1 /* $Id: sync.c,v 1.38 2005/04/14 21:57:58 titer Exp $
3 This file is part of the HandBrake source code.
4 Homepage: <http://handbrake.fr/>.
5 It may be used under the terms of the GNU General Public License. */
10 #include "samplerate.h"
13 #undef INT64_MIN /* Because it isn't defined correctly in Zeta */
15 #define INT64_MIN (-9223372036854775807LL-1)
17 #define AC3_SAMPLES_PER_FRAME 1536
23 int64_t next_start; /* start time of next output frame */
24 int64_t next_pts; /* start time of next input frame */
25 int64_t first_drop; /* PTS of first 'went backwards' frame dropped */
26 int drop_count; /* count of 'time went backwards' drops */
38 struct hb_work_private_s
41 int busy; // bitmask with one bit for each active input
42 // (bit 0 = video; 1 = audio 0, 2 = audio 1, ...
43 // appropriate bit is cleared when input gets
44 // an eof buf. syncWork returns done when all
48 int64_t next_start; /* start time of next output frame */
49 int64_t next_pts; /* start time of next input frame */
50 int64_t first_drop; /* PTS of first 'went backwards' frame dropped */
51 int drop_count; /* count of 'time went backwards' drops */
52 int drops; /* frames dropped to make a cbr video stream */
53 int dups; /* frames duplicated to make a cbr video stream */
57 int chap_mark; /* to propagate chapter mark across a drop */
58 hb_buffer_t * cur; /* The next picture to process */
61 hb_sync_audio_t sync_audio[8];
62 int64_t audio_passthru_slip;
65 uint64_t st_counts[4];
70 /***********************************************************************
72 **********************************************************************/
73 static void InitAudio( hb_work_object_t * w, int i );
74 static void SyncVideo( hb_work_object_t * w );
75 static void SyncAudio( hb_work_object_t * w, int i );
76 static void InsertSilence( hb_work_object_t * w, int i, int64_t d );
77 static void UpdateState( hb_work_object_t * w );
79 /***********************************************************************
81 ***********************************************************************
82 * Initialize the work object
83 **********************************************************************/
84 int syncInit( hb_work_object_t * w, hb_job_t * job )
86 hb_title_t * title = job->title;
87 hb_chapter_t * chapter;
90 hb_work_private_t * pv;
92 pv = calloc( 1, sizeof( hb_work_private_t ) );
96 pv->pts_offset = INT64_MIN;
98 /* Calculate how many video frames we are expecting */
101 duration = job->pts_to_stop + 90000;
103 else if( job->frame_to_stop )
105 /* Set the duration to a rough estimate */
106 duration = ( job->frame_to_stop / ( job->vrate / job->vrate_base ) ) * 90000;
111 for( i = job->chapter_start; i <= job->chapter_end; i++ )
113 chapter = hb_list_item( title->list_chapter, i - 1 );
114 duration += chapter->duration;
117 /* 1 second safety so we're sure we won't miss anything */
119 pv->count_frames_max = duration * job->vrate / job->vrate_base / 90000;
121 hb_log( "sync: expecting %d video frames", pv->count_frames_max );
124 /* Initialize libsamplerate for every audio track we have */
125 if ( ! job->indepth_scan )
127 for( i = 0; i < hb_list_count( title->list_audio ) && i < 8; i++ )
129 pv->busy |= ( 1 << (i + 1) );
137 /***********************************************************************
139 ***********************************************************************
141 **********************************************************************/
142 void syncClose( hb_work_object_t * w )
144 hb_work_private_t * pv = w->private_data;
145 hb_job_t * job = pv->job;
146 hb_title_t * title = job->title;
147 hb_audio_t * audio = NULL;
152 hb_buffer_close( &pv->cur );
155 hb_log( "sync: got %d frames, %d expected",
156 pv->count_frames, pv->count_frames_max );
158 if (pv->drops || pv->dups )
160 hb_log( "sync: %d frames dropped, %d duplicated", pv->drops, pv->dups );
163 for( i = 0; i < hb_list_count( title->list_audio ); i++ )
165 audio = hb_list_item( title->list_audio, i );
166 if( audio->config.out.codec == HB_ACODEC_AC3 )
168 free( pv->sync_audio[i].ac3_buf );
172 src_delete( pv->sync_audio[i].state );
177 w->private_data = NULL;
180 /***********************************************************************
182 ***********************************************************************
183 * The root routine of this work abject
185 * The way this works is that we are syncing the audio to the PTS of
186 * the last video that we processed. That's why we skip the audio sync
187 * if we haven't got a valid PTS from the video yet.
