ON(cvt_idx < 0)) - return -EINVAL; + if (snd_BUG_ON(cvt_idx < 0)) { + err = -EINVAL; + goto unlock; + } per_cvt = get_cvt(spec, cvt_idx); - snd_BUG_ON(!per_cvt->assigned); per_cvt->assigned = 0; hinfo->nid = 0; - mutex_lock(&spec->pcm_lock); snd_hda_spdif_ctls_unassign(codec, pcm_idx); clear_bit(pcm_idx, &spec->pcm_in_use); pin_idx = hinfo_to_pin_index(codec, hinfo); - if (spec->dyn_pcm_assign && pin_idx < 0) { - mutex_unlock(&spec->pcm_lock); - return 0; - } + if (spec->dyn_pcm_assign && pin_idx < 0) + goto unlock; if (snd_BUG_ON(pin_idx < 0)) { - mutex_unlock(&spec->pcm_lock); - return -EINVAL; + err = -EINVAL; + goto unlock; } per_pin = get_pin(spec, pin_idx); @@ -1996,10 +1999,12 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo, per_pin->setup = false; per_pin->channels = 0; mutex_unlock(&per_pin->lock); - mutex_unlock(&spec->pcm_lock); } - return 0; +unlock: + mutex_unlock(&spec->pcm_lock); + + return err; } static const struct hda_pcm_ops generic_ops = { @@ -2210,7 +2215,9 @@ static int generic_hdmi_build_controls(struct hda_codec *codec) for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); + struct hdmi_eld *pin_eld = &per_pin->sink_eld; + pin_eld->eld_valid = false; hdmi_present_sense(per_pin, 0); } @@ -2317,6 +2324,18 @@ static void generic_hdmi_free(struct hda_codec *codec) } #ifdef CONFIG_PM +static int generic_hdmi_suspend(struct hda_codec *codec) +{ + struct hdmi_spec *spec = codec->spec; + int pin_idx; + + for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { + struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); + cancel_delayed_work_sync(&per_pin->work); + } + return 0; +} + static int generic_hdmi_resume(struct hda_codec *codec) { struct hdmi_spec *spec = codec->spec; @@ -2340,6 +2359,7 @@ static const struct hda_codec_ops generic_hdmi_patch_ops = { .build_controls = generic_hdmi_build_controls, .unsol_event = hdmi_unsol_event, #ifdef CONFIG_PM + .suspend = generic_hdmi_suspend, .resume = generic_hdmi_resume, #endif }; @@ -2542,6 +2562,7 @@ static void i915_pin_cvt_fixup(struct hda_codec *codec, hda_nid_t cvt_nid) { if (per_pin) { + haswell_verify_D0(codec, per_pin->cvt_nid, per_pin->pin_nid); snd_hda_set_dev_select(codec, per_pin->pin_nid, per_pin->dev_id); intel_verify_pin_cvt_connect(codec, per_pin); @@ -3394,6 +3415,7 @@ static int tegra_hdmi_build_pcms(struct hda_codec *codec) static int patch_tegra_hdmi(struct hda_codec *codec) { + struct hdmi_spec *spec; int err; err = patch_generic_hdmi(codec); @@ -3401,6 +3423,10 @@ static int patch_tegra_hdmi(struct hda_codec *codec) return err; codec->patch_ops.build_pcms = tegra_hdmi_build_pcms; + spec = codec->spec; + spec->chmap.ops.chmap_cea_alloc_validate_get_type = + nvhdmi_chmap_cea_alloc_validate_get_type; + spec->chmap.ops.chmap_validate = nvhdmi_chmap_validate; return 0; } @@ -3857,6 +3883,11 @@ HDA_CODEC_ENTRY(0x10de0095, "GPU 95 HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de0097, "GPU 97 HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de0098, "GPU 98 HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de0099, "GPU 99 HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de009a, "GPU 9a HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de009d, "GPU 9d HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de009e, "GPU 9e HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de009f, "GPU 9f HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de00a0, "GPU a0 HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de8001, "MCP73 HDMI", patch_nvhdmi_2ch), HDA_CODEC_ENTRY(0x10de8067, "MCP67/68 HDMI", patch_nvhdmi_2ch), HDA_CODEC_ENTRY(0x11069f80, "VX900 HDMI/DP", patch_via_hdmi), diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b500dad33ea9..1f954d3ce499 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -333,7 +333,10 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0215: case 0x10ec0233: case 0x10ec0235: + case 0x10ec0236: + case 0x10ec0245: case 0x10ec0255: + case 0x10ec0256: case 0x10ec0257: case 0x10ec0282: case 0x10ec0283: @@ -345,14 +348,13 @@ 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; + case 0x10ec0287: + alc_update_coef_idx(codec, 0x10, 1<<9, 0); + alc_write_coef_idx(codec, 0x8, 0x4ab7); + break; case 0x10ec0293: alc_update_coef_idx(codec, 0xa, 1<<13, 0); break; @@ -393,6 +395,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) alc_update_coef_idx(codec, 0x7, 1<<5, 0); break; case 0x10ec0892: + case 0x10ec0897: alc_update_coef_idx(codec, 0x7, 1<<5, 0); break; case 0x10ec0899: @@ -1792,6 +1795,7 @@ enum { ALC889_FIXUP_FRONT_HP_NO_PRESENCE, ALC889_FIXUP_VAIO_TT, ALC888_FIXUP_EEE1601, + ALC886_FIXUP_EAPD, ALC882_FIXUP_EAPD, ALC883_FIXUP_EAPD, ALC883_FIXUP_ACER_EAPD, @@ -2100,6 +2104,15 @@ static const struct hda_fixup alc882_fixups[] = { { } } }, + [ALC886_FIXUP_EAPD] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + /* change to EAPD mode */ + { 0x20, AC_VERB_SET_COEF_INDEX, 0x07 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0068 }, + { } + } + }, [ALC882_FIXUP_EAPD] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -2291,13 +2304,13 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { ALC882_FIXUP_ACER_ASPIRE_8930G), SND_PCI_QUIRK(0x1025, 0x0146, "Acer Aspire 6935G", ALC882_FIXUP_ACER_ASPIRE_8930G), + SND_PCI_QUIRK(0x1025, 0x0142, "Acer Aspire 7730G", + ALC882_FIXUP_ACER_ASPIRE_4930G), + SND_PCI_QUIRK(0x1025, 0x0155, "Packard-Bell M5120", ALC882_FIXUP_PB_M5210), SND_PCI_QUIRK(0x1025, 0x015e, "Acer Aspire 6930G", ALC882_FIXUP_ACER_ASPIRE_4930G), SND_PCI_QUIRK(0x1025, 0x0166, "Acer Aspire 6530G", ALC882_FIXUP_ACER_ASPIRE_4930G), - SND_PCI_QUIRK(0x1025, 0x0142, "Acer Aspire 7730G", - ALC882_FIXUP_ACER_ASPIRE_4930G), - SND_PCI_QUIRK(0x1025, 0x0155, "Packard-Bell M5120", ALC882_FIXUP_PB_M5210), SND_PCI_QUIRK(0x1025, 0x021e, "Acer Aspire 5739G", ALC882_FIXUP_ACER_ASPIRE_4930G), SND_PCI_QUIRK(0x1025, 0x0259, "Acer Aspire 5935", ALC889_FIXUP_DAC_ROUTE), @@ -2309,11 +2322,11 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_FIXUP_EEE1601), SND_PCI_QUIRK(0x1043, 0x84bc, "ASUS ET2700", ALC887_FIXUP_ASUS_BASS), SND_PCI_QUIRK(0x1043, 0x8691, "ASUS ROG Ranger VIII", ALC882_FIXUP_GPIO3), + SND_PCI_QUIRK(0x104d, 0x9043, "Sony Vaio VGC-LN51JGB", ALC882_FIXUP_NO_PRIMARY_HP), + SND_PCI_QUIRK(0x104d, 0x9044, "Sony VAIO AiO", ALC882_FIXUP_NO_PRIMARY_HP), SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC889_FIXUP_VAIO_TT), SND_PCI_QUIRK(0x104d, 0x905a, "Sony Vaio Z", ALC882_FIXUP_NO_PRIMARY_HP), SND_PCI_QUIRK(0x104d, 0x9060, "Sony Vaio VPCL14M1R", ALC882_FIXUP_NO_PRIMARY_HP), - SND_PCI_QUIRK(0x104d, 0x9043, "Sony Vaio VGC-LN51JGB", ALC882_FIXUP_NO_PRIMARY_HP), - SND_PCI_QUIRK(0x104d, 0x9044, "Sony VAIO AiO", ALC882_FIXUP_NO_PRIMARY_HP), /* All Apple entries are in codec SSIDs */ SND_PCI_QUIRK(0x106b, 0x00a0, "MacBookPro 3,1", ALC889_FIXUP_MBP_VREF), @@ -2340,6 +2353,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x106b, 0x4a00, "Macbook 5,2", ALC889_FIXUP_MBA11_VREF), SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC882_FIXUP_EAPD), + SND_PCI_QUIRK(0x13fe, 0x1009, "Advantech MIT-W101", ALC886_FIXUP_EAPD), SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte EP45-DS3/Z87X-UD3H", ALC889_FIXUP_FRONT_HP_NO_PRESENCE), SND_PCI_QUIRK(0x1458, 0xa0b8, "Gigabyte AZ370-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS), SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD), @@ -3122,7 +3136,13 @@ 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*/ + /* + * Expose headphone mic (or possibly Line In on some machines) instead + * of PC Beep on 1Ah, and disable 1Ah loopback for all outputs. See + * Documentation/sound/hd-audio/realtek-pc-beep.rst for details of + * this register. + */ + alc_write_coef_idx(codec, 0x36, 0x5757); } static void alc256_shutup(struct hda_codec *codec) @@ -3149,7 +3169,11 @@ static void alc256_shutup(struct hda_codec *codec) /* 3k pull low control for Headset jack. */ /* NOTE: call this before clearing the pin, otherwise codec stalls */ - alc_update_coef_idx(codec, 0x46, 0, 3 << 12); + /* If disable 3k pulldown control for alc257, the Mic detection will not work correctly + * when booting with headset plugged. So skip setting it for the codec alc257 + */ + if (codec->core.vendor_id != 0x10ec0257) + alc_update_coef_idx(codec, 0x46, 0, 3 << 12); snd_hda_codec_write(codec, hp_pin, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); @@ -4621,7 +4645,7 @@ static void alc_determine_headset_type(struct hda_codec *codec) case 0x10ec0274: case 0x10ec0294: alc_process_coef_fw(codec, coef0274); - msleep(80); + msleep(850); val = alc_read_coef_idx(codec, 0x46); is_ctia = (val & 0x00f0) == 0x00f0; break; @@ -4687,8 +4711,6 @@ static void alc_determine_headset_type(struct hda_codec *codec) is_ctia = (val & 0x1c02) == 0x1c02; break; case 0x10ec0225: - codec->power_save_node = 1; - /* fall through */ case 0x10ec0295: case 0x10ec0299: alc_process_coef_fw(codec, alc225_pre_hsmode); @@ -4792,6 +4814,7 @@ static void alc_update_headset_jack_cb(struct hda_codec *codec, struct alc_spec *spec = codec->spec; spec->current_headset_type = ALC_HEADSET_TYPE_UNKNOWN; snd_hda_gen_hp_automute(codec, jack); + alc_update_headset_mode(codec); } static void alc_probe_headset_mode(struct hda_codec *codec) @@ -5350,6 +5373,15 @@ static void alc233_alc662_fixup_lenovo_dual_codecs(struct hda_codec *codec, } } +static void alc225_fixup_s3_pop_noise(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action != HDA_FIXUP_ACT_PRE_PROBE) + return; + + codec->power_save_node = 1; +} + /* Forcibly assign NID 0x03 to HP/LO while NID 0x02 to SPK for EQ */ static void alc274_fixup_bind_dacs(struct hda_codec *codec, const struct hda_fixup *fix, int action) @@ -5420,6 +5452,7 @@ enum { ALC269_FIXUP_HP_LINE1_MIC1_LED, ALC269_FIXUP_INV_DMIC, ALC269_FIXUP_LENOVO_DOCK, + ALC269_FIXUP_LENOVO_DOCK_LIMIT_BOOST, ALC269_FIXUP_NO_SHUTUP, ALC286_FIXUP_SONY_MIC_NO_PRESENCE, ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT, @@ -5492,6 +5525,7 @@ enum { ALC221_FIXUP_HP_FRONT_MIC, ALC292_FIXUP_TPT460, ALC298_FIXUP_SPK_VOLUME, + ALC298_FIXUP_LENOVO_SPK_VOLUME, ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER, ALC269_FIXUP_ATIV_BOOK_8, ALC221_FIXUP_HP_MIC_NO_PRESENCE, @@ -5503,6 +5537,7 @@ enum { ALC233_FIXUP_LENOVO_MULTI_CODECS, ALC294_FIXUP_LENOVO_MIC_LOCATION, ALC225_FIXUP_DELL_WYSE_MIC_NO_PRESENCE, + ALC225_FIXUP_S3_POP_NOISE, ALC700_FIXUP_INTEL_REFERENCE, ALC274_FIXUP_DELL_BIND_DACS, ALC274_FIXUP_DELL_AIO_LINEOUT_VERB, @@ -5712,6 +5747,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT }, + [ALC269_FIXUP_LENOVO_DOCK_LIMIT_BOOST] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_limit_int_mic_boost, + .chained = true, + .chain_id = ALC269_FIXUP_LENOVO_DOCK, + }, [ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_pincfg_no_hp_to_lineout, @@ -6236,6 +6277,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC298_FIXUP_DELL_AIO_MIC_NO_PRESENCE, }, + [ALC298_FIXUP_LENOVO_SPK_VOLUME] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc298_fixup_speaker_volume, + }, [ALC295_FIXUP_DISABLE_DAC3] = { .type = HDA_FIXUP_FUNC, .v.func = alc295_fixup_disable_dac3, @@ -6335,6 +6380,12 @@ static const struct hda_fixup alc269_fixups[] = { { } }, .chained = true, + .chain_id = ALC225_FIXUP_S3_POP_NOISE + }, + [ALC225_FIXUP_S3_POP_NOISE] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc225_fixup_s3_pop_noise, + .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC }, [ALC700_FIXUP_INTEL_REFERENCE] = { @@ -6559,6 +6610,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1043, 0x1bbd, "ASUS Z550MA", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1c23, "Asus X55U", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x1043, 0x125e, "ASUS Q524UQK", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1ccd, "ASUS X555UB", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x3030, "ASUS ZN270IE", ALC256_FIXUP_ASUS_AIO_GPIO2), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), @@ -6566,12 +6618,12 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x8516, "ASUS X101CH", ALC269_FIXUP_ASUS_X101), - SND_PCI_QUIRK(0x104d, 0x90b5, "Sony VAIO Pro 11", ALC286_FIXUP_SONY_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x104d, 0x90b6, "Sony VAIO Pro 13", ALC286_FIXUP_SONY_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x104d, 0x9073, "Sony VAIO", ALC275_FIXUP_SONY_VAIO_GPIO2), SND_PCI_QUIRK(0x104d, 0x907b, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), SND_PCI_QUIRK(0x104d, 0x9099, "Sony VAIO S13", ALC275_FIXUP_SONY_DISABLE_AAMIX), + SND_PCI_QUIRK(0x104d, 0x90b5, "Sony VAIO Pro 11", ALC286_FIXUP_SONY_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x104d, 0x90b6, "Sony VAIO Pro 13", ALC286_FIXUP_SONY_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook", ALC269_FIXUP_LIFEBOOK), SND_PCI_QUIRK(0x10cf, 0x159f, "Lifebook E780", ALC269_FIXUP_LIFEBOOK_NO_HP_TO_LINEOUT), SND_PCI_QUIRK(0x10cf, 0x15dc, "Lifebook T731", ALC269_FIXUP_LIFEBOOK_HP_PIN), @@ -6586,14 +6638,15 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb171, "Cubi N 8GL (MS-B171)", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x17aa, 0x1036, "Lenovo P520", ALC233_FIXUP_LENOVO_MULTI_CODECS), + SND_PCI_QUIRK(0x17aa, 0x1048, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x21ca, "Thinkpad L412", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x21e9, "Thinkpad Edge 15", ALC269_FIXUP_SKU_IGNORE), - SND_PCI_QUIRK(0x17aa, 0x21f6, "Thinkpad T530", ALC269_FIXUP_LENOVO_DOCK), - SND_PCI_QUIRK(0x17aa, 0x21fa, "Thinkpad X230", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x21f3, "Thinkpad T430", ALC269_FIXUP_LENOVO_DOCK), + SND_PCI_QUIRK(0x17aa, 0x21f6, "Thinkpad T530", ALC269_FIXUP_LENOVO_DOCK_LIMIT_BOOST), + SND_PCI_QUIRK(0x17aa, 0x21fa, "Thinkpad X230", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2208, "Thinkpad T431s", ALC269_FIXUP_LENOVO_DOCK), @@ -6629,9 +6682,11 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3151, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x17aa, 