diff options
Diffstat (limited to 'sound')
141 files changed, 1296 insertions, 520 deletions
diff --git a/sound/ac97/bus.c b/sound/ac97/bus.c index 7985dd8198b6..99e1728b52ae 100644 --- a/sound/ac97/bus.c +++ b/sound/ac97/bus.c @@ -520,7 +520,7 @@ static int ac97_bus_remove(struct device *dev) struct ac97_codec_driver *adrv = to_ac97_driver(dev->driver); int ret; - ret = pm_runtime_get_sync(dev); + ret = pm_runtime_resume_and_get(dev); if (ret < 0) return ret; diff --git a/sound/core/Makefile b/sound/core/Makefile index ee4a4a6b99ba..d123587c0fd8 100644 --- a/sound/core/Makefile +++ b/sound/core/Makefile @@ -9,7 +9,9 @@ ifneq ($(CONFIG_SND_PROC_FS),) snd-y += info.o snd-$(CONFIG_SND_OSSEMUL) += info_oss.o endif +ifneq ($(CONFIG_M68K),y) snd-$(CONFIG_ISA_DMA_API) += isadma.o +endif snd-$(CONFIG_SND_OSSEMUL) += sound_oss.o snd-$(CONFIG_SND_VMASTER) += vmaster.o snd-$(CONFIG_SND_JACK) += ctljack.o jack.o diff --git a/sound/core/control_compat.c b/sound/core/control_compat.c index d55be1db1a8a..cca3ed9b0629 100644 --- a/sound/core/control_compat.c +++ b/sound/core/control_compat.c @@ -266,6 +266,7 @@ static int copy_ctl_value_to_user(void __user *userdata, struct snd_ctl_elem_value *data, int type, int count) { + struct snd_ctl_elem_value32 __user *data32 = userdata; int i, size; if (type == SNDRV_CTL_ELEM_TYPE_BOOLEAN || @@ -282,6 +283,8 @@ static int copy_ctl_value_to_user(void __user *userdata, if (copy_to_user(valuep, data->value.bytes.data, size)) return -EFAULT; } + if (copy_to_user(&data32->id, &data->id, sizeof(data32->id))) + return -EFAULT; return 0; } diff --git a/sound/core/jack.c b/sound/core/jack.c index fb26196571a7..b00ae6f39f05 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -54,10 +54,13 @@ static int snd_jack_dev_free(struct snd_device *device) struct snd_card *card = device->card; struct snd_jack_kctl *jack_kctl, *tmp_jack_kctl; + down_write(&card->controls_rwsem); list_for_each_entry_safe(jack_kctl, tmp_jack_kctl, &jack->kctl_list, list) { list_del_init(&jack_kctl->list); snd_ctl_remove(card, jack_kctl->kctl); } + up_write(&card->controls_rwsem); + if (jack->private_free) jack->private_free(jack); @@ -220,6 +223,10 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, return -ENOMEM; jack->id = kstrdup(id, GFP_KERNEL); + if (jack->id == NULL) { + kfree(jack); + return -ENOMEM; + } /* don't creat input device for phantom jack */ if (!phantom_jack) { diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index 7eb54df5556d..50ec8b8ff68c 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -130,11 +130,13 @@ 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; } @@ -146,11 +148,13 @@ 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; } @@ -161,6 +165,7 @@ 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 { @@ -172,6 +177,7 @@ static int snd_mixer_oss_recmask(struct snd_mixer_oss_file *fmixer) result |= 1 << chn; } } + mutex_unlock(&mixer->reg_mutex); return result; } @@ -182,11 +188,12 @@ 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; @@ -201,7 +208,10 @@ 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) @@ -214,6 +224,7 @@ 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; @@ -239,6 +250,7 @@ static int snd_mixer_oss_set_recsrc(struct snd_mixer_oss_file *fmixer, int recsr } } } + mutex_unlock(&mixer->reg_mutex); return result; } @@ -250,6 +262,7 @@ 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]; @@ -257,15 +270,21 @@ 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; } @@ -278,6 +297,7 @@ 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; @@ -288,10 +308,13 @@ 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/pcm_oss.c b/sound/core/oss/pcm_oss.c index 0b03777d0111..ad4e0af2d0d0 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -147,7 +147,7 @@ snd_pcm_hw_param_value_min(const struct snd_pcm_hw_params *params, * * Return the maximum value for field PAR. */ -static unsigned int +static int snd_pcm_hw_param_value_max(const struct snd_pcm_hw_params *params, snd_pcm_hw_param_t var, int *dir) { @@ -682,18 +682,24 @@ static int snd_pcm_oss_period_size(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *oss_params, struct snd_pcm_hw_params *slave_params) { - size_t s; - size_t oss_buffer_size, oss_period_size, oss_periods; - size_t min_period_size, max_period_size; + ssize_t s; + ssize_t oss_buffer_size; + ssize_t oss_period_size, oss_periods; + ssize_t min_period_size, max_period_size; struct snd_pcm_runtime *runtime = substream->runtime; size_t oss_frame_size; oss_frame_size = snd_pcm_format_physical_width(params_format(oss_params)) * params_channels(oss_params) / 8; + oss_buffer_size = snd_pcm_hw_param_value_max(slave_params, + SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + NULL); + if (oss_buffer_size <= 0) + return -EINVAL; oss_buffer_size = snd_pcm_plug_client_size(substream, - snd_pcm_hw_param_value_max(slave_params, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, NULL)) * oss_frame_size; - if (!oss_buffer_size) + oss_buffer_size * oss_frame_size); + if (oss_buffer_size <= 0) return -EINVAL; oss_buffer_size = rounddown_pow_of_two(oss_buffer_size); if (atomic_read(&substream->mmap_count)) { @@ -730,7 +736,7 @@ static int snd_pcm_oss_period_size(struct snd_pcm_substream *substream, min_period_size = snd_pcm_plug_client_size(substream, snd_pcm_hw_param_value_min(slave_params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, NULL)); - if (min_period_size) { + if (min_period_size > 0) { min_period_size *= oss_frame_size; min_period_size = roundup_pow_of_two(min_period_size); if (oss_period_size < min_period_size) @@ -739,7 +745,7 @@ static int snd_pcm_oss_period_size(struct snd_pcm_substream *substream, max_period_size = snd_pcm_plug_client_size(substream, snd_pcm_hw_param_value_max(slave_params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, NULL)); - if (max_period_size) { + if (max_period_size > 0) { max_period_size *= oss_frame_size; max_period_size = rounddown_pow_of_two(max_period_size); if (oss_period_size > max_period_size) @@ -752,7 +758,7 @@ static int snd_pcm_oss_period_size(struct snd_pcm_substream *substream, oss_periods = substream->oss.setup.periods; s = snd_pcm_hw_param_value_max(slave_params, SNDRV_PCM_HW_PARAM_PERIODS, NULL); - if (runtime->oss.maxfrags && s > runtime->oss.maxfrags) + if (s > 0 && runtime->oss.maxfrags && s > runtime->oss.maxfrags) s = runtime->oss.maxfrags; if (oss_periods > s) oss_periods = s; @@ -768,6 +774,11 @@ static int snd_pcm_oss_period_size(struct snd_pcm_substream *substream, if (oss_period_size < 16) return -EINVAL; + + /* don't allocate too large period; 1MB period must be enough */ + if (oss_period_size > 1024 * 1024) + return -ENOMEM; + runtime->oss.period_bytes = oss_period_size; runtime->oss.period_frames = 1; runtime->oss.periods = oss_periods; @@ -878,8 +889,15 @@ static int snd_pcm_oss_change_params_locked(struct snd_pcm_substream *substream) err = -EINVAL; goto failure; } - choose_rate(substream, sparams, runtime->oss.rate); - snd_pcm_hw_param_near(substream, sparams, SNDRV_PCM_HW_PARAM_CHANNELS, runtime->oss.channels, NULL); + + err = choose_rate(substream, sparams, runtime->oss.rate); + if (err < 0) + goto failure; + err = snd_pcm_hw_param_near(substream, sparams, + SNDRV_PCM_HW_PARAM_CHANNELS, + runtime->oss.channels, NULL); + if (err < 0) + goto failure; format = snd_pcm_oss_format_from(runtime->oss.format); @@ -1032,10 +1050,9 @@ static int snd_pcm_oss_change_params_locked(struct snd_pcm_substream *substream) goto failure; } #endif - oss_period_size *= oss_frame_size; - - oss_buffer_size = oss_period_size * runtime->oss.periods; - if (oss_buffer_size < 0) { + oss_period_size = array_size(oss_period_size, oss_frame_size); + oss_buffer_size = array_size(oss_period_size, runtime->oss.periods); + if (oss_buffer_size <= 0) { err = -EINVAL; goto failure; } @@ -1946,7 +1963,7 @@ static int snd_pcm_oss_set_fragment1(struct snd_pcm_substream *substream, unsign if (runtime->oss.subdivision || runtime->oss.fragshift) return -EINVAL; fragshift = val & 0xffff; - if (fragshift >= 31) + if (fragshift >= 25) /* should be large enough */ return -EINVAL; runtime->oss.fragshift = fragshift; runtime->oss.maxfrags = (val >> 16) & 0xffff; @@ -2042,7 +2059,7 @@ static int snd_pcm_oss_set_trigger(struct snd_pcm_oss_file *pcm_oss_file, int tr int err, cmd; #ifdef OSS_DEBUG - pcm_dbg(substream->pcm, "pcm_oss: trigger = 0x%x\n", trigger); + pr_debug("pcm_oss: trigger = 0x%x\n", trigger); #endif psubstream = pcm_oss_file->streams[SNDRV_PCM_STREAM_PLAYBACK]; diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index da400da1fafe..8b7bbabeea24 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -61,7 +61,10 @@ static int snd_pcm_plugin_alloc(struct snd_pcm_plugin *plugin, snd_pcm_uframes_t } if ((width = snd_pcm_format_physical_width(format->format)) < 0) return width; - size = frames * format->channels * width; + size = array3_size(frames, format->channels, width); + /* check for too large period size once again */ + if (size > 1024 * 1024) + return -ENOMEM; if (snd_BUG_ON(size % 8)) return -ENXIO; size /= 8; diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 9a72d641743d..3561cdceaadc 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -810,7 +810,11 @@ EXPORT_SYMBOL(snd_pcm_new_internal); static void free_chmap(struct snd_pcm_str *pstr) { if (pstr->chmap_kctl) { - snd_ctl_remove(pstr->pcm->card, pstr->chmap_kctl); + struct snd_card *card = pstr->pcm->card; + + down_write(&card->controls_rwsem); + snd_ctl_remove(card, pstr->chmap_kctl); + up_write(&card->controls_rwsem); pstr->chmap_kctl = NULL; } } @@ -965,6 +969,8 @@ int snd_pcm_attach_substream(struct snd_pcm *pcm, int stream, init_waitqueue_head(&runtime->tsleep); runtime->status->state = SNDRV_PCM_STATE_OPEN; + mutex_init(&runtime->buffer_mutex); + atomic_set(&runtime->buffer_accessing, 0); substream->runtime = runtime; substream->private_data = pcm->private_data; @@ -996,6 +1002,7 @@ void snd_pcm_detach_substream(struct snd_pcm_substream *substream) substream->runtime = NULL; if (substream->timer) spin_unlock_irq(&substream->timer->lock); + mutex_destroy(&runtime->buffer_mutex); kfree(runtime); put_pid(substream->pid); substream->pid = NULL; diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 1662573a4030..1bce55533519 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1736,7 +1736,7 @@ 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; } @@ -2211,10 +2211,15 @@ snd_pcm_sframes_t __snd_pcm_lib_xfer(struct snd_pcm_substream *substream, err = -EINVAL; goto _end_unlock; } + if (!atomic_inc_unless_negative(&runtime->buffer_accessing)) { + err = -EBUSY; + goto _end_unlock; + } snd_pcm_stream_unlock_irq(substream); err = writer(substream, appl_ofs, data, offset, frames, transfer); snd_pcm_stream_lock_irq(substream); + atomic_dec(&runtime->buffer_accessing); if (err < 0) goto _end_unlock; err = pcm_accessible_state(runtime); diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c index 7600dcdf5fd4..9aea1d6fb054 100644 --- a/sound/core/pcm_memory.c +++ b/sound/core/pcm_memory.c @@ -133,19 +133,20 @@ static void snd_pcm_lib_preallocate_proc_write(struct snd_info_entry *entry, size_t size; struct snd_dma_buffer new_dmab; + mutex_lock(&substream->pcm->open_mutex); if (substream->runtime) { buffer->error = -EBUSY; - return; + goto unlock; } if (!snd_info_get_line(buffer, line, sizeof(line))) { snd_info_get_str(str, line, sizeof(str)); size = simple_strtoul(str, NULL, 10) * 1024; if ((size != 0 && size < 8192) || size > substream->dma_max) { buffer->error = -EINVAL; - return; + goto unlock; } if (substream->dma_buffer.bytes == size) - return; + goto unlock; memset(&new_dmab, 0, sizeof(new_dmab)); new_dmab.dev = substream->dma_buffer.dev; if (size > 0) { @@ -153,7 +154,7 @@ static void snd_pcm_lib_preallocate_proc_write(struct snd_info_entry *entry, substream->dma_buffer.dev.dev, size, &new_dmab) < 0) { buffer->error = -ENOMEM; - return; + goto unlock; } substream->buffer_bytes_max = size; } else { @@ -165,6 +166,8 @@ static void snd_pcm_lib_preallocate_proc_write(struct snd_info_entry *entry, } else { buffer->error = -EINVAL; } + unlock: + mutex_unlock(&substream->pcm->open_mutex); } static inline void preallocate_info_init(struct snd_pcm_substream *substream) diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c index c4eb561d2008..0956be39b035 100644 --- a/sound/core/pcm_misc.c +++ b/sound/core/pcm_misc.c @@ -423,7 +423,7 @@ int snd_pcm_format_set_silence(snd_pcm_format_t format, void *data, unsigned int return 0; width = pcm_formats[(INT)format].phys; /* physical width */ pat = pcm_formats[(INT)format].silence; - if (! width) + if (!width || !pat) return -EINVAL; /* signed or 1 byte data */ if (pcm_formats[(INT)format].signd == 1 || width <= 8) { diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 0c5b7a54ca81..57a4991fa0f3 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -630,6 +630,30 @@ static int snd_pcm_hw_params_choose(struct snd_pcm_substream *pcm, return 0; } +/* acquire buffer_mutex; if it's in r/w operation, return -EBUSY, otherwise + * block the further r/w operations + */ +static int snd_pcm_buffer_access_lock(struct snd_pcm_runtime *runtime) +{ + if (!atomic_dec_unless_positive(&runtime->buffer_accessing)) + return -EBUSY; + mutex_lock(&runtime->buffer_mutex); + return 0; /* keep buffer_mutex, unlocked by below */ +} + +/* release buffer_mutex and clear r/w access flag */ +static void snd_pcm_buffer_access_unlock(struct snd_pcm_runtime *runtime) +{ + mutex_unlock(&runtime->buffer_mutex); + atomic_inc(&runtime->buffer_accessing); +} + +#if IS_ENABLED(CONFIG_SND_PCM_OSS) +#define is_oss_stream(substream) ((substream)->oss.oss) +#else +#define is_oss_stream(substream) false +#endif + static int snd_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -641,22 +665,25 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, if (PCM_RUNTIME_CHECK(substream)) return -ENXIO; runtime = substream->runtime; + err = snd_pcm_buffer_access_lock(runtime); + if (err < 0) + return err; snd_pcm_stream_lock_irq(substream); switch (runtime->status->state) { case SNDRV_PCM_STATE_OPEN: case SNDRV_PCM_STATE_SETUP: case SNDRV_PCM_STATE_PREPARED: + if (!is_oss_stream(substream) && + atomic_read(&substream->mmap_count)) + err = -EBADFD; break; default: - snd_pcm_stream_unlock_irq(substream); - return -EBADFD; + err = -EBADFD; + break; } snd_pcm_stream_unlock_irq(substream); -#if IS_ENABLED(CONFIG_SND_PCM_OSS) - if (!substream->oss.oss) -#endif - if (atomic_read(&substream->mmap_count)) - return -EBADFD; + if (err) + goto unlock; params->rmask = ~0U; err = snd_pcm_hw_refine(substream, params); @@ -733,14 +760,19 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, if ((usecs = period_to_usecs(runtime)) >= 0) pm_qos_add_request(&substream->latency_pm_qos_req, PM_QOS_CPU_DMA_LATENCY, usecs); - return 0; + err = 0; _error: - /* hardware might be unusable from this time, - so we force application to retry to set - the correct hardware parameter settings */ - snd_pcm_set_state(substream, SNDRV_PCM_STATE_OPEN); - if (substream->ops->hw_free != NULL) - substream->ops->hw_free(substream); + if (err) { + /* hardware might be unusable from this time, + * so we force application to retry to set + * the correct hardware parameter settings + */ + snd_pcm_set_state(substream, SNDRV_PCM_STATE_OPEN); + if (substream->ops->hw_free != NULL) + substream->ops->hw_free(substream); + } + unlock: + snd_pcm_buffer_access_unlock(runtime); return err; } @@ -773,22 +805,29 @@ static int snd_pcm_hw_free(struct snd_pcm_substream *substream) if (PCM_RUNTIME_CHECK(substream)) return -ENXIO; runtime = substream->runtime; + result = snd_pcm_buffer_access_lock(runtime); + if (result < 0) + return result; snd_pcm_stream_lock_irq(substream); switch (runtime->status->state) { case SNDRV_PCM_STATE_SETUP: case SNDRV_PCM_STATE_PREPARED: + if (atomic_read(&substream->mmap_count)) + result = -EBADFD; break; default: - snd_pcm_stream_unlock_irq(substream); - return -EBADFD; + result = -EBADFD; + break; } snd_pcm_stream_unlock_irq(substream); - if (atomic_read(&substream->mmap_count)) - return -EBADFD; + if (result) + goto unlock; if (substream->ops->hw_free) result = substream->ops->hw_free(substream); snd_pcm_set_state(substream, SNDRV_PCM_STATE_OPEN); pm_qos_remove_request(&substream->latency_pm_qos_req); + unlock: + snd_pcm_buffer_access_unlock(runtime); return result; } @@ -1025,15 +1064,17 @@ struct action_ops { */ static int snd_pcm_action_group(const struct action_ops *ops, struct snd_pcm_substream *substream, - int state, int do_lock) + int state, int stream_lock) { struct snd_pcm_substream *s = NULL; struct snd_pcm_substream *s1; int res = 0, depth = 1; snd_pcm_group_for_each_entry(s, substream) { - if (do_lock && s != substream) { - if (s->pcm->nonatomic) + if (s != substream) { + if (!stream_lock) + mutex_lock_nested(&s->runtime->buffer_mutex, depth); + else if (s->pcm->nonatomic) mutex_lock_nested(&s->self_group.mutex, depth); else spin_lock_nested(&s->self_group.lock, depth); @@ -1061,18 +1102,18 @@ static int snd_pcm_action_group(const struct action_ops *ops, ops->post_action(s, state); } _unlock: - if (do_lock) { - /* unlock streams */ - snd_pcm_group_for_each_entry(s1, substream) { - if (s1 != substream) { - if (s1->pcm->nonatomic) - mutex_unlock(&s1->self_group.mutex); - else - spin_unlock(&s1->self_group.lock); - } - if (s1 == s) /* end */ - break; + /* unlock streams */ + snd_pcm_group_for_each_entry(s1, substream) { + if (s1 != substream) { + if (!stream_lock) + mutex_unlock(&s1->runtime->buffer_mutex); + else if (s1->pcm->nonatomic) + mutex_unlock(&s1->self_group.mutex); + else + spin_unlock(&s1->self_group.lock); } + if (s1 == s) /* end */ + break; } return res; } @@ -1202,10 +1243,15 @@ static int snd_pcm_action_nonatomic(const struct action_ops *ops, /* Guarantee the group members won't change during non-atomic action */ down_read(&snd_pcm_link_rwsem); + res = snd_pcm_buffer_access_lock(substream->runtime); + if (res < 0) + goto unlock; if (snd_pcm_stream_linked(substream)) res = snd_pcm_action_group(ops, substream, state, 0); else res = snd_pcm_action_single(ops, substream, state); + snd_pcm_buffer_access_unlock(substream->runtime); + unlock: up_read(&snd_pcm_link_rwsem); return res; } @@ -1656,21 +1702,25 @@ static int snd_pcm_do_reset(struct snd_pcm_substream *substream, int state) int err = substream->ops->ioctl(substream, SNDRV_PCM_IOCTL1_RESET, NULL); if (err < 0) return err; + snd_pcm_stream_lock_irq(substream); runtime->hw_ptr_base = 0; runtime->hw_ptr_interrupt = runtime->status->hw_ptr - runtime->status->hw_ptr % runtime->period_size; runtime->silence_start = runtime->status->hw_ptr; runtime->silence_filled = 0; + snd_pcm_stream_unlock_irq(substream); return 0; } static void snd_pcm_post_reset(struct snd_pcm_substream *substream, int state) { struct snd_pcm_runtime *runtime = substream->runtime; + snd_pcm_stream_lock_irq(substream); runtime->control->appl_ptr = runtime->status->hw_ptr; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) snd_pcm_playback_silence(substream, ULONG_MAX); + snd_pcm_stream_unlock_irq(substream); } static const struct action_ops snd_pcm_action_reset = { diff --git a/sound/core/seq/seq_ports.c b/sound/core/seq/seq_ports.c index 83be6b982a87..97e8eb38b096 100644 --- a/sound/core/seq/seq_ports.c +++ b/sound/core/seq/seq_ports.c @@ -514,10 +514,11 @@ 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; @@ -525,7 +526,6 @@ 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) @@ -535,6 +535,18 @@ 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); } @@ -590,27 +602,30 @@ 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.c b/sound/core/seq/seq_queue.c index 71a6ea62c3be..4ff0b927230c 100644 --- a/sound/core/seq/seq_queue.c +++ b/sound/core/seq/seq_queue.c @@ -234,12 +234,15 @@ struct snd_seq_queue *snd_seq_queue_find_name(char *name) /* -------------------------------------------------------- */ +#define MAX_CELL_PROCESSES_IN_QUEUE 1000 + void snd_seq_check_queue(struct snd_seq_queue *q, int atomic, int hop) { unsigned long flags; struct snd_seq_event_cell *cell; snd_seq_tick_time_t cur_tick; snd_seq_real_time_t cur_time; + int processed = 0; if (q == NULL) return; @@ -262,6 +265,8 @@ void snd_seq_check_queue(struct snd_seq_queue *q, int atomic, int hop) if (!cell) break; snd_seq_dispatch_event(cell, atomic, hop); + if (++processed >= MAX_CELL_PROCESSES_IN_QUEUE) + goto out; /* the rest processed at the next batch */ } /* Process time queue... */ @@ -271,14 +276,19 @@ void snd_seq_check_queue(struct snd_seq_queue *q, int atomic, int hop) if (!cell) break; snd_seq_dispatch_event(cell, atomic, hop); + if (++processed >= MAX_CELL_PROCESSES_IN_QUEUE) + goto out; /* the rest processed at the next batch */ } + out: /* free lock */ spin_lock_irqsave(&q->check_lock, flags); if (q->check_again) { q->check_again = 0; - spin_unlock_irqrestore(&q->check_lock, flags); - goto __again; + if (processed < MAX_CELL_PROCESSES_IN_QUEUE) { + spin_unlock_irqrestore(&q->check_lock, flags); + goto __again; + } } q->check_blocked = 0; spin_unlock_irqrestore(&q->check_lock, flags); diff --git a/sound/core/seq_device.c b/sound/core/seq_device.c index e9dbad93f9d0..c9223049551c 100644 --- a/sound/core/seq_device.c +++ b/sound/core/seq_device.c @@ -147,6 +147,8 @@ 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; } @@ -174,11 +176,7 @@ 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 b5a0ba79bf74..d684aa4150aa 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -595,13 +595,13 @@ 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) { @@ -636,23 +636,22 @@ 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/opl3/opl3_midi.c b/sound/drivers/opl3/opl3_midi.c index 280cc79870cf..ce38ec09d408 100644 --- a/sound/drivers/opl3/opl3_midi.c +++ b/sound/drivers/opl3/opl3_midi.c @@ -398,7 +398,7 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan) } if (instr_4op) { vp2 = &opl3->voices[voice + 3]; - if (vp->state > 0) { + if (vp2->state > 0) { opl3_reg = reg_side | (OPL3_REG_KEYON_BLOCK + voice_offset + 3); reg_val = vp->keyon_reg & ~OPL3_KEYON_BIT; diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig index e6b4ca469b2a..e469375e2f2a 100644 --- a/sound/firewire/Kconfig +++ b/sound/firewire/Kconfig @@ -38,7 +38,7 @@ config SND_OXFW * Mackie(Loud) Onyx 1640i (former model) * Mackie(Loud) Onyx Satellite * Mackie(Loud) Tapco Link.Firewire - * Mackie(Loud) d.4 pro + * Mackie(Loud) d.2 pro/d.4 pro (built-in FireWire card with OXFW971 ASIC) * Mackie(Loud) U.420/U.420d * TASCAM FireOne * Stanton Controllers & Systems 1 Deck/Mixer @@ -84,7 +84,7 @@ 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) and d.2 Pro + * Mackie d.2 (optional FireWire card with DM1000 ASIC) * Stanton FinalScratch 2 (ScratchAmp) * Tascam IF-FW/DM * Behringer XENIX UFX 1204/1604 @@ -110,6 +110,7 @@ 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 441c58283b47..d58f4fe2be8c 100644 --- a/sound/firewire/bebob/bebob.c +++ b/sound/firewire/bebob/bebob.c @@ -59,6 +59,7 @@ 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 @@ -387,7 +388,7 @@ 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 card) and d.2 Pro (the card is built-in). + // Mackie, d.2 (optional Firewire card with DM1000). SND_BEBOB_DEV_ENTRY(VEN_MACKIE1, 0x00010067, &spec_normal), /* Stanton, ScratchAmp */ SND_BEBOB_DEV_ENTRY(VEN_STANTON, 0x00000001, &spec_normal), @@ -486,6 +487,8 @@ 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/fcp.c b/sound/firewire/fcp.c index bbfbebf4affb..df44dd5dc4b2 100644 --- a/sound/firewire/fcp.c +++ b/sound/firewire/fcp.c @@ -240,9 +240,7 @@ int fcp_avc_transaction(struct fw_unit *unit, t.response_match_bytes = response_match_bytes; t.state = STATE_PENDING; init_waitqueue_head(&t.wait); - - if (*(const u8 *)command == 0x00 || *(const u8 *)command == 0x03) - t.deferrable = true; + t.deferrable = (*(const u8 *)command == 0x00 || *(const u8 *)command == 0x03); spin_lock_irq(&transactions_lock); list_add_tail(&t.list, &transactions); diff --git a/sound/firewire/fireworks/fireworks_hwdep.c b/sound/firewire/fireworks/fireworks_hwdep.c index e93eb4616c5f..c739173c668f 100644 --- a/sound/firewire/fireworks/fireworks_hwdep.c +++ b/sound/firewire/fireworks/fireworks_hwdep.c @@ -34,6 +34,7 @@ hwdep_read_resp_buf(struct snd_efw *efw, char __user *buf, long remained, type = SNDRV_FIREWIRE_EVENT_EFW_RESPONSE; if (copy_to_user(buf, &type, sizeof(type))) return -EFAULT; + count += sizeof(type); remained -= sizeof(type); buf += sizeof(type); diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index 6184a7c8f2b3..bebb2b8296cb 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -350,7 +350,7 @@ 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.4 pro: Unknown + * d.2 pro/d.4 pro (built-in card): Unknown * U.420: Unknown * U.420d: Unknown */ diff --git a/sound/hda/ext/hdac_ext_stream.c b/sound/hda/ext/hdac_ext_stream.c index 6b1b4b834bae..04f4070fbf36 100644 --- a/sound/hda/ext/hdac_ext_stream.c +++ b/sound/hda/ext/hdac_ext_stream.c @@ -106,20 +106,14 @@ void snd_hdac_stream_free_all(struct hdac_bus *bus) } EXPORT_SYMBOL_GPL(snd_hdac_stream_free_all); -/** - * snd_hdac_ext_stream_decouple - decouple the hdac stream - * @bus: HD-audio core bus - * @stream: HD-audio ext core stream object to initialize - * @decouple: flag to decouple - */ -void snd_hdac_ext_stream_decouple(struct hdac_bus *bus, - struct hdac_ext_stream *stream, bool decouple) +void snd_hdac_ext_stream_decouple_locked(struct hdac_bus *bus, + struct hdac_ext_stream *stream, + bool decouple) { struct hdac_stream *hstream = &stream->hstream; u32 val; int mask = AZX_PPCTL_PROCEN(hstream->index); - spin_lock_irq(&bus->reg_lock); val = readw(bus->ppcap + AZX_REG_PP_PPCTL) & mask; if (decouple && !val) @@ -128,6 +122,20 @@ void snd_hdac_ext_stream_decouple(struct hdac_bus *bus, snd_hdac_updatel(bus->ppcap, AZX_REG_PP_PPCTL, mask, 0); stream->decoupled = decouple; +} +EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_decouple_locked); + +/** + * snd_hdac_ext_stream_decouple - decouple the hdac stream + * @bus: HD-audio core bus + * @stream: HD-audio ext core stream object to initialize + * @decouple: flag to decouple + */ +void snd_hdac_ext_stream_decouple(struct hdac_bus *bus, + struct hdac_ext_stream *stream, bool decouple) +{ + spin_lock_irq(&bus->reg_lock); + snd_hdac_ext_stream_decouple_locked(bus, stream, decouple); spin_unlock_irq(&bus->reg_lock); } EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_decouple); @@ -252,6 +260,7 @@ hdac_ext_link_stream_assign(struct hdac_bus *bus, return NULL; } + spin_lock_irq(&bus->reg_lock); list_for_each_entry(stream, &bus->stream_list, list) { struct hdac_ext_stream *hstream = container_of(stream, struct hdac_ext_stream, @@ -266,17 +275,16 @@ hdac_ext_link_stream_assign(struct hdac_bus *bus, } if (!hstream->link_locked) { - snd_hdac_ext_stream_decouple(bus, hstream, true); + snd_hdac_ext_stream_decouple_locked(bus, hstream, true); res = hstream; break; } } if (res) { - spin_lock_irq(&bus->reg_lock); res->link_locked = 1; res->link_substream = substream; - spin_unlock_irq(&bus->reg_lock); } + spin_unlock_irq(&bus->reg_lock); return res; } @@ -292,6 +300,7 @@ hdac_ext_host_stream_assign(struct hdac_bus *bus, return NULL; } + spin_lock_irq(&bus->reg_lock); list_for_each_entry(stream, &bus->stream_list, list) { struct hdac_ext_stream *hstream = container_of(stream, struct hdac_ext_stream, @@ -301,18 +310,17 @@ hdac_ext_host_stream_assign(struct hdac_bus *bus, if (!stream->opened) { if (!hstream->decoupled) - snd_hdac_ext_stream_decouple(bus, hstream, true); + snd_hdac_ext_stream_decouple_locked(bus, hstream, true); res = hstream; break; } } if (res) { - spin_lock_irq(&bus->reg_lock); res->hstream.opened = 1; res->hstream.running = 0; res->hstream.substream = substream; - spin_unlock_irq(&bus->reg_lock); } + spin_unlock_irq(&bus->reg_lock); return res; } @@ -378,15 +386,17 @@ void snd_hdac_ext_stream_release(struct hdac_ext_stream *stream, int type) break; case HDAC_EXT_STREAM_TYPE_HOST: + spin_lock_irq(&bus->reg_lock); if (stream->decoupled && !stream->link_locked) - snd_hdac_ext_stream_decouple(bus, stream, false); + snd_hdac_ext_stream_decouple_locked(bus, stream, false); + spin_unlock_irq(&bus->reg_lock); snd_hdac_stream_release(&stream->hstream); break; case HDAC_EXT_STREAM_TYPE_LINK: - if (stream->decoupled && !stream->hstream.opened) - snd_hdac_ext_stream_decouple(bus, stream, false); spin_lock_irq(&bus->reg_lock); + if (stream->decoupled && !stream->hstream.opened) + snd_hdac_ext_stream_decouple_locked(bus, stream, false); stream->link_locked = 0; stream->link_substream = NULL; spin_unlock_irq(&bus->reg_lock); diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index 7e7be8e4dcf9..87ba66dcfd47 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -395,8 +395,9 @@ 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_stream.c b/sound/hda/hdac_stream.c index 682ed39f79b0..b299b8b7f871 100644 --- a/sound/hda/hdac_stream.c +++ b/sound/hda/hdac_stream.c @@ -289,6 +289,7 @@ struct hdac_stream *snd_hdac_stream_assign(struct hdac_bus *bus, int key = (substream->pcm->device << 16) | (substream->number << 2) | (substream->stream + 1); + spin_lock_irq(&bus->reg_lock); list_for_each_entry(azx_dev, &bus->stream_list, list) { if (azx_dev->direction != substream->stream) continue; @@ -302,13 +303,12 @@ struct hdac_stream *snd_hdac_stream_assign(struct hdac_bus *bus, res = azx_dev; } if (res) { - spin_lock_irq(&bus->reg_lock); res->opened = 1; res->running = 0; res->assigned_key = key; res->substream = substream; - spin_unlock_irq(&bus->reg_lock); } + spin_unlock_irq(&bus->reg_lock); return res; } EXPORT_SYMBOL_GPL(snd_hdac_stream_assign); diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index b690ed937cbe..df2e45c8814e 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -22,7 +22,7 @@ config SND_SB16_DSP menuconfig SND_ISA bool "ISA sound devices" depends on ISA || COMPILE_TEST - depends on ISA_DMA_API + depends on ISA_DMA_API && !M68K default y help Support for sound devices connected via the ISA bus. diff --git a/sound/isa/cmi8330.c b/sound/isa/cmi8330.c index bb7d4940ac25..281ecd0eea48 100644 --- a/sound/isa/cmi8330.c +++ b/sound/isa/cmi8330.c @@ -549,7 +549,7 @@ 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/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c index fa3c39cff5f8..9ee3a312c679 100644 --- a/sound/isa/cs423x/cs4236.c +++ b/sound/isa/cs423x/cs4236.c @@ -544,7 +544,7 @@ static int snd_cs423x_pnpbios_detect(struct pnp_dev *pdev, static int dev; int err; struct snd_card *card; - struct pnp_dev *cdev; + struct pnp_dev *cdev, *iter; char cid[PNP_ID_LEN]; if (pnp_device_is_isapnp(pdev)) @@ -560,9 +560,11 @@ static int snd_cs423x_pnpbios_detect(struct pnp_dev *pdev, strcpy(cid, pdev->id[0].id); cid[5] = '1'; cdev = NULL; - list_for_each_entry(cdev, &(pdev->protocol->devices), protocol_list) { - if (!strcmp(cdev->id[0].id, cid)) + list_for_each_entry(iter, &(pdev->protocol->devices), protocol_list) { + if (!strcmp(iter->id[0].id, cid)) { + cdev = iter; break; + } } err = snd_cs423x_card_new(&pdev->dev, dev, &card); if (err < 0) diff --git a/sound/isa/gus/gus_dma.c b/sound/isa/gus/gus_dma.c index a1c770d826dd..6d664dd8dde0 100644 --- a/sound/isa/gus/gus_dma.c +++ b/sound/isa/gus/gus_dma.c @@ -126,6 +126,8 @@ static void snd_gf1_dma_interrupt(struct snd_gus_card * gus) } block = snd_gf1_dma_next_block(gus); spin_unlock(&gus->dma_lock); + if (!block) + return; snd_gf1_dma_program(gus, block->addr, block->buf_addr, block->count, (unsigned short) block->cmd); kfree(block); #if 0 diff --git a/sound/isa/sb/sb16_csp.c b/sound/isa/sb/sb16_csp.c index 69960cf1bb51..30021ab5e0e9 100644 --- a/sound/isa/sb/sb16_csp.c +++ b/sound/isa/sb/sb16_csp.c @@ -814,6 +814,7 @@ 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 */ @@ -853,6 +854,7 @@ 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); @@ -878,6 +880,7 @@ 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) { @@ -892,6 +895,7 @@ 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); @@ -1072,10 +1076,14 @@ 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/pci/Kconfig b/sound/pci/Kconfig index 7630f808d087..6edde2f14502 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -279,6 +279,7 @@ config SND_CS46XX_NEW_DSP config SND_CS5530 tristate "CS5530 Audio" depends on ISA_DMA_API && (X86_32 || COMPILE_TEST) + depends on !M68K select SND_SB16_DSP help Say Y here to include support for audio on Cyrix/NatSemi CS5530 chips. diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 66f6c3bf08e3..6fb192a94762 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -938,8 +938,8 @@ static int snd_ac97_ad18xx_pcm_get_volume(struct snd_kcontrol *kcontrol, struct int codec = kcontrol->private_value & 3; mutex_lock(&ac97->page_mutex); - ucontrol->value.integer.value[0] = 31 - ((ac97->spec.ad18xx.pcmreg[codec] >> 0) & 31); - ucontrol->value.integer.value[1] = 31 - ((ac97->spec.ad18xx.pcmreg[codec] >> 8) & 31); + ucontrol->value.integer.value[0] = 31 - ((ac97->spec.ad18xx.pcmreg[codec] >> 8) & 31); + ucontrol->value.integer.value[1] = 31 - ((ac97->spec.ad18xx.pcmreg[codec] >> 0) & 31); mutex_unlock(&ac97->page_mutex); return 0; } diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index df720881eb99..db9d89ba3658 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -302,7 +302,6 @@ MODULE_PARM_DESC(joystick_port, "Joystick port address."); #define CM_MICGAINZ 0x01 /* mic boost */ #define CM_MICGAINZ_SHIFT 0 -#define CM_REG_MIXER3 0x24 #define CM_REG_AUX_VOL 0x26 #define CM_VAUXL_MASK 0xf0 #define CM_VAUXR_MASK 0x0f @@ -3310,7 +3309,7 @@ static void snd_cmipci_remove(struct pci_dev *pci) */ static unsigned char saved_regs[] = { CM_REG_FUNCTRL1, CM_REG_CHFORMAT, CM_REG_LEGACY_CTRL, CM_REG_MISC_CTRL, - CM_REG_MIXER0, CM_REG_MIXER1, CM_REG_MIXER2, CM_REG_MIXER3, CM_REG_PLL, + CM_REG_MIXER0, CM_REG_MIXER1, CM_REG_MIXER2, CM_REG_AUX_VOL, CM_REG_PLL, CM_REG_CH0_FRAME1, CM_REG_CH0_FRAME2, CM_REG_CH1_FRAME1, CM_REG_CH1_FRAME2, CM_REG_EXT_MISC, CM_REG_INT_STATUS, CM_REG_INT_HLDCLR, CM_REG_FUNCTRL0, diff --git a/sound/pci/ctxfi/ctamixer.c b/sound/pci/ctxfi/ctamixer.c index d4ff377eb3a3..6d636bdcaa5a 100644 --- a/sound/pci/ctxfi/ctamixer.c +++ b/sound/pci/ctxfi/ctamixer.c @@ -23,16 +23,15 @@ #define BLANK_SLOT 4094 -static int amixer_master(struct rsc *rsc) +static void amixer_master(struct rsc *rsc) { rsc->conj = 0; - return rsc->idx = container_of(rsc, struct amixer, rsc)->idx[0]; + rsc->idx = container_of(rsc, struct amixer, rsc)->idx[0]; } -static int amixer_next_conj(struct rsc *rsc) +static void amixer_next_conj(struct rsc *rsc) { rsc->conj++; - return container_of(rsc, struct amixer, rsc)->idx[rsc->conj]; } static int amixer_index(const struct rsc *rsc) @@ -331,16 +330,15 @@ int amixer_mgr_destroy(struct amixer_mgr *amixer_mgr) /* SUM resource management */ -static int sum_master(struct rsc *rsc) +static void sum_master(struct rsc *rsc) { rsc->conj = 0; - return rsc->idx = container_of(rsc, struct sum, rsc)->idx[0]; + rsc->idx = container_of(rsc, struct sum, rsc)->idx[0]; } -static int sum_next_conj(struct rsc *rsc) +static void sum_next_conj(struct rsc *rsc) { rsc->conj++; - return container_of(rsc, struct sum, rsc)->idx[rsc->conj]; } static int sum_index(const struct rsc *rsc) diff --git a/sound/pci/ctxfi/ctdaio.c b/sound/pci/ctxfi/ctdaio.c index 27441d498968..b5e1296af09e 100644 --- a/sound/pci/ctxfi/ctdaio.c +++ b/sound/pci/ctxfi/ctdaio.c @@ -51,12 +51,12 @@ static struct daio_rsc_idx idx_20k2[NUM_DAIOTYP] = { [SPDIFIO] = {.left = 0x05, .right = 0x85}, }; -static int daio_master(struct rsc *rsc) +static void daio_master(struct rsc *rsc) { /* Actually, this is not the resource index of DAIO. * For DAO, it is the input mapper index. And, for DAI, * it is the output time-slot index. */ - return rsc->conj = rsc->idx; + rsc->conj = rsc->idx; } static int daio_index(const struct rsc *rsc) @@ -64,19 +64,19 @@ static int daio_index(const struct rsc *rsc) return rsc->conj; } -static int daio_out_next_conj(struct rsc *rsc) +static void daio_out_next_conj(struct rsc *rsc) { - return rsc->conj += 2; + rsc->conj += 2; } -static int daio_in_next_conj_20k1(struct rsc *rsc) +static void daio_in_next_conj_20k1(struct rsc *rsc) { - return rsc->conj += 0x200; + rsc->conj += 0x200; } -static int daio_in_next_conj_20k2(struct rsc *rsc) +static void daio_in_next_conj_20k2(struct rsc *rsc) { - return rsc->conj += 0x100; + rsc->conj += 0x100; } static const struct rsc_ops daio_out_rsc_ops = { diff --git a/sound/pci/ctxfi/ctresource.c b/sound/pci/ctxfi/ctresource.c index 0bb5696e44b3..ec5f597b580a 100644 --- a/sound/pci/ctxfi/ctresource.c +++ b/sound/pci/ctxfi/ctresource.c @@ -109,18 +109,17 @@ static int audio_ring_slot(const struct rsc *rsc) return (rsc->conj << 4) + offset_in_audio_slot_block[rsc->type]; } -static int rsc_next_conj(struct rsc *rsc) +static void rsc_next_conj(struct rsc *rsc) { unsigned int i; for (i = 0; (i < 8) && (!