diff options
Diffstat (limited to 'sound')
118 files changed, 8602 insertions, 1423 deletions
diff --git a/sound/Kconfig b/sound/Kconfig index 6833db9002ec..1140e9988fc5 100644 --- a/sound/Kconfig +++ b/sound/Kconfig @@ -96,6 +96,8 @@ source "sound/x86/Kconfig" source "sound/synth/Kconfig" +source "sound/xen/Kconfig" + endif # SND endif # !UML diff --git a/sound/Makefile b/sound/Makefile index 99d8c31262c8..797ecdcd35e2 100644 --- a/sound/Makefile +++ b/sound/Makefile @@ -5,7 +5,7 @@ obj-$(CONFIG_SOUND) += soundcore.o obj-$(CONFIG_DMASOUND) += oss/dmasound/ obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ sh/ synth/ usb/ \ - firewire/ sparc/ spi/ parisc/ pcmcia/ mips/ soc/ atmel/ hda/ x86/ + firewire/ sparc/ spi/ parisc/ pcmcia/ mips/ soc/ atmel/ hda/ x86/ xen/ obj-$(CONFIG_SND_AOA) += aoa/ # This one must be compilable even if sound is configured out diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 4563432badba..4b01a37c836e 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -1001,7 +1001,7 @@ static int snd_compress_proc_init(struct snd_compr *compr) compr->card->proc_root); if (!entry) return -ENOMEM; - entry->mode = S_IFDIR | S_IRUGO | S_IXUGO; + entry->mode = S_IFDIR | 0555; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); return -ENOMEM; diff --git a/sound/core/device.c b/sound/core/device.c index cb0e46f66cc9..535102d564e3 100644 --- a/sound/core/device.c +++ b/sound/core/device.c @@ -240,6 +240,15 @@ void snd_device_free_all(struct snd_card *card) if (snd_BUG_ON(!card)) return; + list_for_each_entry_safe_reverse(dev, next, &card->devices, list) { + /* exception: free ctl and lowlevel stuff later */ + if (dev->type == SNDRV_DEV_CONTROL || + dev->type == SNDRV_DEV_LOWLEVEL) + continue; + __snd_device_free(dev); + } + + /* free all */ list_for_each_entry_safe_reverse(dev, next, &card->devices, list) __snd_device_free(dev); } diff --git a/sound/core/info.c b/sound/core/info.c index 4b36767af9e1..fe502bc5e6d2 100644 --- a/sound/core/info.c +++ b/sound/core/info.c @@ -454,7 +454,7 @@ static struct snd_info_entry *create_subdir(struct module *mod, entry = snd_info_create_module_entry(mod, name, NULL); if (!entry) return NULL; - entry->mode = S_IFDIR | S_IRUGO | S_IXUGO; + entry->mode = S_IFDIR | 0555; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); return NULL; @@ -470,7 +470,7 @@ int __init snd_info_init(void) snd_proc_root = snd_info_create_entry("asound", NULL); if (!snd_proc_root) return -ENOMEM; - snd_proc_root->mode = S_IFDIR | S_IRUGO | S_IXUGO; + snd_proc_root->mode = S_IFDIR | 0555; snd_proc_root->p = proc_mkdir("asound", NULL); if (!snd_proc_root->p) goto error; @@ -716,7 +716,7 @@ snd_info_create_entry(const char *name, struct snd_info_entry *parent) kfree(entry); return NULL; } - entry->mode = S_IFREG | S_IRUGO; + entry->mode = S_IFREG | 0444; entry->content = SNDRV_INFO_CONTENT_TEXT; mutex_init(&entry->access); INIT_LIST_HEAD(&entry->children); diff --git a/sound/core/init.c b/sound/core/init.c index 79b4df1c1dc7..4849c611c0fe 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -703,7 +703,7 @@ card_id_store_attr(struct device *dev, struct device_attribute *attr, return count; } -static DEVICE_ATTR(id, S_IRUGO | S_IWUSR, card_id_show_attr, card_id_store_attr); +static DEVICE_ATTR(id, 0644, card_id_show_attr, card_id_store_attr); static ssize_t card_number_show_attr(struct device *dev, @@ -713,7 +713,7 @@ card_number_show_attr(struct device *dev, return scnprintf(buf, PAGE_SIZE, "%i\n", card->number); } -static DEVICE_ATTR(number, S_IRUGO, card_number_show_attr, NULL); +static DEVICE_ATTR(number, 0444, card_number_show_attr, NULL); static struct attribute *card_dev_attrs[] = { &dev_attr_id.attr, diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index 379bf486ccc7..64d904bee8bb 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -1247,7 +1247,7 @@ static void snd_mixer_oss_proc_init(struct snd_mixer_oss *mixer) if (! entry) return; entry->content = SNDRV_INFO_CONTENT_TEXT; - entry->mode = S_IFREG | S_IRUGO | S_IWUSR; + entry->mode = S_IFREG | 0644; entry->c.text.read = snd_mixer_oss_proc_read; entry->c.text.write = snd_mixer_oss_proc_write; entry->private_data = mixer; diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 1980f68246cb..905a53c1cde5 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -3045,7 +3045,7 @@ static void snd_pcm_oss_proc_init(struct snd_pcm *pcm) continue; if ((entry = snd_info_create_card_entry(pcm->card, "oss", pstr->proc_root)) != NULL) { entry->content = SNDRV_INFO_CONTENT_TEXT; - entry->mode = S_IFREG | S_IRUGO | S_IWUSR; + entry->mode = S_IFREG | 0644; entry->c.text.read = snd_pcm_oss_proc_read; entry->c.text.write = snd_pcm_oss_proc_write; entry->private_data = pstr; diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 66ac89aad681..c352bfb973cc 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -530,7 +530,7 @@ static int snd_pcm_stream_proc_init(struct snd_pcm_str *pstr) pcm->card->proc_root); if (!entry) return -ENOMEM; - entry->mode = S_IFDIR | S_IRUGO | S_IXUGO; + entry->mode = S_IFDIR | 0555; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); return -ENOMEM; @@ -552,7 +552,7 @@ static int snd_pcm_stream_proc_init(struct snd_pcm_str *pstr) if (entry) { entry->c.text.read = snd_pcm_xrun_debug_read; entry->c.text.write = snd_pcm_xrun_debug_write; - entry->mode |= S_IWUSR; + entry->mode |= 0200; entry->private_data = pstr; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); @@ -590,7 +590,7 @@ static int snd_pcm_substream_proc_init(struct snd_pcm_substream *substream) substream->pstr->proc_root); if (!entry) return -ENOMEM; - entry->mode = S_IFDIR | S_IRUGO | S_IXUGO; + entry->mode = S_IFDIR | 0555; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); return -ENOMEM; @@ -647,7 +647,7 @@ static int snd_pcm_substream_proc_init(struct snd_pcm_substream *substream) entry->private_data = substream; entry->c.text.read = NULL; entry->c.text.write = snd_pcm_xrun_injection_write; - entry->mode = S_IFREG | S_IWUSR; + entry->mode = S_IFREG | 0200; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); entry = NULL; @@ -1087,7 +1087,7 @@ static ssize_t show_pcm_class(struct device *dev, return snprintf(buf, PAGE_SIZE, "%s\n", str); } -static DEVICE_ATTR(pcm_class, S_IRUGO, show_pcm_class, NULL); +static DEVICE_ATTR(pcm_class, 0444, show_pcm_class, NULL); static struct attribute *pcm_dev_attrs[] = { &dev_attr_pcm_class.attr, NULL diff --git a/sound/core/pcm_compat.c b/sound/core/pcm_compat.c index 6491afbb5fd5..39d853bfa5ac 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 f4a19509cccf..44b5ae833082 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 16f254732b2a..7a499d02df6c 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_memory.c b/sound/core/pcm_memory.c index ae33e456708c..4b5356a10315 100644 --- a/sound/core/pcm_memory.c +++ b/sound/core/pcm_memory.c @@ -201,7 +201,7 @@ static inline void preallocate_info_init(struct snd_pcm_substream *substream) if ((entry = snd_info_create_card_entry(substream->pcm->card, "prealloc", substream->proc_root)) != NULL) { entry->c.text.read = snd_pcm_lib_preallocate_proc_read; entry->c.text.write = snd_pcm_lib_preallocate_proc_write; - entry->mode |= S_IWUSR; + entry->mode |= 0200; entry->private_data = substream; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 0e875d5a9e86..04c6301394d0 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; - - poll_wait(file, &runtime->sleep, wait); + return ok | EPOLLERR; - snd_pcm_stream_lock_irq(substream); - avail = snd_pcm_playback_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 = 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); + mask = 0; snd_pcm_stream_lock_irq(substream); - avail = snd_pcm_capture_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 = 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/seq/seq_ports.c b/sound/core/seq/seq_ports.c index d21ece9f8d73..24d90abfc64d 100644 --- a/sound/core/seq/seq_ports.c +++ b/sound/core/seq/seq_ports.c @@ -669,7 +669,7 @@ int snd_seq_event_port_attach(int client, /* Set up the port */ memset(&portinfo, 0, sizeof(portinfo)); portinfo.addr.client = client; - strlcpy(portinfo.name, portname ? portname : "Unamed port", + strlcpy(portinfo.name, portname ? portname : "Unnamed port", sizeof(portinfo.name)); portinfo.capability = cap; diff --git a/sound/core/seq/seq_timer.c b/sound/core/seq/seq_timer.c index 23167578231f..f587d0e27476 100644 --- a/sound/core/seq/seq_timer.c +++ b/sound/core/seq/seq_timer.c @@ -371,9 +371,7 @@ static int initialize_timer(struct snd_seq_timer *tmr) tmr->ticks = 1; if (!(t->hw.flags & SNDRV_TIMER_HW_SLAVE)) { - unsigned long r = t->hw.resolution; - if (! r && t->hw.c_resolution) - r = t->hw.c_resolution(t); + unsigned long r = snd_timer_resolution(tmr->timeri); if (r) { tmr->ticks = (unsigned int)(1000000000uL / (r * freq)); if (! tmr->ticks) diff --git a/sound/core/timer.c b/sound/core/timer.c index 0ddcae495838..665089c45560 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -427,25 +427,35 @@ int snd_timer_close(struct snd_timer_instance *timeri) } EXPORT_SYMBOL(snd_timer_close); +static unsigned long snd_timer_hw_resolution(struct snd_timer *timer) +{ + if (timer->hw.c_resolution) + return timer->hw.c_resolution(timer); + else + return timer->hw.resolution; +} + unsigned long snd_timer_resolution(struct snd_timer_instance *timeri) { struct snd_timer * timer; + unsigned long ret = 0; + unsigned long flags; if (timeri == NULL) return 0; timer = timeri->timer; if (timer) { - if (timer->hw.c_resolution) - return timer->hw.c_resolution(timer); - return timer->hw.resolution; + spin_lock_irqsave(&timer->lock, flags); + ret = snd_timer_hw_resolution(timer); + spin_unlock_irqrestore(&timer->lock, flags); } - return 0; + return ret; } EXPORT_SYMBOL(snd_timer_resolution); static void snd_timer_notify1(struct snd_timer_instance *ti, int event) { - struct snd_timer *timer; + struct snd_timer *timer = ti->timer; unsigned long resolution = 0; struct snd_timer_instance *ts; struct timespec tstamp; @@ -457,14 +467,14 @@ static void snd_timer_notify1(struct snd_timer_instance *ti, int event) if (snd_BUG_ON(event < SNDRV_TIMER_EVENT_START || event > SNDRV_TIMER_EVENT_PAUSE)) return; - if (event == SNDRV_TIMER_EVENT_START || - event == SNDRV_TIMER_EVENT_CONTINUE) - resolution = snd_timer_resolution(ti); + if (timer && + (event == SNDRV_TIMER_EVENT_START || + event == SNDRV_TIMER_EVENT_CONTINUE)) + resolution = snd_timer_hw_resolution(timer); if (ti->ccallback) ti->ccallback(ti, event, &tstamp, resolution); if (ti->flags & SNDRV_TIMER_IFLG_SLAVE) return; - timer = ti->timer; if (timer == NULL) return; if (timer->hw.flags & SNDRV_TIMER_HW_SLAVE) @@ -771,10 +781,7 @@ void snd_timer_interrupt(struct snd_timer * timer, unsigned long ticks_left) spin_lock_irqsave(&timer->lock, flags); /* remember the current resolution */ - if (timer->hw.c_resolution) - resolution = timer->hw.c_resolution(timer); - else - resolution = timer->hw.resolution; + resolution = snd_timer_hw_resolution(timer); /* loop for all active instances * Here we cannot use list_for_each_entry because the active_list of a @@ -1014,12 +1021,8 @@ void snd_timer_notify(struct snd_timer *timer, int event, struct timespec *tstam spin_lock_irqsave(&timer->lock, flags); if (event == SNDRV_TIMER_EVENT_MSTART || event == SNDRV_TIMER_EVENT_MCONTINUE || - event == SNDRV_TIMER_EVENT_MRESUME) { - if (timer->hw.c_resolution) - resolution = timer->hw.c_resolution(timer); - else - resolution = timer->hw.resolution; - } + event == SNDRV_TIMER_EVENT_MRESUME) + resolution = snd_timer_hw_resolution(timer); list_for_each_entry(ti, &timer->active_list_head, active_list) { if (ti->ccallback) ti->ccallback(ti, event, tstamp, resolution); @@ -1656,10 +1659,8 @@ static int snd_timer_user_gstatus(struct file *file, mutex_lock(®ister_mutex); t = snd_timer_find(&tid); if (t != NULL) { - if (t->hw.c_resolution) - gstatus.resolution = t->hw.c_resolution(t); - else - gstatus.resolution = t->hw.resolution; + spin_lock_irq(&t->lock); + gstatus.resolution = snd_timer_hw_resolution(t); if (t->hw.precise_resolution) { t->hw.precise_resolution(t, &gstatus.resolution_num, &gstatus.resolution_den); @@ -1667,6 +1668,7 @@ static int snd_timer_user_gstatus(struct file *file, gstatus.resolution_num = gstatus.resolution; gstatus.resolution_den = 1000000000uL; } + spin_unlock_irq(&t->lock); } else { err = -ENODEV; } diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c index 9e96186742d0..58fa3f94722a 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/drivers/aloop.c b/sound/drivers/aloop.c index eab7f594ebe7..78a2fdc38531 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -768,20 +768,7 @@ static int loopback_close(struct snd_pcm_substream *substream) return 0; } -static const struct snd_pcm_ops loopback_playback_ops = { - .open = loopback_open, - .close = loopback_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = loopback_hw_params, - .hw_free = loopback_hw_free, - .prepare = loopback_prepare, - .trigger = loopback_trigger, - .pointer = loopback_pointer, - .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, -}; - -static const struct snd_pcm_ops loopback_capture_ops = { +static const struct snd_pcm_ops loopback_pcm_ops = { .open = loopback_open, .close = loopback_close, .ioctl = snd_pcm_lib_ioctl, @@ -804,8 +791,8 @@ static int loopback_pcm_new(struct loopback *loopback, substreams, substreams, &pcm); if (err < 0) return err; - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &loopback_playback_ops); - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &loopback_capture_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &loopback_pcm_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &loopback_pcm_ops); pcm->private_data = loopback; pcm->info_flags = 0; diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index 8fb9a54fe8ba..9af154db530a 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -1042,7 +1042,7 @@ static void dummy_proc_init(struct snd_dummy *chip) if (!snd_card_proc_new(chip->card, "dummy_pcm", &entry)) { snd_info_set_text_ops(entry, chip, dummy_proc_read); entry->c.text.write = dummy_proc_write; - entry->mode |= S_IWUSR; + entry->mode |= 0200; entry->private_data = chip; } } diff --git a/sound/drivers/mts64.c b/sound/drivers/mts64.c index f32e81342247..b68e71ca7abd 100644 --- a/sound/drivers/mts64.c +++ b/sound/drivers/mts64.c @@ -41,11 +41,11 @@ static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; static struct platform_device *platform_devices[SNDRV_CARDS]; static int device_count; -module_param_array(index, int, NULL, S_IRUGO); +module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for " CARD_NAME " soundcard."); -module_param_array(id, charp, NULL, S_IRUGO); +module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for " CARD_NAME " soundcard."); -module_param_array(enable, bool, NULL, S_IRUGO); +module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable " CARD_NAME " soundcard."); MODULE_AUTHOR("Matthias Koenig <mk@phasorlab.de>"); diff --git a/sound/drivers/opl4/opl4_proc.c b/sound/drivers/opl4/opl4_proc.c index cd2c07fa2ef4..16b24091d799 100644 --- a/sound/drivers/opl4/opl4_proc.c +++ b/sound/drivers/opl4/opl4_proc.c @@ -104,7 +104,7 @@ int snd_opl4_create_proc(struct snd_opl4 *opl4) if (entry) { if (opl4->hardware < OPL3_HW_OPL4_ML) { /* OPL4 can access 4 MB external ROM/SRAM */ - entry->mode |= S_IWUSR; + entry->mode |= 0200; entry->size = 4 * 1024 * 1024; } else { /* OPL4-ML has 1 MB internal ROM */ diff --git a/sound/drivers/portman2x4.c b/sound/drivers/portman2x4.c index ec8a94325ef6..3cdf0a88d71b 100644 --- a/sound/drivers/portman2x4.c +++ b/sound/drivers/portman2x4.c @@ -60,11 +60,11 @@ static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; static struct platform_device *platform_devices[SNDRV_CARDS]; static int device_count; -module_param_array(index, int, NULL, S_IRUGO); +module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for " CARD_NAME " soundcard."); -module_param_array(id, charp, NULL, S_IRUGO); +module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for " CARD_NAME " soundcard."); -module_param_array(enable, bool, NULL, S_IRUGO); +module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable " CARD_NAME " soundcard."); MODULE_AUTHOR("Levent Guendogdu, Tobias Gehrig, Matthias Koenig"); diff --git a/sound/firewire/bebob/bebob_proc.c b/sound/firewire/bebob/bebob_proc.c index ec24f96794f5..8096891af913 100644 --- a/sound/firewire/bebob/bebob_proc.c +++ b/sound/firewire/bebob/bebob_proc.c @@ -183,7 +183,7 @@ void snd_bebob_proc_init(struct snd_bebob *bebob) bebob->card->proc_root); if (root == NULL) return; - root->mode = S_IFDIR | S_IRUGO | S_IXUGO; + root->mode = S_IFDIR | 0555; if (snd_info_register(root) < 0) { snd_info_free_entry(root); return; diff --git a/sound/firewire/dice/Makefile b/sound/firewire/dice/Makefile index 55b4be9b0034..37062a233f6a 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 dice-mytek.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 000000000000..b2efb1c71a98 --- /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 000000000000..a63fcbc875ad --- /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 15a484b05298..9cad3d608229 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 8ff6da3c51f7..84eca8a51a02 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-mytek.c b/sound/firewire/dice/dice-mytek.c new file mode 100644 index 000000000000..eb7d5492d10b --- /dev/null +++ b/sound/firewire/dice/dice-mytek.c @@ -0,0 +1,46 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * dice-mytek.c - a part of driver for DICE based devices + * + * Copyright (c) 2018 Melvin Vermeeren + */ + +#include "dice.h" + +struct dice_mytek_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]; +}; + +static const struct dice_mytek_spec stereo_192_dsd_dac = { + /* AES, TOSLINK, SPDIF, ADAT inputs on device */ + .tx_pcm_chs = {{8, 8, 8}, {0, 0, 0} }, + /* PCM 44.1-192, native DSD64/DSD128 to device */ + .rx_pcm_chs = {{4, 4, 4}, {0, 0, 0} } +}; + +/* + * Mytek has a few other firewire-capable devices, though newer models appear + * to lack the port more often than not. As I don't have access to any of them + * they are missing here. An example is the Mytek 8x192 ADDA, which is DICE. + */ + +int snd_dice_detect_mytek_formats(struct snd_dice *dice) +{ + int i; + const struct dice_mytek_spec *dev; + + dev = &stereo_192_dsd_dac; + + memcpy(dice->tx_pcm_chs, dev->tx_pcm_chs, + MAX_STREAMS * SND_DICE_RATE_MODE_COUNT * sizeof(unsigned int)); + memcpy(dice->rx_pcm_chs, dev->rx_pcm_chs, + MAX_STREAMS * SND_DICE_RATE_MODE_COUNT * sizeof(unsigned int)); + + for (i = 0; i < MAX_STREAMS; ++i) { + dice->tx_midi_ports[i] = 0; + dice->rx_midi_ports[i] = 0; + } + + return 0; +} diff --git a/sound/firewire/dice/dice-pcm.c b/sound/firewire/dice/dice-pcm.c index 7cb9e9713ac3..80351b29fe0d 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; - - /* - * 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)); + unsigned int *pcm_channels; + unsigned int i; + + 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 f5c1d1bced59..bb870fc73f99 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 | 0555; + 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 928a255bfc35..c3c892c5c7ff 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 (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; - } + if (i < MAX_STREAMS) + goto restart; - /* 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 000000000000..a8875d24ba2a --- /dev/null +++ b/sound/firewire/dice/dice-tcelectronic.c @@ -0,0 +1,104 @@ +// 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, 8}, {16, 16, 7} }, + .rx_pcm_chs = {{16, 16, 8}, {14, 14, 7} }, + .has_midi = true, +}; + +static const struct dice_tc_spec digital_konnekt_x32 = { + .tx_pcm_chs = {{16, 16, 4}, {0, 0, 0} }, + .rx_pcm_chs = {{16, 16, 4}, {0, 0, 0} }, + .has_midi = false, +}; + +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}, + {0x00000030, &digital_konnekt_x32}, + }; + 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; + } + } + + for (i = 0; i < ARRAY_SIZE(entries); ++i) { + entry = entries + i; + if (entry->model_id == model_id) + break; + } + if (i == ARRAY_SIZE(entries)) + 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)); + + if (entry->spec->has_midi) { + dice->tx_midi_ports[0] = 1; + dice->rx_midi_ports[0] = 1; + } + + return 0; +} diff --git a/sound/firewire/dice/dice-transaction.c b/sound/firewire/dice/dice-transaction.c index 0f0350320ae8..b7e138b5abcf 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 96bb01b6b751..774eb2205668 100644 --- a/sound/firewire/dice/dice.c +++ b/sound/firewire/dice/dice.c @@ -15,40 +15,15 @@ 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 OUI_MYTEK 0x001ee8 #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 +50,6 @@ 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 +156,6 @@ 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 +166,10 @@ 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 +210,17 @@ error: "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 +230,13 @@ 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); @@ -313,16 +294,97 @@ 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_VERSION, - .version = DICE_INTERFACE, + .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 610/2626 has a different value in version field. */ + /* M-Audio Profire 610 has a different value in version field. */ { .match_flags = IEEE1394_MATCH_VENDOR_ID | - IEEE1394_MATCH_SPECIFIER_ID, - .vendor_id = 0x000d6c, - .specifier_id = 0x000d6c, + 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, + }, + /* TC Electronic Digital Konnekt x32. */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_TCELECTRONIC, + .model_id = 0x000030, + .