0x3176, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x17aa, 0x3178, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), + SND_PCI_QUIRK(0x17aa, 0x3818, "Lenovo C940", ALC298_FIXUP_LENOVO_SPK_VOLUME), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI), + SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x501a, "Thinkpad", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x501e, "Thinkpad L440", ALC292_FIXUP_TPT440_DOCK), @@ -6650,7 +6705,6 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x511e, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x511f, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), - SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ @@ -6723,6 +6777,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC269_FIXUP_HEADSET_MODE, .name = "headset-mode"}, {.id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC, .name = "headset-mode-no-hp-mic"}, {.id = ALC269_FIXUP_LENOVO_DOCK, .name = "lenovo-dock"}, + {.id = ALC269_FIXUP_LENOVO_DOCK_LIMIT_BOOST, .name = "lenovo-dock-limit-boost"}, {.id = ALC269_FIXUP_HP_GPIO_LED, .name = "hp-gpio-led"}, {.id = ALC269_FIXUP_HP_DOCK_GPIO_MIC1_LED, .name = "hp-dock-gpio-mic1-led"}, {.id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "dell-headset-multi"}, @@ -7103,6 +7158,12 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { ALC225_STANDARD_PINS, {0x12, 0xb7a60130}, {0x17, 0x90170110}), + SND_HDA_PIN_QUIRK(0x10ec0623, 0x17aa, "Lenovo", ALC283_FIXUP_HEADSET_MIC, + {0x14, 0x01014010}, + {0x17, 0x90170120}, + {0x18, 0x02a11030}, + {0x19, 0x02a1103f}, + {0x21, 0x0221101f}), {} }; @@ -7261,7 +7322,9 @@ static int patch_alc269(struct hda_codec *codec) spec->gen.mixer_nid = 0; break; case 0x10ec0215: + case 0x10ec0245: case 0x10ec0285: + case 0x10ec0287: case 0x10ec0289: spec->codec_variant = ALC269_TYPE_ALC215; spec->gen.mixer_nid = 0; @@ -7410,8 +7473,7 @@ static const struct snd_pci_quirk alc861_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1393, "ASUS A6Rp", ALC861_FIXUP_ASUS_A6RP), SND_PCI_QUIRK_VENDOR(0x1043, "ASUS laptop", ALC861_FIXUP_AMP_VREF_0F), SND_PCI_QUIRK(0x1462, 0x7254, "HP DX2200", ALC861_FIXUP_NO_JACK_DETECT), - SND_PCI_QUIRK(0x1584, 0x2b01, "Haier W18", ALC861_FIXUP_AMP_VREF_0F), - SND_PCI_QUIRK(0x1584, 0x0000, "Uniwill ECS M31EI", ALC861_FIXUP_AMP_VREF_0F), + SND_PCI_QUIRK_VENDOR(0x1584, "Haier/Uniwill", ALC861_FIXUP_AMP_VREF_0F), SND_PCI_QUIRK(0x1734, 0x10c7, "FSC Amilo Pi1505", ALC861_FIXUP_FSC_AMILO_PI1505), {} }; @@ -8341,6 +8403,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0234, "ALC234", patch_alc269), HDA_CODEC_ENTRY(0x10ec0235, "ALC233", patch_alc269), HDA_CODEC_ENTRY(0x10ec0236, "ALC236", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0245, "ALC245", patch_alc269), HDA_CODEC_ENTRY(0x10ec0255, "ALC255", patch_alc269), HDA_CODEC_ENTRY(0x10ec0256, "ALC256", patch_alc269), HDA_CODEC_ENTRY(0x10ec0257, "ALC257", patch_alc269), @@ -8360,6 +8423,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0284, "ALC284", patch_alc269), HDA_CODEC_ENTRY(0x10ec0285, "ALC285", patch_alc269), HDA_CODEC_ENTRY(0x10ec0286, "ALC286", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0287, "ALC287", patch_alc269), HDA_CODEC_ENTRY(0x10ec0288, "ALC288", patch_alc269), HDA_CODEC_ENTRY(0x10ec0289, "ALC289", patch_alc269), HDA_CODEC_ENTRY(0x10ec0290, "ALC290", patch_alc269), @@ -8401,6 +8465,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0888, "ALC888", patch_alc882), HDA_CODEC_ENTRY(0x10ec0889, "ALC889", patch_alc882), HDA_CODEC_ENTRY(0x10ec0892, "ALC892", patch_alc662), + HDA_CODEC_ENTRY(0x10ec0897, "ALC897", patch_alc662), HDA_CODEC_ENTRY(0x10ec0899, "ALC898", patch_alc882), HDA_CODEC_ENTRY(0x10ec0900, "ALC1150", patch_alc882), HDA_CODEC_ENTRY(0x10ec0b00, "ALCS1200A", patch_alc882), diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 7cd147411b22..f7896a9ae3d6 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -863,7 +863,7 @@ static int stac_auto_create_beep_ctls(struct hda_codec *codec, static struct snd_kcontrol_new beep_vol_ctl = HDA_CODEC_VOLUME(NULL, 0, 0, 0); - /* check for mute support for the the amp */ + /* check for mute support for the amp */ if ((caps & AC_AMPCAP_MUTE) >> AC_AMPCAP_MUTE_SHIFT) { const struct snd_kcontrol_new *temp; if (spec->anabeep_nid == nid) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index fc30d1e8aa76..9dd104c308e1 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -135,6 +135,7 @@ static struct via_spec *via_new_spec(struct hda_codec *codec) spec->codec_type = VT1708S; spec->gen.indep_hp = 1; spec->gen.keep_eapd_on = 1; + spec->gen.dac_min_mute = 1; spec->gen.pcm_playback_hook = via_playback_pcm_hook; spec->gen.add_stereo_mix_input = HDA_HINT_STEREO_MIX_AUTO; codec->power_save_node = 1; diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 0e66afa403a3..5a7928e1b29e 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -2377,7 +2377,8 @@ static int snd_ice1712_chip_init(struct snd_ice1712 *ice) pci_write_config_byte(ice->pci, 0x61, ice->eeprom.data[ICE_EEP1_ACLINK]); pci_write_config_byte(ice->pci, 0x62, ice->eeprom.data[ICE_EEP1_I2SID]); pci_write_config_byte(ice->pci, 0x63, ice->eeprom.data[ICE_EEP1_SPDIF]); - if (ice->eeprom.subvendor != ICE1712_SUBDEVICE_STDSP24) { + if (ice->eeprom.subvendor != ICE1712_SUBDEVICE_STDSP24 && + ice->eeprom.subvendor != ICE1712_SUBDEVICE_STAUDIO_ADCIII) { ice->gpio.write_mask = ice->eeprom.gpiomask; ice->gpio.direction = ice->eeprom.gpiodir; snd_ice1712_write(ice, ICE1712_IREG_GPIO_WRITE_MASK, diff --git a/sound/pci/ice1712/prodigy192.c b/sound/pci/ice1712/prodigy192.c index 3919aed39ca0..5e52086d7b98 100644 --- a/sound/pci/ice1712/prodigy192.c +++ b/sound/pci/ice1712/prodigy192.c @@ -31,7 +31,7 @@ * Experimentally I found out that only a combination of * OCKS0=1, OCKS1=1 (128fs, 64fs output) and ice1724 - * VT1724_MT_I2S_MCLK_128X=0 (256fs input) yields correct - * sampling rate. That means the the FPGA doubles the + * sampling rate. That means that the FPGA doubles the * MCK01 rate. * * Copyright (c) 2003 Takashi Iwai <tiwai@suse.de> diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c index 2697402b5195..41f6450a2539 100644 --- a/sound/pci/ice1712/prodigy_hifi.c +++ b/sound/pci/ice1712/prodigy_hifi.c @@ -569,7 +569,7 @@ static int wm_adc_mux_enum_get(struct snd_kcontrol *kcontrol, struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); mutex_lock(&ice->gpio_mutex); - ucontrol->value.integer.value[0] = wm_get(ice, WM_ADC_MUX) & 0x1f; + ucontrol->value.enumerated.item[0] = wm_get(ice, WM_ADC_MUX) & 0x1f; mutex_unlock(&ice->gpio_mutex); return 0; } @@ -583,7 +583,7 @@ static int wm_adc_mux_enum_put(struct snd_kcontrol *kcontrol, mutex_lock(&ice->gpio_mutex); oval = wm_get(ice, WM_ADC_MUX); - nval = (oval & 0xe0) | ucontrol->value.integer.value[0]; + nval = (oval & 0xe0) | ucontrol->value.enumerated.item[0]; if (nval != oval) { wm_put(ice, WM_ADC_MUX, nval); change = 1; diff --git a/sound/pci/mixart/mixart_core.c b/sound/pci/mixart/mixart_core.c index 8bf2ce32d4a8..2ea693ee33a1 100644 --- a/sound/pci/mixart/mixart_core.c +++ b/sound/pci/mixart/mixart_core.c @@ -83,7 +83,6 @@ static int get_msg(struct mixart_mgr *mgr, struct mixart_msg *resp, unsigned int i; #endif - mutex_lock(&mgr->msg_lock); err = 0; /* copy message descriptor from miXart to driver */ @@ -132,8 +131,6 @@ static int get_msg(struct mixart_mgr *mgr, struct mixart_msg *resp, writel_be(headptr, MIXART_MEM(mgr, MSG_OUTBOUND_FREE_HEAD)); _clean_exit: - mutex_unlock(&mgr->msg_lock); - return err; } @@ -271,7 +268,9 @@ int snd_mixart_send_msg(struct mixart_mgr *mgr, struct mixart_msg *request, int resp.data = resp_data; resp.size = max_resp_size; + mutex_lock(&mgr->msg_lock); err = get_msg(mgr, &resp, msg_frame); + mutex_unlock(&mgr->msg_lock); if( request->message_id != resp.message_id ) dev_err(&mgr->pci->dev, "RESPONSE ERROR!\n"); diff --git a/sound/pci/oxygen/xonar_dg.c b/sound/pci/oxygen/xonar_dg.c index 4cf3200e988b..df44135e1b0c 100644 --- a/sound/pci/oxygen/xonar_dg.c +++ b/sound/pci/oxygen/xonar_dg.c @@ -39,7 +39,7 @@ * GPIO 4 <- headphone detect * GPIO 5 -> enable ADC analog circuit for the left channel * GPIO 6 -> enable ADC analog circuit for the right channel - * GPIO 7 -> switch green rear output jack between CS4245 and and the first + * GPIO 7 -> switch green rear output jack between CS4245 and the first * channel of CS4361 (mechanical relay) * GPIO 8 -> enable output to speakers * diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index e41bb4100306..edd359772f1f 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -5343,7 +5343,8 @@ static int snd_hdsp_free(struct hdsp *hdsp) if (hdsp->port) pci_release_regions(hdsp->pci); - pci_disable_device(hdsp->pci); + if (pci_is_enabled(hdsp->pci)) + pci_disable_device(hdsp->pci); return 0; } diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 343f533906ba..5bbbbba0817b 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -6913,7 +6913,8 @@ static int snd_hdspm_free(struct hdspm * hdspm) if (hdspm->port) pci_release_regions(hdspm->pci); - pci_disable_device(hdspm->pci); + if (pci_is_enabled(hdspm->pci)) + pci_disable_device(hdspm->pci); return 0; } diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c index edd765e22377..f82fa5be7d33 100644 --- a/sound/pci/rme9652/rme9652.c +++ b/sound/pci/rme9652/rme9652.c @@ -1761,7 +1761,8 @@ static int snd_rme9652_free(struct snd_rme9652 *rme9652) if (rme9652->port) pci_release_regions(rme9652->pci); - pci_disable_device(rme9652->pci); + if (pci_is_enabled(rme9652->pci)) + pci_disable_device(rme9652->pci); return 0; } diff --git a/sound/ppc/powermac.c b/sound/ppc/powermac.c index 33c6be9fb388..7c70ba5e2540 100644 --- a/sound/ppc/powermac.c +++ b/sound/ppc/powermac.c @@ -90,7 +90,11 @@ static int snd_pmac_probe(struct platform_device *devptr) sprintf(card->shortname, "PowerMac %s", name_ext); sprintf(card->longname, "%s (Dev %d) Sub-frame %d", card->shortname, chip->device_id, chip->subframe); - if ( snd_pmac_tumbler_init(chip) < 0 || snd_pmac_tumbler_post_init() < 0) + err = snd_pmac_tumbler_init(chip); + if (err < 0) + goto __error; + err = snd_pmac_tumbler_post_init(); + if (err < 0) goto __error; break; case PMAC_AWACS: diff --git a/sound/soc/codecs/cs35l33.c b/sound/soc/codecs/cs35l33.c index 854cf8f27605..e2c1194ea61a 100644 --- a/sound/soc/codecs/cs35l33.c +++ b/sound/soc/codecs/cs35l33.c @@ -1206,6 +1206,7 @@ static int cs35l33_i2c_probe(struct i2c_client *i2c_client, dev_err(&i2c_client->dev, "CS35L33 Device ID (%X). Expected ID %X\n", devid, CS35L33_CHIP_ID); + ret = -EINVAL; goto err_enable; } diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index a2324a0e72ee..7ff8b9f26971 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -95,7 +95,7 @@ static const struct reg_default cs42l42_reg_defaults[] = { { CS42L42_ASP_RX_INT_MASK, 0x1F }, { CS42L42_ASP_TX_INT_MASK, 0x0F }, { CS42L42_CODEC_INT_MASK, 0x03 }, - { CS42L42_SRCPL_INT_MASK, 0xFF }, + { CS42L42_SRCPL_INT_MASK, 0x7F }, { CS42L42_VPMON_INT_MASK, 0x01 }, { CS42L42_PLL_LOCK_INT_MASK, 0x01 }, { CS42L42_TSRS_PLUG_INT_MASK, 0x0F }, @@ -132,7 +132,7 @@ static const struct reg_default cs42l42_reg_defaults[] = { { CS42L42_MIXER_CHA_VOL, 0x3F }, { CS42L42_MIXER_ADC_VOL, 0x3F }, { CS42L42_MIXER_CHB_VOL, 0x3F }, - { CS42L42_EQ_COEF_IN0, 0x22 }, + { CS42L42_EQ_COEF_IN0, 0x00 }, { CS42L42_EQ_COEF_IN1, 0x00 }, { CS42L42_EQ_COEF_IN2, 0x00 }, { CS42L42_EQ_COEF_IN3, 0x00 }, @@ -404,8 +404,8 @@ static const struct regmap_config cs42l42_regmap = { .cache_type = REGCACHE_RBTREE, }; -static DECLARE_TLV_DB_SCALE(adc_tlv, -9600, 100, false); -static DECLARE_TLV_DB_SCALE(mixer_tlv, -6200, 100, false); +static DECLARE_TLV_DB_SCALE(adc_tlv, -9700, 100, true); +static DECLARE_TLV_DB_SCALE(mixer_tlv, -6300, 100, true); static const char * const cs42l42_hpf_freq_text[] = { "1.86Hz", "120Hz", "235Hz", "466Hz" @@ -424,34 +424,23 @@ static SOC_ENUM_SINGLE_DECL(cs42l42_wnf3_freq_enum, CS42L42_ADC_WNF_HPF_CTL, CS42L42_ADC_WNF_CF_SHIFT, cs42l42_wnf3_freq_text); -static const char * const cs42l42_wnf05_freq_text[] = { - "280Hz", "315Hz", "350Hz", "385Hz", - "420Hz", "455Hz", "490Hz", "525Hz" -}; - -static SOC_ENUM_SINGLE_DECL(cs42l42_wnf05_freq_enum, CS42L42_ADC_WNF_HPF_CTL, - CS42L42_ADC_WNF_CF_SHIFT, - cs42l42_wnf05_freq_text); - static const struct snd_kcontrol_new cs42l42_snd_controls[] = { /* ADC Volume and Filter Controls */ SOC_SINGLE("ADC Notch Switch", CS42L42_ADC_CTL, - CS42L42_ADC_NOTCH_DIS_SHIFT, true, false), + CS42L42_ADC_NOTCH_DIS_SHIFT, true, true), SOC_SINGLE("ADC Weak Force Switch", CS42L42_ADC_CTL, CS42L42_ADC_FORCE_WEAK_VCM_SHIFT, true, false), SOC_SINGLE("ADC Invert Switch", CS42L42_ADC_CTL, CS42L42_ADC_INV_SHIFT, true, false), SOC_SINGLE("ADC Boost Switch", CS42L42_ADC_CTL, CS42L42_ADC_DIG_BOOST_SHIFT, true, false), - SOC_SINGLE_SX_TLV("ADC Volume", CS42L42_ADC_VOLUME, - CS42L42_ADC_VOL_SHIFT, 0xA0, 0x6C, adc_tlv), + SOC_SINGLE_S8_TLV("ADC Volume", CS42L42_ADC_VOLUME, -97, 12, adc_tlv), SOC_SINGLE("ADC WNF Switch", CS42L42_ADC_WNF_HPF_CTL, CS42L42_ADC_WNF_EN_SHIFT, true, false), SOC_SINGLE("ADC HPF Switch", CS42L42_ADC_WNF_HPF_CTL, CS42L42_ADC_HPF_EN_SHIFT, true, false), SOC_ENUM("HPF Corner Freq", cs42l42_hpf_freq_enum), SOC_ENUM("WNF 3dB Freq", cs42l42_wnf3_freq_enum), - SOC_ENUM("WNF 05dB Freq", cs42l42_wnf05_freq_enum), /* DAC Volume and Filter Controls */ SOC_SINGLE("DACA Invert Switch", CS42L42_DAC_CTL1, @@ -462,7 +451,7 @@ static const struct snd_kcontrol_new cs42l42_snd_controls[] = { CS42L42_DAC_HPF_EN_SHIFT, true, false), SOC_DOUBLE_R_TLV("Mixer Volume", CS42L42_MIXER_CHA_VOL, CS42L42_MIXER_CHB_VOL, CS42L42_MIXER_CH_VOL_SHIFT, - 0x3e, 1, mixer_tlv) + 0x3f, 1, mixer_tlv) }; static int cs42l42_hpdrv_evt(struct snd_soc_dapm_widget *w, @@ -794,7 +783,6 @@ static int cs42l42_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) /* interface format */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - case