diff options
50 files changed, 2086 insertions, 876 deletions
diff --git a/include/drm/drm_audio_component.h b/include/drm/drm_audio_component.h index a45f93487039..0d36bfd1a4cd 100644 --- a/include/drm/drm_audio_component.h +++ b/include/drm/drm_audio_component.h @@ -117,6 +117,10 @@ struct drm_audio_component { * @audio_ops: Ops implemented by hda driver, called by DRM driver */ const struct drm_audio_component_audio_ops *audio_ops; + /** + * @master_bind_complete: completion held during component master binding + */ + struct completion master_bind_complete; }; #endif /* _DRM_AUDIO_COMPONENT_H_ */ diff --git a/include/sound/pcm_params.h b/include/sound/pcm_params.h index 36f94735d23d..ba184f49f7e1 100644 --- a/include/sound/pcm_params.h +++ b/include/sound/pcm_params.h @@ -23,11 +23,6 @@ int snd_pcm_hw_param_value(const struct snd_pcm_hw_params *params, #define MASK_OFS(i) ((i) >> 5) #define MASK_BIT(i) (1U << ((i) & 31)) -static inline size_t snd_mask_sizeof(void) -{ - return sizeof(struct snd_mask); -} - static inline void snd_mask_none(struct snd_mask *mask) { memset(mask, 0, sizeof(*mask)); diff --git a/include/sound/timer.h b/include/sound/timer.h index 23e885d31525..760e132cc0cd 100644 --- a/include/sound/timer.h +++ b/include/sound/timer.h @@ -21,13 +21,13 @@ #define SNDRV_TIMER_HW_STOP 0x00000002 /* call stop before start */ #define SNDRV_TIMER_HW_SLAVE 0x00000004 /* only slave timer (variable resolution) */ #define SNDRV_TIMER_HW_FIRST 0x00000008 /* first tick can be incomplete */ -#define SNDRV_TIMER_HW_TASKLET 0x00000010 /* timer is called from tasklet */ +#define SNDRV_TIMER_HW_WORK 0x00000010 /* timer is called from work */ #define SNDRV_TIMER_IFLG_SLAVE 0x00000001 #define SNDRV_TIMER_IFLG_RUNNING 0x00000002 #define SNDRV_TIMER_IFLG_START 0x00000004 #define SNDRV_TIMER_IFLG_AUTO 0x00000008 /* auto restart */ -#define SNDRV_TIMER_IFLG_FAST 0x00000010 /* fast callback (do not use tasklet) */ +#define SNDRV_TIMER_IFLG_FAST 0x00000010 /* fast callback (do not use work) */ #define SNDRV_TIMER_IFLG_CALLBACK 0x00000020 /* timer callback is active */ #define SNDRV_TIMER_IFLG_EXCLUSIVE 0x00000040 /* exclusive owner - no more instances */ #define SNDRV_TIMER_IFLG_EARLY_EVENT 0x00000080 /* write early event to the poll queue */ @@ -74,7 +74,7 @@ struct snd_timer { struct list_head active_list_head; struct list_head ack_list_head; struct list_head sack_list_head; /* slow ack list head */ - struct tasklet_struct task_queue; + struct work_struct task_work; int max_instances; /* upper limit of timer instances */ int num_instances; /* current number of timer instances */ }; @@ -96,7 +96,7 @@ struct snd_timer_instance { unsigned long ticks; /* auto-load ticks when expired */ unsigned long cticks; /* current ticks */ unsigned long pticks; /* accumulated ticks for callback */ - unsigned long resolution; /* current resolution for tasklet */ + unsigned long resolution; /* current resolution for work */ unsigned long lost; /* lost ticks */ int slave_class; unsigned int slave_id; diff --git a/sound/ac97/ac97_core.h b/sound/ac97/ac97_core.h index 0c5956e4b2f3..5a9677c3d4c3 100644 --- a/sound/ac97/ac97_core.h +++ b/sound/ac97/ac97_core.h @@ -3,7 +3,7 @@ * Copyright (C) 2016 Robert Jarzmik <robert.jarzmik@free.fr> */ -unsigned int snd_ac97_bus_scan_one(struct ac97_controller *ac97, +unsigned int snd_ac97_bus_scan_one(struct ac97_controller *adrv, unsigned int codec_num); static inline bool ac97_ids_match(unsigned int id1, unsigned int id2, diff --git a/sound/aoa/soundbus/i2sbus/pcm.c b/sound/aoa/soundbus/i2sbus/pcm.c index d350dbd24305..1c8e8131a716 100644 --- a/sound/aoa/soundbus/i2sbus/pcm.c +++ b/sound/aoa/soundbus/i2sbus/pcm.c @@ -254,12 +254,11 @@ static void i2sbus_wait_for_stop(struct i2sbus_dev *i2sdev, struct pcm_info *pi) { unsigned long flags; - struct completion done; + DECLARE_COMPLETION_ONSTACK(done); long timeout; spin_lock_irqsave(&i2sdev->low_lock, flags); if (pi->dbdma_ring.stopping) { - init_completion(&done); pi->stop_completion = &done; spin_unlock_irqrestore(&i2sdev->low_lock, flags); timeout = wait_for_completion_timeout(&done, HZ); diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index 1006458f7f85..66ecbd4d034e 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -475,12 +475,12 @@ static irqreturn_t atmel_ac97c_interrupt(int irq, void *dev) struct snd_pcm_runtime *runtime; int offset, next_period, block_size; dev_dbg(&chip->pdev->dev, "channel A event%s%s%s%s%s%s\n", - casr & AC97C_CSR_OVRUN ? " OVRUN" : "", - casr & AC97C_CSR_RXRDY ? " RXRDY" : "", - casr & AC97C_CSR_UNRUN ? " UNRUN" : "", - casr & AC97C_CSR_TXEMPTY ? " TXEMPTY" : "", - casr & AC97C_CSR_TXRDY ? " TXRDY" : "", - !casr ? " NONE" : ""); + (casr & AC97C_CSR_OVRUN) ? " OVRUN" : "", + (casr & AC97C_CSR_RXRDY) ? " RXRDY" : "", + (casr & AC97C_CSR_UNRUN) ? " UNRUN" : "", + (casr & AC97C_CSR_TXEMPTY) ? " TXEMPTY" : "", + (casr & AC97C_CSR_TXRDY) ? " TXRDY" : "", + !casr ? " NONE" : ""); if ((casr & camr) & AC97C_CSR_ENDTX) { runtime = chip->playback_substream->runtime; block_size = frames_to_bytes(runtime, runtime->period_size); @@ -521,11 +521,11 @@ static irqreturn_t atmel_ac97c_interrupt(int irq, void *dev) if (sr & AC97C_SR_COEVT) { dev_info(&chip->pdev->dev, "codec channel event%s%s%s%s%s\n", - cosr & AC97C_CSR_OVRUN ? " OVRUN" : "", - cosr & AC97C_CSR_RXRDY ? " RXRDY" : "", - cosr & AC97C_CSR_TXEMPTY ? " TXEMPTY" : "", - cosr & AC97C_CSR_TXRDY ? " TXRDY" : "", - !cosr ? " NONE" : ""); + (cosr & AC97C_CSR_OVRUN) ? " OVRUN" : "", + (cosr & AC97C_CSR_RXRDY) ? " RXRDY" : "", + (cosr & AC97C_CSR_TXEMPTY) ? " TXEMPTY" : "", + (cosr & AC97C_CSR_TXRDY) ? " TXRDY" : "", + !cosr ? " NONE" : ""); retval = IRQ_HANDLED; } diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 0e53f6f31916..c1fec932c49d 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -513,10 +513,11 @@ EXPORT_SYMBOL(snd_compr_malloc_pages); int snd_compr_free_pages(struct snd_compr_stream *stream) { - struct snd_compr_runtime *runtime = stream->runtime; + struct snd_compr_runtime *runtime; if (snd_BUG_ON(!(stream) || !(stream)->runtime)) return -EINVAL; + runtime = stream->runtime; if (runtime->dma_area == NULL) return 0; if (runtime->dma_buffer_p != &stream->dma_buffer) { @@ -1031,7 +1032,7 @@ static const struct file_operations snd_compr_file_ops = { static int snd_compress_dev_register(struct snd_device *device) { - int ret = -EINVAL; + int ret; struct snd_compr *compr; if (snd_BUG_ON(!device || !device->device_data)) diff --git a/sound/core/control.c b/sound/core/control.c index aa0c0cf182af..421ddc76f264 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -150,14 +150,14 @@ void snd_ctl_notify(struct snd_card *card, unsigned int mask, return; if (card->shutdown) return; - read_lock(&card->ctl_files_rwlock); + read_lock_irqsave(&card->ctl_files_rwlock, flags); #if IS_ENABLED(CONFIG_SND_MIXER_OSS) card->mixer_oss_change_count++; #endif list_for_each_entry(ctl, &card->ctl_files, list) { if (!ctl->subscribed) continue; - spin_lock_irqsave(&ctl->read_lock, flags); + spin_lock(&ctl->read_lock); list_for_each_entry(ev, &ctl->events, list) { if (ev->id.numid == id->numid) { ev->mask |= mask; @@ -174,10 +174,10 @@ void snd_ctl_notify(struct snd_card *card, unsigned int mask, } _found: wake_up(&ctl->change_sleep); - spin_unlock_irqrestore(&ctl->read_lock, flags); + spin_unlock(&ctl->read_lock); kill_fasync(&ctl->fasync, SIGIO, POLL_IN); } - read_unlock(&card->ctl_files_rwlock); + read_unlock_irqrestore(&card->ctl_files_rwlock, flags); } EXPORT_SYMBOL(snd_ctl_notify); @@ -717,22 +717,19 @@ static int snd_ctl_card_info(struct snd_card *card, struct snd_ctl_file * ctl, } static int snd_ctl_elem_list(struct snd_card *card, - struct snd_ctl_elem_list __user *_list) + struct snd_ctl_elem_list *list) { - struct snd_ctl_elem_list list; struct snd_kcontrol *kctl; struct snd_ctl_elem_id id; unsigned int offset, space, jidx; int err = 0; - if (copy_from_user(&list, _list, sizeof(list))) - return -EFAULT; - offset = list.offset; - space = list.space; + offset = list->offset; + space = list->space; down_read(&card->controls_rwsem); - list.count = card->controls_count; - list.used = 0; + list->count = card->controls_count; + list->used = 0; if (space > 0) { list_for_each_entry(kctl, &card->controls, list) { if (offset >= kctl->count) { @@ -741,12 +738,12 @@ static int snd_ctl_elem_list(struct snd_card *card, } for (jidx = offset; jidx < kctl->count; jidx++) { snd_ctl_build_ioff(&id, kctl, jidx); - if (copy_to_user(list.pids + list.used, &id, + if (copy_to_user(list->pids + list->used, &id, sizeof(id))) { err = -EFAULT; goto out; } - list.used++; + list->used++; if (!--space) goto out; } @@ -755,11 +752,26 @@ static int snd_ctl_elem_list(struct snd_card *card, } out: up_read(&card->controls_rwsem); - if (!err && copy_to_user(_list, &list, sizeof(list))) - err = -EFAULT; return err; } +static int snd_ctl_elem_list_user(struct snd_card *card, + struct snd_ctl_elem_list __user *_list) +{ + struct snd_ctl_elem_list list; + int err; + + if (copy_from_user(&list, _list, sizeof(list))) + return -EFAULT; + err = snd_ctl_elem_list(card, &list); + if (err) + return err; + if (copy_to_user(_list, &list, sizeof(list))) + return -EFAULT; + + return 0; +} + /* Check whether the given kctl info is valid */ static int snd_ctl_check_elem_info(struct snd_card *card, const struct snd_ctl_elem_info *info) @@ -1703,7 +1715,7 @@ static long snd_ctl_ioctl(struct file *file, unsigned int cmd, unsigned long arg case SNDRV_CTL_IOCTL_CARD_INFO: return snd_ctl_card_info(card, ctl, cmd, argp); case SNDRV_CTL_IOCTL_ELEM_LIST: - return snd_ctl_elem_list(card, argp); + return snd_ctl_elem_list_user(card, argp); case SNDRV_CTL_IOCTL_ELEM_INFO: return snd_ctl_elem_info_user(ctl, argp); case SNDRV_CTL_IOCTL_ELEM_READ: @@ -1939,8 +1951,9 @@ int snd_ctl_get_preferred_subdevice(struct snd_card *card, int type) { struct snd_ctl_file *kctl; int subdevice = -1; + unsigned long flags; - read_lock(&card->ctl_files_rwlock); + read_lock_irqsave(&card->ctl_files_rwlock, flags); list_for_each_entry(kctl, &card->ctl_files, list) { if (kctl->pid == task_pid(current)) { subdevice = kctl->preferred_subdevice[type]; @@ -1948,7 +1961,7 @@ int snd_ctl_get_preferred_subdevice(struct snd_card *card, int type) break; } } - read_unlock(&card->ctl_files_rwlock); + read_unlock_irqrestore(&card->ctl_files_rwlock, flags); return subdevice; } EXPORT_SYMBOL_GPL(snd_ctl_get_preferred_subdevice); @@ -1997,13 +2010,14 @@ static int snd_ctl_dev_disconnect(struct snd_device *device) { struct snd_card *card = device->device_data; struct snd_ctl_file *ctl; + unsigned long flags; - read_lock(&card->ctl_files_rwlock); + read_lock_irqsave(&card->ctl_files_rwlock, flags); list_for_each_entry(ctl, &card->ctl_files, list) { wake_up(&ctl->change_sleep); kill_fasync(&ctl->fasync, SIGIO, POLL_ERR); } - read_unlock(&card->ctl_files_rwlock); + read_unlock_irqrestore(&card->ctl_files_rwlock, flags); return snd_unregister_device(&card->ctl_dev); } diff --git a/sound/core/control_compat.c b/sound/core/control_compat.c index 02df1d7db9a1..1d708aab9c98 100644 --- a/sound/core/control_compat.c +++ b/sound/core/control_compat.c @@ -22,24 +22,22 @@ struct snd_ctl_elem_list32 { static int snd_ctl_elem_list_compat(struct snd_card *card, struct snd_ctl_elem_list32 __user *data32) { - struct snd_ctl_elem_list __user *data; + struct snd_ctl_elem_list data = {}; compat_caddr_t ptr; int err; - data = compat_alloc_user_space(sizeof(*data)); - /* offset, space, used, count */ - if (copy_in_user(data, data32, 4 * sizeof(u32))) + if (copy_from_user(&data, data32, 4 * sizeof(u32))) return -EFAULT; /* pids */ - if (get_user(ptr, &data32->pids) || - put_user(compat_ptr(ptr), &data->pids)) + if (get_user(ptr, &data32->pids)) return -EFAULT; - err = snd_ctl_elem_list(card, data); + data.pids = compat_ptr(ptr); + err = snd_ctl_elem_list(card, &data); if (err < 0) return err; /* copy the result */ - if (copy_in_user(data32, data, 4 * sizeof(u32))) + if (copy_to_user(data32, &data, 4 * sizeof(u32))) return -EFAULT; return 0; } diff --git a/sound/core/hrtimer.c b/sound/core/hrtimer.c index c61ba52a530a..e97ff8cccb64 100644 --- a/sound/core/hrtimer.c +++ b/sound/core/hrtimer.c @@ -114,7 +114,7 @@ static int snd_hrtimer_stop(struct snd_timer *t) } static const struct snd_timer_hardware hrtimer_hw __initconst = { - .flags = SNDRV_TIMER_HW_AUTO | SNDRV_TIMER_HW_TASKLET, + .flags = SNDRV_TIMER_HW_AUTO | SNDRV_TIMER_HW_WORK, .open = snd_hrtimer_open, .close = snd_hrtimer_close, .start = snd_hrtimer_start, diff --git a/sound/core/hwdep.c b/sound/core/hwdep.c index 21edb8ac95eb..0c029892880a 100644 --- a/sound/core/hwdep.c +++ b/sound/core/hwdep.c @@ -203,28 +203,35 @@ static int snd_hwdep_dsp_status(struct snd_hwdep *hw, } static int snd_hwdep_dsp_load(struct snd_hwdep *hw, - struct snd_hwdep_dsp_image __user *_info) + struct snd_hwdep_dsp_image *info) { - struct snd_hwdep_dsp_image info; int err; if (! hw->ops.dsp_load) return -ENXIO; - memset(&info, 0, sizeof(info)); - if (copy_from_user(&info, _info, sizeof(info))) - return -EFAULT; - if (info.index >= 32) + if (info->index >= 32) return -EINVAL; /* check whether the dsp was already loaded */ - if (hw->dsp_loaded & (1u << info.index)) + if (hw->dsp_loaded & (1u << info->index)) return -EBUSY; - err = hw->ops.dsp_load(hw, &info); + err = hw->ops.dsp_load(hw, info); if (err < 0) return err; - hw->dsp_loaded |= (1u << info.index); + hw->dsp_loaded |= (1u << info->index); return 0; } +static int snd_hwdep_dsp_load_user(struct snd_hwdep *hw, + struct snd_hwdep_dsp_image __user *_info) +{ + struct snd_hwdep_dsp_image info = {}; + + if (copy_from_user(&info, _info, sizeof(info))) + return -EFAULT; + return snd_hwdep_dsp_load(hw, &info); +} + + static long snd_hwdep_ioctl(struct file * file, unsigned int cmd, unsigned long arg) { @@ -238,7 +245,7 @@ static long snd_hwdep_ioctl(struct file * file, unsigned int cmd, case SNDRV_HWDEP_IOCTL_DSP_STATUS: return snd_hwdep_dsp_status(hw, argp); case SNDRV_HWDEP_IOCTL_DSP_LOAD: - return snd_hwdep_dsp_load(hw, argp); + return snd_hwdep_dsp_load_user(hw, argp); } if (hw->ops.ioctl) return hw->ops.ioctl(hw, file, cmd, arg); diff --git a/sound/core/hwdep_compat.c b/sound/core/hwdep_compat.c index bc81db9cb3d4..a0b76706c083 100644 --- a/sound/core/hwdep_compat.c +++ b/sound/core/hwdep_compat.c @@ -19,26 +19,17 @@ struct snd_hwdep_dsp_image32 { static int snd_hwdep_dsp_load_compat(struct snd_hwdep *hw, struct snd_hwdep_dsp_image32 __user *src) { - struct snd_hwdep_dsp_image __user *dst; + struct snd_hwdep_dsp_image info = {}; compat_caddr_t ptr; - u32 val; - dst = compat_alloc_user_space(sizeof(*dst)); - - /* index and name */ - if (copy_in_user(dst, src, 4 + 64)) - return -EFAULT; - if (get_user(ptr, &src->image) || - put_user(compat_ptr(ptr), &dst->image)) - return -EFAULT; - if (get_user(val, &src->length) || - put_user(val, &dst->length)) - return -EFAULT; - if (get_user(val, &src->driver_data) || - put_user(val, &dst->driver_data)) + if (copy_from_user(&info, src, 4 + 64) || + get_user(ptr, &src->image) || + get_user(info.length, &src->length) || + get_user(info.driver_data, &src->driver_data)) return -EFAULT; + info.image = compat_ptr(ptr); - return snd_hwdep_dsp_load(hw, dst); + return snd_hwdep_dsp_load(hw, &info); } enum { diff --git a/sound/core/init.c b/sound/core/init.c index 0478847ba2b8..764dbe673d48 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -519,10 +519,9 @@ EXPORT_SYMBOL(snd_card_free_when_closed); */ int snd_card_free(struct snd_card *card) { - struct completion released; + DECLARE_COMPLETION_ONSTACK(released); int ret; - init_completion(&released); card->release_completion = &released; ret = snd_card_free_when_closed(card); if (ret) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index ad74ea9cbff5..0aeeb6244ff6 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -157,8 +157,8 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size, * so if we fail to malloc, try to fetch memory traditionally. */ dmab->dev.type = SNDRV_DMA_TYPE_DEV; -#endif /* CONFIG_GENERIC_ALLOCATOR */ fallthrough; +#endif /* CONFIG_GENERIC_ALLOCATOR */ case SNDRV_DMA_TYPE_DEV: case SNDRV_DMA_TYPE_DEV_UC: snd_malloc_dev_pages(dmab, size); diff --git a/sound/core/pcm.c b/sound/core/pcm.c index b6d2331a82f7..be5714f1bb58 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -991,11 +991,13 @@ void snd_pcm_detach_substream(struct snd_pcm_substream *substream) PAGE_ALIGN(sizeof(struct snd_pcm_mmap_control))); kfree(runtime->hw_constraints.rules); /* Avoid concurrent access to runtime via PCM timer interface */ - if (substream->timer) + if (substream->timer) { spin_lock_irq(&substream->timer->lock); - substream->runtime = NULL; - if (substream->timer) + substream->runtime = NULL; spin_unlock_irq(&substream->timer->lock); + } else { + substream->runtime = NULL; + } kfree(runtime); put_pid(substream->pid); substream->pid = NULL; diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c index 1bf6a3d9e0c2..4f03ba8ed0ae 100644 --- a/sound/core/pcm_memory.c +++ b/sound/core/pcm_memory.c @@ -377,7 +377,7 @@ struct page *snd_pcm_sgbuf_ops_page(struct snd_pcm_substream *substream, unsigne */ int snd_pcm_lib_malloc_pages(struct snd_pcm_substream *substream, size_t size) { - struct snd_card *card = substream->pcm->card; + struct snd_card *card; struct snd_pcm_runtime *runtime; struct snd_dma_buffer *dmab = NULL; @@ -387,6 +387,7 @@ int snd_pcm_lib_malloc_pages(struct snd_pcm_substream *substream, size_t size) SNDRV_DMA_TYPE_UNKNOWN)) return -EINVAL; runtime = substream->runtime; + card = substream->pcm->card; if (runtime->dma_buffer_p) { /* perphaps, we might free the large DMA memory region diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 2a688b711a9a..c78720a3299c 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -35,7 +35,7 @@ module_param_array(amidi_map, int, NULL, 0444); MODULE_PARM_DESC(amidi_map, "Raw MIDI device number assigned to 2nd OSS device."); #endif /* CONFIG_SND_OSSEMUL */ -static int snd_rawmidi_free(struct snd_rawmidi *rawmidi); +static int snd_rawmidi_free(struct snd_rawmidi *rmidi); static int snd_rawmidi_dev_free(struct snd_device *device); static int snd_rawmidi_dev_register(struct snd_device *device); static int snd_rawmidi_dev_disconnect(struct snd_device *device); 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/timer.c b/sound/core/timer.c index 6e27d87b18ed..765ea66665a8 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -173,7 +173,7 @@ EXPORT_SYMBOL(snd_timer_instance_free); */ static struct snd_timer *snd_timer_find(struct snd_timer_id *tid) { - struct snd_timer *timer = NULL; + struct snd_timer *timer; list_for_each_entry(timer, &snd_timer_list, device_list) { if (timer->tmr_class != tid->dev_class) @@ -813,12 +813,12 @@ static void snd_timer_clear_callbacks(struct snd_timer *timer, } /* - * timer tasklet + * timer work * */ -static void snd_timer_tasklet(struct tasklet_struct *t) +static void snd_timer_work(struct work_struct *work) { - struct snd_timer *timer = from_tasklet(timer, t, task_queue); + struct snd_timer *timer = container_of(work, struct snd_timer, task_work); unsigned long flags; if (timer->card && timer->card->shutdown) { @@ -843,7 +843,7 @@ void snd_timer_interrupt(struct snd_timer * timer, unsigned long ticks_left) unsigned long resolution; struct list_head *ack_list_head; unsigned long flags; - int use_tasklet = 