(rsc->msr & (0x1 << i))); ) i++; rsc->conj += (AUDIO_SLOT_BLOCK_NUM >> i); - return rsc->conj; } -static int rsc_master(struct rsc *rsc) +static void rsc_master(struct rsc *rsc) { - return rsc->conj = rsc->idx; + rsc->conj = rsc->idx; } static const struct rsc_ops rsc_generic_ops = { diff --git a/sound/pci/ctxfi/ctresource.h b/sound/pci/ctxfi/ctresource.h index 93e47488a1c1..92146054af58 100644 --- a/sound/pci/ctxfi/ctresource.h +++ b/sound/pci/ctxfi/ctresource.h @@ -39,8 +39,8 @@ struct rsc { }; struct rsc_ops { - int (*master)(struct rsc *rsc); /* Move to master resource */ - int (*next_conj)(struct rsc *rsc); /* Move to next conjugate resource */ + void (*master)(struct rsc *rsc); /* Move to master resource */ + void (*next_conj)(struct rsc *rsc); /* Move to next conjugate resource */ int (*index)(const struct rsc *rsc); /* Return the index of resource */ /* Return the output slot number */ int (*output_slot)(const struct rsc *rsc); diff --git a/sound/pci/ctxfi/ctsrc.c b/sound/pci/ctxfi/ctsrc.c index 37c18ce84974..7d2bda0c3d3d 100644 --- a/sound/pci/ctxfi/ctsrc.c +++ b/sound/pci/ctxfi/ctsrc.c @@ -590,16 +590,15 @@ int src_mgr_destroy(struct src_mgr *src_mgr) /* SRCIMP resource manager operations */ -static int srcimp_master(struct rsc *rsc) +static void srcimp_master(struct rsc *rsc) { rsc->conj = 0; - return rsc->idx = container_of(rsc, struct srcimp, rsc)->idx[0]; + rsc->idx = container_of(rsc, struct srcimp, rsc)->idx[0]; } -static int srcimp_next_conj(struct rsc *rsc) +static void srcimp_next_conj(struct rsc *rsc) { rsc->conj++; - return container_of(rsc, struct srcimp, rsc)->idx[rsc->conj]; } static int srcimp_index(const struct rsc *rsc) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 326f95ce5ceb..c8847de8388f 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1721,8 +1721,11 @@ void snd_hda_ctls_clear(struct hda_codec *codec) { int i; struct hda_nid_item *items = codec->mixers.list; + + down_write(&codec->card->controls_rwsem); for (i = 0; i < codec->mixers.used; i++) snd_ctl_remove(codec->card, items[i].kctl); + up_write(&codec->card->controls_rwsem); snd_array_free(&codec->mixers); snd_array_free(&codec->nids); } diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 7ac3f04ca8c0..e92fcb150e57 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -3458,7 +3458,7 @@ 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 +3468,13 @@ 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_intel.c b/sound/pci/hda/hda_intel.c index ebb1ee69dd0c..b8fe0ec5d624 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -671,13 +671,17 @@ static int azx_position_check(struct azx *chip, struct azx_dev *azx_dev) * 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) @@ -714,6 +718,24 @@ 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 */ } @@ -907,11 +929,7 @@ 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); } @@ -1590,6 +1608,7 @@ static struct snd_pci_quirk probe_mask_list[] = { /* forced codec slots */ SND_PCI_QUIRK(0x1043, 0x1262, "ASUS W5Fm", 0x103), SND_PCI_QUIRK(0x1046, 0x1262, "ASUS W5F", 0x103), + SND_PCI_QUIRK(0x1558, 0x0351, "Schenker Dock 15", 0x105), /* WinFast VP200 H (Teradici) user reported broken communication */ SND_PCI_QUIRK(0x3a21, 0x040d, "WinFast VP200 H", 0x101), {} @@ -1775,8 +1794,6 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci, assign_position_fix(chip, check_position_fix(chip, position_fix[dev])); - check_probe_mask(chip, dev); - if (single_cmd < 0) /* allow fallback to single_cmd at errors */ chip->fallback_to_single_cmd = 1; else /* explicitly set to single_cmd or not */ @@ -1808,6 +1825,8 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci, chip->bus.needs_damn_long_delay = 1; } + check_probe_mask(chip, dev); + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); if (err < 0) { dev_err(card->dev, "Error creating device [card]!\n"); diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index e378cb33c69d..2971b34c87c1 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -292,6 +292,9 @@ static int hda_tegra_first_init(struct azx *chip, struct platform_device *pdev) const char *sname, *drv_name = "tegra-hda"; struct device_node *np = pdev->dev.of_node; + if (irq_id < 0) + return irq_id; + err = hda_tegra_init_chip(chip, pdev); if (err) return err; diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index f620b402b309..5128a5df16fd 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1820,6 +1820,7 @@ static int hdmi_add_cvt(struct hda_codec *codec, hda_nid_t cvt_nid) static const struct snd_pci_quirk force_connect_list[] = { SND_PCI_QUIRK(0x103c, 0x870f, "HP", 1), SND_PCI_QUIRK(0x103c, 0x871a, "HP", 1), + SND_PCI_QUIRK(0x1462, 0xec94, "MS-7C94", 1), {} }; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index de40bb99b679..851ea79da31c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -375,6 +375,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) alc_update_coef_idx(codec, 0x67, 0xf000, 0x3000); /* fallthrough */ case 0x10ec0215: + case 0x10ec0230: case 0x10ec0233: case 0x10ec0235: case 0x10ec0236: @@ -516,6 +517,8 @@ static void alc_shutup_pins(struct hda_codec *codec) struct alc_spec *spec = codec->spec; switch (codec->core.vendor_id) { + case 0x10ec0236: + case 0x10ec0256: case 0x10ec0283: case 0x10ec0286: case 0x10ec0288: @@ -1923,6 +1926,7 @@ enum { ALC887_FIXUP_ASUS_BASS, ALC887_FIXUP_BASS_CHMAP, ALC1220_FIXUP_GB_DUAL_CODECS, + ALC1220_FIXUP_GB_X570, ALC1220_FIXUP_CLEVO_P950, ALC1220_FIXUP_CLEVO_PB51ED, ALC1220_FIXUP_CLEVO_PB51ED_PINS, @@ -2112,6 +2116,30 @@ static void alc1220_fixup_gb_dual_codecs(struct hda_codec *codec, } } +static void alc1220_fixup_gb_x570(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + static const hda_nid_t conn1[] = { 0x0c }; + static const struct coef_fw gb_x570_coefs[] = { + WRITE_COEF(0x07, 0x03c0), + WRITE_COEF(0x1a, 0x01c1), + WRITE_COEF(0x1b, 0x0202), + WRITE_COEF(0x43, 0x3005), + {} + }; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_override_conn_list(codec, 0x14, ARRAY_SIZE(conn1), conn1); + snd_hda_override_conn_list(codec, 0x1b, ARRAY_SIZE(conn1), conn1); + break; + case HDA_FIXUP_ACT_INIT: + alc_process_coef_fw(codec, gb_x570_coefs); + break; + } +} + static void alc1220_fixup_clevo_p950(struct hda_codec *codec, const struct hda_fixup *fix, int action) @@ -2414,6 +2442,10 @@ static const struct hda_fixup alc882_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc1220_fixup_gb_dual_codecs, }, + [ALC1220_FIXUP_GB_X570] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc1220_fixup_gb_x570, + }, [ALC1220_FIXUP_CLEVO_P950] = { .type = HDA_FIXUP_FUNC, .v.func = alc1220_fixup_clevo_p950, @@ -2516,8 +2548,9 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x13fe, 0x1009, "Advantech MIT-W101", ALC886_FIXUP_EAPD), SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte EP45-DS3/Z87X-UD3H", ALC889_FIXUP_FRONT_HP_NO_PRESENCE), SND_PCI_QUIRK(0x1458, 0xa0b8, "Gigabyte AZ370-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS), - SND_PCI_QUIRK(0x1458, 0xa0cd, "Gigabyte X570 Aorus Master", ALC1220_FIXUP_CLEVO_P950), - SND_PCI_QUIRK(0x1458, 0xa0ce, "Gigabyte X570 Aorus Xtreme", ALC1220_FIXUP_CLEVO_P950), + SND_PCI_QUIRK(0x1458, 0xa0cd, "Gigabyte X570 Aorus Master", ALC1220_FIXUP_GB_X570), + SND_PCI_QUIRK(0x1458, 0xa0ce, "Gigabyte X570 Aorus Xtreme", ALC1220_FIXUP_GB_X570), + SND_PCI_QUIRK(0x1458, 0xa0d5, "Gigabyte X570S Aorus Master", ALC1220_FIXUP_GB_X570), SND_PCI_QUIRK(0x1462, 0x11f7, "MSI-GE63", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1228, "MSI-GP63", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1229, "MSI-GP73", ALC1220_FIXUP_CLEVO_P950), @@ -2534,11 +2567,15 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x65d2, "Clevo PB51R[CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x65e1, "Clevo PB51[ED][DF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x65e5, "Clevo PC50D[PRS](?:-D|-G)?", ALC1220_FIXUP_CLEVO_PB51ED_PINS), + SND_PCI_QUIRK(0x1558, 0x65f1, "Clevo PC50HS", ALC1220_FIXUP_CLEVO_PB51ED_PINS), + SND_PCI_QUIRK(0x1558, 0x65f5, "Clevo PD50PN[NRT]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x67d1, "Clevo PB71[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x67e1, "Clevo PB71[DE][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x67e5, "Clevo PC70D[PRS](?:-D|-G)?", ALC1220_FIXUP_CLEVO_PB51ED_PINS), + SND_PCI_QUIRK(0x1558, 0x67f1, "Clevo PC70H[PRS]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x70d1, "Clevo PC70[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), - SND_PCI_QUIRK(0x1558, 0x7714, "Clevo X170", ALC1220_FIXUP_CLEVO_PB51ED_PINS), + SND_PCI_QUIRK(0x1558, 0x7714, "Clevo X170SM", ALC1220_FIXUP_CLEVO_PB51ED_PINS), + SND_PCI_QUIRK(0x1558, 0x7715, "Clevo X170KM-G", ALC1220_FIXUP_CLEVO_PB51ED), SND_PCI_QUIRK(0x1558, 0x9501, "Clevo P950HR", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1558, 0x9506, "Clevo P955HQ", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1558, 0x950a, "Clevo P955H[PR]", ALC1220_FIXUP_CLEVO_P950), @@ -2589,6 +2626,7 @@ static const struct hda_model_fixup alc882_fixup_models[] = { {.id = ALC882_FIXUP_NO_PRIMARY_HP, .name = "no-primary-hp"}, {.id = ALC887_FIXUP_ASUS_BASS, .name = "asus-bass"}, {.id = ALC1220_FIXUP_GB_DUAL_CODECS, .name = "dual-codecs"}, + {.id = ALC1220_FIXUP_GB_X570, .name = "gb-x570"}, {.id = ALC1220_FIXUP_CLEVO_P950, .name = "clevo-p950"}, {} }; @@ -3143,6 +3181,7 @@ static void alc_disable_headset_jack_key(struct hda_codec *codec) alc_update_coef_idx(codec, 0x49, 0x0045, 0x0); alc_update_coef_idx(codec, 0x44, 0x0045 << 8, 0x0); break; + case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: alc_write_coef_idx(codec, 0x48, 0x0); @@ -3170,6 +3209,7 @@ static void alc_enable_headset_jack_key(struct hda_codec *codec) alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); alc_update_coef_idx(codec, 0x44, 0x007f << 8, 0x0045 << 8); break; + case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: alc_write_coef_idx(codec, 0x48, 0xd011); @@ -3518,7 +3558,8 @@ static void alc256_shutup(struct hda_codec *codec) /* If disable 3k pulldown control for alc257, the Mic detection will not work correctly * when booting with headset plugged. So skip setting it for the codec alc257 */ - if (codec->core.vendor_id != 0x10ec0257) + if (codec->core.vendor_id != 0x10ec0236 && + codec->core.vendor_id != 0x10ec0257) alc_update_coef_idx(codec, 0x46, 0, 3 << 12); if (!spec->no_shutup_pins) @@ -4630,6 +4671,7 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) case 0x10ec0255: alc_process_coef_fw(codec, coef0255); break; + case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: alc_process_coef_fw(codec, coef0256); @@ -4744,6 +4786,7 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, alc_process_coef_fw(codec, coef0255); snd_hda_set_pin_ctl_cache(codec, mic_pin, PIN_VREF50); break; + case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: alc_write_coef_idx(codec, 0x45, 0xc489); @@ -4893,6 +4936,7 @@ static void alc_headset_mode_default(struct hda_codec *codec) case 0x10ec0255: alc_process_coef_fw(codec, coef0255); break; + case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: alc_write_coef_idx(codec, 0x1b, 0x0e4b); @@ -4991,6 +5035,7 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) case 0x10ec0255: alc_process_coef_fw(codec, coef0255); break; + case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: alc_process_coef_fw(codec, coef0256); @@ -5104,6 +5149,7 @@ static void alc_headset_mode_omtp(struct hda_codec *codec) case 0x10ec0255: alc_process_coef_fw(codec, coef0255); break; + case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: alc_process_coef_fw(codec, coef0256); @@ -5199,6 +5245,7 @@ static void alc_determine_headset_type(struct hda_codec *codec) val = alc_read_coef_idx(codec, 0x46); is_ctia = (val & 0x0070) == 0x0070; break; + case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: alc_write_coef_idx(codec, 0x1b, 0x0e4b); @@ -5492,6 +5539,7 @@ static void alc255_set_default_jack_type(struct hda_codec *codec) case 0x10ec0255: alc_process_coef_fw(codec, alc255fw); break; + case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: alc_process_coef_fw(codec, alc256fw); @@ -6092,6 +6140,7 @@ static void alc_combo_jack_hp_jd_restart(struct hda_codec *codec) alc_update_coef_idx(codec, 0x4a, 0x8000, 1 << 15); /* Reset HP JD */ alc_update_coef_idx(codec, 0x4a, 0x8000, 0 << 15); break; + case 0x10ec0230: case 0x10ec0235: case 0x10ec0236: case 0x10ec0255: @@ -6207,6 +6256,24 @@ static void alc274_fixup_hp_headset_mic(struct hda_codec *codec, } } +static void alc285_fixup_hp_spectre_x360(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + static const hda_nid_t conn[] = { 0x02 }; + static const struct hda_pintbl pincfgs[] = { + { 0x14, 0x90170110 }, /* rear speaker */ + { } + }; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_apply_pincfgs(codec, pincfgs); + /* force front speaker to DAC1 */ + snd_hda_override_conn_list(codec, 0x17, ARRAY_SIZE(conn), conn); + break; + } +} + /* for hda_fixup_thinkpad_acpi() */ #include "thinkpad_helper.c" @@ -6390,6 +6457,7 @@ enum { ALC285_FIXUP_HP_MUTE_LED, ALC236_FIXUP_HP_MUTE_LED, ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, + ALC256_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, ALC295_FIXUP_ASUS_MIC_NO_PRESENCE, ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS, ALC269VC_FIXUP_ACER_HEADSET_MIC, @@ -7652,6 +7720,8 @@ static const struct hda_fixup alc269_fixups[] = { { 0x20, AC_VERB_SET_PROC_COEF, 0x4e4b }, { } }, + .chained = true, + .chain_id = ALC289_FIXUP_ASUS_GA401, }, [ALC285_FIXUP_HP_GPIO_LED] = { .type = HDA_FIXUP_FUNC, @@ -7672,6 +7742,14 @@ static const struct hda_fixup alc269_fixups[] = { { } }, }, + [ALC256_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + { 0x20, AC_VERB_SET_COEF_INDEX, 0x08}, + { 0x20, AC_VERB_SET_PROC_COEF, 0x2fcf}, + { } + }, + }, [ALC295_FIXUP_ASUS_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -7905,13 +7983,8 @@ static const struct hda_fixup alc269_fixups[] = { .chain_id = ALC269_FIXUP_HP_LINE1_MIC1_LED, }, [ALC285_FIXUP_HP_SPECTRE_X360] = { - .type = HDA_FIXUP_PINS, - .v.pins = (const struct hda_pintbl[]) { - { 0x14, 0x90170110 }, /* enable top speaker */ - {} - }, - .chained = true, - .chain_id = ALC285_FIXUP_SPEAKER2_TO_DAC1, + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_hp_spectre_x360, }, }; @@ -7945,6 +8018,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x1308, "Acer Aspire Z24-890", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x132a, "Acer TravelMate B114-21", ALC233_FIXUP_ACER_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1330, "Acer TravelMate X514-51T", ALC255_FIXUP_ACER_HEADSET_MIC), + SND_PCI_QUIRK(0x1025, 0x141f, "Acer Spin SP513-54N", ALC255_FIXUP_ACER_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1025, 0x142b, "Acer Swift SF314-42", ALC255_FIXUP_ACER_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x1430, "Acer TravelMate B311R-31", ALC256_FIXUP_ACER_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), SND_PCI_QUIRK(0x1028, 0x054b, "Dell XPS one 2710", ALC275_FIXUP_DELL_XPS), @@ -8067,8 +8142,11 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x84da, "HP OMEN dc0019-ur", ALC295_FIXUP_HP_OMEN), SND_PCI_QUIRK(0x103c, 0x84e7, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x8519, "HP Spectre x360 15-df0xxx", ALC285_FIXUP_HP_SPECTRE_X360), + SND_PCI_QUIRK(0x103c, 0x860f, "HP ZBook 15 G6", ALC285_FIXUP_HP_GPIO_AMP_INIT), + SND_PCI_QUIRK(0x103c, 0x861f, "HP Elite Dragonfly G1", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x869d, "HP", ALC236_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x8724, "HP