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|
/* reader.c
Copyright (c) 2003-2016 HandBrake Team
This file is part of the HandBrake source code
Homepage: <http://handbrake.fr/>.
It may be used under the terms of the GNU General Public License v2.
For full terms see the file COPYING file or visit http://www.gnu.org/licenses/gpl-2.0.html
*/
#include "hb.h"
static int reader_init( hb_work_object_t * w, hb_job_t * job );
static void reader_close( hb_work_object_t * w );
static int reader_work( hb_work_object_t * w, hb_buffer_t ** buf_in,
hb_buffer_t ** buf_out);
hb_work_object_t hb_reader =
{
.id = WORK_READER,
.name = "Reader",
.init = reader_init,
.work = reader_work,
.close = reader_close,
.info = NULL,
.bsinfo = NULL,
.flush = NULL
};
typedef struct
{
int startup;
double average; // average time between packets
double filtered_average; // average time between packets
int64_t last; // last timestamp seen on this stream
int id; // stream id
int is_audio; // != 0 if this is an audio stream
int valid; // Stream timing is not valid until next scr.
} stream_timing_t;
typedef struct
{
int id;
hb_buffer_list_t list;
} buffer_splice_list_t;
struct hb_work_private_s
{
hb_handle_t * h;
hb_job_t * job;
hb_title_t * title;
volatile int * die;
hb_bd_t * bd;
hb_dvd_t * dvd;
hb_stream_t * stream;
stream_timing_t *stream_timing;
int64_t scr_offset;
int sub_scr_set;
hb_psdemux_t demux;
int scr_changes;
uint32_t sequence;
uint8_t st_slots; // size (in slots) of stream_timing array
uint8_t saw_video; // != 0 if we've seen video
uint8_t saw_audio; // != 0 if we've seen audio
int start_found; // found pts_to_start point
int64_t pts_to_start;
int chapter_end;
uint64_t st_first;
uint64_t duration;
hb_fifo_t ** fifos;
buffer_splice_list_t * splice_list;
int splice_list_size;
};
/***********************************************************************
* Local prototypes
**********************************************************************/
static hb_fifo_t ** GetFifoForId( hb_work_private_t * r, int id );
static void UpdateState( hb_work_private_t * r, int64_t start);
static hb_buffer_list_t * get_splice_list(hb_work_private_t * r, int id);
/***********************************************************************
* reader_init
***********************************************************************
*
**********************************************************************/
static int hb_reader_open( hb_work_private_t * r )
{
if ( r->title->type == HB_BD_TYPE )
{
if ( !( r->bd = hb_bd_init( r->h, r->title->path ) ) )
return 1;
if(!hb_bd_start(r->bd, r->title))
{
hb_bd_close(&r->bd);
return 1;
}
if (r->job->start_at_preview)
{
// XXX code from DecodePreviews - should go into its own routine
hb_bd_seek(r->bd, (float)r->job->start_at_preview /
(r->job->seek_points ? (r->job->seek_points + 1.0)
: 11.0));
}
else if (r->job->pts_to_start)
{
// Note, bd seeks always put us to an i-frame. no need
// to start decoding early using r->pts_to_start
hb_bd_seek_pts(r->bd, r->job->pts_to_start);
r->duration -= r->job->pts_to_start;
r->job->pts_to_start = 0;
r->start_found = 1;
}
else
{
hb_bd_seek_chapter(r->bd, r->job->chapter_start);
}
if (r->job->angle > 1)
{
hb_bd_set_angle(r->bd, r->job->angle - 1);
}
}
else if (r->title->type == HB_DVD_TYPE)
{
if ( !( r->dvd = hb_dvd_init( r->h, r->title->path ) ) )
return 1;
if(!hb_dvd_start( r->dvd, r->title, r->job->chapter_start))
{
hb_dvd_close(&r->dvd);
return 1;
}
if (r->job->angle)
{
hb_dvd_set_angle(r->dvd, r->job->angle);
}
if (r->job->start_at_preview)
{
hb_dvd_seek(r->dvd, (float)r->job->start_at_preview /
(r->job->seek_points ? (r->job->seek_points + 1.0)
: 11.0));
}
}
else if (r->title->type == HB_STREAM_TYPE ||
r->title->type == HB_FF_STREAM_TYPE)
{
if (!(r->stream = hb_stream_open(r->h, r->title->path, r->title, 0)))
return 1;
if (r->job->start_at_preview)
{
hb_stream_seek(r->stream, (float)(r->job->start_at_preview - 1) /
(r->job->seek_points ? (r->job->seek_points + 1.0)
: 11.0));
}
else if (r->job->pts_to_start)
{
if (hb_stream_seek_ts( r->stream, r->job->pts_to_start ) >= 0)
{
// Seek takes us to the nearest I-frame before the timestamp
// that we want. So we will retrieve the start time of the
// first packet we get, subtract that from pts_to_start, and
// inspect the reset of the frames in sync.
r->start_found = 2;
r->duration -= r->job->pts_to_start;
}
// hb_stream_seek_ts does nothing for TS streams and will return
// an error.
