goto error;
}
- date_Init (&owner->sync.date, owner->mixer_format.i_rate, 1);
- date_Set (&owner->sync.date, VLC_TS_INVALID);
+ owner->sync.end = VLC_TS_INVALID;
owner->sync.resamp_type = AOUT_RESAMPLING_NONE;
+ owner->sync.discontinuity = true;
aout_unlock( p_aout );
atomic_init (&owner->buffers_lost, 0);
aout_volume_SetFormat (owner->volume, owner->mixer_format.i_format);
}
+ owner->sync.end = VLC_TS_INVALID;
owner->sync.resamp_type = AOUT_RESAMPLING_NONE;
if (aout_FiltersNew (aout, &owner->input_format, &owner->mixer_format,
aout_FiltersAdjustResampling (aout, 0);
}
+static void aout_DecSilence (audio_output_t *aout, mtime_t length)
+{
+ aout_owner_t *owner = aout_owner (aout);
+ const audio_sample_format_t *fmt = &owner->mixer_format;
+ size_t frames = (fmt->i_rate * length) / CLOCK_FREQ;
+ block_t *block;
+
+ if (AOUT_FMT_SPDIF(fmt))
+ block = block_Alloc (4 * frames);
+ else
+ block = block_Alloc (frames * fmt->i_bytes_per_frame);
+ if (unlikely(block == NULL))
+ return; /* uho! */
+
+ msg_Dbg (aout, "inserting %zu zeroes", frames);
+ memset (block->p_buffer, 0, block->i_buffer);
+ block->i_nb_samples = frames;
+ block->i_length = length;
+ /* FIXME: PTS... */
+ aout_OutputPlay (aout, block);
+}
+
+static void aout_DecSynchronize (audio_output_t *aout, mtime_t dec_pts,
+ int input_rate)
+{
+ aout_owner_t *owner = aout_owner (aout);
+ mtime_t aout_pts, drift;
+
+retry:
+ /**
+ * 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, &aout_pts) != 0)
+ return; /* nothing can be done if timing is unknown */
+
+ drift = aout_pts - dec_pts;
+
+ if (drift < (owner->sync.discontinuity ? 0
+ : -3 * input_rate * AOUT_MAX_PTS_ADVANCE / INPUT_RATE_DEFAULT))
+ { /* If the audio output is very early (which is rare other than during
+ * prebuffering), hold with silence. */
+ if (!owner->sync.discontinuity)
+ msg_Err (aout, "playback way too early (%"PRId64"): "
+ "playing silence", drift);
+ aout_StopResampling (aout);
+ aout_DecSilence (aout, -drift);
+ owner->sync.discontinuity = false;
+ drift = 0;
+ }
+ else
+ if (drift > (owner->sync.discontinuity ? 0
+ : +3 * input_rate * AOUT_MAX_PTS_DELAY / INPUT_RATE_DEFAULT))
+ { /* If the audio output is very late, drop the buffers.
+ * This should make some room and advance playback quickly. */
+ if (!owner->sync.discontinuity)
+ msg_Err (aout, "playback way too late (%"PRId64"): "
+ "flushing buffers", drift);
+ aout_StopResampling (aout);
+ owner->sync.end = VLC_TS_INVALID;
+ aout_OutputFlush (aout, false);
+ goto retry; /* may be too early now... retry */
+ }
+
+ if (drift < -AOUT_MAX_PTS_ADVANCE)
+ {
+ if (owner->sync.resamp_type == AOUT_RESAMPLING_NONE)
+ {
+ msg_Warn (aout, "playback too early (%"PRId64"): down-sampling",
+ drift);
+ owner->sync.resamp_start_drift = -drift;
+ }
+ owner->sync.resamp_type = AOUT_RESAMPLING_DOWN;
+ }
+ else
+ if (drift > +AOUT_MAX_PTS_DELAY)
+ {
+ if (owner->sync.resamp_type == AOUT_RESAMPLING_NONE)
+ {
+ msg_Warn (aout, "playback too late (%"PRId64"): up-sampling",
+ drift);
+ owner->sync.resamp_start_drift = +drift;
+ }
+ owner->sync.resamp_type = AOUT_RESAMPLING_UP;
+ }
+
+ if (owner->sync.resamp_type == AOUT_RESAMPLING_NONE)
+ return; /* Everything is fine. Nothing to do. */
+
+ /* 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. */
+ const int adj = (owner->sync.resamp_type == AOUT_RESAMPLING_UP) ? +2 : -2;
+
+ /* Check if everything is back to normal, then stop resampling. */
+ if (!aout_FiltersAdjustResampling (aout, adj))
+ {
+ owner->sync.resamp_type = AOUT_RESAMPLING_NONE;
+ msg_Dbg (aout, "resampling stopped (drift: %"PRId64" us)", drift);
+ }
+ else