0; + bool use_work = false; if (timer == NULL) return; @@ -884,7 +884,7 @@ void snd_timer_interrupt(struct snd_timer * timer, unsigned long ticks_left) --timer->running; list_del_init(&ti->active_list); } - if ((timer->hw.flags & SNDRV_TIMER_HW_TASKLET) || + if ((timer->hw.flags & SNDRV_TIMER_HW_WORK) || (ti->flags & SNDRV_TIMER_IFLG_FAST)) ack_list_head = &timer->ack_list_head; else @@ -919,11 +919,11 @@ void snd_timer_interrupt(struct snd_timer * timer, unsigned long ticks_left) snd_timer_process_callbacks(timer, &timer->ack_list_head); /* do we have any slow callbacks? */ - use_tasklet = !list_empty(&timer->sack_list_head); + use_work = !list_empty(&timer->sack_list_head); spin_unlock_irqrestore(&timer->lock, flags); - if (use_tasklet) - tasklet_schedule(&timer->task_queue); + if (use_work) + queue_work(system_highpri_wq, &timer->task_work); } EXPORT_SYMBOL(snd_timer_interrupt); @@ -967,7 +967,7 @@ int snd_timer_new(struct snd_card *card, char *id, struct snd_timer_id *tid, INIT_LIST_HEAD(&timer->ack_list_head); INIT_LIST_HEAD(&timer->sack_list_head); spin_lock_init(&timer->lock); - tasklet_setup(&timer->task_queue, snd_timer_tasklet); + INIT_WORK(&timer->task_work, snd_timer_work); timer->max_instances = 1000; /* default limit per timer */ if (card != NULL) { timer->module = card->module; @@ -1200,7 +1200,7 @@ static int snd_timer_s_close(struct snd_timer *timer) static const struct snd_timer_hardware snd_timer_system = { - .flags = SNDRV_TIMER_HW_FIRST | SNDRV_TIMER_HW_TASKLET, + .flags = SNDRV_TIMER_HW_FIRST | SNDRV_TIMER_HW_WORK, .resolution = 1000000000L / HZ, .ticks = 10000000L, .close = snd_timer_s_close, @@ -1280,8 +1280,8 @@ static void snd_timer_proc_read(struct snd_info_entry *entry, list_for_each_entry(ti, &timer->open_list_head, open_list) snd_iprintf(buffer, " Client %s : %s\n", ti->owner ? ti->owner : "unknown", - ti->flags & (SNDRV_TIMER_IFLG_START | - SNDRV_TIMER_IFLG_RUNNING) + (ti->flags & (SNDRV_TIMER_IFLG_START | + SNDRV_TIMER_IFLG_RUNNING)) ? "running" : "stopped"); } mutex_unlock(®ister_mutex); diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 251eaf1152e2..c91356326699 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -110,7 +110,7 @@ struct loopback_cable { struct { int stream; struct snd_timer_id id; - struct tasklet_struct event_tasklet; + struct work_struct event_work; struct snd_timer_instance *instance; } snd_timer; }; @@ -309,8 +309,8 @@ static int loopback_snd_timer_close_cable(struct loopback_pcm *dpcm) */ snd_timer_close(cable->snd_timer.instance); - /* wait till drain tasklet has finished if requested */ - tasklet_kill(&cable->snd_timer.event_tasklet); + /* wait till drain work has finished if requested */ + cancel_work_sync(&cable->snd_timer.event_work); snd_timer_instance_free(cable->snd_timer.instance); memset(&cable->snd_timer, 0, sizeof(cable->snd_timer)); @@ -794,11 +794,11 @@ static void loopback_snd_timer_function(struct snd_timer_instance *timeri, resolution); } -static void loopback_snd_timer_tasklet(unsigned long arg) +static void loopback_snd_timer_work(struct work_struct *work) { - struct snd_timer_instance *timeri = (struct snd_timer_instance *)arg; - struct loopback_cable *cable = timeri->callback_data; + struct loopback_cable *cable; + cable = container_of(work, struct loopback_cable, snd_timer.event_work); loopback_snd_timer_period_elapsed(cable, SNDRV_TIMER_EVENT_MSTOP, 0); } @@ -828,9 +828,9 @@ static void loopback_snd_timer_event(struct snd_timer_instance *timeri, * state the streaming will be aborted by the usual timeout. It * should not be aborted here because may be the timer sound * card does only a recovery and the timer is back soon. - * This tasklet triggers loopback_snd_timer_tasklet() + * This work triggers loopback_snd_timer_work() */ - tasklet_schedule(&cable->snd_timer.event_tasklet); + schedule_work(&cable->snd_timer.event_work); } } @@ -1124,7 +1124,7 @@ static int loopback_snd_timer_open(struct loopback_pcm *dpcm) err = -ENOMEM; goto exit; } - /* The callback has to be called from another tasklet. If + /* The callback has to be called from another work. If * SNDRV_TIMER_IFLG_FAST is specified it will be called from the * snd_pcm_period_elapsed() call of the selected sound card. * snd_pcm_period_elapsed() helds snd_pcm_stream_lock_irqsave(). @@ -1137,9 +1137,8 @@ static int loopback_snd_timer_open(struct loopback_pcm *dpcm) timeri->callback_data = (void *)cable; timeri->ccallback = loopback_snd_timer_event; - /* initialise a tasklet used for draining */ - tasklet_init(&cable->snd_timer.event_tasklet, - loopback_snd_timer_tasklet, (unsigned long)timeri); + /* initialise a work used for draining */ + INIT_WORK(&cable->snd_timer.event_work, loopback_snd_timer_work); /* The mutex loopback->cable_lock is kept locked. * Therefore snd_timer_open() cannot be called a second time diff --git a/sound/drivers/pcsp/pcsp_lib.c b/sound/drivers/pcsp/pcsp_lib.c index 4e79293d7f11..ed40d0f7432c 100644 --- a/sound/drivers/pcsp/pcsp_lib.c +++ b/sound/drivers/pcsp/pcsp_lib.c @@ -23,10 +23,10 @@ MODULE_PARM_DESC(nforce_wa, "Apply NForce chipset workaround " #define DMIX_WANTS_S16 1 /* - * Call snd_pcm_period_elapsed in a tasklet + * Call snd_pcm_period_elapsed in a work * This avoids spinlock messes and long-running irq contexts */ -static void pcsp_call_pcm_elapsed(unsigned long priv) +static void pcsp_call_pcm_elapsed(struct work_struct *work) { if (atomic_read(&pcsp_chip.timer_active)) { struct snd_pcm_substream *substream; @@ -36,7 +36,7 @@ static void pcsp_call_pcm_elapsed(unsigned long priv) } } -static DECLARE_TASKLET_OLD(pcsp_pcm_tasklet, pcsp_call_pcm_elapsed); +static DECLARE_WORK(pcsp_pcm_work, pcsp_call_pcm_elapsed); /* write the port and returns the next expire time in ns; * called at the trigger-start and in hrtimer callback @@ -119,11 +119,9 @@ static void pcsp_pointer_update(struct snd_pcsp *chip) if (periods_elapsed) { chip->period_ptr += periods_elapsed * period_bytes; chip->period_ptr %= buffer_bytes; + queue_work(system_highpri_wq, &pcsp_pcm_work); } spin_unlock_irqrestore(&chip->substream_lock, flags); - - if (periods_elapsed) - tasklet_schedule(&pcsp_pcm_tasklet); } enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle) @@ -196,7 +194,7 @@ void pcsp_sync_stop(struct snd_pcsp *chip) pcsp_stop_playing(chip); local_irq_enable(); hrtimer_cancel(&chip->timer); - tasklet_kill(&pcsp_pcm_tasklet); + cancel_work_sync(&pcsp_pcm_work); } static int snd_pcsp_playback_close(struct snd_pcm_substream *substream) diff --git a/sound/drivers/portman2x4.c b/sound/drivers/portman2x4.c index 38603cb2bd5b..c876cf9b5005 100644 --- a/sound/drivers/portman2x4.c +++ b/sound/drivers/portman2x4.c @@ -467,7 +467,7 @@ static int portman_probe(struct parport *p) parport_write_control(p, 0); /* Reset Strobe=0. */ /* Check if Tx circuitry is functioning properly. If initialized - * unit TxEmpty is false, send out char and see if if goes true. + * unit TxEmpty is false, send out char and see if it goes true. */ /* 8 */ parport_write_control(p, TXDATA0); /* Tx channel 0, strobe off. */ diff --git a/sound/drivers/vx/vx_core.c b/sound/drivers/vx/vx_core.c index 26d591fe6a6b..d5c65cab195b 100644 --- a/sound/drivers/vx/vx_core.c +++ b/sound/drivers/vx/vx_core.c @@ -597,9 +597,9 @@ static void vx_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *b snd_iprintf(buffer, "%s\n", chip->card->longname); snd_iprintf(buffer, "Xilinx Firmware: %s\n", - chip->chip_status & VX_STAT_XILINX_LOADED ? "Loaded" : "No"); + (chip->chip_status & VX_STAT_XILINX_LOADED) ? "Loaded" : "No"); snd_iprintf(buffer, "Device Initialized: %s\n", - chip->chip_status & VX_STAT_DEVICE_INIT ? "Yes" : "No"); + (chip->chip_status & VX_STAT_DEVICE_INIT) ? "Yes" : "No"); snd_iprintf(buffer, "DSP audio info:"); if (chip->audio_info & VX_AUDIO_INFO_REAL_TIME) snd_iprintf(buffer, " realtime"); diff --git a/sound/drivers/vx/vx_pcm.c b/sound/drivers/vx/vx_pcm.c index 664b9efa9a50..3d2e3bcafca8 100644 --- a/sound/drivers/vx/vx_pcm.c +++ b/sound/drivers/vx/vx_pcm.c @@ -60,7 +60,6 @@ static void vx_pcm_read_per_bytes(struct vx_core *chip, struct snd_pcm_runtime * *buf++ = vx_inb(chip, RXL); if (++offset >= pipe->buffer_bytes) { offset = 0; - buf = (unsigned char *)runtime->dma_area; } pipe->hw_ptr = offset; } @@ -530,7 +529,6 @@ static int vx_pcm_playback_open(struct snd_pcm_substream *subs) err = vx_alloc_pipe(chip, 0, audio, 2, &pipe); /* stereo playback */ if (err < 0) return err; - chip->playback_pipes[audio] = pipe; } /* open for playback */ pipe->references++; diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index ee1c428b1fd3..4e2f2bb7879f 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -64,7 +64,7 @@ #define IT_PKT_HEADER_SIZE_CIP 8 // For 2 CIP header. #define IT_PKT_HEADER_SIZE_NO_CIP 0 // Nothing. -static void pcm_period_tasklet(struct tasklet_struct *t); +static void pcm_period_work(struct work_struct *work); /** * amdtp_stream_init - initialize an AMDTP stream structure @@ -94,7 +94,7 @@ int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit, s->flags = flags; s->context = ERR_PTR(-1); mutex_init(&s->mutex); - tasklet_setup(&s->period_tasklet, pcm_period_tasklet); + INIT_WORK(&s->period_work, pcm_period_work); s->packet_index = 0; init_waitqueue_head(&s->callback_wait); @@ -203,7 +203,7 @@ int amdtp_stream_add_pcm_hw_constraints(struct amdtp_stream *s, // Linux driver for 1394 OHCI controller voluntarily flushes isoc // context when total size of accumulated context header reaches - // PAGE_SIZE. This kicks tasklet for the isoc context and brings + // PAGE_SIZE. This kicks work for the isoc context and brings // callback in the middle of scheduled interrupts. // Although AMDTP streams in the same domain use the same events per // IRQ, use the largest size of context header between IT/IR contexts. @@ -333,7 +333,7 @@ EXPORT_SYMBOL(amdtp_stream_get_max_payload); */ void amdtp_stream_pcm_prepare(struct amdtp_stream *s) { - tasklet_kill(&s->period_tasklet); + cancel_work_sync(&s->period_work); s->pcm_buffer_pointer = 0; s->pcm_period_pointer = 0; } @@ -437,13 +437,14 @@ static void update_pcm_pointers(struct amdtp_stream *s, s->pcm_period_pointer += frames; if (s->pcm_period_pointer >= pcm->runtime->period_size) { s->pcm_period_pointer -= pcm->runtime->period_size; - tasklet_hi_schedule(&s->period_tasklet); + queue_work(system_highpri_wq, &s->period_work); } } -static void pcm_period_tasklet(struct tasklet_struct *t) +static void pcm_period_work(struct work_struct *work) { - struct amdtp_stream *s = from_tasklet(s, t, period_tasklet); + struct amdtp_stream *s = container_of(work, struct amdtp_stream, + period_work); struct snd_pcm_substream *pcm = READ_ONCE(s->pcm); if (pcm) @@ -794,7 +795,7 @@ static void generate_pkt_descs(struct amdtp_stream *s, struct pkt_desc *descs, static inline void cancel_stream(struct amdtp_stream *s) { s->packet_index = -1; - if (in_interrupt()) + if (current_work() == &s->period_work) amdtp_stream_pcm_abort(s); WRITE_ONCE(s->pcm_buffer_pointer, SNDRV_PCM_POS_XRUN); } @@ -1184,7 +1185,7 @@ unsigned long amdtp_domain_stream_pcm_pointer(struct amdtp_domain *d, if (irq_target && amdtp_stream_running(irq_target)) { // This function is called in software IRQ context of - // period_tasklet or process context. + // period_work or process context. // // When the software IRQ context was scheduled by software IRQ // context of IT contexts, queued packets were already handled. @@ -1195,9 +1196,9 @@ unsigned long amdtp_domain_stream_pcm_pointer(struct amdtp_domain *d, // immediately to keep better granularity of PCM pointer. // // Later, the process context will sometimes schedules software - // IRQ context of the period_tasklet. Then, no need to flush the + // IRQ context of the period_work. Then, no need to flush the // queue by the same reason as described in the above - if (!in_interrupt()) { + if (current_work() != &s->period_work) { // Queued packet should be processed without any kernel // preemption to keep latency against bus cycle. preempt_disable(); @@ -1263,7 +1264,7 @@ static void amdtp_stream_stop(struct amdtp_stream *s) return; } - tasklet_kill(&s->period_tasklet); + cancel_work_sync(&s->period_work); fw_iso_context_stop(s->context); fw_iso_context_destroy(s->context); s->context = ERR_PTR(-1); diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h index 703b710aaf7f..2ceb57d1d58e 100644 --- a/sound/firewire/amdtp-stream.h +++ b/sound/firewire/amdtp-stream.h @@ -163,7 +163,7 @@ struct amdtp_stream { /* For a PCM substream processing. */ struct snd_pcm_substream *pcm; - struct tasklet_struct period_tasklet; + struct work_struct period_work; snd_pcm_uframes_t pcm_buffer_pointer; unsigned int pcm_period_pointer; diff --git a/sound/hda/hdac_component.c b/sound/hda/hdac_component.c index 89126c6fd216..bb37e7e0bd79 100644 --- a/sound/hda/hdac_component.c +++ b/sound/hda/hdac_component.c @@ -210,12 +210,14 @@ static int hdac_component_master_bind(struct device *dev) goto module_put; } + complete_all(&acomp->master_bind_complete); return 0; module_put: module_put(acomp->ops->owner); out_unbind: component_unbind_all(dev, acomp); + complete_all(&acomp->master_bind_complete); return ret; } @@ -296,6 +298,7 @@ int snd_hdac_acomp_init(struct hdac_bus *bus, if (!acomp) return -ENOMEM; acomp->audio_ops = aops; + init_completion(&acomp->master_bind_complete); bus->audio_component = acomp; devres_add(dev, acomp); diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index 3c2db3816029..454474ac5716 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -11,9 +11,7 @@ #include <sound/hda_i915.h> #include <sound/hda_register.h> -static struct completion bind_complete; - -#define CONTROLLER_IN_GPU(pci) (((pci)->device == 0x0a0c) || \ +#define IS_HSW_CONTROLLER(pci) (((pci)->device == 0x0a0c) || \ ((pci)->device == 0x0c0c) || \ ((pci)->device == 0x0d0c) || \ ((pci)->device == 0x160c)) @@ -41,7 +39,7 @@ void snd_hdac_i915_set_bclk(struct hdac_bus *bus) if (!acomp || !acomp->ops || !acomp->ops->get_cdclk_freq) return; /* only for i915 binding */ - if (!CONTROLLER_IN_GPU(pci)) + if (!IS_HSW_CONTROLLER(pci)) return; /* only HSW/BDW */ cdclk_freq = acomp->ops->get_cdclk_freq(acomp->dev); @@ -73,11 +71,49 @@ void snd_hdac_i915_set_bclk(struct hdac_bus *bus) } EXPORT_SYMBOL_GPL(snd_hdac_i915_set_bclk); +/* returns true if the devices can be connected for audio */ +static bool connectivity_check(struct pci_dev *i915, struct pci_dev *hdac) +{ + struct pci_bus *bus_a = i915->bus, *bus_b = hdac->bus; + + /* directly connected on the same bus */ + if (bus_a == bus_b) + return true; + + /* + * on i915 discrete GPUs with embedded HDA audio, the two + * devices are connected via 2nd level PCI bridge + */ + bus_a = bus_a->parent; + bus_b = bus_b->parent; + if (!bus_a || !bus_b) + return false; + bus_a = bus_a->parent; + bus_b = bus_b->parent; + if (bus_a && bus_a == bus_b) + return true; + + return false; +} + static int i915_component_master_match(struct device *dev, int subcomponent, void *data) { - return !strcmp(dev->driver->name, "i915") && - subcomponent == I915_COMPONENT_AUDIO; + struct pci_dev *hdac_pci, *i915_pci; + struct hdac_bus *bus = data; + + if (!dev_is_pci(dev)) + return 0; + + hdac_pci = to_pci_dev(bus->dev); + i915_pci = to_pci_dev(dev); + + if (!strcmp(dev->driver->name, "i915") && + subcomponent == I915_COMPONENT_AUDIO && + connectivity_check(i915_pci, hdac_pci)) + return 1; + + return 0; } /* check whether intel graphics is present */ @@ -92,19 +128,6 @@ static bool i915_gfx_present(void) return pci_dev_present(ids); } -static int i915_master_bind(struct device *dev, - struct drm_audio_component *acomp) -{ - complete_all(&bind_complete); - /* clear audio_ops here as it was needed only for completion call */ - acomp->audio_ops = NULL; - return 0; -} - -static const struct drm_audio_component_audio_ops i915_init_ops = { - .master_bind = i915_master_bind -}; - /** * snd_hdac_i915_init - Initialize i915 audio component * @bus: HDA core bus @@ -125,9 +148,7 @@ int snd_hdac_i915_init(struct hdac_bus *bus) if (!i915_gfx_present()) return -ENODEV; - init_completion(&bind_complete); - - err = snd_hdac_acomp_init(bus, &i915_init_ops, + err = snd_hdac_acomp_init(bus, NULL, i915_component_master_match, sizeof(struct i915_audio_component) - sizeof(*acomp)); if (err < 0) @@ -139,8 +160,8 @@ int snd_hdac_i915_init(struct hdac_bus *bus) if (!IS_ENABLED(CONFIG_MODULES) || !request_module("i915")) { /* 60s timeout */ - wait_for_completion_timeout(&bind_complete, - msecs_to_jiffies(60 * 1000)); + wait_for_completion_timeout(&acomp->master_bind_complete, + msecs_to_jiffies(60 * 1000)); } } if (!acomp->ops) { diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index 35e76480306e..5e1f9f10051b 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -117,7 +117,6 @@ struct snd_card_asihpi { * snd_card_asihpi_timer_function(). */ struct snd_card_asihpi_pcm *llmode_streampriv; - struct tasklet_struct t; void (*pcm_start)(struct snd_pcm_substream *substream); void (*pcm_stop)(struct snd_pcm_substream *substream); @@ -258,15 +257,6 @@ static inline u16 hpi_stream_group_reset(u32 h_stream) return hpi_instream_group_reset(h_stream); } -static inline u16 hpi_stream_group_get_map( - u32 h_stream, u32 *mo, u32 *mi) -{ - if (hpi_handle_object(h_stream) == HPI_OBJ_OSTREAM) - return hpi_outstream_group_get_map(h_stream, mo, mi); - else - return hpi_instream_group_get_map(h_stream, mo, mi); -} - static u16 handle_error(u16 err, int line, char *filename) { if (err) @@ -547,9 +537,7 @@ static void snd_card_asihpi_pcm_int_start(struct snd_pcm_substream *substream) card = snd_pcm_substream_chip(substream); WARN_ON(in_interrupt()); - tasklet_disable(&card->t); card->llmode_streampriv = dpcm; - tasklet_enable(&card->t); hpi_handle_error(hpi_adapter_set_property(card->hpi->adapter->index, HPI_ADAPTER_PROPERTY_IRQ_RATE, @@ -565,13 +553,7 @@ static void snd_card_asihpi_pcm_int_stop(struct snd_pcm_substream *substream) hpi_handle_error(hpi_adapter_set_property(card->hpi->adapter->index, HPI_ADAPTER_PROPERTY_IRQ_RATE, 0, 0)); - if (in_interrupt()) - card->llmode_streampriv = NULL; - else { - tasklet_disable(&card->t); - card->llmode_streampriv = NULL; - tasklet_enable(&card->t); - } + card->llmode_streampriv = NULL; } static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream, @@ -921,10 +903,9 @@ static void snd_card_asihpi_timer_function(struct timer_list *t) add_timer(&dpcm->timer); } -static void snd_card_asihpi_int_task(struct tasklet_struct *t) +static void snd_card_asihpi_isr(struct hpi_adapter *a) { - struct snd_card_asihpi *asihpi = from_tasklet(asihpi, t, t); - struct hpi_adapter *a = asihpi->hpi; + struct snd_card_asihpi *asihpi; WARN_ON(!a || !a->snd_card || !a->snd_card->private_data); asihpi = (struct snd_card_asihpi *)a->snd_card->private_data; @@ -933,15 +914,6 @@ static void snd_card_asihpi_int_task(struct tasklet_struct *t) &asihpi->llmode_streampriv->timer); } -static void snd_card_asihpi_isr(struct hpi_adapter *a) -{ - struct snd_card_asihpi *asihpi; - - WARN_ON(!a || !a->snd_card || !a->snd_card->private_data); - asihpi = (struct snd_card_asihpi *)a->snd_card->private_data; - tasklet_schedule(&asihpi->t); -} - /***************************** PLAYBACK OPS ****************/ static int snd_card_asihpi_playback_prepare(struct snd_pcm_substream * substream) @@ -2871,7 +2843,6 @@ static int snd_asihpi_probe(struct pci_dev *pci_dev, if (hpi->interrupt_mode) { asihpi->pcm_start = snd_card_asihpi_pcm_int_start; asihpi->pcm_stop = snd_card_asihpi_pcm_int_stop; - tasklet_setup(&asihpi->t, snd_card_asihpi_int_task); hpi->interrupt_callback = snd_card_asihpi_isr; } else { asihpi->pcm_start = snd_card_asihpi_pcm_timer_start; @@ -2960,14 +2931,12 @@ __nodev: static void snd_asihpi_remove(struct pci_dev *pci_dev) { struct hpi_adapter *hpi = pci_get_drvdata(pci_dev); - struct snd_card_asihpi *asihpi = hpi->snd_card->private_data; /* Stop interrupts */ if (hpi->interrupt_mode) { hpi->interrupt_callback = NULL; hpi_handle_error(hpi_adapter_set_property(hpi->adapter->index, HPI_ADAPTER_PROPERTY_IRQ_RATE, 0, 0)); - tasklet_kill(&asihpi->t); } snd_card_free(hpi->snd_card); diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c index 9790f5108a16..bb31b7fe867d 100644 --- a/sound/pci/asihpi/hpioctl.c +++ b/sound/pci/asihpi/hpioctl.c @@ -329,11 +329,20 @@ static irqreturn_t asihpi_isr(int irq, void *dev_id) asihpi_irq_count, a->adapter->type, a->adapter->index); */ if (a->interrupt_callback) - a->interrupt_callback(a); + return IRQ_WAKE_THREAD; return IRQ_HANDLED; } +static irqreturn_t asihpi_isr_thread(int irq, void *dev_id) +{ + struct hpi_adapter *a = dev_id; + + if (a->interrupt_callback) + a->interrupt_callback(a); + return IRQ_HANDLED; +} + int asihpi_adapter_probe(struct pci_dev *pci_dev, const struct pci_device_id *pci_id) { @@ -478,8 +487,9 @@ int asihpi_adapter_probe(struct pci_dev *pci_dev, } /* Note: request_irq calls asihpi_isr here */ - if (request_irq(pci_dev->irq, asihpi_isr, IRQF_SHARED, - "asihpi", &adapters[adapter_index])) { + if (request_threaded_irq(pci_dev->irq, asihpi_isr, + asihpi_isr_thread, IRQF_SHARED, + "asihpi", &adapters[adapter_index])) { dev_err(&pci_dev->dev, "request_irq(%d) failed\n", pci_dev->irq); goto err; diff --git a/sound/pci/asihpi/hpios.h b/sound/pci/asihpi/hpios.h index 26f7cf455a1e..9e551bc46264 100644 --- a/sound/pci/asihpi/hpios.h +++ b/sound/pci/asihpi/hpios.h @@ -67,7 +67,7 @@ struct hpi_ioctl_linux { }; /* Conflict?: H is already used by a number of drivers hid, bluetooth hci, - and some sound drivers sb16, hdsp, emu10k. AFAIK 0xFC is ununsed command + and some sound drivers sb16, hdsp, emu10k. AFAIK 0xFC is unused command */ #define HPI_IOCTL_LINUX _IOWR('H', 0xFC, struct hpi_ioctl_linux) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 824f4ac1a8ce..4dc01647753c 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -350,7 +350,7 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec, */ if (!cfg->line_outs && cfg->hp_outs > 1 && !