}
else
{
//
// Standard stream, seek to the starting chapter, if set,
// and track the end chapter so that we end at the right time.
hb_chapter_t *chap;
int start = r->job->chapter_start;
chap = hb_list_item(r->job->list_chapter, r->job->chapter_end - 1);
r->chapter_end = chap->index;
if (start > 1)
{
chap = hb_list_item(r->job->list_chapter, start - 1);
start = chap->index;
}
/*
* Seek to the start chapter.
*/
hb_stream_seek_chapter(r->stream, start);
}
}
else
{
// Unknown type, should never happen
return 1;
}
return 0;
}
static int reader_init( hb_work_object_t * w, hb_job_t * job )
{
hb_work_private_t * r;
r = calloc( sizeof( hb_work_private_t ), 1 );
w->private_data = r;
r->h = job->h;
r->job = job;
r->title = job->title;
r->die = job->die;
r->sequence = 0;
r->st_slots = 4;
r->stream_timing = calloc( sizeof(stream_timing_t), r->st_slots );
r->stream_timing[0].id = r->title->video_id;
r->stream_timing[0].average = 90000. * (double)job->vrate.den /
job->vrate.num;
r->stream_timing[0].filtered_average = r->stream_timing[0].average;
r->stream_timing[0].last = -r->stream_timing[0].average;
r->stream_timing[0].valid = 1;
r->stream_timing[0].startup = 10;
r->stream_timing[1].id = -1;
r->demux.last_scr = AV_NOPTS_VALUE;
r->chapter_end = job->chapter_end;
if ( !job->pts_to_start )
r->start_found = 1;
else
{
// The frame at the actual start time may not be an i-frame
// so can't be decoded without starting a little early.
// sync.c will drop early frames.
// Starting a little over 10 seconds early
r->pts_to_start = MAX(0, job->pts_to_start - 1000000);
}
if (job->pts_to_stop)
{
r->duration = job->pts_to_start + job->pts_to_stop;
}
else if (job->frame_to_stop)
{
int frames = job->frame_to_start + job->frame_to_stop;
r->duration = (int64_t)frames * job->title->vrate.den * 90000 /
job->title->vrate.num;
}
else
{
hb_chapter_t *chapter;
int ii;
r->duration = 0;
for (ii = job->chapter_start; ii < job->chapter_end; ii++)
{
chapter = hb_list_item( job->title->list_chapter, ii - 1);
r->duration += chapter->duration;
}
}
// Count number of splice lists needed for merging buffers
// that have been split
int count = 1; // 1 for video
count += hb_list_count( job->list_subtitle );
count += hb_list_count( job->list_audio );
r->splice_list_size = count;
r->splice_list = calloc(count, sizeof(buffer_splice_list_t));
// Initialize stream id's of splice lists
int ii, jj = 0;
r->splice_list[jj++].id = r->title->video_id;
for (ii = 0; ii < hb_list_count(job->list_subtitle); ii++)
{
hb_subtitle_t * subtitle = hb_list_item(job->list_subtitle, ii);
r->splice_list[jj++].id = subtitle->id;
}
for (ii = 0; ii < hb_list_count(job->list_audio); ii++)
{
hb_audio_t * audio = hb_list_item(job->list_audio, ii);
r->splice_list[jj++].id = audio->id;
}
// count also happens to be the upper bound for the number of
// fifos that will be needed (+1 for null terminator)
r->fifos = calloc(count + 1, sizeof(hb_fifo_t*));
// The stream needs to be open before starting the reader thead
// to prevent a race with decoders that may share information
// with the reader. Specifically avcodec needs this.