+ 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. */
+ if (owner->sync.resamp_type == AOUT_RESAMPLING_UP)
+ owner->sync.resamp_type = AOUT_RESAMPLING_DOWN;
+ else
+ owner->sync.resamp_type = AOUT_RESAMPLING_UP;
+ owner->sync.resamp_start_drift = 0;
+ }
+ else
+ 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_Err (aout, "timing screwed (drift: %"PRId64" us): "
+ "stopping resampling", drift);
+ aout_StopResampling (aout);
+ }
+}
+
/*****************************************************************************
* aout_DecPlay : filter & mix the decoded buffer
*****************************************************************************/
if (unlikely(aout_CheckRestart (aout)))
goto drop; /* Pipeline is unrecoverably broken :-( */
- /* We don't care if someone changes the start date behind our back after
- * this. We'll deal with that when pushing the buffer, and compensate
- * with the next incoming buffer. */
- mtime_t start_date = date_Get (&owner->sync.date);
- const mtime_t now = mdate ();
-
- if (start_date != VLC_TS_INVALID && start_date < now)
- { /* The decoder is _very_ late. This can only happen if the user
- * pauses the stream (or if the decoder is buggy, which cannot
- * happen :). */
- msg_Warn (aout, "computed PTS is out of range (%"PRId64"), "
- "clearing out", now - start_date);
- aout_OutputFlush (aout, false);
- if (owner->sync.resamp_type != AOUT_RESAMPLING_NONE)
- msg_Warn (aout, "timing screwed, stopping resampling");
- aout_StopResampling (aout);
- block->i_flags |= BLOCK_FLAG_DISCONTINUITY;
- start_date = VLC_TS_INVALID;
- }
-
- if (block->i_pts < now + AOUT_MIN_PREPARE_TIME)
- { /* The decoder gives us f*cked up PTS. It's its business, but we
- * can't present it anyway, so drop the buffer. */
- msg_Warn (aout, "PTS is out of range (%"PRId64"), dropping buffer",
- now - block->i_pts);
- aout_StopResampling (aout);
+ const mtime_t now = mdate (), advance = block->i_pts - now;
+ if (advance < AOUT_MIN_PREPARE_TIME)
+ { /* Late buffer can be caused by bugs in the decoder, by scheduling
+ * latency spikes (excessive load, SIGSTOP, etc.) or if buffering is
+ * insufficient. We assume the PTS is wrong and play the buffer anyway:
+ * Hopefully video has encountered a similar PTS problem as audio. */
+ msg_Warn (aout, "buffer too late (%"PRId64" us): dropped", advance);
goto drop;
}
-
- /* If the audio drift is too big then it's not worth trying to resample
- * the audio. */
- if (start_date == VLC_TS_INVALID)
- {
- start_date = block->i_pts;
- date_Set (&owner->sync.date, start_date);
- }
-
- mtime_t drift = start_date - block->i_pts;
- if (drift < -input_rate * 3 * AOUT_MAX_PTS_ADVANCE / INPUT_RATE_DEFAULT)
- {
- msg_Warn (aout, "buffer way too early (%"PRId64"), clearing queue",
- drift);
- aout_OutputFlush (aout, false);
- if (owner->sync.resamp_type != AOUT_RESAMPLING_NONE)
- msg_Warn (aout, "timing screwed, stopping resampling");
- aout_StopResampling (aout);
- block->i_flags |= BLOCK_FLAG_DISCONTINUITY;
- start_date = block->i_pts;
- date_Set (&owner->sync.date, start_date);
- drift = 0;
- }
- else
- if (drift > +input_rate * 3 * AOUT_MAX_PTS_DELAY / INPUT_RATE_DEFAULT)
- {
- msg_Warn (aout, "buffer way too late (%"PRId64"), dropping buffer",
- drift);
+ if (advance > AOUT_MAX_ADVANCE_TIME)
+ { /* Early buffers can only be caused by bugs in the decoder. */
+ msg_Err (aout, "buffer too early (%"PRId64" us): dropped", advance);
goto drop;
}
block = aout_FiltersPlay (aout, block, input_rate);
if (block == NULL)
- {
- atomic_fetch_add(&owner->buffers_lost, 1);
- goto out;
- }
-
- /* Adjust the resampler if needed.
