diff options
Diffstat (limited to 'sound')
256 files changed, 3301 insertions, 1210 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.c b/sound/core/control.c index 08ca7666e84c..15efa4b6ea94 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1350,7 +1350,7 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, unlock: up_write(&card->controls_rwsem); - return 0; + return err; } static int snd_ctl_elem_add_user(struct snd_ctl_file *file, diff --git a/sound/core/control_compat.c b/sound/core/control_compat.c index 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/init.c b/sound/core/init.c index db99b7fad6ad..45bbc4884ef0 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -384,10 +384,8 @@ int snd_card_disconnect(struct snd_card *card) return 0; } card->shutdown = 1; - spin_unlock(&card->files_lock); /* replace file->f_op with special dummy operations */ - spin_lock(&card->files_lock); list_for_each_entry(mfile, &card->files_list, list) { /* it's critical part, use endless loop */ /* we have no room to fail */ diff --git a/sound/core/jack.c b/sound/core/jack.c index fb26196571a7..e7ac82d46821 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -34,8 +34,11 @@ static int snd_jack_dev_disconnect(struct snd_device *device) #ifdef CONFIG_SND_JACK_INPUT_DEV struct snd_jack *jack = device->device_data; - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return 0; + } /* If the input device is registered with the input subsystem * then we need to use a different deallocator. */ @@ -44,6 +47,7 @@ static int snd_jack_dev_disconnect(struct snd_device *device) else input_free_device(jack->input_dev); jack->input_dev = NULL; + mutex_unlock(&jack->input_dev_lock); #endif /* CONFIG_SND_JACK_INPUT_DEV */ return 0; } @@ -54,10 +58,13 @@ 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); @@ -79,8 +86,11 @@ static int snd_jack_dev_register(struct snd_device *device) snprintf(jack->name, sizeof(jack->name), "%s %s", card->shortname, jack->id); - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return 0; + } jack->input_dev->name = jack->name; @@ -105,6 +115,7 @@ static int snd_jack_dev_register(struct snd_device *device) if (err == 0) jack->registered = 1; + mutex_unlock(&jack->input_dev_lock); return err; } #endif /* CONFIG_SND_JACK_INPUT_DEV */ @@ -220,10 +231,16 @@ 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) { #ifdef CONFIG_SND_JACK_INPUT_DEV + mutex_init(&jack->input_dev_lock); + + /* don't create input device for phantom jack */ + if (!phantom_jack) { int i; jack->input_dev = input_allocate_device(); @@ -241,8 +258,8 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, input_set_capability(jack->input_dev, EV_SW, jack_switch_types[i]); -#endif /* CONFIG_SND_JACK_INPUT_DEV */ } +#endif /* CONFIG_SND_JACK_INPUT_DEV */ err = snd_device_new(card, SNDRV_DEV_JACK, jack, &ops); if (err < 0) @@ -282,10 +299,14 @@ EXPORT_SYMBOL(snd_jack_new); void snd_jack_set_parent(struct snd_jack *jack, struct device *parent) { WARN_ON(jack->registered); - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return; + } jack->input_dev->dev.parent = parent; + mutex_unlock(&jack->input_dev_lock); } EXPORT_SYMBOL(snd_jack_set_parent); @@ -333,6 +354,8 @@ EXPORT_SYMBOL(snd_jack_set_key); /** * snd_jack_report - Report the current status of a jack + * Note: This function uses mutexes and should be called from a + * context which can sleep (such as a workqueue). * * @jack: The jack to report status for * @status: The current status of the jack @@ -352,8 +375,11 @@ void snd_jack_report(struct snd_jack *jack, int status) status & jack_kctl->mask_bits); #ifdef CONFIG_SND_JACK_INPUT_DEV - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return; + } for (i = 0; i < ARRAY_SIZE(jack->key); i++) { int testbit = SND_JACK_BTN_0 >> i; @@ -372,6 +398,7 @@ void snd_jack_report(struct snd_jack *jack, int status) } input_sync(jack->input_dev); + mutex_unlock(&jack->input_dev_lock); #endif /* CONFIG_SND_JACK_INPUT_DEV */ } EXPORT_SYMBOL(snd_jack_report); diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 6850d13aa98c..fe1ea03582cb 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -76,7 +76,8 @@ static void snd_malloc_dev_iram(struct snd_dma_buffer *dmab, size_t size) /* Assign the pool into private_data field */ dmab->private_data = pool; - dmab->area = gen_pool_dma_alloc(pool, size, &dmab->addr); + dmab->area = gen_pool_dma_alloc_align(pool, size, &dmab->addr, + PAGE_SIZE); } /** diff --git a/sound/core/oss/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 f57c610d7523..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) { @@ -682,17 +682,25 @@ static int snd_pcm_oss_period_size(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *oss_params, struct snd_pcm_hw_params *slave_params) { - size_t s; - size_t oss_buffer_size, oss_period_size, oss_periods; - size_t min_period_size, max_period_size; + ssize_t s; + ssize_t oss_buffer_size; + ssize_t oss_period_size, oss_periods; + ssize_t min_period_size, max_period_size; struct snd_pcm_runtime *runtime = substream->runtime; size_t oss_frame_size; oss_frame_size = snd_pcm_format_physical_width(params_format(oss_params)) * params_channels(oss_params) / 8; + oss_buffer_size = snd_pcm_hw_param_value_max(slave_params, + SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + NULL); + if (oss_buffer_size <= 0) + return -EINVAL; oss_buffer_size = snd_pcm_plug_client_size(substream, - snd_pcm_hw_param_value_max(slave_params, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, NULL)) * oss_frame_size; + oss_buffer_size * oss_frame_size); + if (oss_buffer_size <= 0) + return -EINVAL; oss_buffer_size = rounddown_pow_of_two(oss_buffer_size); if (atomic_read(&substream->mmap_count)) { if (oss_buffer_size > runtime->oss.mmap_bytes) @@ -728,17 +736,21 @@ static int snd_pcm_oss_period_size(struct snd_pcm_substream *substream, min_period_size = snd_pcm_plug_client_size(substream, snd_pcm_hw_param_value_min(slave_params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, NULL)); - min_period_size *= oss_frame_size; - min_period_size = roundup_pow_of_two(min_period_size); - if (oss_period_size < min_period_size) - oss_period_size = min_period_size; + if (min_period_size > 0) { + min_period_size *= oss_frame_size; + min_period_size = roundup_pow_of_two(min_period_size); + if (oss_period_size < min_period_size) + oss_period_size = min_period_size; + } max_period_size = snd_pcm_plug_client_size(substream, snd_pcm_hw_param_value_max(slave_params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, NULL)); - max_period_size *= oss_frame_size; - max_period_size = rounddown_pow_of_two(max_period_size); - if (oss_period_size > max_period_size) - oss_period_size = max_period_size; + if (max_period_size > 0) { + max_period_size *= oss_frame_size; + max_period_size = rounddown_pow_of_two(max_period_size); + if (oss_period_size > max_period_size) + oss_period_size = max_period_size; + } oss_periods = oss_buffer_size / oss_period_size; @@ -746,7 +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; @@ -762,6 +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; @@ -872,8 +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); @@ -1026,10 +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; } @@ -1934,11 +1957,15 @@ static int snd_pcm_oss_set_subdivide(struct snd_pcm_oss_file *pcm_oss_file, int static int snd_pcm_oss_set_fragment1(struct snd_pcm_substream *substream, unsigned int val) { struct snd_pcm_runtime *runtime; + int fragshift; runtime = substream->runtime; if (runtime->oss.subdivision || runtime->oss.fragshift) return -EINVAL; - runtime->oss.fragshift = val & 0xffff; + fragshift = val & 0xffff; + if (fragshift >= 25) /* should be large enough */ + return -EINVAL; + runtime->oss.fragshift = fragshift; runtime->oss.maxfrags = (val >> 16) & 0xffff; if (runtime->oss.fragshift < 4) /* < 16 */ runtime->oss.fragshift = 4; @@ -2032,7 +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 ec501fbaabe4..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); @@ -717,8 +744,13 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, runtime->boundary *= 2; /* clear the buffer for avoiding possible kernel info leaks */ - if (runtime->dma_area && !substream->ops->copy_user) - memset(runtime->dma_area, 0, runtime->dma_bytes); + if (runtime->dma_area && !substream->ops->copy_user) { + size_t size = runtime->dma_bytes; + + if (runtime->info & SNDRV_PCM_INFO_MMAP) + size = PAGE_ALIGN(size); + memset(runtime->dma_area, 0, size); + } snd_pcm_timer_resolution_change(substream); snd_pcm_set_state(substream, SNDRV_PCM_STATE_SETUP); @@ -728,14 +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; } @@ -768,22 +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; } @@ -1020,15 +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); @@ -1056,18 +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; } @@ -1197,10 +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; } @@ -1651,21 +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/rawmidi.c b/sound/core/rawmidi.c index 94db4683cfaf..6a3543b8455f 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -72,11 +72,21 @@ static inline unsigned short snd_rawmidi_file_flags(struct file *file) } } -static inline int snd_rawmidi_ready(struct snd_rawmidi_substream *substream) +static inline bool __snd_rawmidi_ready(struct snd_rawmidi_runtime *runtime) +{ + return runtime->avail >= runtime->avail_min; +} + +static bool snd_rawmidi_ready(struct snd_rawmidi_substream *substream) { struct snd_rawmidi_runtime *runtime = substream->runtime; + unsigned long flags; + bool ready; - return runtime->avail >= runtime->avail_min; + spin_lock_irqsave(&runtime->lock, flags); + ready = __snd_rawmidi_ready(runtime); + spin_unlock_irqrestore(&runtime->lock, flags); + return ready; } static inline int snd_rawmidi_ready_append(struct snd_rawmidi_substream *substream, @@ -945,7 +955,7 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream, if (result > 0) { if (runtime->event) schedule_work(&runtime->event_work); - else if (snd_rawmidi_ready(substream)) + else if (__snd_rawmidi_ready(runtime)) wake_up(&runtime->sleep); } spin_unlock_irqrestore(&runtime->lock, flags); @@ -1024,7 +1034,7 @@ static ssize_t snd_rawmidi_read(struct file *file, char __user *buf, size_t coun result = 0; while (count > 0) { spin_lock_irq(&runtime->lock); - while (!snd_rawmidi_ready(substream)) { + while (!__snd_rawmidi_ready(runtime)) { wait_queue_entry_t wait; if ((file->f_flags & O_NONBLOCK) != 0 || result > 0) { @@ -1041,9 +1051,11 @@ static ssize_t snd_rawmidi_read(struct file *file, char __user *buf, size_t coun return -ENODEV; if (signal_pending(current)) return result > 0 ? result : -ERESTARTSYS; - if (!runtime->avail) - return result > 0 ? result : -EIO; spin_lock_irq(&runtime->lock); + if (!runtime->avail) { + spin_unlock_irq(&runtime->lock); + return result > 0 ? result : -EIO; + } } spin_unlock_irq(&runtime->lock); count1 = snd_rawmidi_kernel_read1(substream, @@ -1181,7 +1193,7 @@ int __snd_rawmidi_transmit_ack(struct snd_rawmidi_substream *substream, int coun runtime->avail += count; substream->bytes += count; if (count > 0) { - if (runtime->drain || snd_rawmidi_ready(substream)) + if (runtime->drain || __snd_rawmidi_ready(runtime)) wake_up(&runtime->sleep); } return count; @@ -1370,9 +1382,11 @@ static ssize_t snd_rawmidi_write(struct file *file, const char __user *buf, return -ENODEV; if (signal_pending(current)) return result > 0 ? result : -ERESTARTSYS; - if (!runtime->avail && !timeout) - return result > 0 ? result : -EIO; spin_lock_irq(&runtime->lock); + if (!runtime->avail && !timeout) { + spin_unlock_irq(&runtime->lock); + return result > 0 ? result : -EIO; + } } spin_unlock_irq(&runtime->lock); count1 = snd_rawmidi_kernel_write1(substream, buf, NULL, count); @@ -1452,6 +1466,7 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, struct snd_rawmidi *rmidi; struct snd_rawmidi_substream *substream; struct snd_rawmidi_runtime *runtime; + unsigned long buffer_size, avail, xruns; rmidi = entry->private_data; snd_iprintf(buffer, "%s\n\n", rmidi->name); @@ -1470,13 +1485,16 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, " Owner PID : %d\n", pid_vnr(substream->pid)); runtime = substream->runtime; + spin_lock_irq(&runtime->lock); + buffer_size = runtime->buffer_size; + avail = runtime->avail; + spin_unlock_irq(&runtime->lock); snd_iprintf(buffer, " Mode : %s\n" " Buffer size : %lu\n" " Avail : %lu\n", runtime->oss ? "OSS compatible" : "native", - (unsigned long) runtime->buffer_size, - (unsigned long) runtime->avail); + buffer_size, avail); } } } @@ -1494,13 +1512,16 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, " Owner PID : %d\n", pid_vnr(substream->pid)); runtime = substream->runtime; + spin_lock_irq(&runtime->lock); + buffer_size = runtime->buffer_size; + avail = runtime->avail; + xruns = runtime->xruns; + spin_unlock_irq(&runtime->lock); snd_iprintf(buffer, " Buffer size : %lu\n" " Avail : %lu\n" " Overruns : %lu\n", - (unsigned long) runtime->buffer_size, - (unsigned long) runtime->avail, - (unsigned long) runtime->xruns); + buffer_size, avail, xruns); } } } diff --git a/sound/core/seq/oss/seq_oss.c b/sound/core/seq/oss/seq_oss.c index c8b9c0b315d8..250a92b18726 100644 --- a/sound/core/seq/oss/seq_oss.c +++ b/sound/core/seq/oss/seq_oss.c @@ -174,9 +174,12 @@ odev_ioctl(struct file *file, unsigned int cmd, unsigned long arg) if (snd_BUG_ON(!dp)) return -ENXIO; - mutex_lock(®ister_mutex); + if (cmd != SNDCTL_SEQ_SYNC && + mutex_lock_interruptible(®ister_mutex)) + return -ERESTARTSYS; rc = snd_seq_oss_ioctl(dp, cmd, arg); - mutex_unlock(®ister_mutex); + if (cmd != SNDCTL_SEQ_SYNC) + mutex_unlock(®ister_mutex); return rc; } diff --git a/sound/core/seq/oss/seq_oss_synth.c b/sound/core/seq/oss/seq_oss_synth.c index 11554d0412f0..1b8409ec2c97 100644 --- a/sound/core/seq/oss/seq_oss_synth.c +++ b/sound/core/seq/oss/seq_oss_synth.c @@ -611,7 +611,8 @@ snd_seq_oss_synth_make_info(struct seq_oss_devinfo *dp, int dev, struct synth_in if (info->is_midi) { struct midi_info minf; - snd_seq_oss_midi_make_info(dp, info->midi_mapped, &minf); + if (snd_seq_oss_midi_make_info(dp, info->midi_mapped, &minf)) + return -ENXIO; inf->synth_type = SYNTH_TYPE_MIDI; inf->synth_subtype = 0; inf->nr_voices = 16; diff --git a/sound/core/seq/seq_ports.c b/sound/core/seq/seq_ports.c index 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/seq_queue.h b/sound/core/seq/seq_queue.h index 9254c8dbe5e3..25d2d6b61007 100644 --- a/sound/core/seq/seq_queue.h +++ b/sound/core/seq/seq_queue.h @@ -26,10 +26,10 @@ struct snd_seq_queue { struct snd_seq_timer *timer; /* time keeper for this queue */ int owner; /* client that 'owns' the timer */ - unsigned int locked:1, /* timer is only accesibble by owner if set */ - klocked:1, /* kernel lock (after START) */ - check_again:1, - check_blocked:1; + bool locked; /* timer is only accesibble by owner if set */ + bool klocked; /* kernel lock (after START) */ + bool check_again; /* concurrent access happened during check */ + bool check_blocked; /* queue being checked */ unsigned int flags; /* status flags */ unsigned int info_flags; /* info for sync */ diff --git a/sound/core/seq_device.c b/sound/core/seq_device.c index 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 013f0e69ff0f..d684aa4150aa 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -491,9 +491,10 @@ static void snd_timer_notify1(struct snd_timer_instance *ti, int event) return; if (timer->hw.flags & SNDRV_TIMER_HW_SLAVE) return; + event += 10; /* convert to SNDRV_TIMER_EVENT_MXXX */ list_for_each_entry(ts, &ti->slave_active_head, active_list) if (ts->ccallback) - ts->ccallback(ts, event + 100, &tstamp, resolution); + ts->ccallback(ts, event, &tstamp, resolution); } /* start/continue a master timer */ @@ -594,13 +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) { @@ -635,23 +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/aloop.c b/sound/drivers/aloop.c index 9ccdad89c288..452b9eaca815 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -1035,6 +1035,14 @@ static int loopback_mixer_new(struct loopback *loopback, int notify) return -ENOMEM; kctl->id.device = dev; kctl->id.subdevice = substr; + + /* Add the control before copying the id so that + * the numid field of the id is set in the copy. + */ + err = snd_ctl_add(card, kctl); + if (err < 0) + return err; + switch (idx) { case ACTIVE_IDX: setup->active_id = kctl->id; @@ -1051,9 +1059,6 @@ static int loopback_mixer_new(struct loopback *loopback, int notify) default: break; } - err = snd_ctl_add(card, kctl); - if (err < 0) - return err; } } } diff --git a/sound/drivers/opl3/opl3_midi.c b/sound/drivers/opl3/opl3_midi.c index 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 b0a904cdb932..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.2 pro/d.4 pro + * Mackie(Loud) d.2 pro/d.4 pro (built-in FireWire card with OXFW971 ASIC) * Mackie(Loud) U.420/U.420d * TASCAM FireOne * Stanton Controllers & Systems 1 Deck/Mixer @@ -84,7 +84,7 @@ config SND_BEBOB * PreSonus FIREBOX/FIREPOD/FP10/Inspire1394 * BridgeCo RDAudio1/Audio5 * Mackie Onyx 1220/1620/1640 (FireWire I/O Card) - * Mackie d.2 (FireWire Option) + * Mackie d.2 (optional FireWire card with DM1000 ASIC) * Stanton FinalScratch 2 (ScratchAmp) * Tascam IF-FW/DM * Behringer XENIX UFX 1204/1604 @@ -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/amdtp-stream.c b/sound/firewire/amdtp-stream.c index e50e28f77e74..07e58dcad4af 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -617,18 +617,24 @@ static int parse_ir_ctx_header(struct amdtp_stream *s, unsigned int cycle, unsigned int *syt, unsigned int index) { const __be32 *cip_header; + unsigned int cip_header_size; int err; *payload_length = be32_to_cpu(ctx_header[0]) >> ISO_DATA_LENGTH_SHIFT; - if (*payload_length > s->ctx_data.tx.ctx_header_size + - s->ctx_data.tx.max_ctx_payload_length) { + + if (!(s->flags & CIP_NO_HEADER)) + cip_header_size = 8; + else + cip_header_size = 0; + + if (*payload_length > cip_header_size + s->ctx_data.tx.max_ctx_payload_length) { dev_err(&s->unit->device, "Detect jumbo payload: %04x %04x\n", - *payload_length, s->ctx_data.tx.max_ctx_payload_length); + *payload_length, cip_header_size + s->ctx_data.tx.max_ctx_payload_length); return -EIO; } - if (!(s->flags & CIP_NO_HEADER)) { + if (cip_header_size > 0) { cip_header = ctx_header + 2; err = check_cip_header(s, cip_header, *payload_length, data_blocks, data_block_counter, syt); @@ -932,23 +938,22 @@ static int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed) s->ctx_data.rx.last_syt_offset = TICKS_PER_CYCLE; } - /* initialize packet buffer */ + // initialize packet buffer. + max_ctx_payload_size = amdtp_stream_get_max_payload(s); if (s->direction == AMDTP_IN_STREAM) { dir = DMA_FROM_DEVICE; type = FW_ISO_CONTEXT_RECEIVE; - if (!(s->flags & CIP_NO_HEADER)) + if (!(s->flags & CIP_NO_HEADER)) { + max_ctx_payload_size -= 8; ctx_header_size = IR_CTX_HEADER_SIZE_CIP; - else + } else { ctx_header_size = IR_CTX_HEADER_SIZE_NO_CIP; - - max_ctx_payload_size = amdtp_stream_get_max_payload(s) - - ctx_header_size; + } } else { dir = DMA_TO_DEVICE; type = FW_ISO_CONTEXT_TRANSMIT; ctx_header_size = 0; // No effect for IT context. - max_ctx_payload_size = amdtp_stream_get_max_payload(s); if (!(s->flags & CIP_NO_HEADER)) max_ctx_payload_size -= IT_PKT_HEADER_SIZE_CIP; } diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c index 976d8cb9a34f..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) */ + // 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/bebob/bebob_hwdep.c b/sound/firewire/bebob/bebob_hwdep.c index 45b740f44c45..c362eb38ab90 100644 --- a/sound/firewire/bebob/bebob_hwdep.c +++ b/sound/firewire/bebob/bebob_hwdep.c @@ -36,12 +36,11 @@ hwdep_read(struct snd_hwdep *hwdep, char __user *buf, long count, } memset(&event, 0, sizeof(event)); + count = min_t(long, count, sizeof(event.lock_status)); if (bebob->dev_lock_changed) { event.lock_status.type = SNDRV_FIREWIRE_EVENT_LOCK_STATUS; event.lock_status.status = (bebob->dev_lock_count > 0); bebob->dev_lock_changed = false; - - count = min_t(long, count, sizeof(event.lock_status)); } spin_unlock_irq(&bebob->lock); diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c index ce07ea0d4e71..3935e90c8e8f 100644 --- a/sound/firewire/bebob/bebob_stream.c +++ b/sound/firewire/bebob/bebob_stream.c @@ -534,20 +534,22 @@ int snd_bebob_stream_init_duplex(struct snd_bebob *bebob) static int keep_resources(struct snd_bebob *bebob, struct amdtp_stream *stream, unsigned int rate, unsigned int index) { - struct snd_bebob_stream_formation *formation; + unsigned int pcm_channels; + unsigned int midi_ports; struct cmp_connection *conn; int err; if (stream == &bebob->tx_stream) { - formation = bebob->tx_stream_formations + index; + pcm_channels = bebob->tx_stream_formations[index].pcm; + midi_ports = bebob->midi_input_ports; conn = &bebob->out_conn; } else { - formation = bebob->rx_stream_formations + index; + pcm_channels = bebob->rx_stream_formations[index].pcm; + midi_ports = bebob->midi_output_ports; conn = &bebob->in_conn; } - err = amdtp_am824_set_parameters(stream, rate, formation->pcm, - formation->midi, false); + err = amdtp_am824_set_parameters(stream, rate, pcm_channels, midi_ports, false); if (err < 0) return err; diff --git a/sound/firewire/dice/dice-alesis.c b/sound/firewire/dice/dice-alesis.c index f5b325263b67..39a4ef8c8543 100644 --- a/sound/firewire/dice/dice-alesis.c +++ b/sound/firewire/dice/dice-alesis.c @@ -16,7 +16,7 @@ alesis_io14_tx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT] = { static const unsigned int alesis_io26_tx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT] = { {10, 10, 4}, /* Tx0 = Analog + S/PDIF. */ - {16, 8, 0}, /* Tx1 = ADAT1 + ADAT2. */ + {16, 4, 0}, /* Tx1 = ADAT1 + ADAT2 (available at low rate). */ }; int snd_dice_detect_alesis_formats(struct snd_dice *dice) diff --git a/sound/firewire/dice/dice-stream.c b/sound/firewire/dice/dice-stream.c index f6a8627ae5a2..0aa3c56cf390 100644 --- a/sound/firewire/dice/dice-stream.c +++ b/sound/firewire/dice/dice-stream.c @@ -489,11 +489,10 @@ void snd_dice_stream_stop_duplex(struct snd_dice *dice) struct reg_params tx_params, rx_params; if (dice->substreams_counter == 0) { - if (get_register_params(dice, &tx_params, &rx_params) >= 0) { - amdtp_domain_stop(&dice->domain); + if (get_register_params(dice, &tx_params, &rx_params) >= 0) finish_session(dice, &tx_params, &rx_params); - } + amdtp_domain_stop(&dice->domain); release_resources(dice); } } diff --git a/sound/firewire/dice/dice-tcelectronic.c b/sound/firewire/dice/dice-tcelectronic.c index a8875d24ba2a..43a3bcb15b3d 100644 --- a/sound/firewire/dice/dice-tcelectronic.c +++ b/sound/firewire/dice/dice-tcelectronic.c @@ -38,8 +38,8 @@ static const struct dice_tc_spec konnekt_24d = { }; static const struct dice_tc_spec konnekt_live = { - .tx_pcm_chs = {{16, 16, 16}, {0, 0, 0} }, - .rx_pcm_chs = {{16, 16, 16}, {0, 0, 0} }, + .tx_pcm_chs = {{16, 16, 6}, {0, 0, 0} }, + .rx_pcm_chs = {{16, 16, 6}, {0, 0, 0} }, .has_midi = true, }; diff --git a/sound/firewire/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/fireface/ff-transaction.c b/sound/firewire/fireface/ff-transaction.c index 7f82762ccc8c..ee7122c461d4 100644 --- a/sound/firewire/fireface/ff-transaction.c +++ b/sound/firewire/fireface/ff-transaction.c @@ -88,7 +88,7 @@ static void transmit_midi_msg(struct snd_ff *ff, unsigned int port) /* Set interval to next transaction. */ ff->next_ktime[port] = ktime_add_ns(ktime_get(), - ff->rx_bytes[port] * 8 * NSEC_PER_SEC / 31250); + ff->rx_bytes[port] * 8 * (NSEC_PER_SEC / 31250)); if (quad_count == 1) tcode = TCODE_WRITE_QUADLET_REQUEST; diff --git a/sound/firewire/fireworks/fireworks_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/fireworks/fireworks_transaction.c b/sound/firewire/fireworks/fireworks_transaction.c index 0f533f5bd960..9f8c53b39f95 100644 --- a/sound/firewire/fireworks/fireworks_transaction.c +++ b/sound/firewire/fireworks/fireworks_transaction.c @@ -123,7 +123,7 @@ copy_resp_to_buf(struct snd_efw *efw, void *data, size_t length, int *rcode) t = (struct snd_efw_transaction *)data; length = min_t(size_t, be32_to_cpu(t->length) * sizeof(u32), length); - spin_lock_irq(&efw->lock); + spin_lock(&efw->lock); if (efw->push_ptr < efw->pull_ptr) capacity = (unsigned int)(efw->pull_ptr - efw->push_ptr); @@ -190,7 +190,7 @@ handle_resp_for_user(struct fw_card *card, int generation, int source, copy_resp_to_buf(efw, data, length, rcode); end: - spin_unlock_irq(&instances_lock); + spin_unlock(&instances_lock); } static void diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index fb6df3fc018e..bebb2b8296cb 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -350,8 +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.2 pro: Unknown - * d.4 pro: Unknown + * d.2 pro/d.4 pro (built-in card): Unknown * U.420: Unknown * U.420d: Unknown */ diff --git a/sound/firewire/tascam/tascam-transaction.c b/sound/firewire/tascam/tascam-transaction.c index 90288b4b4637..a073cece4a7d 100644 --- a/sound/firewire/tascam/tascam-transaction.c +++ b/sound/firewire/tascam/tascam-transaction.c @@ -209,7 +209,7 @@ static void midi_port_work(struct work_struct *work) /* Set interval to next transaction. */ port->next_ktime = ktime_add_ns(ktime_get(), - port->consume_bytes * 8 * NSEC_PER_SEC / 31250); + port->consume_bytes * 8 * (NSEC_PER_SEC / 31250)); /* Start this transaction. */ port->idling = false; diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c index 09ff209df4a3..c87187f63573 100644 --- a/sound/hda/ext/hdac_ext_controller.c +++ b/sound/hda/ext/hdac_ext_controller.c @@ -148,6 +148,8 @@ struct hdac_ext_link *snd_hdac_ext_bus_get_link(struct hdac_bus *bus, return NULL; if (bus->idx != bus_idx) return NULL; + if (addr < 0 || addr > 31) + return NULL; list_for_each_entry(hlink, &bus->hlink_list, list) { for (i = 0; i < HDA_MAX_CODECS; i++) { 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_device.c b/sound/hda/hdac_device.c index b84e12f4f804..489f996d86bc 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -656,6 +656,7 @@ static struct hda_vendor_id hda_vendor_ids[] = { { 0x14f1, "Conexant" }, { 0x17e8, "Chrontel" }, { 0x1854, "LG" }, + { 0x19e5, "Huawei" }, { 0x1aec, "Wolfson Microelectronics" }, { 0x1af4, "QEMU" }, { 0x434d, "C-Media" }, 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/hda/intel-nhlt.c b/sound/hda/intel-nhlt.c index daede96f28ee..6ed80a4cba01 100644 --- a/sound/hda/intel-nhlt.c +++ b/sound/hda/intel-nhlt.c @@ -64,18 +64,49 @@ int intel_nhlt_get_dmic_geo(struct device *dev, struct nhlt_acpi_table *nhlt) struct nhlt_endpoint *epnt; struct nhlt_dmic_array_config *cfg; struct nhlt_vendor_dmic_array_config *cfg_vendor; + struct nhlt_fmt *fmt_configs; unsigned int dmic_geo = 0; - u8 j; + u16 max_ch = 0; + u8 i, j; if (!nhlt) return 0; - epnt = (struct nhlt_endpoint *)nhlt->desc; + if (nhlt->header.length <= sizeof(struct acpi_table_header)) { + dev_warn(dev, "Invalid DMIC description table\n"); + return 0; + } + + for (j = 0, epnt = nhlt->desc; j < nhlt->endpoint_count; j++, + epnt = (struct nhlt_endpoint *)((u8 *)epnt + epnt->length)) { + + if (epnt->linktype != NHLT_LINK_DMIC) + continue; + + cfg = (struct nhlt_dmic_array_config *)(epnt->config.caps); + fmt_configs = (struct nhlt_fmt *)(epnt->config.caps + epnt->config.size); + + /* find max number of channels based on format_configuration */ + if (fmt_configs->fmt_count) { + dev_dbg(dev, "%s: found %d format definitions\n", + __func__, fmt_configs->fmt_count); + + for (i = 0; i < fmt_configs->fmt_count; i++) { + struct wav_fmt_ext *fmt_ext; + + fmt_ext = &fmt_configs->fmt_config[i].fmt_ext; - for (j = 0; j < nhlt->endpoint_count; j++) { - if (epnt->linktype == NHLT_LINK_DMIC) { - cfg = (struct nhlt_dmic_array_config *) - (epnt->config.caps); + if (fmt_ext->fmt.channels > max_ch) + max_ch = fmt_ext->fmt.channels; + } + dev_dbg(dev, "%s: max channels found %d\n", __func__, max_ch); + } else { + dev_dbg(dev, "%s: No format information found\n", __func__); + } + + if (cfg->device_config.config_type != NHLT_CONFIG_TYPE_MIC_ARRAY) { + dmic_geo = max_ch; + } else { switch (cfg->array_type) { case NHLT_MIC_ARRAY_2CH_SMALL: case NHLT_MIC_ARRAY_2CH_BIG: @@ -92,13 +123,23 @@ int intel_nhlt_get_dmic_geo(struct device *dev, struct nhlt_acpi_table *nhlt) dmic_geo = cfg_vendor->nb_mics; break; default: - dev_warn(dev, "undefined DMIC array_type 0x%0x\n", - cfg->array_type); + dev_warn(dev, "%s: undefined DMIC array_type 0x%0x\n", + __func__, cfg->array_type); + } + + if (dmic_geo > 0) { + dev_dbg(dev, "%s: Array with %d dmics\n", __func__, dmic_geo); + } + if (max_ch > dmic_geo) { + dev_dbg(dev, "%s: max channels %d exceed dmic number %d\n", + __func__, max_ch, dmic_geo); } } - epnt = (struct nhlt_endpoint *)((u8 *)epnt + epnt->length); } + dev_dbg(dev, "%s: dmic number %d max_ch %d\n", + __func__, dmic_geo, max_ch); + return dmic_geo; } EXPORT_SYMBOL_GPL(intel_nhlt_get_dmic_geo); diff --git a/sound/isa/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/gus/gus_main.c b/sound/isa/gus/gus_main.c index af6b4d89d695..39911a637e80 100644 --- a/sound/isa/gus/gus_main.c +++ b/sound/isa/gus/gus_main.c @@ -77,17 +77,8 @@ static const struct snd_kcontrol_new snd_gus_joystick_control = { static void snd_gus_init_control(struct snd_gus_card *gus) { - int ret; - - if (!gus->ace_flag) { - ret = - snd_ctl_add(gus->card, - snd_ctl_new1(&snd_gus_joystick_control, - gus)); - if (ret) - snd_printk(KERN_ERR "gus: snd_ctl_add failed: %d\n", - ret); - } + if (!gus->ace_flag) + snd_ctl_add(gus->card, snd_ctl_new1(&snd_gus_joystick_control, gus)); } /* diff --git a/sound/isa/sb/emu8000.c b/sound/isa/sb/emu8000.c index 433e32e254f9..91c96fad0618 100644 --- a/sound/isa/sb/emu8000.c +++ b/sound/isa/sb/emu8000.c @@ -1029,8 +1029,10 @@ snd_emu8000_create_mixer(struct snd_card *card, struct snd_emu8000 *emu) memset(emu->controls, 0, sizeof(emu->controls)); for (i = 0; i < EMU8000_NUM_CONTROLS; i++) { - if ((err = snd_ctl_add(card, emu->controls[i] = snd_ctl_new1(mixer_defs[i], emu))) < 0) + if ((err = snd_ctl_add(card, emu->controls[i] = snd_ctl_new1(mixer_defs[i], emu))) < 0) { + emu->controls[i] = NULL; goto __error; + } } return 0; diff --git a/sound/isa/sb/sb16_csp.c b/sound/isa/sb/sb16_csp.c index 4ad0ff0c4508..