EliteBook 850 G7", ALC285_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8728, "HP EliteBook 840 G7", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8729, "HP", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8736, "HP", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8760, "HP", ALC285_FIXUP_HP_MUTE_LED), @@ -8094,10 +8172,12 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x1740, "ASUS UX430UA", ALC295_FIXUP_ASUS_DACS), SND_PCI_QUIRK(0x1043, 0x17d1, "ASUS UX431FL", ALC294_FIXUP_ASUS_DUAL_SPK), + SND_PCI_QUIRK(0x1043, 0x1662, "ASUS GV301QH", ALC294_FIXUP_ASUS_DUAL_SPK), SND_PCI_QUIRK(0x1043, 0x1881, "ASUS Zephyrus S/M", ALC294_FIXUP_ASUS_GX502_PINS), SND_PCI_QUIRK(0x1043, 0x18b1, "Asus MJ401TA", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x18f1, "Asus FX505DT", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x194e, "ASUS UX563FD", ALC294_FIXUP_ASUS_HPE), + SND_PCI_QUIRK(0x1043, 0x1970, "ASUS UX550VE", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1982, "ASUS B1400CEPE", ALC256_FIXUP_ASUS_HPE), SND_PCI_QUIRK(0x1043, 0x19ce, "ASUS B9450FA", ALC294_FIXUP_ASUS_HPE), SND_PCI_QUIRK(0x1043, 0x19e1, "ASUS UX581LV", ALC295_FIXUP_ASUS_MIC_NO_PRESENCE), @@ -8113,6 +8193,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1e51, "ASUS Zephyrus M15", ALC294_FIXUP_ASUS_GU502_PINS), SND_PCI_QUIRK(0x1043, 0x1e8e, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_GA401), + SND_PCI_QUIRK(0x1043, 0x1d42, "ASUS Zephyrus G14 2022", ALC289_FIXUP_ASUS_GA401), + SND_PCI_QUIRK(0x1043, 0x16b2, "ASUS GU603", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x3030, "ASUS ZN270IE", ALC256_FIXUP_ASUS_AIO_GPIO2), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC), @@ -8145,6 +8227,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x144d, 0xc740, "Samsung Ativ book 8 (NP870Z5G)", ALC269_FIXUP_ATIV_BOOK_8), SND_PCI_QUIRK(0x144d, 0xc812, "Samsung Notebook Pen S (NT950SBE-X58)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x144d, 0xc830, "Samsung Galaxy Book Ion (NT950XCJ-X716A)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), + SND_PCI_QUIRK(0x144d, 0xc832, "Samsung Galaxy Book Flex Alpha (NP730QCJ)", ALC256_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb171, "Cubi N 8GL (MS-B171)", ALC283_FIXUP_HEADSET_MIC), @@ -8468,6 +8551,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC298_FIXUP_HUAWEI_MBX_STEREO, .name = "huawei-mbx-stereo"}, {.id = ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE, .name = "alc256-medion-headset"}, {.id = ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, .name = "alc298-samsung-headphone"}, + {.id = ALC256_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, .name = "alc256-samsung-headphone"}, {.id = ALC255_FIXUP_XIAOMI_HEADSET_MIC, .name = "alc255-xiaomi-headset"}, {.id = ALC274_FIXUP_HP_MIC, .name = "alc274-hp-mic-detect"}, {.id = ALC295_FIXUP_HP_OMEN, .name = "alc295-hp-omen"}, @@ -9063,6 +9147,7 @@ static int patch_alc269(struct hda_codec *codec) spec->shutup = alc256_shutup; spec->init_hook = alc256_init; break; + case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: spec->codec_variant = ALC269_TYPE_ALC256; @@ -9130,6 +9215,16 @@ static int patch_alc269(struct hda_codec *codec) snd_hda_pick_fixup(codec, alc269_fixup_models, alc269_fixup_tbl, alc269_fixups); + /* FIXME: both TX300 and ROG Strix G17 have the same SSID, and + * the quirk breaks the latter (bko#214101). + * Clear the wrong entry. + */ + if (codec->fixup_id == ALC282_FIXUP_ASUS_TX300 && + codec->core.vendor_id == 0x10ec0294) { + codec_dbg(codec, "Clear wrong fixup for ASUS ROG Strix G17\n"); + codec->fixup_id = HDA_FIXUP_ID_NOT_SET; + } + snd_hda_pick_pin_fixup(codec, alc269_pin_fixup_tbl, alc269_fixups, true); snd_hda_pick_pin_fixup(codec, alc269_fallback_pin_fixup_tbl, alc269_fixups, false); snd_hda_pick_fixup(codec, NULL, alc269_fixup_vendor_tbl, @@ -9571,6 +9666,27 @@ static void alc671_fixup_hp_headset_mic2(struct hda_codec *codec, } } +static void alc897_hp_automute_hook(struct hda_codec *codec, + struct hda_jack_callback *jack) +{ + struct alc_spec *spec = codec->spec; + int vref; + + snd_hda_gen_hp_automute(codec, jack); + vref = spec->gen.hp_jack_present ? (PIN_HP | AC_PINCTL_VREF_100) : PIN_HP; + snd_hda_codec_write(codec, 0x1b, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + vref); +} + +static void alc897_fixup_lenovo_headset_mic(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->gen.hp_automute_hook = alc897_hp_automute_hook; + } +} + static const struct coef_fw alc668_coefs[] = { WRITE_COEF(0x01, 0xbebe), WRITE_COEF(0x02, 0xaaaa), WRITE_COEF(0x03, 0x0), WRITE_COEF(0x04, 0x0180), WRITE_COEF(0x06, 0x0), WRITE_COEF(0x07, 0x0f80), @@ -9648,6 +9764,11 @@ enum { ALC671_FIXUP_HP_HEADSET_MIC2, ALC662_FIXUP_ACER_X2660G_HEADSET_MODE, ALC662_FIXUP_ACER_NITRO_HEADSET_MODE, + ALC668_FIXUP_ASUS_NO_HEADSET_MIC, + ALC668_FIXUP_HEADSET_MIC, + ALC668_FIXUP_MIC_DET_COEF, + ALC897_FIXUP_LENOVO_HEADSET_MIC, + ALC897_FIXUP_HEADSET_MIC_PIN, }; static const struct hda_fixup alc662_fixups[] = { @@ -10031,6 +10152,42 @@ static const struct hda_fixup alc662_fixups[] = { .chained = true, .chain_id = ALC662_FIXUP_USI_FUNC }, + [ALC668_FIXUP_ASUS_NO_HEADSET_MIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1b, 0x04a1112c }, + { } + }, + .chained = true, + .chain_id = ALC668_FIXUP_HEADSET_MIC + }, + [ALC668_FIXUP_HEADSET_MIC] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_headset_mic, + .chained = true, + .chain_id = ALC668_FIXUP_MIC_DET_COEF + }, + [ALC668_FIXUP_MIC_DET_COEF] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + { 0x20, AC_VERB_SET_COEF_INDEX, 0x15 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0d60 }, + {} + }, + }, + [ALC897_FIXUP_LENOVO_HEADSET_MIC] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc897_fixup_lenovo_headset_mic, + }, + [ALC897_FIXUP_HEADSET_MIC_PIN] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1a, 0x03a11050 }, + { } + }, + .chained = true, + .chain_id = ALC897_FIXUP_LENOVO_HEADSET_MIC + }, }; static const struct snd_pci_quirk alc662_fixup_tbl[] = { @@ -10057,6 +10214,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x069f, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), SND_PCI_QUIRK(0x103c, 0x873e, "HP", ALC671_FIXUP_HP_HEADSET_MIC2), + SND_PCI_QUIRK(0x103c, 0x885f, "HP 288 Pro G8", ALC671_FIXUP_HP_HEADSET_MIC2), SND_PCI_QUIRK(0x1043, 0x1080, "Asus UX501VW", ALC668_FIXUP_HEADSET_MODE), SND_PCI_QUIRK(0x1043, 0x11cd, "Asus N550", ALC662_FIXUP_ASUS_Nx50), SND_PCI_QUIRK(0x1043, 0x129d, "Asus N750", ALC662_FIXUP_ASUS_Nx50), @@ -10066,6 +10224,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x15a7, "ASUS UX51VZH", ALC662_FIXUP_BASS_16), SND_PCI_QUIRK(0x1043, 0x177d, "ASUS N551", ALC668_FIXUP_ASUS_Nx51), SND_PCI_QUIRK(0x1043, 0x17bd, "ASUS N751", ALC668_FIXUP_ASUS_Nx51), + SND_PCI_QUIRK(0x1043, 0x185d, "ASUS G551JW", ALC668_FIXUP_ASUS_NO_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1963, "ASUS X71SL", ALC662_FIXUP_ASUS_MODE8), SND_PCI_QUIRK(0x1043, 0x1b73, "ASUS N55SF", ALC662_FIXUP_BASS_16), SND_PCI_QUIRK(0x1043, 0x1bf3, "ASUS N76VZ", ALC662_FIXUP_BASS_MODE4_CHMAP), @@ -10074,6 +10233,10 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD), SND_PCI_QUIRK(0x14cd, 0x5003, "USI", ALC662_FIXUP_USI_HEADSET_MODE), SND_PCI_QUIRK(0x17aa, 0x1036, "Lenovo P520", ALC662_FIXUP_LENOVO_MULTI_CODECS), + SND_PCI_QUIRK(0x17aa, 0x32ca, "Lenovo ThinkCentre M80", ALC897_FIXUP_HEADSET_MIC_PIN), + SND_PCI_QUIRK(0x17aa, 0x32cb, "Lenovo ThinkCentre M70", ALC897_FIXUP_HEADSET_MIC_PIN), + SND_PCI_QUIRK(0x17aa, 0x32cf, "Lenovo ThinkCentre M950", ALC897_FIXUP_HEADSET_MIC_PIN), + SND_PCI_QUIRK(0x17aa, 0x32f7, "Lenovo ThinkCentre M90", ALC897_FIXUP_HEADSET_MIC_PIN), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo Ideapad Y550P", ALC662_FIXUP_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Ideapad Y550", ALC662_FIXUP_IDEAPAD), SND_PCI_QUIRK(0x1849, 0x5892, "ASRock B150M", ALC892_FIXUP_ASROCK_MOBO), @@ -10354,6 +10517,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0221, "ALC221", patch_alc269), HDA_CODEC_ENTRY(0x10ec0222, "ALC222", patch_alc269), HDA_CODEC_ENTRY(0x10ec0225, "ALC225", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0230, "ALC236", patch_alc269), HDA_CODEC_ENTRY(0x10ec0231, "ALC231", patch_alc269), HDA_CODEC_ENTRY(0x10ec0233, "ALC233", patch_alc269), HDA_CODEC_ENTRY(0x10ec0234, "ALC234", patch_alc269), diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 2a73fc4fd019..5150e8d38975 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -715,7 +715,7 @@ static inline void snd_intel8x0_update(struct intel8x0 *chip, struct ichdev *ich int status, civ, i, step; int ack = 0; - if (!ichdev->prepared || ichdev->suspended) + if (!(ichdev->prepared || chip->in_measurement) || ichdev->suspended) return; spin_lock_irqsave(&chip->reg_lock, flags); diff --git a/sound/ppc/powermac.c b/sound/ppc/powermac.c index 96ef55082bf9..b135d114ce89 100644 --- a/sound/ppc/powermac.c +++ b/sound/ppc/powermac.c @@ -77,7 +77,11 @@ 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/atmel/Kconfig b/sound/soc/atmel/Kconfig index 71f2d42188c4..51e75b781968 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -11,7 +11,6 @@ if SND_ATMEL_SOC config SND_ATMEL_SOC_PDC bool - depends on HAS_DMA config SND_ATMEL_SOC_DMA bool diff --git a/sound/soc/atmel/atmel-i2s.c b/sound/soc/atmel/atmel-i2s.c index bbe2b638abb5..d870f56c44cf 100644 --- a/sound/soc/atmel/atmel-i2s.c +++ b/sound/soc/atmel/atmel-i2s.c @@ -200,6 +200,7 @@ struct atmel_i2s_dev { unsigned int fmt; const struct atmel_i2s_gck_param *gck_param; const struct atmel_i2s_caps *caps; + int clk_use_no; }; static irqreturn_t atmel_i2s_interrupt(int irq, void *dev_id) @@ -321,9 +322,16 @@ static int atmel_i2s_hw_params(struct snd_pcm_substream *substream, { struct atmel_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); bool is_playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); - unsigned int mr = 0; + unsigned int mr = 0, mr_mask; int ret; + mr_mask = ATMEL_I2SC_MR_FORMAT_MASK | ATMEL_I2SC_MR_MODE_MASK | + ATMEL_I2SC_MR_DATALENGTH_MASK; + if (is_playback) + mr_mask |= ATMEL_I2SC_MR_TXMONO; + else + mr_mask |= ATMEL_I2SC_MR_RXMONO; + switch (dev->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: mr |= ATMEL_I2SC_MR_FORMAT_I2S; @@ -402,7 +410,7 @@ static int atmel_i2s_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - return regmap_write(dev->regmap, ATMEL_I2SC_MR, mr); + return regmap_update_bits(dev->regmap, ATMEL_I2SC_MR, mr_mask, mr); } static int atmel_i2s_switch_mck_generator(struct atmel_i2s_dev *dev, @@ -495,18 +503,28 @@ static int atmel_i2s_trigger(struct snd_pcm_substream *substream, int cmd, is_master = (mr & ATMEL_I2SC_MR_MODE_MASK) == ATMEL_I2SC_MR_MODE_MASTER; /* If master starts, enable the audio clock. */ - if (is_master && mck_enabled) - err = atmel_i2s_switch_mck_generator(dev, true); - if (err) - return err; + if (is_master && mck_enabled) { + if (!dev->clk_use_no) { + err = atmel_i2s_switch_mck_generator(dev, true); + if (err) + return err; + } + dev->clk_use_no++; + } err = regmap_write(dev->regmap, ATMEL_I2SC_CR, cr); if (err) return err; /* If master stops, disable the audio clock. */ - if (is_master && !mck_enabled) - err = atmel_i2s_switch_mck_generator(dev, false); + if (is_master && !mck_enabled) { + if (dev->clk_use_no == 1) { + err = atmel_i2s_switch_mck_generator(dev, false); + if (err) + return err; + } + dev->clk_use_no--; + } return err; } diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index ca603397651c..1e0973322cd0 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -280,7 +280,10 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, /* Enable PMC peripheral clock for this SSC */ pr_debug("atmel_ssc_dai: Starting clock\n"); - clk_enable(ssc_p->ssc->clk); + ret = clk_enable(ssc_p->ssc->clk); + if (ret) + return ret; + ssc_p->mck_rate = clk_get_rate(ssc_p->ssc->clk); /* Reset the SSC unless initialized to keep it in a clean state */ diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index b1bef2bf142d..d1579896f3a1 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -46,35 +46,6 @@ */ #undef ENABLE_MIC_INPUT -static struct clk *mclk; - -static int at91sam9g20ek_set_bias_level(struct snd_soc_card *card, - struct snd_soc_dapm_context *dapm, - enum snd_soc_bias_level level) -{ - static int mclk_on; - int ret = 0; - - switch (level) { - case SND_SOC_BIAS_ON: - case SND_SOC_BIAS_PREPARE: - if (!mclk_on) - ret = clk_enable(mclk); - if (ret == 0) - mclk_on = 1; - break; - - case SND_SOC_BIAS_OFF: - case SND_SOC_BIAS_STANDBY: - if (mclk_on) - clk_disable(mclk); - mclk_on = 0; - break; - } - - return ret; -} - static const struct snd_soc_dapm_widget at91sam9g20ek_dapm_widgets[] = { SND_SOC_DAPM_MIC("Int Mic", NULL), SND_SOC_DAPM_SPK("Ext Spk", NULL), @@ -135,7 +106,6 @@ static struct snd_soc_card snd_soc_at91sam9g20ek = { .owner = THIS_MODULE, .dai_link = &at91sam9g20ek_dai, .num_links = 1, - .set_bias_level = at91sam9g20ek_set_bias_level, .dapm_widgets = at91sam9g20ek_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(at91sam9g20ek_dapm_widgets), @@ -148,7 +118,6 @@ static int at91sam9g20ek_audio_probe(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; struct device_node *codec_np, *cpu_np; - struct clk *pllb; struct snd_soc_card *card = &snd_soc_at91sam9g20ek; int ret; @@ -162,31 +131,6 @@ static int at91sam9g20ek_audio_probe(struct platform_device *pdev) return -EINVAL; } - /* - * Codec MCLK is supplied by PCK0 - set it up. - */ - mclk = clk_get(NULL, "pck0"); - if (IS_ERR(mclk)) { - dev_err(&pdev->dev, "Failed to get MCLK\n"); - ret = PTR_ERR(mclk); - goto err; - } - - pllb = clk_get(NULL, "pllb"); - if (IS_ERR(pllb)) { - dev_err(&pdev->dev, "Failed to get PLLB\n"); - ret = PTR_ERR(pllb); - goto err_mclk; - } - ret = clk_set_parent(mclk, pllb); - clk_put(pllb); - if (ret != 0) { - dev_err(&pdev->dev, "Failed to set MCLK parent\n"); - goto err_mclk; - } - - clk_set_rate(mclk, MCLK_RATE); - card->dev = &pdev->dev; /* Parse device node info */ @@ -214,6 +158,7 @@ static int at91sam9g20ek_audio_probe(struct platform_device *pdev) cpu_np = of_parse_phandle(np, "atmel,ssc-controller", 0); if (!cpu_np) { dev_err(&pdev->dev, "dai and pcm info missing\n"); + of_node_put(codec_np); return -EINVAL; } at91sam9g20ek_dai.cpus->of_node = cpu_np; @@ -229,9 +174,6 @@ static int at91sam9g20ek_audio_probe(struct platform_device *pdev) return ret; -err_mclk: - clk_put(mclk); - mclk = NULL; err: atmel_ssc_put_audio(0); return ret; @@ -241,8 +183,6 @@ static int at91sam9g20ek_audio_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); - clk_disable(mclk); - mclk = NULL; snd_soc_unregister_card(card); atmel_ssc_put_audio(0); diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 12008d3f38a7..e83c333a81cb 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -590,21 +590,26 @@ config SND_SOC_CS4349 config SND_SOC_CS47L15 tristate + depends on MFD_CS47L15 config SND_SOC_CS47L24 tristate config SND_SOC_CS47L35 tristate + depends on MFD_CS47L35 config SND_SOC_CS47L85 tristate + depends on MFD_CS47L85 config SND_SOC_CS47L90 tristate + depends on MFD_CS47L90 config SND_SOC_CS47L92 tristate + depends on MFD_CS47L92 # Cirrus Logic Quad-Channel ADC config SND_SOC_CS53L30 diff --git a/sound/soc/codecs/cpcap.c b/sound/soc/codecs/cpcap.c index 1902689c5ea2..acd88fe38cd4 100644 --- a/sound/soc/codecs/cpcap.c +++ b/sound/soc/codecs/cpcap.c @@ -1541,6 +1541,8 @@ static int cpcap_codec_probe(struct platform_device *pdev) { struct device_node *codec_node = of_get_child_by_name(pdev->dev.parent->of_node, "audio-codec"); + if (!codec_node) + return -ENODEV; pdev->dev.of_node = codec_node; diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index 2fb65f246b0c..77af5b67b9bb 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -150,7 +150,6 @@ static const struct snd_kcontrol_new cs4265_snd_controls[] = { SOC_SINGLE("E to F Buffer Disable Switch", CS4265_SPDIF_CTL1, 6, 1, 0), SOC_ENUM("C Data Access", cam_mode_enum), - SOC_SINGLE("SPDIF Switch", CS4265_SPDIF_CTL2, 5, 1, 1), SOC_SINGLE("Validity Bit Control Switch", CS4265_SPDIF_CTL2, 3, 1, 0), SOC_ENUM("SPDIF Mono/Stereo", spdif_mono_stereo_enum), @@ -186,7 +185,7 @@ static const struct snd_soc_dapm_widget cs4265_dapm_widgets[] = { SND_SOC_DAPM_SWITCH("Loopback", SND_SOC_NOPM, 0, 0, &loopback_ctl), - SND_SOC_DAPM_SWITCH("SPDIF", SND_SOC_NOPM, 0, 0, + SND_SOC_DAPM_SWITCH("SPDIF", CS4265_SPDIF_CTL2, 5, 1, &spdif_switch), SND_SOC_DAPM_SWITCH("DAC", CS4265_PWRCTL, 1, 1, &dac_switch), diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index 5faf8877137a..ebee58eca4d5 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -91,7 +91,7 @@ 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 }, @@ -128,7 +128,7 @@ 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 }, @@ -403,7 +403,7 @@ static const struct regmap_config cs42l42_regmap = { .use_single_write = true, }; -static DECLARE_TLV_DB_SCALE(adc_tlv, -9600, 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[] = { @@ -423,34 +423,23 @@ 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, @@ -669,15 +658,6 @@ static int cs42l42_pll_config(struct snd_soc_component *component) CS42L42_FSYNC_PULSE_WIDTH_MASK, CS42L42_FRAC1_VAL(fsync - 1) << CS42L42_FSYNC_PULSE_WIDTH_SHIFT); - snd_soc_component_update_bits(component, - CS42L42_ASP_FRM_CFG, - CS42L42_ASP_5050_MASK, - CS42L42_ASP_5050_MASK); - /* Set the frame delay to 1.0 SCLK clocks */ - snd_soc_component_update_bits(component, CS42L42_ASP_FRM_CFG, - CS42L42_ASP_FSD_MASK, - CS42L42_ASP_FSD_1_0 << - CS42L42_ASP_FSD_SHIFT); /* Set the sample rates (96k or lower) */ snd_soc_component_update_bits(component, CS42L42_FS_RATE_EN, CS42L42_FS_EN_MASK, @@ -773,7 +753,18 @@ 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: + /* + * 5050 mode, frame starts on falling edge of LRCLK, + * frame delayed by 1.0 SCLKs + */ + snd_soc_component_update_bits(component, + CS42L42_ASP_FRM_CFG, + CS42L42_ASP_STP_MASK | + CS42L42_ASP_5050_MASK | + CS42L42_ASP_FSD_MASK, + CS42L42_ASP_5050_MASK | + (CS42L42_ASP_FSD_1_0 << + CS42L42_ASP_FSD_SHIFT)); break; default: return -EINVAL; @@ -1807,8 +1798,9 @@ 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); diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h index 866d7c873e3c..ca2019732013 100644 --- a/sound/soc/codecs/cs42l42.h +++ b/sound/soc/codecs/cs42l42.h @@ -77,7 +77,7 @@ #define CS42L42_HP_PDN_SHIFT 3 #define CS42L42_HP_PDN_MASK (1 << CS42L42_HP_PDN_SHIFT) #define CS42L42_ADC_PDN_SHIFT 2 -#define CS42L42_ADC_PDN_MASK (1 << CS42L42_HP_PDN_SHIFT) +#define CS42L42_ADC_PDN_MASK (1 << CS42L42_ADC_PDN_SHIFT) #define CS42L42_PDN_ALL_SHIFT 0 #define CS42L42_PDN_ALL_MASK (1 << CS42L42_PDN_ALL_SHIFT) diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index f83a6eaba12c..ef8bd9e04637 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -446,7 +446,7 @@ static int da7219_tonegen_freq_put(struct snd_kcontrol *kcontrol, struct soc_mixer_control *mixer_ctrl = (struct soc_mixer_control *) kcontrol->private_value; unsigned int reg = mixer_ctrl->reg; - __le16 val; + __le16 val_new, val_old; int ret; /* @@ -454,13 +454,19 @@ static int da7219_tonegen_freq_put(struct snd_kcontrol *kcontrol, * Therefore we need to convert to little endian here to align with * HW registers. */ - val = cpu_to_le16(ucontrol->value.integer.value[0]); + val_new = cpu_to_le16(ucontrol->value.integer.value[0]); mutex_lock(&da7219->ctrl_lock); - ret = regmap_raw_write(da7219->regmap, reg, &val, sizeof(val)); + ret = regmap_raw_read(da7219->regmap, reg, &val_old, sizeof(val_old)); + if (ret == 0 && (val_old != val_new)) + ret = regmap_raw_write(da7219->regmap, reg, + &val_new, sizeof(val_new)); mutex_unlock(&da7219->ctrl_lock); - return ret; + if (ret < 0) + return ret; + + return val_old != val_new; } diff --git a/sound/soc/codecs/max9759.c b/sound/soc/codecs/max9759.c index 00e9d4fd1651..0c261335c8a1 100644 --- a/sound/soc/codecs/max9759.c +++ b/sound/soc/codecs/max9759.c @@ -64,7 +64,8 @@ static int speaker_gain_control_put(struct snd_kcontrol *kcontrol, struct snd_soc_component *c = snd_soc_kcontrol_component(kcontrol); struct max9759 *priv = snd_soc_component_get_drvdata(c); - if (ucontrol->value.integer.value[0] > 3) + if (ucontrol->value.integer.value[0] < 0 || + ucontrol->value.integer.value[0] > 3) return -EINVAL; priv->gain = ucontrol->value.integer.value[0]; diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index cf6516693e4e..5a8eedea6be0 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -1196,8 +1196,8 @@ static int pm8916_wcd_analog_spmi_probe(struct platform_device *pdev) irq = platform_get_irq_byname(pdev, "mbhc_switch_int"); if (irq < 0) { - dev_err(dev, "failed to get mbhc switch irq\n"); - return irq; + ret = irq; + goto err_disable_clk; } ret = devm_request_threaded_irq(dev, irq, NULL, @@ -1211,8 +1211,8 @@ static int pm8916_wcd_analog_spmi_probe(struct platform_device *pdev) if (priv->mbhc_btn_enabled) { irq = platform_get_irq_byname(pdev, "mbhc_but_press_det"); if (irq < 0) { - dev_err(dev, "failed to get button press irq\n"); - return irq; + ret = irq; + goto err_disable_clk; } ret = devm_request_threaded_irq(dev, irq, NULL, @@ -1225,8 +1225,8 @@ static int pm8916_wcd_analog_spmi_probe(struct platform_device *pdev) irq = platform_get_irq_byname(pdev, "mbhc_but_rel_det"); if (irq < 0) { - dev_err(dev, "failed to get button release irq\n"); - return irq; + ret = irq; + goto err_disable_clk; } ret = devm_request_threaded_irq(dev, irq, NULL, @@ -1244,6 +1244,10 @@ static int pm8916_wcd_analog_spmi_probe(struct platform_device *pdev) return devm_snd_soc_register_component(dev, &pm8916_wcd_analog, pm8916_wcd_analog_dai, ARRAY_SIZE(pm8916_wcd_analog_dai)); + +err_disable_clk: + clk_disable_unprepare(priv->mclk); + return ret; } static int pm8916_wcd_analog_spmi_remove(struct platform_device *pdev) diff --git a/sound/soc/codecs/msm8916-wcd-digital.c b/sound/soc/codecs/msm8916-wcd-digital.c index 09fccacadd6b..e4cde214b7b2 100644 --- a/sound/soc/codecs/msm8916-wcd-digital.c +++ b/sound/soc/codecs/msm8916-wcd-digital.c @@ -1201,14 +1201,24 @@ static int msm8916_wcd_digital_probe(struct platform_device *pdev) ret = clk_prepare_enable(priv->mclk); if (ret < 0) { dev_err(dev, "failed to enable mclk %d\n", ret); - return ret; + goto err_clk; } dev_set_drvdata(dev, priv); - return devm_snd_soc_register_component(dev, &msm8916_wcd_digital, + ret = devm_snd_soc_register_component(dev, &msm8916_wcd_digital, msm8916_wcd_digital_dai, ARRAY_SIZE(msm8916_wcd_digital_dai)); + if (ret) + goto err_mclk; + + return 0; + +err_mclk: + clk_disable_unprepare(priv->mclk); +err_clk: + clk_disable_unprepare(priv->ahbclk); + return ret; } static int msm8916_wcd_digital_remove(struct platform_device *pdev) diff --git a/sound/soc/codecs/mt6358.c b/sound/soc/codecs/mt6358.c index bb737fd678cc..494ba0eeb433 100644 --- a/sound/soc/codecs/mt6358.c +++ b/sound/soc/codecs/mt6358.c @@ -103,6 +103,7 @@ int mt6358_set_mtkaif_protocol(struct snd_soc_component *cmpnt, priv->mtkaif_protocol = mtkaif_protocol; return 0; } +EXPORT_SYMBOL_GPL(mt6358_set_mtkaif_protocol); static void playback_gpio_set(struct mt6358_priv *priv) { @@ -269,6 +270,7 @@ int mt6358_mtkaif_calibration_enable(struct snd_soc_component *cmpnt) 1 << RG_AUD_PAD_TOP_DAT_MISO_LOOPBACK_SFT); return 0; } +EXPORT_SYMBOL_GPL(mt6358_mtkaif_calibration_enable); int mt6358_mtkaif_calibration_disable(struct snd_soc_component *cmpnt) { @@ -292,6 +294,7 @@ int mt6358_mtkaif_calibration_disable(struct snd_soc_component *cmpnt) capture_gpio_reset(priv); return 0; } +EXPORT_SYMBOL_GPL(mt6358_mtkaif_calibration_disable); int mt6358_set_mtkaif_calibration_phase(struct snd_soc_component *cmpnt, int phase_1, int phase_2) @@ -306,6 +309,7 @@ int mt6358_set_mtkaif_calibration_phase(struct snd_soc_component *cmpnt, phase_2 << RG_AUD_PAD_TOP_PHASE_MODE2_SFT); return 0; } +EXPORT_SYMBOL_GPL(mt6358_set_mtkaif_calibration_phase); /* dl pga gain */ enum { diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index 15bd8335f667..c8ccfa2fff84 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -8,6 +8,7 @@ #include <linux/module.h> #include <linux/delay.h> +#include <linux/dmi.h> #include <linux/init.h> #include <linux/i2c.h> #include <linux/regmap.h> @@ -27,6 +28,12 @@ #include "nau8824.h" +#define NAU8824_JD_ACTIVE_HIGH BIT(0) + +static int nau8824_quirk; +static int quirk_override = -1; +module_param_named(quirk, quirk_override, uint, 0444); +MODULE_PARM_DESC(quirk, "Board-specific quirk override"); static int nau8824_config_sysclk(struct nau8824 *nau8824, int clk_id, unsigned int freq); @@ -1875,6 +1882,34 @@ static int nau8824_read_device_properties(struct device *dev, return 0; } +/* Please keep this list alphabetically sorted */ +static const struct dmi_system_id nau8824_quirk_table[] = { + { + /* Cyberbook T116 rugged tablet */ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "Default string"), + DMI_EXACT_MATCH(DMI_BOARD_NAME, "Cherry Trail CR"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "20170531"), + }, + .driver_data = (void *)(NAU8824_JD_ACTIVE_HIGH), + }, + {} +}; + +static void nau8824_check_quirks(void) +{ + const struct dmi_system_id *dmi_id; + + if (quirk_override != -1) { + nau8824_quirk = quirk_override; + return; + } + + dmi_id = dmi_first_match(nau8824_quirk_table); + if (dmi_id) + nau8824_quirk = (unsigned long)dmi_id->driver_data; +} + static int nau8824_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1899,6 +1934,11 @@ static int nau8824_i2c_probe(struct i2c_client *i2c, nau8824->irq = i2c->irq; sema_init(&nau8824->jd_sem, 1); + nau8824_check_quirks(); + + if (nau8824_quirk & NAU8824_JD_ACTIVE_HIGH) + nau8824->jkdet_polarity = 0; + nau8824_print_device_properties(nau8824); ret = regmap_read(nau8824->regmap, NAU8824_REG_I2C_DEVICE_ID, &value); diff --git a/sound/soc/codecs/rk3328_codec.c b/sound/soc/codecs/rk3328_codec.c index 287c962ba00d..514ebe16bbfa 100644 --- a/sound/soc/codecs/rk3328_codec.c +++ b/sound/soc/codecs/rk3328_codec.c @@ -472,7 +472,8 @@ static int rk3328_platform_probe(struct platform_device *pdev) rk3328->pclk = devm_clk_get(&pdev->dev, "pclk"); if (IS_ERR(rk3328->pclk)) { dev_err(&pdev->dev, "can't get acodec pclk\n"); - return PTR_ERR(rk3328->pclk); + ret = PTR_ERR(rk3328->pclk); + goto err_unprepare_mclk; } ret = clk_prepare_enable(rk3328->pclk); @@ -482,19 +483,34 @@ static int rk3328_platform_probe(struct platform_device *pdev) } base = devm_platform_ioremap_resource(pdev, 0); - if (IS_ERR(base)) - return PTR_ERR(base); + if (IS_ERR(base)) { + ret = PTR_ERR(base); + goto err_unprepare_pclk; + } rk3328->regmap = devm_regmap_init_mmio(&pdev->dev, base, &rk3328_codec_regmap_config); - if (IS_ERR(rk3328->regmap)) - return PTR_ERR(rk3328->regmap); + if (IS_ERR(rk3328->regmap)) { + ret = PTR_ERR(rk3328->regmap); + goto err_unprepare_pclk; + } platform_set_drvdata(pdev, rk3328); - return devm_snd_soc_register_component(&pdev->dev, &soc_codec_rk3328, + ret = devm_snd_soc_register_component(&pdev->dev, &soc_codec_rk3328, rk3328_dai, ARRAY_SIZE(rk3328_dai)); + if (ret) + goto err_unprepare_pclk; + + return 0; + +err_unprepare_pclk: + clk_disable_unprepare(rk3328->pclk); + +err_unprepare_mclk: + clk_disable_unprepare(rk3328->mclk); + return ret; } static const struct of_device_id rk3328_codec_of_match[] = { diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index f70b9f7e68bb..281957a8fa86 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -1691,6 +1691,8 @@ static const struct regmap_config rt5631_regmap_config = { .reg_defaults = rt5631_reg, .num_reg_defaults = ARRAY_SIZE(rt5631_reg), .cache_type = REGCACHE_RBTREE, + .use_single_read = true, + .use_single_write = true, }; static int rt5631_i2c_probe(struct i2c_client *i2c, diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index 2943692f66ed..19e2f622718d 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -3461,6 +3461,7 @@ static void rt5663_calibrate(struct rt5663_priv *rt5663) static int rt5663_parse_dp(struct rt5663_priv *rt5663, struct device *dev) { int table_size; + int ret; device_property_read_u32(dev, "realtek,dc_offset_l_manual", &rt5663->pdata.dc_offset_l_manual); @@ -3477,9 +3478,13 @@ static int rt5663_parse_dp(struct rt5663_priv *rt5663, struct device *dev) table_size = sizeof(struct impedance_mapping_table) * rt5663->pdata.impedance_sensing_num; rt5663->imp_table = devm_kzalloc(dev, table_size, GFP_KERNEL); - device_property_read_u32_array(dev, + if (!rt5663->imp_table) + return -ENOMEM; + ret = device_property_read_u32_array(dev, "realtek,impedance_sensing_table", (u32 *)rt5663->imp_table, table_size); + if (ret) + return ret; } return 0; @@ -3504,8 +3509,11 @@ static int rt5663_i2c_probe(struct i2c_client *i2c, if (pdata) rt5663->pdata = *pdata; - else - rt5663_parse_dp(rt5663, &i2c->dev); + else { + ret = rt5663_parse_dp(rt5663, &i2c->dev); + if (ret) + return ret; + } for (i = 0; i < ARRAY_SIZE(rt5663->supplies); i++) rt5663->supplies[i].supply = rt5663_supply_names[i]; diff --git a/sound/soc/codecs/rt5668.c b/sound/soc/codecs/rt5668.c index 5716cede99cb..acc2b34ca334 100644 --- a/sound/soc/codecs/rt5668.c +++ b/sound/soc/codecs/rt5668.c @@ -1022,11 +1022,13 @@ static void rt5668_jack_detect_handler(struct work_struct *work) container_of(work, struct rt5668_priv, jack_detect_work.work); int val, btn_type; - while (!rt5668->component) - usleep_range(10000, 15000); - - while (!rt5668->component->card->instantiated) - usleep_range(10000, 15000); + if (!rt5668->component || !rt5668->component->card || + !rt5668->component->card->instantiated) { + /* card not yet ready, try later */ + mod_delayed_work(system_power_efficient_wq, + &rt5668->jack_detect_work, msecs_to_jiffies(15)); + return; + } mutex_lock(&rt5668->calibrate_mutex); diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 05e883a65d7a..a8cf4c745130 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -1052,11 +1052,13 @@ static void rt5682_jack_detect_handler(struct work_struct *work) container_of(work, struct rt5682_priv, jack_detect_work.work); int val, btn_type; - while (!rt5682->component) - usleep_range(10000, 15000); - - while (!rt5682->component->card->instantiated) - usleep_range(10000, 15000); + if (!rt5682->component || !rt5682->component->card || + !rt5682->component->card->instantiated) { + /* card not yet ready, try later */ + mod_delayed_work(system_power_efficient_wq, + &rt5682->jack_detect_work, msecs_to_jiffies(15)); + return; + } mutex_lock(&rt5682->calibrate_mutex); diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 130efc243b38..385a885dbc3d 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1814,6 +1814,9 @@ static int sgtl5000_i2c_remove(struct i2c_client *client) { struct sgtl5000_priv *sgtl5000 = i2c_get_clientdata(client); + regmap_write(sgtl5000->regmap, SGTL5000_CHIP_DIG_POWER, SGTL5000_DIG_POWER_DEFAULT); + regmap_write(sgtl5000->regmap, SGTL5000_CHIP_ANA_POWER, SGTL5000_ANA_POWER_DEFAULT); + clk_disable_unprepare(sgtl5000->mclk); regulator_bulk_disable(sgtl5000->num_supplies, sgtl5000->supplies); regulator_bulk_free(sgtl5000->num_supplies, sgtl5000->supplies); @@ -1821,6 +1824,11 @@ static int sgtl5000_i2c_remove(struct i2c_client *client) return 0; } +static void sgtl5000_i2c_shutdown(struct i2c_client *client) +{ + sgtl5000_i2c_remove(client); +} + static const struct i2c_device_id sgtl5000_id[] = { {"sgtl5000", 0}, {}, @@ -1841,6 +1849,7 @@ static struct i2c_driver sgtl5000_i2c_driver = { }, .probe = sgtl5000_i2c_probe, .remove = sgtl5000_i2c_remove, + .shutdown = sgtl5000_i2c_shutdown, .id_table = sgtl5000_id, }; diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h index 56ec5863f250..3a808c762299 100644 --- a/sound/soc/codecs/sgtl5000.h +++ b/sound/soc/codecs/sgtl5000.h @@ -80,6 +80,7 @@ /* * SGTL5000_CHIP_DIG_POWER */ +#define SGTL5000_DIG_POWER_DEFAULT 0x0000 #define SGTL5000_ADC_EN 0x0040 #define SGTL5000_DAC_EN 0x0020 #define SGTL5000_DAP_POWERUP 0x0010 diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h index cb024955c978..73c5f6c8ed69 100644 --- a/sound/soc/codecs/tlv320aic31xx.h +++ b/sound/soc/codecs/tlv320aic31xx.h @@ -151,8 +151,8 @@ struct aic31xx_pdata { #define AIC31XX_WORD_LEN_24BITS 0x02 #define AIC31XX_WORD_LEN_32BITS 0x03 #define AIC31XX_IFACE1_MASTER_MASK GENMASK(3, 2) -#define AIC31XX_BCLK_MASTER BIT(2) -#define AIC31XX_WCLK_MASTER BIT(3) +#define AIC31XX_BCLK_MASTER BIT(3) +#define AIC31XX_WCLK_MASTER BIT(2) /* AIC31XX_DATA_OFFSET */ #define AIC31XX_DATA_OFFSET_MASK GENMASK(7, 0) diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index 81906c25e4a8..016aff97e2fb 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -4076,6 +4076,16 @@ static int wcd9335_setup_irqs(struct wcd9335_codec *wcd) return ret; } +static void wcd9335_teardown_irqs(struct wcd9335_codec *wcd) +{ + int i; + + /* disable interrupts on all slave ports */ + for (i = 0; i < WCD9335_SLIM_NUM_PORT_REG; i++) + regmap_write(wcd->if_regmap, WCD9335_SLIM_PGD_PORT_INT_EN0 + i, + 0x00); +} + static void wcd9335_cdc_sido_ccl_enable(struct wcd9335_codec *wcd, bool ccl_flag) { @@ -4844,6 +4854,7 @@ static void wcd9335_codec_init(struct snd_soc_component *component) static int wcd9335_codec_probe(struct snd_soc_component *component) { struct wcd9335_codec *wcd = dev_get_drvdata(component->dev); + int ret; int i; snd_soc_component_init_regmap(component, wcd->regmap); @@ -4861,7 +4872,15 @@ static int wcd9335_codec_probe(struct snd_soc_component *component) for (i = 0; i < NUM_CODEC_DAIS; i++) INIT_LIST_HEAD(&wcd->dai[i].slim_ch_list); - return wcd9335_setup_irqs(wcd); + ret = wcd9335_setup_irqs(wcd); + if (ret) + goto free_clsh_ctrl; + + return 0; + +free_clsh_ctrl: + wcd_clsh_ctrl_free(wcd->clsh_ctrl); + return ret; } static void wcd9335_codec_remove(struct snd_soc_component *comp) @@ -4869,7 +4888,7 @@ static void wcd9335_codec_remove(struct snd_soc_component *comp) struct wcd9335_codec *wcd = dev_get_drvdata(comp->dev); wcd_clsh_ctrl_free(wcd->clsh_ctrl); - free_irq(regmap_irq_get_virq(wcd->irq_data, WCD9335_IRQ_SLIMBUS), wcd); + wcd9335_teardown_irqs(wcd); } static int wcd9335_codec_set_sysclk(struct snd_soc_component *comp, diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index fe99584c917f..9cd91bb0a902 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1535,18 +1535,38 @@ static int wm8350_component_probe(struct snd_soc_component *component) wm8350_clear_bits(wm8350, WM8350_JACK_DETECT, WM8350_JDL_ENA | WM8350_JDR_ENA); - wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L, + ret = wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L, wm8350_hpl_jack_handler, 0, "Left jack detect", priv); - wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R, + if (ret != 0) + goto err; + + ret = wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R, wm8350_hpr_jack_handler, 0, "Right jack detect", priv); - wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_MICSCD, + if (ret != 0) + goto free_jck_det_l; + + ret = wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_MICSCD, wm8350_mic_handler, 0, "Microphone short", priv); - wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_MICD, + if (ret != 0) + goto free_jck_det_r; + + ret = wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_MICD, wm8350_mic_handler, 0, "Microphone detect", priv); + if (ret != 0) + goto free_micscd; return 0; + +free_micscd: + wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_MICSCD, priv); +free_jck_det_r: + wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R, priv); +free_jck_det_l: + wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L, priv); +err: + return ret; } static void wm8350_component_remove(struct snd_soc_component *component) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 6fd1bef848ed..fa55d79b39b6 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -601,7 +601,7 @@ static int wm8731_hw_init(struct device *dev, struct wm8731_priv *wm8731) ret = wm8731_reset(wm8731->regmap); if (ret < 0) { dev_err(dev, "Failed to issue reset: %d\n", ret); - goto err_regulator_enable; + goto err; } /* Clear POWEROFF, keep everything else disabled */ @@ -618,10 +618,7 @@ static int wm8731_hw_init(struct device *dev, struct wm8731_priv *wm8731) regcache_mark_dirty(wm8731->regmap); -err_regulator_enable: - /* Regulators will be enabled by bias management */ - regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); - +err: return ret; } @@ -765,21 +762,27 @@ static int wm8731_i2c_probe(struct i2c_client *i2c, ret = PTR_ERR(wm8731->regmap); dev_err(&i2c->dev, "Failed to allocate register map: %d\n", ret); - return ret; + goto err_regulator_enable; } ret = wm8731_hw_init(&i2c->dev, wm8731); if (ret != 0) - return ret; + goto err_regulator_enable; ret = devm_snd_soc_register_component(&i2c->dev, &soc_component_dev_wm8731, &wm8731_dai, 1); if (ret != 0) { dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret); - return ret; + goto err_regulator_enable; } return 0; + +err_regulator_enable: + /* Regulators will be enabled by bias management */ + regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); + + return ret; } static int wm8731_i2c_remove(struct i2c_client *client) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index b174a9381c0c..149cfa594b76 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -697,6 +697,7 @@ static int out_pga_event(struct snd_soc_dapm_widget *w, int dcs_mask; int dcs_l, dcs_r; int dcs_l_reg, dcs_r_reg; + int an_out_reg; int timeout; int pwr_reg; @@ -712,6 +713,7 @@ static int out_pga_event(struct snd_soc_dapm_widget *w, dcs_mask = WM8904_DCS_ENA_CHAN_0 | WM8904_DCS_ENA_CHAN_1; dcs_r_reg = WM8904_DC_SERVO_8; dcs_l_reg = WM8904_DC_SERVO_9; + an_out_reg = WM8904_ANALOGUE_OUT1_LEFT; dcs_l = 0; dcs_r = 1; break; @@ -720,6 +722,7 @@ static int out_pga_event(struct snd_soc_dapm_widget *w, dcs_mask = WM8904_DCS_ENA_CHAN_2 | WM8904_DCS_ENA_CHAN_3; dcs_r_reg = WM8904_DC_SERVO_6; dcs_l_reg = WM8904_DC_SERVO_7; + an_out_reg = WM8904_ANALOGUE_OUT2_LEFT; dcs_l = 2; dcs_r = 3; break; @@ -792,6 +795,10 @@ static int out_pga_event(struct snd_soc_dapm_widget *w, snd_soc_component_update_bits(component, reg, WM8904_HPL_ENA_OUTP | WM8904_HPR_ENA_OUTP, WM8904_HPL_ENA_OUTP | WM8904_HPR_ENA_OUTP); + + /* Update volume, requires PGA to be powered */ + val = snd_soc_component_read32(component, an_out_reg); + snd_soc_component_write(component, an_out_reg, val); break; case SND_SOC_DAPM_POST_PMU: diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index 04f23477039a..c677c068b05e 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -534,7 +534,7 @@ static int wm8958_mbc_put(struct snd_kcontrol *kcontrol, wm8958_dsp_apply(component, mbc, wm8994->mbc_ena[mbc]); - return 0; + return 1; } #define WM8958_MBC_SWITCH(xname, xval) {\ @@ -660,7 +660,7 @@ static int wm8958_vss_put(struct snd_kcontrol *kcontrol, wm8958_dsp_apply(component, vss, wm8994->vss_ena[vss]); - return 0; + return 1; } @@ -734,7 +734,7 @@ static int wm8958_hpf_put(struct snd_kcontrol *kcontrol, wm8958_dsp_apply(component, hpf % 3, ucontrol->value.integer.value[0]); - return 0; + return 1; } #define WM8958_HPF_SWITCH(xname, xval) {\ @@ -828,7 +828,7 @@ static int wm8958_enh_eq_put(struct snd_kcontrol *kcontrol, wm8958_dsp_apply(component, eq, ucontrol->value.integer.value[0]); - return 0; + return 1; } #define WM8958_ENH_EQ_SWITCH(xname, xval) {\ diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 5ead3633f794..cf338ad9cddd 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -730,9 +730,16 @@ static int wm8960_configure_clocking(struct snd_soc_component *component) int i, j, k; int ret; - if (!(iface1 & (1<<6))) { - dev_dbg(component->dev, - "Codec is slave mode, no need to configure clock\n"); + /* + * For Slave mode clocking should still be configured, + * so this if statement should be removed, but some platform + * may not work if the sysclk is not configured, to avoid such + * compatible issue, just add '!wm8960->sysclk' condition in + * this if statement. + */ + if (!(iface1 & (1 << 6)) && !wm8960->sysclk) { + dev_warn(component->dev, + "slave mode, but proceeding with no clock configuration\n"); return 0; } diff --git a/sound/soc/fsl/imx-es8328.c b/sound/soc/fsl/imx-es8328.c index fad1eb6253d5..9e602c345619 100644 --- a/sound/soc/fsl/imx-es8328.c +++ b/sound/soc/fsl/imx-es8328.c @@ -87,6 +87,7 @@ static int imx_es8328_probe(struct platform_device *pdev) if (int_port > MUX_PORT_MAX || int_port == 0) { dev_err(dev, "mux-int-port: hardware only has %d mux ports\n", MUX_PORT_MAX); + ret = -EINVAL; goto fail; } diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c index af3c3b90c0ac..83b4a22bf15a 100644 --- a/sound/soc/fsl/pcm030-audio-fabric.c +++ b/sound/soc/fsl/pcm030-audio-fabric.c @@ -93,16 +93,21 @@ static int pcm030_fabric_probe(struct platform_device *op) dev_err(&op->dev, "platform_device_alloc() failed\n"); ret = platform_device_add(pdata->codec_device); - if (ret) + if (ret) { dev_err(&op->dev, "platform_device_add() failed: %d\n", ret); + platform_device_put(pdata->codec_device); + } ret = snd_soc_register_card(card); - if (ret) + if (ret) { dev_err(&op->dev, "snd_soc_register_card() failed: %d\n", ret); + platform_device_del(pdata->codec_device); + platform_device_put(pdata->codec_device); + } platform_set_drvdata(op, pdata); - return ret; + } static int pcm030_fabric_remove(struct platform_device *op) diff --git a/sound/soc/hisilicon/hi6210-i2s.c b/sound/soc/hisilicon/hi6210-i2s.c index ab3b76d298b3..03470e8f3008 100644 --- a/sound/soc/hisilicon/hi6210-i2s.c +++ b/sound/soc/hisilicon/hi6210-i2s.c @@ -102,18 +102,15 @@ 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 */ @@ -165,6 +162,11 @@ 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/img/img-i2s-in.c b/sound/soc/img/img-i2s-in.c index bb668551dd4b..243f916355ee 100644 --- a/sound/soc/img/img-i2s-in.c +++ b/sound/soc/img/img-i2s-in.c @@ -464,7 +464,7 @@ static int img_i2s_in_probe(struct platform_device *pdev) if (ret) goto err_pm_disable; } - ret = pm_runtime_get_sync(&pdev->dev); + ret = pm_runtime_resume_and_get(&pdev->dev); if (ret < 0) goto err_suspend; diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index c3ff203c3f44..7d59846808b5 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -127,7 +127,7 @@ 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; @@ -233,7 +233,6 @@ 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; diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index c67b86e2d0c0..7830d014d924 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -284,9 +284,6 @@ static const struct snd_soc_dapm_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"}, @@ -294,19 +291,23 @@ 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"}, }; @@ -348,6 +349,7 @@ 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"}, @@ -355,6 +357,7 @@ 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/kbl_da7219_max98357a.c b/sound/soc/intel/boards/kbl_da7219_max98357a.c index 537a88932bb6..69362eae65be 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98357a.c +++ b/sound/soc/intel/boards/kbl_da7219_max98357a.c @@ -607,7 +607,7 @@ static int kabylake_audio_probe(struct platform_device *pdev) static const struct platform_device_id kbl_board_ids[] = { { - .name = "kbl_da7219_max98357a", + .name = "kbl_da7219_mx98357a", .driver_data = (kernel_ulong_t)&kabylake_audio_card_da7219_m98357a, }, @@ -629,4 +629,4 @@ module_platform_driver(kabylake_audio) MODULE_DESCRIPTION("Audio Machine driver-DA7219 & MAX98357A in I2S mode"); MODULE_AUTHOR("Naveen Manohar <naveen.m@intel.com>"); MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:kbl_da7219_max98357a"); +MODULE_ALIAS("platform:kbl_da7219_mx98357a"); diff --git a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c index e200baa11011..df7f82e55a5a 100644 --- a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c @@ -113,7 +113,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[] = { }, { .id = "DLGS7219", - .drv_name = "kbl_da7219_max98373", + .drv_name = "kbl_da7219_mx98373", .fw_filename = "intel/dsp_fw_kbl.bin", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &kbl_7219_98373_codecs, diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c index 476ef1897961..79c6cf2c14bf 100644 --- a/sound/soc/intel/skylake/skl-messages.c +++ b/sound/soc/intel/skylake/skl-messages.c @@ -802,9 +802,12 @@ static u16 skl_get_module_param_size(struct skl_dev *skl, case SKL_MODULE_TYPE_BASE_OUTFMT: case SKL_MODULE_TYPE_MIC_SELECT: - case SKL_MODULE_TYPE_KPB: return sizeof(struct skl_base_outfmt_cfg); + case SKL_MODULE_TYPE_MIXER: + case SKL_MODULE_TYPE_KPB: + return sizeof(struct skl_base_cfg); + default: /* * return only base cfg when no specific module type is @@ -857,10 +860,14 @@ static int skl_set_module_format(struct skl_dev *skl, case SKL_MODULE_TYPE_BASE_OUTFMT: case SKL_MODULE_TYPE_MIC_SELECT: - case SKL_MODULE_TYPE_KPB: skl_set_base_outfmt_format(skl, module_config, *param_data); break; + case SKL_MODULE_TYPE_MIXER: + case SKL_MODULE_TYPE_KPB: + skl_set_base_module_format(skl, module_config, *param_data); + break; + default: skl_set_base_module_format(skl, module_config, *param_data); break; diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 7f287424af9b..439dd4ba690c 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -1333,21 +1333,6 @@ static int skl_get_module_info(struct skl_dev *skl, return -EIO; } - list_for_each_entry(module, &skl->uuid_list, list) { - if (guid_equal(uuid_mod, &module->uuid)) { - mconfig->id.module_id = module->id; - if (mconfig->module) - mconfig->module->loadable = module->is_loadable; - ret = 0; - break; - } - } - - if (ret) - return ret; - - uuid_mod = &module->uuid; - ret = -EIO; for (i = 0; i < skl->nr_modules; i++) { skl_module = skl->modules[i]; uuid_tplg = &skl_module->uuid; @@ -1357,10 +1342,18 @@ static int skl_get_module_info(struct skl_dev *skl, break; } } + if (skl->nr_modules && ret) return ret; + ret = -EIO; list_for_each_entry(module, &skl->uuid_list, list) { + if (guid_equal(uuid_mod, &module->uuid)) { + mconfig->id.module_id = module->id; + mconfig->module->loadable = module->is_loadable; + ret = 0; + } + for (i = 0; i < MAX_IN_QUEUE; i++) { pin_id = &mconfig->m_in_pin[i].id; if (guid_equal(&pin_id->mod_uuid, &module->uuid)) @@ -1374,7 +1367,7 @@ static int skl_get_module_info(struct skl_dev *skl, } } - return 0; + return ret; } static int skl_populate_modules(struct skl_dev *skl) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 1940b17f27ef..254b796e635d 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -113,7 +113,7 @@ static int is_skl_dsp_widget_type(struct snd_soc_dapm_widget *w, static void skl_dump_mconfig(struct skl_dev *skl, struct skl_module_cfg *mcfg) { - struct skl_module_iface *iface = &mcfg->module->formats[0]; + struct skl_module_iface *iface = &mcfg->module->formats[mcfg->fmt_idx]; dev_dbg(skl->dev, "Dumping config\n"); dev_dbg(skl->dev, "Input Format:\n"); @@ -195,8 +195,8 @@ static void skl_tplg_update_params_fixup(struct skl_module_cfg *m_cfg, struct skl_module_fmt *in_fmt, *out_fmt; /* Fixups will be applied to pin 0 only */ - in_fmt = &m_cfg->module->formats[0].inputs[0].fmt; - out_fmt = &m_cfg->module->formats[0].outputs[0].fmt; + in_fmt = &m_cfg->module->formats[m_cfg->fmt_idx].inputs[0].fmt; + out_fmt = &m_cfg->module->formats[m_cfg->fmt_idx].outputs[0].fmt; if (params->stream == SNDRV_PCM_STREAM_PLAYBACK) { if (is_fe) { @@ -239,9 +239,9 @@ static void skl_tplg_update_buffer_size(struct skl_dev *skl, /* Since fixups is applied to pin 0 only, ibs, obs needs * change for pin 0 only */ - res = &mcfg->module->resources[0]; - in_fmt = &mcfg->module->formats[0].inputs[0].fmt; - out_fmt = &mcfg->module->formats[0].outputs[0].fmt; + res = &mcfg->module->resources[mcfg->res_idx]; + in_fmt = &mcfg->module->formats[mcfg->fmt_idx].inputs[0].fmt; + out_fmt = &mcfg->module->formats[mcfg->fmt_idx].outputs[0].fmt; if (mcfg->m_type == SKL_MODULE_TYPE_SRCINT) multiplier = 5; @@ -1463,12 +1463,6 @@ static int skl_tplg_tlv_control_set(struct snd_kcontrol *kcontrol, struct skl_dev *skl = get_skl_ctx(w->dapm->dev); if (ac->params) { - /* - * Widget data is expected to be stripped of T and L - */ - size -= 2 * sizeof(unsigned int); - data += 2; - if (size > ac->max) return -EINVAL; ac->size = size; @@ -1637,11 +1631,12 @@ int skl_tplg_update_pipe_params(struct device *dev, struct skl_module_cfg *mconfig, struct skl_pipe_params *params) { - struct skl_module_res *res = &mconfig->module->resources[0]; + struct skl_module_res *res; struct skl_dev *skl = get_skl_ctx(dev); struct skl_module_fmt *format = NULL; u8 cfg_idx = mconfig->pipe->cur_config_idx; + res = &mconfig->module->resources[mconfig->res_idx]; skl_tplg_fill_dma_id(mconfig, params); mconfig->fmt_idx = mconfig->mod_cfg[cfg_idx].fmt_idx; mconfig->res_idx = mconfig->mod_cfg[cfg_idx].res_idx; @@ -1650,9 +1645,9 @@ int skl_tplg_update_pipe_params(struct device *dev, return 0; if (params->stream == SNDRV_PCM_STREAM_PLAYBACK) - format = &mconfig->module->formats[0].inputs[0].fmt; + format = &mconfig->module->formats[mconfig->fmt_idx].inputs[0].fmt; else - format = &mconfig->module->formats[0].outputs[0].fmt; + format = &mconfig->module->formats[mconfig->fmt_idx].outputs[0].fmt; /* set the hw_params */ format->s_freq = params->s_freq; diff --git a/sound/soc/mediatek/common/mtk-btcvsd.c b/sound/soc/mediatek/common/mtk-btcvsd.c index c7a81c4be068..5b47cf5d7ead 100644 --- a/sound/soc/mediatek/common/mtk-btcvsd.c +++ b/sound/soc/mediatek/common/mtk-btcvsd.c @@ -1302,7 +1302,7 @@ static const struct snd_soc_component_driver mtk_btcvsd_snd_platform = { static int mtk_btcvsd_snd_probe(struct platform_device *pdev) { - int ret = 0; + int ret; int irq_id; u32 offset[5] = {0, 0, 0, 0, 0}; struct mtk_btcvsd_snd *btcvsd; @@ -1360,7 +1360,8 @@ static int mtk_btcvsd_snd_probe(struct platform_device *pdev) btcvsd->bt_sram_bank2_base = of_iomap(dev->of_node, 1); if (!btcvsd->bt_sram_bank2_base) { dev_err(dev, "iomap bt_sram_bank2_base fail\n"); - return -EIO; + ret = -EIO; + goto unmap_pkv_err; } btcvsd->infra = syscon_regmap_lookup_by_phandle(dev->of_node, @@ -1368,7 +1369,8 @@ static int mtk_btcvsd_snd_probe(struct platform_device *pdev) if (IS_ERR(btcvsd->infra)) { dev_err(dev, "cannot find infra controller: %ld\n", PTR_ERR(btcvsd->infra)); - return PTR_ERR(btcvsd->infra); + ret = PTR_ERR(btcvsd->infra); + goto unmap_bank2_err; } /* get offset */ @@ -1377,7 +1379,7 @@ static int mtk_btcvsd_snd_probe(struct platform_device *pdev) ARRAY_SIZE(offset)); if (ret) { dev_warn(dev, "%s(), get offset fail, ret %d\n", __func__, ret); - return ret; + goto unmap_bank2_err; } btcvsd->infra_misc_offset = offset[0]; btcvsd->conn_bt_cvsd_mask = offset[1]; @@ -1396,8 +1398,18 @@ static int mtk_btcvsd_snd_probe(struct platform_device *pdev) mtk_btcvsd_snd_set_state(btcvsd, btcvsd->tx, BT_SCO_STATE_IDLE); mtk_btcvsd_snd_set_state(btcvsd, btcvsd->rx, BT_SCO_STATE_IDLE); - return devm_snd_soc_register_component(dev, &mtk_btcvsd_snd_platform, - NULL, 0); + ret = devm_snd_soc_register_component(dev, &mtk_btcvsd_snd_platform, + NULL, 0); + if (ret) + goto unmap_bank2_err; + + return 0; + +unmap_bank2_err: + iounmap(btcvsd->bt_sram_bank2_base); +unmap_pkv_err: + iounmap(btcvsd->bt_pkv_base); + return ret; } static int mtk_btcvsd_snd_remove(struct platform_device *pdev) diff --git a/sound/soc/mediatek/mt8173/mt8173-max98090.c b/sound/soc/mediatek/mt8173/mt8173-max98090.c index 22c00600c999..de1410c2c446 100644 --- a/sound/soc/mediatek/mt8173/mt8173-max98090.c +++ b/sound/soc/mediatek/mt8173/mt8173-max98090.c @@ -180,6 +180,9 @@ static int mt8173_max98090_dev_probe(struct platform_device *pdev) if (ret) dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", __func__, ret); + + of_node_put(codec_node); + of_node_put(platform_node); return ret; } diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c index 8717e87bfe26..6f8542329bab 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c @@ -218,6 +218,8 @@ static int mt8173_rt5650_rt5514_dev_probe(struct platform_device *pdev) if (ret) dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", __func__, ret); + + of_node_put(platform_node); return ret; } diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c index 9d4dd9721154..727ff0f7f20b 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c @@ -285,6 +285,8 @@ static int mt8173_rt5650_rt5676_dev_probe(struct platform_device *pdev) if (ret) dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", __func__, ret); + + of_node_put(platform_node); return