[alsa-devel] [PATCH 1/2] ALSA: core: add hooks for audio timestamps
ALSA did not provide any direct means to infer the audio time for A/V sync and system/audio time correlations (eg. PulseAudio). Applications had to track the number of samples read/written and add/subtract the number of samples queued in the ring buffer. This accounting led to small errors, typically several samples, due to the two-step process. Computing the audio time in the kernel is more direct, as all the information is available in the same routines.
Also add new .audio_wallclock routine to enable fine-grain synchronization between monotonic system time and audio hardware time. Using the wallclock, if supported in hardware, allows for a much better sub-microsecond precision and a common drift tracking for all devices sharing the same wall clock (master clock).
Signed-off-by: Pierre-Louis Bossart pierre-louis.bossart@linux.intel.com --- include/sound/asound.h | 7 +++++-- include/sound/pcm.h | 2 ++ sound/core/pcm_compat.c | 13 +++++++++++-- sound/core/pcm_lib.c | 29 +++++++++++++++++++++++++++-- sound/core/pcm_native.c | 2 ++ 5 files changed, 47 insertions(+), 6 deletions(-)
diff --git a/include/sound/asound.h b/include/sound/asound.h index 0876a1e..2d1ba63 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -152,7 +152,7 @@ struct snd_hwdep_dsp_image { * * *****************************************************************************/
-#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 10) +#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 11)
typedef unsigned long snd_pcm_uframes_t; typedef signed long snd_pcm_sframes_t; @@ -274,6 +274,7 @@ typedef int __bitwise snd_pcm_subformat_t; #define SNDRV_PCM_INFO_JOINT_DUPLEX 0x00200000 /* playback and capture stream are somewhat correlated */ #define SNDRV_PCM_INFO_SYNC_START 0x00400000 /* pcm support some kind of sync go */ #define SNDRV_PCM_INFO_NO_PERIOD_WAKEUP 0x00800000 /* period wakeup can be disabled */ +#define SNDRV_PCM_INFO_HAS_WALL_CLOCK 0x01000000 /* has audio wall clock for audio/system time sync */ #define SNDRV_PCM_INFO_FIFO_IN_FRAMES 0x80000000 /* internal kernel flag - FIFO size is in frames */
typedef int __bitwise snd_pcm_state_t; @@ -422,7 +423,8 @@ struct snd_pcm_status { snd_pcm_uframes_t avail_max; /* max frames available on hw since last status */ snd_pcm_uframes_t overrange; /* count of ADC (capture) overrange detections from last status */ snd_pcm_state_t suspended_state; /* suspended stream state */ - unsigned char reserved[60]; /* must be filled with zero */ + struct timespec audio_tstamp; /* from sample counter or wall clock */ + unsigned char reserved[60-sizeof(struct timespec)]; /* must be filled with zero */ };
struct snd_pcm_mmap_status { @@ -431,6 +433,7 @@ struct snd_pcm_mmap_status { snd_pcm_uframes_t hw_ptr; /* RO: hw ptr (0...boundary-1) */ struct timespec tstamp; /* Timestamp */ snd_pcm_state_t suspended_state; /* RO: suspended stream state */ + struct timespec audio_tstamp; /* from sample counter or wall clock */ };
struct snd_pcm_mmap_control { diff --git a/include/sound/pcm.h b/include/sound/pcm.h index cdca2ab..35c2bbf 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -71,6 +71,8 @@ struct snd_pcm_ops { int (*prepare)(struct snd_pcm_substream *substream); int (*trigger)(struct snd_pcm_substream *substream, int cmd); snd_pcm_uframes_t (*pointer)(struct snd_pcm_substream *substream); + int (*wall_clock)(struct snd_pcm_substream *substream, + struct timespec *audio_ts); int (*copy)(struct snd_pcm_substream *substream, int channel, snd_pcm_uframes_t pos, void __user *buf, snd_pcm_uframes_t count); diff --git a/sound/core/pcm_compat.c b/sound/core/pcm_compat.c index 91cdf94..ab4c953 100644 --- a/sound/core/pcm_compat.c +++ b/sound/core/pcm_compat.c @@ -190,7 +190,8 @@ struct snd_pcm_status32 { u32 avail_max; u32 overrange; s32 suspended_state; - unsigned char reserved[60]; + struct timespec audio_tstamp; + unsigned char reserved[60-sizeof(struct timespec)]; } __attribute__((packed));