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); @@ -1045,10 +1049,14 @@ static int snd_sb_qsound_build(struct snd_sb_csp * p) spin_lock_init(&p->q_lock); - if ((err = snd_ctl_add(card, p->qsound_switch = snd_ctl_new1(&snd_sb_qsound_switch, p))) < 0) + if ((err = snd_ctl_add(card, p->qsound_switch = snd_ctl_new1(&snd_sb_qsound_switch, p))) < 0) { + p->qsound_switch = NULL; goto __error; - if ((err = snd_ctl_add(card, p->qsound_space = snd_ctl_new1(&snd_sb_qsound_space, p))) < 0) + } + if ((err = snd_ctl_add(card, p->qsound_space = snd_ctl_new1(&snd_sb_qsound_space, p))) < 0) { + p->qsound_space = NULL; goto __error; + } return 0; @@ -1068,10 +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/isa/sb/sb16_main.c b/sound/isa/sb/sb16_main.c index 0768bbf8fd71..679f9f48370f 100644 --- a/sound/isa/sb/sb16_main.c +++ b/sound/isa/sb/sb16_main.c @@ -864,14 +864,10 @@ int snd_sb16dsp_pcm(struct snd_sb *chip, int device) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_sb16_playback_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_sb16_capture_ops); - if (chip->dma16 >= 0 && chip->dma8 != chip->dma16) { - err = snd_ctl_add(card, snd_ctl_new1( - &snd_sb16_dma_control, chip)); - if (err) - return err; - } else { + if (chip->dma16 >= 0 && chip->dma8 != chip->dma16) + snd_ctl_add(card, snd_ctl_new1(&snd_sb16_dma_control, chip)); + else pcm->info_flags = SNDRV_PCM_INFO_HALF_DUPLEX; - } snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, card->dev, diff --git a/sound/isa/sb/sb8.c b/sound/isa/sb/sb8.c index d67eae3988bd..6f1fc8789676 100644 --- a/sound/isa/sb/sb8.c +++ b/sound/isa/sb/sb8.c @@ -96,10 +96,6 @@ static int snd_sb8_probe(struct device *pdev, unsigned int dev) /* block the 0x388 port to avoid PnP conflicts */ acard->fm_res = request_region(0x388, 4, "SoundBlaster FM"); - if (!acard->fm_res) { - err = -EBUSY; - goto _err; - } if (port[dev] != SNDRV_AUTO_PORT) { if ((err = snd_sbdsp_create(card, port[dev], irq[dev], diff --git a/sound/isa/wavefront/wavefront_synth.c b/sound/isa/wavefront/wavefront_synth.c index d6420d224d09..09b368761cc0 100644 --- a/sound/isa/wavefront/wavefront_synth.c +++ b/sound/isa/wavefront/wavefront_synth.c @@ -1088,7 +1088,8 @@ wavefront_send_sample (snd_wavefront_t *dev, if (dataptr < data_end) { - __get_user (sample_short, dataptr); + if (get_user(sample_short, dataptr)) + return -EFAULT; dataptr += skip; if (data_is_unsigned) { /* GUS ? */ 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/cthw20k2.c b/sound/pci/ctxfi/cthw20k2.c index 3cd4b7dad945..b1cc4cdc6c41 100644 --- a/sound/pci/ctxfi/cthw20k2.c +++ b/sound/pci/ctxfi/cthw20k2.c @@ -991,7 +991,7 @@ static int daio_mgr_dao_init(void *blk, unsigned int idx, unsigned int conf) if (idx < 4) { /* S/PDIF output */ - switch ((conf & 0x7)) { + switch ((conf & 0xf)) { case 1: set_field(&ctl->txctl[idx], ATXCTL_NUC, 0); break; diff --git a/sound/pci/ctxfi/ctresource.c b/sound/pci/ctxfi/ctresource.c index 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_bind.c b/sound/pci/hda/hda_bind.c index 6a8564566375..17a25e453f60 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -47,6 +47,10 @@ static void hda_codec_unsol_event(struct hdac_device *dev, unsigned int ev) if (codec->bus->shutdown) return; + /* ignore unsol events during system suspend/resume */ + if (codec->core.dev.power.power_state.event != PM_EVENT_ON) + return; + if (codec->patch_ops.unsol_event) codec->patch_ops.unsol_event(codec, ev); } diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 6da296def283..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); } @@ -1798,7 +1801,7 @@ int snd_hda_codec_reset(struct hda_codec *codec) return -EBUSY; /* OK, let it free */ - snd_hdac_device_unregister(&codec->core); + device_release_driver(hda_codec_dev(codec)); /* allow device access again */ snd_hda_unlock_devices(bus); diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 5e6081750bd9..6a159c6c2f54 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -613,13 +613,6 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) 20, 178000000); - /* by some reason, the playback stream stalls on PulseAudio with - * tsched=1 when a capture stream triggers. Until we figure out the - * real cause, disable tsched mode by telling the PCM info flag. - */ - if (chip->driver_caps & AZX_DCAPS_AMD_WORKAROUND) - runtime->hw.info |= SNDRV_PCM_INFO_BATCH; - if (chip->align_buffer_size) /* constrain buffer sizes to be multiple of 128 bytes. This is more efficient in terms of memory diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index a356fb0e5773..9da7a06d024f 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -41,7 +41,7 @@ /* 24 unused */ #define AZX_DCAPS_COUNT_LPIB_DELAY (1 << 25) /* Take LPIB as delay */ #define AZX_DCAPS_PM_RUNTIME (1 << 26) /* runtime PM support */ -#define AZX_DCAPS_SUSPEND_SPURIOUS_WAKEUP (1 << 27) /* Workaround for spurious wakeups after suspend */ +/* 27 unused */ #define AZX_DCAPS_CORBRP_SELF_CLEAR (1 << 28) /* CORBRP clears itself after reset */ #define AZX_DCAPS_NO_MSI64 (1 << 29) /* Stick to 32-bit MSIs */ #define AZX_DCAPS_SEPARATE_STREAM_TAG (1 << 30) /* capture and playback use separate stream tag */ @@ -143,6 +143,7 @@ struct azx { unsigned int align_buffer_size:1; unsigned int region_requested:1; unsigned int disabled:1; /* disabled by vga_switcheroo */ + unsigned int pm_prepared:1; /* GTS present */ unsigned int gts_present:1; diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index e1750bdbe51f..e92fcb150e57 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -1202,11 +1202,17 @@ static const char *get_line_out_pfx(struct hda_codec *codec, int ch, *index = ch; return "Headphone"; case AUTO_PIN_LINE_OUT: - /* This deals with the case where we have two DACs and - * one LO, one HP and one Speaker */ - if (!ch && cfg->speaker_outs && cfg->hp_outs) { - bool hp_lo_shared = !path_has_mixer(codec, spec->hp_paths[0], ctl_type); - bool spk_lo_shared = !path_has_mixer(codec, spec->speaker_paths[0], ctl_type); + /* This deals with the case where one HP or one Speaker or + * one HP + one Speaker need to share the DAC with LO + */ + if (!ch) { + bool hp_lo_shared = false, spk_lo_shared = false; + + if (cfg->speaker_outs) + spk_lo_shared = !path_has_mixer(codec, + spec->speaker_paths[0], ctl_type); + if (cfg->hp_outs) + hp_lo_shared = !path_has_mixer(codec, spec->hp_paths[0], ctl_type); if (hp_lo_shared && spk_lo_shared) return spec->vmaster_mute.hook ? "PCM" : "Master"; if (hp_lo_shared) @@ -1364,16 +1370,20 @@ static int try_assign_dacs(struct hda_codec *codec, int num_outs, struct nid_path *path; hda_nid_t pin = pins[i]; - path = snd_hda_get_path_from_idx(codec, path_idx[i]); - if (path) { - badness += assign_out_path_ctls(codec, path); - continue; + if (!spec->obey_preferred_dacs) { + path = snd_hda_get_path_from_idx(codec, path_idx[i]); + if (path) { + badness += assign_out_path_ctls(codec, path); + continue; + } } dacs[i] = get_preferred_dac(codec, pin); if (dacs[i]) { if (is_dac_already_used(codec, dacs[i])) badness += bad->shared_primary; + } else if (spec->obey_preferred_dacs) { + badness += BAD_NO_PRIMARY_DAC; } if (!dacs[i]) @@ -3448,7 +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; @@ -3458,9 +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) @@ -4013,7 +4027,7 @@ int snd_hda_gen_add_micmute_led(struct hda_codec *codec, spec->micmute_led.led_mode = MICMUTE_LED_FOLLOW_MUTE; spec->micmute_led.capture = 0; - spec->micmute_led.led_value = 0; + spec->micmute_led.led_value = -1; spec->micmute_led.old_hook = spec->cap_sync_hook; spec->micmute_led.update = hook; spec->cap_sync_hook = update_micmute_led; diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index fb9f1a90238b..e728df6145ad 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -236,6 +236,7 @@ struct hda_gen_spec { unsigned int power_down_unused:1; /* power down unused widgets */ unsigned int dac_min_mute:1; /* minimal = mute for DACs */ unsigned int suppress_vmaster:1; /* don't create vmaster kctls */ + unsigned int obey_preferred_dacs:1; /* obey preferred_dacs assignment */ /* other internal flags */ unsigned int no_analog:1; /* digital I/O only */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 590ea262f2e2..b8fe0ec5d624 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -295,8 +295,7 @@ enum { /* PCH for HSW/BDW; with runtime PM */ /* no i915 binding for this as HSW/BDW has another controller for HDMI */ #define AZX_DCAPS_INTEL_PCH \ - (AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_PM_RUNTIME |\ - AZX_DCAPS_SUSPEND_SPURIOUS_WAKEUP) + (AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_PM_RUNTIME) /* HSW HDMI */ #define AZX_DCAPS_INTEL_HASWELL \ @@ -672,13 +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) @@ -715,6 +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 */ } @@ -908,11 +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); } @@ -984,7 +1001,7 @@ static void __azx_runtime_suspend(struct azx *chip) display_power(chip, false); } -static void __azx_runtime_resume(struct azx *chip, bool from_rt) +static void __azx_runtime_resume(struct azx *chip) { struct hda_intel *hda = container_of(chip, struct hda_intel, chip); struct hdac_bus *bus = azx_bus(chip); @@ -1001,12 +1018,15 @@ static void __azx_runtime_resume(struct azx *chip, bool from_rt) azx_init_pci(chip); hda_intel_init_chip(chip, true); - if (status && from_rt) { - list_for_each_codec(codec, &chip->bus) - if (!codec->relaxed_resume && - (status & (1 << codec->addr))) - schedule_delayed_work(&codec->jackpoll_work, - codec->jackpoll_interval); + /* Avoid codec resume if runtime resume is for system suspend */ + if (!chip->pm_prepared) { + list_for_each_codec(codec, &chip->bus) { + if (codec->relaxed_resume) + continue; + + if (codec->forced_resume || (status & (1 << codec->addr))) + pm_request_resume(hda_codec_dev(codec)); + } } /* power down again for link-controlled chips */ @@ -1015,6 +1035,39 @@ static void __azx_runtime_resume(struct azx *chip, bool from_rt) } #ifdef CONFIG_PM_SLEEP +static int azx_prepare(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip; + + if (!azx_is_pm_ready(card)) + return 0; + + chip = card->private_data; + chip->pm_prepared = 1; + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + + flush_work(&azx_bus(chip)->unsol_work); + + /* HDA controller always requires different WAKEEN for runtime suspend + * and system suspend, so don't use direct-complete here. + */ + return 0; +} + +static void azx_complete(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip; + + if (!azx_is_pm_ready(card)) + return; + + chip = card->private_data; + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + chip->pm_prepared = 0; +} + static int azx_suspend(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); @@ -1026,15 +1079,7 @@ static int azx_suspend(struct device *dev) chip = card->private_data; bus = azx_bus(chip); - snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); - /* An ugly workaround: direct call of __azx_runtime_suspend() and - * __azx_runtime_resume() for old Intel platforms that suffer from - * spurious wakeups after S3 suspend - */ - if (chip->driver_caps & AZX_DCAPS_SUSPEND_SPURIOUS_WAKEUP) - __azx_runtime_suspend(chip); - else - pm_runtime_force_suspend(dev); + __azx_runtime_suspend(chip); if (bus->irq >= 0) { free_irq(bus->irq, chip); bus->irq = -1; @@ -1062,11 +1107,7 @@ static int azx_resume(struct device *dev) if (azx_acquire_irq(chip, 1) < 0) return -EIO; - if (chip->driver_caps & AZX_DCAPS_SUSPEND_SPURIOUS_WAKEUP) - __azx_runtime_resume(chip, false); - else - pm_runtime_force_resume(dev); - snd_power_change_state(card, SNDRV_CTL_POWER_D0); + __azx_runtime_resume(chip); trace_azx_resume(chip); return 0; @@ -1114,10 +1155,7 @@ static int azx_runtime_suspend(struct device *dev) chip = card->private_data; /* enable controller wake up event */ - if (snd_power_get_state(card) == SNDRV_CTL_POWER_D0) { - azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) | - STATESTS_INT_MASK); - } + azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) | STATESTS_INT_MASK); __azx_runtime_suspend(chip); trace_azx_runtime_suspend(chip); @@ -1128,18 +1166,14 @@ static int azx_runtime_resume(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); struct azx *chip; - bool from_rt = snd_power_get_state(card) == SNDRV_CTL_POWER_D0; if (!azx_is_pm_ready(card)) return 0; chip = card->private_data; - __azx_runtime_resume(chip, from_rt); + __azx_runtime_resume(chip); /* disable controller Wake Up event*/ - if (from_rt) { - azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) & - ~STATESTS_INT_MASK); - } + azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) & ~STATESTS_INT_MASK); trace_azx_runtime_resume(chip); return 0; @@ -1173,6 +1207,8 @@ static int azx_runtime_idle(struct device *dev) static const struct dev_pm_ops azx_pm = { SET_SYSTEM_SLEEP_PM_OPS(azx_suspend, azx_resume) #ifdef CONFIG_PM_SLEEP + .prepare = azx_prepare, + .complete = azx_complete, .freeze_noirq = azx_freeze_noirq, .thaw_noirq = azx_thaw_noirq, #endif @@ -1572,6 +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), {} @@ -1757,8 +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 */ @@ -1790,6 +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"); @@ -2178,8 +2215,6 @@ static struct snd_pci_quirk power_save_blacklist[] = { SND_PCI_QUIRK(0x1849, 0x7662, "Asrock H81M-HDS", 0), /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ SND_PCI_QUIRK(0x1043, 0x8733, "Asus Prime X370-Pro", 0), - /* https://bugzilla.redhat.com/show_bug.cgi?id=1581607 */ - SND_PCI_QUIRK(0x1558, 0x3501, "Clevo W35xSS_370SS", 0), /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ SND_PCI_QUIRK(0x1558, 0x6504, "Clevo W65_67SB", 0), /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ @@ -2320,6 +2355,7 @@ static int azx_probe_continue(struct azx *chip) if (azx_has_pm_runtime(chip)) { pm_runtime_use_autosuspend(&pci->dev); + pm_runtime_allow(&pci->dev); pm_runtime_put_autosuspend(&pci->dev); } @@ -2440,9 +2476,14 @@ static const struct pci_device_id azx_ids[] = { /* CometLake-H */ { PCI_DEVICE(0x8086, 0x06C8), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + { PCI_DEVICE(0x8086, 0xf1c8), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* CometLake-S */ { PCI_DEVICE(0x8086, 0xa3f0), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* CometLake-R */ + { PCI_DEVICE(0x8086, 0xf0c8), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Icelake */ { PCI_DEVICE(0x8086, 0x34c8), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, diff --git a/sound/pci/hda/hda_sysfs.c b/sound/pci/hda/hda_sysfs.c index 6dbe99131bc4..91b4a29a8c36 100644 --- a/sound/pci/hda/hda_sysfs.c +++ b/sound/pci/hda/hda_sysfs.c @@ -139,7 +139,7 @@ static int reconfig_codec(struct hda_codec *codec) "The codec is being used, can't reconfigure.\n"); goto error; } - err = snd_hda_codec_configure(codec); + err = device_reprobe(hda_codec_dev(codec)); if (err < 0) goto error; err = snd_card_register(codec->card); 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_ca0132.c b/sound/pci/hda/patch_ca0132.c index 6aa39339db0a..9412bdda85c8 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -93,7 +93,7 @@ enum { }; /* Strings for Input Source Enum Control */ -static const char *const in_src_str[3] = {"Rear Mic", "Line", "Front Mic" }; +static const char *const in_src_str[3] = { "Microphone", "Line In", "Front Microphone" }; #define IN_SRC_NUM_OF_INPUTS 3 enum { REAR_MIC, @@ -1065,6 +1065,7 @@ enum { QUIRK_R3DI, QUIRK_R3D, QUIRK_AE5, + QUIRK_AE7, }; #ifdef CONFIG_PCI @@ -1146,7 +1147,7 @@ static const struct hda_pintbl ae5_pincfgs[] = { { 0x0e, 0x01c510f0 }, /* SPDIF In */ { 0x0f, 0x01017114 }, /* Port A -- Rear L/R. */ { 0x10, 0x01017012 }, /* Port D -- Center/LFE or FP Hp */ - { 0x11, 0x01a170ff }, /* Port B -- LineMicIn2 / Rear Headphone */ + { 0x11, 0x012170ff }, /* Port B -- LineMicIn2 / Rear Headphone */ { 0x12, 0x01a170f0 }, /* Port C -- LineIn1 */ { 0x13, 0x908700f0 }, /* What U Hear In*/ { 0x18, 0x50d000f0 }, /* N/A */ @@ -1184,6 +1185,8 @@ static const struct snd_pci_quirk ca0132_quirks[] = { SND_PCI_QUIRK(0x1102, 0x0013, "Recon3D", QUIRK_R3D), SND_PCI_QUIRK(0x1102, 0x0018, "Recon3D", QUIRK_R3D), SND_PCI_QUIRK(0x1102, 0x0051, "Sound Blaster AE-5", QUIRK_AE5), + SND_PCI_QUIRK(0x1102, 0x0191, "Sound Blaster AE-5 Plus", QUIRK_AE5), + SND_PCI_QUIRK(0x1102, 0x0081, "Sound Blaster AE-7", QUIRK_AE7), {} }; @@ -4674,6 +4677,15 @@ static int ca0132_alt_select_in(struct hda_codec *codec) ca0113_mmio_command_set(codec, 0x30, 0x28, 0x00); tmp = FLOAT_THREE; break; + case QUIRK_AE7: + ca0113_mmio_command_set(codec, 0x30, 0x28, 0x00); + tmp = FLOAT_THREE; + chipio_set_conn_rate(codec, MEM_CONNID_MICIN2, + SR_96_000); + chipio_set_conn_rate(codec, MEM_CONNID_MICOUT2, + SR_96_000); + dspio_set_uint_param(codec, 0x80, 0x01, FLOAT_ZERO); + break; default: tmp = FLOAT_ONE; break; @@ -4719,6 +4731,14 @@ static int ca0132_alt_select_in(struct hda_codec *codec) case QUIRK_AE5: ca0113_mmio_command_set(codec, 0x30, 0x28, 0x00); break; + case QUIRK_AE7: + ca0113_mmio_command_set(codec, 0x30, 0x28, 0x3f); + chipio_set_conn_rate(codec, MEM_CONNID_MICIN2, + SR_96_000); + chipio_set_conn_rate(codec, MEM_CONNID_MICOUT2, + SR_96_000); + dspio_set_uint_param(codec, 0x80, 0x01, FLOAT_ZERO); + break; default: break; } @@ -4728,7 +4748,10 @@ static int ca0132_alt_select_in(struct hda_codec *codec) if (ca0132_quirk(spec) == QUIRK_R3DI) chipio_set_conn_rate(codec, 0x0F, SR_96_000); - tmp = FLOAT_ZERO; + if (ca0132_quirk(spec) == QUIRK_AE7) + tmp = FLOAT_THREE; + else + tmp = FLOAT_ZERO; dspio_set_uint_param(codec, 0x80, 0x00, tmp); switch (ca0132_quirk(spec)) { diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 396b5503038a..b81b0a1bada5 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -898,18 +898,18 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x103c, 0x8079, "HP EliteBook 840 G3", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x807C, "HP EliteBook 820 G3", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x80FD, "HP ProBook 640 G2", CXT_FIXUP_HP_DOCK), - SND_PCI_QUIRK(0x103c, 0x828c, "HP EliteBook 840 G4", CXT_FIXUP_HP_DOCK), - SND_PCI_QUIRK(0x103c, 0x83b2, "HP EliteBook 840 G5", CXT_FIXUP_HP_DOCK), - SND_PCI_QUIRK(0x103c, 0x83b3, "HP EliteBook 830 G5", CXT_FIXUP_HP_DOCK), - SND_PCI_QUIRK(0x103c, 0x83d3, "HP ProBook 640 G4", CXT_FIXUP_HP_DOCK), - SND_PCI_QUIRK(0x103c, 0x8174, "HP Spectre x360", CXT_FIXUP_HP_SPECTRE), SND_PCI_QUIRK(0x103c, 0x8115, "HP Z1 Gen3", CXT_FIXUP_HP_GATE_MIC), SND_PCI_QUIRK(0x103c, 0x814f, "HP ZBook 15u G3", CXT_FIXUP_MUTE_LED_GPIO), + SND_PCI_QUIRK(0x103c, 0x8174, "HP Spectre x360", CXT_FIXUP_HP_SPECTRE), SND_PCI_QUIRK(0x103c, 0x822e, "HP ProBook 440 G4", CXT_FIXUP_MUTE_LED_GPIO), - SND_PCI_QUIRK(0x103c, 0x836e, "HP ProBook 455 G5", CXT_FIXUP_MUTE_LED_GPIO), - SND_PCI_QUIRK(0x103c, 0x837f, "HP ProBook 470 G5", CXT_FIXUP_MUTE_LED_GPIO), + SND_PCI_QUIRK(0x103c, 0x828c, "HP EliteBook 840 G4", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x8299, "HP 800 G3 SFF", CXT_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x829a, "HP 800 G3 DM", CXT_FIXUP_HP_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x103c, 0x836e, "HP ProBook 455 G5", CXT_FIXUP_MUTE_LED_GPIO), + SND_PCI_QUIRK(0x103c, 0x837f, "HP ProBook 470 G5", CXT_FIXUP_MUTE_LED_GPIO), + SND_PCI_QUIRK(0x103c, 0x83b2, "HP EliteBook 840 G5", CXT_FIXUP_HP_DOCK), + SND_PCI_QUIRK(0x103c, 0x83b3, "HP EliteBook 830 G5", CXT_FIXUP_HP_DOCK), + SND_PCI_QUIRK(0x103c, 0x83d3, "HP ProBook 640 G4", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x8402, "HP ProBook 645 G4", CXT_FIXUP_MUTE_LED_GPIO), SND_PCI_QUIRK(0x103c, 0x8455, "HP Z2 G4", CXT_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x8456, "HP Z2 G4 SFF", CXT_FIXUP_HP_MIC_NO_PRESENCE), @@ -1012,6 +1012,13 @@ static int patch_conexant_auto(struct hda_codec *codec) snd_hda_pick_fixup(codec, cxt5051_fixup_models, cxt5051_fixups, cxt_fixups); break; + case 0x14f15098: + codec->pin_amp_workaround = 1; + spec->gen.mixer_nid = 0x22; + spec->gen.add_stereo_mix_input = HDA_HINT_STEREO_MIX_AUTO; + snd_hda_pick_fixup(codec, cxt5066_fixup_models, + cxt5066_fixups, cxt_fixups); + break; case 0x14f150f2: codec->power_save_node = 1; /* Fall through */ @@ -1041,11 +1048,11 @@ static int patch_conexant_auto(struct hda_codec *codec) if (err < 0) goto error; - err = snd_hda_gen_parse_auto_config(codec, &spec->gen.autocfg); + err = cx_auto_parse_beep(codec); if (err < 0) goto error; - err = cx_auto_parse_beep(codec); + err = snd_hda_gen_parse_auto_config(codec, &spec->gen.autocfg); if (err < 0) goto error; @@ -1075,6 +1082,7 @@ static int patch_conexant_auto(struct hda_codec *codec) static const struct hda_device_id snd_hda_id_conexant[] = { HDA_CODEC_ENTRY(0x14f11f86, "CX8070", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f12008, "CX8200", patch_conexant_auto), + HDA_CODEC_ENTRY(0x14f120d0, "CX11970", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f15045, "CX20549 (Venice)", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f15047, "CX20551 (Waikiki)", patch_conexant_auto), HDA_CODEC_ENTRY(0x14f15051, "CX20561 (Hermosa)", patch_conexant_auto), diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 51798632d334..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), {} }; @@ -2001,22 +2002,25 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo, int pinctl; int err = 0; + mutex_lock(&spec->pcm_lock); if (hinfo->nid) { pcm_idx = hinfo_to_pcm_index(codec, hinfo); - if (snd_BUG_ON(pcm_idx < 0)) - return -EINVAL; + if (snd_BUG_ON(pcm_idx < 0)) { + err = -EINVAL; + goto unlock; + } cvt_idx = cvt_nid_to_cvt_index(codec, hinfo->nid); - if (snd_BUG_ON(cvt_idx < 0)) - return -EINVAL; + if (snd_BUG_ON(cvt_idx < 0)) { + err = -EINVAL; + goto unlock; + } per_cvt = get_cvt(spec, cvt_idx); - snd_BUG_ON(!per_cvt->assigned); per_cvt->assigned = 0; hinfo->nid = 0; azx_stream(get_azx_dev(substream))->stripe = 0; - mutex_lock(&spec->pcm_lock); snd_hda_spdif_ctls_unassign(codec, pcm_idx); clear_bit(pcm_idx, &spec->pcm_in_use); pin_idx = hinfo_to_pin_index(codec, hinfo); @@ -2044,10 +2048,11 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo, per_pin->setup = false; per_pin->channels = 0; mutex_unlock(&per_pin->lock); - unlock: - mutex_unlock(&spec->pcm_lock); } +unlock: + mutex_unlock(&spec->pcm_lock); + return err; } @@ -2378,6 +2383,18 @@ static void generic_hdmi_free(struct hda_codec *codec) } #ifdef CONFIG_PM +static int generic_hdmi_suspend(struct hda_codec *codec) +{ + struct hdmi_spec *spec = codec->spec; + int pin_idx; + + for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { + struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); + cancel_delayed_work_sync(&per_pin->work); + } + return 0; +} + static int generic_hdmi_resume(struct hda_codec *codec) { struct hdmi_spec *spec = codec->spec; @@ -2401,6 +2418,7 @@ static const struct hda_codec_ops generic_hdmi_patch_ops = { .build_controls = generic_hdmi_build_controls, .unsol_event = hdmi_unsol_event, #ifdef CONFIG_PM + .suspend = generic_hdmi_suspend, .resume = generic_hdmi_resume, #endif }; @@ -2550,7 +2568,7 @@ static void generic_acomp_pin_eld_notify(void *audio_ptr, int port, int dev_id) /* skip notification during system suspend (but not in runtime PM); * the state will be updated at resume */ - if (snd_power_get_state(codec->card) != SNDRV_CTL_POWER_D0) + if (codec->core.dev.power.power_state.event == PM_EVENT_SUSPEND) return; /* ditto during suspend/resume process itself */ if (snd_hdac_is_in_pm(&codec->core)) @@ -2755,7 +2773,7 @@ static void intel_pin_eld_notify(void *audio_ptr, int port, int pipe) /* skip notification during system suspend (but not in runtime PM); * the state will be updated at resume */ - if (snd_power_get_state(codec->card) != SNDRV_CTL_POWER_D0) + if (codec->core.dev.power.power_state.event == PM_EVENT_SUSPEND) return; /* ditto during suspend/resume process itself */ if (snd_hdac_is_in_pm(&codec->core)) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4dfd714f718b..804fc0d331b4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -375,17 +375,18 @@ 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: case 0x10ec0245: case 0x10ec0255: case 0x10ec0256: + case 0x19e58326: case 0x10ec0257: case 0x10ec0282: case 0x10ec0283: case 0x10ec0286: - case 0x10ec0287: case 0x10ec0288: case 0x10ec0285: case 0x10ec0298: @@ -396,6 +397,10 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0275: alc_update_coef_idx(codec, 0xe, 0, 1<<0); break; + case 0x10ec0287: + alc_update_coef_idx(codec, 0x10, 1<<9, 0); + alc_write_coef_idx(codec, 0x8, 0x4ab7); + break; case 0x10ec0293: alc_update_coef_idx(codec, 0xa, 1<<13, 0); break; @@ -436,6 +441,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) alc_update_coef_idx(codec, 0x7, 1<<5, 0); break; case 0x10ec0892: + case 0x10ec0897: alc_update_coef_idx(codec, 0x7, 1<<5, 0); break; case 0x10ec0899: @@ -512,6 +518,9 @@ 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 0x19e58326: case 0x10ec0283: case 0x10ec0286: case 0x10ec0288: @@ -1141,6 +1150,7 @@ static int alc_alloc_spec(struct hda_codec *codec, hda_nid_t mixer_nid) codec->single_adc_amp = 1; /* FIXME: do we need this for all Realtek codec models? */ codec->spdif_status_reset = 1; + codec->forced_resume = 1; codec->patch_ops = alc_patch_ops; err = alc_codec_rename_from_preset(codec); @@ -1893,6 +1903,7 @@ enum { ALC889_FIXUP_FRONT_HP_NO_PRESENCE, ALC889_FIXUP_VAIO_TT, ALC888_FIXUP_EEE1601, + ALC886_FIXUP_EAPD, ALC882_FIXUP_EAPD, ALC883_FIXUP_EAPD, ALC883_FIXUP_ACER_EAPD, @@ -1917,9 +1928,13 @@ 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, + ALC887_FIXUP_ASUS_AUDIO, + ALC887_FIXUP_ASUS_HMIC, + ALCS1200A_FIXUP_MIC_VREF, }; static void alc889_fixup_coef(struct hda_codec *codec, @@ -2104,6 +2119,30 @@ 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) @@ -2132,6 +2171,31 @@ static void alc1220_fixup_clevo_pb51ed(struct hda_codec *codec, alc_fixup_headset_mode_no_hp_mic(codec, fix, action); } +static void alc887_asus_hp_automute_hook(struct hda_codec *codec, + struct hda_jack_callback *jack) +{ + struct alc_spec *spec = codec->spec; + unsigned int vref; + + snd_hda_gen_hp_automute(codec, jack); + + if (spec->gen.hp_jack_present) + vref = AC_PINCTL_VREF_80; + else + vref = AC_PINCTL_VREF_HIZ; + snd_hda_set_pin_ctl(codec, 0x19, PIN_HP | vref); +} + +static void alc887_fixup_asus_jack(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + if (action != HDA_FIXUP_ACT_PROBE) + return; + snd_hda_set_pin_ctl_cache(codec, 0x1b, PIN_HP); + spec->gen.hp_automute_hook = alc887_asus_hp_automute_hook; +} + static const struct hda_fixup alc882_fixups[] = { [ALC882_FIXUP_ABIT_AW9D_MAX] = { .type = HDA_FIXUP_PINS, @@ -2199,6 +2263,15 @@ static const struct hda_fixup alc882_fixups[] = { { } } }, + [ALC886_FIXUP_EAPD] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + /* change to EAPD mode */ + { 0x20, AC_VERB_SET_COEF_INDEX, 0x07 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0068 }, + { } + } + }, [ALC882_FIXUP_EAPD] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -2372,6 +2445,10 @@ 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, @@ -2389,6 +2466,28 @@ static const struct hda_fixup alc882_fixups[] = { .chained = true, .chain_id = ALC1220_FIXUP_CLEVO_PB51ED, }, + [ALC887_FIXUP_ASUS_AUDIO] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x15, 0x02a14150 }, /* use as headset mic, without its own jack detect */ + { 0x19, 0x22219420 }, + {} + }, + }, + [ALC887_FIXUP_ASUS_HMIC] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc887_fixup_asus_jack, + .chained = true, + .chain_id = ALC887_FIXUP_ASUS_AUDIO, + }, + [ALCS1200A_FIXUP_MIC_VREF] = { + .type = HDA_FIXUP_PINCTLS, + .v.pins = (const struct hda_pintbl[]) { + { 0x18, PIN_VREF50 }, /* rear mic */ + { 0x19, PIN_VREF50 }, /* front mic */ + {} + } + }, }; static const struct snd_pci_quirk alc882_fixup_tbl[] = { @@ -2407,13 +2506,13 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { ALC882_FIXUP_ACER_ASPIRE_8930G), SND_PCI_QUIRK(0x1025, 0x0146, "Acer Aspire 6935G", ALC882_FIXUP_ACER_ASPIRE_8930G), + SND_PCI_QUIRK(0x1025, 0x0142, "Acer Aspire 7730G", + ALC882_FIXUP_ACER_ASPIRE_4930G), + SND_PCI_QUIRK(0x1025, 0x0155, "Packard-Bell M5120", ALC882_FIXUP_PB_M5210), SND_PCI_QUIRK(0x1025, 0x015e, "Acer Aspire 6930G", ALC882_FIXUP_ACER_ASPIRE_4930G), SND_PCI_QUIRK(0x1025, 0x0166, "Acer Aspire 6530G", ALC882_FIXUP_ACER_ASPIRE_4930G), - SND_PCI_QUIRK(0x1025, 0x0142, "Acer Aspire 7730G", - ALC882_FIXUP_ACER_ASPIRE_4930G), - SND_PCI_QUIRK(0x1025, 0x0155, "Packard-Bell M5120", ALC882_FIXUP_PB_M5210), SND_PCI_QUIRK(0x1025, 0x021e, "Acer Aspire 5739G", ALC882_FIXUP_ACER_ASPIRE_4930G), SND_PCI_QUIRK(0x1025, 0x0259, "Acer Aspire 5935", ALC889_FIXUP_DAC_ROUTE), @@ -2422,14 +2521,16 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x13c2, "Asus A7M", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x1043, 0x1873, "ASUS W90V", ALC882_FIXUP_ASUS_W90V), SND_PCI_QUIRK(0x1043, 0x1971, "Asus W2JC", ALC882_FIXUP_ASUS_W2JC), + SND_PCI_QUIRK(0x1043, 0x2390, "Asus D700SA", ALC887_FIXUP_ASUS_HMIC), SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_FIXUP_EEE1601), SND_PCI_QUIRK(0x1043, 0x84bc, "ASUS ET2700", ALC887_FIXUP_ASUS_BASS), SND_PCI_QUIRK(0x1043, 0x8691, "ASUS ROG Ranger VIII", ALC882_FIXUP_GPIO3), + SND_PCI_QUIRK(0x1043, 0x8797, "ASUS TUF B550M-PLUS", ALCS1200A_FIXUP_MIC_VREF), + SND_PCI_QUIRK(0x104d, 0x9043, "Sony Vaio VGC-LN51JGB", ALC882_FIXUP_NO_PRIMARY_HP), + SND_PCI_QUIRK(0x104d, 0x9044, "Sony VAIO AiO", ALC882_FIXUP_NO_PRIMARY_HP), SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC889_FIXUP_VAIO_TT), SND_PCI_QUIRK(0x104d, 0x905a, "Sony Vaio Z", ALC882_FIXUP_NO_PRIMARY_HP), SND_PCI_QUIRK(0x104d, 0x9060, "Sony Vaio VPCL14M1R", ALC882_FIXUP_NO_PRIMARY_HP), - SND_PCI_QUIRK(0x104d, 0x9043, "Sony Vaio VGC-LN51JGB", ALC882_FIXUP_NO_PRIMARY_HP), - SND_PCI_QUIRK(0x104d, 0x9044, "Sony VAIO AiO", ALC882_FIXUP_NO_PRIMARY_HP), /* All Apple entries are in codec SSIDs */ SND_PCI_QUIRK(0x106b, 0x00a0, "MacBookPro 3,1", ALC889_FIXUP_MBP_VREF), @@ -2456,29 +2557,50 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x106b, 0x4a00, "Macbook 5,2", ALC889_FIXUP_MBA11_VREF), SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC882_FIXUP_EAPD), + SND_PCI_QUIRK(0x13fe, 0x1009, "Advantech MIT-W101", ALC886_FIXUP_EAPD), SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte EP45-DS3/Z87X-UD3H", ALC889_FIXUP_FRONT_HP_NO_PRESENCE), SND_PCI_QUIRK(0x1458, 0xa0b8, "Gigabyte AZ370-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS), - SND_PCI_QUIRK(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), SND_PCI_QUIRK(0x1462, 0x1275, "MSI-GL63", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1276, "MSI-GL73", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1293, "MSI-GP65", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD), + SND_PCI_QUIRK(0x1462, 0xcc34, "MSI Godlike X570", ALC1220_FIXUP_GB_DUAL_CODECS), SND_PCI_QUIRK(0x1462, 0xda57, "MSI Z270-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS), SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3), SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX), + SND_PCI_QUIRK(0x1558, 0x50d3, "Clevo PC50[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), + SND_PCI_QUIRK(0x1558, 0x65d1, "Clevo PB51[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), + SND_PCI_QUIRK(0x1558, 0x65d2, "Clevo PB51R[CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), + SND_PCI_QUIRK(0x1558, 0x65e1, "Clevo PB51[ED][DF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), + SND_PCI_QUIRK(0x1558, 0x65e5, "Clevo PC50D[PRS](?:-D|-G)?", ALC1220_FIXUP_CLEVO_PB51ED_PINS), + SND_PCI_QUIRK(0x1558, 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, 0x67f5, "Clevo PD70PN[NRT]", 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 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), SND_PCI_QUIRK(0x1558, 0x95e1, "Clevo P95xER", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1558, 0x95e2, "Clevo P950ER", ALC1220_FIXUP_CLEVO_P950), + SND_PCI_QUIRK(0x1558, 0x95e3, "Clevo P955[ER]T", ALC1220_FIXUP_CLEVO_P950), + SND_PCI_QUIRK(0x1558, 0x95e4, "Clevo P955ER", ALC1220_FIXUP_CLEVO_P950), + SND_PCI_QUIRK(0x1558, 0x95e5, "Clevo P955EE6", ALC1220_FIXUP_CLEVO_P950), + SND_PCI_QUIRK(0x1558, 0x95e6, "Clevo P950R[CDF]", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1558, 0x96e1, "Clevo P960[ER][CDFN]-K", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1558, 0x97e1, "Clevo P970[ER][CDFN]", ALC1220_FIXUP_CLEVO_P950), - SND_PCI_QUIRK(0x1558, 0x65d1, "Clevo PB51[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), - SND_PCI_QUIRK(0x1558, 0x67d1, "Clevo PB71[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), - SND_PCI_QUIRK(0x1558, 0x50d3, "Clevo PC50[ER][CDF]", 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, 0x97e2, "Clevo P970RC-M", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_FIXUP_EAPD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Y530", ALC882_FIXUP_LENOVO_Y530), @@ -2517,10 +2639,33 @@ 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"}, {} }; +static const struct snd_hda_pin_quirk alc882_pin_fixup_tbl[] = { + SND_HDA_PIN_QUIRK(0x10ec1220, 0x1043, "ASUS", ALC1220_FIXUP_CLEVO_P950, + {0x14, 0x01014010}, + {0x15, 0x01011012}, + {0x16, 0x01016011}, + {0x18, 0x01a19040}, + {0x19, 0x02a19050}, + {0x1a, 0x0181304f}, + {0x1b, 0x0221401f}, + {0x1e, 0x01456130}), + SND_HDA_PIN_QUIRK(0x10ec1220, 