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, + }, + /* Mytek Stereo 192 DSD-DAC. */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_MYTEK, + .model_id = 0x000002, + .driver_data = (kernel_ulong_t)snd_dice_detect_mytek_formats, + }, + { + .match_flags = IEEE1394_MATCH_VERSION, + .version = DICE_INTERFACE, }, { } }; diff --git a/sound/firewire/dice/dice.h b/sound/firewire/dice/dice.h index da00e75e09d4..83353a3559e8 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 +223,9 @@ 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); +int snd_dice_detect_mytek_formats(struct snd_dice *dice); + #endif diff --git a/sound/firewire/digi00x/digi00x-proc.c b/sound/firewire/digi00x/digi00x-proc.c index a1d601f31165..6996d5a6ff5f 100644 --- a/sound/firewire/digi00x/digi00x-proc.c +++ b/sound/firewire/digi00x/digi00x-proc.c @@ -79,7 +79,7 @@ void snd_dg00x_proc_init(struct snd_dg00x *dg00x) if (root == NULL) return; - root->mode = S_IFDIR | S_IRUGO | S_IXUGO; + root->mode = S_IFDIR | 0555; if (snd_info_register(root) < 0) { snd_info_free_entry(root); return; diff --git a/sound/firewire/fireface/ff-proc.c b/sound/firewire/fireface/ff-proc.c index 69441d121f71..40ccbfd8ef89 100644 --- a/sound/firewire/fireface/ff-proc.c +++ b/sound/firewire/fireface/ff-proc.c @@ -52,7 +52,7 @@ void snd_ff_proc_init(struct snd_ff *ff) ff->card->proc_root); if (root == NULL) return; - root->mode = S_IFDIR | S_IRUGO | S_IXUGO; + root->mode = S_IFDIR | 0555; if (snd_info_register(root) < 0) { snd_info_free_entry(root); return; diff --git a/sound/firewire/fireworks/fireworks_proc.c b/sound/firewire/fireworks/fireworks_proc.c index 9c21f31b8b21..779ecec5af62 100644 --- a/sound/firewire/fireworks/fireworks_proc.c +++ b/sound/firewire/fireworks/fireworks_proc.c @@ -219,7 +219,7 @@ void snd_efw_proc_init(struct snd_efw *efw) efw->card->proc_root); if (root == NULL) return; - root->mode = S_IFDIR | S_IRUGO | S_IXUGO; + root->mode = S_IFDIR | 0555; if (snd_info_register(root) < 0) { snd_info_free_entry(root); return; diff --git a/sound/firewire/isight.c b/sound/firewire/isight.c index 46092fa3ff9b..3919e186a30b 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/firewire/motu/motu-proc.c b/sound/firewire/motu/motu-proc.c index 4edc064999ed..ab6830a6d242 100644 --- a/sound/firewire/motu/motu-proc.c +++ b/sound/firewire/motu/motu-proc.c @@ -107,7 +107,7 @@ void snd_motu_proc_init(struct snd_motu *motu) motu->card->proc_root); if (root == NULL) return; - root->mode = S_IFDIR | S_IRUGO | S_IXUGO; + root->mode = S_IFDIR | 0555; if (snd_info_register(root) < 0) { snd_info_free_entry(root); return; diff --git a/sound/firewire/oxfw/oxfw-proc.c b/sound/firewire/oxfw/oxfw-proc.c index 8ba4f9f262b8..27dac071bc73 100644 --- a/sound/firewire/oxfw/oxfw-proc.c +++ b/sound/firewire/oxfw/oxfw-proc.c @@ -103,7 +103,7 @@ void snd_oxfw_proc_init(struct snd_oxfw *oxfw) oxfw->card->proc_root); if (root == NULL) return; - root->mode = S_IFDIR | S_IRUGO | S_IXUGO; + root->mode = S_IFDIR | 0555; if (snd_info_register(root) < 0) { snd_info_free_entry(root); return; diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index 413ab6313bb6..1e5b2c802635 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -49,7 +49,6 @@ static bool detect_loud_models(struct fw_unit *unit) "Tapco LINK.firewire 4x6", "U.420"}; char model[32]; - unsigned int i; int err; err = fw_csr_string(unit->directory, CSR_MODEL, @@ -57,12 +56,7 @@ static bool detect_loud_models(struct fw_unit *unit) 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)); + return match_string(models, ARRAY_SIZE(models), model) >= 0; } static int name_card(struct snd_oxfw *oxfw) diff --git a/sound/firewire/tascam/tascam-proc.c b/sound/firewire/tascam/tascam-proc.c index bfd4a4c06914..fee3bf32a0da 100644 --- a/sound/firewire/tascam/tascam-proc.c +++ b/sound/firewire/tascam/tascam-proc.c @@ -78,7 +78,7 @@ void snd_tscm_proc_init(struct snd_tscm *tscm) tscm->card->proc_root); if (root == NULL) return; - root->mode = S_IFDIR | S_IRUGO | S_IXUGO; + root->mode = S_IFDIR | 0555; if (snd_info_register(root) < 0) { snd_info_free_entry(root); return; diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index 47a358fab132..419e285e0226 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 d09e456107ad..de6ef1b1cf0e 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/msnd/msnd_pinnacle.c b/sound/isa/msnd/msnd_pinnacle.c index 45e561c425bf..6c584d9b6c42 100644 --- a/sound/isa/msnd/msnd_pinnacle.c +++ b/sound/isa/msnd/msnd_pinnacle.c @@ -757,9 +757,9 @@ static int snd_msnd_pinnacle_cfg_reset(int cfg) static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ -module_param_array(index, int, NULL, S_IRUGO); +module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for msnd_pinnacle soundcard."); -module_param_array(id, charp, NULL, S_IRUGO); +module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for msnd_pinnacle soundcard."); static long io[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; @@ -801,22 +801,22 @@ MODULE_LICENSE("GPL"); MODULE_FIRMWARE(INITCODEFILE); MODULE_FIRMWARE(PERMCODEFILE); -module_param_hw_array(io, long, ioport, NULL, S_IRUGO); +module_param_hw_array(io, long, ioport, NULL, 0444); MODULE_PARM_DESC(io, "IO port #"); -module_param_hw_array(irq, int, irq, NULL, S_IRUGO); -module_param_hw_array(mem, long, iomem, NULL, S_IRUGO); -module_param_array(write_ndelay, int, NULL, S_IRUGO); -module_param(calibrate_signal, int, S_IRUGO); +module_param_hw_array(irq, int, irq, NULL, 0444); +module_param_hw_array(mem, long, iomem, NULL, 0444); +module_param_array(write_ndelay, int, NULL, 0444); +module_param(calibrate_signal, int, 0444); #ifndef MSND_CLASSIC -module_param_array(digital, int, NULL, S_IRUGO); -module_param_hw_array(cfg, long, ioport, NULL, S_IRUGO); -module_param_array(reset, int, 0, S_IRUGO); -module_param_hw_array(mpu_io, long, ioport, NULL, S_IRUGO); -module_param_hw_array(mpu_irq, int, irq, NULL, S_IRUGO); -module_param_hw_array(ide_io0, long, ioport, NULL, S_IRUGO); -module_param_hw_array(ide_io1, long, ioport, NULL, S_IRUGO); -module_param_hw_array(ide_irq, int, irq, NULL, S_IRUGO); -module_param_hw_array(joystick_io, long, ioport, NULL, S_IRUGO); +module_param_array(digital, int, NULL, 0444); +module_param_hw_array(cfg, long, ioport, NULL, 0444); +module_param_array(reset, int, 0, 0444); +module_param_hw_array(mpu_io, long, ioport, NULL, 0444); +module_param_hw_array(mpu_irq, int, irq, NULL, 0444); +module_param_hw_array(ide_io0, long, ioport, NULL, 0444); +module_param_hw_array(ide_io1, long, ioport, NULL, 0444); +module_param_hw_array(ide_irq, int, irq, NULL, 0444); +module_param_hw_array(joystick_io, long, ioport, NULL, 0444); #endif diff --git a/sound/isa/sc6000.c b/sound/isa/sc6000.c index c09d9b914efe..a985e9183be9 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/ac97/ac97_proc.c b/sound/pci/ac97/ac97_proc.c index 6320bf084e47..e120a11c69e8 100644 --- a/sound/pci/ac97/ac97_proc.c +++ b/sound/pci/ac97/ac97_proc.c @@ -448,7 +448,7 @@ void snd_ac97_proc_init(struct snd_ac97 * ac97) if ((entry = snd_info_create_card_entry(ac97->bus->card, name, ac97->bus->proc)) != NULL) { snd_info_set_text_ops(entry, ac97, snd_ac97_proc_regs_read); #ifdef CONFIG_SND_DEBUG - entry->mode |= S_IWUSR; + entry->mode |= 0200; entry->c.text.write = snd_ac97_proc_regs_write; #endif if (snd_info_register(entry) < 0) { @@ -474,7 +474,7 @@ void snd_ac97_bus_proc_init(struct snd_ac97_bus * bus) sprintf(name, "codec97#%d", bus->num); if ((entry = snd_info_create_card_entry(bus->card, name, bus->card->proc_root)) != NULL) { - entry->mode = S_IFDIR | S_IRUGO | S_IXUGO; + entry->mode = S_IFDIR | 0555; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); entry = NULL; diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c index 0bf2c04eeada..d9c54c08e2db 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/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index 720361455c60..64e0961f93ba 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -69,27 +69,27 @@ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; static bool enable_hpi_hwdep = 1; -module_param_array(index, int, NULL, S_IRUGO); +module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "ALSA index value for AudioScience soundcard."); -module_param_array(id, charp, NULL, S_IRUGO); +module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ALSA ID string for AudioScience soundcard."); -module_param_array(enable, bool, NULL, S_IRUGO); +module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "ALSA enable AudioScience soundcard."); -module_param(enable_hpi_hwdep, bool, S_IRUGO|S_IWUSR); +module_param(enable_hpi_hwdep, bool, 0644); MODULE_PARM_DESC(enable_hpi_hwdep, "ALSA enable HPI hwdep for AudioScience soundcard "); /* identify driver */ #ifdef KERNEL_ALSA_BUILD static char *build_info = "Built using headers from kernel source"; -module_param(build_info, charp, S_IRUGO); +module_param(build_info, charp, 0444); MODULE_PARM_DESC(build_info, "Built using headers from kernel source"); #else static char *build_info = "Built within ALSA source"; -module_param(build_info, charp, S_IRUGO); +module_param(build_info, charp, 0444); MODULE_PARM_DESC(build_info, "Built within ALSA source"); #endif diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c index b1a2a7ea4172..7d049569012c 100644 --- a/sound/pci/asihpi/hpioctl.c +++ b/sound/pci/asihpi/hpioctl.c @@ -46,14 +46,14 @@ MODULE_FIRMWARE("asihpi/dsp8900.bin"); #endif static int prealloc_stream_buf; -module_param(prealloc_stream_buf, int, S_IRUGO); +module_param(prealloc_stream_buf, int, 0444); MODULE_PARM_DESC(prealloc_stream_buf, "Preallocate size for per-adapter stream buffer"); /* Allow the debug level to be changed after module load. E.g. echo 2 > /sys/module/asihpi/parameters/hpiDebugLevel */ -module_param(hpi_debug_level, int, S_IRUGO | S_IWUSR); +module_param(hpi_debug_level, int, 0644); MODULE_PARM_DESC(hpi_debug_level, "debug verbosity 0..5"); /* List of adapters found */ diff --git a/sound/pci/ca0106/ca0106_proc.c b/sound/pci/ca0106/ca0106_proc.c index 9b2b8b38122f..a2c85cc37972 100644 --- a/sound/pci/ca0106/ca0106_proc.c +++ b/sound/pci/ca0106/ca0106_proc.c @@ -431,7 +431,7 @@ int snd_ca0106_proc_init(struct snd_ca0106 *emu) if(! snd_card_proc_new(emu->card, "ca0106_reg32", &entry)) { snd_info_set_text_ops(entry, emu, snd_ca0106_proc_reg_read32); entry->c.text.write = snd_ca0106_proc_reg_write32; - entry->mode |= S_IWUSR; + entry->mode |= 0200; } if(! snd_card_proc_new(emu->card, "ca0106_reg16", &entry)) snd_info_set_text_ops(entry, emu, snd_ca0106_proc_reg_read16); @@ -440,12 +440,12 @@ int snd_ca0106_proc_init(struct snd_ca0106 *emu) if(! snd_card_proc_new(emu->card, "ca0106_regs1", &entry)) { snd_info_set_text_ops(entry, emu, snd_ca0106_proc_reg_read1); entry->c.text.write = snd_ca0106_proc_reg_write; - entry->mode |= S_IWUSR; + entry->mode |= 0200; } if(! snd_card_proc_new(emu->card, "ca0106_i2c", &entry)) { entry->c.text.write = snd_ca0106_proc_i2c_write; entry->private_data = emu; - entry->mode |= S_IWUSR; + entry->mode |= 0200; } if(! snd_card_proc_new(emu->card, "ca0106_regs2", &entry)) snd_info_set_text_ops(entry, emu, snd_ca0106_proc_reg_read2); diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index 26a657870664..452cc79b44af 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 655fbea1692c..4910d3f46d4b 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/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index 0020fd0efc46..ed1251c5f449 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -2849,7 +2849,7 @@ static int snd_cs46xx_proc_init(struct snd_card *card, struct snd_cs46xx *chip) entry->private_data = chip; entry->c.ops = &snd_cs46xx_proc_io_ops; entry->size = region->size; - entry->mode = S_IFREG | S_IRUSR; + entry->mode = S_IFREG | 0400; } } #ifdef CONFIG_SND_CS46XX_NEW_DSP diff --git a/sound/pci/cs46xx/dsp_spos.c b/sound/pci/cs46xx/dsp_spos.c index aa61615288ff..c44eadef64ae 100644 --- a/sound/pci/cs46xx/dsp_spos.c +++ b/sound/pci/cs46xx/dsp_spos.c @@ -798,7 +798,7 @@ int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip) if ((entry = snd_info_create_card_entry(card, "dsp", card->proc_root)) != NULL) { entry->content = SNDRV_INFO_CONTENT_TEXT; - entry->mode = S_IFDIR | S_IRUGO | S_IXUGO; + entry->mode = S_IFDIR | 0555; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); @@ -814,7 +814,7 @@ int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip) if ((entry = snd_info_create_card_entry(card, "spos_symbols", ins->proc_dsp_dir)) != NULL) { entry->content = SNDRV_INFO_CONTENT_TEXT; entry->private_data = chip; - entry->mode = S_IFREG | S_IRUGO | S_IWUSR; + entry->mode = S_IFREG | 0644; entry->c.text.read = cs46xx_dsp_proc_symbol_table_read; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); @@ -826,7 +826,7 @@ int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip) if ((entry = snd_info_create_card_entry(card, "spos_modules", ins->proc_dsp_dir)) != NULL) { entry->content = SNDRV_INFO_CONTENT_TEXT; entry->private_data = chip; - entry->mode = S_IFREG | S_IRUGO | S_IWUSR; + entry->mode = S_IFREG | 0644; entry->c.text.read = cs46xx_dsp_proc_modules_read; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); @@ -838,7 +838,7 @@ int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip) if ((entry = snd_info_create_card_entry(card, "parameter", ins->proc_dsp_dir)) != NULL) { entry->content = SNDRV_INFO_CONTENT_TEXT; entry->private_data = chip; - entry->mode = S_IFREG | S_IRUGO | S_IWUSR; + entry->mode = S_IFREG | 0644; entry->c.text.read = cs46xx_dsp_proc_parameter_dump_read; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); @@ -850,7 +850,7 @@ int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip) if ((entry = snd_info_create_card_entry(card, "sample", ins->proc_dsp_dir)) != NULL) { entry->content = SNDRV_INFO_CONTENT_TEXT; entry->private_data = chip; - entry->mode = S_IFREG | S_IRUGO | S_IWUSR; + entry->mode = S_IFREG | 0644; entry->c.text.read = cs46xx_dsp_proc_sample_dump_read; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); @@ -862,7 +862,7 @@ int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip) if ((entry = snd_info_create_card_entry(card, "task_tree", ins->proc_dsp_dir)) != NULL) { entry->content = SNDRV_INFO_CONTENT_TEXT; entry->private_data = chip; - entry->mode = S_IFREG | S_IRUGO | S_IWUSR; + entry->mode = S_IFREG | 0644; entry->c.text.read = cs46xx_dsp_proc_task_tree_read; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); @@ -874,7 +874,7 @@ int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip) if ((entry = snd_info_create_card_entry(card, "scb_info", ins->proc_dsp_dir)) != NULL) { entry->content = SNDRV_INFO_CONTENT_TEXT; entry->private_data = chip; - entry->mode = S_IFREG | S_IRUGO | S_IWUSR; + entry->mode = S_IFREG | 0644; entry->c.text.read = cs46xx_dsp_proc_scb_read; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); diff --git a/sound/pci/cs46xx/dsp_spos_scb_lib.c b/sound/pci/cs46xx/dsp_spos_scb_lib.c index 7488e1b7a770..abb01ce66983 100644 --- a/sound/pci/cs46xx/dsp_spos_scb_lib.c +++ b/sound/pci/cs46xx/dsp_spos_scb_lib.c @@ -271,7 +271,7 @@ void cs46xx_dsp_proc_register_scb_desc (struct snd_cs46xx *chip, entry->content = SNDRV_INFO_CONTENT_TEXT; entry->private_data = scb_info; - entry->mode = S_IFREG | S_IRUGO | S_IWUSR; + entry->mode = S_IFREG | 0644; entry->c.text.read = cs46xx_dsp_proc_scb_info_read; diff --git a/sound/pci/ctxfi/cttimer.c b/sound/pci/ctxfi/cttimer.c index 08e874e9a7f6..2099e9ce441a 100644 --- a/sound/pci/ctxfi/cttimer.c +++ b/sound/pci/ctxfi/cttimer.c @@ -17,7 +17,7 @@ static bool use_system_timer; MODULE_PARM_DESC(use_system_timer, "Force to use system-timer"); -module_param(use_system_timer, bool, S_IRUGO); +module_param(use_system_timer, bool, 0444); struct ct_timer_ops { void (*init)(struct ct_timer_instance *); diff --git a/sound/pci/ctxfi/xfi.c b/sound/pci/ctxfi/xfi.c index f2f32779de98..b2874220be1b 100644 --- a/sound/pci/ctxfi/xfi.c +++ b/sound/pci/ctxfi/xfi.c @@ -26,9 +26,9 @@ MODULE_SUPPORTED_DEVICE("{{Creative Labs, Sound Blaster X-Fi}"); static unsigned int reference_rate = 48000; static unsigned int multiple = 2; MODULE_PARM_DESC(reference_rate, "Reference rate (default=48000)"); -module_param(reference_rate, uint, S_IRUGO); +module_param(reference_rate, uint, 0444); MODULE_PARM_DESC(multiple, "Rate multiplier (default=2)"); -module_param(multiple, uint, S_IRUGO); +module_param(multiple, uint, 0444); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 0935a5c8741f..358ef7dcf410 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -59,7 +59,7 @@ static int get_firmware(const struct firmware **fw_entry, dev_dbg(chip->card->dev, "firmware requested: %s\n", card_fw[fw_index].data); snprintf(name, sizeof(name), "ea/%s", card_fw[fw_index].data); - err = request_firmware(fw_entry, name, pci_device(chip)); + err = request_firmware(fw_entry, name, &chip->pci->dev); if (err < 0) dev_err(chip->card->dev, "get_firmware(): Firmware not available (%d)\n", err); diff --git a/sound/pci/echoaudio/echoaudio.h b/sound/pci/echoaudio/echoaudio.h index 152ce158583c..44b390a667d5 100644 --- a/sound/pci/echoaudio/echoaudio.h +++ b/sound/pci/echoaudio/echoaudio.h @@ -559,10 +559,4 @@ static inline int monitor_index(const struct echoaudio *chip, int out, int in) return out * num_busses_in(chip) + in; } - -#ifndef pci_device -#define pci_device(chip) (&chip->pci->dev) -#endif - - #endif /* _ECHOAUDIO_H_ */ diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index 2c2b12a06177..611589cbdad6 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -1070,7 +1070,7 @@ static int snd_emu10k1x_proc_init(struct emu10k1x *emu) if(! snd_card_proc_new(emu->card, "emu10k1x_regs", &entry)) { snd_info_set_text_ops(entry, emu, snd_emu10k1x_proc_reg_read); entry->c.text.write = snd_emu10k1x_proc_reg_write; - entry->mode |= S_IWUSR; + entry->mode |= 0200; entry->private_data = emu; } diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index a2b56b188be4..b45a01bb73e5 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -170,7 +170,7 @@ static char *audigy_outs[32] = { /* 0x0f */ "Rear Right", /* 0x10 */ "AC97 Front Left", /* 0x11 */ "AC97 Front Right", - /* 0x12 */ "ADC Caputre Left", + /* 0x12 */ "ADC Capture Left", /* 0x13 */ "ADC Capture Right", /* 0x14 */ NULL, /* 0x15 */ NULL, @@ -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 cefe613ef7b7..d39458ab251f 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/emu10k1/emuproc.c b/sound/pci/emu10k1/emuproc.c index bde0d1954f56..b57008031792 100644 --- a/sound/pci/emu10k1/emuproc.c +++ b/sound/pci/emu10k1/emuproc.c @@ -135,7 +135,7 @@ static void snd_emu10k1_proc_read(struct snd_info_entry *entry, /* 15 */ "Rear Right", /* 16 */ "AC97 Front Left", /* 17 */ "AC97 Front Right", - /* 18 */ "ADC Caputre Left", + /* 18 */ "ADC Capture Left", /* 19 */ "ADC Capture Right", /* 20 */ "???", /* 21 */ "???", @@ -574,32 +574,32 @@ int snd_emu10k1_proc_init(struct snd_emu10k1 *emu) if (! snd_card_proc_new(emu->card, "io_regs", &entry)) { snd_info_set_text_ops(entry, emu, snd_emu_proc_io_reg_read); entry->c.text.write = snd_emu_proc_io_reg_write; - entry->mode |= S_IWUSR; + entry->mode |= 0200; } if (! snd_card_proc_new(emu->card, "ptr_regs00a", &entry)) { snd_info_set_text_ops(entry, emu, snd_emu_proc_ptr_reg_read00a); entry->c.text.write = snd_emu_proc_ptr_reg_write00; - entry->mode |= S_IWUSR; + entry->mode |= 0200; } if (! snd_card_proc_new(emu->card, "ptr_regs00b", &entry)) { snd_info_set_text_ops(entry, emu, snd_emu_proc_ptr_reg_read00b); entry->c.text.write = snd_emu_proc_ptr_reg_write00; - entry->mode |= S_IWUSR; + entry->mode |= 0200; } if (! snd_card_proc_new(emu->card, "ptr_regs20a", &entry)) { snd_info_set_text_ops(entry, emu, snd_emu_proc_ptr_reg_read20a); entry->c.text.write = snd_emu_proc_ptr_reg_write20; - entry->mode |= S_IWUSR; + entry->mode |= 0200; } if (! snd_card_proc_new(emu->card, "ptr_regs20b", &entry)) { snd_info_set_text_ops(entry, emu, snd_emu_proc_ptr_reg_read20b); entry->c.text.write = snd_emu_proc_ptr_reg_write20; - entry->mode |= S_IWUSR; + entry->mode |= 0200; } if (! snd_card_proc_new(emu->card, "ptr_regs20c", &entry)) { snd_info_set_text_ops(entry, emu, snd_emu_proc_ptr_reg_read20c); entry->c.text.write = snd_emu_proc_ptr_reg_write20; - entry->mode |= S_IWUSR; + entry->mode |= 0200; } #endif @@ -621,35 +621,35 @@ int snd_emu10k1_proc_init(struct snd_emu10k1 *emu) if (! snd_card_proc_new(emu->card, "fx8010_gpr", &entry)) { entry->content = SNDRV_INFO_CONTENT_DATA; entry->private_data = emu; - entry->mode = S_IFREG | S_IRUGO /*| S_IWUSR*/; + entry->mode = S_IFREG | 0444 /*| S_IWUSR*/; entry->size = emu->audigy ? A_TOTAL_SIZE_GPR : TOTAL_SIZE_GPR; entry->c.ops = &snd_emu10k1_proc_ops_fx8010; } if (! snd_card_proc_new(emu->card, "fx8010_tram_data", &entry)) { entry->content = SNDRV_INFO_CONTENT_DATA; entry->private_data = emu; - entry->mode = S_IFREG | S_IRUGO /*| S_IWUSR*/; + entry->mode = S_IFREG | 0444 /*| S_IWUSR*/; entry->size = emu->audigy ? A_TOTAL_SIZE_TANKMEM_DATA : TOTAL_SIZE_TANKMEM_DATA ; entry->c.ops = &snd_emu10k1_proc_ops_fx8010; } if (! snd_card_proc_new(emu->card, "fx8010_tram_addr", &entry)) { entry->content = SNDRV_INFO_CONTENT_DATA; entry->private_data = emu; - entry->mode = S_IFREG | S_IRUGO /*| S_IWUSR*/; + entry->mode = S_IFREG | 0444 /*| S_IWUSR*/; entry->size = emu->audigy ? A_TOTAL_SIZE_TANKMEM_ADDR : TOTAL_SIZE_TANKMEM_ADDR ; entry->c.ops = &snd_emu10k1_proc_ops_fx8010; } if (! snd_card_proc_new(emu->card, "fx8010_code", &entry)) { entry->content = SNDRV_INFO_CONTENT_DATA; entry->private_data = emu; - entry->mode = S_IFREG | S_IRUGO /*| S_IWUSR*/; + entry->mode = S_IFREG | 0444 /*| S_IWUSR*/; entry->size = emu->audigy ? A_TOTAL_SIZE_CODE : TOTAL_SIZE_CODE; entry->c.ops = &snd_emu10k1_proc_ops_fx8010; } if (! snd_card_proc_new(emu->card, "fx8010_acode", &entry)) { entry->content = SNDRV_INFO_CONTENT_TEXT; entry->private_data = emu; - entry->mode = S_IFREG | S_IRUGO /*| S_IWUSR*/; + entry->mode = S_IFREG | 0444 /*| S_IWUSR*/; entry->c.text.read = snd_emu10k1_proc_acode_read; } return 0; diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c index 5865f3b90b34..dbc7d8d0e1c4 100644 --- a/sound/pci/emu10k1/memory.c +++ b/sound/pci/emu10k1/memory.c @@ -248,13 +248,13 @@ __found_pages: static int is_valid_page(struct snd_emu10k1 *emu, dma_addr_t addr) { if (addr & ~emu->dma_mask) { - dev_err(emu->card->dev, + dev_err_ratelimited(emu->card->dev, "max memory size is 0x%lx (addr = 0x%lx)!!