@@ -215,7 +216,9 @@ static int snd_pcm_status_user_compat(struct snd_pcm_substream *substream, put_user(status.avail, &src->avail) || put_user(status.avail_max, &src->avail_max) || put_user(status.overrange, &src->overrange) || - put_user(status.suspended_state, &src->suspended_state)) + put_user(status.suspended_state, &src->suspended_state) || + put_user(status.audio_tstamp.tv_sec, &src->audio_tstamp.tv_sec) || + put_user(status.audio_tstamp.tv_nsec, &src->audio_tstamp.tv_nsec)) return -EFAULT;
return err; @@ -364,6 +367,7 @@ struct snd_pcm_mmap_status32 { u32 hw_ptr; struct compat_timespec tstamp; s32 suspended_state; + struct compat_timespec audio_tstamp; } __attribute__((packed));
struct snd_pcm_mmap_control32 { @@ -426,12 +430,17 @@ static int snd_pcm_ioctl_sync_ptr_compat(struct snd_pcm_substream *substream, sstatus.hw_ptr = status->hw_ptr % boundary; sstatus.tstamp = status->tstamp; sstatus.suspended_state = status->suspended_state; + sstatus.audio_tstamp = status->audio_tstamp; snd_pcm_stream_unlock_irq(substream); if (put_user(sstatus.state, &src->s.status.state) || put_user(sstatus.hw_ptr, &src->s.status.hw_ptr) || put_user(sstatus.tstamp.tv_sec, &src->s.status.tstamp.tv_sec) || put_user(sstatus.tstamp.tv_nsec, &src->s.status.tstamp.tv_nsec) || put_user(sstatus.suspended_state, &src->s.status.suspended_state) || + put_user(sstatus.audio_tstamp.tv_sec, + &src->s.status.audio_tstamp.tv_sec) || + put_user(sstatus.audio_tstamp.tv_nsec, + &src->s.status.audio_tstamp.tv_nsec) || put_user(scontrol.appl_ptr, &src->c.control.appl_ptr) || put_user(scontrol.avail_min, &src->c.control.avail_min)) return -EFAULT; diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 7ae6719..ee0c931 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -315,6 +315,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, unsigned long jdelta; unsigned long curr_jiffies; struct timespec curr_tstamp; + struct timespec audio_tstamp;
old_hw_ptr = runtime->status->hw_ptr;
@@ -326,9 +327,14 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, */ pos = substream->ops->pointer(substream); curr_jiffies = jiffies; - if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) + if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) { snd_pcm_gettime(runtime, (struct timespec *)&curr_tstamp);
+ if ((runtime->hw.info & SNDRV_PCM_INFO_HAS_WALL_CLOCK) && + (substream->ops->wall_clock)) + substream->ops->wall_clock(substream, &audio_tstamp); + } + if (pos == SNDRV_PCM_POS_XRUN) { xrun(substream); return -EPIPE; @@ -506,8 +512,27 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, runtime->hw_ptr_base = hw_base; runtime->status->hw_ptr = new_hw_ptr; runtime->hw_ptr_jiffies = curr_jiffies; - if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) + if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) { runtime->status->tstamp = curr_tstamp; + if (!(runtime->hw.info & SNDRV_PCM_INFO_HAS_WALL_CLOCK)) { + /* + * no wall clock available, provide audio timestamp + * derived from pointer position+delay + */ + u64 audio_frames, audio_nsecs; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + audio_frames = runtime->status->hw_ptr + - runtime->delay; + else + audio_frames = runtime->status->hw_ptr + + runtime->delay; + audio_nsecs = audio_frames * 1000000000LL / + runtime->rate; + audio_tstamp = ns_to_timespec(audio_nsecs); + } + runtime->status->audio_tstamp = audio_tstamp; + }
return snd_pcm_update_state(substream, runtime); } diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 53b5ada..c5206f4 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -594,6 +594,8 @@ int snd_pcm_status(struct snd_pcm_substream *substream, snd_pcm_update_hw_ptr(substream); if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) { status->tstamp = runtime->status->tstamp; + status->audio_tstamp = + runtime->status->audio_tstamp; goto _tstamp_end; } }
Reuse code from clocksource to handle wall clock counter. Since wrapparound occurs, the audio timestamp is reinitialized to zero on a trigger. Synchronized linked devices will start counting from same reference to avoid any drift.