if ( hb_reader_open( r ) )
{
free( r->stream_timing );
free( r );
return 1;
}
return 0;
}
static void reader_close( hb_work_object_t * w )
{
hb_work_private_t * r = w->private_data;
if ( r == NULL )
{
return;
}
if (r->bd)
{
hb_bd_stop( r->bd );
hb_bd_close( &r->bd );
}
else if (r->dvd)
{
hb_dvd_stop( r->dvd );
hb_dvd_close( &r->dvd );
}
else if (r->stream)
{
hb_stream_close(&r->stream);
}
if ( r->stream_timing )
{
free( r->stream_timing );
}
int ii;
for (ii = 0; ii < r->splice_list_size; ii++)
{
hb_buffer_list_close(&r->splice_list[ii].list);
}
free(r->fifos);
free(r->splice_list);
free(r);
}
static hb_buffer_t * splice_discontinuity( hb_work_private_t *r, hb_buffer_t *buf )
{
// Handle buffers that were split across a PCR discontinuity.
// Rejoin them into a single buffer.
hb_buffer_list_t * list = get_splice_list(r, buf->s.id);
if (list != NULL)
{
hb_buffer_list_append(list, buf);
if (buf->s.split)
{
return NULL;
}
int count = hb_buffer_list_count(list);
if (count > 1)
{
int size = hb_buffer_list_size(list);
hb_buffer_t * b = hb_buffer_init(size);
buf = hb_buffer_list_head(list);
b->s = buf->s;
int pos = 0;
while ((buf = hb_buffer_list_rem_head(list)) != NULL)
{
memcpy(b->data + pos, buf->data, buf->size);
pos += buf->size;
hb_buffer_close(&buf);
}
buf = b;
}
else
{
buf = hb_buffer_list_clear(list);
}
}
return buf;
}
static void push_buf( hb_work_private_t *r, hb_fifo_t *fifo, hb_buffer_t *buf )
{
while ( !*r->die && !r->job->done )
{
if ( hb_fifo_full_wait( fifo ) )
{
hb_fifo_push( fifo, buf );
buf = NULL;
break;
}
}
if ( buf )
{
hb_buffer_close( &buf );
}
}
static int is_audio( hb_work_private_t *r, int id )
{
int i;
hb_audio_t *audio;
for( i = 0; ( audio = hb_list_item( r->title->list_audio, i ) ); ++i )
{
if ( audio->id == id )
{
return 1;
}
}
return 0;
}
static int is_subtitle( hb_work_private_t *r, int id )
{
int i;
hb_subtitle_t *sub;
for( i = 0; ( sub = hb_list_item( r->title->list_subtitle, i ) ); ++i )
{
if ( sub->id == id )
{
return 1;
}
}
return 0;
}
// The MPEG STD (Standard Target Decoder) essentially requires that we keep
// per-stream timing so that when there's a timing discontinuity we can
// seemlessly join packets on either side of the discontinuity. This join
// requires that we know the timestamp of the previous packet and the
// average inter-packet time (since we position the new packet at the end
// of the previous packet). The next four routines keep track of this
// per-stream timing.
// find or create the per-stream timing state for 'buf'
static stream_timing_t *id_to_st( hb_work_private_t *r, const hb_buffer_t *buf, int valid )
{
stream_timing_t *st = r->stream_timing;
while ( st->id != buf->s.id && st->id != -1)
{
++st;
}
// if we haven't seen this stream add it.
if ( st->id == -1 )
{
// we keep the steam timing info in an array with some power-of-two
// number of slots. If we don't have two slots left (one for our new
// entry plus one for the "-1" eol) we need to expand the array.
int slot = st - r->stream_timing;
if ( slot + 1 >= r->st_slots )
{
r->st_slots *= 2;
r->stream_timing = realloc( r->stream_timing, r->st_slots *
sizeof(*r->stream_timing) );
st = r->stream_timing + slot;
}
st->id = buf->s.id;
st->average = 30.*90.;
st->filtered_average = st->average;
st->startup = 10;
st->last = -st->average;
if ( ( st->is_audio = is_audio( r, buf->s.id ) ) != 0 )
{
r->saw_audio = 1;
}
st[1].id = -1;
st->valid = valid;
}
return st;
}
// update the average inter-packet time of the stream associated with 'buf'
// using a recursive low-pass filter with a 16 packet time constant.
