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authorMichael Niedermayer <michaelni@gmx.at>2004-05-29 02:06:32 +0000
committerMichael Niedermayer <michaelni@gmx.at>2004-05-29 02:06:32 +0000
commite928649b0bf6c8c7b87eb09d5e393a70387b10e9 (patch)
tree3f065adcb0e5c14127d01162d7cc8a0d934c073f /libavformat/utils.c
parenta7b2871cd1401ce7be59b153eed3f25565b0bb23 (diff)
downloadffmpeg-e928649b0bf6c8c7b87eb09d5e393a70387b10e9.tar.gz
pass AVPacket into av_write_frame()
fixes the random dts/pts during encoding asf preroll fix no more initial zero frames for b frame encoding mpeg-es dts during demuxing fixed .ffm timestamp scale fixed, ffm is still broken though Originally committed as revision 3168 to svn://svn.ffmpeg.org/ffmpeg/trunk
Diffstat (limited to 'libavformat/utils.c')
-rw-r--r--libavformat/utils.c113
1 files changed, 94 insertions, 19 deletions
diff --git a/libavformat/utils.c b/libavformat/utils.c
index b60367c090..3232f0276f 100644
--- a/libavformat/utils.c
+++ b/libavformat/utils.c
@@ -569,10 +569,12 @@ static void compute_pkt_fields(AVFormatContext *s, AVStream *st,
int num, den, presentation_delayed;
/* handle wrapping */
- if(pkt->pts != AV_NOPTS_VALUE)
- pkt->pts= lsb2full(pkt->pts, st->cur_dts, st->pts_wrap_bits);
- if(pkt->dts != AV_NOPTS_VALUE)
- pkt->dts= lsb2full(pkt->dts, st->cur_dts, st->pts_wrap_bits);
+ if(st->cur_dts != AV_NOPTS_VALUE){
+ if(pkt->pts != AV_NOPTS_VALUE)
+ pkt->pts= lsb2full(pkt->pts, st->cur_dts, st->pts_wrap_bits);
+ if(pkt->dts != AV_NOPTS_VALUE)
+ pkt->dts= lsb2full(pkt->dts, st->cur_dts, st->pts_wrap_bits);
+ }
if (pkt->duration == 0) {
compute_frame_duration(&num, &den, s, st, pc, pkt);
@@ -596,7 +598,13 @@ static void compute_pkt_fields(AVFormatContext *s, AVStream *st,
if(pkt->dts != AV_NOPTS_VALUE && pkt->pts != AV_NOPTS_VALUE && pkt->pts > pkt->dts)
presentation_delayed = 1;
}
+
+ if(st->cur_dts == AV_NOPTS_VALUE){
+ if(presentation_delayed) st->cur_dts = -pkt->duration;
+ else st->cur_dts = 0;
+ }
+// av_log(NULL, AV_LOG_DEBUG, "IN delayed:%d pts:%lld, dts:%lld cur_dts:%lld\n", presentation_delayed, pkt->pts, pkt->dts, st->cur_dts);
/* interpolate PTS and DTS if they are not present */
if (presentation_delayed) {
/* DTS = decompression time stamp */
@@ -637,6 +645,7 @@ static void compute_pkt_fields(AVFormatContext *s, AVStream *st,
}
st->cur_dts += pkt->duration;
}
+// av_log(NULL, AV_LOG_DEBUG, "OUTdelayed:%d pts:%lld, dts:%lld cur_dts:%lld\n", presentation_delayed, pkt->pts, pkt->dts, st->cur_dts);
/* update flags */
if (pc) {
@@ -1672,6 +1681,30 @@ int av_find_stream_info(AVFormatContext *ic)
}
av_estimate_timings(ic);
+#if 0
+ /* correct DTS for b frame streams with no timestamps */
+ for(i=0;i<ic->nb_streams;i++) {
+ st = ic->streams[i];
+ if (st->codec.codec_type == CODEC_TYPE_VIDEO) {
+ if(b-frames){
+ ppktl = &ic->packet_buffer;
+ while(ppkt1){
+ if(ppkt1->stream_index != i)
+ continue;
+ if(ppkt1->pkt->dts < 0)
+ break;
+ if(ppkt1->pkt->pts != AV_NOPTS_VALUE)
+ break;
+ ppkt1->pkt->dts -= delta;
+ ppkt1= ppkt1->next;
+ }
+ if(ppkt1)
+ continue;
+ st->cur_dts -= delta;
+ }
+ }
+ }
+#endif
return ret;
}
@@ -1764,6 +1797,7 @@ AVStream *av_new_stream(AVFormatContext *s, int id)
st->id = id;
st->start_time = AV_NOPTS_VALUE;
st->duration = AV_NOPTS_VALUE;
+ st->cur_dts = AV_NOPTS_VALUE;
/* default pts settings is MPEG like */
av_set_pts_info(st, 33, 1, 90000);
@@ -1836,27 +1870,68 @@ int av_write_header(AVFormatContext *s)
* one audio or video frame.
