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.
419 if( sub->size == 0 || sub->start < cur->start )
421 sub = hb_fifo_get( subtitle->fifo_raw );
422 sub->start = pv->next_start;
423 hb_fifo_push( subtitle->fifo_out, sub );
431 if( subtitle->source == VOBSUB )
434 while( ( sub = hb_fifo_see( subtitle->fifo_raw ) ) )
439 * EOF, pass it through immediately.
444 /* If two subtitles overlap, make the first one stop
445 when the second one starts */
446 sub2 = hb_fifo_see2( subtitle->fifo_raw );
447 if( sub2 && sub->stop > sub2->start )
448 sub->stop = sub2->start;
450 // hb_log("0x%x: video seq: %lld subtitle sequence: %lld",
451 // sub, cur->sequence, sub->sequence);
453 if( sub->sequence > cur->sequence )
456 * The video is behind where we are, so wait until
457 * it catches up to the same reader point on the
458 * DVD. Then our PTS should be in the same region
465 if( sub->stop > cur->start ) {
467 * The stop time is in the future, so fall through
468 * and we'll deal with it in the next block of
475 * The subtitle is older than this picture, trash it
477 sub = hb_fifo_get( subtitle->fifo_raw );
478 hb_buffer_close( &sub );
481 if( sub && sub->size == 0 )
484 * Continue immediately on subtitle EOF
490 * There is a valid subtitle, is it time to display it?
494 if( sub->stop > sub->start)
497 * Normal subtitle which ends after it starts, check to
498 * see that the current video is between the start and end.
500 if( cur->start > sub->start &&
501 cur->start < sub->stop )
504 * We should be playing this, so leave the
507 * fall through to display
509 if( ( sub->stop - sub->start ) < ( 3 * 90000 ) )
512 * Subtitle is on for less than three seconds, extend
513 * the time that it is displayed to make it easier
514 * to read. Make it 3 seconds or until the next
515 * subtitle is displayed.
517 * This is in response to Indochine which only
518 * displays subs for 1 second - too fast to read.
520 sub->stop = sub->start + ( 3 * 90000 );
522 sub2 = hb_fifo_see2( subtitle->fifo_raw );
524 if( sub2 && sub->stop > sub2->start )
526 sub->stop = sub2->start;
533 * Defer until the play point is within the subtitle
541 * The end of the subtitle is less than the start, this is a
542 * sign of a PTS discontinuity.
544 if( sub->start > cur->start )
547 * we haven't reached the start time yet, or
548 * we have jumped backwards after having
549 * already started this subtitle.
551 if( cur->start < sub->stop )
554 * We have jumped backwards and so should
555 * continue displaying this subtitle.
557 * fall through to display.
563 * Defer until the play point is within the subtitle
569 * Play this subtitle as the start is greater than our
572 * fall through to display/
581 * Got a sub to display...
588 * Adjust the pts of the current frame so that it's contiguous
589 * with the previous frame. The start time of the current frame
590 * has to be the end time of the previous frame and the stop
591 * time has to be the start of the next frame. We don't
592 * make any adjustments to the source timestamps other than removing
593 * the clock offsets (which also removes pts discontinuities).
594 * This means we automatically encode at the source's frame rate.
595 * MP2 uses an implicit duration (frames end when the next frame
596 * starts) but more advanced containers like MP4 use an explicit
597 * duration. Since we're looking ahead one frame we set the
598 * explicit stop time from the start time of the next frame.
601 pv->cur = cur = hb_fifo_get( job->fifo_raw );
602 pv->next_pts = cur->start;
603 int64_t duration = cur->start - buf_tmp->start;
606 hb_log( "sync: invalid video duration %lld, start %lld, next %lld",
607 duration, buf_tmp->start, next->start );
610 buf_tmp->start = pv->next_start;
611 pv->next_start += duration;
612 buf_tmp->stop = pv->next_start;
616 // we have a pending chapter mark from a recent drop - put it on this
617 // buffer (this may make it one frame late but we can't do any better).
