Merge branch 'for-linus' into for-next
Back-merge of UAC3 fixes for applying further enhancements.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
diff --git a/include/linux/usb/audio-v3.h b/include/linux/usb/audio-v3.h
index a8959aa..38add1d 100644
--- a/include/linux/usb/audio-v3.h
+++ b/include/linux/usb/audio-v3.h
@@ -392,4 +392,30 @@ struct uac3_interrupt_data_msg {
#define UAC3_AC_ACTIVE_INTERFACE_CONTROL 0x01
#define UAC3_AC_POWER_DOMAIN_CONTROL 0x02
+/* BADD predefined Unit/Terminal values */
+#define UAC3_BADD_IT_ID1 1 /* Input Terminal ID1: bTerminalID = 1 */
+#define UAC3_BADD_FU_ID2 2 /* Feature Unit ID2: bUnitID = 2 */
+#define UAC3_BADD_OT_ID3 3 /* Output Terminal ID3: bTerminalID = 3 */
+#define UAC3_BADD_IT_ID4 4 /* Input Terminal ID4: bTerminalID = 4 */
+#define UAC3_BADD_FU_ID5 5 /* Feature Unit ID5: bUnitID = 5 */
+#define UAC3_BADD_OT_ID6 6 /* Output Terminal ID6: bTerminalID = 6 */
+#define UAC3_BADD_FU_ID7 7 /* Feature Unit ID7: bUnitID = 7 */
+#define UAC3_BADD_MU_ID8 8 /* Mixer Unit ID8: bUnitID = 8 */
+#define UAC3_BADD_CS_ID9 9 /* Clock Source Entity ID9: bClockID = 9 */
+#define UAC3_BADD_PD_ID10 10 /* Power Domain ID10: bPowerDomainID = 10 */
+#define UAC3_BADD_PD_ID11 11 /* Power Domain ID11: bPowerDomainID = 11 */
+
+/* BADD wMaxPacketSize of AS endpoints */
+#define UAC3_BADD_EP_MAXPSIZE_SYNC_MONO_16 0x0060
+#define UAC3_BADD_EP_MAXPSIZE_ASYNC_MONO_16 0x0062
+#define UAC3_BADD_EP_MAXPSIZE_SYNC_MONO_24 0x0090
+#define UAC3_BADD_EP_MAXPSIZE_ASYNC_MONO_24 0x0093
+#define UAC3_BADD_EP_MAXPSIZE_SYNC_STEREO_16 0x00C0
+#define UAC3_BADD_EP_MAXPSIZE_ASYNC_STEREO_16 0x00C4
+#define UAC3_BADD_EP_MAXPSIZE_SYNC_STEREO_24 0x0120
+#define UAC3_BADD_EP_MAXPSIZE_ASYNC_STEREO_24 0x0126
+
+/* BADD sample rate is always fixed to 48kHz */
+#define UAC3_BADD_SAMPLING_RATE 48000
+
#endif /* __LINUX_USB_AUDIO_V3_H */
diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h
index 5ebcc51..8c1572d 100644
--- a/include/sound/emu10k1.h
+++ b/include/sound/emu10k1.h
@@ -1610,7 +1610,7 @@ struct snd_emu10k1_fx8010_pcm {
struct snd_pcm_indirect pcm_rec;
unsigned int tram_pos;
unsigned int tram_shift;
- struct snd_emu10k1_fx8010_irq *irq;
+ struct snd_emu10k1_fx8010_irq irq;
};
struct snd_emu10k1_fx8010 {
@@ -1902,7 +1902,7 @@ int snd_emu10k1_fx8010_register_irq_handler(struct snd_emu10k1 *emu,
snd_fx8010_irq_handler_t *handler,
unsigned char gpr_running,
void *private_data,
- struct snd_emu10k1_fx8010_irq **r_irq);
+ struct snd_emu10k1_fx8010_irq *irq);
int snd_emu10k1_fx8010_unregister_irq_handler(struct snd_emu10k1 *emu,
struct snd_emu10k1_fx8010_irq *irq);
diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h
index 06536e0..c052afc 100644
--- a/include/sound/hdaudio.h
+++ b/include/sound/hdaudio.h
@@ -571,4 +571,9 @@ static inline unsigned int snd_array_index(struct snd_array *array, void *ptr)
return (unsigned long)(ptr - array->list) / array->elem_size;
}
+/* a helper macro to iterate for each snd_array element */
+#define snd_array_for_each(array, idx, ptr) \
+ for ((idx) = 0, (ptr) = (array)->list; (idx) < (array)->used; \
+ (ptr) = snd_array_elem(array, ++(idx)))
+
#endif /* __SOUND_HDAUDIO_H */
diff --git a/include/uapi/sound/tlv.h b/include/uapi/sound/tlv.h
index be5371f..7d6d65f 100644
--- a/include/uapi/sound/tlv.h
+++ b/include/uapi/sound/tlv.h
@@ -42,6 +42,10 @@
#define SNDRV_CTL_TLVD_LENGTH(...) \
((unsigned int)sizeof((const unsigned int[]) { __VA_ARGS__ }))
+/* Accessor offsets for TLV data items */
+#define SNDRV_CTL_TLVO_TYPE 0
+#define SNDRV_CTL_TLVO_LEN 1
+
#define SNDRV_CTL_TLVD_CONTAINER_ITEM(...) \
SNDRV_CTL_TLVD_ITEM(SNDRV_CTL_TLVT_CONTAINER, __VA_ARGS__)
#define SNDRV_CTL_TLVD_DECLARE_CONTAINER(name, ...) \
@@ -61,6 +65,10 @@
SNDRV_CTL_TLVD_DB_SCALE_ITEM(min, step, mute) \
}
+/* Accessor offsets for min, mute and step items in dB scale type TLV */
+#define SNDRV_CTL_TLVO_DB_SCALE_MIN 2
+#define SNDRV_CTL_TLVO_DB_SCALE_MUTE_AND_STEP 3
+
/* dB scale specified with min/max values instead of step */
#define SNDRV_CTL_TLVD_DB_MINMAX_ITEM(min_dB, max_dB) \
SNDRV_CTL_TLVD_ITEM(SNDRV_CTL_TLVT_DB_MINMAX, (min_dB), (max_dB))
@@ -75,6 +83,10 @@
SNDRV_CTL_TLVD_DB_MINMAX_MUTE_ITEM(min_dB, max_dB) \
}
+/* Accessor offsets for min, max items in db-minmax types of TLV. */
+#define SNDRV_CTL_TLVO_DB_MINMAX_MIN 2
+#define SNDRV_CTL_TLVO_DB_MINMAX_MAX 3
+
/* linear volume between min_dB and max_dB (.01dB unit) */
#define SNDRV_CTL_TLVD_DB_LINEAR_ITEM(min_dB, max_dB) \
SNDRV_CTL_TLVD_ITEM(SNDRV_CTL_TLVT_DB_LINEAR, (min_dB), (max_dB))
@@ -83,6 +95,10 @@
SNDRV_CTL_TLVD_DB_LINEAR_ITEM(min_dB, max_dB) \
}
+/* Accessor offsets for min, max items in db-linear type of TLV. */
+#define SNDRV_CTL_TLVO_DB_LINEAR_MIN 2
+#define SNDRV_CTL_TLVO_DB_LINEAR_MAX 3
+
/* dB range container:
* Items in dB range container must be ordered by their values and by their
* dB values. This implies that larger values must correspond with larger
diff --git a/sound/core/pcm_compat.c b/sound/core/pcm_compat.c
index 6491afb..39d853b 100644
--- a/sound/core/pcm_compat.c
+++ b/sound/core/pcm_compat.c
@@ -45,10 +45,7 @@ static int snd_pcm_ioctl_rewind_compat(struct snd_pcm_substream *substream,
if (get_user(frames, src))
return -EFAULT;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- err = snd_pcm_playback_rewind(substream, frames);
- else
- err = snd_pcm_capture_rewind(substream, frames);
+ err = snd_pcm_rewind(substream, frames);
if (put_user(err, src))
return -EFAULT;
return err < 0 ? err : 0;
@@ -62,10 +59,7 @@ static int snd_pcm_ioctl_forward_compat(struct snd_pcm_substream *substream,
if (get_user(frames, src))
return -EFAULT;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- err = snd_pcm_playback_forward(substream, frames);
- else
- err = snd_pcm_capture_forward(substream, frames);
+ err = snd_pcm_forward(substream, frames);
if (put_user(err, src))
return -EFAULT;
return err < 0 ? err : 0;
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index f4a1950..44b5ae8 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -191,10 +191,7 @@ int snd_pcm_update_state(struct snd_pcm_substream *substream,
{
snd_pcm_uframes_t avail;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- avail = snd_pcm_playback_avail(runtime);
- else
- avail = snd_pcm_capture_avail(runtime);
+ avail = snd_pcm_avail(substream);
if (avail > runtime->avail_max)
runtime->avail_max = avail;
if (runtime->status->state == SNDRV_PCM_STATE_DRAINING) {
@@ -1856,10 +1853,7 @@ static int wait_for_avail(struct snd_pcm_substream *substream,
* This check must happen after been added to the waitqueue
* and having current state be INTERRUPTIBLE.
*/
- if (is_playback)
- avail = snd_pcm_playback_avail(runtime);
- else
- avail = snd_pcm_capture_avail(runtime);
+ avail = snd_pcm_avail(substream);
if (avail >= runtime->twake)
break;
snd_pcm_stream_unlock_irq(substream);
@@ -2175,10 +2169,7 @@ snd_pcm_sframes_t __snd_pcm_lib_xfer(struct snd_pcm_substream *substream,
runtime->twake = runtime->control->avail_min ? : 1;
if (runtime->status->state == SNDRV_PCM_STATE_RUNNING)
snd_pcm_update_hw_ptr(substream);
- if (is_playback)
- avail = snd_pcm_playback_avail(runtime);
- else
- avail = snd_pcm_capture_avail(runtime);
+ avail = snd_pcm_avail(substream);
while (size > 0) {
snd_pcm_uframes_t frames, appl_ptr, appl_ofs;
snd_pcm_uframes_t cont;
diff --git a/sound/core/pcm_local.h b/sound/core/pcm_local.h
index 16f2547..7a499d0 100644
--- a/sound/core/pcm_local.h
+++ b/sound/core/pcm_local.h
@@ -36,6 +36,24 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream);
void snd_pcm_playback_silence(struct snd_pcm_substream *substream,
snd_pcm_uframes_t new_hw_ptr);
+static inline snd_pcm_uframes_t
+snd_pcm_avail(struct snd_pcm_substream *substream)
+{
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ return snd_pcm_playback_avail(substream->runtime);
+ else
+ return snd_pcm_capture_avail(substream->runtime);
+}
+
+static inline snd_pcm_uframes_t
+snd_pcm_hw_avail(struct snd_pcm_substream *substream)
+{
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ return snd_pcm_playback_hw_avail(substream->runtime);
+ else
+ return snd_pcm_capture_hw_avail(substream->runtime);
+}
+
#ifdef CONFIG_SND_PCM_TIMER
void snd_pcm_timer_resolution_change(struct snd_pcm_substream *substream);
void snd_pcm_timer_init(struct snd_pcm_substream *substream);
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 0e875d5..04c6301 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -99,6 +99,57 @@ static inline void down_write_nonblock(struct rw_semaphore *lock)
cond_resched();
}
+#define PCM_LOCK_DEFAULT 0
+#define PCM_LOCK_IRQ 1
+#define PCM_LOCK_IRQSAVE 2
+
+static unsigned long __snd_pcm_stream_lock_mode(struct snd_pcm_substream *substream,
+ unsigned int mode)
+{
+ unsigned long flags = 0;
+ if (substream->pcm->nonatomic) {
+ down_read_nested(&snd_pcm_link_rwsem, SINGLE_DEPTH_NESTING);
+ mutex_lock(&substream->self_group.mutex);
+ } else {
+ switch (mode) {
+ case PCM_LOCK_DEFAULT:
+ read_lock(&snd_pcm_link_rwlock);
+ break;
+ case PCM_LOCK_IRQ:
+ read_lock_irq(&snd_pcm_link_rwlock);
+ break;
+ case PCM_LOCK_IRQSAVE:
+ read_lock_irqsave(&snd_pcm_link_rwlock, flags);
+ break;
+ }
+ spin_lock(&substream->self_group.lock);
+ }
+ return flags;
+}
+
+static void __snd_pcm_stream_unlock_mode(struct snd_pcm_substream *substream,
+ unsigned int mode, unsigned long flags)
+{
+ if (substream->pcm->nonatomic) {
+ mutex_unlock(&substream->self_group.mutex);
+ up_read(&snd_pcm_link_rwsem);
+ } else {
+ spin_unlock(&substream->self_group.lock);
+
+ switch (mode) {
+ case PCM_LOCK_DEFAULT:
+ read_unlock(&snd_pcm_link_rwlock);
+ break;
+ case PCM_LOCK_IRQ:
+ read_unlock_irq(&snd_pcm_link_rwlock);
+ break;
+ case PCM_LOCK_IRQSAVE:
+ read_unlock_irqrestore(&snd_pcm_link_rwlock, flags);
+ break;
+ }
+ }
+}
+
/**
* snd_pcm_stream_lock - Lock the PCM stream
* @substream: PCM substream
@@ -109,13 +160,7 @@ static inline void down_write_nonblock(struct rw_semaphore *lock)
*/
void snd_pcm_stream_lock(struct snd_pcm_substream *substream)
{
- if (substream->pcm->nonatomic) {
- down_read_nested(&snd_pcm_link_rwsem, SINGLE_DEPTH_NESTING);
- mutex_lock(&substream->self_group.mutex);
- } else {
- read_lock(&snd_pcm_link_rwlock);
- spin_lock(&substream->self_group.lock);
- }
+ __snd_pcm_stream_lock_mode(substream, PCM_LOCK_DEFAULT);
}
EXPORT_SYMBOL_GPL(snd_pcm_stream_lock);
@@ -127,13 +172,7 @@ EXPORT_SYMBOL_GPL(snd_pcm_stream_lock);
*/
void snd_pcm_stream_unlock(struct snd_pcm_substream *substream)
{
- if (substream->pcm->nonatomic) {
- mutex_unlock(&substream->self_group.mutex);
- up_read(&snd_pcm_link_rwsem);
- } else {
- spin_unlock(&substream->self_group.lock);
- read_unlock(&snd_pcm_link_rwlock);
- }
+ __snd_pcm_stream_unlock_mode(substream, PCM_LOCK_DEFAULT, 0);
}
EXPORT_SYMBOL_GPL(snd_pcm_stream_unlock);
@@ -147,9 +186,7 @@ EXPORT_SYMBOL_GPL(snd_pcm_stream_unlock);
*/
void snd_pcm_stream_lock_irq(struct snd_pcm_substream *substream)
{
- if (!substream->pcm->nonatomic)
- local_irq_disable();
- snd_pcm_stream_lock(substream);
+ __snd_pcm_stream_lock_mode(substream, PCM_LOCK_IRQ);
}
EXPORT_SYMBOL_GPL(snd_pcm_stream_lock_irq);
@@ -161,19 +198,13 @@ EXPORT_SYMBOL_GPL(snd_pcm_stream_lock_irq);
*/
void snd_pcm_stream_unlock_irq(struct snd_pcm_substream *substream)
{
- snd_pcm_stream_unlock(substream);
- if (!substream->pcm->nonatomic)
- local_irq_enable();
+ __snd_pcm_stream_unlock_mode(substream, PCM_LOCK_IRQ, 0);
}
EXPORT_SYMBOL_GPL(snd_pcm_stream_unlock_irq);
unsigned long _snd_pcm_stream_lock_irqsave(struct snd_pcm_substream *substream)
{
- unsigned long flags = 0;
- if (!substream->pcm->nonatomic)
- local_irq_save(flags);
- snd_pcm_stream_lock(substream);
- return flags;
+ return __snd_pcm_stream_lock_mode(substream, PCM_LOCK_IRQSAVE);
}
EXPORT_SYMBOL_GPL(_snd_pcm_stream_lock_irqsave);
@@ -187,9 +218,7 @@ EXPORT_SYMBOL_GPL(_snd_pcm_stream_lock_irqsave);
void snd_pcm_stream_unlock_irqrestore(struct snd_pcm_substream *substream,
unsigned long flags)
{
- snd_pcm_stream_unlock(substream);
- if (!substream->pcm->nonatomic)
- local_irq_restore(flags);
+ __snd_pcm_stream_unlock_mode(substream, PCM_LOCK_IRQSAVE, flags);
}
EXPORT_SYMBOL_GPL(snd_pcm_stream_unlock_irqrestore);
@@ -857,6 +886,18 @@ static int snd_pcm_sw_params_user(struct snd_pcm_substream *substream,
return err;
}
+static inline snd_pcm_uframes_t
+snd_pcm_calc_delay(struct snd_pcm_substream *substream)
+{
+ snd_pcm_uframes_t delay;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ delay = snd_pcm_playback_hw_avail(substream->runtime);
+ else
+ delay = snd_pcm_capture_avail(substream->runtime);
+ return delay + substream->runtime->delay;
+}
+
int snd_pcm_status(struct snd_pcm_substream *substream,
struct snd_pcm_status *status)
{
@@ -908,21 +949,9 @@ int snd_pcm_status(struct snd_pcm_substream *substream,
_tstamp_end:
status->appl_ptr = runtime->control->appl_ptr;
status->hw_ptr = runtime->status->hw_ptr;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- status->avail = snd_pcm_playback_avail(runtime);
- if (runtime->status->state == SNDRV_PCM_STATE_RUNNING ||
- runtime->status->state == SNDRV_PCM_STATE_DRAINING) {
- status->delay = runtime->buffer_size - status->avail;
- status->delay += runtime->delay;
- } else
- status->delay = 0;
- } else {
- status->avail = snd_pcm_capture_avail(runtime);
- if (runtime->status->state == SNDRV_PCM_STATE_RUNNING)
- status->delay = status->avail + runtime->delay;
- else
- status->delay = 0;
- }
+ status->avail = snd_pcm_avail(substream);
+ status->delay = snd_pcm_running(substream) ?
+ snd_pcm_calc_delay(substream) : 0;
status->avail_max = runtime->avail_max;
status->overrange = runtime->overrange;
runtime->avail_max = 0;
@@ -2610,10 +2639,9 @@ static snd_pcm_sframes_t rewind_appl_ptr(struct snd_pcm_substream *substream,
return ret < 0 ? 0 : frames;
}
-static snd_pcm_sframes_t snd_pcm_playback_rewind(struct snd_pcm_substream *substream,
- snd_pcm_uframes_t frames)
+static snd_pcm_sframes_t snd_pcm_rewind(struct snd_pcm_substream *substream,
+ snd_pcm_uframes_t frames)
{
- struct snd_pcm_runtime *runtime = substream->runtime;
snd_pcm_sframes_t ret;
if (frames == 0)
@@ -2623,33 +2651,14 @@ static snd_pcm_sframes_t snd_pcm_playback_rewind(struct snd_pcm_substream *subst
ret = do_pcm_hwsync(substream);
if (!ret)
ret = rewind_appl_ptr(substream, frames,
- snd_pcm_playback_hw_avail(runtime));
+ snd_pcm_hw_avail(substream));
snd_pcm_stream_unlock_irq(substream);
return ret;
}
-static snd_pcm_sframes_t snd_pcm_capture_rewind(struct snd_pcm_substream *substream,
- snd_pcm_uframes_t frames)
+static snd_pcm_sframes_t snd_pcm_forward(struct snd_pcm_substream *substream,
+ snd_pcm_uframes_t frames)
{
- struct snd_pcm_runtime *runtime = substream->runtime;
- snd_pcm_sframes_t ret;
-
- if (frames == 0)
- return 0;
-
- snd_pcm_stream_lock_irq(substream);
- ret = do_pcm_hwsync(substream);
- if (!ret)
- ret = rewind_appl_ptr(substream, frames,
- snd_pcm_capture_hw_avail(runtime));
- snd_pcm_stream_unlock_irq(substream);
- return ret;
-}
-
-static snd_pcm_sframes_t snd_pcm_playback_forward(struct snd_pcm_substream *substream,
- snd_pcm_uframes_t frames)
-{
- struct snd_pcm_runtime *runtime = substream->runtime;
snd_pcm_sframes_t ret;
if (frames == 0)
@@ -2659,25 +2668,7 @@ static snd_pcm_sframes_t snd_pcm_playback_forward(struct snd_pcm_substream *subs
ret = do_pcm_hwsync(substream);
if (!ret)
ret = forward_appl_ptr(substream, frames,
- snd_pcm_playback_avail(runtime));
- snd_pcm_stream_unlock_irq(substream);
- return ret;
-}
-
-static snd_pcm_sframes_t snd_pcm_capture_forward(struct snd_pcm_substream *substream,
- snd_pcm_uframes_t frames)
-{
- struct snd_pcm_runtime *runtime = substream->runtime;
- snd_pcm_sframes_t ret;
-
- if (frames == 0)
- return 0;
-
- snd_pcm_stream_lock_irq(substream);
- ret = do_pcm_hwsync(substream);
- if (!ret)
- ret = forward_appl_ptr(substream, frames,
- snd_pcm_capture_avail(runtime));
+ snd_pcm_avail(substream));
snd_pcm_stream_unlock_irq(substream);
return ret;
}
@@ -2695,19 +2686,13 @@ static int snd_pcm_hwsync(struct snd_pcm_substream *substream)
static int snd_pcm_delay(struct snd_pcm_substream *substream,
snd_pcm_sframes_t *delay)
{
- struct snd_pcm_runtime *runtime = substream->runtime;
int err;
snd_pcm_sframes_t n = 0;
snd_pcm_stream_lock_irq(substream);
err = do_pcm_hwsync(substream);
- if (!err) {
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- n = snd_pcm_playback_hw_avail(runtime);
- else
- n = snd_pcm_capture_avail(runtime);
- n += runtime->delay;
- }
+ if (!err)
+ n = snd_pcm_calc_delay(substream);
snd_pcm_stream_unlock_irq(substream);
if (!err)
*delay = n;
@@ -2834,10 +2819,7 @@ static int snd_pcm_rewind_ioctl(struct snd_pcm_substream *substream,
return -EFAULT;
if (put_user(0, _frames))
return -EFAULT;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- result = snd_pcm_playback_rewind(substream, frames);
- else
- result = snd_pcm_capture_rewind(substream, frames);
+ result = snd_pcm_rewind(substream, frames);
__put_user(result, _frames);
return result < 0 ? result : 0;
}
@@ -2852,10 +2834,7 @@ static int snd_pcm_forward_ioctl(struct snd_pcm_substream *substream,
return -EFAULT;
if (put_user(0, _frames))
return -EFAULT;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- result = snd_pcm_playback_forward(substream, frames);
- else
- result = snd_pcm_capture_forward(substream, frames);
+ result = snd_pcm_forward(substream, frames);
__put_user(result, _frames);
return result < 0 ? result : 0;
}
@@ -2998,7 +2977,7 @@ int snd_pcm_kernel_ioctl(struct snd_pcm_substream *substream,
/* provided only for OSS; capture-only and no value returned */
if (substream->stream != SNDRV_PCM_STREAM_CAPTURE)
return -EINVAL;
- result = snd_pcm_capture_forward(substream, *frames);
+ result = snd_pcm_forward(substream, *frames);
return result < 0 ? result : 0;
}
case SNDRV_PCM_IOCTL_HW_PARAMS:
@@ -3140,82 +3119,46 @@ static ssize_t snd_pcm_writev(struct kiocb *iocb, struct iov_iter *from)
return result;
}
-static __poll_t snd_pcm_playback_poll(struct file *file, poll_table * wait)
+static __poll_t snd_pcm_poll(struct file *file, poll_table *wait)
{
struct snd_pcm_file *pcm_file;
struct snd_pcm_substream *substream;
struct snd_pcm_runtime *runtime;
- __poll_t mask;
+ __poll_t mask, ok;
snd_pcm_uframes_t avail;
pcm_file = file->private_data;
substream = pcm_file->substream;
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ok = EPOLLOUT | EPOLLWRNORM;
+ else
+ ok = EPOLLIN | EPOLLRDNORM;
if (PCM_RUNTIME_CHECK(substream))
- return EPOLLOUT | EPOLLWRNORM | EPOLLERR;
- runtime = substream->runtime;
+ return ok | EPOLLERR;
+ runtime = substream->runtime;
poll_wait(file, &runtime->sleep, wait);
+ mask = 0;
snd_pcm_stream_lock_irq(substream);
- avail = snd_pcm_playback_avail(runtime);
+ avail = snd_pcm_avail(substream);
switch (runtime->status->state) {
case SNDRV_PCM_STATE_RUNNING:
case SNDRV_PCM_STATE_PREPARED:
case SNDRV_PCM_STATE_PAUSED:
- if (avail >= runtime->control->avail_min) {
- mask = EPOLLOUT | EPOLLWRNORM;
- break;
- }
- /* Fall through */
- case SNDRV_PCM_STATE_DRAINING:
- mask = 0;
- break;
- default:
- mask = EPOLLOUT | EPOLLWRNORM | EPOLLERR;
- break;
- }
- snd_pcm_stream_unlock_irq(substream);
- return mask;
-}
-
-static __poll_t snd_pcm_capture_poll(struct file *file, poll_table * wait)
-{
- struct snd_pcm_file *pcm_file;
- struct snd_pcm_substream *substream;
- struct snd_pcm_runtime *runtime;
- __poll_t mask;
- snd_pcm_uframes_t avail;
-
- pcm_file = file->private_data;
-
- substream = pcm_file->substream;
- if (PCM_RUNTIME_CHECK(substream))
- return EPOLLIN | EPOLLRDNORM | EPOLLERR;
- runtime = substream->runtime;
-
- poll_wait(file, &runtime->sleep, wait);
-
- snd_pcm_stream_lock_irq(substream);
- avail = snd_pcm_capture_avail(runtime);
- switch (runtime->status->state) {
- case SNDRV_PCM_STATE_RUNNING:
- case SNDRV_PCM_STATE_PREPARED:
- case SNDRV_PCM_STATE_PAUSED:
- if (avail >= runtime->control->avail_min) {
- mask = EPOLLIN | EPOLLRDNORM;
- break;
- }
- mask = 0;
+ if (avail >= runtime->control->avail_min)
+ mask = ok;
break;
case SNDRV_PCM_STATE_DRAINING:
- if (avail > 0) {
- mask = EPOLLIN | EPOLLRDNORM;
- break;
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ mask = ok;
+ if (!avail)
+ mask |= EPOLLERR;
}
- /* Fall through */
+ break;
default:
- mask = EPOLLIN | EPOLLRDNORM | EPOLLERR;
+ mask = ok | EPOLLERR;
break;
}
snd_pcm_stream_unlock_irq(substream);
@@ -3707,7 +3650,7 @@ const struct file_operations snd_pcm_f_ops[2] = {
.open = snd_pcm_playback_open,
.release = snd_pcm_release,
.llseek = no_llseek,
- .poll = snd_pcm_playback_poll,
+ .poll = snd_pcm_poll,
.unlocked_ioctl = snd_pcm_ioctl,
.compat_ioctl = snd_pcm_ioctl_compat,
.mmap = snd_pcm_mmap,
@@ -3721,7 +3664,7 @@ const struct file_operations snd_pcm_f_ops[2] = {
.open = snd_pcm_capture_open,
.release = snd_pcm_release,
.llseek = no_llseek,
- .poll = snd_pcm_capture_poll,
+ .poll = snd_pcm_poll,
.unlocked_ioctl = snd_pcm_ioctl,
.compat_ioctl = snd_pcm_ioctl_compat,
.mmap = snd_pcm_mmap,
diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c
index 9e96186..58fa3f9 100644
--- a/sound/core/vmaster.c
+++ b/sound/core/vmaster.c
@@ -421,13 +421,15 @@ struct snd_kcontrol *snd_ctl_make_virtual_master(char *name,
kctl->private_free = master_free;
/* additional (constant) TLV read */
- if (tlv &&
- (tlv[0] == SNDRV_CTL_TLVT_DB_SCALE ||
- tlv[0] == SNDRV_CTL_TLVT_DB_MINMAX ||
- tlv[0] == SNDRV_CTL_TLVT_DB_MINMAX_MUTE)) {
- kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ;
- memcpy(master->tlv, tlv, sizeof(master->tlv));
- kctl->tlv.p = master->tlv;
+ if (tlv) {
+ unsigned int type = tlv[SNDRV_CTL_TLVO_TYPE];
+ if (type == SNDRV_CTL_TLVT_DB_SCALE ||
+ type == SNDRV_CTL_TLVT_DB_MINMAX ||
+ type == SNDRV_CTL_TLVT_DB_MINMAX_MUTE) {
+ kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ;
+ memcpy(master->tlv, tlv, sizeof(master->tlv));
+ kctl->tlv.p = master->tlv;
+ }
}
return kctl;
diff --git a/sound/firewire/dice/Makefile b/sound/firewire/dice/Makefile
index 55b4be9..7b7997a 100644
--- a/sound/firewire/dice/Makefile
+++ b/sound/firewire/dice/Makefile
@@ -1,3 +1,4 @@
snd-dice-objs := dice-transaction.o dice-stream.o dice-proc.o dice-midi.o \
- dice-pcm.o dice-hwdep.o dice.o
+ dice-pcm.o dice-hwdep.o dice.o dice-tcelectronic.o \
+ dice-alesis.o dice-extension.o
obj-$(CONFIG_SND_DICE) += snd-dice.o
diff --git a/sound/firewire/dice/dice-alesis.c b/sound/firewire/dice/dice-alesis.c
new file mode 100644
index 0000000..b2efb1c
--- /dev/null
+++ b/sound/firewire/dice/dice-alesis.c
@@ -0,0 +1,52 @@
+// SPDX-License-Identifier: GPL-2.0
+/*
+ * dice-alesis.c - a part of driver for DICE based devices
+ *
+ * Copyright (c) 2018 Takashi Sakamoto
+ */
+
+#include "dice.h"
+
+static const unsigned int
+alesis_io14_tx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT] = {
+ {6, 6, 4}, /* Tx0 = Analog + S/PDIF. */
+ {8, 4, 0}, /* Tx1 = ADAT1. */
+};
+
+static const unsigned int
+alesis_io26_tx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT] = {
+ {10, 10, 8}, /* Tx0 = Analog + S/PDIF. */
+ {16, 8, 0}, /* Tx1 = ADAT1 + ADAT2. */
+};
+
+int snd_dice_detect_alesis_formats(struct snd_dice *dice)
+{
+ __be32 reg;
+ u32 data;
+ int i;
+ int err;
+
+ err = snd_dice_transaction_read_tx(dice, TX_NUMBER_AUDIO, ®,
+ sizeof(reg));
+ if (err < 0)
+ return err;
+ data = be32_to_cpu(reg);
+
+ if (data == 4 || data == 6) {
+ memcpy(dice->tx_pcm_chs, alesis_io14_tx_pcm_chs,
+ MAX_STREAMS * SND_DICE_RATE_MODE_COUNT *
+ sizeof(unsigned int));
+ } else {
+ memcpy(dice->rx_pcm_chs, alesis_io26_tx_pcm_chs,
+ MAX_STREAMS * SND_DICE_RATE_MODE_COUNT *
+ sizeof(unsigned int));
+ }
+
+ for (i = 0; i < SND_DICE_RATE_MODE_COUNT; ++i)
+ dice->rx_pcm_chs[0][i] = 8;
+
+ dice->tx_midi_ports[0] = 1;
+ dice->rx_midi_ports[0] = 1;
+
+ return 0;
+}
diff --git a/sound/firewire/dice/dice-extension.c b/sound/firewire/dice/dice-extension.c
new file mode 100644
index 0000000..a63fcbc
--- /dev/null
+++ b/sound/firewire/dice/dice-extension.c
@@ -0,0 +1,172 @@
+// SPDX-License-Identifier: GPL-2.0
+/*
+ * dice-extension.c - a part of driver for DICE based devices
+ *
+ * Copyright (c) 2018 Takashi Sakamoto
+ */
+
+#include "dice.h"
+
+/* For TCD2210/2220, TCAT defines extension of application protocol. */
+
+#define DICE_EXT_APP_SPACE 0xffffe0200000uLL
+
+#define DICE_EXT_APP_CAPS_OFFSET 0x00
+#define DICE_EXT_APP_CAPS_SIZE 0x04
+#define DICE_EXT_APP_CMD_OFFSET 0x08
+#define DICE_EXT_APP_CMD_SIZE 0x0c
+#define DICE_EXT_APP_MIXER_OFFSET 0x10
+#define DICE_EXT_APP_MIXER_SIZE 0x14
+#define DICE_EXT_APP_PEAK_OFFSET 0x18
+#define DICE_EXT_APP_PEAK_SIZE 0x1c
+#define DICE_EXT_APP_ROUTER_OFFSET 0x20
+#define DICE_EXT_APP_ROUTER_SIZE 0x24
+#define DICE_EXT_APP_STREAM_OFFSET 0x28
+#define DICE_EXT_APP_STREAM_SIZE 0x2c
+#define DICE_EXT_APP_CURRENT_OFFSET 0x30
+#define DICE_EXT_APP_CURRENT_SIZE 0x34
+#define DICE_EXT_APP_STANDALONE_OFFSET 0x38
+#define DICE_EXT_APP_STANDALONE_SIZE 0x3c
+#define DICE_EXT_APP_APPLICATION_OFFSET 0x40
+#define DICE_EXT_APP_APPLICATION_SIZE 0x44
+
+#define EXT_APP_STREAM_TX_NUMBER 0x0000
+#define EXT_APP_STREAM_RX_NUMBER 0x0004
+#define EXT_APP_STREAM_ENTRIES 0x0008
+#define EXT_APP_STREAM_ENTRY_SIZE 0x010c
+#define EXT_APP_NUMBER_AUDIO 0x0000
+#define EXT_APP_NUMBER_MIDI 0x0004
+#define EXT_APP_NAMES 0x0008
+#define EXT_APP_NAMES_SIZE 256
+#define EXT_APP_AC3 0x0108
+
+#define EXT_APP_CONFIG_LOW_ROUTER 0x0000
+#define EXT_APP_CONFIG_LOW_STREAM 0x1000
+#define EXT_APP_CONFIG_MIDDLE_ROUTER 0x2000
+#define EXT_APP_CONFIG_MIDDLE_STREAM 0x3000
+#define EXT_APP_CONFIG_HIGH_ROUTER 0x4000
+#define EXT_APP_CONFIG_HIGH_STREAM 0x5000
+
+static inline int read_transaction(struct snd_dice *dice, u64 section_addr,
+ u32 offset, void *buf, size_t len)
+{
+ return snd_fw_transaction(dice->unit,
+ len == 4 ? TCODE_READ_QUADLET_REQUEST :
+ TCODE_READ_BLOCK_REQUEST,
+ section_addr + offset, buf, len, 0);
+}
+
+static int read_stream_entries(struct snd_dice *dice, u64 section_addr,
+ u32 base_offset, unsigned int stream_count,
+ unsigned int mode,
+ unsigned int pcm_channels[MAX_STREAMS][3],
+ unsigned int midi_ports[MAX_STREAMS])
+{
+ u32 entry_offset;
+ __be32 reg[2];
+ int err;
+ int i;
+
+ for (i = 0; i < stream_count; ++i) {
+ entry_offset = base_offset + i * EXT_APP_STREAM_ENTRY_SIZE;
+ err = read_transaction(dice, section_addr,
+ entry_offset + EXT_APP_NUMBER_AUDIO,
+ reg, sizeof(reg));
+ if (err < 0)
+ return err;
+ pcm_channels[i][mode] = be32_to_cpu(reg[0]);
+ midi_ports[i] = max(midi_ports[i], be32_to_cpu(reg[1]));
+ }
+
+ return 0;
+}
+
+static int detect_stream_formats(struct snd_dice *dice, u64 section_addr)
+{
+ u32 base_offset;
+ __be32 reg[2];
+ unsigned int stream_count;
+ int mode;
+ int err = 0;
+
+ for (mode = 0; mode < SND_DICE_RATE_MODE_COUNT; ++mode) {
+ unsigned int cap;
+
+ /*
+ * Some models report stream formats at highest mode, however
+ * they don't support the mode. Check clock capabilities.
+ */
+ if (mode == 2) {
+ cap = CLOCK_CAP_RATE_176400 | CLOCK_CAP_RATE_192000;
+ } else if (mode == 1) {
+ cap = CLOCK_CAP_RATE_88200 | CLOCK_CAP_RATE_96000;
+ } else {
+ cap = CLOCK_CAP_RATE_32000 | CLOCK_CAP_RATE_44100 |
+ CLOCK_CAP_RATE_48000;
+ }
+ if (!(cap & dice->clock_caps))
+ continue;
+
+ base_offset = 0x2000 * mode + 0x1000;
+
+ err = read_transaction(dice, section_addr,
+ base_offset + EXT_APP_STREAM_TX_NUMBER,
+ ®, sizeof(reg));
+ if (err < 0)
+ break;
+
+ base_offset += EXT_APP_STREAM_ENTRIES;
+ stream_count = be32_to_cpu(reg[0]);
+ err = read_stream_entries(dice, section_addr, base_offset,
+ stream_count, mode,
+ dice->tx_pcm_chs,
+ dice->tx_midi_ports);
+ if (err < 0)
+ break;
+
+ base_offset += stream_count * EXT_APP_STREAM_ENTRY_SIZE;
+ stream_count = be32_to_cpu(reg[1]);
+ err = read_stream_entries(dice, section_addr, base_offset,
+ stream_count,
+ mode, dice->rx_pcm_chs,
+ dice->rx_midi_ports);
+ if (err < 0)
+ break;
+ }
+
+ return err;
+}
+
+int snd_dice_detect_extension_formats(struct snd_dice *dice)
+{
+ __be32 *pointers;
+ unsigned int i;
+ u64 section_addr;
+ int err;
+
+ pointers = kmalloc_array(9, sizeof(__be32) * 2, GFP_KERNEL);
+ if (pointers == NULL)
+ return -ENOMEM;
+
+ err = snd_fw_transaction(dice->unit, TCODE_READ_BLOCK_REQUEST,
+ DICE_EXT_APP_SPACE, pointers,
+ 9 * sizeof(__be32) * 2, 0);
+ if (err < 0)
+ goto end;
+
+ /* Check two of them for offset have the same value or not. */
+ for (i = 0; i < 9; ++i) {
+ int j;
+
+ for (j = i + 1; j < 9; ++j) {
+ if (pointers[i * 2] == pointers[j * 2])
+ goto end;
+ }
+ }
+
+ section_addr = DICE_EXT_APP_SPACE + be32_to_cpu(pointers[12]) * 4;
+ err = detect_stream_formats(dice, section_addr);
+end:
+ kfree(pointers);
+ return err;
+}
diff --git a/sound/firewire/dice/dice-interface.h b/sound/firewire/dice/dice-interface.h
index 15a484b..9cad3d6 100644
--- a/sound/firewire/dice/dice-interface.h
+++ b/sound/firewire/dice/dice-interface.h
@@ -175,13 +175,18 @@
#define GLOBAL_SAMPLE_RATE 0x05c
/*
+ * Some old firmware versions do not have the following global registers.
+ * Windows drivers produced by TCAT lost backward compatibility in its
+ * early release because they can handle firmware only which supports the
+ * following registers.
+ */
+
+/*
* The version of the DICE driver specification that this device conforms to;
* read-only.
*/
#define GLOBAL_VERSION 0x060
-/* Some old firmware versions do not have the following global registers: */
-
/*
* Supported sample rates and clock sources; read-only.
*/
diff --git a/sound/firewire/dice/dice-midi.c b/sound/firewire/dice/dice-midi.c
index 8ff6da3..84eca8a 100644
--- a/sound/firewire/dice/dice-midi.c
+++ b/sound/firewire/dice/dice-midi.c
@@ -101,27 +101,18 @@ int snd_dice_create_midi(struct snd_dice *dice)
.close = midi_close,
.trigger = midi_playback_trigger,
};
- __be32 reg;
struct snd_rawmidi *rmidi;
struct snd_rawmidi_str *str;
unsigned int midi_in_ports, midi_out_ports;
+ int i;
int err;
- /*
- * Use the number of MIDI conformant data channel at current sampling
- * transfer frequency.