static void update_ipt( hb_work_private_t *r, const hb_buffer_t *buf )
{
stream_timing_t *st = id_to_st( r, buf, 1 );
if (buf->s.renderOffset == AV_NOPTS_VALUE)
{
st->last += st->filtered_average;
return;
}
double dt = buf->s.renderOffset - st->last;
// Protect against spurious bad timestamps
// timestamps should only move forward and by reasonable increments
if ( dt > 0 && dt < 5 * 90000LL )
{
if( st->startup )
{
st->average += ( dt - st->average ) * (1./4.);
st->startup--;
}
else
{
st->average += ( dt - st->average ) * (1./32.);
}
// Ignore outliers
if (dt < 1.5 * st->average)
{
st->filtered_average += ( dt - st->filtered_average ) * (1./32.);
}
}
st->last = buf->s.renderOffset;
st->valid = 1;
}
// use the per-stream state associated with 'buf' to compute a new scr_offset
// such that 'buf' will follow the previous packet of this stream separated
// by the average packet time of the stream.
static void new_scr_offset( hb_work_private_t *r, hb_buffer_t *buf )
{
stream_timing_t *st = id_to_st( r, buf, 1 );
int64_t last;
if ( !st->valid )
{
// !valid means we've not received any previous data
// for this stream. There is no 'last' packet time.
// So approximate it with video's last time.
last = r->stream_timing[0].last;
st->valid = 1;
}
else
{
last = st->last;
}
int64_t nxt = last + st->filtered_average;
r->scr_offset = buf->s.renderOffset - nxt;
// This log is handy when you need to debug timing problems...
//hb_log("id %x last %"PRId64" avg %g nxt %"PRId64" renderOffset %"PRId64
// " scr_offset %"PRId64"",
// buf->s.id, last, st->filtered_average, nxt,
// buf->s.renderOffset, r->scr_offset);
r->scr_changes = r->demux.scr_changes;
}
static void reader_send_eof( hb_work_private_t * r )
{
int ii;
// send eof buffers downstream to decoders to signal we're done.
push_buf(r, r->job->fifo_mpeg2, hb_buffer_eof_init());
hb_audio_t *audio;
for (ii = 0; (audio = hb_list_item(r->job->list_audio, ii)); ++ii)
{
if (audio->priv.fifo_in)
push_buf(r, audio->priv.fifo_in, hb_buffer_eof_init());
}
hb_subtitle_t *subtitle;
for (ii = 0; (subtitle = hb_list_item(r->job->list_subtitle, ii)); ++ii)
{
if (subtitle->fifo_in && subtitle->source != SRTSUB)
push_buf(r, subtitle->fifo_in, hb_buffer_eof_init());
}
hb_log("reader: done. %d scr changes", r->demux.scr_changes);
}
static int reader_work( hb_work_object_t * w, hb_buffer_t ** buf_in,
hb_buffer_t ** buf_out)
{
hb_work_private_t * r = w->private_data;
hb_fifo_t ** fifos;
hb_buffer_t * buf;
hb_buffer_list_t list;
int ii, chapter = -1;
hb_buffer_list_clear(&list);
if (r->bd)
chapter = hb_bd_chapter( r->bd );
else if (r->dvd)
chapter = hb_dvd_chapter( r->dvd );
else if (r->stream)
chapter = hb_stream_chapter( r->stream );
if( chapter < 0 )
{
hb_log( "reader: end of the title reached" );
reader_send_eof(r);
return HB_WORK_DONE;
}
if( chapter > r->chapter_end )
{
hb_log("reader: end of chapter %d (media %d) reached at media chapter %d",
r->job->chapter_end, r->chapter_end, chapter);
reader_send_eof(r);
return HB_WORK_DONE;
}
if (r->bd)
{
if( (buf = hb_bd_read( r->bd )) == NULL )
{
reader_send_eof(r);
return HB_WORK_DONE;
}
}
else if (r->dvd)
{
if( (buf = hb_dvd_read( r->dvd )) == NULL )
{
reader_send_eof(r);
return HB_WORK_DONE;
}
}
else if (r->stream)
{
if ( (buf = hb_stream_read( r->stream )) == NULL )
{
reader_send_eof(r);
return HB_WORK_DONE;
}
}
(hb_demux[r->title->demuxer])(buf, &list, &r->demux);
while ((buf = hb_buffer_list_rem_head(&list)) != NULL)
{
fifos = GetFifoForId( r, buf->s.id );
if (fifos && r->stream && r->start_found == 2 )
{
// We will inspect the timestamps of each frame in sync
// to skip from this seek point to the timestamp we
// want to start at.