(cond_flags & HDA_PINCFG_NO_HP_FIXUP)) { - int i = 0; + i = 0; while (i < cfg->hp_outs) { /* The real HPs should have the sequence 0x0f */ if ((hp_out[i].seq & 0x0f) == 0x0f) { diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 36a9dbc33aa0..61e495187b1a 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -368,7 +368,8 @@ enum { #define CONTROLLER_IN_GPU(pci) (((pci)->device == 0x0a0c) || \ ((pci)->device == 0x0c0c) || \ ((pci)->device == 0x0d0c) || \ - ((pci)->device == 0x160c)) + ((pci)->device == 0x160c) || \ + ((pci)->device == 0x490d)) #define IS_BXT(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x5a98) @@ -2493,6 +2494,9 @@ static const struct pci_device_id azx_ids[] = { /* Tigerlake-H */ { PCI_DEVICE(0x8086, 0x43c8), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* DG1 */ + { PCI_DEVICE(0x8086, 0x490d), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Elkhart Lake */ { PCI_DEVICE(0x8086, 0x4b55), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h index 727b6d3ba454..8ceaf0ef5df1 100644 --- a/sound/pci/hda/hda_jack.h +++ b/sound/pci/hda/hda_jack.h @@ -77,7 +77,7 @@ int snd_hda_jack_detect_enable(struct hda_codec *codec, hda_nid_t nid, struct hda_jack_callback * snd_hda_jack_detect_enable_callback_mst(struct hda_codec *codec, hda_nid_t nid, - int dev_id, hda_jack_callback_fn cb); + int dev_id, hda_jack_callback_fn func); /** * snd_hda_jack_detect_enable - enable the jack-detection diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 8c28b1022f49..5beb8aa44ecd 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -100,7 +100,7 @@ int snd_hda_mixer_amp_volume_get(struct snd_kcontrol *kcontrol, int snd_hda_mixer_amp_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag, - unsigned int size, unsigned int __user *tlv); + unsigned int size, unsigned int __user *_tlv); int snd_hda_mixer_amp_switch_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); int snd_hda_mixer_amp_switch_get(struct snd_kcontrol *kcontrol, @@ -119,7 +119,7 @@ int snd_hda_mixer_amp_switch_put_beep(struct snd_kcontrol *kcontrol, int snd_hda_codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch, int dir, int idx, int mask, int val); int snd_hda_codec_amp_stereo(struct hda_codec *codec, hda_nid_t nid, - int dir, int idx, int mask, int val); + int direction, int idx, int mask, int val); int snd_hda_codec_amp_init(struct hda_codec *codec, hda_nid_t nid, int ch, int direction, int idx, int mask, int val); int snd_hda_codec_amp_init_stereo(struct hda_codec *codec, hda_nid_t nid, @@ -198,7 +198,7 @@ int snd_hda_input_mux_put(struct hda_codec *codec, unsigned int *cur_val); int snd_hda_add_imux_item(struct hda_codec *codec, struct hda_input_mux *imux, const char *label, - int index, int *type_index_ret); + int index, int *type_idx); /* * Multi-channel / digital-out PCM helper @@ -642,7 +642,7 @@ unsigned int snd_hda_codec_eapd_power_filter(struct hda_codec *codec, */ int snd_hda_enum_helper_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo, - int num_entries, const char * const *texts); + int num_items, const char * const *texts); #define snd_hda_enum_bool_helper_info(kcontrol, uinfo) \ snd_hda_enum_helper_info(kcontrol, uinfo, 0, NULL) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index b7dbf2e7f77a..9779978e4bc7 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -38,6 +38,8 @@ #define FLOAT_ONE 0x3f800000 #define FLOAT_TWO 0x40000000 #define FLOAT_THREE 0x40400000 +#define FLOAT_FIVE 0x40a00000 +#define FLOAT_SIX 0x40c00000 #define FLOAT_EIGHT 0x41000000 #define FLOAT_MINUS_5 0xc0a00000 @@ -80,11 +82,11 @@ MODULE_FIRMWARE(R3DI_EFX_FILE); static const char *const dirstr[2] = { "Playback", "Capture" }; -#define NUM_OF_OUTPUTS 3 +#define NUM_OF_OUTPUTS 2 +static const char *const out_type_str[2] = { "Speakers", "Headphone" }; enum { SPEAKER_OUT, HEADPHONE_OUT, - SURROUND_OUT }; enum { @@ -143,7 +145,12 @@ enum { MIC_BOOST_ENUM, AE5_HEADPHONE_GAIN_ENUM, AE5_SOUND_FILTER_ENUM, - ZXR_HEADPHONE_GAIN + ZXR_HEADPHONE_GAIN, + SPEAKER_CHANNEL_CFG_ENUM, + SPEAKER_FULL_RANGE_FRONT, + SPEAKER_FULL_RANGE_REAR, + BASS_REDIRECTION, + BASS_REDIRECTION_XOVER, #define EFFECTS_COUNT (EFFECT_END_NID - EFFECT_START_NID) }; @@ -589,46 +596,108 @@ static const struct ct_eq_preset ca0132_alt_eq_presets[] = { } }; -/* DSP command sequences for ca0132_alt_select_out */ -#define ALT_OUT_SET_MAX_COMMANDS 9 /* Max number of commands in sequence */ -struct ca0132_alt_out_set { - char *name; /*preset name*/ - unsigned char commands; - unsigned int mids[ALT_OUT_SET_MAX_COMMANDS]; - unsigned int reqs[ALT_OUT_SET_MAX_COMMANDS]; - unsigned int vals[ALT_OUT_SET_MAX_COMMANDS]; +/* + * DSP reqs for handling full-range speakers/bass redirection. If a speaker is + * set as not being full range, and bass redirection is enabled, all + * frequencies below the crossover frequency are redirected to the LFE + * channel. If the surround configuration has no LFE channel, this can't be + * enabled. X-Bass must be disabled when using these. + */ +enum speaker_range_reqs { + SPEAKER_BASS_REDIRECT = 0x15, + SPEAKER_BASS_REDIRECT_XOVER_FREQ = 0x16, + /* Between 0x16-0x1a are the X-Bass reqs. */ + SPEAKER_FULL_RANGE_FRONT_L_R = 0x1a, + SPEAKER_FULL_RANGE_CENTER_LFE = 0x1b, + SPEAKER_FULL_RANGE_REAR_L_R = 0x1c, + SPEAKER_FULL_RANGE_SURROUND_L_R = 0x1d, + SPEAKER_BASS_REDIRECT_SUB_GAIN = 0x1e, +}; + +/* + * Definitions for the DSP req's to handle speaker tuning. These all belong to + * module ID 0x96, the output effects module. + */ +enum speaker_tuning_reqs { + /* + * Currently, this value is always set to 0.0f. However, on Windows, + * when selecting certain headphone profiles on the new Sound Blaster + * connect software, the QUERY_SPEAKER_EQ_ADDRESS req on mid 0x80 is + * sent. This gets the speaker EQ address area, which is then used to + * send over (presumably) an equalizer profile for the specific + * headphone setup. It is sent using the same method the DSP + * firmware is uploaded with, which I believe is why the 'ctspeq.bin' + * file exists in linux firmware tree but goes unused. It would also + * explain why the QUERY_SPEAKER_EQ_ADDRESS req is defined but unused. + * Once this profile is sent over, SPEAKER_TUNING_USE_SPEAKER_EQ is + * set to 1.0f. + */ + SPEAKER_TUNING_USE_SPEAKER_EQ = 0x1f, + SPEAKER_TUNING_ENABLE_CENTER_EQ = 0x20, + SPEAKER_TUNING_FRONT_LEFT_VOL_LEVEL = 0x21, + SPEAKER_TUNING_FRONT_RIGHT_VOL_LEVEL = 0x22, + SPEAKER_TUNING_CENTER_VOL_LEVEL = 0x23, + SPEAKER_TUNING_LFE_VOL_LEVEL = 0x24, + SPEAKER_TUNING_REAR_LEFT_VOL_LEVEL = 0x25, + SPEAKER_TUNING_REAR_RIGHT_VOL_LEVEL = 0x26, + SPEAKER_TUNING_SURROUND_LEFT_VOL_LEVEL = 0x27, + SPEAKER_TUNING_SURROUND_RIGHT_VOL_LEVEL = 0x28, + /* + * Inversion is used when setting headphone virtualization to line + * out. Not sure why this is, but it's the only place it's ever used. + */ + SPEAKER_TUNING_FRONT_LEFT_INVERT = 0x29, + SPEAKER_TUNING_FRONT_RIGHT_INVERT = 0x2a, + SPEAKER_TUNING_CENTER_INVERT = 0x2b, + SPEAKER_TUNING_LFE_INVERT = 0x2c, + SPEAKER_TUNING_REAR_LEFT_INVERT = 0x2d, + SPEAKER_TUNING_REAR_RIGHT_INVERT = 0x2e, + SPEAKER_TUNING_SURROUND_LEFT_INVERT = 0x2f, + SPEAKER_TUNING_SURROUND_RIGHT_INVERT = 0x30, + /* Delay is used when setting surround speaker distance in Windows. */ + SPEAKER_TUNING_FRONT_LEFT_DELAY = 0x31, + SPEAKER_TUNING_FRONT_RIGHT_DELAY = 0x32, + SPEAKER_TUNING_CENTER_DELAY = 0x33, + SPEAKER_TUNING_LFE_DELAY = 0x34, + SPEAKER_TUNING_REAR_LEFT_DELAY = 0x35, + SPEAKER_TUNING_REAR_RIGHT_DELAY = 0x36, + SPEAKER_TUNING_SURROUND_LEFT_DELAY = 0x37, + SPEAKER_TUNING_SURROUND_RIGHT_DELAY = 0x38, + /* Of these two, only mute seems to ever be used. */ + SPEAKER_TUNING_MAIN_VOLUME = 0x39, + SPEAKER_TUNING_MUTE = 0x3a, +}; + +/* Surround output channel count configuration structures. */ +#define SPEAKER_CHANNEL_CFG_COUNT 5 +enum { + SPEAKER_CHANNELS_2_0, + SPEAKER_CHANNELS_2_1, + SPEAKER_CHANNELS_4_0, + SPEAKER_CHANNELS_4_1, + SPEAKER_CHANNELS_5_1, +}; + +struct ca0132_alt_speaker_channel_cfg { + char *name; + unsigned int val; }; -static const struct ca0132_alt_out_set alt_out_presets[] = { - { .name = "Line Out", - .commands = 7, - .mids = { 0x96, 0x96, 0x96, 0x8F, - 0x96, 0x96, 0x96 }, - .reqs = { 0x19, 0x17, 0x18, 0x01, - 0x1F, 0x15, 0x3A }, - .vals = { 0x3F000000, 0x42A00000, 0x00000000, - 0x00000000, 0x00000000, 0x00000000, - 0x00000000 } +static const struct ca0132_alt_speaker_channel_cfg speaker_channel_cfgs[] = { + { .name = "2.0", + .val = FLOAT_ONE }, - { .name = "Headphone", - .commands = 7, - .mids = { 0x96, 0x96, 0x96, 0x8F, - 0x96, 0x96, 0x96 }, - .reqs = { 0x19, 0x17, 0x18, 0x01, - 0x1F, 0x15, 0x3A }, - .vals = { 0x3F000000, 0x42A00000, 0x00000000, - 0x00000000, 0x00000000, 0x00000000, - 0x00000000 } + { .name = "2.1", + .val = FLOAT_TWO }, - { .name = "Surround", - .commands = 8, - .mids = { 0x96, 0x8F, 0x96, 0x96, - 0x96, 0x96, 0x96, 0x96 }, - .reqs = { 0x18, 0x01, 0x1F, 0x15, - 0x3A, 0x1A, 0x1B, 0x1C }, - .vals = { 0x00000000, 0x00000000, 0x00000000, - 0x00000000, 0x00000000, 0x00000000, - 0x00000000, 0x00000000 } + { .name = "4.0", + .val = FLOAT_FIVE + }, + { .name = "4.1", + .val = FLOAT_SIX + }, + { .name = "5.1", + .val = FLOAT_EIGHT } }; @@ -658,26 +727,29 @@ static const struct ct_dsp_volume_ctl ca0132_alt_vol_ctls[] = { }; /* Values for ca0113_mmio_command_set for selecting output. */ -#define AE5_CA0113_OUT_SET_COMMANDS 6 -struct ae5_ca0113_output_set { - unsigned int group[AE5_CA0113_OUT_SET_COMMANDS]; - unsigned int target[AE5_CA0113_OUT_SET_COMMANDS]; - unsigned int vals[AE5_CA0113_OUT_SET_COMMANDS]; +#define AE_CA0113_OUT_SET_COMMANDS 6 +struct ae_ca0113_output_set { + unsigned int group[AE_CA0113_OUT_SET_COMMANDS]; + unsigned int target[AE_CA0113_OUT_SET_COMMANDS]; + unsigned int vals[NUM_OF_OUTPUTS][AE_CA0113_OUT_SET_COMMANDS]; }; -static const struct ae5_ca0113_output_set ae5_ca0113_output_presets[] = { - { .group = { 0x30, 0x30, 0x48, 0x48, 0x48, 0x30 }, - .target = { 0x2e, 0x30, 0x0d, 0x17, 0x19, 0x32 }, - .vals = { 0x00, 0x00, 0x40, 0x00, 0x00, 0x3f } - }, - { .group = { 0x30, 0x30, 0x48, 0x48, 0x48, 0x30 }, - .target = { 0x2e, 0x30, 0x0d, 0x17, 0x19, 0x32 }, - .vals = { 0x3f, 0x3f, 0x00, 0x00, 0x00, 0x00 } - }, - { .group = { 0x30, 0x30, 0x48, 0x48, 0x48, 0x30 }, - .target = { 0x2e, 0x30, 0x0d, 0x17, 0x19, 0x32 }, - .vals = { 0x00, 0x00, 0x40, 0x00, 0x00, 0x3f } - } +static const struct ae_ca0113_output_set ae5_ca0113_output_presets = { + .group = { 0x30, 0x30, 0x48, 0x48, 0x48, 0x30 }, + .target = { 0x2e, 0x30, 0x0d, 0x17, 0x19, 0x32 }, + /* Speakers. */ + .vals = { { 0x00, 0x00, 0x40, 0x00, 0x00, 0x3f }, + /* Headphones. */ + { 0x3f, 0x3f, 0x00, 0x00, 0x00, 0x00 } }, +}; + +static const struct ae_ca0113_output_set ae7_ca0113_output_presets = { + .group = { 0x30, 0x30, 0x48, 0x48, 0x48, 0x30 }, + .target = { 0x2e, 0x30, 0x0d, 0x17, 0x19, 0x32 }, + /* Speakers. */ + .vals = { { 0x00, 0x00, 0x40, 0x00, 0x00, 0x3f }, + /* Headphones. */ + { 0x3f, 0x3f, 0x00, 0x00, 0x02, 0x00 } }, }; /* ae5 ca0113 command sequences to set headphone gain levels. */ @@ -1009,8 +1081,12 @@ struct ca0132_spec { /* ca0132_alt control related values */ unsigned char in_enum_val; unsigned char out_enum_val; + unsigned char channel_cfg_val; + unsigned char speaker_range_val[2]; unsigned char mic_boost_enum_val; unsigned char smart_volume_setting; + unsigned char bass_redirection_val; + long bass_redirect_xover_freq; long fx_ctl_val[EFFECT_LEVEL_SLIDERS]; long xbass_xover_freq; long eq_preset_val; @@ -1065,6 +1141,7 @@ enum { QUIRK_R3DI, QUIRK_R3D, QUIRK_AE5, + QUIRK_AE7, }; #ifdef CONFIG_PCI @@ -1168,6 +1245,20 @@ static const struct hda_pintbl r3di_pincfgs[] = { {} }; +static const struct hda_pintbl ae7_pincfgs[] = { + { 0x0b, 0x01017010 }, + { 0x0c, 0x014510f0 }, + { 0x0d, 0x414510f0 }, + { 0x0e, 0x01c520f0 }, + { 0x0f, 0x01017114 }, + { 0x10, 0x01017011 }, + { 0x11, 0x018170ff }, + { 0x12, 0x01a170f0 }, + { 0x13, 0x908700f0 }, + { 0x18, 0x500000f0 }, + {} +}; + static const struct snd_pci_quirk ca0132_quirks[] = { SND_PCI_QUIRK(0x1028, 0x057b, "Alienware M17x R4", QUIRK_ALIENWARE_M17XR4), SND_PCI_QUIRK(0x1028, 0x0685, "Alienware 15 2015", QUIRK_ALIENWARE), @@ -1184,9 +1275,203 @@ 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, 0x0081, "Sound Blaster AE-7", QUIRK_AE7), {} }; +/* Output selection quirk info structures. */ +#define MAX_QUIRK_MMIO_GPIO_SET_VALS 3 +#define MAX_QUIRK_SCP_SET_VALS 2 +struct ca0132_alt_out_set_info { + unsigned int dac2port; /* ParamID 0x0d value. */ + + bool has_hda_gpio; + char hda_gpio_pin; + char hda_gpio_set; + + unsigned int mmio_gpio_count; + char mmio_gpio_pin[MAX_QUIRK_MMIO_GPIO_SET_VALS]; + char mmio_gpio_set[MAX_QUIRK_MMIO_GPIO_SET_VALS]; + + unsigned int scp_cmds_count; + unsigned int scp_cmd_mid[MAX_QUIRK_SCP_SET_VALS]; + unsigned int scp_cmd_req[MAX_QUIRK_SCP_SET_VALS]; + unsigned int scp_cmd_val[MAX_QUIRK_SCP_SET_VALS]; + + bool has_chipio_write; + unsigned int chipio_write_addr; + unsigned int chipio_write_data; +}; + +struct ca0132_alt_out_set_quirk_data { + int quirk_id; + + bool has_headphone_gain; + bool is_ae_series; + + struct ca0132_alt_out_set_info out_set_info[NUM_OF_OUTPUTS]; +}; + +static const struct ca0132_alt_out_set_quirk_data quirk_out_set_data[] = { + { .quirk_id = QUIRK_R3DI, + .has_headphone_gain = false, + .is_ae_series = false, + .out_set_info = { + /* Speakers. */ + { .dac2port = 0x24, + .has_hda_gpio = true, + .hda_gpio_pin = 2, + .hda_gpio_set = 1, + .mmio_gpio_count = 0, + .scp_cmds_count = 0, + .has_chipio_write = false, + }, + /* Headphones. */ + { .dac2port = 0x21, + .has_hda_gpio = true, + .hda_gpio_pin = 2, + .hda_gpio_set = 0, + .mmio_gpio_count = 0, + .scp_cmds_count = 0, + .has_chipio_write = false, + } }, + }, + { .quirk_id = QUIRK_R3D, + .has_headphone_gain = false, + .is_ae_series = false, + .out_set_info = { + /* Speakers. */ + { .dac2port = 0x24, + .has_hda_gpio = false, + .mmio_gpio_count = 1, + .mmio_gpio_pin = { 1 }, + .mmio_gpio_set = { 1 }, + .scp_cmds_count = 0, + .has_chipio_write = false, + }, + /* Headphones. */ + { .dac2port = 0x21, + .has_hda_gpio = false, + .mmio_gpio_count = 1, + .mmio_gpio_pin = { 1 }, + .mmio_gpio_set = { 0 }, + .scp_cmds_count = 0, + .has_chipio_write = false, + } }, + }, + { .quirk_id = QUIRK_SBZ, + .has_headphone_gain = false, + .is_ae_series = false, + .out_set_info = { + /* Speakers. */ + { .dac2port = 0x18, + .has_hda_gpio = false, + .mmio_gpio_count = 3, + .mmio_gpio_pin = { 7, 4, 1 }, + .mmio_gpio_set = { 0, 1, 1 }, + .scp_cmds_count = 0, + .has_chipio_write = false, }, + /* Headphones. */ + { .dac2port = 0x12, + .has_hda_gpio = false, + .mmio_gpio_count = 3, + .mmio_gpio_pin = { 7, 4, 1 }, + .mmio_gpio_set = { 1, 1, 0 }, + .scp_cmds_count = 0, + .has_chipio_write = false, + } }, + }, + { .quirk_id = QUIRK_ZXR, + .has_headphone_gain = true, + .is_ae_series = false, + .out_set_info = { + /* Speakers. */ + { .dac2port = 0x24, + .has_hda_gpio = false, + .mmio_gpio_count = 3, + .mmio_gpio_pin = { 2, 3, 5 }, + .mmio_gpio_set = { 1, 1, 0 }, + .scp_cmds_count = 0, + .has_chipio_write = false, + }, + /* Headphones. */ + { .dac2port = 0x21, + .has_hda_gpio = false, + .mmio_gpio_count = 3, + .mmio_gpio_pin = { 2, 3, 5 }, + .mmio_gpio_set = { 0, 1, 1 }, + .scp_cmds_count = 0, + .has_chipio_write = false, + } }, + }, + { .quirk_id = QUIRK_AE5, + .has_headphone_gain = true, + .is_ae_series = true, + .out_set_info = { + /* Speakers. */ + { .dac2port = 0xa4, + .has_hda_gpio = false, + .mmio_gpio_count = 0, + .scp_cmds_count = 2, + .scp_cmd_mid = { 0x96, 0x96 }, + .scp_cmd_req = { SPEAKER_TUNING_FRONT_LEFT_INVERT, + SPEAKER_TUNING_FRONT_RIGHT_INVERT }, + .scp_cmd_val = { FLOAT_ZERO, FLOAT_ZERO }, + .has_chipio_write = true, + .chipio_write_addr = 0x0018b03c, + .chipio_write_data = 0x00000012 + }, + /* Headphones. */ + { .dac2port = 0xa1, + .has_hda_gpio = false, + .mmio_gpio_count = 0, + .scp_cmds_count = 2, + .scp_cmd_mid = { 0x96, 0x96 }, + .scp_cmd_req = { SPEAKER_TUNING_FRONT_LEFT_INVERT, + SPEAKER_TUNING_FRONT_RIGHT_INVERT }, + .scp_cmd_val = { FLOAT_ONE, FLOAT_ONE }, + .has_chipio_write = true, + .chipio_write_addr = 0x0018b03c, + .chipio_write_data = 0x00000012 + } }, + }, + { .quirk_id = QUIRK_AE7, + .has_headphone_gain = true, + .is_ae_series = true, + .out_set_info = { + /* Speakers. */ + { .dac2port = 0x58, + .has_hda_gpio = false, + .mmio_gpio_count = 1, + .mmio_gpio_pin = { 0 }, + .mmio_gpio_set = { 1 }, + .scp_cmds_count = 2, + .scp_cmd_mid = { 0x96, 0x96 }, + .scp_cmd_req = { SPEAKER_TUNING_FRONT_LEFT_INVERT, + SPEAKER_TUNING_FRONT_RIGHT_INVERT }, + .scp_cmd_val = { FLOAT_ZERO, FLOAT_ZERO }, + .has_chipio_write = true, + .chipio_write_addr = 0x0018b03c, + .chipio_write_data = 0x00000000 + }, + /* Headphones. */ + { .dac2port = 0x58, + .has_hda_gpio = false, + .mmio_gpio_count = 1, + .mmio_gpio_pin = { 0 }, + .mmio_gpio_set = { 1 }, + .scp_cmds_count = 2, + .scp_cmd_mid = { 0x96, 0x96 }, + .scp_cmd_req = { SPEAKER_TUNING_FRONT_LEFT_INVERT, + SPEAKER_TUNING_FRONT_RIGHT_INVERT }, + .scp_cmd_val = { FLOAT_ONE, FLOAT_ONE }, + .has_chipio_write = true, + .chipio_write_addr = 0x0018b03c, + .chipio_write_data = 0x00000010 + } }, + } +}; + /* * CA0132 codec access */ @@ -3339,6 +3624,7 @@ static void ca0132_gpio_init(struct hda_codec *codec) switch (ca0132_quirk(spec)) { case QUIRK_SBZ: case QUIRK_AE5: + case QUIRK_AE7: snd_hda_codec_write(codec, 0x01, 0, 0x793, 0x00); snd_hda_codec_write(codec, 0x01, 0, 0x794, 0x53); snd_hda_codec_write(codec, 0x01, 0, 0x790, 0x23); @@ -3444,26 +3730,6 @@ static void r3di_gpio_mic_set(struct hda_codec *codec, AC_VERB_SET_GPIO_DATA, cur_gpio); } -static void r3di_gpio_out_set(struct hda_codec *codec, - enum r3di_out_select cur_out) -{ - unsigned int cur_gpio; - - /* Get the current GPIO Data setup */ - cur_gpio = snd_hda_codec_read(codec, 0x01, 0, AC_VERB_GET_GPIO_DATA, 0); - - switch (cur_out) { - case R3DI_HEADPHONE_OUT: - cur_gpio &= ~(1 << R3DI_OUT_SELECT_BIT); - break; - case R3DI_LINE_OUT: - cur_gpio |= (1 << R3DI_OUT_SELECT_BIT); - break; - } - snd_hda_codec_write(codec, codec->core.afg, 0, - AC_VERB_SET_GPIO_DATA, cur_gpio); -} - static void r3di_gpio_dsp_status_set(struct hda_codec *codec, enum r3di_dsp_status dsp_status) { @@ -4159,135 +4425,198 @@ static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val); static void ae5_mmio_select_out(struct hda_codec *codec) { struct ca0132_spec *spec = codec->spec; + const struct ae_ca0113_output_set *out_cmds; unsigned int i; - for (i = 0; i < AE5_CA0113_OUT_SET_COMMANDS; i++) - ca0113_mmio_command_set(codec, - ae5_ca0113_output_presets[spec->cur_out_type].group[i], - ae5_ca0113_output_presets[spec->cur_out_type].target[i], - ae5_ca0113_output_presets[spec->cur_out_type].vals[i]); + if (ca0132_quirk(spec) == QUIRK_AE5) + out_cmds = &ae5_ca0113_output_presets; + else + out_cmds = &ae7_ca0113_output_presets; + + for (i = 0; i < AE_CA0113_OUT_SET_COMMANDS; i++) + ca0113_mmio_command_set(codec, out_cmds->group[i], + out_cmds->target[i], + out_cmds->vals[spec->cur_out_type][i]); +} + +static int ca0132_alt_set_full_range_speaker(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + int quirk = ca0132_quirk(spec); + unsigned int tmp; + int err; + + /* 2.0/4.0 setup has no LFE channel, so setting full-range does nothing. */ + if (spec->channel_cfg_val == SPEAKER_CHANNELS_4_0 + || spec->channel_cfg_val == SPEAKER_CHANNELS_2_0) + return 0; + + /* Set front L/R full range. Zero for full-range, one for redirection. */ + tmp = spec->speaker_range_val[0] ? FLOAT_ZERO : FLOAT_ONE; + err = dspio_set_uint_param(codec, 0x96, + SPEAKER_FULL_RANGE_FRONT_L_R, tmp); + if (err < 0) + return err; + + /* When setting full-range rear, both rear and center/lfe are set. */ + tmp = spec->speaker_range_val[1] ? FLOAT_ZERO : FLOAT_ONE; + err = dspio_set_uint_param(codec, 0x96, + SPEAKER_FULL_RANGE_CENTER_LFE, tmp); + if (err < 0) + return err; + + err = dspio_set_uint_param(codec, 0x96, + SPEAKER_FULL_RANGE_REAR_L_R, tmp); + if (err < 0) + return err; + + /* + * Only the AE series cards set this value when setting full-range, + * and it's always 1.0f. + */ + if (quirk == QUIRK_AE5 || quirk == QUIRK_AE7) { + err = dspio_set_uint_param(codec, 0x96, + SPEAKER_FULL_RANGE_SURROUND_L_R, FLOAT_ONE); + if (err < 0) + return err; + } + + return 0; +} + +static int ca0132_alt_surround_set_bass_redirection(struct hda_codec *codec, + bool val) +{ + struct ca0132_spec *spec = codec->spec; + unsigned int tmp; + int err; + + if (val && spec->channel_cfg_val != SPEAKER_CHANNELS_4_0 && + spec->channel_cfg_val != SPEAKER_CHANNELS_2_0) + tmp = FLOAT_ONE; + else + tmp = FLOAT_ZERO; + + err = dspio_set_uint_param(codec, 0x96, SPEAKER_BASS_REDIRECT, tmp); + if (err < 0) + return err; + + /* If it is enabled, make sure to set the crossover frequency. */ + if (tmp) { + tmp = float_xbass_xover_lookup[spec->xbass_xover_freq]; + err = dspio_set_uint_param(codec, 0x96, + SPEAKER_BASS_REDIRECT_XOVER_FREQ, tmp); + if (err < 0) + return err; + } + + return 0; } /* * These are the commands needed to setup output on each of the different card * types. */ -static void ca0132_alt_select_out_quirk_handler(struct hda_codec *codec) +static void ca0132_alt_select_out_get_quirk_data(struct hda_codec *codec, + const struct ca0132_alt_out_set_quirk_data **quirk_data) { struct ca0132_spec *spec = codec->spec; - unsigned int tmp; + int quirk = ca0132_quirk(spec); + unsigned int i; - switch (spec->cur_out_type) { - case SPEAKER_OUT: - switch (ca0132_quirk(spec)) { - case QUIRK_SBZ: - ca0113_mmio_gpio_set(codec, 7, false); - ca0113_mmio_gpio_set(codec, 4, true); - ca0113_mmio_gpio_set(codec, 1, true); - chipio_set_control_param(codec, 0x0d, 0x18); - break; - case QUIRK_ZXR: - ca0113_mmio_gpio_set(codec, 2, true); - ca0113_mmio_gpio_set(codec, 3, true); - ca0113_mmio_gpio_set(codec, 5, false); - zxr_headphone_gain_set(codec, 0); - chipio_set_control_param(codec, 0x0d, 0x24); - break; - case QUIRK_R3DI: - chipio_set_control_param(codec, 0x0d, 0x24); - r3di_gpio_out_set(codec, R3DI_LINE_OUT); - break; - case QUIRK_R3D: - chipio_set_control_param(codec, 0x0d, 0x24); - ca0113_mmio_gpio_set(codec, 1, true); - break; - case QUIRK_AE5: - ae5_mmio_select_out(codec); - ae5_headphone_gain_set(codec, 2); - tmp = FLOAT_ZERO; - dspio_set_uint_param(codec, 0x96, 0x29, tmp); - dspio_set_uint_param(codec, 0x96, 0x2a, tmp); - chipio_set_control_param(codec, 0x0d, 0xa4); - chipio_write(codec, 0x18b03c, 0x00000012); - break; - default: - break; + *quirk_data = NULL; + for (i = 0; i < ARRAY_SIZE(quirk_out_set_data); i++) { + if (quirk_out_set_data[i].quirk_id == quirk) { + *quirk_data = &quirk_out_set_data[i]; + return; } - break; - case HEADPHONE_OUT: - switch (ca0132_quirk(spec)) { - case QUIRK_SBZ: - ca0113_mmio_gpio_set(codec, 7, true); - ca0113_mmio_gpio_set(codec, 4, true); - ca0113_mmio_gpio_set(codec, 1, false); - chipio_set_control_param(codec, 0x0d, 0x12); - break; - case QUIRK_ZXR: - ca0113_mmio_gpio_set(codec, 2, false); - ca0113_mmio_gpio_set(codec, 3, false); - ca0113_mmio_gpio_set(codec, 5, true); - zxr_headphone_gain_set(codec, spec->zxr_gain_set); - chipio_set_control_param(codec, 0x0d, 0x21); - break; - case QUIRK_R3DI: - chipio_set_control_param(codec, 0x0d, 0x21); - r3di_gpio_out_set(codec, R3DI_HEADPHONE_OUT); - break; - case QUIRK_R3D: - chipio_set_control_param(codec, 0x0d, 0x21); - ca0113_mmio_gpio_set(codec, 0x1, false); - break; - case QUIRK_AE5: - ae5_mmio_select_out(codec); - ae5_headphone_gain_set(codec, - spec->ae5_headphone_gain_val); - tmp = FLOAT_ONE; - dspio_set_uint_param(codec, 0x96, 0x29, tmp); - dspio_set_uint_param(codec, 0x96, 0x2a, tmp); - chipio_set_control_param(codec, 0x0d, 0xa1); - chipio_write(codec, 0x18b03c, 0x00000012); - break; - default: - break; + } +} + +static int ca0132_alt_select_out_quirk_set(struct hda_codec *codec) +{ + const struct ca0132_alt_out_set_quirk_data *quirk_data; + const struct ca0132_alt_out_set_info *out_info; + struct ca0132_spec *spec = codec->spec; + unsigned int i, gpio_data; + int err; + + ca0132_alt_select_out_get_quirk_data(codec, &quirk_data); + if (!quirk_data) + return 0; + + out_info = &quirk_data->out_set_info[spec->cur_out_type]; + if (quirk_data->is_ae_series) + ae5_mmio_select_out(codec); + + if (out_info->has_hda_gpio) { + gpio_data = snd_hda_codec_read(codec, codec->core.afg, 0, + AC_VERB_GET_GPIO_DATA, 0); + + if (out_info->hda_gpio_set) + gpio_data |= (1 << out_info->hda_gpio_pin); + else + gpio_data &= ~(1 << out_info->hda_gpio_pin); + + snd_hda_codec_write(codec, codec->core.afg, 0, + AC_VERB_SET_GPIO_DATA, gpio_data); + } + + if (out_info->mmio_gpio_count) { + for (i = 0; i < out_info->mmio_gpio_count; i++) { + ca0113_mmio_gpio_set(codec, out_info->mmio_gpio_pin[i], + out_info->mmio_gpio_set[i]); } - break; - case SURROUND_OUT: - switch (ca0132_quirk(spec)) { - case QUIRK_SBZ: - ca0113_mmio_gpio_set(codec, 7, false); - ca0113_mmio_gpio_set(codec, 4, true); - ca0113_mmio_gpio_set(codec, 1, true); - chipio_set_control_param(codec, 0x0d, 0x18); - break; - case QUIRK_ZXR: - ca0113_mmio_gpio_set(codec, 2, true); - ca0113_mmio_gpio_set(codec, 3, true); - ca0113_mmio_gpio_set(codec, 5, false); - zxr_headphone_gain_set(codec, 0); - chipio_set_control_param(codec, 0x0d, 0x24); - break; - case QUIRK_R3DI: - chipio_set_control_param(codec, 0x0d, 0x24); - r3di_gpio_out_set(codec, R3DI_LINE_OUT); - break; - case QUIRK_R3D: - ca0113_mmio_gpio_set(codec, 1, true); - chipio_set_control_param(codec, 0x0d, 0x24); - break; - case QUIRK_AE5: - ae5_mmio_select_out(codec); - ae5_headphone_gain_set(codec, 2); - tmp = FLOAT_ZERO; - dspio_set_uint_param(codec, 0x96, 0x29, tmp); - dspio_set_uint_param(codec, 0x96, 0x2a, tmp); - chipio_set_control_param(codec, 0x0d, 0xa4); - chipio_write(codec, 0x18b03c, 0x00000012); - break; - default: - break; + } + + if (out_info->scp_cmds_count) { + for (i = 0; i < out_info->scp_cmds_count; i++) { + err = dspio_set_uint_param(codec, + out_info->scp_cmd_mid[i], + out_info->scp_cmd_req[i], + out_info->scp_cmd_val[i]); + if (err < 0) + return err; } - break; } + + chipio_set_control_param(codec, 0x0d, out_info->dac2port); + + if (out_info->has_chipio_write) { + chipio_write(codec, out_info->chipio_write_addr, + out_info->chipio_write_data); + } + + if (quirk_data->has_headphone_gain) { + if (spec->cur_out_type != HEADPHONE_OUT) { + if (quirk_data->is_ae_series) + ae5_headphone_gain_set(codec, 2); + else + zxr_headphone_gain_set(codec, 0); + } else { + if (quirk_data->is_ae_series) + ae5_headphone_gain_set(codec, + spec->ae5_headphone_gain_val); + else + zxr_headphone_gain_set(codec, + spec->zxr_gain_set); + } + } + + return 0; +} + +static void ca0132_set_out_node_pincfg(struct hda_codec *codec, hda_nid_t nid, + bool out_enable, bool hp_enable) +{ + unsigned int pin_ctl; + + pin_ctl = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + + pin_ctl = hp_enable ? pin_ctl | PIN_HP_AMP : pin_ctl & ~PIN_HP_AMP; + pin_ctl = out_enable ? pin_ctl | PIN_OUT : pin_ctl & ~PIN_OUT; + snd_hda_set_pin_ctl(codec, nid, pin_ctl); } /* @@ -4296,18 +4625,14 @@ static void ca0132_alt_select_out_quirk_handler(struct hda_codec *codec) * output with an enumerated control "output source" if the auto detect * mute switch is set to off. If the auto detect mute switch is enabled, it * will detect either headphone or lineout(SPEAKER_OUT) from jack detection. - * It also adds the ability to auto-detect the front headphone port. The only - * way to select surround is to disable auto detect, and set Surround with the - * enumerated control. + * It also adds the ability to auto-detect the front headphone port. */ static int ca0132_alt_select_out(struct hda_codec *codec) { struct ca0132_spec *spec = codec->spec; - unsigned int pin_ctl; + unsigned int tmp, outfx_set; int jack_present; int auto_jack; - unsigned int i; - unsigned int tmp; int err; /* Default Headphone is rear headphone */ hda_nid_t headphone_nid = spec->out_pins[1]; @@ -4334,115 +4659,112 @@ static int ca0132_alt_select_out(struct hda_codec *codec) } else spec->cur_out_type = spec->out_enum_val; - /* Begin DSP output switch */ - tmp = FLOAT_ONE; - err = dspio_set_uint_param(codec, 0x96, 0x3A, tmp); + outfx_set = spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID]; + + /* Begin DSP output switch, mute DSP volume. */ + err = dspio_set_uint_param(codec, 0x96, SPEAKER_TUNING_MUTE, FLOAT_ONE); if (err < 0) goto exit; - ca0132_alt_select_out_quirk_handler(codec); + if (ca0132_alt_select_out_quirk_set(codec) < 0) + goto exit; switch (spec->cur_out_type) { case SPEAKER_OUT: codec_dbg(codec, "%s speaker\n", __func__); - /* disable headphone node */ - pin_ctl = snd_hda_codec_read(codec, spec->out_pins[1], 0, - AC_VERB_GET_PIN_WIDGET_CONTROL, 0); - snd_hda_set_pin_ctl(codec, spec->out_pins[1], - pin_ctl & ~PIN_HP); - /* enable line-out node */ - pin_ctl = snd_hda_codec_read(codec, spec->out_pins[0], 0, - AC_VERB_GET_PIN_WIDGET_CONTROL, 0); - snd_hda_set_pin_ctl(codec, spec->out_pins[0], - pin_ctl | PIN_OUT); /* Enable EAPD */ snd_hda_codec_write(codec, spec->out_pins[0], 0, AC_VERB_SET_EAPD_BTLENABLE, 0x01); - /* If PlayEnhancement is enabled, set different source */ - if (spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID]) - dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_ONE); + /* Disable headphone node. */ + ca0132_set_out_node_pincfg(codec, spec->out_pins[1], 0, 0); + /* Set front L-R to output. */ + ca0132_set_out_node_pincfg(codec, spec->out_pins[0], 1, 0); + /* Set Center/LFE to output. */ + ca0132_set_out_node_pincfg(codec, spec->out_pins[2], 1, 0); + /* Set rear surround to output. */ + ca0132_set_out_node_pincfg(codec, spec->out_pins[3], 1, 0); + + /* + * Without PlayEnhancement being enabled, if we've got a 2.0 + * setup, set it to floating point eight to disable any DSP + * processing effects. + */ + if (!outfx_set && spec->channel_cfg_val == SPEAKER_CHANNELS_2_0) + tmp = FLOAT_EIGHT; else - dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_EIGHT); + tmp = speaker_channel_cfgs[spec->channel_cfg_val].val; + + err = dspio_set_uint_param(codec, 0x80, 0x04, tmp); + if (err < 0) + goto exit; + break; case HEADPHONE_OUT: codec_dbg(codec, "%s hp\n", __func__); - snd_hda_codec_write(codec, spec->out_pins[0], 0, AC_VERB_SET_EAPD_BTLENABLE, 0x00); - /* disable speaker*/ - pin_ctl = snd_hda_codec_read(codec, spec->out_pins[0], 0, - AC_VERB_GET_PIN_WIDGET_CONTROL, 0); - snd_hda_set_pin_ctl(codec, spec->out_pins[0], - pin_ctl & ~PIN_HP); + /* Disable all speaker nodes. */ + ca0132_set_out_node_pincfg(codec, spec->out_pins[0], 0, 0); + ca0132_set_out_node_pincfg(codec, spec->out_pins[2], 0, 0); + ca0132_set_out_node_pincfg(codec, spec->out_pins[3], 0, 0); /* enable headphone, either front or rear */ - if (snd_hda_jack_detect(codec, spec->unsol_tag_front_hp)) headphone_nid = spec->out_pins[2]; else if (snd_hda_jack_detect(codec, spec->unsol_tag_hp)) headphone_nid = spec->out_pins[1]; - pin_ctl = snd_hda_codec_read(codec, headphone_nid, 0, - AC_VERB_GET_PIN_WIDGET_CONTROL, 0); - snd_hda_set_pin_ctl(codec, headphone_nid, - pin_ctl | PIN_HP); + ca0132_set_out_node_pincfg(codec, headphone_nid, 1, 1); - if (spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID]) - dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_ONE); + if (outfx_set) + err = dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_ONE); else - dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_ZERO); - break; - case SURROUND_OUT: - codec_dbg(codec, "%s surround\n", __func__); + err = dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_ZERO); - /* enable line out node */ - pin_ctl = snd_hda_codec_read(codec, spec->out_pins[0], 0, - AC_VERB_GET_PIN_WIDGET_CONTROL, 0); - snd_hda_set_pin_ctl(codec, spec->out_pins[0], - pin_ctl | PIN_OUT); - /* Disable headphone out */ - pin_ctl = snd_hda_codec_read(codec, spec->out_pins[1], 0, - AC_VERB_GET_PIN_WIDGET_CONTROL, 0); - snd_hda_set_pin_ctl(codec, spec->out_pins[1], - pin_ctl & ~PIN_HP); - /* Enable EAPD on line out */ - snd_hda_codec_write(codec, spec->out_pins[0], 0, - AC_VERB_SET_EAPD_BTLENABLE, 0x01); - /* enable center/lfe out node */ - pin_ctl = snd_hda_codec_read(codec, spec->out_pins[2], 0, - AC_VERB_GET_PIN_WIDGET_CONTROL, 0); - snd_hda_set_pin_ctl(codec, spec->out_pins[2], - pin_ctl | PIN_OUT); - /* Now set rear surround node as out. */ - pin_ctl = snd_hda_codec_read(codec, spec->out_pins[3], 0, - AC_VERB_GET_PIN_WIDGET_CONTROL, 0); - snd_hda_set_pin_ctl(codec, spec->out_pins[3], - pin_ctl | PIN_OUT); - - dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_EIGHT); + if (err < 0) + goto exit; break; } /* - * Surround always sets it's scp command to req 0x04 to FLOAT_EIGHT. - * With this set though, X_BASS cannot be enabled. So, if we have OutFX - * enabled, we need to make sure X_BASS is off, otherwise everything - * sounds all muffled. Running ca0132_effects_set with X_BASS as the - * effect should sort this out. + * If output effects are enabled, set the X-Bass effect value again to + * make sure that it's properly enabled/disabled for speaker + * configurations with an LFE channel. */ - if (spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID]) + if (outfx_set) ca0132_effects_set(codec, X_BASS, spec->effects_switch[X_BASS - EFFECT_START_NID]); - /* run through the output dsp commands for the selected output. */ - for (i = 0; i < alt_out_presets[spec->cur_out_type].commands; i++) { - err = dspio_set_uint_param(codec, - alt_out_presets[spec->cur_out_type].mids[i], - alt_out_presets[spec->cur_out_type].reqs[i], - alt_out_presets[spec->cur_out_type].vals[i]); + /* Set speaker EQ bypass attenuation to 0. */ + err = dspio_set_uint_param(codec, 0x8f, 0x01, FLOAT_ZERO); + if (err < 0) + goto exit; + + /* + * Although unused on all cards but the AE series, this is always set + * to zero when setting the output. + */ + err = dspio_set_uint_param(codec, 0x96, + SPEAKER_TUNING_USE_SPEAKER_EQ, FLOAT_ZERO); + if (err < 0) + goto exit; + + if (spec->cur_out_type == SPEAKER_OUT) + err = ca0132_alt_surround_set_bass_redirection(codec, + spec->bass_redirection_val); + else + err = ca0132_alt_surround_set_bass_redirection(codec, 0); + + /* Unmute DSP now that we're done with output selection. */ + err = dspio_set_uint_param(codec, 0x96, + SPEAKER_TUNING_MUTE, FLOAT_ZERO); + if (err < 0) + goto exit; + if (spec->cur_out_type == SPEAKER_OUT) { + err = ca0132_alt_set_full_range_speaker(codec); if (err < 0) goto exit; } @@ -4675,6 +4997,15 @@ 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; @@ -4720,6 +5051,14 @@ 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; } @@ -4729,7 +5068,10 @@ 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)) { @@ -4852,7 +5194,7 @@ static int ca0132_voicefx_set(struct hda_codec *codec, int enable) static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val) { struct ca0132_spec *spec = codec->spec; - unsigned int on, tmp; + unsigned int on, tmp, channel_cfg; int num_fx = OUT_EFFECTS_COUNT + IN_EFFECTS_COUNT; int err = 0; int idx = nid - EFFECT_START_NID; @@ -4865,8 +5207,12 @@ static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val) /* if PE if off, turn off out effects. */ if (!spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID]) val = 0; - if (spec->cur_out_type == SURROUND_OUT && nid == X_BASS) - val = 0; + if (spec->cur_out_type == SPEAKER_OUT && nid == X_BASS) { + channel_cfg = spec->channel_cfg_val; + if (channel_cfg != SPEAKER_CHANNELS_2_0 && + channel_cfg != SPEAKER_CHANNELS_4_0) + val = 0; + } } /* for in effect, qualify with CrystalVoice */ @@ -5122,6 +5468,18 @@ static int ca0132_vnode_switch_set(struct snd_kcontrol *kcontrol, return ret; } /* End of control change helpers. */ + +static void ca0132_alt_bass_redirection_xover_set(struct hda_codec *codec, + long idx) +{ + snd_hda_power_up(codec); + + dspio_set_param(codec, 0x96, 0x20, SPEAKER_BASS_REDIRECT_XOVER_FREQ, + &(float_xbass_xover_lookup[idx]), sizeof(unsigned int)); + + snd_hda_power_down(codec); +} + /* * Below I've added controls to mess with the effect levels, I've only enabled * them on the Sound Blaster Z, but they would probably also work on the @@ -5130,6 +5488,7 @@ static int ca0132_vnode_switch_set(struct snd_kcontrol *kcontrol, */ /* Sets DSP effect level from the sliders above the controls */ + static int ca0132_alt_slider_ctl_set(struct