- */
- err = snd_dice_transaction_read_tx(dice, TX_NUMBER_MIDI,
- ®, sizeof(reg));
- if (err < 0)
- return err;
- midi_in_ports = be32_to_cpu(reg);
-
- err = snd_dice_transaction_read_rx(dice, RX_NUMBER_MIDI,
- ®, sizeof(reg));
- if (err < 0)
- return err;
- midi_out_ports = be32_to_cpu(reg);
+ midi_in_ports = 0;
+ midi_out_ports = 0;
+ for (i = 0; i < MAX_STREAMS; ++i) {
+ midi_in_ports = max(midi_in_ports, dice->tx_midi_ports[i]);
+ midi_out_ports = max(midi_out_ports, dice->rx_midi_ports[i]);
+ }
if (midi_in_ports + midi_out_ports == 0)
return 0;
diff --git a/sound/firewire/dice/dice-pcm.c b/sound/firewire/dice/dice-pcm.c
index 7cb9e97..80351b2 100644
--- a/sound/firewire/dice/dice-pcm.c
+++ b/sound/firewire/dice/dice-pcm.c
@@ -9,43 +9,115 @@
#include "dice.h"
+static int dice_rate_constraint(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct snd_pcm_substream *substream = rule->private;
+ struct snd_dice *dice = substream->private_data;
+ unsigned int index = substream->pcm->device;
+
+ const struct snd_interval *c =
+ hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+ struct snd_interval *r =
+ hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
+ struct snd_interval rates = {
+ .min = UINT_MAX, .max = 0, .integer = 1
+ };
+ unsigned int *pcm_channels;
+ enum snd_dice_rate_mode mode;
+ unsigned int i, rate;
+
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ pcm_channels = dice->tx_pcm_chs[index];
+ else
+ pcm_channels = dice->rx_pcm_chs[index];
+
+ for (i = 0; i < ARRAY_SIZE(snd_dice_rates); ++i) {
+ rate = snd_dice_rates[i];
+ if (snd_dice_stream_get_rate_mode(dice, rate, &mode) < 0)
+ continue;
+
+ if (!snd_interval_test(c, pcm_channels[mode]))
+ continue;
+
+ rates.min = min(rates.min, rate);
+ rates.max = max(rates.max, rate);
+ }
+
+ return snd_interval_refine(r, &rates);
+}
+
+static int dice_channels_constraint(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct snd_pcm_substream *substream = rule->private;
+ struct snd_dice *dice = substream->private_data;
+ unsigned int index = substream->pcm->device;
+
+ const struct snd_interval *r =
+ hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_RATE);
+ struct snd_interval *c =
+ hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+ struct snd_interval channels = {
+ .min = UINT_MAX, .max = 0, .integer = 1
+ };
+ unsigned int *pcm_channels;
+ enum snd_dice_rate_mode mode;
+ unsigned int i, rate;
+
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ pcm_channels = dice->tx_pcm_chs[index];
+ else
+ pcm_channels = dice->rx_pcm_chs[index];
+
+ for (i = 0; i < ARRAY_SIZE(snd_dice_rates); ++i) {
+ rate = snd_dice_rates[i];
+ if (snd_dice_stream_get_rate_mode(dice, rate, &mode) < 0)
+ continue;
+
+ if (!snd_interval_test(r, rate))
+ continue;
+
+ channels.min = min(channels.min, pcm_channels[mode]);
+ channels.max = max(channels.max, pcm_channels[mode]);
+ }
+
+ return snd_interval_refine(c, &channels);
+}
+
static int limit_channels_and_rates(struct snd_dice *dice,
struct snd_pcm_runtime *runtime,
enum amdtp_stream_direction dir,
- unsigned int index, unsigned int size)
+ unsigned int index)
{
struct snd_pcm_hardware *hw = &runtime->hw;
- struct amdtp_stream *stream;
- unsigned int rate;
- __be32 reg;
- int err;
+ unsigned int *pcm_channels;
+ unsigned int i;
- /*
- * Retrieve current Multi Bit Linear Audio data channel and limit to
- * it.
- */
- if (dir == AMDTP_IN_STREAM) {
- stream = &dice->tx_stream[index];
- err = snd_dice_transaction_read_tx(dice,
- size * index + TX_NUMBER_AUDIO,
- ®, sizeof(reg));
- } else {
- stream = &dice->rx_stream[index];
- err = snd_dice_transaction_read_rx(dice,
- size * index + RX_NUMBER_AUDIO,
- ®, sizeof(reg));
+ if (dir == AMDTP_IN_STREAM)
+ pcm_channels = dice->tx_pcm_chs[index];
+ else
+ pcm_channels = dice->rx_pcm_chs[index];
+
+ hw->channels_min = UINT_MAX;
+ hw->channels_max = 0;
+
+ for (i = 0; i < ARRAY_SIZE(snd_dice_rates); ++i) {
+ enum snd_dice_rate_mode mode;
+ unsigned int rate, channels;
+
+ rate = snd_dice_rates[i];
+ if (snd_dice_stream_get_rate_mode(dice, rate, &mode) < 0)
+ continue;
+ hw->rates |= snd_pcm_rate_to_rate_bit(rate);
+
+ channels = pcm_channels[mode];
+ if (channels == 0)
+ continue;
+ hw->channels_min = min(hw->channels_min, channels);
+ hw->channels_max = max(hw->channels_max, channels);
}
- if (err < 0)
- return err;
- hw->channels_min = hw->channels_max = be32_to_cpu(reg);
-
- /* Retrieve current sampling transfer frequency and limit to it. */
- err = snd_dice_transaction_get_rate(dice, &rate);
- if (err < 0)
- return err;
-
- hw->rates = snd_pcm_rate_to_rate_bit(rate);
snd_pcm_limit_hw_rates(runtime);
return 0;
@@ -56,36 +128,34 @@ static int init_hw_info(struct snd_dice *dice,
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_pcm_hardware *hw = &runtime->hw;
+ unsigned int index = substream->pcm->device;
enum amdtp_stream_direction dir;
struct amdtp_stream *stream;
- __be32 reg[2];
- unsigned int count, size;
int err;
if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
hw->formats = AM824_IN_PCM_FORMAT_BITS;
dir = AMDTP_IN_STREAM;
- stream = &dice->tx_stream[substream->pcm->device];
- err = snd_dice_transaction_read_tx(dice, TX_NUMBER, reg,
- sizeof(reg));
+ stream = &dice->tx_stream[index];
} else {
hw->formats = AM824_OUT_PCM_FORMAT_BITS;
dir = AMDTP_OUT_STREAM;
- stream = &dice->rx_stream[substream->pcm->device];
- err = snd_dice_transaction_read_rx(dice, RX_NUMBER, reg,
- sizeof(reg));
+ stream = &dice->rx_stream[index];
}
+ err = limit_channels_and_rates(dice, substream->runtime, dir,
+ index);
if (err < 0)
return err;
- count = min_t(unsigned int, be32_to_cpu(reg[0]), MAX_STREAMS);
- if (substream->pcm->device >= count)
- return -ENXIO;
-
- size = be32_to_cpu(reg[1]) * 4;
- err = limit_channels_and_rates(dice, substream->runtime, dir,
- substream->pcm->device, size);
+ err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ dice_rate_constraint, substream,
+ SNDRV_PCM_HW_PARAM_CHANNELS, -1);
+ if (err < 0)
+ return err;
+ err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+ dice_channels_constraint, substream,
+ SNDRV_PCM_HW_PARAM_RATE, -1);
if (err < 0)
return err;
@@ -95,6 +165,8 @@ static int init_hw_info(struct snd_dice *dice,
static int pcm_open(struct snd_pcm_substream *substream)
{
struct snd_dice *dice = substream->private_data;
+ unsigned int source;
+ bool internal;
int err;
err = snd_dice_stream_lock_try(dice);
@@ -105,6 +177,43 @@ static int pcm_open(struct snd_pcm_substream *substream)
if (err < 0)
goto err_locked;
+ err = snd_dice_transaction_get_clock_source(dice, &source);
+ if (err < 0)
+ goto err_locked;
+ switch (source) {
+ case CLOCK_SOURCE_AES1:
+ case CLOCK_SOURCE_AES2:
+ case CLOCK_SOURCE_AES3:
+ case CLOCK_SOURCE_AES4:
+ case CLOCK_SOURCE_AES_ANY:
+ case CLOCK_SOURCE_ADAT:
+ case CLOCK_SOURCE_TDIF:
+ case CLOCK_SOURCE_WC:
+ internal = false;
+ break;
+ default:
+ internal = true;
+ break;
+ }
+
+ /*
+ * When source of clock is not internal or any PCM streams are running,
+ * available sampling rate is limited at current sampling rate.
+ */
+ if (!internal ||
+ amdtp_stream_pcm_running(&dice->tx_stream[0]) ||
+ amdtp_stream_pcm_running(&dice->tx_stream[1]) ||
+ amdtp_stream_pcm_running(&dice->rx_stream[0]) ||
+ amdtp_stream_pcm_running(&dice->rx_stream[1])) {
+ unsigned int rate;
+
+ err = snd_dice_transaction_get_rate(dice, &rate);
+ if (err < 0)
+ goto err_locked;
+ substream->runtime->hw.rate_min = rate;
+ substream->runtime->hw.rate_max = rate;
+ }
+
snd_pcm_set_sync(substream);
end:
return err;
@@ -318,37 +427,19 @@ int snd_dice_create_pcm(struct snd_dice *dice)
.page = snd_pcm_lib_get_vmalloc_page,
.mmap = snd_pcm_lib_mmap_vmalloc,
};
- __be32 reg;
struct snd_pcm *pcm;
- unsigned int i, max_capture, max_playback, capture, playback;
+ unsigned int capture, playback;
+ int i, j;
int err;
- /* Check whether PCM substreams are required. */
- if (dice->force_two_pcms) {
- max_capture = max_playback = 2;
- } else {
- max_capture = max_playback = 0;
- err = snd_dice_transaction_read_tx(dice, TX_NUMBER, ®,
- sizeof(reg));
- if (err < 0)
- return err;
- max_capture = min_t(unsigned int, be32_to_cpu(reg), MAX_STREAMS);
-
- err = snd_dice_transaction_read_rx(dice, RX_NUMBER, ®,
- sizeof(reg));
- if (err < 0)
- return err;
- max_playback = min_t(unsigned int, be32_to_cpu(reg), MAX_STREAMS);
- }
-
for (i = 0; i < MAX_STREAMS; i++) {
capture = playback = 0;
- if (i < max_capture)
- capture = 1;
- if (i < max_playback)
- playback = 1;
- if (capture == 0 && playback == 0)
- break;
+ for (j = 0; j < SND_DICE_RATE_MODE_COUNT; ++j) {
+ if (dice->tx_pcm_chs[i][j] > 0)
+ capture = 1;
+ if (dice->rx_pcm_chs[i][j] > 0)
+ playback = 1;
+ }
err = snd_pcm_new(dice->card, "DICE", i, playback, capture,
&pcm);
diff --git a/sound/firewire/dice/dice-proc.c b/sound/firewire/dice/dice-proc.c
index f5c1d1b..faf8c85 100644
--- a/sound/firewire/dice/dice-proc.c
+++ b/sound/firewire/dice/dice-proc.c
@@ -148,12 +148,12 @@ static void dice_proc_read(struct snd_info_entry *entry,
>> CLOCK_RATE_SHIFT));
snd_iprintf(buffer, " ext status: %08x\n", buf.global.extended_status);
snd_iprintf(buffer, " sample rate: %u\n", buf.global.sample_rate);
- snd_iprintf(buffer, " version: %u.%u.%u.%u\n",
- (buf.global.version >> 24) & 0xff,
- (buf.global.version >> 16) & 0xff,
- (buf.global.version >> 8) & 0xff,
- (buf.global.version >> 0) & 0xff);
if (quadlets >= 90) {
+ snd_iprintf(buffer, " version: %u.%u.%u.%u\n",
+ (buf.global.version >> 24) & 0xff,
+ (buf.global.version >> 16) & 0xff,
+ (buf.global.version >> 8) & 0xff,
+ (buf.global.version >> 0) & 0xff);
snd_iprintf(buffer, " clock caps:");
for (i = 0; i <= 6; ++i)
if (buf.global.clock_caps & (1 << i))
@@ -243,10 +243,74 @@ static void dice_proc_read(struct snd_info_entry *entry,
}
}
-void snd_dice_create_proc(struct snd_dice *dice)
+static void dice_proc_read_formation(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ static const char *const rate_labels[] = {
+ [SND_DICE_RATE_MODE_LOW] = "low",
+ [SND_DICE_RATE_MODE_MIDDLE] = "middle",
+ [SND_DICE_RATE_MODE_HIGH] = "high",
+ };
+ struct snd_dice *dice = entry->private_data;
+ int i, j;
+
+ snd_iprintf(buffer, "Output stream from unit:\n");
+ for (i = 0; i < SND_DICE_RATE_MODE_COUNT; ++i)
+ snd_iprintf(buffer, "\t%s", rate_labels[i]);
+ snd_iprintf(buffer, "\tMIDI\n");
+ for (i = 0; i < MAX_STREAMS; ++i) {
+ snd_iprintf(buffer, "Tx %u:", i);
+ for (j = 0; j < SND_DICE_RATE_MODE_COUNT; ++j)
+ snd_iprintf(buffer, "\t%u", dice->tx_pcm_chs[i][j]);
+ snd_iprintf(buffer, "\t%u\n", dice->tx_midi_ports[i]);
+ }
+
+ snd_iprintf(buffer, "Input stream to unit:\n");
+ for (i = 0; i < SND_DICE_RATE_MODE_COUNT; ++i)
+ snd_iprintf(buffer, "\t%s", rate_labels[i]);
+ snd_iprintf(buffer, "\n");
+ for (i = 0; i < MAX_STREAMS; ++i) {
+ snd_iprintf(buffer, "Rx %u:", i);
+ for (j = 0; j < SND_DICE_RATE_MODE_COUNT; ++j)
+ snd_iprintf(buffer, "\t%u", dice->rx_pcm_chs[i][j]);
+ snd_iprintf(buffer, "\t%u\n", dice->rx_midi_ports[i]);
+ }
+}
+
+static void add_node(struct snd_dice *dice, struct snd_info_entry *root,
+ const char *name,
+ void (*op)(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer))
{
struct snd_info_entry *entry;
- if (!snd_card_proc_new(dice->card, "dice", &entry))
- snd_info_set_text_ops(entry, dice, dice_proc_read);
+ entry = snd_info_create_card_entry(dice->card, name, root);
+ if (!entry)
+ return;
+
+ snd_info_set_text_ops(entry, dice, op);
+ if (snd_info_register(entry) < 0)
+ snd_info_free_entry(entry);
+}
+
+void snd_dice_create_proc(struct snd_dice *dice)
+{
+ struct snd_info_entry *root;
+
+ /*
+ * All nodes are automatically removed at snd_card_disconnect(),
+ * by following to link list.
+ */
+ root = snd_info_create_card_entry(dice->card, "firewire",
+ dice->card->proc_root);
+ if (!root)
+ return;
+ root->mode = S_IFDIR | S_IRUGO | S_IXUGO;
+ if (snd_info_register(root) < 0) {
+ snd_info_free_entry(root);
+ return;
+ }
+
+ add_node(dice, root, "dice", dice_proc_read);
+ add_node(dice, root, "formation", dice_proc_read_formation);
}
diff --git a/sound/firewire/dice/dice-stream.c b/sound/firewire/dice/dice-stream.c
index 928a255..c3c892c 100644
--- a/sound/firewire/dice/dice-stream.c
+++ b/sound/firewire/dice/dice-stream.c
@@ -30,13 +30,43 @@ const unsigned int snd_dice_rates[SND_DICE_RATES_COUNT] = {
[6] = 192000,
};
+int snd_dice_stream_get_rate_mode(struct snd_dice *dice, unsigned int rate,
+ enum snd_dice_rate_mode *mode)
+{
+ /* Corresponding to each entry in snd_dice_rates. */
+ static const enum snd_dice_rate_mode modes[] = {
+ [0] = SND_DICE_RATE_MODE_LOW,
+ [1] = SND_DICE_RATE_MODE_LOW,
+ [2] = SND_DICE_RATE_MODE_LOW,
+ [3] = SND_DICE_RATE_MODE_MIDDLE,
+ [4] = SND_DICE_RATE_MODE_MIDDLE,
+ [5] = SND_DICE_RATE_MODE_HIGH,
+ [6] = SND_DICE_RATE_MODE_HIGH,
+ };
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(snd_dice_rates); i++) {
+ if (!(dice->clock_caps & BIT(i)))
+ continue;
+ if (snd_dice_rates[i] != rate)
+ continue;
+
+ *mode = modes[i];
+ return 0;
+ }
+
+ return -EINVAL;
+}
+
/*
* This operation has an effect to synchronize GLOBAL_STATUS/GLOBAL_SAMPLE_RATE
* to GLOBAL_STATUS. Especially, just after powering on, these are different.
*/
-static int ensure_phase_lock(struct snd_dice *dice)
+static int ensure_phase_lock(struct snd_dice *dice, unsigned int rate)
{
__be32 reg, nominal;
+ u32 data;
+ int i;
int err;
err = snd_dice_transaction_read_global(dice, GLOBAL_CLOCK_SELECT,
@@ -44,9 +74,21 @@ static int ensure_phase_lock(struct snd_dice *dice)
if (err < 0)
return err;
+ data = be32_to_cpu(reg);
+
+ data &= ~CLOCK_RATE_MASK;
+ for (i = 0; i < ARRAY_SIZE(snd_dice_rates); ++i) {
+ if (snd_dice_rates[i] == rate)
+ break;
+ }
+ if (i == ARRAY_SIZE(snd_dice_rates))
+ return -EINVAL;
+ data |= i << CLOCK_RATE_SHIFT;
+
if (completion_done(&dice->clock_accepted))
reinit_completion(&dice->clock_accepted);
+ reg = cpu_to_be32(data);
err = snd_dice_transaction_write_global(dice, GLOBAL_CLOCK_SELECT,
®, sizeof(reg));
if (err < 0)
@@ -192,6 +234,7 @@ static int start_streams(struct snd_dice *dice, enum amdtp_stream_direction dir,
unsigned int rate, struct reg_params *params)
{
__be32 reg[2];
+ enum snd_dice_rate_mode mode;
unsigned int i, pcm_chs, midi_ports;
struct amdtp_stream *streams;
struct fw_iso_resources *resources;
@@ -206,12 +249,23 @@ static int start_streams(struct snd_dice *dice, enum amdtp_stream_direction dir,
resources = dice->rx_resources;
}
+ err = snd_dice_stream_get_rate_mode(dice, rate, &mode);
+ if (err < 0)
+ return err;
+
for (i = 0; i < params->count; i++) {
+ unsigned int pcm_cache;
+ unsigned int midi_cache;
+
if (dir == AMDTP_IN_STREAM) {
+ pcm_cache = dice->tx_pcm_chs[i][mode];
+ midi_cache = dice->tx_midi_ports[i];
err = snd_dice_transaction_read_tx(dice,
params->size * i + TX_NUMBER_AUDIO,
reg, sizeof(reg));
} else {
+ pcm_cache = dice->rx_pcm_chs[i][mode];
+ midi_cache = dice->rx_midi_ports[i];
err = snd_dice_transaction_read_rx(dice,
params->size * i + RX_NUMBER_AUDIO,
reg, sizeof(reg));
@@ -221,6 +275,14 @@ static int start_streams(struct snd_dice *dice, enum amdtp_stream_direction dir,
pcm_chs = be32_to_cpu(reg[0]);
midi_ports = be32_to_cpu(reg[1]);
+ /* These are important for developer of this driver. */
+ if (pcm_chs != pcm_cache || midi_ports != midi_cache) {
+ dev_info(&dice->unit->device,
+ "cache mismatch: pcm: %u:%u, midi: %u:%u\n",
+ pcm_chs, pcm_cache, midi_ports, midi_cache);
+ return -EPROTO;
+ }
+
err = keep_resources(dice, dir, i, rate, pcm_chs, midi_ports);
if (err < 0)
return err;
@@ -256,6 +318,68 @@ static int start_streams(struct snd_dice *dice, enum amdtp_stream_direction dir,
return err;
}
+static int start_duplex_streams(struct snd_dice *dice, unsigned int rate)
+{
+ struct reg_params tx_params, rx_params;
+ int i;
+ int err;
+
+ err = get_register_params(dice, &tx_params, &rx_params);
+ if (err < 0)
+ return err;
+
+ /* Stop transmission. */
+ stop_streams(dice, AMDTP_IN_STREAM, &tx_params);
+ stop_streams(dice, AMDTP_OUT_STREAM, &rx_params);
+ snd_dice_transaction_clear_enable(dice);
+ release_resources(dice);
+
+ err = ensure_phase_lock(dice, rate);
+ if (err < 0) {
+ dev_err(&dice->unit->device, "fail to ensure phase lock\n");
+ return err;
+ }
+
+ /* Likely to have changed stream formats. */
+ err = get_register_params(dice, &tx_params, &rx_params);
+ if (err < 0)
+ return err;
+
+ /* Start both streams. */
+ err = start_streams(dice, AMDTP_IN_STREAM, rate, &tx_params);
+ if (err < 0)
+ goto error;
+ err = start_streams(dice, AMDTP_OUT_STREAM, rate, &rx_params);
+ if (err < 0)
+ goto error;
+
+ err = snd_dice_transaction_set_enable(dice);
+ if (err < 0) {
+ dev_err(&dice->unit->device, "fail to enable interface\n");
+ goto error;
+ }
+
+ for (i = 0; i < MAX_STREAMS; i++) {
+ if ((i < tx_params.count &&
+ !amdtp_stream_wait_callback(&dice->tx_stream[i],
+ CALLBACK_TIMEOUT)) ||
+ (i < rx_params.count &&
+ !amdtp_stream_wait_callback(&dice->rx_stream[i],
+ CALLBACK_TIMEOUT))) {
+ err = -ETIMEDOUT;
+ goto error;
+ }
+ }
+
+ return 0;
+error:
+ stop_streams(dice, AMDTP_IN_STREAM, &tx_params);
+ stop_streams(dice, AMDTP_OUT_STREAM, &rx_params);
+ snd_dice_transaction_clear_enable(dice);
+ release_resources(dice);
+ return err;
+}
+
/*
* MEMO: After this function, there're two states of streams:
* - None streams are running.
@@ -265,17 +389,13 @@ int snd_dice_stream_start_duplex(struct snd_dice *dice, unsigned int rate)
{
unsigned int curr_rate;
unsigned int i;
- struct reg_params tx_params, rx_params;
- bool need_to_start;
+ enum snd_dice_rate_mode mode;
int err;
if (dice->substreams_counter == 0)
return -EIO;
- err = get_register_params(dice, &tx_params, &rx_params);
- if (err < 0)
- return err;
-
+ /* Check sampling transmission frequency. */
err = snd_dice_transaction_get_rate(dice, &curr_rate);
if (err < 0) {
dev_err(&dice->unit->device,
@@ -285,72 +405,36 @@ int snd_dice_stream_start_duplex(struct snd_dice *dice, unsigned int rate)
if (rate == 0)
rate = curr_rate;
if (rate != curr_rate)
- return -EINVAL;
+ goto restart;
- /* Judge to need to restart streams. */
- for (i = 0; i < MAX_STREAMS; i++) {
- if (i < tx_params.count) {
- if (amdtp_streaming_error(&dice->tx_stream[i]) ||
- !amdtp_stream_running(&dice->tx_stream[i]))
- break;
- }
- if (i < rx_params.count) {
- if (amdtp_streaming_error(&dice->rx_stream[i]) ||
- !amdtp_stream_running(&dice->rx_stream[i]))
- break;
- }
+ /* Check error of packet streaming. */
+ for (i = 0; i < MAX_STREAMS; ++i) {
+ if (amdtp_streaming_error(&dice->tx_stream[i]))
+ break;
+ if (amdtp_streaming_error(&dice->rx_stream[i]))
+ break;
}
- need_to_start = (i < MAX_STREAMS);
+ if (i < MAX_STREAMS)
+ goto restart;
- if (need_to_start) {
- /* Stop transmission. */
- snd_dice_transaction_clear_enable(dice);
- stop_streams(dice, AMDTP_IN_STREAM, &tx_params);
- stop_streams(dice, AMDTP_OUT_STREAM, &rx_params);
- release_resources(dice);
-
- err = ensure_phase_lock(dice);
- if (err < 0) {
- dev_err(&dice->unit->device,
- "fail to ensure phase lock\n");
- return err;
- }
-
- /* Start both streams. */
- err = start_streams(dice, AMDTP_IN_STREAM, rate, &tx_params);
- if (err < 0)
- goto error;
- err = start_streams(dice, AMDTP_OUT_STREAM, rate, &rx_params);
- if (err < 0)
- goto error;
-
- err = snd_dice_transaction_set_enable(dice);
- if (err < 0) {
- dev_err(&dice->unit->device,
- "fail to enable interface\n");
- goto error;
- }
-
- for (i = 0; i < MAX_STREAMS; i++) {
- if ((i < tx_params.count &&
- !amdtp_stream_wait_callback(&dice->tx_stream[i],
- CALLBACK_TIMEOUT)) ||
- (i < rx_params.count &&
- !amdtp_stream_wait_callback(&dice->rx_stream[i],
- CALLBACK_TIMEOUT))) {
- err = -ETIMEDOUT;
- goto error;
- }
- }
+ /* Check required streams are running or not. */
+ err = snd_dice_stream_get_rate_mode(dice, rate, &mode);
+ if (err < 0)
+ return err;
+ for (i = 0; i < MAX_STREAMS; ++i) {
+ if (dice->tx_pcm_chs[i][mode] > 0 &&
+ !amdtp_stream_running(&dice->tx_stream[i]))
+ break;
+ if (dice->rx_pcm_chs[i][mode] > 0 &&
+ !amdtp_stream_running(&dice->rx_stream[i]))
+ break;
}
+ if (i < MAX_STREAMS)
+ goto restart;
- return err;
-error:
- snd_dice_transaction_clear_enable(dice);
- stop_streams(dice, AMDTP_IN_STREAM, &tx_params);
- stop_streams(dice, AMDTP_OUT_STREAM, &rx_params);
- release_resources(dice);
- return err;
+ return 0;
+restart:
+ return start_duplex_streams(dice, rate);
}
/*
@@ -484,6 +568,69 @@ void snd_dice_stream_update_duplex(struct snd_dice *dice)
}
}
+int snd_dice_stream_detect_current_formats(struct snd_dice *dice)
+{
+ unsigned int rate;
+ enum snd_dice_rate_mode mode;
+ __be32 reg[2];
+ struct reg_params tx_params, rx_params;
+ int i;
+ int err;
+
+ /* If extended protocol is available, detect detail spec. */
+ err = snd_dice_detect_extension_formats(dice);
+ if (err >= 0)
+ return err;
+
+ /*
+ * Available stream format is restricted at current mode of sampling
+ * clock.
+ */
+ err = snd_dice_transaction_get_rate(dice, &rate);
+ if (err < 0)
+ return err;
+
+ err = snd_dice_stream_get_rate_mode(dice, rate, &mode);
+ if (err < 0)
+ return err;
+
+ /*
+ * Just after owning the unit (GLOBAL_OWNER), the unit can return
+ * invalid stream formats. Selecting clock parameters have an effect
+ * for the unit to refine it.
+ */
+ err = ensure_phase_lock(dice, rate);
+ if (err < 0)
+ return err;
+
+ err = get_register_params(dice, &tx_params, &rx_params);
+ if (err < 0)
+ return err;
+
+ for (i = 0; i < tx_params.count; ++i) {
+ err = snd_dice_transaction_read_tx(dice,
+ tx_params.size * i + TX_NUMBER_AUDIO,
+ reg, sizeof(reg));
+ if (err < 0)
+ return err;
+ dice->tx_pcm_chs[i][mode] = be32_to_cpu(reg[0]);
+ dice->tx_midi_ports[i] = max_t(unsigned int,
+ be32_to_cpu(reg[1]), dice->tx_midi_ports[i]);
+ }
+ for (i = 0; i < rx_params.count; ++i) {
+ err = snd_dice_transaction_read_rx(dice,
+ rx_params.size * i + RX_NUMBER_AUDIO,
+ reg, sizeof(reg));
+ if (err < 0)
+ return err;
+ dice->rx_pcm_chs[i][mode] = be32_to_cpu(reg[0]);
+ dice->rx_midi_ports[i] = max_t(unsigned int,
+ be32_to_cpu(reg[1]), dice->rx_midi_ports[i]);
+ }
+
+ return 0;
+}
+
static void dice_lock_changed(struct snd_dice *dice)
{
dice->dev_lock_changed = true;
diff --git a/sound/firewire/dice/dice-tcelectronic.c b/sound/firewire/dice/dice-tcelectronic.c
new file mode 100644
index 0000000..af8203b
--- /dev/null
+++ b/sound/firewire/dice/dice-tcelectronic.c
@@ -0,0 +1,100 @@
+// SPDX-License-Identifier: GPL-2.0
+/*
+ * dice-tc_electronic.c - a part of driver for DICE based devices
+ *
+ * Copyright (c) 2018 Takashi Sakamoto
+ */
+
+#include "dice.h"
+
+struct dice_tc_spec {
+ unsigned int tx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT];
+ unsigned int rx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT];
+ bool has_midi;
+};
+
+static const struct dice_tc_spec desktop_konnekt6 = {
+ .tx_pcm_chs = {{6, 6, 2}, {0, 0, 0} },
+ .rx_pcm_chs = {{6, 6, 4}, {0, 0, 0} },
+ .has_midi = false,
+};
+
+static const struct dice_tc_spec impact_twin = {
+ .tx_pcm_chs = {{14, 10, 6}, {0, 0, 0} },
+ .rx_pcm_chs = {{14, 10, 6}, {0, 0, 0} },
+ .has_midi = true,
+};
+
+static const struct dice_tc_spec konnekt_8 = {
+ .tx_pcm_chs = {{4, 4, 3}, {0, 0, 0} },
+ .rx_pcm_chs = {{4, 4, 3}, {0, 0, 0} },
+ .has_midi = true,
+};
+
+static const struct dice_tc_spec konnekt_24d = {
+ .tx_pcm_chs = {{16, 16, 6}, {0, 0, 0} },
+ .rx_pcm_chs = {{16, 16, 6}, {0, 0, 0} },
+ .has_midi = true,
+};
+
+static const struct dice_tc_spec konnekt_live = {
+ .tx_pcm_chs = {{16, 16, 16}, {0, 0, 0} },
+ .rx_pcm_chs = {{16, 16, 16}, {0, 0, 0} },
+ .has_midi = true,
+};
+
+static const struct dice_tc_spec studio_konnekt_48 = {
+ .tx_pcm_chs = {{16, 16, 16}, {16, 16, 0} },
+ .rx_pcm_chs = {{16, 16, 16}, {16, 16, 0} },
+ .has_midi = true,
+};
+
+int snd_dice_detect_tcelectronic_formats(struct snd_dice *dice)
+{
+ static const struct {
+ u32 model_id;
+ const struct dice_tc_spec *spec;
+ } *entry, entries[] = {
+ {0x00000020, &konnekt_24d},
+ {0x00000021, &konnekt_8},
+ {0x00000022, &studio_konnekt_48},
+ {0x00000023, &konnekt_live},
+ {0x00000024, &desktop_konnekt6},
+ {0x00000027, &impact_twin},
+ };
+ struct fw_csr_iterator it;
+ int key, val, model_id;
+ int i;
+
+ model_id = 0;
+ fw_csr_iterator_init(&it, dice->unit->directory);
+ while (fw_csr_iterator_next(&it, &key, &val)) {
+ if (key == CSR_MODEL) {
+ model_id = val;
+ break;
+ }
+ }
+
+ entry = NULL;
+ for (i = 0; i < ARRAY_SIZE(entries); ++i) {
+ entry = entries + i;
+ if (entry->model_id == model_id)
+ break;
+ }
+ if (!entry)
+ return -ENODEV;
+
+ memcpy(dice->tx_pcm_chs, entry->spec->tx_pcm_chs,
+ MAX_STREAMS * SND_DICE_RATE_MODE_COUNT * sizeof(unsigned int));
+ memcpy(dice->rx_pcm_chs, entry->spec->rx_pcm_chs,
+ MAX_STREAMS * SND_DICE_RATE_MODE_COUNT * sizeof(unsigned int));
+
+ for (i = 0; i < MAX_STREAMS; ++i) {
+ if (entry->spec->has_midi) {
+ dice->tx_midi_ports[i] = 1;
+ dice->rx_midi_ports[i] = 1;
+ }
+ }
+
+ return 0;
+}
diff --git a/sound/firewire/dice/dice-transaction.c b/sound/firewire/dice/dice-transaction.c
index 0f03503..b7e138b 100644
--- a/sound/firewire/dice/dice-transaction.c
+++ b/sound/firewire/dice/dice-transaction.c
@@ -265,7 +265,7 @@ int snd_dice_transaction_reinit(struct snd_dice *dice)
static int get_subaddrs(struct snd_dice *dice)
{
static const int min_values[10] = {
- 10, 0x64 / 4,
+ 10, 0x60 / 4,
10, 0x18 / 4,
10, 0x18 / 4,
0, 0,
@@ -301,33 +301,40 @@ static int get_subaddrs(struct snd_dice *dice)
}
}
- /*
- * Check that the implemented DICE driver specification major version
- * number matches.
- */
- err = snd_fw_transaction(dice->unit, TCODE_READ_QUADLET_REQUEST,
- DICE_PRIVATE_SPACE +
- be32_to_cpu(pointers[0]) * 4 + GLOBAL_VERSION,
- &version, sizeof(version), 0);
- if (err < 0)
- goto end;
+ if (be32_to_cpu(pointers[1]) > 0x18) {
+ /*
+ * Check that the implemented DICE driver specification major
+ * version number matches.
+ */
+ err = snd_fw_transaction(dice->unit, TCODE_READ_QUADLET_REQUEST,
+ DICE_PRIVATE_SPACE +
+ be32_to_cpu(pointers[0]) * 4 + GLOBAL_VERSION,
+ &version, sizeof(version), 0);
+ if (err < 0)
+ goto end;
- if ((version & cpu_to_be32(0xff000000)) != cpu_to_be32(0x01000000)) {
- dev_err(&dice->unit->device,
- "unknown DICE version: 0x%08x\n", be32_to_cpu(version));
- err = -ENODEV;
- goto end;
+ if ((version & cpu_to_be32(0xff000000)) !=
+ cpu_to_be32(0x01000000)) {
+ dev_err(&dice->unit->device,
+ "unknown DICE version: 0x%08x\n",
+ be32_to_cpu(version));
+ err = -ENODEV;
+ goto end;
+ }
+
+ /* Set up later. */
+ dice->clock_caps = 1;
}
dice->global_offset = be32_to_cpu(pointers[0]) * 4;
dice->tx_offset = be32_to_cpu(pointers[2]) * 4;
dice->rx_offset = be32_to_cpu(pointers[4]) * 4;
- dice->sync_offset = be32_to_cpu(pointers[6]) * 4;
- dice->rsrv_offset = be32_to_cpu(pointers[8]) * 4;
- /* Set up later. */
- if (be32_to_cpu(pointers[1]) * 4 >= GLOBAL_CLOCK_CAPABILITIES + 4)
- dice->clock_caps = 1;
+ /* Old firmware doesn't support these fields. */
+ if (pointers[7])
+ dice->sync_offset = be32_to_cpu(pointers[6]) * 4;
+ if (pointers[9])
+ dice->rsrv_offset = be32_to_cpu(pointers[8]) * 4;
end:
kfree(pointers);
return err;
diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c
index 96bb01b..40f7a32 100644
--- a/sound/firewire/dice/dice.c
+++ b/sound/firewire/dice/dice.c
@@ -15,40 +15,14 @@ MODULE_LICENSE("GPL v2");
#define OUI_LOUD 0x000ff2
#define OUI_FOCUSRITE 0x00130e
#define OUI_TCELECTRONIC 0x000166
+#define OUI_ALESIS 0x000595
+#define OUI_MAUDIO 0x000d6c
#define DICE_CATEGORY_ID 0x04
#define WEISS_CATEGORY_ID 0x00
#define LOUD_CATEGORY_ID 0x10
-/*
- * Some models support several isochronous channels, while these streams are not
- * always available. In this case, add the model name to this list.