618 buf_tmp->new_chap = pv->chap_mark;
622 /* If we have a subtitle for this picture, copy it */
623 /* FIXME: we should avoid this memcpy */
624 if( sub && subtitle &&
625 subtitle->format == PICTURESUB )
629 if( subtitle->dest == RENDERSUB )
632 * Tack onto the video buffer for rendering
634 buf_tmp->sub = hb_buffer_init( sub->size );
635 buf_tmp->sub->x = sub->x;
636 buf_tmp->sub->y = sub->y;
637 buf_tmp->sub->width = sub->width;
638 buf_tmp->sub->height = sub->height;
639 memcpy( buf_tmp->sub->data, sub->data, sub->size );
642 * Pass-Through, pop it off of the raw queue, rewrite times and
643 * make it available to be reencoded.
645 uint64_t sub_duration;
646 sub = hb_fifo_get( subtitle->fifo_raw );
647 sub_duration = sub->stop - sub->start;
648 sub->start = buf_tmp->start;
649 sub->stop = sub->start + duration;
650 hb_fifo_push( subtitle->fifo_sync, sub );
654 * EOF - consume for rendered, else pass through
656 if( subtitle->dest == RENDERSUB )
658 sub = hb_fifo_get( subtitle->fifo_raw );
659 hb_buffer_close( &sub );
661 sub = hb_fifo_get( subtitle->fifo_raw );
662 hb_fifo_push( subtitle->fifo_out, sub );
667 /* Push the frame to the renderer */
668 hb_fifo_push( job->fifo_sync, buf_tmp );
673 if( job->frame_to_stop && pv->count_frames > job->frame_to_stop )
675 // Drop an empty buffer into our output to ensure that things
676 // get flushed all the way out.
677 hb_fifo_push( job->fifo_sync, hb_buffer_init( 0 ) );
679 hb_log( "sync: reached %d frames, exiting early (%i busy)",
680 pv->count_frames, pv->busy );
684 /* Make sure we won't get more frames then expected */
685 if( pv->count_frames >= pv->count_frames_max * 2)
687 hb_log( "sync: got too many frames (%d), exiting early",
690 // Drop an empty buffer into our output to ensure that things
691 // get flushed all the way out.
692 hb_fifo_push( job->fifo_sync, hb_buffer_init( 0 ) );
699 static void OutputAudioFrame( hb_job_t *job, hb_audio_t *audio, hb_buffer_t *buf,
700 hb_sync_audio_t *sync, hb_fifo_t *fifo, int i )
702 int64_t start = sync->next_start;
703 int64_t duration = buf->stop - buf->start;
705 sync->next_pts += duration;
707 if( audio->config.in.samplerate == audio->config.out.samplerate ||
708 audio->config.out.codec == HB_ACODEC_AC3 ||
709 audio->config.out.codec == HB_ACODEC_DCA )
712 * If we don't have to do sample rate conversion or this audio is
713 * pass-thru just send the input buffer downstream after adjusting
714 * its timestamps to make the output stream continuous.
719 /* Not pass-thru - do sample rate conversion */
720 int count_in, count_out;
721 hb_buffer_t * buf_raw = buf;
722 int channel_count = HB_AMIXDOWN_GET_DISCRETE_CHANNEL_COUNT(audio->config.out.mixdown) *
725 count_in = buf_raw->size / channel_count;
727 * When using stupid rates like 44.1 there will always be some
728 * truncation error. E.g., a 1536 sample AC3 frame will turn into a
729 * 1536*44.1/48.0 = 1411.2 sample frame. If we just truncate the .2
730 * the error will build up over time and eventually the audio will
731 * substantially lag the video. libsamplerate will keep track of the
732 * fractional sample & give it to us when appropriate if we give it
733 * an extra sample of space in the output buffer.