189 **********************************************************************/
190 int syncWork( hb_work_object_t * w, hb_buffer_t ** unused1,
191 hb_buffer_t ** unused2 )
193 hb_work_private_t * pv = w->private_data;
199 for( i = 0; i < hb_list_count( pv->job->title->list_audio ); i++ )
201 if ( pv->busy & ( 1 << (i + 1) ) )
205 return ( pv->busy? HB_WORK_OK : HB_WORK_DONE );
208 hb_work_object_t hb_sync =
217 static void InitAudio( hb_work_object_t * w, int i )
219 hb_work_private_t * pv = w->private_data;
220 hb_job_t * job = pv->job;
221 hb_title_t * title = job->title;
222 hb_sync_audio_t * sync;
224 sync = &pv->sync_audio[i];
225 sync->audio = hb_list_item( title->list_audio, i );
227 if( sync->audio->config.out.codec == HB_ACODEC_AC3 )
229 /* Have a silent AC-3 frame ready in case we have to fill a
235 codec = avcodec_find_encoder( CODEC_ID_AC3 );
236 c = avcodec_alloc_context();
238 c->bit_rate = sync->audio->config.in.bitrate;
239 c->sample_rate = sync->audio->config.in.samplerate;
240 c->channels = HB_INPUT_CH_LAYOUT_GET_DISCRETE_COUNT( sync->audio->config.in.channel_layout );
242 if( hb_avcodec_open( c, codec ) < 0 )
244 hb_log( "sync: avcodec_open failed" );
248 zeros = calloc( AC3_SAMPLES_PER_FRAME *
249 sizeof( short ) * c->channels, 1 );
250 sync->ac3_size = sync->audio->config.in.bitrate * AC3_SAMPLES_PER_FRAME /
251 sync->audio->config.in.samplerate / 8;
252 sync->ac3_buf = malloc( sync->ac3_size );
254 if( avcodec_encode_audio( c, sync->ac3_buf, sync->ac3_size,
255 zeros ) != sync->ac3_size )
257 hb_log( "sync: avcodec_encode_audio failed" );
261 hb_avcodec_close( c );
266 /* Initialize libsamplerate */
268 sync->state = src_new( SRC_SINC_MEDIUM_QUALITY, HB_AMIXDOWN_GET_DISCRETE_CHANNEL_COUNT(sync->audio->config.out.mixdown), &error );
269 sync->data.end_of_input = 0;
273 /***********************************************************************
275 ***********************************************************************
277 **********************************************************************/
278 static void SyncVideo( hb_work_object_t * w )
280 hb_work_private_t * pv = w->private_data;
281 hb_buffer_t * cur, * next, * sub = NULL;
282 hb_job_t * job = pv->job;
283 hb_subtitle_t *subtitle;
286 if( !pv->cur && !( pv->cur = hb_fifo_get( job->fifo_raw ) ) )
288 /* We haven't even got a frame yet */
294 /* we got an end-of-stream. Feed it downstream & signal that we're done. */
295 hb_fifo_push( job->fifo_sync, hb_buffer_init( 0 ) );
300 /* At this point we have a frame to process. Let's check
301 1) if we will be able to push into the fifo ahead
302 2) if the next frame is there already, since we need it to
303 compute the duration of the current frame*/
304 while( !hb_fifo_is_full( job->fifo_sync ) &&
305 ( next = hb_fifo_see( job->fifo_raw ) ) )
307 hb_buffer_t * buf_tmp;
309 if( next->size == 0 )
311 /* we got an end-of-stream. Feed it downstream & signal that
312 * we're done. Note that this means we drop the final frame of
313 * video (we don't know its duration). On DVDs the final frame
314 * is often strange and dropping it seems to be a good idea. */
315 hb_fifo_push( job->fifo_sync, hb_buffer_init( 0 ) );
319 if( pv->pts_offset == INT64_MIN )
321 /* This is our first frame */
323 if ( cur->start != 0 )