- */
-static bool force_two_pcm_support(struct fw_unit *unit)
-{
- static const char *const models[] = {
- /* TC Electronic models. */
- "StudioKonnekt48",
- /* Focusrite models. */
- "SAFFIRE_PRO_40",
- "LIQUID_SAFFIRE_56",
- "SAFFIRE_PRO_40_1",
- };
- char model[32];
- unsigned int i;
- int err;
-
- err = fw_csr_string(unit->directory, CSR_MODEL, model, sizeof(model));
- if (err < 0)
- return false;
-
- for (i = 0; i < ARRAY_SIZE(models); i++) {
- if (strcmp(models[i], model) == 0)
- break;
- }
-
- return i < ARRAY_SIZE(models);
-}
+#define MODEL_ALESIS_IO_BOTH 0x000001
static int check_dice_category(struct fw_unit *unit)
{
@@ -75,11 +49,6 @@ static int check_dice_category(struct fw_unit *unit)
}
}
- if (vendor == OUI_FOCUSRITE || vendor == OUI_TCELECTRONIC) {
- if (force_two_pcm_support(unit))
- return 0;
- }
-
if (vendor == OUI_WEISS)
category = WEISS_CATEGORY_ID;
else if (vendor == OUI_LOUD)
@@ -186,9 +155,6 @@ static void do_registration(struct work_struct *work)
if (err < 0)
return;
- if (force_two_pcm_support(dice->unit))
- dice->force_two_pcms = true;
-
err = snd_dice_transaction_init(dice);
if (err < 0)
goto error;
@@ -199,6 +165,10 @@ static void do_registration(struct work_struct *work)
dice_card_strings(dice);
+ err = dice->detect_formats(dice);
+ if (err < 0)
+ goto error;
+
err = snd_dice_stream_init_duplex(dice);
if (err < 0)
goto error;
@@ -239,14 +209,17 @@ static void do_registration(struct work_struct *work)
"Sound card registration failed: %d\n", err);
}
-static int dice_probe(struct fw_unit *unit, const struct ieee1394_device_id *id)
+static int dice_probe(struct fw_unit *unit,
+ const struct ieee1394_device_id *entry)
{
struct snd_dice *dice;
int err;
- err = check_dice_category(unit);
- if (err < 0)
- return -ENODEV;
+ if (!entry->driver_data) {
+ err = check_dice_category(unit);
+ if (err < 0)
+ return -ENODEV;
+ }
/* Allocate this independent of sound card instance. */
dice = kzalloc(sizeof(struct snd_dice), GFP_KERNEL);
@@ -256,6 +229,13 @@ static int dice_probe(struct fw_unit *unit, const struct ieee1394_device_id *id)
dice->unit = fw_unit_get(unit);
dev_set_drvdata(&unit->device, dice);
+ if (!entry->driver_data) {
+ dice->detect_formats = snd_dice_stream_detect_current_formats;
+ } else {
+ dice->detect_formats =
+ (snd_dice_detect_formats_t)entry->driver_data;
+ }
+
spin_lock_init(&dice->lock);
mutex_init(&dice->mutex);
init_completion(&dice->clock_accepted);
@@ -313,17 +293,82 @@ static void dice_bus_reset(struct fw_unit *unit)
#define DICE_INTERFACE 0x000001
static const struct ieee1394_device_id dice_id_table[] = {
+ /* M-Audio Profire 2626 has a different value in version field. */
+ {
+ .match_flags = IEEE1394_MATCH_VENDOR_ID |
+ IEEE1394_MATCH_MODEL_ID,
+ .vendor_id = OUI_MAUDIO,
+ .model_id = 0x000010,
+ .driver_data = (kernel_ulong_t)snd_dice_detect_extension_formats,
+ },
+ /* M-Audio Profire 610 has a different value in version field. */
+ {
+ .match_flags = IEEE1394_MATCH_VENDOR_ID |
+ IEEE1394_MATCH_MODEL_ID,
+ .vendor_id = OUI_MAUDIO,
+ .model_id = 0x000011,
+ .driver_data = (kernel_ulong_t)snd_dice_detect_extension_formats,
+ },
+ /* TC Electronic Konnekt 24D. */
+ {
+ .match_flags = IEEE1394_MATCH_VENDOR_ID |
+ IEEE1394_MATCH_MODEL_ID,
+ .vendor_id = OUI_TCELECTRONIC,
+ .model_id = 0x000020,
+ .driver_data = (kernel_ulong_t)snd_dice_detect_tcelectronic_formats,
+ },
+ /* TC Electronic Konnekt 8. */
+ {
+ .match_flags = IEEE1394_MATCH_VENDOR_ID |
+ IEEE1394_MATCH_MODEL_ID,
+ .vendor_id = OUI_TCELECTRONIC,
+ .model_id = 0x000021,
+ .driver_data = (kernel_ulong_t)snd_dice_detect_tcelectronic_formats,
+ },
+ /* TC Electronic Studio Konnekt 48. */
+ {
+ .match_flags = IEEE1394_MATCH_VENDOR_ID |
+ IEEE1394_MATCH_MODEL_ID,
+ .vendor_id = OUI_TCELECTRONIC,
+ .model_id = 0x000022,
+ .driver_data = (kernel_ulong_t)snd_dice_detect_tcelectronic_formats,
+ },
+ /* TC Electronic Konnekt Live. */
+ {
+ .match_flags = IEEE1394_MATCH_VENDOR_ID |
+ IEEE1394_MATCH_MODEL_ID,
+ .vendor_id = OUI_TCELECTRONIC,
+ .model_id = 0x000023,
+ .driver_data = (kernel_ulong_t)snd_dice_detect_tcelectronic_formats,
+ },
+ /* TC Electronic Desktop Konnekt 6. */
+ {
+ .match_flags = IEEE1394_MATCH_VENDOR_ID |
+ IEEE1394_MATCH_MODEL_ID,
+ .vendor_id = OUI_TCELECTRONIC,
+ .model_id = 0x000024,
+ .driver_data = (kernel_ulong_t)snd_dice_detect_tcelectronic_formats,
+ },
+ /* TC Electronic Impact Twin. */
+ {
+ .match_flags = IEEE1394_MATCH_VENDOR_ID |
+ IEEE1394_MATCH_MODEL_ID,
+ .vendor_id = OUI_TCELECTRONIC,
+ .model_id = 0x000027,
+ .driver_data = (kernel_ulong_t)snd_dice_detect_tcelectronic_formats,
+ },
+ /* Alesis iO14/iO26. */
+ {
+ .match_flags = IEEE1394_MATCH_VENDOR_ID |
+ IEEE1394_MATCH_MODEL_ID,
+ .vendor_id = OUI_ALESIS,
+ .model_id = MODEL_ALESIS_IO_BOTH,
+ .driver_data = (kernel_ulong_t)snd_dice_detect_alesis_formats,
+ },
{
.match_flags = IEEE1394_MATCH_VERSION,
.version = DICE_INTERFACE,
},
- /* M-Audio Profire 610/2626 has a different value in version field. */
- {
- .match_flags = IEEE1394_MATCH_VENDOR_ID |
- IEEE1394_MATCH_SPECIFIER_ID,
- .vendor_id = 0x000d6c,
- .specifier_id = 0x000d6c,
- },
{ }
};
MODULE_DEVICE_TABLE(ieee1394, dice_id_table);
diff --git a/sound/firewire/dice/dice.h b/sound/firewire/dice/dice.h
index da00e75..505b79f 100644
--- a/sound/firewire/dice/dice.h
+++ b/sound/firewire/dice/dice.h
@@ -63,6 +63,16 @@
*/
#define MAX_STREAMS 2
+enum snd_dice_rate_mode {
+ SND_DICE_RATE_MODE_LOW = 0,
+ SND_DICE_RATE_MODE_MIDDLE,
+ SND_DICE_RATE_MODE_HIGH,
+ SND_DICE_RATE_MODE_COUNT,
+};
+
+struct snd_dice;
+typedef int (*snd_dice_detect_formats_t)(struct snd_dice *dice);
+
struct snd_dice {
struct snd_card *card;
struct fw_unit *unit;
@@ -80,6 +90,11 @@ struct snd_dice {
unsigned int rsrv_offset;
unsigned int clock_caps;
+ unsigned int tx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT];
+ unsigned int rx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT];
+ unsigned int tx_midi_ports[MAX_STREAMS];
+ unsigned int rx_midi_ports[MAX_STREAMS];
+ snd_dice_detect_formats_t detect_formats;
struct fw_address_handler notification_handler;
int owner_generation;
@@ -98,8 +113,6 @@ struct snd_dice {
bool global_enabled;
struct completion clock_accepted;
unsigned int substreams_counter;
-
- bool force_two_pcms;
};
enum snd_dice_addr_type {
@@ -190,11 +203,14 @@ void snd_dice_transaction_destroy(struct snd_dice *dice);
#define SND_DICE_RATES_COUNT 7
extern const unsigned int snd_dice_rates[SND_DICE_RATES_COUNT];
+int snd_dice_stream_get_rate_mode(struct snd_dice *dice, unsigned int rate,
+ enum snd_dice_rate_mode *mode);
int snd_dice_stream_start_duplex(struct snd_dice *dice, unsigned int rate);
void snd_dice_stream_stop_duplex(struct snd_dice *dice);
int snd_dice_stream_init_duplex(struct snd_dice *dice);
void snd_dice_stream_destroy_duplex(struct snd_dice *dice);
void snd_dice_stream_update_duplex(struct snd_dice *dice);
+int snd_dice_stream_detect_current_formats(struct snd_dice *dice);
int snd_dice_stream_lock_try(struct snd_dice *dice);
void snd_dice_stream_lock_release(struct snd_dice *dice);
@@ -207,4 +223,8 @@ void snd_dice_create_proc(struct snd_dice *dice);
int snd_dice_create_midi(struct snd_dice *dice);
+int snd_dice_detect_tcelectronic_formats(struct snd_dice *dice);
+int snd_dice_detect_alesis_formats(struct snd_dice *dice);
+int snd_dice_detect_extension_formats(struct snd_dice *dice);
+
#endif
diff --git a/sound/firewire/isight.c b/sound/firewire/isight.c
index 46092fa..3919e18 100644
--- a/sound/firewire/isight.c
+++ b/sound/firewire/isight.c
@@ -569,18 +569,20 @@ static int isight_create_mixer(struct isight *isight)
return err;
isight->gain_max = be32_to_cpu(value);
- isight->gain_tlv[0] = SNDRV_CTL_TLVT_DB_MINMAX;
- isight->gain_tlv[1] = 2 * sizeof(unsigned int);
+ isight->gain_tlv[SNDRV_CTL_TLVO_TYPE] = SNDRV_CTL_TLVT_DB_MINMAX;
+ isight->gain_tlv[SNDRV_CTL_TLVO_LEN] = 2 * sizeof(unsigned int);
err = reg_read(isight, REG_GAIN_DB_START, &value);
if (err < 0)
return err;
- isight->gain_tlv[2] = (s32)be32_to_cpu(value) * 100;
+ isight->gain_tlv[SNDRV_CTL_TLVO_DB_MINMAX_MIN] =
+ (s32)be32_to_cpu(value) * 100;
err = reg_read(isight, REG_GAIN_DB_END, &value);
if (err < 0)
return err;
- isight->gain_tlv[3] = (s32)be32_to_cpu(value) * 100;
+ isight->gain_tlv[SNDRV_CTL_TLVO_DB_MINMAX_MAX] =
+ (s32)be32_to_cpu(value) * 100;
ctl = snd_ctl_new1(&gain_control, isight);
if (ctl)
diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c
index 47a358f..419e285 100644
--- a/sound/hda/hdac_regmap.c
+++ b/sound/hda/hdac_regmap.c
@@ -65,10 +65,10 @@ static bool hda_writeable_reg(struct device *dev, unsigned int reg)
{
struct hdac_device *codec = dev_to_hdac_dev(dev);
unsigned int verb = get_verb(reg);
+ const unsigned int *v;
int i;
- for (i = 0; i < codec->vendor_verbs.used; i++) {
- unsigned int *v = snd_array_elem(&codec->vendor_verbs, i);
+ snd_array_for_each(&codec->vendor_verbs, i, v) {
if (verb == *v)
return true;
}
diff --git a/sound/isa/cmi8328.c b/sound/isa/cmi8328.c
index d09e456..de6ef1b 100644
--- a/sound/isa/cmi8328.c
+++ b/sound/isa/cmi8328.c
@@ -192,7 +192,7 @@ static int snd_cmi8328_mixer(struct snd_wss *chip)
}
/* find index of an item in "-1"-ended array */
-int array_find(int array[], int item)
+static int array_find(int array[], int item)
{
int i;
@@ -203,7 +203,7 @@ int array_find(int array[], int item)
return -1;
}
/* the same for long */
-int array_find_l(long array[], long item)
+static int array_find_l(long array[], long item)
{
int i;
diff --git a/sound/isa/sc6000.c b/sound/isa/sc6000.c
index c09d9b9..a985e91 100644
--- a/sound/isa/sc6000.c
+++ b/sound/isa/sc6000.c
@@ -592,7 +592,7 @@ static int snd_sc6000_probe(struct device *devptr, unsigned int dev)
*vport = devm_ioport_map(devptr, port[dev], 0x10);
if (*vport == NULL) {
snd_printk(KERN_ERR PFX
- "I/O port cannot be iomaped.\n");
+ "I/O port cannot be iomapped.\n");
err = -EBUSY;
goto err_unmap1;
}
@@ -607,7 +607,7 @@ static int snd_sc6000_probe(struct device *devptr, unsigned int dev)
vmss_port = devm_ioport_map(devptr, mss_port[dev], 4);
if (!vmss_port) {
snd_printk(KERN_ERR PFX
- "MSS port I/O cannot be iomaped.\n");
+ "MSS port I/O cannot be iomapped.\n");
err = -EBUSY;
goto err_unmap2;
}
diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c
index 0bf2c04..d9c54c0 100644
--- a/sound/pci/ad1889.c
+++ b/sound/pci/ad1889.c
@@ -258,7 +258,7 @@ snd_ad1889_ac97_ready(struct snd_ad1889 *chip)
while (!(ad1889_readw(chip, AD_AC97_ACIC) & AD_AC97_ACIC_ACRDY)
&& --retry)
- mdelay(1);
+ usleep_range(1000, 2000);
if (!retry) {
dev_err(chip->card->dev, "[%s] Link is not ready.\n",
__func__);
@@ -872,7 +872,7 @@ snd_ad1889_init(struct snd_ad1889 *chip)
ad1889_writew(chip, AD_DS_CCS, AD_DS_CCS_CLKEN); /* turn on clock */
ad1889_readw(chip, AD_DS_CCS); /* flush posted write */
- mdelay(10);
+ usleep_range(10000, 11000);
/* enable Master and Target abort interrupts */
ad1889_writel(chip, AD_DMA_DISR, AD_DMA_DISR_PMAE | AD_DMA_DISR_PTAE);
diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c
index 26a6578..452cc79 100644
--- a/sound/pci/cmipci.c
+++ b/sound/pci/cmipci.c
@@ -1139,7 +1139,7 @@ static int save_mixer_state(struct cmipci *cm)
struct snd_ctl_elem_value *val;
unsigned int i;
- val = kmalloc(sizeof(*val), GFP_ATOMIC);
+ val = kmalloc(sizeof(*val), GFP_KERNEL);
if (!val)
return -ENOMEM;
for (i = 0; i < CM_SAVED_MIXERS; i++) {
diff --git a/sound/pci/cs46xx/cs46xx.c b/sound/pci/cs46xx/cs46xx.c
index 655fbea..4910d3f 100644
--- a/sound/pci/cs46xx/cs46xx.c
+++ b/sound/pci/cs46xx/cs46xx.c
@@ -58,7 +58,7 @@ MODULE_PARM_DESC(id, "ID string for the CS46xx soundcard.");
module_param_array(enable, bool, NULL, 0444);
MODULE_PARM_DESC(enable, "Enable CS46xx soundcard.");
module_param_array(external_amp, bool, NULL, 0444);
-MODULE_PARM_DESC(external_amp, "Force to enable external amplifer.");
+MODULE_PARM_DESC(external_amp, "Force to enable external amplifier.");
module_param_array(thinkpad, bool, NULL, 0444);
MODULE_PARM_DESC(thinkpad, "Force to enable Thinkpad's CLKRUN control.");
module_param_array(mmap_valid, bool, NULL, 0444);
diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c
index a2b56b1..608ff48 100644
--- a/sound/pci/emu10k1/emufx.c
+++ b/sound/pci/emu10k1/emufx.c
@@ -421,14 +421,10 @@ int snd_emu10k1_fx8010_register_irq_handler(struct snd_emu10k1 *emu,
snd_fx8010_irq_handler_t *handler,
unsigned char gpr_running,
void *private_data,
- struct snd_emu10k1_fx8010_irq **r_irq)
+ struct snd_emu10k1_fx8010_irq *irq)
{
- struct snd_emu10k1_fx8010_irq *irq;
unsigned long flags;
- irq = kmalloc(sizeof(*irq), GFP_ATOMIC);
- if (irq == NULL)
- return -ENOMEM;
irq->handler = handler;
irq->gpr_running = gpr_running;
irq->private_data = private_data;
@@ -443,8 +439,6 @@ int snd_emu10k1_fx8010_register_irq_handler(struct snd_emu10k1 *emu,
emu->fx8010.irq_handlers = irq;
}
spin_unlock_irqrestore(&emu->fx8010.irq_lock, flags);
- if (r_irq)
- *r_irq = irq;
return 0;
}
@@ -468,7 +462,6 @@ int snd_emu10k1_fx8010_unregister_irq_handler(struct snd_emu10k1 *emu,
tmp->next = tmp->next->next;
}
spin_unlock_irqrestore(&emu->fx8010.irq_lock, flags);
- kfree(irq);
return 0;
}
diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c
index cefe613..d39458a 100644
--- a/sound/pci/emu10k1/emupcm.c
+++ b/sound/pci/emu10k1/emupcm.c
@@ -1724,7 +1724,7 @@ static int snd_emu10k1_fx8010_playback_trigger(struct snd_pcm_substream *substre
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
case SNDRV_PCM_TRIGGER_SUSPEND:
- snd_emu10k1_fx8010_unregister_irq_handler(emu, pcm->irq); pcm->irq = NULL;
+ snd_emu10k1_fx8010_unregister_irq_handler(emu, &pcm->irq);
snd_emu10k1_ptr_write(emu, emu->gpr_base + pcm->gpr_trigger, 0, 0);
pcm->tram_pos = INITIAL_TRAM_POS(pcm->buffer_size);
pcm->tram_shift = 0;
diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig
index f7a492c..4235907 100644
--- a/sound/pci/hda/Kconfig
+++ b/sound/pci/hda/Kconfig
@@ -127,11 +127,15 @@
config SND_HDA_CODEC_HDMI
tristate "Build HDMI/DisplayPort HD-audio codec support"
+ select SND_DYNAMIC_MINORS
help
Say Y or M here to include HDMI and DisplayPort HD-audio codec
support in snd-hda-intel driver. This includes all AMD/ATI,
Intel and Nvidia HDMI/DisplayPort codecs.
+ Note that this option mandatorily enables CONFIG_SND_DYNAMIC_MINORS
+ to assure the multiple streams for DP-MST support.
+
comment "Set to Y if you want auto-loading the codec driver"
depends on SND_HDA=y && SND_HDA_CODEC_HDMI=m
diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c
index d3ea731..b9a6b66 100644
--- a/sound/pci/hda/hda_auto_parser.c
+++ b/sound/pci/hda/hda_auto_parser.c
@@ -793,11 +793,11 @@ EXPORT_SYMBOL_GPL(snd_hda_add_verbs);
*/
void snd_hda_apply_verbs(struct hda_codec *codec)
{
+ const struct hda_verb **v;
int i;
- for (i = 0; i < codec->verbs.used; i++) {
- struct hda_verb **v = snd_array_elem(&codec->verbs, i);
+
+ snd_array_for_each(&codec->verbs, i, v)
snd_hda_sequence_write(codec, *v);
- }
}
EXPORT_SYMBOL_GPL(snd_hda_apply_verbs);
@@ -890,10 +890,10 @@ EXPORT_SYMBOL_GPL(snd_hda_apply_fixup);
static bool pin_config_match(struct hda_codec *codec,
const struct hda_pintbl *pins)
{
+ const struct hda_pincfg *pin;
int i;
- for (i = 0; i < codec->init_pins.used; i++) {
- struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i);
+ snd_array_for_each(&codec->init_pins, i, pin) {
hda_nid_t nid = pin->nid;
u32 cfg = pin->cfg;
const struct hda_pintbl *t_pins;
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 5bc3a74..08151f3 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -481,9 +481,10 @@ static struct hda_pincfg *look_up_pincfg(struct hda_codec *codec,
struct snd_array *array,
hda_nid_t nid)
{
+ struct hda_pincfg *pin;
int i;
- for (i = 0; i < array->used; i++) {
- struct hda_pincfg *pin = snd_array_elem(array, i);
+
+ snd_array_for_each(array, i, pin) {
if (pin->nid == nid)
return pin;
}
@@ -618,14 +619,15 @@ EXPORT_SYMBOL_GPL(snd_hda_codec_get_pin_target);
*/
void snd_hda_shutup_pins(struct hda_codec *codec)
{
+ const struct hda_pincfg *pin;
int i;
+
/* don't shut up pins when unloading the driver; otherwise it breaks
* the default pin setup at the next load of the driver
*/
if (codec->bus->shutdown)
return;
- for (i = 0; i < codec->init_pins.used; i++) {
- struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i);
+ snd_array_for_each(&codec->init_pins, i, pin) {
/* use read here for syncing after issuing each verb */
snd_hda_codec_read(codec, pin->nid, 0,
AC_VERB_SET_PIN_WIDGET_CONTROL, 0);
@@ -638,13 +640,14 @@ EXPORT_SYMBOL_GPL(snd_hda_shutup_pins);
/* Restore the pin controls cleared previously via snd_hda_shutup_pins() */
static void restore_shutup_pins(struct hda_codec *codec)
{
+ const struct hda_pincfg *pin;
int i;
+
if (!codec->pins_shutup)
return;
if (codec->bus->shutdown)
return;
- for (i = 0; i < codec->init_pins.used; i++) {
- struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i);
+ snd_array_for_each(&codec->init_pins, i, pin) {
snd_hda_codec_write(codec, pin->nid, 0,
AC_VERB_SET_PIN_WIDGET_CONTROL,
pin->ctrl);
@@ -697,8 +700,7 @@ get_hda_cvt_setup(struct hda_codec *codec, hda_nid_t nid)
struct hda_cvt_setup *p;
int i;
- for (i = 0; i < codec->cvt_setups.used; i++) {
- p = snd_array_elem(&codec->cvt_setups, i);
+ snd_array_for_each(&codec->cvt_setups, i, p) {
if (p->nid == nid)
return p;
}
@@ -1076,8 +1078,7 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid,
/* make other inactive cvts with the same stream-tag dirty */
type = get_wcaps_type(get_wcaps(codec, nid));
list_for_each_codec(c, codec->bus) {
- for (i = 0; i < c->cvt_setups.used; i++) {
- p = snd_array_elem(&c->cvt_setups, i);
+ snd_array_for_each(&c->cvt_setups, i, p) {
if (!p->active && p->stream_tag == stream_tag &&
get_wcaps_type(get_wcaps(c, p->nid)) == type)
p->dirty = 1;
@@ -1140,12 +1141,11 @@ static void really_cleanup_stream(struct hda_codec *codec,
static void purify_inactive_streams(struct hda_codec *codec)
{
struct hda_codec *c;
+ struct hda_cvt_setup *p;
int i;
list_for_each_codec(c, codec->bus) {
- for (i = 0; i < c->cvt_setups.used; i++) {
- struct hda_cvt_setup *p;
- p = snd_array_elem(&c->cvt_setups, i);
+ snd_array_for_each(&c->cvt_setups, i, p) {
if (p->dirty)
really_cleanup_stream(c, p);
}
@@ -1156,10 +1156,10 @@ static void purify_inactive_streams(struct hda_codec *codec)
/* clean up all streams; called from suspend */
static void hda_cleanup_all_streams(struct hda_codec *codec)
{
+ struct hda_cvt_setup *p;
int i;
- for (i = 0; i < codec->cvt_setups.used; i++) {
- struct hda_cvt_setup *p = snd_array_elem(&codec->cvt_setups, i);
+ snd_array_for_each(&codec->cvt_setups, i, p) {
if (p->stream_tag)
really_cleanup_stream(codec, p);
}
@@ -1493,10 +1493,10 @@ static void get_ctl_amp_tlv(struct snd_kcontrol *kcontrol, unsigned int *tlv)
val1 = ((int)val1) * ((int)val2);
if (min_mute || (caps & AC_AMPCAP_MIN_MUTE))
val2 |= TLV_DB_SCALE_MUTE;
- tlv[0] = SNDRV_CTL_TLVT_DB_SCALE;
- tlv[1] = 2 * sizeof(unsigned int);
- tlv[2] = val1;
- tlv[3] = val2;
+ tlv[SNDRV_CTL_TLVO_TYPE] = SNDRV_CTL_TLVT_DB_SCALE;
+ tlv[SNDRV_CTL_TLVO_LEN] = 2 * sizeof(unsigned int);
+ tlv[SNDRV_CTL_TLVO_DB_SCALE_MIN] = val1;
+ tlv[SNDRV_CTL_TLVO_DB_SCALE_MUTE_AND_STEP] = val2;
}
/**
@@ -1544,10 +1544,10 @@ void snd_hda_set_vmaster_tlv(struct hda_codec *codec, hda_nid_t nid, int dir,
nums = (caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT;
step = (caps & AC_AMPCAP_STEP_SIZE) >> AC_AMPCAP_STEP_SIZE_SHIFT;
step = (step + 1) * 25;
- tlv[0] = SNDRV_CTL_TLVT_DB_SCALE;
- tlv[1] = 2 * sizeof(unsigned int);
- tlv[2] = -nums * step;
- tlv[3] = step;
+ tlv[SNDRV_CTL_TLVO_TYPE] = SNDRV_CTL_TLVT_DB_SCALE;
+ tlv[SNDRV_CTL_TLVO_LEN] = 2 * sizeof(unsigned int);
+ tlv[SNDRV_CTL_TLVO_DB_SCALE_MIN] = -nums * step;
+ tlv[SNDRV_CTL_TLVO_DB_SCALE_MUTE_AND_STEP] = step;
}
EXPORT_SYMBOL_GPL(snd_hda_set_vmaster_tlv);
@@ -1845,10 +1845,10 @@ static int init_slave_0dB(struct snd_kcontrol *slave,
} else if (kctl->vd[0].access & SNDRV_CTL_ELEM_ACCESS_TLV_READ)
tlv = kctl->tlv.p;
- if (!tlv || tlv[0] != SNDRV_CTL_TLVT_DB_SCALE)
+ if (!tlv || tlv[SNDRV_CTL_TLVO_TYPE] != SNDRV_CTL_TLVT_DB_SCALE)
return 0;
- step = tlv[3];
+ step = tlv[SNDRV_CTL_TLVO_DB_SCALE_MUTE_AND_STEP];
step &= ~TLV_DB_SCALE_MUTE;
if (!step)
return 0;
@@ -1860,7 +1860,7 @@ static int init_slave_0dB(struct snd_kcontrol *slave,
}
arg->step = step;
- val = -tlv[2] / step;
+ val = -tlv[SNDRV_CTL_TLVO_DB_SCALE_MIN] / step;
if (val > 0) {
put_kctl_with_value(slave, val);
return val;
@@ -2175,6 +2175,8 @@ static int snd_hda_spdif_default_get(struct snd_kcontrol *kcontrol,
int idx = kcontrol->private_value;
struct hda_spdif_out *spdif;
+ if (WARN_ON(codec->spdif_out.used <= idx))
+ return -EINVAL;
mutex_lock(&codec->spdif_mutex);
spdif = snd_array_elem(&codec->spdif_out, idx);
ucontrol->value.iec958.status[0] = spdif->status & 0xff;
@@ -2282,6 +2284,8 @@ static int snd_hda_spdif_default_put(struct snd_kcontrol *kcontrol,
unsigned short val;
int change;
+ if (WARN_ON(codec->spdif_out.used <= idx))
+ return -EINVAL;
mutex_lock(&codec->spdif_mutex);
spdif = snd_array_elem(&codec->spdif_out, idx);
nid = spdif->nid;
@@ -2308,6 +2312,8 @@ static int snd_hda_spdif_out_switch_get(struct snd_kcontrol *kcontrol,
int idx = kcontrol->private_value;
struct hda_spdif_out *spdif;
+ if (WARN_ON(codec->spdif_out.used <= idx))
+ return -EINVAL;
mutex_lock(&codec->spdif_mutex);
spdif = snd_array_elem(&codec->spdif_out, idx);
ucontrol->value.integer.value[0] = spdif->ctls & AC_DIG1_ENABLE;
@@ -2336,6 +2342,8 @@ static int snd_hda_spdif_out_switch_put(struct snd_kcontrol *kcontrol,
unsigned short val;
int change;
+ if (WARN_ON(codec->spdif_out.used <= idx))
+ return -EINVAL;
mutex_lock(&codec->spdif_mutex);
spdif = snd_array_elem(&codec->spdif_out, idx);
nid = spdif->nid;
@@ -2461,10 +2469,10 @@ EXPORT_SYMBOL_GPL(snd_hda_create_dig_out_ctls);
struct hda_spdif_out *snd_hda_spdif_out_of_nid(struct hda_codec *codec,
hda_nid_t nid)
{
+ struct hda_spdif_out *spdif;
int i;
- for (i = 0; i < codec->spdif_out.used; i++) {
- struct hda_spdif_out *spdif =
- snd_array_elem(&codec->spdif_out, i);
+
+ snd_array_for_each(&codec->spdif_out, i, spdif) {
if (spdif->nid == nid)
return spdif;
}
@@ -2483,6 +2491,8 @@ void snd_hda_spdif_ctls_unassign(struct hda_codec *codec, int idx)
{
struct hda_spdif_out *spdif;
+ if (WARN_ON(codec->spdif_out.used <= idx))
+ return;
mutex_lock(&codec->spdif_mutex);
spdif = snd_array_elem(&codec->spdif_out, idx);
spdif->nid = (u16)-1;
@@ -2503,6 +2513,8 @@ void snd_hda_spdif_ctls_assign(struct hda_codec *codec, int idx, hda_nid_t nid)
struct hda_spdif_out *spdif;
unsigned short val;
+ if (WARN_ON(codec->spdif_out.used <= idx))
+ return;
mutex_lock(&codec->spdif_mutex);
spdif = snd_array_elem(&codec->spdif_out, idx);
if (spdif->nid != nid) {
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 5cc6509..db773e2 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -264,10 +264,10 @@ static struct nid_path *get_nid_path(struct hda_codec *codec,
int anchor_nid)
{
struct hda_gen_spec *spec = codec->spec;
+ struct nid_path *path;
int i;
- for (i = 0; i < spec->paths.used; i++) {
- struct nid_path *path = snd_array_elem(&spec->paths, i);
+ snd_array_for_each(&spec->paths, i, path) {
if (path->depth <= 0)
continue;
if ((!from_nid || path->path[0] == from_nid) &&
@@ -325,10 +325,10 @@ EXPORT_SYMBOL_GPL(snd_hda_get_path_from_idx);
static bool is_dac_already_used(struct hda_codec *codec, hda_nid_t nid)
{
struct hda_gen_spec *spec = codec->spec;
+ const struct nid_path *path;
int i;
- for (i = 0; i < spec->paths.used; i++) {
- struct nid_path *path = snd_array_elem(&spec->paths, i);
+ snd_array_for_each(&spec->paths, i, path) {
if (path->path[0] == nid)
return true;
}
@@ -351,11 +351,11 @@ static bool is_reachable_path(struct hda_codec *codec,
static bool is_ctl_used(struct hda_codec *codec, unsigned int val, int type)
{
struct hda_gen_spec *spec = codec->spec;
+ const struct nid_path *path;
int i;
val &= AMP_VAL_COMPARE_MASK;
- for (i = 0; i < spec->paths.used; i++) {
- struct nid_path *path = snd_array_elem(&spec->paths, i);
+ snd_array_for_each(&spec->paths, i, path) {
if ((path->ctls[type] & AMP_VAL_COMPARE_MASK) == val)
return true;
}
@@ -638,13 +638,13 @@ static bool is_active_nid(struct hda_codec *codec, hda_nid_t nid,
{
struct hda_gen_spec *spec = codec->spec;
int type = get_wcaps_type(get_wcaps(codec, nid));
+ const struct nid_path *path;
int i, n;
if (nid == codec->core.afg)
return true;
- for (n = 0; n < spec->paths.used; n++) {
- struct nid_path *path = snd_array_elem(&spec->paths, n);
+ snd_array_for_each(&spec->paths, n, path) {
if (!path->active)
continue;
if (codec->power_save_node) {
@@ -2065,7 +2065,7 @@ static int parse_output_paths(struct hda_codec *codec)
snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid,
HDA_OUTPUT, spec->vmaster_tlv);
if (spec->dac_min_mute)
- spec->vmaster_tlv[3] |= TLV_DB_SCALE_MUTE;
+ spec->vmaster_tlv[SNDRV_CTL_TLVO_DB_SCALE_MUTE_AND_STEP] |= TLV_DB_SCALE_MUTE;
}
}
@@ -2696,10 +2696,10 @@ static const struct snd_kcontrol_new out_jack_mode_enum = {
static bool find_kctl_name(struct hda_codec *codec, const char *name, int idx)
{
struct hda_gen_spec *spec = codec->spec;
+ const struct snd_kcontrol_new *kctl;
int i;
- for (i = 0; i < spec->kctls.used; i++) {
- struct snd_kcontrol_new *kctl = snd_array_elem(&spec->kctls, i);
+ snd_array_for_each(&spec->kctls, i, kctl) {
if (!strcmp(kctl->name, name) && kctl->index == idx)
return true;
}
@@ -4021,8 +4021,7 @@ static hda_nid_t set_path_power(struct hda_codec *codec, hda_nid_t nid,
struct nid_path *path;
int n;
- for (n = 0; n < spec->paths.used; n++) {
- path = snd_array_elem(&spec->paths, n);
+ snd_array_for_each(&spec->paths, n, path) {
if (!path->depth)
continue;
if (path->path[0] == nid ||
@@ -5831,10 +5830,10 @@ static void init_digital(struct hda_codec *codec)
*/
static void clear_unsol_on_unused_pins(struct hda_codec *codec)
{
+ const struct hda_pincfg *pin;
int i;
- for (i = 0; i < codec->init_pins.used; i++) {
- struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i);
+ snd_array_for_each(&codec->init_pins, i, pin) {
hda_nid_t nid = pin->nid;
if (is_jack_detectable(codec, nid) &&
!snd_hda_jack_tbl_get(codec, nid))
diff --git a/sound/pci/hda/hda_sysfs.c b/sound/pci/hda/hda_sysfs.c
index 9b7efec..6ec79c5 100644
--- a/sound/pci/hda/hda_sysfs.c
+++ b/sound/pci/hda/hda_sysfs.c
@@ -80,10 +80,10 @@ static ssize_t pin_configs_show(struct hda_codec *codec,
struct snd_array *list,
char *buf)
{
+ const struct hda_pincfg *pin;
int i, len = 0;
mutex_lock(&codec->user_mutex);
- for (i = 0; i < list->used; i++) {
- struct hda_pincfg *pin = snd_array_elem(list, i);
+ snd_array_for_each(list, i, pin) {
len += sprintf(buf + len, "0x%02x 0x%08x\n",
pin->nid, pin->cfg);
}
@@ -217,10 +217,10 @@ static ssize_t init_verbs_show(struct device *dev,
char *buf)
{
struct hda_codec *codec = dev_get_drvdata(dev);
+ const struct hda_verb *v;
int i, len = 0;
mutex_lock(&codec->user_mutex);
- for (i = 0; i < codec->init_verbs.used; i++) {
- struct hda_verb *v = snd_array_elem(&codec->init_verbs, i);
+ snd_array_for_each(&codec->init_verbs, i, v) {
len += snprintf(buf + len, PAGE_SIZE - len,
"0x%02x 0x%03x 0x%04x\n",
v->nid, v->verb, v->param);
@@ -267,10 +267,10 @@ static ssize_t hints_show(struct device *dev,
char *buf)
{
struct hda_codec *codec = dev_get_drvdata(dev);
+ const struct hda_hint *hint;
int i, len = 0;
mutex_lock(&codec->user_mutex);
- for (i = 0; i < codec->hints.used; i++) {
- struct hda_hint *hint = snd_array_elem(&codec->hints, i);
+ snd_array_for_each(&codec->hints, i, hint) {
len += snprintf(buf + len, PAGE_SIZE - len,
"%s = %s\n", hint->key, hint->val);
}
@@ -280,10 +280,10 @@ static ssize_t hints_show(struct device *dev,
static struct hda_hint *get_hint(struct hda_codec *codec, const char *key)
{
+ struct hda_hint *hint;
int i;
- for (i = 0; i < codec->hints.used; i++) {
- struct hda_hint *hint = snd_array_elem(&codec->hints, i);
+ snd_array_for_each(&codec->hints, i, hint) {
if (!strcmp(hint->key, key))
return hint;
}
@@ -783,13 +783,13 @@ void snd_hda_sysfs_init(struct hda_codec *codec)
void snd_hda_sysfs_clear(struct hda_codec *codec)
{
#ifdef CONFIG_SND_HDA_RECONFIG
+ struct hda_hint *hint;
int i;
/* clear init verbs */
snd_array_free(&codec->init_verbs);
/* clear hints */
- for (i = 0; i < codec->hints.used; i++) {
- struct hda_hint *hint = snd_array_elem(&codec->hints, i);
+ snd_array_for_each(&codec->hints, i, hint) {
kfree(hint->key); /* we don't need to free hint->val */
}
snd_array_free(&codec->hints);
diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 768ea86..1cc4f51 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -28,6 +28,9 @@
#include <linux/module.h>
#include <linux/firmware.h>
#include <linux/kernel.h>
+#include <linux/types.h>
+#include <linux/io.h>
+#include <linux/pci.h>
#include <sound/core.h>
#include "hda_codec.h"
#include "hda_local.h"
@@ -39,9 +42,15 @@
/* Enable this to see controls for tuning purpose. */
/*#define ENABLE_TUNING_CONTROLS*/
+#ifdef ENABLE_TUNING_CONTROLS
+#include <sound/tlv.h>
+#endif
+
#define FLOAT_ZERO 0x00000000
#define FLOAT_ONE 0x3f800000
#define FLOAT_TWO 0x40000000
+#define FLOAT_THREE 0x40400000
+#define FLOAT_EIGHT 0x41000000
#define FLOAT_MINUS_5 0xc0a00000
#define UNSOL_TAG_DSP 0x16
@@ -72,16 +81,22 @@
#define SCP_GET 1
#define EFX_FILE "ctefx.bin"
+#define SBZ_EFX_FILE "ctefx-sbz.bin"
+#define R3DI_EFX_FILE "ctefx-r3di.bin"
#ifdef CONFIG_SND_HDA_CODEC_CA0132_DSP
MODULE_FIRMWARE(EFX_FILE);
+MODULE_FIRMWARE(SBZ_EFX_FILE);
+MODULE_FIRMWARE(R3DI_EFX_FILE);
#endif
-static char *dirstr[2] = { "Playback", "Capture" };
+static const char *dirstr[2] = { "Playback", "Capture" };
+#define NUM_OF_OUTPUTS 3
enum {
SPEAKER_OUT,
- HEADPHONE_OUT
+ HEADPHONE_OUT,
+ SURROUND_OUT
};
enum {
@@ -89,6 +104,15 @@ enum {
LINE_MIC_IN
};
+/* Strings for Input Source Enum Control */
+static const char *in_src_str[3] = {"Rear Mic", "Line", "Front Mic" };
+#define IN_SRC_NUM_OF_INPUTS 3
+enum {
+ REAR_MIC,
+ REAR_LINE_IN,
+ FRONT_MIC,
+};
+
enum {
#define VNODE_START_NID 0x80
VNID_SPK = VNODE_START_NID, /* Speaker vnid */
@@ -122,13 +146,28 @@ enum {
VOICEFX = IN_EFFECT_END_NID,
PLAY_ENHANCEMENT,
CRYSTAL_VOICE,
- EFFECT_END_NID
+ EFFECT_END_NID,
+ OUTPUT_SOURCE_ENUM,
+ INPUT_SOURCE_ENUM,
+ XBASS_XOVER,
+ EQ_PRESET_ENUM,
+ SMART_VOLUME_ENUM,
+ MIC_BOOST_ENUM
#define EFFECTS_COUNT (EFFECT_END_NID - EFFECT_START_NID)
};
/* Effects values size*/
#define EFFECT_VALS_MAX_COUNT 12
+/*
+ * Default values for the effect slider controls, they are in order of their
+ * effect NID's. Surround, Crystalizer, Dialog Plus, Smart Volume, and then
+ * X-bass.
+ */
+static const unsigned int effect_slider_defaults[] = {67, 65, 50, 74, 50};
+/* Amount of effect level sliders for ca0132_alt controls. */
+#define EFFECT_LEVEL_SLIDERS 5
+
/* Latency introduced by DSP blocks in milliseconds. */
#define DSP_CAPTURE_INIT_LATENCY 0
#define DSP_CRYSTAL_VOICE_LATENCY 124
@@ -472,6 +511,161 @@ static struct ct_voicefx_preset ca0132_voicefx_presets[] = {
}
};
+/* ca0132 EQ presets, taken from Windows Sound Blaster Z Driver */
+
+#define EQ_PRESET_MAX_PARAM_COUNT 11
+
+struct ct_eq {
+ char *name;
+ hda_nid_t nid;
+ int mid;
+ int reqs[EQ_PRESET_MAX_PARAM_COUNT]; /*effect module request*/
+};
+
+struct ct_eq_preset {
+ char *name; /*preset name*/
+ unsigned int vals[EQ_PRESET_MAX_PARAM_COUNT];
+};
+
+static struct ct_eq ca0132_alt_eq_enum = {
+ .name = "FX: Equalizer Preset Switch",
+ .nid = EQ_PRESET_ENUM,
+ .mid = 0x96,
+ .reqs = {10, 11, 12, 13, 14, 15, 16, 17, 18, 19, 20}
+};
+
+
+static struct ct_eq_preset ca0132_alt_eq_presets[] = {
+ { .name = "Flat",
+ .vals = { 0x00000000, 0x00000000, 0x00000000,
+ 0x00000000, 0x00000000, 0x00000000,
+ 0x00000000, 0x00000000, 0x00000000,
+ 0x00000000, 0x00000000 }
+ },
+ { .name = "Acoustic",
+ .vals = { 0x00000000, 0x00000000, 0x3F8CCCCD,
+ 0x40000000, 0x00000000, 0x00000000,
+ 0x00000000, 0x00000000, 0x40000000,
+ 0x40000000, 0x40000000 }
+ },
+ { .name = "Classical",
+ .vals = { 0x00000000, 0x00000000, 0x40C00000,
+ 0x40C00000, 0x40466666, 0x00000000,
+ 0x00000000, 0x00000000, 0x00000000,
+ 0x40466666, 0x40466666 }
+ },
+ { .name = "Country",
+ .vals = { 0x00000000, 0xBF99999A, 0x00000000,
+ 0x3FA66666, 0x3FA66666, 0x3F8CCCCD,
+ 0x00000000, 0x00000000, 0x40000000,
+ 0x40466666, 0x40800000 }
+ },
+ { .name = "Dance",
+ .vals = { 0x00000000, 0xBF99999A, 0x40000000,
+ 0x40466666, 0x40866666, 0xBF99999A,
+ 0xBF99999A, 0x00000000, 0x00000000,
+ 0x40800000, 0x40800000 }
+ },
+ { .name = "Jazz",
+ .vals = { 0x00000000, 0x00000000, 0x00000000,
+ 0x3F8CCCCD, 0x40800000, 0x40800000,
+ 0x40800000, 0x00000000, 0x3F8CCCCD,
+ 0x40466666, 0x40466666 }
+ },
+ { .name = "New Age",
+ .vals = { 0x00000000, 0x00000000, 0x40000000,
+ 0x40000000, 0x00000000, 0x00000000,
+ 0x00000000, 0x3F8CCCCD, 0x40000000,
+ 0x40000000, 0x40000000 }
+ },
+ { .name = "Pop",
+ .vals = { 0x00000000, 0xBFCCCCCD, 0x00000000,
+ 0x40000000, 0x40000000, 0x00000000,
+ 0xBF99999A, 0xBF99999A, 0x00000000,
+ 0x40466666, 0x40C00000 }
+ },
+ { .name = "Rock",
+ .vals = { 0x00000000, 0xBF99999A, 0xBF99999A,
+ 0x3F8CCCCD, 0x40000000, 0xBF99999A,
+ 0xBF99999A, 0x00000000, 0x00000000,
+ 0x40800000, 0x40800000 }
+ },
+ { .name = "Vocal",
+ .vals = { 0x00000000, 0xC0000000, 0xBF99999A,
+ 0xBF99999A, 0x00000000, 0x40466666,
+ 0x40800000, 0x40466666, 0x00000000,
+ 0x00000000, 0x3F8CCCCD }
+ }
+};
+
+/* 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];
+};
+
+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 }
+ },
+ { .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 = "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 }
+ }
+};
+
+/*
+ * DSP volume setting structs. Req 1 is left volume, req 2 is right volume,
+ * and I don't know what the third req is, but it's always zero. I assume it's
+ * some sort of update or set command to tell the DSP there's new volume info.