735 count_out = ( duration * audio->config.out.samplerate ) / 90000 + 1;
737 sync->data.input_frames = count_in;
738 sync->data.output_frames = count_out;
739 sync->data.src_ratio = (double)audio->config.out.samplerate /
740 (double)audio->config.in.samplerate;
742 buf = hb_buffer_init( count_out * channel_count );
743 sync->data.data_in = (float *) buf_raw->data;
744 sync->data.data_out = (float *) buf->data;
745 if( src_process( sync->state, &sync->data ) )
747 /* XXX If this happens, we're screwed */
748 hb_log( "sync: audio %d src_process failed", i );
750 hb_buffer_close( &buf_raw );
752 buf->size = sync->data.output_frames_gen * channel_count;
753 duration = ( sync->data.output_frames_gen * 90000 ) /
754 audio->config.out.samplerate;
756 buf->frametype = HB_FRAME_AUDIO;
758 buf->stop = start + duration;
759 sync->next_start = start + duration;
760 hb_fifo_push( fifo, buf );
763 /***********************************************************************
765 ***********************************************************************
767 **********************************************************************/
768 static void SyncAudio( hb_work_object_t * w, int i )
770 hb_work_private_t * pv = w->private_data;
771 hb_job_t * job = pv->job;
772 hb_sync_audio_t * sync = &pv->sync_audio[i];
773 hb_audio_t * audio = sync->audio;
778 if( audio->config.out.codec == HB_ACODEC_AC3 ||
779 audio->config.out.codec == HB_ACODEC_DCA )
781 fifo = audio->priv.fifo_out;
785 fifo = audio->priv.fifo_sync;
788 while( !hb_fifo_is_full( fifo ) && ( buf = hb_fifo_see( audio->priv.fifo_raw ) ) )
790 start = buf->start - pv->audio_passthru_slip;
791 /* if the next buffer is an eof send it downstream */
792 if ( buf->size <= 0 )
794 buf = hb_fifo_get( audio->priv.fifo_raw );
795 hb_fifo_push( fifo, buf );
796 pv->busy &=~ (1 << (i + 1) );
799 if( job->frame_to_stop && pv->count_frames >= job->frame_to_stop )
801 hb_fifo_push( fifo, hb_buffer_init(0) );
802 pv->busy &=~ (1 << (i + 1) );
805 if ( (int64_t)( start - sync->next_pts ) < 0 )
807 // audio time went backwards.
808 // If our output clock is more than a half frame ahead of the
809 // input clock drop this frame to move closer to sync.
810 // Otherwise drop frames until the input clock matches the output clock.
811 if ( sync->first_drop || sync->next_start - start > 90*15 )
813 // Discard data that's in the past.
814 if ( sync->first_drop == 0 )
816 sync->first_drop = sync->next_pts;
819 buf = hb_fifo_get( audio->priv.fifo_raw );
820 hb_buffer_close( &buf );
823 sync->next_pts = start;
825 if ( sync->first_drop )
827 // we were dropping old data but input buf time is now current
828 hb_log( "sync: audio %d time went backwards %d ms, dropped %d frames "
829 "(next %lld, current %lld)", i,
830 (int)( sync->next_pts - sync->first_drop ) / 90,
831 sync->drop_count, sync->first_drop, sync->next_pts );
832 sync->first_drop = 0;
833 sync->drop_count = 0;
834 sync->next_pts = start;
836 if ( start - sync->next_pts >= (90 * 70) )
838 if ( start - sync->next_pts > (90000LL * 60) )
840 // there's a gap of more than a minute between the last
841 // frame and this. assume we got a corrupted timestamp
842 // and just drop the next buf.
843 hb_log( "sync: %d minute time gap in audio %d - dropping buf"
844 " start %lld, next %lld",
845 (int)((start - sync->next_pts) / (90000*60)),
846 i, start, sync->next_pts );
847 buf = hb_fifo_get( audio->priv.fifo_raw );
848 hb_buffer_close( &buf );
852 * there's a gap of at least 70ms between the last
853 * frame we processed & the next. Fill it with silence.