326 * The first pts from a dvd should always be zero but
327 * can be non-zero with a transport or program stream since
328 * we're not guaranteed to start on an IDR frame. If we get
329 * a non-zero initial PTS extend its duration so it behaves
330 * as if it started at zero so that our audio timing will
333 hb_log( "sync: first pts is %lld", cur->start );
338 if( cur->new_chap ) {
339 hb_log("sync got new chapter %d", cur->new_chap );
343 * since the first frame is always 0 and the upstream reader code
344 * is taking care of adjusting for pts discontinuities, we just have
345 * to deal with the next frame's start being in the past. This can
346 * happen when the PTS is adjusted after data loss but video frame
347 * reordering causes some frames with the old clock to appear after
348 * the clock change. This creates frames that overlap in time which
349 * looks to us like time going backward. The downstream muxing code
350 * can deal with overlaps of up to a frame time but anything larger
351 * we handle by dropping frames here.
353 if ( (int64_t)( next->start - cur->start ) <= 0 ||
354 (int64_t)( (cur->start - pv->audio_passthru_slip ) - pv->next_pts ) < 0 )
356 if ( pv->first_drop == 0 )
358 pv->first_drop = next->start;
361 buf_tmp = hb_fifo_get( job->fifo_raw );
362 if ( buf_tmp->new_chap )
364 // don't drop a chapter mark when we drop the buffer
365 pv->chap_mark = buf_tmp->new_chap;
367 hb_buffer_close( &buf_tmp );
370 if ( pv->first_drop )
372 hb_log( "sync: video time didn't advance - dropped %d frames "
373 "(delta %d ms, current %lld, next %lld, dur %d)",
374 pv->drop_count, (int)( cur->start - pv->first_drop ) / 90,
375 cur->start, next->start, (int)( next->start - cur->start ) );
381 * Track the video sequence number localy so that we can sync the audio
382 * to it using the sequence number as well as the PTS.
384 pv->video_sequence = cur->sequence;
387 * Look for a subtitle for this frame.
389 * If found then it will be tagged onto a video buffer of the correct time and
390 * sent in to the render pipeline. This only needs to be done for VOBSUBs which
391 * get rendered, other types of subtitles can just sit in their raw_queue until
392 * delt with at muxing.
394 for( i = 0; i < hb_list_count( job->list_subtitle ); i++)
396 subtitle = hb_list_item( job->list_subtitle, i );
399 * Rewrite timestamps on subtitles that need it (on raw queue).
401 if( subtitle->source == CCSUB )
404 * Rewrite timestamps on subtitles that came from Closed Captions
405 * since they are using the MPEG2 timestamps.
407 while( ( sub = hb_fifo_see( subtitle->fifo_raw ) ) )
410 * Rewrite the timestamps as and when the video
411 * (cur->start) reaches the same timestamp as a
412 * closed caption (sub->start).
414 * What about discontinuity boundaries - not delt
417 * Bypass the sync fifo altogether.
421 sub = hb_fifo_get( subtitle->fifo_raw );
422 hb_fifo_push( subtitle->fifo_out, sub );
426 if( sub->start < cur->start )
429 duration = sub->stop - sub->start;
430 sub = hb_fifo_get( subtitle->fifo_raw );
431 sub->start = pv->next_start;
432 sub->stop = sub->start + duration;
433 hb_fifo_push( subtitle->fifo_out, sub );
442 if( subtitle->source == VOBSUB )
445 while( ( sub = hb_fifo_see( subtitle->fifo_raw ) ) )
450 * EOF, pass it through immediately.
