/* $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 */
typedef struct
{
hb_audio_t * audio;
- int64_t count_frames;
-
+
+ int64_t next_start; /* start time of next output frame */
+ 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 */
+
/* Raw */
SRC_STATE * state;
SRC_DATA data;
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 pts_offset_old;
- int64_t count_frames;
- int64_t count_frames_max;
- int64_t video_sequence;
+ int64_t next_start; /* start time of next output frame */
+ 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;
+ int chap_mark; /* to propagate chapter mark across a drop */
hb_buffer_t * cur; /* The next picture to process */
/* Audio */
hb_sync_audio_t sync_audio[8];
+ int64_t audio_passthru_slip;
/* Statistics */
uint64_t st_counts[4];
* 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 * );
-static void InsertSilence( hb_work_object_t * w, 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->pts_offset_old = 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)
+ {
+ duration = job->pts_to_stop + 90000;
+ }
+ else if( job->frame_to_stop )
{
- chapter = hb_list_item( title->list_chapter, i - 1 );
- duration += chapter->duration;
- }
- duration += 90000;
+ /* Set the duration to a rough estimate */
+ duration = ( job->frame_to_stop / ( job->vrate / job->vrate_base ) ) * 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 %lld video frames", pv->count_frames_max );
+ 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 < 8; 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_work_private_t * pv = w->private_data;
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 );
+
+ if (pv->drops || pv->dups )
+ {
+ hb_log( "sync: %d frames dropped, %d duplicated", pv->drops, pv->dups );
+ }
for( i = 0; i < hb_list_count( title->list_audio ); i++ )
{
- if( job->acodec & HB_ACODEC_AC3 )
+ audio = hb_list_item( title->list_audio, i );
+ if( audio->config.out.codec == HB_ACODEC_AC3 )
{
free( pv->sync_audio[i].ac3_buf );
}
src_delete( pv->sync_audio[i].state );
}
}
-
+
free( pv );
w->private_data = NULL;
}
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 =
sync = &pv->sync_audio[i];
sync->audio = hb_list_item( title->list_audio, i );
- if( job->acodec & HB_ACODEC_AC3 )
+ if( sync->audio->config.out.codec == HB_ACODEC_AC3 )
{
/* Have a silent AC-3 frame ready in case we have to fill a
gap */
codec = avcodec_find_encoder( CODEC_ID_AC3 );
c = avcodec_alloc_context();
- c->bit_rate = sync->audio->bitrate;
- c->sample_rate = sync->audio->rate;
- c->channels = 2;
+ c->bit_rate = sync->audio->config.in.bitrate;
+ 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;
zeros = calloc( AC3_SAMPLES_PER_FRAME *
sizeof( short ) * c->channels, 1 );
- sync->ac3_size = sync->audio->bitrate * AC3_SAMPLES_PER_FRAME /
- sync->audio->rate / 8;
+ sync->ac3_size = sync->audio->config.in.bitrate * AC3_SAMPLES_PER_FRAME /
+ sync->audio->config.in.samplerate / 8;
sync->ac3_buf = malloc( sync->ac3_size );
if( avcodec_encode_audio( c, sync->ac3_buf, sync->ac3_size,
{
hb_log( "sync: avcodec_encode_audio failed" );
}
-
+
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->amixdown), &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;
}
}
-
-
-#define PTS_DISCONTINUITY_TOLERANCE 90000
-
/***********************************************************************
* SyncVideo
***********************************************************************
- *
+ *
**********************************************************************/
-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;
- int64_t pts_expected;
- int chap_break;
+ hb_subtitle_t *subtitle;
+ int i;
- if( pv->done )
+ if( !pv->cur && !( pv->cur = hb_fifo_get( job->fifo_raw ) ) )
{
- return HB_WORK_DONE;
+ /* We haven't even got a frame yet */
+ return;
}
-
- if( hb_thread_has_exited( job->reader ) &&
- !hb_fifo_size( job->fifo_mpeg2 ) &&
- !hb_fifo_size( job->fifo_raw ) )
+ cur = pv->cur;
+ if( cur->size == 0 )
{
- /* All video data has been processed already, we won't get
- more */
- hb_log( "sync: got %lld frames, %lld expected",
- pv->count_frames, pv->count_frames_max );
- pv->done = 1;
-
- hb_buffer_t * buf_tmp;
+ /* we got an end-of-stream. Feed it downstream & signal that we're done. */
+ hb_fifo_push( job->fifo_sync, hb_buffer_init( 0 ) );
- // 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;
- }
+ /*
+ * Push through any subtitle EOFs in case they were not synced through.
