/* $Id: sync.c,v 1.38 2005/04/14 21:57:58 titer Exp $
This file is part of the HandBrake source code.
- Homepage: <http://handbrake.m0k.org/>.
+ Homepage: <http://handbrake.fr/>.
It may be used under the terms of the GNU General Public License. */
#include "hb.h"
-
+#include "hbffmpeg.h"
+#include <stdio.h>
#include "samplerate.h"
-#include "ffmpeg/avcodec.h"
#ifdef INT64_MIN
#undef INT64_MIN /* Because it isn't defined correctly in Zeta */
int64_t next_start; /* start time of next output frame */
int64_t next_pts; /* start time of next input frame */
- int64_t start_silence; /* if we're inserting silence, the time we started */
int64_t first_drop; /* PTS of first 'went backwards' frame dropped */
int drop_count; /* count of 'time went backwards' drops */
- int inserting_silence;
/* Raw */
SRC_STATE * state;
struct hb_work_private_s
{
hb_job_t * job;
- int done;
-
+ int busy; // bitmask with one bit for each active input
+ // (bit 0 = video; 1 = audio 0, 2 = audio 1, ...
+ // appropriate bit is cleared when input gets
+ // an eof buf. syncWork returns done when all
+ // bits are clear.
/* Video */
hb_subtitle_t * subtitle;
int64_t pts_offset;
int64_t next_pts; /* start time of next input frame */
int64_t first_drop; /* PTS of first 'went backwards' frame dropped */
int drop_count; /* count of 'time went backwards' drops */
+ int drops; /* frames dropped to make a cbr video stream */
+ int dups; /* frames duplicated to make a cbr video stream */
int video_sequence;
int count_frames;
int count_frames_max;
* Local prototypes
**********************************************************************/
static void InitAudio( hb_work_object_t * w, int i );
-static int SyncVideo( hb_work_object_t * w );
+static void SyncVideo( hb_work_object_t * w );
static void SyncAudio( hb_work_object_t * w, int i );
-static int NeedSilence( hb_work_object_t * w, hb_audio_t *, int i );
static void InsertSilence( hb_work_object_t * w, int i, int64_t d );
static void UpdateState( hb_work_object_t * w );
pv->job = job;
pv->pts_offset = INT64_MIN;
- pv->count_frames = 0;
/* Calculate how many video frames we are expecting */
- duration = 0;
- for( i = job->chapter_start; i <= job->chapter_end; i++ )
+ if (job->pts_to_stop)
{
- chapter = hb_list_item( title->list_chapter, i - 1 );
- duration += chapter->duration;
+ duration = job->pts_to_stop + 90000;
}
- duration += 90000;
+ else
+ {
+ duration = 0;
+ for( i = job->chapter_start; i <= job->chapter_end; i++ )
+ {
+ chapter = hb_list_item( title->list_chapter, i - 1 );
+ duration += chapter->duration;
+ }
+ duration += 90000;
/* 1 second safety so we're sure we won't miss anything */
+ }
pv->count_frames_max = duration * job->vrate / job->vrate_base / 90000;
hb_log( "sync: expecting %d video frames", pv->count_frames_max );
+ pv->busy |= 1;
/* Initialize libsamplerate for every audio track we have */
- for( i = 0; i < hb_list_count( title->list_audio ); i++ )
+ if ( ! job->indepth_scan )
{
- InitAudio( w, i );
+ for( i = 0; i < hb_list_count( title->list_audio ); i++ )
+ {
+ pv->busy |= ( 1 << (i + 1) );
+ InitAudio( w, i );
+ }
}
/* Get subtitle info, if any */
pv->subtitle = hb_list_item( title->list_subtitle, 0 );
- pv->video_sequence = 0;
-
return 0;
}
hb_job_t * job = pv->job;
hb_title_t * title = job->title;
hb_audio_t * audio = NULL;
-
int i;
- if( pv->cur ) hb_buffer_close( &pv->cur );
+ if( pv->cur )
+ {
+ hb_buffer_close( &pv->cur );
+ }
+
+ hb_log( "sync: got %d frames, %d expected",
+ pv->count_frames, pv->count_frames_max );
- for( i = 0; i < hb_list_count( title->list_audio ); i++ )
+ if (pv->drops || pv->dups )
{
- if ( pv->sync_audio[i].start_silence )
- {
- hb_log( "sync: added %d ms of silence to audio %d",