455 /* If two subtitles overlap, make the first one stop
456 when the second one starts */
457 sub2 = hb_fifo_see2( subtitle->fifo_raw );
458 if( sub2 && sub->stop > sub2->start )
459 sub->stop = sub2->start;
461 // hb_log("0x%x: video seq: %lld subtitle sequence: %lld",
462 // sub, cur->sequence, sub->sequence);
464 if( sub->sequence > cur->sequence )
467 * The video is behind where we are, so wait until
468 * it catches up to the same reader point on the
469 * DVD. Then our PTS should be in the same region
476 if( sub->stop > cur->start ) {
478 * The stop time is in the future, so fall through
479 * and we'll deal with it in the next block of
486 * The subtitle is older than this picture, trash it
488 sub = hb_fifo_get( subtitle->fifo_raw );
489 hb_buffer_close( &sub );
492 if( sub && sub->size == 0 )
495 * Continue immediately on subtitle EOF
501 * There is a valid subtitle, is it time to display it?
505 if( sub->stop > sub->start)
508 * Normal subtitle which ends after it starts, check to
509 * see that the current video is between the start and end.
511 if( cur->start > sub->start &&
512 cur->start < sub->stop )
515 * We should be playing this, so leave the
518 * fall through to display
520 if( ( sub->stop - sub->start ) < ( 3 * 90000 ) )
523 * Subtitle is on for less than three seconds, extend
524 * the time that it is displayed to make it easier
525 * to read. Make it 3 seconds or until the next
526 * subtitle is displayed.
528 * This is in response to Indochine which only
529 * displays subs for 1 second - too fast to read.
531 sub->stop = sub->start + ( 3 * 90000 );
533 sub2 = hb_fifo_see2( subtitle->fifo_raw );
535 if( sub2 && sub->stop > sub2->start )
537 sub->stop = sub2->start;
544 * Defer until the play point is within the subtitle
552 * The end of the subtitle is less than the start, this is a
553 * sign of a PTS discontinuity.
555 if( sub->start > cur->start )
558 * we haven't reached the start time yet, or
559 * we have jumped backwards after having
560 * already started this subtitle.
562 if( cur->start < sub->stop )
565 * We have jumped backwards and so should
566 * continue displaying this subtitle.
568 * fall through to display.
574 * Defer until the play point is within the subtitle
580 * Play this subtitle as the start is greater than our
583 * fall through to display/
592 * Got a sub to display...
599 * Adjust the pts of the current frame so that it's contiguous
600 * with the previous frame. The start time of the current frame
601 * has to be the end time of the previous frame and the stop
602 * time has to be the start of the next frame. We don't
603 * make any adjustments to the source timestamps other than removing
604 * the clock offsets (which also removes pts discontinuities).
605 * This means we automatically encode at the source's frame rate.
606 * MP2 uses an implicit duration (frames end when the next frame
607 * starts) but more advanced containers like MP4 use an explicit
608 * duration. Since we're looking ahead one frame we set the
609 * explicit stop time from the start time of the next frame.
612 pv->cur = cur = hb_fifo_get( job->fifo_raw );
613 pv->next_pts = cur->start;
614 int64_t duration = cur->start - buf_tmp->start;
617 hb_log( "sync: invalid video duration %lld, start %lld, next %lld",
618 duration, buf_tmp->start, next->start );
621 buf_tmp->start = pv->next_start;
622 pv->next_start += duration;
623 buf_tmp->stop = pv->next_start;
627 // we have a pending chapter mark from a recent drop - put it on this
628 // buffer (this may make it one frame late but we can't do any better).
