void aout_DecFlush (audio_output_t *aout)
{
aout_owner_t *owner = aout_owner (aout);
aout_lock (aout);
date_Set (&owner->sync.date, VLC_TS_INVALID);
aout_OutputFlush (aout, false);
aout_unlock (aout);
}
void aout_DecFlush (audio_output_t *aout)
{
aout_owner_t *owner = aout_owner (aout);
aout_OutputLock (aout);
owner->sync.end = VLC_TS_INVALID;
if (owner->mixer_format.i_format)
aout_OutputFlush (aout, false);
aout_OutputUnlock (aout);
}
bool aout_DecIsEmpty (audio_output_t *aout)
{
aout_owner_t *owner = aout_owner (aout);
mtime_t now = mdate ();
bool empty = true;
aout_OutputLock (aout);
if (owner->sync.end != VLC_TS_INVALID)
empty = owner->sync.end <= now;
if (empty && owner->mixer_format.i_format)
/* The last PTS has elapsed already. So the underlying audio output
* buffer should be empty or almost. Thus draining should be fast
* and will not block the caller too long. */
aout_OutputFlush (aout, true);
aout_OutputUnlock (aout);
return empty;
}
bool aout_DecIsEmpty (audio_output_t *aout)
{
aout_owner_t *owner = aout_owner (aout);
mtime_t end_date, now = mdate ();
bool empty;
aout_lock (aout);
end_date = date_Get (&owner->sync.date);
empty = end_date == VLC_TS_INVALID || end_date <= now;
if (empty)
/* The last PTS has elapsed already. So the underlying audio output
* buffer should be empty or almost. Thus draining should be fast
* and will not block the caller too long. */
aout_OutputFlush (aout, true);
aout_unlock (aout);
return empty;
}
static void aout_DecSynchronize (audio_output_t *aout, mtime_t dec_pts,
int input_rate)
{
aout_owner_t *owner = aout_owner (aout);
mtime_t drift;
/**
* Depending on the drift between the actual and intended playback times,
* the audio core may ignore the drift, trigger upsampling or downsampling,
* insert silence or even discard samples.
* Future VLC versions may instead adjust the input rate.
*
* The audio output plugin is responsible for estimating its actual
* playback time, or rather the estimated time when the next sample will
* be played. (The actual playback time is always the current time, that is
* to say mdate(). It is not an useful statistic.)
*
* Most audio output plugins can estimate the delay until playback of
* the next sample to be written to the buffer, or equally the time until
* all samples in the buffer will have been played. Then:
* pts = mdate() + delay
*/
if (aout_OutputTimeGet (aout, &drift) != 0)
return; /* nothing can be done if timing is unknown */
drift += mdate () - dec_pts;
/* Late audio output.
* This can happen due to insufficient caching, scheduling jitter
* or bug in the decoder. Ideally, the output would seek backward. But that
* is not portable, not supported by some hardware and often unsafe/buggy
* where supported. The other alternative is to flush the buffers
* completely. */
if (drift > (owner->sync.discontinuity ? 0
: +3 * input_rate * AOUT_MAX_PTS_DELAY / INPUT_RATE_DEFAULT))
{
if (!owner->sync.discontinuity)
msg_Warn (aout, "playback way too late (%"PRId64"): "
"flushing buffers", drift);
else
msg_Dbg (aout, "playback too late (%"PRId64"): "
"flushing buffers", drift);
aout_OutputFlush (aout, false);
aout_StopResampling (aout);
owner->sync.end = VLC_TS_INVALID;
owner->sync.discontinuity = true;
/* Now the output might be too early... Recheck. */
if (aout_OutputTimeGet (aout, &drift) != 0)
return; /* nothing can be done if timing is unknown */
drift += mdate () - dec_pts;
}
/* Early audio output.
* This is rare except at startup when the buffers are still empty. */
if (drift < (owner->sync.discontinuity ? 0
: -3 * input_rate * AOUT_MAX_PTS_ADVANCE / INPUT_RATE_DEFAULT))
{
if (!owner->sync.discontinuity)
msg_Warn (aout, "playback way too early (%"PRId64"): "
"playing silence", drift);
aout_DecSilence (aout, -drift, dec_pts);
aout_StopResampling (aout);
owner->sync.discontinuity = true;
drift = 0;
}
if (!aout_FiltersCanResample(owner->filters))
return;
/* Resampling */
if (drift > +AOUT_MAX_PTS_DELAY
&& owner->sync.resamp_type != AOUT_RESAMPLING_UP)
{
msg_Warn (aout, "playback too late (%"PRId64"): up-sampling",
drift);
owner->sync.resamp_type = AOUT_RESAMPLING_UP;
owner->sync.resamp_start_drift = +drift;
}
if (drift < -AOUT_MAX_PTS_ADVANCE
&& owner->sync.resamp_type != AOUT_RESAMPLING_DOWN)
{
msg_Warn (aout, "playback too early (%"PRId64"): down-sampling",
drift);
owner->sync.resamp_type = AOUT_RESAMPLING_DOWN;
owner->sync.resamp_start_drift = -drift;
}
if (owner->sync.resamp_type == AOUT_RESAMPLING_NONE)
return; /* Everything is fine. Nothing to do. */
if (llabs (drift) > 2 * owner->sync.resamp_start_drift)
{ /* If the drift is ever increasing, then something is seriously wrong.
* Cease resampling and hope for the best. */
msg_Warn (aout, "timing screwed (drift: %"PRId64" us): "
"stopping resampling", drift);
aout_StopResampling (aout);
return;
}
/* Resampling has been triggered earlier. This checks if it needs to be
* increased or decreased. Resampling rate changes must be kept slow for
* the comfort of listeners. */
int adj = (owner->sync.resamp_type == AOUT_RESAMPLING_UP) ? +2 : -2;
if (2 * llabs (drift) <= owner->sync.resamp_start_drift)
/* If the drift has been reduced from more than half its initial
* value, then it is time to switch back the resampling direction. */
adj *= -1;
if (!aout_FiltersAdjustResampling (owner->filters, adj))
{ /* Everything is back to normal: stop resampling. */
owner->sync.resamp_type = AOUT_RESAMPLING_NONE;
msg_Dbg (aout, "resampling stopped (drift: %"PRId64" us)", drift);
}
}