- (int)((pv->sync_audio[i].next_pts -
- pv->sync_audio[i].start_silence) / 90), i );
- }
+ hb_log( "sync: %d frames dropped, %d duplicated", pv->drops, pv->dups );
+ }
+ for( i = 0; i < hb_list_count( title->list_audio ); i++ )
+ {
audio = hb_list_item( title->list_audio, i );
if( audio->config.out.codec == HB_ACODEC_AC3 )
{
hb_work_private_t * pv = w->private_data;
int i;
- /* If we ever got a video frame, handle audio now */
- if( pv->pts_offset != INT64_MIN )
+ if ( pv->busy & 1 )
+ SyncVideo( w );
+
+ for( i = 0; i < hb_list_count( pv->job->title->list_audio ); i++ )
{
- for( i = 0; i < hb_list_count( pv->job->title->list_audio ); i++ )
- {
+ if ( pv->busy & ( 1 << (i + 1) ) )
SyncAudio( w, i );
- }
}
- /* Handle video */
- return SyncVideo( w );
+ return ( pv->busy? HB_WORK_OK : HB_WORK_DONE );
}
hb_work_object_t hb_sync =
c->sample_rate = sync->audio->config.in.samplerate;
c->channels = HB_INPUT_CH_LAYOUT_GET_DISCRETE_COUNT( sync->audio->config.in.channel_layout );
- if( avcodec_open( c, codec ) < 0 )
+ if( hb_avcodec_open( c, codec ) < 0 )
{
hb_log( "sync: avcodec_open failed" );
return;
}
free( zeros );
- avcodec_close( c );
+ hb_avcodec_close( c );
av_free( c );
}
else
{
/* Initialize libsamplerate */
int error;
- sync->state = src_new( SRC_LINEAR, HB_AMIXDOWN_GET_DISCRETE_CHANNEL_COUNT(sync->audio->config.out.mixdown), &error );
+ sync->state = src_new( SRC_SINC_MEDIUM_QUALITY, HB_AMIXDOWN_GET_DISCRETE_CHANNEL_COUNT(sync->audio->config.out.mixdown), &error );
sync->data.end_of_input = 0;
}
}
***********************************************************************
*
**********************************************************************/
-static int SyncVideo( hb_work_object_t * w )
+static void SyncVideo( hb_work_object_t * w )
{
hb_work_private_t * pv = w->private_data;
hb_buffer_t * cur, * next, * sub = NULL;
hb_job_t * job = pv->job;
- if( pv->done )
- {
- return HB_WORK_DONE;
- }
-
- if( hb_thread_has_exited( job->reader ) &&
- !hb_fifo_size( job->fifo_mpeg2 ) &&
- !hb_fifo_size( job->fifo_raw ) )
- {
- /* All video data has been processed already, we won't get
- more */
- hb_log( "sync: got %d frames, %d expected",
- pv->count_frames, pv->count_frames_max );
- pv->done = 1;
-
- hb_buffer_t * buf_tmp;
-
- // Drop an empty buffer into our output to ensure that things
- // get flushed all the way out.
- buf_tmp = hb_buffer_init(0); // Empty end buffer
- hb_fifo_push( job->fifo_sync, buf_tmp );
-
- return HB_WORK_DONE;
- }
-
if( !pv->cur && !( pv->cur = hb_fifo_get( job->fifo_raw ) ) )
{
/* We haven't even got a frame yet */
- return HB_WORK_OK;
+ return;
}
cur = pv->cur;
+ if( cur->size == 0 )
+ {
+ /* we got an end-of-stream. Feed it downstream & signal that we're done. */
+ hb_fifo_push( job->fifo_sync, hb_buffer_init( 0 ) );
+ pv->busy &=~ 1;
+ return;
+ }
/* At this point we have a frame to process. Let's check
1) if we will be able to push into the fifo ahead
{
hb_buffer_t * buf_tmp;
+ if( next->size == 0 )
+ {
+ /* we got an end-of-stream. Feed it downstream & signal that
+ * we're done. Note that this means we drop the final frame of
+ * video (we don't know its duration). On DVDs the final frame
+ * is often strange and dropping it seems to be a good idea. */
+ hb_fifo_push( job->fifo_sync, hb_buffer_init( 0 ) );
+ pv->busy &=~ 1;
+ return;
+ }
if( pv->pts_offset == INT64_MIN )
{
/* This is our first frame */
}
}
+ if( cur->new_chap ) {
+ hb_log("sync got new chapter %d", cur->new_chap );
+ }
+
/*
* since the first frame is always 0 and the upstream reader code
* is taking care of adjusting for pts discontinuities, we just have
* can deal with overlaps of up to a frame time but anything larger
* we handle by dropping frames here.