629 buf_tmp->new_chap = pv->chap_mark;
633 /* If we have a subtitle for this picture, copy it */
634 /* FIXME: we should avoid this memcpy */
635 if( sub && subtitle &&
636 subtitle->format == PICTURESUB )
640 if( subtitle->dest == RENDERSUB )
643 * Tack onto the video buffer for rendering
645 buf_tmp->sub = hb_buffer_init( sub->size );
646 buf_tmp->sub->x = sub->x;
647 buf_tmp->sub->y = sub->y;
648 buf_tmp->sub->width = sub->width;
649 buf_tmp->sub->height = sub->height;
650 memcpy( buf_tmp->sub->data, sub->data, sub->size );
653 * Pass-Through, pop it off of the raw queue, rewrite times and
654 * make it available to be reencoded.
656 uint64_t sub_duration;
657 sub = hb_fifo_get( subtitle->fifo_raw );
658 sub_duration = sub->stop - sub->start;
659 sub->start = buf_tmp->start;
660 sub->stop = sub->start + duration;
661 hb_fifo_push( subtitle->fifo_sync, sub );
665 * EOF - consume for rendered, else pass through
667 if( subtitle->dest == RENDERSUB )
669 sub = hb_fifo_get( subtitle->fifo_raw );
670 hb_buffer_close( &sub );
672 sub = hb_fifo_get( subtitle->fifo_raw );
673 hb_fifo_push( subtitle->fifo_out, sub );
678 /* Push the frame to the renderer */
679 hb_fifo_push( job->fifo_sync, buf_tmp );
684 if( job->frame_to_stop && pv->count_frames > job->frame_to_stop )
686 // Drop an empty buffer into our output to ensure that things
687 // get flushed all the way out.
688 hb_fifo_push( job->fifo_sync, hb_buffer_init( 0 ) );
690 hb_log( "sync: reached %d frames, exiting early (%i busy)",
691 pv->count_frames, pv->busy );
695 /* Make sure we won't get more frames then expected */
696 if( pv->count_frames >= pv->count_frames_max * 2)
698 hb_log( "sync: got too many frames (%d), exiting early",
701 // Drop an empty buffer into our output to ensure that things
702 // get flushed all the way out.
703 hb_fifo_push( job->fifo_sync, hb_buffer_init( 0 ) );
710 static void OutputAudioFrame( hb_job_t *job, hb_audio_t *audio, hb_buffer_t *buf,
711 hb_sync_audio_t *sync, hb_fifo_t *fifo, int i )
713 int64_t start = sync->next_start;
714 int64_t duration = buf->stop - buf->start;
716 sync->next_pts += duration;
718 if( audio->config.in.samplerate == audio->config.out.samplerate ||
719 audio->config.out.codec == HB_ACODEC_AC3 ||
720 audio->config.out.codec == HB_ACODEC_DCA )
723 * If we don't have to do sample rate conversion or this audio is
724 * pass-thru just send the input buffer downstream after adjusting
725 * its timestamps to make the output stream continuous.
730 /* Not pass-thru - do sample rate conversion */
731 int count_in, count_out;
732 hb_buffer_t * buf_raw = buf;
733 int channel_count = HB_AMIXDOWN_GET_DISCRETE_CHANNEL_COUNT(audio->config.out.mixdown) *
736 count_in = buf_raw->size / channel_count;
738 * When using stupid rates like 44.1 there will always be some
739 * truncation error. E.g., a 1536 sample AC3 frame will turn into a
740 * 1536*44.1/48.0 = 1411.2 sample frame. If we just truncate the .2
741 * the error will build up over time and eventually the audio will
742 * substantially lag the video. libsamplerate will keep track of the
743 * fractional sample & give it to us when appropriate if we give it
744 * an extra sample of space in the output buffer.
