/* $Id: sync.c,v 1.38 2005/04/14 21:57:58 titer Exp $
This file is part of the HandBrake source code.
Homepage: .
It may be used under the terms of the GNU General Public License. */
#include "hb.h"
#include
#include "samplerate.h"
#include "libavcodec/avcodec.h"
#ifdef INT64_MIN
#undef INT64_MIN /* Because it isn't defined correctly in Zeta */
#endif
#define INT64_MIN (-9223372036854775807LL-1)
#define AC3_SAMPLES_PER_FRAME 1536
typedef struct
{
hb_audio_t * audio;
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;
/* AC-3 */
int ac3_size;
uint8_t * ac3_buf;
} hb_sync_audio_t;
struct hb_work_private_s
{
hb_job_t * job;
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 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];
/* Statistics */
uint64_t st_counts[4];
uint64_t st_dates[4];
uint64_t st_first;
};
/***********************************************************************
* Local prototypes
**********************************************************************/
static void InitAudio( hb_work_object_t * w, int i );
static void SyncVideo( hb_work_object_t * w );
static void SyncAudio( 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 );
/***********************************************************************
* hb_work_sync_init
***********************************************************************
* Initialize the work object
**********************************************************************/
int syncInit( hb_work_object_t * w, hb_job_t * job )
{
hb_title_t * title = job->title;
hb_chapter_t * chapter;
int i;
uint64_t duration;
hb_work_private_t * pv;
pv = calloc( 1, sizeof( hb_work_private_t ) );
w->private_data = pv;
pv->job = job;
pv->pts_offset = INT64_MIN;
/* Calculate how many video frames we are expecting */
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 %d video frames", pv->count_frames_max );
pv->busy |= 1;
/* Initialize libsamplerate for every audio track we have */
if ( ! job->indepth_scan )
{
for( i = 0; i < hb_list_count( title->list_audio ); i++ )
{
pv->busy |= ( 1 << (i + 1) );
InitAudio( w, i );
}
}
/* Get subtitle info, if any */
pv->subtitle = hb_list_item( title->list_subtitle, 0 );
return 0;
}
/***********************************************************************
* Close
***********************************************************************
*
**********************************************************************/
void syncClose( hb_work_object_t * w )
{
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 );
}
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++ )
{
audio = hb_list_item( title->list_audio, i );
if( audio->config.out.codec == HB_ACODEC_AC3 )
{
free( pv->sync_audio[i].ac3_buf );
}
else
{
src_delete( pv->sync_audio[i].state );
}
}
free( pv );
w->private_data = NULL;
}
/***********************************************************************
* Work
***********************************************************************
* The root routine of this work abject
*
* The way this works is that we are syncing the audio to the PTS of
* the last video that we processed. That's why we skip the audio sync
* if we haven't got a valid PTS from the video yet.