+ */
+ for( i = 0; i < hb_list_count( job->list_subtitle ); i++)
+ {
+ subtitle = hb_list_item( job->list_subtitle, i );
+ if( subtitle->dest == PASSTHRUSUB )
+ {
+ hb_fifo_push( subtitle->fifo_out, hb_buffer_init( 0 ) );
+ }
+ }
- if( !pv->cur && !( pv->cur = hb_fifo_get( job->fifo_raw ) ) )
- {
- /* We haven't even got a frame yet */
- return HB_WORK_OK;
+ pv->busy &=~ 1;
+ return;
}
- cur = pv->cur;
/* At this point we have a frame to process. Let's check
1) if we will be able to push into the fifo ahead
2) if the next frame is there already, since we need it to
- know whether we'll have to repeat the current frame or not */
+ compute the duration of the current frame*/
while( !hb_fifo_is_full( job->fifo_sync ) &&
( next = hb_fifo_see( job->fifo_raw ) ) )
{
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 ) );
+
+ /*
+ * Push through any subtitle EOFs in case they were not synced through.
+ */
+ for( i = 0; i < hb_list_count( job->list_subtitle ); i++)
+ {
+ subtitle = hb_list_item( job->list_subtitle, i );
+ if( subtitle->dest == PASSTHRUSUB )
+ {
+ hb_fifo_push( subtitle->fifo_out, hb_buffer_init( 0 ) );
+ }
+ }
+ pv->busy &=~ 1;
+ return;
+ }
if( pv->pts_offset == INT64_MIN )
{
/* This is our first frame */
- hb_log( "sync: first pts is %lld", cur->start );
- pv->pts_offset = cur->start;
+ pv->pts_offset = 0;
+ if ( cur->start != 0 )
+ {
+ /*
+ * The first pts from a dvd should always be zero but
+ * can be non-zero with a transport or program stream since
+ * we're not guaranteed to start on an IDR frame. If we get
+ * a non-zero initial PTS extend its duration so it behaves
+ * as if it started at zero so that our audio timing will
+ * be in sync.
+ */
+ hb_log( "sync: first pts is %lld", cur->start );
+ cur->start = 0;
+ }
+ }
+
+ 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
+ * to deal with the next frame's start being in the past. This can
+ * happen when the PTS is adjusted after data loss but video frame
+ * reordering causes some frames with the old clock to appear after
+ * the clock change. This creates frames that overlap in time which
+ * looks to us like time going backward. The downstream muxing code
+ * can deal with overlaps of up to a frame time but anything larger
+ * we handle by dropping frames here.
+ */
+ if ( (int64_t)( next->start - cur->start ) <= 0 ||
+ (int64_t)( (cur->start - pv->audio_passthru_slip ) - pv->next_pts ) < 0 )
+ {
+ if ( pv->first_drop == 0 )
+ {
+ pv->first_drop = next->start;
+ }
+ ++pv->drop_count;
+ buf_tmp = hb_fifo_get( job->fifo_raw );
+ if ( buf_tmp->new_chap )
+ {
+ // don't drop a chapter mark when we drop the buffer
+ pv->chap_mark = buf_tmp->new_chap;
+ }
+ hb_buffer_close( &buf_tmp );
+ continue;
+ }
+ if ( pv->first_drop )
+ {
+ hb_log( "sync: video time didn't advance - dropped %d frames "
+ "(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;
+ }
+ /*
* Track the video sequence number localy so that we can sync the audio
* to it using the sequence number as well as the PTS.
*/
pv->video_sequence = cur->sequence;
- /* Check for PTS jumps over 0.5 second */
- if( next->start < cur->start - PTS_DISCONTINUITY_TOLERANCE ||
- next->start > cur->start + PTS_DISCONTINUITY_TOLERANCE )
+ /*
+ * Look for a subtitle for this frame.
+ *
+ * If found then it will be tagged onto a video buffer of the correct time and
+ * sent in to the render pipeline. This only needs to be done for VOBSUBs which
+ * get rendered, other types of subtitles can just sit in their raw_queue until
+ * delt with at muxing.
+ */
+ for( i = 0; i < hb_list_count( job->list_subtitle ); i++)
{
- hb_log( "Sync: Video PTS discontinuity (current buffer start=%lld, next buffer start=%lld), trash Video",
- cur->start, next->start );
-
+ subtitle = hb_list_item( job->list_subtitle, i );
+
/*
- * Do we need to trash the subtitle, is it from the next->start period
- * or is it from our old position. If the latter then trash it.
+ * Rewrite timestamps on subtitles that need it (on raw queue).