ret; } diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650.c b/sound/soc/mediatek/mt8173/mt8173-rt5650.c index ef6f23675286..21e7d4d3ded5 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650.c @@ -309,6 +309,8 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) if (ret) dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", __func__, ret); + + of_node_put(platform_node); return ret; } diff --git a/sound/soc/meson/g12a-tohdmitx.c b/sound/soc/meson/g12a-tohdmitx.c index 9cfbd343a00c..cbe47e0cae42 100644 --- a/sound/soc/meson/g12a-tohdmitx.c +++ b/sound/soc/meson/g12a-tohdmitx.c @@ -127,7 +127,7 @@ static int g12a_tohdmitx_i2s_mux_put_enum(struct snd_kcontrol *kcontrol, snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL); - return 0; + return 1; } static const struct snd_kcontrol_new g12a_tohdmitx_i2s_mux = diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index a2c79426513b..d7d272bbebb2 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -455,7 +455,10 @@ static int mxs_saif_hw_params(struct snd_pcm_substream *substream, * basic clock which should be fast enough for the internal * logic. */ - clk_enable(saif->clk); + ret = clk_enable(saif->clk); + if (ret) + return ret; + ret = clk_set_rate(saif->clk, 24000000); clk_disable(saif->clk); if (ret) diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c index 9841e1da9782..8282fe6d00dd 100644 --- a/sound/soc/mxs/mxs-sgtl5000.c +++ b/sound/soc/mxs/mxs-sgtl5000.c @@ -118,6 +118,9 @@ static int mxs_sgtl5000_probe(struct platform_device *pdev) codec_np = of_parse_phandle(np, "audio-codec", 0); if (!saif_np[0] || !saif_np[1] || !codec_np) { dev_err(&pdev->dev, "phandle missing or invalid\n"); + of_node_put(codec_np); + of_node_put(saif_np[0]); + of_node_put(saif_np[1]); return -EINVAL; } diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c index 745cc9dd14f3..bc65009be875 100644 --- a/sound/soc/qcom/qdsp6/q6routing.c +++ b/sound/soc/qcom/qdsp6/q6routing.c @@ -440,9 +440,15 @@ static int msm_routing_put_audio_mixer(struct snd_kcontrol *kcontrol, struct session_data *session = &data->sessions[session_id]; if (ucontrol->value.integer.value[0]) { + if (session->port_id == be_id) + return 0; + session->port_id = be_id; snd_soc_dapm_mixer_update_power(dapm, kcontrol, 1, update); } else { + if (session->port_id == -1 || session->port_id != be_id) + return 0; + session->port_id = -1; snd_soc_dapm_mixer_update_power(dapm, kcontrol, 0, update); } diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 61c984f10d8e..086c90e09577 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -186,7 +186,9 @@ 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: @@ -199,7 +201,8 @@ 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); @@ -213,7 +216,8 @@ 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); @@ -229,14 +233,15 @@ 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); @@ -252,19 +257,23 @@ 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/samsung/idma.c b/sound/soc/samsung/idma.c index 65497cd477a5..47f6f5d70853 100644 --- a/sound/soc/samsung/idma.c +++ b/sound/soc/samsung/idma.c @@ -363,6 +363,8 @@ static int preallocate_idma_buffer(struct snd_pcm *pcm, int stream) buf->addr = idma.lp_tx_addr; buf->bytes = idma_hardware.buffer_bytes_max; buf->area = (unsigned char * __force)ioremap(buf->addr, buf->bytes); + if (!buf->area) + return -ENOMEM; return 0; } diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 3447dbdba1f1..6ac7df30a289 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -816,14 +816,27 @@ static int fsi_clk_enable(struct device *dev, return ret; } - clk_enable(clock->xck); - clk_enable(clock->ick); - clk_enable(clock->div); + ret = clk_enable(clock->xck); + if (ret) + goto err; + ret = clk_enable(clock->ick); + if (ret) + goto disable_xck; + ret = clk_enable(clock->div); + if (ret) + goto disable_ick; clock->count++; } return ret; + +disable_ick: + clk_disable(clock->ick); +disable_xck: + clk_disable(clock->xck); +err: + return ret; } static int fsi_clk_disable(struct device *dev, diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index b9aacf3d3b29..7532ab27a48d 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -289,7 +289,6 @@ static void rsnd_adg_set_ssi_clk(struct rsnd_mod *ssi_mod, u32 val) int rsnd_adg_clk_query(struct rsnd_priv *priv, unsigned int rate) { struct rsnd_adg *adg = rsnd_priv_to_adg(priv); - struct clk *clk; int i; int sel_table[] = { [CLKA] = 0x1, @@ -302,10 +301,9 @@ int rsnd_adg_clk_query(struct rsnd_priv *priv, unsigned int rate) * find suitable clock from * AUDIO_CLKA/AUDIO_CLKB/AUDIO_CLKC/AUDIO_CLKI. */ - for_each_rsnd_clk(clk, adg, i) { + for (i = 0; i < CLKMAX; i++) if (rate == adg->clk_rate[i]) return sel_table[i]; - } /* * find divided clock from BRGA/BRGB diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 9e54d8ae6d2c..da6e40aef7b6 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -871,6 +871,11 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) return -EINVAL; } + if (!codec_dai) { + dev_err(rtd->card->dev, "Missing codec\n"); + return -EINVAL; + } + /* check client and interface hw capabilities */ if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_PLAYBACK) && snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_PLAYBACK)) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a856eabf5f99..66a99d6f9434 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3180,7 +3180,7 @@ 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++) { @@ -3364,7 +3364,7 @@ int snd_soc_get_dai_name(struct of_phandle_args *args, for_each_component(pos) { component_of_node = soc_component_to_node(pos); - if (component_of_node != args->np) + if (component_of_node != args->np || !pos->num_dai) continue; ret = snd_soc_component_of_xlate_dai_name(pos, args, dai_name); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7c4d5963692d..1c09dfb0c0f0 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1676,8 +1676,7 @@ static void dapm_seq_run(struct snd_soc_card *card, switch (w->id) { case snd_soc_dapm_pre: if (!w->event) - list_for_each_entry_safe_continue(w, n, list, - power_list); + continue; if (event == SND_SOC_DAPM_STREAM_START) ret = w->event(w, @@ -1689,8 +1688,7 @@ static void dapm_seq_run(struct snd_soc_card *card, case snd_soc_dapm_post: if (!w->event) - list_for_each_entry_safe_continue(w, n, list, - power_list); + continue; if (event == SND_SOC_DAPM_STREAM_START) ret = w->event(w, @@ -2542,10 +2540,16 @@ 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); @@ -2558,13 +2562,26 @@ 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; } /** @@ -3580,14 +3597,15 @@ 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); @@ -4029,7 +4047,7 @@ static int snd_soc_dapm_dai_link_put(struct snd_kcontrol *kcontrol, rtd->params_select = ucontrol->value.enumerated.item[0]; - return 0; + return 1; } static void diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 95fc24580f85..c88bc6bb41cf 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -314,7 +314,7 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, unsigned int sign_bit = mc->sign_bit; unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; - int err; + int err, ret; bool type_2r = false; unsigned int val2 = 0; unsigned int val, val_mask; @@ -322,13 +322,27 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, if (sign_bit) mask = BIT(sign_bit + 1) - 1; - val = ((ucontrol->value.integer.value[0] + min) & mask); + val = ucontrol->value.integer.value[0]; + if (mc->platform_max && ((int)val + min) > mc->platform_max) + return -EINVAL; + if (val > max - min) + return -EINVAL; + if (val < 0) + return -EINVAL; + val = (val + min) & mask; if (invert) val = max - val; val_mask = mask << shift; val = val << shift; if (snd_soc_volsw_is_stereo(mc)) { - val2 = ((ucontrol->value.integer.value[1] + min) & mask); + val2 = ucontrol->value.integer.value[1]; + if (mc->platform_max && ((int)val2 + min) > mc->platform_max) + return -EINVAL; + if (val2 > max - min) + return -EINVAL; + if (val2 < 0) + return -EINVAL; + val2 = (val2 + min) & mask; if (invert) val2 = max - val2; if (reg == reg2) { @@ -342,12 +356,18 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, err = snd_soc_component_update_bits(component, reg, val_mask, val); if (err < 0) return err; + ret = err; - if (type_2r) + if (type_2r) { err = snd_soc_component_update_bits(component, reg2, val_mask, - val2); + val2); + /* Don't discard any error code or drop change flag */ + if (ret == 0 || err < 0) { + ret = err; + } + } - return err; + return ret; } EXPORT_SYMBOL_GPL(snd_soc_put_volsw); @@ -422,8 +442,15 @@ int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, int err = 0; unsigned int val, val_mask, val2 = 0; + val = ucontrol->value.integer.value[0]; + if (mc->platform_max && val > mc->platform_max) + return -EINVAL; + if (val > max - min) + return -EINVAL; + if (val < 0) + return -EINVAL; val_mask = mask << shift; - val = (ucontrol->value.integer.value[0] + min) & mask; + val = (val + min) & mask; val = val << shift; err = snd_soc_component_update_bits(component, reg, val_mask, val); @@ -496,7 +523,7 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; unsigned int val, val_mask; - int ret; + int err, ret; if (invert) val = (max - ucontrol->value.integer.value[0]) & mask; @@ -505,9 +532,10 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, val_mask = mask << shift; val = val << shift; - ret = snd_soc_component_update_bits(component, reg, val_mask, val); - if (ret < 0) - return ret; + err = snd_soc_component_update_bits(component, reg, val_mask, val); + if (err < 0) + return err; + ret = err; if (snd_soc_volsw_is_stereo(mc)) { if (invert) @@ -517,8 +545,12 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, val_mask = mask << shift; val = val << shift; - ret = snd_soc_component_update_bits(component, rreg, val_mask, + err = snd_soc_component_update_bits(component, rreg, val_mask, val); + /* Don't discard any error code or drop change flag */ + if (ret == 0 || err < 0) { + ret = err; + } } return ret; @@ -889,6 +921,8 @@ int snd_soc_put_xr_sx(struct snd_kcontrol *kcontrol, unsigned int i, regval, regmask; int err; + if (val < mc->min || val > mc->max) + return -EINVAL; if (invert) val = max - val; val &= mask; diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index c367609433bf..870b00229353 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -587,7 +587,8 @@ static int soc_tplg_kcontrol_bind_io(struct snd_soc_tplg_ctl_hdr *hdr, if (le32_to_cpu(hdr->ops.info) == SND_SOC_TPLG_CTL_BYTES && k->iface & SNDRV_CTL_ELEM_IFACE_MIXER - && k->access & SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE + && (k->access & SNDRV_CTL_ELEM_ACCESS_TLV_READ + || k->access & SNDRV_CTL_ELEM_ACCESS_TLV_WRITE) && k->access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK) { struct soc_bytes_ext *sbe; struct snd_soc_tplg_bytes_control *be; @@ -2777,6 +2778,7 @@ EXPORT_SYMBOL_GPL(snd_soc_tplg_widget_remove_all); /* remove dynamic controls from the component driver */ int snd_soc_tplg_component_remove(struct snd_soc_component *comp, u32 index) { + struct snd_card *card = comp->card->snd_card; struct snd_soc_dobj *dobj, *next_dobj; int pass = SOC_TPLG_PASS_END; @@ -2784,6 +2786,7 @@ int snd_soc_tplg_component_remove(struct snd_soc_component *comp, u32 index) while (pass >= SOC_TPLG_PASS_START) { /* remove mixer controls */ + down_write(&card->controls_rwsem); list_for_each_entry_safe(dobj, next_dobj, &comp->dobj_list, list) { @@ -2827,6 +2830,7 @@ int snd_soc_tplg_component_remove(struct snd_soc_component *comp, u32 index) break; } } + up_write(&card->controls_rwsem); pass--; } diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index 3f645200d3a5..b3cdd10c83ae 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -67,6 +67,7 @@ static struct hdac_ext_stream * return NULL; } + spin_lock_irq(&bus->reg_lock); list_for_each_entry(stream, &bus->stream_list, list) { struct hdac_ext_stream *hstream = stream_to_hdac_ext_stream(stream); @@ -106,12 +107,12 @@ static struct hdac_ext_stream * * is updated in snd_hdac_ext_stream_decouple(). */ if (!res->decoupled) - snd_hdac_ext_stream_decouple(bus, res, true); - spin_lock_irq(&bus->reg_lock); + snd_hdac_ext_stream_decouple_locked(bus, res, true); + res->link_locked = 1; res->link_substream = substream; - spin_unlock_irq(&bus->reg_lock); } + spin_unlock_irq(&bus->reg_lock); return res; } diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c index 356bb134ae93..7573f3f9f0f2 100644 --- a/sound/soc/sof/intel/hda-loader.c +++ b/sound/soc/sof/intel/hda-loader.c @@ -50,7 +50,7 @@ static int cl_stream_prepare(struct snd_sof_dev *sdev, unsigned int format, ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV_SG, &pci->dev, size, dmab); if (ret < 0) { dev_err(sdev->dev, "error: memory alloc failed: %x\n", ret); - goto error; + goto out_put; } hstream->period_bytes = 0;/* initialize period_bytes */ @@ -60,16 +60,17 @@ static int cl_stream_prepare(struct snd_sof_dev *sdev, unsigned int format, ret = hda_dsp_stream_hw_params(sdev, dsp_stream, dmab, NULL); if (ret < 0) { dev_err(sdev->dev, "error: hdac prepare failed: %x\n", ret); - goto error; + goto out_free; } hda_dsp_stream_spib_config(sdev, dsp_stream, HDA_DSP_SPIB_ENABLE, size); return hstream->stream_tag; -error: - hda_dsp_stream_put(sdev, direction, hstream->stream_tag); +out_free: snd_dma_free_pages(dmab); +out_put: + hda_dsp_stream_put(sdev, direction, hstream->stream_tag); return ret; } diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c index 2ed92c990b97..dd9013c47664 100644 --- a/sound/soc/sti/uniperif_player.c +++ b/sound/soc/sti/uniperif_player.c @@ -91,7 +91,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id) SET_UNIPERIF_ITM_BCLR_FIFO_ERROR(player); /* Stop the player */ - snd_pcm_stop_xrun(player->substream); + snd_pcm_stop(player->substream, SNDRV_PCM_STATE_XRUN); } ret = IRQ_HANDLED; @@ -105,7 +105,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id) SET_UNIPERIF_ITM_BCLR_DMA_ERROR(player); /* Stop the player */ - snd_pcm_stop_xrun(player->substream); + snd_pcm_stop(player->substream, SNDRV_PCM_STATE_XRUN); ret = IRQ_HANDLED; } @@ -138,7 +138,7 @@ static irqreturn_t uni_player_irq_handler(int irq, void *dev_id) dev_err(player->dev, "Underflow recovery failed\n"); /* Stop the player */ - snd_pcm_stop_xrun(player->substream); + snd_pcm_stop(player->substream, SNDRV_PCM_STATE_XRUN); ret = IRQ_HANDLED; } diff --git a/sound/soc/sti/uniperif_reader.c b/sound/soc/sti/uniperif_reader.c index 136059331211..065c5f0d1f5f 100644 --- a/sound/soc/sti/uniperif_reader.c +++ b/sound/soc/sti/uniperif_reader.c @@ -65,7 +65,7 @@ static irqreturn_t uni_reader_irq_handler(int irq, void *dev_id) if (unlikely(status & UNIPERIF_ITS_FIFO_ERROR_MASK(reader))) { dev_err(reader->dev, "FIFO error detected\n"); - snd_pcm_stop_xrun(reader->substream); + snd_pcm_stop(reader->substream, SNDRV_PCM_STATE_XRUN); ret = IRQ_HANDLED; } diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index 9e8b1497efd3..a281ceb3c67e 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -139,6 +139,7 @@ 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 4954a33ff46b..30edd70e8183 100644 --- a/sound/soc/tegra/tegra_max98090.c +++ b/sound/soc/tegra/tegra_max98090.c @@ -182,6 +182,7 @@ 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 d46915a3ec4c..3d68a41040ed 100644 --- a/sound/soc/tegra/tegra_rt5640.c +++ b/sound/soc/tegra/tegra_rt5640.c @@ -132,6 +132,7 @@ 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 81cb6cc6236e..ae150ade9441 100644 --- a/sound/soc/tegra/tegra_rt5677.c +++ b/sound/soc/tegra/tegra_rt5677.c @@ -175,6 +175,7 @@ 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 e13b81d29cf3..fe21d9eff8c0 100644 --- a/sound/soc/tegra/tegra_sgtl5000.c +++ b/sound/soc/tegra/tegra_sgtl5000.c @@ -97,6 +97,7 @@ 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 f6dd790dad71..a2362a2189dc 100644 --- a/sound/soc/tegra/tegra_wm8753.c +++ b/sound/soc/tegra/tegra_wm8753.c @@ -101,6 +101,7 @@ 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 0fa01cacfec9..08bcc94dcff8 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -217,6 +217,7 @@ 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 b85bd9f89073..232eac58373a 100644 --- a/sound/soc/tegra/tegra_wm9712.c +++ b/sound/soc/tegra/tegra_wm9712.c @@ -54,6 +54,7 @@ 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 3f67ddd13674..5086bc2446d2 100644 --- a/sound/soc/tegra/trimslice.c +++ b/sound/soc/tegra/trimslice.c @@ -94,6 +94,7 @@ 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/soc/ti/davinci-i2s.c b/sound/soc/ti/davinci-i2s.c index d89b5c928c4d..b2b2dcdb05d4 100644 --- a/sound/soc/ti/davinci-i2s.c +++ b/sound/soc/ti/davinci-i2s.c @@ -708,7 +708,9 @@ static int davinci_i2s_probe(struct platform_device *pdev) dev->clk = clk_get(&pdev->dev, NULL); if (IS_ERR(dev->clk)) return -ENODEV; - clk_enable(dev->clk); + ret = clk_enable(dev->clk); + if (ret) + goto err_put_clk; dev->dev = &pdev->dev; dev_set_drvdata(&pdev->dev, dev); @@ -730,6 +732,7 @@ err_unregister_component: snd_soc_unregister_component(&pdev->dev); err_release_clk: clk_disable(dev->clk); +err_put_clk: clk_put(dev->clk); return ret; } diff --git a/sound/soc/uniphier/Kconfig b/sound/soc/uniphier/Kconfig index aa3592ee1358..ddfa6424c656 100644 --- a/sound/soc/uniphier/Kconfig +++ b/sound/soc/uniphier/Kconfig @@ -23,7 +23,6 @@ config SND_SOC_UNIPHIER_LD11 tristate "UniPhier LD11/LD20 Device Driver" depends on SND_SOC_UNIPHIER select SND_SOC_UNIPHIER_AIO - select