+ */
+#define DSP_VOL_OUT 0
+#define DSP_VOL_IN 1
+
+struct ct_dsp_volume_ctl {
+ hda_nid_t vnid;
+ int mid; /* module ID*/
+ unsigned int reqs[3]; /* scp req ID */
+};
+
+static struct ct_dsp_volume_ctl ca0132_alt_vol_ctls[] = {
+ { .vnid = VNID_SPK,
+ .mid = 0x32,
+ .reqs = {3, 4, 2}
+ },
+ { .vnid = VNID_MIC,
+ .mid = 0x37,
+ .reqs = {2, 3, 1}
+ }
+};
+
enum hda_cmd_vendor_io {
/* for DspIO node */
VENDOR_DSPIO_SCP_WRITE_DATA_LOW = 0x000,
@@ -703,6 +897,7 @@ struct ca0132_spec {
const struct hda_verb *base_init_verbs;
const struct hda_verb *base_exit_verbs;
const struct hda_verb *chip_init_verbs;
+ const struct hda_verb *sbz_init_verbs;
struct hda_verb *spec_init_verbs;
struct auto_pin_cfg autocfg;
@@ -719,6 +914,7 @@ struct ca0132_spec {
hda_nid_t shared_mic_nid;
hda_nid_t shared_out_nid;
hda_nid_t unsol_tag_hp;
+ hda_nid_t unsol_tag_front_hp; /* for desktop ca0132 codecs */
hda_nid_t unsol_tag_amic1;
/* chip access */
@@ -734,6 +930,9 @@ struct ca0132_spec {
unsigned int scp_resp_header;
unsigned int scp_resp_data[4];
unsigned int scp_resp_count;
+ bool alt_firmware_present;
+ bool startup_check_entered;
+ bool dsp_reload;
/* mixer and effects related */
unsigned char dmic_ctl;
@@ -746,6 +945,17 @@ struct ca0132_spec {
long effects_switch[EFFECTS_COUNT];
long voicefx_val;
long cur_mic_boost;
+ /* ca0132_alt control related values */
+ unsigned char in_enum_val;
+ unsigned char out_enum_val;
+ unsigned char mic_boost_enum_val;
+ unsigned char smart_volume_setting;
+ long fx_ctl_val[EFFECT_LEVEL_SLIDERS];
+ long xbass_xover_freq;
+ long eq_preset_val;
+ unsigned int tlv[4];
+ struct hda_vmaster_mute_hook vmaster_mute;
+
struct hda_codec *codec;
struct delayed_work unsol_hp_work;
@@ -754,6 +964,25 @@ struct ca0132_spec {
#ifdef ENABLE_TUNING_CONTROLS
long cur_ctl_vals[TUNING_CTLS_COUNT];
#endif
+ /*
+ * Sound Blaster Z PCI region 2 iomem, used for input and output
+ * switching, and other unknown commands.
+ */
+ void __iomem *mem_base;
+
+ /*
+ * Whether or not to use the alt functions like alt_select_out,
+ * alt_select_in, etc. Only used on desktop codecs for now, because of
+ * surround sound support.
+ */
+ bool use_alt_functions;
+
+ /*
+ * Whether or not to use alt controls: volume effect sliders, EQ
+ * presets, smart volume presets, and new control names with FX prefix.
+ * Renames PlayEnhancement and CrystalVoice too.
+ */
+ bool use_alt_controls;
};
/*
@@ -762,6 +991,8 @@ struct ca0132_spec {
enum {
QUIRK_NONE,
QUIRK_ALIENWARE,
+ QUIRK_SBZ,
+ QUIRK_R3DI,
};
static const struct hda_pintbl alienware_pincfgs[] = {
@@ -778,10 +1009,44 @@ static const struct hda_pintbl alienware_pincfgs[] = {
{}
};
+/* Sound Blaster Z pin configs taken from Windows Driver */
+static const struct hda_pintbl sbz_pincfgs[] = {
+ { 0x0b, 0x01017010 }, /* Port G -- Lineout FRONT L/R */
+ { 0x0c, 0x014510f0 }, /* SPDIF Out 1 */
+ { 0x0d, 0x014510f0 }, /* Digital Out */
+ { 0x0e, 0x01c510f0 }, /* SPDIF In */
+ { 0x0f, 0x0221701f }, /* Port A -- BackPanel HP */
+ { 0x10, 0x01017012 }, /* Port D -- Center/LFE or FP Hp */
+ { 0x11, 0x01017014 }, /* Port B -- LineMicIn2 / Rear L/R */
+ { 0x12, 0x01a170f0 }, /* Port C -- LineIn1 */
+ { 0x13, 0x908700f0 }, /* What U Hear In*/
+ { 0x18, 0x50d000f0 }, /* N/A */
+ {}
+};
+
+/* Recon3D integrated pin configs taken from Windows Driver */
+static const struct hda_pintbl r3di_pincfgs[] = {
+ { 0x0b, 0x01014110 }, /* Port G -- Lineout FRONT L/R */
+ { 0x0c, 0x014510f0 }, /* SPDIF Out 1 */
+ { 0x0d, 0x014510f0 }, /* Digital Out */
+ { 0x0e, 0x41c520f0 }, /* SPDIF In */
+ { 0x0f, 0x0221401f }, /* Port A -- BackPanel HP */
+ { 0x10, 0x01016011 }, /* Port D -- Center/LFE or FP Hp */
+ { 0x11, 0x01011014 }, /* Port B -- LineMicIn2 / Rear L/R */
+ { 0x12, 0x02a090f0 }, /* Port C -- LineIn1 */
+ { 0x13, 0x908700f0 }, /* What U Hear In*/
+ { 0x18, 0x500000f0 }, /* N/A */
+ {}
+};
+
static const struct snd_pci_quirk ca0132_quirks[] = {
SND_PCI_QUIRK(0x1028, 0x0685, "Alienware 15 2015", QUIRK_ALIENWARE),
SND_PCI_QUIRK(0x1028, 0x0688, "Alienware 17 2015", QUIRK_ALIENWARE),
SND_PCI_QUIRK(0x1028, 0x0708, "Alienware 15 R2 2016", QUIRK_ALIENWARE),
+ SND_PCI_QUIRK(0x1102, 0x0010, "Sound Blaster Z", QUIRK_SBZ),
+ SND_PCI_QUIRK(0x1102, 0x0023, "Sound Blaster Z", QUIRK_SBZ),
+ SND_PCI_QUIRK(0x1458, 0xA016, "Recon3Di", QUIRK_R3DI),
+ SND_PCI_QUIRK(0x1458, 0xA036, "Recon3Di", QUIRK_R3DI),
{}
};
@@ -965,6 +1230,29 @@ static int chipio_write(struct hda_codec *codec,
}
/*
+ * Write given value to the given address through the chip I/O widget.
+ * not protected by the Mutex
+ */
+static int chipio_write_no_mutex(struct hda_codec *codec,
+ unsigned int chip_addx, const unsigned int data)
+{
+ int err;
+
+
+ /* write the address, and if successful proceed to write data */
+ err = chipio_write_address(codec, chip_addx);
+ if (err < 0)
+ goto exit;
+
+ err = chipio_write_data(codec, data);
+ if (err < 0)
+ goto exit;
+
+exit:
+ return err;
+}
+
+/*
* Write multiple values to the given address through the chip I/O widget.
* protected by the Mutex
*/
@@ -1058,6 +1346,81 @@ static void chipio_set_control_param(struct hda_codec *codec,
}
/*
+ * Set chip parameters through the chip I/O widget. NO MUTEX.
+ */
+static void chipio_set_control_param_no_mutex(struct hda_codec *codec,
+ enum control_param_id param_id, int param_val)
+{
+ int val;
+
+ if ((param_id < 32) && (param_val < 8)) {
+ val = (param_val << 5) | (param_id);
+ snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+ VENDOR_CHIPIO_PARAM_SET, val);
+ } else {
+ if (chipio_send(codec, VENDOR_CHIPIO_STATUS, 0) == 0) {
+ snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+ VENDOR_CHIPIO_PARAM_EX_ID_SET,
+ param_id);
+ snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+ VENDOR_CHIPIO_PARAM_EX_VALUE_SET,
+ param_val);
+ }
+ }
+}
+/*
+ * Connect stream to a source point, and then connect
+ * that source point to a destination point.
+ */
+static void chipio_set_stream_source_dest(struct hda_codec *codec,
+ int streamid, int source_point, int dest_point)
+{
+ chipio_set_control_param_no_mutex(codec,
+ CONTROL_PARAM_STREAM_ID, streamid);
+ chipio_set_control_param_no_mutex(codec,
+ CONTROL_PARAM_STREAM_SOURCE_CONN_POINT, source_point);
+ chipio_set_control_param_no_mutex(codec,
+ CONTROL_PARAM_STREAM_DEST_CONN_POINT, dest_point);
+}
+
+/*
+ * Set number of channels in the selected stream.
+ */
+static void chipio_set_stream_channels(struct hda_codec *codec,
+ int streamid, unsigned int channels)
+{
+ chipio_set_control_param_no_mutex(codec,
+ CONTROL_PARAM_STREAM_ID, streamid);
+ chipio_set_control_param_no_mutex(codec,
+ CONTROL_PARAM_STREAMS_CHANNELS, channels);
+}
+
+/*
+ * Enable/Disable audio stream.
+ */
+static void chipio_set_stream_control(struct hda_codec *codec,
+ int streamid, int enable)
+{
+ chipio_set_control_param_no_mutex(codec,
+ CONTROL_PARAM_STREAM_ID, streamid);
+ chipio_set_control_param_no_mutex(codec,
+ CONTROL_PARAM_STREAM_CONTROL, enable);
+}
+
+
+/*
+ * Set sampling rate of the connection point. NO MUTEX.
+ */
+static void chipio_set_conn_rate_no_mutex(struct hda_codec *codec,
+ int connid, enum ca0132_sample_rate rate)
+{
+ chipio_set_control_param_no_mutex(codec,
+ CONTROL_PARAM_CONN_POINT_ID, connid);
+ chipio_set_control_param_no_mutex(codec,
+ CONTROL_PARAM_CONN_POINT_SAMPLE_RATE, rate);
+}
+
+/*
* Set sampling rate of the connection point.
*/
static void chipio_set_conn_rate(struct hda_codec *codec,
@@ -1420,8 +1783,8 @@ static int dspio_send_scp_message(struct hda_codec *codec,
* Returns zero or a negative error code.
*/
static int dspio_scp(struct hda_codec *codec,
- int mod_id, int req, int dir, void *data, unsigned int len,
- void *reply, unsigned int *reply_len)
+ int mod_id, int src_id, int req, int dir, const void *data,
+ unsigned int len, void *reply, unsigned int *reply_len)
{
int status = 0;
struct scp_msg scp_send, scp_reply;
@@ -1445,7 +1808,7 @@ static int dspio_scp(struct hda_codec *codec,
return -EINVAL;
}
- scp_send.hdr = make_scp_header(mod_id, 0x20, (dir == SCP_GET), req,
+ scp_send.hdr = make_scp_header(mod_id, src_id, (dir == SCP_GET), req,
0, 0, 0, len/sizeof(unsigned int));
if (data != NULL && len > 0) {
len = min((unsigned int)(sizeof(scp_send.data)), len);
@@ -1502,15 +1865,24 @@ static int dspio_scp(struct hda_codec *codec,
* Set DSP parameters
*/
static int dspio_set_param(struct hda_codec *codec, int mod_id,
- int req, void *data, unsigned int len)
+ int src_id, int req, const void *data, unsigned int len)
{
- return dspio_scp(codec, mod_id, req, SCP_SET, data, len, NULL, NULL);
+ return dspio_scp(codec, mod_id, src_id, req, SCP_SET, data, len, NULL,
+ NULL);
}
static int dspio_set_uint_param(struct hda_codec *codec, int mod_id,
- int req, unsigned int data)
+ int req, const unsigned int data)
{
- return dspio_set_param(codec, mod_id, req, &data, sizeof(unsigned int));
+ return dspio_set_param(codec, mod_id, 0x20, req, &data,
+ sizeof(unsigned int));
+}
+
+static int dspio_set_uint_param_no_source(struct hda_codec *codec, int mod_id,
+ int req, const unsigned int data)
+{
+ return dspio_set_param(codec, mod_id, 0x00, req, &data,
+ sizeof(unsigned int));
}
/*
@@ -1522,8 +1894,9 @@ static int dspio_alloc_dma_chan(struct hda_codec *codec, unsigned int *dma_chan)
unsigned int size = sizeof(dma_chan);
codec_dbg(codec, " dspio_alloc_dma_chan() -- begin\n");
- status = dspio_scp(codec, MASTERCONTROL, MASTERCONTROL_ALLOC_DMA_CHAN,
- SCP_GET, NULL, 0, dma_chan, &size);
+ status = dspio_scp(codec, MASTERCONTROL, 0x20,
+ MASTERCONTROL_ALLOC_DMA_CHAN, SCP_GET, NULL, 0,
+ dma_chan, &size);
if (status < 0) {
codec_dbg(codec, "dspio_alloc_dma_chan: SCP Failed\n");
@@ -1552,8 +1925,9 @@ static int dspio_free_dma_chan(struct hda_codec *codec, unsigned int dma_chan)
codec_dbg(codec, " dspio_free_dma_chan() -- begin\n");
codec_dbg(codec, "dspio_free_dma_chan: chan=%d\n", dma_chan);
- status = dspio_scp(codec, MASTERCONTROL, MASTERCONTROL_ALLOC_DMA_CHAN,
- SCP_SET, &dma_chan, sizeof(dma_chan), NULL, &dummy);
+ status = dspio_scp(codec, MASTERCONTROL, 0x20,
+ MASTERCONTROL_ALLOC_DMA_CHAN, SCP_SET, &dma_chan,
+ sizeof(dma_chan), NULL, &dummy);
if (status < 0) {
codec_dbg(codec, "dspio_free_dma_chan: SCP Failed\n");
@@ -2575,14 +2949,16 @@ static int dspxfr_image(struct hda_codec *codec,
*/
static void dspload_post_setup(struct hda_codec *codec)
{
+ struct ca0132_spec *spec = codec->spec;
codec_dbg(codec, "---- dspload_post_setup ------\n");
+ if (!spec->use_alt_functions) {
+ /*set DSP speaker to 2.0 configuration*/
+ chipio_write(codec, XRAM_XRAM_INST_OFFSET(0x18), 0x08080080);
+ chipio_write(codec, XRAM_XRAM_INST_OFFSET(0x19), 0x3f800000);
- /*set DSP speaker to 2.0 configuration*/
- chipio_write(codec, XRAM_XRAM_INST_OFFSET(0x18), 0x08080080);
- chipio_write(codec, XRAM_XRAM_INST_OFFSET(0x19), 0x3f800000);
-
- /*update write pointer*/
- chipio_write(codec, XRAM_XRAM_INST_OFFSET(0x29), 0x00000002);
+ /*update write pointer*/
+ chipio_write(codec, XRAM_XRAM_INST_OFFSET(0x29), 0x00000002);
+ }
}
/**
@@ -2690,6 +3066,170 @@ static bool dspload_wait_loaded(struct hda_codec *codec)
}
/*
+ * Setup GPIO for the other variants of Core3D.
+ */
+
+/*
+ * Sets up the GPIO pins so that they are discoverable. If this isn't done,
+ * the card shows as having no GPIO pins.
+ */
+static void ca0132_gpio_init(struct hda_codec *codec)
+{
+ struct ca0132_spec *spec = codec->spec;
+
+ switch (spec->quirk) {
+ case QUIRK_SBZ:
+ 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);
+ break;
+ case QUIRK_R3DI:
+ snd_hda_codec_write(codec, 0x01, 0, 0x793, 0x00);
+ snd_hda_codec_write(codec, 0x01, 0, 0x794, 0x5B);
+ break;
+ }
+
+}
+
+/* Sets the GPIO for audio output. */
+static void ca0132_gpio_setup(struct hda_codec *codec)
+{
+ struct ca0132_spec *spec = codec->spec;
+
+ switch (spec->quirk) {
+ case QUIRK_SBZ:
+ snd_hda_codec_write(codec, 0x01, 0,
+ AC_VERB_SET_GPIO_DIRECTION, 0x07);
+ snd_hda_codec_write(codec, 0x01, 0,
+ AC_VERB_SET_GPIO_MASK, 0x07);
+ snd_hda_codec_write(codec, 0x01, 0,
+ AC_VERB_SET_GPIO_DATA, 0x04);
+ snd_hda_codec_write(codec, 0x01, 0,
+ AC_VERB_SET_GPIO_DATA, 0x06);
+ break;
+ case QUIRK_R3DI:
+ snd_hda_codec_write(codec, 0x01, 0,
+ AC_VERB_SET_GPIO_DIRECTION, 0x1E);
+ snd_hda_codec_write(codec, 0x01, 0,
+ AC_VERB_SET_GPIO_MASK, 0x1F);
+ snd_hda_codec_write(codec, 0x01, 0,
+ AC_VERB_SET_GPIO_DATA, 0x0C);
+ break;
+ }
+}
+
+/*
+ * GPIO control functions for the Recon3D integrated.
+ */
+
+enum r3di_gpio_bit {
+ /* Bit 1 - Switch between front/rear mic. 0 = rear, 1 = front */
+ R3DI_MIC_SELECT_BIT = 1,
+ /* Bit 2 - Switch between headphone/line out. 0 = Headphone, 1 = Line */
+ R3DI_OUT_SELECT_BIT = 2,
+ /*
+ * I dunno what this actually does, but it stays on until the dsp
+ * is downloaded.
+ */
+ R3DI_GPIO_DSP_DOWNLOADING = 3,
+ /*
+ * Same as above, no clue what it does, but it comes on after the dsp
+ * is downloaded.
+ */
+ R3DI_GPIO_DSP_DOWNLOADED = 4
+};
+
+enum r3di_mic_select {
+ /* Set GPIO bit 1 to 0 for rear mic */
+ R3DI_REAR_MIC = 0,
+ /* Set GPIO bit 1 to 1 for front microphone*/
+ R3DI_FRONT_MIC = 1
+};
+
+enum r3di_out_select {
+ /* Set GPIO bit 2 to 0 for headphone */
+ R3DI_HEADPHONE_OUT = 0,
+ /* Set GPIO bit 2 to 1 for speaker */
+ R3DI_LINE_OUT = 1
+};
+enum r3di_dsp_status {
+ /* Set GPIO bit 3 to 1 until DSP is downloaded */
+ R3DI_DSP_DOWNLOADING = 0,
+ /* Set GPIO bit 4 to 1 once DSP is downloaded */
+ R3DI_DSP_DOWNLOADED = 1
+};
+
+
+static void r3di_gpio_mic_set(struct hda_codec *codec,
+ enum r3di_mic_select cur_mic)
+{
+ 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_mic) {
+ case R3DI_REAR_MIC:
+ cur_gpio &= ~(1 << R3DI_MIC_SELECT_BIT);
+ break;
+ case R3DI_FRONT_MIC:
+ cur_gpio |= (1 << R3DI_MIC_SELECT_BIT);
+ break;
+ }
+ snd_hda_codec_write(codec, codec->core.afg, 0,
+ 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)
+{
+ 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 (dsp_status) {
+ case R3DI_DSP_DOWNLOADING:
+ cur_gpio |= (1 << R3DI_GPIO_DSP_DOWNLOADING);
+ snd_hda_codec_write(codec, codec->core.afg, 0,
+ AC_VERB_SET_GPIO_DATA, cur_gpio);
+ break;
+ case R3DI_DSP_DOWNLOADED:
+ /* Set DOWNLOADING bit to 0. */
+ cur_gpio &= ~(1 << R3DI_GPIO_DSP_DOWNLOADING);
+
+ snd_hda_codec_write(codec, codec->core.afg, 0,
+ AC_VERB_SET_GPIO_DATA, cur_gpio);
+
+ cur_gpio |= (1 << R3DI_GPIO_DSP_DOWNLOADED);
+ break;
+ }
+
+ snd_hda_codec_write(codec, codec->core.afg, 0,
+ AC_VERB_SET_GPIO_DATA, cur_gpio);
+}
+
+/*
* PCM callbacks
*/
static int ca0132_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
@@ -2852,6 +3392,24 @@ static unsigned int ca0132_capture_pcm_delay(struct hda_pcm_stream *info,
.tlv = { .c = ca0132_volume_tlv }, \
.private_value = HDA_COMPOSE_AMP_VAL(nid, channel, 0, dir) }
+/*
+ * Creates a mixer control that uses defaults of HDA_CODEC_VOL except for the
+ * volume put, which is used for setting the DSP volume. This was done because
+ * the ca0132 functions were taking too much time and causing lag.
+ */
+#define CA0132_ALT_CODEC_VOL_MONO(xname, nid, channel, dir) \
+ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+ .name = xname, \
+ .subdevice = HDA_SUBDEV_AMP_FLAG, \
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
+ SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, \
+ .info = snd_hda_mixer_amp_volume_info, \
+ .get = snd_hda_mixer_amp_volume_get, \
+ .put = ca0132_alt_volume_put, \
+ .tlv = { .c = snd_hda_mixer_amp_tlv }, \
+ .private_value = HDA_COMPOSE_AMP_VAL(nid, channel, 0, dir) }
+
#define CA0132_CODEC_MUTE_MONO(xname, nid, channel, dir) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
.name = xname, \
@@ -2864,9 +3422,88 @@ static unsigned int ca0132_capture_pcm_delay(struct hda_pcm_stream *info,
/* stereo */
#define CA0132_CODEC_VOL(xname, nid, dir) \
CA0132_CODEC_VOL_MONO(xname, nid, 3, dir)
+#define CA0132_ALT_CODEC_VOL(xname, nid, dir) \
+ CA0132_ALT_CODEC_VOL_MONO(xname, nid, 3, dir)
#define CA0132_CODEC_MUTE(xname, nid, dir) \
CA0132_CODEC_MUTE_MONO(xname, nid, 3, dir)
+/* lookup tables */
+/*
+ * Lookup table with decibel values for the DSP. When volume is changed in
+ * Windows, the DSP is also sent the dB value in floating point. In Windows,
+ * these values have decimal points, probably because the Windows driver
+ * actually uses floating point. We can't here, so I made a lookup table of
+ * values -90 to 9. -90 is the lowest decibel value for both the ADC's and the
+ * DAC's, and 9 is the maximum.
+ */
+static const unsigned int float_vol_db_lookup[] = {
+0xC2B40000, 0xC2B20000, 0xC2B00000, 0xC2AE0000, 0xC2AC0000, 0xC2AA0000,
+0xC2A80000, 0xC2A60000, 0xC2A40000, 0xC2A20000, 0xC2A00000, 0xC29E0000,
+0xC29C0000, 0xC29A0000, 0xC2980000, 0xC2960000, 0xC2940000, 0xC2920000,
+0xC2900000, 0xC28E0000, 0xC28C0000, 0xC28A0000, 0xC2880000, 0xC2860000,
+0xC2840000, 0xC2820000, 0xC2800000, 0xC27C0000, 0xC2780000, 0xC2740000,
+0xC2700000, 0xC26C0000, 0xC2680000, 0xC2640000, 0xC2600000, 0xC25C0000,
+0xC2580000, 0xC2540000, 0xC2500000, 0xC24C0000, 0xC2480000, 0xC2440000,
+0xC2400000, 0xC23C0000, 0xC2380000, 0xC2340000, 0xC2300000, 0xC22C0000,
+0xC2280000, 0xC2240000, 0xC2200000, 0xC21C0000, 0xC2180000, 0xC2140000,
+0xC2100000, 0xC20C0000, 0xC2080000, 0xC2040000, 0xC2000000, 0xC1F80000,
+0xC1F00000, 0xC1E80000, 0xC1E00000, 0xC1D80000, 0xC1D00000, 0xC1C80000,
+0xC1C00000, 0xC1B80000, 0xC1B00000, 0xC1A80000, 0xC1A00000, 0xC1980000,
+0xC1900000, 0xC1880000, 0xC1800000, 0xC1700000, 0xC1600000, 0xC1500000,
+0xC1400000, 0xC1300000, 0xC1200000, 0xC1100000, 0xC1000000, 0xC0E00000,
+0xC0C00000, 0xC0A00000, 0xC0800000, 0xC0400000, 0xC0000000, 0xBF800000,
+0x00000000, 0x3F800000, 0x40000000, 0x40400000, 0x40800000, 0x40A00000,
+0x40C00000, 0x40E00000, 0x41000000, 0x41100000
+};
+
+/*
+ * This table counts from float 0 to 1 in increments of .01, which is
+ * useful for a few different sliders.
+ */
+static const unsigned int float_zero_to_one_lookup[] = {
+0x00000000, 0x3C23D70A, 0x3CA3D70A, 0x3CF5C28F, 0x3D23D70A, 0x3D4CCCCD,
+0x3D75C28F, 0x3D8F5C29, 0x3DA3D70A, 0x3DB851EC, 0x3DCCCCCD, 0x3DE147AE,
+0x3DF5C28F, 0x3E051EB8, 0x3E0F5C29, 0x3E19999A, 0x3E23D70A, 0x3E2E147B,
+0x3E3851EC, 0x3E428F5C, 0x3E4CCCCD, 0x3E570A3D, 0x3E6147AE, 0x3E6B851F,
+0x3E75C28F, 0x3E800000, 0x3E851EB8, 0x3E8A3D71, 0x3E8F5C29, 0x3E947AE1,
+0x3E99999A, 0x3E9EB852, 0x3EA3D70A, 0x3EA8F5C3, 0x3EAE147B, 0x3EB33333,
+0x3EB851EC, 0x3EBD70A4, 0x3EC28F5C, 0x3EC7AE14, 0x3ECCCCCD, 0x3ED1EB85,
+0x3ED70A3D, 0x3EDC28F6, 0x3EE147AE, 0x3EE66666, 0x3EEB851F, 0x3EF0A3D7,
+0x3EF5C28F, 0x3EFAE148, 0x3F000000, 0x3F028F5C, 0x3F051EB8, 0x3F07AE14,
+0x3F0A3D71, 0x3F0CCCCD, 0x3F0F5C29, 0x3F11EB85, 0x3F147AE1, 0x3F170A3D,
+0x3F19999A, 0x3F1C28F6, 0x3F1EB852, 0x3F2147AE, 0x3F23D70A, 0x3F266666,
+0x3F28F5C3, 0x3F2B851F, 0x3F2E147B, 0x3F30A3D7, 0x3F333333, 0x3F35C28F,
+0x3F3851EC, 0x3F3AE148, 0x3F3D70A4, 0x3F400000, 0x3F428F5C, 0x3F451EB8,
+0x3F47AE14, 0x3F4A3D71, 0x3F4CCCCD, 0x3F4F5C29, 0x3F51EB85, 0x3F547AE1,
+0x3F570A3D, 0x3F59999A, 0x3F5C28F6, 0x3F5EB852, 0x3F6147AE, 0x3F63D70A,
+0x3F666666, 0x3F68F5C3, 0x3F6B851F, 0x3F6E147B, 0x3F70A3D7, 0x3F733333,
+0x3F75C28F, 0x3F7851EC, 0x3F7AE148, 0x3F7D70A4, 0x3F800000
+};
+
+/*
+ * This table counts from float 10 to 1000, which is the range of the x-bass
+ * crossover slider in Windows.
+ */
+static const unsigned int float_xbass_xover_lookup[] = {
+0x41200000, 0x41A00000, 0x41F00000, 0x42200000, 0x42480000, 0x42700000,
+0x428C0000, 0x42A00000, 0x42B40000, 0x42C80000, 0x42DC0000, 0x42F00000,
+0x43020000, 0x430C0000, 0x43160000, 0x43200000, 0x432A0000, 0x43340000,
+0x433E0000, 0x43480000, 0x43520000, 0x435C0000, 0x43660000, 0x43700000,
+0x437A0000, 0x43820000, 0x43870000, 0x438C0000, 0x43910000, 0x43960000,
+0x439B0000, 0x43A00000, 0x43A50000, 0x43AA0000, 0x43AF0000, 0x43B40000,
+0x43B90000, 0x43BE0000, 0x43C30000, 0x43C80000, 0x43CD0000, 0x43D20000,
+0x43D70000, 0x43DC0000, 0x43E10000, 0x43E60000, 0x43EB0000, 0x43F00000,
+0x43F50000, 0x43FA0000, 0x43FF0000, 0x44020000, 0x44048000, 0x44070000,
+0x44098000, 0x440C0000, 0x440E8000, 0x44110000, 0x44138000, 0x44160000,
+0x44188000, 0x441B0000, 0x441D8000, 0x44200000, 0x44228000, 0x44250000,
+0x44278000, 0x442A0000, 0x442C8000, 0x442F0000, 0x44318000, 0x44340000,
+0x44368000, 0x44390000, 0x443B8000, 0x443E0000, 0x44408000, 0x44430000,
+0x44458000, 0x44480000, 0x444A8000, 0x444D0000, 0x444F8000, 0x44520000,
+0x44548000, 0x44570000, 0x44598000, 0x445C0000, 0x445E8000, 0x44610000,
+0x44638000, 0x44660000, 0x44688000, 0x446B0000, 0x446D8000, 0x44700000,
+0x44728000, 0x44750000, 0x44778000, 0x447A0000
+};
+
/* The following are for tuning of products */
#ifdef ENABLE_TUNING_CONTROLS
@@ -2942,7 +3579,7 @@ static int tuning_ctl_set(struct hda_codec *codec, hda_nid_t nid,
break;
snd_hda_power_up(codec);
- dspio_set_param(codec, ca0132_tuning_ctls[i].mid,
+ dspio_set_param(codec, ca0132_tuning_ctls[i].mid, 0x20,
ca0132_tuning_ctls[i].req,
&(lookup[idx]), sizeof(unsigned int));
snd_hda_power_down(codec);
@@ -3068,8 +3705,8 @@ static int equalizer_ctl_put(struct snd_kcontrol *kcontrol,
return 1;
}
-static const DECLARE_TLV_DB_SCALE(voice_focus_db_scale, 2000, 100, 0);
-static const DECLARE_TLV_DB_SCALE(eq_db_scale, -2400, 100, 0);
+static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(voice_focus_db_scale, 2000, 100, 0);
+static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(eq_db_scale, -2400, 100, 0);
static int add_tuning_control(struct hda_codec *codec,
hda_nid_t pnid, hda_nid_t nid,
@@ -3207,7 +3844,7 @@ static int ca0132_select_out(struct hda_codec *codec)
pin_ctl & ~PIN_HP);
/* enable speaker node */
pin_ctl = snd_hda_codec_read(codec, spec->out_pins[0], 0,
- AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+ AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
snd_hda_set_pin_ctl(codec, spec->out_pins[0],
pin_ctl | PIN_OUT);
} else {
@@ -3251,13 +3888,209 @@ static int ca0132_select_out(struct hda_codec *codec)
return err < 0 ? err : 0;
}
+/*
+ * This function behaves similarly to the ca0132_select_out funciton above,
+ * except with a few differences. It adds the ability to select the current
+ * 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.
+ */
+static int ca0132_alt_select_out(struct hda_codec *codec)
+{
+ struct ca0132_spec *spec = codec->spec;
+ unsigned int pin_ctl;
+ 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];
+
+ codec_dbg(codec, "%s\n", __func__);
+
+ snd_hda_power_up_pm(codec);
+
+ auto_jack = spec->vnode_lswitch[VNID_HP_ASEL - VNODE_START_NID];
+
+ /*
+ * If headphone rear or front is plugged in, set to headphone.
+ * If neither is plugged in, set to rear line out. Only if
+ * hp/speaker auto detect is enabled.
+ */
+ if (auto_jack) {
+ jack_present = snd_hda_jack_detect(codec, spec->unsol_tag_hp) ||
+ snd_hda_jack_detect(codec, spec->unsol_tag_front_hp);
+
+ if (jack_present)
+ spec->cur_out_type = HEADPHONE_OUT;
+ else
+ spec->cur_out_type = SPEAKER_OUT;
+ } 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);
+ if (err < 0)
+ goto exit;
+
+ switch (spec->cur_out_type) {
+ case SPEAKER_OUT:
+ codec_dbg(codec, "%s speaker\n", __func__);
+ /*speaker out config*/
+ switch (spec->quirk) {
+ case QUIRK_SBZ:
+ writew(0x0007, spec->mem_base + 0x320);
+ writew(0x0104, spec->mem_base + 0x320);
+ writew(0x0101, spec->mem_base + 0x320);
+ chipio_set_control_param(codec, 0x0D, 0x18);
+ break;
+ case QUIRK_R3DI:
+ chipio_set_control_param(codec, 0x0D, 0x24);
+ r3di_gpio_out_set(codec, R3DI_LINE_OUT);
+ break;
+ }
+
+ /* 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);
+ else
+ dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_EIGHT);
+ break;
+ case HEADPHONE_OUT:
+ codec_dbg(codec, "%s hp\n", __func__);
+ /* Headphone out config*/
+ switch (spec->quirk) {
+ case QUIRK_SBZ:
+ writew(0x0107, spec->mem_base + 0x320);
+ writew(0x0104, spec->mem_base + 0x320);
+ writew(0x0001, spec->mem_base + 0x320);
+ chipio_set_control_param(codec, 0x0D, 0x12);
+ break;
+ case QUIRK_R3DI:
+ chipio_set_control_param(codec, 0x0D, 0x21);
+ r3di_gpio_out_set(codec, R3DI_HEADPHONE_OUT);
+ break;
+ }
+
+ 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);
+
+ /* 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);
+
+ if (spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID])
+ 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__);
+ /* Surround out config*/
+ switch (spec->quirk) {
+ case QUIRK_SBZ:
+ writew(0x0007, spec->mem_base + 0x320);
+ writew(0x0104, spec->mem_base + 0x320);
+ writew(0x0101, spec->mem_base + 0x320);
+ chipio_set_control_param(codec, 0x0D, 0x18);
+ break;
+ case QUIRK_R3DI:
+ chipio_set_control_param(codec, 0x0D, 0x24);
+ r3di_gpio_out_set(codec, R3DI_LINE_OUT);
+ break;
+ }
+ /* 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);
+
+ if (spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID])
+ dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_ONE);
+ else
+ dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_EIGHT);
+ break;
+ }
+
+ /* run through the output dsp commands for line-out */
+ 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]);
+
+ if (err < 0)
+ goto exit;
+ }
+
+exit:
+ snd_hda_power_down_pm(codec);
+
+ return err < 0 ? err : 0;
+}
+
static void ca0132_unsol_hp_delayed(struct work_struct *work)
{
struct ca0132_spec *spec = container_of(
to_delayed_work(work), struct ca0132_spec, unsol_hp_work);
struct hda_jack_tbl *jack;
- ca0132_select_out(spec->codec);
+ if (spec->use_alt_functions)
+ ca0132_alt_select_out(spec->codec);
+ else
+ ca0132_select_out(spec->codec);
+
jack = snd_hda_jack_tbl_get(spec->codec, spec->unsol_tag_hp);
if (jack) {
jack->block_report = 0;
@@ -3268,6 +4101,10 @@ static void ca0132_unsol_hp_delayed(struct work_struct *work)
static void ca0132_set_dmic(struct hda_codec *codec, int enable);
static int ca0132_mic_boost_set(struct hda_codec *codec, long val);
static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val);
+static void resume_mic1(struct hda_codec *codec, unsigned int oldval);
+static int stop_mic1(struct hda_codec *codec);
+static int ca0132_cvoice_switch_set(struct hda_codec *codec);
+static int ca0132_alt_mic_boost_set(struct hda_codec *codec, long val);
/*
* Select the active VIP source
@@ -3310,6 +4147,71 @@ static int ca0132_set_vipsource(struct hda_codec *codec, int val)
return 1;
}
+static int ca0132_alt_set_vipsource(struct hda_codec *codec, int val)
+{
+ struct ca0132_spec *spec = codec->spec;
+ unsigned int tmp;
+
+ if (spec->dsp_state != DSP_DOWNLOADED)
+ return 0;
+
+ codec_dbg(codec, "%s\n", __func__);
+
+ chipio_set_stream_control(codec, 0x03, 0);
+ chipio_set_stream_control(codec, 0x04, 0);
+
+ /* if CrystalVoice is off, vipsource should be 0 */
+ if (!spec->effects_switch[CRYSTAL_VOICE - EFFECT_START_NID] ||
+ (val == 0) || spec->in_enum_val == REAR_LINE_IN) {
+ codec_dbg(codec, "%s: off.", __func__);
+ chipio_set_control_param(codec, CONTROL_PARAM_VIP_SOURCE, 0);
+
+ tmp = FLOAT_ZERO;
+ dspio_set_uint_param(codec, 0x80, 0x05, tmp);
+
+ chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000);
+ chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000);
+ if (spec->quirk == QUIRK_R3DI)
+ chipio_set_conn_rate(codec, 0x0F, SR_96_000);
+
+
+ if (spec->in_enum_val == REAR_LINE_IN)
+ tmp = FLOAT_ZERO;
+ else {
+ if (spec->quirk == QUIRK_SBZ)
+ tmp = FLOAT_THREE;
+ else
+ tmp = FLOAT_ONE;
+ }
+
+ dspio_set_uint_param(codec, 0x80, 0x00, tmp);
+
+ } else {
+ codec_dbg(codec, "%s: on.", __func__);
+ chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_16_000);
+ chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_16_000);
+ if (spec->quirk == QUIRK_R3DI)
+ chipio_set_conn_rate(codec, 0x0F, SR_16_000);
+
+ if (spec->effects_switch[VOICE_FOCUS - EFFECT_START_NID])
+ tmp = FLOAT_TWO;
+ else
+ tmp = FLOAT_ONE;
+ dspio_set_uint_param(codec, 0x80, 0x00, tmp);
+
+ tmp = FLOAT_ONE;
+ dspio_set_uint_param(codec, 0x80, 0x05, tmp);
+
+ msleep(20);
+ chipio_set_control_param(codec, CONTROL_PARAM_VIP_SOURCE, val);
+ }
+
+ chipio_set_stream_control(codec, 0x03, 1);
+ chipio_set_stream_control(codec, 0x04, 1);
+
+ return 1;
+}
+
/*
* Select the active microphone.
* If autodetect is enabled, mic will be selected based on jack detection.
@@ -3363,6 +4265,125 @@ static int ca0132_select_mic(struct hda_codec *codec)
}
/*
+ * Select the active input.
+ * Mic detection isn't used, because it's kind of pointless on the SBZ.
+ * The front mic has no jack-detection, so the only way to switch to it
+ * is to do it manually in alsamixer.
+ */
+static int ca0132_alt_select_in(struct hda_codec *codec)
+{
+ struct ca0132_spec *spec = codec->spec;
+ unsigned int tmp;
+
+ codec_dbg(codec, "%s\n", __func__);
+
+ snd_hda_power_up_pm(codec);
+
+ chipio_set_stream_control(codec, 0x03, 0);
+ chipio_set_stream_control(codec, 0x04, 0);
+
+ spec->cur_mic_type = spec->in_enum_val;
+
+ switch (spec->cur_mic_type) {
+ case REAR_MIC:
+ switch (spec->quirk) {
+ case QUIRK_SBZ:
+ writew(0x0000, spec->mem_base + 0x320);
+ tmp = FLOAT_THREE;
+ break;
+ case QUIRK_R3DI:
+ r3di_gpio_mic_set(codec, R3DI_REAR_MIC);
+ tmp = FLOAT_ONE;
+ break;
+ default:
+ tmp = FLOAT_ONE;
+ break;
+ }
+
+ chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000);
+ chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000);
+ if (spec->quirk == QUIRK_R3DI)
+ chipio_set_conn_rate(codec, 0x0F, SR_96_000);
+
+ dspio_set_uint_param(codec, 0x80, 0x00, tmp);
+
+ chipio_set_stream_control(codec, 0x03, 1);
+ chipio_set_stream_control(codec, 0x04, 1);
+
+ if (spec->quirk == QUIRK_SBZ) {
+ chipio_write(codec, 0x18B098, 0x0000000C);
+ chipio_write(codec, 0x18B09C, 0x0000000C);
+ }
+ ca0132_alt_mic_boost_set(codec, spec->mic_boost_enum_val);
+ break;
+ case REAR_LINE_IN:
+ ca0132_mic_boost_set(codec, 0);
+ switch (spec->quirk) {
+ case QUIRK_SBZ:
+ writew(0x0000, spec->mem_base + 0x320);
+ break;
+ case QUIRK_R3DI:
+ r3di_gpio_mic_set(codec, R3DI_REAR_MIC);
+ break;
+ }
+
+ chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000);
+ chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000);
+ if (spec->quirk == QUIRK_R3DI)
+ chipio_set_conn_rate(codec, 0x0F, SR_96_000);
+
+ tmp = FLOAT_ZERO;
+ dspio_set_uint_param(codec, 0x80, 0x00, tmp);
+
+ if (spec->quirk == QUIRK_SBZ) {
+ chipio_write(codec, 0x18B098, 0x00000000);
+ chipio_write(codec, 0x18B09C, 0x00000000);
+ }
+
+ chipio_set_stream_control(codec, 0x03, 1);
+ chipio_set_stream_control(codec, 0x04, 1);
+ break;
+ case FRONT_MIC:
+ switch (spec->quirk) {
+ case QUIRK_SBZ:
+ writew(0x0100, spec->mem_base + 0x320);
+ writew(0x0005, spec->mem_base + 0x320);
+ tmp = FLOAT_THREE;
+ break;
+ case QUIRK_R3DI:
+ r3di_gpio_mic_set(codec, R3DI_FRONT_MIC);
+ tmp = FLOAT_ONE;
+ break;
+ default:
+ tmp = FLOAT_ONE;
+ break;
+ }
+
+ chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000);
+ chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000);
+ if (spec->quirk == QUIRK_R3DI)
+ chipio_set_conn_rate(codec, 0x0F, SR_96_000);
+
+ dspio_set_uint_param(codec, 0x80, 0x00, tmp);
+
+ chipio_set_stream_control(codec, 0x03, 1);
+ chipio_set_stream_control(codec, 0x04, 1);
+
+ if (spec->quirk == QUIRK_SBZ) {
+ chipio_write(codec, 0x18B098, 0x0000000C);
+ chipio_write(codec, 0x18B09C, 0x000000CC);
+ }
+ ca0132_alt_mic_boost_set(codec, spec->mic_boost_enum_val);
+ break;
+ }
+ ca0132_cvoice_switch_set(codec);
+
+ snd_hda_power_down_pm(codec);
+ return 0;
+
+}
+
+/*
* Check if VNODE settings take effect immediately.