*/
- if( pv->subtitle )
+ if( subtitle->source == CC608SUB ||
+ subtitle->source == CC708SUB )
{
- while( ( sub = hb_fifo_see( pv->subtitle->fifo_raw ) ) )
+ /*
+ * Rewrite timestamps on subtitles that came from Closed Captions
+ * since they are using the MPEG2 timestamps.
+ */
+ while( ( sub = hb_fifo_see( subtitle->fifo_raw ) ) )
{
- if( ( sub->start > ( cur->start - PTS_DISCONTINUITY_TOLERANCE ) ) &&
- ( sub->start < ( cur->start + PTS_DISCONTINUITY_TOLERANCE ) ) )
+ /*
+ * Rewrite the timestamps as and when the video
+ * (cur->start) reaches the same timestamp as a
+ * closed caption (sub->start).
+ *
+ * What about discontinuity boundaries - not delt
+ * with here - Van?
+ *
+ * Bypass the sync fifo altogether.
+ */
+ if( sub->size <= 0 )
{
+ sub = hb_fifo_get( subtitle->fifo_raw );
+ hb_fifo_push( subtitle->fifo_out, sub );
+ sub = NULL;
+ break;
+ } else {
/*
- * The subtitle is from our current time region which we are
- * jumping from. So trash it as we are about to jump backwards
- * or forwards and don't want it blocking the subtitle fifo.
+ * Sync the subtitles to the incoming video, and use
+ * the matching converted video timestamp.
+ *
+ * Note that it doesn't appear that we need to convert
+ * timestamps, I guess that they were already correct,
+ * so just push them through for rendering.
+ *
*/
- hb_log("Trashing subtitle 0x%x due to PTS discontinuity", sub);
- sub = hb_fifo_get( pv->subtitle->fifo_raw );
- hb_buffer_close( &sub );
+ if( sub->start < cur->start )
+ {
+ uint64_t duration;
+ duration = sub->stop - sub->start;
+ sub = hb_fifo_get( subtitle->fifo_raw );
+ hb_fifo_push( subtitle->fifo_out, sub );
+ } else {
+ sub = NULL;
+ break;
+ }
}
}
}
- /* Trash current picture */
- /* Also, make sure we don't trash a chapter break */
- chap_break = cur->new_chap;
- hb_buffer_close( &cur );
- pv->cur = cur = hb_fifo_get( job->fifo_raw );
- cur->new_chap |= chap_break; // Don't stomp existing chapter breaks
-
- /* Calculate new offset */
- pv->pts_offset_old = pv->pts_offset;
- pv->pts_offset = cur->start -
- pv->count_frames * pv->job->vrate_base / 300;
-
- pv->video_sequence = cur->sequence;
- continue;
- }
-
- /* Look for a subtitle for this frame */
- if( pv->subtitle )
- {
- hb_buffer_t * sub2;
- while( ( sub = hb_fifo_see( pv->subtitle->fifo_raw ) ) )
+ if( subtitle->source == VOBSUB )
{
- /* If two subtitles overlap, make the first one stop
- when the second one starts */
- sub2 = hb_fifo_see2( pv->subtitle->fifo_raw );
- if( sub2 && sub->stop > sub2->start )
- sub->stop = sub2->start;
-
- // hb_log("0x%x: video seq: %lld subtitle sequence: %lld",
- // sub, cur->sequence, sub->sequence);
-
- if( sub->sequence > cur->sequence )
+ hb_buffer_t * sub2;
+ while( ( sub = hb_fifo_see( subtitle->fifo_raw ) ) )
{
- /*
- * The video is behind where we are, so wait until
- * it catches up to the same reader point on the
- * DVD. Then our PTS should be in the same region
- * as the video.
- */
- sub = NULL;
- break;
- }
-
- if( sub->stop > cur->start ) {
- /*
- * The stop time is in the future, so fall through
- * and we'll deal with it in the next block of
- * code.
- */
- break;
- }
- else
- {
- /*
- * The stop time is in the past. But is it due to
- * it having been played already, or has the PTS
- * been reset to 0?
- */
- if( ( cur->start - sub->stop ) > PTS_DISCONTINUITY_TOLERANCE ) {
+ if( sub->size == 0 )
+ {
/*
- * There is a lot of time between our current
- * video and where this subtitle is ending,
- * assume that we are about to reset the PTS
- * and do not throw away this subtitle.
+ * EOF, pass it through immediately.
*/
break;
}
- }
- /*
- * The subtitle is older than this picture, trash it
- */
- sub = hb_fifo_get( pv->subtitle->fifo_raw );
- hb_buffer_close( &sub );
- }
-
- /*
- * There is a valid subtitle, is it time to display it?
- */
- if( sub )
- {
- if( sub->stop > sub->start)
- {
- /*
- * Normal subtitle which ends after it starts, check to
- * see that the current video is between the start and end.
