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authorprigaux <[email protected]>2007-03-06 21:15:55 +0000
committerprigaux <[email protected]>2007-03-06 21:15:55 +0000
commitc4a642b4648d486777af91ccc2894355df447089 (patch)
tree4ff9eee3e39a530cb649d9db43456e0ab3d712a3 /libhb/sync.c
parentff5ed5d2a2009504f1da3cf348689a59a82b27f1 (diff)
Merge the 5.1 branch into the trunk.
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@422 b64f7644-9d1e-0410-96f1-a4d463321fa5
Diffstat (limited to 'libhb/sync.c')
-rw-r--r--libhb/sync.c679
1 files changed, 679 insertions, 0 deletions
diff --git a/libhb/sync.c b/libhb/sync.c
new file mode 100644
index 000000000..24ee4a8af
--- /dev/null
+++ b/libhb/sync.c
@@ -0,0 +1,679 @@
+/* $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/>.
+ It may be used under the terms of the GNU General Public License. */
+
+#include "hb.h"
+
+#include "samplerate.h"
+#include "ffmpeg/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 count_frames;
+
+ /* 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 done;
+
+ /* Video */
+ hb_subtitle_t * subtitle;
+ int64_t pts_offset;
+ int64_t pts_offset_old;
+ int64_t count_frames;
+ int64_t count_frames_max;
+ 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, int channelsused );
+static int SyncVideo( hb_work_object_t * w );
+static void SyncAudio( hb_work_object_t * w, int i, int channelsused );
+static int NeedSilence( hb_work_object_t * w, hb_audio_t * );
+static void InsertSilence( hb_work_object_t * w, int i, int channelsused );
+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;
+ 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++ )
+ {
+ 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 );
+
+ /* Initialize libsamplerate for every audio track we have */
+ for( i = 0; i < hb_list_count( title->list_audio ); i++ )
+ {
+ /* this should really pass in a channelsused value for this audio
+ but for now, it uses the global job value of channelsused */
+ InitAudio( w, i, job->channelsused );
+ }
+
+ /* 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;
+
+ int i;
+
+ if( pv->cur ) hb_buffer_close( &pv->cur );
+
+ for( i = 0; i < hb_list_count( title->list_audio ); i++ )
+ {
+ if( job->acodec & HB_ACODEC_AC3 )
+ {
+ free( pv->sync_audio[i].ac3_buf );
+ }
+ else
+ {
+ src_delete( pv->sync_audio[i].state );
+ }
+ }
+}
+
+/***********************************************************************
+ * Work
+ ***********************************************************************
+ * The root routine of this work abject
+ **********************************************************************/
+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 we ever got a video frame, handle audio now */
+ if( pv->pts_offset != INT64_MIN )
+ {
+ for( i = 0; i < hb_list_count( pv->job->title->list_audio ); i++ )
+ {
+ /* this should really pass in a channelsused value for this audio
+ but for now, it uses the global job value of channelsused */
+ SyncAudio( w, i, pv->job->channelsused );
+ }
+ }
+
+ /* Handle video */
+ return SyncVideo( w );
+}
+
+hb_work_object_t hb_sync =
+{
+ WORK_SYNC,
+ "Synchronization",
+ syncInit,
+ syncWork,
+ syncClose
+};
+
+static void InitAudio( hb_work_object_t * w, int i, int channelsused )
+{
+ 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( job->acodec & 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->bitrate;
+ c->sample_rate = sync->audio->rate;
+ c->channels = sync->audio->channels;
+
+ 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->bitrate * AC3_SAMPLES_PER_FRAME /
+ sync->audio->rate / 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_LINEAR, channelsused, &error );
+ sync->data.end_of_input = 0;
+ }
+}
+
+
+
+#define PTS_DISCONTINUITY_TOLERANCE 90000
+
+/***********************************************************************
+ * SyncVideo
+ ***********************************************************************
+ *
+ **********************************************************************/
+static int 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;
+
+ if( pv->done )
+ {
+ return HB_WORK_DONE;
+ }
+
+ if( hb_thread_has_exited( job->reader ) &&
+ !hb_fifo_size( job->fifo_mpeg2 ) &&
+ !hb_fifo_size( job->fifo_raw ) )
+ {
+ /* 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;
+ return HB_WORK_DONE;
+ }
+
+ if( !pv->cur && !( pv->cur = hb_fifo_get( job->fifo_raw ) ) )
+ {
+ /* We haven't even got a frame yet */
+ return HB_WORK_OK;
+ }
+ 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 */
+ while( !hb_fifo_is_full( job->fifo_sync ) &&
+ ( next = hb_fifo_see( job->fifo_raw ) ) )
+ {
+ hb_buffer_t * buf_tmp;
+
+ 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;
+ }
+
+ /* Check for PTS jumps over 0.5 second */
+ if( next->start < cur->start - PTS_DISCONTINUITY_TOLERANCE ||
+ next->start > cur->start + PTS_DISCONTINUITY_TOLERANCE )
+ {
