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 if( sub->size == 0 || sub->start < cur->start )
419 sub = hb_fifo_get( subtitle->fifo_raw );
420 sub->start = pv->next_start;
421 hb_fifo_push( subtitle->fifo_out, sub );
429 if( subtitle->source == VOBSUB )
432 while( ( sub = hb_fifo_see( subtitle->fifo_raw ) ) )
437 * EOF, pass it through immediately.
442 /* If two subtitles overlap, make the first one stop
443 when the second one starts */
444 sub2 = hb_fifo_see2( subtitle->fifo_raw );
445 if( sub2 && sub->stop > sub2->start )
446 sub->stop = sub2->start;
448 // hb_log("0x%x: video seq: %lld subtitle sequence: %lld",
449 // sub, cur->sequence, sub->sequence);
451 if( sub->sequence > cur->sequence )
454 * The video is behind where we are, so wait until
455 * it catches up to the same reader point on the
456 * DVD. Then our PTS should be in the same region
463 if( sub->stop > cur->start ) {
465 * The stop time is in the future, so fall through
466 * and we'll deal with it in the next block of
473 * The subtitle is older than this picture, trash it
475 sub = hb_fifo_get( subtitle->fifo_raw );
476 hb_buffer_close( &sub );
479 if( sub && sub->size == 0 )
482 * Continue immediately on subtitle EOF
488 * There is a valid subtitle, is it time to display it?
492 if( sub->stop > sub->start)
495 * Normal subtitle which ends after it starts, check to
496 * see that the current video is between the start and end.
498 if( cur->start > sub->start &&
499 cur->start < sub->stop )
502 * We should be playing this, so leave the
505 * fall through to display
507 if( ( sub->stop - sub->start ) < ( 3 * 90000 ) )
510 * Subtitle is on for less than three seconds, extend
511 * the time that it is displayed to make it easier
512 * to read. Make it 3 seconds or until the next
513 * subtitle is displayed.
515 * This is in response to Indochine which only
516 * displays subs for 1 second - too fast to read.
518 sub->stop = sub->start + ( 3 * 90000 );
520 sub2 = hb_fifo_see2( subtitle->fifo_raw );
522 if( sub2 && sub->stop > sub2->start )
524 sub->stop = sub2->start;
531 * Defer until the play point is within the subtitle
539 * The end of the subtitle is less than the start, this is a
540 * sign of a PTS discontinuity.
542 if( sub->start > cur->start )
545 * we haven't reached the start time yet, or
546 * we have jumped backwards after having
547 * already started this subtitle.
549 if( cur->start < sub->stop )
552 * We have jumped backwards and so should
553 * continue displaying this subtitle.
555 * fall through to display.
561 * Defer until the play point is within the subtitle
567 * Play this subtitle as the start is greater than our
570 * fall through to display/
579 * Got a sub to display...
586 * Adjust the pts of the current frame so that it's contiguous
587 * with the previous frame. The start time of the current frame
588 * has to be the end time of the previous frame and the stop
589 * time has to be the start of the next frame. We don't
590 * make any adjustments to the source timestamps other than removing
591 * the clock offsets (which also removes pts discontinuities).
592 * This means we automatically encode at the source's frame rate.
593 * MP2 uses an implicit duration (frames end when the next frame
594 * starts) but more advanced containers like MP4 use an explicit
595 * duration. Since we're looking ahead one frame we set the
596 * explicit stop time from the start time of the next frame.
599 pv->cur = cur = hb_fifo_get( job->fifo_raw );
600 pv->next_pts = cur->start;
601 int64_t duration = cur->start - buf_tmp->start;
604 hb_log( "sync: invalid video duration %lld, start %lld, next %lld",
605 duration, buf_tmp->start, next->start );
608 buf_tmp->start = pv->next_start;
609 pv->next_start += duration;
610 buf_tmp->stop = pv->next_start;
614 // we have a pending chapter mark from a recent drop - put it on this
615 // buffer (this may make it one frame late but we can't do any better).