- */
- if( cur->start > sub->start &&
- cur->start < sub->stop )
+ /* If two subtitles overlap, make the first one stop
+ when the second one starts */
+ sub2 = hb_fifo_see2( subtitle->fifo_raw );
+ if( sub2 && sub->stop > sub2->start )
+ sub->stop = sub2->start;
+
+ // hb_log("0x%x: video seq: %lld subtitle sequence: %lld",
+ // sub, cur->sequence, sub->sequence);
+
+ if( sub->sequence > cur->sequence )
{
/*
- * We should be playing this, so leave the
- * subtitle in place.
- *
- * fall through to display
+ * The video is behind where we are, so wait until
+ * it catches up to the same reader point on the
+ * DVD. Then our PTS should be in the same region
+ * as the video.
*/
- }
- else
- {
+ sub = NULL;
+ break;
+ }
+
+ if( sub->stop > cur->start ) {
/*
- * Defer until the play point is within the subtitle
+ * The stop time is in the future, so fall through
+ * and we'll deal with it in the next block of
+ * code.
*/
- sub = NULL;
+ break;
}
+
+ /*
+ * The subtitle is older than this picture, trash it
+ */
+ sub = hb_fifo_get( subtitle->fifo_raw );
+ hb_buffer_close( &sub );
}
- else
+
+ if( sub && sub->size == 0 )
{
- /*
- * The end of the subtitle is less than the start, this is a
- * sign of a PTS discontinuity.
+ /*
+ * Continue immediately on subtitle EOF
*/
- if( sub->start > cur->start )
+ break;
+ }
+
+ /*
+ * There is a valid subtitle, is it time to display it?
+ */
+ if( sub )
+ {
+ if( sub->stop > sub->start)
{
/*
- * we haven't reached the start time yet, or
- * we have jumped backwards after having
- * already started this subtitle.
+ * Normal subtitle which ends after it starts, check to
+ * see that the current video is between the start and end.
*/
- if( cur->start < sub->stop )
+ if( cur->start > sub->start &&
+ cur->start < sub->stop )
{
/*
- * We have jumped backwards and so should
- * continue displaying this subtitle.
+ * We should be playing this, so leave the
+ * subtitle in place.
*
- * fall through to display.
+ * fall through to display
*/
- }
- else
+ if( ( sub->stop - sub->start ) < ( 3 * 90000 ) )
+ {
+ /*
+ * Subtitle is on for less than three seconds, extend
+ * the time that it is displayed to make it easier
+ * to read. Make it 3 seconds or until the next
+ * subtitle is displayed.
+ *
+ * This is in response to Indochine which only
+ * displays subs for 1 second - too fast to read.
+ */
+ sub->stop = sub->start + ( 3 * 90000 );
+
+ sub2 = hb_fifo_see2( subtitle->fifo_raw );
+
+ if( sub2 && sub->stop > sub2->start )
+ {
+ sub->stop = sub2->start;
+ }
+ }
+ }
+ else
{
/*
* Defer until the play point is within the subtitle
*/
sub = NULL;
}
- } else {
+ }
+ else
+ {
/*
- * Play this subtitle as the start is greater than our
- * video point.
- *
- * fall through to display/
+ * The end of the subtitle is less than the start, this is a
+ * sign of a PTS discontinuity.
*/
+ if( sub->start > cur->start )
+ {
+ /*
+ * we haven't reached the start time yet, or
+ * we have jumped backwards after having
+ * already started this subtitle.
+ */
+ if( cur->start < sub->stop )
+ {
+ /*
+ * We have jumped backwards and so should
+ * continue displaying this subtitle.
+ *
+ * fall through to display.
+ */
+ }
+ else
+ {
+ /*
+ * Defer until the play point is within the subtitle
+ */
+ sub = NULL;
+ }
+ } else {
+ /*
+ * Play this subtitle as the start is greater than our
+ * video point.
+ *
+ * fall through to display/
+ */
+ }
}
}
}
- }
-
- /* The PTS of the frame we are expecting now */
- pts_expected = pv->pts_offset +
- pv->count_frames * pv->job->vrate_base / 300;
-
- //hb_log("Video expecting PTS %lld, current frame: %lld, next frame: %lld, cf: %lld",
- // pts_expected, cur->start, next->start, pv->count_frames * pv->job->vrate_base / 300 );
+ if( sub )
+ {
+ /*
+ * Got a sub to display...