\n", emu->dma_mask, (unsigned long)addr); return 0; } if (addr & (EMUPAGESIZE-1)) { - dev_err(emu->card->dev, "page is not aligned\n"); + dev_err_ratelimited(emu->card->dev, "page is not aligned\n"); return 0; } return 1; @@ -345,7 +345,7 @@ snd_emu10k1_alloc_pages(struct snd_emu10k1 *emu, struct snd_pcm_substream *subst else addr = snd_pcm_sgbuf_get_addr(substream, ofs); if (! is_valid_page(emu, addr)) { - dev_err(emu->card->dev, + dev_err_ratelimited(emu->card->dev, "emu: failure page = %d\n", idx); mutex_unlock(&hdr->block_mutex); return NULL; diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index f7a492c382d9..4235907b7858 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -127,11 +127,15 @@ comment "Set to Y if you want auto-loading the codec driver" 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 d3ea73171a3d..b9a6b66aeb0e 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 5bc3a7468e17..08151f3c0b13 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_controller.c b/sound/pci/hda/hda_controller.c index d1eb14842340..a12e594d4e3b 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -748,8 +748,10 @@ int snd_hda_attach_pcm_stream(struct hda_bus *_bus, struct hda_codec *codec, return err; strlcpy(pcm->name, cpcm->name, sizeof(pcm->name)); apcm = kzalloc(sizeof(*apcm), GFP_KERNEL); - if (apcm == NULL) + if (apcm == NULL) { + snd_device_free(chip->card, pcm); return -ENOMEM; + } apcm->chip = chip; apcm->pcm = pcm; apcm->codec = codec; diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 5cc65093d941..db773e219aaa 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_intel.c b/sound/pci/hda/hda_intel.c index a0c93b9c9a28..1ae1850b3bfd 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2209,7 +2209,18 @@ static struct snd_pci_quirk power_save_blacklist[] = { /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ SND_PCI_QUIRK(0x1849, 0x0c0c, "Asrock B85M-ITX", 0), /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ + SND_PCI_QUIRK(0x1849, 0x7662, "Asrock H81M-HDS", 0), + /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ SND_PCI_QUIRK(0x1043, 0x8733, "Asus Prime X370-Pro", 0), + /* https://bugzilla.redhat.com/show_bug.cgi?id=1581607 */ + SND_PCI_QUIRK(0x1558, 0x3501, "Clevo W35xSS_370SS", 0), + /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ + /* Note the P55A-UD3 and Z87-D3HP share the subsys id for the HDA dev */ + SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte P55A-UD3 / Z87-D3HP", 0), + /* https://bugzilla.kernel.org/show_bug.cgi?id=199607 */ + SND_PCI_QUIRK(0x8086, 0x2057, "Intel NUC5i7RYB", 0), + /* https://bugzilla.redhat.com/show_bug.cgi?id=1520902 */ + SND_PCI_QUIRK(0x8086, 0x2068, "Intel NUC7i3BNB", 0), /* https://bugzilla.redhat.com/show_bug.cgi?id=1572975 */ SND_PCI_QUIRK(0x17aa, 0x36a7, "Lenovo C50 All in one", 0), /* https://bugzilla.kernel.org/show_bug.cgi?id=198611 */ diff --git a/sound/pci/hda/hda_sysfs.c b/sound/pci/hda/hda_sysfs.c index 9b7efece4484..6ec79c58d48d 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/hp_x360_helper.c b/sound/pci/hda/hp_x360_helper.c new file mode 100644 index 000000000000..969542c57358 --- /dev/null +++ b/sound/pci/hda/hp_x360_helper.c @@ -0,0 +1,95 @@ +// SPDX-License-Identifier: GPL-2.0 +/* Fixes for HP X360 laptops with top B&O speakers + * to be included from codec driver + */ + +static void alc295_fixup_hp_top_speakers(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + static const struct hda_pintbl pincfgs[] = { + { 0x17, 0x90170110 }, + { } + }; + static const struct coef_fw alc295_hp_speakers_coefs[] = { + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0000), WRITE_COEF(0x28, 0x0000), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x003f), WRITE_COEF(0x28, 0x1000), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0004), WRITE_COEF(0x28, 0x0600), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x006a), WRITE_COEF(0x28, 0x0006), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x006c), WRITE_COEF(0x28, 0xc0c0), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0008), WRITE_COEF(0x28, 0xb000), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x002e), WRITE_COEF(0x28, 0x0800), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x006a), WRITE_COEF(0x28, 0x00c1), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x006c), WRITE_COEF(0x28, 0x0320), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0039), WRITE_COEF(0x28, 0x0000), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x003b), WRITE_COEF(0x28, 0xffff), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x003c), WRITE_COEF(0x28, 0xffd0), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x003a), WRITE_COEF(0x28, 0x1dfe), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0080), WRITE_COEF(0x28, 0x0880), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x003a), WRITE_COEF(0x28, 0x0dfe), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0018), WRITE_COEF(0x28, 0x0219), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x006a), WRITE_COEF(0x28, 0x005d), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x006c), WRITE_COEF(0x28, 0x9142), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00c0), WRITE_COEF(0x28, 0x01ce), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00c1), WRITE_COEF(0x28, 0xed0c), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00c2), WRITE_COEF(0x28, 0x1c00), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00c3), WRITE_COEF(0x28, 0x0000), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00c4), WRITE_COEF(0x28, 0x0200), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00c5), WRITE_COEF(0x28, 0x0000), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00c6), WRITE_COEF(0x28, 0x0399), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00c7), WRITE_COEF(0x28, 0x2330), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00c8), WRITE_COEF(0x28, 0x1e5d), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00c9), WRITE_COEF(0x28, 0x6eff), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00ca), WRITE_COEF(0x28, 0x01c0), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00cb), WRITE_COEF(0x28, 0xed0c), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00cc), WRITE_COEF(0x28, 0x1c00), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00cd), WRITE_COEF(0x28, 0x0000), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00ce), WRITE_COEF(0x28, 0x0200), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00cf), WRITE_COEF(0x28, 0x0000), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00d0), WRITE_COEF(0x28, 0x0399), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00d1), WRITE_COEF(0x28, 0x2330), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00d2), WRITE_COEF(0x28, 0x1e5d), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00d3), WRITE_COEF(0x28, 0x6eff), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0062), WRITE_COEF(0x28, 0x8000), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0063), WRITE_COEF(0x28, 0x5f5f), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0064), WRITE_COEF(0x28, 0x1000), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0065), WRITE_COEF(0x28, 0x0000), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0066), WRITE_COEF(0x28, 0x4004), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0067), WRITE_COEF(0x28, 0x0802), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0068), WRITE_COEF(0x28, 0x890f), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0069), WRITE_COEF(0x28, 0xe021), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0070), WRITE_COEF(0x28, 0x8012), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0071), WRITE_COEF(0x28, 0x3450), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0072), WRITE_COEF(0x28, 0x0123), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0073), WRITE_COEF(0x28, 0x4543), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0074), WRITE_COEF(0x28, 0x2100), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0075), WRITE_COEF(0x28, 0x4321), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0076), WRITE_COEF(0x28, 0x0000), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0050), WRITE_COEF(0x28, 0x8200), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x003a), WRITE_COEF(0x28, 0x1dfe), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0051), WRITE_COEF(0x28, 0x0707), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0052), WRITE_COEF(0x28, 0x4090), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x006a), WRITE_COEF(0x28, 0x0090), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x006c), WRITE_COEF(0x28, 0x721f), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0012), WRITE_COEF(0x28, 0xebeb), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x009e), WRITE_COEF(0x28, 0x0000), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0060), WRITE_COEF(0x28, 0x2213), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x006a), WRITE_COEF(0x28, 0x0006), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x006c), WRITE_COEF(0x28, 0x0000), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x003f), WRITE_COEF(0x28, 0x3000), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0004), WRITE_COEF(0x28, 0x0500), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0040), WRITE_COEF(0x28, 0x800c), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0046), WRITE_COEF(0x28, 0xc22e), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x004b), WRITE_COEF(0x28, 0x0000), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0050), WRITE_COEF(0x28, 0x82ec), WRITE_COEF(0x29, 0xb024), + }; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_apply_pincfgs(codec, pincfgs); + alc295_fixup_disable_dac3(codec, fix, action); + break; + case HDA_FIXUP_ACT_INIT: + alc_process_coef_fw(codec, alc295_hp_speakers_coefs); + break; + } +} diff --git a/sound/pci/hda/local.h b/sound/pci/hda/local.h deleted file mode 100644 index 3b8b7d78f9e0..000000000000 --- a/sound/pci/hda/local.h +++ /dev/null @@ -1,40 +0,0 @@ -/* SPDX-License-Identifier: GPL-2.0 */ -/* - */ - -#ifndef __HDAC_LOCAL_H -#define __HDAC_LOCAL_H - -int hdac_read_parm(struct hdac_device *codec, hda_nid_t nid, int parm); - -#define get_wcaps(codec, nid) \ - hdac_read_parm(codec, nid, AC_PAR_AUDIO_WIDGET_CAP) -/* get the widget type from widget capability bits */ -static inline int get_wcaps_type(unsigned int wcaps) -{ - if (!wcaps) - return -1; /* invalid type */ - return (wcaps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; -} - -#define get_pin_caps(codec, nid) \ - hdac_read_parm(codec, nid, AC_PAR_PIN_CAP) - -static inline -unsigned int get_pin_cfg(struct hdac_device *codec, hda_nid_t nid) -{ - unsigned int val; - - if (snd_hdac_read(codec, nid, AC_VERB_GET_CONFIG_DEFAULT, 0, &val)) - return -1; - return val; -} - -#define get_amp_caps(codec, nid, dir) \ - hdac_read_parm(codec, nid, (dir) == HDA_OUTPUT ? \ - AC_PAR_AMP_OUT_CAP : AC_PAR_AMP_IN_CAP) - -#define get_power_caps(codec, nid) \ - hdac_read_parm(codec, nid, AC_PAR_POWER_STATE) - -#endif /* __HDAC_LOCAL_H */ diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 768ea8651993..292e2c592c17 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 *const 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 *const 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 @@ -150,7 +189,7 @@ struct ct_effect { #define EFX_DIR_OUT 0 #define EFX_DIR_IN 1 -static struct ct_effect ca0132_effects[EFFECTS_COUNT] = { +static const struct ct_effect ca0132_effects[EFFECTS_COUNT] = { { .name = "Surround", .nid = SURROUND, .mid = 0x96, @@ -277,7 +316,7 @@ struct ct_tuning_ctl { unsigned int def_val;/*effect default values*/ }; -static struct ct_tuning_ctl ca0132_tuning_ctls[] = { +static const struct ct_tuning_ctl ca0132_tuning_ctls[] = { { .name = "Wedge Angle", .parent_nid = VOICE_FOCUS, .nid = WEDGE_ANGLE, @@ -392,14 +431,14 @@ struct ct_voicefx_preset { unsigned int vals[VOICEFX_MAX_PARAM_COUNT]; }; -static struct ct_voicefx ca0132_voicefx = { +static const struct ct_voicefx ca0132_voicefx = { .name = "VoiceFX Capture Switch", .nid = VOICEFX, .mid = 0x95, .reqs = {10, 11, 12, 13, 14, 15, 16, 17, 18} }; -static struct ct_voicefx_preset ca0132_voicefx_presets[] = { +static const struct ct_voicefx_preset ca0132_voicefx_presets[] = { { .name = "Neutral", .vals = { 0x00000000, 0x43C80000, 0x44AF0000, 0x44FA0000, 0x3F800000, 0x3F800000, @@ -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 const 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 const 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 const 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, @@ -698,11 +892,12 @@ enum dsp_download_state { */ struct ca0132_spec { - struct snd_kcontrol_new *mixers[5]; + const struct snd_kcontrol_new *mixers[5]; unsigned int num_mixers; 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 @@ exit: } /* + * 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 @@ exit: */ 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 @@ exit: 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 *const 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 @@ exit: /* * 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 +5502,59 @@ 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 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, "FX: %s %s Volume", 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 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, "FX: %s %s Switch", pfx, dirstr[dir]); + else + sprintf(namestr, "%s %s Switch", pfx, dirstr[dir]); + return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec)); } @@ -3929,11 +5569,141 @@ 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. */ -static struct snd_kcontrol_new ca0132_mixer[] = { +static const struct snd_kcontrol_new ca0132_mixer[] = { CA0132_CODEC_VOL("Master Playback Volume", VNID_SPK, HDA_OUTPUT), CA0132_CODEC_MUTE("Master Playback Switch", VNID_SPK, HDA_OUTPUT), CA0132_CODEC_VOL("Capture Volume", VNID_MIC, HDA_INPUT), @@ -3955,10 +5725,55 @@ 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 const 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 const 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 +5782,94 @@ 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 +5894,10 @@ 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 +5952,11 @@ 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 +5965,16 @@ 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 +6181,196 @@ 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 +6415,159 @@ 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 +6575,16 @@ 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 +6606,49 @@ 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 +6676,17 @@ 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 +6732,10 @@ 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 +6758,8 @@ 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 +6789,27 @@ 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 +6823,11 @@ 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 +6845,15 @@ 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 +6864,120 @@ 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 +6986,264 @@ 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 +7255,45 @@ 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 +7305,39 @@ 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 +7351,14 @@ 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 +7378,71 @@ 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 +7469,7 @@ 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 +7480,8 @@ 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 +7545,46 @@ 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 5b4dbcec6de8..dbf9910c5269 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; @@ -965,6 +964,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x103c, 0x822e, "HP ProBook 440 G4", CXT_FIXUP_MUTE_LED_GPIO), SND_PCI_QUIRK(0x103c, 0x8299, "HP 800 G3 SFF", CXT_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x829a, "HP 800 G3 DM", CXT_FIXUP_HP_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x103c, 0x8455, "HP Z2 G4", CXT_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN), SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT_FIXUP_OLPC_XO), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410), @@ -998,6 +998,7 @@ static const struct hda_model_fixup cxt5066_fixup_models[] = { { .id = CXT_FIXUP_MUTE_LED_EAPD, .name = "mute-led-eapd" }, { .id = CXT_FIXUP_HP_DOCK, .name = "hp-dock" }, { .id = CXT_FIXUP_MUTE_LED_GPIO, .name = "mute-led-gpio" }, + { .id = CXT_FIXUP_HP_MIC_NO_PRESENCE, .name = "hp-mic-fix" }, {} }; diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 7d7eb1354eee..8840daf9c6a3 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -510,7 +510,7 @@ static int eld_proc_new(struct hdmi_spec_per_pin *per_pin, int index) snd_info_set_text_ops(entry, per_pin, print_eld_info); entry->c.text.write = write_eld_info; - entry->mode |= S_IWUSR; + entry->mode |= 0200; per_pin->proc_entry = entry; return 0; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 01a6643fc7d4..d64dcb9a4c99 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, @@ -3653,30 +3653,37 @@ static void alc269_fixup_hp_mute_led(struct hda_codec *codec, } } -static void alc269_fixup_hp_mute_led_mic1(struct hda_codec *codec, - const struct hda_fixup *fix, int action) +static void alc269_fixup_hp_mute_led_micx(struct hda_codec *codec, + const struct hda_fixup *fix, + int action, hda_nid_t pin) { struct alc_spec *spec = codec->spec; + if (action == HDA_FIXUP_ACT_PRE_PROBE) { spec->mute_led_polarity = 0; - spec->mute_led_nid = 0x18; + spec->mute_led_nid = pin; spec->gen.vmaster_mute.hook = alc269_fixup_mic_mute_hook; spec->gen.vmaster_mute_enum = 1; codec->power_filter = led_power_filter; } } +static void alc269_fixup_hp_mute_led_mic1(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + alc269_fixup_hp_mute_led_micx(codec, fix, action, 0x18); +} + static void alc269_fixup_hp_mute_led_mic2(struct hda_codec *codec, const struct hda_fixup *fix, int action) { - struct alc_spec *spec = codec->spec; - if (action == HDA_FIXUP_ACT_PRE_PROBE) { - spec->mute_led_polarity = 0; - spec->mute_led_nid = 0x19; - spec->gen.vmaster_mute.hook = alc269_fixup_mic_mute_hook; - spec->gen.vmaster_mute_enum = 1; - codec->power_filter = led_power_filter; - } + alc269_fixup_hp_mute_led_micx(codec, fix, action, 0x19); +} + +static void alc269_fixup_hp_mute_led_mic3(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + alc269_fixup_hp_mute_led_micx(codec, fix, action, 0x1b); } /* update LED status via GPIO */ @@ -5387,6 +5394,9 @@ static void alc274_fixup_bind_dacs(struct hda_codec *codec, /* for dell wmi mic mute led */ #include "dell_wmi_helper.c" +/* for alc295_fixup_hp_top_speakers */ +#include "hp_x360_helper.c" + enum { ALC269_FIXUP_SONY_VAIO, ALC275_FIXUP_SONY_VAIO_GPIO2, @@ -5413,6 +5423,7 @@ enum { ALC269_FIXUP_HP_MUTE_LED, ALC269_FIXUP_HP_MUTE_LED_MIC1, ALC269_FIXUP_HP_MUTE_LED_MIC2, + ALC269_FIXUP_HP_MUTE_LED_MIC3, ALC269_FIXUP_HP_GPIO_LED, ALC269_FIXUP_HP_GPIO_MIC1_LED, ALC269_FIXUP_HP_LINE1_MIC1_LED, @@ -5506,6 +5517,7 @@ enum { ALC298_FIXUP_TPT470_DOCK, ALC255_FIXUP_DUMMY_LINEOUT_VERB, ALC255_FIXUP_DELL_HEADSET_MIC, + ALC295_FIXUP_HP_X360, }; static const struct hda_fixup alc269_fixups[] = { @@ -5672,6 +5684,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_hp_mute_led_mic2, }, + [ALC269_FIXUP_HP_MUTE_LED_MIC3] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_hp_mute_led_mic3, + }, [ALC269_FIXUP_HP_GPIO_LED] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_hp_gpio_led, @@ -6375,6 +6391,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MIC }, + [ALC295_FIXUP_HP_X360] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc295_fixup_hp_top_speakers, + .chained = true, + .chain_id = ALC269_FIXUP_HP_MUTE_LED_MIC3 + } }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -6494,6 +6516,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2337, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x221c, "HP EliteBook 755 G2", ALC280_FIXUP_HP_HEADSET_MIC), SND_PCI_QUIRK(0x103c, 0x8256, "HP", ALC221_FIXUP_HP_FRONT_MIC), + SND_PCI_QUIRK(0x103c, 0x827e, "HP x360", ALC295_FIXUP_HP_X360), SND_PCI_QUIRK(0x103c, 0x82bf, "HP", ALC221_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x82c0, "HP", ALC221_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), @@ -6580,7 +6603,6 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x312f, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), SND_PCI_QUIRK(0x17aa, 0x3138, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), SND_PCI_QUIRK(0x17aa, 0x313c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), - SND_PCI_QUIRK(0x17aa, 