*/
- if ( (int64_t)( next->start - pv->next_pts ) <= 0 )
+ if ( (int64_t)( next->start - cur->start ) <= 0 )
{
if ( pv->first_drop == 0 )
{
if ( pv->first_drop )
{
hb_log( "sync: video time didn't advance - dropped %d frames "
- "(delta %d ms, current %lld, next %lld)",
- pv->drop_count, (int)( pv->next_pts - pv->first_drop ) / 90,
- pv->next_pts, pv->first_drop );
+ "(delta %d ms, current %lld, next %lld, dur %d)",
+ pv->drop_count, (int)( cur->start - pv->first_drop ) / 90,
+ cur->start, next->start, (int)( next->start - cur->start ) );
pv->first_drop = 0;
pv->drop_count = 0;
}
}
}
- /*
- * Adjust the pts of the current frame so that it's contiguous
- * with the previous frame. The start time of the current frame
- * has to be the end time of the previous frame and the stop
- * time has to be the start of the next frame. We don't
- * make any adjustments to the source timestamps other than removing
- * the clock offsets (which also removes pts discontinuities).
- * This means we automatically encode at the source's frame rate.
- * MP2 uses an implicit duration (frames end when the next frame
- * starts) but more advanced containers like MP4 use an explicit
- * duration. Since we're looking ahead one frame we set the
- * explicit stop time from the start time of the next frame.
- */
- buf_tmp = cur;
- pv->cur = cur = hb_fifo_get( job->fifo_raw );
- pv->next_pts = next->start;
- int64_t duration = next->start - buf_tmp->start;
- if ( duration <= 0 )
+ int64_t duration;
+ if ( job->mux & HB_MUX_AVI || job->cfr )
{
- hb_log( "sync: invalid video duration %lld, start %lld, next %lld",
- duration, buf_tmp->start, next->start );
+ /*
+ * The concept of variable frame rate video was a bit too advanced
+ * for Microsoft so AVI doesn't support it. Since almost all dvd
+ * video is VFR we have to convert it to constant frame rate to
+ * put it in an AVI container. So here we duplicate, drop and
+ * otherwise trash video frames to appease the gods of Redmond.
+ */
+
+ /* mpeg durations are exact when expressed in ticks of the
+ * 27MHz System clock but not in HB's 90KHz PTS clock. To avoid
+ * a truncation bias that will eventually cause the audio to desync
+ * we compute the duration of the next frame using 27MHz ticks
+ * then truncate it to 90KHz. */
+ duration = ( (int64_t)(pv->count_frames + 1 ) * job->vrate_base ) / 300 -
+ pv->next_start;
+
+ /* We don't want the input & output clocks to be exactly in phase
+ * otherwise small variations in the time will cause us to think
+ * we're a full frame off & there will be lots of drops and dups.
+ * We offset the input clock by half the duration so it's maximally
+ * out of phase with the output clock. */
+ if( cur->start < pv->next_start - ( duration >> 1 ) )
+ {
+ /* current frame too old - drop it */
+ if ( cur->new_chap )
+ {
+ pv->chap_mark = cur->new_chap;
+ }
+ hb_buffer_close( &cur );
+ pv->cur = cur = hb_fifo_get( job->fifo_raw );
+ pv->next_pts = next->start;
+ ++pv->drops;
+ continue;
+ }
+
+ if( next->start > pv->next_start + duration + ( duration >> 1 ) )
+ {
+ /* next frame too far ahead - dup current frame */
+ buf_tmp = hb_buffer_init( cur->size );
+ hb_buffer_copy_settings( buf_tmp, cur );
+ memcpy( buf_tmp->data, cur->data, cur->size );
+ buf_tmp->sequence = cur->sequence;
+ ++pv->dups;
+ }
+ else
+ {
+ /* this frame in our time window & doesn't need to be duped */
+ buf_tmp = cur;
+ pv->cur = cur = hb_fifo_get( job->fifo_raw );
+ pv->next_pts = next->start;
+ }
+ }
+ else
+ {
+ /*
+ * Adjust the pts of the current frame so that it's contiguous
+ * with the previous frame. The start time of the current frame
+ * has to be the end time of the previous frame and the stop
+ * time has to be the start of the next frame. We don't
+ * make any adjustments to the source timestamps other than removing
+ * the clock offsets (which also removes pts discontinuities).