+ */
+ break;
+ }
+ } // end subtitles
- if( cur->start < pts_expected - pv->job->vrate_base / 300 / 2 &&
- next->start < pts_expected + pv->job->vrate_base / 300 / 2 )
+ /*
+ * 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;
+ int64_t duration = cur->start - buf_tmp->start;
+ if ( duration <= 0 )
{
- /* The current frame is too old but the next one matches,
- let's trash */
- /* Also, make sure we don't trash a chapter break */
- chap_break = cur->new_chap;
- hb_buffer_close( &cur );
- pv->cur = cur = hb_fifo_get( job->fifo_raw );
- cur->new_chap |= chap_break; // Make sure we don't stomp the existing one.
-
- continue;
+ hb_log( "sync: invalid video duration %lld, start %lld, next %lld",
+ duration, buf_tmp->start, next->start );
}
- if( next->start > pts_expected + 3 * pv->job->vrate_base / 300 / 2 )
- {
- /* We'll need the current frame more than one time. Make a
- copy of it and keep it */
- buf_tmp = hb_buffer_init( cur->size );
- memcpy( buf_tmp->data, cur->data, cur->size );
- buf_tmp->sequence = cur->sequence;
- }
- else
+ buf_tmp->start = pv->next_start;
+ pv->next_start += duration;
+ buf_tmp->stop = pv->next_start;
+
+ if ( pv->chap_mark )
{
- /* The frame has the expected date and won't have to be
- duplicated, just put it through */
- buf_tmp = cur;
- pv->cur = cur = hb_fifo_get( job->fifo_raw );
+ // we have a pending chapter mark from a recent drop - put it on this
+ // buffer (this may make it one frame late but we can't do any better).
+ buf_tmp->new_chap = pv->chap_mark;
+ pv->chap_mark = 0;
}
-
- /* Replace those MPEG-2 dates with our dates */
- buf_tmp->start = (uint64_t) pv->count_frames *
- pv->job->vrate_base / 300;
- buf_tmp->stop = (uint64_t) ( pv->count_frames + 1 ) *
- pv->job->vrate_base / 300;
/* If we have a subtitle for this picture, copy it */
/* FIXME: we should avoid this memcpy */
- if( sub )
+ if( sub && subtitle &&
+ subtitle->format == PICTURESUB )
{
- buf_tmp->sub = hb_buffer_init( sub->size );
- buf_tmp->sub->x = sub->x;
- buf_tmp->sub->y = sub->y;
- buf_tmp->sub->width = sub->width;
- buf_tmp->sub->height = sub->height;
- memcpy( buf_tmp->sub->data, sub->data, sub->size );
+ if( sub->size > 0 )
+ {
+ if( subtitle->dest == RENDERSUB )
+ {
+ /*
+ * Tack onto the video buffer for rendering
+ */
+ buf_tmp->sub = hb_buffer_init( sub->size );
+ buf_tmp->sub->x = sub->x;
+ buf_tmp->sub->y = sub->y;
+ buf_tmp->sub->width = sub->width;
+ buf_tmp->sub->height = sub->height;
+ memcpy( buf_tmp->sub->data, sub->data, sub->size );
+ } else {
+ /*
+ * Pass-Through, pop it off of the raw queue, rewrite times and
+ * make it available to be reencoded.
+ */
+ uint64_t sub_duration;
+ sub = hb_fifo_get( subtitle->fifo_raw );
+ sub_duration = sub->stop - sub->start;
+ sub->start = buf_tmp->start;
+ sub->stop = sub->start + duration;
+ hb_fifo_push( subtitle->fifo_sync, sub );
+ }
+ } else {
+ /*
+ * EOF - consume for rendered, else pass through
+ */
+ if( subtitle->dest == RENDERSUB )
+ {
+ sub = hb_fifo_get( subtitle->fifo_raw );
+ hb_buffer_close( &sub );
+ } else {
+ sub = hb_fifo_get( subtitle->fifo_raw );
+ hb_fifo_push( subtitle->fifo_out, sub );
+ }
+ }
}
/* Push the frame to the renderer */
/* Update UI */
UpdateState( w );
+
+ if( job->frame_to_stop && pv->count_frames > job->frame_to_stop )
+ {
+ // 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;
+ hb_log( "sync: reached %d frames, exiting early (%i busy)",
+ pv->count_frames, pv->busy );
+ return;
+ }
/* Make sure we won't get more frames then expected */
- if( pv->count_frames >= pv->count_frames_max )
+ if( pv->count_frames >= pv->count_frames_max * 2)
{
- hb_log( "sync: got %lld frames", 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 );
-
- break;
+ hb_log( "sync: got too many frames (%d), exiting early",
+ pv->count_frames );
+
+ // 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;
}
}
+}
+
+static void OutputAudioFrame( hb_job_t *job, hb_audio_t *audio, hb_buffer_t *buf,
+ hb_sync_audio_t *sync, hb_fifo_t *fifo, int i )
+{
+ int64_t start = sync->next_start;
+ int64_t duration = buf->stop - buf->start;
+
+ sync->next_pts += duration;
- return HB_WORK_OK;
+ 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
+ * pass-thru just send the input buffer downstream after adjusting
+ * its timestamps to make the output stream continuous.