*/
static bool ca0132_is_vnode_effective(struct hda_codec *codec,
@@ -3418,7 +4439,7 @@ 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;
+ unsigned int on, tmp;
int num_fx = OUT_EFFECTS_COUNT + IN_EFFECTS_COUNT;
int err = 0;
int idx = nid - EFFECT_START_NID;
@@ -3442,6 +4463,46 @@ static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val)
/* Voice Focus applies to 2-ch Mic, Digital Mic */
if ((nid == VOICE_FOCUS) && (spec->cur_mic_type != DIGITAL_MIC))
val = 0;
+
+ /* If Voice Focus on SBZ, set to two channel. */
+ if ((nid == VOICE_FOCUS) && (spec->quirk == QUIRK_SBZ)
+ && (spec->cur_mic_type != REAR_LINE_IN)) {
+ if (spec->effects_switch[CRYSTAL_VOICE -
+ EFFECT_START_NID]) {
+
+ if (spec->effects_switch[VOICE_FOCUS -
+ EFFECT_START_NID]) {
+ tmp = FLOAT_TWO;
+ val = 1;
+ } else
+ tmp = FLOAT_ONE;
+
+ dspio_set_uint_param(codec, 0x80, 0x00, tmp);
+ }
+ }
+ /*
+ * For SBZ noise reduction, there's an extra command
+ * to module ID 0x47. No clue why.
+ */
+ if ((nid == NOISE_REDUCTION) && (spec->quirk == QUIRK_SBZ)
+ && (spec->cur_mic_type != REAR_LINE_IN)) {
+ if (spec->effects_switch[CRYSTAL_VOICE -
+ EFFECT_START_NID]) {
+ if (spec->effects_switch[NOISE_REDUCTION -
+ EFFECT_START_NID])
+ tmp = FLOAT_ONE;
+ else
+ tmp = FLOAT_ZERO;
+ } else
+ tmp = FLOAT_ZERO;
+
+ dspio_set_uint_param(codec, 0x47, 0x00, tmp);
+ }
+
+ /* If rear line in disable effects. */
+ if (spec->use_alt_functions &&
+ spec->in_enum_val == REAR_LINE_IN)
+ val = 0;
}
codec_dbg(codec, "ca0132_effect_set: nid=0x%x, val=%ld\n",
@@ -3469,6 +4530,9 @@ static int ca0132_pe_switch_set(struct hda_codec *codec)
codec_dbg(codec, "ca0132_pe_switch_set: val=%ld\n",
spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID]);
+ if (spec->use_alt_functions)
+ ca0132_alt_select_out(codec);
+
i = OUT_EFFECT_START_NID - EFFECT_START_NID;
nid = OUT_EFFECT_START_NID;
/* PE affects all out effects */
@@ -3526,7 +4590,10 @@ static int ca0132_cvoice_switch_set(struct hda_codec *codec)
/* set correct vipsource */
oldval = stop_mic1(codec);
- ret |= ca0132_set_vipsource(codec, 1);
+ if (spec->use_alt_functions)
+ ret |= ca0132_alt_set_vipsource(codec, 1);
+ else
+ ret |= ca0132_set_vipsource(codec, 1);
resume_mic1(codec, oldval);
return ret;
}
@@ -3546,6 +4613,16 @@ static int ca0132_mic_boost_set(struct hda_codec *codec, long val)
return ret;
}
+static int ca0132_alt_mic_boost_set(struct hda_codec *codec, long val)
+{
+ struct ca0132_spec *spec = codec->spec;
+ int ret = 0;
+
+ ret = snd_hda_codec_amp_update(codec, spec->input_pins[0], 0,
+ HDA_INPUT, 0, HDA_AMP_VOLMASK, val);
+ return ret;
+}
+
static int ca0132_vnode_switch_set(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
@@ -3560,8 +4637,12 @@ static int ca0132_vnode_switch_set(struct snd_kcontrol *kcontrol,
if (nid == VNID_HP_SEL) {
auto_jack =
spec->vnode_lswitch[VNID_HP_ASEL - VNODE_START_NID];
- if (!auto_jack)
- ca0132_select_out(codec);
+ if (!auto_jack) {
+ if (spec->use_alt_functions)
+ ca0132_alt_select_out(codec);
+ else
+ ca0132_select_out(codec);
+ }
return 1;
}
@@ -3574,7 +4655,10 @@ static int ca0132_vnode_switch_set(struct snd_kcontrol *kcontrol,
}
if (nid == VNID_HP_ASEL) {
- ca0132_select_out(codec);
+ if (spec->use_alt_functions)
+ ca0132_alt_select_out(codec);
+ else
+ ca0132_select_out(codec);
return 1;
}
@@ -3602,6 +4686,432 @@ static int ca0132_vnode_switch_set(struct snd_kcontrol *kcontrol,
return ret;
}
/* End of control change helpers. */
+/*
+ * 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
+ * Chromebook. I figured they were probably tuned specifically for it, and left
+ * out for a reason.
+ */
+
+/* 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)
+{
+ int i = 0;
+ unsigned int y;
+ /*
+ * For X_BASS, req 2 is actually crossover freq instead of
+ * effect level
+ */
+ if (nid == X_BASS)
+ y = 2;
+ else
+ y = 1;
+
+ snd_hda_power_up(codec);
+ if (nid == XBASS_XOVER) {
+ for (i = 0; i < OUT_EFFECTS_COUNT; i++)
+ if (ca0132_effects[i].nid == X_BASS)
+ break;
+
+ dspio_set_param(codec, ca0132_effects[i].mid, 0x20,
+ ca0132_effects[i].reqs[1],
+ &(lookup[idx - 1]), sizeof(unsigned int));
+ } else {
+ /* Find the actual effect structure */
+ for (i = 0; i < OUT_EFFECTS_COUNT; i++)
+ if (nid == ca0132_effects[i].nid)
+ break;
+
+ dspio_set_param(codec, ca0132_effects[i].mid, 0x20,
+ ca0132_effects[i].reqs[y],
+ &(lookup[idx]), sizeof(unsigned int));
+ }
+
+ snd_hda_power_down(codec);
+
+ return 0;
+}
+
+static int ca0132_alt_xbass_xover_slider_ctl_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;
+ long *valp = ucontrol->value.integer.value;
+
+ *valp = spec->xbass_xover_freq;
+ return 0;
+}
+
+static int ca0132_alt_slider_ctl_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;
+ hda_nid_t nid = get_amp_nid(kcontrol);
+ long *valp = ucontrol->value.integer.value;
+ int idx = nid - OUT_EFFECT_START_NID;
+
+ *valp = spec->fx_ctl_val[idx];
+ return 0;
+}
+
+/*
+ * The X-bass crossover starts at 10hz, so the min is 1. The
+ * frequency is set in multiples of 10.
+ */
+static int ca0132_alt_xbass_xover_slider_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 1;
+ uinfo->value.integer.max = 100;
+ uinfo->value.integer.step = 1;
+
+ return 0;
+}
+
+static int ca0132_alt_effect_slider_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ int chs = get_amp_channels(kcontrol);
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = chs == 3 ? 2 : 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 100;
+ uinfo->value.integer.step = 1;
+
+ return 0;
+}
+
+static int ca0132_alt_xbass_xover_slider_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;
+ hda_nid_t nid = get_amp_nid(kcontrol);
+ long *valp = ucontrol->value.integer.value;
+ int idx;
+
+ /* any change? */
+ if (spec->xbass_xover_freq == *valp)
+ return 0;
+
+ spec->xbass_xover_freq = *valp;
+
+ idx = *valp;
+ ca0132_alt_slider_ctl_set(codec, nid, float_xbass_xover_lookup, idx);
+
+ return 0;
+}
+
+static int ca0132_alt_effect_slider_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;
+ hda_nid_t nid = get_amp_nid(kcontrol);
+ long *valp = ucontrol->value.integer.value;
+ int idx;
+
+ idx = nid - EFFECT_START_NID;
+ /* any change? */
+ if (spec->fx_ctl_val[idx] == *valp)
+ return 0;
+
+ spec->fx_ctl_val[idx] = *valp;
+
+ idx = *valp;
+ ca0132_alt_slider_ctl_set(codec, nid, float_zero_to_one_lookup, idx);
+
+ return 0;
+}
+
+
+/*
+ * Mic Boost Enum for alternative ca0132 codecs. I didn't like that the original
+ * only has off or full 30 dB, and didn't like making a volume slider that has
+ * traditional 0-100 in alsamixer that goes in big steps. I like enum better.
+ */
+#define MIC_BOOST_NUM_OF_STEPS 4
+#define MIC_BOOST_ENUM_MAX_STRLEN 10
+
+static int ca0132_alt_mic_boost_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ char *sfx = "dB";
+ char namestr[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = MIC_BOOST_NUM_OF_STEPS;
+ if (uinfo->value.enumerated.item >= MIC_BOOST_NUM_OF_STEPS)
+ uinfo->value.enumerated.item = MIC_BOOST_NUM_OF_STEPS - 1;
+ sprintf(namestr, "%d %s", (uinfo->value.enumerated.item * 10), sfx);
+ strcpy(uinfo->value.enumerated.name, namestr);
+ return 0;
+}
+
+static int ca0132_alt_mic_boost_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->mic_boost_enum_val;
+ return 0;
+}
+
+static int ca0132_alt_mic_boost_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 = MIC_BOOST_NUM_OF_STEPS;
+
+ if (sel >= items)
+ return 0;
+
+ codec_dbg(codec, "ca0132_alt_mic_boost: boost=%d\n",
+ sel);
+
+ spec->mic_boost_enum_val = sel;
+
+ if (spec->in_enum_val != REAR_LINE_IN)
+ ca0132_alt_mic_boost_set(codec, spec->mic_boost_enum_val);
+
+ return 1;
+}
+
+
+/*
+ * Input Select Control for alternative ca0132 codecs. This exists because
+ * front microphone has no auto-detect, and we need a way to set the rear
+ * as line-in
+ */
+static int ca0132_alt_input_source_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = IN_SRC_NUM_OF_INPUTS;
+ if (uinfo->value.enumerated.item >= IN_SRC_NUM_OF_INPUTS)
+ uinfo->value.enumerated.item = IN_SRC_NUM_OF_INPUTS - 1;
+ strcpy(uinfo->value.enumerated.name,
+ in_src_str[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int ca0132_alt_input_source_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->in_enum_val;
+ return 0;
+}
+
+static int ca0132_alt_input_source_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 = IN_SRC_NUM_OF_INPUTS;
+
+ if (sel >= items)
+ return 0;
+
+ codec_dbg(codec, "ca0132_alt_input_select: sel=%d, preset=%s\n",
+ sel, in_src_str[sel]);
+
+ spec->in_enum_val = sel;
+
+ ca0132_alt_select_in(codec);
+
+ return 1;
+}
+
+/* Sound Blaster Z Output Select Control */
+static int ca0132_alt_output_select_get_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = NUM_OF_OUTPUTS;
+ 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);
+ return 0;
+}
+
+static int ca0132_alt_output_select_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->out_enum_val;
+ return 0;
+}
+
+static int ca0132_alt_output_select_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 = NUM_OF_OUTPUTS;
+ unsigned int auto_jack;
+
+ if (sel >= items)
+ return 0;
+
+ codec_dbg(codec, "ca0132_alt_output_select: sel=%d, preset=%s\n",
+ sel, alt_out_presets[sel].name);
+
+ spec->out_enum_val = sel;
+
+ auto_jack = spec->vnode_lswitch[VNID_HP_ASEL - VNODE_START_NID];
+
+ if (!auto_jack)
+ 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
+ * and night, disregard the slider value and have uneditable values.
+ */
+#define NUM_OF_SVM_SETTINGS 3
+static const char *out_svm_set_enum_str[3] = {"Normal", "Loud", "Night" };
+
+static int ca0132_alt_svm_setting_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = NUM_OF_SVM_SETTINGS;
+ if (uinfo->value.enumerated.item >= NUM_OF_SVM_SETTINGS)
+ uinfo->value.enumerated.item = NUM_OF_SVM_SETTINGS - 1;
+ strcpy(uinfo->value.enumerated.name,
+ out_svm_set_enum_str[uinfo->value.enumerated.item]);
+ return 0;
+}
+
+static int ca0132_alt_svm_setting_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->smart_volume_setting;
+ return 0;
+}
+
+static int ca0132_alt_svm_setting_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 = NUM_OF_SVM_SETTINGS;
+ unsigned int idx = SMART_VOLUME - EFFECT_START_NID;
+ unsigned int tmp;
+
+ if (sel >= items)
+ return 0;
+
+ codec_dbg(codec, "ca0132_alt_svm_setting: sel=%d, preset=%s\n",
+ sel, out_svm_set_enum_str[sel]);
+
+ spec->smart_volume_setting = sel;
+
+ switch (sel) {
+ case 0:
+ tmp = FLOAT_ZERO;
+ break;
+ case 1:
+ tmp = FLOAT_ONE;
+ break;
+ case 2:
+ tmp = FLOAT_TWO;
+ break;
+ default:
+ tmp = FLOAT_ZERO;
+ break;
+ }
+ /* Req 2 is the Smart Volume Setting req. */
+ dspio_set_uint_param(codec, ca0132_effects[idx].mid,
+ ca0132_effects[idx].reqs[2], tmp);
+ return 1;
+}
+
+/* Sound Blaster Z EQ preset controls */
+static int ca0132_alt_eq_preset_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ unsigned int items = ARRAY_SIZE(ca0132_alt_eq_presets);
+
+ 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,
+ ca0132_alt_eq_presets[uinfo->value.enumerated.item].name);
+ return 0;
+}
+
+static int ca0132_alt_eq_preset_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->eq_preset_val;
+ return 0;
+}
+
+static int ca0132_alt_eq_preset_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 i, err = 0;
+ int sel = ucontrol->value.enumerated.item[0];
+ unsigned int items = ARRAY_SIZE(ca0132_alt_eq_presets);
+
+ if (sel >= items)
+ return 0;
+
+ codec_dbg(codec, "%s: sel=%d, preset=%s\n", __func__, sel,
+ ca0132_alt_eq_presets[sel].name);
+ /*
+ * Idx 0 is default.
+ * Default needs to qualify with CrystalVoice state.
+ */
+ for (i = 0; i < EQ_PRESET_MAX_PARAM_COUNT; i++) {
+ err = dspio_set_uint_param(codec, ca0132_alt_eq_enum.mid,
+ ca0132_alt_eq_enum.reqs[i],
+ ca0132_alt_eq_presets[sel].vals[i]);
+ if (err < 0)
+ break;
+ }
+
+ if (err >= 0)
+ spec->eq_preset_val = sel;
+
+ return 1;
+}
static int ca0132_voicefx_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
@@ -3753,10 +5263,15 @@ static int ca0132_switch_put(struct snd_kcontrol *kcontrol,
/* mic boost */
if (nid == spec->input_pins[0]) {
spec->cur_mic_boost = *valp;
+ if (spec->use_alt_functions) {
+ if (spec->in_enum_val != REAR_LINE_IN)
+ changed = ca0132_mic_boost_set(codec, *valp);
+ } else {
+ /* Mic boost does not apply to Digital Mic */
+ if (spec->cur_mic_type != DIGITAL_MIC)
+ changed = ca0132_mic_boost_set(codec, *valp);
+ }
- /* Mic boost does not apply to Digital Mic */
- if (spec->cur_mic_type != DIGITAL_MIC)
- changed = ca0132_mic_boost_set(codec, *valp);
goto exit;
}
@@ -3768,6 +5283,41 @@ static int ca0132_switch_put(struct snd_kcontrol *kcontrol,
/*
* Volume related
*/
+/*
+ * Sets the internal DSP decibel level to match the DAC for output, and the
+ * ADC for input. Currently only the SBZ sets dsp capture volume level, and
+ * all alternative codecs set DSP playback volume.
+ */
+static void ca0132_alt_dsp_volume_put(struct hda_codec *codec, hda_nid_t nid)
+{
+ struct ca0132_spec *spec = codec->spec;
+ unsigned int dsp_dir;
+ unsigned int lookup_val;
+
+ if (nid == VNID_SPK)
+ dsp_dir = DSP_VOL_OUT;
+ else
+ dsp_dir = DSP_VOL_IN;
+
+ lookup_val = spec->vnode_lvol[nid - VNODE_START_NID];
+
+ dspio_set_uint_param(codec,
+ ca0132_alt_vol_ctls[dsp_dir].mid,
+ ca0132_alt_vol_ctls[dsp_dir].reqs[0],
+ float_vol_db_lookup[lookup_val]);
+
+ lookup_val = spec->vnode_rvol[nid - VNODE_START_NID];
+
+ dspio_set_uint_param(codec,
+ ca0132_alt_vol_ctls[dsp_dir].mid,
+ ca0132_alt_vol_ctls[dsp_dir].reqs[1],
+ float_vol_db_lookup[lookup_val]);
+
+ dspio_set_uint_param(codec,
+ ca0132_alt_vol_ctls[dsp_dir].mid,
+ ca0132_alt_vol_ctls[dsp_dir].reqs[2], FLOAT_ZERO);
+}
+
static int ca0132_volume_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
@@ -3869,6 +5419,51 @@ static int ca0132_volume_put(struct snd_kcontrol *kcontrol,
return changed;
}
+/*
+ * This function is the same as the one above, because using an if statement
+ * inside of the above volume control for the DSP volume would cause too much
+ * lag. This is a lot more smooth.
+ */
+static int ca0132_alt_volume_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;
+ hda_nid_t nid = get_amp_nid(kcontrol);
+ int ch = get_amp_channels(kcontrol);
+ long *valp = ucontrol->value.integer.value;
+ hda_nid_t vnid = 0;
+ int changed = 1;
+
+ switch (nid) {
+ case 0x02:
+ vnid = VNID_SPK;
+ break;
+ case 0x07:
+ vnid = VNID_MIC;
+ break;
+ }
+
+ /* store the left and right volume */
+ if (ch & 1) {
+ spec->vnode_lvol[vnid - VNODE_START_NID] = *valp;
+ valp++;
+ }
+ if (ch & 2) {
+ spec->vnode_rvol[vnid - VNODE_START_NID] = *valp;
+ valp++;
+ }
+
+ snd_hda_power_up(codec);
+ ca0132_alt_dsp_volume_put(codec, vnid);
+ mutex_lock(&codec->control_mutex);
+ changed = snd_hda_mixer_amp_volume_put(kcontrol, ucontrol);
+ mutex_unlock(&codec->control_mutex);
+ snd_hda_power_down(codec);
+
+ return changed;
+}
+
static int ca0132_volume_tlv(struct snd_kcontrol *kcontrol, int op_flag,
unsigned int size, unsigned int __user *tlv)
{
@@ -3907,14 +5502,61 @@ static int ca0132_volume_tlv(struct snd_kcontrol *kcontrol, int op_flag,
return err;
}
+/* Add volume slider control for effect level */
+static int ca0132_alt_add_effect_slider(struct hda_codec *codec, hda_nid_t nid,
+ const char *pfx, int dir)
+{
+ char *fx = "FX:";
+ char namestr[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
+ int type = dir ? HDA_INPUT : HDA_OUTPUT;
+ struct snd_kcontrol_new knew =
+ HDA_CODEC_VOLUME_MONO(namestr, nid, 1, 0, type);
+
+ sprintf(namestr, "%s %s %s Volume", fx, pfx, dirstr[dir]);
+
+ knew.tlv.c = 0;
+ knew.tlv.p = 0;
+
+ switch (nid) {
+ case XBASS_XOVER:
+ 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;
+ break;
+ default:
+ knew.info = ca0132_alt_effect_slider_info;
+ knew.get = ca0132_alt_slider_ctl_get;
+ knew.put = ca0132_alt_effect_slider_put;
+ knew.private_value =
+ HDA_COMPOSE_AMP_VAL(nid, 1, 0, type);
+ break;
+ }
+
+ return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec));
+}
+
+/*
+ * Added FX: prefix for the alternative codecs, because otherwise the surround
+ * effect would conflict with the Surround sound volume control. Also seems more
+ * clear as to what the switches do. Left alone for others.
+ */
static int add_fx_switch(struct hda_codec *codec, hda_nid_t nid,
const char *pfx, int dir)
{
+ struct ca0132_spec *spec = codec->spec;
+ char *fx = "FX:";
char namestr[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
int type = dir ? HDA_INPUT : HDA_OUTPUT;
struct snd_kcontrol_new knew =
CA0132_CODEC_MUTE_MONO(namestr, nid, 1, type);
- sprintf(namestr, "%s %s Switch", pfx, dirstr[dir]);
+ /* If using alt_controls, add FX: prefix. But, don't add FX:
+ * prefix to OutFX or InFX enable controls.
+ */
+ if ((spec->use_alt_controls) && (nid <= IN_EFFECT_END_NID))
+ sprintf(namestr, "%s %s %s Switch", fx, pfx, dirstr[dir]);
+ else
+ sprintf(namestr, "%s %s Switch", pfx, dirstr[dir]);
+
return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec));
}
@@ -3929,6 +5571,136 @@ static int add_voicefx(struct hda_codec *codec)
return snd_hda_ctl_add(codec, VOICEFX, snd_ctl_new1(&knew, codec));
}
+/* Create the EQ Preset control */
+static int add_ca0132_alt_eq_presets(struct hda_codec *codec)
+{
+ struct snd_kcontrol_new knew =
+ HDA_CODEC_MUTE_MONO(ca0132_alt_eq_enum.name,
+ EQ_PRESET_ENUM, 1, 0, HDA_OUTPUT);
+ knew.info = ca0132_alt_eq_preset_info;
+ knew.get = ca0132_alt_eq_preset_get;
+ knew.put = ca0132_alt_eq_preset_put;
+ return snd_hda_ctl_add(codec, EQ_PRESET_ENUM,
+ snd_ctl_new1(&knew, codec));
+}
+
+/*
+ * Add enumerated control for the three different settings of the smart volume
+ * output effect. Normal just uses the slider value, and loud and night are
+ * their own things that ignore that value.
+ */
+static int ca0132_alt_add_svm_enum(struct hda_codec *codec)
+{
+ struct snd_kcontrol_new knew =
+ HDA_CODEC_MUTE_MONO("FX: Smart Volume Setting",
+ SMART_VOLUME_ENUM, 1, 0, HDA_OUTPUT);
+ knew.info = ca0132_alt_svm_setting_info;
+ knew.get = ca0132_alt_svm_setting_get;
+ knew.put = ca0132_alt_svm_setting_put;
+ return snd_hda_ctl_add(codec, SMART_VOLUME_ENUM,
+ snd_ctl_new1(&knew, codec));
+
+}
+
+/*
+ * Create an Output Select enumerated control for codecs with surround
+ * out capabilities.
+ */
+static int ca0132_alt_add_output_enum(struct hda_codec *codec)
+{
+ struct snd_kcontrol_new knew =
+ HDA_CODEC_MUTE_MONO("Output Select",
+ OUTPUT_SOURCE_ENUM, 1, 0, HDA_OUTPUT);
+ knew.info = ca0132_alt_output_select_get_info;
+ knew.get = ca0132_alt_output_select_get;
+ knew.put = ca0132_alt_output_select_put;
+ return snd_hda_ctl_add(codec, OUTPUT_SOURCE_ENUM,
+ 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.
+ */
+static int ca0132_alt_add_input_enum(struct hda_codec *codec)
+{
+ struct snd_kcontrol_new knew =
+ HDA_CODEC_MUTE_MONO("Input Source",
+ INPUT_SOURCE_ENUM, 1, 0, HDA_INPUT);
+ knew.info = ca0132_alt_input_source_info;
+ knew.get = ca0132_alt_input_source_get;
+ knew.put = ca0132_alt_input_source_put;
+ return snd_hda_ctl_add(codec, INPUT_SOURCE_ENUM,
+ snd_ctl_new1(&knew, codec));
+}
+
+/*
+ * Add mic boost enumerated control. Switches through 0dB to 30dB. This adds
+ * more control than the original mic boost, which is either full 30dB or off.
+ */
+static int ca0132_alt_add_mic_boost_enum(struct hda_codec *codec)
+{
+ struct snd_kcontrol_new knew =
+ HDA_CODEC_MUTE_MONO("Mic Boost Capture Switch",
+ MIC_BOOST_ENUM, 1, 0, HDA_INPUT);
+ knew.info = ca0132_alt_mic_boost_info;
+ knew.get = ca0132_alt_mic_boost_get;
+ knew.put = ca0132_alt_mic_boost_put;
+ return snd_hda_ctl_add(codec, MIC_BOOST_ENUM,
+ snd_ctl_new1(&knew, codec));
+
+}
+
+/*
+ * Need to create slave controls for the alternate codecs that have surround
+ * capabilities.
+ */
+static const char * const ca0132_alt_slave_pfxs[] = {
+ "Front", "Surround", "Center", "LFE", NULL,
+};
+
+/*
+ * Also need special channel map, because the default one is incorrect.
+ * I think this has to do with the pin for rear surround being 0x11,
+ * and the center/lfe being 0x10. Usually the pin order is the opposite.
+ */
+const struct snd_pcm_chmap_elem ca0132_alt_chmaps[] = {
+ { .channels = 2,
+ .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR } },
+ { .channels = 4,
+ .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR,
+ SNDRV_CHMAP_RL, SNDRV_CHMAP_RR } },
+ { .channels = 6,
+ .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR,
+ SNDRV_CHMAP_FC, SNDRV_CHMAP_LFE,
+ SNDRV_CHMAP_RL, SNDRV_CHMAP_RR } },
+ { }
+};
+
+/* Add the correct chmap for streams with 6 channels. */
+static void ca0132_alt_add_chmap_ctls(struct hda_codec *codec)
+{
+ int err = 0;
+ struct hda_pcm *pcm;
+
+ list_for_each_entry(pcm, &codec->pcm_list_head, list) {
+ struct hda_pcm_stream *hinfo =
+ &pcm->stream[SNDRV_PCM_STREAM_PLAYBACK];
+ struct snd_pcm_chmap *chmap;
+ const struct snd_pcm_chmap_elem *elem;
+
+ elem = ca0132_alt_chmaps;
+ if (hinfo->channels_max == 6) {
+ err = snd_pcm_add_chmap_ctls(pcm->pcm,
+ SNDRV_PCM_STREAM_PLAYBACK,
+ elem, hinfo->channels_max, 0, &chmap);
+ if (err < 0)
+ codec_dbg(codec, "snd_pcm_add_chmap_ctls failed!");
+ }
+ }
+}
+
/*
* When changing Node IDs for Mixer Controls below, make sure to update
* Node IDs in ca0132_config() as well.
@@ -3955,10 +5727,55 @@ static struct snd_kcontrol_new ca0132_mixer[] = {
{ } /* end */
};
+/*
+ * SBZ specific control mixer. Removes auto-detect for mic, and adds surround
+ * controls. Also sets both the Front Playback and Capture Volume controls to
+ * alt so they set the DSP's decibel level.
+ */
+static struct snd_kcontrol_new sbz_mixer[] = {
+ CA0132_ALT_CODEC_VOL("Front Playback Volume", 0x02, HDA_OUTPUT),
+ CA0132_CODEC_MUTE("Front Playback Switch", VNID_SPK, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x04, 0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Surround Playback Switch", 0x04, 0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x03, 1, 0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x03, 1, 0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x03, 2, 0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x03, 2, 0, HDA_OUTPUT),
+ CA0132_ALT_CODEC_VOL("Capture Volume", 0x07, HDA_INPUT),
+ CA0132_CODEC_MUTE("Capture Switch", VNID_MIC, HDA_INPUT),
+ HDA_CODEC_VOLUME("What U Hear Capture Volume", 0x0a, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("What U Hear Capture Switch", 0x0a, 0, HDA_INPUT),
+ CA0132_CODEC_MUTE_MONO("HP/Speaker Auto Detect Playback Switch",
+ VNID_HP_ASEL, 1, HDA_OUTPUT),
+ { } /* end */
+};
+
+/*
+ * Same as the Sound Blaster Z, except doesn't use the alt volume for capture
+ * because it doesn't set decibel levels for the DSP for capture.
+ */
+static struct snd_kcontrol_new r3di_mixer[] = {
+ CA0132_ALT_CODEC_VOL("Front Playback Volume", 0x02, HDA_OUTPUT),
+ CA0132_CODEC_MUTE("Front Playback Switch", VNID_SPK, HDA_OUTPUT),
+ HDA_CODEC_VOLUME("Surround Playback Volume", 0x04, 0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("Surround Playback Switch", 0x04, 0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x03, 1, 0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x03, 1, 0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x03, 2, 0, HDA_OUTPUT),
+ HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x03, 2, 0, HDA_OUTPUT),
+ CA0132_CODEC_VOL("Capture Volume", VNID_MIC, HDA_INPUT),
+ CA0132_CODEC_MUTE("Capture Switch", VNID_MIC, HDA_INPUT),
+ HDA_CODEC_VOLUME("What U Hear Capture Volume", 0x0a, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("What U Hear Capture Switch", 0x0a, 0, HDA_INPUT),
+ CA0132_CODEC_MUTE_MONO("HP/Speaker Auto Detect Playback Switch",
+ VNID_HP_ASEL, 1, HDA_OUTPUT),
+ { } /* end */
+};
+
static int ca0132_build_controls(struct hda_codec *codec)
{
struct ca0132_spec *spec = codec->spec;
- int i, num_fx;
+ int i, num_fx, num_sliders;
int err = 0;
/* Add Mixer controls */
@@ -3967,29 +5784,94 @@ static int ca0132_build_controls(struct hda_codec *codec)
if (err < 0)
return err;
}
+ /* Setup vmaster with surround slaves for desktop ca0132 devices */
+ if (spec->use_alt_functions) {
+ snd_hda_set_vmaster_tlv(codec, spec->dacs[0], HDA_OUTPUT,
+ spec->tlv);
+ snd_hda_add_vmaster(codec, "Master Playback Volume",
+ spec->tlv, ca0132_alt_slave_pfxs,
+ "Playback Volume");
+ err = __snd_hda_add_vmaster(codec, "Master Playback Switch",
+ NULL, ca0132_alt_slave_pfxs,
+ "Playback Switch",
+ true, &spec->vmaster_mute.sw_kctl);
+
+ }
/* Add in and out effects controls.
* VoiceFX, PE and CrystalVoice are added separately.
*/
num_fx = OUT_EFFECTS_COUNT + IN_EFFECTS_COUNT;
for (i = 0; i < num_fx; i++) {
+ /* SBZ breaks if Echo Cancellation is used */
+ if (spec->quirk == QUIRK_SBZ) {
+ if (i == (ECHO_CANCELLATION - IN_EFFECT_START_NID +
+ OUT_EFFECTS_COUNT))
+ continue;
+ }
+
err = add_fx_switch(codec, ca0132_effects[i].nid,
ca0132_effects[i].name,
ca0132_effects[i].direct);
if (err < 0)
return err;
}
+ /*
+ * If codec has use_alt_controls set to true, add effect level sliders,
+ * EQ presets, and Smart Volume presets. Also, change names to add FX
+ * prefix, and change PlayEnhancement and CrystalVoice to match.
+ */
+ if (spec->use_alt_controls) {
+ ca0132_alt_add_svm_enum(codec);
+ add_ca0132_alt_eq_presets(codec);
+ err = add_fx_switch(codec, PLAY_ENHANCEMENT,
+ "Enable OutFX", 0);
+ if (err < 0)
+ return err;
- err = add_fx_switch(codec, PLAY_ENHANCEMENT, "PlayEnhancement", 0);
- if (err < 0)
- return err;
+ err = add_fx_switch(codec, CRYSTAL_VOICE,
+ "Enable InFX", 1);
+ if (err < 0)
+ return err;
- err = add_fx_switch(codec, CRYSTAL_VOICE, "CrystalVoice", 1);
- if (err < 0)
- return err;
+ num_sliders = OUT_EFFECTS_COUNT - 1;
+ for (i = 0; i < num_sliders; i++) {
+ err = ca0132_alt_add_effect_slider(codec,
+ ca0132_effects[i].nid,
+ ca0132_effects[i].name,
+ ca0132_effects[i].direct);
+ if (err < 0)
+ return err;
+ }
+ err = ca0132_alt_add_effect_slider(codec, XBASS_XOVER,
+ "X-Bass Crossover", EFX_DIR_OUT);
+
+ if (err < 0)
+ return err;
+ } else {
+ err = add_fx_switch(codec, PLAY_ENHANCEMENT,
+ "PlayEnhancement", 0);
+ if (err < 0)
+ return err;
+
+ err = add_fx_switch(codec, CRYSTAL_VOICE,
+ "CrystalVoice", 1);
+ if (err < 0)
+ return err;
+ }
add_voicefx(codec);
+ /*
+ * If the codec uses alt_functions, you need the enumerated controls
+ * to select the new outputs and inputs, plus add the new mic boost
+ * setting control.
+ */
+ if (spec->use_alt_functions) {
+ ca0132_alt_add_output_enum(codec);
+ ca0132_alt_add_input_enum(codec);
+ ca0132_alt_add_mic_boost_enum(codec);
+ }
#ifdef ENABLE_TUNING_CONTROLS
add_tuning_ctls(codec);
#endif
@@ -4014,6 +5896,10 @@ static int ca0132_build_controls(struct hda_codec *codec)
if (err < 0)
return err;
}
+
+ if (spec->use_alt_functions)
+ ca0132_alt_add_chmap_ctls(codec);
+
return 0;
}
@@ -4068,6 +5954,11 @@ static int ca0132_build_pcms(struct hda_codec *codec)
info = snd_hda_codec_pcm_new(codec, "CA0132 Analog");
if (!info)
return -ENOMEM;
+ if (spec->use_alt_functions) {
+ info->own_chmap = true;
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK].chmap
+ = ca0132_alt_chmaps;
+ }
info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ca0132_pcm_analog_playback;
info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->dacs[0];
info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max =
@@ -4076,12 +5967,16 @@ static int ca0132_build_pcms(struct hda_codec *codec)
info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = 1;
info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adcs[0];
- info = snd_hda_codec_pcm_new(codec, "CA0132 Analog Mic-In2");
- if (!info)
- return -ENOMEM;
- info->stream[SNDRV_PCM_STREAM_CAPTURE] = ca0132_pcm_analog_capture;
- info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = 1;
- info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adcs[1];
+ /* With the DSP enabled, desktops don't use this ADC. */
+ if (spec->use_alt_functions) {
+ info = snd_hda_codec_pcm_new(codec, "CA0132 Analog Mic-In2");
+ if (!info)
+ return -ENOMEM;
+ info->stream[SNDRV_PCM_STREAM_CAPTURE] =
+ ca0132_pcm_analog_capture;
+ info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = 1;
+ info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adcs[1];
+ }
info = snd_hda_codec_pcm_new(codec, "CA0132 What U Hear");
if (!info)
@@ -4288,6 +6183,196 @@ static void ca0132_refresh_widget_caps(struct hda_codec *codec)
}
/*
+ * Recon3Di r3di_setup_defaults sub functions.
+ */
+
+static void r3di_dsp_scp_startup(struct hda_codec *codec)
+{
+ unsigned int tmp;
+
+ tmp = 0x00000000;
+ dspio_set_uint_param_no_source(codec, 0x80, 0x0A, tmp);
+
+ tmp = 0x00000001;
+ dspio_set_uint_param_no_source(codec, 0x80, 0x0B, tmp);
+
+ tmp = 0x00000004;
+ dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
+
+ tmp = 0x00000005;
+ dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
+
+ tmp = 0x00000000;
+ dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
+
+}
+
+static void r3di_dsp_initial_mic_setup(struct hda_codec *codec)
+{
+ unsigned int tmp;
+
+ /* Mic 1 Setup */
+ chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000);
+ chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000);
+ /* This ConnPointID is unique to Recon3Di. Haven't seen it elsewhere */
+ chipio_set_conn_rate(codec, 0x0F, SR_96_000);
+ tmp = FLOAT_ONE;
+ dspio_set_uint_param(codec, 0x80, 0x00, tmp);
+
+ /* Mic 2 Setup, even though it isn't connected on SBZ */
+ chipio_set_conn_rate(codec, MEM_CONNID_MICIN2, SR_96_000);
+ chipio_set_conn_rate(codec, MEM_CONNID_MICOUT2, SR_96_000);
+ chipio_set_conn_rate(codec, 0x0F, SR_96_000);
+ tmp = FLOAT_ZERO;
+ dspio_set_uint_param(codec, 0x80, 0x01, tmp);
+}
+
+/*
+ * Initialize Sound Blaster Z analog microphones.
+ */
+static void sbz_init_analog_mics(struct hda_codec *codec)
+{
+ unsigned int tmp;
+
+ /* Mic 1 Setup */
+ chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000);
+ chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000);
+ tmp = FLOAT_THREE;
+ dspio_set_uint_param(codec, 0x80, 0x00, tmp);
+
+ /* Mic 2 Setup, even though it isn't connected on SBZ */
+ chipio_set_conn_rate(codec, MEM_CONNID_MICIN2, SR_96_000);
+ chipio_set_conn_rate(codec, MEM_CONNID_MICOUT2, SR_96_000);
+ tmp = FLOAT_ZERO;
+ dspio_set_uint_param(codec, 0x80, 0x01, tmp);
+
+}
+
+/*
+ * Sets the source of stream 0x14 to connpointID 0x48, and the destination
+ * connpointID to 0x91. If this isn't done, the destination is 0x71, and
+ * you get no sound. I'm guessing this has to do with the Sound Blaster Z
+ * having an updated DAC, which changes the destination to that DAC.
+ */
+static void sbz_connect_streams(struct hda_codec *codec)
+{
+ struct ca0132_spec *spec = codec->spec;
+
+ mutex_lock(&spec->chipio_mutex);
+
+ codec_dbg(codec, "Connect Streams entered, mutex locked and loaded.\n");
+
+ chipio_set_stream_channels(codec, 0x0C, 6);
+ chipio_set_stream_control(codec, 0x0C, 1);
+
+ /* This value is 0x43 for 96khz, and 0x83 for 192khz. */
+ chipio_write_no_mutex(codec, 0x18a020, 0x00000043);
+
+ /* Setup stream 0x14 with it's source and destination points */
+ chipio_set_stream_source_dest(codec, 0x14, 0x48, 0x91);
+ chipio_set_conn_rate_no_mutex(codec, 0x48, SR_96_000);
+ chipio_set_conn_rate_no_mutex(codec, 0x91, SR_96_000);
+ chipio_set_stream_channels(codec, 0x14, 2);
+ chipio_set_stream_control(codec, 0x14, 1);
+
+ codec_dbg(codec, "Connect Streams exited, mutex released.\n");
+
+ mutex_unlock(&spec->chipio_mutex);
+
+}
+
+/*
+ * Write data through ChipIO to setup proper stream destinations.
+ * Not sure how it exactly works, but it seems to direct data
+ * to different destinations. Example is f8 to c0, e0 to c0.
+ * All I know is, if you don't set these, you get no sound.