SND_SOC_DAIFMT_LEFT_J: break; default: return -EINVAL; @@ -1809,7 +1797,7 @@ static int cs42l42_i2c_probe(struct i2c_client *i2c_client, dev_dbg(&i2c_client->dev, "Found reset GPIO\n"); gpiod_set_value_cansleep(cs42l42->reset_gpio, 1); } - mdelay(3); + usleep_range(CS42L42_BOOT_TIME_US, CS42L42_BOOT_TIME_US * 2); /* Request IRQ */ ret = devm_request_threaded_irq(&i2c_client->dev, @@ -1817,8 +1805,9 @@ static int cs42l42_i2c_probe(struct i2c_client *i2c_client, NULL, cs42l42_irq_thread, IRQF_ONESHOT | IRQF_TRIGGER_LOW, "cs42l42", cs42l42); - - if (ret != 0) + if (ret == -EPROBE_DEFER) + goto err_disable; + else if (ret != 0) dev_err(&i2c_client->dev, "Failed to request IRQ: %d\n", ret); @@ -1936,6 +1925,7 @@ static int cs42l42_runtime_resume(struct device *dev) } gpiod_set_value_cansleep(cs42l42->reset_gpio, 1); + usleep_range(CS42L42_BOOT_TIME_US, CS42L42_BOOT_TIME_US * 2); regcache_cache_only(cs42l42->regmap, false); regcache_sync(cs42l42->regmap); diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h index d87a0a5322d5..72d3778e10ad 100644 --- a/sound/soc/codecs/cs42l42.h +++ b/sound/soc/codecs/cs42l42.h @@ -81,7 +81,7 @@ #define CS42L42_HP_PDN_SHIFT 3 #define CS42L42_HP_PDN_MASK (1 << CS42L42_HP_PDN_SHIFT) #define CS42L42_ADC_PDN_SHIFT 2 -#define CS42L42_ADC_PDN_MASK (1 << CS42L42_HP_PDN_SHIFT) +#define CS42L42_ADC_PDN_MASK (1 << CS42L42_ADC_PDN_SHIFT) #define CS42L42_PDN_ALL_SHIFT 0 #define CS42L42_PDN_ALL_MASK (1 << CS42L42_PDN_ALL_SHIFT) @@ -743,6 +743,7 @@ #define CS42L42_FRAC2_VAL(val) (((val) & 0xff0000) >> 16) #define CS42L42_NUM_SUPPLIES 5 +#define CS42L42_BOOT_TIME_US 3000 static const char *const cs42l42_supply_names[CS42L42_NUM_SUPPLIES] = { "VA", diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index cb6ca85f1536..52858b6c95a6 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -1266,6 +1266,7 @@ static int cs42l56_i2c_probe(struct i2c_client *i2c_client, dev_err(&i2c_client->dev, "CS42L56 Device ID (%X). Expected %X\n", devid, CS42L56_DEVID); + ret = -EINVAL; goto err_enable; } alpha_rev = reg & CS42L56_AREV_MASK; @@ -1323,7 +1324,7 @@ static int cs42l56_i2c_probe(struct i2c_client *i2c_client, ret = snd_soc_register_codec(&i2c_client->dev, &soc_codec_dev_cs42l56, &cs42l56_dai, 1); if (ret < 0) - return ret; + goto err_enable; return 0; diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index 949dbdc0445e..0410f2e5183c 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -56,13 +56,8 @@ static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(adc_pga_gain_tlv, 1, 1, TLV_DB_SCALE_ITEM(0, 0, 0), 2, 2, TLV_DB_SCALE_ITEM(250, 0, 0), 3, 3, TLV_DB_SCALE_ITEM(450, 0, 0), - 4, 4, TLV_DB_SCALE_ITEM(700, 0, 0), - 5, 5, TLV_DB_SCALE_ITEM(1000, 0, 0), - 6, 6, TLV_DB_SCALE_ITEM(1300, 0, 0), - 7, 7, TLV_DB_SCALE_ITEM(1600, 0, 0), - 8, 8, TLV_DB_SCALE_ITEM(1800, 0, 0), - 9, 9, TLV_DB_SCALE_ITEM(2100, 0, 0), - 10, 10, TLV_DB_SCALE_ITEM(2400, 0, 0), + 4, 7, TLV_DB_SCALE_ITEM(700, 300, 0), + 8, 10, TLV_DB_SCALE_ITEM(1800, 300, 0), ); static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(hpout_vol_tlv, diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 1c3626347e12..aeeec1144558 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -142,14 +142,14 @@ static struct hdac_hdmi_pcm * hdac_hdmi_get_pcm_from_cvt(struct hdac_hdmi_priv *hdmi, struct hdac_hdmi_cvt *cvt) { - struct hdac_hdmi_pcm *pcm = NULL; + struct hdac_hdmi_pcm *pcm; list_for_each_entry(pcm, &hdmi->pcm_list, head) { if (pcm->cvt == cvt) - break; + return pcm; } - return pcm; + return NULL; } static void hdac_hdmi_jack_report(struct hdac_hdmi_pcm *pcm, diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index 3633eb30dd13..4f949ad50d6a 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -16,8 +16,8 @@ #define CDC_D_REVISION1 (0xf000) #define CDC_D_PERPH_SUBTYPE (0xf005) -#define CDC_D_INT_EN_SET (0x015) -#define CDC_D_INT_EN_CLR (0x016) +#define CDC_D_INT_EN_SET (0xf015) +#define CDC_D_INT_EN_CLR (0xf016) #define MBHC_SWITCH_INT BIT(7) #define MBHC_MIC_ELECTRICAL_INS_REM_DET BIT(6) #define MBHC_BUTTON_PRESS_DET BIT(5) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index af6325c78292..ce3865a8ddc2 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -174,6 +174,9 @@ static bool rt286_readable_register(struct device *dev, unsigned int reg) case RT286_PROC_COEF: case RT286_SET_AMP_GAIN_ADC_IN1: case RT286_SET_AMP_GAIN_ADC_IN2: + case RT286_SET_GPIO_MASK: + case RT286_SET_GPIO_DIRECTION: + case RT286_SET_GPIO_DATA: case RT286_SET_POWER(RT286_DAC_OUT1): case RT286_SET_POWER(RT286_DAC_OUT2): case RT286_SET_POWER(RT286_ADC_IN1): @@ -1119,12 +1122,11 @@ static const struct dmi_system_id force_combo_jack_table[] = { { } }; -static const struct dmi_system_id dmi_dell_dino[] = { +static const struct dmi_system_id dmi_dell[] = { { - .ident = "Dell Dino", + .ident = "Dell", .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc."), - DMI_MATCH(DMI_PRODUCT_NAME, "XPS 13 9343") } }, { } @@ -1135,7 +1137,7 @@ static int rt286_i2c_probe(struct i2c_client *i2c, { struct rt286_platform_data *pdata = dev_get_platdata(&i2c->dev); struct rt286_priv *rt286; - int i, ret, val; + int i, ret, vendor_id; rt286 = devm_kzalloc(&i2c->dev, sizeof(*rt286), GFP_KERNEL); @@ -1151,14 +1153,15 @@ static int rt286_i2c_probe(struct i2c_client *i2c, } ret = regmap_read(rt286->regmap, - RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &val); + RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &vendor_id); if (ret != 0) { dev_err(&i2c->dev, "I2C error %d\n", ret); return ret; } - if (val != RT286_VENDOR_ID && val != RT288_VENDOR_ID) { + if (vendor_id != RT286_VENDOR_ID && vendor_id != RT288_VENDOR_ID) { dev_err(&i2c->dev, - "Device with ID register %#x is not rt286\n", val); + "Device with ID register %#x is not rt286\n", + vendor_id); return -ENODEV; } @@ -1182,8 +1185,8 @@ static int rt286_i2c_probe(struct i2c_client *i2c, if (pdata) rt286->pdata = *pdata; - if (dmi_check_system(force_combo_jack_table) || - dmi_check_system(dmi_dell_dino)) + if ((vendor_id == RT288_VENDOR_ID && dmi_check_system(dmi_dell)) || + dmi_check_system(force_combo_jack_table)) rt286->pdata.cbj_en = true; regmap_write(rt286->regmap, RT286_SET_AUDIO_POWER, AC_PWRST_D3); @@ -1222,7 +1225,7 @@ static int rt286_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL3, 0xf777, 0x4737); regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL4, 0x00ff, 0x003f); - if (dmi_check_system(dmi_dell_dino)) { + if (vendor_id == RT288_VENDOR_ID && dmi_check_system(dmi_dell)) { regmap_update_bits(rt286->regmap, RT286_SET_GPIO_MASK, 0x40, 0x40); regmap_update_bits(rt286->regmap, diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 438fe52a12df..5af5dfc0fd46 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -341,9 +341,9 @@ static bool rt5640_readable_register(struct device *dev, unsigned int reg) } static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0); -static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); +static const DECLARE_TLV_DB_MINMAX(dac_vol_tlv, -6562, 0); static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); -static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); +static const DECLARE_TLV_DB_MINMAX(adc_vol_tlv, -1762, 3000); static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); /* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */ diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index 57c2add323c4..38510fd06458 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -287,9 +287,9 @@ static bool rt5651_readable_register(struct device *dev, unsigned int reg) } static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0); -static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); +static const DECLARE_TLV_DB_MINMAX(dac_vol_tlv, -6562, 0); static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); -static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); +static const DECLARE_TLV_DB_MINMAX(adc_vol_tlv, -1762, 3000); static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); /* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */ diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c index fa66b11df8d4..ae626d57c1ad 100644 --- a/sound/soc/codecs/rt5659.c +++ b/sound/soc/codecs/rt5659.c @@ -3391,12 +3391,17 @@ static int rt5659_set_dai_sysclk(struct snd_soc_dai *dai, struct snd_soc_codec *codec = dai->codec; struct rt5659_priv *rt5659 = snd_soc_codec_get_drvdata(codec); unsigned int reg_val = 0; + int ret; if (freq == rt5659->sysclk && clk_id == rt5659->sysclk_src) return 0; switch (clk_id) { case RT5659_SCLK_S_MCLK: + ret = clk_set_rate(rt5659->mclk, freq); + if (ret) + return ret; + reg_val |= RT5659_SCLK_SRC_MCLK; break; case RT5659_SCLK_S_PLL1: diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h index 5ba485cae4e6..06d7c0aaeb61 100644 --- a/sound/soc/codecs/rt5670.h +++ b/sound/soc/codecs/rt5670.h @@ -760,7 +760,7 @@ #define RT5670_PWR_VREF2_BIT 4 #define RT5670_PWR_FV2 (0x1 << 3) #define RT5670_PWR_FV2_BIT 3 -#define RT5670_LDO_SEL_MASK (0x3) +#define RT5670_LDO_SEL_MASK (0x7) #define RT5670_LDO_SEL_SFT 0 /* Power Management for Analog 2 (0x64) */ diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index ca8a70ab22a8..b7a0002d9872 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -75,7 +75,7 @@ static const struct reg_default sgtl5000_reg_defaults[] = { { SGTL5000_DAP_EQ_BASS_BAND4, 0x002f }, { SGTL5000_DAP_MAIN_CHAN, 0x8000 }, { SGTL5000_DAP_MIX_CHAN, 0x0000 }, - { SGTL5000_DAP_AVC_CTRL, 0x0510 }, + { SGTL5000_DAP_AVC_CTRL, 0x5100 }, { SGTL5000_DAP_AVC_THRESHOLD, 0x1473 }, { SGTL5000_DAP_AVC_ATTACK, 0x0028 }, { SGTL5000_DAP_AVC_DECAY, 0x0050 }, @@ -1563,6 +1563,40 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, dev_err(&client->dev, "Error %d initializing CHIP_CLK_CTRL\n", ret); + /* Mute everything to avoid pop from the following power-up */ + ret = regmap_write(sgtl5000->regmap, SGTL5000_CHIP_ANA_CTRL, + SGTL5000_CHIP_ANA_CTRL_DEFAULT); + if (ret) { + dev_err(&client->dev, + "Error %d muting outputs via CHIP_ANA_CTRL\n", ret); + goto disable_clk; + } + + /* + * If VAG is powered-on (e.g. from previous boot), it would be disabled + * by the write to ANA_POWER in later steps of the probe code. This + * may create a loud pop even with all outputs muted. The proper way + * to circumvent this is disabling the bit first and waiting the proper + * cool-down time. + */ + ret = regmap_read(sgtl5000->regmap, SGTL5000_CHIP_ANA_POWER, &value); + if (ret) { + dev_err(&client->dev, "Failed to read ANA_POWER: %d\n", ret); + goto disable_clk; + } + if (value & SGTL5000_VAG_POWERUP) { + ret = regmap_update_bits(sgtl5000->regmap, + SGTL5000_CHIP_ANA_POWER, + SGTL5000_VAG_POWERUP, + 0); + if (ret) { + dev_err(&client->dev, "Error %d disabling VAG\n", ret); + goto disable_clk; + } + + msleep(SGTL5000_VAG_POWERDOWN_DELAY); + } + /* Follow section 2.2.1.1 of AN3663 */ ana_pwr = SGTL5000_ANA_POWER_DEFAULT; if (sgtl5000->num_supplies <= VDDD) { diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h index 22f3442af982..9ea41749d037 100644 --- a/sound/soc/codecs/sgtl5000.h +++ b/sound/soc/codecs/sgtl5000.h @@ -236,6 +236,7 @@ /* * SGTL5000_CHIP_ANA_CTRL */ +#define SGTL5000_CHIP_ANA_CTRL_DEFAULT 0x0133 #define SGTL5000_LINE_OUT_MUTE 0x0100 #define SGTL5000_HP_SEL_MASK 0x0040 #define SGTL5000_HP_SEL_SHIFT 6 diff --git a/sound/soc/codecs/sti-sas.c b/sound/soc/codecs/sti-sas.c index 62c618765224..730dd453a744 100644 --- a/sound/soc/codecs/sti-sas.c +++ b/sound/soc/codecs/sti-sas.c @@ -407,6 +407,7 @@ static const struct of_device_id sti_sas_dev_match[] = { }, {}, }; +MODULE_DEVICE_TABLE(of, sti_sas_dev_match); static int sti_sas_driver_probe(struct platform_device *pdev) { diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 9ed455700954..228ab7bd314d 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -710,7 +710,13 @@ int wm8960_configure_pll(struct snd_soc_codec *codec, int freq_in, best_freq_out = -EINVAL; *sysclk_idx = *dac_idx = *bclk_idx = -1; - for (i = 0; i < ARRAY_SIZE(sysclk_divs); ++i) { + /* + * From Datasheet, the PLL performs best when f2 is between + * 90MHz and 100MHz, the desired sysclk output is 11.2896MHz + * or 12.288MHz, then sysclkdiv = 2 is the best choice. + * So search sysclk_divs from 2 to 1 other than from 1 to 2. + */ + for (i = ARRAY_SIZE(sysclk_divs) - 1; i >= 0; --i) { if (sysclk_divs[i] == -1) continue; for (j = 0; j < ARRAY_SIZE(dac_divs); ++j) { diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index 49401a8aae64..19c963801e26 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -1179,6 +1179,8 @@ static int wm8997_probe(struct platform_device *pdev) goto err_spk_irqs; } + return ret; + err_spk_irqs: arizona_free_spk_irqs(arizona); diff --git a/sound/soc/codecs/wm8998.c b/sound/soc/codecs/wm8998.c index 44f447136e22..a94e0aeb2e19 100644 --- a/sound/soc/codecs/wm8998.c +++ b/sound/soc/codecs/wm8998.c @@ -1425,7 +1425,7 @@ static int wm8998_probe(struct platform_device *pdev) ret = arizona_init_spk_irqs(arizona); if (ret < 0) - return ret; + goto err_pm_disable; ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wm8998, wm8998_dai, ARRAY_SIZE(wm8998_dai)); @@ -1438,6 +1438,8 @@ static int wm8998_probe(struct platform_device *pdev) err_spk_irqs: arizona_free_spk_irqs(arizona); +err_pm_disable: + pm_runtime_disable(&pdev->dev); return ret; } diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 158ce68bc9bf..1516252aa0a5 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1391,7 +1391,7 @@ static int wm_adsp_create_control(struct wm_adsp *dsp, ctl_work = kzalloc(sizeof(*ctl_work), GFP_KERNEL); if (!ctl_work) { ret = -ENOMEM; - goto err_ctl_cache; + goto err_list_del; } ctl_work->dsp = dsp; @@ -1401,7 +1401,8 @@ static int wm_adsp_create_control(struct wm_adsp *dsp, return 0; -err_ctl_cache: +err_list_del: + list_del(&ctl->list); kfree(ctl->cache); err_ctl_name: kfree(ctl->name); diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index