0x1462, "MS-7C35", ALC1220_FIXUP_CLEVO_P950, + {0x14, 0x01015010}, + {0x15, 0x01011012}, + {0x16, 0x01011011}, + {0x18, 0x01a11040}, + {0x19, 0x02a19050}, + {0x1a, 0x0181104f}, + {0x1b, 0x0221401f}, + {0x1e, 0x01451130}), + {} +}; + /* * BIOS auto configuration */ @@ -2562,6 +2707,7 @@ static int patch_alc882(struct hda_codec *codec) snd_hda_pick_fixup(codec, alc882_fixup_models, alc882_fixup_tbl, alc882_fixups); + snd_hda_pick_pin_fixup(codec, alc882_pin_fixup_tbl, alc882_fixups, true); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); alc_auto_parse_customize_define(codec); @@ -3040,6 +3186,7 @@ static void alc_disable_headset_jack_key(struct hda_codec *codec) case 0x10ec0215: case 0x10ec0225: case 0x10ec0285: + case 0x10ec0287: case 0x10ec0295: case 0x10ec0289: case 0x10ec0299: @@ -3047,8 +3194,10 @@ 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: + case 0x19e58326: alc_write_coef_idx(codec, 0x48, 0x0); alc_update_coef_idx(codec, 0x49, 0x0045, 0x0); break; @@ -3066,6 +3215,7 @@ static void alc_enable_headset_jack_key(struct hda_codec *codec) case 0x10ec0215: case 0x10ec0225: case 0x10ec0285: + case 0x10ec0287: case 0x10ec0295: case 0x10ec0289: case 0x10ec0299: @@ -3073,8 +3223,10 @@ 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: + case 0x19e58326: alc_write_coef_idx(codec, 0x48, 0xd011); alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); break; @@ -3421,7 +3573,8 @@ 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) @@ -3848,6 +4001,15 @@ static void alc271_fixup_dmic(struct hda_codec *codec, snd_hda_sequence_write(codec, verbs); } +/* Fix the speaker amp after resume, etc */ +static void alc269vb_fixup_aspire_e1_coef(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + if (action == HDA_FIXUP_ACT_INIT) + alc_update_coef_idx(codec, 0x0d, 0x6000, 0x6000); +} + static void alc269_fixup_pcm_44k(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -4135,6 +4297,12 @@ static void alc_fixup_hp_gpio_led(struct hda_codec *codec, } } +static void alc236_fixup_hp_gpio_led(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + alc_fixup_hp_gpio_led(codec, action, 0x02, 0x01); +} + static void alc269_fixup_hp_gpio_led(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -4524,8 +4692,10 @@ 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: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -4638,8 +4808,10 @@ 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: + case 0x19e58326: alc_write_coef_idx(codec, 0x45, 0xc489); snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); alc_process_coef_fw(codec, coef0256); @@ -4787,8 +4959,10 @@ 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: + case 0x19e58326: alc_write_coef_idx(codec, 0x1b, 0x0e4b); alc_write_coef_idx(codec, 0x45, 0xc089); msleep(50); @@ -4885,8 +5059,10 @@ 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: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -4998,8 +5174,10 @@ 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: + case 0x19e58326: alc_process_coef_fw(codec, coef0256); break; case 0x10ec0234: @@ -5093,8 +5271,10 @@ 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: + case 0x19e58326: alc_write_coef_idx(codec, 0x1b, 0x0e4b); alc_write_coef_idx(codec, 0x06, 0x6104); alc_write_coefex_idx(codec, 0x57, 0x3, 0x09a3); @@ -5123,7 +5303,7 @@ static void alc_determine_headset_type(struct hda_codec *codec) case 0x10ec0274: case 0x10ec0294: alc_process_coef_fw(codec, coef0274); - msleep(80); + msleep(850); val = alc_read_coef_idx(codec, 0x46); is_ctia = (val & 0x00f0) == 0x00f0; break; @@ -5307,6 +5487,7 @@ static void alc_update_headset_jack_cb(struct hda_codec *codec, struct hda_jack_callback *jack) { snd_hda_gen_hp_automute(codec, jack); + alc_update_headset_mode(codec); } static void alc_probe_headset_mode(struct hda_codec *codec) @@ -5385,8 +5566,10 @@ 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: + case 0x19e58326: alc_process_coef_fw(codec, alc256fw); break; } @@ -5524,6 +5707,18 @@ static void alc_fixup_tpt470_dacs(struct hda_codec *codec, spec->gen.preferred_dacs = preferred_pairs; } +static void alc295_fixup_asus_dacs(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + static const hda_nid_t preferred_pairs[] = { + 0x17, 0x02, 0x21, 0x03, 0 + }; + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) + spec->gen.preferred_dacs = preferred_pairs; +} + static void alc_shutup_dell_xps13(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -5937,6 +6132,21 @@ static void alc274_fixup_bind_dacs(struct hda_codec *codec, codec->power_save_node = 0; } +/* avoid DAC 0x06 for bass speaker 0x17; it has no volume control */ +static void alc289_fixup_asus_ga401(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + static const hda_nid_t preferred_pairs[] = { + 0x14, 0x02, 0x17, 0x02, 0x21, 0x03, 0 + }; + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->gen.preferred_dacs = preferred_pairs; + spec->gen.obey_preferred_dacs = 1; + } +} + /* The DAC of NID 0x3 will introduce click/pop noise on headphones, so invalidate it */ static void alc285_fixup_invalidate_dacs(struct hda_codec *codec, const struct hda_fixup *fix, int action) @@ -5947,6 +6157,29 @@ static void alc285_fixup_invalidate_dacs(struct hda_codec *codec, snd_hda_override_wcaps(codec, 0x03, 0); } +static void alc_combo_jack_hp_jd_restart(struct hda_codec *codec) +{ + switch (codec->core.vendor_id) { + case 0x10ec0274: + case 0x10ec0294: + case 0x10ec0225: + case 0x10ec0295: + case 0x10ec0299: + alc_update_coef_idx(codec, 0x4a, 0x8000, 1 << 15); /* Reset HP JD */ + alc_update_coef_idx(codec, 0x4a, 0x8000, 0 << 15); + break; + case 0x10ec0230: + case 0x10ec0235: + case 0x10ec0236: + case 0x10ec0255: + case 0x10ec0256: + case 0x19e58326: + alc_update_coef_idx(codec, 0x1b, 0x8000, 1 << 15); /* Reset HP JD */ + alc_update_coef_idx(codec, 0x1b, 0x8000, 0 << 15); + break; + } +} + static void alc295_fixup_chromebook(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -5957,16 +6190,7 @@ static void alc295_fixup_chromebook(struct hda_codec *codec, spec->ultra_low_power = true; break; case HDA_FIXUP_ACT_INIT: - switch (codec->core.vendor_id) { - case 0x10ec0295: - alc_update_coef_idx(codec, 0x4a, 0x8000, 1 << 15); /* Reset HP JD */ - alc_update_coef_idx(codec, 0x4a, 0x8000, 0 << 15); - break; - case 0x10ec0236: - alc_update_coef_idx(codec, 0x1b, 0x8000, 1 << 15); /* Reset HP JD */ - alc_update_coef_idx(codec, 0x1b, 0x8000, 0 << 15); - break; - } + alc_combo_jack_hp_jd_restart(codec); break; } } @@ -6012,6 +6236,35 @@ static void alc294_fixup_gx502_hp(struct hda_codec *codec, } } +static void alc294_gu502_toggle_output(struct hda_codec *codec, + struct hda_jack_callback *cb) +{ + /* Windows sets 0x10 to 0x8420 for Node 0x20 which is + * responsible from changes between speakers and headphones + */ + if (snd_hda_jack_detect_state(codec, 0x21) == HDA_JACK_PRESENT) + alc_write_coef_idx(codec, 0x10, 0x8420); + else + alc_write_coef_idx(codec, 0x10, 0x0a20); +} + +static void alc294_fixup_gu502_hp(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (!is_jack_detectable(codec, 0x21)) + return; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_jack_detect_enable_callback(codec, 0x21, + alc294_gu502_toggle_output); + break; + case HDA_FIXUP_ACT_INIT: + alc294_gu502_toggle_output(codec, NULL); + break; + } +} + static void alc285_fixup_hp_gpio_amp_init(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -6022,6 +6275,34 @@ static void alc285_fixup_hp_gpio_amp_init(struct hda_codec *codec, alc_write_coef_idx(codec, 0x65, 0x0); } +static void alc274_fixup_hp_headset_mic(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + switch (action) { + case HDA_FIXUP_ACT_INIT: + alc_combo_jack_hp_jd_restart(codec); + break; + } +} + +static void alc285_fixup_hp_spectre_x360(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + static const hda_nid_t conn[] = { 0x02 }; + static const struct hda_pintbl pincfgs[] = { + { 0x14, 0x90170110 }, /* rear speaker */ + { } + }; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_apply_pincfgs(codec, pincfgs); + /* force front speaker to DAC1 */ + snd_hda_override_conn_list(codec, 0x17, ARRAY_SIZE(conn), conn); + break; + } +} + /* for hda_fixup_thinkpad_acpi() */ #include "thinkpad_helper.c" @@ -6113,6 +6394,7 @@ enum { ALC283_FIXUP_HEADSET_MIC, ALC255_FIXUP_MIC_MUTE_LED, ALC282_FIXUP_ASPIRE_V5_PINS, + ALC269VB_FIXUP_ASPIRE_E1_COEF, ALC280_FIXUP_HP_GPIO4, ALC286_FIXUP_HP_GPIO_LED, ALC280_FIXUP_HP_GPIO2_MIC_HOTKEY, @@ -6142,6 +6424,7 @@ enum { ALC221_FIXUP_HP_FRONT_MIC, ALC292_FIXUP_TPT460, ALC298_FIXUP_SPK_VOLUME, + ALC298_FIXUP_LENOVO_SPK_VOLUME, ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER, ALC269_FIXUP_ATIV_BOOK_8, ALC221_FIXUP_HP_MIC_NO_PRESENCE, @@ -6196,10 +6479,15 @@ enum { ALC294_FIXUP_ASUS_GX502_HP, ALC294_FIXUP_ASUS_GX502_PINS, ALC294_FIXUP_ASUS_GX502_VERBS, + ALC294_FIXUP_ASUS_GU502_HP, + ALC294_FIXUP_ASUS_GU502_PINS, + ALC294_FIXUP_ASUS_GU502_VERBS, ALC285_FIXUP_HP_GPIO_LED, ALC285_FIXUP_HP_MUTE_LED, + ALC236_FIXUP_HP_GPIO_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, @@ -6214,7 +6502,17 @@ enum { ALC269_FIXUP_LEMOTE_A1802, ALC269_FIXUP_LEMOTE_A190X, ALC256_FIXUP_INTEL_NUC8_RUGGED, + ALC233_FIXUP_INTEL_NUC8_DMIC, + ALC233_FIXUP_INTEL_NUC8_BOOST, + ALC256_FIXUP_INTEL_NUC10, ALC255_FIXUP_XIAOMI_HEADSET_MIC, + ALC274_FIXUP_HP_MIC, + ALC274_FIXUP_HP_HEADSET_MIC, + ALC256_FIXUP_ASUS_HPE, + ALC285_FIXUP_THINKPAD_NO_BASS_SPK_HEADSET_JACK, + ALC295_FIXUP_ASUS_DACS, + ALC295_FIXUP_HP_OMEN, + ALC285_FIXUP_HP_SPECTRE_X360, }; static const struct hda_fixup alc269_fixups[] = { @@ -6777,6 +7075,10 @@ static const struct hda_fixup alc269_fixups[] = { { }, }, }, + [ALC269VB_FIXUP_ASPIRE_E1_COEF] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269vb_fixup_aspire_e1_coef, + }, [ALC280_FIXUP_HP_GPIO4] = { .type = HDA_FIXUP_FUNC, .v.func = alc280_fixup_hp_gpio4, @@ -6916,6 +7218,16 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc233_fixup_lenovo_line2_mic_hotkey, }, + [ALC233_FIXUP_INTEL_NUC8_DMIC] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_inv_dmic, + .chained = true, + .chain_id = ALC233_FIXUP_INTEL_NUC8_BOOST, + }, + [ALC233_FIXUP_INTEL_NUC8_BOOST] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_limit_int_mic_boost + }, [ALC255_FIXUP_DELL_SPK_NOISE] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_disable_aamix, @@ -6964,6 +7276,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC298_FIXUP_DELL_AIO_MIC_NO_PRESENCE, }, + [ALC298_FIXUP_LENOVO_SPK_VOLUME] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc298_fixup_speaker_volume, + }, [ALC295_FIXUP_DISABLE_DAC3] = { .type = HDA_FIXUP_FUNC, .v.func = alc295_fixup_disable_dac3, @@ -7395,6 +7711,35 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc294_fixup_gx502_hp, }, + [ALC294_FIXUP_ASUS_GU502_PINS] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x01a11050 }, /* rear HP mic */ + { 0x1a, 0x01a11830 }, /* rear external mic */ + { 0x21, 0x012110f0 }, /* rear HP out */ + { } + }, + .chained = true, + .chain_id = ALC294_FIXUP_ASUS_GU502_VERBS + }, + [ALC294_FIXUP_ASUS_GU502_VERBS] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + /* set 0x15 to HP-OUT ctrl */ + { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 }, + /* unmute the 0x15 amp */ + { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000 }, + /* set 0x1b to HP-OUT */ + { 0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, + { } + }, + .chained = true, + .chain_id = ALC294_FIXUP_ASUS_GU502_HP + }, + [ALC294_FIXUP_ASUS_GU502_HP] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc294_fixup_gu502_hp, + }, [ALC294_FIXUP_ASUS_COEF_1B] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -7405,6 +7750,8 @@ 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, @@ -7414,6 +7761,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc285_fixup_hp_mute_led, }, + [ALC236_FIXUP_HP_GPIO_LED] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc236_fixup_hp_gpio_led, + }, [ALC236_FIXUP_HP_MUTE_LED] = { .type = HDA_FIXUP_FUNC, .v.func = alc236_fixup_hp_mute_led, @@ -7425,6 +7776,14 @@ 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[]) { @@ -7464,11 +7823,10 @@ static const struct hda_fixup alc269_fixups[] = { .chain_id = ALC269_FIXUP_HEADSET_MIC }, [ALC289_FIXUP_ASUS_GA401] = { - .type = HDA_FIXUP_PINS, - .v.pins = (const struct hda_pintbl[]) { - { 0x19, 0x03a11020 }, /* headset mic with jack detect */ - { } - }, + .type = HDA_FIXUP_FUNC, + .v.func = alc289_fixup_asus_ga401, + .chained = true, + .chain_id = ALC289_FIXUP_ASUS_GA502, }, [ALC289_FIXUP_ASUS_GA502] = { .type = HDA_FIXUP_PINS, @@ -7584,6 +7942,15 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE }, + [ALC256_FIXUP_INTEL_NUC10] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE + }, [ALC255_FIXUP_XIAOMI_HEADSET_MIC] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -7592,7 +7959,66 @@ static const struct hda_fixup alc269_fixups[] = { { } }, .chained = true, - .chain_id = ALC289_FIXUP_ASUS_GA401 + .chain_id = ALC289_FIXUP_ASUS_GA502 + }, + [ALC274_FIXUP_HP_MIC] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + { 0x20, AC_VERB_SET_COEF_INDEX, 0x45 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x5089 }, + { } + }, + }, + [ALC274_FIXUP_HP_HEADSET_MIC] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc274_fixup_hp_headset_mic, + .chained = true, + .chain_id = ALC274_FIXUP_HP_MIC + }, + [ALC256_FIXUP_ASUS_HPE] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + /* Set EAPD high */ + { 0x20, AC_VERB_SET_COEF_INDEX, 0x0f }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x7778 }, + { } + }, + .chained = true, + .chain_id = ALC294_FIXUP_ASUS_HEADSET_MIC + }, + [ALC285_FIXUP_THINKPAD_NO_BASS_SPK_HEADSET_JACK] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_headset_jack, + .chained = true, + .chain_id = ALC269_FIXUP_THINKPAD_ACPI + }, + [ALC295_FIXUP_ASUS_DACS] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc295_fixup_asus_dacs, + }, + [ALC295_FIXUP_HP_OMEN] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x12, 0xb7a60130 }, + { 0x13, 0x40000000 }, + { 0x14, 0x411111f0 }, + { 0x16, 0x411111f0 }, + { 0x17, 0x90170110 }, + { 0x18, 0x411111f0 }, + { 0x19, 0x02a11030 }, + { 0x1a, 0x411111f0 }, + { 0x1b, 0x04a19030 }, + { 0x1d, 0x40600001 }, + { 0x1e, 0x411111f0 }, + { 0x21, 0x03211020 }, + {} + }, + .chained = true, + .chain_id = ALC269_FIXUP_HP_LINE1_MIC1_LED, + }, + [ALC285_FIXUP_HP_SPECTRE_X360] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_hp_spectre_x360, }, }; @@ -7602,17 +8028,21 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0349, "Acer AOD260", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x047c, "Acer AC700", ALC269_FIXUP_ACER_AC700), SND_PCI_QUIRK(0x1025, 0x072d, "Acer Aspire V5-571G", ALC269_FIXUP_ASPIRE_HEADSET_MIC), - SND_PCI_QUIRK(0x1025, 0x080d, "Acer Aspire V5-122P", ALC269_FIXUP_ASPIRE_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x0740, "Acer AO725", ALC271_FIXUP_HP_GATE_MIC_JACK), SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK), SND_PCI_QUIRK(0x1025, 0x0762, "Acer Aspire E1-472", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572), SND_PCI_QUIRK(0x1025, 0x0775, "Acer Aspire E1-572", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572), SND_PCI_QUIRK(0x1025, 0x079b, "Acer Aspire V5-573G", ALC282_FIXUP_ASPIRE_V5_PINS), + SND_PCI_QUIRK(0x1025, 0x080d, "Acer Aspire V5-122P", ALC269_FIXUP_ASPIRE_HEADSET_MIC), + SND_PCI_QUIRK(0x1025, 0x0840, "Acer Aspire E1", ALC269VB_FIXUP_ASPIRE_E1_COEF), + SND_PCI_QUIRK(0x1025, 0x101c, "Acer Veriton N2510G", ALC269_FIXUP_LIFEBOOK), SND_PCI_QUIRK(0x1025, 0x102b, "Acer Aspire C24-860", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x1065, "Acer Aspire C20-820", ALC269VC_FIXUP_ACER_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x106d, "Acer Cloudbook 14", ALC283_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0x1025, 0x1099, "Acer Aspire E5-523G", ALC255_FIXUP_ACER_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x110e, "Acer Aspire ES1-432", ALC255_FIXUP_ACER_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1025, 0x1166, "Acer Veriton N4640G", ALC269_FIXUP_LIFEBOOK), + SND_PCI_QUIRK(0x1025, 0x1167, "Acer Veriton N6640G", ALC269_FIXUP_LIFEBOOK), SND_PCI_QUIRK(0x1025, 0x1246, "Acer Predator Helios 500", ALC299_FIXUP_PREDATOR_SPK), SND_PCI_QUIRK(0x1025, 0x1247, "Acer vCopperbox", ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS), SND_PCI_QUIRK(0x1025, 0x1248, "Acer Veriton N4660G", ALC269VC_FIXUP_ACER_MIC_NO_PRESENCE), @@ -7622,6 +8052,8 @@ 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), @@ -7656,8 +8088,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0738, "Dell Precision 5820", ALC269_FIXUP_NO_SHUTUP), SND_PCI_QUIRK(0x1028, 0x075c, "Dell XPS 27 7760", ALC298_FIXUP_SPK_VOLUME), SND_PCI_QUIRK(0x1028, 0x075d, "Dell AIO", ALC298_FIXUP_SPK_VOLUME), - SND_PCI_QUIRK(0x1028, 0x07b0, "Dell Precision 7520", ALC295_FIXUP_DISABLE_DAC3), SND_PCI_QUIRK(0x1028, 0x0798, "Dell Inspiron 17 7000 Gaming", ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER), + SND_PCI_QUIRK(0x1028, 0x07b0, "Dell Precision 7520", ALC295_FIXUP_DISABLE_DAC3), SND_PCI_QUIRK(0x1028, 0x080c, "Dell WYSE", ALC225_FIXUP_DELL_WYSE_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x084b, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB), SND_PCI_QUIRK(0x1028, 0x084e, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB), @@ -7667,8 +8099,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x08ad, "Dell WYSE AIO", ALC225_FIXUP_DELL_WYSE_AIO_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x08ae, "Dell WYSE NB", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0935, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB), - SND_PCI_QUIRK(0x1028, 0x097e, "Dell Precision", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x097d, "Dell Precision", ALC289_FIXUP_DUAL_SPK), + SND_PCI_QUIRK(0x1028, 0x097e, "Dell Precision", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x098d, "Dell Precision", ALC233_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x09bf, "Dell Precision", ALC233_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), @@ -7676,35 +8108,18 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), SND_PCI_QUIRK(0x103c, 0x18e6, "HP", ALC269_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x218b, "HP", ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED), - SND_PCI_QUIRK(0x103c, 0x225f, "HP", ALC280_FIXUP_HP_GPIO2_MIC_HOTKEY), - /* ALC282 */ SND_PCI_QUIRK(0x103c, 0x21f9, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2210, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2214, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x221b, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x221c, "HP EliteBook 755 G2", ALC280_FIXUP_HP_HEADSET_MIC), + SND_PCI_QUIRK(0x103c, 0x2221, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x2225, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2236, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2237, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2238, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2239, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x224b, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED), - SND_PCI_QUIRK(0x103c, 0x2268, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), - SND_PCI_QUIRK(0x103c, 0x226a, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), - SND_PCI_QUIRK(0x103c, 0x226b, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), - SND_PCI_QUIRK(0x103c, 0x226e, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), - SND_PCI_QUIRK(0x103c, 0x2271, "HP", ALC286_FIXUP_HP_GPIO_LED), - SND_PCI_QUIRK(0x103c, 0x2272, "HP", ALC280_FIXUP_HP_DOCK_PINS), - SND_PCI_QUIRK(0x103c, 0x2273, "HP", ALC280_FIXUP_HP_DOCK_PINS), - SND_PCI_QUIRK(0x103c, 0x229e, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), - SND_PCI_QUIRK(0x103c, 0x22b2, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), - SND_PCI_QUIRK(0x103c, 0x22b7, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), - SND_PCI_QUIRK(0x103c, 0x22bf, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), - SND_PCI_QUIRK(0x103c, 0x22cf, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), - SND_PCI_QUIRK(0x103c, 0x22db, "HP", ALC280_FIXUP_HP_9480M), - SND_PCI_QUIRK(0x103c, 0x22dc, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), - SND_PCI_QUIRK(0x103c, 0x22fb, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), - /* ALC290 */ - SND_PCI_QUIRK(0x103c, 0x221b, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), - SND_PCI_QUIRK(0x103c, 0x2221, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), - SND_PCI_QUIRK(0x103c, 0x2225, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2253, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2254, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2255, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), @@ -7712,41 +8127,67 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2257, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2259, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x225a, "HP", ALC269_FIXUP_HP_DOCK_GPIO_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x225f, "HP", ALC280_FIXUP_HP_GPIO2_MIC_HOTKEY), SND_PCI_QUIRK(0x103c, 0x2260, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2263, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2264, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2265, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x2268, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x226a, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x226b, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x226e, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x2271, "HP", ALC286_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x2272, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x2272, "HP", ALC280_FIXUP_HP_DOCK_PINS), SND_PCI_QUIRK(0x103c, 0x2273, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x2273, "HP", ALC280_FIXUP_HP_DOCK_PINS), SND_PCI_QUIRK(0x103c, 0x2278, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x227f, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2282, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x228b, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x228e, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x229e, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22b2, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22b7, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22bf, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22c4, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x22c5, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x22c7, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x22c8, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), - SND_PCI_QUIRK(0x103c, 0x22c4, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22cf, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22db, "HP", ALC280_FIXUP_HP_9480M), + SND_PCI_QUIRK(0x103c, 0x22dc, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), + SND_PCI_QUIRK(0x103c, 0x22fb, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2334, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2335, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2336, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2337, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), - SND_PCI_QUIRK(0x103c, 0x221c, "HP EliteBook 755 G2", ALC280_FIXUP_HP_HEADSET_MIC), SND_PCI_QUIRK(0x103c, 0x802e, "HP Z240 SFF", ALC221_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x802f, "HP Z240", ALC221_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x820d, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x8256, "HP", ALC221_FIXUP_HP_FRONT_MIC), SND_PCI_QUIRK(0x103c, 0x827e, "HP x360", ALC295_FIXUP_HP_X360), + SND_PCI_QUIRK(0x103c, 0x827f, "HP x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x82bf, "HP G3 mini", ALC221_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x82c0, "HP G3 mini premium", ALC221_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x83b9, "HP Spectre x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), + SND_PCI_QUIRK(0x103c, 0x841c, "HP Pavilion 15-CK0xx", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x8497, "HP Envy x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), + SND_PCI_QUIRK(0x103c, 0x84da, "HP OMEN dc0019-ur", ALC295_FIXUP_HP_OMEN), SND_PCI_QUIRK(0x103c, 0x84e7, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3), + SND_PCI_QUIRK(0x103c, 0x8519, "HP Spectre x360 15-df0xxx", ALC285_FIXUP_HP_SPECTRE_X360), + SND_PCI_QUIRK(0x103c, 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), SND_PCI_QUIRK(0x103c, 0x877a, "HP", ALC285_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x877d, "HP", ALC236_FIXUP_HP_MUTE_LED), + SND_PCI_QUIRK(0x103c, 0x87e5, "HP ProBook 440 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x89aa, "HP EliteBook 630 G9", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), @@ -7755,18 +8196,25 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x10d0, "ASUS X540LA/X540LJ", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x115d, "Asus 1015E", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x11c0, "ASUS X556UR", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1043, 0x125e, "ASUS Q524UQK", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1043, 0x1271, "ASUS X430UN", ALC256_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1290, "ASUS X441SA", ALC233_FIXUP_EAPD_COEF_AND_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x12a0, "ASUS X441UV", ALC233_FIXUP_EAPD_COEF_AND_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1043, 0x12f0, "ASUS X541UV", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x12e0, "ASUS X541SA", ALC256_FIXUP_ASUS_MIC), + SND_PCI_QUIRK(0x1043, 0x12f0, "ASUS X541UV", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x13b0, "ASUS Z550SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_ASUS_ZENBOOK), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A), SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), + SND_PCI_QUIRK(0x1043, 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), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), @@ -7776,46 +8224,109 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1bbd, "ASUS Z550MA", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1c23, "Asus X55U", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x1ccd, "ASUS X555UB", ALC256_FIXUP_ASUS_MIC), + SND_PCI_QUIRK(0x1043, 0x1d4e, "ASUS TM420", ALC256_FIXUP_ASUS_HPE), SND_PCI_QUIRK(0x1043, 0x1e11, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA502), + SND_PCI_QUIRK(0x1043, 0x1e51, "ASUS Zephyrus M15", ALC294_FIXUP_ASUS_GU502_PINS), + SND_PCI_QUIRK(0x1043, 0x1e8e, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_GA401), - SND_PCI_QUIRK(0x1043, 0x1881, "ASUS Zephyrus S/M", ALC294_FIXUP_ASUS_GX502_PINS), + 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), SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x8516, "ASUS X101CH", ALC269_FIXUP_ASUS_X101), - SND_PCI_QUIRK(0x104d, 0x90b5, "Sony VAIO Pro 11", ALC286_FIXUP_SONY_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x104d, 0x90b6, "Sony VAIO Pro 13", ALC286_FIXUP_SONY_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x104d, 0x9073, "Sony VAIO", ALC275_FIXUP_SONY_VAIO_GPIO2), SND_PCI_QUIRK(0x104d, 0x907b, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), SND_PCI_QUIRK(0x104d, 0x9099, "Sony VAIO S13", ALC275_FIXUP_SONY_DISABLE_AAMIX), + SND_PCI_QUIRK(0x104d, 0x90b5, "Sony VAIO Pro 11", ALC286_FIXUP_SONY_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x104d, 0x90b6, "Sony VAIO Pro 13", ALC286_FIXUP_SONY_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook", ALC269_FIXUP_LIFEBOOK), SND_PCI_QUIRK(0x10cf, 0x159f, "Lifebook E780", ALC269_FIXUP_LIFEBOOK_NO_HP_TO_LINEOUT), SND_PCI_QUIRK(0x10cf, 0x15dc, "Lifebook T731", ALC269_FIXUP_LIFEBOOK_HP_PIN), - SND_PCI_QUIRK(0x10cf, 0x1757, "Lifebook E752", ALC269_FIXUP_LIFEBOOK_HP_PIN), SND_PCI_QUIRK(0x10cf, 0x1629, "Lifebook U7x7", ALC255_FIXUP_LIFEBOOK_U7x7_HEADSET_MIC), + SND_PCI_QUIRK(0x10cf, 0x1757, "Lifebook E752", ALC269_FIXUP_LIFEBOOK_HP_PIN), SND_PCI_QUIRK(0x10cf, 0x1845, "Lifebook U904", ALC269_FIXUP_LIFEBOOK_EXTMIC), SND_PCI_QUIRK(0x10ec, 0x10f2, "Intel Reference board", ALC700_FIXUP_INTEL_REFERENCE), + SND_PCI_QUIRK(0x10ec, 0x118c, "Medion EE4254 MD62100", ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE), SND_PCI_QUIRK(0x10ec, 0x1230, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), + SND_PCI_QUIRK(0x10ec, 0x1252, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), + SND_PCI_QUIRK(0x10ec, 0x1254, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0x10f7, 0x8338, "Panasonic CF-SZ6", ALC269_FIXUP_HEADSET_MODE), SND_PCI_QUIRK(0x144d, 0xc109, "Samsung Ativ book 9 (NP900X3G)", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x144d, 0xc176, "Samsung Notebook 9 Pro (NP930MBE-K04US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x144d, 0xc189, "Samsung Galaxy Flex Book (NT950QCG-X716)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x144d, 0xc18a, "Samsung Galaxy Book Ion (NP930XCJ-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), - SND_PCI_QUIRK(0x144d, 0xc830, "Samsung Galaxy Book Ion (NT950XCJ-X716A)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x144d, 0xc740, "Samsung Ativ book 8 (NP870Z5G)", ALC269_FIXUP_ATIV_BOOK_8), SND_PCI_QUIRK(0x144d, 0xc812, "Samsung Notebook Pen S (NT950SBE-X58)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), + SND_PCI_QUIRK(0x144d, 0xc830, "Samsung Galaxy Book Ion (NT950XCJ-X716A)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), + SND_PCI_QUIRK(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), + SND_PCI_QUIRK(0x152d, 0x1082, "Quanta NL3", ALC269_FIXUP_LIFEBOOK), + SND_PCI_QUIRK(0x1558, 0x1323, "Clevo N130ZU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x1325, "System76 Darter Pro (darp5)", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x1401, "Clevo L140[CZ]U", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x1403, "Clevo N140CU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x1404, "Clevo N150CU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x14a1, "Clevo L141MU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x4018, "Clevo NV40M[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x4019, "Clevo NV40MZ", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x4020, "Clevo NV40MB", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x40a1, "Clevo NL40GU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x40c1, "Clevo NL40[CZ]U", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x40d1, "Clevo NL41DU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x50a3, "Clevo NJ51GU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x50b3, "Clevo NK50S[BEZ]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x50b6, "Clevo NK50S5", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x50b8, "Clevo NK50SZ", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x50d5, "Clevo NP50D5", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x50f0, "Clevo NH50A[CDF]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x50f2, "Clevo NH50E[PR]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x50f3, "Clevo NH58DPQ", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x50f5, "Clevo NH55EPY", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x50f6, "Clevo NH55DPQ", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x5101, "Clevo S510WU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x5157, "Clevo W517GU1", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x51a1, "Clevo NS50MU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x70a1, "Clevo NB70T[HJK]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x70b3, "Clevo NK70SB", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x70f2, "Clevo NH79EPY", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x70f3, "Clevo