+ */
+static void sbz_chipio_startup_data(struct hda_codec *codec)
+{
+ struct ca0132_spec *spec = codec->spec;
+
+ mutex_lock(&spec->chipio_mutex);
+ codec_dbg(codec, "Startup Data entered, mutex locked and loaded.\n");
+
+ /* These control audio output */
+ chipio_write_no_mutex(codec, 0x190060, 0x0001f8c0);
+ chipio_write_no_mutex(codec, 0x190064, 0x0001f9c1);
+ chipio_write_no_mutex(codec, 0x190068, 0x0001fac6);
+ chipio_write_no_mutex(codec, 0x19006c, 0x0001fbc7);
+ /* Signal to update I think */
+ chipio_write_no_mutex(codec, 0x19042c, 0x00000001);
+
+ chipio_set_stream_channels(codec, 0x0C, 6);
+ chipio_set_stream_control(codec, 0x0C, 1);
+ /* No clue what these control */
+ chipio_write_no_mutex(codec, 0x190030, 0x0001e0c0);
+ chipio_write_no_mutex(codec, 0x190034, 0x0001e1c1);
+ chipio_write_no_mutex(codec, 0x190038, 0x0001e4c2);
+ chipio_write_no_mutex(codec, 0x19003c, 0x0001e5c3);
+ chipio_write_no_mutex(codec, 0x190040, 0x0001e2c4);
+ chipio_write_no_mutex(codec, 0x190044, 0x0001e3c5);
+ chipio_write_no_mutex(codec, 0x190048, 0x0001e8c6);
+ chipio_write_no_mutex(codec, 0x19004c, 0x0001e9c7);
+ chipio_write_no_mutex(codec, 0x190050, 0x0001ecc8);
+ chipio_write_no_mutex(codec, 0x190054, 0x0001edc9);
+ chipio_write_no_mutex(codec, 0x190058, 0x0001eaca);
+ chipio_write_no_mutex(codec, 0x19005c, 0x0001ebcb);
+
+ chipio_write_no_mutex(codec, 0x19042c, 0x00000001);
+
+ codec_dbg(codec, "Startup Data exited, mutex released.\n");
+ mutex_unlock(&spec->chipio_mutex);
+}
+
+/*
+ * Sound Blaster Z uses these after DSP is loaded. Weird SCP commands
+ * without a 0x20 source like normal.
+ */
+static void sbz_dsp_scp_startup(struct hda_codec *codec)
+{
+ unsigned int tmp;
+
+ tmp = 0x00000003;
+ dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
+
+ tmp = 0x00000000;
+ dspio_set_uint_param_no_source(codec, 0x80, 0x0A, tmp);
+
+ tmp = 0x00000001;
+ dspio_set_uint_param_no_source(codec, 0x80, 0x0B, tmp);
+
+ tmp = 0x00000004;
+ dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
+
+ tmp = 0x00000005;
+ dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
+
+ tmp = 0x00000000;
+ dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
+
+}
+
+static void sbz_dsp_initial_mic_setup(struct hda_codec *codec)
+{
+ unsigned int tmp;
+
+ chipio_set_stream_control(codec, 0x03, 0);
+ chipio_set_stream_control(codec, 0x04, 0);
+
+ chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000);
+ chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000);
+
+ tmp = FLOAT_THREE;
+ dspio_set_uint_param(codec, 0x80, 0x00, tmp);
+
+ chipio_set_stream_control(codec, 0x03, 1);
+ chipio_set_stream_control(codec, 0x04, 1);
+
+ chipio_write(codec, 0x18b098, 0x0000000c);
+ chipio_write(codec, 0x18b09C, 0x0000000c);
+}
+
+/*
* Setup default parameters for DSP
*/
static void ca0132_setup_defaults(struct hda_codec *codec)
@@ -4332,16 +6417,159 @@ static void ca0132_setup_defaults(struct hda_codec *codec)
}
/*
+ * Setup default parameters for Recon3Di DSP.
+ */
+
+static void r3di_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;
+
+ r3di_dsp_scp_startup(codec);
+
+ r3di_dsp_initial_mic_setup(codec);
+
+ /*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);
+
+ r3di_gpio_dsp_status_set(codec, R3DI_DSP_DOWNLOADED);
+
+ /* Setup effect defaults */
+ 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]);
+ }
+ }
+
+}
+
+/*
+ * Setup default parameters for the Sound Blaster Z DSP. A lot more going on
+ * than the Chromebook setup.
+ */
+static void sbz_setup_defaults(struct hda_codec *codec)
+{
+ struct ca0132_spec *spec = codec->spec;
+ unsigned int tmp, stream_format;
+ int num_fx;
+ int idx, i;
+
+ if (spec->dsp_state != DSP_DOWNLOADED)
+ return;
+
+ sbz_dsp_scp_startup(codec);
+
+ sbz_init_analog_mics(codec);
+
+ sbz_connect_streams(codec);
+
+ sbz_chipio_startup_data(codec);
+
+ chipio_set_stream_control(codec, 0x03, 1);
+ chipio_set_stream_control(codec, 0x04, 1);
+
+ /*
+ * Sets internal input loopback to off, used to have a switch to
+ * enable input loopback, but turned out to be way too buggy.
+ */
+ 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);
+
+ sbz_dsp_initial_mic_setup(codec);
+
+
+ /* 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]);
+ }
+ }
+
+ /*
+ * Have to make a stream to bind the sound output to, otherwise
+ * you'll get dead audio. Before I did this, it would bind to an
+ * audio input, and would never work
+ */
+ stream_format = snd_hdac_calc_stream_format(48000, 2,
+ SNDRV_PCM_FORMAT_S32_LE, 32, 0);
+
+ snd_hda_codec_setup_stream(codec, spec->dacs[0], spec->dsp_stream_id,
+ 0, stream_format);
+
+ snd_hda_codec_cleanup_stream(codec, spec->dacs[0]);
+
+ snd_hda_codec_setup_stream(codec, spec->dacs[0], spec->dsp_stream_id,
+ 0, stream_format);
+
+ snd_hda_codec_cleanup_stream(codec, spec->dacs[0]);
+}
+
+/*
* Initialization of flags in chip
*/
static void ca0132_init_flags(struct hda_codec *codec)
{
- chipio_set_control_flag(codec, CONTROL_FLAG_IDLE_ENABLE, 0);
- chipio_set_control_flag(codec, CONTROL_FLAG_PORT_A_COMMON_MODE, 0);
- chipio_set_control_flag(codec, CONTROL_FLAG_PORT_D_COMMON_MODE, 0);
- chipio_set_control_flag(codec, CONTROL_FLAG_PORT_A_10KOHM_LOAD, 0);
- chipio_set_control_flag(codec, CONTROL_FLAG_PORT_D_10KOHM_LOAD, 0);
- chipio_set_control_flag(codec, CONTROL_FLAG_ADC_C_HIGH_PASS, 1);
+ struct ca0132_spec *spec = codec->spec;
+
+ if (spec->use_alt_functions) {
+ chipio_set_control_flag(codec, CONTROL_FLAG_DSP_96KHZ, 1);
+ chipio_set_control_flag(codec, CONTROL_FLAG_DAC_96KHZ, 1);
+ chipio_set_control_flag(codec, CONTROL_FLAG_ADC_B_96KHZ, 1);
+ chipio_set_control_flag(codec, CONTROL_FLAG_ADC_C_96KHZ, 1);
+ chipio_set_control_flag(codec, CONTROL_FLAG_SRC_RATE_96KHZ, 1);
+ chipio_set_control_flag(codec, CONTROL_FLAG_IDLE_ENABLE, 0);
+ chipio_set_control_flag(codec, CONTROL_FLAG_SPDIF2OUT, 0);
+ chipio_set_control_flag(codec,
+ CONTROL_FLAG_PORT_D_10KOHM_LOAD, 0);
+ chipio_set_control_flag(codec,
+ CONTROL_FLAG_PORT_A_10KOHM_LOAD, 1);
+ } else {
+ chipio_set_control_flag(codec, CONTROL_FLAG_IDLE_ENABLE, 0);
+ chipio_set_control_flag(codec,
+ CONTROL_FLAG_PORT_A_COMMON_MODE, 0);
+ chipio_set_control_flag(codec,
+ CONTROL_FLAG_PORT_D_COMMON_MODE, 0);
+ chipio_set_control_flag(codec,
+ CONTROL_FLAG_PORT_A_10KOHM_LOAD, 0);
+ chipio_set_control_flag(codec,
+ CONTROL_FLAG_PORT_D_10KOHM_LOAD, 0);
+ chipio_set_control_flag(codec, CONTROL_FLAG_ADC_C_HIGH_PASS, 1);
+ }
}
/*
@@ -4349,6 +6577,16 @@ static void ca0132_init_flags(struct hda_codec *codec)
*/
static void ca0132_init_params(struct hda_codec *codec)
{
+ struct ca0132_spec *spec = codec->spec;
+
+ if (spec->use_alt_functions) {
+ chipio_set_conn_rate(codec, MEM_CONNID_WUH, SR_48_000);
+ chipio_set_conn_rate(codec, 0x0B, SR_48_000);
+ chipio_set_control_param(codec, CONTROL_PARAM_SPDIF1_SOURCE, 0);
+ chipio_set_control_param(codec, 0, 0);
+ chipio_set_control_param(codec, CONTROL_PARAM_VIP_SOURCE, 0);
+ }
+
chipio_set_control_param(codec, CONTROL_PARAM_PORTA_160OHM_GAIN, 6);
chipio_set_control_param(codec, CONTROL_PARAM_PORTD_160OHM_GAIN, 6);
}
@@ -4370,11 +6608,49 @@ static void ca0132_set_dsp_msr(struct hda_codec *codec, bool is96k)
static bool ca0132_download_dsp_images(struct hda_codec *codec)
{
bool dsp_loaded = false;
+ struct ca0132_spec *spec = codec->spec;
const struct dsp_image_seg *dsp_os_image;
const struct firmware *fw_entry;
-
- if (request_firmware(&fw_entry, EFX_FILE, codec->card->dev) != 0)
- return false;
+ /*
+ * Alternate firmwares for different variants. The Recon3Di apparently
+ * can use the default firmware, but I'll leave the option in case
+ * it needs it again.
+ */
+ switch (spec->quirk) {
+ case QUIRK_SBZ:
+ if (request_firmware(&fw_entry, SBZ_EFX_FILE,
+ codec->card->dev) != 0) {
+ codec_dbg(codec, "SBZ alt firmware not detected. ");
+ spec->alt_firmware_present = false;
+ } else {
+ codec_dbg(codec, "Sound Blaster Z firmware selected.");
+ spec->alt_firmware_present = true;
+ }
+ break;
+ case QUIRK_R3DI:
+ if (request_firmware(&fw_entry, R3DI_EFX_FILE,
+ codec->card->dev) != 0) {
+ codec_dbg(codec, "Recon3Di alt firmware not detected.");
+ spec->alt_firmware_present = false;
+ } else {
+ codec_dbg(codec, "Recon3Di firmware selected.");
+ spec->alt_firmware_present = true;
+ }
+ break;
+ default:
+ spec->alt_firmware_present = false;
+ break;
+ }
+ /*
+ * Use default ctefx.bin if no alt firmware is detected, or if none
+ * exists for your particular codec.
+ */
+ if (!spec->alt_firmware_present) {
+ codec_dbg(codec, "Default firmware selected.");
+ if (request_firmware(&fw_entry, EFX_FILE,
+ codec->card->dev) != 0)
+ return false;
+ }
dsp_os_image = (struct dsp_image_seg *)(fw_entry->data);
if (dspload_image(codec, dsp_os_image, 0, 0, true, 0)) {
@@ -4402,13 +6678,17 @@ static void ca0132_download_dsp(struct hda_codec *codec)
return; /* don't retry failures */
chipio_enable_clocks(codec);
- spec->dsp_state = DSP_DOWNLOADING;
- if (!ca0132_download_dsp_images(codec))
- spec->dsp_state = DSP_DOWNLOAD_FAILED;
- else
- spec->dsp_state = DSP_DOWNLOADED;
+ if (spec->dsp_state != DSP_DOWNLOADED) {
+ spec->dsp_state = DSP_DOWNLOADING;
- if (spec->dsp_state == DSP_DOWNLOADED)
+ if (!ca0132_download_dsp_images(codec))
+ spec->dsp_state = DSP_DOWNLOAD_FAILED;
+ else
+ spec->dsp_state = DSP_DOWNLOADED;
+ }
+
+ /* For codecs using alt functions, this is already done earlier */
+ if (spec->dsp_state == DSP_DOWNLOADED && (!spec->use_alt_functions))
ca0132_set_dsp_msr(codec, true);
}
@@ -4454,6 +6734,10 @@ static void ca0132_init_unsol(struct hda_codec *codec)
amic_callback);
snd_hda_jack_detect_enable_callback(codec, UNSOL_TAG_DSP,
ca0132_process_dsp_response);
+ /* Front headphone jack detection */
+ if (spec->use_alt_functions)
+ snd_hda_jack_detect_enable_callback(codec,
+ spec->unsol_tag_front_hp, hp_callback);
}
/*
@@ -4476,7 +6760,8 @@ static struct hda_verb ca0132_base_exit_verbs[] = {
{}
};
-/* Other verbs tables. Sends after DSP download. */
+/* Other verbs tables. Sends after DSP download. */
+
static struct hda_verb ca0132_init_verbs0[] = {
/* chip init verbs */
{0x15, 0x70D, 0xF0},
@@ -4506,8 +6791,27 @@ static struct hda_verb ca0132_init_verbs0[] = {
{0x15, 0x546, 0xC9},
{0x15, 0x53B, 0xCE},
{0x15, 0x5E8, 0xC9},
- {0x15, 0x717, 0x0D},
- {0x15, 0x718, 0x20},
+ {}
+};
+
+/* Extra init verbs for SBZ */
+static struct hda_verb sbz_init_verbs[] = {
+ {0x15, 0x70D, 0x20},
+ {0x15, 0x70E, 0x19},
+ {0x15, 0x707, 0x00},
+ {0x15, 0x539, 0xCE},
+ {0x15, 0x546, 0xC9},
+ {0x15, 0x70D, 0xB7},
+ {0x15, 0x70E, 0x09},
+ {0x15, 0x707, 0x10},
+ {0x15, 0x70D, 0xAF},
+ {0x15, 0x70E, 0x09},
+ {0x15, 0x707, 0x01},
+ {0x15, 0x707, 0x05},
+ {0x15, 0x70D, 0x73},
+ {0x15, 0x70E, 0x09},
+ {0x15, 0x707, 0x14},
+ {0x15, 0x6FF, 0xC4},
{}
};
@@ -4521,7 +6825,11 @@ static void ca0132_init_chip(struct hda_codec *codec)
mutex_init(&spec->chipio_mutex);
spec->cur_out_type = SPEAKER_OUT;
- spec->cur_mic_type = DIGITAL_MIC;
+ if (!spec->use_alt_functions)
+ spec->cur_mic_type = DIGITAL_MIC;
+ else
+ spec->cur_mic_type = REAR_MIC;
+
spec->cur_mic_boost = 0;
for (i = 0; i < VNODES_COUNT; i++) {
@@ -4539,6 +6847,15 @@ static void ca0132_init_chip(struct hda_codec *codec)
on = (unsigned int)ca0132_effects[i].reqs[0];
spec->effects_switch[i] = on ? 1 : 0;
}
+ /*
+ * Sets defaults for the effect slider controls, only for alternative
+ * ca0132 codecs. Also sets x-bass crossover frequency to 80hz.
+ */
+ if (spec->use_alt_controls) {
+ spec->xbass_xover_freq = 8;
+ for (i = 0; i < EFFECT_LEVEL_SLIDERS; i++)
+ spec->fx_ctl_val[i] = effect_slider_defaults[i];
+ }
spec->voicefx_val = 0;
spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID] = 1;
@@ -4549,6 +6866,120 @@ static void ca0132_init_chip(struct hda_codec *codec)
#endif
}
+/*
+ * Recon3Di exit specific commands.
+ */
+/* prevents popping noise on shutdown */
+static void r3di_gpio_shutdown(struct hda_codec *codec)
+{
+ snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 0x00);
+}
+
+/*
+ * Sound Blaster Z exit specific commands.
+ */
+static void sbz_region2_exit(struct hda_codec *codec)
+{
+ struct ca0132_spec *spec = codec->spec;
+ unsigned int i;
+
+ for (i = 0; i < 4; i++)
+ writeb(0x0, spec->mem_base + 0x100);
+ for (i = 0; i < 8; i++)
+ writeb(0xb3, spec->mem_base + 0x304);
+ /*
+ * I believe these are GPIO, with the right most hex digit being the
+ * gpio pin, and the second digit being on or off. We see this more in
+ * the input/output select functions.
+ */
+ writew(0x0000, spec->mem_base + 0x320);
+ writew(0x0001, spec->mem_base + 0x320);
+ writew(0x0104, spec->mem_base + 0x320);
+ writew(0x0005, spec->mem_base + 0x320);
+ writew(0x0007, spec->mem_base + 0x320);
+}
+
+static void sbz_set_pin_ctl_default(struct hda_codec *codec)
+{
+ hda_nid_t pins[5] = {0x0B, 0x0C, 0x0E, 0x12, 0x13};
+ unsigned int i;
+
+ snd_hda_codec_write(codec, 0x11, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40);
+
+ for (i = 0; i < 5; i++)
+ snd_hda_codec_write(codec, pins[i], 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00);
+}
+
+static void sbz_clear_unsolicited(struct hda_codec *codec)
+{
+ hda_nid_t pins[7] = {0x0B, 0x0E, 0x0F, 0x10, 0x11, 0x12, 0x13};
+ unsigned int i;
+
+ for (i = 0; i < 7; i++) {
+ snd_hda_codec_write(codec, pins[i], 0,
+ AC_VERB_SET_UNSOLICITED_ENABLE, 0x00);
+ }
+}
+
+/* On shutdown, sends commands in sets of three */
+static void sbz_gpio_shutdown_commands(struct hda_codec *codec, int dir,
+ int mask, int data)
+{
+ if (dir >= 0)
+ snd_hda_codec_write(codec, 0x01, 0,
+ AC_VERB_SET_GPIO_DIRECTION, dir);
+ if (mask >= 0)
+ snd_hda_codec_write(codec, 0x01, 0,
+ AC_VERB_SET_GPIO_MASK, mask);
+
+ if (data >= 0)
+ snd_hda_codec_write(codec, 0x01, 0,
+ AC_VERB_SET_GPIO_DATA, data);
+}
+
+static void sbz_exit_chip(struct hda_codec *codec)
+{
+ chipio_set_stream_control(codec, 0x03, 0);
+ chipio_set_stream_control(codec, 0x04, 0);
+
+ /* Mess with GPIO */
+ sbz_gpio_shutdown_commands(codec, 0x07, 0x07, -1);
+ sbz_gpio_shutdown_commands(codec, 0x07, 0x07, 0x05);
+ sbz_gpio_shutdown_commands(codec, 0x07, 0x07, 0x01);
+
+ chipio_set_stream_control(codec, 0x14, 0);
+ chipio_set_stream_control(codec, 0x0C, 0);
+
+ chipio_set_conn_rate(codec, 0x41, SR_192_000);
+ chipio_set_conn_rate(codec, 0x91, SR_192_000);
+
+ chipio_write(codec, 0x18a020, 0x00000083);
+
+ sbz_gpio_shutdown_commands(codec, 0x07, 0x07, 0x03);
+ sbz_gpio_shutdown_commands(codec, 0x07, 0x07, 0x07);
+ sbz_gpio_shutdown_commands(codec, 0x07, 0x07, 0x06);
+
+ chipio_set_stream_control(codec, 0x0C, 0);
+
+ chipio_set_control_param(codec, 0x0D, 0x24);
+
+ sbz_clear_unsolicited(codec);
+ sbz_set_pin_ctl_default(codec);
+
+ snd_hda_codec_write(codec, 0x0B, 0,
+ AC_VERB_SET_EAPD_BTLENABLE, 0x00);
+
+ if (dspload_is_loaded(codec))
+ dsp_reset(codec);
+
+ snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+ VENDOR_CHIPIO_CT_EXTENSIONS_ENABLE, 0x00);
+
+ sbz_region2_exit(codec);
+}
+
static void ca0132_exit_chip(struct hda_codec *codec)
{
/* put any chip cleanup stuffs here. */
@@ -4557,28 +6988,264 @@ static void ca0132_exit_chip(struct hda_codec *codec)
dsp_reset(codec);
}
+/*
+ * This fixes a problem that was hard to reproduce. Very rarely, I would
+ * boot up, and there would be no sound, but the DSP indicated it had loaded
+ * properly. I did a few memory dumps to see if anything was different, and
+ * there were a few areas of memory uninitialized with a1a2a3a4. This function
+ * checks if those areas are uninitialized, and if they are, it'll attempt to
+ * reload the card 3 times. Usually it fixes by the second.
+ */
+static void sbz_dsp_startup_check(struct hda_codec *codec)
+{
+ struct ca0132_spec *spec = codec->spec;
+ unsigned int dsp_data_check[4];
+ unsigned int cur_address = 0x390;
+ unsigned int i;
+ unsigned int failure = 0;
+ unsigned int reload = 3;
+
+ if (spec->startup_check_entered)
+ return;
+
+ spec->startup_check_entered = true;
+
+ for (i = 0; i < 4; i++) {
+ chipio_read(codec, cur_address, &dsp_data_check[i]);
+ cur_address += 0x4;
+ }
+ for (i = 0; i < 4; i++) {
+ if (dsp_data_check[i] == 0xa1a2a3a4)
+ failure = 1;
+ }
+
+ codec_dbg(codec, "Startup Check: %d ", failure);
+ if (failure)
+ codec_info(codec, "DSP not initialized properly. Attempting to fix.");
+ /*
+ * While the failure condition is true, and we haven't reached our
+ * three reload limit, continue trying to reload the driver and
+ * fix the issue.
+ */
+ while (failure && (reload != 0)) {
+ codec_info(codec, "Reloading... Tries left: %d", reload);
+ sbz_exit_chip(codec);
+ spec->dsp_state = DSP_DOWNLOAD_INIT;
+ codec->patch_ops.init(codec);
+ failure = 0;
+ for (i = 0; i < 4; i++) {
+ chipio_read(codec, cur_address, &dsp_data_check[i]);
+ cur_address += 0x4;
+ }
+ for (i = 0; i < 4; i++) {
+ if (dsp_data_check[i] == 0xa1a2a3a4)
+ failure = 1;
+ }
+ reload--;
+ }
+
+ if (!failure && reload < 3)
+ codec_info(codec, "DSP fixed.");
+
+ if (!failure)
+ return;
+
+ codec_info(codec, "DSP failed to initialize properly. Either try a full shutdown or a suspend to clear the internal memory.");
+}
+
+/*
+ * This is for the extra volume verbs 0x797 (left) and 0x798 (right). These add
+ * extra precision for decibel values. If you had the dB value in floating point
+ * you would take the value after the decimal point, multiply by 64, and divide
+ * by 2. So for 8.59, it's (59 * 64) / 100. Useful if someone wanted to
+ * implement fixed point or floating point dB volumes. For now, I'll set them
+ * to 0 just incase a value has lingered from a boot into Windows.
+ */
+static void ca0132_alt_vol_setup(struct hda_codec *codec)
+{
+ snd_hda_codec_write(codec, 0x02, 0, 0x797, 0x00);
+ snd_hda_codec_write(codec, 0x02, 0, 0x798, 0x00);
+ snd_hda_codec_write(codec, 0x03, 0, 0x797, 0x00);
+ snd_hda_codec_write(codec, 0x03, 0, 0x798, 0x00);
+ snd_hda_codec_write(codec, 0x04, 0, 0x797, 0x00);
+ snd_hda_codec_write(codec, 0x04, 0, 0x798, 0x00);
+ snd_hda_codec_write(codec, 0x07, 0, 0x797, 0x00);
+ snd_hda_codec_write(codec, 0x07, 0, 0x798, 0x00);
+}
+
+/*
+ * Extra commands that don't really fit anywhere else.
+ */
+static void sbz_pre_dsp_setup(struct hda_codec *codec)
+{
+ struct ca0132_spec *spec = codec->spec;
+
+ writel(0x00820680, spec->mem_base + 0x01C);
+ writel(0x00820680, spec->mem_base + 0x01C);
+
+ snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xfc);
+ snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xfd);
+ snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xfe);
+ snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xff);
+
+ chipio_write(codec, 0x18b0a4, 0x000000c2);
+
+ snd_hda_codec_write(codec, 0x11, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, 0x44);
+}
+
+/*
+ * Extra commands that don't really fit anywhere else.
+ */
+static void r3di_pre_dsp_setup(struct hda_codec *codec)
+{
+ chipio_write(codec, 0x18b0a4, 0x000000c2);
+
+ snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+ VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x1E);
+ snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+ VENDOR_CHIPIO_8051_ADDRESS_HIGH, 0x1C);
+ snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+ VENDOR_CHIPIO_8051_DATA_WRITE, 0x5B);
+
+ snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+ VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x20);
+ snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+ VENDOR_CHIPIO_8051_ADDRESS_HIGH, 0x19);
+ snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+ VENDOR_CHIPIO_8051_DATA_WRITE, 0x00);
+ snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+ VENDOR_CHIPIO_8051_DATA_WRITE, 0x40);
+
+ snd_hda_codec_write(codec, 0x11, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, 0x04);
+}
+
+
+/*
+ * These are sent before the DSP is downloaded. Not sure
+ * what they do, or if they're necessary. Could possibly
+ * be removed. Figure they're better to leave in.
+ */
+static void sbz_region2_startup(struct hda_codec *codec)
+{
+ struct ca0132_spec *spec = codec->spec;
+
+ writel(0x00000000, spec->mem_base + 0x400);
+ writel(0x00000000, spec->mem_base + 0x408);
+ writel(0x00000000, spec->mem_base + 0x40C);
+ writel(0x00880680, spec->mem_base + 0x01C);
+ writel(0x00000083, spec->mem_base + 0xC0C);
+ writel(0x00000030, spec->mem_base + 0xC00);
+ writel(0x00000000, spec->mem_base + 0xC04);
+ writel(0x00000003, spec->mem_base + 0xC0C);
+ writel(0x00000003, spec->mem_base + 0xC0C);
+ writel(0x00000003, spec->mem_base + 0xC0C);
+ writel(0x00000003, spec->mem_base + 0xC0C);
+ 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);
+}
+
+/*
+ * Extra init functions for alternative ca0132 codecs. Done
+ * here so they don't clutter up the main ca0132_init function
+ * anymore than they have to.
+ */
+static void ca0132_alt_init(struct hda_codec *codec)
+{
+ struct ca0132_spec *spec = codec->spec;
+
+ ca0132_alt_vol_setup(codec);
+
+ switch (spec->quirk) {
+ case QUIRK_SBZ:
+ codec_dbg(codec, "SBZ alt_init");
+ ca0132_gpio_init(codec);
+ sbz_pre_dsp_setup(codec);
+ snd_hda_sequence_write(codec, spec->chip_init_verbs);
+ snd_hda_sequence_write(codec, spec->sbz_init_verbs);
+ break;
+ case QUIRK_R3DI:
+ codec_dbg(codec, "R3DI alt_init");
+ ca0132_gpio_init(codec);
+ ca0132_gpio_setup(codec);
+ r3di_gpio_dsp_status_set(codec, R3DI_DSP_DOWNLOADING);
+ r3di_pre_dsp_setup(codec);
+ snd_hda_sequence_write(codec, spec->chip_init_verbs);
+ snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, 0x6FF, 0xC4);
+ break;
+ }
+}
+
static int ca0132_init(struct hda_codec *codec)
{
struct ca0132_spec *spec = codec->spec;
struct auto_pin_cfg *cfg = &spec->autocfg;
int i;
+ bool dsp_loaded;
+
+ /*
+ * If the DSP is already downloaded, and init has been entered again,
+ * there's only two reasons for it. One, the codec has awaken from a
+ * suspended state, and in that case dspload_is_loaded will return
+ * false, and the init will be ran again. The other reason it gets
+ * re entered is on startup for some reason it triggers a suspend and
+ * resume state. In this case, it will check if the DSP is downloaded,
+ * and not run the init function again. For codecs using alt_functions,
+ * it will check if the DSP is loaded properly.
+ */
+ if (spec->dsp_state == DSP_DOWNLOADED) {
+ dsp_loaded = dspload_is_loaded(codec);
+ if (!dsp_loaded) {
+ spec->dsp_reload = true;
+ spec->dsp_state = DSP_DOWNLOAD_INIT;
+ } else {
+ if (spec->quirk == QUIRK_SBZ)
+ sbz_dsp_startup_check(codec);
+ return 0;
+ }
+ }
if (spec->dsp_state != DSP_DOWNLOAD_FAILED)
spec->dsp_state = DSP_DOWNLOAD_INIT;
spec->curr_chip_addx = INVALID_CHIP_ADDRESS;
+ if (spec->quirk == QUIRK_SBZ)
+ sbz_region2_startup(codec);
+
snd_hda_power_up_pm(codec);
ca0132_init_unsol(codec);
-
ca0132_init_params(codec);
ca0132_init_flags(codec);
+
snd_hda_sequence_write(codec, spec->base_init_verbs);
+
+ if (spec->quirk != QUIRK_NONE)
+ ca0132_alt_init(codec);
+
ca0132_download_dsp(codec);
+
ca0132_refresh_widget_caps(codec);
- ca0132_setup_defaults(codec);
- ca0132_init_analog_mic2(codec);
- ca0132_init_dmic(codec);
+
+ if (spec->quirk == QUIRK_SBZ)
+ writew(0x0107, spec->mem_base + 0x320);
+
+ switch (spec->quirk) {
+ case QUIRK_R3DI:
+ r3di_setup_defaults(codec);
+ break;
+ case QUIRK_NONE:
+ case QUIRK_ALIENWARE:
+ ca0132_setup_defaults(codec);
+ ca0132_init_analog_mic2(codec);
+ ca0132_init_dmic(codec);
+ break;
+ }
for (i = 0; i < spec->num_outputs; i++)
init_output(codec, spec->out_pins[i], spec->dacs[0]);
@@ -4590,14 +7257,45 @@ static int ca0132_init(struct hda_codec *codec)
init_input(codec, cfg->dig_in_pin, spec->dig_in);
- snd_hda_sequence_write(codec, spec->chip_init_verbs);
- snd_hda_sequence_write(codec, spec->spec_init_verbs);
+ if (!spec->use_alt_functions) {
+ snd_hda_sequence_write(codec, spec->chip_init_verbs);
+ snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+ VENDOR_CHIPIO_PARAM_EX_ID_SET, 0x0D);
+ snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+ VENDOR_CHIPIO_PARAM_EX_VALUE_SET, 0x20);
+ }
- ca0132_select_out(codec);
- ca0132_select_mic(codec);
+ if (spec->quirk == QUIRK_SBZ)
+ ca0132_gpio_setup(codec);
+
+ snd_hda_sequence_write(codec, spec->spec_init_verbs);
+ switch (spec->quirk) {
+ case QUIRK_SBZ:
+ sbz_setup_defaults(codec);
+ ca0132_alt_select_out(codec);
+ ca0132_alt_select_in(codec);
+ break;
+ case QUIRK_R3DI:
+ ca0132_alt_select_out(codec);
+ ca0132_alt_select_in(codec);
+ break;
+ default:
+ ca0132_select_out(codec);
+ ca0132_select_mic(codec);
+ break;
+ }
snd_hda_jack_report_sync(codec);
+ /*
+ * Re set the PlayEnhancement switch on a resume event, because the
+ * controls will not be reloaded.
+ */
+ if (spec->dsp_reload) {
+ spec->dsp_reload = false;
+ ca0132_pe_switch_set(codec);
+ }
+
snd_hda_power_down_pm(codec);
return 0;
@@ -4609,19 +7307,39 @@ static void ca0132_free(struct hda_codec *codec)
cancel_delayed_work_sync(&spec->unsol_hp_work);
snd_hda_power_up(codec);
- snd_hda_sequence_write(codec, spec->base_exit_verbs);
- ca0132_exit_chip(codec);
+ switch (spec->quirk) {
+ case QUIRK_SBZ:
+ sbz_exit_chip(codec);
+ break;
+ case QUIRK_R3DI:
+ r3di_gpio_shutdown(codec);
+ snd_hda_sequence_write(codec, spec->base_exit_verbs);
+ ca0132_exit_chip(codec);
+ break;
+ default:
+ snd_hda_sequence_write(codec, spec->base_exit_verbs);
+ ca0132_exit_chip(codec);
+ break;
+ }
snd_hda_power_down(codec);
+ if (spec->mem_base)
+ iounmap(spec->mem_base);
kfree(spec->spec_init_verbs);
kfree(codec->spec);
}
+static void ca0132_reboot_notify(struct hda_codec *codec)
+{
+ codec->patch_ops.free(codec);
+}
+
static const struct hda_codec_ops ca0132_patch_ops = {
.build_controls = ca0132_build_controls,
.build_pcms = ca0132_build_pcms,
.init = ca0132_init,
.free = ca0132_free,
.unsol_event = snd_hda_jack_unsol_event,
+ .reboot_notify = ca0132_reboot_notify,
};
static void ca0132_config(struct hda_codec *codec)
@@ -4635,9 +7353,14 @@ static void ca0132_config(struct hda_codec *codec)
spec->multiout.dac_nids = spec->dacs;
spec->multiout.num_dacs = 3;
- spec->multiout.max_channels = 2;
- if (spec->quirk == QUIRK_ALIENWARE) {
+ if (!spec->use_alt_functions)
+ spec->multiout.max_channels = 2;
+ else
+ spec->multiout.max_channels = 6;
+
+ switch (spec->quirk) {
+ case QUIRK_ALIENWARE:
codec_dbg(codec, "ca0132_config: QUIRK_ALIENWARE applied.\n");
snd_hda_apply_pincfgs(codec, alienware_pincfgs);
@@ -4657,7 +7380,71 @@ static void ca0132_config(struct hda_codec *codec)
spec->input_pins[2] = 0x13;
spec->shared_mic_nid = 0x7;
spec->unsol_tag_amic1 = 0x11;
- } else {
+ break;
+ case QUIRK_SBZ:
+ codec_dbg(codec, "%s: QUIRK_SBZ applied.\n", __func__);
+ snd_hda_apply_pincfgs(codec, sbz_pincfgs);
+
+ spec->num_outputs = 2;
+ spec->out_pins[0] = 0x0B; /* Line out */
+ spec->out_pins[1] = 0x0F; /* Rear headphone out */
+ spec->out_pins[2] = 0x10; /* Front Headphone / Center/LFE*/
+ spec->out_pins[3] = 0x11; /* Rear surround */
+ spec->shared_out_nid = 0x2;
+ spec->unsol_tag_hp = spec->out_pins[1];
+ spec->unsol_tag_front_hp = spec->out_pins[2];
+
+ spec->adcs[0] = 0x7; /* Rear Mic / Line-in */
+ spec->adcs[1] = 0x8; /* Front Mic, but only if no DSP */
+ spec->adcs[2] = 0xa; /* what u hear */
+
+ spec->num_inputs = 2;
+ spec->input_pins[0] = 0x12; /* Rear Mic / Line-in */
+ spec->input_pins[1] = 0x13; /* What U Hear */
+ spec->shared_mic_nid = 0x7;
+ spec->unsol_tag_amic1 = spec->input_pins[0];
+
+ /* SPDIF I/O */
+ spec->dig_out = 0x05;
+ spec->multiout.dig_out_nid = spec->dig_out;
+ cfg->dig_out_pins[0] = 0x0c;
+ cfg->dig_outs = 1;
+ cfg->dig_out_type[0] = HDA_PCM_TYPE_SPDIF;
+ spec->dig_in = 0x09;
+ cfg->dig_in_pin = 0x0e;
+ cfg->dig_in_type = HDA_PCM_TYPE_SPDIF;
+ break;
+ case QUIRK_R3DI:
+ codec_dbg(codec, "%s: QUIRK_R3DI applied.\n", __func__);
+ snd_hda_apply_pincfgs(codec, r3di_pincfgs);
+
+ spec->num_outputs = 2;
+ spec->out_pins[0] = 0x0B; /* Line out */
+ spec->out_pins[1] = 0x0F; /* Rear headphone out */
+ spec->out_pins[2] = 0x10; /* Front Headphone / Center/LFE*/
+ spec->out_pins[3] = 0x11; /* Rear surround */
+ spec->shared_out_nid = 0x2;
+ spec->unsol_tag_hp = spec->out_pins[1];
+ spec->unsol_tag_front_hp = spec->out_pins[2];
+
+ spec->adcs[0] = 0x07; /* Rear Mic / Line-in */
+ spec->adcs[1] = 0x08; /* Front Mic, but only if no DSP */
+ spec->adcs[2] = 0x0a; /* what u hear */
+
+ spec->num_inputs = 2;
+ spec->input_pins[0] = 0x12; /* Rear Mic / Line-in */
+ spec->input_pins[1] = 0x13; /* What U Hear */
+ spec->shared_mic_nid = 0x7;
+ spec->unsol_tag_amic1 = spec->input_pins[0];
+
+ /* SPDIF I/O */
+ spec->dig_out = 0x05;
+ spec->multiout.dig_out_nid = spec->dig_out;
+ cfg->dig_out_pins[0] = 0x0c;
+ cfg->dig_outs = 1;
+ cfg->dig_out_type[0] = HDA_PCM_TYPE_SPDIF;
+ break;
+ default:
spec->num_outputs = 2;
spec->out_pins[0] = 0x0b; /* speaker out */
spec->out_pins[1] = 0x10; /* headphone out */
@@ -4684,6 +7471,7 @@ static void ca0132_config(struct hda_codec *codec)
spec->dig_in = 0x09;
cfg->dig_in_pin = 0x0e;
cfg->dig_in_type = HDA_PCM_TYPE_SPDIF;
+ break;
}
}
@@ -4694,6 +7482,8 @@ static int ca0132_prepare_verbs(struct hda_codec *codec)
struct ca0132_spec *spec = codec->spec;
spec->chip_init_verbs = ca0132_init_verbs0;
+ if (spec->quirk == QUIRK_SBZ)
+ spec->sbz_init_verbs = sbz_init_verbs;
spec->spec_init_verbs = kzalloc(sizeof(struct hda_verb) * NUM_SPEC_VERBS, GFP_KERNEL);
if (!spec->spec_init_verbs)
return -ENOMEM;
@@ -4757,9 +7547,46 @@ static int patch_ca0132(struct hda_codec *codec)
else
spec->quirk = QUIRK_NONE;
+ /* Setup BAR Region 2 for Sound Blaster Z */
+ if (spec->quirk == QUIRK_SBZ) {
+ spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20);
+ if (spec->mem_base == NULL) {
+ codec_warn(codec, "pci_iomap failed!");
+ codec_info(codec, "perhaps this is not an SBZ?");
+ spec->quirk = QUIRK_NONE;
+ }
+ }
+
spec->dsp_state = DSP_DOWNLOAD_INIT;
spec->num_mixers = 1;
- spec->mixers[0] = ca0132_mixer;
+
+ /* Set which mixers each quirk uses. */
+ switch (spec->quirk) {
+ case QUIRK_SBZ:
+ spec->mixers[0] = sbz_mixer;
+ snd_hda_codec_set_name(codec, "Sound Blaster Z");
+ break;
+ case QUIRK_R3DI:
+ spec->mixers[0] = r3di_mixer;
+ snd_hda_codec_set_name(codec, "Recon3Di");
+ break;
+ default:
+ spec->mixers[0] = ca0132_mixer;
+ break;
+ }
+
+ /* Setup whether or not to use alt functions/controls */
+ switch (spec->quirk) {
+ case QUIRK_SBZ:
+ case QUIRK_R3DI:
+ spec->use_alt_controls = true;
+ spec->use_alt_functions = true;
+ break;
+ default:
+ spec->use_alt_controls = false;
+ spec->use_alt_functions = false;
+ break;
+ }
spec->base_init_verbs = ca0132_base_init_verbs;
spec->base_exit_verbs = ca0132_base_exit_verbs;
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 5b4dbce..093d2a9 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -588,6 +588,7 @@ static void cxt_fixup_olpc_xo(struct hda_codec *codec,
const struct hda_fixup *fix, int action)
{
struct conexant_spec *spec = codec->spec;
+ struct snd_kcontrol_new *kctl;
int i;
if (action != HDA_FIXUP_ACT_PROBE)
@@ -606,9 +607,7 @@ static void cxt_fixup_olpc_xo(struct hda_codec *codec,
snd_hda_codec_set_pin_target(codec, 0x1a, PIN_VREF50);
/* override mic boost control */
- for (i = 0; i < spec->gen.kctls.used; i++) {
- struct snd_kcontrol_new *kctl =
- snd_array_elem(&spec->gen.kctls, i);
+ snd_array_for_each(&spec->gen.kctls, i, kctl) {
if (!strcmp(kctl->name, "Mic Boost Volume")) {
kctl->put = olpc_xo_mic_boost_put;
break;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 01a6643..3af9384 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -2830,6 +2830,7 @@ static int find_ext_mic_pin(struct hda_codec *codec);
static void alc286_shutup(struct hda_codec *codec)
{
+ const struct hda_pincfg *pin;
int i;
int mic_pin = find_ext_mic_pin(codec);
/* don't shut up pins when unloading the driver; otherwise it breaks
@@ -2837,8 +2838,7 @@ static void alc286_shutup(struct hda_codec *codec)
*/
if (codec->bus->shutdown)
return;
- for (i = 0; i < codec->init_pins.used; i++) {
- struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i);
+ snd_array_for_each(&codec->init_pins, i, pin) {
/* use read here for syncing after issuing each verb */
if (pin->nid != mic_pin)
snd_hda_codec_read(codec, pin->nid, 0,
diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c
index abc7bd5..f0e7135 100644
--- a/sound/sparc/dbri.c
+++ b/sound/sparc/dbri.c
@@ -2542,7 +2542,7 @@ static int snd_dbri_create(struct snd_card *card,
dbri->irq = irq;
dbri->dma = dma_zalloc_coherent(&op->dev, sizeof(struct dbri_dma),
- &dbri->dma_dvma, GFP_ATOMIC);
+ &dbri->dma_dvma, GFP_KERNEL);
if (!dbri->dma)
return -ENOMEM;
diff --git a/sound/usb/card.c b/sound/usb/card.c
index 4a1c6bb..f6c3c1c 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -221,32 +221,13 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif)
struct usb_device *dev = chip->dev;
struct usb_host_interface *host_iface;
struct usb_interface_descriptor *altsd;
- void *control_header;
int i, protocol;
- int rest_bytes;
/* find audiocontrol interface */
host_iface = &usb_ifnum_to_if(dev, ctrlif)->altsetting[0];
- control_header = snd_usb_find_csint_desc(host_iface->extra,
- host_iface->extralen,
- NULL, UAC_HEADER);
altsd = get_iface_desc(host_iface);
protocol = altsd->bInterfaceProtocol;
- if (!control_header) {
- dev_err(&dev->dev, "cannot find UAC_HEADER\n");
- return -EINVAL;
- }
-
- rest_bytes = (void *)(host_iface->extra + host_iface->extralen) -
- control_header;
-
- /* just to be sure -- this shouldn't hit at all */
- if (rest_bytes <= 0) {
- dev_err(&dev->dev, "invalid control header\n");
- return -EINVAL;
- }
-
switch (protocol) {
default:
dev_warn(&dev->dev,
@@ -255,7 +236,25 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif)
/* fall through */
case UAC_VERSION_1: {
- struct uac1_ac_header_descriptor *h1 = control_header;
+ struct uac1_ac_header_descriptor *h1;
+ int rest_bytes;
+
+ h1 = snd_usb_find_csint_desc(host_iface->extra,
+ host_iface->extralen,
+ NULL, UAC_HEADER);
+ if (!h1) {
+ dev_err(&dev->dev, "cannot find UAC_HEADER\n");
+ return -EINVAL;
+ }
+
+ rest_bytes = (void *)(host_iface->extra +
+ host_iface->extralen) - (void *)h1;
+
+ /* just to be sure -- this shouldn't hit at all */
+ if (rest_bytes <= 0) {
+ dev_err(&dev->dev, "invalid control header\n");
+ return -EINVAL;
+ }
if (rest_bytes < sizeof(*h1)) {
dev_err(&dev->dev, "too short v1 buffer descriptor\n");
@@ -308,6 +307,20 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif)
return -EINVAL;
}
+ if (protocol == UAC_VERSION_3) {
+ int badd = assoc->bFunctionSubClass;
+
+ if (badd != UAC3_FUNCTION_SUBCLASS_FULL_ADC_3_0 &&
+ (badd < UAC3_FUNCTION_SUBCLASS_GENERIC_IO ||
+ badd > UAC3_FUNCTION_SUBCLASS_SPEAKERPHONE)) {
+ dev_err(&dev->dev,
+ "Unsupported UAC3 BADD profile\n");
+ return -EINVAL;
+ }
+
+ chip->badd_profile = badd;
+ }
+
for (i = 0; i < assoc->bInterfaceCount; i++) {
int intf = assoc->bFirstInterface + i;
@@ -349,6 +362,90 @@ static int snd_usb_audio_dev_free(struct snd_device *device)
return snd_usb_audio_free(chip);
}
+static void usb_audio_make_shortname(struct usb_device *dev,
+ struct snd_usb_audio *chip,
+ 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));
+ return;
+ }
+
+ /* retrieve the device string as shortname */
+ if (!dev->descriptor.iProduct ||
+ usb_string(dev, dev->descriptor.iProduct,
+ card->shortname, sizeof(card->shortname)) <= 0) {
+ /* no name available from anywhere, so use ID */
+ sprintf(card->shortname, "USB Device %#04x:%#04x",
+ USB_ID_VENDOR(chip->usb_id),
+ USB_ID_PRODUCT(chip->usb_id));
+ }
+
+ strim(card->shortname);
+}
+
+static void usb_audio_make_longname(struct usb_device *dev,
+ struct snd_usb_audio *chip,
+ const struct snd_usb_audio_quirk *quirk)
+{
+ struct snd_card *card = chip->card;
+ int len;
+
+ /* 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));
+ return;
+ }
+
+ if (quirk && quirk->vendor_name && *quirk->vendor_name) {
+ len = strlcpy(card->longname, quirk->vendor_name, sizeof(card->longname));
+ } else {
+ /* retrieve the vendor and device strings as longname */
+ if (dev->descriptor.iManufacturer)
+ len = usb_string(dev, dev->descriptor.iManufacturer,
+ card->longname, sizeof(card->longname));
+ else
+ len = 0;
+ /* we don't really care if there isn't any vendor string */
+ }
+ if (len > 0) {
+ strim(card->longname);
+ if (*card->longname)
+ strlcat(card->longname, " ", sizeof(card->longname));
+ }
+
+ strlcat(card->longname, card->shortname, sizeof(card->longname));
+
+ len = strlcat(card->longname, " at ", sizeof(card->longname));
+
+ if (len < sizeof(card->longname))
+ usb_make_path(dev, card->longname + len, sizeof(card->longname) - len);
+
+ switch (snd_usb_get_speed(dev)) {
+ case USB_SPEED_LOW:
+ strlcat(card->longname, ", low speed", sizeof(card->longname));
+ break;
+ case USB_SPEED_FULL:
+ strlcat(card->longname, ", full speed", sizeof(card->longname));
+ break;
+ case USB_SPEED_HIGH:
+ strlcat(card->longname, ", high speed", sizeof(card->longname));
+ break;
+ case USB_SPEED_SUPER:
+ strlcat(card->longname, ", super speed", sizeof(card->longname));
+ break;
+ case USB_SPEED_SUPER_PLUS:
+ strlcat(card->longname, ", super speed plus", sizeof(card->longname));
+ break;
+ default:
+ break;
+ }
+}
+
/*
* create a chip instance and set its names.