616 buf_tmp->new_chap = pv->chap_mark;
620 /* If we have a subtitle for this picture, copy it */
621 /* FIXME: we should avoid this memcpy */
622 if( sub && subtitle &&
623 subtitle->format == PICTURESUB )
627 if( subtitle->dest == RENDERSUB )
630 * Tack onto the video buffer for rendering
632 buf_tmp->sub = hb_buffer_init( sub->size );
633 buf_tmp->sub->x = sub->x;
634 buf_tmp->sub->y = sub->y;
635 buf_tmp->sub->width = sub->width;
636 buf_tmp->sub->height = sub->height;
637 memcpy( buf_tmp->sub->data, sub->data, sub->size );
640 * Pass-Through, pop it off of the raw queue, rewrite times and
641 * make it available to be muxed.
643 uint64_t sub_duration;
644 sub = hb_fifo_get( subtitle->fifo_raw );
645 sub_duration = sub->stop - sub->start;
646 sub->start = buf_tmp->start;
647 sub->stop = sub->start + duration;
648 hb_fifo_push( subtitle->fifo_out, sub );
652 * EOF - consume for rendered, else pass through
654 if( subtitle->dest == RENDERSUB )
656 sub = hb_fifo_get( subtitle->fifo_raw );
657 hb_buffer_close( &sub );
659 sub = hb_fifo_get( subtitle->fifo_raw );
660 hb_fifo_push( subtitle->fifo_out, sub );
665 /* Push the frame to the renderer */
666 hb_fifo_push( job->fifo_sync, buf_tmp );
671 if( job->frame_to_stop && pv->count_frames > job->frame_to_stop )
673 // Drop an empty buffer into our output to ensure that things
674 // get flushed all the way out.
675 hb_fifo_push( job->fifo_sync, hb_buffer_init( 0 ) );
677 hb_log( "sync: reached %d frames, exiting early (%i busy)",
678 pv->count_frames, pv->busy );
682 /* Make sure we won't get more frames then expected */
683 if( pv->count_frames >= pv->count_frames_max * 2)
685 hb_log( "sync: got too many frames (%d), exiting early",
688 // Drop an empty buffer into our output to ensure that things
689 // get flushed all the way out.
690 hb_fifo_push( job->fifo_sync, hb_buffer_init( 0 ) );
697 static void OutputAudioFrame( hb_job_t *job, hb_audio_t *audio, hb_buffer_t *buf,
698 hb_sync_audio_t *sync, hb_fifo_t *fifo, int i )
700 int64_t start = sync->next_start;
701 int64_t duration = buf->stop - buf->start;
703 sync->next_pts += duration;
705 if( audio->config.in.samplerate == audio->config.out.samplerate ||
706 audio->config.out.codec == HB_ACODEC_AC3 ||
707 audio->config.out.codec == HB_ACODEC_DCA )
710 * If we don't have to do sample rate conversion or this audio is
711 * pass-thru just send the input buffer downstream after adjusting
712 * its timestamps to make the output stream continuous.
717 /* Not pass-thru - do sample rate conversion */
718 int count_in, count_out;
719 hb_buffer_t * buf_raw = buf;
720 int channel_count = HB_AMIXDOWN_GET_DISCRETE_CHANNEL_COUNT(audio->config.out.mixdown) *
723 count_in = buf_raw->size / channel_count;
725 * When using stupid rates like 44.1 there will always be some
726 * truncation error. E.g., a 1536 sample AC3 frame will turn into a
727 * 1536*44.1/48.0 = 1411.2 sample frame. If we just truncate the .2
728 * the error will build up over time and eventually the audio will
729 * substantially lag the video. libsamplerate will keep track of the
730 * fractional sample & give it to us when appropriate if we give it
731 * an extra sample of space in the output buffer.