+ * This means we automatically encode at the source's frame rate.
+ * MP2 uses an implicit duration (frames end when the next frame
+ * starts) but more advanced containers like MP4 use an explicit
+ * duration. Since we're looking ahead one frame we set the
+ * explicit stop time from the start time of the next frame.
+ */
+ buf_tmp = cur;
+ pv->cur = cur = hb_fifo_get( job->fifo_raw );
+ pv->next_pts = cur->start;
+ duration = cur->start - buf_tmp->start;
+ if ( duration <= 0 )
+ {
+ hb_log( "sync: invalid video duration %lld, start %lld, next %lld",
+ duration, buf_tmp->start, next->start );
+ }
}
+
buf_tmp->start = pv->next_start;
pv->next_start += duration;
buf_tmp->stop = pv->next_start;
+
if ( pv->chap_mark )
{
// we have a pending chapter mark from a recent drop - put it on this
/* Make sure we won't get more frames then expected */
if( pv->count_frames >= pv->count_frames_max * 2)
{
- hb_log( "sync: got too many frames (%d), exiting early", pv->count_frames );
- pv->done = 1;
-
- // Drop an empty buffer into our output to ensure that things
- // get flushed all the way out.
- buf_tmp = hb_buffer_init(0); // Empty end buffer
- hb_fifo_push( job->fifo_sync, buf_tmp );
+ hb_log( "sync: got too many frames (%d), exiting early",
+ pv->count_frames );
- break;
+ // Drop an empty buffer into our output to ensure that things
+ // get flushed all the way out.
+ hb_fifo_push( job->fifo_sync, hb_buffer_init( 0 ) );
+ pv->busy &=~ 1;
+ return;
}
}
-
- return HB_WORK_OK;
}
static void OutputAudioFrame( hb_job_t *job, hb_audio_t *audio, hb_buffer_t *buf,
{
int64_t start = sync->next_start;
int64_t duration = buf->stop - buf->start;
- if (duration <= 0 ||
- duration > ( 90000 * AC3_SAMPLES_PER_FRAME ) / audio->config.out.samplerate )
- {
- hb_log("sync: audio %d weird duration %lld, start %lld, stop %lld, next %lld",
- i, duration, buf->start, buf->stop, sync->next_pts);
- if ( duration <= 0 )
- {
- duration = ( 90000 * AC3_SAMPLES_PER_FRAME ) / audio->config.out.samplerate;
- buf->stop = buf->start + duration;
- }
- }
+
sync->next_pts += duration;
- if( /* audio->rate == job->arate || This should work but doesn't */
+ if( audio->config.in.samplerate == audio->config.out.samplerate ||
audio->config.out.codec == HB_ACODEC_AC3 ||
audio->config.out.codec == HB_ACODEC_DCA )
{
/*
- * If we don't have to do sample rate conversion or this audio is AC3
+ * If we don't have to do sample rate conversion or this audio is
* pass-thru just send the input buffer downstream after adjusting
* its timestamps to make the output stream continuous.
*/
sizeof( float );
count_in = buf_raw->size / channel_count;
- count_out = ( buf_raw->stop - buf_raw->start ) * audio->config.out.samplerate / 90000;
+ /*
+ * When using stupid rates like 44.1 there will always be some
+ * truncation error. E.g., a 1536 sample AC3 frame will turn into a
+ * 1536*44.1/48.0 = 1411.2 sample frame. If we just truncate the .2
+ * the error will build up over time and eventually the audio will
+ * substantially lag the video. libsamplerate will keep track of the
+ * fractional sample & give it to us when appropriate if we give it
+ * an extra sample of space in the output buffer.