NH77DPQ", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x70f4, "Clevo NH77EPY", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x70f6, "Clevo NH77DPQ-Y", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x8228, "Clevo NR40BU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x8520, "Clevo NH50D[CD]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x8521, "Clevo NH77D[CD]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x8535, "Clevo NH50D[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x8536, "Clevo NH79D[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8550, "System76 Gazelle (gaze14)", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8551, "System76 Gazelle (gaze14)", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8560, "System76 Gazelle (gaze14)", ALC269_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1558, 0x8561, "System76 Gazelle (gaze14)", ALC269_FIXUP_HEADSET_MIC), + SND_PCI_QUIRK(0x1558, 0x8562, "Clevo NH[5|7][0-9]RZ[Q]", ALC269_FIXUP_DMIC), + SND_PCI_QUIRK(0x1558, 0x8668, "Clevo NP50B[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x8680, "Clevo NJ50LU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x8686, "Clevo NH50[CZ]U", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x8a20, "Clevo NH55DCQ-Y", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x8a51, "Clevo NH70RCQ-Y", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x8d50, "Clevo NH55RCQ-M", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x951d, "Clevo N950T[CDF]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x9600, "Clevo N960K[PR]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x961d, "Clevo N960S[CDF]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x971d, "Clevo N970T[CDF]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0xa500, "Clevo NL53RU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0xa600, "Clevo NL5XNU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0xb018, "Clevo NP50D[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0xb019, "Clevo NH77D[BE]Q", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0xb022, "Clevo NH77D[DC][QW]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0xc018, "Clevo NP50D[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0xc019, "Clevo NH77D[BE]Q", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0xc022, "Clevo NH77[DC][QW]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x17aa, 0x1036, "Lenovo P520", ALC233_FIXUP_LENOVO_MULTI_CODECS), SND_PCI_QUIRK(0x17aa, 0x1048, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE), @@ -7823,9 +8334,9 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x21ca, "Thinkpad L412", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x21e9, "Thinkpad Edge 15", ALC269_FIXUP_SKU_IGNORE), + SND_PCI_QUIRK(0x17aa, 0x21f3, "Thinkpad T430", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x21f6, "Thinkpad T530", ALC269_FIXUP_LENOVO_DOCK_LIMIT_BOOST), SND_PCI_QUIRK(0x17aa, 0x21fa, "Thinkpad X230", ALC269_FIXUP_LENOVO_DOCK), - SND_PCI_QUIRK(0x17aa, 0x21f3, "Thinkpad T430", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2208, "Thinkpad T431s", ALC269_FIXUP_LENOVO_DOCK), @@ -7853,6 +8364,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x225d, "Thinkpad T480", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x2292, "Thinkpad X1 Carbon 7th", ALC285_FIXUP_THINKPAD_HEADSET_JACK), SND_PCI_QUIRK(0x17aa, 0x22be, "Thinkpad X1 Carbon 8th", ALC285_FIXUP_THINKPAD_HEADSET_JACK), + SND_PCI_QUIRK(0x17aa, 0x22c1, "Thinkpad P1 Gen 3", ALC285_FIXUP_THINKPAD_NO_BASS_SPK_HEADSET_JACK), + SND_PCI_QUIRK(0x17aa, 0x22c2, "Thinkpad X1 Extreme Gen 3", ALC285_FIXUP_THINKPAD_NO_BASS_SPK_HEADSET_JACK), SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x310c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), @@ -7863,9 +8376,11 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3151, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x17aa, 0x3176, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x17aa, 0x3178, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), + SND_PCI_QUIRK(0x17aa, 0x3818, "Lenovo C940", ALC298_FIXUP_LENOVO_SPK_VOLUME), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI), + SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x501a, "Thinkpad", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x501e, "Thinkpad L440", ALC292_FIXUP_TPT440_DOCK), @@ -7884,19 +8399,21 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x511e, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x511f, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), - SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MACH-WX9", ALC256_FIXUP_HUAWEI_MACH_WX9_PINS), SND_PCI_QUIRK(0x1b35, 0x1235, "CZC B20", ALC269_FIXUP_CZC_B20), SND_PCI_QUIRK(0x1b35, 0x1236, "CZC TMI", ALC269_FIXUP_CZC_TMI), SND_PCI_QUIRK(0x1b35, 0x1237, "CZC L101", ALC269_FIXUP_CZC_L101), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ - SND_PCI_QUIRK(0x1d72, 0x1602, "RedmiBook", ALC255_FIXUP_XIAOMI_HEADSET_MIC), - SND_PCI_QUIRK(0x1d72, 0x1901, "RedmiBook 14", ALC256_FIXUP_ASUS_HEADSET_MIC), - SND_PCI_QUIRK(0x10ec, 0x118c, "Medion EE4254 MD62100", ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE), SND_PCI_QUIRK(0x1c06, 0x2013, "Lemote A1802", ALC269_FIXUP_LEMOTE_A1802), SND_PCI_QUIRK(0x1c06, 0x2015, "Lemote A190X", ALC269_FIXUP_LEMOTE_A190X), + SND_PCI_QUIRK(0x1d72, 0x1602, "RedmiBook", ALC255_FIXUP_XIAOMI_HEADSET_MIC), + SND_PCI_QUIRK(0x1d72, 0x1701, "XiaomiNotebook Pro", ALC298_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1d72, 0x1901, "RedmiBook 14", ALC256_FIXUP_ASUS_HEADSET_MIC), + SND_PCI_QUIRK(0x1d72, 0x1947, "RedmiBook Air", ALC255_FIXUP_XIAOMI_HEADSET_MIC), + SND_PCI_QUIRK(0x8086, 0x2074, "Intel NUC 8", ALC233_FIXUP_INTEL_NUC8_DMIC), SND_PCI_QUIRK(0x8086, 0x2080, "Intel NUC 8 Rugged", ALC256_FIXUP_INTEL_NUC8_RUGGED), + SND_PCI_QUIRK(0x8086, 0x2081, "Intel NUC 10", ALC256_FIXUP_INTEL_NUC10), #if 0 /* Below is a quirk table taken from the old code. @@ -8026,6 +8543,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC283_FIXUP_HEADSET_MIC, .name = "alc283-headset"}, {.id = ALC255_FIXUP_MIC_MUTE_LED, .name = "alc255-dell-mute"}, {.id = ALC282_FIXUP_ASPIRE_V5_PINS, .name = "aspire-v5"}, + {.id = ALC269VB_FIXUP_ASPIRE_E1_COEF, .name = "aspire-e1-coef"}, {.id = ALC280_FIXUP_HP_GPIO4, .name = "hp-gpio4"}, {.id = ALC286_FIXUP_HP_GPIO_LED, .name = "hp-gpio-led"}, {.id = ALC280_FIXUP_HP_GPIO2_MIC_HOTKEY, .name = "hp-gpio2-hotkey"}, @@ -8069,7 +8587,11 @@ 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"}, + {.id = ALC285_FIXUP_HP_SPECTRE_X360, .name = "alc285-hp-spectre-x360"}, {} }; #define ALC225_STANDARD_PINS \ @@ -8298,6 +8820,10 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x1a, 0x90a70130}, {0x1b, 0x90170110}, {0x21, 0x03211020}), + SND_HDA_PIN_QUIRK(0x10ec0274, 0x103c, "HP", ALC274_FIXUP_HP_HEADSET_MIC, + {0x17, 0x90170110}, + {0x19, 0x03a11030}, + {0x21, 0x03211020}), SND_HDA_PIN_QUIRK(0x10ec0280, 0x103c, "HP", ALC280_FIXUP_HP_GPIO4, {0x12, 0x90a60130}, {0x14, 0x90170110}, @@ -8353,6 +8879,20 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x14, 0x90170110}, {0x19, 0x04a11040}, {0x21, 0x04211020}), + SND_HDA_PIN_QUIRK(0x10ec0285, 0x17aa, "Lenovo", ALC285_FIXUP_LENOVO_PC_BEEP_IN_NOISE, + {0x14, 0x90170110}, + {0x19, 0x04a11040}, + {0x1d, 0x40600001}, + {0x21, 0x04211020}), + SND_HDA_PIN_QUIRK(0x10ec0285, 0x17aa, "Lenovo", ALC285_FIXUP_THINKPAD_NO_BASS_SPK_HEADSET_JACK, + {0x14, 0x90170110}, + {0x19, 0x04a11040}, + {0x21, 0x04211020}), + SND_HDA_PIN_QUIRK(0x10ec0287, 0x17aa, "Lenovo", ALC285_FIXUP_THINKPAD_HEADSET_JACK, + {0x14, 0x90170110}, + {0x17, 0x90170111}, + {0x19, 0x03a11030}, + {0x21, 0x03211020}), SND_HDA_PIN_QUIRK(0x10ec0286, 0x1025, "Acer", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE, {0x12, 0x90a60130}, {0x17, 0x90170110}, @@ -8416,6 +8956,9 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { SND_HDA_PIN_QUIRK(0x10ec0293, 0x1028, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE, ALC292_STANDARD_PINS, {0x13, 0x90a60140}), + SND_HDA_PIN_QUIRK(0x10ec0294, 0x1043, "ASUS", ALC294_FIXUP_ASUS_HPE, + {0x17, 0x90170110}, + {0x21, 0x04211020}), SND_HDA_PIN_QUIRK(0x10ec0294, 0x1043, "ASUS", ALC294_FIXUP_ASUS_MIC, {0x14, 0x90170110}, {0x1b, 0x90a70130}, @@ -8640,8 +9183,10 @@ static int patch_alc269(struct hda_codec *codec) spec->shutup = alc256_shutup; spec->init_hook = alc256_init; break; + case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: + case 0x19e58326: spec->codec_variant = ALC269_TYPE_ALC256; spec->shutup = alc256_shutup; spec->init_hook = alc256_init; @@ -8707,6 +9252,16 @@ 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, @@ -8824,8 +9379,7 @@ static const struct snd_pci_quirk alc861_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1393, "ASUS A6Rp", ALC861_FIXUP_ASUS_A6RP), SND_PCI_QUIRK_VENDOR(0x1043, "ASUS laptop", ALC861_FIXUP_AMP_VREF_0F), SND_PCI_QUIRK(0x1462, 0x7254, "HP DX2200", ALC861_FIXUP_NO_JACK_DETECT), - SND_PCI_QUIRK(0x1584, 0x2b01, "Haier W18", ALC861_FIXUP_AMP_VREF_0F), - SND_PCI_QUIRK(0x1584, 0x0000, "Uniwill ECS M31EI", ALC861_FIXUP_AMP_VREF_0F), + SND_PCI_QUIRK_VENDOR(0x1584, "Haier/Uniwill", ALC861_FIXUP_AMP_VREF_0F), SND_PCI_QUIRK(0x1734, 0x10c7, "FSC Amilo Pi1505", ALC861_FIXUP_FSC_AMILO_PI1505), {} }; @@ -9149,6 +9703,27 @@ 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), @@ -9226,6 +9801,12 @@ 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, + ALC897_FIXUP_HP_HSMIC_VERB, }; static const struct hda_fixup alc662_fixups[] = { @@ -9609,6 +10190,49 @@ 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 + }, + [ALC897_FIXUP_HP_HSMIC_VERB] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { } + }, + }, }; static const struct snd_pci_quirk alc662_fixup_tbl[] = { @@ -9620,6 +10244,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0349, "eMachines eM250", ALC662_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x034a, "Gateway LT27", ALC662_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE), + SND_PCI_QUIRK(0x1025, 0x0566, "Acer Aspire Ethos 8951G", ALC669_FIXUP_ACER_ASPIRE_ETHOS), SND_PCI_QUIRK(0x1025, 0x123c, "Acer Nitro N50-600", ALC662_FIXUP_ACER_NITRO_HEADSET_MODE), SND_PCI_QUIRK(0x1025, 0x124e, "Acer 2660G", ALC662_FIXUP_ACER_X2660G_HEADSET_MODE), SND_PCI_QUIRK(0x1028, 0x05d8, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), @@ -9633,15 +10258,19 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0698, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x069f, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), + SND_PCI_QUIRK(0x103c, 0x8719, "HP", ALC897_FIXUP_HP_HSMIC_VERB), + 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, 0x13df, "Asus N550JX", ALC662_FIXUP_BASS_1A), SND_PCI_QUIRK(0x1043, 0x129d, "Asus N750", ALC662_FIXUP_ASUS_Nx50), SND_PCI_QUIRK(0x1043, 0x12ff, "ASUS G751", ALC668_FIXUP_ASUS_G751), + SND_PCI_QUIRK(0x1043, 0x13df, "Asus N550JX", ALC662_FIXUP_BASS_1A), SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_BASS_MODE4_CHMAP), SND_PCI_QUIRK(0x1043, 0x15a7, "ASUS UX51VZH", ALC662_FIXUP_BASS_16), SND_PCI_QUIRK(0x1043, 0x177d, "ASUS N551", ALC668_FIXUP_ASUS_Nx51), 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), @@ -9650,6 +10279,11 @@ 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, 0x1057, "Lenovo P360", ALC897_FIXUP_HEADSET_MIC_PIN), + 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), @@ -9657,7 +10291,6 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1b0a, 0x01b8, "ACER Veriton", ALC662_FIXUP_ACER_VERITON), SND_PCI_QUIRK(0x1b35, 0x1234, "CZC ET26", ALC662_FIXUP_CZC_ET26), SND_PCI_QUIRK(0x1b35, 0x2206, "CZC P10T", ALC662_FIXUP_CZC_P10T), - SND_PCI_QUIRK(0x1025, 0x0566, "Acer Aspire Ethos 8951G", ALC669_FIXUP_ACER_ASPIRE_ETHOS), #if 0 /* Below is a quirk table taken from the old code. @@ -9931,6 +10564,7 @@ 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), @@ -9998,11 +10632,13 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0888, "ALC888", patch_alc882), HDA_CODEC_ENTRY(0x10ec0889, "ALC889", patch_alc882), HDA_CODEC_ENTRY(0x10ec0892, "ALC892", patch_alc662), + HDA_CODEC_ENTRY(0x10ec0897, "ALC897", patch_alc662), HDA_CODEC_ENTRY(0x10ec0899, "ALC898", patch_alc882), HDA_CODEC_ENTRY(0x10ec0900, "ALC1150", patch_alc882), HDA_CODEC_ENTRY(0x10ec0b00, "ALCS1200A", patch_alc882), HDA_CODEC_ENTRY(0x10ec1168, "ALC1220", patch_alc882), HDA_CODEC_ENTRY(0x10ec1220, "ALC1220", patch_alc882), + HDA_CODEC_ENTRY(0x19e58326, "HW8326", patch_alc269), {} /* terminator */ }; MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_realtek); diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 7ef8f3105cdb..3edb4e25797d 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -113,6 +113,7 @@ static struct via_spec *via_new_spec(struct hda_codec *codec) spec->codec_type = VT1708S; spec->gen.indep_hp = 1; spec->gen.keep_eapd_on = 1; + spec->gen.dac_min_mute = 1; spec->gen.pcm_playback_hook = via_playback_pcm_hook; spec->gen.add_stereo_mix_input = HDA_HINT_STEREO_MIX_AUTO; codec->power_save_node = 1; @@ -519,11 +520,11 @@ static int via_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; - err = snd_hda_gen_parse_auto_config(codec, &spec->gen.autocfg); + err = auto_parse_beep(codec); if (err < 0) return err; - err = auto_parse_beep(codec); + err = snd_hda_gen_parse_auto_config(codec, &spec->gen.autocfg); if (err < 0) return err; @@ -1002,6 +1003,7 @@ static const struct hda_verb vt1802_init_verbs[] = { enum { VIA_FIXUP_INTMIC_BOOST, VIA_FIXUP_ASUS_G75, + VIA_FIXUP_POWER_SAVE, }; static void via_fixup_intmic_boost(struct hda_codec *codec, @@ -1011,6 +1013,13 @@ static void via_fixup_intmic_boost(struct hda_codec *codec, override_mic_boost(codec, 0x30, 0, 2, 40); } +static void via_fixup_power_save(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) + codec->power_save_node = 0; +} + static const struct hda_fixup via_fixups[] = { [VIA_FIXUP_INTMIC_BOOST] = { .type = HDA_FIXUP_FUNC, @@ -1025,11 +1034,16 @@ static const struct hda_fixup via_fixups[] = { { } } }, + [VIA_FIXUP_POWER_SAVE] = { + .type = HDA_FIXUP_FUNC, + .v.func = via_fixup_power_save, + }, }; static const struct snd_pci_quirk vt2002p_fixups[] = { SND_PCI_QUIRK(0x1043, 0x1487, "Asus G75", VIA_FIXUP_ASUS_G75), SND_PCI_QUIRK(0x1043, 0x8532, "Asus X202E", VIA_FIXUP_INTMIC_BOOST), + SND_PCI_QUIRK_VENDOR(0x1558, "Clevo", VIA_FIXUP_POWER_SAVE), {} }; diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 6ff94d8ad86e..5150e8d38975 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -354,6 +354,7 @@ struct ichdev { unsigned int ali_slot; /* ALI DMA slot */ struct ac97_pcm *pcm; int pcm_open_flag; + unsigned int prepared:1; unsigned int suspended: 1; }; @@ -714,6 +715,9 @@ static inline void snd_intel8x0_update(struct intel8x0 *chip, struct ichdev *ich int status, civ, i, step; int ack = 0; + if (!(ichdev->prepared || chip->in_measurement) || ichdev->suspended) + return; + spin_lock_irqsave(&chip->reg_lock, flags); status = igetbyte(chip, port + ichdev->roff_sr); civ = igetbyte(chip, port + ICH_REG_OFF_CIV); @@ -907,6 +911,7 @@ static int snd_intel8x0_hw_params(struct snd_pcm_substream *substream, if (ichdev->pcm_open_flag) { snd_ac97_pcm_close(ichdev->pcm); ichdev->pcm_open_flag = 0; + ichdev->prepared = 0; } err = snd_ac97_pcm_open(ichdev->pcm, params_rate(hw_params), params_channels(hw_params), @@ -928,6 +933,7 @@ static int snd_intel8x0_hw_free(struct snd_pcm_substream *substream) if (ichdev->pcm_open_flag) { snd_ac97_pcm_close(ichdev->pcm); ichdev->pcm_open_flag = 0; + ichdev->prepared = 0; } return snd_pcm_lib_free_pages(substream); } @@ -1002,6 +1008,7 @@ static int snd_intel8x0_pcm_prepare(struct snd_pcm_substream *substream) ichdev->pos_shift = (runtime->sample_bits > 16) ? 2 : 1; } snd_intel8x0_setup_periods(chip, ichdev); + ichdev->prepared = 1; return 0; } diff --git a/sound/pci/mixart/mixart_core.c b/sound/pci/mixart/mixart_core.c index 048a2660d18d..363b26e85af5 100644 --- a/sound/pci/mixart/mixart_core.c +++ b/sound/pci/mixart/mixart_core.c @@ -70,7 +70,6 @@ static int get_msg(struct mixart_mgr *mgr, struct mixart_msg *resp, unsigned int i; #endif - mutex_lock(&mgr->msg_lock); err = 0; /* copy message descriptor from miXart to driver */ @@ -119,8 +118,6 @@ static int get_msg(struct mixart_mgr *mgr, struct mixart_msg *resp, writel_be(headptr, MIXART_MEM(mgr, MSG_OUTBOUND_FREE_HEAD)); _clean_exit: - mutex_unlock(&mgr->msg_lock); - return err; } @@ -258,7 +255,9 @@ int snd_mixart_send_msg(struct mixart_mgr *mgr, struct mixart_msg *request, int resp.data = resp_data; resp.size = max_resp_size; + mutex_lock(&mgr->msg_lock); err = get_msg(mgr, &resp, msg_frame); + mutex_unlock(&mgr->msg_lock); if( request->message_id != resp.message_id ) dev_err(&mgr->pci->dev, "RESPONSE ERROR!\n"); diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 5cbdc9be9c7e..c7b3e76ea2d2 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -5326,7 +5326,8 @@ static int snd_hdsp_free(struct hdsp *hdsp) if (hdsp->port) pci_release_regions(hdsp->pci); - pci_disable_device(hdsp->pci); + if (pci_is_enabled(hdsp->pci)) + pci_disable_device(hdsp->pci); return 0; } diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 81a6f4b2bd3c..e34f07c9ff47 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -6889,7 +6889,8 @@ static int snd_hdspm_free(struct hdspm * hdspm) if (hdspm->port) pci_release_regions(hdspm->pci); - pci_disable_device(hdspm->pci); + if (pci_is_enabled(hdspm->pci)) + pci_disable_device(hdspm->pci); return 0; } diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c index 4c851f8dcaf8..73ad6e74aac9 100644 --- a/sound/pci/rme9652/rme9652.c +++ b/sound/pci/rme9652/rme9652.c @@ -1745,7 +1745,8 @@ static int snd_rme9652_free(struct snd_rme9652 *rme9652) if (rme9652->port) pci_release_regions(rme9652->pci); - pci_disable_device(rme9652->pci); + if (pci_is_enabled(rme9652->pci)) + pci_disable_device(rme9652->pci); return 0; } diff --git a/sound/ppc/powermac.c b/sound/ppc/powermac.c index 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/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index f4ee6798154a..1612ec65aaf6 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -73,8 +73,13 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) return ret; } - da7219_dai_wclk = clk_get(component->dev, "da7219-dai-wclk"); - da7219_dai_bclk = clk_get(component->dev, "da7219-dai-bclk"); + da7219_dai_wclk = devm_clk_get(component->dev, "da7219-dai-wclk"); + if (IS_ERR(da7219_dai_wclk)) + return PTR_ERR(da7219_dai_wclk); + + da7219_dai_bclk = devm_clk_get(component->dev, "da7219-dai-bclk"); + if (IS_ERR(da7219_dai_bclk)) + return PTR_ERR(da7219_dai_bclk); ret = snd_soc_card_jack_new(card, "Headset Jack", SND_JACK_HEADSET | SND_JACK_LINEOUT | diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index 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 229cc89f8c5a..dfc536cd9d2f 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -586,21 +586,26 @@ 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 @@ -754,7 +759,6 @@ config SND_SOC_MAX98095 config SND_SOC_MAX98357A tristate "Maxim MAX98357A CODEC" - depends on GPIOLIB config SND_SOC_MAX98371 tristate diff --git a/sound/soc/codecs/ak4458.c b/sound/soc/codecs/ak4458.c index 71562154c0b1..eca5fc559d69 100644 --- a/sound/soc/codecs/ak4458.c +++ b/sound/soc/codecs/ak4458.c @@ -523,18 +523,10 @@ static struct snd_soc_dai_driver ak4497_dai = { .ops = &ak4458_dai_ops, }; -static void ak4458_power_off(struct ak4458_priv *ak4458) +static void ak4458_reset(struct ak4458_priv *ak4458, bool active) { if (ak4458->reset_gpiod) { - gpiod_set_value_cansleep(ak4458->reset_gpiod, 0); - usleep_range(1000, 2000); - } -} - -static void ak4458_power_on(struct ak4458_priv *ak4458) -{ - if (ak4458->reset_gpiod) { - gpiod_set_value_cansleep(ak4458->reset_gpiod, 1); + gpiod_set_value_cansleep(ak4458->reset_gpiod, active); usleep_range(1000, 2000); } } @@ -548,7 +540,7 @@ static int ak4458_init(struct snd_soc_component *component) if (ak4458->mute_gpiod) gpiod_set_value_cansleep(ak4458->mute_gpiod, 1); - ak4458_power_on(ak4458); + ak4458_reset(ak4458, false); ret = snd_soc_component_update_bits(component, AK4458_00_CONTROL1, 0x80, 0x80); /* ACKS bit = 1; 10000000 */ @@ -571,7 +563,7 @@ static void ak4458_remove(struct snd_soc_component *component) { struct ak4458_priv *ak4458 = snd_soc_component_get_drvdata(component); - ak4458_power_off(ak4458); + ak4458_reset(ak4458, true); } #ifdef CONFIG_PM @@ -581,7 +573,7 @@ static int __maybe_unused ak4458_runtime_suspend(struct device *dev) regcache_cache_only(ak4458->regmap, true); - ak4458_power_off(ak4458); + ak4458_reset(ak4458, true); if (ak4458->mute_gpiod) gpiod_set_value_cansleep(ak4458->mute_gpiod, 0); @@ -596,8 +588,8 @@ static int __maybe_unused ak4458_runtime_resume(struct device *dev) if (ak4458->mute_gpiod) gpiod_set_value_cansleep(ak4458->mute_gpiod, 1); - ak4458_power_off(ak4458); - ak4458_power_on(ak4458); + ak4458_reset(ak4458, true); + ak4458_reset(ak4458, false); regcache_cache_only(ak4458->regmap, false); regcache_mark_dirty(ak4458->regmap); @@ -715,6 +707,7 @@ static const struct of_device_id ak4458_of_match[] = { { .compatible = "asahi-kasei,ak4497", .data = &ak4497_drvdata}, { }, }; +MODULE_DEVICE_TABLE(of, ak4458_of_match); static struct i2c_driver ak4458_i2c_driver = { .driver = { diff --git a/sound/soc/codecs/ak5558.c b/sound/soc/codecs/ak5558.c index 8179512129d3..e98312a6b840 100644 --- a/sound/soc/codecs/ak5558.c +++ b/sound/soc/codecs/ak5558.c @@ -264,7 +264,7 @@ static void ak5558_power_off(struct ak5558_priv *ak5558) if (!ak5558->reset_gpiod) return; - gpiod_set_value_cansleep(ak5558->reset_gpiod, 0); + gpiod_set_value_cansleep(ak5558->reset_gpiod, 1); usleep_range(1000, 2000); } @@ -273,7 +273,7 @@ static void ak5558_power_on(struct ak5558_priv *ak5558) if (!ak5558->reset_gpiod) return; - gpiod_set_value_cansleep(ak5558->reset_gpiod, 1); + gpiod_set_value_cansleep(ak5558->reset_gpiod, 0); usleep_range(1000, 2000); } @@ -389,6 +389,7 @@ static const struct of_device_id ak5558_i2c_dt_ids[] = { { .compatible = "asahi-kasei,ak5558"}, { } }; +MODULE_DEVICE_TABLE(of, ak5558_i2c_dt_ids); static struct i2c_driver ak5558_i2c_driver = { .driver = { diff --git a/sound/soc/codecs/cpcap.c b/sound/soc/codecs/cpcap.c index d7f05b384f1f..acd88fe38cd4 100644 --- a/sound/soc/codecs/cpcap.c +++ b/sound/soc/codecs/cpcap.c @@ -1263,12 +1263,12 @@ static int cpcap_voice_hw_params(struct snd_pcm_substream *substream, if (direction == SNDRV_PCM_STREAM_CAPTURE) { mask = 0x0000; - mask |= CPCAP_BIT_MIC1_RX_TIMESLOT0; - mask |= CPCAP_BIT_MIC1_RX_TIMESLOT1; - mask |= CPCAP_BIT_MIC1_RX_TIMESLOT2; - mask |= CPCAP_BIT_MIC2_TIMESLOT0; - mask |= CPCAP_BIT_MIC2_TIMESLOT1; - mask |= CPCAP_BIT_MIC2_TIMESLOT2; + mask |= BIT(CPCAP_BIT_MIC1_RX_TIMESLOT0); + mask |= BIT(CPCAP_BIT_MIC1_RX_TIMESLOT1); + mask |= BIT(CPCAP_BIT_MIC1_RX_TIMESLOT2); + mask |= BIT(CPCAP_BIT_MIC2_TIMESLOT0); + mask |= BIT(CPCAP_BIT_MIC2_TIMESLOT1); + mask |= BIT(CPCAP_BIT_MIC2_TIMESLOT2); val = 0x0000; if (channels >= 2) val = BIT(CPCAP_BIT_MIC1_RX_TIMESLOT0); @@ -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/cs35l33.c b/sound/soc/codecs/cs35l33.c index 6042194d95d3..8894369e329a 100644 --- a/sound/soc/codecs/cs35l33.c +++ b/sound/soc/codecs/cs35l33.c @@ -1201,6 +1201,7 @@ static int cs35l33_i2c_probe(struct i2c_client *i2c_client, dev_err(&i2c_client->dev, "CS35L33 Device ID (%X). Expected ID %X\n", devid, CS35L33_CHIP_ID); + ret = -EINVAL; goto err_enable; } diff --git a/sound/soc/codecs/cs35l36.c b/sound/soc/codecs/cs35l36.c index e9b5f76f27a8..aa32b8c26578 100644 --- a/sound/soc/codecs/cs35l36.c +++ b/sound/soc/codecs/cs35l36.c @@ -444,7 +444,8 @@ static bool cs35l36_volatile_reg(struct device *dev, unsigned int reg) } } -static DECLARE_TLV_DB_SCALE(dig_vol_tlv, -10200, 25, 0); +static const DECLARE_TLV_DB_RANGE(dig_vol_tlv, 0, 912, + TLV_DB_MINMAX_ITEM(-10200, 1200)); static DECLARE_TLV_DB_SCALE(amp_gain_tlv, 0, 1, 1); static const char * const cs35l36_pcm_sftramp_text[] = { 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 5125bb9b37b5..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 }, @@ -398,10 +398,13 @@ static const struct regmap_config cs42l42_regmap = { .reg_defaults = cs42l42_reg_defaults, .num_reg_defaults = ARRAY_SIZE(cs42l42_reg_defaults), .cache_type = REGCACHE_RBTREE, + + .use_single_read = true, + .use_single_write = true, }; -static DECLARE_TLV_DB_SCALE(adc_tlv, -9600, 100, false); -static DECLARE_TLV_DB_SCALE(mixer_tlv, -6200, 100, false); +static DECLARE_TLV_DB_SCALE(adc_tlv, -9700, 100, true); +static DECLARE_TLV_DB_SCALE(mixer_tlv, -6300, 100, true); static const char * const cs42l42_hpf_freq_text[] = { "1.86Hz", "120Hz", "235Hz", "466Hz" @@ -420,34 +423,23 @@ 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, @@ -458,7 +450,7 @@ static const struct snd_kcontrol_new cs42l42_snd_controls[] = { CS42L42_DAC_HPF_EN_SHIFT, true, false), SOC_DOUBLE_R_TLV("Mixer Volume", CS42L42_MIXER_CHA_VOL, CS42L42_MIXER_CHB_VOL, CS42L42_MIXER_CH_VOL_SHIFT, - 0x3e, 1, mixer_tlv) + 0x3f, 1, mixer_tlv) }; static int cs42l42_hpdrv_evt(struct snd_soc_dapm_widget *w, @@ -666,15 +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, @@ -691,24 +674,6 @@ static int cs42l42_pll_config(struct snd_soc_component *component) CS42L42_CLK_OASRC_SEL_MASK, CS42L42_CLK_OASRC_SEL_12 << CS42L42_CLK_OASRC_SEL_SHIFT); - /* channel 1 on low LRCLK, 32 bit */ - snd_soc_component_update_bits(component, - CS42L42_ASP_RX_DAI0_CH1_AP_RES, - CS42L42_ASP_RX_CH_AP_MASK | - CS42L42_ASP_RX_CH_RES_MASK, - (CS42L42_ASP_RX_CH_AP_LOW << - CS42L42_ASP_RX_CH_AP_SHIFT) | - (CS42L42_ASP_RX_CH_RES_32 << - CS42L42_ASP_RX_CH_RES_SHIFT)); - /* Channel 2 on high LRCLK, 32 bit */ - snd_soc_component_update_bits(component, - CS42L42_ASP_RX_DAI0_CH2_AP_RES, - CS42L42_ASP_RX_CH_AP_MASK | - CS42L42_ASP_RX_CH_RES_MASK, - (CS42L42_ASP_RX_CH_AP_HI << - CS42L42_ASP_RX_CH_AP_SHIFT) | - (CS42L42_ASP_RX_CH_RES_32 << - CS42L42_ASP_RX_CH_RES_SHIFT)); if (pll_ratio_table[i].mclk_src_sel == 0) { /* Pass the clock straight through */ snd_soc_component_update_bits(component, @@ -788,7 +753,18 @@ 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; @@ -797,27 +773,23 @@ static int cs42l42_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) /* Bitclock/frame inversion */ switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: + asp_cfg_val |= CS42L42_ASP_SCPOL_NOR << CS42L42_ASP_SCPOL_SHIFT; break; case SND_SOC_DAIFMT_NB_IF: - asp_cfg_val |= CS42L42_ASP_POL_INV << - CS42L42_ASP_LCPOL_IN_SHIFT; + asp_cfg_val |= CS42L42_ASP_SCPOL_NOR << CS42L42_ASP_SCPOL_SHIFT; + asp_cfg_val |= CS42L42_ASP_LCPOL_INV << CS42L42_ASP_LCPOL_SHIFT; break; case SND_SOC_DAIFMT_IB_NF: - asp_cfg_val |= CS42L42_ASP_POL_INV << - CS42L42_ASP_SCPOL_IN_DAC_SHIFT; break; case SND_SOC_DAIFMT_IB_IF: - asp_cfg_val |= CS42L42_ASP_POL_INV << - CS42L42_ASP_LCPOL_IN_SHIFT; - asp_cfg_val |= CS42L42_ASP_POL_INV << - CS42L42_ASP_SCPOL_IN_DAC_SHIFT; + asp_cfg_val |= CS42L42_ASP_LCPOL_INV << CS42L42_ASP_LCPOL_SHIFT; break; } - snd_soc_component_update_bits(component, CS42L42_ASP_CLK_CFG, - CS42L42_ASP_MODE_MASK | - CS42L42_ASP_SCPOL_IN_DAC_MASK | - CS42L42_ASP_LCPOL_IN_MASK, asp_cfg_val); + snd_soc_component_update_bits(component, CS42L42_ASP_CLK_CFG, CS42L42_ASP_MODE_MASK | + CS42L42_ASP_SCPOL_MASK | + CS42L42_ASP_LCPOL_MASK, + asp_cfg_val); return 0; } @@ -828,14 +800,29 @@ static int cs42l42_pcm_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_component *component = dai->component; struct cs42l42_private *cs42l42 = snd_soc_component_get_drvdata(component); - int retval; + unsigned int width = (params_width(params) / 8) - 1; + unsigned int val = 0; cs42l42->srate = params_rate(params); - cs42l42->swidth = params_width(params); - retval = cs42l42_pll_config(component); + switch(substream->stream) { + case SNDRV_PCM_STREAM_PLAYBACK: + val |= width << CS42L42_ASP_RX_CH_RES_SHIFT; + /* channel 1 on low LRCLK */ + snd_soc_component_update_bits(component, CS42L42_ASP_RX_DAI0_CH1_AP_RES, + CS42L42_ASP_RX_CH_AP_MASK | + CS42L42_ASP_RX_CH_RES_MASK, val); + /* Channel 2 on high LRCLK */ + val |= CS42L42_ASP_RX_CH_AP_HI << CS42L42_ASP_RX_CH_AP_SHIFT; + snd_soc_component_update_bits(component, CS42L42_ASP_RX_DAI0_CH2_AP_RES, + CS42L42_ASP_RX_CH_AP_MASK | + CS42L42_ASP_RX_CH_RES_MASK, val); + break; + default: + break; + } - return retval; + return cs42l42_pll_config(component); } static int cs42l42_set_sysclk(struct snd_soc_dai *dai, @@ -900,9 +887,9 @@ static int cs42l42_digital_mute(struct snd_soc_dai *dai, int mute) return 0; } -#define CS42L42_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S18_3LE | \ - SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE | \ - SNDRV_PCM_FMTBIT_S32_LE) +#define CS42L42_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE ) static const struct snd_soc_dai_ops cs42l42_ops = { @@ -1803,7 +1790,7 @@ static int cs42l42_i2c_probe(struct i2c_client *i2c_client, dev_dbg(&i2c_client->dev, "Found reset GPIO\n"); gpiod_set_value_cansleep(cs42l42->reset_gpio, 1); } - mdelay(3); + usleep_range(CS42L42_BOOT_TIME_US, CS42L42_BOOT_TIME_US * 2); /* Request IRQ */ ret = devm_request_threaded_irq(&i2c_client->dev, @@ -1811,8 +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); @@ -1928,6 +1916,7 @@ static int cs42l42_runtime_resume(struct device *dev) } gpiod_set_value_cansleep(cs42l42->reset_gpio, 1); + usleep_range(CS42L42_BOOT_TIME_US, CS42L42_BOOT_TIME_US * 2); regcache_cache_only(cs42l42->regmap, false); regcache_sync(cs42l42->regmap); diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h index 9e3cc528dcff..