733 count_out = ( duration * audio->config.out.samplerate ) / 90000 + 1;
735 sync->data.input_frames = count_in;
736 sync->data.output_frames = count_out;
737 sync->data.src_ratio = (double)audio->config.out.samplerate /
738 (double)audio->config.in.samplerate;
740 buf = hb_buffer_init( count_out * channel_count );
741 sync->data.data_in = (float *) buf_raw->data;
742 sync->data.data_out = (float *) buf->data;
743 if( src_process( sync->state, &sync->data ) )
745 /* XXX If this happens, we're screwed */
746 hb_log( "sync: audio %d src_process failed", i );
748 hb_buffer_close( &buf_raw );
750 buf->size = sync->data.output_frames_gen * channel_count;
751 duration = ( sync->data.output_frames_gen * 90000 ) /
752 audio->config.out.samplerate;
754 buf->frametype = HB_FRAME_AUDIO;
756 buf->stop = start + duration;
757 sync->next_start = start + duration;
758 hb_fifo_push( fifo, buf );
761 /***********************************************************************
763 ***********************************************************************
765 **********************************************************************/
766 static void SyncAudio( hb_work_object_t * w, int i )
768 hb_work_private_t * pv = w->private_data;
769 hb_job_t * job = pv->job;
770 hb_sync_audio_t * sync = &pv->sync_audio[i];
771 hb_audio_t * audio = sync->audio;
776 if( audio->config.out.codec == HB_ACODEC_AC3 ||
777 audio->config.out.codec == HB_ACODEC_DCA )
779 fifo = audio->priv.fifo_out;
783 fifo = audio->priv.fifo_sync;
786 while( !hb_fifo_is_full( fifo ) && ( buf = hb_fifo_see( audio->priv.fifo_raw ) ) )
788 start = buf->start - pv->audio_passthru_slip;
789 /* if the next buffer is an eof send it downstream */
790 if ( buf->size <= 0 )
792 buf = hb_fifo_get( audio->priv.fifo_raw );
793 hb_fifo_push( fifo, buf );
794 pv->busy &=~ (1 << (i + 1) );
797 if( job->frame_to_stop && pv->count_frames >= job->frame_to_stop )
799 hb_fifo_push( fifo, hb_buffer_init(0) );
800 pv->busy &=~ (1 << (i + 1) );
803 if ( (int64_t)( start - sync->next_pts ) < 0 )
805 // audio time went backwards.
806 // If our output clock is more than a half frame ahead of the
807 // input clock drop this frame to move closer to sync.
808 // Otherwise drop frames until the input clock matches the output clock.
809 if ( sync->first_drop || sync->next_start - start > 90*15 )
811 // Discard data that's in the past.
812 if ( sync->first_drop == 0 )
814 sync->first_drop = sync->next_pts;
817 buf = hb_fifo_get( audio->priv.fifo_raw );
818 hb_buffer_close( &buf );
821 sync->next_pts = start;
823 if ( sync->first_drop )
825 // we were dropping old data but input buf time is now current
826 hb_log( "sync: audio %d time went backwards %d ms, dropped %d frames "
827 "(next %lld, current %lld)", i,
828 (int)( sync->next_pts - sync->first_drop ) / 90,
829 sync->drop_count, sync->first_drop, sync->next_pts );
830 sync->first_drop = 0;
831 sync->drop_count = 0;
832 sync->next_pts = start;
834 if ( start - sync->next_pts >= (90 * 70) )
836 if ( start - sync->next_pts > (90000LL * 60) )
838 // there's a gap of more than a minute between the last
839 // frame and this. assume we got a corrupted timestamp
840 // and just drop the next buf.
841 hb_log( "sync: %d minute time gap in audio %d - dropping buf"
842 " start %lld, next %lld",
843 (int)((start - sync->next_pts) / (90000*60)),
844 i, start, sync->next_pts );
845 buf = hb_fifo_get( audio->priv.fifo_raw );
846 hb_buffer_close( &buf );
850 * there's a gap of at least 70ms between the last
851 * frame we processed & the next. Fill it with silence.