+ */
+ count_out = ( duration * audio->config.out.samplerate ) / 90000 + 1;
sync->data.input_frames = count_in;
sync->data.output_frames = count_out;
- sync->data.src_ratio = (double)count_out / (double)count_in;
+ sync->data.src_ratio = (double)audio->config.out.samplerate /
+ (double)audio->config.in.samplerate;
buf = hb_buffer_init( count_out * channel_count );
sync->data.data_in = (float *) buf_raw->data;
hb_buffer_close( &buf_raw );
buf->size = sync->data.output_frames_gen * channel_count;
+ duration = ( sync->data.output_frames_gen * 90000 ) /
+ audio->config.out.samplerate;
}
+ buf->frametype = HB_FRAME_AUDIO;
buf->start = start;
buf->stop = start + duration;
- buf->frametype = HB_FRAME_AUDIO;
sync->next_start = start + duration;
hb_fifo_push( fifo, buf );
}
hb_audio_t * audio = sync->audio;
hb_buffer_t * buf;
hb_fifo_t * fifo;
- int rate;
if( audio->config.out.codec == HB_ACODEC_AC3 )
{
fifo = audio->priv.fifo_out;
- rate = audio->config.in.samplerate;
}
else
{
fifo = audio->priv.fifo_sync;
- rate = audio->config.out.samplerate;
}
while( !hb_fifo_is_full( fifo ) && ( buf = hb_fifo_see( audio->priv.fifo_raw ) ) )
{
+ /* if the next buffer is an eof send it downstream */
+ if ( buf->size <= 0 )
+ {
+ buf = hb_fifo_get( audio->priv.fifo_raw );
+ hb_fifo_push( fifo, buf );
+ pv->busy &=~ (1 << (i + 1) );
+ return;
+ }
if ( (int64_t)( buf->start - sync->next_pts ) < 0 )
{
- /*
- * audio time went backwards by more than a frame time (this can
- * happen when we reset the PTS because of lost data).
- * Discard data that's in the past.
- */
- if ( sync->first_drop == 0 )
+ // audio time went backwards.
+ // If our output clock is more than a half frame ahead of the
+ // input clock drop this frame to move closer to sync.
+ // Otherwise drop frames until the input clock matches the output clock.
+ if ( sync->first_drop || sync->next_start - buf->start > 90*15 )
{
- sync->first_drop = buf->start;
+ // Discard data that's in the past.
+ if ( sync->first_drop == 0 )
+ {
+ sync->first_drop = sync->next_pts;
+ }
+ ++sync->drop_count;
+ buf = hb_fifo_get( audio->priv.fifo_raw );
+ hb_buffer_close( &buf );
+ continue;
}
- ++sync->drop_count;
- buf = hb_fifo_get( audio->priv.fifo_raw );
- hb_buffer_close( &buf );
- continue;
+ sync->next_pts = buf->start;
}
if ( sync->first_drop )
{
+ // we were dropping old data but input buf time is now current
hb_log( "sync: audio %d time went backwards %d ms, dropped %d frames "
"(next %lld, current %lld)", i,
(int)( sync->next_pts - sync->first_drop ) / 90,
sync->drop_count, sync->first_drop, sync->next_pts );
sync->first_drop = 0;
sync->drop_count = 0;
+ sync->next_pts = buf->start;
}
-
- if ( sync->inserting_silence && (int64_t)(buf->start - sync->next_pts) > 0 )
+ if ( buf->start - sync->next_pts >= (90 * 70) )
{
- /*
- * if we're within one frame time of the amount of silence
- * we need, insert just what we need otherwise insert a frame time.
- */
- int64_t framedur = buf->stop - buf->start;
- if ( buf->start - sync->next_pts <= framedur )
+ if ( buf->start - sync->next_pts > (90000LL * 60) )
{
- InsertSilence( w, i, buf->start - sync->next_pts );
- sync->inserting_silence = 0;
- }
- else
- {
- InsertSilence( w, i, framedur );
+ // there's a gap of more than a minute between the last
+ // frame and this. assume we got a corrupted timestamp
+ // and just drop the next buf.
+ hb_log( "sync: %d minute time gap in audio %d - dropping buf"
+ " start %lld, next %lld",
+ (int)((buf->start - sync->next_pts) / (90000*60)),
+ i, buf->start, sync->next_pts );
+ buf = hb_fifo_get( audio->priv.fifo_raw );
+ hb_buffer_close( &buf );
+ continue;
}
- continue;
- }
- if ( buf->start - sync->next_pts >= (90 * 100) )
- {
/*
- * there's a gap of at least 100ms between the last
+ * there's a gap of at least 70ms between the last
* frame we processed & the next. Fill it with silence.