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) @@ -258,11 +258,12 @@ #define CS42L42_ASP_SLAVE_MODE 0x00 #define CS42L42_ASP_MODE_SHIFT 4 #define CS42L42_ASP_MODE_MASK (1 << CS42L42_ASP_MODE_SHIFT) -#define CS42L42_ASP_SCPOL_IN_DAC_SHIFT 2 -#define CS42L42_ASP_SCPOL_IN_DAC_MASK (1 << CS42L42_ASP_SCPOL_IN_DAC_SHIFT) -#define CS42L42_ASP_LCPOL_IN_SHIFT 0 -#define CS42L42_ASP_LCPOL_IN_MASK (1 << CS42L42_ASP_LCPOL_IN_SHIFT) -#define CS42L42_ASP_POL_INV 1 +#define CS42L42_ASP_SCPOL_SHIFT 2 +#define CS42L42_ASP_SCPOL_MASK (3 << CS42L42_ASP_SCPOL_SHIFT) +#define CS42L42_ASP_SCPOL_NOR 3 +#define CS42L42_ASP_LCPOL_SHIFT 0 +#define CS42L42_ASP_LCPOL_MASK (3 << CS42L42_ASP_LCPOL_SHIFT) +#define CS42L42_ASP_LCPOL_INV 3 #define CS42L42_ASP_FRM_CFG (CS42L42_PAGE_12 + 0x08) #define CS42L42_ASP_STP_SHIFT 4 @@ -739,6 +740,7 @@ #define CS42L42_FRAC2_VAL(val) (((val) & 0xff0000) >> 16) #define CS42L42_NUM_SUPPLIES 5 +#define CS42L42_BOOT_TIME_US 3000 static const char *const cs42l42_supply_names[CS42L42_NUM_SUPPLIES] = { "VA", @@ -756,7 +758,6 @@ struct cs42l42_private { struct completion pdn_done; u32 sclk; u32 srate; - u32 swidth; u8 plug_state; u8 hs_type; u8 ts_inv; diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 55408c8fcb4e..cdd7ae90c2b5 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -247,8 +247,28 @@ static const struct snd_soc_dapm_widget cs42l51_dapm_widgets[] = { &cs42l51_adcr_mux_controls), }; +static int mclk_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm); + struct cs42l51_private *cs42l51 = snd_soc_component_get_drvdata(comp); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + return clk_prepare_enable(cs42l51->mclk_handle); + case SND_SOC_DAPM_POST_PMD: + /* Delay mclk shutdown to fulfill power-down sequence requirements */ + msleep(20); + clk_disable_unprepare(cs42l51->mclk_handle); + break; + } + + return 0; +} + static const struct snd_soc_dapm_widget cs42l51_dapm_mclk_widgets[] = { - SND_SOC_DAPM_CLOCK_SUPPLY("MCLK") + SND_SOC_DAPM_SUPPLY("MCLK", SND_SOC_NOPM, 0, 0, mclk_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), }; static const struct snd_soc_dapm_route cs42l51_routes[] = { diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 2ea4cba3be2a..6c054b357205 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -137,7 +137,9 @@ static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0); static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); -static DECLARE_TLV_DB_SCALE(mix_tlv, -50, 50, 0); +static DECLARE_TLV_DB_SCALE(pass_tlv, -6000, 50, 0); + +static DECLARE_TLV_DB_SCALE(mix_tlv, -5150, 50, 0); static DECLARE_TLV_DB_SCALE(beep_tlv, -56, 200, 0); @@ -351,7 +353,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { CS42L52_SPKB_VOL, 0, 0x40, 0xC0, hl_tlv), SOC_DOUBLE_R_SX_TLV("Bypass Volume", CS42L52_PASSTHRUA_VOL, - CS42L52_PASSTHRUB_VOL, 0, 0x88, 0x90, pga_tlv), + CS42L52_PASSTHRUB_VOL, 0, 0x88, 0x90, pass_tlv), SOC_DOUBLE("Bypass Mute", CS42L52_MISC_CTL, 4, 5, 1, 0), @@ -364,7 +366,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { CS42L52_ADCB_VOL, 0, 0xA0, 0x78, ipd_tlv), SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", CS42L52_ADCA_MIXER_VOL, CS42L52_ADCB_MIXER_VOL, - 0, 0x19, 0x7F, ipd_tlv), + 0, 0x19, 0x7F, mix_tlv), SOC_DOUBLE("ADC Switch", CS42L52_ADC_MISC_CTL, 0, 1, 1, 0), diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index ac569ab3d30f..8be7d83f0ce9 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -391,9 +391,9 @@ static const struct snd_kcontrol_new cs42l56_snd_controls[] = { SOC_DOUBLE("ADC Boost Switch", CS42L56_GAIN_BIAS_CTL, 3, 2, 1, 1), SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L56_HPA_VOLUME, - CS42L56_HPB_VOLUME, 0, 0x84, 0x48, hl_tlv), + CS42L56_HPB_VOLUME, 0, 0x44, 0x48, hl_tlv), SOC_DOUBLE_R_SX_TLV("LineOut Volume", CS42L56_LOA_VOLUME, - CS42L56_LOB_VOLUME, 0, 0x84, 0x48, hl_tlv), + CS42L56_LOB_VOLUME, 0, 0x44, 0x48, hl_tlv), SOC_SINGLE_TLV("Bass Shelving Volume", CS42L56_TONE_CTL, 0, 0x00, 1, tone_tlv), @@ -1248,6 +1248,7 @@ static int cs42l56_i2c_probe(struct i2c_client *i2c_client, dev_err(&i2c_client->dev, "CS42L56 Device ID (%X). Expected %X\n", devid, CS42L56_DEVID); + ret = -EINVAL; goto err_enable; } alpha_rev = reg & CS42L56_AREV_MASK; @@ -1305,7 +1306,7 @@ static int cs42l56_i2c_probe(struct i2c_client *i2c_client, ret = devm_snd_soc_register_component(&i2c_client->dev, &soc_component_dev_cs42l56, &cs42l56_dai, 1); if (ret < 0) - return ret; + goto err_enable; return 0; diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index 7fb34422a2a4..8f70dee95878 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -1735,6 +1735,14 @@ static DEVICE_ATTR(hpload_dc_r, 0444, cs43130_show_dc_r, NULL); static DEVICE_ATTR(hpload_ac_l, 0444, cs43130_show_ac_l, NULL); static DEVICE_ATTR(hpload_ac_r, 0444, cs43130_show_ac_r, NULL); +static struct attribute *hpload_attrs[] = { + &dev_attr_hpload_dc_l.attr, + &dev_attr_hpload_dc_r.attr, + &dev_attr_hpload_ac_l.attr, + &dev_attr_hpload_ac_r.attr, +}; +ATTRIBUTE_GROUPS(hpload); + static struct reg_sequence hp_en_cal_seq[] = { {CS43130_INT_MASK_4, CS43130_INT_MASK_ALL}, {CS43130_HP_MEAS_LOAD_1, 0}, @@ -2302,25 +2310,15 @@ static int cs43130_probe(struct snd_soc_component *component) cs43130->hpload_done = false; if (cs43130->dc_meas) { - ret = device_create_file(component->dev, &dev_attr_hpload_dc_l); - if (ret < 0) - return ret; - - ret = device_create_file(component->dev, &dev_attr_hpload_dc_r); - if (ret < 0) - return ret; - - ret = device_create_file(component->dev, &dev_attr_hpload_ac_l); - if (ret < 0) - return ret; - - ret = device_create_file(component->dev, &dev_attr_hpload_ac_r); - if (ret < 0) + ret = sysfs_create_groups(&component->dev->kobj, hpload_groups); + if (ret) return ret; cs43130->wq = create_singlethread_workqueue("cs43130_hp"); - if (!cs43130->wq) + if (!cs43130->wq) { + sysfs_remove_groups(&component->dev->kobj, hpload_groups); return -ENOMEM; + } INIT_WORK(&cs43130->work, cs43130_imp_meas); } diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c index ed22361b35c1..a5a383b92305 100644 --- a/sound/soc/codecs/cs53l30.c +++ b/sound/soc/codecs/cs53l30.c @@ -347,22 +347,22 @@ static const struct snd_kcontrol_new cs53l30_snd_controls[] = { SOC_ENUM("ADC2 NG Delay", adc2_ng_delay_enum), SOC_SINGLE_SX_TLV("ADC1A PGA Volume", - CS53L30_ADC1A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC1A_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC1B PGA Volume", - CS53L30_ADC1B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC1B_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC2A PGA Volume", - CS53L30_ADC2A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC2A_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC2B PGA Volume", - CS53L30_ADC2B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + CS53L30_ADC2B_AFE_CTL, 0, 0x34, 0x24, pga_tlv), SOC_SINGLE_SX_TLV("ADC1A Digital Volume", - CS53L30_ADC1A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC1A_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC1B Digital Volume", - CS53L30_ADC1B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC1B_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC2A Digital Volume", - CS53L30_ADC2A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC2A_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), SOC_SINGLE_SX_TLV("ADC2B Digital Volume", - CS53L30_ADC2B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + CS53L30_ADC2B_DIG_VOL, 0, 0xA0, 0x6C, dig_tlv), }; static const struct snd_soc_dapm_widget cs53l30_dapm_widgets[] = { diff --git a/sound/soc/codecs/cx2072x.c b/sound/soc/codecs/cx2072x.c index 1c1ba7bea4d8..8ee4b2e1ff68 100644 --- a/sound/soc/codecs/cx2072x.c +++ b/sound/soc/codecs/cx2072x.c @@ -1579,7 +1579,7 @@ static struct snd_soc_dai_driver soc_codec_cx2072x_dai[] = { .id = CX2072X_DAI_DSP, .probe = cx2072x_dsp_dai_probe, .playback = { - .stream_name = "Playback", + .stream_name = "DSP Playback", .channels_min = 2, .channels_max = 2, .rates = CX2072X_RATES_DSP, @@ -1591,7 +1591,7 @@ static struct snd_soc_dai_driver soc_codec_cx2072x_dai[] = { .name = "cx2072x-aec", .id = 3, .capture = { - .stream_name = "Capture", + .stream_name = "AEC Capture", .channels_min = 2, .channels_max = 2, .rates = CX2072X_RATES_DSP, diff --git a/sound/soc/codecs/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/es8316.c b/sound/soc/codecs/es8316.c index 36eef1fb3d18..b781b28de012 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -63,13 +63,8 @@ static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(adc_pga_gain_tlv, 1, 1, TLV_DB_SCALE_ITEM(0, 0, 0), 2, 2, TLV_DB_SCALE_ITEM(250, 0, 0), 3, 3, TLV_DB_SCALE_ITEM(450, 0, 0), - 4, 4, TLV_DB_SCALE_ITEM(700, 0, 0), - 5, 5, TLV_DB_SCALE_ITEM(1000, 0, 0), - 6, 6, TLV_DB_SCALE_ITEM(1300, 0, 0), - 7, 7, TLV_DB_SCALE_ITEM(1600, 0, 0), - 8, 8, TLV_DB_SCALE_ITEM(1800, 0, 0), - 9, 9, TLV_DB_SCALE_ITEM(2100, 0, 0), - 10, 10, TLV_DB_SCALE_ITEM(2400, 0, 0), + 4, 7, TLV_DB_SCALE_ITEM(700, 300, 0), + 8, 10, TLV_DB_SCALE_ITEM(1800, 300, 0), ); static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(hpout_vol_tlv, diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index fdf64c29f563..4117ab6e9b6f 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -161,13 +161,16 @@ static int es8328_put_deemph(struct snd_kcontrol *kcontrol, if (deemph > 1) return -EINVAL; + if (es8328->deemph == deemph) + return 0; + ret = es8328_set_deemph(component); if (ret < 0) return ret; es8328->deemph = deemph; - return 0; + return 1; } 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/max98088.c b/sound/soc/codecs/max98088.c index f031d2caa8b7..fa4cdbfd0b80 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -41,6 +41,7 @@ struct max98088_priv { enum max98088_type devtype; struct max98088_pdata *pdata; struct clk *mclk; + unsigned char mclk_prescaler; unsigned int sysclk; struct max98088_cdata dai[2]; int eq_textcnt; @@ -998,13 +999,16 @@ static int max98088_dai1_hw_params(struct snd_pcm_substream *substream, /* Configure NI when operating as master */ if (snd_soc_component_read32(component, M98088_REG_14_DAI1_FORMAT) & M98088_DAI_MAS) { + unsigned long pclk; + if (max98088->sysclk == 0) { dev_err(component->dev, "Invalid system clock frequency\n"); return -EINVAL; } ni = 65536ULL * (rate < 50000 ? 96ULL : 48ULL) * (unsigned long long int)rate; - do_div(ni, (unsigned long long int)max98088->sysclk); + pclk = DIV_ROUND_CLOSEST(max98088->sysclk, max98088->mclk_prescaler); + ni = DIV_ROUND_CLOSEST_ULL(ni, pclk); snd_soc_component_write(component, M98088_REG_12_DAI1_CLKCFG_HI, (ni >> 8) & 0x7F); snd_soc_component_write(component, M98088_REG_13_DAI1_CLKCFG_LO, @@ -1065,13 +1069,16 @@ static int max98088_dai2_hw_params(struct snd_pcm_substream *substream, /* Configure NI when operating as master */ if (snd_soc_component_read32(component, M98088_REG_1C_DAI2_FORMAT) & M98088_DAI_MAS) { + unsigned long pclk; + if (max98088->sysclk == 0) { dev_err(component->dev, "Invalid system clock frequency\n"); return -EINVAL; } ni = 65536ULL * (rate < 50000 ? 96ULL : 48ULL) * (unsigned long long int)rate; - do_div(ni, (unsigned long long int)max98088->sysclk); + pclk = DIV_ROUND_CLOSEST(max98088->sysclk, max98088->mclk_prescaler); + ni = DIV_ROUND_CLOSEST_ULL(ni, pclk); snd_soc_component_write(component, M98088_REG_1A_DAI2_CLKCFG_HI, (ni >> 8) & 0x7F); snd_soc_component_write(component, M98088_REG_1B_DAI2_CLKCFG_LO, @@ -1113,8 +1120,10 @@ static int max98088_dai_set_sysclk(struct snd_soc_dai *dai, */ if ((freq >= 10000000) && (freq < 20000000)) { snd_soc_component_write(component, M98088_REG_10_SYS_CLK, 0x10); + max98088->mclk_prescaler = 1; } else if ((freq >= 20000000) && (freq < 30000000)) { snd_soc_component_write(component, M98088_REG_10_SYS_CLK, 0x20); + max98088->mclk_prescaler = 2; } else { dev_err(component->dev, "Invalid master clock frequency\n"); return -EINVAL; diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 6b9d326e11b0..ce9f99dd3e87 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -413,6 +413,9 @@ static int max98090_put_enab_tlv(struct snd_kcontrol *kcontrol, val = (val >> mc->shift) & mask; + if (sel < 0 || sel > mc->max) + return -EINVAL; + *select = sel; /* Setting a volume is only valid if it is already On */ @@ -427,7 +430,7 @@ static int max98090_put_enab_tlv(struct snd_kcontrol *kcontrol, mask << mc->shift, sel << mc->shift); - return 0; + return *select != val; } static const char *max98090_perf_pwr_text[] = diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index 96718e3a1ad0..16fbc9faed90 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -410,11 +410,13 @@ static int max98373_dac_event(struct snd_soc_dapm_widget *w, regmap_update_bits(max98373->regmap, MAX98373_R20FF_GLOBAL_SHDN, MAX98373_GLOBAL_EN_MASK, 1); + usleep_range(30000, 31000); break; case SND_SOC_DAPM_POST_PMD: regmap_update_bits(max98373->regmap, MAX98373_R20FF_GLOBAL_SHDN, MAX98373_GLOBAL_EN_MASK, 0); + usleep_range(30000, 31000); max98373->tdm_mode = false; break; default: diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index 337bddb7c2a4..5a8eedea6be0 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -1195,8 +1195,10 @@ static int pm8916_wcd_analog_spmi_probe(struct platform_device *pdev) } irq = platform_get_irq_byname(pdev, "mbhc_switch_int"); - if (irq < 0) - return irq; + if (irq < 0) { + ret = irq; + goto err_disable_clk; + } ret = devm_request_threaded_irq(dev, irq, NULL, pm8916_mbhc_switch_irq_handler, @@ -1208,8 +1210,10 @@ 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) - return irq; + if (irq < 0) { + ret = irq; + goto err_disable_clk; + } ret = devm_request_threaded_irq(dev, irq, NULL, mbhc_btn_press_irq_handler, @@ -1220,8 +1224,10 @@ static int pm8916_wcd_analog_spmi_probe(struct platform_device *pdev) dev_err(dev, "cannot request mbhc button press irq\n"); irq = platform_get_irq_byname(pdev, "mbhc_but_rel_det"); - if (irq < 0) - return irq; + if (irq < 0) { + ret = irq; + goto err_disable_clk; + } ret = devm_request_threaded_irq(dev, irq, NULL, mbhc_btn_release_irq_handler, @@ -1238,6 +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/nau8822.c b/sound/soc/codecs/nau8822.c index 78db3bd0b3bc..cd163978792e 100644 --- a/sound/soc/codecs/nau8822.c +++ b/sound/soc/codecs/nau8822.c @@ -741,6 +741,8 @@ static int nau8822_set_pll(struct snd_soc_dai *dai, int pll_id, int source, pll_param->mclk_scaler, pll_param->pre_factor); snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, NAU8822_PLL_EN_MASK, NAU8822_PLL_OFF); + snd_soc_component_update_bits(component, NAU8822_REG_PLL_N, NAU8822_PLLMCLK_DIV2 | NAU8822_PLLN_MASK, (pll_param->pre_factor ? NAU8822_PLLMCLK_DIV2 : 0) | pll_param->pll_int); @@ -757,6 +759,8 @@ static int nau8822_set_pll(struct snd_soc_dai *dai, int pll_id, int source, pll_param->mclk_scaler << NAU8822_MCLKSEL_SFT); snd_soc_component_update_bits(component, NAU8822_REG_CLOCKING, NAU8822_CLKM_MASK, NAU8822_CLKM_PLL); + snd_soc_component_update_bits(component, + NAU8822_REG_POWER_MANAGEMENT_1, NAU8822_PLL_EN_MASK, NAU8822_PLL_ON); return 0; } diff --git a/sound/soc/codecs/nau8822.h b/sound/soc/codecs/nau8822.h index 489191ff187e..b45d42c15de6 100644 --- a/sound/soc/codecs/nau8822.h +++ b/sound/soc/codecs/nau8822.h @@ -90,6 +90,9 @@ #define NAU8822_REFIMP_3K 0x3 #define NAU8822_IOBUF_EN (0x1 << 2) #define NAU8822_ABIAS_EN (0x1 << 3) +#define NAU8822_PLL_EN_MASK (0x1 << 5) +#define NAU8822_PLL_ON (0x1 << 5) +#define NAU8822_PLL_OFF (0x0 << 5) /* NAU8822_REG_AUDIO_INTERFACE (0x4) */ #define NAU8822_AIFMT_MASK (0x3 << 3) 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..4e71ecf54af7 100644 --- a/sound/soc/codecs/rk3328_codec.c +++ b/sound/soc/codecs/rk3328_codec.c @@ -472,29 +472,45 @@ 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); if (ret < 0) { dev_err(&pdev->dev, "failed to enable acodec pclk\n"); - return ret; + goto err_unprepare_mclk; } base = devm_platform_ioremap_resource(pdev, 0); - if (IS_ERR(base)) - return PTR_ERR(base); + if (IS_ERR(base)) { + ret = PTR_ERR(base); + goto err_unprepare_pclk; + } rk3328->regmap = devm_regmap_init_mmio(&pdev->dev, base, &rk3328_codec_regmap_config); - if (IS_ERR(rk3328->regmap)) - return PTR_ERR(rk3328->regmap); + if (IS_ERR(rk3328->regmap)) { + ret = PTR_ERR(rk3328->regmap); + goto err_unprepare_pclk; + } platform_set_drvdata(pdev, rk3328); - return devm_snd_soc_register_component(&pdev->dev, &soc_codec_rk3328, + ret = devm_snd_soc_register_component(&pdev->dev, &soc_codec_rk3328, rk3328_dai, ARRAY_SIZE(rk3328_dai)); + if (ret) + goto err_unprepare_pclk; + + return 0; + +err_unprepare_pclk: + clk_disable_unprepare(rk3328->pclk); + +err_unprepare_mclk: + clk_disable_unprepare(rk3328->mclk); + return ret; } static const struct of_device_id rk3328_codec_of_match[] = { diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 9593a9a27bf8..d8ab8af2c786 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -171,6 +171,9 @@ static bool rt286_readable_register(struct device *dev, unsigned int reg) case RT286_PROC_COEF: case RT286_SET_AMP_GAIN_ADC_IN1: case RT286_SET_AMP_GAIN_ADC_IN2: + case RT286_SET_GPIO_MASK: + case RT286_SET_GPIO_DIRECTION: + case RT286_SET_GPIO_DATA: case RT286_SET_POWER(RT286_DAC_OUT1): case RT286_SET_POWER(RT286_DAC_OUT2): case RT286_SET_POWER(RT286_ADC_IN1): @@ -1115,12 +1118,11 @@ static const struct dmi_system_id force_combo_jack_table[] = { { } }; -static const struct dmi_system_id dmi_dell_dino[] = { +static const struct dmi_system_id dmi_dell[] = { { - .ident = "Dell Dino", + .ident = "Dell", .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc."), - DMI_MATCH(DMI_PRODUCT_NAME, "XPS 13 9343") } }, { } @@ -1131,7 +1133,7 @@ static int rt286_i2c_probe(struct i2c_client *i2c, { struct rt286_platform_data *pdata = dev_get_platdata(&i2c->dev); struct rt286_priv *rt286; - int i, ret, val; + int i, ret, vendor_id; rt286 = devm_kzalloc(&i2c->dev, sizeof(*rt286), GFP_KERNEL); @@ -1147,14 +1149,15 @@ static int rt286_i2c_probe(struct i2c_client *i2c, } ret = regmap_read(rt286->regmap, - RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &val); + RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &vendor_id); if (ret != 0) { dev_err(&i2c->dev, "I2C error %d\n", ret); return ret; } - if (val != RT286_VENDOR_ID && val != RT288_VENDOR_ID) { + if (vendor_id != RT286_VENDOR_ID && vendor_id != RT288_VENDOR_ID) { dev_err(&i2c->dev, - "Device with ID register %#x is not rt286\n", val); + "Device with ID register %#x is not rt286\n", + vendor_id); return -ENODEV; } @@ -1178,8 +1181,8 @@ static int rt286_i2c_probe(struct i2c_client *i2c, if (pdata) rt286->pdata = *pdata; - if (dmi_check_system(force_combo_jack_table) || - dmi_check_system(dmi_dell_dino)) + if ((vendor_id == RT288_VENDOR_ID && dmi_check_system(dmi_dell)) || + dmi_check_system(force_combo_jack_table)) rt286->pdata.cbj_en = true; regmap_write(rt286->regmap, RT286_SET_AUDIO_POWER, AC_PWRST_D3); @@ -1218,7 +1221,7 @@ static int rt286_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL3, 0xf777, 0x4737); regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL4, 0x00ff, 0x003f); - if (dmi_check_system(dmi_dell_dino)) { + if (vendor_id == RT288_VENDOR_ID && dmi_check_system(dmi_dell)) { regmap_update_bits(rt286->regmap, RT286_SET_GPIO_MASK, 0x40, 0x40); regmap_update_bits(rt286->regmap, diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 7081142a355e..c444a56df95b 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -419,7 +419,7 @@ static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol, } } - return 0; + return 1; } static const struct snd_kcontrol_new rt5514_snd_controls[] = { 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/rt5640.c b/sound/soc/codecs/rt5640.c index 747ca248bf10..3bc63fbcb188 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -339,9 +339,9 @@ static bool rt5640_readable_register(struct device *dev, unsigned int reg) } static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0); -static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); +static const DECLARE_TLV_DB_MINMAX(dac_vol_tlv, -6562, 0); static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); -static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); +static const DECLARE_TLV_DB_MINMAX(adc_vol_tlv, -1762, 3000); static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); /* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */ diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index c83f7f5da96b..a66e93a3af74 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -4074,9 +4074,14 @@ static int rt5645_i2c_remove(struct i2c_client *i2c) if (i2c->irq) free_irq(i2c->irq, rt5645); + /* + * Since the rt5645_btn_check_callback() can queue jack_detect_work, + * the timer need to be delted first + */ + del_timer_sync(&rt5645->btn_check_timer); + cancel_delayed_work_sync(&rt5645->jack_detect_work); cancel_delayed_work_sync(&rt5645->rcclock_work); - del_timer_sync(&rt5645->btn_check_timer); regulator_bulk_disable(ARRAY_SIZE(rt5645->supplies), rt5645->supplies); diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index c506c9305043..829cf552fe3e 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -285,9 +285,9 @@ static bool rt5651_readable_register(struct device *dev, unsigned int reg) } static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0); -static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); +static const DECLARE_TLV_DB_MINMAX(dac_vol_tlv, -6562, 0); static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); -static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); +static const DECLARE_TLV_DB_MINMAX(adc_vol_tlv, -1762, 3000); static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); /* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */ diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c index e66d08398f74..a28afb480060 100644 --- a/sound/soc/codecs/rt5659.c +++ b/sound/soc/codecs/rt5659.c @@ -2470,13 +2470,18 @@ static int set_dmic_power(struct snd_soc_dapm_widget *w, return 0; } -static const struct snd_soc_dapm_widget rt5659_dapm_widgets[] = { +static const struct snd_soc_dapm_widget rt5659_particular_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("LDO2", RT5659_PWR_ANLG_3, RT5659_PWR_LDO2_BIT, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("PLL", RT5659_PWR_ANLG_3, RT5659_PWR_PLL_BIT, 0, - NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5659_PWR_ANLG_2, RT5659_PWR_MB1_BIT, + 0, NULL, 0), SND_SOC_DAPM_SUPPLY("Mic Det Power", RT5659_PWR_VOL, RT5659_PWR_MIC_DET_BIT, 0, NULL, 0), +}; + +static const struct snd_soc_dapm_widget rt5659_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("PLL", RT5659_PWR_ANLG_3, RT5659_PWR_PLL_BIT, 0, + NULL, 0), SND_SOC_DAPM_SUPPLY("Mono Vref", RT5659_PWR_ANLG_1, RT5659_PWR_VREF3_BIT, 0, NULL, 0), @@ -2501,8 +2506,6 @@ static const struct snd_soc_dapm_widget rt5659_dapm_widgets[] = { RT5659_ADC_MONO_R_ASRC_SFT, 0, NULL, 0), /* Input Side */ - SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5659_PWR_ANLG_2, RT5659_PWR_MB1_BIT, - 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MICBIAS2", RT5659_PWR_ANLG_2, RT5659_PWR_MB2_BIT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MICBIAS3", RT5659_PWR_ANLG_2, RT5659_PWR_MB3_BIT, @@ -3463,12 +3466,17 @@ static int rt5659_set_component_sysclk(struct snd_soc_component *component, int { struct rt5659_priv *rt5659 = snd_soc_component_get_drvdata(component); unsigned int reg_val = 0; + int ret; if (freq == rt5659->sysclk && clk_id == rt5659->sysclk_src) return 0; switch (clk_id) { case RT5659_SCLK_S_MCLK: + ret = clk_set_rate(rt5659->mclk, freq); + if (ret) + return ret; + reg_val |= RT5659_SCLK_SRC_MCLK; break; case RT5659_SCLK_S_PLL1: @@ -3692,10 +3700,23 @@ static int rt5659_set_bias_level(struct snd_soc_component *component, static int rt5659_probe(struct snd_soc_component *component) { + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); struct rt5659_priv *rt5659 = snd_soc_component_get_drvdata(component); rt5659->component = component; + switch (rt5659->pdata.jd_src) { + case RT5659_JD_HDA_HEADER: + break; + + default: + snd_soc_dapm_new_controls(dapm, + rt5659_particular_dapm_widgets, + ARRAY_SIZE(rt5659_particular_dapm_widgets)); + break; + } + return 0; } diff --git a/sound/soc/codecs/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 f5b59305c957..8a1e485982d8 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -71,7 +71,7 @@ static const struct reg_default sgtl5000_reg_defaults[] = { { SGTL5000_DAP_EQ_BASS_BAND4, 0x002f }, { SGTL5000_DAP_MAIN_CHAN, 0x8000 }, { SGTL5000_DAP_MIX_CHAN, 0x0000 }, - { SGTL5000_DAP_AVC_CTRL, 0x0510 }, + { SGTL5000_DAP_AVC_CTRL, 0x5100 }, { SGTL5000_DAP_AVC_THRESHOLD, 0x1473 }, { SGTL5000_DAP_AVC_ATTACK, 0x0028 }, { SGTL5000_DAP_AVC_DECAY, 0x0050 }, diff --git a/sound/soc/codecs/sti-sas.c b/sound/soc/codecs/sti-sas.c index ec9933b054ad..423daac9d5a9 100644 --- a/sound/soc/codecs/sti-sas.c +++ b/sound/soc/codecs/sti-sas.c @@ -411,6 +411,7 @@ static const struct of_device_id sti_sas_dev_match[] = { }, {}, }; +MODULE_DEVICE_TABLE(of, sti_sas_dev_match); static int sti_sas_driver_probe(struct platform_device *pdev) { diff --git a/sound/soc/codecs/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/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 68165de1c8de..7a1ffbaf48be 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -662,7 +662,7 @@ static int aic32x4_set_processing_blocks(struct snd_soc_component *component, } static int aic32x4_setup_clocks(struct snd_soc_component *component, - unsigned int sample_rate) + unsigned int sample_rate, unsigned int channels) { u8 aosr; u16 dosr; @@ -750,7 +750,9 @@ static int aic32x4_setup_clocks(struct snd_soc_component *component, dosr); clk_set_rate(clocks[5].clk, - sample_rate * 32); + sample_rate * 32 * + channels); + return 0; } } @@ -772,7 +774,8 @@ static int aic32x4_hw_params(struct snd_pcm_substream *substream, u8 iface1_reg = 0; u8 dacsetup_reg = 0; - aic32x4_setup_clocks(component, params_rate(params)); + aic32x4_setup_clocks(component, params_rate(params), + params_channels(params)); switch (params_width(params)) { case 16: diff --git a/sound/soc/codecs/tscs454.c b/sound/soc/codecs/tscs454.c index c3587af9985c..3d981441b8d1 100644 --- a/sound/soc/codecs/tscs454.c +++ b/sound/soc/codecs/tscs454.c @@ -3128,18 +3128,17 @@ static int set_aif_sample_format(struct snd_soc_component *component, unsigned int width; int ret; - switch (format) { - case SNDRV_PCM_FORMAT_S16_LE: + switch (snd_pcm_format_width(format)) { + case 16: width = FV_WL_16; break; - case SNDRV_PCM_FORMAT_S20_3LE: + case 20: width = FV_WL_20; break; - case SNDRV_PCM_FORMAT_S24_3LE: + case 24: width = FV_WL_24; break; - case SNDRV_PCM_FORMAT_S24_LE: - case SNDRV_PCM_FORMAT_S32_LE: + case 32: width = FV_WL_32; break; default: @@ -3337,6 +3336,7 @@ static const struct snd_soc_component_driver soc_component_dev_tscs454 = { .num_dapm_routes = ARRAY_SIZE(tscs454_intercon), .controls = tscs454_snd_controls, .num_controls = ARRAY_SIZE(tscs454_snd_controls), + .endianness = 1, }; #define TSCS454_RATES SNDRV_PCM_RATE_8000_96000 diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index f318403133e9..016aff97e2fb 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -618,7 +618,7 @@ static const char * const sb_tx8_mux_text[] = { "ZERO", "RX_MIX_TX8", "DEC8", "DEC8_192" }; -static const DECLARE_TLV_DB_SCALE(digital_gain, 0, 1, 0); +static const DECLARE_TLV_DB_SCALE(digital_gain, -8400, 100, -8400); static const DECLARE_TLV_DB_SCALE(line_gain, 0, 7, 1); static const DECLARE_TLV_DB_SCALE(analog_gain, 0, 25, 1); static const DECLARE_TLV_DB_SCALE(ear_pa_gain, 0, 150, 0); @@ -4076,6 +4076,16 @@ 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/wm2000.c b/sound/soc/codecs/wm2000.c index 72e165cc6443..97ece3114b3d 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -536,7 +536,7 @@ static int wm2000_anc_transition(struct wm2000_priv *wm2000, { struct i2c_client *i2c = wm2000->i2c; int i, j; - int ret; + int ret = 0; if (wm2000->anc_mode == mode) return 0; @@ -566,13 +566,13 @@ static int wm2000_anc_transition(struct wm2000_priv *wm2000, ret = anc_transitions[i].step[j](i2c, anc_transitions[i].analogue); if (ret != 0) - return ret; + break; } if (anc_transitions[i].dest == ANC_OFF) clk_disable_unprepare(wm2000->mclk); - return 0; + return ret; } static int wm2000_anc_set_mode(struct wm2000_priv *wm2000) diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index fe99584c917f..9cd91bb0a902 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1535,18 +1535,38 @@ static int wm8350_component_probe(struct snd_soc_component *component) wm8350_clear_bits(wm8350, WM8350_JACK_DETECT, WM8350_JDL_ENA | WM8350_JDR_ENA); - wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L, + ret = wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L, wm8350_hpl_jack_handler, 0, "Left jack detect", priv); - wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R, + if (ret != 0) + goto err; + + ret = wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R, wm8350_hpr_jack_handler, 0, "Right jack detect", priv); - wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_MICSCD, + if (ret != 0) + goto free_jck_det_l; + + ret = wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_MICSCD, wm8350_mic_handler, 0, "Microphone short", priv); - wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_MICD, + if (ret != 0) + goto free_jck_det_r; + + ret = wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_MICD, wm8350_mic_handler, 0, "Microphone detect", priv); + if (ret != 0) + goto free_micscd; return 0; + +free_micscd: + wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_MICSCD, priv); +free_jck_det_r: + wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R, priv); +free_jck_det_l: + wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L, priv); +err: + return ret; } static void wm8350_component_remove(struct snd_soc_component *component) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 6fd1bef848ed..fa55d79b39b6 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -601,7 +601,7 @@ static int wm8731_hw_init(struct device *dev, struct wm8731_priv *wm8731) ret = wm8731_reset(wm8731->regmap); if (ret < 0) { dev_err(dev, "Failed to issue reset: %d\n", ret); - goto err_regulator_enable; + goto err; } /* Clear POWEROFF, keep everything else disabled */ @@ -618,10 +618,7 @@ static int wm8731_hw_init(struct device *dev, struct wm8731_priv *wm8731) regcache_mark_dirty(wm8731->regmap); -err_regulator_enable: - /* Regulators will be enabled by bias management */ - regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); - +err: return ret; } @@ -765,21 +762,27 @@ static int wm8731_i2c_probe(struct i2c_client *i2c, ret = PTR_ERR(wm8731->regmap); dev_err(&i2c->dev, "Failed to allocate register map: %d\n", ret); - return ret; + goto err_regulator_enable; } ret = wm8731_hw_init(&i2c->dev, wm8731); if (ret != 0) - return ret; + goto err_regulator_enable; ret = devm_snd_soc_register_component(&i2c->dev, &soc_component_dev_wm8731, &wm8731_dai, 1); if (ret != 0) { dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret); - return ret; + goto err_regulator_enable; } return 0; + +err_regulator_enable: + /* Regulators will be enabled by bias management */ + regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); + + return ret; } static int wm8731_i2c_remove(struct i2c_client *client) diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index 04f23477039a..c677c068b05e 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -534,7 +534,7 @@ static int wm8958_mbc_put(struct snd_kcontrol *kcontrol, wm8958_dsp_apply(component, mbc, wm8994->mbc_ena[mbc]); - return 0; + return 1; } #define WM8958_MBC_SWITCH(xname, xval) {\ @@ -660,7 +660,7 @@ static int wm8958_vss_put(struct snd_kcontrol *kcontrol, wm8958_dsp_apply(component, vss, wm8994->vss_ena[vss]); - return 0; + return 1; } @@ -734,7 +734,7 @@ static int wm8958_hpf_put(struct snd_kcontrol *kcontrol, wm8958_dsp_apply(component, hpf % 3, ucontrol->value.integer.value[0]); - return 0; + return 1; } #define WM8958_HPF_SWITCH(xname, xval) {\ @@ -828,7 +828,7 @@ static int wm8958_enh_eq_put(struct snd_kcontrol *kcontrol, wm8958_dsp_apply(component, eq, ucontrol->value.integer.value[0]); - return 0; + return 1; } #define WM8958_ENH_EQ_SWITCH(xname, xval) {\ diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 6cf0f6612bda..512f8899dcbb 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -707,7 +707,13 @@ int wm8960_configure_pll(struct snd_soc_component *component, int freq_in, best_freq_out = -EINVAL; *sysclk_idx = *dac_idx = *bclk_idx = -1; - for (i = 0; i < ARRAY_SIZE(sysclk_divs); ++i) { + /* + * From Datasheet, the PLL performs best when f2 is between + * 90MHz and 100MHz, the desired sysclk output is 11.2896MHz + * or 12.288MHz, then sysclkdiv = 2 is the best choice. + * So search sysclk_divs from 2 to 1 other than from 1 to 2. + */ + for (i = ARRAY_SIZE(sysclk_divs) - 1; i >= 0; --i) { if (sysclk_divs[i] == -1) continue; for (j = 0; j < ARRAY_SIZE(dac_divs); ++j) { @@ -746,9 +752,16 @@ 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/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index d9d59f45833f..ebaee468057b 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3854,6 +3854,7 @@ static int wm8962_runtime_suspend(struct device *dev) #endif static const struct dev_pm_ops wm8962_pm = { + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, pm_runtime_force_resume) SET_RUNTIME_PM_OPS(wm8962_runtime_suspend, wm8962_runtime_resume, NULL) }; diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index 37e4bb3dbd8a..229f2986cd96 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -1177,6 +1177,8 @@ static int wm8997_probe(struct platform_device *pdev) goto err_spk_irqs; } + return ret; + err_spk_irqs: arizona_free_spk_irqs(arizona); diff --git a/sound/soc/codecs/wm8998.c b/sound/soc/codecs/wm8998.c index 7c1899219573..817ccddd6344 100644 --- a/sound/soc/codecs/wm8998.c +++ b/sound/soc/codecs/wm8998.c @@ -1375,7 +1375,7 @@ static int wm8998_probe(struct platform_device *pdev) ret = arizona_init_spk_irqs(arizona); if (ret < 0) - return ret; + goto err_pm_disable; ret = devm_snd_soc_register_component(&pdev->dev, &soc_component_dev_wm8998, @@ -1390,6 +1390,8 @@ static int wm8998_probe(struct platform_device *pdev) err_spk_irqs: arizona_free_spk_irqs(arizona); +err_pm_disable: + pm_runtime_disable(&pdev->dev); return ret; } diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 9b8bb7bbe945..aedfa6b2895b 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -791,7 +791,7 @@ int wm_adsp_fw_put(struct snd_kcontrol *kcontrol, struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; struct wm_adsp *dsp = snd_soc_component_get_drvdata(component); - int ret = 0; + int ret = 1; if (ucontrol->value.enumerated.item[0] == dsp[e->shift_l].fw) return 0; @@ -1496,7 +1496,7 @@ static int wm_adsp_create_control(struct wm_adsp *dsp, ctl_work = kzalloc(sizeof(*ctl_work), GFP_KERNEL); if (!ctl_work) { ret = -ENOMEM; - goto err_ctl_cache; + goto err_list_del; } ctl_work->dsp = dsp; @@ -1506,7 +1506,8 @@ static int wm_adsp_create_control(struct wm_adsp *dsp, return 0; -err_ctl_cache: +err_list_del: + list_del(&ctl->list); kfree(ctl->cache); err_ctl_name: kfree(ctl->name); @@ -1912,6 +1913,7 @@ static int wm_adsp_load(struct wm_adsp *dsp) mem = wm_adsp_find_region(dsp, type); if (!mem) { adsp_err(dsp, "No region of type: %x\n", type); + ret = -EINVAL; goto out_fw; } diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 84290be778f0..33ade79fa032 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -494,11 +494,13 @@ static int fsl_esai_startup(struct snd_pcm_substream *substream, ESAI_SAICR_SYNC, esai_priv->synchronous ? ESAI_SAICR_SYNC : 0); - /* Set a default slot number -- 2 */ + /* Set slots count */ regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR, - ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(2)); + ESAI_xCCR_xDC_MASK, + ESAI_xCCR_xDC(esai_priv->slots)); regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, - ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(2)); + ESAI_xCCR_xDC_MASK, + ESAI_xCCR_xDC(esai_priv->slots)); } return 0; diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 914b75c23d1b..027259695551 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -694,7 +694,7 @@ static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai) return 0; } -static struct snd_soc_dai_driver fsl_sai_dai = { +static struct snd_soc_dai_driver fsl_sai_dai_template = { .probe = fsl_sai_dai_probe, .playback = { .stream_name = "CPU-Playback", @@ -965,12 +965,15 @@ static int fsl_sai_probe(struct platform_device *pdev) return ret; } + memcpy(&sai->cpu_dai_drv, &fsl_sai_dai_template, + sizeof(fsl_sai_dai_template)); + /* Sync Tx with Rx as default by following old DT binding */ sai->synchronous[RX] = true; sai->synchronous[TX] = false; - fsl_sai_dai.symmetric_rates = 1; - fsl_sai_dai.symmetric_channels = 1; - fsl_sai_dai.symmetric_samplebits = 1; + sai->cpu_dai_drv.symmetric_rates = 1; + sai->cpu_dai_drv.symmetric_channels = 1; + sai->cpu_dai_drv.symmetric_samplebits = 1; if (of_find_property(np, "fsl,sai-synchronous-rx", NULL) && of_find_property(np, "fsl,sai-asynchronous", NULL)) { @@ -987,9 +990,9 @@ static int fsl_sai_probe(struct platform_device *pdev) /* Discard all settings for asynchronous mode */ sai->synchronous[RX] = false; sai->synchronous[TX] = false; - fsl_sai_dai.symmetric_rates = 0; - fsl_sai_dai.symmetric_channels = 0; - fsl_sai_dai.symmetric_samplebits = 0; + sai->cpu_dai_drv.symmetric_rates = 0; + sai->cpu_dai_drv.symmetric_channels = 0; + sai->cpu_dai_drv.symmetric_samplebits = 0; } if (of_find_property(np, "fsl,sai-mclk-direction-output", NULL) && @@ -1018,7 +1021,7 @@ static int fsl_sai_probe(struct platform_device *pdev) pm_runtime_enable(&pdev->dev); ret = devm_snd_soc_register_component(&pdev->dev, &fsl_component, - &fsl_sai_dai, 1); + &sai->cpu_dai_drv, 1); if (ret) goto err_pm_disable; diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 6aba7d28f5f3..afaef2027234 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -67,8 +67,8 @@ #define FSL_SAI_xCR3(tx, ofs) (tx ? FSL_SAI_TCR3(ofs) : FSL_SAI_RCR3(ofs)) #define FSL_SAI_xCR4(tx, ofs) (tx ? FSL_SAI_TCR4(ofs) : FSL_SAI_RCR4(ofs)) #define FSL_SAI_xCR5(tx, ofs) (tx ? FSL_SAI_TCR5(ofs) : FSL_SAI_RCR5(ofs)) -#define FSL_SAI_xDR(tx, ofs) (tx ? FSL_SAI_TDR(ofs) : FSL_SAI_RDR(ofs)) -#define FSL_SAI_xFR(tx, ofs) (tx ? FSL_SAI_TFR(ofs) : FSL_SAI_RFR(ofs)) +#define FSL_SAI_xDR0(tx) (tx ? FSL_SAI_TDR0 : FSL_SAI_RDR0) +#define FSL_SAI_xFR0(tx) (tx ? FSL_SAI_TFR0 : FSL_SAI_RFR0) #define FSL_SAI_xMR(tx) (tx ? FSL_SAI_TMR : FSL_SAI_RMR) /* SAI Transmit/Receive Control Register */ @@ -180,6 +180,7 @@ struct fsl_sai { unsigned int bclk_ratio; const struct fsl_sai_soc_data *soc_data; + struct snd_soc_dai_driver cpu_dai_drv; struct snd_dmaengine_dai_dma_data dma_params_rx; struct snd_dmaengine_dai_dma_data dma_params_tx; }; diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index a4ebd6ddaba1..ed18bc69e095 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -873,6 +873,7 @@ static int fsl_ssi_hw_free(struct snd_pcm_substream *substream, static int _fsl_ssi_set_dai_fmt(struct fsl_ssi *ssi, unsigned int fmt) { u32 strcr = 0, scr = 0, stcr, srcr, mask; + unsigned int slots; ssi->dai_fmt = fmt; @@ -904,10 +905,11 @@ static int _fsl_ssi_set_dai_fmt(struct fsl_ssi *ssi, unsigned int fmt) return -EINVAL; } + slots = ssi->slots ? : 2; regmap_update_bits(ssi->regs, REG_SSI_STCCR, - SSI_SxCCR_DC_MASK, SSI_SxCCR_DC(2)); + SSI_SxCCR_DC_MASK, SSI_SxCCR_DC(slots)); regmap_update_bits(ssi->regs, REG_SSI_SRCCR, - SSI_SxCCR_DC_MASK, SSI_SxCCR_DC(2)); + SSI_SxCCR_DC_MASK, SSI_SxCCR_DC(slots)); /* Data on rising edge of bclk, frame low, 1clk before data */ strcr |= SSI_STCR_TFSI | SSI_STCR_TSCKP | SSI_STCR_TEFS; diff --git a/sound/soc/fsl/imx-es8328.c b/sound/soc/fsl/imx-es8328.c index 15a27a2cd0ca..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; } @@ -145,13 +146,13 @@ static int imx_es8328_probe(struct platform_device *pdev) data = devm_kzalloc(dev, sizeof(*data), GFP_KERNEL); if (!data) { ret = -ENOMEM; - goto fail; + goto put_device; } comp = devm_kzalloc(dev, 3 * sizeof(*comp), GFP_KERNEL); if (!comp) { ret = -ENOMEM; - goto fail; + goto put_device; } data->dev = dev; @@ -182,12 +183,12 @@ static int imx_es8328_probe(struct platform_device *pdev) ret = snd_soc_of_parse_card_name(&data->card, "model"); if (ret) { dev_err(dev, "Unable to parse card name\n"); - goto fail; + goto put_device; } ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); if (ret) { dev_err(dev, "Unable to parse routing: %d\n", ret); - goto fail; + goto put_device; } data->card.num_links = 1; data->card.owner = THIS_MODULE; @@ -196,10 +197,12 @@ static int imx_es8328_probe(struct platform_device *pdev) ret = snd_soc_register_card(&data->card); if (ret) { dev_err(dev, "Unable to register: %d\n", ret); - goto fail; + goto put_device; } platform_set_drvdata(pdev, data); +put_device: + put_device(&ssi_pdev->dev); fail: of_node_put(ssi_np); of_node_put(codec_np); diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 15e8b9343c35..7106d56a3346 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -120,19 +120,19 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); if (!data) { ret = -ENOMEM; - goto fail; + goto put_device; } comp = devm_kzalloc(&pdev->dev, 3 * sizeof(*comp), GFP_KERNEL); if (!comp) { ret = -ENOMEM; - goto fail; + goto put_device; } data->codec_clk = clk_get(&codec_dev->dev, NULL); if (IS_ERR(data->codec_clk)) { ret = PTR_ERR(data->codec_clk); - goto fail; + goto put_device; } data->clk_frequency = clk_get_rate(data->codec_clk); @@ -158,10 +158,10 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) data->card.dev = &pdev->dev; ret = snd_soc_of_parse_card_name(&data->card, "model"); if (ret) - goto fail; + goto put_device; ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); if (ret) - goto fail; + goto put_device; data->card.num_links = 1; data->card.owner = THIS_MODULE; data->card.dai_link = &data->dai; @@ -176,7 +176,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) if (ret != -EPROBE_DEFER) dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); - goto fail; + goto put_device; } of_node_put(ssi_np); @@ -184,6 +184,8 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) return 0; +put_device: + put_device(&codec_dev->dev); fail: if (data && !IS_ERR(data->codec_clk)) clk_put(data->codec_clk); 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/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 6007e6305735..1bc498124689 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -340,7 +340,7 @@ static int graph_dai_link_of(struct asoc_simple_priv *priv, struct device_node *top = dev->of_node; struct asoc_simple_dai *cpu_dai; struct asoc_simple_dai *codec_dai; - int ret, single_cpu; + int ret, single_cpu = 0; /* Do it only CPU turn */ if (!li->cpu) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index fc9c753db8dd..4484c40c7a39 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -258,7 +258,7 @@ static int simple_dai_link_of(struct asoc_simple_priv *priv, struct device_node *plat = NULL; char prop[128]; char *prefix = ""; - int ret, single_cpu; + int ret, single_cpu = 0; /* * |CPU |Codec : turn diff --git a/sound/soc/hisilicon/hi6210-i2s.c b/sound/soc/hisilicon/hi6210-i2s.c index 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 31f1dd6541aa..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; @@ -500,14 +499,14 @@ static struct snd_soc_dai_driver sst_platform_dai[] = { .channels_min = SST_STEREO, .channels_max = SST_STEREO, .rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { .stream_name = "Headset Capture", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + .formats = SNDRV_PCM_FMTBIT_S16_LE, }, }, { @@ -518,7 +517,7 @@ static struct snd_soc_dai_driver sst_platform_dai[] = { .channels_min = SST_STEREO, .channels_max = SST_STEREO, .rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + .formats = SNDRV_PCM_FMTBIT_S16_LE, }, }, { diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 6012367f6fe4..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"}, }; @@ -400,6 +403,19 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { /* Acer One 10 S1002 */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Acer"), + DMI_MATCH(DMI_PRODUCT_NAME, "One S1002"), + }, + .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF2 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Acer"), @@ -424,6 +440,18 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { }, { .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ARCHOS"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "ARCHOS 140 CESIUM"), + }, + .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, + { + .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "ME176C"), }, @@ -451,6 +479,9 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "T100TAF"), }, .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | BYT_RT5640_MONO_SPEAKER | BYT_RT5640_DIFF_MIC | BYT_RT5640_SSP0_AIF2 | @@ -485,6 +516,23 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_MCLK_EN), }, { + /* Chuwi Hi8 (CWI509) */ + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Hampoo"), + DMI_MATCH(DMI_BOARD_NAME, "BYT-PA03C"), + DMI_MATCH(DMI_SYS_VENDOR, "ilife"), + DMI_MATCH(DMI_PRODUCT_NAME, "S806"), + }, + .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, + { .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Circuitco"), DMI_MATCH(DMI_PRODUCT_NAME, "Minnowboard Max B3 PLATFORM"), @@ -513,6 +561,27 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_MONO_SPEAKER | BYT_RT5640_MCLK_EN), }, + { /* Estar Beauty HD MID 7316R */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Estar"), + DMI_MATCH(DMI_PRODUCT_NAME, "eSTAR BEAUTY HD Intel Quad core"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, + { /* Glavey TM800A550L */ + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "AMI Corporation"), + DMI_MATCH(DMI_BOARD_NAME, "Aptio CRB"), + /* Above strings are too generic, also match on BIOS version */ + DMI_MATCH(DMI_BIOS_VERSION, "ZY-8-BI-PX4S70VTR400-X423B-005-D"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Hewlett-Packard"), @@ -580,6 +649,20 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_MONO_SPEAKER | BYT_RT5640_MCLK_EN), }, + { /* Lenovo Miix 3-830 */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "LENOVO"), + DMI_EXACT_MATCH(DMI_PRODUCT_VERSION, "Lenovo MIIX 3-830"), + }, + .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { /* Linx Linx7 tablet */ .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "LINX"), @@ -776,6 +859,20 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF2 | BYT_RT5640_MCLK_EN), }, + { /* Voyo Winpad A15 */ + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "AMI Corporation"), + DMI_MATCH(DMI_BOARD_NAME, "Aptio CRB"), + /* Above strings are too generic, also match on BIOS date */ + DMI_MATCH(DMI_BIOS_DATE, "11/20/2014"), + }, + .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_MCLK_EN), + }, { /* Catch-all for generic Insyde tablets, must be last */ .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Insyde"), diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 4606f6f582d6..921c09cdb480 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -436,6 +436,19 @@ static const struct dmi_system_id byt_rt5651_quirk_table[] = { BYT_RT5651_MONO_SPEAKER), }, { + /* Jumper EZpad 7 */ + .callback = byt_rt5651_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Jumper"), + DMI_MATCH(DMI_PRODUCT_NAME, "EZpad"), + /* Jumper12x.WJ2012.bsBKRCP05 with the version dropped */ + DMI_MATCH(DMI_BIOS_VERSION, "Jumper12x.WJ2012.bsBKRCP"), + }, + .driver_data = (void *)(BYT_RT5651_DEFAULT_QUIRKS | + BYT_RT5651_IN2_MAP | + BYT_RT5651_JD_NOT_INV), + }, + { /* KIANO SlimNote 14.2 */ .callback = byt_rt5651_quirk_cb, .matches = { diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c index 3dadf9bff796..cf47fd9cd506 100644 --- a/sound/soc/intel/boards/haswell.c +++ b/sound/soc/intel/boards/haswell.c @@ -206,6 +206,7 @@ static struct platform_driver haswell_audio = { .probe = haswell_audio_probe, .driver = { .name = "haswell-audio", + .pm = &snd_soc_pm_ops, }, }; diff --git a/sound/soc/intel/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/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index 829f95fc4179..16996f6aa2dc 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -282,12 +282,34 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_interval *channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - struct snd_soc_dpcm *dpcm = container_of( - params, struct snd_soc_dpcm, hw_params); - struct snd_soc_dai_link *fe_dai_link = dpcm->fe->dai_link; - struct snd_soc_dai_link *be_dai_link = dpcm->be->dai_link; + struct snd_soc_dpcm *dpcm, *rtd_dpcm = NULL; /* + * The following loop will be called only for playback stream + * In this platform, there is only one playback device on every SSP + */ + for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_PLAYBACK, dpcm) { + rtd_dpcm = dpcm; + break; + } + + /* + * This following loop will be called only for capture stream + * In this platform, there is only one capture device on every SSP + */ + for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_CAPTURE, dpcm) { + rtd_dpcm = dpcm; + break; + } + + if (!rtd_dpcm) + return -EINVAL; + + /* + * The above 2 loops are mutually exclusive based on the stream direction, + * thus rtd_dpcm variable will never be overwritten + */ + /* * Topology for kblda7219m98373 & kblmax98373 supports only S24_LE, * where as kblda7219m98927 & kblmax98927 supports S16_LE by default. * Skipping the port wise FE and BE configuration for kblda7219m98373 & @@ -309,9 +331,9 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, /* * The ADSP will convert the FE rate to 48k, stereo, 24 bit */ - if (!strcmp(fe_dai_link->name, "Kbl Audio Port") || - !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") || - !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) { + if (!strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Port") || + !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Headset Playback") || + !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Capture Port")) { rate->min = rate->max = 48000; channels->min = channels->max = 2; snd_mask_none(fmt); @@ -322,7 +344,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, * The speaker on the SSP0 supports S16_LE and not S24_LE. * thus changing the mask here */ - if (!strcmp(be_dai_link->name, "SSP0-Codec")) + if (!strcmp(rtd_dpcm->be->dai_link->name, "SSP0-Codec")) snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE); return 0; diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index 7cefda341fbf..a540a2dad80c 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -401,17 +401,40 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_interval *channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - struct snd_soc_dpcm *dpcm = container_of( - params, struct snd_soc_dpcm, hw_params); - struct snd_soc_dai_link *fe_dai_link = dpcm->fe->dai_link; - struct snd_soc_dai_link *be_dai_link = dpcm->be->dai_link; + struct snd_soc_dpcm *dpcm, *rtd_dpcm = NULL; + + /* + * The following loop will be called only for playback stream + * In this platform, there is only one playback device on every SSP + */ + for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_PLAYBACK, dpcm) { + rtd_dpcm = dpcm; + break; + } + + /* + * This following loop will be called only for capture stream + * In this platform, there is only one capture device on every SSP + */ + for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_CAPTURE, dpcm) { + rtd_dpcm = dpcm; + break; + } + + if (!rtd_dpcm) + return -EINVAL; + + /* + * The above 2 loops are mutually exclusive based on the stream direction, + * thus rtd_dpcm variable will never be overwritten + */ /* * The ADSP will convert the FE rate to 48k, stereo, 24 bit */ - if (!strcmp(fe_dai_link->name, "Kbl Audio Port") || - !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") || - !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) { + if (!strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Port") || + !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Headset Playback") || + !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Capture Port")) { rate->min = rate->max = 48000; channels->min = channels->max = 2; snd_mask_none(fmt); @@ -421,7 +444,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, * The speaker on the SSP0 supports S16_LE and not S24_LE. * thus changing the mask here */ - if (!strcmp(be_dai_link->name, "SSP0-Codec")) + if (!strcmp(rtd_dpcm->be->dai_link->name, "SSP0-Codec")) snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE); return 0; diff --git a/sound/soc/intel/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/common/soc-intel-quirks.h b/sound/soc/intel/common/soc-intel-quirks.h index 863a477d3405..645baf0ed3dd 100644 --- a/sound/soc/intel/common/soc-intel-quirks.h +++ b/sound/soc/intel/common/soc-intel-quirks.h @@ -11,6 +11,7 @@ #if IS_ENABLED(CONFIG_X86) +#include <linux/dmi.h> #include <asm/cpu_device_id.h> #include <asm/intel-family.h> #include <asm/iosf_mbi.h> @@ -40,12 +41,36 @@ SOC_INTEL_IS_CPU(cml, INTEL_FAM6_KABYLAKE_L); static inline bool soc_intel_is_byt_cr(struct platform_device *pdev) { + /* + * List of systems which: + * 1. Use a non CR version of the Bay Trail SoC + * 2. Contain at least 6 interrupt resources so that the + * platform_get_resource(pdev, IORESOURCE_IRQ, 5) check below + * succeeds + * 3. Despite 1. and 2. still have their IPC IRQ at index 0 rather then 5 + * + * This needs to be here so that it can be shared between the SST and + * SOF drivers. We rely on the compiler to optimize this out in files + * where soc_intel_is_byt_cr is not used. + */ + static const struct dmi_system_id force_bytcr_table[] = { + { /* Lenovo Yoga Tablet 2 series */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_FAMILY, "YOGATablet2"), + }, + }, + {} + }; struct device *dev = &pdev->dev; int status = 0; if (!soc_intel_is_byt()) return false; + if (dmi_check_system(force_bytcr_table)) + return true; + if (iosf_mbi_available()) { u32 bios_status; diff --git a/sound/soc/intel/skylake/bxt-sst.c b/sound/soc/intel/skylake/bxt-sst.c index 92a82e6b5fe6..38b9d7494083 100644 --- a/sound/soc/intel/skylake/bxt-sst.c +++ b/sound/soc/intel/skylake/bxt-sst.c @@ -17,7 +17,6 @@ #include "skl.h" #define BXT_BASEFW_TIMEOUT 3000 -#define BXT_INIT_TIMEOUT 300 #define BXT_ROM_INIT_TIMEOUT 70 #define BXT_IPC_PURGE_FW 0x01004000 @@ -38,8 +37,6 @@ /* Delay before scheduling D0i3 entry */ #define BXT_D0I3_DELAY 5000 -#define BXT_FW_ROM_INIT_RETRY 3 - static unsigned int bxt_get_errorcode(struct sst_dsp *ctx) { return sst_dsp_shim_read(ctx, BXT_ADSP_ERROR_CODE); diff --git a/sound/soc/intel/skylake/cnl-sst.c b/sound/soc/intel/skylake/cnl-sst.c index 4f64f097e9ae..e808f62960ba 100644 --- a/sound/soc/intel/skylake/cnl-sst.c +++ b/sound/soc/intel/skylake/cnl-sst.c @@ -57,18 +57,34 @@ static int cnl_prepare_fw(struct sst_dsp *ctx, const void *fwdata, u32 fwsize) ctx->dsp_ops.stream_tag = stream_tag; memcpy(ctx->dmab.area, fwdata, fwsize); + ret = skl_dsp_core_power_up(ctx, SKL_DSP_CORE0_MASK); + if (ret < 0) { + dev_err(ctx->dev, "dsp core0 power up failed\n"); + ret = -EIO; + goto base_fw_load_failed; + } + /* purge FW request */ sst_dsp_shim_write(ctx, CNL_ADSP_REG_HIPCIDR, CNL_ADSP_REG_HIPCIDR_BUSY | (CNL_IPC_PURGE | ((stream_tag - 1) << CNL_ROM_CTRL_DMA_ID))); - ret = cnl_dsp_enable_core(ctx, SKL_DSP_CORE0_MASK); + ret = skl_dsp_start_core(ctx, SKL_DSP_CORE0_MASK); if (ret < 0) { - dev_err(ctx->dev, "dsp boot core failed ret: %d\n", ret); + dev_err(ctx->dev, "Start dsp core failed ret: %d\n", ret); ret = -EIO; goto base_fw_load_failed; } + ret = sst_dsp_register_poll(ctx, CNL_ADSP_REG_HIPCIDA, + CNL_ADSP_REG_HIPCIDA_DONE, + CNL_ADSP_REG_HIPCIDA_DONE, + BXT_INIT_TIMEOUT, "HIPCIDA Done"); + if (ret < 0) { + dev_err(ctx->dev, "timeout for purge request: %d\n", ret); + goto base_fw_load_failed; + } + /* enable interrupt */ cnl_ipc_int_enable(ctx); cnl_ipc_op_int_enable(ctx); @@ -109,7 +125,7 @@ static int cnl_load_base_firmware(struct sst_dsp *ctx) { struct firmware stripped_fw; struct skl_dev *cnl = ctx->thread_context; - int ret; + int ret, i; if (!ctx->fw) { ret = request_firmware(&ctx->fw, ctx->fw_name, ctx->dev); @@ -131,12 +147,16 @@ static int cnl_load_base_firmware(struct sst_dsp *ctx) stripped_fw.size = ctx->fw->size; skl_dsp_strip_extended_manifest(&stripped_fw); - ret = cnl_prepare_fw(ctx, stripped_fw.data, stripped_fw.size); - if (ret < 0) { - dev_err(ctx->dev, "prepare firmware failed: %d\n", ret); - goto cnl_load_base_firmware_failed; + for (i = 0; i < BXT_FW_ROM_INIT_RETRY; i++) { + ret = cnl_prepare_fw(ctx, stripped_fw.data, stripped_fw.size); + if (!ret) + break; + dev_dbg(ctx->dev, "prepare firmware failed: %d\n", ret); } + if (ret < 0) + goto cnl_load_base_firmware_failed; + ret = sst_transfer_fw_host_dma(ctx); if (ret < 0) { dev_err(ctx->dev, "transfer firmware failed: %d\n", ret); @@ -158,6 +178,7 @@ static int cnl_load_base_firmware(struct sst_dsp *ctx) return 0; cnl_load_base_firmware_failed: + dev_err(ctx->dev, "firmware load failed: %d\n", ret); release_firmware(ctx->fw); ctx->fw = NULL; @@ -203,6 +224,7 @@ static int cnl_set_dsp_D0(struct sst_dsp *ctx, unsigned int core_id) "dsp boot timeout, status=%#x error=%#x\n", sst_dsp_shim_read(ctx, CNL_ADSP_FW_STATUS), sst_dsp_shim_read(ctx, CNL_ADSP_ERROR_CODE)); + ret = -ETIMEDOUT; goto err; } } else { diff --git a/sound/soc/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-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c index 19f328d71f24..d9c8f5cb389e 100644 --- a/sound/soc/intel/skylake/skl-nhlt.c +++ b/sound/soc/intel/skylake/skl-nhlt.c @@ -182,7 +182,8 @@ void skl_nhlt_remove_sysfs(struct skl_dev *skl) { struct device *dev = &skl->pci->dev; - sysfs_remove_file(&dev->kobj, &dev_attr_platform_id.attr); + if (skl->nhlt) + sysfs_remove_file(&dev->kobj, &dev_attr_platform_id.attr); } /* diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 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-sst-dsp.h b/sound/soc/intel/skylake/skl-sst-dsp.h index cdfec0fca577..1df9ef422f61 100644 --- a/sound/soc/intel/skylake/skl-sst-dsp.h +++ b/sound/soc/intel/skylake/skl-sst-dsp.h @@ -67,6 +67,8 @@ struct skl_dev; #define SKL_FW_INIT 0x1 #define SKL_FW_RFW_START 0xf +#define BXT_FW_ROM_INIT_RETRY 3 +#define BXT_INIT_TIMEOUT 300 #define SKL_ADSPIC_IPC 1 #define SKL_ADSPIS_IPC 1 diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 69cd7a81bf2a..254b796e635d 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -14,6 +14,7 @@ #include <linux/uuid.h> #include <sound/intel-nhlt.h> #include <sound/soc.h> +#include <sound/soc-acpi.h> #include <sound/soc-topology.h> #include <uapi/sound/snd_sst_tokens.h> #include <uapi/sound/skl-tplg-interface.h> @@ -112,7 +113,7 @@ static 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"); @@ -194,8 +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) { @@ -238,9 +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; @@ -578,6 +579,38 @@ static int skl_tplg_unload_pipe_modules(struct skl_dev *skl, return ret; } +static bool skl_tplg_is_multi_fmt(struct skl_dev *skl, struct skl_pipe *pipe) +{ + struct skl_pipe_fmt *cur_fmt; + struct skl_pipe_fmt *next_fmt; + int i; + + if (pipe->nr_cfgs <= 1) + return false; + + if (pipe->conn_type != SKL_PIPE_CONN_TYPE_FE) + return true; + + for (i = 0; i < pipe->nr_cfgs - 1; i++) { + if (pipe->direction == SNDRV_PCM_STREAM_PLAYBACK) { + cur_fmt = &pipe->configs[i].out_fmt; + next_fmt = &pipe->configs[i + 1].out_fmt; + } else { + cur_fmt = &pipe->configs[i].in_fmt; + next_fmt = &pipe->configs[i + 1].in_fmt; + } + + if (!CHECK_HW_PARAMS(cur_fmt->channels, cur_fmt->freq, + cur_fmt->bps, + next_fmt->channels, + next_fmt->freq, + next_fmt->bps)) + return true; + } + + return false; +} + /* * Here, we select pipe format based on the pipe type and pipe * direction to determine the current config index for the pipeline. @@ -600,6 +633,14 @@ skl_tplg_get_pipe_config(struct skl_dev *skl, struct skl_module_cfg *mconfig) return 0; } + if (skl_tplg_is_multi_fmt(skl, pipe)) { + pipe->cur_config_idx = pipe->pipe_config_idx; + pipe->memory_pages = pconfig->mem_pages; + dev_dbg(skl->dev, "found pipe config idx:%d\n", + pipe->cur_config_idx); + return 0; + } + if (pipe->conn_type == SKL_PIPE_CONN_TYPE_NONE) { dev_dbg(skl->dev, "No conn_type detected, take 0th config\n"); pipe->cur_config_idx = 0; @@ -1314,6 +1355,68 @@ static int skl_tplg_pga_event(struct snd_soc_dapm_widget *w, return 0; } +static int skl_tplg_multi_config_set_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol, + bool is_set) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct hdac_bus *bus = snd_soc_component_get_drvdata(component); + struct skl_dev *skl = bus_to_skl(bus); + struct skl_pipeline *ppl; + struct skl_pipe *pipe = NULL; + struct soc_enum *ec = (struct soc_enum *)kcontrol->private_value; + u32 *pipe_id; + + if (!ec) + return -EINVAL; + + if (is_set && ucontrol->value.enumerated.item[0] > ec->items) + return -EINVAL; + + pipe_id = ec->dobj.private; + + list_for_each_entry(ppl, &skl->ppl_list, node) { + if (ppl->pipe->ppl_id == *pipe_id) { + pipe = ppl->pipe; + break; + } + } + if (!pipe) + return -EIO; + + if (is_set) + pipe->pipe_config_idx = ucontrol->value.enumerated.item[0]; + else + ucontrol->value.enumerated.item[0] = pipe->pipe_config_idx; + + return 0; +} + +static int skl_tplg_multi_config_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return skl_tplg_multi_config_set_get(kcontrol, ucontrol, false); +} + +static int skl_tplg_multi_config_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return skl_tplg_multi_config_set_get(kcontrol, ucontrol, true); +} + +static int skl_tplg_multi_config_get_dmic(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return skl_tplg_multi_config_set_get(kcontrol, ucontrol, false); +} + +static int skl_tplg_multi_config_set_dmic(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return skl_tplg_multi_config_set_get(kcontrol, ucontrol, true); +} + static int skl_tplg_tlv_control_get(struct snd_kcontrol *kcontrol, unsigned int __user *data, unsigned int size) { @@ -1360,12 +1463,6 @@ 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; @@ -1534,11 +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; @@ -1547,9 +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; @@ -1853,6 +1951,16 @@ static const struct snd_soc_tplg_kcontrol_ops skl_tplg_kcontrol_ops[] = { .get = skl_tplg_mic_control_get, .put = skl_tplg_mic_control_set, }, + { + .id = SKL_CONTROL_TYPE_MULTI_IO_SELECT, + .get = skl_tplg_multi_config_get, + .put = skl_tplg_multi_config_set, + }, + { + .id = SKL_CONTROL_TYPE_MULTI_IO_SELECT_DMIC, + .get = skl_tplg_multi_config_get_dmic, + .put = skl_tplg_multi_config_set_dmic, + } }; static int skl_tplg_fill_pipe_cfg(struct device *dev, @@ -3013,12 +3121,21 @@ static int skl_tplg_control_load(struct snd_soc_component *cmpnt, case SND_SOC_TPLG_CTL_ENUM: tplg_ec = container_of(hdr, struct snd_soc_tplg_enum_control, hdr); - if (kctl->access & SNDRV_CTL_ELEM_ACCESS_READWRITE) { + if (kctl->access & SNDRV_CTL_ELEM_ACCESS_READ) { se = (struct soc_enum *)kctl->private_value; if (tplg_ec->priv.size) - return skl_init_enum_data(bus->dev, se, - tplg_ec); + skl_init_enum_data(bus->dev, se, tplg_ec); } + + /* + * now that the control initializations are done, remove + * write permission for the DMIC configuration enums to + * avoid conflicts between NHLT settings and user interaction + */ + + if (hdr->ops.get == SKL_CONTROL_TYPE_MULTI_IO_SELECT_DMIC) + kctl->access = SNDRV_CTL_ELEM_ACCESS_READ; + break; default: @@ -3488,6 +3605,38 @@ static int skl_manifest_load(struct snd_soc_component *cmpnt, int index, return 0; } +static void skl_tplg_complete(struct snd_soc_component *component) +{ + struct snd_soc_dobj *dobj; + struct snd_soc_acpi_mach *mach = + dev_get_platdata(component->card->dev); + int i; + + list_for_each_entry(dobj, &component->dobj_list, list) { + struct snd_kcontrol *kcontrol = dobj->control.kcontrol; + struct soc_enum *se; + char **texts; + char chan_text[4]; + + if (dobj->type != SND_SOC_DOBJ_ENUM || !kcontrol || + kcontrol->put != skl_tplg_multi_config_set_dmic) + continue; + + se = (struct soc_enum *)kcontrol->private_value; + texts = dobj->control.dtexts; + sprintf(chan_text, "c%d", mach->mach_params.dmic_num); + + for (i = 0; i < se->items; i++) { + struct snd_ctl_elem_value val = {}; + + if (strstr(texts[i], chan_text)) { + val.value.enumerated.item[0] = i; + kcontrol->put(kcontrol, &val); + } + } + } +} + static struct snd_soc_tplg_ops skl_tplg_ops = { .widget_load = skl_tplg_widget_load, .control_load = skl_tplg_control_load, @@ -3497,6 +3646,7 @@ static struct snd_soc_tplg_ops skl_tplg_ops = { .io_ops_count = ARRAY_SIZE(skl_tplg_kcontrol_ops), .manifest = skl_manifest_load, .dai_load = skl_dai_load, + .complete = skl_tplg_complete, }; /* @@ -3565,8 +3715,20 @@ int skl_tplg_init(struct snd_soc_component *component, struct hdac_bus *bus) ret = request_firmware(&fw, skl->tplg_name, bus->dev); if (ret < 0) { - dev_info(bus->dev, "tplg fw %s load failed with %d, falling back to dfw_sst.bin", - skl->tplg_name, ret); + char alt_tplg_name[64]; + + snprintf(alt_tplg_name, sizeof(alt_tplg_name), "%s-tplg.bin", + skl->mach->drv_name); + dev_info(bus->dev, "tplg fw %s load failed with %d, trying alternative tplg name %s", + skl->tplg_name, ret, alt_tplg_name); + + ret = request_firmware(&fw, alt_tplg_name, bus->dev); + if (!ret) + goto component_load; + + dev_info(bus->dev, "tplg %s failed with %d, falling back to dfw_sst.bin", + alt_tplg_name, ret); + ret = request_firmware(&fw, "dfw_sst.bin", bus->dev); if (ret < 0) { dev_err(bus->dev, "Fallback tplg fw %s load failed with %d\n", @@ -3575,6 +3737,8 @@ int skl_tplg_init(struct snd_soc_component *component, struct hdac_bus *bus) } } +component_load: + /* * The complete tplg for SKL is loaded as index 0, we don't use * any other index diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h index e967800dbb62..06576147cc29 100644 --- a/sound/soc/intel/skylake/skl-topology.h +++ b/sound/soc/intel/skylake/skl-topology.h @@ -306,6 +306,7 @@ struct skl_pipe { struct skl_path_config configs[SKL_MAX_PATH_CONFIGS]; struct list_head w_list; bool passthru; + u32 pipe_config_idx; }; enum skl_module_state { diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 141dbbf975ac..2e5fbd220923 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -129,6 +129,7 @@ static int skl_init_chip(struct hdac_bus *bus, bool full_reset) struct hdac_ext_link *hlink; int ret; + snd_hdac_set_codec_wakeup(bus, true); skl_enable_miscbdcge(bus->dev, false); ret = snd_hdac_bus_init_chip(bus, full_reset); @@ -137,6 +138,7 @@ static int skl_init_chip(struct hdac_bus *bus, bool full_reset) writel(0, hlink->ml_addr + AZX_REG_ML_LOSIDV); skl_enable_miscbdcge(bus->dev, true); + snd_hdac_set_codec_wakeup(bus, false); return ret; } @@ -480,13 +482,8 @@ static struct skl_ssp_clk skl_ssp_clks[] = { static struct snd_soc_acpi_mach *skl_find_hda_machine(struct skl_dev *skl, struct snd_soc_acpi_mach *machines) { - struct hdac_bus *bus = skl_to_bus(skl); struct snd_soc_acpi_mach *mach; - /* check if we have any codecs detected on bus */ - if (bus->codec_mask == 0) - return NULL; - /* point to common table */ mach = snd_soc_acpi_intel_hda_machines; @@ -635,6 +632,9 @@ static int skl_clock_device_register(struct skl_dev *skl) struct platform_device_info pdevinfo = {NULL}; struct skl_clk_pdata *clk_pdata; + if (!skl->nhlt) + return 0; + clk_pdata = devm_kzalloc(&skl->pci->dev, sizeof(*clk_pdata), GFP_KERNEL); if (!clk_pdata) @@ -807,6 +807,9 @@ static void skl_probe_work(struct work_struct *work) return; } + skl_init_pci(skl); + skl_dum_set(bus); + err = skl_init_chip(bus, true); if (err < 0) { dev_err(bus->dev, "Init chip failed with err: %d\n", err); @@ -922,8 +925,6 @@ static int skl_first_init(struct hdac_bus *bus) return -ENXIO; } - snd_hdac_bus_reset_link(bus, true); - snd_hdac_bus_parse_capabilities(bus); /* check if PPCAP exists */ @@ -971,11 +972,7 @@ static int skl_first_init(struct hdac_bus *bus) if (err < 0) return err; - /* initialize chip */ - skl_init_pci(skl); - skl_dum_set(bus); - - return skl_init_chip(bus, true); + return 0; } static int skl_probe(struct pci_dev *pci, @@ -1080,8 +1077,6 @@ static int skl_probe(struct pci_dev *pci, if (bus->mlcap) snd_hdac_ext_bus_get_ml_capabilities(bus); - snd_hdac_bus_stop_chip(bus); - /* create device for soc dmic */ err = skl_dmic_device_register(skl); if (err < 0) { @@ -1098,7 +1093,8 @@ out_dsp_free: out_clk_free: skl_clock_device_unregister(skl); out_nhlt_free: - intel_nhlt_free(skl->nhlt); + if (skl->nhlt) + intel_nhlt_free(skl->nhlt); out_free: skl_free(bus); @@ -1147,7 +1143,8 @@ static void skl_remove(struct pci_dev *pci) skl_dmic_device_unregister(skl); skl_clock_device_unregister(skl); skl_nhlt_remove_sysfs(skl); - intel_nhlt_free(skl->nhlt); + if (skl->nhlt) + intel_nhlt_free(skl->nhlt); skl_free(bus); dev_set_drvdata(&pci->dev, NULL); } diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index 0bbd86390be5..9bfd2aabbfe6 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -309,10 +309,14 @@ static int jz4740_i2s_set_sysclk(struct snd_soc_dai *dai, int clk_id, switch (clk_id) { case JZ4740_I2S_CLKSRC_EXT: parent = clk_get(NULL, "ext"); + if (IS_ERR(parent)) + return PTR_ERR(parent); clk_set_parent(i2s->clk_i2s, parent); break; case JZ4740_I2S_CLKSRC_PLL: parent = clk_get(NULL, "pll half"); + if (IS_ERR(parent)) + return PTR_ERR(parent); clk_set_parent(i2s->clk_i2s, parent); ret = clk_set_rate(i2s->clk_i2s, freq); break; diff --git a/sound/soc/mediatek/common/mtk-btcvsd.c b/sound/soc/mediatek/common/mtk-btcvsd.c index d00608c73c6e..b66f7dee1e14 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; @@ -1358,7 +1358,8 @@ 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, @@ -1366,7 +1367,8 @@ 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 */ @@ -1375,7 +1377,7 @@ 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]; @@ -1394,8 +1396,18 @@ 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/mt2701/mt2701-wm8960.c b/sound/soc/mediatek/mt2701/mt2701-wm8960.c index 8c4c89e4c616..b9ad42112ea1 100644 --- a/sound/soc/mediatek/mt2701/mt2701-wm8960.c +++ b/sound/soc/mediatek/mt2701/mt2701-wm8960.c @@ -129,7 +129,8 @@ static int mt2701_wm8960_machine_probe(struct platform_device *pdev) if (!codec_node) { dev_err(&pdev->dev, "Property 'audio-codec' missing or invalid\n"); - return -EINVAL; + ret = -EINVAL; + goto put_platform_node; } for_each_card_prelinks(card, i, dai_link) { if (dai_link->codecs->name) @@ -140,7 +141,7 @@ static int mt2701_wm8960_machine_probe(struct platform_device *pdev) ret = snd_soc_of_parse_audio_routing(card, "audio-routing"); if (ret) { dev_err(&pdev->dev, "failed to parse audio-routing: %d\n", ret); - return ret; + goto put_codec_node; } ret = devm_snd_soc_register_card(&pdev->dev, card); @@ -148,6 +149,10 @@ static int mt2701_wm8960_machine_probe(struct platform_device *pdev) dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", __func__, ret); +put_codec_node: + of_node_put(codec_node); +put_platform_node: + of_node_put(platform_node); return ret; } diff --git a/sound/soc/mediatek/mt8173/mt8173-max98090.c b/sound/soc/mediatek/mt8173/mt8173-max98090.c index 22c00600c999..32df18180114 100644 --- a/sound/soc/mediatek/mt8173/mt8173-max98090.c +++ b/sound/soc/mediatek/mt8173/mt8173-max98090.c @@ -167,7 +167,8 @@ static int mt8173_max98090_dev_probe(struct platform_device *pdev) if (!codec_node) { dev_err(&pdev->dev, "Property 'audio-codec' missing or invalid\n"); - return -EINVAL; + ret = -EINVAL; + goto put_platform_node; } for_each_card_prelinks(card, i, dai_link) { if (dai_link->codecs->name) @@ -180,6 +181,11 @@ 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); + +put_platform_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/Kconfig b/sound/soc/meson/Kconfig index 2e3676147cea..e0d24592ebd7 100644 --- a/sound/soc/meson/Kconfig +++ b/sound/soc/meson/Kconfig @@ -1,6 +1,6 @@ # SPDX-License-Identifier: GPL-2.0-only menu "ASoC support for Amlogic platforms" - depends on ARCH_MESON || COMPILE_TEST + depends on ARCH_MESON || (COMPILE_TEST && COMMON_CLK) config SND_MESON_AXG_FIFO tristate diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c index e25336f73912..f5a431b8de6c 100644 --- a/sound/soc/meson/axg-tdm-interface.c +++ b/sound/soc/meson/axg-tdm-interface.c @@ -467,8 +467,20 @@ static int axg_tdm_iface_set_bias_level(struct snd_soc_component *component, return ret; } +static const struct snd_soc_dapm_widget axg_tdm_iface_dapm_widgets[] = { + SND_SOC_DAPM_SIGGEN("Playback Signal"), +}; + +static const struct snd_soc_dapm_route axg_tdm_iface_dapm_routes[] = { + { "Loopback", NULL, "Playback Signal" }, +}; + static const struct snd_soc_component_driver axg_tdm_iface_component_drv = { - .set_bias_level = axg_tdm_iface_set_bias_level, + .dapm_widgets = axg_tdm_iface_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(axg_tdm_iface_dapm_widgets), + .dapm_routes = axg_tdm_iface_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(axg_tdm_iface_dapm_routes), + .set_bias_level = axg_tdm_iface_set_bias_level, }; static const struct of_device_id axg_tdm_iface_of_match[] = { diff --git a/sound/soc/meson/axg-tdmin.c b/sound/soc/meson/axg-tdmin.c index 88ed95ae886b..b4faf9d5c1aa 100644 --- a/sound/soc/meson/axg-tdmin.c +++ b/sound/soc/meson/axg-tdmin.c @@ -228,15 +228,6 @@ static const struct axg_tdm_formatter_driver axg_tdmin_drv = { .regmap_cfg = &axg_tdmin_regmap_cfg, .ops = &axg_tdmin_ops, .quirks = &(const struct axg_tdm_formatter_hw) { - .skew_offset = 2, - }, -}; - -static const struct axg_tdm_formatter_driver g12a_tdmin_drv = { - .component_drv = &axg_tdmin_component_drv, - .regmap_cfg = &axg_tdmin_regmap_cfg, - .ops = &axg_tdmin_ops, - .quirks = &(const struct axg_tdm_formatter_hw) { .skew_offset = 3, }, }; @@ -247,10 +238,10 @@ static const struct of_device_id axg_tdmin_of_match[] = { .data = &axg_tdmin_drv, }, { .compatible = "amlogic,g12a-tdmin", - .data = &g12a_tdmin_drv, + .data = &axg_tdmin_drv, }, { .compatible = "amlogic,sm1-tdmin", - .data = &g12a_tdmin_drv, + .data = &axg_tdmin_drv, }, {} }; MODULE_DEVICE_TABLE(of, axg_tdmin_of_match); 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 1e38ce858326..c899a05e896f 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) @@ -764,6 +767,7 @@ static int mxs_saif_probe(struct platform_device *pdev) saif->master_id = saif->id; } else { ret = of_alias_get_id(master, "saif"); + of_node_put(master); if (ret < 0) return ret; else 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/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index dbce7e92baf3..c5d6952a4a33 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -174,21 +174,6 @@ static int lpass_cpu_daiops_hw_params(struct snd_pcm_substream *substream, return 0; } -static int lpass_cpu_daiops_hw_free(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); - int ret; - - ret = regmap_write(drvdata->lpaif_map, - LPAIF_I2SCTL_REG(drvdata->variant, dai->driver->id), - 0); - if (ret) - dev_err(dai->dev, "error writing to i2sctl reg: %d\n", ret); - - return ret; -} - static int lpass_cpu_daiops_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -269,7 +254,6 @@ const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops = { .startup = lpass_cpu_daiops_startup, .shutdown = lpass_cpu_daiops_shutdown, .hw_params = lpass_cpu_daiops_hw_params, - .hw_free = lpass_cpu_daiops_hw_free, .prepare = lpass_cpu_daiops_prepare, .trigger = lpass_cpu_daiops_trigger, }; diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index 4c745baa39f7..b1981d84ac18 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -61,7 +61,7 @@ static int lpass_platform_pcmops_open(struct snd_pcm_substream *substream) int ret, dma_ch, dir = substream->stream; struct lpass_pcm_data *data; - data = devm_kzalloc(soc_runtime->dev, sizeof(*data), GFP_KERNEL); + data = kzalloc(sizeof(*data), GFP_KERNEL); if (!data) return -ENOMEM; @@ -73,8 +73,10 @@ static int lpass_platform_pcmops_open(struct snd_pcm_substream *substream) else dma_ch = 0; - if (dma_ch < 0) + if (dma_ch < 0) { + kfree(data); return dma_ch; + } drvdata->substream[dma_ch] = substream; @@ -95,6 +97,7 @@ static int lpass_platform_pcmops_open(struct snd_pcm_substream *substream) ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); if (ret < 0) { + kfree(data); dev_err(soc_runtime->dev, "setting constraints failed: %d\n", ret); return -EINVAL; @@ -119,6 +122,7 @@ static int lpass_platform_pcmops_close(struct snd_pcm_substream *substream) if (v->free_dma_channel) v->free_dma_channel(drvdata, data->dma_ch); + kfree(data); return 0; } diff --git a/sound/soc/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/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index 7e6c41e63d8e..23e1de61e92e 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -16,6 +16,7 @@ #include "qdsp6/q6afe.h" #include "../codecs/rt5663.h" +#define DRIVER_NAME "sdm845" #define DEFAULT_SAMPLE_RATE_48K 48000 #define DEFAULT_MCLK_RATE 24576000 #define TDM_BCLK_RATE 6144000 @@ -407,6 +408,7 @@ static int sdm845_snd_platform_probe(struct platform_device *pdev) goto data_alloc_fail; } + card->driver_name = DRIVER_NAME; card->dapm_widgets = sdm845_snd_widgets; card->num_dapm_widgets = ARRAY_SIZE(sdm845_snd_widgets); card->dev = dev; 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/samsung/tm2_wm5110.c b/sound/soc/samsung/tm2_wm5110.c index bb9910d4cbe2..058847071584 100644 --- a/sound/soc/samsung/tm2_wm5110.c +++ b/sound/soc/samsung/tm2_wm5110.c @@ -553,7 +553,7 @@ static int tm2_probe(struct platform_device *pdev) ret = of_parse_phandle_with_args(dev->of_node, "i2s-controller", cells_name, i, &args); - if (!args.np) { + if (ret) { dev_err(dev, "i2s-controller property parse error: %d\n", i); ret = -EINVAL; goto dai_node_put; diff --git a/sound/soc/sh/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/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index a6c1cf987e6e..df8d7b53b760 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1426,8 +1426,75 @@ static int rsnd_hw_params(struct snd_pcm_substream *substream, } if (io->converted_chan) dev_dbg(dev, "convert channels = %d\n", io->converted_chan); - if (io->converted_rate) + if (io->converted_rate) { + /* + * SRC supports convert rates from params_rate(hw_params)/k_down + * to params_rate(hw_params)*k_up, where k_up is always 6, and + * k_down depends on number of channels and SRC unit. + * So all SRC units can upsample audio up to 6 times regardless + * its number of channels. And all SRC units can downsample + * 2 channel audio up to 6 times too. + */ + int k_up = 6; + int k_down = 6; + int channel; + struct rsnd_mod *src_mod = rsnd_io_to_mod_src(io); + dev_dbg(dev, "convert rate = %d\n", io->converted_rate); + + channel = io->converted_chan ? io->converted_chan : + params_channels(hw_params); + + switch (rsnd_mod_id(src_mod)) { + /* + * SRC0 can downsample 4, 6 and 8 channel audio up to 4 times. + * SRC1, SRC3 and SRC4 can downsample 4 channel audio + * up to 4 times. + * SRC1, SRC3 and SRC4 can downsample 6 and 8 channel audio + * no more than twice. + */ + case 1: + case 3: + case 4: + if (channel > 4) { + k_down = 2; + break; + } + fallthrough; + case 0: + if (channel > 2) + k_down = 4; + break; + + /* Other SRC units do not support more than 2 channels */ + default: + if (channel > 2) + return -EINVAL; + } + + if (params_rate(hw_params) > io->converted_rate * k_down) { + hw_param_interval(hw_params, SNDRV_PCM_HW_PARAM_RATE)->min = + io->converted_rate * k_down; + hw_param_interval(hw_params, SNDRV_PCM_HW_PARAM_RATE)->max = + io->converted_rate * k_down; + hw_params->cmask |= SNDRV_PCM_HW_PARAM_RATE; + } else if (params_rate(hw_params) * k_up < io->converted_rate) { + hw_param_interval(hw_params, SNDRV_PCM_HW_PARAM_RATE)->min = + (io->converted_rate + k_up - 1) / k_up; + hw_param_interval(hw_params, SNDRV_PCM_HW_PARAM_RATE)->max = + (io->converted_rate + k_up - 1) / k_up; + hw_params->cmask |= SNDRV_PCM_HW_PARAM_RATE; + } + + /* + * TBD: Max SRC input and output rates also depend on number + * of channels and SRC unit: + * SRC1, SRC3 and SRC4 do not support more than 128kHz + * for 6 channel and 96kHz for 8 channel audio. + * Perhaps this function should return EINVAL if the input or + * the output rate exceeds the limitation. + */ + } } ret = rsnd_dai_call(hw_params, io, substream, hw_params); diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 47d5ddb526f2..09af402ca31f 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -507,10 +507,15 @@ static int rsnd_ssi_init(struct rsnd_mod *mod, struct rsnd_priv *priv) { struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); + int ret; if (!rsnd_ssi_is_run_mods(mod, io)) return 0; + ret = rsnd_ssi_master_clk_start(mod, io); + if (ret < 0) + return ret; + ssi->usrcnt++; rsnd_mod_power_on(mod); @@ -792,7 +797,6 @@ static void __rsnd_ssi_interrupt(struct rsnd_mod *mod, SSI_SYS_STATUS(i * 2), 0xf << (id * 4)); stop = true; - break; } } break; @@ -810,7 +814,6 @@ static void __rsnd_ssi_interrupt(struct rsnd_mod *mod, SSI_SYS_STATUS((i * 2) + 1), 0xf << 4); stop = true; - break; } } break; @@ -1060,13 +1063,6 @@ static int rsnd_ssi_pio_pointer(struct rsnd_mod *mod, return 0; } -static int rsnd_ssi_prepare(struct rsnd_mod *mod, - struct rsnd_dai_stream *io, - struct rsnd_priv *priv) -{ - return rsnd_ssi_master_clk_start(mod, io); -} - static struct rsnd_mod_ops rsnd_ssi_pio_ops = { .name = SSI_NAME, .probe = rsnd_ssi_common_probe, @@ -1079,7 +1075,6 @@ static struct rsnd_mod_ops rsnd_ssi_pio_ops = { .pointer = rsnd_ssi_pio_pointer, .pcm_new = rsnd_ssi_pcm_new, .hw_params = rsnd_ssi_hw_params, - .prepare = rsnd_ssi_prepare, .get_status = rsnd_ssi_get_status, }; @@ -1166,7 +1161,6 @@ static struct rsnd_mod_ops rsnd_ssi_dma_ops = { .pcm_new = rsnd_ssi_pcm_new, .fallback = rsnd_ssi_fallback, .hw_params = rsnd_ssi_hw_params, - .prepare = rsnd_ssi_prepare, .get_status = rsnd_ssi_get_status, }; diff --git a/sound/soc/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 9df20768a8f2..093ab32ea2c3 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3178,7 +3178,7 @@ 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++) { @@ -3362,7 +3362,7 @@ 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 06aa39379749..56c9c4189f26 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, @@ -2484,6 +2482,7 @@ void snd_soc_dapm_free_widget(struct snd_soc_dapm_widget *w) enum snd_soc_dapm_direction dir; list_del(&w->list); + list_del(&w->dirty); /* * remove source and sink paths associated to this widget. * While removing the path, remove reference to it from both @@ -2541,10 +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); @@ -2557,13 +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; } /** @@ -3403,7 +3421,6 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, update.val = val; card->update = &update; } - change |= reg_change; ret = soc_dapm_mixer_update_power(card, kcontrol, connect, rconnect); @@ -3509,7 +3526,6 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, update.val = val; card->update = &update; } - change |= reg_change; ret = soc_dapm_mux_update_power(card, kcontrol, item[0], e); @@ -3579,14 +3595,15 @@ 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); @@ -4028,7 +4045,7 @@ 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..7a37312c8e0c 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 +523,15 @@ 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, tmp; + + tmp = ucontrol->value.integer.value[0]; + if (tmp < 0) + return -EINVAL; + if (mc->platform_max && tmp > mc->platform_max) + return -EINVAL; + if (tmp > mc->max - mc->min + 1) + return -EINVAL; if (invert) val = (max - ucontrol->value.integer.value[0]) & mask; @@ -505,11 +540,20 @@ 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)) { + tmp = ucontrol->value.integer.value[1]; + if (tmp < 0) + return -EINVAL; + if (mc->platform_max && tmp > mc->platform_max) + return -EINVAL; + if (tmp > mc->max - mc->min + 1) + return -EINVAL; + if (invert) val = (max - ucontrol->value.integer.value[1]) & mask; else @@ -517,8 +561,12 @@ 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 +937,8 @@ 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-pcm.c b/sound/soc/soc-pcm.c index cc4e9aa80fb0..1196167364d4 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2346,6 +2346,7 @@ static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_DRAIN: ret = dpcm_dai_trigger_fe_be(substream, cmd, true); break; case SNDRV_PCM_TRIGGER_STOP: @@ -2363,6 +2364,7 @@ static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_DRAIN: ret = dpcm_dai_trigger_fe_be(substream, cmd, false); break; case SNDRV_PCM_TRIGGER_STOP: diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 0100f123484e..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; @@ -982,7 +983,7 @@ static int soc_tplg_denum_create_values(struct soc_enum *se, return -EINVAL; se->dobj.control.dvalues = kzalloc(le32_to_cpu(ec->items) * - sizeof(u32), + sizeof(*se->dobj.control.dvalues), GFP_KERNEL); if (!se->dobj.control.dvalues) return -ENOMEM; @@ -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/debug.c b/sound/soc/sof/debug.c index 5529e8eeca46..08726034ff09 100644 --- a/sound/soc/sof/debug.c +++ b/sound/soc/sof/debug.c @@ -135,7 +135,7 @@ static ssize_t sof_dfsentry_write(struct file *file, const char __user *buffer, char *string; int ret; - string = kzalloc(count, GFP_KERNEL); + string = kzalloc(count+1, GFP_KERNEL); if (!string) return -ENOMEM; diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index 9e8233c10d86..df38616c431a 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -68,8 +68,7 @@ void hda_codec_jack_check(struct snd_sof_dev *sdev) * has been recorded in STATESTS */ if (codec->jacktbl.used) - schedule_delayed_work(&codec->jackpoll_work, - codec->jackpoll_interval); + pm_request_resume(&codec->core.dev); } #else void hda_codec_jack_wake_enable(struct snd_sof_dev *sdev) {} 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-dsp.c b/sound/soc/sof/intel/hda-dsp.c index fb55a3c5afd0..06715b3d8c31 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -179,7 +179,7 @@ int hda_dsp_core_power_down(struct snd_sof_dev *sdev, unsigned int core_mask) return snd_sof_dsp_read_poll_timeout(sdev, HDA_DSP_BAR, HDA_DSP_REG_ADSPCS, adspcs, - !(adspcs & HDA_DSP_ADSPCS_SPA_MASK(core_mask)), + !(adspcs & HDA_DSP_ADSPCS_CPA_MASK(core_mask)), HDA_DSP_REG_POLL_INTERVAL_US, HDA_DSP_PD_TIMEOUT * USEC_PER_MSEC); } @@ -192,10 +192,17 @@ bool hda_dsp_core_is_enabled(struct snd_sof_dev *sdev, val = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_ADSPCS); - is_enable = ((val & HDA_DSP_ADSPCS_CPA_MASK(core_mask)) && - (val & HDA_DSP_ADSPCS_SPA_MASK(core_mask)) && - !(val & HDA_DSP_ADSPCS_CRST_MASK(core_mask)) && - !(val & HDA_DSP_ADSPCS_CSTALL_MASK(core_mask))); +#define MASK_IS_EQUAL(v, m, field) ({ \ + u32 _m = field(m); \ + ((v) & _m) == _m; \ +}) + + is_enable = MASK_IS_EQUAL(val, core_mask, HDA_DSP_ADSPCS_CPA_MASK) && + MASK_IS_EQUAL(val, core_mask, HDA_DSP_ADSPCS_SPA_MASK) && + !(val & HDA_DSP_ADSPCS_CRST_MASK(core_mask)) && + !(val & HDA_DSP_ADSPCS_CSTALL_MASK(core_mask)); + +#undef MASK_IS_EQUAL dev_dbg(sdev->dev, "DSP core(s) enabled? %d : core_mask %x\n", is_enable, core_mask); diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c index 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/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 3c4b604412f0..a10166741865 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -672,6 +672,7 @@ free_streams: /* dsp_unmap: not currently used */ iounmap(sdev->bar[HDA_DSP_BAR]); hdac_bus_unmap: + platform_device_unregister(hdev->dmic_dev); iounmap(bus->remap_addr); err: return ret; diff --git a/sound/soc/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/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index ee448d5e07a6..c4021d6ac9df 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -1364,6 +1364,7 @@ static struct snd_soc_card *sun4i_codec_create_card(struct device *dev) return ERR_PTR(-ENOMEM); card->dev = dev; + card->owner = THIS_MODULE; card->name = "sun4i-codec"; card->dapm_widgets = sun4i_codec_card_dapm_widgets; card->num_dapm_widgets = ARRAY_SIZE(sun4i_codec_card_dapm_widgets); @@ -1396,6 +1397,7 @@ static struct snd_soc_card *sun6i_codec_create_card(struct device *dev) return ERR_PTR(-ENOMEM); card->dev = dev; + card->owner = THIS_MODULE; card->name = "A31 Audio Codec"; card->dapm_widgets = sun6i_codec_card_dapm_widgets; card->num_dapm_widgets = ARRAY_SIZE(sun6i_codec_card_dapm_widgets); @@ -1449,6 +1451,7 @@ static struct snd_soc_card *sun8i_a23_codec_create_card(struct device *dev) return ERR_PTR(-ENOMEM); card->dev = dev; + card->owner = THIS_MODULE; card->name = "A23 Audio Codec"; card->dapm_widgets = sun6i_codec_card_dapm_widgets; card->num_dapm_widgets = ARRAY_SIZE(sun6i_codec_card_dapm_widgets); @@ -1487,6 +1490,7 @@ static struct snd_soc_card *sun8i_h3_codec_create_card(struct device *dev) return ERR_PTR(-ENOMEM); card->dev = dev; + card->owner = THIS_MODULE; card->name = "H3 Audio Codec"; card->dapm_widgets = sun6i_codec_card_dapm_widgets; card->num_dapm_widgets = ARRAY_SIZE(sun6i_codec_card_dapm_widgets); @@ -1525,6 +1529,7 @@ static struct snd_soc_card *sun8i_v3s_codec_create_card(struct device *dev) return ERR_PTR(-ENOMEM); card->dev = dev; + card->owner = THIS_MODULE; card->name = "V3s Audio Codec"; card->dapm_widgets = sun6i_codec_card_dapm_widgets; card->num_dapm_widgets = ARRAY_SIZE(sun6i_codec_card_dapm_widgets); diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index d0a8d5810c0a..9655dec4166b 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -442,11 +442,11 @@ static int sun8i_i2s_set_chan_cfg(const struct sun4i_i2s *i2s, switch (i2s->format & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_A: case SND_SOC_DAIFMT_DSP_B: - case SND_SOC_DAIFMT_LEFT_J: - case SND_SOC_DAIFMT_RIGHT_J: lrck_period = params_physical_width(params) * slots; break; + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_RIGHT_J: case SND_SOC_DAIFMT_I2S: lrck_period = params_physical_width(params); break; diff --git a/sound/soc/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 48970efe7838..4caee57114b8 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/card.c b/sound/usb/card.c index 230d862cfa3a..c2dd18c5cadb 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -176,9 +176,8 @@ static int snd_usb_create_stream(struct snd_usb_audio *chip, int ctrlif, int int ctrlif, interface); return -EINVAL; } - usb_driver_claim_interface(&usb_audio_driver, iface, (void *)-1L); - - return 0; + return usb_driver_claim_interface(&usb_audio_driver, iface, + USB_AUDIO_IFACE_UNUSED); } if ((altsd->bInterfaceClass != USB_CLASS_AUDIO && @@ -198,7 +197,8 @@ static int snd_usb_create_stream(struct snd_usb_audio *chip, int ctrlif, int int if (! snd_usb_parse_audio_interface(chip, interface)) { usb_set_interface(dev, interface, 0); /* reset the current interface */ - usb_driver_claim_interface(&usb_audio_driver, iface, (void *)-1L); + return usb_driver_claim_interface(&usb_audio_driver, iface, + USB_AUDIO_IFACE_UNUSED); } return 0; @@ -703,7 +703,7 @@ static void usb_audio_disconnect(struct usb_interface *intf) struct snd_card *card; struct list_head *p; - if (chip == (void *)-1L) + if (chip == USB_AUDIO_IFACE_UNUSED) return; card = chip->card; @@ -811,7 +811,7 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message) struct usb_mixer_interface *mixer; struct list_head *p; - if (chip == (void *)-1L) + if (chip == USB_AUDIO_IFACE_UNUSED) return 0; if (!chip->num_suspended_intf++) { @@ -842,7 +842,7 @@ static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume) struct list_head *p; int err = 0; - if (chip == (void *)-1L) + if (chip == USB_AUDIO_IFACE_UNUSED) return 0; atomic_inc(&chip->active); /* avoid autopm */ diff --git a/sound/usb/clock.c b/sound/usb/clock.c index b118cf97607f..3d1c0ec11753 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -296,7 +296,7 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, selector = snd_usb_find_clock_selector(chip->ctrl_intf, entity_id); if (selector) { - int ret, i, cur; + int ret, i, cur, err; /* the entity ID we are looking for is a selector. * find out what it currently selects */ @@ -318,13 +318,23 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, ret = __uac_clock_find_source(chip, fmt, selector->baCSourceID[ret - 1], visited, validate); + if (ret > 0) { + /* + * For Samsung USBC Headset (AKG), setting clock selector again + * will result in incorrect default clock setting problems + */ + if (chip->usb_id == USB_ID(0x04e8, 0xa051)) + return ret; + err = uac_clock_selector_set_val(chip, entity_id, cur); + if (err < 0) + return err; + } + if (!validate || ret > 0 || !chip->autoclock) return ret; /* The current clock source is invalid, try others. */ for (i = 1; i <= selector->bNrInPins; i++) { - int err; - if (i == cur) continue; @@ -390,7 +400,7 @@ static int __uac3_clock_find_source(struct snd_usb_audio *chip, selector = snd_usb_find_clock_selector_v3(chip->ctrl_intf, entity_id); if (selector) { - int ret, i, cur; + int ret, i, cur, err; /* the entity ID we are looking for is a selector. * find out what it currently selects */ @@ -412,6 +422,12 @@ static int __uac3_clock_find_source(struct snd_usb_audio *chip, ret = __uac3_clock_find_source(chip, fmt, selector->baCSourceID[ret - 1], visited, validate); + if (ret > 0) { + err = uac_clock_selector_set_val(chip, entity_id, cur); + if (err < 0) + return err; + } + if (!validate || ret > 0 || !chip->autoclock) return ret; @@ -531,6 +547,12 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface, } crate = data[0] | (data[1] << 8) | (data[2] << 16); + if (!crate) { + dev_info(&dev->dev, "failed to read current rate; disabling the check\n"); + chip->sample_rate_read_error = 3; /* three strikes, see above */ + return 0; + } + if (crate != rate) { dev_warn(&dev->dev, "current rate %d is different from the runtime rate %d\n", crate, rate); // runtime->rate = crate; diff --git a/sound/usb/format.c b/sound/usb/format.c index 1f9ea513230a..84b66f7c627c 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -40,6 +40,8 @@ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip, case UAC_VERSION_1: default: { struct uac_format_type_i_discrete_descriptor *fmt = _fmt; + if (format >= 64) + return 0; /* invalid format */ sample_width = fmt->bBitResolution; sample_bytes = fmt->bSubframeSize; format = 1ULL << format; @@ -193,9 +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 0caf53f5764c..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, @@ -690,6 +690,10 @@ static int line6_init_cap_control(struct usb_line6 *line6) line6->buffer_message = kmalloc(LINE6_MIDI_MESSAGE_MAXLEN, GFP_KERNEL); if (!line6->buffer_message) return -ENOMEM; + + ret = line6_init_midi(line6); + if (ret < 0) + return ret; } else { ret = line6_hwdep_init(line6); if (ret < 0) diff --git a/sound/usb/line6/driver.h b/sound/usb/line6/driver.h index 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/pod.c b/sound/usb/line6/pod.c index ee4c9d220fdf..b667bb8ad938 100644 --- a/sound/usb/line6/pod.c +++ b/sound/usb/line6/pod.c @@ -376,11 +376,6 @@ static int pod_init(struct usb_line6 *line6, if (err < 0) return err; - /* initialize MIDI subsystem: */ - err = line6_init_midi(line6); - if (err < 0) - return err; - /* initialize PCM subsystem: */ err = line6_init_pcm(line6, &pod_pcm_properties); if (err < 0) diff --git a/sound/usb/line6/podhd.c b/sound/usb/line6/podhd.c index 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/line6/variax.c b/sound/usb/line6/variax.c index ed158f04de80..c2245aa93b08 100644 --- a/sound/usb/line6/variax.c +++ b/sound/usb/line6/variax.c @@ -159,7 +159,6 @@ static int variax_init(struct usb_line6 *line6, const struct usb_device_id *id) { struct usb_line6_variax *variax = line6_to_variax(line6); - int err; line6->process_message = line6_variax_process_message; line6->disconnect = line6_variax_disconnect; @@ -172,11 +171,6 @@ static int variax_init(struct usb_line6 *line6, if (variax->buffer_activate == NULL) return -ENOMEM; - /* initialize MIDI subsystem: */ - err = line6_init_midi(&variax->line6); - if (err < 0) - return err; - /* initiate startup procedure: */ schedule_delayed_work(&line6->startup_work, msecs_to_jiffies(VARIAX_STARTUP_DELAY1)); diff --git a/sound/usb/midi.c b/sound/usb/midi.c index bc9068b616bb..