e10e03800cce..6991718d7c8a 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1747,8 +1747,10 @@ static int davinci_mcasp_get_dma_type(struct davinci_mcasp *mcasp) PTR_ERR(chan)); return PTR_ERR(chan); } - if (WARN_ON(!chan->device || !chan->device->dev)) + if (WARN_ON(!chan->device || !chan->device->dev)) { + dma_release_channel(chan); return -EINVAL; + } if (chan->device->dev->of_node) ret = of_property_read_string(chan->device->dev->of_node, diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index e1b97e59275a..15d7e6da0555 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -243,6 +243,7 @@ static int fsl_asrc_dma_hw_params(struct snd_pcm_substream *substream, ret = dmaengine_slave_config(pair->dma_chan[dir], &config_be); if (ret) { dev_err(dev, "failed to config DMA channel for Back-End\n"); + dma_release_channel(pair->dma_chan[dir]); return ret; } diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 6152ae24772b..3ac87f7843f6 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -494,11 +494,13 @@ static int fsl_esai_startup(struct snd_pcm_substream *substream, ESAI_SAICR_SYNC, esai_priv->synchronous ? ESAI_SAICR_SYNC : 0); - /* Set a default slot number -- 2 */ + /* Set slots count */ regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR, - ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(2)); + ESAI_xCCR_xDC_MASK, + ESAI_xCCR_xDC(esai_priv->slots)); regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, - ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(2)); + ESAI_xCCR_xDC_MASK, + ESAI_xCCR_xDC(esai_priv->slots)); } return 0; diff --git a/sound/soc/hisilicon/hi6210-i2s.c b/sound/soc/hisilicon/hi6210-i2s.c index 0c8f86d4020e..d8d14cdee786 100644 --- a/sound/soc/hisilicon/hi6210-i2s.c +++ b/sound/soc/hisilicon/hi6210-i2s.c @@ -111,18 +111,15 @@ static int hi6210_i2s_startup(struct snd_pcm_substream *substream, for (n = 0; n < i2s->clocks; n++) { ret = clk_prepare_enable(i2s->clk[n]); - if (ret) { - while (n--) - clk_disable_unprepare(i2s->clk[n]); - return ret; - } + if (ret) + goto err_unprepare_clk; } ret = clk_set_rate(i2s->clk[CLK_I2S_BASE], 49152000); if (ret) { dev_err(i2s->dev, "%s: setting 49.152MHz base rate failed %d\n", __func__, ret); - return ret; + goto err_unprepare_clk; } /* enable clock before frequency division */ @@ -174,6 +171,11 @@ static int hi6210_i2s_startup(struct snd_pcm_substream *substream, hi6210_write_reg(i2s, HII2S_SW_RST_N, val); return 0; + +err_unprepare_clk: + while (n--) + clk_disable_unprepare(i2s->clk[n]); + return ret; } static void hi6210_i2s_shutdown(struct snd_pcm_substream *substream, diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c index 0f3604b55942..999eb3ba7867 100644 --- a/sound/soc/intel/atom/sst-atom-controls.c +++ b/sound/soc/intel/atom/sst-atom-controls.c @@ -974,7 +974,9 @@ static int sst_set_be_modules(struct snd_soc_dapm_widget *w, dev_dbg(c->dev, "Enter: widget=%s\n", w->name); if (SND_SOC_DAPM_EVENT_ON(event)) { + mutex_lock(&drv->lock); ret = sst_send_slot_map(drv); + mutex_unlock(&drv->lock); if (ret) return ret; ret = sst_send_pipe_module_params(w, k); @@ -1341,7 +1343,7 @@ int sst_send_pipe_gains(struct snd_soc_dai *dai, int stream, int mute) dai->capture_widget->name); w = dai->capture_widget; snd_soc_dapm_widget_for_each_source_path(w, p) { - if (p->connected && !p->connected(w, p->sink)) + if (p->connected && !p->connected(w, p->source)) continue; if (p->connect && p->source->power && diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 4558c8b93036..96f7facd0fa0 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -135,7 +135,7 @@ static void sst_fill_alloc_params(struct snd_pcm_substream *substream, snd_pcm_uframes_t period_size; ssize_t periodbytes; ssize_t buffer_bytes = snd_pcm_lib_buffer_bytes(substream); - u32 buffer_addr = virt_to_phys(substream->dma_buffer.area); + u32 buffer_addr = virt_to_phys(substream->runtime->dma_area); channels = substream->runtime->channels; period_size = substream->runtime->period_size; @@ -241,7 +241,6 @@ static int sst_platform_alloc_stream(struct snd_pcm_substream *substream, /* set codec params and inform SST driver the same */ sst_fill_pcm_params(substream, ¶m); sst_fill_alloc_params(substream, &alloc_params); - substream->runtime->dma_area = substream->dma_buffer.area; str_params.sparams = param; str_params.aparams = alloc_params; str_params.codec = SST_CODEC_TYPE_PCM; @@ -339,7 +338,7 @@ static int sst_media_open(struct snd_pcm_substream *substream, ret_val = power_up_sst(stream); if (ret_val < 0) - return ret_val; + goto out_power_up; /* Make sure, that the period size is always even */ snd_pcm_hw_constraint_step(substream->runtime, 0, @@ -348,8 +347,9 @@ static int sst_media_open(struct snd_pcm_substream *substream, return snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); out_ops: - kfree(stream); mutex_unlock(&sst_lock); +out_power_up: + kfree(stream); return ret_val; } @@ -507,14 +507,14 @@ static struct snd_soc_dai_driver sst_platform_dai[] = { .channels_min = SST_STEREO, .channels_max = SST_STEREO, .rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { .stream_name = "Headset Capture", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + .formats = SNDRV_PCM_FMTBIT_S16_LE, }, }, { @@ -525,7 +525,7 @@ static struct snd_soc_dai_driver sst_platform_dai[] = { .channels_min = SST_STEREO, .channels_max = SST_STEREO, .rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + .formats = SNDRV_PCM_FMTBIT_S16_LE, }, }, { diff --git a/sound/soc/intel/atom/sst/sst_pci.c b/sound/soc/intel/atom/sst/sst_pci.c index 6906ee624cf6..438c7bcd8c4c 100644 --- a/sound/soc/intel/atom/sst/sst_pci.c +++ b/sound/soc/intel/atom/sst/sst_pci.c @@ -107,7 +107,7 @@ static int sst_platform_get_resources(struct intel_sst_drv *ctx) dev_dbg(ctx->dev, "DRAM Ptr %p\n", ctx->dram); do_release_regions: pci_release_regions(pci); - return 0; + return ret; } /* diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c index 7843104fadcb..1b01bc318fd2 100644 --- a/sound/soc/intel/boards/bxt_rt298.c +++ b/sound/soc/intel/boards/bxt_rt298.c @@ -529,6 +529,7 @@ static int bxt_card_late_probe(struct snd_soc_card *card) /* broxton audio machine driver for SPT + RT298S */ static struct snd_soc_card broxton_rt298 = { .name = "broxton-rt298", + .owner = THIS_MODULE, .dai_link = broxton_rt298_dais, .num_links = ARRAY_SIZE(broxton_rt298_dais), .controls = broxton_controls, @@ -544,6 +545,7 @@ static struct snd_soc_card broxton_rt298 = { static struct snd_soc_card geminilake_rt298 = { .name = "geminilake-rt298", + .owner = THIS_MODULE, .dai_link = broxton_rt298_dais, .num_links = ARRAY_SIZE(broxton_rt298_dais), .controls = broxton_controls, diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 4a76b099a508..e389ecf06e63 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -226,9 +226,6 @@ static const struct snd_soc_dapm_widget byt_rt5640_widgets[] = { static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = { {"Headphone", NULL, "Platform Clock"}, {"Headset Mic", NULL, "Platform Clock"}, - {"Internal Mic", NULL, "Platform Clock"}, - {"Speaker", NULL, "Platform Clock"}, - {"Headset Mic", NULL, "MICBIAS1"}, {"IN2P", NULL, "Headset Mic"}, {"Headphone", NULL, "HPOL"}, @@ -236,19 +233,23 @@ static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = { }; static const struct snd_soc_dapm_route byt_rt5640_intmic_dmic1_map[] = { + {"Internal Mic", NULL, "Platform Clock"}, {"DMIC1", NULL, "Internal Mic"}, }; static const struct snd_soc_dapm_route byt_rt5640_intmic_dmic2_map[] = { + {"Internal Mic", NULL, "Platform Clock"}, {"DMIC2", NULL, "Internal Mic"}, }; static const struct snd_soc_dapm_route byt_rt5640_intmic_in1_map[] = { + {"Internal Mic", NULL, "Platform Clock"}, {"Internal Mic", NULL, "MICBIAS1"}, {"IN1P", NULL, "Internal Mic"}, }; static const struct snd_soc_dapm_route byt_rt5640_intmic_in3_map[] = { + {"Internal Mic", NULL, "Platform Clock"}, {"Internal Mic", NULL, "MICBIAS1"}, {"IN3P", NULL, "Internal Mic"}, }; @@ -290,6 +291,7 @@ static const struct snd_soc_dapm_route byt_rt5640_ssp0_aif2_map[] = { }; static const struct snd_soc_dapm_route byt_rt5640_stereo_spk_map[] = { + {"Speaker", NULL, "Platform Clock"}, {"Speaker", NULL, "SPOLP"}, {"Speaker", NULL, "SPOLN"}, {"Speaker", NULL, "SPORP"}, @@ -297,6 +299,7 @@ static const struct snd_soc_dapm_route byt_rt5640_stereo_spk_map[] = { }; static const struct snd_soc_dapm_route byt_rt5640_mono_spk_map[] = { + {"Speaker", NULL, "Platform Clock"}, {"Speaker", NULL, "SPOLP"}, {"Speaker", NULL, "SPOLN"}, }; diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c index 8158409921e0..c6007aa95fff 100644 --- a/sound/soc/intel/boards/haswell.c +++ b/sound/soc/intel/boards/haswell.c @@ -197,6 +197,7 @@ static struct platform_driver haswell_audio = { .probe = haswell_audio_probe, .driver = { .name = "haswell-audio", + .pm = &snd_soc_pm_ops, }, }; diff --git a/sound/soc/intel/skylake/cnl-sst.c b/sound/soc/intel/skylake/cnl-sst.c index 387de388ce29..6a5080361887 100644 --- a/sound/soc/intel/skylake/cnl-sst.c +++ b/sound/soc/intel/skylake/cnl-sst.c @@ -212,6 +212,7 @@ static int cnl_set_dsp_D0(struct sst_dsp *ctx, unsigned int core_id) "dsp boot timeout, status=%#x error=%#x\n", sst_dsp_shim_read(ctx, CNL_ADSP_FW_STATUS), sst_dsp_shim_read(ctx, CNL_ADSP_ERROR_CODE)); + ret = -ETIMEDOUT; goto err; } } else { diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index 99394c036998..2c6b0ac97c68 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -92,7 +92,7 @@ #define JZ_AIC_I2S_STATUS_BUSY BIT(2) #define JZ_AIC_CLK_DIV_MASK 0xf -#define I2SDIV_DV_SHIFT 8 +#define I2SDIV_DV_SHIFT 0 #define I2SDIV_DV_MASK (0xf << I2SDIV_DV_SHIFT) #define I2SDIV_IDV_SHIFT 8 #define I2SDIV_IDV_MASK (0xf << I2SDIV_IDV_SHIFT) @@ -318,10 +318,14 @@ static int jz4740_i2s_set_sysclk(struct snd_soc_dai *dai, int clk_id, switch (clk_id) { case JZ4740_I2S_CLKSRC_EXT: parent = clk_get(NULL, "ext"); + if (IS_ERR(parent)) + return PTR_ERR(parent); clk_set_parent(i2s->clk_i2s, parent); break; case JZ4740_I2S_CLKSRC_PLL: parent = clk_get(NULL, "pll half"); + if (IS_ERR(parent)) + return PTR_ERR(parent); clk_set_parent(i2s->clk_i2s, parent); ret = clk_set_rate(i2s->clk_i2s, freq); break; diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index cf23af159acf..35ca8e8bb5e5 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -136,7 +136,7 @@ static int kirkwood_dma_open(struct snd_pcm_substream *substream) err = request_irq(priv->irq, kirkwood_dma_irq, IRQF_SHARED, "kirkwood-i2s", priv); if (err) - return -EBUSY; + return err; /* * Enable Error interrupts. We're only ack'ing them but diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index 292b103abada..475579a9830a 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -182,21 +182,6 @@ static int lpass_cpu_daiops_hw_params(struct snd_pcm_substream *substream, return 0; } -static int lpass_cpu_daiops_hw_free(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); - int ret; - - ret = regmap_write(drvdata->lpaif_map, - LPAIF_I2SCTL_REG(drvdata->variant, dai->driver->id), - 0); - if (ret) - dev_err(dai->dev, "error writing to i2sctl reg: %d\n", ret); - - return ret; -} - static int lpass_cpu_daiops_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -277,7 +262,6 @@ const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops = { .startup = lpass_cpu_daiops_startup, .shutdown = lpass_cpu_daiops_shutdown, .hw_params = lpass_cpu_daiops_hw_params, - .hw_free = lpass_cpu_daiops_hw_free, .prepare = lpass_cpu_daiops_prepare, .trigger = lpass_cpu_daiops_trigger, }; diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index e1945e1772cd..35c49fc9602b 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -67,7 +67,7 @@ static int lpass_platform_pcmops_open(struct snd_pcm_substream *substream) int ret, dma_ch, dir = substream->stream; struct lpass_pcm_data *data; - data = devm_kzalloc(soc_runtime->dev, sizeof(*data), GFP_KERNEL); + data = kzalloc(sizeof(*data), GFP_KERNEL); if (!data) return -ENOMEM; @@ -80,8 +80,10 @@ static int lpass_platform_pcmops_open(struct snd_pcm_substream *substream) else dma_ch = 0; - if (dma_ch < 0) + if (dma_ch < 0) { + kfree(data); return dma_ch; + } drvdata->substream[dma_ch] = substream; @@ -102,6 +104,7 @@ static int lpass_platform_pcmops_open(struct snd_pcm_substream *substream) ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); if (ret < 0) { + kfree(data); dev_err(soc_runtime->dev, "setting constraints failed: %d\n", ret); return -EINVAL; @@ -127,6 +130,7 @@ static int lpass_platform_pcmops_close(struct snd_pcm_substream *substream) if (v->free_dma_channel) v->free_dma_channel(drvdata, data->dma_ch); + kfree(data); return 0; } diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 0e07e3dea7de..8d1a7114f6c2 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -188,7 +188,9 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, { struct rk_i2s_dev *i2s = to_info(cpu_dai); unsigned int mask = 0, val = 0; + int ret = 0; + pm_runtime_get_sync(cpu_dai->dev); mask = I2S_CKR_MSS_MASK; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: @@ -201,7 +203,8 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, i2s->is_master_mode = false; break; default: - return -EINVAL; + ret = -EINVAL; + goto err_pm_put; } regmap_update_bits(i2s->regmap, I2S_CKR, mask, val); @@ -215,7 +218,8 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, val = I2S_CKR_CKP_POS; break; default: - return -EINVAL; + ret = -EINVAL; + goto err_pm_put; } regmap_update_bits(i2s->regmap, I2S_CKR, mask, val); @@ -231,14 +235,15 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, case SND_SOC_DAIFMT_I2S: val = I2S_TXCR_IBM_NORMAL; break; - case SND_SOC_DAIFMT_DSP_A: /* PCM no delay mode */ - val = I2S_TXCR_TFS_PCM; - break; - case SND_SOC_DAIFMT_DSP_B: /* PCM delay 1 mode */ + case SND_SOC_DAIFMT_DSP_A: /* PCM delay 1 bit mode */ val = I2S_TXCR_TFS_PCM | I2S_TXCR_PBM_MODE(1); break; + case