Max buffer time is limited to 178 seconds to make sure wall clock counter does not overflow
Signed-off-by: Pierre-Louis Bossart pierre-louis.bossart@linux.intel.com --- sound/pci/hda/hda_intel.c | 89 +++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 89 insertions(+)
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index b682442..208f73d 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -46,6 +46,10 @@ #include <linux/mutex.h> #include <linux/reboot.h> #include <linux/io.h> + +#include <linux/clocksource.h> +#include <linux/time.h> + #ifdef CONFIG_X86 /* for snoop control */ #include <asm/pgtable.h> @@ -406,6 +410,9 @@ struct azx_dev { */ unsigned int insufficient :1; unsigned int wc_marked:1; + + struct timecounter azx_tc; + struct cyclecounter azx_cc; };
/* CORB/RIRB */ @@ -1703,6 +1710,64 @@ static inline void azx_release_device(struct azx_dev *azx_dev) azx_dev->opened = 0; }
+static cycle_t azx_cc_read(const struct cyclecounter *cc) +{ + struct azx_dev *azx_dev = container_of(cc, struct azx_dev, azx_cc); + struct snd_pcm_substream *substream = azx_dev->substream; + struct azx_pcm *apcm = snd_pcm_substream_chip(substream); + struct azx *chip = apcm->chip; + + return azx_readl(chip, WALLCLK); +} + +static void azx_timecounter_init(struct snd_pcm_substream *substream, + bool force, cycle_t last) +{ + struct azx_dev *azx_dev = get_azx_dev(substream); + struct timecounter *tc = &azx_dev->azx_tc; + struct cyclecounter *cc = &azx_dev->azx_cc; + u64 nsec; + + cc->read = azx_cc_read; + cc->mask = CLOCKSOURCE_MASK(32); + + /* + * Converting from 24 MHz to ns means applying a 125/3 factor. + * To avoid any saturation issues in intermediate operations, + * the 125 factor is applied first. The division is applied + * last after reading the timecounter value. + * Applying the 1/3 factor as part of the multiplication + * requires at least 20 bits for a decent precision, however + * overflows occur after about 4 hours or less, not a option. + */ + + cc->mult = 125; /* saturation after 195 years */ + cc->shift = 0; + + nsec = 0; /* audio time is elapsed time since trigger */ + timecounter_init(tc, cc, nsec); + if (force) + /* + * force timecounter to use predefined value, + * used for synchronized starts + */ + tc->cycle_last = last; +} + +static int azx_get_wallclock_tstamp(struct snd_pcm_substream *substream, + struct timespec *ts) +{ + struct azx_dev *azx_dev = get_azx_dev(substream); + u64 nsec; + + nsec = timecounter_read(&azx_dev->azx_tc); + nsec /= 3; /* can be optimized */ + + *ts = ns_to_timespec(nsec); + + return 0; +} + static struct snd_pcm_hardware azx_pcm_hw = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | @@ -1712,6 +1777,7 @@ static struct snd_pcm_hardware azx_pcm_hw = { /* SNDRV_PCM_INFO_RESUME |*/ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_SYNC_START | + SNDRV_PCM_INFO_HAS_WALL_CLOCK | SNDRV_PCM_INFO_NO_PERIOD_WAKEUP), .formats = SNDRV_PCM_FMTBIT_S16_LE, .rates = SNDRV_PCM_RATE_48000, @@ -1751,6 +1817,12 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) runtime->hw.rates = hinfo->rates; snd_pcm_limit_hw_rates(runtime); snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); + + /* avoid wrap-around with wall-clock */ + snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_TIME, + 20, + 178000000); + if (chip->align_buffer_size) /* constrain buffer sizes to be multiple of 128 bytes. This is more efficient in terms of memory @@ -2024,6 +2096,22 @@ static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) azx_readl(chip, OLD_SSYNC) & ~sbits); else azx_writel(chip, SSYNC, azx_readl(chip, SSYNC) & ~sbits); + if (start) { + azx_timecounter_init(substream, 0, 0); + if (nsync > 1) { + cycle_t cycle_last; + + /* same start cycle for master and group */ + azx_dev = get_azx_dev(substream); + cycle_last = azx_dev->azx_tc.cycle_last; + + snd_pcm_group_for_each_entry(s, substream) { + if (s->pcm->card != substream->pcm->card) + continue; + azx_timecounter_init(s, 1, cycle_last); + } + } + } spin_unlock(&chip->reg_lock); return 0; } @@ -2260,6 +2348,7 @@ static struct snd_pcm_ops azx_pcm_ops = { .prepare = azx_pcm_prepare, .trigger = azx_pcm_trigger, .pointer = azx_pcm_pointer, + .wall_clock = azx_get_wallclock_tstamp, .mmap = azx_pcm_mmap, .page = snd_pcm_sgbuf_ops_page, };
Pierre-Louis Bossart wrote:
@@ -1712,6 +1777,7 @@ static struct snd_pcm_hardware azx_pcm_hw = { /* SNDRV_PCM_INFO_RESUME |*/ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_SYNC_START |
SNDRV_PCM_INFO_HAS_WALL_CLOCK |
... but not if this device is connected to an S/PDIF input.