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); } @@ -1332,7 +1333,7 @@ static int snd_usbmidi_in_endpoint_create(struct snd_usb_midi *umidi, error: snd_usbmidi_in_endpoint_delete(ep); - return -ENOMEM; + return err; } /* @@ -1889,6 +1890,12 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi *umidi, ms_ep = find_usb_ms_endpoint_descriptor(hostep); if (!ms_ep) continue; + if (ms_ep->bLength <= sizeof(*ms_ep)) + continue; + if (ms_ep->bNumEmbMIDIJack > 0x10) + continue; + if (ms_ep->bLength < sizeof(*ms_ep) + ms_ep->bNumEmbMIDIJack) + continue; if (usb_endpoint_dir_out(ep)) { if (endpoints[epidx].out_ep) { if (++epidx >= MIDI_MAX_ENDPOINTS) { @@ -2141,6 +2148,8 @@ static int snd_usbmidi_detect_roland(struct snd_usb_midi *umidi, cs_desc[1] == USB_DT_CS_INTERFACE && cs_desc[2] == 0xf1 && cs_desc[3] == 0x02) { + if (cs_desc[4] > 0x10 || cs_desc[5] > 0x10) + continue; endpoint->in_cables = (1 << cs_desc[4]) - 1; endpoint->out_cables = (1 << cs_desc[5]) - 1; return snd_usbmidi_detect_endpoints(umidi, endpoint, 1); 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 8aa96ed0b1b5..67eb1293fa15 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1030,7 +1030,7 @@ struct usb_feature_control_info { int type_uac2; /* data type for uac2 if different from uac1, else -1 */ }; -static struct usb_feature_control_info audio_feature_info[] = { +static const struct usb_feature_control_info audio_feature_info[] = { { UAC_FU_MUTE, "Mute", USB_MIXER_INV_BOOLEAN, -1 }, { UAC_FU_VOLUME, "Volume", USB_MIXER_S16, -1 }, { UAC_FU_BASS, "Tone Control - Bass", USB_MIXER_S8, -1 }, @@ -1544,7 +1544,7 @@ static void check_no_speaker_on_headset(struct snd_kcontrol *kctl, strlcpy(kctl->id.name, "Headphone", sizeof(kctl->id.name)); } -static struct usb_feature_control_info *get_feature_control_info(int control) +static const struct usb_feature_control_info *get_feature_control_info(int control) { int i; @@ -1562,7 +1562,7 @@ static void __build_feature_ctl(struct usb_mixer_interface *mixer, struct usb_audio_term *oterm, int unitid, int nameid, int readonly_mask) { - struct usb_feature_control_info *ctl_info; + const struct usb_feature_control_info *ctl_info; unsigned int len = 0; int mapped_name = 0; struct snd_kcontrol *kctl; @@ -2237,7 +2237,7 @@ static const struct snd_kcontrol_new mixer_procunit_ctl = { */ struct procunit_value_info { int control; - char *suffix; + const char *suffix; int val_type; int min_value; }; @@ -2245,44 +2245,44 @@ struct procunit_value_info { struct procunit_info { int type; char *name; - struct procunit_value_info *values; + const struct procunit_value_info *values; }; -static struct procunit_value_info undefined_proc_info[] = { +static const struct procunit_value_info undefined_proc_info[] = { { 0x00, "Control Undefined", 0 }, { 0 } }; -static struct procunit_value_info updown_proc_info[] = { +static const struct procunit_value_info updown_proc_info[] = { { UAC_UD_ENABLE, "Switch", USB_MIXER_BOOLEAN }, { UAC_UD_MODE_SELECT, "Mode Select", USB_MIXER_U8, 1 }, { 0 } }; -static struct procunit_value_info prologic_proc_info[] = { +static const struct procunit_value_info prologic_proc_info[] = { { UAC_DP_ENABLE, "Switch", USB_MIXER_BOOLEAN }, { UAC_DP_MODE_SELECT, "Mode Select", USB_MIXER_U8, 1 }, { 0 } }; -static struct procunit_value_info threed_enh_proc_info[] = { +static const struct procunit_value_info threed_enh_proc_info[] = { { UAC_3D_ENABLE, "Switch", USB_MIXER_BOOLEAN }, { UAC_3D_SPACE, "Spaciousness", USB_MIXER_U8 }, { 0 } }; -static struct procunit_value_info reverb_proc_info[] = { +static const struct procunit_value_info reverb_proc_info[] = { { UAC_REVERB_ENABLE, "Switch", USB_MIXER_BOOLEAN }, { UAC_REVERB_LEVEL, "Level", USB_MIXER_U8 }, { UAC_REVERB_TIME, "Time", USB_MIXER_U16 }, { UAC_REVERB_FEEDBACK, "Feedback", USB_MIXER_U8 }, { 0 } }; -static struct procunit_value_info chorus_proc_info[] = { +static const struct procunit_value_info chorus_proc_info[] = { { UAC_CHORUS_ENABLE, "Switch", USB_MIXER_BOOLEAN }, { UAC_CHORUS_LEVEL, "Level", USB_MIXER_U8 }, { UAC_CHORUS_RATE, "Rate", USB_MIXER_U16 }, { UAC_CHORUS_DEPTH, "Depth", USB_MIXER_U16 }, { 0 } }; -static struct procunit_value_info dcr_proc_info[] = { +static const struct procunit_value_info dcr_proc_info[] = { { UAC_DCR_ENABLE, "Switch", USB_MIXER_BOOLEAN }, { UAC_DCR_RATE, "Ratio", USB_MIXER_U16 }, { UAC_DCR_MAXAMPL, "Max Amp", USB_MIXER_S16 }, @@ -2292,7 +2292,7 @@ static struct procunit_value_info dcr_proc_info[] = { { 0 } }; -static struct procunit_info procunits[] = { +static const struct procunit_info procunits[] = { { UAC_PROCESS_UP_DOWNMIX, "Up Down", updown_proc_info }, { UAC_PROCESS_DOLBY_PROLOGIC, "Dolby Prologic", prologic_proc_info }, { UAC_PROCESS_STEREO_EXTENDER, "3D Stereo Extender", threed_enh_proc_info }, @@ -2302,16 +2302,16 @@ static struct procunit_info procunits[] = { { 0 }, }; -static struct procunit_value_info uac3_updown_proc_info[] = { +static const struct procunit_value_info uac3_updown_proc_info[] = { { UAC3_UD_MODE_SELECT, "Mode Select", USB_MIXER_U8, 1 }, { 0 } }; -static struct procunit_value_info uac3_stereo_ext_proc_info[] = { +static const struct procunit_value_info uac3_stereo_ext_proc_info[] = { { UAC3_EXT_WIDTH_CONTROL, "Width Control", USB_MIXER_U8 }, { 0 } }; -static struct procunit_info uac3_procunits[] = { +static const struct procunit_info uac3_procunits[] = { { UAC3_PROCESS_UP_DOWNMIX, "Up Down", uac3_updown_proc_info }, { UAC3_PROCESS_STEREO_EXTENDER, "3D Stereo Extender", uac3_stereo_ext_proc_info }, { UAC3_PROCESS_MULTI_FUNCTION, "Multi-Function", undefined_proc_info }, @@ -2321,23 +2321,23 @@ static struct procunit_info uac3_procunits[] = { /* * predefined data for extension units */ -static struct procunit_value_info clock_rate_xu_info[] = { +static const struct procunit_value_info clock_rate_xu_info[] = { { USB_XU_CLOCK_RATE_SELECTOR, "Selector", USB_MIXER_U8, 0 }, { 0 } }; -static struct procunit_value_info clock_source_xu_info[] = { +static const struct procunit_value_info clock_source_xu_info[] = { { USB_XU_CLOCK_SOURCE_SELECTOR, "External", USB_MIXER_BOOLEAN }, { 0 } }; -static struct procunit_value_info spdif_format_xu_info[] = { +static const struct procunit_value_info spdif_format_xu_info[] = { { USB_XU_DIGITAL_FORMAT_SELECTOR, "SPDIF/AC3", USB_MIXER_BOOLEAN }, { 0 } }; -static struct procunit_value_info soft_limit_xu_info[] = { +static const struct procunit_value_info soft_limit_xu_info[] = { { USB_XU_SOFT_LIMIT_SELECTOR, " ", USB_MIXER_BOOLEAN }, { 0 } }; -static struct procunit_info extunits[] = { +static const struct procunit_info extunits[] = { { USB_XU_CLOCK_RATE, "Clock rate", clock_rate_xu_info }, { USB_XU_CLOCK_SOURCE, "DigitalIn CLK source", clock_source_xu_info }, { USB_XU_DIGITAL_IO_STATUS, "DigitalOut format:", spdif_format_xu_info }, @@ -2349,7 +2349,7 @@ static struct procunit_info extunits[] = { * build a processing/extension unit */ static int build_audio_procunit(struct mixer_build *state, int unitid, - void *raw_desc, struct procunit_info *list, + void *raw_desc, const struct procunit_info *list, bool extension_unit) { struct uac_processing_unit_descriptor *desc = raw_desc; @@ -2357,14 +2357,14 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, struct usb_mixer_elem_info *cval; struct snd_kcontrol *kctl; int i, err, nameid, type, len; - struct procunit_info *info; - struct procunit_value_info *valinfo; + const struct procunit_info *info; + const struct procunit_value_info *valinfo; const struct usbmix_name_map *map; - static struct procunit_value_info default_value_info[] = { + static const struct procunit_value_info default_value_info[] = { { 0x01, "Switch", USB_MIXER_BOOLEAN }, { 0 } }; - static struct procunit_info default_info = { + static const struct procunit_info default_info = { 0, NULL, default_value_info }; const char *name = extension_unit ? @@ -2842,7 +2842,7 @@ struct uac3_badd_profile { int st_chmask; /* side tone mixing channel mask */ }; -static struct uac3_badd_profile uac3_badd_profiles[] = { +static const struct uac3_badd_profile uac3_badd_profiles[] = { { /* * BAIF, BAOF or combination of both @@ -2903,7 +2903,7 @@ static struct uac3_badd_profile uac3_badd_profiles[] = { }; static bool uac3_badd_func_has_valid_channels(struct usb_mixer_interface *mixer, - struct uac3_badd_profile *f, + const struct uac3_badd_profile *f, int c_chmask, int p_chmask) { /* @@ -2947,7 +2947,7 @@ static int snd_usb_mixer_controls_badd(struct usb_mixer_interface *mixer, struct usb_device *dev = mixer->chip->dev; struct usb_interface_assoc_descriptor *assoc; int badd_profile = mixer->chip->badd_profile; - struct uac3_badd_profile *f; + const struct uac3_badd_profile *f; const struct usbmix_ctl_map *map; int p_chmask = 0, c_chmask = 0, st_chmask = 0; int i; @@ -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 char *val_types[] = {"BOOLEAN", "INV_BOOLEAN", - "S8", "U8", "S16", "U16"}; + static const char * const val_types[] = { + [USB_MIXER_BOOLEAN] = "BOOLEAN", + [USB_MIXER_INV_BOOLEAN] = "INV_BOOLEAN", + [USB_MIXER_S8] = "S8", + [USB_MIXER_U8] = "U8", + [USB_MIXER_S16] = "S16", + [USB_MIXER_U16] = "U16", + [USB_MIXER_S32] = "S32", + [USB_MIXER_U32] = "U32", + [USB_MIXER_BESPOKEN] = "BESPOKEN", + }; snd_iprintf(buffer, " Info: id=%i, control=%i, cmask=0x%x, " "channels=%i, type=\"%s\"\n", cval->head.id, cval->control, cval->cmask, cval->channels, @@ -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_maps.c b/sound/usb/mixer_maps.c index ac84f0b2b0bc..dda13624a28c 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -14,7 +14,7 @@ struct usbmix_name_map { int id; const char *name; int control; - struct usbmix_dB_map *dB; + const struct usbmix_dB_map *dB; }; struct usbmix_selector_map { @@ -53,7 +53,7 @@ Mic-IN[9] --+->FU[10]----------------------------+ | ++--+->SU[11]-->FU[12] --------------------------------------------------------------------------------------> USB_OUT[13] */ -static struct usbmix_name_map extigy_map[] = { +static const struct usbmix_name_map extigy_map[] = { /* 1: IT pcm */ { 2, "PCM Playback" }, /* FU */ /* 3: IT pcm */ @@ -94,12 +94,12 @@ static struct usbmix_name_map extigy_map[] = { * e.g. no Master and fake PCM volume * Pavel Mihaylov <bin@bash.info> */ -static struct usbmix_dB_map mp3plus_dB_1 = {.min = -4781, .max = 0}; +static const struct usbmix_dB_map mp3plus_dB_1 = {.min = -4781, .max = 0}; /* just guess */ -static struct usbmix_dB_map mp3plus_dB_2 = {.min = -1781, .max = 618}; +static const struct usbmix_dB_map mp3plus_dB_2 = {.min = -1781, .max = 618}; /* just guess */ -static struct usbmix_name_map mp3plus_map[] = { +static const struct usbmix_name_map mp3plus_map[] = { /* 1: IT pcm */ /* 2: IT mic */ /* 3: IT line */ @@ -140,7 +140,7 @@ Lin_IN[7]-+--->FU[8]---+ +->EU[23]->FU[28]------------->Spk_OUT[19] | ^ +->FU[13]--------------------------------------+ */ -static struct usbmix_name_map audigy2nx_map[] = { +static const struct usbmix_name_map audigy2nx_map[] = { /* 1: IT pcm playback */ /* 4: IT digital in */ { 6, "Digital In Playback" }, /* FU */ @@ -168,12 +168,12 @@ static struct usbmix_name_map audigy2nx_map[] = { { 0 } /* terminator */ }; -static struct usbmix_name_map mbox1_map[] = { +static const struct usbmix_name_map mbox1_map[] = { { 1, "Clock" }, { 0 } /* terminator */ }; -static struct usbmix_selector_map c400_selectors[] = { +static const struct usbmix_selector_map c400_selectors[] = { { .id = 0x80, .count = 2, @@ -182,7 +182,7 @@ static struct usbmix_selector_map c400_selectors[] = { { 0 } /* terminator */ }; -static struct usbmix_selector_map audigy2nx_selectors[] = { +static const struct usbmix_selector_map audigy2nx_selectors[] = { { .id = 14, /* Capture Source */ .count = 3, @@ -202,21 +202,21 @@ static struct usbmix_selector_map audigy2nx_selectors[] = { }; /* Creative SoundBlaster Live! 24-bit External */ -static struct usbmix_name_map live24ext_map[] = { +static const struct usbmix_name_map live24ext_map[] = { /* 2: PCM Playback Volume */ { 5, "Mic Capture" }, /* FU, default PCM Capture Volume */ { 0 } /* terminator */ }; /* LineX FM Transmitter entry - needed to bypass controls bug */ -static struct usbmix_name_map linex_map[] = { +static const struct usbmix_name_map linex_map[] = { /* 1: IT pcm */ /* 2: OT Speaker */ { 3, "Master" }, /* FU: master volume - left / right / mute */ { 0 } /* terminator */ }; -static struct usbmix_name_map maya44_map[] = { +static const struct usbmix_name_map maya44_map[] = { /* 1: IT line */ { 2, "Line Playback" }, /* FU */ /* 3: IT line */ @@ -239,7 +239,7 @@ static struct usbmix_name_map maya44_map[] = { * so this map removes all unwanted sliders from alsamixer */ -static struct usbmix_name_map justlink_map[] = { +static const struct usbmix_name_map justlink_map[] = { /* 1: IT pcm playback */ /* 2: Not present */ { 3, NULL}, /* IT mic (No mic input on device) */ @@ -256,7 +256,7 @@ static struct usbmix_name_map justlink_map[] = { }; /* TerraTec Aureon 5.1 MkII USB */ -static struct usbmix_name_map aureon_51_2_map[] = { +static const struct usbmix_name_map aureon_51_2_map[] = { /* 1: IT USB */ /* 2: IT Mic */ /* 3: IT Line */ @@ -275,7 +275,7 @@ static struct usbmix_name_map aureon_51_2_map[] = { {} /* terminator */ }; -static struct usbmix_name_map scratch_live_map[] = { +static const struct usbmix_name_map scratch_live_map[] = { /* 1: IT Line 1 (USB streaming) */ /* 2: OT Line 1 (Speaker) */ /* 3: IT Line 1 (Line connector) */ @@ -291,7 +291,7 @@ static struct usbmix_name_map scratch_live_map[] = { { 0 } /* terminator */ }; -static struct usbmix_name_map ebox44_map[] = { +static const struct usbmix_name_map ebox44_map[] = { { 4, NULL }, /* FU */ { 6, NULL }, /* MU */ { 7, NULL }, /* FU */ @@ -306,7 +306,7 @@ static struct usbmix_name_map ebox44_map[] = { * FIXME: or mp3plus_map should use "Capture Source" too, * so this maps can be merget */ -static struct usbmix_name_map hercules_usb51_map[] = { +static const struct usbmix_name_map hercules_usb51_map[] = { { 8, "Capture Source" }, /* SU, default "PCM Capture Source" */ { 9, "Master Playback" }, /* FU, default "Speaker Playback" */ { 10, "Mic Boost", 7 }, /* FU, default "Auto Gain Input" */ @@ -317,7 +317,7 @@ static struct usbmix_name_map hercules_usb51_map[] = { }; /* Plantronics Gamecom 780 has a broken volume control, better to disable it */ -static struct usbmix_name_map gamecom780_map[] = { +static const struct usbmix_name_map gamecom780_map[] = { { 9, NULL }, /* FU, speaker out */ {} }; @@ -331,12 +331,19 @@ static const struct usbmix_name_map scms_usb3318_map[] = { }; /* Bose companion 5, the dB conversion factor is 16 instead of 256 */ -static struct usbmix_dB_map bose_companion5_dB = {-5006, -6}; -static struct usbmix_name_map bose_companion5_map[] = { +static const struct usbmix_dB_map bose_companion5_dB = {-5006, -6}; +static const struct usbmix_name_map bose_companion5_map[] = { { 3, NULL, .dB = &bose_companion5_dB }, { 0 } /* terminator */ }; +/* Sennheiser Communications Headset [PC 8], the dB value is reported as -6 negative maximum */ +static const struct usbmix_dB_map sennheiser_pc8_dB = {-9500, 0}; +static const struct usbmix_name_map sennheiser_pc8_map[] = { + { 9, NULL, .dB = &sennheiser_pc8_dB }, + { 0 } /* terminator */ +}; + /* * Dell usb dock with ALC4020 codec had a firmware problem where it got * screwed up when zero volume is passed; just skip it as a workaround @@ -406,7 +413,7 @@ static const struct usbmix_name_map aorus_master_alc1220vb_map[] = { * Control map entries */ -static struct usbmix_ctl_map usbmix_ctl_maps[] = { +static const struct usbmix_ctl_map usbmix_ctl_maps[] = { { .id = USB_ID(0x041e, 0x3000), .map = extigy_map, @@ -560,37 +567,37 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = { * Control map entries for UAC3 BADD profiles */ -static struct usbmix_name_map uac3_badd_generic_io_map[] = { +static const struct usbmix_name_map uac3_badd_generic_io_map[] = { { UAC3_BADD_FU_ID2, "Generic Out Playback" }, { UAC3_BADD_FU_ID5, "Generic In Capture" }, { 0 } /* terminator */ }; -static struct usbmix_name_map uac3_badd_headphone_map[] = { +static const struct usbmix_name_map uac3_badd_headphone_map[] = { { UAC3_BADD_FU_ID2, "Headphone Playback" }, { 0 } /* terminator */ }; -static struct usbmix_name_map uac3_badd_speaker_map[] = { +static const struct usbmix_name_map uac3_badd_speaker_map[] = { { UAC3_BADD_FU_ID2, "Speaker Playback" }, { 0 } /* terminator */ }; -static struct usbmix_name_map uac3_badd_microphone_map[] = { +static const struct usbmix_name_map uac3_badd_microphone_map[] = { { UAC3_BADD_FU_ID5, "Mic Capture" }, { 0 } /* terminator */ }; /* Covers also 'headset adapter' profile */ -static struct usbmix_name_map uac3_badd_headset_map[] = { +static const struct usbmix_name_map uac3_badd_headset_map[] = { { UAC3_BADD_FU_ID2, "Headset Playback" }, { UAC3_BADD_FU_ID5, "Headset Capture" }, { UAC3_BADD_FU_ID7, "Sidetone Mixing" }, { 0 } /* terminator */ }; -static struct usbmix_name_map uac3_badd_speakerphone_map[] = { +static const struct usbmix_name_map uac3_badd_speakerphone_map[] = { { UAC3_BADD_FU_ID2, "Speaker Playback" }, { UAC3_BADD_FU_ID5, "Mic Capture" }, { 0 } /* terminator */ }; -static struct usbmix_ctl_map uac3_badd_usbmix_ctl_maps[] = { +static const struct usbmix_ctl_map uac3_badd_usbmix_ctl_maps[] = { { .id = UAC3_FUNCTION_SUBCLASS_GENERIC_IO, .map = uac3_badd_generic_io_map, @@ -619,5 +626,10 @@ static struct usbmix_ctl_map uac3_badd_usbmix_ctl_maps[] = { .id = UAC3_FUNCTION_SUBCLASS_SPEAKERPHONE, .map = uac3_badd_speakerphone_map, }, + { + /* Sennheiser Communications Headset [PC 8] */ + .id = USB_ID(0x1395, 0x0025), + .map = sennheiser_pc8_map, + }, { 0 } /* terminator */ }; diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 49f0dc0e3e4d..1f7c80541d03 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -119,7 +119,7 @@ static int snd_create_std_mono_ctl(struct usb_mixer_interface *mixer, * Create a set of standard UAC controls from a table */ static int snd_create_std_mono_table(struct usb_mixer_interface *mixer, - struct std_mono_table *t) + const struct std_mono_table *t) { int err; @@ -1388,7 +1388,7 @@ static int snd_c400_create_mixer(struct usb_mixer_interface *mixer) * are valid they presents mono controls as L and R channels of * stereo. So we provide a good mixer here. */ -static struct std_mono_table ebox44_table[] = { +static const struct std_mono_table ebox44_table[] = { { .unitid = 4, .control = 1, @@ -1697,7 +1697,7 @@ static struct snd_kcontrol_new snd_microii_mixer_spdif[] = { static int snd_microii_controls_create(struct usb_mixer_interface *mixer) { int err, i; - static usb_mixer_elem_resume_func_t resume_funcs[] = { + static const usb_mixer_elem_resume_func_t resume_funcs[] = { snd_microii_spdif_default_update, NULL, snd_microii_spdif_switch_update @@ -2268,7 +2268,7 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x1235, 0x8203): /* Focusrite Scarlett 6i6 2nd Gen */ case USB_ID(0x1235, 0x8204): /* Focusrite Scarlett 18i8 2nd Gen */ case USB_ID(0x1235, 0x8201): /* Focusrite Scarlett 18i20 2nd Gen */ - err = snd_scarlett_gen2_controls_create(mixer); + err = snd_scarlett_gen2_init(mixer); break; case USB_ID(0x041e, 0x323b): /* Creative Sound Blaster E1 */ @@ -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.c b/sound/usb/mixer_scarlett.c index 83715fd8dfd6..6217424e973f 100644 --- a/sound/usb/mixer_scarlett.c +++ b/sound/usb/mixer_scarlett.c @@ -623,7 +623,7 @@ static int add_output_ctls(struct usb_mixer_interface *mixer, /********************** device-specific config *************************/ /* untested... */ -static struct scarlett_device_info s6i6_info = { +static const struct scarlett_device_info s6i6_info = { .matrix_in = 18, .matrix_out = 8, .input_len = 6, @@ -665,7 +665,7 @@ static struct scarlett_device_info s6i6_info = { }; /* untested... */ -static struct scarlett_device_info s8i6_info = { +static const struct scarlett_device_info s8i6_info = { .matrix_in = 18, .matrix_out = 6, .input_len = 8, @@ -704,7 +704,7 @@ static struct scarlett_device_info s8i6_info = { } }; -static struct scarlett_device_info s18i6_info = { +static const struct scarlett_device_info s18i6_info = { .matrix_in = 18, .matrix_out = 6, .input_len = 18, @@ -741,7 +741,7 @@ static struct scarlett_device_info s18i6_info = { } }; -static struct scarlett_device_info s18i8_info = { +static const struct scarlett_device_info s18i8_info = { .matrix_in = 18, .matrix_out = 8, .input_len = 18, @@ -783,7 +783,7 @@ static struct scarlett_device_info s18i8_info = { } }; -static struct scarlett_device_info s18i20_info = { +static const struct scarlett_device_info s18i20_info = { .matrix_in = 18, .matrix_out = 8, .input_len = 18, @@ -833,7 +833,7 @@ static struct scarlett_device_info s18i20_info = { static int scarlett_controls_create_generic(struct usb_mixer_interface *mixer, - struct scarlett_device_info *info) + const struct scarlett_device_info *info) { int i, err; char mx[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; @@ -896,7 +896,7 @@ int snd_scarlett_controls_create(struct usb_mixer_interface *mixer) { int err, i, o; char mx[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; - struct scarlett_device_info *info; + const struct scarlett_device_info *info; struct usb_mixer_elem_info *elem; static char sample_rate_buffer[4] = { '\x80', '\xbb', '\x00', '\x00' }; diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index 74c00c905d24..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" @@ -635,7 +635,7 @@ static int scarlett2_usb( /* send a second message to get the response */ err = snd_usb_ctl_msg(mixer->chip->dev, - usb_sndctrlpipe(mixer->chip->dev, 0), + usb_rcvctrlpipe(mixer->chip->dev, 0), SCARLETT2_USB_VENDOR_SPECIFIC_CMD_RESP, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, 0, @@ -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); @@ -1997,38 +2007,11 @@ static int scarlett2_mixer_status_create(struct usb_mixer_interface *mixer) return usb_submit_urb(mixer->urb, GFP_KERNEL); } -/* Entry point */ -int snd_scarlett_gen2_controls_create(struct usb_mixer_interface *mixer) +static int snd_scarlett_gen2_controls_create(struct usb_mixer_interface *mixer, + const struct scarlett2_device_info *info) { - const struct scarlett2_device_info *info; int err; - /* only use UAC_VERSION_2 */ - if (!mixer->protocol) - return 0; - - switch (mixer->chip->usb_id) { - case USB_ID(0x1235, 0x8203): - info = &s6i6_gen2_info; - break; - case USB_ID(0x1235, 0x8204): - info = &s18i8_gen2_info; - break; - case USB_ID(0x1235, 0x8201): - info = &s18i20_gen2_info; - break; - default: /* device not (yet) supported */ - return -EINVAL; - } - - if (!(mixer->chip->setup & SCARLETT2_ENABLE)) { - usb_audio_err(mixer->chip, - "Focusrite Scarlett Gen 2 Mixer Driver disabled; " - "use options snd_usb_audio device_setup=1 " - "to enable and report any issues to g@b4.vu"); - return 0; - } - /* Initialise private data, routing, sequence number */ err = scarlett2_init_private(mixer, info); if (err < 0) @@ -2073,3 +2056,51 @@ int snd_scarlett_gen2_controls_create(struct usb_mixer_interface *mixer) return 0; } + +int snd_scarlett_gen2_init(struct usb_mixer_interface *mixer) +{ + struct snd_usb_audio *chip = mixer->chip; + const struct scarlett2_device_info *info; + int err; + + /* only use UAC_VERSION_2 */ + if (!mixer->protocol) + return 0; + + switch (chip->usb_id) { + case USB_ID(0x1235, 0x8203): + info = &s6i6_gen2_info; + break; + case USB_ID(0x1235, 0x8204): + info = &s18i8_gen2_info; + break; + case USB_ID(0x1235, 0x8201): + info = &s18i20_gen2_info; + break; + default: /* device not (yet) supported */ + return -EINVAL; + } + + if (!(chip->setup & SCARLETT2_ENABLE)) { + usb_audio_info(chip, + "Focusrite Scarlett Gen 2 Mixer Driver disabled; " + "use options snd_usb_audio vid=0x%04x pid=0x%04x " + "device_setup=1 to enable and report any issues " + "to g@b4.vu", + USB_ID_VENDOR(chip->usb_id), + USB_ID_PRODUCT(chip->usb_id)); + return 0; + } + + usb_audio_info(chip, + "Focusrite Scarlett Gen 2 Mixer Driver enabled pid=0x%04x", + USB_ID_PRODUCT(chip->usb_id)); + + err = snd_scarlett_gen2_controls_create(mixer, info); + if (err < 0) + usb_audio_err(mixer->chip, + "Error initialising Scarlett Mixer Driver: %d", + err); + + return err; +} diff --git a/sound/usb/mixer_scarlett_gen2.h b/sound/usb/mixer_scarlett_gen2.h index 52e1dad77afd..668c6b0cb50a 100644 --- a/sound/usb/mixer_scarlett_gen2.h +++ b/sound/usb/mixer_scarlett_gen2.h @@ -2,6 +2,6 @@ #ifndef __USB_MIXER_SCARLETT_GEN2_H #define __USB_MIXER_SCARLETT_GEN2_H -int snd_scarlett_gen2_controls_create(struct usb_mixer_interface *mixer); +int snd_scarlett_gen2_init(struct usb_mixer_interface *mixer); #endif /* __USB_MIXER_SCARLETT_GEN2_H */ diff --git a/sound/usb/mixer_us16x08.c b/sound/usb/mixer_us16x08.c index f0e8e1539450..c6c834ac83ac 100644 --- a/sound/usb/mixer_us16x08.c +++ b/sound/usb/mixer_us16x08.c @@ -607,7 +607,7 @@ static int snd_us16x08_eq_put(struct snd_kcontrol *kcontrol, static int snd_us16x08_meter_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - uinfo->count = 1; + uinfo->count = 34; uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->value.integer.max = 0x7FFF; uinfo->value.integer.min = 0; diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 878f1201aad6..75218539fb10 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -323,6 +323,7 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, switch (subs->stream->chip->usb_id) { case USB_ID(0x0763, 0x2030): /* M-Audio Fast Track C400 */ case USB_ID(0x0763, 0x2031): /* M-Audio Fast Track C600 */ + case USB_ID(0x22f0, 0x0006): /* Allen&Heath Qu-16 */ ep = 0x81; ifnum = 3; goto add_sync_ep_from_ifnum; @@ -332,6 +333,7 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, ifnum = 2; goto add_sync_ep_from_ifnum; case USB_ID(0x2466, 0x8003): /* Fractal Audio Axe-Fx II */ + case USB_ID(0x0499, 0x172a): /* Yamaha MODX */ ep = 0x86; ifnum = 2; goto add_sync_ep_from_ifnum; @@ -339,6 +341,10 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, ep = 0x81; ifnum = 2; goto add_sync_ep_from_ifnum; + case USB_ID(0x1686, 0xf029): /* Zoom UAC-2 */ + ep = 0x82; + ifnum = 2; + goto add_sync_ep_from_ifnum; case USB_ID(0x1397, 0x0001): /* Behringer UFX1604 */ case USB_ID(0x1397, 0x0002): /* Behringer UFX1204 */ ep = 0x81; @@ -1879,7 +1885,7 @@ void snd_usb_preallocate_buffer(struct snd_usb_substream *subs) { struct snd_pcm *pcm = subs->stream->pcm; struct snd_pcm_substream *s = pcm->streams[subs->direction].substream; - struct device *dev = subs->dev->bus->controller; + struct device *dev = subs->dev->bus->sysdev; if (!snd_usb_use_vmalloc) snd_pcm_lib_preallocate_pages(s, SNDRV_DMA_TYPE_DEV_SG, diff --git a/sound/usb/proc.c b/sound/usb/proc.c index 49e3f176aaf5..ffbf4bd9208c 100644 --- a/sound/usb/proc.c +++ b/sound/usb/proc.c @@ -60,7 +60,7 @@ void snd_usb_audio_create_proc(struct snd_usb_audio *chip) static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct snd_info_buffer *buffer) { struct audioformat *fp; - static char *sync_types[4] = { + static const char * const sync_types[4] = { "NONE", "ASYNC", "ADAPTIVE", "SYNC" }; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 8c3b3a291ddb..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. @@ -2485,6 +2537,16 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +{ + USB_DEVICE_VENDOR_SPEC(0x0944, 0x0204), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "KORG, Inc.", + /* .product_name = "ToneLab EX", */ + .ifnum = 3, + .type = QUIRK_MIDI_STANDARD_INTERFACE, + } +}, + /* AKAI devices */ { USB_DEVICE(0x09e8, 0x0062), @@ -3754,5 +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 cc75d9749e9f..72223545abfd 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -55,8 +55,12 @@ static int create_composite_quirk(struct snd_usb_audio *chip, if (!iface) continue; if (quirk->ifnum != probed_ifnum && - !usb_interface_claimed(iface)) - usb_driver_claim_interface(driver, iface, (void *)-1L); + !usb_interface_claimed(iface)) { + err = usb_driver_claim_interface(driver, iface, + USB_AUDIO_IFACE_UNUSED); + if (err < 0) + return err; + } } return 0; @@ -390,8 +394,12 @@ static int create_autodetect_quirks(struct snd_usb_audio *chip, continue; err = create_autodetect_quirk(chip, iface, driver); - if (err >= 0) - usb_driver_claim_interface(driver, iface, (void *)-1L); + if (err >= 0) { + err = usb_driver_claim_interface(driver, iface, + USB_AUDIO_IFACE_UNUSED); + if (err < 0) + return err; + } } return 0; @@ -1452,6 +1460,8 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) case USB_ID(0x1901, 0x0191): /* GE B850V3 CP2114 audio interface */ case USB_ID(0x21B4, 0x0081): /* AudioQuest DragonFly */ case USB_ID(0x2912, 0x30c8): /* Audioengine D1 */ + case USB_ID(0x413c, 0xa506): /* Dell AE515 sound bar */ + case USB_ID(0x046d, 0x084c): /* Logitech ConferenceCam Connect */ return true; } @@ -1604,13 +1614,21 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) msleep(20); - /* Zoom R16/24, Logitech H650e/H570e, Jabra 550a, Kingston HyperX - * needs a tiny delay here, otherwise requests like get/set - * frequency return as failed despite actually succeeding. + /* + * Plantronics headsets (C320, C320-M, etc) need a delay to avoid + * random microhpone failures. + */ + if (USB_ID_VENDOR(chip->usb_id) == 0x047f && + (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) + msleep(20); + + /* Zoom R16/24, many Logitech(at least H650e/H570e/BCC950), + * Jabra 550a, Kingston HyperX needs a tiny delay here, + * otherwise requests like get/set frequency return + * as failed despite actually succeeding. */ if ((chip->usb_id == USB_ID(0x1686, 0x00dd) || - chip->usb_id == USB_ID(0x046d, 0x0a46) || - chip->usb_id == USB_ID(0x046d, 0x0a56) || + USB_ID_VENDOR(chip->usb_id) == 0x046d || /* Logitech */ chip->usb_id == USB_ID(0x0b0e, 0x0349) || chip->usb_id == USB_ID(0x0951, 0x16ad)) && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) @@ -1731,7 +1749,9 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, case 0x25ce: /* Mytek devices */ case 0x278b: /* Rotel? */ case 0x292b: /* Gustard/Ess based devices */ + case 0x2972: /* FiiO devices */ case 0x2ab6: /* T+A devices */ + case 0x3353: /* Khadas devices */ case 0x3842: /* EVGA */ case 0xc502: /* HiBy devices */ if (fp->dsd_raw) @@ -1820,6 +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/stream.c b/sound/usb/stream.c index d01edd5da6cf..eff1ac1dc9ba 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -193,16 +193,16 @@ static int usb_chmap_ctl_get(struct snd_kcontrol *kcontrol, struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); struct snd_usb_substream *subs = info->private_data; struct snd_pcm_chmap_elem *chmap = NULL; - int i; + int i = 0; - memset(ucontrol->value.integer.value, 0, - sizeof(ucontrol->value.integer.value)); if (subs->cur_audiofmt) chmap = subs->cur_audiofmt->chmap; if (chmap) { for (i = 0; i < chmap->channels; i++) ucontrol->value.integer.value[i] = chmap->map[i]; } + for (; i < subs->channels_max; i++) + ucontrol->value.integer.value[i] = 0; return 0; } @@ -240,7 +240,7 @@ static int add_chmap(struct snd_pcm *pcm, int stream, static struct snd_pcm_chmap_elem *convert_chmap(int channels, unsigned int bits, int protocol) { - static unsigned int uac1_maps[] = { + static const unsigned int uac1_maps[] = { SNDRV_CHMAP_FL, /* left front */ SNDRV_CHMAP_FR, /* right front */ SNDRV_CHMAP_FC, /* center front */ @@ -255,7 +255,7 @@ static struct snd_pcm_chmap_elem *convert_chmap(int channels, unsigned int bits, SNDRV_CHMAP_TC, /* top */ 0 /* terminator */ }; - static unsigned int uac2_maps[] = { + static const unsigned int uac2_maps[] = { SNDRV_CHMAP_FL, /* front left */ SNDRV_CHMAP_FR, /* front right */ SNDRV_CHMAP_FC, /* front center */ diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 55a2119c2411..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)) @@ -59,6 +59,8 @@ struct snd_usb_audio { struct media_intf_devnode *ctl_intf_media_devnode; }; +#define USB_AUDIO_IFACE_UNUSED ((void *)-1L) + #define usb_audio_err(chip, fmt, args...) \ dev_err(&(chip)->dev->dev, fmt, ##args) #define usb_audio_warn(chip, fmt, args...) \ 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/usb/validate.c b/sound/usb/validate.c index 5a3c4f7882b0..89a48d731719 100644 --- a/sound/usb/validate.c +++ b/sound/usb/validate.c @@ -233,7 +233,7 @@ static bool validate_midi_out_jack(const void *p, #define FIXED(p, t, s) { .protocol = (p), .type = (t), .size = sizeof(s) } #define FUNC(p, t, f) { .protocol = (p), .type = (t), .func = (f) } -static struct usb_desc_validator audio_validators[] = { +static const struct usb_desc_validator audio_validators[] = { /* UAC1 */ FUNC(UAC_VERSION_1, UAC_HEADER, validate_uac1_header), FIXED(UAC_VERSION_1, UAC_INPUT_TERMINAL, @@ -288,7 +288,7 @@ static struct usb_desc_validator audio_validators[] = { { } /* terminator */ }; -static struct usb_desc_validator midi_validators[] = { +static const struct usb_desc_validator midi_validators[] = { FIXED(UAC_VERSION_ALL, USB_MS_HEADER, struct usb_ms_header_descriptor), FIXED(UAC_VERSION_ALL, USB_MS_MIDI_IN_JACK, 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); } |