SND_SOC_DAIFMT_DSP_B: /* PCM no delay mode */ + val = I2S_TXCR_TFS_PCM; + break; default: - return -EINVAL; + ret = -EINVAL; + goto err_pm_put; } regmap_update_bits(i2s->regmap, I2S_TXCR, mask, val); @@ -254,19 +259,23 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, case SND_SOC_DAIFMT_I2S: val = I2S_RXCR_IBM_NORMAL; break; - case SND_SOC_DAIFMT_DSP_A: /* PCM no delay mode */ - val = I2S_RXCR_TFS_PCM; - break; - case SND_SOC_DAIFMT_DSP_B: /* PCM delay 1 mode */ + case SND_SOC_DAIFMT_DSP_A: /* PCM delay 1 bit mode */ val = I2S_RXCR_TFS_PCM | I2S_RXCR_PBM_MODE(1); break; + case SND_SOC_DAIFMT_DSP_B: /* PCM no delay mode */ + val = I2S_RXCR_TFS_PCM; + break; default: - return -EINVAL; + ret = -EINVAL; + goto err_pm_put; } regmap_update_bits(i2s->regmap, I2S_RXCR, mask, val); - return 0; +err_pm_put: + pm_runtime_put(cpu_dai->dev); + + return ret; } static int rockchip_i2s_hw_params(struct snd_pcm_substream *substream, diff --git a/sound/soc/rockchip/rockchip_pdm.c b/sound/soc/rockchip/rockchip_pdm.c index 8a2e3bbce3a1..ad16c8310dd3 100644 --- a/sound/soc/rockchip/rockchip_pdm.c +++ b/sound/soc/rockchip/rockchip_pdm.c @@ -478,8 +478,10 @@ static int rockchip_pdm_resume(struct device *dev) int ret; ret = pm_runtime_get_sync(dev); - if (ret < 0) + if (ret < 0) { + pm_runtime_put(dev); return ret; + } ret = regcache_sync(pdm->regmap); diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index 4d948757d300..5e5ed5475473 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -172,7 +172,7 @@ static int rsnd_ssiu_init_gen2(struct rsnd_mod *mod, i; for_each_rsnd_mod_array(i, pos, io, rsnd_ssi_array) { - shift = (i * 4) + 16; + shift = (i * 4) + 20; val = (val & ~(0xF << shift)) | rsnd_mod_id(pos) << shift; } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 42c2a3065b77..2a172de37466 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4046,7 +4046,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, if (!routes) { dev_err(card->dev, "ASoC: Could not allocate DAPM route table\n"); - return -EINVAL; + return -ENOMEM; } for (i = 0; i < num_routes; i++) { diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index fb2fef166672..f72fe0cba30d 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -413,7 +413,7 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, memset(&template, 0, sizeof(template)); template.reg = e->reg; - template.mask = e->mask << e->shift_l; + template.mask = e->mask; template.shift = e->shift_l; template.off_val = snd_soc_enum_item_to_val(e, 0); template.on_val = template.off_val; @@ -539,8 +539,22 @@ static bool dapm_kcontrol_set_value(const struct snd_kcontrol *kcontrol, if (data->value == value) return false; - if (data->widget) - data->widget->on_val = value; + if (data->widget) { + switch (dapm_kcontrol_get_wlist(kcontrol)->widgets[0]->id) { + case snd_soc_dapm_switch: + case snd_soc_dapm_mixer: + case snd_soc_dapm_mixer_named_ctl: + data->widget->on_val = value & data->widget->mask; + break; + case snd_soc_dapm_demux: + case snd_soc_dapm_mux: + data->widget->on_val = value >> data->widget->shift; + break; + default: + data->widget->on_val = value; + break; + } + } data->value = value; @@ -799,7 +813,13 @@ static void dapm_set_mixer_path_status(struct snd_soc_dapm_path *p, int i, val = max - val; p->connect = !!val; } else { - p->connect = 0; + /* since a virtual mixer has no backing registers to + * decide which path to connect, it will try to match + * with initial state. This is to ensure + * that the default mixer choice will be + * correctly powered up during initialization. + */ + p->connect = invert; } } @@ -2414,6 +2434,7 @@ void snd_soc_dapm_free_widget(struct snd_soc_dapm_widget *w) enum snd_soc_dapm_direction dir; list_del(&w->list); + list_del(&w->dirty); /* * remove source and sink paths associated to this widget. * While removing the path, remove reference to it from both @@ -2470,10 +2491,16 @@ static struct snd_soc_dapm_widget *dapm_find_widget( return NULL; } -static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, - const char *pin, int status) +/* + * set the DAPM pin status: + * returns 1 when the value has been updated, 0 when unchanged, or a negative + * error code; called from kcontrol put callback + */ +static int __snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, + const char *pin, int status) { struct snd_soc_dapm_widget *w = dapm_find_widget(dapm, pin, true); + int ret = 0; dapm_assert_locked(dapm); @@ -2486,13 +2513,26 @@ static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, dapm_mark_dirty(w, "pin configuration"); dapm_widget_invalidate_input_paths(w); dapm_widget_invalidate_output_paths(w); + ret = 1; } w->connected = status; if (status == 0) w->force = 0; - return 0; + return ret; +} + +/* + * similar as __snd_soc_dapm_set_pin(), but returns 0 when successful; + * called from several API functions below + */ +static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, + const char *pin, int status) +{ + int ret = __snd_soc_dapm_set_pin(dapm, pin, status); + + return ret < 0 ? ret : 0; } /** @@ -3420,14 +3460,15 @@ int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol, { struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); const char *pin = (const char *)kcontrol->private_value; + int ret; - if (ucontrol->value.integer.value[0]) - snd_soc_dapm_enable_pin(&card->dapm, pin); - else - snd_soc_dapm_disable_pin(&card->dapm, pin); + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + ret = __snd_soc_dapm_set_pin(&card->dapm, pin, + !!ucontrol->value.integer.value[0]); + mutex_unlock(&card->dapm_mutex); snd_soc_dapm_sync(&card->dapm); - return 0; + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_pin_switch); @@ -3803,7 +3844,7 @@ static int snd_soc_dapm_dai_link_put(struct snd_kcontrol *kcontrol, w->params_select = ucontrol->value.enumerated.item[0]; - return 0; + return 1; } int snd_soc_dapm_new_pcm(struct snd_soc_card *card, diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 500f98c730b9..d5ef627e93be 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -837,7 +837,7 @@ int snd_soc_get_xr_sx(struct snd_kcontrol *kcontrol, unsigned int regbase = mc->regbase; unsigned int regcount = mc->regcount; unsigned int regwshift = component->val_bytes * BITS_PER_BYTE; - unsigned int regwmask = (1<<regwshift)-1; + unsigned int regwmask = (1UL<<regwshift)-1; unsigned int invert = mc->invert; unsigned long mask = (1UL<<mc->nbits)-1; long min = mc->min; @@ -886,7 +886,7 @@ int snd_soc_put_xr_sx(struct snd_kcontrol *kcontrol, unsigned int regbase = mc->regbase; unsigned int regcount = mc->regcount; unsigned int regwshift = component->val_bytes * BITS_PER_BYTE; - unsigned int regwmask = (1<<regwshift)-1; + unsigned int regwmask = (1UL<<regwshift)-1; unsigned int invert = mc->invert; unsigned long mask = (1UL<<mc->nbits)-1; long max = mc->max; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index e75822dd9930..e995e96ab903 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2048,7 +2048,8 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, switch (cmd) { case SNDRV_PCM_TRIGGER_START: if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && - (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); @@ -2078,7 +2079,8 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; case SNDRV_PCM_TRIGGER_STOP: - if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) @@ -2170,6 +2172,7 @@ static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_DRAIN: ret = dpcm_dai_trigger_fe_be(substream, cmd, true); break; case SNDRV_PCM_TRIGGER_STOP: @@ -2187,6 +2190,7 @@ static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_DRAIN: ret = dpcm_dai_trigger_fe_be(substream, cmd, false); break; case SNDRV_PCM_TRIGGER_STOP: diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 1a912f72bddd..0fbe50502699 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -421,7 +421,7 @@ static int soc_tplg_add_kcontrol(struct soc_tplg *tplg, struct snd_soc_component *comp = tplg->comp; return soc_tplg_add_dcontrol(comp->card->snd_card, - comp->dev, k, NULL, comp, kcontrol); + comp->dev, k, comp->name_prefix, comp, kcontrol); } /* remove a mixer kcontrol */ @@ -1954,7 +1954,9 @@ static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg, _pcm = pcm; } else { abi_match = false; - pcm_new_ver(tplg, pcm, &_pcm); + ret = pcm_new_ver(tplg, pcm, &_pcm); + if (ret < 0) + return ret; } /* create the FE DAIs and DAI links */ @@ -2583,6 +2585,7 @@ EXPORT_SYMBOL_GPL(snd_soc_tplg_widget_remove_all); /* remove dynamic controls from the component driver */ int snd_soc_tplg_component_remove(struct snd_soc_component *comp, u32 index) { + struct snd_card *card = comp->card->snd_card; struct snd_soc_dobj *dobj, *next_dobj; int pass = SOC_TPLG_PASS_END; @@ -2590,6 +2593,7 @@ int snd_soc_tplg_component_remove(struct snd_soc_component *comp, u32 index) while (pass >= SOC_TPLG_PASS_START) { /* remove mixer controls */ + down_write(&card->controls_rwsem); list_for_each_entry_safe(dobj, next_dobj, &comp->dobj_list, list) { @@ -2623,6 +2627,7 @@ int snd_soc_tplg_component_remove(struct snd_soc_component *comp, u32 index) break; } } + up_write(&card->controls_rwsem); pass--; } diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index baa9007464ed..700779ca82d0 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -1199,6 +1199,7 @@ static struct snd_soc_card *sun4i_codec_create_card(struct device *dev) return ERR_PTR(-ENOMEM); card->dev = dev; + card->owner = THIS_MODULE; card->name = "sun4i-codec"; card->dapm_widgets = sun4i_codec_card_dapm_widgets; card->num_dapm_widgets = ARRAY_SIZE(sun4i_codec_card_dapm_widgets); @@ -1231,6 +1232,7 @@ static struct snd_soc_card *sun6i_codec_create_card(struct device *dev) return ERR_PTR(-ENOMEM); card->dev = dev; + card->owner = THIS_MODULE; card->name = "A31 Audio Codec"; card->dapm_widgets = sun6i_codec_card_dapm_widgets; card->num_dapm_widgets = ARRAY_SIZE(sun6i_codec_card_dapm_widgets); @@ -1284,6 +1286,7 @@ static struct snd_soc_card *sun8i_a23_codec_create_card(struct device *dev) return ERR_PTR(-ENOMEM); card->dev = dev; + card->owner = THIS_MODULE; card->name = "A23 Audio Codec"; card->dapm_widgets = sun6i_codec_card_dapm_widgets; card->num_dapm_widgets = ARRAY_SIZE(sun6i_codec_card_dapm_widgets); @@ -1322,6 +1325,7 @@ static struct snd_soc_card *sun8i_h3_codec_create_card(struct device *dev) return ERR_PTR(-ENOMEM); card->dev = dev; + card->owner = THIS_MODULE; card->name = "H3 Audio Codec"; card->dapm_widgets = sun6i_codec_card_dapm_widgets; card->num_dapm_widgets = ARRAY_SIZE(sun6i_codec_card_dapm_widgets); @@ -1360,6 +1364,7 @@ static struct snd_soc_card *sun8i_v3s_codec_create_card(struct device *dev) return ERR_PTR(-ENOMEM); card->dev = dev; + card->owner = THIS_MODULE; card->name = "V3s Audio Codec"; card->dapm_widgets = sun6i_codec_card_dapm_widgets; card->num_dapm_widgets = ARRAY_SIZE(sun6i_codec_card_dapm_widgets); diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c index 43679aeeb12b..88e838ac937d 100644 --- a/sound/soc/tegra/tegra30_ahub.c +++ b/sound/soc/tegra/tegra30_ahub.c @@ -655,8 +655,10 @@ static int tegra30_ahub_resume(struct device *dev) int ret; ret = pm_runtime_get_sync(dev); - if (ret < 0) + if (ret < 0) { + pm_runtime_put(dev); return ret; + } ret = regcache_sync(ahub->regmap_ahub); ret |= regcache_sync(ahub->regmap_apbif); pm_runtime_put(dev); diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c index 0b176ea24914..bf155c5092f0 100644 --- a/sound/soc/tegra/tegra30_i2s.c +++ b/sound/soc/tegra/tegra30_i2s.c @@ -551,8 +551,10 @@ static int tegra30_i2s_resume(struct device *dev) int ret; ret = pm_runtime_get_sync(dev); - if (ret < 0) + if (ret < 0) { + pm_runtime_put(dev); return ret; + } ret = regcache_sync(i2s->regmap); pm_runtime_put(dev); diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index 5197d6b18cb6..f9536876223f 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -137,6 +137,7 @@ static struct snd_soc_dai_link tegra_alc5632_dai = { static struct snd_soc_card snd_soc_tegra_alc5632 = { .name = "tegra-alc5632", + .driver_name = "tegra", .owner = THIS_MODULE, .dai_link = &tegra_alc5632_dai, .num_links = 1, diff --git a/sound/soc/tegra/tegra_max98090.c b/sound/soc/tegra/tegra_max98090.c index cf142e2c7bd7..10998d703dcd 100644 --- a/sound/soc/tegra/tegra_max98090.c +++ b/sound/soc/tegra/tegra_max98090.c @@ -188,6 +188,7 @@ static struct snd_soc_dai_link tegra_max98090_dai = { static struct snd_soc_card snd_soc_tegra_max98090 = { .name = "tegra-max98090", + .driver_name = "tegra", .owner = THIS_MODULE, .dai_link = &tegra_max98090_dai, .num_links = 1, diff --git a/sound/soc/tegra/tegra_rt5640.c b/sound/soc/tegra/tegra_rt5640.c index fc81b48aa9d6..e0cbe85b6d46 100644 --- a/sound/soc/tegra/tegra_rt5640.c +++ b/sound/soc/tegra/tegra_rt5640.c @@ -138,6 +138,7 @@ static struct snd_soc_dai_link tegra_rt5640_dai = { static struct snd_soc_card snd_soc_tegra_rt5640 = { .name = "tegra-rt5640", + .driver_name = "tegra", .owner = THIS_MODULE, .dai_link = &tegra_rt5640_dai, .num_links = 1, diff --git a/sound/soc/tegra/tegra_rt5677.c b/sound/soc/tegra/tegra_rt5677.c index 0e4805c7b4ca..50e5f9769eed 100644 --- a/sound/soc/tegra/tegra_rt5677.c +++ b/sound/soc/tegra/tegra_rt5677.c @@ -181,6 +181,7 @@ static struct snd_soc_dai_link tegra_rt5677_dai = { static struct snd_soc_card snd_soc_tegra_rt5677 = { .name = "tegra-rt5677", + .driver_name = "tegra", .owner = THIS_MODULE, .dai_link = &tegra_rt5677_dai, .num_links = 1, diff --git a/sound/soc/tegra/tegra_sgtl5000.c b/sound/soc/tegra/tegra_sgtl5000.c index 901457da25ec..e6cbc89eaa92 100644 --- a/sound/soc/tegra/tegra_sgtl5000.c +++ b/sound/soc/tegra/tegra_sgtl5000.c @@ -103,6 +103,7 @@ static struct snd_soc_dai_link tegra_sgtl5000_dai = { static struct snd_soc_card snd_soc_tegra_sgtl5000 = { .name = "tegra-sgtl5000", + .driver_name = "tegra", .owner = THIS_MODULE, .dai_link = &tegra_sgtl5000_dai, .num_links = 1, diff --git a/sound/soc/tegra/tegra_wm8753.c b/sound/soc/tegra/tegra_wm8753.c index 23a810e3bacc..3fa0e991308a 100644 --- a/sound/soc/tegra/tegra_wm8753.c +++ b/sound/soc/tegra/tegra_wm8753.c @@ -110,6 +110,7 @@ static struct snd_soc_dai_link tegra_wm8753_dai = { static struct snd_soc_card snd_soc_tegra_wm8753 = { .name = "tegra-wm8753", + .driver_name = "tegra", .owner = THIS_MODULE, .dai_link = &tegra_wm8753_dai, .num_links = 1, diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 18bdae59a4df..161e53029ae8 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -222,6 +222,7 @@ static struct snd_soc_dai_link tegra_wm8903_dai = { static struct snd_soc_card snd_soc_tegra_wm8903 = { .name = "tegra-wm8903", + .driver_name = "tegra", .owner = THIS_MODULE, .dai_link = &tegra_wm8903_dai, .num_links = 1, diff --git a/sound/soc/tegra/tegra_wm9712.c b/sound/soc/tegra/tegra_wm9712.c index 864a3345972e..7175e6eea911 100644 --- a/sound/soc/tegra/tegra_wm9712.c +++ b/sound/soc/tegra/tegra_wm9712.c @@ -59,6 +59,7 @@ static struct snd_soc_dai_link tegra_wm9712_dai = { static struct snd_soc_card snd_soc_tegra_wm9712 = { .name = "tegra-wm9712", + .driver_name = "tegra", .owner = THIS_MODULE, .dai_link = &tegra_wm9712_dai, .num_links = 1, diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c index 99bcdd979eb2..47ef6d6f4ae1 100644 --- a/sound/soc/tegra/trimslice.c +++ b/sound/soc/tegra/trimslice.c @@ -103,6 +103,7 @@ static struct snd_soc_dai_link trimslice_tlv320aic23_dai = { static struct snd_soc_card snd_soc_trimslice = { .name = "tegra-trimslice", + .driver_name = "tegra", .owner = THIS_MODULE, .dai_link = &trimslice_tlv320aic23_dai, .num_links = 1, diff --git a/sound/synth/emux/emux.c b/sound/synth/emux/emux.c index b9981e8c0027..82b587afa615 100644 --- a/sound/synth/emux/emux.c +++ b/sound/synth/emux/emux.c @@ -101,7 +101,7 @@ int snd_emux_register(struct snd_emux *emu, struct snd_card *card, int index, ch emu->name = kstrdup(name, GFP_KERNEL); emu->voices = kcalloc(emu->max_voices, sizeof(struct snd_emux_voice), GFP_KERNEL); - if (emu->voices == NULL) + if (emu->name == NULL || emu->voices == NULL) return -ENOMEM; /* create soundfont list */ diff --git a/sound/usb/6fire/comm.c b/sound/usb/6fire/comm.c index 161215d78d95..f29c115b9d56 100644 --- a/sound/usb/6fire/comm.c +++ b/sound/usb/6fire/comm.c @@ -99,7 +99,7 @@ static int usb6fire_comm_send_buffer(u8 *buffer, struct usb_device *dev) int actual_len; ret = usb_interrupt_msg(dev, usb_sndintpipe(dev, COMM_EP), - buffer, buffer[1] + 2, &actual_len, HZ); + buffer, buffer[1] + 2, &actual_len, 1000); if (ret < 0) return ret; else if (actual_len != buffer[1] + 2) diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c index 9520b4cd7038..7a89111041ed 100644 --- a/sound/usb/6fire/firmware.c +++ b/sound/usb/6fire/firmware.c @@ -166,7 +166,7 @@ static int usb6fire_fw_ezusb_write(struct usb_device *device, ret = usb_control_msg(device, usb_sndctrlpipe(device, 0), type, USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_DEVICE, - value, 0, data, len, HZ); + value, 0, data, len, 1000); if (ret < 0) return ret; else if (ret != len) @@ -179,7 +179,7 @@ static int usb6fire_fw_ezusb_read(struct usb_device *device, { int ret = usb_control_msg(device, usb_rcvctrlpipe(device, 0), type, USB_DIR_IN | USB_TYPE_VENDOR | USB_RECIP_DEVICE, value, - 0, data, len, HZ); + 0, data, len, 1000); if (ret < 0) return ret; else if (ret != len) @@ -194,7 +194,7 @@ static int usb6fire_fw_fpga_write(struct usb_device *device, int ret; ret = usb_bulk_msg(device, usb_sndbulkpipe(device, FPGA_EP), data, len, - &actual_len, HZ); + &actual_len, 1000); if (ret < 0) return ret; else if (actual_len != len) diff --git a/sound/usb/card.c b/sound/usb/card.c index 4169c71f8a32..1a1cb73360a4 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -183,9 +183,8 @@ static int snd_usb_create_stream(struct snd_usb_audio *chip, int ctrlif, int int ctrlif, interface); return -EINVAL; } - usb_driver_claim_interface(&usb_audio_driver, iface, (void *)-1L); - - return 0; + return usb_driver_claim_interface(&usb_audio_driver, iface, + USB_AUDIO_IFACE_UNUSED); } if ((altsd->bInterfaceClass != USB_CLASS_AUDIO && @@ -205,7 +204,8 @@ static int snd_usb_create_stream(struct snd_usb_audio *chip, int ctrlif, int int if (! snd_usb_parse_audio_interface(chip, interface)) { usb_set_interface(dev, interface, 0); /* reset the current interface */ - usb_driver_claim_interface(&usb_audio_driver, iface, (void *)-1L); + return usb_driver_claim_interface(&usb_audio_driver, iface, + USB_AUDIO_IFACE_UNUSED); } return 0; @@ -665,7 +665,7 @@ static void usb_audio_disconnect(struct usb_interface *intf) struct snd_card *card; struct list_head *p; - if (chip == (void *)-1L) + if (chip == USB_AUDIO_IFACE_UNUSED) return; card = chip->card; @@ -765,12 +765,9 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message) struct usb_mixer_interface *mixer; struct list_head *p; - if (chip == (void *)-1L) + if (chip == USB_AUDIO_IFACE_UNUSED) return 0; - chip->autosuspended = !!PMSG_IS_AUTO(message); - if (!chip->autosuspended) - snd_power_change_state(chip->card, SNDRV_CTL_POWER_D3hot); if (!chip->num_suspended_intf++) { list_for_each_entry(as, &chip->pcm_list, list) { snd_pcm_suspend_all(as->pcm); @@ -783,6 +780,11 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message) snd_usb_mixer_suspend(mixer); } + if (!PMSG_IS_AUTO(message) && !chip->system_suspend) { + snd_power_change_state(chip->card, SNDRV_CTL_POWER_D3hot); + chip->system_suspend = chip->num_suspended_intf; + } + return 0; } @@ -793,12 +795,13 @@ static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume) struct list_head *p; int err = 0; - if (chip == (void *)-1L) - return 0; - if (--chip->num_suspended_intf) + if (chip == USB_AUDIO_IFACE_UNUSED) return 0; atomic_inc(&chip->active); /* avoid autopm */ + if (chip->num_suspended_intf > 1) + goto out; + /* * ALSA leaves material resumption to user space * we just notify and restart the mixers @@ -813,9 +816,12 @@ static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume) snd_usbmidi_resume(p); } - if (!chip->autosuspended) + out: + if (chip->num_suspended_intf == chip->system_suspend) { snd_power_change_state(chip->card, SNDRV_CTL_POWER_D0); - chip->autosuspended = 0; + chip->system_suspend = 0; + } + chip->num_suspended_intf--; err_out: atomic_dec(&chip->active); /* allow autopm after this point */ diff --git a/sound/usb/card.h b/sound/usb/card.h index ed87cc83eb47..0cde519bfa42 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -126,6 +126,7 @@ struct snd_usb_substream { unsigned int tx_length_quirk:1; /* add length specifier to transfers */ unsigned int fmt_type; /* USB audio format type (1-3) */ unsigned int pkt_offset_adj; /* Bytes to drop from beginning of packets (for non-compliant devices) */ + unsigned int stream_offset_adj; /* Bytes to drop from beginning of stream (for non-compliant devices) */ unsigned int running: 1; /* running status */ diff --git a/sound/usb/clock.c b/sound/usb/clock.c index eb3396ffba4c..70e74895b113 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -327,6 +327,12 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface, } crate = data[0] | (data[1] << 8) | (data[2] << 16); + if (!crate) { + dev_info(&dev->dev, "failed to read current rate; disabling the check\n"); + chip->sample_rate_read_error = 3; /* three strikes, see above */ + return 0; + } + if (crate != rate) { dev_warn(&dev->dev, "current rate %d is different from the runtime rate %d\n", crate, rate); // runtime->rate = crate; diff --git a/sound/usb/format.c b/sound/usb/format.c index 2c44386e5569..56b5baee6552 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -52,6 +52,8 @@ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip, case UAC_VERSION_1: default: { struct uac_format_type_i_discrete_descriptor *fmt = _fmt; + if (format >= 64) + return 0; /* invalid format */ sample_width = fmt->bBitResolution; sample_bytes = fmt->bSubframeSize; format = 1 << format; @@ -187,9 +189,11 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof continue; /* C-Media CM6501 mislabels its 96 kHz altsetting */ /* Terratec Aureon 7.1 USB C-Media 6206, too */ + /* Ozone Z90 USB C-Media, too */ if (rate == 48000 && nr_rates == 1 && (chip->usb_id == USB_ID(0x0d8c, 0x0201) || chip->usb_id == USB_ID(0x0d8c, 0x0102) || + chip->usb_id == USB_ID(0x0d8c, 0x0078) || chip->usb_id == USB_ID(0x0ccd, 0x00b1)) && fp->altsetting == 5 && fp->maxpacksize == 392) rate = 96000; @@ -221,6 +225,52 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof } /* + * Many Focusrite devices supports a limited set of sampling rates per + * altsetting. Maximum rate is exposed in the last 4 bytes of Format Type + * descriptor which has a non-standard bLength = 10. + */ +static bool focusrite_valid_sample_rate(struct snd_usb_audio *chip, + struct audioformat *fp, + unsigned int rate) +{ + struct usb_interface *iface; + struct usb_host_interface *alts; + unsigned char *fmt; + unsigned int max_rate; + + iface = usb_ifnum_to_if(chip->dev, fp->iface); + if (!iface) + return true; + + alts = &iface->altsetting[fp->altset_idx]; + fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen, + NULL, UAC_FORMAT_TYPE); + if (!fmt) + return true; + + if (fmt[0] == 10) { /* bLength */ + max_rate = combine_quad(&fmt[6]); + + /* Validate max rate */ + if (max_rate != 48000 && + max_rate != 96000 && + max_rate != 192000 && + max_rate != 384000) { + + usb_audio_info(chip, + "%u:%d : unexpected max rate: %u\n", + fp->iface, fp->altsetting, max_rate); + + return true; + } + + return rate <= max_rate; + } + + return true; +} + +/* * Helper function to walk the array of sample rate triplets reported by * the device. The problem is that we need to parse whole array first to * get to know how many sample rates we have to expect. @@ -256,6 +306,11 @@ static int parse_uac2_sample_rate_range(struct snd_usb_audio *chip, } for (rate = min; rate <= max; rate += res) { + /* Filter out invalid rates on Focusrite devices */ + if (USB_ID_VENDOR(chip->usb_id) == 0x1235 && + !focusrite_valid_sample_rate(chip, fp, rate)) + goto skip_rate; + if (fp->rate_table) fp->rate_table[nr_rates] = rate; if (!fp->rate_min || rate < fp->rate_min) @@ -270,6 +325,7 @@ static int parse_uac2_sample_rate_range(struct snd_usb_audio *chip, break; } +skip_rate: /* avoid endless loop */ if (res == 0) break; diff --git a/sound/usb/line6/capture.c b/sound/usb/line6/capture.c index 7c812565f90d..a65a82d5791d 100644 --- a/sound/usb/line6/capture.c +++ b/sound/usb/line6/capture.c @@ -291,6 +291,8 @@ int line6_create_audio_in_urbs(struct snd_line6_pcm *line6pcm) urb->interval = LINE6_ISO_INTERVAL; urb->error_count = 0; urb->complete = audio_in_callback; + if (usb_urb_ep_type_check(urb)) + return -EINVAL; } return 0; diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c index bf4eacc53a7d..c629a2bf6d2c 100644 --- a/sound/usb/line6/driver.c +++ b/sound/usb/line6/driver.c @@ -117,12 +117,12 @@ static int line6_send_raw_message(struct usb_line6 *line6, const char *buffer, retval = usb_interrupt_msg(line6->usbdev, usb_sndintpipe(line6->usbdev, properties->ep_ctrl_w), (char *)frag_buf, frag_size, - &partial, LINE6_TIMEOUT * HZ); + &partial, LINE6_TIMEOUT); } else { retval = usb_bulk_msg(line6->usbdev, usb_sndbulkpipe(line6->usbdev, properties->ep_ctrl_w), (char *)frag_buf, frag_size, - &partial, LINE6_TIMEOUT * HZ); + &partial, LINE6_TIMEOUT); } if (retval) { @@ -358,7 +358,7 @@ int line6_read_data(struct usb_line6 *line6, unsigned address, void *data, ret = usb_control_msg(usbdev, usb_sndctrlpipe(usbdev, 0), 0x67, USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_OUT, (datalen << 8) | 0x21, address, - NULL, 0, LINE6_TIMEOUT * HZ); + NULL, 0, LINE6_TIMEOUT); if (ret < 0) { dev_err(line6->ifcdev, "read request failed (error %d)\n", ret); @@ -373,7 +373,7 @@ int line6_read_data(struct usb_line6 *line6, unsigned address, void *data, USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_IN, 0x0012, 0x0000, len, 1, - LINE6_TIMEOUT * HZ); + LINE6_TIMEOUT); if (ret < 0) { dev_err(line6->ifcdev, "receive length failed (error %d)\n", ret); @@ -401,7 +401,7 @@ int line6_read_data(struct usb_line6 *line6, unsigned address, void *data, ret = usb_control_msg(usbdev, usb_rcvctrlpipe(usbdev, 0), 0x67, USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_IN, 0x0013, 0x0000, data, datalen, - LINE6_TIMEOUT * HZ); + LINE6_TIMEOUT); if (ret < 0) dev_err(line6->ifcdev, "read failed (error %d)\n", ret); @@ -433,7 +433,7 @@ int line6_write_data(struct usb_line6 *line6, unsigned address, void *data, ret = usb_control_msg(usbdev, usb_sndctrlpipe(usbdev, 0), 0x67, USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_OUT, 0x0022, address, data, datalen, - LINE6_TIMEOUT * HZ); + LINE6_TIMEOUT); if (ret < 0) { dev_err(line6->ifcdev, @@ -449,7 +449,7 @@ int line6_write_data(struct usb_line6 *line6, unsigned address, void *data, USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_IN, 0x0012, 0x0000, - status, 1, LINE6_TIMEOUT * HZ); + status, 1, LINE6_TIMEOUT); if (ret < 0) { dev_err(line6->ifcdev, @@ -698,6 +698,10 @@ static int line6_init_cap_control(struct usb_line6 *line6) line6->buffer_message = kmalloc(LINE6_MIDI_MESSAGE_MAXLEN, GFP_KERNEL); if (!line6->buffer_message) return -ENOMEM; + + ret = line6_init_midi(line6); + if (ret < 0) + return ret; } else { ret = line6_hwdep_init(line6); if (ret < 0) diff --git a/sound/usb/line6/driver.h b/sound/usb/line6/driver.h index dc97895547be..80598698d706 100644 --- a/sound/usb/line6/driver.h +++ b/sound/usb/line6/driver.h @@ -31,7 +31,7 @@ #define LINE6_FALLBACK_INTERVAL 10 #define LINE6_FALLBACK_MAXPACKETSIZE 16 -#define LINE6_TIMEOUT 1 +#define LINE6_TIMEOUT 1000 #define LINE6_BUFSIZE_LISTEN 64 #define LINE6_MIDI_MESSAGE_MAXLEN 256 diff --git a/sound/usb/line6/playback.c b/sound/usb/line6/playback.c index 812d18191e01..1736eb3ee98e 100644 --- a/sound/usb/line6/playback.c +++ b/sound/usb/line6/playback.c @@ -436,6 +436,8 @@ int line6_create_audio_out_urbs(struct snd_line6_pcm *line6pcm) urb->interval = LINE6_ISO_INTERVAL; urb->error_count = 0; urb->complete = audio_out_callback; + if (usb_urb_ep_type_check(urb)) + return -EINVAL; } return 0; diff --git a/sound/usb/line6/pod.c b/sound/usb/line6/pod.c index 358224cc5638..73e6dc7d6314 100644 --- a/sound/usb/line6/pod.c +++ b/sound/usb/line6/pod.c @@ -421,11 +421,6 @@ static int pod_init(struct usb_line6 *line6, if (err < 0) return err; - /* initialize MIDI subsystem: */ - err = line6_init_midi(line6); - if (err < 0) - return err; - /* initialize PCM subsystem: */ err = line6_init_pcm(line6, &pod_pcm_properties); if (err < 0) diff --git a/sound/usb/line6/podhd.c b/sound/usb/line6/podhd.c index 1513fbaf70c2..b5573eb49cb4 100644 --- a/sound/usb/line6/podhd.c +++ b/sound/usb/line6/podhd.c @@ -235,7 +235,7 @@ static int podhd_dev_start(struct usb_line6_podhd *pod) ret = usb_control_msg(usbdev, usb_sndctrlpipe(usbdev, 0), 0x67, USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_OUT, 0x11, 0, - NULL, 0, LINE6_TIMEOUT * HZ); + NULL, 0, LINE6_TIMEOUT); if (ret < 0) { dev_err(pod->line6.ifcdev, "read request failed (error %d)\n", ret); goto exit; @@ -245,7 +245,7 @@ static int podhd_dev_start(struct usb_line6_podhd *pod) ret = usb_control_msg(usbdev, usb_rcvctrlpipe(usbdev, 0), 0x67, USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_IN, 0x11, 0x0, - init_bytes, 3, LINE6_TIMEOUT * HZ); + init_bytes, 3, LINE6_TIMEOUT); if (ret < 0) { dev_err(pod->line6.ifcdev, "receive length failed (error %d)\n", ret); @@ -265,7 +265,7 @@ static int podhd_dev_start(struct usb_line6_podhd *pod) USB_REQ_SET_FEATURE, USB_TYPE_STANDARD | USB_RECIP_DEVICE | USB_DIR_OUT, 1, 0, - NULL, 0, LINE6_TIMEOUT * HZ); + NULL, 0, LINE6_TIMEOUT); exit: kfree(init_bytes); return ret; diff --git a/sound/usb/line6/toneport.c b/sound/usb/line6/toneport.c index 4bdedfa87487..a4fc8dc2baf3 100644 --- a/sound/usb/line6/toneport.c +++ b/sound/usb/line6/toneport.c @@ -133,7 +133,7 @@ static int toneport_send_cmd(struct usb_device *usbdev, int cmd1, int cmd2) ret = usb_control_msg(usbdev, usb_sndctrlpipe(usbdev, 0), 0x67, USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_OUT, - cmd1, cmd2, NULL, 0, LINE6_TIMEOUT * HZ); + cmd1, cmd2, NULL, 0, LINE6_TIMEOUT); if (ret < 0) { dev_err(&usbdev->dev, "send failed (error %d)\n", ret); diff --git a/sound/usb/line6/variax.c b/sound/usb/line6/variax.c index 0c4512d0382e..a911cff0cec8 100644 --- a/sound/usb/line6/variax.c +++ b/sound/usb/line6/variax.c @@ -217,7 +217,6 @@ static int variax_init(struct usb_line6 *line6, const struct usb_device_id *id) { struct usb_line6_variax *variax = (struct usb_line6_variax *) line6; - int err; line6->process_message = line6_variax_process_message; line6->disconnect = line6_variax_disconnect; @@ -233,11 +232,6 @@ static int variax_init(struct usb_line6 *line6, if (variax->buffer_activate == NULL) return -ENOMEM; - /* initialize MIDI subsystem: */ - err = line6_init_midi(&variax->line6); - if (err < 0) - return err; - /* initiate startup procedure: */ variax_startup1(variax); return 0; diff --git a/sound/usb/midi.c b/sound/usb/midi.c index a92e2b2a91ec..5f5a6b7ef1cf 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -1477,6 +1477,8 @@ void snd_usbmidi_disconnect(struct list_head *p) spin_unlock_irq(&umidi->disc_lock); up_write(&umidi->disc_rwsem); + del_timer_sync(&umidi->error_timer); + for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i) { struct snd_usb_midi_endpoint *ep = &umidi->endpoints[i]; if (ep->out) @@ -1503,7 +1505,6 @@ void snd_usbmidi_disconnect(struct list_head *p) ep->in = NULL; } } - del_timer_sync(&umidi->error_timer); } EXPORT_SYMBOL(snd_usbmidi_disconnect); @@ -1804,6 +1805,28 @@ static int snd_usbmidi_create_endpoints(struct snd_usb_midi *umidi, return 0; } +static struct usb_ms_endpoint_descriptor *find_usb_ms_endpoint_descriptor( + struct usb_host_endpoint *hostep) +{ + unsigned char *extra = hostep->extra; + int extralen = hostep->extralen; + + while (extralen > 3) { + struct usb_ms_endpoint_descriptor *ms_ep = + (struct usb_ms_endpoint_descriptor *)extra; + + if (ms_ep->bLength > 3 && + ms_ep->bDescriptorType == USB_DT_CS_ENDPOINT && + ms_ep->bDescriptorSubtype == UAC_MS_GENERAL) + return ms_ep; + if (!extra[0]) + break; + extralen -= extra[0]; + extra += extra[0]; + } + return NULL; +} + /* * Returns MIDIStreaming device capabilities. */ @@ -1841,11 +1864,14 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi *umidi, ep = get_ep_desc(hostep); if (!usb_endpoint_xfer_bulk(ep) && !usb_endpoint_xfer_int(ep)) continue; - ms_ep = (struct usb_ms_endpoint_descriptor *)hostep->extra; - if (hostep->extralen < 4 || - ms_ep->bLength < 4 || - ms_ep->bDescriptorType != USB_DT_CS_ENDPOINT || - ms_ep->bDescriptorSubtype != UAC_MS_GENERAL) + ms_ep = find_usb_ms_endpoint_descriptor(hostep); + if (!ms_ep) + continue; + if (ms_ep->bLength <= sizeof(*ms_ep)) + continue; + if (ms_ep->bNumEmbMIDIJack > 0x10) + continue; + if (ms_ep->bLength < sizeof(*ms_ep) + ms_ep->bNumEmbMIDIJack) continue; if (usb_endpoint_dir_out(ep)) { if (endpoints[epidx].out_ep) { @@ -2099,6 +2125,8 @@ static int snd_usbmidi_detect_roland(struct snd_usb_midi *umidi, cs_desc[1] == USB_DT_CS_INTERFACE && cs_desc[2] == 0xf1 && cs_desc[3] == 0x02) { + if (cs_desc[4] > 0x10 || cs_desc[5] > 0x10) + continue; endpoint->in_cables = (1 << cs_desc[4]) - 1; endpoint->out_cables = (1 << cs_desc[5]) - 1; return snd_usbmidi_detect_endpoints(umidi, endpoint, 1); @@ -2260,16 +2288,22 @@ void snd_usbmidi_input_stop(struct list_head *p) } EXPORT_SYMBOL(snd_usbmidi_input_stop); -static void snd_usbmidi_input_start_ep(struct snd_usb_midi_in_endpoint *ep) +static void snd_usbmidi_input_start_ep(struct snd_usb_midi *umidi, + struct snd_usb_midi_in_endpoint *ep) { unsigned int i; + unsigned long flags; if (!ep) return; for (i = 0; i < INPUT_URBS; ++i) { struct urb *urb = ep->urbs[i]; - urb->dev = ep->umidi->dev; - snd_usbmidi_submit_urb(urb, GFP_KERNEL); + spin_lock_irqsave(&umidi->disc_lock, flags); + if (!atomic_read(&urb->use_count)) { + urb->dev = ep->umidi->dev; + snd_usbmidi_submit_urb(urb, GFP_ATOMIC); + } + spin_unlock_irqrestore(&umidi->disc_lock, flags); } } @@ -2285,7 +2319,7 @@ void snd_usbmidi_input_start(struct list_head *p) if (umidi->input_running || !umidi->opened[1]) return; for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i) - snd_usbmidi_input_start_ep(umidi->endpoints[i].in); + snd_usbmidi_input_start_ep(umidi, umidi->endpoints[i].in); umidi->input_running = 1; } EXPORT_SYMBOL(snd_usbmidi_input_start); diff --git a/sound/usb/misc/ua101.c b/sound/usb/misc/ua101.c index 386fbfd5c617..1aeddab02aad 100644 --- a/sound/usb/misc/ua101.c +++ b/sound/usb/misc/ua101.c @@ -1032,7 +1032,7 @@ static int detect_usb_format(struct ua101 *ua) fmt_playback->bSubframeSize * ua->playback.channels; epd = &ua->intf[INTF_CAPTURE]->altsetting[1].endpoint[0].desc; - if (!usb_endpoint_is_isoc_in(epd)) { + if (!usb_endpoint_is_isoc_in(epd) || usb_endpoint_maxp(epd) == 0) { dev_err(&ua->dev->dev, "invalid capture endpoint\n"); return -ENXIO; } @@ -1040,7 +1040,7 @@ static int detect_usb_format(struct ua101 *ua) ua->capture.max_packet_bytes = usb_endpoint_maxp(epd); epd = &ua->intf[INTF_PLAYBACK]->altsetting[1].endpoint[0].desc; - if (!usb_endpoint_is_isoc_out(epd)) { + if (!usb_endpoint_is_isoc_out(epd) || usb_endpoint_maxp(epd) == 0) { dev_err(&ua->dev->dev, "invalid playback endpoint\n"); return -ENXIO; } diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index e6e4c3b9d9d3..b29a3546ab6a 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -591,8 +591,9 @@ static int check_matrix_bitmap(unsigned char *bmap, * if failed, give up and free the control instance. */ -int snd_usb_mixer_add_control(struct usb_mixer_elem_list *list, - struct snd_kcontrol *kctl) +int snd_usb_mixer_add_list(struct usb_mixer_elem_list *list, + struct snd_kcontrol *kctl, + bool is_std_info) { struct usb_mixer_interface *mixer = list->mixer; int err; @@ -605,6 +606,7 @@ int snd_usb_mixer_add_control(struct usb_mixer_elem_list *list, return err; } list->kctl = kctl; + list->is_std_info = is_std_info; list->next_id_elem = mixer->id_elems[list->id]; mixer->id_elems[list->id] = list; return 0; @@ -986,6 +988,14 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval, cval->res = 384; } break; + case USB_ID(0x0495, 0x3042): /* ESS Technology Asus USB DAC */ + if ((strstr(kctl->id.name, "Playback Volume") != NULL) || + strstr(kctl->id.name, "Capture Volume") != NULL) { + cval->min >>= 8; + cval->max = 0; + cval->res = 1; + } + break; } } @@ -2342,7 +2352,7 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) if (map->id == state.chip->usb_id) { state.map = map->map; state.selector_map = map->selector_map; - mixer->ignore_ctl_error = map->ignore_ctl_error; + mixer->ignore_ctl_error |= map->ignore_ctl_error; break; } } @@ -2395,15 +2405,23 @@ void snd_usb_mixer_notify_id(struct usb_mixer_interface *mixer, int unitid) { struct usb_mixer_elem_list *list; - for (list = mixer->id_elems[unitid]; list; list = list->next_id_elem) + for_each_mixer_elem(list, mixer, unitid) { + struct usb_mixer_elem_info *info; + + if (!list->is_std_info) + continue; + info = mixer_elem_list_to_info(list); + /* invalidate cache, so the value is read from the device */ + info->cached = 0; snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &list->kctl->id); + } } static void snd_usb_mixer_dump_cval(struct snd_info_buffer *buffer, struct usb_mixer_elem_list *list) { - struct usb_mixer_elem_info *cval = (struct usb_mixer_elem_info *)list; + struct usb_mixer_elem_info *cval = mixer_elem_list_to_info(list); static char *val_types[] = {"BOOLEAN", "INV_BOOLEAN", "S8", "U8", "S16", "U16"}; snd_iprintf(buffer, " Info: id=%i, control=%i, cmask=0x%x, " @@ -2429,8 +2447,7 @@ static void snd_usb_mixer_proc_read(struct snd_info_entry *entry, mixer->ignore_ctl_error); snd_iprintf(buffer, "Card: %s\n", chip->card->longname); for (unitid = 0; unitid < MAX_ID_ELEMS; unitid++) { - for (list = mixer->id_elems[unitid]; list; - list = list->next_id_elem) { + for_each_mixer_elem(list, mixer, unitid) { snd_iprintf(buffer, " Unit: %i\n", list->id); if (list->kctl) snd_iprintf(buffer, @@ -2460,19 +2477,21 @@ static void snd_usb_mixer_interrupt_v2(struct usb_mixer_interface *mixer, return; } - for (list = mixer->id_elems[unitid]; list; list = list->next_id_elem) + for_each_mixer_elem(list, mixer, unitid) count++; if (count == 0) return; - for (list = mixer->id_elems[unitid]; list; list = list->next_id_elem) { + for_each_mixer_elem(list, mixer, unitid) { struct usb_mixer_elem_info *info; if (!list->kctl) continue; + if (!list->is_std_info) + continue; - info = (struct usb_mixer_elem_info *)list; + info = mixer_elem_list_to_info(list); if (count > 1 && info->control != control) continue; @@ -2692,7 +2711,7 @@ int snd_usb_mixer_suspend(struct usb_mixer_interface *mixer) static int restore_mixer_value(struct usb_mixer_elem_list *list) { - struct usb_mixer_elem_info *cval = (struct usb_mixer_elem_info *)list; + struct usb_mixer_elem_info *cval = mixer_elem_list_to_info(list); int c, err, idx; if (cval->cmask) { @@ -2728,8 +2747,7 @@ int snd_usb_mixer_resume(struct usb_mixer_interface *mixer, bool reset_resume) if (reset_resume) { /* restore cached mixer values */ for (id = 0; id < MAX_ID_ELEMS; id++) { - for (list = mixer->id_elems[id]; list; - list = list->next_id_elem) { + for_each_mixer_elem(list, mixer, id) { if (list->resume) { err = list->resume(list); if (err < 0) diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index ba27f7ade670..7c824a44589b 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -49,10 +49,17 @@ struct usb_mixer_elem_list { struct usb_mixer_elem_list *next_id_elem; /* list of controls with same id */ struct snd_kcontrol *kctl; unsigned int id; + bool is_std_info; usb_mixer_elem_dump_func_t dump; usb_mixer_elem_resume_func_t resume; }; +/* iterate over mixer element list of the given unit id */ +#define for_each_mixer_elem(list, mixer, id) \ + for ((list) = (mixer)->id_elems[id]; (list); (list) = (list)->next_id_elem) +#define mixer_elem_list_to_info(list) \ + container_of(list, struct usb_mixer_elem_info, head) + struct usb_mixer_elem_info { struct usb_mixer_elem_list head; unsigned int control; /* CS or ICN (high byte) */ @@ -80,8 +87,12 @@ void snd_usb_mixer_notify_id(struct usb_mixer_interface *mixer, int unitid); int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, int request, int validx, int value_set); -int snd_usb_mixer_add_control(struct usb_mixer_elem_list *list, - struct snd_kcontrol *kctl); +int snd_usb_mixer_add_list(struct usb_mixer_elem_list *list, + struct snd_kcontrol *kctl, + bool is_std_info); + +#define snd_usb_mixer_add_control(list, kctl) \ + snd_usb_mixer_add_list(list, kctl, true) void snd_usb_mixer_elem_init_std(struct usb_mixer_elem_list *list, struct usb_mixer_interface *mixer, diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index eaa03acd4686..26ce6838e842 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -363,6 +363,14 @@ static const struct usbmix_name_map dell_alc4020_map[] = { { 0 } }; +/* Some mobos shipped with a dummy HD-audio show the invalid GET_MIN/GET_MAX + * response for Input Gain Pad (id=19, control=12). Skip it. + */ +static const struct usbmix_name_map asus_rog_map[] = { + { 19, NULL, 12 }, /* FU, Input Gain Pad */ + {} +}; + /* * Control map entries */ @@ -482,6 +490,26 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = { .id = USB_ID(0x05a7, 0x1020), .map = bose_companion5_map, }, + { /* Gigabyte TRX40 Aorus Pro WiFi */ + .id = USB_ID(0x0414, 0xa002), + .map = asus_rog_map, + }, + { /* ASUS ROG Zenith II */ + .id = USB_ID(0x0b05, 0x1916), + .map = asus_rog_map, + }, + { /* ASUS ROG Strix */ + .id = USB_ID(0x0b05, 0x1917), + .map = asus_rog_map, + }, + { /* MSI TRX40 Creator */ + .id = USB_ID(0x0db0, 0x0d64), + .map = asus_rog_map, + }, + { /* MSI TRX40 */ + .id = USB_ID(0x0db0, 0x543d), + .map = asus_rog_map, + }, { 0 } /* terminator */ }; diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index b54f7dab8372..d7878ed5ecc0 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -169,7 +169,8 @@ static int add_single_ctl_with_resume(struct usb_mixer_interface *mixer, return -ENOMEM; } kctl->private_free = snd_usb_mixer_elem_free; - return snd_usb_mixer_add_control(list, kctl); + /* don't use snd_usb_mixer_add_control() here, this is a special list element */ + return snd_usb_mixer_add_list(list, kctl, false); } /* @@ -195,6 +196,7 @@ static const struct rc_config { { USB_ID(0x041e, 0x3042), 0, 1, 1, 1, 1, 0x000d }, /* Usb X-Fi S51 */ { USB_ID(0x041e, 0x30df), 0, 1, 1, 1, 1, 0x000d }, /* Usb X-Fi S51 Pro */ { USB_ID(0x041e, 0x3237), 0, 1, 1, 1, 1, 0x000d }, /* Usb X-Fi S51 Pro */ + { USB_ID(0x041e, 0x3263), 0, 1, 1, 1, 1, 0x000d }, /* Usb X-Fi S51 Pro */ { USB_ID(0x041e, 0x3048), 2, 2, 6, 6, 2, 0x6e91 }, /* Toshiba SB0500 */ }; @@ -1171,7 +1173,7 @@ void snd_emuusb_set_samplerate(struct snd_usb_audio *chip, int unitid = 12; /* SamleRate ExtensionUnit ID */ list_for_each_entry(mixer, &chip->mixer_list, list) { - cval = (struct usb_mixer_elem_info *)mixer->id_elems[unitid]; + cval = mixer_elem_list_to_info(mixer->id_elems[unitid]); if (cval) { snd_usb_mixer_set_ctl_value(cval, UAC_SET_CUR, cval->control << 8, @@ -1520,11 +1522,15 @@ static int snd_microii_spdif_default_get(struct snd_kcontrol *kcontrol, /* use known values for that card: interface#1 altsetting#1 */ iface = usb_ifnum_to_if(chip->dev, 1); - if (!iface || iface->num_altsetting < 2) - return -EINVAL; + if (!iface || iface->num_altsetting < 2) { + err = -EINVAL; + goto end; + } alts = &iface->altsetting[1]; - if (get_iface_desc(alts)->bNumEndpoints < 1) - return -EINVAL; + if (get_iface_desc(alts)->bNumEndpoints < 1) { + err = -EINVAL; + goto end; + } ep = get_endpoint(alts, 0)->bEndpointAddress; err = snd_usb_ctl_msg(chip->dev, diff --git a/sound/usb/mixer_scarlett.c b/sound/usb/mixer_scarlett.c index c33e2378089d..4aeb9488a0c9 100644 --- a/sound/usb/mixer_scarlett.c +++ b/sound/usb/mixer_scarlett.c @@ -287,8 +287,7 @@ static int scarlett_ctl_switch_put(struct snd_kcontrol *kctl, static int scarlett_ctl_resume(struct usb_mixer_elem_list *list) { - struct usb_mixer_elem_info *elem = - container_of(list, struct usb_mixer_elem_info, head); + struct usb_mixer_elem_info *elem = mixer_elem_list_to_info(list); int i; for (i = 0; i < elem->channels; i++) @@ -447,8 +446,7 @@ static int scarlett_ctl_enum_put(struct snd_kcontrol *kctl, static int scarlett_ctl_enum_resume(struct usb_mixer_elem_list *list) { - struct usb_mixer_elem_info *elem = - container_of(list, struct usb_mixer_elem_info, head); + struct usb_mixer_elem_info *elem = mixer_elem_list_to_info(list); if (elem->cached) snd_usb_set_cur_mix_value(elem, 0, 0, *elem->cache_val); diff --git a/sound/usb/mixer_us16x08.c b/sound/usb/mixer_us16x08.c index 26ed23b18b77..7db3032e723a 100644 --- a/sound/usb/mixer_us16x08.c +++ b/sound/usb/mixer_us16x08.c @@ -617,7 +617,7 @@ static int snd_us16x08_eq_put(struct snd_kcontrol *kcontrol, static int snd_us16x08_meter_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - uinfo->count = 1; + uinfo->count = 34; uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->value.integer.max = 0x7FFF; uinfo->value.integer.min = 0; diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index ff38fca1781b..ecdbdb26164e 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -324,6 +324,7 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, struct usb_host_interface *alts; struct usb_interface *iface; unsigned int ep; + unsigned int ifnum; /* Implicit feedback sync EPs consumers are always playback EPs */ if (subs->direction != SNDRV_PCM_STREAM_PLAYBACK) @@ -332,45 +333,25 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, switch (subs->stream->chip->usb_id) { case USB_ID(0x0763, 0x2030): /* M-Audio Fast Track C400 */ case USB_ID(0x0763, 0x2031): /* M-Audio Fast Track C600 */ + case USB_ID(0x22f0, 0x0006): /* Allen&Heath Qu-16 */ ep = 0x81; - iface = usb_ifnum_to_if(dev, 3); - - if (!iface || iface->num_altsetting == 0) - return -EINVAL; - - alts = &iface->altsetting[1]; - goto add_sync_ep; - break; + ifnum = 3; + goto add_sync_ep_from_ifnum; case USB_ID(0x0763, 0x2080): /* M-Audio FastTrack Ultra */ case USB_ID(0x0763, 0x2081): ep = 0x81; - iface = usb_ifnum_to_if(dev, 2); - - if (!iface || iface->num_altsetting == 0) - return -EINVAL; - - alts = &iface->altsetting[1]; - goto add_sync_ep; - case USB_ID(0x2466, 0x8003): + ifnum = 2; + goto add_sync_ep_from_ifnum; + case USB_ID(0x2466, 0x8003): /* Fractal Audio Axe-Fx II */ ep = 0x86; - iface = usb_ifnum_to_if(dev, 2); - - if (!iface || iface->num_altsetting == 0) - return -EINVAL; - - alts = &iface->altsetting[1]; - goto add_sync_ep; - case USB_ID(0x1397, 0x0002): + ifnum = 2; + goto add_sync_ep_from_ifnum; + case USB_ID(0x1397, 0x0002): /* Behringer UFX1204 */ ep = 0x81; - iface = usb_ifnum_to_if(dev, 1); - - if (!iface || iface->num_altsetting == 0) - return -EINVAL; - - alts = &iface->altsetting[1]; - goto add_sync_ep; - + ifnum = 1; + goto add_sync_ep_from_ifnum; } + if (attr == USB_ENDPOINT_SYNC_ASYNC && altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC && altsd->bInterfaceProtocol == 2 && @@ -385,6 +366,14 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, /* No quirk */ return 0; +add_sync_ep_from_ifnum: + iface = usb_ifnum_to_if(dev, ifnum); + + if (!iface || iface->num_altsetting < 2) + return -EINVAL; + + alts = &iface->altsetting[1]; + add_sync_ep: subs->sync_endpoint = snd_usb_add_endpoint(subs->stream->chip, alts, ep, !subs->direction, @@ -1313,6 +1302,12 @@ static void retire_capture_urb(struct snd_usb_substream *subs, // continue; } bytes = urb->iso_frame_desc[i].actual_length; + if (subs->stream_offset_adj > 0) { + unsigned int adj = min(subs->stream_offset_adj, bytes); + cp += adj; + bytes -= adj; + subs->stream_offset_adj -= adj; + } frames = bytes / stride; if (!subs->txfr_quirk) bytes = frames * stride; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index c892b4d1e733..1904fc542025 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2479,6 +2479,16 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +{ + USB_DEVICE_VENDOR_SPEC(0x0944, 0x0204), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "KORG, Inc.", + /* .product_name = "ToneLab EX", */ + .ifnum = 3, + .type = QUIRK_MIDI_STANDARD_INTERFACE, + } +}, + /* AKAI devices */ { USB_DEVICE(0x09e8, 0x0062), @@ -3323,4 +3333,150 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } }, +/* + * MacroSilicon MS2109 based HDMI capture cards + * + * These claim 96kHz 1ch in the descriptors, but are actually 48kHz 2ch. + * They also need QUIRK_AUDIO_ALIGN_TRANSFER, which makes one wonder if + * they pretend to be 96kHz mono as a workaround for stereo being broken + * by that... + * + * They also have an issue with initial stream alignment that causes the + * channels to be swapped and out of phase, which is dealt with in quirks.c. + */ +{ + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .idVendor = 0x534d, + .idProduct = 0x2109, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "MacroSilicon", + .product_name = "MS2109", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = &(const struct snd_usb_audio_quirk[]) { + { + .ifnum = 2, + .type = QUIRK_AUDIO_ALIGN_TRANSFER, + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_MIXER, + }, + { + .ifnum = 3, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .channels = 2, + .iface = 3, + .altsetting = 1, + .altset_idx = 1, + .attributes = 0, + .endpoint = 0x82, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 48000, + .rate_max = 48000, + } + }, + { + .ifnum = -1 + } + } + } +}, +{ + /* + * PIONEER DJ DDJ-RB + * PCM is 4 channels out, 2 dummy channels in @ 44.1 fixed + * The feedback for the output is the dummy input. + */ + USB_DEVICE_VENDOR_SPEC(0x2b73, 0x000e), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 4, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_44100, + .rate_min = 44100, + .rate_max = 44100, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 44100 } + } + }, + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 2, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x82, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC| + USB_ENDPOINT_USAGE_IMPLICIT_FB, + .rates = SNDRV_PCM_RATE_44100, + .rate_min = 44100, + .rate_max = 44100, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 44100 } + } + }, + { + .ifnum = -1 + } + } + } +}, +{ + /* + * Sennheiser GSP670 + * Change order of interfaces loaded + */ + USB_DEVICE(0x1395, 0x0300), + .bInterfaceClass = USB_CLASS_PER_INTERFACE, + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = &(const struct snd_usb_audio_quirk[]) { + // Communication + { + .ifnum = 3, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + // Recording + { + .ifnum = 4, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + // Main + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = -1 + } + } + } +}, + #undef USB_DEVICE_VENDOR_SPEC diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 4872c27f6054..182c9de4f255 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -66,8 +66,12 @@ static int create_composite_quirk(struct snd_usb_audio *chip, if (!iface) continue; if (quirk->ifnum != probed_ifnum && - !usb_interface_claimed(iface)) - usb_driver_claim_interface(driver, iface, (void *)-1L); + !usb_interface_claimed(iface)) { + err = usb_driver_claim_interface(driver, iface, + USB_AUDIO_IFACE_UNUSED); + if (err < 0) + return err; + } } return 0; @@ -398,8 +402,12 @@ static int create_autodetect_quirks(struct snd_usb_audio *chip, continue; err = create_autodetect_quirk(chip, iface, driver); - if (err >= 0) - usb_driver_claim_interface(driver, iface, (void *)-1L); + if (err >= 0) { + err = usb_driver_claim_interface(driver, iface, + USB_AUDIO_IFACE_UNUSED); + if (err < 0) + return err; + } } return 0; @@ -1120,6 +1128,9 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, case USB_ID(0x041e, 0x3f19): /* E-Mu 0204 USB */ set_format_emu_quirk(subs, fmt); break; + case USB_ID(0x534d, 0x2109): /* MacroSilicon MS2109 */ + subs->stream_offset_adj = 2; + break; } } @@ -1152,6 +1163,8 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) case USB_ID(0x1de7, 0x0114): /* Phoenix Audio MT202pcs */ case USB_ID(0x21B4, 0x0081): /* AudioQuest DragonFly */ case USB_ID(0x2912, 0x30c8): /* Audioengine D1 */ + case USB_ID(0x413c, 0xa506): /* Dell AE515 sound bar */ + case USB_ID(0x046d, 0x084c): /* Logitech ConferenceCam Connect */ return true; } return false; @@ -1164,6 +1177,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) static bool is_itf_usb_dsd_2alts_dac(unsigned int id) { switch (id) { + case USB_ID(0x154e, 0x1002): /* Denon DCD-1500RE */ case USB_ID(0x154e, 0x1003): /* Denon DA-300USB */ case USB_ID(0x154e, 0x3005): /* Marantz HD-DAC1 */ case USB_ID(0x154e, 0x3006): /* Marantz SA-14S1 */ @@ -1318,13 +1332,15 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) mdelay(20); - /* Zoom R16/24, Logitech H650e, Jabra 550a needs a tiny delay here, - * otherwise requests like get/set frequency return as failed despite - * actually succeeding. + /* Zoom R16/24, Logitech H650e/H570e, Jabra 550a, Kingston HyperX + * needs a tiny delay here, otherwise requests like get/set + * frequency return as failed despite actually succeeding. */ if ((chip->usb_id == USB_ID(0x1686, 0x00dd) || chip->usb_id == USB_ID(0x046d, 0x0a46) || - chip->usb_id == USB_ID(0x0b0e, 0x0349)) && + chip->usb_id == USB_ID(0x046d, 0x0a56) || + chip->usb_id == USB_ID(0x0b0e, 0x0349) || + chip->usb_id == USB_ID(0x0951, 0x16ad)) && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) mdelay(1); } diff --git a/sound/usb/stream.c b/sound/usb/stream.c index d1776e5517ff..7b86bf38f10e 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -95,6 +95,7 @@ static void snd_usb_init_substream(struct snd_usb_stream *as, subs->tx_length_quirk = as->chip->tx_length_quirk; subs->speed = snd_usb_get_speed(subs->dev); subs->pkt_offset_adj = 0; + subs->stream_offset_adj = 0; snd_usb_set_pcm_ops(as->pcm, stream); @@ -184,16 +185,16 @@ static int usb_chmap_ctl_get(struct snd_kcontrol *kcontrol, struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); struct snd_usb_substream *subs = info->private_data; struct snd_pcm_chmap_elem *chmap = NULL; - int i; + int i = 0; - memset(ucontrol->value.integer.value, 0, - sizeof(ucontrol->value.integer.value)); if (subs->cur_audiofmt) chmap = subs->cur_audiofmt->chmap; if (chmap) { for (i = 0; i < chmap->channels; i++) ucontrol->value.integer.value[i] = chmap->map[i]; } + for (; i < subs->channels_max; i++) + ucontrol->value.integer.value[i] = 0; return 0; } diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 4d5c89a7ba2b..62456a806bb4 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -37,7 +37,7 @@ struct snd_usb_audio { struct usb_interface *pm_intf; u32 usb_id; struct mutex mutex; - unsigned int autosuspended:1; + unsigned int system_suspend; atomic_t active; atomic_t shutdown; atomic_t usage_count; @@ -63,6 +63,8 @@ struct snd_usb_audio { struct usb_host_interface *ctrl_intf; /* the audio control interface */ }; +#define USB_AUDIO_IFACE_UNUSED ((void *)-1L) + #define usb_audio_err(chip, fmt, args...) \ dev_err(&(chip)->dev->dev, fmt, ##args) #define usb_audio_warn(chip, fmt, args...) \ diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index f93b355756e6..2dfc0abf2e37 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -689,6 +689,8 @@ static int usX2Y_rate_set(struct usX2Ydev *usX2Y, int rate) us->submitted = 2*NOOF_SETRATE_URBS; for (i = 0; i < NOOF_SETRATE_URBS; ++i) { struct urb *urb = us->urb[i]; + if (!urb) + continue; if (urb->status) { if (!err) err = -ENODEV; |