hda_codec *codec, hda_nid_t nid, const unsigned int *lookup, int idx) { @@ -5175,8 +5534,13 @@ static int ca0132_alt_xbass_xover_slider_ctl_get(struct snd_kcontrol *kcontrol, struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct ca0132_spec *spec = codec->spec; long *valp = ucontrol->value.integer.value; + hda_nid_t nid = get_amp_nid(kcontrol); + + if (nid == BASS_REDIRECTION_XOVER) + *valp = spec->bass_redirect_xover_freq; + else + *valp = spec->xbass_xover_freq; - *valp = spec->xbass_xover_freq; return 0; } @@ -5230,16 +5594,25 @@ static int ca0132_alt_xbass_xover_slider_put(struct snd_kcontrol *kcontrol, struct ca0132_spec *spec = codec->spec; hda_nid_t nid = get_amp_nid(kcontrol); long *valp = ucontrol->value.integer.value; + long *cur_val; int idx; + if (nid == BASS_REDIRECTION_XOVER) + cur_val = &spec->bass_redirect_xover_freq; + else + cur_val = &spec->xbass_xover_freq; + /* any change? */ - if (spec->xbass_xover_freq == *valp) + if (*cur_val == *valp) return 0; - spec->xbass_xover_freq = *valp; + *cur_val = *valp; idx = *valp; - ca0132_alt_slider_ctl_set(codec, nid, float_xbass_xover_lookup, idx); + if (nid == BASS_REDIRECTION_XOVER) + ca0132_alt_bass_redirection_xover_set(codec, *cur_val); + else + ca0132_alt_slider_ctl_set(codec, nid, float_xbass_xover_lookup, idx); return 0; } @@ -5466,6 +5839,13 @@ static int ca0132_alt_input_source_put(struct snd_kcontrol *kcontrol, int sel = ucontrol->value.enumerated.item[0]; unsigned int items = IN_SRC_NUM_OF_INPUTS; + /* + * The AE-7 has no front microphone, so limit items to 2: rear mic and + * line-in. + */ + if (ca0132_quirk(spec) == QUIRK_AE7) + items = 2; + if (sel >= items) return 0; @@ -5489,7 +5869,7 @@ static int ca0132_alt_output_select_get_info(struct snd_kcontrol *kcontrol, if (uinfo->value.enumerated.item >= NUM_OF_OUTPUTS) uinfo->value.enumerated.item = NUM_OF_OUTPUTS - 1; strcpy(uinfo->value.enumerated.name, - alt_out_presets[uinfo->value.enumerated.item].name); + out_type_str[uinfo->value.enumerated.item]); return 0; } @@ -5516,7 +5896,7 @@ static int ca0132_alt_output_select_put(struct snd_kcontrol *kcontrol, return 0; codec_dbg(codec, "ca0132_alt_output_select: sel=%d, preset=%s\n", - sel, alt_out_presets[sel].name); + sel, out_type_str[sel]); spec->out_enum_val = sel; @@ -5528,6 +5908,54 @@ static int ca0132_alt_output_select_put(struct snd_kcontrol *kcontrol, return 1; } +/* Select surround output type: 2.1, 4.0, 4.1, or 5.1. */ +static int ca0132_alt_speaker_channel_cfg_get_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + unsigned int items = SPEAKER_CHANNEL_CFG_COUNT; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = items; + if (uinfo->value.enumerated.item >= items) + uinfo->value.enumerated.item = items - 1; + strcpy(uinfo->value.enumerated.name, + speaker_channel_cfgs[uinfo->value.enumerated.item].name); + return 0; +} + +static int ca0132_alt_speaker_channel_cfg_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + + ucontrol->value.enumerated.item[0] = spec->channel_cfg_val; + return 0; +} + +static int ca0132_alt_speaker_channel_cfg_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + int sel = ucontrol->value.enumerated.item[0]; + unsigned int items = SPEAKER_CHANNEL_CFG_COUNT; + + if (sel >= items) + return 0; + + codec_dbg(codec, "ca0132_alt_speaker_channels: sel=%d, channels=%s\n", + sel, speaker_channel_cfgs[sel].name); + + spec->channel_cfg_val = sel; + + if (spec->out_enum_val == SPEAKER_OUT) + ca0132_alt_select_out(codec); + + return 1; +} + /* * Smart Volume output setting control. Three different settings, Normal, * which takes the value from the smart volume slider. The two others, loud @@ -5754,6 +6182,16 @@ static int ca0132_switch_get(struct snd_kcontrol *kcontrol, return 0; } + if (nid == SPEAKER_FULL_RANGE_FRONT || nid == SPEAKER_FULL_RANGE_REAR) { + *valp = spec->speaker_range_val[nid - SPEAKER_FULL_RANGE_FRONT]; + return 0; + } + + if (nid == BASS_REDIRECTION) { + *valp = spec->bass_redirection_val; + return 0; + } + return 0; } @@ -5832,6 +6270,22 @@ static int ca0132_switch_put(struct snd_kcontrol *kcontrol, goto exit; } + if (nid == SPEAKER_FULL_RANGE_FRONT || nid == SPEAKER_FULL_RANGE_REAR) { + spec->speaker_range_val[nid - SPEAKER_FULL_RANGE_FRONT] = *valp; + if (spec->cur_out_type == SPEAKER_OUT) + ca0132_alt_set_full_range_speaker(codec); + + changed = 0; + } + + if (nid == BASS_REDIRECTION) { + spec->bass_redirection_val = *valp; + if (spec->cur_out_type == SPEAKER_OUT) + ca0132_alt_surround_set_bass_redirection(codec, *valp); + + changed = 0; + } + exit: snd_hda_power_down(codec); return changed; @@ -6173,6 +6627,81 @@ static int ca0132_alt_add_output_enum(struct hda_codec *codec) } /* + * Add a control for selecting channel count on speaker output. Setting this + * allows the DSP to do bass redirection and channel upmixing on surround + * configurations. + */ +static int ca0132_alt_add_speaker_channel_cfg_enum(struct hda_codec *codec) +{ + struct snd_kcontrol_new knew = + HDA_CODEC_MUTE_MONO("Surround Channel Config", + SPEAKER_CHANNEL_CFG_ENUM, 1, 0, HDA_OUTPUT); + knew.info = ca0132_alt_speaker_channel_cfg_get_info; + knew.get = ca0132_alt_speaker_channel_cfg_get; + knew.put = ca0132_alt_speaker_channel_cfg_put; + return snd_hda_ctl_add(codec, SPEAKER_CHANNEL_CFG_ENUM, + snd_ctl_new1(&knew, codec)); +} + +/* + * Full range front stereo and rear surround switches. When these are set to + * full range, the lower frequencies from these channels are no longer + * redirected to the LFE channel. + */ +static int ca0132_alt_add_front_full_range_switch(struct hda_codec *codec) +{ + struct snd_kcontrol_new knew = + CA0132_CODEC_MUTE_MONO("Full-Range Front Speakers", + SPEAKER_FULL_RANGE_FRONT, 1, HDA_OUTPUT); + + return snd_hda_ctl_add(codec, SPEAKER_FULL_RANGE_FRONT, + snd_ctl_new1(&knew, codec)); +} + +static int ca0132_alt_add_rear_full_range_switch(struct hda_codec *codec) +{ + struct snd_kcontrol_new knew = + CA0132_CODEC_MUTE_MONO("Full-Range Rear Speakers", + SPEAKER_FULL_RANGE_REAR, 1, HDA_OUTPUT); + + return snd_hda_ctl_add(codec, SPEAKER_FULL_RANGE_REAR, + snd_ctl_new1(&knew, codec)); +} + +/* + * Bass redirection redirects audio below the crossover frequency to the LFE + * channel on speakers that are set as not being full-range. On configurations + * without an LFE channel, it does nothing. Bass redirection seems to be the + * replacement for X-Bass on configurations with an LFE channel. + */ +static int ca0132_alt_add_bass_redirection_crossover(struct hda_codec *codec) +{ + const char *namestr = "Bass Redirection Crossover"; + struct snd_kcontrol_new knew = + HDA_CODEC_VOLUME_MONO(namestr, BASS_REDIRECTION_XOVER, 1, 0, + HDA_OUTPUT); + + knew.tlv.c = NULL; + knew.info = ca0132_alt_xbass_xover_slider_info; + knew.get = ca0132_alt_xbass_xover_slider_ctl_get; + knew.put = ca0132_alt_xbass_xover_slider_put; + + return snd_hda_ctl_add(codec, BASS_REDIRECTION_XOVER, + snd_ctl_new1(&knew, codec)); +} + +static int ca0132_alt_add_bass_redirection_switch(struct hda_codec *codec) +{ + const char *namestr = "Bass Redirection"; + struct snd_kcontrol_new knew = + CA0132_CODEC_MUTE_MONO(namestr, BASS_REDIRECTION, 1, + HDA_OUTPUT); + + return snd_hda_ctl_add(codec, BASS_REDIRECTION, + snd_ctl_new1(&knew, codec)); +} + +/* * Create an Input Source enumerated control for the alternate ca0132 codecs * because the front microphone has no auto-detect, and Line-in has to be set * somehow. @@ -6478,6 +7007,21 @@ static int ca0132_build_controls(struct hda_codec *codec) err = ca0132_alt_add_output_enum(codec); if (err < 0) return err; + err = ca0132_alt_add_speaker_channel_cfg_enum(codec); + if (err < 0) + return err; + err = ca0132_alt_add_front_full_range_switch(codec); + if (err < 0) + return err; + err = ca0132_alt_add_rear_full_range_switch(codec); + if (err < 0) + return err; + err = ca0132_alt_add_bass_redirection_crossover(codec); + if (err < 0) + return err; + err = ca0132_alt_add_bass_redirection_switch(codec); + if (err < 0) + return err; err = ca0132_alt_add_mic_boost_enum(codec); if (err < 0) return err; @@ -6492,20 +7036,25 @@ static int ca0132_build_controls(struct hda_codec *codec) } } - if (ca0132_quirk(spec) == QUIRK_AE5) { + switch (ca0132_quirk(spec)) { + case QUIRK_AE5: + case QUIRK_AE7: err = ae5_add_headphone_gain_enum(codec); if (err < 0) return err; err = ae5_add_sound_filter_enum(codec); if (err < 0) return err; - } - - if (ca0132_quirk(spec) == QUIRK_ZXR) { + break; + case QUIRK_ZXR: err = zxr_add_headphone_gain_switch(codec); if (err < 0) return err; + break; + default: + break; } + #ifdef ENABLE_TUNING_CONTROLS add_tuning_ctls(codec); #endif @@ -6875,6 +7424,68 @@ static void ca0132_refresh_widget_caps(struct hda_codec *codec) } /* + * Default speaker tuning values setup for alternative codecs. + */ +static const unsigned int sbz_default_delay_values[] = { + /* Non-zero values are floating point 0.000198. */ + 0x394f9e38, 0x394f9e38, 0x00000000, 0x00000000, 0x00000000, 0x00000000 +}; + +static const unsigned int zxr_default_delay_values[] = { + /* Non-zero values are floating point 0.000220. */ + 0x00000000, 0x00000000, 0x3966afcd, 0x3966afcd, 0x3966afcd, 0x3966afcd +}; + +static const unsigned int ae5_default_delay_values[] = { + /* Non-zero values are floating point 0.000100. */ + 0x00000000, 0x00000000, 0x38d1b717, 0x38d1b717, 0x38d1b717, 0x38d1b717 +}; + +/* + * If we never change these, probably only need them on initialization. + */ +static void ca0132_alt_init_speaker_tuning(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + unsigned int i, tmp, start_req, end_req; + const unsigned int *values; + + switch (ca0132_quirk(spec)) { + case QUIRK_SBZ: + values = sbz_default_delay_values; + break; + case QUIRK_ZXR: + values = zxr_default_delay_values; + break; + case QUIRK_AE5: + case QUIRK_AE7: + values = ae5_default_delay_values; + break; + default: + values = sbz_default_delay_values; + break; + } + + tmp = FLOAT_ZERO; + dspio_set_uint_param(codec, 0x96, SPEAKER_TUNING_ENABLE_CENTER_EQ, tmp); + + start_req = SPEAKER_TUNING_FRONT_LEFT_VOL_LEVEL; + end_req = SPEAKER_TUNING_REAR_RIGHT_VOL_LEVEL; + for (i = start_req; i < end_req + 1; i++) + dspio_set_uint_param(codec, 0x96, i, tmp); + + start_req = SPEAKER_TUNING_FRONT_LEFT_INVERT; + end_req = SPEAKER_TUNING_REAR_RIGHT_INVERT; + for (i = start_req; i < end_req + 1; i++) + dspio_set_uint_param(codec, 0x96, i, tmp); + + + for (i = 0; i < 6; i++) + dspio_set_uint_param(codec, 0x96, + SPEAKER_TUNING_FRONT_LEFT_DELAY + i, values[i]); +} + +/* * Creates a dummy stream to bind the output to. This seems to have to be done * after changing the main outputs source and destination streams. */ @@ -7021,6 +7632,7 @@ static void ca0132_alt_dsp_scp_startup(struct hda_codec *codec) switch (ca0132_quirk(spec)) { case QUIRK_SBZ: case QUIRK_AE5: + case QUIRK_AE7: tmp = 0x00000003; dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); tmp = 0x00000000; @@ -7230,6 +7842,206 @@ static void ae5_post_dsp_startup_data(struct hda_codec *codec) mutex_unlock(&spec->chipio_mutex); } +static const unsigned int ae7_port_set_data[] = { + 0x0001e0c0, 0x0001e1c1, 0x0001e4c2, 0x0001e5c3, 0x0001e2c4, 0x0001e3c5, + 0x0001e8c6, 0x0001e9c7, 0x0001ecc8, 0x0001edc9, 0x0001eaca, 0x0001ebcb +}; + +static void ae7_post_dsp_setup_ports(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + unsigned int i, count, addr; + + mutex_lock(&spec->chipio_mutex); + + chipio_set_stream_channels(codec, 0x0c, 6); + chipio_set_stream_control(codec, 0x0c, 1); + + count = ARRAY_SIZE(ae7_port_set_data); + addr = 0x190030; + for (i = 0; i < count; i++) { + chipio_write_no_mutex(codec, addr, ae7_port_set_data[i]); + + /* Addresses are incremented by 4-bytes. */ + addr += 0x04; + } + + /* + * Port setting always ends with a write of 0x1 to address 0x19042c. + */ + chipio_write_no_mutex(codec, 0x19042c, 0x00000001); + + ca0113_mmio_command_set(codec, 0x30, 0x30, 0x00); + ca0113_mmio_command_set(codec, 0x48, 0x0d, 0x40); + ca0113_mmio_command_set(codec, 0x48, 0x17, 0x00); + ca0113_mmio_command_set(codec, 0x48, 0x19, 0x00); + ca0113_mmio_command_set(codec, 0x48, 0x11, 0xff); + ca0113_mmio_command_set(codec, 0x48, 0x12, 0xff); + ca0113_mmio_command_set(codec, 0x48, 0x13, 0xff); + ca0113_mmio_command_set(codec, 0x48, 0x14, 0x7f); + + mutex_unlock(&spec->chipio_mutex); +} + +static void ae7_post_dsp_asi_stream_setup(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + + mutex_lock(&spec->chipio_mutex); + + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, 0x725, 0x81); + ca0113_mmio_command_set(codec, 0x30, 0x2b, 0x00); + + chipio_set_conn_rate_no_mutex(codec, 0x70, SR_96_000); + chipio_set_stream_channels(codec, 0x0c, 6); + chipio_set_stream_control(codec, 0x0c, 1); + + chipio_set_stream_source_dest(codec, 0x05, 0x43, 0x00); + chipio_set_stream_source_dest(codec, 0x18, 0x09, 0xd0); + + chipio_set_conn_rate_no_mutex(codec, 0xd0, SR_96_000); + chipio_set_stream_channels(codec, 0x18, 6); + chipio_set_stream_control(codec, 0x18, 1); + + chipio_set_control_param_no_mutex(codec, CONTROL_PARAM_ASI, 4); + + mutex_unlock(&spec->chipio_mutex); +} + +static void ae7_post_dsp_pll_setup(struct hda_codec *codec) +{ + const unsigned int addr[] = { 0x41, 0x45, 0x40, 0x43, 0x51 }; + const unsigned int data[] = { 0xc8, 0xcc, 0xcb, 0xc7, 0x8d }; + unsigned int i; + + for (i = 0; i < ARRAY_SIZE(addr); i++) { + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_8051_ADDRESS_LOW, addr[i]); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_PLL_PMU_WRITE, data[i]); + } +} + +static void ae7_post_dsp_asi_setup_ports(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + const unsigned int target[] = { 0x0b, 0x04, 0x06, 0x0a, 0x0c, 0x11, + 0x12, 0x13, 0x14 }; + const unsigned int data[] = { 0x12, 0x00, 0x48, 0x05, 0x5f, 0xff, + 0xff, 0xff, 0x7f }; + unsigned int i; + + mutex_lock(&spec->chipio_mutex); + + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x43); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc7); + + chipio_write_no_mutex(codec, 0x189000, 0x0001f101); + chipio_write_no_mutex(codec, 0x189004, 0x0001f101); + chipio_write_no_mutex(codec, 0x189024, 0x00014004); + chipio_write_no_mutex(codec, 0x189028, 0x0002000f); + + ae7_post_dsp_pll_setup(codec); + chipio_set_control_param_no_mutex(codec, CONTROL_PARAM_ASI, 7); + + for (i = 0; i < ARRAY_SIZE(target); i++) + ca0113_mmio_command_set(codec, 0x48, target[i], data[i]); + + ca0113_mmio_command_set_type2(codec, 0x48, 0x07, 0x83); + ca0113_mmio_command_set(codec, 0x48, 0x0f, 0x00); + ca0113_mmio_command_set(codec, 0x48, 0x10, 0x00); + + chipio_set_stream_source_dest(codec, 0x21, 0x64, 0x56); + chipio_set_stream_channels(codec, 0x21, 2); + chipio_set_conn_rate_no_mutex(codec, 0x56, SR_8_000); + + chipio_set_control_param_no_mutex(codec, CONTROL_PARAM_NODE_ID, 0x09); + /* + * In the 8051's memory, this param is referred to as 'n2sid', which I + * believe is 'node to streamID'. It seems to be a way to assign a + * stream to a given HDA node. + */ + chipio_set_control_param_no_mutex(codec, 0x20, 0x21); + + chipio_write_no_mutex(codec, 0x18b038, 0x00000088); + + /* + * Now, at this point on Windows, an actual stream is setup and + * seemingly sends data to the HDA node 0x09, which is the digital + * audio input node. This is left out here, because obviously I don't + * know what data is being sent. Interestingly, the AE-5 seems to go + * through the motions of getting here and never actually takes this + * step, but the AE-7 does. + */ + + ca0113_mmio_gpio_set(codec, 0, 1); + ca0113_mmio_gpio_set(codec, 1, 1); + + ca0113_mmio_command_set_type2(codec, 0x48, 0x07, 0x83); + chipio_write_no_mutex(codec, 0x18b03c, 0x00000000); + ca0113_mmio_command_set(codec, 0x48, 0x0f, 0x00); + ca0113_mmio_command_set(codec, 0x48, 0x10, 0x00); + + chipio_set_stream_source_dest(codec, 0x05, 0x43, 0x00); + chipio_set_stream_source_dest(codec, 0x18, 0x09, 0xd0); + + chipio_set_conn_rate_no_mutex(codec, 0xd0, SR_96_000); + chipio_set_stream_channels(codec, 0x18, 6); + + /* + * Runs again, this has been repeated a few times, but I'm just + * following what the Windows driver does. + */ + ae7_post_dsp_pll_setup(codec); + chipio_set_control_param_no_mutex(codec, CONTROL_PARAM_ASI, 7); + + mutex_unlock(&spec->chipio_mutex); +} + +/* + * The Windows driver has commands that seem to setup ASI, which I believe to + * be some sort of audio serial interface. My current speculation is that it's + * related to communicating with the new DAC. + */ +static void ae7_post_dsp_asi_setup(struct hda_codec *codec) +{ + chipio_8051_write_direct(codec, 0x93, 0x10); + + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x44); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc2); + + ca0113_mmio_command_set_type2(codec, 0x48, 0x07, 0x83); + ca0113_mmio_command_set(codec, 0x30, 0x2e, 0x3f); + + chipio_set_control_param(codec, 3, 3); + chipio_set_control_flag(codec, CONTROL_FLAG_ASI_96KHZ, 1); + + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, 0x724, 0x83); + chipio_set_control_param(codec, CONTROL_PARAM_ASI, 0); + snd_hda_codec_write(codec, 0x17, 0, 0x794, 0x00); + + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x92); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_8051_ADDRESS_HIGH, 0xfa); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_8051_DATA_WRITE, 0x22); + + ae7_post_dsp_pll_setup(codec); + ae7_post_dsp_asi_stream_setup(codec); + + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x43); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc7); + + ae7_post_dsp_asi_setup_ports(codec); +} + /* * Setup default parameters for DSP */ @@ -7306,6 +8118,12 @@ static void r3d_setup_defaults(struct hda_codec *codec) if (ca0132_quirk(spec) == QUIRK_R3DI) r3di_gpio_dsp_status_set(codec, R3DI_DSP_DOWNLOADED); + /* Disable mute on Center/LFE. */ + if (ca0132_quirk(spec) == QUIRK_R3D) { + ca0113_mmio_gpio_set(codec, 2, false); + ca0113_mmio_gpio_set(codec, 4, true); + } + /* Setup effect defaults */ num_fx = OUT_EFFECTS_COUNT + IN_EFFECTS_COUNT + 1; for (idx = 0; idx < num_fx; idx++) { @@ -7373,6 +8191,8 @@ static void sbz_setup_defaults(struct hda_codec *codec) } } + ca0132_alt_init_speaker_tuning(codec); + ca0132_alt_create_dummy_stream(codec); } @@ -7440,6 +8260,93 @@ static void ae5_setup_defaults(struct hda_codec *codec) } } + ca0132_alt_init_speaker_tuning(codec); + + ca0132_alt_create_dummy_stream(codec); +} + +/* + * Setup default parameters for the Sound Blaster AE-7 DSP. + */ +static void ae7_setup_defaults(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + unsigned int tmp; + int num_fx; + int idx, i; + + if (spec->dsp_state != DSP_DOWNLOADED) + return; + + ca0132_alt_dsp_scp_startup(codec); + ca0132_alt_init_analog_mics(codec); + ae7_post_dsp_setup_ports(codec); + + tmp = FLOAT_ZERO; + dspio_set_uint_param(codec, 0x96, + SPEAKER_TUNING_FRONT_LEFT_INVERT, tmp); + dspio_set_uint_param(codec, 0x96, + SPEAKER_TUNING_FRONT_RIGHT_INVERT, tmp); + + ca0113_mmio_command_set(codec, 0x30, 0x2e, 0x3f); + + /* New, unknown SCP req's */ + dspio_set_uint_param(codec, 0x80, 0x0d, tmp); + dspio_set_uint_param(codec, 0x80, 0x0e, tmp); + + ca0113_mmio_gpio_set(codec, 0, false); + + /* Internal loopback off */ + tmp = FLOAT_ONE; + dspio_set_uint_param(codec, 0x37, 0x08, tmp); + dspio_set_uint_param(codec, 0x37, 0x10, tmp); + + /*remove DSP headroom*/ + tmp = FLOAT_ZERO; + dspio_set_uint_param(codec, 0x96, 0x3C, tmp); + + /* set