Regards, Clemens
On 09/28/2012 04:33 AM, Clemens Ladisch wrote:
Pierre-Louis Bossart wrote:
@@ -1712,6 +1777,7 @@ static struct snd_pcm_hardware azx_pcm_hw = { /* SNDRV_PCM_INFO_RESUME |*/ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_SYNC_START |
SNDRV_PCM_INFO_HAS_WALL_CLOCK |... but not if this device is connected to an S/PDIF input.
Yeah, but I can't figure out what happens here. HDAudio is the clock master in all cases, unless the audio codec does an ASRC I don't think this works. I've looked at the spec several times, asked around, this case looks wild.
Pierre-Louis Bossart wrote:
On 09/28/2012 04:33 AM, Clemens Ladisch wrote:
Pierre-Louis Bossart wrote:
@@ -1712,6 +1777,7 @@ static struct snd_pcm_hardware azx_pcm_hw = { /* SNDRV_PCM_INFO_RESUME |*/ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_SYNC_START |
SNDRV_PCM_INFO_HAS_WALL_CLOCK |... but not if this device is connected to an S/PDIF input.
Yeah, but I can't figure out what happens here. HDAudio is the clock master in all cases
Except for those cases described in section 5.4.3 of the HDA spec.
Finding out whether a device is connected to an S/PDIF input is not easy and might change dynamically.
It might be easiest to omit HAS_WALL_CLOCK for inputs ...
Regards, Clemens
Reuse code from clocksource to handle wall clock counter. Since wrapparound occurs, the audio timestamp is reinitialized to zero on a trigger. Synchronized linked devices will start counting from same reference to avoid any drift.
@@ -2024,6 +2096,22 @@ static int azx_pcm_trigger(struct
snd_pcm_substream *substream, int cmd)
azx_readl(chip, OLD_SSYNC) & ~sbits); else azx_writel(chip, SSYNC, azx_readl(chip, SSYNC) & ~sbits);
if (start) {azx_timecounter_init(substream, 0, 0);if (nsync > 1) {cycle_t cycle_last;/* same start cycle for master and group */azx_dev = get_azx_dev(substream);cycle_last = azx_dev->azx_tc.cycle_last;snd_pcm_group_for_each_entry(s, substream) {if (s->pcm->card != substream->pcm->card)continue;azx_timecounter_init(s, 1, cycle_last);}}} spin_unlock(&chip->reg_lock); return 0;}
Will the timer reinitialise when multistreaming using front panel headphone , rear panel speakers , digital output and hdmi ?
Pierre-Louis Bossart wrote:
@@ -506,8 +512,27 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream,
if (!(runtime->hw.info & SNDRV_PCM_INFO_HAS_WALL_CLOCK)) {/** no wall clock available, provide audio timestamp* derived from pointer position+delay*/u64 audio_frames, audio_nsecs;if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)audio_frames = runtime->status->hw_ptr- runtime->delay;elseaudio_frames = runtime->status->hw_ptr+ runtime->delay;audio_nsecs = audio_frames * 1000000000LL /runtime->rate;
This looks like a 64-bit division.
And what happens if audio_frames becomes negative?
Regards, Clemens
audio_frames = runtime->status->hw_ptr+ runtime->delay;audio_nsecs = audio_frames * 1000000000LL /runtime->rate;This looks like a 64-bit division.
And what happens if audio_frames becomes negative?
It's my understanding that hw_ptr represents the cumulative frames played since the beginning, not sure why it'd become negative, ever. I know this deserves more love, I don't understand the notion of 'boundary' and the use of the hw_ptr_base, I figured smarter people than me would help. Thanks, -Pierre
Pierre-Louis Bossart wrote:
audio_frames = runtime->status->hw_ptr+ runtime->delay;audio_nsecs = audio_frames * 1000000000LL /runtime->rate;This looks like a 64-bit division.
... which needs to be handled with a function from <linux/math64.h>.
And what happens if audio_frames becomes negative?
It's my understanding that hw_ptr represents the cumulative frames played since the beginning, not sure why it'd become negative, ever.
2^32 / 192 kHz = 6.2 h
I don't understand the notion of 'boundary'
This is where hw_ptr wraps around. It's a multiple of the buffer size to make some calculations easier.
and the use of the hw_ptr_base
It's the hw_ptr corresponding to the start of the buffer.
I guess audio_tstamp isn't supposed to wrap around?
Regards, Clemens
participants (3)
-
Clemens Ladisch -
Pierre-Louis Bossart -
Raymond Yau