*
**********************************************************************/
int syncWork( hb_work_object_t * w, hb_buffer_t ** unused1,
hb_buffer_t ** unused2 )
{
hb_work_private_t * pv = w->private_data;
int i;
if ( pv->busy & 1 )
SyncVideo( w );
for( i = 0; i < hb_list_count( pv->job->title->list_audio ); i++ )
{
if ( pv->busy & ( 1 << (i + 1) ) )
SyncAudio( w, i );
}
return ( pv->busy? HB_WORK_OK : HB_WORK_DONE );
}
hb_work_object_t hb_sync =
{
WORK_SYNC,
"Synchronization",
syncInit,
syncWork,
syncClose
};
static void InitAudio( hb_work_object_t * w, int i )
{
hb_work_private_t * pv = w->private_data;
hb_job_t * job = pv->job;
hb_title_t * title = job->title;
hb_sync_audio_t * sync;
sync = &pv->sync_audio[i];
sync->audio = hb_list_item( title->list_audio, i );
if( sync->audio->config.out.codec == HB_ACODEC_AC3 )
{
/* Have a silent AC-3 frame ready in case we have to fill a
gap */
AVCodec * codec;
AVCodecContext * c;
short * zeros;
codec = avcodec_find_encoder( CODEC_ID_AC3 );
c = avcodec_alloc_context();
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 )
{
hb_log( "sync: avcodec_open failed" );
return;
}
zeros = calloc( AC3_SAMPLES_PER_FRAME *
sizeof( short ) * c->channels, 1 );
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,
zeros ) != sync->ac3_size )
{
hb_log( "sync: avcodec_encode_audio failed" );
}
free( zeros );
avcodec_close( c );
av_free( c );
}
else
{
/* Initialize libsamplerate */
int 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;
}
}
/***********************************************************************
* SyncVideo
***********************************************************************
*
**********************************************************************/
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;
if( !pv->cur && !( pv->cur = hb_fifo_get( job->fifo_raw ) ) )
{
/* We haven't even got a frame yet */
return;
}
cur = pv->cur;
if( cur->size == 0 )
{
/* we got an end-of-stream. Feed it downstream & signal that we're done. */
hb_fifo_push( job->fifo_sync, hb_buffer_init( 0 ) );
pv->busy &=~ 1;
return;
}
/* 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
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 ) );
pv->busy &=~ 1;
return;
}
if( pv->pts_offset == INT64_MIN )
{
/* This is our first frame */
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;
}
}
/*
* 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 )
{
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;
/* Look for a subtitle for this frame */
if( pv->subtitle )
{
hb_buffer_t * sub2;
while( ( sub = hb_fifo_see( pv->subtitle->fifo_raw ) ) )
{
/* 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 )
{
/*
* 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;
}
/*
* 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 )
{
/*
* We should be playing this, so leave the
* subtitle in place.
*
* fall through to display
*/
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( pv->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
{
/*
* 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/
*/
}
}
}
}
int64_t duration;
if ( job->mux & HB_MUX_AVI || job->cfr )
{
/*
* The concept of variable frame rate video was a bit too advanced
* for Microsoft so AVI doesn't support it. Since almost all dvd
* video is VFR we have to convert it to constant frame rate to
* put it in an AVI container. So here we duplicate, drop and
* otherwise trash video frames to appease the gods of Redmond.
*/
/* mpeg durations are exact when expressed in ticks of the
* 27MHz System clock but not in HB's 90KHz PTS clock. To avoid
* a truncation bias that will eventually cause the audio to desync
* we compute the duration of the next frame using 27MHz ticks
* then truncate it to 90KHz. */
duration = ( (int64_t)(pv->count_frames + 1 ) * job->vrate_base ) / 300 -
pv->next_start;
/* We don't want the input & output clocks to be exactly in phase
* otherwise small variations in the time will cause us to think
* we're a full frame off & there will be lots of drops and dups.