*/
- if ( ! sync->inserting_silence )
- {
- hb_log( "sync: adding %d ms of silence to audio %d"
- " start %lld, next %lld",
- (int)((buf->start - sync->next_pts) / 90),
- i, buf->start, sync->next_pts );
- sync->inserting_silence = 1;
- }
- InsertSilence( w, i, buf->stop - buf->start );
- continue;
+ hb_log( "sync: adding %d ms of silence to audio %d"
+ " start %lld, next %lld",
+ (int)((buf->start - sync->next_pts) / 90),
+ i, buf->start, sync->next_pts );
+ InsertSilence( w, i, buf->start - sync->next_pts );
+ return;
}
/*
buf = hb_fifo_get( audio->priv.fifo_raw );
OutputAudioFrame( job, audio, buf, sync, fifo, i );
}
-
- if( NeedSilence( w, audio, i ) )
- {
- InsertSilence( w, i, (90000 * AC3_SAMPLES_PER_FRAME) / sync->audio->config.out.samplerate );
- }
-}
-
-static int NeedSilence( hb_work_object_t * w, hb_audio_t * audio, int i )
-{
- hb_work_private_t * pv = w->private_data;
- hb_job_t * job = pv->job;
- hb_sync_audio_t * sync = &pv->sync_audio[i];
-
- if( hb_fifo_size( audio->priv.fifo_in ) ||
- hb_fifo_size( audio->priv.fifo_raw ) ||
- hb_fifo_size( audio->priv.fifo_sync ) ||
- hb_fifo_size( audio->priv.fifo_out ) )
- {
- /* We have some audio, we are fine */
- return 0;
- }
-
- /* No audio left in fifos */
-
- if( hb_thread_has_exited( job->reader ) )
- {
- /* We might miss some audio to complete encoding and muxing
- the video track */
- if ( sync->start_silence == 0 )
- {
- hb_log("sync: reader has exited, adding silence to audio %d", i);
- sync->start_silence = sync->next_pts;
- }
- return 1;
- }
- return 0;
}
static void InsertSilence( hb_work_object_t * w, int i, int64_t duration )
hb_job_t *job = pv->job;
hb_sync_audio_t *sync = &pv->sync_audio[i];
hb_buffer_t *buf;
+ hb_fifo_t *fifo;
- if( sync->audio->config.out.codec == HB_ACODEC_AC3 )
- {
- buf = hb_buffer_init( sync->ac3_size );
- buf->start = sync->next_pts;
- buf->stop = buf->start + duration;
- memcpy( buf->data, sync->ac3_buf, buf->size );
- OutputAudioFrame( job, sync->audio, buf, sync, sync->audio->priv.fifo_out, i );
- }
- else
+ // to keep pass-thru and regular audio in sync we generate silence in
+ // AC3 frame-sized units. If the silence duration isn't an integer multiple
+ // of the AC3 frame duration we will truncate or round up depending on
+ // which minimizes the timing error.
+ const int frame_dur = ( 90000 * AC3_SAMPLES_PER_FRAME ) /
+ sync->audio->config.in.samplerate;
+ int frame_count = ( duration + (frame_dur >> 1) ) / frame_dur;
+
+ while ( --frame_count >= 0 )
{
- buf = hb_buffer_init( duration * sizeof( float ) *
- HB_AMIXDOWN_GET_DISCRETE_CHANNEL_COUNT(sync->audio->config.out.mixdown) );
- buf->start = sync->next_pts;
- buf->stop = buf->start + duration;
- memset( buf->data, 0, buf->size );
- OutputAudioFrame( job, sync->audio, buf, sync, sync->audio->priv.fifo_sync, i );
+ if( sync->audio->config.out.codec == HB_ACODEC_AC3 )
+ {
+ buf = hb_buffer_init( sync->ac3_size );
+ buf->start = sync->next_pts;
+ buf->stop = buf->start + frame_dur;
+ memcpy( buf->data, sync->ac3_buf, buf->size );
+ fifo = sync->audio->priv.fifo_out;
+ }
+ else
+ {
+ buf = hb_buffer_init( AC3_SAMPLES_PER_FRAME * sizeof( float ) *
+ HB_AMIXDOWN_GET_DISCRETE_CHANNEL_COUNT(
+ sync->audio->config.out.mixdown) );
+ buf->start = sync->next_pts;
+ buf->stop = buf->start + frame_dur;
+ memset( buf->data, 0, buf->size );
+ fifo = sync->audio->priv.fifo_sync;
+ }
+ OutputAudioFrame( job, sync->audio, buf, sync, fifo, i );
}
}
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