SND_SOC_UNIPHIER_AIO_DMA help This adds ASoC driver for Socionext UniPhier LD11/LD20 input and output that can be used with other codecs. @@ -34,7 +33,6 @@ config SND_SOC_UNIPHIER_PXS2 tristate "UniPhier PXs2 Device Driver" depends on SND_SOC_UNIPHIER select SND_SOC_UNIPHIER_AIO - select SND_SOC_UNIPHIER_AIO_DMA help This adds ASoC driver for Socionext UniPhier PXs2 input and output that can be used with other codecs. diff --git a/sound/soc/xilinx/xlnx_formatter_pcm.c b/sound/soc/xilinx/xlnx_formatter_pcm.c index dc8721f4f56b..f6b3a5bdbcea 100644 --- a/sound/soc/xilinx/xlnx_formatter_pcm.c +++ b/sound/soc/xilinx/xlnx_formatter_pcm.c @@ -37,6 +37,7 @@ #define XLNX_AUD_XFER_COUNT 0x28 #define XLNX_AUD_CH_STS_START 0x2C #define XLNX_BYTES_PER_CH 0x44 +#define XLNX_AUD_ALIGN_BYTES 64 #define AUD_STS_IOC_IRQ_MASK BIT(31) #define AUD_STS_CH_STS_MASK BIT(29) @@ -370,12 +371,32 @@ static int xlnx_formatter_pcm_open(struct snd_pcm_substream *substream) snd_soc_set_runtime_hwparams(substream, &xlnx_pcm_hardware); runtime->private_data = stream_data; - /* Resize the period size divisible by 64 */ + /* Resize the period bytes as divisible by 64 */ err = snd_pcm_hw_constraint_step(runtime, 0, - SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64); + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, + XLNX_AUD_ALIGN_BYTES); if (err) { dev_err(component->dev, - "unable to set constraint on period bytes\n"); + "Unable to set constraint on period bytes\n"); + return err; + } + + /* Resize the buffer bytes as divisible by 64 */ + err = snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, + XLNX_AUD_ALIGN_BYTES); + if (err) { + dev_err(component->dev, + "Unable to set constraint on buffer bytes\n"); + return err; + } + + /* Set periods as integer multiple */ + err = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (err < 0) { + dev_err(component->dev, + "Unable to set constraint on periods to be integer\n"); return err; } @@ -461,8 +482,8 @@ static int xlnx_formatter_pcm_hw_params(struct snd_pcm_substream *substream, stream_data->buffer_size = size; - low = lower_32_bits(substream->dma_buffer.addr); - high = upper_32_bits(substream->dma_buffer.addr); + low = lower_32_bits(runtime->dma_addr); + high = upper_32_bits(runtime->dma_addr); writel(low, stream_data->mmio + XLNX_AUD_BUFF_ADDR_LSB); writel(high, stream_data->mmio + XLNX_AUD_BUFF_ADDR_MSB); diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c index 4de1ba9a418d..6e5d315bab59 100644 --- a/sound/spi/at73c213.c +++ b/sound/spi/at73c213.c @@ -218,7 +218,9 @@ static int snd_at73c213_pcm_open(struct snd_pcm_substream *substream) runtime->hw = snd_at73c213_playback_hw; chip->substream = substream; - clk_enable(chip->ssc->clk); + err = clk_enable(chip->ssc->clk); + if (err) + return err; return 0; } @@ -784,7 +786,9 @@ static int snd_at73c213_chip_init(struct snd_at73c213 *chip) goto out; /* Enable DAC master clock. */ - clk_enable(chip->board->dac_clk); + retval = clk_enable(chip->board->dac_clk); + if (retval) + goto out; /* Initialize at73c213 on SPI bus. */ retval = snd_at73c213_write_reg(chip, DAC_RST, 0x04); @@ -897,7 +901,9 @@ static int snd_at73c213_dev_init(struct snd_card *card, chip->card = card; chip->irq = -1; - clk_enable(chip->ssc->clk); + retval = clk_enable(chip->ssc->clk); + if (retval) + return retval; retval = request_irq(irq, snd_at73c213_interrupt, 0, "at73c213", chip); if (retval) { @@ -1016,7 +1022,9 @@ static int snd_at73c213_remove(struct spi_device *spi) int retval; /* Stop playback. */ - clk_enable(chip->ssc->clk); + retval = clk_enable(chip->ssc->clk); + if (retval) + goto out; ssc_writel(chip->ssc->regs, CR, SSC_BIT(CR_TXDIS)); clk_disable(chip->ssc->clk); @@ -1096,9 +1104,16 @@ static int snd_at73c213_resume(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); struct snd_at73c213 *chip = card->private_data; + int retval; - clk_enable(chip->board->dac_clk); - clk_enable(chip->ssc->clk); + retval = clk_enable(chip->board->dac_clk); + if (retval) + return retval; + retval = clk_enable(chip->ssc->clk); + if (retval) { + clk_disable(chip->board->dac_clk); + return retval; + } ssc_writel(chip->ssc->regs, CR, SSC_BIT(CR_TXEN)); return 0; diff --git a/sound/synth/emux/emux.c b/sound/synth/emux/emux.c index f65e6c7b139f..6695530bba9b 100644 --- a/sound/synth/emux/emux.c +++ b/sound/synth/emux/emux.c @@ -88,7 +88,7 @@ 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 43a2a62d66f7..49629d4bb327 100644 --- a/sound/usb/6fire/comm.c +++ b/sound/usb/6fire/comm.c @@ -95,7 +95,7 @@ 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 69137c14d0dc..2333e8ff3411 100644 --- a/sound/usb/6fire/firmware.c +++ b/sound/usb/6fire/firmware.c @@ -162,7 +162,7 @@ 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) @@ -175,7 +175,7 @@ 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) @@ -190,7 +190,7 @@ 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/clock.c b/sound/usb/clock.c index 6a51b9d20eeb..3d1c0ec11753 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -319,6 +319,12 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, selector->baCSourceID[ret - 1], visited, validate); if (ret > 0) { + /* + * For Samsung USBC Headset (AKG), setting clock selector again + * will result in incorrect default clock setting problems + */ + if (chip->usb_id == USB_ID(0x04e8, 0xa051)) + return ret; err = uac_clock_selector_set_val(chip, entity_id, cur); if (err < 0) return err; diff --git a/sound/usb/format.c b/sound/usb/format.c index 9e9d4c10dfac..84b66f7c627c 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -195,9 +195,11 @@ 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; diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c index 1e38cdda2af6..8ca56ba600cf 100644 --- a/sound/usb/line6/driver.c +++ b/sound/usb/line6/driver.c @@ -113,12 +113,12 @@ 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) { @@ -350,7 +350,7 @@ 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); @@ -365,7 +365,7 @@ 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); @@ -393,7 +393,7 @@ 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); @@ -425,7 +425,7 @@ 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, @@ -441,7 +441,7 @@ 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, diff --git a/sound/usb/line6/driver.h b/sound/usb/line6/driver.h index e5e572ed5f30..890c239e3fc0 100644 --- a/sound/usb/line6/driver.h +++ b/sound/usb/line6/driver.h @@ -27,7 +27,7 @@ #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/podhd.c b/sound/usb/line6/podhd.c index 5d9954a2d05e..8b1610bdb8d5 100644 --- a/sound/usb/line6/podhd.c +++ b/sound/usb/line6/podhd.c @@ -190,7 +190,7 @@ 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; @@ -200,7 +200,7 @@ 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); @@ -220,7 +220,7 @@ 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 d0a555dbe324..21f86c71dad7 100644 --- a/sound/usb/line6/toneport.c +++ b/sound/usb/line6/toneport.c @@ -128,7 +128,7 @@ 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/midi.c b/sound/usb/midi.c index 33e9a7f6246f..ce501200e592 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -1210,6 +1210,7 @@ static void snd_usbmidi_output_drain(struct snd_rawmidi_substream *substream) } while (drain_urbs && timeout); finish_wait(&ep->drain_wait, &wait); } + port->active = 0; spin_unlock_irq(&ep->buffer_lock); } diff --git a/sound/usb/misc/ua101.c b/sound/usb/misc/ua101.c index 307b72d5fffa..77304a29a61d 100644 --- a/sound/usb/misc/ua101.c +++ b/sound/usb/misc/ua101.c @@ -1020,7 +1020,7 @@ 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; } @@ -1028,7 +1028,7 @@ 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 f4f8778e907a..67eb1293fa15 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -3241,8 +3241,17 @@ static void snd_usb_mixer_dump_cval(struct snd_info_buffer *buffer, struct usb_mixer_elem_list *list) { struct usb_mixer_elem_info *cval = mixer_elem_list_to_info(list); - static const char * const val_types[] = {"BOOLEAN", "INV_BOOLEAN", - "S8", "U8", "S16", "U16"}; + static const char * const val_types[] = { + [USB_MIXER_BOOLEAN] = "BOOLEAN", + [USB_MIXER_INV_BOOLEAN] = "INV_BOOLEAN", + [USB_MIXER_S8] = "S8", + [USB_MIXER_U8] = "U8", + [USB_MIXER_S16] = "S16", + [USB_MIXER_U16] = "U16", + [USB_MIXER_S32] = "S32", + [USB_MIXER_U32] = "U32", + [USB_MIXER_BESPOKEN] = "BESPOKEN", + }; snd_iprintf(buffer, " Info: id=%i, control=%i, cmask=0x%x, " "channels=%i, type=\"%s\"\n", cval->head.id, cval->control, cval->cmask, cval->channels, @@ -3598,6 +3607,9 @@ static int restore_mixer_value(struct usb_mixer_elem_list *list) struct usb_mixer_elem_info *cval = mixer_elem_list_to_info(list); int c, err, idx; + if (cval->val_type == USB_MIXER_BESPOKEN) + return 0; + if (cval->cmask) { idx = 0; for (c = 0; c < MAX_CHANNELS; c++) { diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index 01b5e5cc2221..0e813cd85bee 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -55,6 +55,7 @@ enum { USB_MIXER_U16, USB_MIXER_S32, USB_MIXER_U32, + USB_MIXER_BESPOKEN, /* non-standard type */ }; typedef void (*usb_mixer_elem_dump_func_t)(struct snd_info_buffer *buffer, diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index d926869c031b..1f7c80541d03 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -2370,9 +2370,10 @@ void snd_usb_mixer_fu_apply_quirk(struct usb_mixer_interface *mixer, if (unitid == 7 && cval->control == UAC_FU_VOLUME) snd_dragonfly_quirk_db_scale(mixer, cval, kctl); break; - /* lowest playback value is muted on C-Media devices */ - case USB_ID(0x0d8c, 0x000c): - case USB_ID(0x0d8c, 0x0014): + /* lowest playback value is muted on some devices */ + case USB_ID(0x0d8c, 0x000c): /* C-Media */ + case USB_ID(0x0d8c, 0x0014): /* C-Media */ + case USB_ID(0x19f7, 0x0003): /* RODE NT-USB */ if (strstr(kctl->id.name, "Playback")) cval->min_mute = 1; break; diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index 7a10c9e22c46..ab7abe360fcf 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -254,10 +254,10 @@ static const struct scarlett2_device_info s6i6_gen2_info = { .pad_input_count = 2, .line_out_descrs = { - "Monitor L", - "Monitor R", - "Headphones L", - "Headphones R", + "Headphones 1 L", + "Headphones 1 R", + "Headphones 2 L", + "Headphones 2 R", }, .ports = { @@ -356,7 +356,7 @@ static const struct scarlett2_device_info s18i8_gen2_info = { }, [SCARLETT2_PORT_TYPE_PCM] = { .id = 0x600, - .num = { 20, 18, 18, 14, 10 }, + .num = { 8, 18, 18, 14, 10 }, .src_descr = "PCM %d", .src_num_offset = 1, .dst_descr = "PCM %02d Capture" @@ -949,10 +949,15 @@ static int scarlett2_add_new_ctl(struct usb_mixer_interface *mixer, if (!elem) return -ENOMEM; + /* We set USB_MIXER_BESPOKEN type, so that the core USB mixer code + * ignores them for resume and other operations. + * Also, the head.id field is set to 0, as we don't use this field. + */ elem->head.mixer = mixer; elem->control = index; - elem->head.id = index; + elem->head.id = 0; elem->channels = channels; + elem->val_type = USB_MIXER_BESPOKEN; kctl = snd_ctl_new1(ncontrol, elem); if (!kctl) { @@ -1028,11 +1033,10 @@ static int scarlett2_master_volume_ctl_get(struct snd_kcontrol *kctl, struct usb_mixer_interface *mixer = elem->head.mixer; struct scarlett2_mixer_data *private = mixer->private_data; - if (private->vol_updated) { - mutex_lock(&private->data_mutex); + mutex_lock(&private->data_mutex); + if (private->vol_updated) scarlett2_update_volumes(mixer); - mutex_unlock(&private->data_mutex); - } + mutex_unlock(&private->data_mutex); ucontrol->value.integer.value[0] = private->master_vol; return 0; @@ -1046,11 +1050,10 @@ static int scarlett2_volume_ctl_get(struct snd_kcontrol *kctl, struct scarlett2_mixer_data *private = mixer->private_data; int index = elem->control; - if (private->vol_updated) { - mutex_lock(&private->data_mutex); + mutex_lock(&private->data_mutex); + if (private->vol_updated) scarlett2_update_volumes(mixer); - mutex_unlock(&private->data_mutex); - } + mutex_unlock(&private->data_mutex); ucontrol->value.integer.value[0] = private->vol[index]; return 0; @@ -1181,6 +1184,8 @@ static int scarlett2_sw_hw_enum_ctl_put(struct snd_kcontrol *kctl, /* Send SW/HW switch change to the device */ err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_SW_HW_SWITCH, index, val); + if (err == 0) + err = 1; unlock: mutex_unlock(&private->data_mutex); @@ -1241,6 +1246,8 @@ static int scarlett2_level_enum_ctl_put(struct snd_kcontrol *kctl, /* Send switch change to the device */ err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_LEVEL_SWITCH, index, val); + if (err == 0) + err = 1; unlock: mutex_unlock(&private->data_mutex); @@ -1291,6 +1298,8 @@ static int scarlett2_pad_ctl_put(struct snd_kcontrol *kctl, /* Send switch change to the device */ err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_PAD_SWITCH, index, val); + if (err == 0) + err = 1; unlock: mutex_unlock(&private->data_mutex); @@ -1314,11 +1323,10 @@ static int scarlett2_button_ctl_get(struct snd_kcontrol *kctl, struct usb_mixer_interface *mixer = elem->head.mixer; struct scarlett2_mixer_data *private = mixer->private_data; - if (private->vol_updated) { - mutex_lock(&private->data_mutex); + mutex_lock(&private->data_mutex); + if (private->vol_updated) scarlett2_update_volumes(mixer); - mutex_unlock(&private->data_mutex); - } + mutex_unlock(&private->data_mutex); ucontrol->value.enumerated.item[0] = private->buttons[elem->control]; return 0; @@ -1347,6 +1355,8 @@ static int scarlett2_button_ctl_put(struct snd_kcontrol *kctl, /* Send switch change to the device */ err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_BUTTONS, index, val); + if (err == 0) + err = 1; unlock: mutex_unlock(&private->data_mutex); diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 441335abb401..c29ccdf9e8bc 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -25,6 +25,16 @@ .idProduct = prod, \ .bInterfaceClass = USB_CLASS_VENDOR_SPEC +/* A standard entry matching with vid/pid and the audio class/subclass */ +#define USB_AUDIO_DEVICE(vend, prod) \ + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | \ + USB_DEVICE_ID_MATCH_INT_CLASS | \ + USB_DEVICE_ID_MATCH_INT_SUBCLASS, \ + .idVendor = vend, \ + .idProduct = prod, \ + .bInterfaceClass = USB_CLASS_AUDIO, \ + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL + /* HP Thunderbolt Dock Audio Headset */ { USB_DEVICE(0x03f0, 0x0269), @@ -126,6 +136,48 @@ }, /* + * Creative Technology, Ltd Live! Cam Sync HD [VF0770] + * The device advertises 8 formats, but only a rate of 48kHz is honored by the + * hardware and 24 bits give chopped audio, so only report the one working + * combination. + */ +{ + USB_DEVICE(0x041e, 0x4095), + .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 = 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, + .fmt_bits = 16, + .iface = 3, + .altsetting = 4, + .altset_idx = 4, + .endpoint = 0x82, + .ep_attr = 0x05, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 48000 }, + }, + }, + { + .ifnum = -1 + }, + }, + }, +}, + +/* * HP Wireless Audio * When not ignored, causes instability issues for some users, forcing them to * blacklist the entire module. @@ -3764,5 +3816,37 @@ ALC1220_VB_DESKTOP(0x26ce, 0x0a01), /* Asrock TRX40 Creator */ } } }, +{ + /* + * 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 186e90e3636c..72223545abfd 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1840,6 +1840,12 @@ static const struct registration_quirk registration_quirks[] = { REG_QUIRK_ENTRY(0x0951, 0x16d8, 2), /* Kingston HyperX AMP */ REG_QUIRK_ENTRY(0x0951, 0x16ed, 2), /* Kingston HyperX Cloud Alpha S */ REG_QUIRK_ENTRY(0x0951, 0x16ea, 2), /* Kingston HyperX Cloud Flight S */ + REG_QUIRK_ENTRY(0x0ecb, 0x1f46, 2), /* JBL Quantum 600 */ + REG_QUIRK_ENTRY(0x0ecb, 0x1f47, 2), /* JBL Quantum 800 */ + REG_QUIRK_ENTRY(0x0ecb, 0x1f4c, 2), /* JBL Quantum 400 */ + REG_QUIRK_ENTRY(0x0ecb, 0x2039, 2), /* JBL Quantum 400 */ + REG_QUIRK_ENTRY(0x0ecb, 0x203c, 2), /* JBL Quantum 600 */ + REG_QUIRK_ENTRY(0x0ecb, 0x203e, 2), /* JBL Quantum 800 */ { 0 } /* terminator */ }; diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index ff97fdcf63bd..b1959e04cbb1 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -8,7 +8,7 @@ */ /* handling of USB vendor/product ID pairs as 32-bit numbers */ -#define USB_ID(vendor, product) (((vendor) << 16) | (product)) +#define USB_ID(vendor, product) (((unsigned int)(vendor) << 16) | (product)) #define USB_ID_VENDOR(id) ((id) >> 16) #define USB_ID_PRODUCT(id) ((u16)(id)) diff --git a/sound/usb/usx2y/usb_stream.c b/sound/usb/usx2y/usb_stream.c index 091c071b270a..cff684942c4f 100644 --- a/sound/usb/usx2y/usb_stream.c +++ b/sound/usb/usx2y/usb_stream.c @@ -142,8 +142,11 @@ void usb_stream_free(struct usb_stream_kernel *sk) if (!s) return; - free_pages_exact(sk->write_page, s->write_size); - sk->write_page = NULL; + if (sk->write_page) { + free_pages_exact(sk->write_page, s->write_size); + sk->write_page = NULL; + } + free_pages_exact(s, s->read_size); sk->s = NULL; } diff --git a/sound/x86/intel_hdmi_audio.c b/sound/x86/intel_hdmi_audio.c index 5fd4e32247a6..a314f13e3292 100644 --- a/sound/x86/intel_hdmi_audio.c +++ b/sound/x86/intel_hdmi_audio.c @@ -1279,7 +1279,7 @@ static int had_pcm_mmap(struct snd_pcm_substream *substream, { vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot); return remap_pfn_range(vma, vma->vm_start, - substream->dma_buffer.addr >> PAGE_SHIFT, + substream->runtime->dma_addr >> PAGE_SHIFT, vma->vm_end - vma->vm_start, vma->vm_page_prot); } |