if (buf->s.start != AV_NOPTS_VALUE &&
buf->s.start < r->job->pts_to_start)
{
r->job->pts_to_start -= buf->s.start;
}
else if ( buf->s.start >= r->job->pts_to_start )
{
r->job->pts_to_start = 0;
}
r->start_found = 1;
}
if ( fifos && ! r->saw_video && !r->job->indepth_scan )
{
// The first data packet with a PTS from an audio or video stream
// that we're decoding defines 'time zero'. Discard packets until
// we get one.
if (buf->s.start != AV_NOPTS_VALUE &&
buf->s.renderOffset != AV_NOPTS_VALUE &&
(buf->s.id == r->title->video_id ||
is_audio( r, buf->s.id)))
{
// force a new scr offset computation
r->scr_changes = r->demux.scr_changes - 1;
// create a stream state if we don't have one so the
// offset will get computed correctly.
id_to_st( r, buf, 1 );
r->saw_video = 1;
hb_log( "reader: first SCR %"PRId64" id 0x%x DTS %"PRId64,
r->demux.last_scr, buf->s.id, buf->s.renderOffset );
}
else
{
fifos = NULL;
}
}
if ( r->job->indepth_scan || fifos )
{
if ( buf->s.renderOffset != AV_NOPTS_VALUE )
{
if ( r->scr_changes != r->demux.scr_changes )
{
// This is the first audio or video packet after an SCR
// change. Compute a new scr offset that would make this
// packet follow the last of this stream with the
// correct average spacing.
stream_timing_t *st = id_to_st( r, buf, 0 );
// if this is the video stream and we don't have
// audio yet or this is an audio stream
// generate a new scr
if ( st->is_audio ||
( st == r->stream_timing && !r->saw_audio ) )
{
new_scr_offset( r, buf );
r->sub_scr_set = 0;
}
else
{
// defer the scr change until we get some
// audio since audio has a timestamp per
// frame but video & subtitles don't. Clear
// the timestamps so the decoder will generate
// them from the frame durations.
if (is_subtitle(r, buf->s.id) &&
buf->s.start != AV_NOPTS_VALUE)
{
if (!r->sub_scr_set)
{
// We can't generate timestamps in the
// subtitle decoder as we can for
// audio & video. So we need to make
// the closest guess that we can
// for the subtitles start time here.
int64_t last = r->stream_timing[0].last;
r->scr_offset = buf->s.start - last;
r->sub_scr_set = 1;
}
}
else
{
buf->s.start = AV_NOPTS_VALUE;
buf->s.renderOffset = AV_NOPTS_VALUE;
}
}
}
}
if ( buf->s.start != AV_NOPTS_VALUE )
{
int64_t start = buf->s.start - r->scr_offset;
if (!r->start_found || r->job->indepth_scan)
{
UpdateState( r, start );
}
if (r->job->indepth_scan && r->job->pts_to_stop &&
start >= r->pts_to_start + r->job->pts_to_stop)
{
// sync normally would terminate p-to-p
// but sync doesn't run during indepth scan
hb_log("reader: reached pts %"PRId64", exiting early", start);
reader_send_eof(r);
hb_buffer_list_close(&list);
return HB_WORK_DONE;
}
if (!r->start_found && start >= r->pts_to_start)
{
// pts_to_start point found
// Note that this code path only gets executed for
// medai where we have not performed an initial seek
// to get close to the start time. So the 'start' time
// is the time since the first frame.
if (r->stream)
{
// libav multi-threaded decoders can get into
// a bad state if the initial data is not
// decodable. So try to improve the chances of
// a good start by waiting for an initial iframe
hb_stream_set_need_keyframe(r->stream, 1);
hb_buffer_close( &buf );
continue;
}
r->start_found = 1;
// sync.c also pays attention to job->pts_to_start
// It eats up the 10 second slack that we build in
// to the start time here in reader (so that video
// decode is clean at the start time).
// sync.c expects pts_to_start to be relative to the
// first timestamp it sees.
if (r->job->pts_to_start > start)
{
r->job->pts_to_start -= start;
}
else
{
r->job->pts_to_start = 0;
}
}
// This log is handy when you need to debug timing problems
//hb_log("id %x scr_offset %"PRId64
// " start %"PRId64" --> %"PRId64"",
// buf->s.id, r->scr_offset, buf->s.start,
// buf->s.start - r->scr_offset);
buf->s.start -= r->scr_offset;
if ( buf->s.stop != AV_NOPTS_VALUE )
{
buf->s.stop -= r->scr_offset;
}
}
if ( buf->s.renderOffset != AV_NOPTS_VALUE )
{
// This packet is referenced to the same SCR as the last.