+ */
+ }
+ else
+ {
+ /* Not pass-thru - do sample rate conversion */
+ int count_in, count_out;
+ hb_buffer_t * buf_raw = buf;
+ int channel_count = HB_AMIXDOWN_GET_DISCRETE_CHANNEL_COUNT(audio->config.out.mixdown) *
+ sizeof( float );
+
+ count_in = buf_raw->size / channel_count;
+ /*
+ * 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)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;
+ sync->data.data_out = (float *) buf->data;
+ if( src_process( sync->state, &sync->data ) )
+ {
+ /* XXX If this happens, we're screwed */
+ hb_log( "sync: audio %d src_process failed", i );
+ }
+ 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;
+ sync->next_start = start + duration;
+ hb_fifo_push( fifo, buf );
}
/***********************************************************************
* SyncAudio
***********************************************************************
- *
+ *
**********************************************************************/
static void SyncAudio( hb_work_object_t * w, int i )
{
hb_work_private_t * pv = w->private_data;
- hb_job_t * job;
- hb_audio_t * audio;
+ hb_job_t * job = pv->job;
+ hb_sync_audio_t * sync = &pv->sync_audio[i];
+ hb_audio_t * audio = sync->audio;
hb_buffer_t * buf;
- hb_sync_audio_t * sync;
-
hb_fifo_t * fifo;
- int rate;
+ int64_t start;
- int64_t pts_expected;
- int64_t start;
-
- job = pv->job;
- sync = &pv->sync_audio[i];
- audio = sync->audio;
-
- if( job->acodec & HB_ACODEC_AC3 )
+ if( audio->config.out.codec == HB_ACODEC_AC3 ||
+ audio->config.out.codec == HB_ACODEC_DCA )
{
- fifo = audio->fifo_out;
- rate = audio->rate;
+ fifo = audio->priv.fifo_out;
}
else
{
- fifo = audio->fifo_sync;
- rate = job->arate;
+ fifo = audio->priv.fifo_sync;
}
- while( !hb_fifo_is_full( fifo ) &&
- ( buf = hb_fifo_see( audio->fifo_raw ) ) )
+ while( !hb_fifo_is_full( fifo ) && ( buf = hb_fifo_see( audio->priv.fifo_raw ) ) )
{
- /* The PTS of the samples we are expecting now */
- pts_expected = pv->pts_offset + sync->count_frames * 90000 / rate;
-
- // hb_log("Video Sequence: %lld, Audio Sequence: %lld", pv->video_sequence, buf->sequence);
- /*
- * Using the same logic as the Video have we crossed a VOB
- * boundary as detected by the expected PTS and the PTS of our
- * audio being out by more than the tolerance value.
- */
- if( ( buf->start > pts_expected + PTS_DISCONTINUITY_TOLERANCE ||
- buf->start < pts_expected - PTS_DISCONTINUITY_TOLERANCE ) &&
- pv->pts_offset_old > INT64_MIN )
+ start = buf->start - pv->audio_passthru_slip;
+ /* if the next buffer is an eof send it downstream */
+ if ( buf->size <= 0 )
{
- /*
- * Useful debug, but too verbose for normal use.
- */
- hb_log("Sync: Audio discontinuity (sequence: vid %lld aud %lld) (pts %lld < %lld < %lld)",
- pv->video_sequence, buf->sequence,
- pts_expected - PTS_DISCONTINUITY_TOLERANCE, buf->start,
- pts_expected + PTS_DISCONTINUITY_TOLERANCE );
-
- /* There has been a PTS discontinuity, and this frame might
- be from before the discontinuity*/
- pts_expected = pv->pts_offset_old + sync->count_frames *
- 90000 / rate;
-
- /*
- * Is the audio from a valid period given the previous
- * Video PTS. I.e. has there just been a video PTS
- * discontinuity and this audio belongs to the vdeo from
- * before?
- */
- if( buf->start > pts_expected + PTS_DISCONTINUITY_TOLERANCE ||
- buf->start < pts_expected - PTS_DISCONTINUITY_TOLERANCE )
- {
- /*
- * It's outside of our tolerance for where the video
- * is now, and it's outside of the tolerance for
- * where we have been in the case of a VOB change.