*/
@@ -360,7 +457,7 @@ static int snd_usb_audio_create(struct usb_interface *intf,
{
struct snd_card *card;
struct snd_usb_audio *chip;
- int err, len;
+ int err;
char component[14];
static struct snd_device_ops ops = {
.dev_free = snd_usb_audio_dev_free,
@@ -422,64 +519,8 @@ static int snd_usb_audio_create(struct usb_interface *intf,
USB_ID_VENDOR(chip->usb_id), USB_ID_PRODUCT(chip->usb_id));
snd_component_add(card, component);
- /* retrieve the device string as shortname */
- if (quirk && quirk->product_name && *quirk->product_name) {
- strlcpy(card->shortname, quirk->product_name, sizeof(card->shortname));
- } else {
- if (!dev->descriptor.iProduct ||
- usb_string(dev, dev->descriptor.iProduct,
- card->shortname, sizeof(card->shortname)) <= 0) {
- /* no name available from anywhere, so use ID */
- sprintf(card->shortname, "USB Device %#04x:%#04x",
- USB_ID_VENDOR(chip->usb_id),
- USB_ID_PRODUCT(chip->usb_id));
- }
- }
- strim(card->shortname);
-
- /* retrieve the vendor and device strings as longname */
- if (quirk && quirk->vendor_name && *quirk->vendor_name) {
- len = strlcpy(card->longname, quirk->vendor_name, sizeof(card->longname));
- } else {
- if (dev->descriptor.iManufacturer)
- len = usb_string(dev, dev->descriptor.iManufacturer,
- card->longname, sizeof(card->longname));
- else
- len = 0;
- /* we don't really care if there isn't any vendor string */
- }
- if (len > 0) {
- strim(card->longname);
- if (*card->longname)
- strlcat(card->longname, " ", sizeof(card->longname));
- }
-
- strlcat(card->longname, card->shortname, sizeof(card->longname));
-
- len = strlcat(card->longname, " at ", sizeof(card->longname));
-
- if (len < sizeof(card->longname))
- usb_make_path(dev, card->longname + len, sizeof(card->longname) - len);
-
- switch (snd_usb_get_speed(dev)) {
- case USB_SPEED_LOW:
- strlcat(card->longname, ", low speed", sizeof(card->longname));
- break;
- case USB_SPEED_FULL:
- strlcat(card->longname, ", full speed", sizeof(card->longname));
- break;
- case USB_SPEED_HIGH:
- strlcat(card->longname, ", high speed", sizeof(card->longname));
- break;
- case USB_SPEED_SUPER:
- strlcat(card->longname, ", super speed", sizeof(card->longname));
- break;
- case USB_SPEED_SUPER_PLUS:
- strlcat(card->longname, ", super speed plus", sizeof(card->longname));
- break;
- default:
- break;
- }
+ usb_audio_make_shortname(dev, chip, quirk);
+ usb_audio_make_longname(dev, chip, quirk);
snd_usb_audio_create_proc(chip);
diff --git a/sound/usb/clock.c b/sound/usb/clock.c
index 0b030d8..17673f3 100644
--- a/sound/usb/clock.c
+++ b/sound/usb/clock.c
@@ -587,8 +587,15 @@ int snd_usb_init_sample_rate(struct snd_usb_audio *chip, int iface,
default:
return set_sample_rate_v1(chip, iface, alts, fmt, rate);
- case UAC_VERSION_2:
case UAC_VERSION_3:
+ if (chip->badd_profile >= UAC3_FUNCTION_SUBCLASS_GENERIC_IO) {
+ if (rate != UAC3_BADD_SAMPLING_RATE)
+ return -ENXIO;
+ else
+ return 0;
+ }
+ /* fall through */
+ case UAC_VERSION_2:
return set_sample_rate_v2v3(chip, iface, alts, fmt, rate);
}
}
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index bb5ab7a..173979e 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -112,14 +112,12 @@ enum {
#include "mixer_maps.c"
static const struct usbmix_name_map *
-find_map(struct mixer_build *state, int unitid, int control)
+find_map(const struct usbmix_name_map *p, int unitid, int control)
{
- const struct usbmix_name_map *p = state->map;
-
if (!p)
return NULL;
- for (p = state->map; p->id; p++) {
+ for (; p->id; p++) {
if (p->id == unitid &&
(!control || !p->control || control == p->control))
return p;
@@ -201,10 +199,10 @@ static void *find_audio_control_unit(struct mixer_build *state,
/*
* copy a string with the given id
*/
-static int snd_usb_copy_string_desc(struct mixer_build *state,
+static int snd_usb_copy_string_desc(struct snd_usb_audio *chip,
int index, char *buf, int maxlen)
{
- int len = usb_string(state->chip->dev, index, buf, maxlen - 1);
+ int len = usb_string(chip->dev, index, buf, maxlen - 1);
if (len < 0)
return 0;
@@ -658,14 +656,14 @@ static struct iterm_name_combo {
{ 0 },
};
-static int get_term_name(struct mixer_build *state, struct usb_audio_term *iterm,
+static int get_term_name(struct snd_usb_audio *chip, struct usb_audio_term *iterm,
unsigned char *name, int maxlen, int term_only)
{
struct iterm_name_combo *names;
int len;
if (iterm->name) {
- len = snd_usb_copy_string_desc(state, iterm->name,
+ len = snd_usb_copy_string_desc(chip, iterm->name,
name, maxlen);
if (len)
return len;
@@ -1341,16 +1339,16 @@ static struct usb_feature_control_info *get_feature_control_info(int control)
return NULL;
}
-static void build_feature_ctl(struct mixer_build *state, void *raw_desc,
- unsigned int ctl_mask, int control,
- struct usb_audio_term *iterm, int unitid,
- int readonly_mask)
+static void __build_feature_ctl(struct usb_mixer_interface *mixer,
+ const struct usbmix_name_map *imap,
+ unsigned int ctl_mask, int control,
+ struct usb_audio_term *iterm,
+ struct usb_audio_term *oterm,
+ int unitid, int nameid, int readonly_mask)
{
- struct uac_feature_unit_descriptor *desc = raw_desc;
struct usb_feature_control_info *ctl_info;
unsigned int len = 0;
int mapped_name = 0;
- int nameid = uac_feature_unit_iFeature(desc);
struct snd_kcontrol *kctl;
struct usb_mixer_elem_info *cval;
const struct usbmix_name_map *map;
@@ -1361,14 +1359,14 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc,
return;
}
- map = find_map(state, unitid, control);
+ map = find_map(imap, unitid, control);
if (check_ignored_ctl(map))
return;
cval = kzalloc(sizeof(*cval), GFP_KERNEL);
if (!cval)
return;
- snd_usb_mixer_elem_init_std(&cval->head, state->mixer, unitid);
+ snd_usb_mixer_elem_init_std(&cval->head, mixer, unitid);
cval->control = control;
cval->cmask = ctl_mask;
@@ -1377,7 +1375,7 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc,
kfree(cval);
return;
}
- if (state->mixer->protocol == UAC_VERSION_1)
+ if (mixer->protocol == UAC_VERSION_1)
cval->val_type = ctl_info->type;
else /* UAC_VERSION_2 */
cval->val_type = ctl_info->type_uac2 >= 0 ?
@@ -1406,7 +1404,7 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc,
kctl = snd_ctl_new1(&usb_feature_unit_ctl, cval);
if (!kctl) {
- usb_audio_err(state->chip, "cannot malloc kcontrol\n");
+ usb_audio_err(mixer->chip, "cannot malloc kcontrol\n");
kfree(cval);
return;
}
@@ -1415,7 +1413,7 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc,
len = check_mapped_name(map, kctl->id.name, sizeof(kctl->id.name));
mapped_name = len != 0;
if (!len && nameid)
- len = snd_usb_copy_string_desc(state, nameid,
+ len = snd_usb_copy_string_desc(mixer->chip, nameid,
kctl->id.name, sizeof(kctl->id.name));
switch (control) {
@@ -1430,10 +1428,12 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc,
* - otherwise, anonymous name.
*/
if (!len) {
- len = get_term_name(state, iterm, kctl->id.name,
- sizeof(kctl->id.name), 1);
- if (!len)
- len = get_term_name(state, &state->oterm,
+ if (iterm)
+ len = get_term_name(mixer->chip, iterm,
+ kctl->id.name,
+ sizeof(kctl->id.name), 1);
+ if (!len && oterm)
+ len = get_term_name(mixer->chip, oterm,
kctl->id.name,
sizeof(kctl->id.name), 1);
if (!len)
@@ -1442,15 +1442,15 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc,
}
if (!mapped_name)
- check_no_speaker_on_headset(kctl, state->mixer->chip->card);
+ check_no_speaker_on_headset(kctl, mixer->chip->card);
/*
* determine the stream direction:
* if the connected output is USB stream, then it's likely a
* capture stream. otherwise it should be playback (hopefully :)
*/
- if (!mapped_name && !(state->oterm.type >> 16)) {
- if ((state->oterm.type & 0xff00) == 0x0100)
+ if (!mapped_name && oterm && !(oterm->type >> 16)) {
+ if ((oterm->type & 0xff00) == 0x0100)
append_ctl_name(kctl, " Capture");
else
append_ctl_name(kctl, " Playback");
@@ -1478,7 +1478,7 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc,
}
}
- snd_usb_mixer_fu_apply_quirk(state->mixer, cval, unitid, kctl);
+ snd_usb_mixer_fu_apply_quirk(mixer, cval, unitid, kctl);
range = (cval->max - cval->min) / cval->res;
/*
@@ -1487,26 +1487,46 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc,
* devices. It will definitively catch all buggy Logitech devices.
*/
if (range > 384) {
- usb_audio_warn(state->chip,
+ usb_audio_warn(mixer->chip,
"Warning! Unlikely big volume range (=%u), cval->res is probably wrong.",
range);
- usb_audio_warn(state->chip,
+ usb_audio_warn(mixer->chip,
"[%d] FU [%s] ch = %d, val = %d/%d/%d",
cval->head.id, kctl->id.name, cval->channels,
cval->min, cval->max, cval->res);
}
- usb_audio_dbg(state->chip, "[%d] FU [%s] ch = %d, val = %d/%d/%d\n",
+ usb_audio_dbg(mixer->chip, "[%d] FU [%s] ch = %d, val = %d/%d/%d\n",
cval->head.id, kctl->id.name, cval->channels,
cval->min, cval->max, cval->res);
snd_usb_mixer_add_control(&cval->head, kctl);
}
+static void build_feature_ctl(struct mixer_build *state, void *raw_desc,
+ unsigned int ctl_mask, int control,
+ struct usb_audio_term *iterm, int unitid,
+ int readonly_mask)
+{
+ struct uac_feature_unit_descriptor *desc = raw_desc;
+ int nameid = uac_feature_unit_iFeature(desc);
+
+ __build_feature_ctl(state->mixer, state->map, ctl_mask, control,
+ iterm, &state->oterm, unitid, nameid, readonly_mask);
+}
+
+static void build_feature_ctl_badd(struct usb_mixer_interface *mixer,
+ unsigned int ctl_mask, int control, int unitid,
+ const struct usbmix_name_map *badd_map)
+{
+ __build_feature_ctl(mixer, badd_map, ctl_mask, control,
+ NULL, NULL, unitid, 0, 0);
+}
+
static void get_connector_control_name(struct mixer_build *state,
struct usb_audio_term *term,
bool is_input, char *name, int name_size)
{
- int name_len = get_term_name(state, term, name, name_size, 0);
+ int name_len = get_term_name(state->chip, term, name, name_size, 0);
if (name_len == 0)
strlcpy(name, "Unknown", name_size);
@@ -1605,7 +1625,7 @@ static int parse_clock_source_unit(struct mixer_build *state, int unitid,
}
kctl->private_free = snd_usb_mixer_elem_free;
- ret = snd_usb_copy_string_desc(state, hdr->iClockSource,
+ ret = snd_usb_copy_string_desc(state->chip, hdr->iClockSource,
name, sizeof(name));
if (ret > 0)
snprintf(kctl->id.name, sizeof(kctl->id.name),
@@ -1815,7 +1835,7 @@ static void build_mixer_unit_ctl(struct mixer_build *state,
struct snd_kcontrol *kctl;
const struct usbmix_name_map *map;
- map = find_map(state, unitid, 0);
+ map = find_map(state->map, unitid, 0);
if (check_ignored_ctl(map))
return;
@@ -1848,7 +1868,7 @@ static void build_mixer_unit_ctl(struct mixer_build *state,
len = check_mapped_name(map, kctl->id.name, sizeof(kctl->id.name));
if (!len)
- len = get_term_name(state, iterm, kctl->id.name,
+ len = get_term_name(state->chip, iterm, kctl->id.name,
sizeof(kctl->id.name), 0);
if (!len)
len = sprintf(kctl->id.name, "Mixer Source %d", in_ch + 1);
@@ -2114,7 +2134,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid,
if (!(controls[valinfo->control / 8] & (1 << ((valinfo->control % 8) - 1))))
continue;
- map = find_map(state, unitid, valinfo->control);
+ map = find_map(state->map, unitid, valinfo->control);
if (check_ignored_ctl(map))
continue;
cval = kzalloc(sizeof(*cval), GFP_KERNEL);
@@ -2162,7 +2182,8 @@ static int build_audio_procunit(struct mixer_build *state, int unitid,
nameid = uac_processing_unit_iProcessing(desc, state->mixer->protocol);
len = 0;
if (nameid)
- len = snd_usb_copy_string_desc(state, nameid,
+ len = snd_usb_copy_string_desc(state->chip,
+ nameid,
kctl->id.name,
sizeof(kctl->id.name));
if (!len)
@@ -2317,7 +2338,7 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid,
if (desc->bNrInPins == 1) /* only one ? nonsense! */
return 0;
- map = find_map(state, unitid, 0);
+ map = find_map(state->map, unitid, 0);
if (check_ignored_ctl(map))
return 0;
@@ -2358,7 +2379,8 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid,
len = check_mapped_selector_name(state, unitid, i, namelist[i],
MAX_ITEM_NAME_LEN);
if (! len && check_input_term(state, desc->baSourceID[i], &iterm) >= 0)
- len = get_term_name(state, &iterm, namelist[i], MAX_ITEM_NAME_LEN, 0);
+ len = get_term_name(state->chip, &iterm, namelist[i],
+ MAX_ITEM_NAME_LEN, 0);
if (! len)
sprintf(namelist[i], "Input %u", i);
}
@@ -2380,12 +2402,12 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid,
/* if iSelector is given, use it */
nameid = uac_selector_unit_iSelector(desc);
if (nameid)
- len = snd_usb_copy_string_desc(state, nameid,
+ len = snd_usb_copy_string_desc(state->chip, nameid,
kctl->id.name,
sizeof(kctl->id.name));
/* ... or pick up the terminal name at next */
if (!len)
- len = get_term_name(state, &state->oterm,
+ len = get_term_name(state->chip, &state->oterm,
kctl->id.name, sizeof(kctl->id.name), 0);
/* ... or use the fixed string "USB" as the last resort */
if (!len)
@@ -2503,6 +2525,246 @@ static int snd_usb_mixer_dev_free(struct snd_device *device)
return 0;
}
+/* UAC3 predefined channels configuration */
+struct uac3_badd_profile {
+ int subclass;
+ const char *name;
+ int c_chmask; /* capture channels mask */
+ int p_chmask; /* playback channels mask */
+ int st_chmask; /* side tone mixing channel mask */
+};
+
+static struct uac3_badd_profile uac3_badd_profiles[] = {
+ {
+ /*
+ * BAIF, BAOF or combination of both
+ * IN: Mono or Stereo cfg, Mono alt possible
+ * OUT: Mono or Stereo cfg, Mono alt possible
+ */
+ .subclass = UAC3_FUNCTION_SUBCLASS_GENERIC_IO,
+ .name = "GENERIC IO",
+ .c_chmask = -1, /* dynamic channels */
+ .p_chmask = -1, /* dynamic channels */
+ },
+ {
+ /* BAOF; Stereo only cfg, Mono alt possible */
+ .subclass = UAC3_FUNCTION_SUBCLASS_HEADPHONE,
+ .name = "HEADPHONE",
+ .p_chmask = 3,
+ },
+ {
+ /* BAOF; Mono or Stereo cfg, Mono alt possible */
+ .subclass = UAC3_FUNCTION_SUBCLASS_SPEAKER,
+ .name = "SPEAKER",
+ .p_chmask = -1, /* dynamic channels */
+ },
+ {
+ /* BAIF; Mono or Stereo cfg, Mono alt possible */
+ .subclass = UAC3_FUNCTION_SUBCLASS_MICROPHONE,
+ .name = "MICROPHONE",
+ .c_chmask = -1, /* dynamic channels */
+ },
+ {
+ /*
+ * BAIOF topology
+ * IN: Mono only
+ * OUT: Mono or Stereo cfg, Mono alt possible
+ */
+ .subclass = UAC3_FUNCTION_SUBCLASS_HEADSET,
+ .name = "HEADSET",
+ .c_chmask = 1,
+ .p_chmask = -1, /* dynamic channels */
+ .st_chmask = 1,
+ },
+ {
+ /* BAIOF; IN: Mono only; OUT: Stereo only, Mono alt possible */
+ .subclass = UAC3_FUNCTION_SUBCLASS_HEADSET_ADAPTER,
+ .name = "HEADSET ADAPTER",
+ .c_chmask = 1,
+ .p_chmask = 3,
+ .st_chmask = 1,
+ },
+ {
+ /* BAIF + BAOF; IN: Mono only; OUT: Mono only */
+ .subclass = UAC3_FUNCTION_SUBCLASS_SPEAKERPHONE,
+ .name = "SPEAKERPHONE",
+ .c_chmask = 1,
+ .p_chmask = 1,
+ },
+ { 0 } /* terminator */
+};
+
+static bool uac3_badd_func_has_valid_channels(struct usb_mixer_interface *mixer,
+ struct uac3_badd_profile *f,
+ int c_chmask, int p_chmask)
+{
+ /*
+ * If both playback/capture channels are dynamic, make sure
+ * at least one channel is present
+ */
+ if (f->c_chmask < 0 && f->p_chmask < 0) {
+ if (!c_chmask && !p_chmask) {
+ usb_audio_warn(mixer->chip, "BAAD %s: no channels?",
+ f->name);
+ return false;
+ }
+ return true;
+ }
+
+ if ((f->c_chmask < 0 && !c_chmask) ||
+ (f->c_chmask >= 0 && f->c_chmask != c_chmask)) {
+ usb_audio_warn(mixer->chip, "BAAD %s c_chmask mismatch",
+ f->name);
+ return false;
+ }
+ if ((f->p_chmask < 0 && !p_chmask) ||
+ (f->p_chmask >= 0 && f->p_chmask != p_chmask)) {
+ usb_audio_warn(mixer->chip, "BAAD %s p_chmask mismatch",
+ f->name);
+ return false;
+ }
+ return true;
+}
+
+/*
+ * create mixer controls for UAC3 BADD profiles
+ *
+ * UAC3 BADD device doesn't contain CS descriptors thus we will guess everything
+ *
+ * BADD device may contain Mixer Unit, which doesn't have any controls, skip it
+ */
+static int snd_usb_mixer_controls_badd(struct usb_mixer_interface *mixer,
+ int ctrlif)
+{
+ struct usb_device *dev = mixer->chip->dev;
+ struct usb_interface_assoc_descriptor *assoc;
+ int badd_profile = mixer->chip->badd_profile;
+ struct uac3_badd_profile *f;
+ const struct usbmix_ctl_map *map;
+ int p_chmask = 0, c_chmask = 0, st_chmask = 0;
+ int i;
+
+ assoc = usb_ifnum_to_if(dev, ctrlif)->intf_assoc;
+
+ /* Detect BADD capture/playback channels from AS EP descriptors */
+ for (i = 0; i < assoc->bInterfaceCount; i++) {
+ int intf = assoc->bFirstInterface + i;
+
+ struct usb_interface *iface;
+ struct usb_host_interface *alts;
+ struct usb_interface_descriptor *altsd;
+ unsigned int maxpacksize;
+ char dir_in;
+ int chmask, num;
+
+ if (intf == ctrlif)
+ continue;
+
+ iface = usb_ifnum_to_if(dev, intf);
+ num = iface->num_altsetting;
+
+ if (num < 2)
+ return -EINVAL;
+
+ /*
+ * The number of Channels in an AudioStreaming interface
+ * and the audio sample bit resolution (16 bits or 24
+ * bits) can be derived from the wMaxPacketSize field in
+ * the Standard AS Audio Data Endpoint descriptor in
+ * Alternate Setting 1
+ */
+ alts = &iface->altsetting[1];
+ altsd = get_iface_desc(alts);
+
+ if (altsd->bNumEndpoints < 1)
+ return -EINVAL;
+
+ /* check direction */
+ dir_in = (get_endpoint(alts, 0)->bEndpointAddress & USB_DIR_IN);
+ maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize);
+
+ switch (maxpacksize) {
+ default:
+ usb_audio_err(mixer->chip,
+ "incorrect wMaxPacketSize 0x%x for BADD profile\n",
+ maxpacksize);
+ return -EINVAL;
+ case UAC3_BADD_EP_MAXPSIZE_SYNC_MONO_16:
+ case UAC3_BADD_EP_MAXPSIZE_ASYNC_MONO_16:
+ case UAC3_BADD_EP_MAXPSIZE_SYNC_MONO_24:
+ case UAC3_BADD_EP_MAXPSIZE_ASYNC_MONO_24:
+ chmask = 1;
+ break;
+ case UAC3_BADD_EP_MAXPSIZE_SYNC_STEREO_16:
+ case UAC3_BADD_EP_MAXPSIZE_ASYNC_STEREO_16:
+ case UAC3_BADD_EP_MAXPSIZE_SYNC_STEREO_24:
+ case UAC3_BADD_EP_MAXPSIZE_ASYNC_STEREO_24:
+ chmask = 3;
+ break;
+ }
+
+ if (dir_in)
+ c_chmask = chmask;
+ else
+ p_chmask = chmask;
+ }
+
+ usb_audio_dbg(mixer->chip,
+ "UAC3 BADD profile 0x%x: detected c_chmask=%d p_chmask=%d\n",
+ badd_profile, c_chmask, p_chmask);
+
+ /* check the mapping table */
+ for (map = uac3_badd_usbmix_ctl_maps; map->id; map++) {
+ if (map->id == badd_profile)
+ break;
+ }
+
+ if (!map->id)
+ return -EINVAL;
+
+ for (f = uac3_badd_profiles; f->name; f++) {
+ if (badd_profile == f->subclass)
+ break;
+ }
+ if (!f->name)
+ return -EINVAL;
+ if (!uac3_badd_func_has_valid_channels(mixer, f, c_chmask, p_chmask))
+ return -EINVAL;
+ st_chmask = f->st_chmask;
+
+ /* Playback */
+ if (p_chmask) {
+ /* Master channel, always writable */
+ build_feature_ctl_badd(mixer, 0, UAC_FU_MUTE,
+ UAC3_BADD_FU_ID2, map->map);
+ /* Mono/Stereo volume channels, always writable */
+ build_feature_ctl_badd(mixer, p_chmask, UAC_FU_VOLUME,
+ UAC3_BADD_FU_ID2, map->map);
+ }
+
+ /* Capture */
+ if (c_chmask) {
+ /* Master channel, always writable */
+ build_feature_ctl_badd(mixer, 0, UAC_FU_MUTE,
+ UAC3_BADD_FU_ID5, map->map);
+ /* Mono/Stereo volume channels, always writable */
+ build_feature_ctl_badd(mixer, c_chmask, UAC_FU_VOLUME,
+ UAC3_BADD_FU_ID5, map->map);
+ }
+
+ /* Side tone-mixing */
+ if (st_chmask) {
+ /* Master channel, always writable */
+ build_feature_ctl_badd(mixer, 0, UAC_FU_MUTE,
+ UAC3_BADD_FU_ID7, map->map);
+ /* Mono volume channel, always writable */
+ build_feature_ctl_badd(mixer, 1, UAC_FU_VOLUME,
+ UAC3_BADD_FU_ID7, map->map);
+ }
+
+ return 0;
+}
+
/*
* create mixer controls
*
@@ -2606,9 +2868,9 @@ void snd_usb_mixer_notify_id(struct usb_mixer_interface *mixer, int unitid)
{
struct usb_mixer_elem_list *list;
- for (list = mixer->id_elems[unitid]; list; list = list->next_id_elem) {
+ for_each_mixer_elem(list, mixer, unitid) {
struct usb_mixer_elem_info *info =
- (struct usb_mixer_elem_info *)list;
+ mixer_elem_list_to_info(list);
/* invalidate cache, so the value is read from the device */
info->cached = 0;
snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE,
@@ -2619,7 +2881,7 @@ void snd_usb_mixer_notify_id(struct usb_mixer_interface *mixer, int unitid)
static void snd_usb_mixer_dump_cval(struct snd_info_buffer *buffer,
struct usb_mixer_elem_list *list)
{
- struct usb_mixer_elem_info *cval = (struct usb_mixer_elem_info *)list;
+ struct usb_mixer_elem_info *cval = mixer_elem_list_to_info(list);
static char *val_types[] = {"BOOLEAN", "INV_BOOLEAN",
"S8", "U8", "S16", "U16"};
snd_iprintf(buffer, " Info: id=%i, control=%i, cmask=0x%x, "
@@ -2645,8 +2907,7 @@ static void snd_usb_mixer_proc_read(struct snd_info_entry *entry,
mixer->ignore_ctl_error);
snd_iprintf(buffer, "Card: %s\n", chip->card->longname);
for (unitid = 0; unitid < MAX_ID_ELEMS; unitid++) {
- for (list = mixer->id_elems[unitid]; list;
- list = list->next_id_elem) {
+ for_each_mixer_elem(list, mixer, unitid) {
snd_iprintf(buffer, " Unit: %i\n", list->id);
if (list->kctl)
snd_iprintf(buffer,
@@ -2676,19 +2937,19 @@ static void snd_usb_mixer_interrupt_v2(struct usb_mixer_interface *mixer,
return;
}
- for (list = mixer->id_elems[unitid]; list; list = list->next_id_elem)
+ for_each_mixer_elem(list, mixer, unitid)
count++;
if (count == 0)
return;
- for (list = mixer->id_elems[unitid]; list; list = list->next_id_elem) {
+ for_each_mixer_elem(list, mixer, unitid) {
struct usb_mixer_elem_info *info;
if (!list->kctl)
continue;
- info = (struct usb_mixer_elem_info *)list;
+ info = mixer_elem_list_to_info(list);
if (count > 1 && info->control != control)
continue;
@@ -2809,6 +3070,48 @@ static int snd_usb_mixer_status_create(struct usb_mixer_interface *mixer)
return 0;
}
+static int keep_iface_ctl_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol);
+
+ ucontrol->value.integer.value[0] = mixer->chip->keep_iface;
+ return 0;
+}
+
+static int keep_iface_ctl_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol);
+ bool keep_iface = !!ucontrol->value.integer.value[0];
+
+ if (mixer->chip->keep_iface == keep_iface)
+ return 0;
+ mixer->chip->keep_iface = keep_iface;
+ return 1;
+}
+
+static const struct snd_kcontrol_new keep_iface_ctl = {
+ .iface = SNDRV_CTL_ELEM_IFACE_CARD,
+ .name = "Keep Interface",
+ .info = snd_ctl_boolean_mono_info,
+ .get = keep_iface_ctl_get,
+ .put = keep_iface_ctl_put,
+};
+
+static int create_keep_iface_ctl(struct usb_mixer_interface *mixer)
+{
+ struct snd_kcontrol *kctl = snd_ctl_new1(&keep_iface_ctl, mixer);
+
+ /* need only one control per card */
+ if (snd_ctl_find_id(mixer->chip->card, &kctl->id)) {
+ snd_ctl_free_one(kctl);
+ return 0;
+ }
+
+ return snd_ctl_add(mixer->chip->card, kctl);
+}
+
int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif,
int ignore_error)
{
@@ -2847,8 +3150,16 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif,
break;
}
- if ((err = snd_usb_mixer_controls(mixer)) < 0 ||
- (err = snd_usb_mixer_status_create(mixer)) < 0)
+ if (mixer->protocol == UAC_VERSION_3 &&
+ chip->badd_profile >= UAC3_FUNCTION_SUBCLASS_GENERIC_IO) {
+ if ((err = snd_usb_mixer_controls_badd(mixer, ctrlif)) < 0)
+ goto _error;
+ } else if ((err = snd_usb_mixer_controls(mixer)) < 0 ||
+ (err = snd_usb_mixer_status_create(mixer)) < 0) {
+ goto _error;
+ }
+ err = create_keep_iface_ctl(mixer);
+ if (err < 0)
goto _error;
snd_usb_mixer_apply_create_quirk(mixer);
@@ -2909,7 +3220,7 @@ int snd_usb_mixer_suspend(struct usb_mixer_interface *mixer)
static int restore_mixer_value(struct usb_mixer_elem_list *list)
{
- struct usb_mixer_elem_info *cval = (struct usb_mixer_elem_info *)list;
+ struct usb_mixer_elem_info *cval = mixer_elem_list_to_info(list);
int c, err, idx;
if (cval->cmask) {
@@ -2945,8 +3256,7 @@ int snd_usb_mixer_resume(struct usb_mixer_interface *mixer, bool reset_resume)
if (reset_resume) {
/* restore cached mixer values */
for (id = 0; id < MAX_ID_ELEMS; id++) {
- for (list = mixer->id_elems[id]; list;
- list = list->next_id_elem) {
+ for_each_mixer_elem(list, mixer, id) {
if (list->resume) {
err = list->resume(list);
if (err < 0)
@@ -2956,6 +3266,8 @@ int snd_usb_mixer_resume(struct usb_mixer_interface *mixer, bool reset_resume)
}
}
+ snd_usb_mixer_resume_quirk(mixer);
+
return snd_usb_mixer_activate(mixer);
}
#endif
diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h
index ba27f7a..e026534 100644
--- a/sound/usb/mixer.h
+++ b/sound/usb/mixer.h
@@ -53,6 +53,12 @@ struct usb_mixer_elem_list {
usb_mixer_elem_resume_func_t resume;
};
+/* iterate over mixer element list of the given unit id */
+#define for_each_mixer_elem(list, mixer, id) \
+ for ((list) = (mixer)->id_elems[id]; (list); (list) = (list)->next_id_elem)
+#define mixer_elem_list_to_info(list) \
+ container_of(list, struct usb_mixer_elem_info, head)
+
struct usb_mixer_elem_info {
struct usb_mixer_elem_list head;
unsigned int control; /* CS or ICN (high byte) */
diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c
index eaa03ac..71069e1 100644
--- a/sound/usb/mixer_maps.c
+++ b/sound/usb/mixer_maps.c
@@ -485,3 +485,68 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = {
{ 0 } /* terminator */
};
+/*
+ * Control map entries for UAC3 BADD profiles
+ */
+
+static struct usbmix_name_map uac3_badd_generic_io_map[] = {
+ { UAC3_BADD_FU_ID2, "Generic Out Playback" },
+ { UAC3_BADD_FU_ID5, "Generic In Capture" },
+ { 0 } /* terminator */
+};
+static struct usbmix_name_map uac3_badd_headphone_map[] = {
+ { UAC3_BADD_FU_ID2, "Headphone Playback" },
+ { 0 } /* terminator */
+};
+static struct usbmix_name_map uac3_badd_speaker_map[] = {
+ { UAC3_BADD_FU_ID2, "Speaker Playback" },
+ { 0 } /* terminator */
+};
+static struct usbmix_name_map uac3_badd_microphone_map[] = {
+ { UAC3_BADD_FU_ID5, "Mic Capture" },
+ { 0 } /* terminator */
+};
+/* Covers also 'headset adapter' profile */
+static struct usbmix_name_map uac3_badd_headset_map[] = {
+ { UAC3_BADD_FU_ID2, "Headset Playback" },
+ { UAC3_BADD_FU_ID5, "Headset Capture" },
+ { UAC3_BADD_FU_ID7, "Sidetone Mixing" },
+ { 0 } /* terminator */
+};
+static struct usbmix_name_map uac3_badd_speakerphone_map[] = {
+ { UAC3_BADD_FU_ID2, "Speaker Playback" },
+ { UAC3_BADD_FU_ID5, "Mic Capture" },
+ { 0 } /* terminator */
+};
+
+static struct usbmix_ctl_map uac3_badd_usbmix_ctl_maps[] = {
+ {
+ .id = UAC3_FUNCTION_SUBCLASS_GENERIC_IO,
+ .map = uac3_badd_generic_io_map,
+ },
+ {
+ .id = UAC3_FUNCTION_SUBCLASS_HEADPHONE,
+ .map = uac3_badd_headphone_map,
+ },
+ {
+ .id = UAC3_FUNCTION_SUBCLASS_SPEAKER,
+ .map = uac3_badd_speaker_map,
+ },
+ {
+ .id = UAC3_FUNCTION_SUBCLASS_MICROPHONE,
+ .map = uac3_badd_microphone_map,
+ },
+ {
+ .id = UAC3_FUNCTION_SUBCLASS_HEADSET,
+ .map = uac3_badd_headset_map,
+ },
+ {
+ .id = UAC3_FUNCTION_SUBCLASS_HEADSET_ADAPTER,
+ .map = uac3_badd_headset_map,
+ },
+ {
+ .id = UAC3_FUNCTION_SUBCLASS_SPEAKERPHONE,
+ .map = uac3_badd_speakerphone_map,
+ },
+ { 0 } /* terminator */
+};
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index 56537a1..1b94387 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -1172,7 +1172,7 @@ void snd_emuusb_set_samplerate(struct snd_usb_audio *chip,
int unitid = 12; /* SamleRate ExtensionUnit ID */
list_for_each_entry(mixer, &chip->mixer_list, list) {
- cval = (struct usb_mixer_elem_info *)mixer->id_elems[unitid];
+ cval = mixer_elem_list_to_info(mixer->id_elems[unitid]);
if (cval) {
snd_usb_mixer_set_ctl_value(cval, UAC_SET_CUR,
cval->control << 8,
@@ -1799,6 +1799,26 @@ static int snd_soundblaster_e1_switch_create(struct usb_mixer_interface *mixer)
NULL);
}
+static void dell_dock_init_vol(struct snd_usb_audio *chip, int ch, int id)
+{
+ u16 buf = 0;
+
+ snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), UAC_SET_CUR,
+ USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT,
+ ch, snd_usb_ctrl_intf(chip) | (id << 8),
+ &buf, 2);
+}
+
+static int dell_dock_mixer_init(struct usb_mixer_interface *mixer)
+{
+ /* fix to 0dB playback volumes */
+ dell_dock_init_vol(mixer->chip, 1, 16);
+ dell_dock_init_vol(mixer->chip, 2, 16);
+ dell_dock_init_vol(mixer->chip, 1, 19);
+ dell_dock_init_vol(mixer->chip, 2, 19);
+ return 0;
+}
+
int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer)
{
int err = 0;
@@ -1884,11 +1904,25 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer)
case USB_ID(0x041e, 0x323b): /* Creative Sound Blaster E1 */
err = snd_soundblaster_e1_switch_create(mixer);
break;
+ case USB_ID(0x0bda, 0x4014): /* Dell WD15 dock */
+ err = dell_dock_mixer_init(mixer);
+ break;
}
return err;
}
+#ifdef CONFIG_PM
+void snd_usb_mixer_resume_quirk(struct usb_mixer_interface *mixer)
+{
+ switch (mixer->chip->usb_id) {
+ case USB_ID(0x0bda, 0x4014): /* Dell WD15 dock */
+ dell_dock_mixer_init(mixer);
+ break;
+ }
+}
+#endif
+
void snd_usb_mixer_rc_memory_change(struct usb_mixer_interface *mixer,
int unitid)
{
diff --git a/sound/usb/mixer_quirks.h b/sound/usb/mixer_quirks.h
index b5abd32..52be26d 100644
--- a/sound/usb/mixer_quirks.h
+++ b/sound/usb/mixer_quirks.h
@@ -14,5 +14,9 @@ void snd_usb_mixer_fu_apply_quirk(struct usb_mixer_interface *mixer,
struct usb_mixer_elem_info *cval, int unitid,
struct snd_kcontrol *kctl);
+#ifdef CONFIG_PM
+void snd_usb_mixer_resume_quirk(struct usb_mixer_interface *mixer);
+#endif
+
#endif /* SND_USB_MIXER_QUIRKS_H */
diff --git a/sound/usb/mixer_scarlett.c b/sound/usb/mixer_scarlett.c
index c33e237..4aeb948 100644
--- a/sound/usb/mixer_scarlett.c
+++ b/sound/usb/mixer_scarlett.c
@@ -287,8 +287,7 @@ static int scarlett_ctl_switch_put(struct snd_kcontrol *kctl,
static int scarlett_ctl_resume(struct usb_mixer_elem_list *list)