* We offset the input clock by half the duration so it's maximally
* out of phase with the output clock. */
if( cur->start < pv->next_start - ( duration >> 1 ) )
{
/* current frame too old - drop it */
if ( cur->new_chap )
{
pv->chap_mark = cur->new_chap;
}
hb_buffer_close( &cur );
pv->cur = cur = hb_fifo_get( job->fifo_raw );
pv->next_pts = next->start;
++pv->drops;
continue;
}
if( next->start > pv->next_start + duration + ( duration >> 1 ) )
{
/* next frame too far ahead - dup current frame */
buf_tmp = hb_buffer_init( cur->size );
hb_buffer_copy_settings( buf_tmp, cur );
memcpy( buf_tmp->data, cur->data, cur->size );
buf_tmp->sequence = cur->sequence;
++pv->dups;
}
else
{
/* this frame in our time window & doesn't need to be duped */
buf_tmp = cur;
pv->cur = cur = hb_fifo_get( job->fifo_raw );
pv->next_pts = next->start;
}
}
else
{
/*
* 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;
duration = cur->start - buf_tmp->start;
if ( duration <= 0 )
{
hb_log( "sync: invalid video duration %lld, start %lld, next %lld",
duration, buf_tmp->start, next->start );
}
}
buf_tmp->start = pv->next_start;
pv->next_start += duration;
buf_tmp->stop = pv->next_start;
if ( pv->chap_mark )
{
// 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;
}
/* If we have a subtitle for this picture, copy it */
/* FIXME: we should avoid this memcpy */
if( sub )
{
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 );
}
/* Push the frame to the renderer */
hb_fifo_push( job->fifo_sync, buf_tmp );
/* Update UI */
UpdateState( w );
/* Make sure we won't get more frames then expected */
if( pv->count_frames >= pv->count_frames_max * 2)
{
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;
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 = pv->job;
hb_sync_audio_t * sync = &pv->sync_audio[i];
hb_audio_t * audio = sync->audio;
hb_buffer_t * buf;
hb_fifo_t * fifo;
if( audio->config.out.codec == HB_ACODEC_AC3 )
{
fifo = audio->priv.fifo_out;
}
else
{
fifo = audio->priv.fifo_sync;
}
while( !hb_fifo_is_full( fifo ) && ( buf = hb_fifo_see( audio->priv.fifo_raw ) ) )
{
/* if the next buffer is an eof send it downstream */
if ( buf->size <= 0 )
{
buf = hb_fifo_get( audio->priv.fifo_raw );
hb_fifo_push( fifo, buf );
pv->busy &=~ (1 << (i + 1) );
return;
}
if ( (int64_t)( buf->start - sync->next_pts ) < 0 )
{
// 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 - buf->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 = buf->start;
}
if ( sync->first_drop )
{
// 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 = buf->start;
}
if ( buf->start - sync->next_pts >= (90 * 70) )
{
if ( buf->start - sync->next_pts > (90000LL * 60) )
{
// 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)((buf->start - sync->next_pts) / (90000*60)),
i, buf->start, sync->next_pts );
buf = hb_fifo_get( audio->priv.fifo_raw );
hb_buffer_close( &buf );
continue;
}
/*
* there's a gap of at least 70ms between the last
* frame we processed & the next. Fill it with silence.
*/
hb_log( "sync: adding %d ms of silence to audio %d"
" start %lld, next %lld",
(int)((buf->start - sync->next_pts) / 90),
i, buf->start, sync->next_pts );
InsertSilence( w, i, buf->start - sync->next_pts );
return;
}
/*
* 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 );
}
}
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 = 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 )
{
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 );
}
}
static void UpdateState( hb_work_object_t * w )
{
hb_work_private_t * pv = w->private_data;
hb_state_t state;
if( !pv->count_frames )
{
pv->st_first = hb_get_date();
}
pv->count_frames++;
if( hb_get_date() > pv->st_dates[3] + 1000 )
{
memmove( &pv->st_dates[0], &pv->st_dates[1],
3 * sizeof( uint64_t ) );
memmove( &pv->st_counts[0], &pv->st_counts[1],
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.rate_cur = 1000.0 *
(float) ( pv->st_counts[3] - pv->st_counts[0] ) /
(float) ( pv->st_dates[3] - pv->st_dates[0] );
if( hb_get_date() > pv->st_first + 4000 )
{
int eta;
p.rate_avg = 1000.0 * (float) pv->st_counts[3] /
(float) ( pv->st_dates[3] - pv->st_first );
eta = (float) ( pv->count_frames_max - pv->st_counts[3] ) /
p.rate_avg;
p.hours = eta / 3600;
p.minutes = ( eta % 3600 ) / 60;
p.seconds = eta % 60;
}
else
{
p.rate_avg = 0.0;
p.hours = -1;
p.minutes = -1;
p.seconds = -1;
}
#undef p
hb_set_state( pv->job->h, &state );
}