+ hb_log( "PTS discontinuity (%lld, %lld)",
+ cur->start, next->start );
+
+ /* Trash all subtitles */
+ if( pv->subtitle )
+ {
+ while( ( sub = hb_fifo_get( pv->subtitle->fifo_raw ) ) )
+ {
+ hb_buffer_close( &sub );
+ }
+ }
+
+ /* Trash current picture */
+ hb_buffer_close( &cur );
+ pv->cur = cur = hb_fifo_get( job->fifo_raw );
+
+ /* Calculate new offset */
+ pv->pts_offset_old = pv->pts_offset;
+ pv->pts_offset = cur->start -
+ pv->count_frames * pv->job->vrate_base / 300;
+ 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 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;
+
+ if( sub->stop > cur->start )
+ break;
+
+ /* The subtitle is older than this picture, trash it */
+ sub = hb_fifo_get( pv->subtitle->fifo_raw );
+ hb_buffer_close( &sub );
+ }
+
+ /* If we have subtitles left in the fifo, check if we should
+ apply the first one to the current frame or if we should
+ keep it for later */
+ if( sub && sub->start > cur->start )
+ {
+ sub = NULL;
+ }
+ }
+
+ /* The PTS of the frame we are expecting now */
+ pts_expected = pv->pts_offset +
+ pv->count_frames * pv->job->vrate_base / 300;
+
+ if( cur->start < pts_expected - pv->job->vrate_base / 300 / 2 &&
+ next->start < pts_expected + pv->job->vrate_base / 300 / 2 )
+ {
+ /* The current frame is too old but the next one matches,
+ let's trash */
+ hb_buffer_close( &cur );
+ pv->cur = cur = hb_fifo_get( job->fifo_raw );
+ continue;
+ }
+
+ 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 );
+ }
+ else
+ {
+ /* 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 );
+ }
+
+ /* 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 )
+ {
+ 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 )
+ {
+ hb_log( "sync: got %lld frames", pv->count_frames );
+ pv->done = 1;
+ break;
+ }
+ }
+
+ return HB_WORK_OK;
+}
+
+/***********************************************************************
+ * SyncAudio
+ ***********************************************************************
+ *
+ **********************************************************************/
+static void SyncAudio( hb_work_object_t * w, int i, int channelsused )
+{
+ hb_work_private_t * pv = w->private_data;
+ hb_job_t * job;
+ hb_audio_t * audio;
+ hb_buffer_t * buf;
+ hb_sync_audio_t * sync;
+
+ hb_fifo_t * fifo;
+ int rate;
+
+ int64_t pts_expected;
+ int64_t start;
+
+ job = pv->job;
+ sync = &pv->sync_audio[i];
+ audio = sync->audio;
+
+ if( job->acodec & HB_ACODEC_AC3 )
+ {
+ fifo = audio->fifo_out;
+ rate = audio->rate;
+ }
+ else
+ {
+ fifo = audio->fifo_sync;
+ rate = job->arate;
+ }
+
+ while( !hb_fifo_is_full( fifo ) &&
+ ( buf = hb_fifo_see( audio->fifo_raw ) ) )
+ {
+ /* The PTS of the samples we are expecting now */
+ pts_expected = pv->pts_offset + sync->count_frames * 90000 / rate;
+
+ if( ( buf->start > pts_expected + PTS_DISCONTINUITY_TOLERANCE ||
+ buf->start < pts_expected - PTS_DISCONTINUITY_TOLERANCE ) &&
+ pv->pts_offset_old > INT64_MIN )
+ {
+ /* 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;
+
+ if( buf->start > pts_expected + PTS_DISCONTINUITY_TOLERANCE ||
+ buf->start < pts_expected - PTS_DISCONTINUITY_TOLERANCE )
+ {
+ /* There is really nothing we can do with it */
+ buf = hb_fifo_get( audio->fifo_raw );
+ hb_buffer_close( &buf );
+ continue;
+ }
+
+ /* Use the older offset */
+ start = pts_expected - pv->pts_offset_old;
+ }
+ else
+ {
+ start = pts_expected - pv->pts_offset;
+ }
+
+ /* Tolerance: 100 ms */
+ if( buf->start < pts_expected - 9000 )
+ {
+ /* Late audio, trash it */
+ hb_log( "sync: trashing late audio" );
+ buf = hb_fifo_get( audio->fifo_raw );
+ hb_buffer_close( &buf );
+ continue;
+ }
+ else if( buf->start > pts_expected + 9000 )
+ {
+ /* Missing audio, send a frame of silence */
+ InsertSilence( w, i, channelsused );
+ continue;
+ }
+
+ if( job->acodec & HB_ACODEC_AC3 )
+ {
+ 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 / channelsused / 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 * channelsused *
+ sizeof( float ) );
+ sync->data.data_out = (float *) buf->data;
+ if( src_process( sync->state, &sync->data ) )
+ {
+ /* XXX If this happens, we're screwed */
+ hb_log( "sync: src_process failed" );
+ }
+ hb_buffer_close( &buf_raw );
+
+ buf->size = sync->data.output_frames_gen * channelsused * 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;
+ }
+
+ buf->key = 1;
+ hb_fifo_push( fifo, buf );
+ }
+
+ if( NeedSilence( w, audio ) )
+ {
+ InsertSilence( w, i, channelsused );
+ }
+}
+
+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 */
+ 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 */
+ return 1;
+ }
+
+ return 0;
+}
+
+static void InsertSilence( hb_work_object_t * w, int i, int channelsused )
+{
+ 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
+ {
+ buf = hb_buffer_init( channelsused * 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;
+ }
+}
+
+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 );
+}