// Adjust timestamp to remove the System Clock Reference
// offset then update the average inter-packet time
// for this stream.
buf->s.renderOffset -= r->scr_offset;
update_ipt( r, buf );
}
#if 0
// JAS: This was added to fix a rare "audio time went backward"
// sync error I found in one sample. But it has a bad side
// effect on DVDs, causing frequent "adding silence" sync
// errors. So I am disabling it.
else
{
update_ipt( r, buf );
}
#endif
}
buf = splice_discontinuity(r, buf);
if( fifos && buf != NULL )
{
if ( !r->start_found )
{
hb_buffer_close( &buf );
continue;
}
buf->sequence = r->sequence++;
/* if there are mutiple output fifos, send a copy of the
* buffer down all but the first (we have to not ship the
* original buffer or we'll race with the thread that's
* consuming the buffer & inject garbage into the data stream). */
for (ii = 1; fifos[ii] != NULL; ii++)
{
hb_buffer_t *buf_copy = hb_buffer_init(buf->size);
buf_copy->s = buf->s;
memcpy(buf_copy->data, buf->data, buf->size);
push_buf(r, fifos[ii], buf_copy);
}
push_buf(r, fifos[0], buf);
buf = NULL;
}
else
{
hb_buffer_close(&buf);
}
}
hb_buffer_list_close(&list);
return HB_WORK_OK;
}
static void UpdateState( hb_work_private_t * r, int64_t start)
{
hb_state_t state;
uint64_t now;
double avg;
now = hb_get_date();
if( !r->st_first )
{
r->st_first = now;
}
hb_get_state2(r->job->h, &state);
#define p state.param.working
if ( !r->job->indepth_scan )
{
state.state = HB_STATE_SEARCHING;
p.progress = (float) start / (float) r->job->pts_to_start;
}
else
{
state.state = HB_STATE_WORKING;
p.progress = (float) start / (float) r->duration;
}
if( p.progress > 1.0 )
{
p.progress = 1.0;
}
p.rate_cur = 0.0;
p.rate_avg = 0.0;
if (now > r->st_first)
{
int eta;
avg = 1000.0 * (double)start / (now - r->st_first);
if ( !r->job->indepth_scan )
eta = ( r->job->pts_to_start - start ) / avg;
else
eta = ( r->duration - start ) / avg;
p.hours = eta / 3600;
p.minutes = ( eta % 3600 ) / 60;
p.seconds = eta % 60;
}
else
{
p.hours = -1;
p.minutes = -1;
p.seconds = -1;
}
#undef p
hb_set_state( r->job->h, &state );
}
/***********************************************************************
* GetFifoForId
***********************************************************************
*
**********************************************************************/
static hb_fifo_t ** GetFifoForId( hb_work_private_t * r, int id )
{
hb_job_t * job = r->job;
hb_title_t * title = job->title;
hb_audio_t * audio;
hb_subtitle_t * subtitle;
int i, n;
if( id == title->video_id )
{
if (job->indepth_scan && !job->frame_to_stop)
{
/*
* Ditch the video here during the indepth scan until
* we can improve the MPEG2 decode performance.
*
* But if we specify a stop frame, we must decode the
* frames in order to count them.
*/
return NULL;
}
else
{
r->fifos[0] = job->fifo_mpeg2;
r->fifos[1] = NULL;
return r->fifos;
}
}
for( i = n = 0; i < hb_list_count( job->list_subtitle ); i++ )
{
subtitle = hb_list_item( job->list_subtitle, i );
if (id == subtitle->id)
{
/* pass the subtitles to be processed */
r->fifos[n++] = subtitle->fifo_in;
}
}
if ( n != 0 )
{
r->fifos[n] = NULL;
return r->fifos;
}
if( !job->indepth_scan )
{
for( i = n = 0; i < hb_list_count( job->list_audio ); i++ )
{
audio = hb_list_item( job->list_audio, i );
if( id == audio->id )
{
r->fifos[n++] = audio->priv.fifo_in;
}
}
if( n != 0 )
{
r->fifos[n] = NULL;
return r->fifos;
}
}
return NULL;
}
static hb_buffer_list_t * get_splice_list(hb_work_private_t * r, int id)
{
int ii;
for (ii = 0; ii < r->splice_list_size; ii++)
{
if (r->splice_list[ii].id == id)
{
return &r->splice_list[ii].list;
}
}
return NULL;
}
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