746 count_out = ( duration * audio->config.out.samplerate ) / 90000 + 1;
748 sync->data.input_frames = count_in;
749 sync->data.output_frames = count_out;
750 sync->data.src_ratio = (double)audio->config.out.samplerate /
751 (double)audio->config.in.samplerate;
753 buf = hb_buffer_init( count_out * channel_count );
754 sync->data.data_in = (float *) buf_raw->data;
755 sync->data.data_out = (float *) buf->data;
756 if( src_process( sync->state, &sync->data ) )
758 /* XXX If this happens, we're screwed */
759 hb_log( "sync: audio %d src_process failed", i );
761 hb_buffer_close( &buf_raw );
763 buf->size = sync->data.output_frames_gen * channel_count;
764 duration = ( sync->data.output_frames_gen * 90000 ) /
765 audio->config.out.samplerate;
767 buf->frametype = HB_FRAME_AUDIO;
769 buf->stop = start + duration;
770 sync->next_start = start + duration;
771 hb_fifo_push( fifo, buf );
774 /***********************************************************************
776 ***********************************************************************
778 **********************************************************************/
779 static void SyncAudio( hb_work_object_t * w, int i )
781 hb_work_private_t * pv = w->private_data;
782 hb_job_t * job = pv->job;
783 hb_sync_audio_t * sync = &pv->sync_audio[i];
784 hb_audio_t * audio = sync->audio;
789 if( audio->config.out.codec == HB_ACODEC_AC3 ||
790 audio->config.out.codec == HB_ACODEC_DCA )
792 fifo = audio->priv.fifo_out;
796 fifo = audio->priv.fifo_sync;
799 while( !hb_fifo_is_full( fifo ) && ( buf = hb_fifo_see( audio->priv.fifo_raw ) ) )
801 start = buf->start - pv->audio_passthru_slip;
802 /* if the next buffer is an eof send it downstream */
803 if ( buf->size <= 0 )
805 buf = hb_fifo_get( audio->priv.fifo_raw );
806 hb_fifo_push( fifo, buf );
807 pv->busy &=~ (1 << (i + 1) );
810 if( job->frame_to_stop && pv->count_frames >= job->frame_to_stop )
812 hb_fifo_push( fifo, hb_buffer_init(0) );
813 pv->busy &=~ (1 << (i + 1) );
816 if ( (int64_t)( start - sync->next_pts ) < 0 )
818 // audio time went backwards.
819 // If our output clock is more than a half frame ahead of the
820 // input clock drop this frame to move closer to sync.
821 // Otherwise drop frames until the input clock matches the output clock.
822 if ( sync->first_drop || sync->next_start - start > 90*15 )
824 // Discard data that's in the past.
825 if ( sync->first_drop == 0 )
827 sync->first_drop = sync->next_pts;
830 buf = hb_fifo_get( audio->priv.fifo_raw );
831 hb_buffer_close( &buf );
834 sync->next_pts = start;
836 if ( sync->first_drop )
838 // we were dropping old data but input buf time is now current
839 hb_log( "sync: audio %d time went backwards %d ms, dropped %d frames "
840 "(next %lld, current %lld)", i,
841 (int)( sync->next_pts - sync->first_drop ) / 90,
842 sync->drop_count, sync->first_drop, sync->next_pts );
843 sync->first_drop = 0;
844 sync->drop_count = 0;
845 sync->next_pts = start;
847 if ( start - sync->next_pts >= (90 * 70) )
849 if ( start - sync->next_pts > (90000LL * 60) )
851 // there's a gap of more than a minute between the last
852 // frame and this. assume we got a corrupted timestamp
853 // and just drop the next buf.
854 hb_log( "sync: %d minute time gap in audio %d - dropping buf"
855 " start %lld, next %lld",
856 (int)((start - sync->next_pts) / (90000*60)),
857 i, start, sync->next_pts );
858 buf = hb_fifo_get( audio->priv.fifo_raw );
859 hb_buffer_close( &buf );
863 * there's a gap of at least 70ms between the last
864 * frame we processed & the next. Fill it with silence.