0x3112, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP), @@ -6752,6 +6774,11 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x1b, 0x01111010}, {0x1e, 0x01451130}, {0x21, 0x02211020}), + SND_HDA_PIN_QUIRK(0x10ec0235, 0x17aa, "Lenovo", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY, + {0x12, 0x90a60140}, + {0x14, 0x90170110}, + {0x19, 0x02a11030}, + {0x21, 0x02211020}), SND_HDA_PIN_QUIRK(0x10ec0236, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, {0x12, 0x90a60140}, {0x14, 0x90170110}, diff --git a/sound/pci/ice1712/pontis.c b/sound/pci/ice1712/pontis.c index 5101f40f6fbd..93b8cfc6636f 100644 --- a/sound/pci/ice1712/pontis.c +++ b/sound/pci/ice1712/pontis.c @@ -662,7 +662,7 @@ static void wm_proc_init(struct snd_ice1712 *ice) struct snd_info_entry *entry; if (! snd_card_proc_new(ice->card, "wm_codec", &entry)) { snd_info_set_text_ops(entry, ice, wm_proc_regs_read); - entry->mode |= S_IWUSR; + entry->mode |= 0200; entry->c.text.write = wm_proc_regs_write; } } diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c index 8dabd4d0211d..d7366ade5a25 100644 --- a/sound/pci/ice1712/prodigy_hifi.c +++ b/sound/pci/ice1712/prodigy_hifi.c @@ -926,7 +926,7 @@ static void wm_proc_init(struct snd_ice1712 *ice) struct snd_info_entry *entry; if (!snd_card_proc_new(ice->card, "wm_codec", &entry)) { snd_info_set_text_ops(entry, ice, wm_proc_regs_read); - entry->mode |= S_IWUSR; + entry->mode |= 0200; entry->c.text.write = wm_proc_regs_write; } } diff --git a/sound/pci/lola/lola_proc.c b/sound/pci/lola/lola_proc.c index c241dc06dd92..904e3c4f4dfe 100644 --- a/sound/pci/lola/lola_proc.c +++ b/sound/pci/lola/lola_proc.c @@ -214,7 +214,7 @@ void lola_proc_debug_new(struct lola *chip) snd_info_set_text_ops(entry, chip, lola_proc_codec_read); if (!snd_card_proc_new(chip->card, "codec_rw", &entry)) { snd_info_set_text_ops(entry, chip, lola_proc_codec_rw_read); - entry->mode |= S_IWUSR; + entry->mode |= 0200; entry->c.text.write = lola_proc_codec_rw_write; } if (!snd_card_proc_new(chip->card, "regs", &entry)) diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index 4ca12665ff73..81af21ac1439 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -1052,10 +1052,10 @@ static int add_controls(struct oxygen *chip, [CONTROL_CD_CAPTURE_SWITCH] = "CD Capture Switch", [CONTROL_AUX_CAPTURE_SWITCH] = "Aux Capture Switch", }; - unsigned int i, j; + unsigned int i; struct snd_kcontrol_new template; struct snd_kcontrol *ctl; - int err; + int j, err; for (i = 0; i < count; ++i) { template = controls[i]; @@ -1086,11 +1086,11 @@ static int add_controls(struct oxygen *chip, err = snd_ctl_add(chip->card, ctl); if (err < 0) return err; - for (j = 0; j < CONTROL_COUNT; ++j) - if (!strcmp(ctl->id.name, known_ctl_names[j])) { - chip->controls[j] = ctl; - ctl->private_free = oxygen_any_ctl_free; - } + j = match_string(known_ctl_names, CONTROL_COUNT, ctl->id.name); + if (j >= 0) { + chip->controls[j] = ctl; + ctl->private_free = oxygen_any_ctl_free; + } } return 0; } diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index f9ae72f28ddc..e57da4036231 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -1465,7 +1465,7 @@ static void pcxhr_proc_init(struct snd_pcxhr *chip) !snd_card_proc_new(chip->card, "gpio", &entry)) { snd_info_set_text_ops(entry, chip, pcxhr_proc_gpio_read); entry->c.text.write = pcxhr_proc_gpo_write; - entry->mode |= S_IWUSR; + entry->mode |= 0200; } if (!snd_card_proc_new(chip->card, "ltc", &entry)) snd_info_set_text_ops(entry, chip, pcxhr_proc_ltc); diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index feca0a672976..80dc42197154 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -1733,10 +1733,10 @@ static ssize_t cs43130_show_ac_r(struct device *dev, return cs43130_show_ac(dev, buf, HP_RIGHT); } -static DEVICE_ATTR(hpload_dc_l, S_IRUGO, cs43130_show_dc_l, NULL); -static DEVICE_ATTR(hpload_dc_r, S_IRUGO, cs43130_show_dc_r, NULL); -static DEVICE_ATTR(hpload_ac_l, S_IRUGO, cs43130_show_ac_l, NULL); -static DEVICE_ATTR(hpload_ac_r, S_IRUGO, cs43130_show_ac_r, NULL); +static DEVICE_ATTR(hpload_dc_l, 0444, cs43130_show_dc_l, NULL); +static DEVICE_ATTR(hpload_dc_r, 0444, cs43130_show_dc_r, NULL); +static DEVICE_ATTR(hpload_ac_l, 0444, cs43130_show_ac_l, NULL); +static DEVICE_ATTR(hpload_ac_r, 0444, cs43130_show_ac_r, NULL); static struct reg_sequence hp_en_cal_seq[] = { {CS43130_INT_MASK_4, CS43130_INT_MASK_ALL}, diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 82b0927e6ed7..af062c4f4017 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -627,22 +627,21 @@ static void wm_adsp2_init_debugfs(struct wm_adsp *dsp, if (!root) goto err; - if (!debugfs_create_bool("booted", S_IRUGO, root, &dsp->booted)) + if (!debugfs_create_bool("booted", 0444, root, &dsp->booted)) goto err; - if (!debugfs_create_bool("running", S_IRUGO, root, &dsp->running)) + if (!debugfs_create_bool("running", 0444, root, &dsp->running)) goto err; - if (!debugfs_create_x32("fw_id", S_IRUGO, root, &dsp->fw_id)) + if (!debugfs_create_x32("fw_id", 0444, root, &dsp->fw_id)) goto err; - if (!debugfs_create_x32("fw_version", S_IRUGO, root, - &dsp->fw_id_version)) + if (!debugfs_create_x32("fw_version", 0444, root, &dsp->fw_id_version)) goto err; for (i = 0; i < ARRAY_SIZE(wm_adsp_debugfs_fops); ++i) { if (!debugfs_create_file(wm_adsp_debugfs_fops[i].name, - S_IRUGO, root, dsp, + 0444, root, dsp, &wm_adsp_debugfs_fops[i].fops)) goto err; } diff --git a/sound/soc/fsl/fsl_ssi_dbg.c b/sound/soc/fsl/fsl_ssi_dbg.c index 7aac63e2c561..0ff469c027dd 100644 --- a/sound/soc/fsl/fsl_ssi_dbg.c +++ b/sound/soc/fsl/fsl_ssi_dbg.c @@ -146,7 +146,7 @@ int fsl_ssi_debugfs_create(struct fsl_ssi_dbg *ssi_dbg, struct device *dev) if (!ssi_dbg->dbg_dir) return -ENOMEM; - ssi_dbg->dbg_stats = debugfs_create_file("stats", S_IRUGO, + ssi_dbg->dbg_stats = debugfs_create_file("stats", 0444, ssi_dbg->dbg_dir, ssi_dbg, &fsl_ssi_stats_ops); if (!ssi_dbg->dbg_stats) { diff --git a/sound/sound_core.c b/sound/sound_core.c index b4efb22db561..40ad000c2e3c 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -413,7 +413,7 @@ int register_sound_special_device(const struct file_operations *fops, int unit, break; } return sound_insert_unit(&chains[chain], fops, -1, unit, max_unit, - name, S_IRUSR | S_IWUSR, dev); + name, 0600, dev); } EXPORT_SYMBOL(register_sound_special_device); @@ -440,7 +440,7 @@ EXPORT_SYMBOL(register_sound_special); int register_sound_mixer(const struct file_operations *fops, int dev) { return sound_insert_unit(&chains[0], fops, dev, 0, 128, - "mixer", S_IRUSR | S_IWUSR, NULL); + "mixer", 0600, NULL); } EXPORT_SYMBOL(register_sound_mixer); @@ -468,7 +468,7 @@ EXPORT_SYMBOL(register_sound_mixer); int register_sound_dsp(const struct file_operations *fops, int dev) { return sound_insert_unit(&chains[3], fops, dev, 3, 131, - "dsp", S_IWUSR | S_IRUSR, NULL); + "dsp", 0600, NULL); } EXPORT_SYMBOL(register_sound_dsp); diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index abc7bd5055eb..7609eceba1a2 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -2518,7 +2518,7 @@ static void snd_dbri_proc(struct snd_card *card) #ifdef DBRI_DEBUG if (!snd_card_proc_new(card, "debug", &entry)) { snd_info_set_text_ops(entry, dbri, dbri_debug_read); - entry->mode = S_IFREG | S_IRUGO; /* Readable only. */ + entry->mode = S_IFREG | 0444; /* Readable only. */ } #endif } @@ -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 4a1c6bb3dfa0..a1ed798a1c6b 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -86,6 +86,8 @@ static bool ignore_ctl_error; static bool autoclock = true; static char *quirk_alias[SNDRV_CARDS]; +bool snd_usb_use_vmalloc = true; + module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for the USB audio adapter."); module_param_array(id, charp, NULL, 0444); @@ -105,6 +107,8 @@ module_param(autoclock, bool, 0444); MODULE_PARM_DESC(autoclock, "Enable auto-clock selection for UAC2 devices (default: yes)."); module_param_array(quirk_alias, charp, NULL, 0444); MODULE_PARM_DESC(quirk_alias, "Quirk aliases, e.g. 0123abcd:5678beef."); +module_param_named(use_vmalloc, snd_usb_use_vmalloc, bool, 0444); +MODULE_PARM_DESC(use_vmalloc, "Use vmalloc for PCM intermediate buffers (default: yes)."); /* * we keep the snd_usb_audio_t instances by ourselves for merging @@ -221,32 +225,13 @@ 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 +240,25 @@ 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 +311,20 @@ 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; @@ -329,8 +346,9 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) * */ -static int snd_usb_audio_free(struct snd_usb_audio *chip) +static void snd_usb_audio_free(struct snd_card *card) { + struct snd_usb_audio *chip = card->private_data; struct snd_usb_endpoint *ep, *n; list_for_each_entry_safe(ep, n, &chip->ep_list, list) @@ -339,14 +357,90 @@ static int snd_usb_audio_free(struct snd_usb_audio *chip) mutex_destroy(&chip->mutex); if (!atomic_read(&chip->shutdown)) dev_set_drvdata(&chip->dev->dev, NULL); - kfree(chip); - return 0; } -static int snd_usb_audio_dev_free(struct snd_device *device) +static void usb_audio_make_shortname(struct usb_device *dev, + struct snd_usb_audio *chip, + const struct snd_usb_audio_quirk *quirk) { - struct snd_usb_audio *chip = device->device_data; - return snd_usb_audio_free(chip); + 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; + } } /* @@ -360,11 +454,8 @@ 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, - }; *rchip = NULL; @@ -382,18 +473,13 @@ static int snd_usb_audio_create(struct usb_interface *intf, } err = snd_card_new(&intf->dev, index[idx], id[idx], THIS_MODULE, - 0, &card); + sizeof(*chip), &card); if (err < 0) { dev_err(&dev->dev, "cannot create card instance %d\n", idx); return err; } - chip = kzalloc(sizeof(*chip), GFP_KERNEL); - if (! chip) { - snd_card_free(card); - return -ENOMEM; - } - + chip = card->private_data; mutex_init(&chip->mutex); init_waitqueue_head(&chip->shutdown_wait); chip->index = idx; @@ -411,75 +497,15 @@ static int snd_usb_audio_create(struct usb_interface *intf, INIT_LIST_HEAD(&chip->midi_list); INIT_LIST_HEAD(&chip->mixer_list); - if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) { - snd_usb_audio_free(chip); - snd_card_free(card); - return err; - } + card->private_free = snd_usb_audio_free; strcpy(card->driver, "USB-Audio"); sprintf(component, "USB%04x:%04x", 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 0b030d8fe3fa..c79749613fa6 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -443,10 +443,11 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface, data[0] = rate; data[1] = rate >> 8; data[2] = rate >> 16; - if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR, - USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_OUT, - UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep, - data, sizeof(data))) < 0) { + err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR, + USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_OUT, + UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep, + data, sizeof(data)); + if (err < 0) { dev_err(&dev->dev, "%d:%d: cannot set freq %d to ep %#x\n", iface, fmt->altsetting, rate, ep); return err; @@ -460,10 +461,11 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface, if (chip->sample_rate_read_error > 2) return 0; - if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC_GET_CUR, - USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_IN, - UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep, - data, sizeof(data))) < 0) { + err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC_GET_CUR, + USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_IN, + UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep, + data, sizeof(data)); + if (err < 0) { dev_err(&dev->dev, "%d:%d: cannot get freq at ep %#x\n", iface, fmt->altsetting, ep); chip->sample_rate_read_error++; @@ -587,8 +589,15 @@ 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/helper.h b/sound/usb/helper.h index 4463e6d6dcb3..d338bd0e0ca6 100644 --- a/sound/usb/helper.h +++ b/sound/usb/helper.h @@ -18,16 +18,12 @@ unsigned char snd_usb_parse_datainterval(struct snd_usb_audio *chip, * retrieve usb_interface descriptor from the host interface * (conditional for compatibility with the older API) */ -#ifndef get_iface_desc #define get_iface_desc(iface) (&(iface)->desc) #define get_endpoint(alt,ep) (&(alt)->endpoint[ep].desc) #define get_ep_desc(ep) (&(ep)->desc) #define get_cfg_desc(cfg) (&(cfg)->desc) -#endif -#ifndef snd_usb_get_speed #define snd_usb_get_speed(dev) ((dev)->speed) -#endif static inline int snd_usb_ctrl_intf(struct snd_usb_audio *chip) { diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index bb5ab7a7dfa5..898afd3001ea 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; @@ -600,7 +598,8 @@ int snd_usb_mixer_add_control(struct usb_mixer_elem_list *list, while (snd_ctl_find_id(mixer->chip->card, &kctl->id)) kctl->id.index++; - if ((err = snd_ctl_add(mixer->chip->card, kctl)) < 0) { + err = snd_ctl_add(mixer->chip->card, kctl); + if (err < 0) { usb_audio_dbg(mixer->chip, "cannot add control (err = %d)\n", err); return err; @@ -658,14 +657,14 @@ 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; @@ -719,6 +718,66 @@ static int get_term_name(struct mixer_build *state, struct usb_audio_term *iterm } /* + * Get logical cluster information for UAC3 devices. + */ +static int get_cluster_channels_v3(struct mixer_build *state, unsigned int cluster_id) +{ + struct uac3_cluster_header_descriptor c_header; + int err; + + err = snd_usb_ctl_msg(state->chip->dev, + usb_rcvctrlpipe(state->chip->dev, 0), + UAC3_CS_REQ_HIGH_CAPABILITY_DESCRIPTOR, + USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, + cluster_id, + snd_usb_ctrl_intf(state->chip), + &c_header, sizeof(c_header)); + if (err < 0) + goto error; + if (err != sizeof(c_header)) { + err = -EIO; + goto error; + } + + return c_header.bNrChannels; + +error: + usb_audio_err(state->chip, "cannot request logical cluster ID: %d (err: %d)\n", cluster_id, err); + return err; +} + +/* + * Get number of channels for a Mixer Unit. + */ +static int uac_mixer_unit_get_channels(struct mixer_build *state, + struct uac_mixer_unit_descriptor *desc) +{ + int mu_channels; + + if (desc->bLength < 11) + return -EINVAL; + if (!desc->bNrInPins) + return -EINVAL; + + switch (state->mixer->protocol) { + case UAC_VERSION_1: + case UAC_VERSION_2: + default: + mu_channels = uac_mixer_unit_bNrChannels(desc); + break; + case UAC_VERSION_3: + mu_channels = get_cluster_channels_v3(state, + uac3_mixer_unit_wClusterDescrID(desc)); + break; + } + + if (!mu_channels) + return -EINVAL; + + return mu_channels; +} + +/* * parse the source unit recursively until it reaches to a terminal * or a branched unit. */ @@ -844,8 +903,12 @@ static int check_input_term(struct mixer_build *state, int id, term->id = id; term->type = le16_to_cpu(d->wTerminalType); - /* REVISIT: UAC3 IT doesn't have channels/cfg */ - term->channels = 0; + err = get_cluster_channels_v3(state, le16_to_cpu(d->wClusterDescrID)); + if (err < 0) + return err; + term->channels = err; + + /* REVISIT: UAC3 IT doesn't have channels cfg */ term->chconfig = 0; term->name = le16_to_cpu(d->wTerminalDescrStr); @@ -865,6 +928,18 @@ static int check_input_term(struct mixer_build *state, int id, term->name = le16_to_cpu(d->wClockSourceStr); return 0; } + case UAC3_MIXER_UNIT: { + struct uac_mixer_unit_descriptor *d = p1; + + err = uac_mixer_unit_get_channels(state, d); + if (err < 0) + return err; + + term->channels = err; + term->type = d->bDescriptorSubtype << 16; /* virtual type */ + + return 0; + } default: return -ENODEV; } @@ -1258,6 +1333,51 @@ static int mixer_ctl_master_bool_get(struct snd_kcontrol *kcontrol, return 0; } +/* get the connectors status and report it as boolean type */ +static int mixer_ctl_connector_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *cval = kcontrol->private_data; + struct snd_usb_audio *chip = cval->head.mixer->chip; + int idx = 0, validx, ret, val; + + validx = cval->control << 8 | 0; + + ret = snd_usb_lock_shutdown(chip) ? -EIO : 0; + if (ret) + goto error; + + idx = snd_usb_ctrl_intf(chip) | (cval->head.id << 8); + if (cval->head.mixer->protocol == UAC_VERSION_2) { + struct uac2_connectors_ctl_blk uac2_conn; + + ret = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), UAC2_CS_CUR, + USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, + validx, idx, &uac2_conn, sizeof(uac2_conn)); + val = !!uac2_conn.bNrChannels; + } else { /* UAC_VERSION_3 */ + struct uac3_insertion_ctl_blk uac3_conn; + + ret = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), UAC2_CS_CUR, + USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, + validx, idx, &uac3_conn, sizeof(uac3_conn)); + val = !!uac3_conn.bmConInserted; + } + + snd_usb_unlock_shutdown(chip); + + if (ret < 0) { +error: + usb_audio_err(chip, + "cannot get connectors status: req = %#x, wValue = %#x, wIndex = %#x, type = %d\n", + UAC_GET_CUR, validx, idx, cval->val_type); + return ret; + } + + ucontrol->value.integer.value[0] = val; + return 0; +} + static struct snd_kcontrol_new usb_feature_unit_ctl = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "", /* will be filled later manually */ @@ -1288,6 +1408,15 @@ static struct snd_kcontrol_new usb_bool_master_control_ctl_ro = { .put = NULL, }; +static const struct snd_kcontrol_new usb_connector_ctl_ro = { + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .name = "", /* will be filled later manually */ + .access = SNDRV_CTL_ELEM_ACCESS_READ, + .info = snd_ctl_boolean_mono_info, + .get = mixer_ctl_connector_get, + .put = NULL, +}; + /* * This symbol is exported in order to allow the mixer quirks to * hook up to the standard feature unit control mechanism @@ -1341,16 +1470,16 @@ 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 +1490,14 @@ 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 +1506,7 @@ 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 +1535,7 @@ 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 +1544,7 @@ 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 +1559,12 @@ 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 +1573,15 @@ 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 +1609,7 @@ 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 +1618,46 @@ 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); @@ -1534,17 +1685,25 @@ static void build_connector_control(struct mixer_build *state, return; snd_usb_mixer_elem_init_std(&cval->head, state->mixer, term->id); /* - * The first byte from reading the UAC2_TE_CONNECTOR control returns the - * number of channels connected. This boolean ctl will simply report - * if any channels are connected or not. - * (Audio20_final.pdf Table 5-10: Connector Control CUR Parameter Block) + * UAC2: The first byte from reading the UAC2_TE_CONNECTOR control returns the + * number of channels connected. + * + * UAC3: The first byte specifies size of bitmap for the inserted controls. The + * following byte(s) specifies which connectors are inserted. + * + * This boolean ctl will simply report if any channels are connected + * or not. */ - cval->control = UAC2_TE_CONNECTOR; + if (state->mixer->protocol == UAC_VERSION_2) + cval->control = UAC2_TE_CONNECTOR; + else /* UAC_VERSION_3 */ + cval->control = UAC3_TE_INSERTION; + cval->val_type = USB_MIXER_BOOLEAN; cval->channels = 1; /* report true if any channel is connected */ cval->min = 0; cval->max = 1; - kctl = snd_ctl_new1(&usb_bool_master_control_ctl_ro, cval); + kctl = snd_ctl_new1(&usb_connector_ctl_ro, cval); if (!kctl) { usb_audio_err(state->chip, "cannot malloc kcontrol\n"); kfree(cval); @@ -1605,7 +1764,7 @@ 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), @@ -1692,7 +1851,8 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, } /* parse the source unit */ - if ((err = parse_audio_unit(state, hdr->bSourceID)) < 0) + err = parse_audio_unit(state, hdr->bSourceID); + if (err < 0) return err; /* determine the input source type and name */ @@ -1806,16 +1966,15 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, */ static void build_mixer_unit_ctl(struct mixer_build *state, struct uac_mixer_unit_descriptor *desc, - int in_pin, int in_ch, int unitid, - struct usb_audio_term *iterm) + int in_pin, int in_ch, int num_outs, + int unitid, struct usb_audio_term *iterm) { struct usb_mixer_elem_info *cval; - unsigned int num_outs = uac_mixer_unit_bNrChannels(desc); unsigned int i, len; 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 +2007,7 @@ 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); @@ -1863,16 +2022,28 @@ static int parse_audio_input_terminal(struct mixer_build *state, int unitid, void *raw_desc) { struct usb_audio_term iterm; - struct uac2_input_terminal_descriptor *d = raw_desc; + unsigned int control, bmctls, term_id; - check_input_term(state, d->bTerminalID, &iterm); if (state->mixer->protocol == UAC_VERSION_2) { - /* Check for jack detection. */ - if (uac_v2v3_control_is_readable(le16_to_cpu(d->bmControls), - UAC2_TE_CONNECTOR)) { - build_connector_control(state, &iterm, true); - } + struct uac2_input_terminal_descriptor *d_v2 = raw_desc; + control = UAC2_TE_CONNECTOR; + term_id = d_v2->bTerminalID; + bmctls = le16_to_cpu(d_v2->bmControls); + } else if (state->mixer->protocol == UAC_VERSION_3) { + struct uac3_input_terminal_descriptor *d_v3 = raw_desc; + control = UAC3_TE_INSERTION; + term_id = d_v3->bTerminalID; + bmctls = le32_to_cpu(d_v3->bmControls); + } else { + return 0; /* UAC1. No Insertion control */ } + + check_input_term(state, term_id, &iterm); + + /* Check for jack detection. */ + if (uac_v2v3_control_is_readable(bmctls, control)) + build_connector_control(state, &iterm, true); + return 0; } @@ -1887,14 +2058,17 @@ static int parse_audio_mixer_unit(struct mixer_build *state, int unitid, int input_pins, num_ins, num_outs; int pin, ich, err; - if (desc->bLength < 11 || !