852 * Or in the case of DCA, skip some frames from the
855 if( sync->audio->config.out.codec == HB_ACODEC_DCA )
857 hb_log( "sync: audio gap %d ms. Skipping frames. Audio %d"
858 " start %lld, next %lld",
859 (int)((start - sync->next_pts) / 90),
860 i, start, sync->next_pts );
861 pv->audio_passthru_slip += (start - sync->next_pts);
864 hb_log( "sync: adding %d ms of silence to audio %d"
865 " start %lld, next %lld",
866 (int)((start - sync->next_pts) / 90),
867 i, start, sync->next_pts );
868 InsertSilence( w, i, start - sync->next_pts );
873 * When we get here we've taken care of all the dups and gaps in the
874 * audio stream and are ready to inject the next input frame into
877 buf = hb_fifo_get( audio->priv.fifo_raw );
878 OutputAudioFrame( job, audio, buf, sync, fifo, i );
882 static void InsertSilence( hb_work_object_t * w, int i, int64_t duration )
884 hb_work_private_t * pv = w->private_data;
885 hb_job_t *job = pv->job;
886 hb_sync_audio_t *sync = &pv->sync_audio[i];
890 // to keep pass-thru and regular audio in sync we generate silence in
891 // AC3 frame-sized units. If the silence duration isn't an integer multiple
892 // of the AC3 frame duration we will truncate or round up depending on
893 // which minimizes the timing error.
894 const int frame_dur = ( 90000 * AC3_SAMPLES_PER_FRAME ) /
895 sync->audio->config.in.samplerate;
896 int frame_count = ( duration + (frame_dur >> 1) ) / frame_dur;
898 while ( --frame_count >= 0 )
900 if( sync->audio->config.out.codec == HB_ACODEC_AC3 )
902 buf = hb_buffer_init( sync->ac3_size );
903 buf->start = sync->next_pts;
904 buf->stop = buf->start + frame_dur;
905 memcpy( buf->data, sync->ac3_buf, buf->size );
906 fifo = sync->audio->priv.fifo_out;
910 buf = hb_buffer_init( AC3_SAMPLES_PER_FRAME * sizeof( float ) *
911 HB_AMIXDOWN_GET_DISCRETE_CHANNEL_COUNT(
912 sync->audio->config.out.mixdown) );
913 buf->start = sync->next_pts;
914 buf->stop = buf->start + frame_dur;
915 memset( buf->data, 0, buf->size );
916 fifo = sync->audio->priv.fifo_sync;
918 OutputAudioFrame( job, sync->audio, buf, sync, fifo, i );
922 static void UpdateState( hb_work_object_t * w )
924 hb_work_private_t * pv = w->private_data;
927 if( !pv->count_frames )
929 pv->st_first = hb_get_date();
933 if( hb_get_date() > pv->st_dates[3] + 1000 )
935 memmove( &pv->st_dates[0], &pv->st_dates[1],
936 3 * sizeof( uint64_t ) );
937 memmove( &pv->st_counts[0], &pv->st_counts[1],
938 3 * sizeof( uint64_t ) );
939 pv->st_dates[3] = hb_get_date();
940 pv->st_counts[3] = pv->count_frames;
943 #define p state.param.working
944 state.state = HB_STATE_WORKING;
945 p.progress = (float) pv->count_frames / (float) pv->count_frames_max;
946 if( p.progress > 1.0 )
950 p.rate_cur = 1000.0 *
951 (float) ( pv->st_counts[3] - pv->st_counts[0] ) /
952 (float) ( pv->st_dates[3] - pv->st_dates[0] );
953 if( hb_get_date() > pv->st_first + 4000 )
956 p.rate_avg = 1000.0 * (float) pv->st_counts[3] /
957 (float) ( pv->st_dates[3] - pv->st_first );
958 eta = (float) ( pv->count_frames_max - pv->st_counts[3] ) /
960 p.hours = eta / 3600;
961 p.minutes = ( eta % 3600 ) / 60;
962 p.seconds = eta % 60;
973 hb_set_state( pv->job->h, &state );