865 * Or in the case of DCA, skip some frames from the
868 if( sync->audio->config.out.codec == HB_ACODEC_DCA )
870 hb_log( "sync: audio gap %d ms. Skipping frames. Audio %d"
871 " start %lld, next %lld",
872 (int)((start - sync->next_pts) / 90),
873 i, start, sync->next_pts );
874 pv->audio_passthru_slip += (start - sync->next_pts);
877 hb_log( "sync: adding %d ms of silence to audio %d"
878 " start %lld, next %lld",
879 (int)((start - sync->next_pts) / 90),
880 i, start, sync->next_pts );
881 InsertSilence( w, i, start - sync->next_pts );
886 * When we get here we've taken care of all the dups and gaps in the
887 * audio stream and are ready to inject the next input frame into
890 buf = hb_fifo_get( audio->priv.fifo_raw );
891 OutputAudioFrame( job, audio, buf, sync, fifo, i );
895 static void InsertSilence( hb_work_object_t * w, int i, int64_t duration )
897 hb_work_private_t * pv = w->private_data;
898 hb_job_t *job = pv->job;
899 hb_sync_audio_t *sync = &pv->sync_audio[i];
903 // to keep pass-thru and regular audio in sync we generate silence in
904 // AC3 frame-sized units. If the silence duration isn't an integer multiple
905 // of the AC3 frame duration we will truncate or round up depending on
906 // which minimizes the timing error.
907 const int frame_dur = ( 90000 * AC3_SAMPLES_PER_FRAME ) /
908 sync->audio->config.in.samplerate;
909 int frame_count = ( duration + (frame_dur >> 1) ) / frame_dur;
911 while ( --frame_count >= 0 )
913 if( sync->audio->config.out.codec == HB_ACODEC_AC3 )
915 buf = hb_buffer_init( sync->ac3_size );
916 buf->start = sync->next_pts;
917 buf->stop = buf->start + frame_dur;
918 memcpy( buf->data, sync->ac3_buf, buf->size );
919 fifo = sync->audio->priv.fifo_out;
923 buf = hb_buffer_init( AC3_SAMPLES_PER_FRAME * sizeof( float ) *
924 HB_AMIXDOWN_GET_DISCRETE_CHANNEL_COUNT(
925 sync->audio->config.out.mixdown) );
926 buf->start = sync->next_pts;
927 buf->stop = buf->start + frame_dur;
928 memset( buf->data, 0, buf->size );
929 fifo = sync->audio->priv.fifo_sync;
931 OutputAudioFrame( job, sync->audio, buf, sync, fifo, i );
935 static void UpdateState( hb_work_object_t * w )
937 hb_work_private_t * pv = w->private_data;
940 if( !pv->count_frames )
942 pv->st_first = hb_get_date();
946 if( hb_get_date() > pv->st_dates[3] + 1000 )
948 memmove( &pv->st_dates[0], &pv->st_dates[1],
949 3 * sizeof( uint64_t ) );
950 memmove( &pv->st_counts[0], &pv->st_counts[1],
951 3 * sizeof( uint64_t ) );
952 pv->st_dates[3] = hb_get_date();
953 pv->st_counts[3] = pv->count_frames;
956 #define p state.param.working
957 state.state = HB_STATE_WORKING;
958 p.progress = (float) pv->count_frames / (float) pv->count_frames_max;
959 if( p.progress > 1.0 )
963 p.rate_cur = 1000.0 *
964 (float) ( pv->st_counts[3] - pv->st_counts[0] ) /
965 (float) ( pv->st_dates[3] - pv->st_dates[0] );
966 if( hb_get_date() > pv->st_first + 4000 )
969 p.rate_avg = 1000.0 * (float) pv->st_counts[3] /
970 (float) ( pv->st_dates[3] - pv->st_first );
971 eta = (float) ( pv->count_frames_max - pv->st_counts[3] ) /
973 p.hours = eta / 3600;
974 p.minutes = ( eta % 3600 ) / 60;
975 p.seconds = eta % 60;
986 hb_set_state( pv->job->h, &state );