- * Try and reconverge regardless. so continue on to
- * our convergence code below which will kick in as
- * it will be more than 100ms out.
- *
- * Note that trashing the Audio could make things
- * worse if the Audio is in front because we will end
- * up diverging even more. We need to hold on to the
- * audio until the video catches up.
- */
- hb_log("Sync: Audio is way out of sync, attempt to reconverge from current video PTS");
-
- /*
- * It wasn't from the old place, so we must be from
- * the new, but just too far out. So attempt to
- * reconverge by resetting the point we want to be to
- * where we are currently wanting to be.
- */
- pts_expected = pv->pts_offset + sync->count_frames * 90000 / rate;
- start = pts_expected - pv->pts_offset;
- } else {
- /* Use the older offset */
- start = pts_expected - pv->pts_offset_old;
- }
+ buf = hb_fifo_get( audio->priv.fifo_raw );
+ hb_fifo_push( fifo, buf );
+ pv->busy &=~ (1 << (i + 1) );
+ return;
}
- else
+ if( job->frame_to_stop && pv->count_frames >= job->frame_to_stop )
{
- start = pts_expected - pv->pts_offset;
+ hb_fifo_push( fifo, hb_buffer_init(0) );
+ pv->busy &=~ (1 << (i + 1) );
+ return;
}
-
- /* Tolerance: 100 ms */
- if( buf->start < pts_expected - 9000 )
+ if ( (int64_t)( start - sync->next_pts ) < 0 )
{
- /* Audio is behind the Video, trash it, can't use it now. */
- hb_log( "Sync: Audio PTS (%lld) < Video PTS (%lld) by greater than 100ms, trashing audio to reconverge",
- buf->start, pts_expected);
- buf = hb_fifo_get( audio->fifo_raw );
- hb_buffer_close( &buf );
- continue;
+ // 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 - start > 90*15 )
+ {
+ // 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->next_pts = start;
}
- else if( buf->start > pts_expected + 9000 )
+ if ( sync->first_drop )
{
- /* Audio is ahead of the Video, insert silence until we catch up*/
- hb_log("Sync: Audio PTS (%lld) > Video PTS (%lld) by greater than 100ms insert silence until reconverged", buf->start, pts_expected);
- InsertSilence( w, i );
- continue;
+ // 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 = start;
}
-
- if( job->acodec & HB_ACODEC_AC3 )
+ if ( start - sync->next_pts >= (90 * 70) )
{
- buf = hb_fifo_get( audio->fifo_raw );
- buf->start = start;
- buf->stop = start + 90000 * AC3_SAMPLES_PER_FRAME / rate;
-
- sync->count_frames += AC3_SAMPLES_PER_FRAME;
- }
- else
- {
- hb_buffer_t * buf_raw = hb_fifo_get( audio->fifo_raw );
-
- int count_in, count_out;
-
- count_in = buf_raw->size / HB_AMIXDOWN_GET_DISCRETE_CHANNEL_COUNT(audio->amixdown) / sizeof( float );
- count_out = ( buf_raw->stop - buf_raw->start ) * job->arate / 90000;
- if( buf->start < pts_expected - 1500 )
- count_out--;
- else if( buf->start > pts_expected + 1500 )
- count_out++;
-
- sync->data.data_in = (float *) buf_raw->data;
- sync->data.input_frames = count_in;
- sync->data.output_frames = count_out;
-
- sync->data.src_ratio = (double) sync->data.output_frames /
- (double) sync->data.input_frames;
-
- buf = hb_buffer_init( sync->data.output_frames * HB_AMIXDOWN_GET_DISCRETE_CHANNEL_COUNT(audio->amixdown) *
- sizeof( float ) );
- sync->data.data_out = (float *) buf->data;
- if( src_process( sync->state, &sync->data ) )
+ if ( start - sync->next_pts > (90000LL * 60) )
{
- /* XXX If this happens, we're screwed */
- hb_log( "sync: src_process failed" );
+ // 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)((start - sync->next_pts) / (90000*60)),
+ i, start, sync->next_pts );
+ buf = hb_fifo_get( audio->priv.fifo_raw );
+ hb_buffer_close( &buf );
+ continue;
}
- hb_buffer_close( &buf_raw );
-
- buf->size = sync->data.output_frames_gen * HB_AMIXDOWN_GET_DISCRETE_CHANNEL_COUNT(audio->amixdown) * sizeof( float );
-
- /* Set dates for resampled data */
- buf->start = start;
- buf->stop = start + sync->data.output_frames_gen *
- 90000 / job->arate;
-
- sync->count_frames += sync->data.output_frames_gen;
+ /*
+ * there's a gap of at least 70ms between the last
+ * frame we processed & the next. Fill it with silence.