(input_pins = desc->bNrInPins) || - !(num_outs = uac_mixer_unit_bNrChannels(desc))) { + err = uac_mixer_unit_get_channels(state, desc); + if (err < 0) { usb_audio_err(state->chip, "invalid MIXER UNIT descriptor %d\n", unitid); - return -EINVAL; + return err; } + num_outs = err; + input_pins = desc->bNrInPins; + num_ins = 0; ich = 0; for (pin = 0; pin < input_pins; pin++) { @@ -1921,7 +2095,7 @@ static int parse_audio_mixer_unit(struct mixer_build *state, int unitid, } } if (ich_has_controls) - build_mixer_unit_ctl(state, desc, pin, ich, + build_mixer_unit_ctl(state, desc, pin, ich, num_outs, unitid, &iterm); } } @@ -2098,7 +2272,8 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, } for (i = 0; i < num_ins; i++) { - if ((err = parse_audio_unit(state, desc->baSourceID[i])) < 0) + err = parse_audio_unit(state, desc->baSourceID[i]); + if (err < 0) return err; } @@ -2114,7 +2289,7 @@ 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 +2337,8 @@ 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) @@ -2310,14 +2486,15 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, } for (i = 0; i < desc->bNrInPins; i++) { - if ((err = parse_audio_unit(state, desc->baSourceID[i])) < 0) + err = parse_audio_unit(state, desc->baSourceID[i]); + if (err < 0) return err; } 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 +2535,8 @@ 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 +2558,12 @@ 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) @@ -2458,7 +2636,7 @@ static int parse_audio_unit(struct mixer_build *state, int unitid) } else { /* UAC_VERSION_3 */ switch (p1[2]) { case UAC_INPUT_TERMINAL: - return 0; /* NOP */ + return parse_audio_input_terminal(state, unitid, p1); case UAC3_MIXER_UNIT: return parse_audio_mixer_unit(state, unitid, p1); case UAC3_CLOCK_SOURCE: @@ -2503,6 +2681,246 @@ 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 * @@ -2596,6 +3014,12 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) err = parse_audio_unit(&state, desc->bCSourceID); if (err < 0 && err != -EINVAL) return err; + + if (uac_v2v3_control_is_readable(le32_to_cpu(desc->bmControls), + UAC3_TE_INSERTION)) { + build_connector_control(&state, &state.oterm, + false); + } } } @@ -2606,9 +3030,9 @@ 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 +3043,7 @@ 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 +3069,7 @@ 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 +3099,19 @@ 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 +3232,48 @@ 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 +3312,21 @@ 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) { + err = snd_usb_mixer_controls_badd(mixer, ctrlif); + if (err < 0) + goto _error; + } else { + err = snd_usb_mixer_controls(mixer); + if (err < 0) + goto _error; + err = snd_usb_mixer_status_create(mixer); + if (err < 0) + goto _error; + } + err = create_keep_iface_ctl(mixer); + if (err < 0) goto _error; snd_usb_mixer_apply_create_quirk(mixer); @@ -2909,7 +3387,7 @@ 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 +3423,7 @@ 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 +3433,8 @@ 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 ba27f7ade670..e02653465e29 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 eaa03acd4686..71069e110897 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 56537a156580..4149543f613e 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, @@ -1799,12 +1799,33 @@ 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; struct snd_info_entry *entry; - if ((err = snd_usb_soundblaster_remote_init(mixer)) < 0) + err = snd_usb_soundblaster_remote_init(mixer); + if (err < 0) return err; switch (mixer->chip->usb_id) { @@ -1884,11 +1905,25 @@ 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 b5abd328a361..52be26db558f 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 c33e2378089d..4aeb9488a0c9 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 3cbfae6604f9..78d1cad08a0a 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -76,10 +76,9 @@ snd_pcm_uframes_t snd_usb_pcm_delay(struct snd_usb_substream *subs, */ static snd_pcm_uframes_t snd_usb_pcm_pointer(struct snd_pcm_substream *substream) { - struct snd_usb_substream *subs; + struct snd_usb_substream *subs = substream->runtime->private_data; unsigned int hwptr_done; - subs = (struct snd_usb_substream *)substream->runtime->private_data; if (atomic_read(&subs->stream->chip->shutdown)) return SNDRV_PCM_POS_XRUN; spin_lock(&subs->lock); @@ -164,10 +163,11 @@ static int init_pitch_v1(struct snd_usb_audio *chip, int iface, ep = get_endpoint(alts, 0)->bEndpointAddress; data[0] = 1; - if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR, - USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_OUT, - UAC_EP_CS_ATTR_PITCH_CONTROL << 8, ep, - data, sizeof(data))) < 0) { + err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR, + USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_OUT, + UAC_EP_CS_ATTR_PITCH_CONTROL << 8, ep, + data, sizeof(data)); + if (err < 0) { usb_audio_err(chip, "%d:%d: cannot set enable PITCH\n", iface, ep); return err; @@ -185,10 +185,11 @@ static int init_pitch_v2(struct snd_usb_audio *chip, int iface, int err; data[0] = 1; - if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC2_CS_CUR, - USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_OUT, - UAC2_EP_CS_PITCH << 8, 0, - data, sizeof(data))) < 0) { + err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC2_CS_CUR, + USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_OUT, + UAC2_EP_CS_PITCH << 8, 0, + data, sizeof(data)); + if (err < 0) { usb_audio_err(chip, "%d:%d: cannot set enable PITCH (v2)\n", iface, fmt->altsetting); return err; @@ -321,6 +322,7 @@ 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 +332,27 @@ 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 +367,14 @@ 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 +500,7 @@ 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 +510,21 @@ 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 +538,11 @@ 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); @@ -736,7 +728,11 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, struct audioformat *fmt; int ret; - ret = snd_pcm_lib_alloc_vmalloc_buffer(substream, + if (snd_usb_use_vmalloc) + ret = snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); + else + ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); if (ret < 0) return ret; @@ -789,7 +785,11 @@ static int snd_usb_hw_free(struct snd_pcm_substream *substream) snd_usb_endpoint_deactivate(subs->data_endpoint); snd_usb_unlock_shutdown(subs->stream->chip); } - return snd_pcm_lib_free_vmalloc_buffer(substream); + + if (snd_usb_use_vmalloc) + return snd_pcm_lib_free_vmalloc_buffer(substream); + else + return snd_pcm_lib_free_pages(substream); } /* @@ -1181,9 +1181,6 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre pt = 125 * (1 << fp->datainterval); ptmin = min(ptmin, pt); } - err = snd_usb_autoresume(subs->stream->chip); - if (err < 0) - return err; param_period_time_if_needed = SNDRV_PCM_HW_PARAM_PERIOD_TIME; if (subs->speed == USB_SPEED_FULL) @@ -1192,30 +1189,37 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre if (ptmin == 1000) /* if period time doesn't go below 1 ms, no rules needed */ param_period_time_if_needed = -1; - snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_TIME, - ptmin, UINT_MAX); - - if ((err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - hw_rule_rate, subs, - SNDRV_PCM_HW_PARAM_FORMAT, - SNDRV_PCM_HW_PARAM_CHANNELS, - param_period_time_if_needed, - -1)) < 0) - goto rep_err; - if ((err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - hw_rule_channels, subs, - SNDRV_PCM_HW_PARAM_FORMAT, - SNDRV_PCM_HW_PARAM_RATE, - param_period_time_if_needed, - -1)) < 0) - goto rep_err; - if ((err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, - hw_rule_format, subs, - SNDRV_PCM_HW_PARAM_RATE, - SNDRV_PCM_HW_PARAM_CHANNELS, - param_period_time_if_needed, - -1)) < 0) - goto rep_err; + + err = snd_pcm_hw_constraint_minmax(runtime, + SNDRV_PCM_HW_PARAM_PERIOD_TIME, + ptmin, UINT_MAX); + if (err < 0) + return err; + + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + hw_rule_rate, subs, + SNDRV_PCM_HW_PARAM_FORMAT, + SNDRV_PCM_HW_PARAM_CHANNELS, + param_period_time_if_needed, + -1); + if (err < 0) + return err; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + hw_rule_channels, subs, + SNDRV_PCM_HW_PARAM_FORMAT, + SNDRV_PCM_HW_PARAM_RATE, + param_period_time_if_needed, + -1); + if (err < 0) + return err; + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, + hw_rule_format, subs, + SNDRV_PCM_HW_PARAM_RATE, + SNDRV_PCM_HW_PARAM_CHANNELS, + param_period_time_if_needed, + -1); + if (err < 0) + return err; if (param_period_time_if_needed >= 0) { err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_TIME, @@ -1225,19 +1229,18 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre SNDRV_PCM_HW_PARAM_RATE, -1); if (err < 0) - goto rep_err; + return err; } - if ((err = snd_usb_pcm_check_knot(runtime, subs)) < 0) - goto rep_err; - return 0; + err = snd_usb_pcm_check_knot(runtime, subs); + if (err < 0) + return err; -rep_err: - snd_usb_autosuspend(subs->stream->chip); - return err; + return snd_usb_autoresume(subs->stream->chip); } -static int snd_usb_pcm_open(struct snd_pcm_substream *substream, int direction) +static int snd_usb_pcm_open(struct snd_pcm_substream *substream) { + int direction = substream->stream; struct snd_usb_stream *as = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; struct snd_usb_substream *subs = &as->substream[direction]; @@ -1257,14 +1260,16 @@ static int snd_usb_pcm_open(struct snd_pcm_substream *substream, int direction) return setup_hw_info(runtime, subs); } -static int snd_usb_pcm_close(struct snd_pcm_substream *substream, int direction) +static int snd_usb_pcm_close(struct snd_pcm_substream *substream) { + int direction = substream->stream; struct snd_usb_stream *as = snd_pcm_substream_chip(substream); struct snd_usb_substream *subs = &as->substream[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; @@ -1311,7 +1316,7 @@ static void retire_capture_urb(struct snd_usb_substream *subs, if (bytes % (runtime->sample_bits >> 3) != 0) { int oldbytes = bytes; bytes = frames * stride; - dev_warn(&subs->dev->dev, + dev_warn_ratelimited(&subs->dev->dev, "Corrected urb data len. %d->%d\n", oldbytes, bytes); } @@ -1619,26 +1624,6 @@ static void retire_playback_urb(struct snd_usb_substream *subs, spin_unlock_irqrestore(&subs->lock, flags); } -static int snd_usb_playback_open(struct snd_pcm_substream *substream) -{ - return snd_usb_pcm_open(substream, SNDRV_PCM_STREAM_PLAYBACK); -} - -static int snd_usb_playback_close(struct snd_pcm_substream *substream) -{ - return snd_usb_pcm_close(substream, SNDRV_PCM_STREAM_PLAYBACK); -} - -static int snd_usb_capture_open(struct snd_pcm_substream *substream) -{ - return snd_usb_pcm_open(substream, SNDRV_PCM_STREAM_CAPTURE); -} - -static int snd_usb_capture_close(struct snd_pcm_substream *substream) -{ - return snd_usb_pcm_close(substream, SNDRV_PCM_STREAM_CAPTURE); -} - static int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substream, int cmd) { @@ -1700,8 +1685,8 @@ static int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream } static const struct snd_pcm_ops snd_usb_playback_ops = { - .open = snd_usb_playback_open, - .close = snd_usb_playback_close, + .open = snd_usb_pcm_open, + .close = snd_usb_pcm_close, .ioctl = snd_pcm_lib_ioctl, .hw_params = snd_usb_hw_params, .hw_free = snd_usb_hw_free, @@ -1713,8 +1698,8 @@ static const struct snd_pcm_ops snd_usb_playback_ops = { }; static const struct snd_pcm_ops snd_usb_capture_ops = { - .open = snd_usb_capture_open, - .close = snd_usb_capture_close, + .open = snd_usb_pcm_open, + .close = snd_usb_pcm_close, .ioctl = snd_pcm_lib_ioctl, .hw_params = snd_usb_hw_params, .hw_free = snd_usb_hw_free, @@ -1725,9 +1710,50 @@ static const struct snd_pcm_ops snd_usb_capture_ops = { .mmap = snd_pcm_lib_mmap_vmalloc, }; +static const struct snd_pcm_ops snd_usb_playback_dev_ops = { + .open = snd_usb_pcm_open, + .close = snd_usb_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_usb_hw_params, + .hw_free = snd_usb_hw_free, + .prepare = snd_usb_pcm_prepare, + .trigger = snd_usb_substream_playback_trigger, + .pointer = snd_usb_pcm_pointer, + .page = snd_pcm_sgbuf_ops_page, +}; + +static const struct snd_pcm_ops snd_usb_capture_dev_ops = { + .open = snd_usb_pcm_open, + .close = snd_usb_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_usb_hw_params, + .hw_free = snd_usb_hw_free, + .prepare = snd_usb_pcm_prepare, + .trigger = snd_usb_substream_capture_trigger, + .pointer = snd_usb_pcm_pointer, + .page = snd_pcm_sgbuf_ops_page, +}; + void snd_usb_set_pcm_ops(struct snd_pcm *pcm, int stream) { - snd_pcm_set_ops(pcm, stream, - stream == SNDRV_PCM_STREAM_PLAYBACK ? - &snd_usb_playback_ops : &snd_usb_capture_ops); + const struct snd_pcm_ops *ops; + + if (snd_usb_use_vmalloc) + ops = stream == SNDRV_PCM_STREAM_PLAYBACK ? + &snd_usb_playback_ops : &snd_usb_capture_ops; + else + ops = stream == SNDRV_PCM_STREAM_PLAYBACK ? + &snd_usb_playback_dev_ops : &snd_usb_capture_dev_ops; + snd_pcm_set_ops(pcm, stream, ops); +} + +void snd_usb_preallocate_buffer(struct snd_usb_substream *subs) +{ + struct snd_pcm *pcm = subs->stream->pcm; + struct snd_pcm_substream *s = pcm->streams[subs->direction].substream; + struct device *dev = subs->dev->bus->controller; + + if (!snd_usb_use_vmalloc) + snd_pcm_lib_preallocate_pages(s, SNDRV_DMA_TYPE_DEV_SG, + dev, 64*1024, 512*1024); } diff --git a/sound/usb/pcm.h b/sound/usb/pcm.h index 35740d5ef268..f77ec58bf1a1 100644 --- a/sound/usb/pcm.h +++ b/sound/usb/pcm.h @@ -10,6 +10,7 @@ void snd_usb_set_pcm_ops(struct snd_pcm *pcm, int stream); int snd_usb_init_pitch(struct snd_usb_audio *chip, int iface, struct usb_host_interface *alts, struct audioformat *fmt); +void snd_usb_preallocate_buffer(struct snd_usb_substream *subs); #endif /* __USBAUDIO_PCM_H */ diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 754e632a27bd..0e37e358ca97 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 acbeb52f6fd6..f4b69173682c 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 5ed334575fc7..729afd808cc4 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -106,6 +106,8 @@ static void snd_usb_init_substream(struct snd_usb_stream *as, subs->ep_num = fp->endpoint; if (fp->channels > subs->channels_max) subs->channels_max = fp->channels; + + snd_usb_preallocate_buffer(subs); } /* kctl callbacks for usb-audio channel maps */ @@ -633,6 +635,395 @@ 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); + kfree(chmap); + return NULL; + +found_clock: + fp = audio_format_alloc_init(chip, alts, UAC_VERSION_3, iface_no, + altset_idx, altno, num_channels, clock); + if (!fp) { + kfree(chmap); + 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->chmap); + 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 +1031,8 @@ 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 +1081,48 @@ 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; - } - - fp = kzalloc(sizeof(*fp), GFP_KERNEL); - if (!fp) - return -ENOMEM; + bm_quirk = 1; - 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; - } + 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; } - /* 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); + if (!fp) + 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 4d5c89a7ba2b..b9faeca645fd 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; @@ -121,4 +127,6 @@ struct snd_usb_audio_quirk { int snd_usb_lock_shutdown(struct snd_usb_audio *chip); void snd_usb_unlock_shutdown(struct snd_usb_audio *chip); +extern bool snd_usb_use_vmalloc; + #endif /* __USBAUDIO_H */ diff --git a/sound/xen/Kconfig b/sound/xen/Kconfig new file mode 100644 index 000000000000..4f1fceea82d2 --- /dev/null +++ b/sound/xen/Kconfig @@ -0,0 +1,10 @@ +# ALSA Xen drivers + +config SND_XEN_FRONTEND + tristate "Xen para-virtualized sound frontend driver" + depends on XEN + select SND_PCM + select XEN_XENBUS_FRONTEND + help + Choose this option if you want to enable a para-virtualized + frontend sound driver for Xen guest OSes. diff --git a/sound/xen/Makefile b/sound/xen/Makefile new file mode 100644 index 000000000000..1e6470ecc2f2 --- /dev/null +++ b/sound/xen/Makefile @@ -0,0 +1,9 @@ +# SPDX-License-Identifier: GPL-2.0 OR MIT + +snd_xen_front-objs := xen_snd_front.o \ + xen_snd_front_cfg.o \ + xen_snd_front_evtchnl.o \ + xen_snd_front_shbuf.o \ + xen_snd_front_alsa.o + +obj-$(CONFIG_SND_XEN_FRONTEND) += snd_xen_front.o diff --git a/sound/xen/xen_snd_front.c b/sound/xen/xen_snd_front.c new file mode 100644 index 000000000000..b089b13b5160 --- /dev/null +++ b/sound/xen/xen_snd_front.c @@ -0,0 +1,397 @@ +// SPDX-License-Identifier: GPL-2.0 OR MIT + +/* + * Xen para-virtual sound device + * + * Copyright (C) 2016-2018 EPAM Systems Inc. + * + * Author: Oleksandr Andrushchenko <oleksandr_andrushchenko@epam.com> + */ + +#include <linux/delay.h> +#include <linux/module.h> + +#include <xen/page.h> +#include <xen/platform_pci.h> +#include <xen/xen.h> +#include <xen/xenbus.h> + +#include <xen/interface/io/sndif.h> + +#include "xen_snd_front.h" +#include "xen_snd_front_alsa.h" +#include "xen_snd_front_evtchnl.h" +#include "xen_snd_front_shbuf.h" + +static struct xensnd_req * +be_stream_prepare_req(struct xen_snd_front_evtchnl *evtchnl, u8 operation) +{ + struct xensnd_req *req; + + req = RING_GET_REQUEST(&evtchnl->u.req.ring, + evtchnl->u.req.ring.req_prod_pvt); + req->operation = operation; + req->id = evtchnl->evt_next_id++; + evtchnl->evt_id = req->id; + return req; +} + +static int be_stream_do_io(struct xen_snd_front_evtchnl *evtchnl) +{ + if (unlikely(evtchnl->state != EVTCHNL_STATE_CONNECTED)) + return -EIO; + + reinit_completion(&evtchnl->u.req.completion); + xen_snd_front_evtchnl_flush(evtchnl); + return 0; +} + +static int be_stream_wait_io(struct xen_snd_front_evtchnl *evtchnl) +{ + if (wait_for_completion_timeout(&evtchnl->u.req.completion, + msecs_to_jiffies(VSND_WAIT_BACK_MS)) <= 0) + return -ETIMEDOUT; + + return evtchnl->u.req.resp_status; +} + +int xen_snd_front_stream_query_hw_param(struct xen_snd_front_evtchnl *evtchnl, + struct xensnd_query_hw_param *hw_param_req, + struct xensnd_query_hw_param *hw_param_resp) +{ + struct xensnd_req *req; + int ret; + + mutex_lock(&evtchnl->u.req.req_io_lock); + + mutex_lock(&evtchnl->ring_io_lock); + req = be_stream_prepare_req(evtchnl, XENSND_OP_HW_PARAM_QUERY); + req->op.hw_param = *hw_param_req; + mutex_unlock(&evtchnl->ring_io_lock); + + ret = be_stream_do_io(evtchnl); + + if (ret == 0) + ret = be_stream_wait_io(evtchnl); + + if (ret == 0) + *hw_param_resp = evtchnl->u.req.resp.hw_param; + + mutex_unlock(&evtchnl->u.req.req_io_lock); + return ret; +} + +int xen_snd_front_stream_prepare(struct xen_snd_front_evtchnl *evtchnl, + struct xen_snd_front_shbuf *sh_buf, + u8 format, unsigned int channels, + unsigned int rate, u32 buffer_sz, + u32 period_sz) +{ + struct xensnd_req *req; + int ret; + + mutex_lock(&evtchnl->u.req.req_io_lock); + + mutex_lock(&evtchnl->ring_io_lock); + req = be_stream_prepare_req(evtchnl, XENSND_OP_OPEN); + req->op.open.pcm_format = format; + req->op.open.pcm_channels = channels; + req->op.open.pcm_rate = rate; + req->op.open.buffer_sz = buffer_sz; + req->op.open.period_sz = period_sz; + req->op.open.gref_directory = xen_snd_front_shbuf_get_dir_start(sh_buf); + mutex_unlock(&evtchnl->ring_io_lock); + + ret = be_stream_do_io(evtchnl); + + if (ret == 0) + ret = be_stream_wait_io(evtchnl); + + mutex_unlock(&evtchnl->u.req.req_io_lock); + return ret; +} + +int xen_snd_front_stream_close(struct xen_snd_front_evtchnl *evtchnl) +{ + struct xensnd_req *req; + int ret; + + mutex_lock(&evtchnl->u.req.req_io_lock); + + mutex_lock(&evtchnl->ring_io_lock); + req = be_stream_prepare_req(evtchnl, XENSND_OP_CLOSE); + mutex_unlock(&evtchnl->ring_io_lock); + + ret = be_stream_do_io(evtchnl); + + if (ret == 0) + ret = be_stream_wait_io(evtchnl); + + mutex_unlock(&evtchnl->u.req.req_io_lock); + return ret; +} + +int xen_snd_front_stream_write(struct xen_snd_front_evtchnl *evtchnl, + unsigned long pos, unsigned long count) +{ + struct xensnd_req *req; + int ret; + + mutex_lock(&evtchnl->u.req.req_io_lock); + + mutex_lock(&evtchnl->ring_io_lock); + req = be_stream_prepare_req(evtchnl, XENSND_OP_WRITE); + req->op.rw.length = count; + req->op.rw.offset = pos; + mutex_unlock(&evtchnl->ring_io_lock); + + ret = be_stream_do_io(evtchnl); + + if (ret == 0) + ret = be_stream_wait_io(evtchnl); + + mutex_unlock(&evtchnl->u.req.req_io_lock); + return ret; +} + +int xen_snd_front_stream_read(struct xen_snd_front_evtchnl *evtchnl, + unsigned long pos, unsigned long count) +{ + struct xensnd_req *req; + int ret; + + mutex_lock(&evtchnl->u.req.req_io_lock); + + mutex_lock(&evtchnl->ring_io_lock); + req = be_stream_prepare_req(evtchnl, XENSND_OP_READ); + req->op.rw.length = count; + req->op.rw.offset = pos; + mutex_unlock(&evtchnl->ring_io_lock); + + ret = be_stream_do_io(evtchnl); + + if (ret == 0) + ret = be_stream_wait_io(evtchnl); + + mutex_unlock(&evtchnl->u.req.req_io_lock); + return ret; +} + +int xen_snd_front_stream_trigger(struct xen_snd_front_evtchnl *evtchnl, + int type) +{ + struct xensnd_req *req; + int ret; + + mutex_lock(&evtchnl->u.req.req_io_lock); + + mutex_lock(&evtchnl->ring_io_lock); + req = be_stream_prepare_req(evtchnl, XENSND_OP_TRIGGER); + req->op.trigger.type = type; + mutex_unlock(&evtchnl->ring_io_lock); + + ret = be_stream_do_io(evtchnl); + + if (ret == 0) + ret = be_stream_wait_io(evtchnl); + + mutex_unlock(&evtchnl->u.req.req_io_lock); + return ret; +} + +static void xen_snd_drv_fini(struct xen_snd_front_info *front_info) +{ + xen_snd_front_alsa_fini(front_info); + xen_snd_front_evtchnl_free_all(front_info); +} + +static int sndback_initwait(struct xen_snd_front_info *front_info) +{ + int num_streams; + int ret; + + ret = xen_snd_front_cfg_card(front_info, &num_streams); + if (ret < 0) + return ret; + + /* create event channels for all streams and publish */ + ret = xen_snd_front_evtchnl_create_all(front_info, num_streams); + if (ret < 0) + return ret; + + return xen_snd_front_evtchnl_publish_all(front_info); +} + +static int sndback_connect(struct xen_snd_front_info *front_info) +{ + return xen_snd_front_alsa_init(front_info); +} + +static void sndback_disconnect(struct xen_snd_front_info *front_info) +{ + xen_snd_drv_fini(front_info); + xenbus_switch_state(front_info->xb_dev, XenbusStateInitialising); +} + +static void sndback_changed(struct xenbus_device *xb_dev, + enum xenbus_state backend_state) +{ + struct xen_snd_front_info *front_info = dev_get_drvdata(&xb_dev->dev); + int ret; + + dev_dbg(&xb_dev->dev, "Backend state is %s, front is %s\n", + xenbus_strstate(backend_state), + xenbus_strstate(xb_dev->state)); + + switch (backend_state) { + case XenbusStateReconfiguring: + /* fall through */ + case XenbusStateReconfigured: + /* fall through */ + case XenbusStateInitialised: + /* fall through */ + break; + + case XenbusStateInitialising: + /* Recovering after backend unexpected closure. */ + sndback_disconnect(front_info); + break; + + case XenbusStateInitWait: + /* Recovering after backend unexpected closure. */ + sndback_disconnect(front_info); + + ret = sndback_initwait(front_info); + if (ret < 0) + xenbus_dev_fatal(xb_dev, ret, "initializing frontend"); + else + xenbus_switch_state(xb_dev, XenbusStateInitialised); + break; + + case XenbusStateConnected: + if (xb_dev->state != XenbusStateInitialised) + break; + + ret = sndback_connect(front_info); + if (ret < 0) + xenbus_dev_fatal(xb_dev, ret, "initializing frontend"); + else + xenbus_switch_state(xb_dev, XenbusStateConnected); + break; + + case XenbusStateClosing: + /* + * In this state backend starts freeing resources, + * so let it go into closed state first, so we can also + * remove ours. + */ + break; + + case XenbusStateUnknown: + /* fall through */ + case XenbusStateClosed: + if (xb_dev->state == XenbusStateClosed) + break; + + sndback_disconnect(front_info); + break; + } +} + +static int xen_drv_probe(struct xenbus_device *xb_dev, + const struct xenbus_device_id *id) +{ + struct xen_snd_front_info *front_info; + + front_info = devm_kzalloc(&xb_dev->dev, + sizeof(*front_info), GFP_KERNEL); + if (!front_info) + return -ENOMEM; + + front_info->xb_dev = xb_dev; + dev_set_drvdata(&xb_dev->dev, front_info); + + return xenbus_switch_state(xb_dev, XenbusStateInitialising); +} + +static int xen_drv_remove(struct xenbus_device *dev) +{ + struct xen_snd_front_info *front_info = dev_get_drvdata(&dev->dev); + int to = 100; + + xenbus_switch_state(dev, XenbusStateClosing); + + /* + * On driver removal it is disconnected from XenBus, + * so no backend state change events come via .otherend_changed + * callback. This prevents us from exiting gracefully, e.g. + * signaling the backend to free event channels, waiting for its + * state to change to XenbusStateClosed and cleaning at our end. + * Normally when front driver removed backend will finally go into + * XenbusStateInitWait state. + * + * Workaround: read backend's state manually and wait with time-out. + */ + while ((xenbus_read_unsigned(front_info->xb_dev->otherend, "state", + XenbusStateUnknown) != XenbusStateInitWait) && + --to) + msleep(10); + + if (!to) { + unsigned int state; + + state = xenbus_read_unsigned(front_info->xb_dev->otherend, + "state", XenbusStateUnknown); + pr_err("Backend state is %s while removing driver\n", + xenbus_strstate(state)); + } + + xen_snd_drv_fini(front_info); + xenbus_frontend_closed(dev); + return 0; +} + +static const struct xenbus_device_id xen_drv_ids[] = { + { XENSND_DRIVER_NAME }, + { "" } +}; + +static struct xenbus_driver xen_driver = { + .ids = xen_drv_ids, + .probe = xen_drv_probe, + .remove = xen_drv_remove, + .otherend_changed = sndback_changed, +}; + +static int __init xen_drv_init(void) +{ + if (!xen_domain()) + return -ENODEV; + + if (!xen_has_pv_devices()) + return -ENODEV; + + /* At the moment we only support case with XEN_PAGE_SIZE == PAGE_SIZE */ + if (XEN_PAGE_SIZE != PAGE_SIZE) { + pr_err(XENSND_DRIVER_NAME ": different kernel and Xen page sizes are not supported: XEN_PAGE_SIZE (%lu) != PAGE_SIZE (%lu)\n", + XEN_PAGE_SIZE, PAGE_SIZE); + return -ENODEV; + } + + pr_info("Initialising Xen " XENSND_DRIVER_NAME " frontend driver\n"); + return xenbus_register_frontend(&xen_driver); +} + +static void __exit xen_drv_fini(void) +{ + pr_info("Unregistering Xen " XENSND_DRIVER_NAME " frontend driver\n"); + xenbus_unregister_driver(&xen_driver); +} + +module_init(xen_drv_init); +module_exit(xen_drv_fini); + +MODULE_DESCRIPTION("Xen virtual sound device frontend"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("xen:" XENSND_DRIVER_NAME); +MODULE_SUPPORTED_DEVICE("{{ALSA,Virtual soundcard}}"); diff --git a/sound/xen/xen_snd_front.h b/sound/xen/xen_snd_front.h new file mode 100644 index 000000000000..a2ea2463bcc5 --- /dev/null +++ b/sound/xen/xen_snd_front.h @@ -0,0 +1,54 @@ +/* SPDX-License-Identifier: GPL-2.0 OR MIT */ + +/* + * Xen para-virtual sound device + * + * Copyright (C) 2016-2018 EPAM Systems Inc. + * + * Author: Oleksandr Andrushchenko <oleksandr_andrushchenko@epam.com> + */ + +#ifndef __XEN_SND_FRONT_H +#define __XEN_SND_FRONT_H + +#include "xen_snd_front_cfg.h" + +struct xen_snd_front_card_info; +struct xen_snd_front_evtchnl; +struct xen_snd_front_evtchnl_pair; +struct xen_snd_front_shbuf; +struct xensnd_query_hw_param; + +struct xen_snd_front_info { + struct xenbus_device *xb_dev; + + struct xen_snd_front_card_info *card_info; + + int num_evt_pairs; + struct xen_snd_front_evtchnl_pair *evt_pairs; + + struct xen_front_cfg_card cfg; +}; + +int xen_snd_front_stream_query_hw_param(struct xen_snd_front_evtchnl *evtchnl, + struct xensnd_query_hw_param *hw_param_req, + struct xensnd_query_hw_param *hw_param_resp); + +int xen_snd_front_stream_prepare(struct xen_snd_front_evtchnl *evtchnl, + struct xen_snd_front_shbuf *sh_buf, + u8 format, unsigned int channels, + unsigned int rate, u32 buffer_sz, + u32 period_sz); + +int xen_snd_front_stream_close(struct xen_snd_front_evtchnl *evtchnl); + +int xen_snd_front_stream_write(struct xen_snd_front_evtchnl *evtchnl, + unsigned long pos, unsigned long count); + +int xen_snd_front_stream_read(struct xen_snd_front_evtchnl *evtchnl, + unsigned long pos, unsigned long count); + +int xen_snd_front_stream_trigger(struct xen_snd_front_evtchnl *evtchnl, + int type); + +#endif /* __XEN_SND_FRONT_H */ diff --git a/sound/xen/xen_snd_front_alsa.c b/sound/xen/xen_snd_front_alsa.c new file mode 100644 index 000000000000..5a2bd70a2fa1 --- /dev/null +++ b/sound/xen/xen_snd_front_alsa.c @@ -0,0 +1,822 @@ +// SPDX-License-Identifier: GPL-2.0 OR MIT + +/* + * Xen para-virtual sound device + * + * Copyright (C) 2016-2018 EPAM Systems Inc. + * + * Author: Oleksandr Andrushchenko <oleksandr_andrushchenko@epam.com> + */ + +#include <linux/platform_device.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> + +#include <xen/xenbus.h> + +#include "xen_snd_front.h" +#include "xen_snd_front_alsa.h" +#include "xen_snd_front_cfg.h" +#include "xen_snd_front_evtchnl.h" +#include "xen_snd_front_shbuf.h" + +struct xen_snd_front_pcm_stream_info { + struct xen_snd_front_info *front_info; + struct xen_snd_front_evtchnl_pair *evt_pair; + struct xen_snd_front_shbuf sh_buf; + int index; + + bool is_open; + struct snd_pcm_hardware pcm_hw; + + /* Number of processed frames as reported by the backend. */ + snd_pcm_uframes_t be_cur_frame; + /* Current HW pointer to be reported via .period callback. */ + atomic_t hw_ptr; + /* Modulo of the number of processed frames - for period detection. */ + u32 out_frames; +}; + +struct xen_snd_front_pcm_instance_info { + struct xen_snd_front_card_info *card_info; + struct snd_pcm *pcm; + struct snd_pcm_hardware pcm_hw; + int num_pcm_streams_pb; + struct xen_snd_front_pcm_stream_info *streams_pb; + int num_pcm_streams_cap; + struct xen_snd_front_pcm_stream_info *streams_cap; +}; + +struct xen_snd_front_card_info { + struct xen_snd_front_info *front_info; + struct snd_card *card; + struct snd_pcm_hardware pcm_hw; + int num_pcm_instances; + struct xen_snd_front_pcm_instance_info *pcm_instances; +}; + +struct alsa_sndif_sample_format { + u8 sndif; + snd_pcm_format_t alsa; +}; + +struct alsa_sndif_hw_param { + u8 sndif; + snd_pcm_hw_param_t alsa; +}; + +static const struct alsa_sndif_sample_format ALSA_SNDIF_FORMATS[] = { + { + .sndif = XENSND_PCM_FORMAT_U8, + .alsa = SNDRV_PCM_FORMAT_U8 + }, + { + .sndif = XENSND_PCM_FORMAT_S8, + .alsa = SNDRV_PCM_FORMAT_S8 + }, + { + .sndif = XENSND_PCM_FORMAT_U16_LE, + .alsa = SNDRV_PCM_FORMAT_U16_LE + }, + { + .sndif = XENSND_PCM_FORMAT_U16_BE, + .alsa = SNDRV_PCM_FORMAT_U16_BE + }, + { + .sndif = XENSND_PCM_FORMAT_S16_LE, + .alsa = SNDRV_PCM_FORMAT_S16_LE + }, + { + .sndif = XENSND_PCM_FORMAT_S16_BE, + .alsa = SNDRV_PCM_FORMAT_S16_BE + }, + { + .sndif = XENSND_PCM_FORMAT_U24_LE, + .alsa = SNDRV_PCM_FORMAT_U24_LE + }, + { + .sndif = XENSND_PCM_FORMAT_U24_BE, + .alsa = SNDRV_PCM_FORMAT_U24_BE + }, + { + .sndif = XENSND_PCM_FORMAT_S24_LE, + .alsa = SNDRV_PCM_FORMAT_S24_LE + }, + { + .sndif = XENSND_PCM_FORMAT_S24_BE, + .alsa = SNDRV_PCM_FORMAT_S24_BE + }, + { + .sndif = XENSND_PCM_FORMAT_U32_LE, + .alsa = SNDRV_PCM_FORMAT_U32_LE + }, + { + .sndif = XENSND_PCM_FORMAT_U32_BE, + .alsa = SNDRV_PCM_FORMAT_U32_BE + }, + { + .sndif = XENSND_PCM_FORMAT_S32_LE, + .alsa = SNDRV_PCM_FORMAT_S32_LE + }, + { + .sndif = XENSND_PCM_FORMAT_S32_BE, + .alsa = SNDRV_PCM_FORMAT_S32_BE + }, + { + .sndif = XENSND_PCM_FORMAT_A_LAW, + .alsa = SNDRV_PCM_FORMAT_A_LAW + }, + { + .sndif = XENSND_PCM_FORMAT_MU_LAW, + .alsa = SNDRV_PCM_FORMAT_MU_LAW + }, + { + .sndif = XENSND_PCM_FORMAT_F32_LE, + .alsa = SNDRV_PCM_FORMAT_FLOAT_LE + }, + { + .sndif = XENSND_PCM_FORMAT_F32_BE, + .alsa = SNDRV_PCM_FORMAT_FLOAT_BE + }, + { + .sndif = XENSND_PCM_FORMAT_F64_LE, + .alsa = SNDRV_PCM_FORMAT_FLOAT64_LE + }, + { + .sndif = XENSND_PCM_FORMAT_F64_BE, + .alsa = SNDRV_PCM_FORMAT_FLOAT64_BE + }, + { + .sndif = XENSND_PCM_FORMAT_IEC958_SUBFRAME_LE, + .alsa = SNDRV_PCM_FORMAT_IEC958_SUBFRAME_LE + }, + { + .sndif = XENSND_PCM_FORMAT_IEC958_SUBFRAME_BE, + .alsa = SNDRV_PCM_FORMAT_IEC958_SUBFRAME_BE + }, + { + .sndif = XENSND_PCM_FORMAT_IMA_ADPCM, + .alsa = SNDRV_PCM_FORMAT_IMA_ADPCM + }, + { + .sndif = XENSND_PCM_FORMAT_MPEG, + .alsa = SNDRV_PCM_FORMAT_MPEG + }, + { + .sndif = XENSND_PCM_FORMAT_GSM, + .alsa = SNDRV_PCM_FORMAT_GSM + }, +}; + +static int to_sndif_format(snd_pcm_format_t format) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(ALSA_SNDIF_FORMATS); i++) + if (ALSA_SNDIF_FORMATS[i].alsa == format) + return ALSA_SNDIF_FORMATS[i].sndif; + + return -EINVAL; +} + +static u64 to_sndif_formats_mask(u64 alsa_formats) +{ + u64 mask; + int i; + + mask = 0; + for (i = 0; i < ARRAY_SIZE(ALSA_SNDIF_FORMATS); i++) + if (1 << ALSA_SNDIF_FORMATS[i].alsa & alsa_formats) + mask |= 1 << ALSA_SNDIF_FORMATS[i].sndif; + + return mask; +} + +static u64 to_alsa_formats_mask(u64 sndif_formats) +{ + u64 mask; + int i; + + mask = 0; + for (i = 0; i < ARRAY_SIZE(ALSA_SNDIF_FORMATS); i++) + if (1 << ALSA_SNDIF_FORMATS[i].sndif & sndif_formats) + mask |= 1 << ALSA_SNDIF_FORMATS[i].alsa; + + return mask; +} + +static void stream_clear(struct xen_snd_front_pcm_stream_info *stream) +{ + stream->is_open = false; + stream->be_cur_frame = 0; + stream->out_frames = 0; + atomic_set(&stream->hw_ptr, 0); + xen_snd_front_evtchnl_pair_clear(stream->evt_pair); + xen_snd_front_shbuf_clear(&stream->sh_buf); +} + +static void stream_free(struct xen_snd_front_pcm_stream_info *stream) +{ + xen_snd_front_shbuf_free(&stream->sh_buf); + stream_clear(stream); +} + +static struct xen_snd_front_pcm_stream_info * +stream_get(struct snd_pcm_substream *substream) +{ + struct xen_snd_front_pcm_instance_info *pcm_instance = + snd_pcm_substream_chip(substream); + struct xen_snd_front_pcm_stream_info *stream; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + stream = &pcm_instance->streams_pb[substream->number]; + else + stream = &pcm_instance->streams_cap[substream->number]; + + return stream; +} + +static int alsa_hw_rule(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct xen_snd_front_pcm_stream_info *stream = rule->private; + struct device *dev = &stream->front_info->xb_dev->dev; + struct snd_mask *formats = + hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + struct snd_interval *rates = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_interval *period = + hw_param_interval(params, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE); + struct snd_interval *buffer = + hw_param_interval(params, + SNDRV_PCM_HW_PARAM_BUFFER_SIZE); + struct xensnd_query_hw_param req; + struct xensnd_query_hw_param resp; + struct snd_interval interval; + struct snd_mask mask; + u64 sndif_formats; + int changed, ret; + + /* Collect all the values we need for the query. */ + + req.formats = to_sndif_formats_mask((u64)formats->bits[0] | + (u64)(formats->bits[1]) << 32); + + req.rates.min = rates->min; + req.rates.max = rates->max; + + req.channels.min = channels->min; + req.channels.max = channels->max; + + req.buffer.min = buffer->min; + req.buffer.max = buffer->max; + + req.period.min = period->min; + req.period.max = period->max; + + ret = xen_snd_front_stream_query_hw_param(&stream->evt_pair->req, + &req, &resp); + if (ret < 0) { + /* Check if this is due to backend communication error. */ + if (ret == -EIO || ret == -ETIMEDOUT) + dev_err(dev, "Failed to query ALSA HW parameters\n"); + return ret; + } + + /* Refine HW parameters after the query. */ + changed = 0; + + sndif_formats = to_alsa_formats_mask(resp.formats); + snd_mask_none(&mask); + mask.bits[0] = (u32)sndif_formats; + mask.bits[1] = (u32)(sndif_formats >> 32); + ret = snd_mask_refine(formats, &mask); + if (ret < 0) + return ret; + changed |= ret; + + interval.openmin = 0; + interval.openmax = 0; + interval.integer = 1; + + interval.min = resp.rates.min; + interval.max = resp.rates.max; + ret = snd_interval_refine(rates, &interval); + if (ret < 0) + return ret; + changed |= ret; + + interval.min = resp.channels.min; + interval.max = resp.channels.max; + ret = snd_interval_refine(channels, &interval); + if (ret < 0) + return ret; + changed |= ret; + + interval.min = resp.buffer.min; + interval.max = resp.buffer.max; + ret = snd_interval_refine(buffer, &interval); + if (ret < 0) + return ret; + changed |= ret; + + interval.min = resp.period.min; + interval.max = resp.period.max; + ret = snd_interval_refine(period, &interval); + if (ret < 0) + return ret; + changed |= ret; + + return changed; +} + +static int alsa_open(struct snd_pcm_substream *substream) +{ + struct xen_snd_front_pcm_instance_info *pcm_instance = + snd_pcm_substream_chip(substream); + struct xen_snd_front_pcm_stream_info *stream = stream_get(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct xen_snd_front_info *front_info = + pcm_instance->card_info->front_info; + struct device *dev = &front_info->xb_dev->dev; + int ret; + + /* + * Return our HW properties: override defaults with those configured + * via XenStore. + */ + runtime->hw = stream->pcm_hw; + runtime->hw.info &= ~(SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_DOUBLE | + SNDRV_PCM_INFO_BATCH | + SNDRV_PCM_INFO_NONINTERLEAVED | + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_PAUSE); + runtime->hw.info |= SNDRV_PCM_INFO_INTERLEAVED; + + stream->evt_pair = &front_info->evt_pairs[stream->index]; + + stream->front_info = front_info; + + stream->evt_pair->evt.u.evt.substream = substream; + + stream_clear(stream); + + xen_snd_front_evtchnl_pair_set_connected(stream->evt_pair, true); + + ret = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, + alsa_hw_rule, stream, + SNDRV_PCM_HW_PARAM_FORMAT, -1); + if (ret) { + dev_err(dev, "Failed to add HW rule for SNDRV_PCM_HW_PARAM_FORMAT\n"); + return ret; + } + + ret = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + alsa_hw_rule, stream, + SNDRV_PCM_HW_PARAM_RATE, -1); + if (ret) { + dev_err(dev, "Failed to add HW rule for SNDRV_PCM_HW_PARAM_RATE\n"); + return ret; + } + + ret = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + alsa_hw_rule, stream, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); + if (ret) { + dev_err(dev, "Failed to add HW rule for SNDRV_PCM_HW_PARAM_CHANNELS\n"); + return ret; + } + + ret = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + alsa_hw_rule, stream, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, -1); + if (ret) { + dev_err(dev, "Failed to add HW rule for SNDRV_PCM_HW_PARAM_PERIOD_SIZE\n"); + return ret; + } + + ret = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + alsa_hw_rule, stream, + SNDRV_PCM_HW_PARAM_BUFFER_SIZE, -1); + if (ret) { + dev_err(dev, "Failed to add HW rule for SNDRV_PCM_HW_PARAM_BUFFER_SIZE\n"); + return ret; + } + + return 0; +} + +static int alsa_close(struct snd_pcm_substream *substream) +{ + struct xen_snd_front_pcm_stream_info *stream = stream_get(substream); + + xen_snd_front_evtchnl_pair_set_connected(stream->evt_pair, false); + return 0; +} + +static int alsa_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct xen_snd_front_pcm_stream_info *stream = stream_get(substream); + int ret; + + /* + * This callback may be called multiple times, + * so free the previously allocated shared buffer if any. + */ + stream_free(stream); + + ret = xen_snd_front_shbuf_alloc(stream->front_info->xb_dev, + &stream->sh_buf, + params_buffer_bytes(params)); + if (ret < 0) { + stream_free(stream); + dev_err(&stream->front_info->xb_dev->dev, + "Failed to allocate buffers for stream with index %d\n", + stream->index); + return ret; + } + + return 0; +} + +static int alsa_hw_free(struct snd_pcm_substream *substream) +{ + struct xen_snd_front_pcm_stream_info *stream = stream_get(substream); + int ret; + + ret = xen_snd_front_stream_close(&stream->evt_pair->req); + stream_free(stream); + return ret; +} + +static int alsa_prepare(struct snd_pcm_substream *substream) +{ + struct xen_snd_front_pcm_stream_info *stream = stream_get(substream); + + if (!stream->is_open) { + struct snd_pcm_runtime *runtime = substream->runtime; + u8 sndif_format; + int ret; + + ret = to_sndif_format(runtime->format); + if (ret < 0) { + dev_err(&stream->front_info->xb_dev->dev, + "Unsupported sample format: %d\n", + runtime->format); + return ret; + } + sndif_format = ret; + + ret = xen_snd_front_stream_prepare(&stream->evt_pair->req, + &stream->sh_buf, + sndif_format, + runtime->channels, + runtime->rate, + snd_pcm_lib_buffer_bytes(substream), + snd_pcm_lib_period_bytes(substream)); + if (ret < 0) + return ret; + + stream->is_open = true; + } + + return 0; +} + +static int alsa_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct xen_snd_front_pcm_stream_info *stream = stream_get(substream); + int type; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + type = XENSND_OP_TRIGGER_START; + break; + + case SNDRV_PCM_TRIGGER_RESUME: + type = XENSND_OP_TRIGGER_RESUME; + break; + + case SNDRV_PCM_TRIGGER_STOP: + type = XENSND_OP_TRIGGER_STOP; + break; + + case SNDRV_PCM_TRIGGER_SUSPEND: + type = XENSND_OP_TRIGGER_PAUSE; + break; + + default: + return -EINVAL; + } + + return xen_snd_front_stream_trigger(&stream->evt_pair->req, type); +} + +void xen_snd_front_alsa_handle_cur_pos(struct xen_snd_front_evtchnl *evtchnl, + u64 pos_bytes) +{ + struct snd_pcm_substream *substream = evtchnl->u.evt.substream; + struct xen_snd_front_pcm_stream_info *stream = stream_get(substream); + snd_pcm_uframes_t delta, new_hw_ptr, cur_frame; + + cur_frame = bytes_to_frames(substream->runtime, pos_bytes); + + delta = cur_frame - stream->be_cur_frame; + stream->be_cur_frame = cur_frame; + + new_hw_ptr = (snd_pcm_uframes_t)atomic_read(&stream->hw_ptr); + new_hw_ptr = (new_hw_ptr + delta) % substream->runtime->buffer_size; + atomic_set(&stream->hw_ptr, (int)new_hw_ptr); + + stream->out_frames += delta; + if (stream->out_frames > substream->runtime->period_size) { + stream->out_frames %= substream->runtime->period_size; + snd_pcm_period_elapsed(substream); + } +} + +static snd_pcm_uframes_t alsa_pointer(struct snd_pcm_substream *substream) +{ + struct xen_snd_front_pcm_stream_info *stream = stream_get(substream); + + return (snd_pcm_uframes_t)atomic_read(&stream->hw_ptr); +} + +static int alsa_pb_copy_user(struct snd_pcm_substream *substream, + int channel, unsigned long pos, void __user *src, + unsigned long count) +{ + struct xen_snd_front_pcm_stream_info *stream = stream_get(substream); + + if (unlikely(pos + count > stream->sh_buf.buffer_sz)) + return -EINVAL; + + if (copy_from_user(stream->sh_buf.buffer + pos, src, count)) + return -EFAULT; + + return xen_snd_front_stream_write(&stream->evt_pair->req, pos, count); +} + +static int alsa_pb_copy_kernel(struct snd_pcm_substream *substream, + int channel, unsigned long pos, void *src, + unsigned long count) +{ + struct xen_snd_front_pcm_stream_info *stream = stream_get(substream); + + if (unlikely(pos + count > stream->sh_buf.buffer_sz)) + return -EINVAL; + + memcpy(stream->sh_buf.buffer + pos, src, count); + + return xen_snd_front_stream_write(&stream->evt_pair->req, pos, count); +} + +static int alsa_cap_copy_user(struct snd_pcm_substream *substream, + int channel, unsigned long pos, void __user *dst, + unsigned long count) +{ + struct xen_snd_front_pcm_stream_info *stream = stream_get(substream); + int ret; + + if (unlikely(pos + count > stream->sh_buf.buffer_sz)) + return -EINVAL; + + ret = xen_snd_front_stream_read(&stream->evt_pair->req, pos, count); + if (ret < 0) + return ret; + + return copy_to_user(dst, stream->sh_buf.buffer + pos, count) ? + -EFAULT : 0; +} + +static int alsa_cap_copy_kernel(struct snd_pcm_substream *substream, + int channel, unsigned long pos, void *dst, + unsigned long count) +{ + struct xen_snd_front_pcm_stream_info *stream = stream_get(substream); + int ret; + + if (unlikely(pos + count > stream->sh_buf.buffer_sz)) + return -EINVAL; + + ret = xen_snd_front_stream_read(&stream->evt_pair->req, pos, count); + if (ret < 0) + return ret; + + memcpy(dst, stream->sh_buf.buffer + pos, count); + + return 0; +} + +static int alsa_pb_fill_silence(struct snd_pcm_substream *substream, + int channel, unsigned long pos, + unsigned long count) +{ + struct xen_snd_front_pcm_stream_info *stream = stream_get(substream); + + if (unlikely(pos + count > stream->sh_buf.buffer_sz)) + return -EINVAL; + + memset(stream->sh_buf.buffer + pos, 0, count); + + return xen_snd_front_stream_write(&stream->evt_pair->req, pos, count); +} + +/* + * FIXME: The mmaped data transfer is asynchronous and there is no + * ack signal from user-space when it is done. This is the + * reason it is not implemented in the PV driver as we do need + * to know when the buffer can be transferred to the backend. + */ + +static struct snd_pcm_ops snd_drv_alsa_playback_ops = { + .open = alsa_open, + .close = alsa_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = alsa_hw_params, + .hw_free = alsa_hw_free, + .prepare = alsa_prepare, + .trigger = alsa_trigger, + .pointer = alsa_pointer, + .copy_user = alsa_pb_copy_user, + .copy_kernel = alsa_pb_copy_kernel, + .fill_silence = alsa_pb_fill_silence, +}; + +static struct snd_pcm_ops snd_drv_alsa_capture_ops = { + .open = alsa_open, + .close = alsa_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = alsa_hw_params, + .hw_free = alsa_hw_free, + .prepare = alsa_prepare, + .trigger = alsa_trigger, + .pointer = alsa_pointer, + .copy_user = alsa_cap_copy_user, + .copy_kernel = alsa_cap_copy_kernel, +}; + +static int new_pcm_instance(struct xen_snd_front_card_info *card_info, + struct xen_front_cfg_pcm_instance *instance_cfg, + struct xen_snd_front_pcm_instance_info *pcm_instance_info) +{ + struct snd_pcm *pcm; + int ret, i; + + dev_dbg(&card_info->front_info->xb_dev->dev, + "New PCM device \"%s\" with id %d playback %d capture %d", + instance_cfg->name, + instance_cfg->device_id, + instance_cfg->num_streams_pb, + instance_cfg->num_streams_cap); + + pcm_instance_info->card_info = card_info; + + pcm_instance_info->pcm_hw = instance_cfg->pcm_hw; + + if (instance_cfg->num_streams_pb) { + pcm_instance_info->streams_pb = + devm_kcalloc(&card_info->card->card_dev, + instance_cfg->num_streams_pb, + sizeof(struct xen_snd_front_pcm_stream_info), + GFP_KERNEL); + if (!pcm_instance_info->streams_pb) + return -ENOMEM; + } + + if (instance_cfg->num_streams_cap) { + pcm_instance_info->streams_cap = + devm_kcalloc(&card_info->card->card_dev, + instance_cfg->num_streams_cap, + sizeof(struct xen_snd_front_pcm_stream_info), + GFP_KERNEL); + if (!pcm_instance_info->streams_cap) + return -ENOMEM; + } + + pcm_instance_info->num_pcm_streams_pb = + instance_cfg->num_streams_pb; + pcm_instance_info->num_pcm_streams_cap = + instance_cfg->num_streams_cap; + + for (i = 0; i < pcm_instance_info->num_pcm_streams_pb; i++) { + pcm_instance_info->streams_pb[i].pcm_hw = + instance_cfg->streams_pb[i].pcm_hw; + pcm_instance_info->streams_pb[i].index = + instance_cfg->streams_pb[i].index; + } + + for (i = 0; i < pcm_instance_info->num_pcm_streams_cap; i++) { + pcm_instance_info->streams_cap[i].pcm_hw = + instance_cfg->streams_cap[i].pcm_hw; + pcm_instance_info->streams_cap[i].index = + instance_cfg->streams_cap[i].index; + } + + ret = snd_pcm_new(card_info->card, instance_cfg->name, + instance_cfg->device_id, + instance_cfg->num_streams_pb, + instance_cfg->num_streams_cap, + &pcm); + if (ret < 0) + return ret; + + pcm->private_data = pcm_instance_info; + pcm->info_flags = 0; + /* we want to handle all PCM operations in non-atomic context */ + pcm->nonatomic = true; + strncpy(pcm->name, "Virtual card PCM", sizeof(pcm->name)); + + if (instance_cfg->num_streams_pb) + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &snd_drv_alsa_playback_ops); + + if (instance_cfg->num_streams_cap) + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, + &snd_drv_alsa_capture_ops); + + pcm_instance_info->pcm = pcm; + return 0; +} + +int xen_snd_front_alsa_init(struct xen_snd_front_info *front_info) +{ + struct device *dev = &front_info->xb_dev->dev; + struct xen_front_cfg_card *cfg = &front_info->cfg; + struct xen_snd_front_card_info *card_info; + struct snd_card *card; + int ret, i; + + dev_dbg(dev, "Creating virtual sound card\n"); + + ret = snd_card_new(dev, 0, XENSND_DRIVER_NAME, THIS_MODULE, + sizeof(struct xen_snd_front_card_info), &card); + if (ret < 0) + return ret; + + card_info = card->private_data; + card_info->front_info = front_info; + front_info->card_info = card_info; + card_info->card = card; + card_info->pcm_instances = + devm_kcalloc(dev, cfg->num_pcm_instances, + sizeof(struct xen_snd_front_pcm_instance_info), + GFP_KERNEL); + if (!card_info->pcm_instances) { + ret = -ENOMEM; + goto fail; + } + + card_info->num_pcm_instances = cfg->num_pcm_instances; + card_info->pcm_hw = cfg->pcm_hw; + + for (i = 0; i < cfg->num_pcm_instances; i++) { + ret = new_pcm_instance(card_info, &cfg->pcm_instances[i], + &card_info->pcm_instances[i]); + if (ret < 0) + goto fail; + } + + strncpy(card->driver, XENSND_DRIVER_NAME, sizeof(card->driver)); + strncpy(card->shortname, cfg->name_short, sizeof(card->shortname)); + strncpy(card->longname, cfg->name_long, sizeof(card->longname)); + + ret = snd_card_register(card); + if (ret < 0) + goto fail; + + return 0; + +fail: + snd_card_free(card); + return ret; +} + +void xen_snd_front_alsa_fini(struct xen_snd_front_info *front_info) +{ + struct xen_snd_front_card_info *card_info; + struct snd_card *card; + + card_info = front_info->card_info; + if (!card_info) + return; + + card = card_info->card; + if (!card) + return; + + dev_dbg(&front_info->xb_dev->dev, "Removing virtual sound card %d\n", + card->number); + snd_card_free(card); + + /* Card_info will be freed when destroying front_info->xb_dev->dev. */ + card_info->card = NULL; +} diff --git a/sound/xen/xen_snd_front_alsa.h b/sound/xen/xen_snd_front_alsa.h new file mode 100644 index 000000000000..18abd9eec967 --- /dev/null +++ b/sound/xen/xen_snd_front_alsa.h @@ -0,0 +1,23 @@ +/* SPDX-License-Identifier: GPL-2.0 OR MIT */ + +/* + * Xen para-virtual sound device + * + * Copyright (C) 2016-2018 EPAM Systems Inc. + * + * Author: Oleksandr Andrushchenko <oleksandr_andrushchenko@epam.com> + */ + +#ifndef __XEN_SND_FRONT_ALSA_H +#define __XEN_SND_FRONT_ALSA_H + +struct xen_snd_front_info; + +int xen_snd_front_alsa_init(struct xen_snd_front_info *front_info); + +void xen_snd_front_alsa_fini(struct xen_snd_front_info *front_info); + +void xen_snd_front_alsa_handle_cur_pos(struct xen_snd_front_evtchnl *evtchnl, + u64 pos_bytes); + +#endif /* __XEN_SND_FRONT_ALSA_H */ diff --git a/sound/xen/xen_snd_front_cfg.c b/sound/xen/xen_snd_front_cfg.c new file mode 100644 index 000000000000..eda077c8087a --- /dev/null +++ b/sound/xen/xen_snd_front_cfg.c @@ -0,0 +1,519 @@ +// SPDX-License-Identifier: GPL-2.0 OR MIT + +/* + * Xen para-virtual sound device + * + * Copyright (C) 2016-2018 EPAM Systems Inc. + * + * Author: Oleksandr Andrushchenko <oleksandr_andrushchenko@epam.com> + */ + +#include <xen/xenbus.h> + +#include <xen/interface/io/sndif.h> + +#include "xen_snd_front.h" +#include "xen_snd_front_cfg.h" + +/* Maximum number of supported streams. */ +#define VSND_MAX_STREAM 8 + +struct cfg_hw_sample_rate { + const char *name; + unsigned int mask; + unsigned int value; +}; + +static const struct cfg_hw_sample_rate CFG_HW_SUPPORTED_RATES[] = { + { .name = "5512", .mask = SNDRV_PCM_RATE_5512, .value = 5512 }, + { .name = "8000", .mask = SNDRV_PCM_RATE_8000, .value = 8000 }, + { .name = "11025", .mask = SNDRV_PCM_RATE_11025, .value = 11025 }, + { .name = "16000", .mask = SNDRV_PCM_RATE_16000, .value = 16000 }, + { .name = "22050", .mask = SNDRV_PCM_RATE_22050, .value = 22050 }, + { .name = "32000", .mask = SNDRV_PCM_RATE_32000, .value = 32000 }, + { .name = "44100", .mask = SNDRV_PCM_RATE_44100, .value = 44100 }, + { .name = "48000", .mask = SNDRV_PCM_RATE_48000, .value = 48000 }, + { .name = "64000", .mask = SNDRV_PCM_RATE_64000, .value = 64000 }, + { .name = "96000", .mask = SNDRV_PCM_RATE_96000, .value = 96000 }, + { .name = "176400", .mask = SNDRV_PCM_RATE_176400, .value = 176400 }, + { .name = "192000", .mask = SNDRV_PCM_RATE_192000, .value = 192000 }, +}; + +struct cfg_hw_sample_format { + const char *name; + u64 mask; +}; + +static const struct cfg_hw_sample_format CFG_HW_SUPPORTED_FORMATS[] = { + { + .name = XENSND_PCM_FORMAT_U8_STR, + .mask = SNDRV_PCM_FMTBIT_U8 + }, + { + .name = XENSND_PCM_FORMAT_S8_STR, + .mask = SNDRV_PCM_FMTBIT_S8 + }, + { + .name = XENSND_PCM_FORMAT_U16_LE_STR, + .mask = SNDRV_PCM_FMTBIT_U16_LE + }, + { + .name = XENSND_PCM_FORMAT_U16_BE_STR, + .mask = SNDRV_PCM_FMTBIT_U16_BE + }, + { + .name = XENSND_PCM_FORMAT_S16_LE_STR, + .mask = SNDRV_PCM_FMTBIT_S16_LE + }, + { + .name = XENSND_PCM_FORMAT_S16_BE_STR, + .mask = SNDRV_PCM_FMTBIT_S16_BE + }, + { + .name = XENSND_PCM_FORMAT_U24_LE_STR, + .mask = SNDRV_PCM_FMTBIT_U24_LE + }, + { + .name = XENSND_PCM_FORMAT_U24_BE_STR, + .mask = SNDRV_PCM_FMTBIT_U24_BE + }, + { + .name = XENSND_PCM_FORMAT_S24_LE_STR, + .mask = SNDRV_PCM_FMTBIT_S24_LE + }, + { + .name = XENSND_PCM_FORMAT_S24_BE_STR, + .mask = SNDRV_PCM_FMTBIT_S24_BE + }, + { + .name = XENSND_PCM_FORMAT_U32_LE_STR, + .mask = SNDRV_PCM_FMTBIT_U32_LE + }, + { + .name = XENSND_PCM_FORMAT_U32_BE_STR, + .mask = SNDRV_PCM_FMTBIT_U32_BE + }, + { + .name = XENSND_PCM_FORMAT_S32_LE_STR, + .mask = SNDRV_PCM_FMTBIT_S32_LE + }, + { + .name = XENSND_PCM_FORMAT_S32_BE_STR, + .mask = SNDRV_PCM_FMTBIT_S32_BE + }, + { + .name = XENSND_PCM_FORMAT_A_LAW_STR, + .mask = SNDRV_PCM_FMTBIT_A_LAW + }, + { + .name = XENSND_PCM_FORMAT_MU_LAW_STR, + .mask = SNDRV_PCM_FMTBIT_MU_LAW + }, + { + .name = XENSND_PCM_FORMAT_F32_LE_STR, + .mask = SNDRV_PCM_FMTBIT_FLOAT_LE + }, + { + .name = XENSND_PCM_FORMAT_F32_BE_STR, + .mask = SNDRV_PCM_FMTBIT_FLOAT_BE + }, + { + .name = XENSND_PCM_FORMAT_F64_LE_STR, + .mask = SNDRV_PCM_FMTBIT_FLOAT64_LE + }, + { + .name = XENSND_PCM_FORMAT_F64_BE_STR, + .mask = SNDRV_PCM_FMTBIT_FLOAT64_BE + }, + { + .name = XENSND_PCM_FORMAT_IEC958_SUBFRAME_LE_STR, + .mask = SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE + }, + { + .name = XENSND_PCM_FORMAT_IEC958_SUBFRAME_BE_STR, + .mask = SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE + }, + { + .name = XENSND_PCM_FORMAT_IMA_ADPCM_STR, + .mask = SNDRV_PCM_FMTBIT_IMA_ADPCM + }, + { + .name = XENSND_PCM_FORMAT_MPEG_STR, + .mask = SNDRV_PCM_FMTBIT_MPEG + }, + { + .name = XENSND_PCM_FORMAT_GSM_STR, + .mask = SNDRV_PCM_FMTBIT_GSM + }, +}; + +static void cfg_hw_rates(char *list, unsigned int len, + const char *path, struct snd_pcm_hardware *pcm_hw) +{ + char *cur_rate; + unsigned int cur_mask; + unsigned int cur_value; + unsigned int rates; + unsigned int rate_min; + unsigned int rate_max; + int i; + + rates = 0; + rate_min = -1; + rate_max = 0; + while ((cur_rate = strsep(&list, XENSND_LIST_SEPARATOR))) { + for (i = 0; i < ARRAY_SIZE(CFG_HW_SUPPORTED_RATES); i++) + if (!strncasecmp(cur_rate, + CFG_HW_SUPPORTED_RATES[i].name, + XENSND_SAMPLE_RATE_MAX_LEN)) { + cur_mask = CFG_HW_SUPPORTED_RATES[i].mask; + cur_value = CFG_HW_SUPPORTED_RATES[i].value; + rates |= cur_mask; + if (rate_min > cur_value) + rate_min = cur_value; + if (rate_max < cur_value) + rate_max = cur_value; + } + } + + if (rates) { + pcm_hw->rates = rates; + pcm_hw->rate_min = rate_min; + pcm_hw->rate_max = rate_max; + } +} + +static void cfg_formats(char *list, unsigned int len, + const char *path, struct snd_pcm_hardware *pcm_hw) +{ + u64 formats; + char *cur_format; + int i; + + formats = 0; + while ((cur_format = strsep(&list, XENSND_LIST_SEPARATOR))) { + for (i = 0; i < ARRAY_SIZE(CFG_HW_SUPPORTED_FORMATS); i++) + if (!strncasecmp(cur_format, + CFG_HW_SUPPORTED_FORMATS[i].name, + XENSND_SAMPLE_FORMAT_MAX_LEN)) + formats |= CFG_HW_SUPPORTED_FORMATS[i].mask; + } + + if (formats) + pcm_hw->formats = formats; +} + +#define MAX_BUFFER_SIZE (64 * 1024) +#define MIN_PERIOD_SIZE 64 +#define MAX_PERIOD_SIZE MAX_BUFFER_SIZE +#define USE_FORMATS (SNDRV_PCM_FMTBIT_U8 | \ + SNDRV_PCM_FMTBIT_S16_LE) +#define USE_RATE (SNDRV_PCM_RATE_CONTINUOUS | \ + SNDRV_PCM_RATE_8000_48000) +#define USE_RATE_MIN 5512 +#define USE_RATE_MAX 48000 +#define USE_CHANNELS_MIN 1 +#define USE_CHANNELS_MAX 2 +#define USE_PERIODS_MIN 2 +#define USE_PERIODS_MAX (MAX_BUFFER_SIZE / MIN_PERIOD_SIZE) + +static const struct snd_pcm_hardware SND_DRV_PCM_HW_DEFAULT = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_MMAP_VALID), + .formats = USE_FORMATS, + .rates = USE_RATE, + .rate_min = USE_RATE_MIN, + .rate_max = USE_RATE_MAX, + .channels_min = USE_CHANNELS_MIN, + .channels_max = USE_CHANNELS_MAX, + .buffer_bytes_max = MAX_BUFFER_SIZE, + .period_bytes_min = MIN_PERIOD_SIZE, + .period_bytes_max = MAX_PERIOD_SIZE, + .periods_min = USE_PERIODS_MIN, + .periods_max = USE_PERIODS_MAX, + .fifo_size = 0, +}; + +static void cfg_read_pcm_hw(const char *path, + struct snd_pcm_hardware *parent_pcm_hw, + struct snd_pcm_hardware *pcm_hw) +{ + char *list; + int val; + size_t buf_sz; + unsigned int len; + + /* Inherit parent's PCM HW and read overrides from XenStore. */ + if (parent_pcm_hw) + *pcm_hw = *parent_pcm_hw; + else + *pcm_hw = SND_DRV_PCM_HW_DEFAULT; + + val = xenbus_read_unsigned(path, XENSND_FIELD_CHANNELS_MIN, 0); + if (val) + pcm_hw->channels_min = val; + + val = xenbus_read_unsigned(path, XENSND_FIELD_CHANNELS_MAX, 0); + if (val) + pcm_hw->channels_max = val; + + list = xenbus_read(XBT_NIL, path, XENSND_FIELD_SAMPLE_RATES, &len); + if (!IS_ERR(list)) { + cfg_hw_rates(list, len, path, pcm_hw); + kfree(list); + } + + list = xenbus_read(XBT_NIL, path, XENSND_FIELD_SAMPLE_FORMATS, &len); + if (!IS_ERR(list)) { + cfg_formats(list, len, path, pcm_hw); + kfree(list); + } + + buf_sz = xenbus_read_unsigned(path, XENSND_FIELD_BUFFER_SIZE, 0); + if (buf_sz) + pcm_hw->buffer_bytes_max = buf_sz; + + /* Update configuration to match new values. */ + if (pcm_hw->channels_min > pcm_hw->channels_max) + pcm_hw->channels_min = pcm_hw->channels_max; + + if (pcm_hw->rate_min > pcm_hw->rate_max) + pcm_hw->rate_min = pcm_hw->rate_max; + + pcm_hw->period_bytes_max = pcm_hw->buffer_bytes_max; + + pcm_hw->periods_max = pcm_hw->period_bytes_max / + pcm_hw->period_bytes_min; +} + +static int cfg_get_stream_type(const char *path, int index, + int *num_pb, int *num_cap) +{ + char *str = NULL; + char *stream_path; + int ret; + + *num_pb = 0; + *num_cap = 0; + stream_path = kasprintf(GFP_KERNEL, "%s/%d", path, index); + if (!stream_path) { + ret = -ENOMEM; + goto fail; + } + + str = xenbus_read(XBT_NIL, stream_path, XENSND_FIELD_TYPE, NULL); + if (IS_ERR(str)) { + ret = PTR_ERR(str); + str = NULL; + goto fail; + } + + if (!strncasecmp(str, XENSND_STREAM_TYPE_PLAYBACK, + sizeof(XENSND_STREAM_TYPE_PLAYBACK))) { + (*num_pb)++; + } else if (!strncasecmp(str, XENSND_STREAM_TYPE_CAPTURE, + sizeof(XENSND_STREAM_TYPE_CAPTURE))) { + (*num_cap)++; + } else { + ret = -EINVAL; + goto fail; + } + ret = 0; + +fail: + kfree(stream_path); + kfree(str); + return ret; +} + +static int cfg_stream(struct xen_snd_front_info *front_info, + struct xen_front_cfg_pcm_instance *pcm_instance, + const char *path, int index, int *cur_pb, int *cur_cap, + int *stream_cnt) +{ + char *str = NULL; + char *stream_path; + struct xen_front_cfg_stream *stream; + int ret; + + stream_path = devm_kasprintf(&front_info->xb_dev->dev, + GFP_KERNEL, "%s/%d", path, index); + if (!stream_path) { + ret = -ENOMEM; + goto fail; + } + + str = xenbus_read(XBT_NIL, stream_path, XENSND_FIELD_TYPE, NULL); + if (IS_ERR(str)) { + ret = PTR_ERR(str); + str = NULL; + goto fail; + } + + if (!strncasecmp(str, XENSND_STREAM_TYPE_PLAYBACK, + sizeof(XENSND_STREAM_TYPE_PLAYBACK))) { + stream = &pcm_instance->streams_pb[(*cur_pb)++]; + } else if (!strncasecmp(str, XENSND_STREAM_TYPE_CAPTURE, + sizeof(XENSND_STREAM_TYPE_CAPTURE))) { + stream = &pcm_instance->streams_cap[(*cur_cap)++]; + } else { + ret = -EINVAL; + goto fail; + } + + /* Get next stream index. */ + stream->index = (*stream_cnt)++; + stream->xenstore_path = stream_path; + /* + * Check XenStore if PCM HW configuration exists for this stream + * and update if so, e.g. we inherit all values from device's PCM HW, + * but can still override some of the values for the stream. + */ + cfg_read_pcm_hw(stream->xenstore_path, + &pcm_instance->pcm_hw, &stream->pcm_hw); + ret = 0; + +fail: + kfree(str); + return ret; +} + +static int cfg_device(struct xen_snd_front_info *front_info, + struct xen_front_cfg_pcm_instance *pcm_instance, + struct snd_pcm_hardware *parent_pcm_hw, + const char *path, int node_index, int *stream_cnt) +{ + char *str; + char *device_path; + int ret, i, num_streams; + int num_pb, num_cap; + int cur_pb, cur_cap; + char node[3]; + + device_path = kasprintf(GFP_KERNEL, "%s/%d", path, node_index); + if (!device_path) + return -ENOMEM; + + str = xenbus_read(XBT_NIL, device_path, XENSND_FIELD_DEVICE_NAME, NULL); + if (!IS_ERR(str)) { + strlcpy(pcm_instance->name, str, sizeof(pcm_instance->name)); + kfree(str); + } + + pcm_instance->device_id = node_index; + + /* + * Check XenStore if PCM HW configuration exists for this device + * and update if so, e.g. we inherit all values from card's PCM HW, + * but can still override some of the values for the device. + */ + cfg_read_pcm_hw(device_path, parent_pcm_hw, &pcm_instance->pcm_hw); + + /* Find out how many streams were configured in Xen store. */ + num_streams = 0; + do { + snprintf(node, sizeof(node), "%d", num_streams); + if (!xenbus_exists(XBT_NIL, device_path, node)) + break; + + num_streams++; + } while (num_streams < VSND_MAX_STREAM); + + pcm_instance->num_streams_pb = 0; + pcm_instance->num_streams_cap = 0; + /* Get number of playback and capture streams. */ + for (i = 0; i < num_streams; i++) { + ret = cfg_get_stream_type(device_path, i, &num_pb, &num_cap); + if (ret < 0) + goto fail; + + pcm_instance->num_streams_pb += num_pb; + pcm_instance->num_streams_cap += num_cap; + } + + if (pcm_instance->num_streams_pb) { + pcm_instance->streams_pb = + devm_kcalloc(&front_info->xb_dev->dev, + pcm_instance->num_streams_pb, + sizeof(struct xen_front_cfg_stream), + GFP_KERNEL); + if (!pcm_instance->streams_pb) { + ret = -ENOMEM; + goto fail; + } + } + + if (pcm_instance->num_streams_cap) { + pcm_instance->streams_cap = + devm_kcalloc(&front_info->xb_dev->dev, + pcm_instance->num_streams_cap, + sizeof(struct xen_front_cfg_stream), + GFP_KERNEL); + if (!pcm_instance->streams_cap) { + ret = -ENOMEM; + goto fail; + } + } + + cur_pb = 0; + cur_cap = 0; + for (i = 0; i < num_streams; i++) { + ret = cfg_stream(front_info, pcm_instance, device_path, i, + &cur_pb, &cur_cap, stream_cnt); + if (ret < 0) + goto fail; + } + ret = 0; + +fail: + kfree(device_path); + return ret; +} + +int xen_snd_front_cfg_card(struct xen_snd_front_info *front_info, + int *stream_cnt) +{ + struct xenbus_device *xb_dev = front_info->xb_dev; + struct xen_front_cfg_card *cfg = &front_info->cfg; + int ret, num_devices, i; + char node[3]; + + *stream_cnt = 0; + num_devices = 0; + do { + snprintf(node, sizeof(node), "%d", num_devices); + if (!xenbus_exists(XBT_NIL, xb_dev->nodename, node)) + break; + + num_devices++; + } while (num_devices < SNDRV_PCM_DEVICES); + + if (!num_devices) { + dev_warn(&xb_dev->dev, + "No devices configured for sound card at %s\n", + xb_dev->nodename); + return -ENODEV; + } + + /* Start from default PCM HW configuration for the card. */ + cfg_read_pcm_hw(xb_dev->nodename, NULL, &cfg->pcm_hw); + + cfg->pcm_instances = + devm_kcalloc(&front_info->xb_dev->dev, num_devices, + sizeof(struct xen_front_cfg_pcm_instance), + GFP_KERNEL); + if (!cfg->pcm_instances) + return -ENOMEM; + + for (i = 0; i < num_devices; i++) { + ret = cfg_device(front_info, &cfg->pcm_instances[i], + &cfg->pcm_hw, xb_dev->nodename, i, stream_cnt); + if (ret < 0) + return ret; + } + cfg->num_pcm_instances = num_devices; + return 0; +} + diff --git a/sound/xen/xen_snd_front_cfg.h b/sound/xen/xen_snd_front_cfg.h new file mode 100644 index 000000000000..2353fcc74889 --- /dev/null +++ b/sound/xen/xen_snd_front_cfg.h @@ -0,0 +1,46 @@ +/* SPDX-License-Identifier: GPL-2.0 OR MIT */ + +/* + * Xen para-virtual sound device + * + * Copyright (C) 2016-2018 EPAM Systems Inc. + * + * Author: Oleksandr Andrushchenko <oleksandr_andrushchenko@epam.com> + */ + +#ifndef __XEN_SND_FRONT_CFG_H +#define __XEN_SND_FRONT_CFG_H + +#include <sound/core.h> +#include <sound/pcm.h> + +struct xen_snd_front_info; + +struct xen_front_cfg_stream { + int index; + char *xenstore_path; + struct snd_pcm_hardware pcm_hw; +}; + +struct xen_front_cfg_pcm_instance { + char name[80]; + int