WUH source */ + tmp = FLOAT_TWO; + dspio_set_uint_param(codec, 0x31, 0x00, tmp); + chipio_set_conn_rate(codec, MEM_CONNID_WUH, SR_48_000); + + /* Set speaker source? */ + dspio_set_uint_param(codec, 0x32, 0x00, tmp); + ca0113_mmio_command_set(codec, 0x30, 0x28, 0x00); + + /* + * This is the second time we've called this, but this is seemingly + * what Windows does. + */ + ca0132_alt_init_analog_mics(codec); + + ae7_post_dsp_asi_setup(codec); + + /* + * Not sure why, but these are both set to 1. They're only set to 0 + * upon shutdown. + */ + ca0113_mmio_gpio_set(codec, 0, true); + ca0113_mmio_gpio_set(codec, 1, true); + + /* Volume control related. */ + ca0113_mmio_command_set(codec, 0x48, 0x0f, 0x04); + ca0113_mmio_command_set(codec, 0x48, 0x10, 0x04); + ca0113_mmio_command_set_type2(codec, 0x48, 0x07, 0x80); + + /* out, in effects + voicefx */ + num_fx = OUT_EFFECTS_COUNT + IN_EFFECTS_COUNT + 1; + for (idx = 0; idx < num_fx; idx++) { + for (i = 0; i <= ca0132_effects[idx].params; i++) { + dspio_set_uint_param(codec, + ca0132_effects[idx].mid, + ca0132_effects[idx].reqs[i], + ca0132_effects[idx].def_vals[i]); + } + } + + ca0132_alt_init_speaker_tuning(codec); + ca0132_alt_create_dummy_stream(codec); } @@ -7757,9 +8664,15 @@ static void ca0132_init_chip(struct hda_codec *codec) * ca0132 codecs. Also sets x-bass crossover frequency to 80hz. */ if (ca0132_use_alt_controls(spec)) { + /* Set speakers to default to full range. */ + spec->speaker_range_val[0] = 1; + spec->speaker_range_val[1] = 1; + spec->xbass_xover_freq = 8; for (i = 0; i < EFFECT_LEVEL_SLIDERS; i++) spec->fx_ctl_val[i] = effect_slider_defaults[i]; + + spec->bass_redirect_xover_freq = 8; } spec->voicefx_val = 0; @@ -7925,6 +8838,32 @@ static void ae5_exit_chip(struct hda_codec *codec) snd_hda_codec_write(codec, 0x01, 0, 0x724, 0x83); } +static void ae7_exit_chip(struct hda_codec *codec) +{ + chipio_set_stream_control(codec, 0x18, 0); + chipio_set_stream_source_dest(codec, 0x21, 0xc8, 0xc8); + chipio_set_stream_channels(codec, 0x21, 0); + chipio_set_control_param(codec, CONTROL_PARAM_NODE_ID, 0x09); + chipio_set_control_param(codec, 0x20, 0x01); + + chipio_set_control_param(codec, CONTROL_PARAM_ASI, 0); + + chipio_set_stream_control(codec, 0x18, 0); + chipio_set_stream_control(codec, 0x0c, 0); + + ca0113_mmio_command_set(codec, 0x30, 0x2b, 0x00); + snd_hda_codec_write(codec, 0x15, 0, 0x724, 0x83); + ca0113_mmio_command_set_type2(codec, 0x48, 0x07, 0x83); + ca0113_mmio_command_set(codec, 0x30, 0x30, 0x00); + ca0113_mmio_command_set(codec, 0x30, 0x2e, 0x00); + ca0113_mmio_gpio_set(codec, 0, false); + ca0113_mmio_gpio_set(codec, 1, false); + ca0113_mmio_command_set(codec, 0x30, 0x32, 0x3f); + + snd_hda_codec_write(codec, 0x01, 0, 0x793, 0x00); + snd_hda_codec_write(codec, 0x01, 0, 0x794, 0x53); +} + static void zxr_exit_chip(struct hda_codec *codec) { chipio_set_stream_control(codec, 0x03, 0); @@ -8108,81 +9047,149 @@ static void r3di_pre_dsp_setup(struct hda_codec *codec) * what they do, or if they're necessary. Could possibly * be removed. Figure they're better to leave in. */ -static void ca0132_mmio_init(struct hda_codec *codec) +static const unsigned int ca0113_mmio_init_address_sbz[] = { + 0x400, 0x408, 0x40c, 0x01c, 0xc0c, 0xc00, 0xc04, 0xc0c, 0xc0c, 0xc0c, + 0xc0c, 0xc08, 0xc08, 0xc08, 0xc08, 0xc08, 0xc04 +}; + +static const unsigned int ca0113_mmio_init_data_sbz[] = { + 0x00000030, 0x00000000, 0x00000003, 0x00000003, 0x00000003, + 0x00000003, 0x000000c1, 0x000000f1, 0x00000001, 0x000000c7, + 0x000000c1, 0x00000080 +}; + +static const unsigned int ca0113_mmio_init_data_zxr[] = { + 0x00000030, 0x00000000, 0x00000000, 0x00000003, 0x00000003, + 0x00000003, 0x00000001, 0x000000f1, 0x00000001, 0x000000c7, + 0x000000c1, 0x00000080 +}; + +static const unsigned int ca0113_mmio_init_address_ae5[] = { + 0x400, 0x42c, 0x46c, 0x4ac, 0x4ec, 0x43c, 0x47c, 0x4bc, 0x4fc, 0x408, + 0x100, 0x410, 0x40c, 0x100, 0x100, 0x830, 0x86c, 0x800, 0x86c, 0x800, + 0x804, 0x20c, 0x01c, 0xc0c, 0xc00, 0xc04, 0xc0c, 0xc0c, 0xc0c, 0xc0c, + 0xc08, 0xc08, 0xc08, 0xc08, 0xc08, 0xc04, 0x01c +}; + +static const unsigned int ca0113_mmio_init_data_ae5[] = { + 0x00000001, 0x00000000, 0x00000000, 0x00000000, 0x00000000, + 0x00000000, 0x00000000, 0x00000000, 0x00000000, 0x00000001, + 0x00000600, 0x00000014, 0x00000001, 0x0000060f, 0x0000070f, + 0x00000aff, 0x00000000, 0x0000006b, 0x00000001, 0x0000006b, + 0x00000057, 0x00800000, 0x00880680, 0x00000080, 0x00000030, + 0x00000000, 0x00000000, 0x00000003, 0x00000003, 0x00000003, + 0x00000001, 0x000000f1, 0x00000001, 0x000000c7, 0x000000c1, + 0x00000080, 0x00880680 +}; + +static void ca0132_mmio_init_sbz(struct hda_codec *codec) { struct ca0132_spec *spec = codec->spec; + unsigned int tmp[2], i, count, cur_addr; + const unsigned int *addr, *data; - if (ca0132_quirk(spec) == QUIRK_AE5) - writel(0x00000001, spec->mem_base + 0x400); - else - writel(0x00000000, spec->mem_base + 0x400); + addr = ca0113_mmio_init_address_sbz; + for (i = 0; i < 3; i++) + writel(0x00000000, spec->mem_base + addr[i]); - if (ca0132_quirk(spec) == QUIRK_AE5) - writel(0x00000001, spec->mem_base + 0x408); - else - writel(0x00000000, spec->mem_base + 0x408); + cur_addr = i; + switch (ca0132_quirk(spec)) { + case QUIRK_ZXR: + tmp[0] = 0x00880480; + tmp[1] = 0x00000080; + break; + case QUIRK_SBZ: + tmp[0] = 0x00820680; + tmp[1] = 0x00000083; + break; + case QUIRK_R3D: + tmp[0] = 0x00880680; + tmp[1] = 0x00000083; + break; + default: + tmp[0] = 0x00000000; + tmp[1] = 0x00000000; + break; + } - if (ca0132_quirk(spec) == QUIRK_AE5) - writel(0x00000001, spec->mem_base + 0x40c); - else - writel(0x00000000, spec->mem_base + 0x40C); + for (i = 0; i < 2; i++) + writel(tmp[i], spec->mem_base + addr[cur_addr + i]); - if (ca0132_quirk(spec) == QUIRK_ZXR) - writel(0x00880640, spec->mem_base + 0x01C); - else - writel(0x00880680, spec->mem_base + 0x01C); + cur_addr += i; - if (ca0132_quirk(spec) == QUIRK_AE5) - writel(0x00000080, spec->mem_base + 0xC0C); - else - writel(0x00000083, spec->mem_base + 0xC0C); + switch (ca0132_quirk(spec)) { + case QUIRK_ZXR: + count = ARRAY_SIZE(ca0113_mmio_init_data_zxr); + data = ca0113_mmio_init_data_zxr; + break; + default: + count = ARRAY_SIZE(ca0113_mmio_init_data_sbz); + data = ca0113_mmio_init_data_sbz; + break; + } - writel(0x00000030, spec->mem_base + 0xC00); - writel(0x00000000, spec->mem_base + 0xC04); + for (i = 0; i < count; i++) + writel(data[i], spec->mem_base + addr[cur_addr + i]); +} - if (ca0132_quirk(spec) == QUIRK_AE5) - writel(0x00000000, spec->mem_base + 0xC0C); - else - writel(0x00000003, spec->mem_base + 0xC0C); +static void ca0132_mmio_init_ae5(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + const unsigned int *addr, *data; + unsigned int i, count; + + addr = ca0113_mmio_init_address_ae5; + data = ca0113_mmio_init_data_ae5; + count = ARRAY_SIZE(ca0113_mmio_init_data_ae5); - writel(0x00000003, spec->mem_base + 0xC0C); - writel(0x00000003, spec->mem_base + 0xC0C); - writel(0x00000003, spec->mem_base + 0xC0C); + if (ca0132_quirk(spec) == QUIRK_AE7) { + writel(0x00000680, spec->mem_base + 0x1c); + writel(0x00880680, spec->mem_base + 0x1c); + } + + for (i = 0; i < count; i++) { + /* + * AE-7 shares all writes with the AE-5, except that it writes + * a different value to 0x20c. + */ + if (i == 21 && ca0132_quirk(spec) == QUIRK_AE7) { + writel(0x00800001, spec->mem_base + addr[i]); + continue; + } + + writel(data[i], spec->mem_base + addr[i]); + } if (ca0132_quirk(spec) == QUIRK_AE5) - writel(0x00000001, spec->mem_base + 0xC08); - else - writel(0x000000C1, spec->mem_base + 0xC08); - - writel(0x000000F1, spec->mem_base + 0xC08); - writel(0x00000001, spec->mem_base + 0xC08); - writel(0x000000C7, spec->mem_base + 0xC08); - writel(0x000000C1, spec->mem_base + 0xC08); - writel(0x00000080, spec->mem_base + 0xC04); - - if (ca0132_quirk(spec) == QUIRK_AE5) { - writel(0x00000000, spec->mem_base + 0x42c); - writel(0x00000000, spec->mem_base + 0x46c); - writel(0x00000000, spec->mem_base + 0x4ac); - writel(0x00000000, spec->mem_base + 0x4ec); - writel(0x00000000, spec->mem_base + 0x43c); - writel(0x00000000, spec->mem_base + 0x47c); - writel(0x00000000, spec->mem_base + 0x4bc); - writel(0x00000000, spec->mem_base + 0x4fc); - writel(0x00000600, spec->mem_base + 0x100); - writel(0x00000014, spec->mem_base + 0x410); - writel(0x0000060f, spec->mem_base + 0x100); - writel(0x0000070f, spec->mem_base + 0x100); - writel(0x00000aff, spec->mem_base + 0x830); - writel(0x00000000, spec->mem_base + 0x86c); - writel(0x0000006b, spec->mem_base + 0x800); - writel(0x00000001, spec->mem_base + 0x86c); - writel(0x0000006b, spec->mem_base + 0x800); - writel(0x00000057, spec->mem_base + 0x804); - writel(0x00800000, spec->mem_base + 0x20c); + writel(0x00880680, spec->mem_base + 0x1c); +} + +static void ca0132_mmio_init(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + + switch (ca0132_quirk(spec)) { + case QUIRK_R3D: + case QUIRK_SBZ: + case QUIRK_ZXR: + ca0132_mmio_init_sbz(codec); + break; + case QUIRK_AE5: + ca0132_mmio_init_ae5(codec); + break; } } +static const unsigned int ca0132_ae5_register_set_addresses[] = { + 0x304, 0x304, 0x304, 0x304, 0x100, 0x304, 0x100, 0x304, 0x100, 0x304, + 0x100, 0x304, 0x86c, 0x800, 0x86c, 0x800, 0x804 +}; + +static const unsigned char ca0132_ae5_register_set_data[] = { + 0x0f, 0x0e, 0x1f, 0x0c, 0x3f, 0x08, 0x7f, 0x00, 0xff, 0x00, 0x6b, + 0x01, 0x6b, 0x57 +}; + /* * This function writes to some SFR's, does some region2 writes, and then * eventually resets the codec with the 0x7ff verb. Not quite sure why it does @@ -8191,6 +9198,18 @@ static void ca0132_mmio_init(struct hda_codec *codec) static void ae5_register_set(struct hda_codec *codec) { struct ca0132_spec *spec = codec->spec; + unsigned int count = ARRAY_SIZE(ca0132_ae5_register_set_addresses); + const unsigned int *addr = ca0132_ae5_register_set_addresses; + const unsigned char *data = ca0132_ae5_register_set_data; + unsigned int i, cur_addr; + unsigned char tmp[3]; + + if (ca0132_quirk(spec) == QUIRK_AE7) { + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x41); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc8); + } chipio_8051_write_direct(codec, 0x93, 0x10); snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, @@ -8198,25 +9217,43 @@ static void ae5_register_set(struct hda_codec *codec) snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc2); - writeb(0x0f, spec->mem_base + 0x304); - writeb(0x0f, spec->mem_base + 0x304); - writeb(0x0f, spec->mem_base + 0x304); - writeb(0x0f, spec->mem_base + 0x304); - writeb(0x0e, spec->mem_base + 0x100); - writeb(0x1f, spec->mem_base + 0x304); - writeb(0x0c, spec->mem_base + 0x100); - writeb(0x3f, spec->mem_base + 0x304); - writeb(0x08, spec->mem_base + 0x100); - writeb(0x7f, spec->mem_base + 0x304); - writeb(0x00, spec->mem_base + 0x100); - writeb(0xff, spec->mem_base + 0x304); + if (ca0132_quirk(spec) == QUIRK_AE7) { + tmp[0] = 0x03; + tmp[1] = 0x03; + tmp[2] = 0x07; + } else { + tmp[0] = 0x0f; + tmp[1] = 0x0f; + tmp[2] = 0x0f; + } - ca0113_mmio_command_set(codec, 0x30, 0x2d, 0x3f); + for (i = cur_addr = 0; i < 3; i++, cur_addr++) + writeb(tmp[i], spec->mem_base + addr[cur_addr]); + + /* + * First writes are in single bytes, final are in 4 bytes. So, we use + * writeb, then writel. + */ + for (i = 0; cur_addr < 12; i++, cur_addr++) + writeb(data[i], spec->mem_base + addr[cur_addr]); + + for (; cur_addr < count; i++, cur_addr++) + writel(data[i], spec->mem_base + addr[cur_addr]); + + writel(0x00800001, spec->mem_base + 0x20c); + + if (ca0132_quirk(spec) == QUIRK_AE7) { + ca0113_mmio_command_set_type2(codec, 0x48, 0x07, 0x83); + ca0113_mmio_command_set(codec, 0x30, 0x2e, 0x3f); + } else { + ca0113_mmio_command_set(codec, 0x30, 0x2d, 0x3f); + } chipio_8051_write_direct(codec, 0x90, 0x00); chipio_8051_write_direct(codec, 0x90, 0x10); - ca0113_mmio_command_set(codec, 0x48, 0x07, 0x83); + if (ca0132_quirk(spec) == QUIRK_AE5) + ca0113_mmio_command_set(codec, 0x48, 0x07, 0x83); chipio_write(codec, 0x18b0a4, 0x000000c2); @@ -8268,6 +9305,19 @@ static void ca0132_alt_init(struct hda_codec *codec) snd_hda_sequence_write(codec, spec->desktop_init_verbs); ca0113_mmio_command_set(codec, 0x30, 0x32, 0x3f); break; + case QUIRK_AE7: + ca0132_gpio_init(codec); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x49); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_PLL_PMU_WRITE, 0x88); + snd_hda_sequence_write(codec, spec->chip_init_verbs); + snd_hda_sequence_write(codec, spec->desktop_init_verbs); + chipio_write(codec, 0x18b008, 0x000000f8); + chipio_write(codec, 0x18b008, 0x000000f0); + chipio_write(codec, 0x18b030, 0x00000020); + ca0113_mmio_command_set(codec, 0x30, 0x32, 0x3f); + break; case QUIRK_ZXR: snd_hda_sequence_write(codec, spec->chip_init_verbs); snd_hda_sequence_write(codec, spec->desktop_init_verbs); @@ -8315,7 +9365,7 @@ static int ca0132_init(struct hda_codec *codec) snd_hda_power_up_pm(codec); - if (ca0132_quirk(spec) == QUIRK_AE5) + if (ca0132_quirk(spec) == QUIRK_AE5 || ca0132_quirk(spec) == QUIRK_AE7) ae5_register_set(codec); ca0132_init_unsol(codec); @@ -8343,6 +9393,9 @@ static int ca0132_init(struct hda_codec *codec) case QUIRK_AE5: ae5_setup_defaults(codec); break; + case QUIRK_AE7: + ae7_setup_defaults(codec); + break; default: ca0132_setup_defaults(codec); ca0132_init_analog_mic2(codec); @@ -8430,6 +9483,9 @@ static void ca0132_free(struct hda_codec *codec) case QUIRK_AE5: ae5_exit_chip(codec); break; + case QUIRK_AE7: + ae7_exit_chip(codec); + break; case QUIRK_R3DI: r3di_gpio_shutdown(codec); break; @@ -8534,6 +9590,10 @@ static void ca0132_config(struct hda_codec *codec) codec_dbg(codec, "%s: QUIRK_AE5 applied.\n", __func__); snd_hda_apply_pincfgs(codec, ae5_pincfgs); break; + case QUIRK_AE7: + codec_dbg(codec, "%s: QUIRK_AE7 applied.\n", __func__); + snd_hda_apply_pincfgs(codec, ae7_pincfgs); + break; default: break; } @@ -8615,6 +9675,7 @@ static void ca0132_config(struct hda_codec *codec) spec->dig_in = 0x09; break; case QUIRK_AE5: + case QUIRK_AE7: spec->num_outputs = 2; spec->out_pins[0] = 0x0B; /* Line out */ spec->out_pins[1] = 0x11; /* Rear headphone out */ @@ -8813,6 +9874,10 @@ static int patch_ca0132(struct hda_codec *codec) spec->mixers[0] = desktop_mixer; snd_hda_codec_set_name(codec, "Sound BlasterX AE-5"); break; + case QUIRK_AE7: + spec->mixers[0] = desktop_mixer; + snd_hda_codec_set_name(codec, "Sound Blaster AE-7"); + break; default: spec->mixers[0] = ca0132_mixer; break; @@ -8823,6 +9888,7 @@ static int patch_ca0132(struct hda_codec *codec) case QUIRK_SBZ: case QUIRK_R3D: case QUIRK_AE5: + case QUIRK_AE7: case QUIRK_ZXR: spec->use_alt_controls = true; spec->use_alt_functions = true; diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 402050088090..055440740184 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -4269,6 +4269,7 @@ HDA_CODEC_ENTRY(0x8086280c, "Cannonlake HDMI", patch_i915_glk_hdmi), HDA_CODEC_ENTRY(0x8086280d, "Geminilake HDMI", patch_i915_glk_hdmi), HDA_CODEC_ENTRY(0x8086280f, "Icelake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x80862812, "Tigerlake HDMI", patch_i915_tgl_hdmi), +HDA_CODEC_ENTRY(0x80862814, "DG1 HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862816, "Rocketlake HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x8086281a, "Jasperlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x8086281b, "Elkhartlake HDMI", patch_i915_icl_hdmi), diff --git a/sound/pci/mixart/mixart.h b/sound/pci/mixart/mixart.h index 42111562e9bc..cbed6d9a9f2e 100644 --- a/sound/pci/mixart/mixart.h +++ b/sound/pci/mixart/mixart.h @@ -69,7 +69,7 @@ struct mixart_mgr { u32 msg_fifo[MSG_FIFO_SIZE]; int msg_fifo_readptr; int msg_fifo_writeptr; - atomic_t msg_processed; /* number of messages to be processed in tasklet */ + atomic_t msg_processed; /* number of messages to be processed in irq thread */ struct mutex lock; /* interrupt lock */ struct mutex msg_lock; /* mailbox lock */ diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index 098c69b3b7aa..fcc2073c5025 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -445,7 +445,6 @@ struct snd_riptide { union firmware_version firmware; spinlock_t lock; - struct tasklet_struct riptide_tq; struct snd_info_entry *proc_entry; unsigned long received_irqs; @@ -1070,9 +1069,9 @@ getmixer(struct cmdif *cif, short num, unsigned short *rval, return 0; } -static void riptide_handleirq(struct tasklet_struct *t) +static irqreturn_t riptide_handleirq(int irq, void *dev_id) { - struct snd_riptide *chip = from_tasklet(chip, t, riptide_tq); + struct snd_riptide *chip = dev_id; struct cmdif *cif = chip->cif; struct snd_pcm_substream *substream[PLAYBACK_SUBSTREAMS + 1]; struct snd_pcm_runtime *runtime; @@ -1083,7 +1082,7 @@ static void riptide_handleirq(struct tasklet_struct *t) unsigned int flag; if (!cif) - return; + return IRQ_HANDLED; for (i = 0; i < PLAYBACK_SUBSTREAMS; i++) substream[i] = chip->playback_substream[i]; @@ -1134,6 +1133,8 @@ static void riptide_handleirq(struct tasklet_struct *t) } } } + + return IRQ_HANDLED; } #ifdef CONFIG_PM_SLEEP @@ -1699,13 +1700,14 @@ snd_riptide_interrupt(int irq, void *dev_id) { struct snd_riptide *chip = dev_id; struct cmdif *cif = chip->cif; + irqreturn_t ret = IRQ_HANDLED; if (cif) { chip->received_irqs++; if (IS_EOBIRQ(cif->hwport) || IS_EOSIRQ(cif->hwport) || IS_EOCIRQ(cif->hwport)) { chip->handled_irqs++; - tasklet_schedule(&chip->riptide_tq); + ret = IRQ_WAKE_THREAD; } if (chip->rmidi && IS_MPUIRQ(cif->hwport)) { chip->handled_irqs++; @@ -1714,7 +1716,7 @@ snd_riptide_interrupt(int irq, void *dev_id) } SET_AIACK(cif->hwport); } - return IRQ_HANDLED; + return ret; } static void @@ -1843,7 +1845,6 @@ snd_riptide_create(struct snd_card *card, struct pci_dev *pci, chip->received_irqs = 0; chip->handled_irqs = 0; chip->cif = NULL; - tasklet_setup(&chip->riptide_tq, riptide_handleirq); if ((chip->res_port = request_region(chip->port, 64, "RIPTIDE")) == NULL) { @@ -1856,8 +1857,9 @@ snd_riptide_create(struct snd_card *card, struct pci_dev *pci, hwport = (struct riptideport *)chip->port; UNSET_AIE(hwport); - if (request_irq(pci->irq, snd_riptide_interrupt, IRQF_SHARED, - KBUILD_MODNAME, chip)) { + if (request_threaded_irq(pci->irq, snd_riptide_interrupt, + riptide_handleirq, IRQF_SHARED, + KBUILD_MODNAME, chip)) { snd_printk(KERN_ERR "Riptide: unable to grab IRQ %d\n", pci->irq); snd_riptide_free(chip); diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index dda56ecfd33b..cea53a878c36 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -447,8 +447,8 @@ struct hdsp { struct snd_pcm_substream *capture_substream; struct snd_pcm_substream *playback_substream; struct hdsp_midi midi[2]; - struct tasklet_struct midi_tasklet; - int use_midi_tasklet; + struct work_struct midi_work; + int use_midi_work; int precise_ptr; u32 control_register; /* cached value */ u32 control2_register; /* cached value */ @@ -1385,7 +1385,6 @@ static void snd_hdsp_midi_input_trigger(struct snd_rawmidi_substream *substream, } } else { hdsp->control_register &= ~ie; - tasklet_kill(&hdsp->midi_tasklet); } hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register); @@ -2542,37 +2541,37 @@ static int snd_hdsp_put_precise_pointer(struct snd_kcontrol *kcontrol, struct sn return change; } -#define HDSP_USE_MIDI_TASKLET(xname, xindex) \ +#define HDSP_USE_MIDI_WORK(xname, xindex) \ { .iface = SNDRV_CTL_ELEM_IFACE_CARD, \ .name = xname, \ .index = xindex, \ - .info = snd_hdsp_info_use_midi_tasklet, \ - .get = snd_hdsp_get_use_midi_tasklet, \ - .put = snd_hdsp_put_use_midi_tasklet \ + .info = snd_hdsp_info_use_midi_work, \ + .get = snd_hdsp_get_use_midi_work, \ + .put = snd_hdsp_put_use_midi_work \ } -static int hdsp_set_use_midi_tasklet(struct hdsp *hdsp, int use_tasklet) +static int hdsp_set_use_midi_work(struct hdsp *hdsp, int use_work) { - if (use_tasklet) - hdsp->use_midi_tasklet = 1; + if (use_work) + hdsp->use_midi_work = 1; else - hdsp->use_midi_tasklet = 0; + hdsp->use_midi_work = 0; return 0; } -#define snd_hdsp_info_use_midi_tasklet snd_ctl_boolean_mono_info +#define snd_hdsp_info_use_midi_work snd_ctl_boolean_mono_info -static int snd_hdsp_get_use_midi_tasklet(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int snd_hdsp_get_use_midi_work(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); spin_lock_irq(&hdsp->lock); - ucontrol->value.integer.value[0] = hdsp->use_midi_tasklet; + ucontrol->value.integer.value[0] = hdsp->use_midi_work; spin_unlock_irq(&hdsp->lock); return 0; } -static int snd_hdsp_put_use_midi_tasklet(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int snd_hdsp_put_use_midi_work(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; @@ -2582,8 +2581,8 @@ static int snd_hdsp_put_use_midi_tasklet(struct snd_kcontrol *kcontrol, struct s return -EBUSY; val = ucontrol->value.integer.value[0] & 1; spin_lock_irq(&hdsp->lock); - change = (int)val != hdsp->use_midi_tasklet; - hdsp_set_use_midi_tasklet(hdsp, val); + change = (int)val != hdsp->use_midi_work; + hdsp_set_use_midi_work(hdsp, val); spin_unlock_irq(&hdsp->lock); return change; } @@ -2950,7 +2949,7 @@ HDSP_SPDIF_SYNC_CHECK("SPDIF Lock Status", 0), HDSP_ADATSYNC_SYNC_CHECK("ADAT Sync Lock Status", 0), HDSP_TOGGLE_SETTING("Line Out", HDSP_LineOut), HDSP_PRECISE_POINTER("Precise Pointer", 0), -HDSP_USE_MIDI_TASKLET("Use Midi Tasklet", 0), +HDSP_USE_MIDI_WORK("Use Midi Tasklet", 0), }; @@ -3370,7 +3369,7 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) snd_iprintf(buffer, "MIDI1 Input status: 0x%x\n", hdsp_read(hdsp, HDSP_midiStatusIn0)); snd_iprintf(buffer, "MIDI2 Output status: 0x%x\n", hdsp_read(hdsp, HDSP_midiStatusOut1)); snd_iprintf(buffer, "MIDI2 Input status: 0x%x\n", hdsp_read(hdsp, HDSP_midiStatusIn1)); - snd_iprintf(buffer, "Use Midi Tasklet: %s\n", hdsp->use_midi_tasklet ? "on" : "off"); + snd_iprintf(buffer, "Use Midi Tasklet: %s\n", hdsp->use_midi_work ? "on" : "off"); snd_iprintf(buffer, "\n"); @@ -3791,9 +3790,9 @@ static int snd_hdsp_set_defaults(struct hdsp *hdsp) return 0; } -static void hdsp_midi_tasklet(struct tasklet_struct *t) +static void hdsp_midi_work(struct work_struct *work) { - struct hdsp *hdsp = from_tasklet(hdsp, t, midi_tasklet); + struct hdsp *hdsp = container_of(work, struct hdsp, midi_work); if (hdsp->midi[0].pending) snd_hdsp_midi_input_read (&hdsp->midi[0]); @@ -3838,7 +3837,7 @@ static irqreturn_t snd_hdsp_interrupt(int irq, void *dev_id) } if (midi0 && midi0status) { - if (hdsp->use_midi_tasklet) { + if (hdsp->use_midi_work) { /* we disable interrupts for this input until processing is done */ hdsp->control_register &= ~HDSP_Midi0InterruptEnable; hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register); @@ -3849,7 +3848,7 @@ static irqreturn_t snd_hdsp_interrupt(int irq, void *dev_id) } } if (hdsp->io_type != Multiface && hdsp->io_type != RPM && hdsp->io_type != H9632 && midi1 && midi1status) { - if (hdsp->use_midi_tasklet) { + if (hdsp->use_midi_work) { /* we disable interrupts for this input until processing is done */ hdsp->control_register &= ~HDSP_Midi1InterruptEnable; hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register); @@ -3859,8 +3858,8 @@ static irqreturn_t snd_hdsp_interrupt(int irq, void *dev_id) snd_hdsp_midi_input_read (&hdsp->midi[1]); } } - if (hdsp->use_midi_tasklet && schedule) - tasklet_schedule(&hdsp->midi_tasklet); + if (hdsp->use_midi_work && schedule) + queue_work(system_highpri_wq, &hdsp->midi_work); return IRQ_HANDLED; } @@ -5182,7 +5181,7 @@ static int snd_hdsp_create(struct snd_card *card, spin_lock_init(&hdsp->lock); - tasklet_setup(&hdsp->midi_tasklet, hdsp_midi_tasklet); + INIT_WORK(&hdsp->midi_work, hdsp_midi_work); pci_read_config_word(hdsp->pci, PCI_CLASS_REVISION, &hdsp->firmware_rev); hdsp->firmware_rev &= 0xff; @@ -5235,7 +5234,7 @@ static int snd_hdsp_create(struct snd_card *card, hdsp->irq = pci->irq; card->sync_irq = hdsp->irq; hdsp->precise_ptr = 0; - hdsp->use_midi_tasklet = 1; + hdsp->use_midi_work = 1; hdsp->dds_value = 0; if ((err = snd_hdsp_initialize_memory(hdsp)) < 0) @@ -5305,7 +5304,7 @@ static int snd_hdsp_free(struct hdsp *hdsp) { if (hdsp->port) { /* stop the audio, and cancel all interrupts */ - tasklet_kill(&hdsp->midi_tasklet); + cancel_work_sync(&hdsp->midi_work); hdsp->control_register &= ~(HDSP_Start|HDSP_AudioInterruptEnable|HDSP_Midi0InterruptEnable|HDSP_Midi1InterruptEnable); hdsp_write (hdsp, HDSP_controlRegister, hdsp->control_register); } diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 572350aaf18d..4a1f576dd9cf 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -997,7 +997,7 @@ struct hdspm { u32 settings_register; /* cached value for AIO / RayDat (sync reference, master/slave) */ struct hdspm_midi midi[4]; - struct tasklet_struct midi_tasklet; + struct work_struct midi_work; size_t period_bytes; unsigned char ss_in_channels; @@ -1217,7 +1217,7 @@ static int snd_hdspm_use_is_exclusive(struct hdspm *hdspm) return ret; } -/* round arbitary sample rates to commonly known rates */ +/* round arbitrary sample rates to commonly known rates */ static int hdspm_round_frequency(int rate) { if (rate < 38050) @@ -2169,9 +2169,9 @@ static int snd_hdspm_create_midi(struct snd_card *card, } -static void hdspm_midi_tasklet(struct tasklet_struct *t) +static void hdspm_midi_work(struct work_struct *work) { - struct hdspm *hdspm = from_tasklet(hdspm, t, midi_tasklet); + struct hdspm *hdspm = container_of(work, struct hdspm, midi_work); int i = 0; while (i < hdspm->midiPorts) { @@ -5449,7 +5449,7 @@ static irqreturn_t snd_hdspm_interrupt(int irq, void *dev_id) } if (schedule) - tasklet_hi_schedule(&hdspm->midi_tasklet); + queue_work(system_highpri_wq, &hdspm->midi_work); } return IRQ_HANDLED; @@ -6538,6 +6538,7 @@ static int snd_hdspm_create(struct snd_card *card, hdspm->card = card; spin_lock_init(&hdspm->lock); + INIT_WORK(&hdspm->midi_work, hdspm_midi_work); pci_read_config_word(hdspm->pci, PCI_CLASS_REVISION, &hdspm->firmware_rev); @@ -6836,9 +6837,6 @@ static int snd_hdspm_create(struct snd_card *card, } - tasklet_setup(&hdspm->midi_tasklet, hdspm_midi_tasklet); - - if (hdspm->io_type != MADIface) { hdspm->serial = (hdspm_read(hdspm, HDSPM_midiStatusIn0)>>8) & 0xFFFFFF; @@ -6873,6 +6871,7 @@ static int snd_hdspm_free(struct hdspm * hdspm) { if (hdspm->port) { + cancel_work_sync(&hdspm->midi_work); /* stop th audio, and cancel all interrupts */ hdspm->control_register &= diff --git a/sound/usb/card.c b/sound/usb/card.c index 696e788c5d31..fa764b61fe9c 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -333,6 +333,106 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) } /* + * Profile name preset table + */ +struct usb_audio_device_name { + u32 id; + const char *vendor_name; + const char *product_name; + const char *profile_name; /* override card->longname */ +}; + +#define PROFILE_NAME(vid, pid, vendor, product, profile) \ + { .id = USB_ID(vid, pid), .vendor_name = (vendor), \ + .product_name = (product), .profile_name = (profile) } +#define DEVICE_NAME(vid, pid, vendor, product) \ + PROFILE_NAME(vid, pid, vendor, product, NULL) + +/* vendor/product and profile name presets, sorted in device id order */ +static const struct usb_audio_device_name usb_audio_names[] = { + /* HP Thunderbolt Dock Audio Headset */ + PROFILE_NAME(0x03f0, 0x0269, "HP", "Thunderbolt Dock Audio Headset", + "HP-Thunderbolt-Dock-Audio-Headset"), + /* HP Thunderbolt Dock Audio Module */ + PROFILE_NAME(0x03f0, 0x0567, "HP", "Thunderbolt Dock Audio Module", + "HP-Thunderbolt-Dock-Audio-Module"), + + /* Two entries for Gigabyte TRX40 Aorus Master: + * TRX40 Aorus Master has two USB-audio devices, one for the front + * headphone with ESS SABRE9218 DAC chip, while another for the rest + * I/O (the rear panel and the front mic) with Realtek ALC1220-VB. + * Here we provide two distinct names for making UCM profiles easier. + */ + PROFILE_NAME(0x0414, 0xa000, "Gigabyte", "Aorus Master Front Headphone", + "Gigabyte-Aorus-Master-Front-Headphone"), + PROFILE_NAME(0x0414, 0xa001, "Gigabyte", "Aorus Master Main Audio", + "Gigabyte-Aorus-Master-Main-Audio"), + + /* Gigabyte TRX40 Aorus Pro WiFi */ + PROFILE_NAME(0x0414, 0xa002, + "Realtek", "ALC1220-VB-DT", "Realtek-ALC1220-VB-Desktop"), + + /* Creative/E-Mu devices */ + DEVICE_NAME(0x041e, 0x3010, "Creative Labs", "Sound Blaster MP3+"), + /* Creative/Toshiba Multimedia Center SB-0500 */ + DEVICE_NAME(0x041e, 0x3048, "Toshiba", "SB-0500"), + + DEVICE_NAME(0x046d, 0x0990, "Logitech, Inc.", "QuickCam Pro 9000"), + + /* Dell WD15 Dock */ + PROFILE_NAME(0x0bda, 0x4014, "Dell", "WD15 Dock", "Dell-WD15-Dock"), + /* Dell WD19 Dock */ + PROFILE_NAME(0x0bda, 0x402e, "Dell", "WD19 Dock", "Dell-WD15-Dock"), + + DEVICE_NAME(0x0ccd, 0x0028, "TerraTec", "Aureon5.1MkII"), + + /* + * The original product_name is "USB Sound Device", however this name + * is also used by the CM106 based cards, so make it unique. + */ + DEVICE_NAME(0x0d8c, 0x0102, NULL, "ICUSBAUDIO7D"), + DEVICE_NAME(0x0d8c, 0x0103, NULL, "Audio Advantage MicroII"), + + /* MSI TRX40 Creator */ + PROFILE_NAME(0x0db0, 0x0d64, + "Realtek", "ALC1220-VB-DT", "Realtek-ALC1220-VB-Desktop"), + /* MSI TRX40 */ + PROFILE_NAME(0x0db0, 0x543d, + "Realtek", "ALC1220-VB-DT", "Realtek-ALC1220-VB-Desktop"), + + /* Stanton/N2IT Final Scratch v1 device ('Scratchamp') */ + DEVICE_NAME(0x103d, 0x0100, "Stanton", "ScratchAmp"), + DEVICE_NAME(0x103d, 0x0101, "Stanton", "ScratchAmp"), + + /* aka. Serato Scratch Live DJ Box */ + DEVICE_NAME(0x13e5, 0x0001, "Rane", "SL-1"), + + /* Lenovo ThinkStation P620 Rear Line-in, Line-out and Microphone */ + PROFILE_NAME(0x17aa, 0x1046, "Lenovo", "ThinkStation P620 Rear", + "Lenovo-ThinkStation-P620-Rear"), + /* Lenovo ThinkStation P620 Internal Speaker + Front Headset */ + PROFILE_NAME(0x17aa, 0x104d, "Lenovo", "ThinkStation P620 Main", + "Lenovo-ThinkStation-P620-Main"), + + /* Asrock TRX40 Creator */ + PROFILE_NAME(0x26ce, 0x0a01, + "Realtek", "ALC1220-VB-DT", "Realtek-ALC1220-VB-Desktop"), + + { } /* terminator */ +}; + +static const struct usb_audio_device_name * +lookup_device_name(u32 id) +{ + static const struct usb_audio_device_name *p; + + for (p = usb_audio_names; p->id; p++) + if (p->id == id) + return p; + return NULL; +} + +/* * free the chip instance * * here we have to do not much, since pcm and controls are already freed @@ -357,10 +457,16 @@ static void usb_audio_make_shortname(struct usb_device *dev, const struct snd_usb_audio_quirk *quirk) { struct snd_card *card = chip->card; - - if (quirk && quirk->product_name && *quirk->product_name) { - strlcpy(card->shortname, quirk->product_name, - sizeof(card->shortname)); + const struct usb_audio_device_name *preset; + const char *s = NULL; + + preset = lookup_device_name(chip->usb_id); + if (preset && preset->product_name) + s = preset->product_name; + else if (quirk && quirk->product_name) + s = quirk->product_name; + if (s && *s) { + strlcpy(card->shortname, s, sizeof(card->shortname)); return; } @@ -382,17 +488,26 @@ static void usb_audio_make_longname(struct usb_device *dev, const struct snd_usb_audio_quirk *quirk) { struct snd_card *card = chip->card; + const struct usb_audio_device_name *preset; + const char *s = NULL; int len; + preset = lookup_device_name(chip->usb_id); + /* shortcut - if any pre-defined string is given, use it */ - if (quirk && quirk->profile_name && *quirk->profile_name) { - strlcpy(card->longname, quirk->profile_name, - sizeof(card->longname)); + if (preset && preset->profile_name) + s = preset->profile_name; + if (s && *s) { + strlcpy(card->longname, s, sizeof(card->longname)); return; } - if (quirk && quirk->vendor_name && *quirk->vendor_name) { - len = strlcpy(card->longname, quirk->vendor_name, sizeof(card->longname)); + if (preset && preset->vendor_name) + s = preset->vendor_name; + else if (quirk && quirk->vendor_name) + s = quirk->vendor_name; + if (s && *s) { + len = strlcpy(card->longname, s, sizeof(card->longname)); } else { /* retrieve the vendor and device strings as longname */ if (dev->descriptor.iManufacturer) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 5fbc8dd2f409..e2f9ce2f5b8b 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -318,7 +318,7 @@ static inline void prepare_inbound_urb(struct snd_usb_endpoint *ep, /* * Send output urbs that have been prepared previously. URBs are dequeued - * from ep->ready_playback_urbs and in case there there aren't any available + * from ep->ready_playback_urbs and in case there aren't any available * or there are no packets that have been prepared, this function does * nothing. * diff --git a/sound/usb/midi.c b/sound/usb/midi.c index e8287a05e36b..c8213652470c 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -142,7 +142,7 @@ struct snd_usb_midi_out_endpoint { unsigned int active_urbs; unsigned int drain_urbs; int max_transfer; /* size of urb buffer */ - struct tasklet_struct tasklet; + struct work_struct work; unsigned int next_urb; spinlock_t buffer_lock; @@ -344,9 +344,10 @@ static void snd_usbmidi_do_output(struct snd_usb_midi_out_endpoint *ep) spin_unlock_irqrestore(&ep->buffer_lock, flags); } -static void snd_usbmidi_out_tasklet(struct tasklet_struct *t) +static void snd_usbmidi_out_work(struct work_struct *work) { - struct snd_usb_midi_out_endpoint *ep = from_tasklet(ep, t, tasklet); + struct snd_usb_midi_out_endpoint *ep = + container_of(work, struct snd_usb_midi_out_endpoint, work); snd_usbmidi_do_output(ep); } @@ -1177,7 +1178,7 @@ static void snd_usbmidi_output_trigger(struct snd_rawmidi_substream *substream, snd_rawmidi_proceed(substream); return; } - tasklet_schedule(&port->ep->tasklet); + queue_work(system_highpri_wq, &port->ep->work); } } @@ -1440,7 +1441,7 @@ static int snd_usbmidi_out_endpoint_create(struct snd_usb_midi *umidi, } spin_lock_init(&ep->buffer_lock); - tasklet_setup(&ep->tasklet, snd_usbmidi_out_tasklet); + INIT_WORK(&ep->work, snd_usbmidi_out_work); init_waitqueue_head(&ep->drain_wait); for (i = 0; i < 0x10; ++i) @@ -1503,7 +1504,7 @@ void snd_usbmidi_disconnect(struct list_head *p) for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i) { struct snd_usb_midi_endpoint *ep = &umidi->endpoints[i]; if (ep->out) - tasklet_kill(&ep->out->tasklet); + cancel_work_sync(&ep->out->work); if (ep->out) { for (j = 0; j < OUTPUT_URBS; ++j) usb_kill_urb(ep->out->urbs[j].urb); diff --git a/sound/usb/misc/ua101.c b/sound/usb/misc/ua101.c index 3b2dce1043f5..6b30155964ec 100644 --- a/sound/usb/misc/ua101.c +++ b/sound/usb/misc/ua101.c @@ -96,7 +96,7 @@ struct ua101 { u8 rate_feedback[MAX_QUEUE_LENGTH]; struct list_head ready_playback_urbs; - struct tasklet_struct playback_tasklet; + struct work_struct playback_work; wait_queue_head_t alsa_capture_wait; wait_queue_head_t rate_feedback_wait; wait_queue_head_t alsa_playback_wait; @@ -188,7 +188,7 @@ static void playback_urb_complete(struct urb *usb_urb) spin_lock_irqsave(&ua->lock, flags); list_add_tail(&urb->ready_list, &ua->ready_playback_urbs); if (ua->rate_feedback_count > 0) - tasklet_schedule(&ua->playback_tasklet); + queue_work(system_highpri_wq, &ua->playback_work); ua->playback.substream->runtime->delay -= urb->urb.iso_frame_desc[0].length / ua->playback.frame_bytes; @@ -247,9 +247,9 @@ static inline void add_with_wraparound(struct ua101 *ua, *value -= ua->playback.queue_length; } -static void playback_tasklet(struct tasklet_struct *t) +static void playback_work(struct work_struct *work) { - struct ua101 *ua = from_tasklet(ua, t, playback_tasklet); + struct ua101 *ua = container_of(work, struct ua101, playback_work); unsigned long flags; unsigned int frames; struct ua101_urb *urb; @@ -401,7 +401,7 @@ static void capture_urb_complete(struct urb *urb) } if (test_bit(USB_PLAYBACK_RUNNING, &ua->states) && !list_empty(&ua->ready_playback_urbs)) - tasklet_schedule(&ua->playback_tasklet); + queue_work(system_highpri_wq, &ua->playback_work); } spin_unlock_irqrestore(&ua->lock, flags); @@ -532,7 +532,7 @@ static void stop_usb_playback(struct ua101 *ua) kill_stream_urbs(&ua->playback); - tasklet_kill(&ua->playback_tasklet); + cancel_work_sync(&ua->playback_work); disable_iso_interface(ua, INTF_PLAYBACK); } @@ -550,7 +550,7 @@ static int start_usb_playback(struct ua101 *ua) return 0; kill_stream_urbs(&ua->playback); - tasklet_kill(&ua->playback_tasklet); + cancel_work_sync(&ua->playback_work); err = enable_iso_interface(ua, INTF_PLAYBACK); if (err < 0) @@ -1218,7 +1218,7 @@ static int ua101_probe(struct usb_interface *interface, spin_lock_init(&ua->lock); mutex_init(&ua->mutex); INIT_LIST_HEAD(&ua->ready_playback_urbs); - tasklet_setup(&ua->playback_tasklet, playback_tasklet); + INIT_WORK(&ua->playback_work, playback_work); init_waitqueue_head(&ua->alsa_capture_wait); init_waitqueue_head(&ua->rate_feedback_wait); init_waitqueue_head(&ua->alsa_playback_wait); diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 199cdbfdc761..df036a359f2f 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -2602,6 +2602,216 @@ static int snd_bbfpro_controls_create(struct usb_mixer_interface *mixer) return 0; } +/* + * Pioneer DJ DJM-250MK2 and maybe other DJM models + * + * For playback, no duplicate mapping should be set. + * There are three mixer stereo channels (CH1, CH2, AUX) + * and three stereo sources (Playback 1-2, Playback 3-4, Playback 5-6). + * Each channel should be mapped just once to one source. + * If mapped multiple times, only one source will play on given channel + * (sources are not mixed together). + * + * For recording, duplicate mapping is OK. We will get the same signal multiple times. + * + * Channels 7-8 are in both directions fixed to FX SEND / FX RETURN. + * + * See also notes in the quirks-table.h file. + */ + +struct snd_pioneer_djm_option { + const u16 wIndex; + const u16 wValue; + const char *name; +}; + +static const struct snd_pioneer_djm_option snd_pioneer_djm_options_capture_level[] = { + { .name = "-5 dB", .wValue = 0x0300, .wIndex = 0x8003 }, + { .name = "-10 dB", .wValue = 0x0200, .wIndex = 0x8003 }, + { .name = "-15 dB", .wValue = 0x0100, .wIndex = 0x8003 }, + { .name = "-19 dB", .wValue = 0x0000, .wIndex = 0x8003 } +}; + +static const struct snd_pioneer_djm_option snd_pioneer_djm_options_capture_ch12[] = { + { .name = "CH1 Control Tone PHONO", .wValue = 0x0103, .wIndex = 0x8002 }, + { .name = "CH1 Control Tone LINE", .wValue = 0x0100, .wIndex = 0x8002 }, + { .name = "Post CH1 Fader", .wValue = 0x0106, .wIndex = 0x8002 }, + { .name = "Cross Fader A", .wValue = 0x0107, .wIndex = 0x8002 }, + { .name = "Cross Fader B", .wValue = 0x0108, .wIndex = 0x8002 }, + { .name = "MIC", .wValue = 0x0109, .wIndex = 0x8002 }, + { .name = "AUX", .wValue = 0x010d, .wIndex = 0x8002 }, + { .name = "REC OUT", .wValue = 0x010a, .wIndex = 0x8002 } +}; + +static const struct snd_pioneer_djm_option snd_pioneer_djm_options_capture_ch34[] = { + { .name = "CH2 Control Tone PHONO", .wValue = 0x0203, .wIndex = 0x8002 }, + { .name = "CH2 Control Tone LINE", .wValue = 0x0200, .wIndex = 0x8002 }, + { .name = "Post CH2 Fader", .wValue = 0x0206, .wIndex = 0x8002 }, + { .name = "Cross Fader A", .wValue = 0x0207, .wIndex = 0x8002 }, + { .name = "Cross Fader B", .wValue = 0x0208, .wIndex = 0x8002 }, + { .name = "MIC", .wValue = 0x0209, .wIndex = 0x8002 }, + { .name = "AUX", .wValue = 0x020d, .wIndex = 0x8002 }, + { .name = "REC OUT", .wValue = 0x020a, .wIndex = 0x8002 } +}; + +static const struct snd_pioneer_djm_option snd_pioneer_djm_options_capture_ch56[] = { + { .name = "REC OUT", .wValue = 0x030a, .wIndex = 0x8002 }, + { .name = "Post CH1 Fader", .wValue = 0x0311, .wIndex = 0x8002 }, + { .name = "Post CH2 Fader", .wValue = 0x0312, .wIndex = 0x8002 }, + { .name = "Cross Fader A", .wValue = 0x0307, .wIndex = 0x8002 }, + { .name = "Cross Fader B", .wValue = 0x0308, .wIndex = 0x8002 }, + { .name = "MIC", .wValue = 0x0309, .wIndex = 0x8002 }, + { .name = "AUX", .wValue = 0x030d, .wIndex = 0x8002 } +}; + +static const struct snd_pioneer_djm_option snd_pioneer_djm_options_playback_12[] = { + { .name = "CH1", .wValue = 0x0100, .wIndex = 0x8016 }, + { .name = "CH2", .wValue = 0x0101, .wIndex = 0x8016 }, + { .name = "AUX", .wValue = 0x0104, .wIndex = 0x8016 } +}; + +static const struct snd_pioneer_djm_option snd_pioneer_djm_options_playback_34[] = { + { .name = "CH1", .wValue = 0x0200, .wIndex = 0x8016 }, + { .name = "CH2", .wValue = 0x0201, .wIndex = 0x8016 }, + { .name = "AUX", .wValue = 0x0204, .wIndex = 0x8016 } +}; + +static const struct snd_pioneer_djm_option snd_pioneer_djm_options_playback_56[] = { + { .name = "CH1", .wValue = 0x0300, .wIndex = 0x8016 }, + { .name = "CH2", .wValue = 0x0301, .wIndex = 0x8016 }, + { .name = "AUX", .wValue = 0x0304, .wIndex = 0x8016 } +}; + +struct snd_pioneer_djm_option_group { + const char *name; + const struct snd_pioneer_djm_option *options; + const size_t count; + const u16 default_value; +}; + +#define snd_pioneer_djm_option_group_item(_name, suffix, _default_value) { \ + .name = _name, \ + .options = snd_pioneer_djm_options_##suffix, \ + .count = ARRAY_SIZE(snd_pioneer_djm_options_##suffix), \ + .default_value = _default_value } + +static const struct snd_pioneer_djm_option_group snd_pioneer_djm_option_groups[] = { + snd_pioneer_djm_option_group_item("Master Capture Level Capture Switch", capture_level, 0), + snd_pioneer_djm_option_group_item("Capture 1-2 Capture Switch", capture_ch12, 2), + snd_pioneer_djm_option_group_item("Capture 3-4 Capture Switch", capture_ch34, 2), + snd_pioneer_djm_option_group_item("Capture 5-6 Capture Switch", capture_ch56, 0), + snd_pioneer_djm_option_group_item("Playback 1-2 Playback Switch", playback_12, 0), + snd_pioneer_djm_option_group_item("Playback 3-4 Playback Switch", playback_34, 1), + snd_pioneer_djm_option_group_item("Playback 5-6 Playback Switch", playback_56, 2) +}; + +// layout of the kcontrol->private_value: +#define SND_PIONEER_DJM_VALUE_MASK 0x0000ffff +#define SND_PIONEER_DJM_GROUP_MASK 0xffff0000 +#define SND_PIONEER_DJM_GROUP_SHIFT 16 + +static int snd_pioneer_djm_controls_info(struct snd_kcontrol *kctl, struct snd_ctl_elem_info *info) +{ + u16 group_index = kctl->private_value >> SND_PIONEER_DJM_GROUP_SHIFT; + size_t count; + const char *name; + const struct snd_pioneer_djm_option_group *group; + + if (group_index >= ARRAY_SIZE(snd_pioneer_djm_option_groups)) + return -EINVAL; + + group = &snd_pioneer_djm_option_groups[group_index]; + count = group->count; + if (info->value.enumerated.item >= count) + info->value.enumerated.item = count - 1; + name = group->options[info->value.enumerated.item].name; + strlcpy(info->value.enumerated.name, name, sizeof(info->value.enumerated.name)); + info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + info->count = 1; + info->value.enumerated.items = count; + return 0; +} + +static int snd_pioneer_djm_controls_update(struct usb_mixer_interface *mixer, u16 group, u16 value) +{ + int err; + + if (group >= ARRAY_SIZE(snd_pioneer_djm_option_groups) + || value >= snd_pioneer_djm_option_groups[group].count) + return -EINVAL; + + err = snd_usb_lock_shutdown(mixer->chip); + if (err) + return err; + + err = snd_usb_ctl_msg( + mixer->chip->dev, usb_sndctrlpipe(mixer->chip->dev, 0), + USB_REQ_SET_FEATURE, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_DEVICE, + snd_pioneer_djm_option_groups[group].options[value].wValue, + snd_pioneer_djm_option_groups[group].options[value].wIndex, + NULL, 0); + + snd_usb_unlock_shutdown(mixer->chip); + return err; +} + +static int snd_pioneer_djm_controls_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *elem) +{ + elem->value.enumerated.item[0] = kctl->private_value & SND_PIONEER_DJM_VALUE_MASK; + return 0; +} + +static int snd_pioneer_djm_controls_put(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *elem) +{ + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kctl); + struct usb_mixer_interface *mixer = list->mixer; + unsigned long private_value = kctl->private_value; + u16 group = (private_value & SND_PIONEER_DJM_GROUP_MASK) >> SND_PIONEER_DJM_GROUP_SHIFT; + u16 value = elem->value.enumerated.item[0]; + + kctl->private_value = (group << SND_PIONEER_DJM_GROUP_SHIFT) | value; + + return snd_pioneer_djm_controls_update(mixer, group, value); +} + +static int snd_pioneer_djm_controls_resume(struct usb_mixer_elem_list *list) +{ + unsigned long private_value = list->kctl->private_value; + u16 group = (private_value & SND_PIONEER_DJM_GROUP_MASK) >> SND_PIONEER_DJM_GROUP_SHIFT; + u16 value = (private_value & SND_PIONEER_DJM_VALUE_MASK); + + return snd_pioneer_djm_controls_update(list->mixer, group, value); +} + +static int snd_pioneer_djm_controls_create(struct usb_mixer_interface *mixer) +{ + int err, i; + const struct snd_pioneer_djm_option_group *group; + struct snd_kcontrol_new knew = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .index = 0, + .info = snd_pioneer_djm_controls_info, + .get = snd_pioneer_djm_controls_get, + .put = snd_pioneer_djm_controls_put + }; + + for (i = 0; i < ARRAY_SIZE(snd_pioneer_djm_option_groups); i++) { + group = &snd_pioneer_djm_option_groups[i]; + knew.name = group->name; + knew.private_value = (i << SND_PIONEER_DJM_GROUP_SHIFT) | group->default_value; + err = snd_pioneer_djm_controls_update(mixer, i, group->default_value); + if (err) + return err; + err = add_single_ctl_with_resume(mixer, 0, snd_pioneer_djm_controls_resume, + &knew, NULL); + if (err) + return err; + } + return 0; +} + int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) { int err = 0; @@ -2706,6 +2916,9 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x2a39, 0x3fb0): /* RME Babyface Pro FS */ err = snd_bbfpro_controls_create(mixer); break; + case USB_ID(0x2b73, 0x0017): /* Pioneer DJ DJM-250MK2 */ + err = snd_pioneer_djm_controls_create(mixer); + break; } return err; diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index 0ffff7640892..d33df146d6ce 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -1946,7 +1946,7 @@ static void scarlett2_mixer_interrupt(struct urb *urb) goto requeue; if (len == 8) { - data = le32_to_cpu(*(u32 *)urb->transfer_buffer); + data = le32_to_cpu(*(__le32 *)urb->transfer_buffer); if (data & SCARLETT2_USB_INTERRUPT_VOL_CHANGE) scarlett2_mixer_interrupt_vol_change(mixer); if (data & SCARLETT2_USB_INTERRUPT_BUTTON_CHANGE) diff --git a/sound/usb/mixer_us16x08.c b/sound/usb/mixer_us16x08.c index a4d4d71db55b..92b1a6d9c931 100644 --- a/sound/usb/mixer_us16x08.c +++ b/sound/usb/mixer_us16x08.c @@ -1109,7 +1109,7 @@ static const struct snd_us16x08_control_params eq_controls[] = { .control_id = SND_US16X08_ID_EQLOWFREQ, .type = USB_MIXER_U8, .num_channels = 16, - .name = "EQ Low Frequence", + .name = "EQ Low Frequency", }, { /* EQ mid low gain */ .kcontrol_new = &snd_us16x08_eq_gain_ctl, @@ -1123,7 +1123,7 @@ static const struct snd_us16x08_control_params eq_controls[] = { .control_id = SND_US16X08_ID_EQLOWMIDFREQ, .type = USB_MIXER_U8, .num_channels = 16, - .name = "EQ MidLow Frequence", + .name = "EQ MidLow Frequency", }, { /* EQ mid low Q */ .kcontrol_new = &snd_us16x08_eq_mid_width_ctl, @@ -1144,7 +1144,7 @@ static const struct snd_us16x08_control_params eq_controls[] = { .control_id = SND_US16X08_ID_EQHIGHMIDFREQ, .type = USB_MIXER_U8, .num_channels = 16, - .name = "EQ MidHigh Frequence", + .name = "EQ MidHigh Frequency", }, { /* EQ mid high Q */ .kcontrol_new = &snd_us16x08_eq_mid_width_ctl, @@ -1165,7 +1165,7 @@ static const struct snd_us16x08_control_params eq_controls[] = { .control_id = SND_US16X08_ID_EQHIGHFREQ, .type = USB_MIXER_U8, .num_channels = 16, - .name = "EQ High Frequence", + .name = "EQ High Frequency", }, }; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 23eafd50126f..3c1697f6b60c 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -25,33 +25,16 @@ .idProduct = prod, \ .bInterfaceClass = USB_CLASS_VENDOR_SPEC -#define QUIRK_RENAME_DEVICE(_vendor, _device) \ - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { \ - .vendor_name = _vendor, \ - .product_name = _device, \ - .ifnum = QUIRK_NO_INTERFACE \ - } - -#define QUIRK_DEVICE_PROFILE(_vendor, _device, _profile) \ - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { \ - .vendor_name = _vendor, \ - .product_name = _device, \ - .profile_name = _profile, \ - .ifnum = QUIRK_NO_INTERFACE \ - } +/* 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), - QUIRK_DEVICE_PROFILE("HP", "Thunderbolt Dock Audio Headset", - "HP-Thunderbolt-Dock-Audio-Headset"), -}, -/* HP Thunderbolt Dock Audio Module */ -{ - USB_DEVICE(0x03f0, 0x0567), - QUIRK_DEVICE_PROFILE("HP", "Thunderbolt Dock Audio Module", - "HP-Thunderbolt-Dock-Audio-Module"), -}, /* FTDI devices */ { USB_DEVICE(0x0403, 0xb8d8), @@ -85,44 +68,14 @@ } }, -/* Creative/E-Mu devices */ -{ - USB_DEVICE(0x041e, 0x3010), - QUIRK_RENAME_DEVICE("Creative Labs", "Sound Blaster MP3+") -}, -/* Creative/Toshiba Multimedia Center SB-0500 */ -{ - USB_DEVICE(0x041e, 0x3048), - QUIRK_RENAME_DEVICE("Toshiba", "SB-0500") -}, -{ - /* E-Mu 0202 USB */ - .match_flags = USB_DEVICE_ID_MATCH_DEVICE, - .idVendor = 0x041e, - .idProduct = 0x3f02, - .bInterfaceClass = USB_CLASS_AUDIO, -}, -{ - /* E-Mu 0404 USB */ - .match_flags = USB_DEVICE_ID_MATCH_DEVICE, - .idVendor = 0x041e, - .idProduct = 0x3f04, - .bInterfaceClass = USB_CLASS_AUDIO, -}, -{ - /* E-Mu Tracker Pre */ - .match_flags = USB_DEVICE_ID_MATCH_DEVICE, - .idVendor = 0x041e, - .idProduct = 0x3f0a, - .bInterfaceClass = USB_CLASS_AUDIO, -}, -{ - /* E-Mu 0204 USB */ - .match_flags = USB_DEVICE_ID_MATCH_DEVICE, - .idVendor = 0x041e, - .idProduct = 0x3f19, - .bInterfaceClass = USB_CLASS_AUDIO, -}, +/* E-Mu 0202 USB */ +{ USB_DEVICE_VENDOR_SPEC(0x041e, 0x3f02) }, +/* E-Mu 0404 USB */ +{ USB_DEVICE_VENDOR_SPEC(0x041e, 0x3f04) }, +/* E-Mu Tracker Pre */ +{ USB_DEVICE_VENDOR_SPEC(0x041e, 0x3f0a) }, +/* E-Mu 0204 USB */ +{ USB_DEVICE_VENDOR_SPEC(0x041e, 0x3f19) }, /* * HP Wireless Audio @@ -164,70 +117,13 @@ * Logitech QuickCam: bDeviceClass is vendor-specific, so generic interface * class matches do not take effect without an explicit ID match. */ -{ - .match_flags = USB_DEVICE_ID_MATCH_DEVICE | - USB_DEVICE_ID_MATCH_INT_CLASS | - USB_DEVICE_ID_MATCH_INT_SUBCLASS, - .idVendor = 0x046d, - .idProduct = 0x0850, - .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL -}, -{ - .match_flags = USB_DEVICE_ID_MATCH_DEVICE | - USB_DEVICE_ID_MATCH_INT_CLASS | - USB_DEVICE_ID_MATCH_INT_SUBCLASS, - .idVendor = 0x046d, - .idProduct = 0x08ae, - .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL -}, -{ - .match_flags = USB_DEVICE_ID_MATCH_DEVICE | - USB_DEVICE_ID_MATCH_INT_CLASS | - USB_DEVICE_ID_MATCH_INT_SUBCLASS, - .idVendor = 0x046d, - .idProduct = 0x08c6, - .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL -}, -{ - .match_flags = USB_DEVICE_ID_MATCH_DEVICE | - USB_DEVICE_ID_MATCH_INT_CLASS | - USB_DEVICE_ID_MATCH_INT_SUBCLASS, - .idVendor = 0x046d, - .idProduct = 0x08f0, - .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL -}, -{ - .match_flags = USB_DEVICE_ID_MATCH_DEVICE | - USB_DEVICE_ID_MATCH_INT_CLASS | - USB_DEVICE_ID_MATCH_INT_SUBCLASS, - .idVendor = 0x046d, - .idProduct = 0x08f5, - .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL -}, -{ - .match_flags = USB_DEVICE_ID_MATCH_DEVICE | - USB_DEVICE_ID_MATCH_INT_CLASS | - USB_DEVICE_ID_MATCH_INT_SUBCLASS, - .idVendor = 0x046d, - .idProduct = 0x08f6, - .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL -}, -{ - .match_flags = USB_DEVICE_ID_MATCH_DEVICE | - USB_DEVICE_ID_MATCH_INT_CLASS | - USB_DEVICE_ID_MATCH_INT_SUBCLASS, - .idVendor = 0x046d, - .idProduct = 0x0990, - .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, - QUIRK_RENAME_DEVICE("Logitech, Inc.", "QuickCam Pro 9000") -}, +{ USB_AUDIO_DEVICE(0x046d, 0x0850) }, +{ USB_AUDIO_DEVICE(0x046d, 0x08ae) }, +{ USB_AUDIO_DEVICE(0x046d, 0x08c6) }, +{ USB_AUDIO_DEVICE(0x046d, 0x08f0) }, +{ USB_AUDIO_DEVICE(0x046d, 0x08f5) }, +{ USB_AUDIO_DEVICE(0x046d, 0x08f6) }, +{ USB_AUDIO_DEVICE(0x046d, 0x0990) }, /* * Yamaha devices @@ -2610,10 +2506,6 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - USB_DEVICE(0x0ccd, 0x0028), - QUIRK_RENAME_DEVICE("TerraTec", "Aureon5.1MkII") -}, -{ USB_DEVICE(0x0ccd, 0x0035), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .vendor_name = "Miditech", @@ -2623,16 +2515,6 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, -/* Stanton/N2IT Final Scratch v1 device ('Scratchamp') */ -{ - USB_DEVICE(0x103d, 0x0100), - QUIRK_RENAME_DEVICE("Stanton", "ScratchAmp") -}, -{ - USB_DEVICE(0x103d, 0x0101), - QUIRK_RENAME_DEVICE("Stanton", "ScratchAmp") -}, - /* Novation EMS devices */ { USB_DEVICE_VENDOR_SPEC(0x1235, 0x0001), @@ -2817,20 +2699,10 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, -/* */ -{ - /* aka. Serato Scratch Live DJ Box */ - USB_DEVICE(0x13e5, 0x0001), - QUIRK_RENAME_DEVICE("Rane", "SL-1") -}, - /* Lenovo ThinkStation P620 Rear Line-in, Line-out and Microphone */ { USB_DEVICE(0x17aa, 0x1046), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .vendor_name = "Lenovo", - .product_name = "ThinkStation P620 Rear", - .profile_name = "Lenovo-ThinkStation-P620-Rear", .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_SETUP_DISABLE_AUTOSUSPEND } @@ -2839,9 +2711,6 @@ YAMAHA_DEVICE(0x7010, "UB99"), { USB_DEVICE(0x17aa, 0x104d), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .vendor_name = "Lenovo", - .product_name = "ThinkStation P620 Main", - .profile_name = "Lenovo-ThinkStation-P620-Main", .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_SETUP_DISABLE_AUTOSUSPEND } @@ -2879,10 +2748,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, /* KeithMcMillen Stringport */ -{ - USB_DEVICE(0x1f38, 0x0001), - .bInterfaceClass = USB_CLASS_AUDIO, -}, +{ USB_DEVICE(0x1f38, 0x0001) }, /* FIXME: should be more restrictive matching */ /* Miditech devices */ { @@ -2913,13 +2779,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), */ #define AU0828_DEVICE(vid, pid, vname, pname) { \ - .idVendor = vid, \ - .idProduct = pid, \ - .match_flags = USB_DEVICE_ID_MATCH_DEVICE | \ - USB_DEVICE_ID_MATCH_INT_CLASS | \ - USB_DEVICE_ID_MATCH_INT_SUBCLASS, \ - .bInterfaceClass = USB_CLASS_AUDIO, \ - .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, \ + USB_AUDIO_DEVICE(vid, pid), \ .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { \ .vendor_name = vname, \ .product_name = pname, \ @@ -2949,13 +2809,7 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), /* Syntek STK1160 */ { - .match_flags = USB_DEVICE_ID_MATCH_DEVICE | - USB_DEVICE_ID_MATCH_INT_CLASS | - USB_DEVICE_ID_MATCH_INT_SUBCLASS, - .idVendor = 0x05e1, - .idProduct = 0x0408, - .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, + USB_AUDIO_DEVICE(0x05e1, 0x0408), .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { .vendor_name = "Syntek", .product_name = "STK1160", @@ -3117,10 +2971,7 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), }, { /* Tascam US122 MKII - playback-only support */ - .match_flags = USB_DEVICE_ID_MATCH_DEVICE, - .idVendor = 0x0644, - .idProduct = 0x8021, - .bInterfaceClass = USB_CLASS_AUDIO, + USB_DEVICE_VENDOR_SPEC(0x0644, 0x8021), .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { .vendor_name = "TASCAM", .product_name = "US122 MKII", @@ -3305,19 +3156,6 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } }, -/* - * The original product_name is "USB Sound Device", however this name - * is also used by the CM106 based cards, so make it unique. - */ -{ - USB_DEVICE(0x0d8c, 0x0102), - QUIRK_RENAME_DEVICE(NULL, "ICUSBAUDIO7D") -}, -{ - USB_DEVICE(0x0d8c, 0x0103), - QUIRK_RENAME_DEVICE(NULL, "Audio Advantage MicroII") -}, - /* disabled due to regression for other devices; * see https://bugzilla.kernel.org/show_bug.cgi?id=199905 */ @@ -3418,18 +3256,10 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } } }, -/* Dell WD15 Dock */ -{ - USB_DEVICE(0x0bda, 0x4014), - QUIRK_DEVICE_PROFILE("Dell", "WD15 Dock", "Dell-WD15-Dock") -}, /* Dell WD19 Dock */ { USB_DEVICE(0x0bda, 0x402e), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .vendor_name = "Dell", - .product_name = "WD19 Dock", - .profile_name = "Dell-WD15-Dock", .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_SETUP_FMT_AFTER_RESUME } @@ -3701,33 +3531,6 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } }, -#define ALC1220_VB_DESKTOP(vend, prod) { \ - USB_DEVICE(vend, prod), \ - QUIRK_DEVICE_PROFILE("Realtek", "ALC1220-VB-DT", \ - "Realtek-ALC1220-VB-Desktop") \ -} -ALC1220_VB_DESKTOP(0x0414, 0xa002), /* Gigabyte TRX40 Aorus Pro WiFi */ -ALC1220_VB_DESKTOP(0x0db0, 0x0d64), /* MSI TRX40 Creator */ -ALC1220_VB_DESKTOP(0x0db0, 0x543d), /* MSI TRX40 */ -ALC1220_VB_DESKTOP(0x26ce, 0x0a01), /* Asrock TRX40 Creator */ -#undef ALC1220_VB_DESKTOP - -/* Two entries for Gigabyte TRX40 Aorus Master: - * TRX40 Aorus Master has two USB-audio devices, one for the front headphone - * with ESS SABRE9218 DAC chip, while another for the rest I/O (the rear - * panel and the front mic) with Realtek ALC1220-VB. - * Here we provide two distinct names for making UCM profiles easier. - */ -{ - USB_DEVICE(0x0414, 0xa000), - QUIRK_DEVICE_PROFILE("Gigabyte", "Aorus Master Front Headphone", - "Gigabyte-Aorus-Master-Front-Headphone") -}, -{ - USB_DEVICE(0x0414, 0xa001), - QUIRK_DEVICE_PROFILE("Gigabyte", "Aorus Master Main Audio", - "Gigabyte-Aorus-Master-Main-Audio") -}, { /* * Pioneer DJ DJM-900NXS2 @@ -3804,13 +3607,7 @@ ALC1220_VB_DESKTOP(0x26ce, 0x0a01), /* Asrock TRX40 Creator */ * channels to be swapped and out of phase, which is dealt with in quirks.c. */ { - .match_flags = USB_DEVICE_ID_MATCH_DEVICE | - USB_DEVICE_ID_MATCH_INT_CLASS | - USB_DEVICE_ID_MATCH_INT_SUBCLASS, - .idVendor = 0x534d, - .idProduct = 0x2109, - .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, + USB_AUDIO_DEVICE(0x534d, 0x2109), .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { .vendor_name = "MacroSilicon", .product_name = "MS2109", @@ -3851,3 +3648,4 @@ ALC1220_VB_DESKTOP(0x26ce, 0x0a01), /* Asrock TRX40 Creator */ }, #undef USB_DEVICE_VENDOR_SPEC +#undef USB_AUDIO_DEVICE diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 6839915a0128..0805b7f21272 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -110,7 +110,6 @@ enum quirk_type { struct snd_usb_audio_quirk { const char *vendor_name; const char *product_name; - const char *profile_name; /* override the card->longname */ int16_t ifnum; uint16_t type; bool shares_media_device; |