854 * Or in the case of DCA, skip some frames from the
857 if( sync->audio->config.out.codec == HB_ACODEC_DCA )
859 hb_log( "sync: audio gap %d ms. Skipping frames. Audio %d"
860 " start %lld, next %lld",
861 (int)((start - sync->next_pts) / 90),
862 i, start, sync->next_pts );
863 pv->audio_passthru_slip += (start - sync->next_pts);
866 hb_log( "sync: adding %d ms of silence to audio %d"
867 " start %lld, next %lld",
868 (int)((start - sync->next_pts) / 90),
869 i, start, sync->next_pts );
870 InsertSilence( w, i, start - sync->next_pts );
875 * When we get here we've taken care of all the dups and gaps in the
876 * audio stream and are ready to inject the next input frame into
879 buf = hb_fifo_get( audio->priv.fifo_raw );
880 OutputAudioFrame( job, audio, buf, sync, fifo, i );
884 static void InsertSilence( hb_work_object_t * w, int i, int64_t duration )
886 hb_work_private_t * pv = w->private_data;
887 hb_job_t *job = pv->job;
888 hb_sync_audio_t *sync = &pv->sync_audio[i];
892 // to keep pass-thru and regular audio in sync we generate silence in
893 // AC3 frame-sized units. If the silence duration isn't an integer multiple
894 // of the AC3 frame duration we will truncate or round up depending on
895 // which minimizes the timing error.
896 const int frame_dur = ( 90000 * AC3_SAMPLES_PER_FRAME ) /
897 sync->audio->config.in.samplerate;
898 int frame_count = ( duration + (frame_dur >> 1) ) / frame_dur;
900 while ( --frame_count >= 0 )
902 if( sync->audio->config.out.codec == HB_ACODEC_AC3 )
904 buf = hb_buffer_init( sync->ac3_size );
905 buf->start = sync->next_pts;
906 buf->stop = buf->start + frame_dur;
907 memcpy( buf->data, sync->ac3_buf, buf->size );
908 fifo = sync->audio->priv.fifo_out;
912 buf = hb_buffer_init( AC3_SAMPLES_PER_FRAME * sizeof( float ) *
913 HB_AMIXDOWN_GET_DISCRETE_CHANNEL_COUNT(
914 sync->audio->config.out.mixdown) );
915 buf->start = sync->next_pts;
916 buf->stop = buf->start + frame_dur;
917 memset( buf->data, 0, buf->size );
918 fifo = sync->audio->priv.fifo_sync;
920 OutputAudioFrame( job, sync->audio, buf, sync, fifo, i );
924 static void UpdateState( hb_work_object_t * w )
926 hb_work_private_t * pv = w->private_data;
929 if( !pv->count_frames )
931 pv->st_first = hb_get_date();
935 if( hb_get_date() > pv->st_dates[3] + 1000 )
937 memmove( &pv->st_dates[0], &pv->st_dates[1],
938 3 * sizeof( uint64_t ) );
939 memmove( &pv->st_counts[0], &pv->st_counts[1],
940 3 * sizeof( uint64_t ) );
941 pv->st_dates[3] = hb_get_date();
942 pv->st_counts[3] = pv->count_frames;
945 #define p state.param.working
946 state.state = HB_STATE_WORKING;
947 p.progress = (float) pv->count_frames / (float) pv->count_frames_max;
948 if( p.progress > 1.0 )
952 p.rate_cur = 1000.0 *
953 (float) ( pv->st_counts[3] - pv->st_counts[0] ) /
954 (float) ( pv->st_dates[3] - pv->st_dates[0] );
955 if( hb_get_date() > pv->st_first + 4000 )
958 p.rate_avg = 1000.0 * (float) pv->st_counts[3] /
959 (float) ( pv->st_dates[3] - pv->st_first );
960 eta = (float) ( pv->count_frames_max - pv->st_counts[3] ) /
962 p.hours = eta / 3600;
963 p.minutes = ( eta % 3600 ) / 60;
964 p.seconds = eta % 60;
975 hb_set_state( pv->job->h, &state );