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authorTakashi Iwai <tiwai@suse.de>2020-10-12 09:51:00 +0300
committerTakashi Iwai <tiwai@suse.de>2020-10-12 09:51:00 +0300
commit4dda3a19141b44102860b46e307153ed8b32ea7b (patch)
treef2326ed10e594f7a7979cb45c77e1b8a52e10090
parent148ebf548a1af366fc797fcc7d03f0bb92b12a79 (diff)
parent96e503f9000f2ad17d550cd884a5e386eb7f532f (diff)
downloadlinux-4dda3a19141b44102860b46e307153ed8b32ea7b.tar.xz
Merge branch 'for-next' into for-linus
-rw-r--r--include/drm/drm_audio_component.h4
-rw-r--r--include/sound/pcm_params.h5
-rw-r--r--include/sound/timer.h8
-rw-r--r--sound/ac97/ac97_core.h2
-rw-r--r--sound/aoa/soundbus/i2sbus/pcm.c3
-rw-r--r--sound/atmel/ac97c.c22
-rw-r--r--sound/core/compress_offload.c5
-rw-r--r--sound/core/control.c56
-rw-r--r--sound/core/control_compat.c14
-rw-r--r--sound/core/hrtimer.c2
-rw-r--r--sound/core/hwdep.c27
-rw-r--r--sound/core/hwdep_compat.c23
-rw-r--r--sound/core/init.c3
-rw-r--r--sound/core/memalloc.c2
-rw-r--r--sound/core/pcm.c8
-rw-r--r--sound/core/pcm_memory.c3
-rw-r--r--sound/core/rawmidi.c2
-rw-r--r--sound/core/seq/oss/seq_oss.c7
-rw-r--r--sound/core/timer.c26
-rw-r--r--sound/drivers/aloop.c23
-rw-r--r--sound/drivers/pcsp/pcsp_lib.c12
-rw-r--r--sound/drivers/portman2x4.c2
-rw-r--r--sound/drivers/vx/vx_core.c4
-rw-r--r--sound/drivers/vx/vx_pcm.c2
-rw-r--r--sound/firewire/amdtp-stream.c25
-rw-r--r--sound/firewire/amdtp-stream.h2
-rw-r--r--sound/hda/hdac_component.c3
-rw-r--r--sound/hda/hdac_i915.c69
-rw-r--r--sound/pci/asihpi/asihpi.c37
-rw-r--r--sound/pci/asihpi/hpioctl.c16
-rw-r--r--sound/pci/asihpi/hpios.h2
-rw-r--r--sound/pci/hda/hda_auto_parser.c2
-rw-r--r--sound/pci/hda/hda_intel.c6
-rw-r--r--sound/pci/hda/hda_jack.h2
-rw-r--r--sound/pci/hda/hda_local.h8
-rw-r--r--sound/pci/hda/patch_ca0132.c1782
-rw-r--r--sound/pci/hda/patch_hdmi.c1
-rw-r--r--sound/pci/mixart/mixart.h2
-rw-r--r--sound/pci/riptide/riptide.c20
-rw-r--r--sound/pci/rme9652/hdsp.c55
-rw-r--r--sound/pci/rme9652/hdspm.c15
-rw-r--r--sound/usb/card.c133
-rw-r--r--sound/usb/endpoint.c2
-rw-r--r--sound/usb/midi.c13
-rw-r--r--sound/usb/misc/ua101.c16
-rw-r--r--sound/usb/mixer_quirks.c213
-rw-r--r--sound/usb/mixer_scarlett_gen2.c2
-rw-r--r--sound/usb/mixer_us16x08.c8
-rw-r--r--sound/usb/quirks-table.h262
-rw-r--r--sound/usb/usbaudio.h1
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(&register_mutex);
+ if (cmd != SNDCTL_SEQ_SYNC &&
+ mutex_lock_interruptible(&register_mutex))
+ return -ERESTARTSYS;
rc = snd_seq_oss_ioctl(dp, cmd, arg);
- mutex_unlock(&register_mutex);
+ if (cmd != SNDCTL_SEQ_SYNC)
+ mutex_unlock(&register_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(&register_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;