- * We first need to calculate the output rate of this resampler. */
- if ((owner->sync.resamp_type == AOUT_RESAMPLING_NONE)
- && (drift < -AOUT_MAX_PTS_ADVANCE || drift > +AOUT_MAX_PTS_DELAY))
- { /* Can happen in several circumstances :
- * 1. A problem at the input (clock drift)
- * 2. A small pause triggered by the user
- * 3. Some delay in the output stage, causing a loss of lip
- * synchronization
- * Solution : resample the buffer to avoid a scratch.
- */
- owner->sync.resamp_start_drift = (int)-drift;
- owner->sync.resamp_type = (drift < 0) ? AOUT_RESAMPLING_DOWN
- : AOUT_RESAMPLING_UP;
- msg_Warn (aout, (drift < 0)
- ? "buffer too early (%"PRId64"), down-sampling"
- : "buffer too late (%"PRId64"), up-sampling", drift);
- }
- if (owner->sync.resamp_type != AOUT_RESAMPLING_NONE)
- { /* Resampling has been triggered previously (because of dates
- * mismatch). We want the resampling to happen progressively so
- * it isn't too audible to the listener. */
- const int adjust = (owner->sync.resamp_type == AOUT_RESAMPLING_UP)
- ? +2 : -2;
- /* Check if everything is back to normal, then stop resampling. */
- if (!aout_FiltersAdjustResampling (aout, adjust))
- {
- owner->sync.resamp_type = AOUT_RESAMPLING_NONE;
- msg_Warn (aout, "resampling stopped (drift: %"PRIi64")",
- block->i_pts - start_date);
- }
- else if (abs ((int)(block->i_pts - start_date))
- < abs (owner->sync.resamp_start_drift) / 2)
- { /* If we reduced the drift from half, then it is time to switch
- * back the resampling direction. */
- if (owner->sync.resamp_type == AOUT_RESAMPLING_UP)
- owner->sync.resamp_type = AOUT_RESAMPLING_DOWN;
- else
- owner->sync.resamp_type = AOUT_RESAMPLING_UP;
- owner->sync.resamp_start_drift = 0;
- }
- else if (owner->sync.resamp_start_drift
- && (abs ((int)(block->i_pts - start_date))
- > abs (owner->sync.resamp_start_drift) * 3 / 2))
- { /* If the drift is increasing and not decreasing, than something
- * is bad. We'd better stop the resampling right now. */
- msg_Warn (aout, "timing screwed, stopping resampling");
- aout_StopResampling (aout);
- block->i_flags |= BLOCK_FLAG_DISCONTINUITY;
- }
- }
-
- block->i_pts = start_date;
- date_Increment (&owner->sync.date, block->i_nb_samples);
+ goto lost;
/* Software volume */
aout_volume_Amplify (owner->volume, block);
+ /* Drift correction */
+ aout_DecSynchronize (aout, block->i_pts, input_rate);
+
/* Output */
+ owner->sync.end = block->i_pts + block->i_length + 1;
aout_OutputPlay (aout, block);
out:
aout_unlock (aout);
return 0;
drop:
block_Release (block);
+lost:
atomic_fetch_add(&owner->buffers_lost, 1);
goto out;
}
aout_owner_t *owner = aout_owner (aout);
aout_lock (aout);
- /* XXX: Should the date be offset by the pause duration instead? */
- date_Set (&owner->sync.date, VLC_TS_INVALID);
+ if (owner->sync.end != VLC_TS_INVALID)
+ {
+ if (paused)
+ owner->sync.end -= date;
+ else
+ owner->sync.end += date;
+ }
aout_OutputPause (aout, paused, date);
aout_unlock (aout);
}
aout_owner_t *owner = aout_owner (aout);
aout_lock (aout);
- date_Set (&owner->sync.date, VLC_TS_INVALID);
+ owner->sync.end = VLC_TS_INVALID;
aout_OutputFlush (aout, false);
aout_unlock (aout);
}
bool aout_DecIsEmpty (audio_output_t *aout)
{
aout_owner_t *owner = aout_owner (aout);
- mtime_t end_date, now = mdate ();
- bool empty;
+ mtime_t now = mdate ();
+ bool empty = true;
aout_lock (aout);
- end_date = date_Get (&owner->sync.date);
- empty = end_date == VLC_TS_INVALID || end_date <= now;
+ if (owner->sync.end != VLC_TS_INVALID)
+ empty = owner->sync.end <= 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
projects / vlc / commitdiff