{
- struct usb_mixer_elem_info *elem =
- container_of(list, struct usb_mixer_elem_info, head);
+ struct usb_mixer_elem_info *elem = mixer_elem_list_to_info(list);
int i;
for (i = 0; i < elem->channels; i++)
@@ -447,8 +446,7 @@ static int scarlett_ctl_enum_put(struct snd_kcontrol *kctl,
static int scarlett_ctl_enum_resume(struct usb_mixer_elem_list *list)
{
- struct usb_mixer_elem_info *elem =
- container_of(list, struct usb_mixer_elem_info, head);
+ struct usb_mixer_elem_info *elem = mixer_elem_list_to_info(list);
if (elem->cached)
snd_usb_set_cur_mix_value(elem, 0, 0, *elem->cache_val);
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 3cbfae6..dc2dfec 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -321,6 +321,7 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs,
struct usb_host_interface *alts;
struct usb_interface *iface;
unsigned int ep;
+ unsigned int ifnum;
/* Implicit feedback sync EPs consumers are always playback EPs */
if (subs->direction != SNDRV_PCM_STREAM_PLAYBACK)
@@ -330,44 +331,27 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs,
case USB_ID(0x0763, 0x2030): /* M-Audio Fast Track C400 */
case USB_ID(0x0763, 0x2031): /* M-Audio Fast Track C600 */
ep = 0x81;
- iface = usb_ifnum_to_if(dev, 3);
-
- if (!iface || iface->num_altsetting == 0)
- return -EINVAL;
-
- alts = &iface->altsetting[1];
- goto add_sync_ep;
- break;
+ ifnum = 3;
+ goto add_sync_ep_from_ifnum;
case USB_ID(0x0763, 0x2080): /* M-Audio FastTrack Ultra */
case USB_ID(0x0763, 0x2081):
ep = 0x81;
- iface = usb_ifnum_to_if(dev, 2);
-
- if (!iface || iface->num_altsetting == 0)
- return -EINVAL;
-
- alts = &iface->altsetting[1];
- goto add_sync_ep;
- case USB_ID(0x2466, 0x8003):
+ ifnum = 2;
+ goto add_sync_ep_from_ifnum;
+ case USB_ID(0x2466, 0x8003): /* Fractal Audio Axe-Fx II */
ep = 0x86;
- iface = usb_ifnum_to_if(dev, 2);
-
- if (!iface || iface->num_altsetting == 0)
- return -EINVAL;
-
- alts = &iface->altsetting[1];
- goto add_sync_ep;
- case USB_ID(0x1397, 0x0002):
+ ifnum = 2;
+ goto add_sync_ep_from_ifnum;
+ case USB_ID(0x2466, 0x8010): /* Fractal Audio Axe-Fx III */
ep = 0x81;
- iface = usb_ifnum_to_if(dev, 1);
-
- if (!iface || iface->num_altsetting == 0)
- return -EINVAL;
-
- alts = &iface->altsetting[1];
- goto add_sync_ep;
-
+ ifnum = 2;
+ goto add_sync_ep_from_ifnum;
+ case USB_ID(0x1397, 0x0002): /* Behringer UFX1204 */
+ ep = 0x81;
+ ifnum = 1;
+ goto add_sync_ep_from_ifnum;
}
+
if (attr == USB_ENDPOINT_SYNC_ASYNC &&
altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC &&
altsd->bInterfaceProtocol == 2 &&
@@ -382,6 +366,14 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs,
/* No quirk */
return 0;
+add_sync_ep_from_ifnum:
+ iface = usb_ifnum_to_if(dev, ifnum);
+
+ if (!iface || iface->num_altsetting == 0)
+ return -EINVAL;
+
+ alts = &iface->altsetting[1];
+
add_sync_ep:
subs->sync_endpoint = snd_usb_add_endpoint(subs->stream->chip,
alts, ep, !subs->direction,
@@ -507,7 +499,7 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt)
iface = usb_ifnum_to_if(dev, fmt->iface);
if (WARN_ON(!iface))
return -EINVAL;
- alts = &iface->altsetting[fmt->altset_idx];
+ alts = usb_altnum_to_altsetting(iface, fmt->altsetting);
altsd = get_iface_desc(alts);
if (WARN_ON(altsd->bAlternateSetting != fmt->altsetting))
return -EINVAL;
@@ -517,21 +509,21 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt)
/* close the old interface */
if (subs->interface >= 0 && subs->interface != fmt->iface) {
- err = usb_set_interface(subs->dev, subs->interface, 0);
- if (err < 0) {
- dev_err(&dev->dev,
- "%d:%d: return to setting 0 failed (%d)\n",
- fmt->iface, fmt->altsetting, err);
- return -EIO;
+ if (!subs->stream->chip->keep_iface) {
+ err = usb_set_interface(subs->dev, subs->interface, 0);
+ if (err < 0) {
+ dev_err(&dev->dev,
+ "%d:%d: return to setting 0 failed (%d)\n",
+ fmt->iface, fmt->altsetting, err);
+ return -EIO;
+ }
}
subs->interface = -1;
subs->altset_idx = 0;
}
/* set interface */
- if (subs->interface != fmt->iface ||
- subs->altset_idx != fmt->altset_idx) {
-
+ if (iface->cur_altsetting != alts) {
err = snd_usb_select_mode_quirk(subs, fmt);
if (err < 0)
return -EIO;
@@ -545,12 +537,11 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt)
}
dev_dbg(&dev->dev, "setting usb interface %d:%d\n",
fmt->iface, fmt->altsetting);
- subs->interface = fmt->iface;
- subs->altset_idx = fmt->altset_idx;
-
snd_usb_set_interface_quirk(dev);
}
+ subs->interface = fmt->iface;
+ subs->altset_idx = fmt->altset_idx;
subs->data_endpoint = snd_usb_add_endpoint(subs->stream->chip,
alts, fmt->endpoint, subs->direction,
SND_USB_ENDPOINT_TYPE_DATA);
@@ -1264,7 +1255,8 @@ static int snd_usb_pcm_close(struct snd_pcm_substream *substream, int direction)
stop_endpoints(subs, true);
- if (subs->interface >= 0 &&
+ if (!as->chip->keep_iface &&
+ subs->interface >= 0 &&
!snd_usb_lock_shutdown(subs->stream->chip)) {
usb_set_interface(subs->dev, subs->interface, 0);
subs->interface = -1;
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 754e632..0e37e35 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -3371,5 +3371,15 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
}
}
},
+/* Dell WD15 Dock */
+{
+ USB_DEVICE(0x0bda, 0x4014),
+ .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+ .vendor_name = "Dell",
+ .product_name = "WD15 Dock",
+ .profile_name = "Dell-WD15-Dock",
+ .ifnum = QUIRK_NO_INTERFACE
+ }
+},
#undef USB_DEVICE_VENDOR_SPEC
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index acbeb52..f4b6917 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -851,6 +851,36 @@ static int snd_usb_mbox2_boot_quirk(struct usb_device *dev)
return 0; /* Successful boot */
}
+static int snd_usb_axefx3_boot_quirk(struct usb_device *dev)
+{
+ int err;
+
+ dev_dbg(&dev->dev, "Waiting for Axe-Fx III to boot up...\n");
+
+ /* If the Axe-Fx III has not fully booted, it will timeout when trying
+ * to enable the audio streaming interface. A more generous timeout is
+ * used here to detect when the Axe-Fx III has finished booting as the
+ * set interface message will be acked once it has
+ */
+ err = usb_control_msg(dev, usb_sndctrlpipe(dev, 0),
+ USB_REQ_SET_INTERFACE, USB_RECIP_INTERFACE,
+ 1, 1, NULL, 0, 120000);
+ if (err < 0) {
+ dev_err(&dev->dev,
+ "failed waiting for Axe-Fx III to boot: %d\n", err);
+ return err;
+ }
+
+ dev_dbg(&dev->dev, "Axe-Fx III is now ready\n");
+
+ err = usb_set_interface(dev, 1, 0);
+ if (err < 0)
+ dev_dbg(&dev->dev,
+ "error stopping Axe-Fx III interface: %d\n", err);
+
+ return 0;
+}
+
/*
* Setup quirks
*/
@@ -1026,6 +1056,8 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev,
return snd_usb_fasttrackpro_boot_quirk(dev);
case USB_ID(0x047f, 0xc010): /* Plantronics Gamecom 780 */
return snd_usb_gamecon780_boot_quirk(dev);
+ case USB_ID(0x2466, 0x8010): /* Fractal Audio Axe-Fx 3 */
+ return snd_usb_axefx3_boot_quirk(dev);
}
return 0;
@@ -1327,20 +1359,47 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip,
/* XMOS based USB DACs */
switch (chip->usb_id) {
+ case USB_ID(0x1511, 0x0037): /* AURALiC VEGA */
+ case USB_ID(0x20b1, 0x0002): /* Wyred 4 Sound DAC-2 DSD */
+ case USB_ID(0x20b1, 0x2004): /* Matrix Audio X-SPDIF 2 */
case USB_ID(0x20b1, 0x3008): /* iFi Audio micro/nano iDSD */
case USB_ID(0x20b1, 0x2008): /* Matrix Audio X-Sabre */
case USB_ID(0x20b1, 0x300a): /* Matrix Audio Mini-i Pro */
case USB_ID(0x22d9, 0x0416): /* OPPO HA-1 */
+ case USB_ID(0x22d9, 0x0436): /* OPPO Sonica */
+ case USB_ID(0x22d9, 0x0461): /* OPPO UDP-205 */
+ case USB_ID(0x2522, 0x0012): /* LH Labs VI DAC Infinity */
+ case USB_ID(0x25ce, 0x001f): /* Mytek Brooklyn DAC */
+ case USB_ID(0x25ce, 0x0021): /* Mytek Manhattan DAC */
+ case USB_ID(0x25ce, 0x8025): /* Mytek Brooklyn DAC+ */
case USB_ID(0x2772, 0x0230): /* Pro-Ject Pre Box S2 Digital */
if (fp->altsetting == 2)
return SNDRV_PCM_FMTBIT_DSD_U32_BE;
break;
+ case USB_ID(0x0d8c, 0x0316): /* Hegel HD12 DSD */
+ case USB_ID(0x16b0, 0x06b2): /* NuPrime DAC-10 */
+ case USB_ID(0x16d0, 0x0733): /* Furutech ADL Stratos */
+ case USB_ID(0x16d0, 0x09db): /* NuPrime Audio DAC-9 */
+ case USB_ID(0x1db5, 0x0003): /* Bryston BDA3 */
case USB_ID(0x20b1, 0x000a): /* Gustard DAC-X20U */
+ case USB_ID(0x20b1, 0x2005): /* Denafrips Ares DAC */
case USB_ID(0x20b1, 0x2009): /* DIYINHK DSD DXD 384kHz USB to I2S/DSD */
case USB_ID(0x20b1, 0x2023): /* JLsounds I2SoverUSB */
+ case USB_ID(0x20b1, 0x3021): /* Eastern El. MiniMax Tube DAC Supreme */
case USB_ID(0x20b1, 0x3023): /* Aune X1S 32BIT/384 DSD DAC */
+ case USB_ID(0x20b1, 0x302d): /* Unison Research Unico CD Due */
+ case USB_ID(0x20b1, 0x3036): /* Holo Springs Level 3 R2R DAC */
+ case USB_ID(0x20b1, 0x307b): /* CH Precision C1 DAC */
+ case USB_ID(0x20b1, 0x3086): /* Singxer F-1 converter board */
+ case USB_ID(0x22d9, 0x0426): /* OPPO HA-2 */
+ case USB_ID(0x22e1, 0xca01): /* HDTA Serenade DSD */
+ case USB_ID(0x249c, 0x9326): /* M2Tech Young MkIII */
case USB_ID(0x2616, 0x0106): /* PS Audio NuWave DAC */
+ case USB_ID(0x2622, 0x0041): /* Audiolab M-DAC+ */
+ case USB_ID(0x27f7, 0x3002): /* W4S DAC-2v2SE */
+ case USB_ID(0x29a2, 0x0086): /* Mutec MC3+ USB */
+ case USB_ID(0x6b42, 0x0042): /* MSB Technology */
if (fp->altsetting == 3)
return SNDRV_PCM_FMTBIT_DSD_U32_BE;
break;
diff --git a/sound/usb/stream.c b/sound/usb/stream.c
index 5ed3345..bce3152 100644
--- a/sound/usb/stream.c
+++ b/sound/usb/stream.c
@@ -633,6 +633,391 @@ snd_usb_find_output_terminal_descriptor(struct usb_host_interface *ctrl_iface,
return NULL;
}
+static struct audioformat *
+audio_format_alloc_init(struct snd_usb_audio *chip,
+ struct usb_host_interface *alts,
+ int protocol, int iface_no, int altset_idx,
+ int altno, int num_channels, int clock)
+{
+ struct audioformat *fp;
+
+ fp = kzalloc(sizeof(*fp), GFP_KERNEL);
+ if (!fp)
+ return NULL;
+
+ fp->iface = iface_no;
+ fp->altsetting = altno;
+ fp->altset_idx = altset_idx;
+ fp->endpoint = get_endpoint(alts, 0)->bEndpointAddress;
+ fp->ep_attr = get_endpoint(alts, 0)->bmAttributes;
+ fp->datainterval = snd_usb_parse_datainterval(chip, alts);
+ fp->protocol = protocol;
+ fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize);
+ fp->channels = num_channels;
+ if (snd_usb_get_speed(chip->dev) == USB_SPEED_HIGH)
+ fp->maxpacksize = (((fp->maxpacksize >> 11) & 3) + 1)
+ * (fp->maxpacksize & 0x7ff);
+ fp->clock = clock;
+ INIT_LIST_HEAD(&fp->list);
+
+ return fp;
+}
+
+static struct audioformat *
+snd_usb_get_audioformat_uac12(struct snd_usb_audio *chip,
+ struct usb_host_interface *alts,
+ int protocol, int iface_no, int altset_idx,
+ int altno, int stream, int bm_quirk)
+{
+ struct usb_device *dev = chip->dev;
+ struct uac_format_type_i_continuous_descriptor *fmt;
+ unsigned int num_channels = 0, chconfig = 0;
+ struct audioformat *fp;
+ int clock = 0;
+ u64 format;
+
+ /* get audio formats */
+ if (protocol == UAC_VERSION_1) {
+ struct uac1_as_header_descriptor *as =
+ snd_usb_find_csint_desc(alts->extra, alts->extralen,
+ NULL, UAC_AS_GENERAL);
+ struct uac_input_terminal_descriptor *iterm;
+
+ if (!as) {
+ dev_err(&dev->dev,
+ "%u:%d : UAC_AS_GENERAL descriptor not found\n",
+ iface_no, altno);
+ return NULL;
+ }
+
+ if (as->bLength < sizeof(*as)) {
+ dev_err(&dev->dev,
+ "%u:%d : invalid UAC_AS_GENERAL desc\n",
+ iface_no, altno);
+ return NULL;
+ }
+
+ format = le16_to_cpu(as->wFormatTag); /* remember the format value */
+
+ iterm = snd_usb_find_input_terminal_descriptor(chip->ctrl_intf,
+ as->bTerminalLink);
+ if (iterm) {
+ num_channels = iterm->bNrChannels;
+ chconfig = le16_to_cpu(iterm->wChannelConfig);
+ }
+ } else { /* UAC_VERSION_2 */
+ struct uac2_input_terminal_descriptor *input_term;
+ struct uac2_output_terminal_descriptor *output_term;
+ struct uac2_as_header_descriptor *as =
+ snd_usb_find_csint_desc(alts->extra, alts->extralen,
+ NULL, UAC_AS_GENERAL);
+
+ if (!as) {
+ dev_err(&dev->dev,
+ "%u:%d : UAC_AS_GENERAL descriptor not found\n",
+ iface_no, altno);
+ return NULL;
+ }
+
+ if (as->bLength < sizeof(*as)) {
+ dev_err(&dev->dev,
+ "%u:%d : invalid UAC_AS_GENERAL desc\n",
+ iface_no, altno);
+ return NULL;
+ }
+
+ num_channels = as->bNrChannels;
+ format = le32_to_cpu(as->bmFormats);
+ chconfig = le32_to_cpu(as->bmChannelConfig);
+
+ /*
+ * lookup the terminal associated to this interface
+ * to extract the clock
+ */
+ input_term = snd_usb_find_input_terminal_descriptor(chip->ctrl_intf,
+ as->bTerminalLink);
+ if (input_term) {
+ clock = input_term->bCSourceID;
+ if (!chconfig && (num_channels == input_term->bNrChannels))
+ chconfig = le32_to_cpu(input_term->bmChannelConfig);
+ goto found_clock;
+ }
+
+ output_term = snd_usb_find_output_terminal_descriptor(chip->ctrl_intf,
+ as->bTerminalLink);
+ if (output_term) {
+ clock = output_term->bCSourceID;
+ goto found_clock;
+ }
+
+ dev_err(&dev->dev,
+ "%u:%d : bogus bTerminalLink %d\n",
+ iface_no, altno, as->bTerminalLink);
+ return NULL;
+ }
+
+found_clock:
+ /* get format type */
+ fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen,
+ NULL, UAC_FORMAT_TYPE);
+ if (!fmt) {
+ dev_err(&dev->dev,
+ "%u:%d : no UAC_FORMAT_TYPE desc\n",
+ iface_no, altno);
+ return NULL;
+ }
+ if (((protocol == UAC_VERSION_1) && (fmt->bLength < 8))
+ || ((protocol == UAC_VERSION_2) &&
+ (fmt->bLength < 6))) {
+ dev_err(&dev->dev,
+ "%u:%d : invalid UAC_FORMAT_TYPE desc\n",
+ iface_no, altno);
+ return NULL;
+ }
+
+ /*
+ * Blue Microphones workaround: The last altsetting is
+ * identical with the previous one, except for a larger
+ * packet size, but is actually a mislabeled two-channel
+ * setting; ignore it.
+ *
+ * Part 2: analyze quirk flag and format
+ */
+ if (bm_quirk && fmt->bNrChannels == 1 && fmt->bSubframeSize == 2)
+ return NULL;
+
+ fp = audio_format_alloc_init(chip, alts, protocol, iface_no,
+ altset_idx, altno, num_channels, clock);
+ if (!fp)
+ return ERR_PTR(-ENOMEM);
+
+ fp->attributes = parse_uac_endpoint_attributes(chip, alts, protocol,
+ iface_no);
+
+ /* some quirks for attributes here */
+ snd_usb_audioformat_attributes_quirk(chip, fp, stream);
+
+ /* ok, let's parse further... */
+ if (snd_usb_parse_audio_format(chip, fp, format,
+ fmt, stream) < 0) {
+ kfree(fp->rate_table);
+ kfree(fp);
+ return NULL;
+ }
+
+ /* Create chmap */
+ if (fp->channels != num_channels)
+ chconfig = 0;
+
+ fp->chmap = convert_chmap(fp->channels, chconfig, protocol);
+
+ return fp;
+}
+
+static struct audioformat *
+snd_usb_get_audioformat_uac3(struct snd_usb_audio *chip,
+ struct usb_host_interface *alts,
+ int iface_no, int altset_idx,
+ int altno, int stream)
+{
+ struct usb_device *dev = chip->dev;
+ struct uac3_input_terminal_descriptor *input_term;
+ struct uac3_output_terminal_descriptor *output_term;
+ struct uac3_cluster_header_descriptor *cluster;
+ struct uac3_as_header_descriptor *as = NULL;
+ struct uac3_hc_descriptor_header hc_header;
+ struct snd_pcm_chmap_elem *chmap;
+ unsigned char badd_profile;
+ u64 badd_formats = 0;
+ unsigned int num_channels;
+ struct audioformat *fp;
+ u16 cluster_id, wLength;
+ int clock = 0;
+ int err;
+
+ badd_profile = chip->badd_profile;
+
+ if (badd_profile >= UAC3_FUNCTION_SUBCLASS_GENERIC_IO) {
+ unsigned int maxpacksize =
+ le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize);
+
+ switch (maxpacksize) {
+ default:
+ dev_err(&dev->dev,
+ "%u:%d : incorrect wMaxPacketSize for BADD profile\n",
+ iface_no, altno);
+ return NULL;
+ case UAC3_BADD_EP_MAXPSIZE_SYNC_MONO_16:
+ case UAC3_BADD_EP_MAXPSIZE_ASYNC_MONO_16:
+ badd_formats = SNDRV_PCM_FMTBIT_S16_LE;
+ num_channels = 1;
+ break;
+ case UAC3_BADD_EP_MAXPSIZE_SYNC_MONO_24:
+ case UAC3_BADD_EP_MAXPSIZE_ASYNC_MONO_24:
+ badd_formats = SNDRV_PCM_FMTBIT_S24_3LE;
+ num_channels = 1;
+ break;
+ case UAC3_BADD_EP_MAXPSIZE_SYNC_STEREO_16:
+ case UAC3_BADD_EP_MAXPSIZE_ASYNC_STEREO_16:
+ badd_formats = SNDRV_PCM_FMTBIT_S16_LE;
+ num_channels = 2;
+ break;
+ case UAC3_BADD_EP_MAXPSIZE_SYNC_STEREO_24:
+ case UAC3_BADD_EP_MAXPSIZE_ASYNC_STEREO_24:
+ badd_formats = SNDRV_PCM_FMTBIT_S24_3LE;
+ num_channels = 2;
+ break;
+ }
+
+ chmap = kzalloc(sizeof(*chmap), GFP_KERNEL);
+ if (!chmap)
+ return ERR_PTR(-ENOMEM);
+
+ if (num_channels == 1) {
+ chmap->map[0] = SNDRV_CHMAP_MONO;
+ } else {
+ chmap->map[0] = SNDRV_CHMAP_FL;
+ chmap->map[1] = SNDRV_CHMAP_FR;
+ }
+
+ chmap->channels = num_channels;
+ clock = UAC3_BADD_CS_ID9;
+ goto found_clock;
+ }
+
+ as = snd_usb_find_csint_desc(alts->extra, alts->extralen,
+ NULL, UAC_AS_GENERAL);
+ if (!as) {
+ dev_err(&dev->dev,
+ "%u:%d : UAC_AS_GENERAL descriptor not found\n",
+ iface_no, altno);
+ return NULL;
+ }
+
+ if (as->bLength < sizeof(*as)) {
+ dev_err(&dev->dev,
+ "%u:%d : invalid UAC_AS_GENERAL desc\n",
+ iface_no, altno);
+ return NULL;
+ }
+
+ cluster_id = le16_to_cpu(as->wClusterDescrID);
+ if (!cluster_id) {
+ dev_err(&dev->dev,
+ "%u:%d : no cluster descriptor\n",
+ iface_no, altno);
+ return NULL;
+ }
+
+ /*
+ * Get number of channels and channel map through
+ * High Capability Cluster Descriptor
+ *
+ * First step: get High Capability header and
+ * read size of Cluster Descriptor
+ */
+ err = snd_usb_ctl_msg(chip->dev,
+ usb_rcvctrlpipe(chip->dev, 0),
+ UAC3_CS_REQ_HIGH_CAPABILITY_DESCRIPTOR,
+ USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN,
+ cluster_id,
+ snd_usb_ctrl_intf(chip),
+ &hc_header, sizeof(hc_header));
+ if (err < 0)
+ return ERR_PTR(err);
+ else if (err != sizeof(hc_header)) {
+ dev_err(&dev->dev,
+ "%u:%d : can't get High Capability descriptor\n",
+ iface_no, altno);
+ return ERR_PTR(-EIO);
+ }
+
+ /*
+ * Second step: allocate needed amount of memory
+ * and request Cluster Descriptor
+ */
+ wLength = le16_to_cpu(hc_header.wLength);
+ cluster = kzalloc(wLength, GFP_KERNEL);
+ if (!cluster)
+ return ERR_PTR(-ENOMEM);
+ err = snd_usb_ctl_msg(chip->dev,
+ usb_rcvctrlpipe(chip->dev, 0),
+ UAC3_CS_REQ_HIGH_CAPABILITY_DESCRIPTOR,
+ USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN,
+ cluster_id,
+ snd_usb_ctrl_intf(chip),
+ cluster, wLength);
+ if (err < 0) {
+ kfree(cluster);
+ return ERR_PTR(err);
+ } else if (err != wLength) {
+ dev_err(&dev->dev,
+ "%u:%d : can't get Cluster Descriptor\n",
+ iface_no, altno);
+ kfree(cluster);
+ return ERR_PTR(-EIO);
+ }
+
+ num_channels = cluster->bNrChannels;
+ chmap = convert_chmap_v3(cluster);
+ kfree(cluster);
+
+ /*
+ * lookup the terminal associated to this interface
+ * to extract the clock
+ */
+ input_term = snd_usb_find_input_terminal_descriptor(chip->ctrl_intf,
+ as->bTerminalLink);
+ if (input_term) {
+ clock = input_term->bCSourceID;
+ goto found_clock;
+ }
+
+ output_term = snd_usb_find_output_terminal_descriptor(chip->ctrl_intf,
+ as->bTerminalLink);
+ if (output_term) {
+ clock = output_term->bCSourceID;
+ goto found_clock;
+ }
+
+ dev_err(&dev->dev, "%u:%d : bogus bTerminalLink %d\n",
+ iface_no, altno, as->bTerminalLink);
+ return NULL;
+
+found_clock:
+ fp = audio_format_alloc_init(chip, alts, UAC_VERSION_3, iface_no,
+ altset_idx, altno, num_channels, clock);
+ if (!fp)
+ return ERR_PTR(-ENOMEM);
+
+ fp->chmap = chmap;
+
+ if (badd_profile >= UAC3_FUNCTION_SUBCLASS_GENERIC_IO) {
+ fp->attributes = 0; /* No attributes */
+
+ fp->fmt_type = UAC_FORMAT_TYPE_I;
+ fp->formats = badd_formats;
+
+ fp->nr_rates = 0; /* SNDRV_PCM_RATE_CONTINUOUS */
+ fp->rate_min = UAC3_BADD_SAMPLING_RATE;
+ fp->rate_max = UAC3_BADD_SAMPLING_RATE;
+ fp->rates = SNDRV_PCM_RATE_CONTINUOUS;
+
+ } else {
+ fp->attributes = parse_uac_endpoint_attributes(chip, alts,
+ UAC_VERSION_3,
+ iface_no);
+ /* ok, let's parse further... */
+ if (snd_usb_parse_audio_format_v3(chip, fp, as, stream) < 0) {
+ kfree(fp->rate_table);
+ kfree(fp);
+ return NULL;
+ }
+ }
+
+ return fp;
+}
+
int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no)
{
struct usb_device *dev;
@@ -640,13 +1025,8 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no)
struct usb_host_interface *alts;
struct usb_interface_descriptor *altsd;
int i, altno, err, stream;
- u64 format = 0;
- unsigned int num_channels = 0;
struct audioformat *fp = NULL;
- int num, protocol, clock = 0;
- struct uac_format_type_i_continuous_descriptor *fmt = NULL;
- struct snd_pcm_chmap_elem *chmap_v3 = NULL;
- unsigned int chconfig;
+ int num, protocol;
dev = chip->dev;
@@ -695,303 +1075,48 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no)
protocol <= 2)
protocol = UAC_VERSION_1;
- chconfig = 0;
- /* get audio formats */
switch (protocol) {
default:
dev_dbg(&dev->dev, "%u:%d: unknown interface protocol %#02x, assuming v1\n",
iface_no, altno, protocol);
protocol = UAC_VERSION_1;
/* fall through */
-
- case UAC_VERSION_1: {
- struct uac1_as_header_descriptor *as =
- snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_AS_GENERAL);
- struct uac_input_terminal_descriptor *iterm;
-
- if (!as) {
- dev_err(&dev->dev,
- "%u:%d : UAC_AS_GENERAL descriptor not found\n",
- iface_no, altno);
- continue;
- }
-
- if (as->bLength < sizeof(*as)) {
- dev_err(&dev->dev,
- "%u:%d : invalid UAC_AS_GENERAL desc\n",
- iface_no, altno);
- continue;
- }
-
- format = le16_to_cpu(as->wFormatTag); /* remember the format value */
-
- iterm = snd_usb_find_input_terminal_descriptor(chip->ctrl_intf,
- as->bTerminalLink);
- if (iterm) {
- num_channels = iterm->bNrChannels;
- chconfig = le16_to_cpu(iterm->wChannelConfig);
- }
-
- break;
- }
-
+ case UAC_VERSION_1:
+ /* fall through */
case UAC_VERSION_2: {
- struct uac2_input_terminal_descriptor *input_term;
- struct uac2_output_terminal_descriptor *output_term;
- struct uac2_as_header_descriptor *as =
- snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_AS_GENERAL);
-
- if (!as) {
- dev_err(&dev->dev,
- "%u:%d : UAC_AS_GENERAL descriptor not found\n",
- iface_no, altno);
- continue;
- }
-
- if (as->bLength < sizeof(*as)) {
- dev_err(&dev->dev,
- "%u:%d : invalid UAC_AS_GENERAL desc\n",
- iface_no, altno);
- continue;
- }
-
- num_channels = as->bNrChannels;
- format = le32_to_cpu(as->bmFormats);
- chconfig = le32_to_cpu(as->bmChannelConfig);
-
- /* lookup the terminal associated to this interface
- * to extract the clock */
- input_term = snd_usb_find_input_terminal_descriptor(chip->ctrl_intf,
- as->bTerminalLink);
- if (input_term) {
- clock = input_term->bCSourceID;
- if (!chconfig && (num_channels == input_term->bNrChannels))
- chconfig = le32_to_cpu(input_term->bmChannelConfig);
- break;
- }
-
- output_term = snd_usb_find_output_terminal_descriptor(chip->ctrl_intf,
- as->bTerminalLink);
- if (output_term) {
- clock = output_term->bCSourceID;
- break;
- }
-
- dev_err(&dev->dev,
- "%u:%d : bogus bTerminalLink %d\n",
- iface_no, altno, as->bTerminalLink);
- continue;
- }
-
- case UAC_VERSION_3: {
- struct uac3_input_terminal_descriptor *input_term;
- struct uac3_output_terminal_descriptor *output_term;
- struct uac3_as_header_descriptor *as;
- struct uac3_cluster_header_descriptor *cluster;
- struct uac3_hc_descriptor_header hc_header;
- u16 cluster_id, wLength;
-
- as = snd_usb_find_csint_desc(alts->extra,
- alts->extralen,
- NULL, UAC_AS_GENERAL);
-
- if (!as) {
- dev_err(&dev->dev,
- "%u:%d : UAC_AS_GENERAL descriptor not found\n",
- iface_no, altno);
- continue;
- }
-
- if (as->bLength < sizeof(*as)) {
- dev_err(&dev->dev,
- "%u:%d : invalid UAC_AS_GENERAL desc\n",
- iface_no, altno);
- continue;
- }
-
- cluster_id = le16_to_cpu(as->wClusterDescrID);
- if (!cluster_id) {
- dev_err(&dev->dev,
- "%u:%d : no cluster descriptor\n",
- iface_no, altno);
- continue;
- }
-
- /*
- * Get number of channels and channel map through
- * High Capability Cluster Descriptor
- *
- * First step: get High Capability header and
- * read size of Cluster Descriptor
- */
- err = snd_usb_ctl_msg(chip->dev,
- usb_rcvctrlpipe(chip->dev, 0),
- UAC3_CS_REQ_HIGH_CAPABILITY_DESCRIPTOR,
- USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN,
- cluster_id,
- snd_usb_ctrl_intf(chip),
- &hc_header, sizeof(hc_header));
- if (err < 0)
- return err;
- else if (err != sizeof(hc_header)) {
- dev_err(&dev->dev,
- "%u:%d : can't get High Capability descriptor\n",
- iface_no, altno);
- return -EIO;
- }
-
- /*
- * Second step: allocate needed amount of memory
- * and request Cluster Descriptor
- */
- wLength = le16_to_cpu(hc_header.wLength);
- cluster = kzalloc(wLength, GFP_KERNEL);
- if (!cluster)
- return -ENOMEM;
- err = snd_usb_ctl_msg(chip->dev,
- usb_rcvctrlpipe(chip->dev, 0),
- UAC3_CS_REQ_HIGH_CAPABILITY_DESCRIPTOR,
- USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN,
- cluster_id,
- snd_usb_ctrl_intf(chip),
- cluster, wLength);
- if (err < 0) {
- kfree(cluster);
- return err;
- } else if (err != wLength) {
- dev_err(&dev->dev,
- "%u:%d : can't get Cluster Descriptor\n",
- iface_no, altno);
- kfree(cluster);
- return -EIO;
- }
-
- num_channels = cluster->bNrChannels;
- chmap_v3 = convert_chmap_v3(cluster);
-
- kfree(cluster);
-
- format = le64_to_cpu(as->bmFormats);
-
- /* lookup the terminal associated to this interface
- * to extract the clock */
- input_term = snd_usb_find_input_terminal_descriptor(
- chip->ctrl_intf,
- as->bTerminalLink);
-
- if (input_term) {
- clock = input_term->bCSourceID;
- break;
- }
-
- output_term = snd_usb_find_output_terminal_descriptor(chip->ctrl_intf,
- as->bTerminalLink);
- if (output_term) {
- clock = output_term->bCSourceID;
- break;
- }
-
- dev_err(&dev->dev,
- "%u:%d : bogus bTerminalLink %d\n",
- iface_no, altno, as->bTerminalLink);
- continue;
- }
- }
-
- if (protocol == UAC_VERSION_1 || protocol == UAC_VERSION_2) {
- /* get format type */
- fmt = snd_usb_find_csint_desc(alts->extra,
- alts->extralen,
- NULL, UAC_FORMAT_TYPE);
- if (!fmt) {
- dev_err(&dev->dev,
- "%u:%d : no UAC_FORMAT_TYPE desc\n",
- iface_no, altno);
- continue;
- }
- if (((protocol == UAC_VERSION_1) && (fmt->bLength < 8))
- || ((protocol == UAC_VERSION_2) &&
- (fmt->bLength < 6))) {
- dev_err(&dev->dev,
- "%u:%d : invalid UAC_FORMAT_TYPE desc\n",
- iface_no, altno);
- continue;
- }
+ int bm_quirk = 0;
/*
* Blue Microphones workaround: The last altsetting is
* identical with the previous one, except for a larger
* packet size, but is actually a mislabeled two-channel
* setting; ignore it.
+ *
+ * Part 1: prepare quirk flag
*/
- if (fmt->bNrChannels == 1 &&
- fmt->bSubframeSize == 2 &&
- altno == 2 && num == 3 &&
+ if (altno == 2 && num == 3 &&
fp && fp->altsetting == 1 && fp->channels == 1 &&
fp->formats == SNDRV_PCM_FMTBIT_S16_LE &&
protocol == UAC_VERSION_1 &&
le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize) ==
fp->maxpacksize * 2)
- continue;
+ bm_quirk = 1;
+
+ fp = snd_usb_get_audioformat_uac12(chip, alts, protocol,
+ iface_no, i, altno,
+ stream, bm_quirk);
+ break;
+ }
+ case UAC_VERSION_3:
+ fp = snd_usb_get_audioformat_uac3(chip, alts,
+ iface_no, i, altno, stream);
+ break;
}
- fp = kzalloc(sizeof(*fp), GFP_KERNEL);
if (!fp)
- return -ENOMEM;
-
- fp->iface = iface_no;
- fp->altsetting = altno;
- fp->altset_idx = i;
- fp->endpoint = get_endpoint(alts, 0)->bEndpointAddress;
- fp->ep_attr = get_endpoint(alts, 0)->bmAttributes;
- fp->datainterval = snd_usb_parse_datainterval(chip, alts);
- fp->protocol = protocol;
- fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize);
- fp->channels = num_channels;
- if (snd_usb_get_speed(dev) == USB_SPEED_HIGH)
- fp->maxpacksize = (((fp->maxpacksize >> 11) & 3) + 1)
- * (fp->maxpacksize & 0x7ff);
- fp->attributes = parse_uac_endpoint_attributes(chip, alts, protocol, iface_no);
- fp->clock = clock;
- INIT_LIST_HEAD(&fp->list);
-
- /* some quirks for attributes here */
- snd_usb_audioformat_attributes_quirk(chip, fp, stream);
-
- /* ok, let's parse further... */
- if (protocol == UAC_VERSION_1 || protocol == UAC_VERSION_2) {
- if (snd_usb_parse_audio_format(chip, fp, format,
- fmt, stream) < 0) {
- kfree(fp->rate_table);
- kfree(fp);
- fp = NULL;
- continue;
- }
- } else {
- struct uac3_as_header_descriptor *as;
-
- as = snd_usb_find_csint_desc(alts->extra,
- alts->extralen,
- NULL, UAC_AS_GENERAL);
-
- if (snd_usb_parse_audio_format_v3(chip, fp, as,
- stream) < 0) {
- kfree(fp->rate_table);
- kfree(fp);
- fp = NULL;
- continue;
- }
- }
-
- /* Create chmap */
- if (fp->channels != num_channels)
- chconfig = 0;
-
- if (protocol == UAC_VERSION_3)
- fp->chmap = chmap_v3;
- else
- fp->chmap = convert_chmap(fp->channels, chconfig,
- protocol);
+ continue;
+ else if (IS_ERR(fp))
+ return PTR_ERR(fp);
dev_dbg(&dev->dev, "%u:%d: add audio endpoint %#x\n", iface_no, altno, fp->endpoint);
err = snd_usb_add_audio_stream(chip, stream, fp);
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index 4d5c89a..7b28cbd 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -49,6 +49,8 @@ struct snd_usb_audio {
int num_suspended_intf;
int sample_rate_read_error;
+ int badd_profile; /* UAC3 BADD profile */
+
struct list_head pcm_list; /* list of pcm streams */
struct list_head ep_list; /* list of audio-related endpoints */
int pcm_devs;
@@ -59,6 +61,9 @@ struct snd_usb_audio {
int setup; /* from the 'device_setup' module param */
bool autoclock; /* from the 'autoclock' module param */
+ bool keep_iface; /* keep interface/altset after closing
+ * or parameter change
+ */
struct usb_host_interface *ctrl_intf; /* the audio control interface */
};
@@ -109,6 +114,7 @@ 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;
const void *data;