+ * Or in the case of DCA, skip some frames from the
+ * other streams.
+ */
+ if( sync->audio->config.out.codec == HB_ACODEC_DCA )
+ {
+ hb_log( "sync: audio gap %d ms. Skipping frames. Audio %d"
+ " start %lld, next %lld",
+ (int)((start - sync->next_pts) / 90),
+ i, start, sync->next_pts );
+ pv->audio_passthru_slip += (start - sync->next_pts);
+ return;
+ }
+ hb_log( "sync: adding %d ms of silence to audio %d"
+ " start %lld, next %lld",
+ (int)((start - sync->next_pts) / 90),
+ i, start, sync->next_pts );
+ InsertSilence( w, i, start - sync->next_pts );
+ return;
}
- buf->frametype = HB_FRAME_AUDIO;
- hb_fifo_push( fifo, buf );
- }
-
- if( NeedSilence( w, audio ) )
- {
- InsertSilence( w, i );
- }
-}
-
-static int NeedSilence( hb_work_object_t * w, hb_audio_t * audio )
-{
- hb_work_private_t * pv = w->private_data;
- hb_job_t * job = pv->job;
-
- if( hb_fifo_size( audio->fifo_in ) ||
- hb_fifo_size( audio->fifo_raw ) ||
- hb_fifo_size( audio->fifo_sync ) ||
- hb_fifo_size( audio->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 */
- hb_log("Reader has exited early, inserting silence.");
- return 1;
- }
-
- if( hb_fifo_is_full( job->fifo_mpeg2 ) &&
- hb_fifo_is_full( job->fifo_raw ) &&
- hb_fifo_is_full( job->fifo_sync ) &&
- hb_fifo_is_full( job->fifo_render ) &&
- hb_fifo_is_full( job->fifo_mpeg4 ) )
- {
- /* Too much video and no audio, oh-oh */
- hb_log("Still got some video - and nothing in the audio fifo, insert silence");
- return 1;
+ /*
+ * When we get here we've taken care of all the dups and gaps in the
+ * audio stream and are ready to inject the next input frame into
+ * the output stream.
+ */
+ buf = hb_fifo_get( audio->priv.fifo_raw );
+ OutputAudioFrame( job, audio, buf, sync, fifo, i );
}
-
- return 0;
}
-static void InsertSilence( hb_work_object_t * w, int i )
+static void InsertSilence( hb_work_object_t * w, int i, int64_t duration )
{
hb_work_private_t * pv = w->private_data;
- hb_job_t * job;
- hb_sync_audio_t * sync;
- hb_buffer_t * buf;
-
- job = pv->job;
- sync = &pv->sync_audio[i];
-
- if( job->acodec & HB_ACODEC_AC3 )
- {
- buf = hb_buffer_init( sync->ac3_size );
- buf->start = sync->count_frames * 90000 / sync->audio->rate;
- buf->stop = buf->start + 90000 * AC3_SAMPLES_PER_FRAME /
- sync->audio->rate;
- memcpy( buf->data, sync->ac3_buf, buf->size );
-
- hb_log( "sync: adding a silent AC-3 frame for track %x",
- sync->audio->id );
- hb_fifo_push( sync->audio->fifo_out, buf );
-
- sync->count_frames += AC3_SAMPLES_PER_FRAME;
-
- }
- else
+ hb_job_t *job = pv->job;
+ hb_sync_audio_t *sync = &pv->sync_audio[i];
+ hb_buffer_t *buf;
+ hb_fifo_t *fifo;
+
+ // 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( HB_AMIXDOWN_GET_DISCRETE_CHANNEL_COUNT(sync->audio->amixdown) * job->arate / 20 *
- sizeof( float ) );
- buf->start = sync->count_frames * 90000 / job->arate;
- buf->stop = buf->start + 90000 / 20;
- memset( buf->data, 0, buf->size );
-
- hb_log( "sync: adding 50 ms of silence for track %x",
- sync->audio->id );
- hb_fifo_push( sync->audio->fifo_sync, buf );
-
- sync->count_frames += job->arate / 20;
+ 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 );
}
}
3 * sizeof( uint64_t ) );
pv->st_dates[3] = hb_get_date();
pv->st_counts[3] = pv->count_frames;
- }
+ }
#define p state.param.working
state.state = HB_STATE_WORKING;
p.progress = (float) pv->count_frames / (float) pv->count_frames_max;
if( p.progress > 1.0 )
{
- p.progress = 1.0;
+ p.progress = 1.0;
}
p.rate_cur = 1000.0 *
(float) ( pv->st_counts[3] - pv->st_counts[0] ) /
OSDN Git Service