device_id; + struct snd_pcm_hardware pcm_hw; + int num_streams_pb; + struct xen_front_cfg_stream *streams_pb; + int num_streams_cap; + struct xen_front_cfg_stream *streams_cap; +}; + +struct xen_front_cfg_card { + char name_short[32]; + char name_long[80]; + struct snd_pcm_hardware pcm_hw; + int num_pcm_instances; + struct xen_front_cfg_pcm_instance *pcm_instances; +}; + +int xen_snd_front_cfg_card(struct xen_snd_front_info *front_info, + int *stream_cnt); + +#endif /* __XEN_SND_FRONT_CFG_H */ diff --git a/sound/xen/xen_snd_front_evtchnl.c b/sound/xen/xen_snd_front_evtchnl.c new file mode 100644 index 000000000000..102d6e096cc8 --- /dev/null +++ b/sound/xen/xen_snd_front_evtchnl.c @@ -0,0 +1,494 @@ +// SPDX-License-Identifier: GPL-2.0 OR MIT + +/* + * Xen para-virtual sound device + * + * Copyright (C) 2016-2018 EPAM Systems Inc. + * + * Author: Oleksandr Andrushchenko <oleksandr_andrushchenko@epam.com> + */ + +#include <xen/events.h> +#include <xen/grant_table.h> +#include <xen/xen.h> +#include <xen/xenbus.h> + +#include "xen_snd_front.h" +#include "xen_snd_front_alsa.h" +#include "xen_snd_front_cfg.h" +#include "xen_snd_front_evtchnl.h" + +static irqreturn_t evtchnl_interrupt_req(int irq, void *dev_id) +{ + struct xen_snd_front_evtchnl *channel = dev_id; + struct xen_snd_front_info *front_info = channel->front_info; + struct xensnd_resp *resp; + RING_IDX i, rp; + + if (unlikely(channel->state != EVTCHNL_STATE_CONNECTED)) + return IRQ_HANDLED; + + mutex_lock(&channel->ring_io_lock); + +again: + rp = channel->u.req.ring.sring->rsp_prod; + /* Ensure we see queued responses up to rp. */ + rmb(); + + /* + * Assume that the backend is trusted to always write sane values + * to the ring counters, so no overflow checks on frontend side + * are required. + */ + for (i = channel->u.req.ring.rsp_cons; i != rp; i++) { + resp = RING_GET_RESPONSE(&channel->u.req.ring, i); + if (resp->id != channel->evt_id) + continue; + switch (resp->operation) { + case XENSND_OP_OPEN: + /* fall through */ + case XENSND_OP_CLOSE: + /* fall through */ + case XENSND_OP_READ: + /* fall through */ + case XENSND_OP_WRITE: + /* fall through */ + case XENSND_OP_TRIGGER: + channel->u.req.resp_status = resp->status; + complete(&channel->u.req.completion); + break; + case XENSND_OP_HW_PARAM_QUERY: + channel->u.req.resp_status = resp->status; + channel->u.req.resp.hw_param = + resp->resp.hw_param; + complete(&channel->u.req.completion); + break; + + default: + dev_err(&front_info->xb_dev->dev, + "Operation %d is not supported\n", + resp->operation); + break; + } + } + + channel->u.req.ring.rsp_cons = i; + if (i != channel->u.req.ring.req_prod_pvt) { + int more_to_do; + + RING_FINAL_CHECK_FOR_RESPONSES(&channel->u.req.ring, + more_to_do); + if (more_to_do) + goto again; + } else { + channel->u.req.ring.sring->rsp_event = i + 1; + } + + mutex_unlock(&channel->ring_io_lock); + return IRQ_HANDLED; +} + +static irqreturn_t evtchnl_interrupt_evt(int irq, void *dev_id) +{ + struct xen_snd_front_evtchnl *channel = dev_id; + struct xensnd_event_page *page = channel->u.evt.page; + u32 cons, prod; + + if (unlikely(channel->state != EVTCHNL_STATE_CONNECTED)) + return IRQ_HANDLED; + + mutex_lock(&channel->ring_io_lock); + + prod = page->in_prod; + /* Ensure we see ring contents up to prod. */ + virt_rmb(); + if (prod == page->in_cons) + goto out; + + /* + * Assume that the backend is trusted to always write sane values + * to the ring counters, so no overflow checks on frontend side + * are required. + */ + for (cons = page->in_cons; cons != prod; cons++) { + struct xensnd_evt *event; + + event = &XENSND_IN_RING_REF(page, cons); + if (unlikely(event->id != channel->evt_id++)) + continue; + + switch (event->type) { + case XENSND_EVT_CUR_POS: + xen_snd_front_alsa_handle_cur_pos(channel, + event->op.cur_pos.position); + break; + } + } + + page->in_cons = cons; + /* Ensure ring contents. */ + virt_wmb(); + +out: + mutex_unlock(&channel->ring_io_lock); + return IRQ_HANDLED; +} + +void xen_snd_front_evtchnl_flush(struct xen_snd_front_evtchnl *channel) +{ + int notify; + + channel->u.req.ring.req_prod_pvt++; + RING_PUSH_REQUESTS_AND_CHECK_NOTIFY(&channel->u.req.ring, notify); + if (notify) + notify_remote_via_irq(channel->irq); +} + +static void evtchnl_free(struct xen_snd_front_info *front_info, + struct xen_snd_front_evtchnl *channel) +{ + unsigned long page = 0; + + if (channel->type == EVTCHNL_TYPE_REQ) + page = (unsigned long)channel->u.req.ring.sring; + else if (channel->type == EVTCHNL_TYPE_EVT) + page = (unsigned long)channel->u.evt.page; + + if (!page) + return; + + channel->state = EVTCHNL_STATE_DISCONNECTED; + if (channel->type == EVTCHNL_TYPE_REQ) { + /* Release all who still waits for response if any. */ + channel->u.req.resp_status = -EIO; + complete_all(&channel->u.req.completion); + } + + if (channel->irq) + unbind_from_irqhandler(channel->irq, channel); + + if (channel->port) + xenbus_free_evtchn(front_info->xb_dev, channel->port); + + /* End access and free the page. */ + if (channel->gref != GRANT_INVALID_REF) + gnttab_end_foreign_access(channel->gref, 0, page); + else + free_page(page); + + memset(channel, 0, sizeof(*channel)); +} + +void xen_snd_front_evtchnl_free_all(struct xen_snd_front_info *front_info) +{ + int i; + + if (!front_info->evt_pairs) + return; + + for (i = 0; i < front_info->num_evt_pairs; i++) { + evtchnl_free(front_info, &front_info->evt_pairs[i].req); + evtchnl_free(front_info, &front_info->evt_pairs[i].evt); + } + + kfree(front_info->evt_pairs); + front_info->evt_pairs = NULL; +} + +static int evtchnl_alloc(struct xen_snd_front_info *front_info, int index, + struct xen_snd_front_evtchnl *channel, + enum xen_snd_front_evtchnl_type type) +{ + struct xenbus_device *xb_dev = front_info->xb_dev; + unsigned long page; + grant_ref_t gref; + irq_handler_t handler; + char *handler_name = NULL; + int ret; + + memset(channel, 0, sizeof(*channel)); + channel->type = type; + channel->index = index; + channel->front_info = front_info; + channel->state = EVTCHNL_STATE_DISCONNECTED; + channel->gref = GRANT_INVALID_REF; + page = get_zeroed_page(GFP_KERNEL); + if (!page) { + ret = -ENOMEM; + goto fail; + } + + handler_name = kasprintf(GFP_KERNEL, "%s-%s", XENSND_DRIVER_NAME, + type == EVTCHNL_TYPE_REQ ? + XENSND_FIELD_RING_REF : + XENSND_FIELD_EVT_RING_REF); + if (!handler_name) { + ret = -ENOMEM; + goto fail; + } + + mutex_init(&channel->ring_io_lock); + + if (type == EVTCHNL_TYPE_REQ) { + struct xen_sndif_sring *sring = (struct xen_sndif_sring *)page; + + init_completion(&channel->u.req.completion); + mutex_init(&channel->u.req.req_io_lock); + SHARED_RING_INIT(sring); + FRONT_RING_INIT(&channel->u.req.ring, sring, XEN_PAGE_SIZE); + + ret = xenbus_grant_ring(xb_dev, sring, 1, &gref); + if (ret < 0) { + channel->u.req.ring.sring = NULL; + goto fail; + } + + handler = evtchnl_interrupt_req; + } else { + ret = gnttab_grant_foreign_access(xb_dev->otherend_id, + virt_to_gfn((void *)page), 0); + if (ret < 0) + goto fail; + + channel->u.evt.page = (struct xensnd_event_page *)page; + gref = ret; + handler = evtchnl_interrupt_evt; + } + + channel->gref = gref; + + ret = xenbus_alloc_evtchn(xb_dev, &channel->port); + if (ret < 0) + goto fail; + + ret = bind_evtchn_to_irq(channel->port); + if (ret < 0) { + dev_err(&xb_dev->dev, + "Failed to bind IRQ for domid %d port %d: %d\n", + front_info->xb_dev->otherend_id, channel->port, ret); + goto fail; + } + + channel->irq = ret; + + ret = request_threaded_irq(channel->irq, NULL, handler, + IRQF_ONESHOT, handler_name, channel); + if (ret < 0) { + dev_err(&xb_dev->dev, "Failed to request IRQ %d: %d\n", + channel->irq, ret); + goto fail; + } + + kfree(handler_name); + return 0; + +fail: + if (page) + free_page(page); + kfree(handler_name); + dev_err(&xb_dev->dev, "Failed to allocate ring: %d\n", ret); + return ret; +} + +int xen_snd_front_evtchnl_create_all(struct xen_snd_front_info *front_info, + int num_streams) +{ + struct xen_front_cfg_card *cfg = &front_info->cfg; + struct device *dev = &front_info->xb_dev->dev; + int d, ret = 0; + + front_info->evt_pairs = + kcalloc(num_streams, + sizeof(struct xen_snd_front_evtchnl_pair), + GFP_KERNEL); + if (!front_info->evt_pairs) + return -ENOMEM; + + /* Iterate over devices and their streams and create event channels. */ + for (d = 0; d < cfg->num_pcm_instances; d++) { + struct xen_front_cfg_pcm_instance *pcm_instance; + int s, index; + + pcm_instance = &cfg->pcm_instances[d]; + + for (s = 0; s < pcm_instance->num_streams_pb; s++) { + index = pcm_instance->streams_pb[s].index; + + ret = evtchnl_alloc(front_info, index, + &front_info->evt_pairs[index].req, + EVTCHNL_TYPE_REQ); + if (ret < 0) { + dev_err(dev, "Error allocating control channel\n"); + goto fail; + } + + ret = evtchnl_alloc(front_info, index, + &front_info->evt_pairs[index].evt, + EVTCHNL_TYPE_EVT); + if (ret < 0) { + dev_err(dev, "Error allocating in-event channel\n"); + goto fail; + } + } + + for (s = 0; s < pcm_instance->num_streams_cap; s++) { + index = pcm_instance->streams_cap[s].index; + + ret = evtchnl_alloc(front_info, index, + &front_info->evt_pairs[index].req, + EVTCHNL_TYPE_REQ); + if (ret < 0) { + dev_err(dev, "Error allocating control channel\n"); + goto fail; + } + + ret = evtchnl_alloc(front_info, index, + &front_info->evt_pairs[index].evt, + EVTCHNL_TYPE_EVT); + if (ret < 0) { + dev_err(dev, "Error allocating in-event channel\n"); + goto fail; + } + } + } + + front_info->num_evt_pairs = num_streams; + return 0; + +fail: + xen_snd_front_evtchnl_free_all(front_info); + return ret; +} + +static int evtchnl_publish(struct xenbus_transaction xbt, + struct xen_snd_front_evtchnl *channel, + const char *path, const char *node_ring, + const char *node_chnl) +{ + struct xenbus_device *xb_dev = channel->front_info->xb_dev; + int ret; + + /* Write control channel ring reference. */ + ret = xenbus_printf(xbt, path, node_ring, "%u", channel->gref); + if (ret < 0) { + dev_err(&xb_dev->dev, "Error writing ring-ref: %d\n", ret); + return ret; + } + + /* Write event channel ring reference. */ + ret = xenbus_printf(xbt, path, node_chnl, "%u", channel->port); + if (ret < 0) { + dev_err(&xb_dev->dev, "Error writing event channel: %d\n", ret); + return ret; + } + + return 0; +} + +int xen_snd_front_evtchnl_publish_all(struct xen_snd_front_info *front_info) +{ + struct xen_front_cfg_card *cfg = &front_info->cfg; + struct xenbus_transaction xbt; + int ret, d; + +again: + ret = xenbus_transaction_start(&xbt); + if (ret < 0) { + xenbus_dev_fatal(front_info->xb_dev, ret, + "starting transaction"); + return ret; + } + + for (d = 0; d < cfg->num_pcm_instances; d++) { + struct xen_front_cfg_pcm_instance *pcm_instance; + int s, index; + + pcm_instance = &cfg->pcm_instances[d]; + + for (s = 0; s < pcm_instance->num_streams_pb; s++) { + index = pcm_instance->streams_pb[s].index; + + ret = evtchnl_publish(xbt, + &front_info->evt_pairs[index].req, + pcm_instance->streams_pb[s].xenstore_path, + XENSND_FIELD_RING_REF, + XENSND_FIELD_EVT_CHNL); + if (ret < 0) + goto fail; + + ret = evtchnl_publish(xbt, + &front_info->evt_pairs[index].evt, + pcm_instance->streams_pb[s].xenstore_path, + XENSND_FIELD_EVT_RING_REF, + XENSND_FIELD_EVT_EVT_CHNL); + if (ret < 0) + goto fail; + } + + for (s = 0; s < pcm_instance->num_streams_cap; s++) { + index = pcm_instance->streams_cap[s].index; + + ret = evtchnl_publish(xbt, + &front_info->evt_pairs[index].req, + pcm_instance->streams_cap[s].xenstore_path, + XENSND_FIELD_RING_REF, + XENSND_FIELD_EVT_CHNL); + if (ret < 0) + goto fail; + + ret = evtchnl_publish(xbt, + &front_info->evt_pairs[index].evt, + pcm_instance->streams_cap[s].xenstore_path, + XENSND_FIELD_EVT_RING_REF, + XENSND_FIELD_EVT_EVT_CHNL); + if (ret < 0) + goto fail; + } + } + ret = xenbus_transaction_end(xbt, 0); + if (ret < 0) { + if (ret == -EAGAIN) + goto again; + + xenbus_dev_fatal(front_info->xb_dev, ret, + "completing transaction"); + goto fail_to_end; + } + return 0; +fail: + xenbus_transaction_end(xbt, 1); +fail_to_end: + xenbus_dev_fatal(front_info->xb_dev, ret, "writing XenStore"); + return ret; +} + +void xen_snd_front_evtchnl_pair_set_connected(struct xen_snd_front_evtchnl_pair *evt_pair, + bool is_connected) +{ + enum xen_snd_front_evtchnl_state state; + + if (is_connected) + state = EVTCHNL_STATE_CONNECTED; + else + state = EVTCHNL_STATE_DISCONNECTED; + + mutex_lock(&evt_pair->req.ring_io_lock); + evt_pair->req.state = state; + mutex_unlock(&evt_pair->req.ring_io_lock); + + mutex_lock(&evt_pair->evt.ring_io_lock); + evt_pair->evt.state = state; + mutex_unlock(&evt_pair->evt.ring_io_lock); +} + +void xen_snd_front_evtchnl_pair_clear(struct xen_snd_front_evtchnl_pair *evt_pair) +{ + mutex_lock(&evt_pair->req.ring_io_lock); + evt_pair->req.evt_next_id = 0; + mutex_unlock(&evt_pair->req.ring_io_lock); + + mutex_lock(&evt_pair->evt.ring_io_lock); + evt_pair->evt.evt_next_id = 0; + mutex_unlock(&evt_pair->evt.ring_io_lock); +} + diff --git a/sound/xen/xen_snd_front_evtchnl.h b/sound/xen/xen_snd_front_evtchnl.h new file mode 100644 index 000000000000..cbe51fd1ec15 --- /dev/null +++ b/sound/xen/xen_snd_front_evtchnl.h @@ -0,0 +1,95 @@ +/* SPDX-License-Identifier: GPL-2.0 OR MIT */ + +/* + * Xen para-virtual sound device + * + * Copyright (C) 2016-2018 EPAM Systems Inc. + * + * Author: Oleksandr Andrushchenko <oleksandr_andrushchenko@epam.com> + */ + +#ifndef __XEN_SND_FRONT_EVTCHNL_H +#define __XEN_SND_FRONT_EVTCHNL_H + +#include <xen/interface/io/sndif.h> + +struct xen_snd_front_info; + +#ifndef GRANT_INVALID_REF +/* + * FIXME: usage of grant reference 0 as invalid grant reference: + * grant reference 0 is valid, but never exposed to a PV driver, + * because of the fact it is already in use/reserved by the PV console. + */ +#define GRANT_INVALID_REF 0 +#endif + +/* Timeout in ms to wait for backend to respond. */ +#define VSND_WAIT_BACK_MS 3000 + +enum xen_snd_front_evtchnl_state { + EVTCHNL_STATE_DISCONNECTED, + EVTCHNL_STATE_CONNECTED, +}; + +enum xen_snd_front_evtchnl_type { + EVTCHNL_TYPE_REQ, + EVTCHNL_TYPE_EVT, +}; + +struct xen_snd_front_evtchnl { + struct xen_snd_front_info *front_info; + int gref; + int port; + int irq; + int index; + /* State of the event channel. */ + enum xen_snd_front_evtchnl_state state; + enum xen_snd_front_evtchnl_type type; + /* Either response id or incoming event id. */ + u16 evt_id; + /* Next request id or next expected event id. */ + u16 evt_next_id; + /* Shared ring access lock. */ + struct mutex ring_io_lock; + union { + struct { + struct xen_sndif_front_ring ring; + struct completion completion; + /* Serializer for backend IO: request/response. */ + struct mutex req_io_lock; + + /* Latest response status. */ + int resp_status; + union { + struct xensnd_query_hw_param hw_param; + } resp; + } req; + struct { + struct xensnd_event_page *page; + /* This is needed to handle XENSND_EVT_CUR_POS event. */ + struct snd_pcm_substream *substream; + } evt; + } u; +}; + +struct xen_snd_front_evtchnl_pair { + struct xen_snd_front_evtchnl req; + struct xen_snd_front_evtchnl evt; +}; + +int xen_snd_front_evtchnl_create_all(struct xen_snd_front_info *front_info, + int num_streams); + +void xen_snd_front_evtchnl_free_all(struct xen_snd_front_info *front_info); + +int xen_snd_front_evtchnl_publish_all(struct xen_snd_front_info *front_info); + +void xen_snd_front_evtchnl_flush(struct xen_snd_front_evtchnl *evtchnl); + +void xen_snd_front_evtchnl_pair_set_connected(struct xen_snd_front_evtchnl_pair *evt_pair, + bool is_connected); + +void xen_snd_front_evtchnl_pair_clear(struct xen_snd_front_evtchnl_pair *evt_pair); + +#endif /* __XEN_SND_FRONT_EVTCHNL_H */ diff --git a/sound/xen/xen_snd_front_shbuf.c b/sound/xen/xen_snd_front_shbuf.c new file mode 100644 index 000000000000..07ac176a41ba --- /dev/null +++ b/sound/xen/xen_snd_front_shbuf.c @@ -0,0 +1,194 @@ +// SPDX-License-Identifier: GPL-2.0 OR MIT + +/* + * Xen para-virtual sound device + * + * Copyright (C) 2016-2018 EPAM Systems Inc. + * + * Author: Oleksandr Andrushchenko <oleksandr_andrushchenko@epam.com> + */ + +#include <linux/kernel.h> +#include <xen/xen.h> +#include <xen/xenbus.h> + +#include "xen_snd_front_shbuf.h" + +grant_ref_t xen_snd_front_shbuf_get_dir_start(struct xen_snd_front_shbuf *buf) +{ + if (!buf->grefs) + return GRANT_INVALID_REF; + + return buf->grefs[0]; +} + +void xen_snd_front_shbuf_clear(struct xen_snd_front_shbuf *buf) +{ + memset(buf, 0, sizeof(*buf)); +} + +void xen_snd_front_shbuf_free(struct xen_snd_front_shbuf *buf) +{ + int i; + + if (buf->grefs) { + for (i = 0; i < buf->num_grefs; i++) + if (buf->grefs[i] != GRANT_INVALID_REF) + gnttab_end_foreign_access(buf->grefs[i], + 0, 0UL); + kfree(buf->grefs); + } + kfree(buf->directory); + free_pages_exact(buf->buffer, buf->buffer_sz); + xen_snd_front_shbuf_clear(buf); +} + +/* + * number of grant references a page can hold with respect to the + * xensnd_page_directory header + */ +#define XENSND_NUM_GREFS_PER_PAGE ((XEN_PAGE_SIZE - \ + offsetof(struct xensnd_page_directory, gref)) / \ + sizeof(grant_ref_t)) + +static void fill_page_dir(struct xen_snd_front_shbuf *buf, + int num_pages_dir) +{ + struct xensnd_page_directory *page_dir; + unsigned char *ptr; + int i, cur_gref, grefs_left, to_copy; + + ptr = buf->directory; + grefs_left = buf->num_grefs - num_pages_dir; + /* + * skip grant references at the beginning, they are for pages granted + * for the page directory itself + */ + cur_gref = num_pages_dir; + for (i = 0; i < num_pages_dir; i++) { + page_dir = (struct xensnd_page_directory *)ptr; + if (grefs_left <= XENSND_NUM_GREFS_PER_PAGE) { + to_copy = grefs_left; + page_dir->gref_dir_next_page = GRANT_INVALID_REF; + } else { + to_copy = XENSND_NUM_GREFS_PER_PAGE; + page_dir->gref_dir_next_page = buf->grefs[i + 1]; + } + + memcpy(&page_dir->gref, &buf->grefs[cur_gref], + to_copy * sizeof(grant_ref_t)); + + ptr += XEN_PAGE_SIZE; + grefs_left -= to_copy; + cur_gref += to_copy; + } +} + +static int grant_references(struct xenbus_device *xb_dev, + struct xen_snd_front_shbuf *buf, + int num_pages_dir, int num_pages_buffer, + int num_grefs) +{ + grant_ref_t priv_gref_head; + unsigned long frame; + int ret, i, j, cur_ref; + int otherend_id; + + ret = gnttab_alloc_grant_references(num_grefs, &priv_gref_head); + if (ret) + return ret; + + buf->num_grefs = num_grefs; + otherend_id = xb_dev->otherend_id; + j = 0; + + for (i = 0; i < num_pages_dir; i++) { + cur_ref = gnttab_claim_grant_reference(&priv_gref_head); + if (cur_ref < 0) { + ret = cur_ref; + goto fail; + } + + frame = xen_page_to_gfn(virt_to_page(buf->directory + + XEN_PAGE_SIZE * i)); + gnttab_grant_foreign_access_ref(cur_ref, otherend_id, frame, 0); + buf->grefs[j++] = cur_ref; + } + + for (i = 0; i < num_pages_buffer; i++) { + cur_ref = gnttab_claim_grant_reference(&priv_gref_head); + if (cur_ref < 0) { + ret = cur_ref; + goto fail; + } + + frame = xen_page_to_gfn(virt_to_page(buf->buffer + + XEN_PAGE_SIZE * i)); + gnttab_grant_foreign_access_ref(cur_ref, otherend_id, frame, 0); + buf->grefs[j++] = cur_ref; + } + + gnttab_free_grant_references(priv_gref_head); + fill_page_dir(buf, num_pages_dir); + return 0; + +fail: + gnttab_free_grant_references(priv_gref_head); + return ret; +} + +static int alloc_int_buffers(struct xen_snd_front_shbuf *buf, + int num_pages_dir, int num_pages_buffer, + int num_grefs) +{ + buf->grefs = kcalloc(num_grefs, sizeof(*buf->grefs), GFP_KERNEL); + if (!buf->grefs) + return -ENOMEM; + + buf->directory = kcalloc(num_pages_dir, XEN_PAGE_SIZE, GFP_KERNEL); + if (!buf->directory) + goto fail; + + buf->buffer_sz = num_pages_buffer * XEN_PAGE_SIZE; + buf->buffer = alloc_pages_exact(buf->buffer_sz, GFP_KERNEL); + if (!buf->buffer) + goto fail; + + return 0; + +fail: + kfree(buf->grefs); + buf->grefs = NULL; + kfree(buf->directory); + buf->directory = NULL; + return -ENOMEM; +} + +int xen_snd_front_shbuf_alloc(struct xenbus_device *xb_dev, + struct xen_snd_front_shbuf *buf, + unsigned int buffer_sz) +{ + int num_pages_buffer, num_pages_dir, num_grefs; + int ret; + + xen_snd_front_shbuf_clear(buf); + + num_pages_buffer = DIV_ROUND_UP(buffer_sz, XEN_PAGE_SIZE); + /* number of pages the page directory consumes itself */ + num_pages_dir = DIV_ROUND_UP(num_pages_buffer, + XENSND_NUM_GREFS_PER_PAGE); + num_grefs = num_pages_buffer + num_pages_dir; + + ret = alloc_int_buffers(buf, num_pages_dir, + num_pages_buffer, num_grefs); + if (ret < 0) + return ret; + + ret = grant_references(xb_dev, buf, num_pages_dir, num_pages_buffer, + num_grefs); + if (ret < 0) + return ret; + + fill_page_dir(buf, num_pages_dir); + return 0; +} diff --git a/sound/xen/xen_snd_front_shbuf.h b/sound/xen/xen_snd_front_shbuf.h new file mode 100644 index 000000000000..d28e97c47b2c --- /dev/null +++ b/sound/xen/xen_snd_front_shbuf.h @@ -0,0 +1,36 @@ +/* SPDX-License-Identifier: GPL-2.0 OR MIT */ + +/* + * Xen para-virtual sound device + * + * Copyright (C) 2016-2018 EPAM Systems Inc. + * + * Author: Oleksandr Andrushchenko <oleksandr_andrushchenko@epam.com> + */ + +#ifndef __XEN_SND_FRONT_SHBUF_H +#define __XEN_SND_FRONT_SHBUF_H + +#include <xen/grant_table.h> + +#include "xen_snd_front_evtchnl.h" + +struct xen_snd_front_shbuf { + int num_grefs; + grant_ref_t *grefs; + u8 *directory; + u8 *buffer; + size_t buffer_sz; +}; + +grant_ref_t xen_snd_front_shbuf_get_dir_start(struct xen_snd_front_shbuf *buf); + +int xen_snd_front_shbuf_alloc(struct xenbus_device *xb_dev, + struct xen_snd_front_shbuf *buf, + unsigned int buffer_sz); + +void xen_snd_front_shbuf_clear(struct xen_snd_front_shbuf *buf); + +void xen_snd_front_shbuf_free(struct xen_snd_front_shbuf *buf); + +#endif /* __XEN_SND_FRONT_SHBUF_H */ |