*
* @param s media file handle
- * @param stream_index stream index
- * @param buf buffer containing the frame data
- * @param size size of buffer
+ * @param pkt the packet, which contains the stream_index, buf/buf_size, dts/pts, ...
* @return < 0 if error, = 0 if OK, 1 if end of stream wanted.
*/
-int av_write_frame(AVFormatContext *s, int stream_index, const uint8_t *buf,
- int size)
+int av_write_frame(AVFormatContext *s, AVPacket *pkt)
{
AVStream *st;
int64_t pts_mask;
int ret, frame_size;
+ int b_frames;
- st = s->streams[stream_index];
- pts_mask = (1LL << st->pts_wrap_bits) - 1;
+ if(pkt->stream_index<0)
+ return -1;
+ st = s->streams[pkt->stream_index];
- /* HACK/FIXME we skip all zero size audio packets so a encoder can pass pts by outputing zero size packets */
- if(st->codec.codec_type==CODEC_TYPE_AUDIO && size==0)
- ret = 0;
- else
- ret = s->oformat->write_packet(s, stream_index, buf, size,
- st->pts.val & pts_mask);
+ b_frames = FFMAX(st->codec.has_b_frames, st->codec.max_b_frames);
+
+ // av_log(s, AV_LOG_DEBUG, "av_write_frame: pts:%lld dts:%lld cur_dts:%lld b:%d size:%d\n", pkt->pts, pkt->dts, st->cur_dts, b_frames, pkt->size);
+
+/* if(pkt->pts == AV_NOPTS_VALUE && pkt->dts == AV_NOPTS_VALUE)
+ return -1;*/
+
+ if(pkt->pts != AV_NOPTS_VALUE)
+ pkt->pts = av_rescale(pkt->pts, st->time_base.den, AV_TIME_BASE * (int64_t)st->time_base.num);
+ if(pkt->dts != AV_NOPTS_VALUE)
+ pkt->dts = av_rescale(pkt->dts, st->time_base.den, AV_TIME_BASE * (int64_t)st->time_base.num);
+
+ /* duration field */
+ pkt->duration = av_rescale(pkt->duration, st->time_base.den, AV_TIME_BASE * (int64_t)st->time_base.num);
+
+ //XXX/FIXME this is a temporary hack until all encoders output pts
+ if((pkt->pts == 0 || pkt->pts == AV_NOPTS_VALUE) && pkt->dts == AV_NOPTS_VALUE && !b_frames){
+ pkt->dts=
+// pkt->pts= st->cur_dts;
+ pkt->pts= st->pts.val;
+ }
+
+ //calculate dts from pts
+ if(pkt->pts != AV_NOPTS_VALUE && pkt->dts == AV_NOPTS_VALUE){
+ if(b_frames){
+ if(st->last_IP_pts == AV_NOPTS_VALUE){
+ st->last_IP_pts= -av_rescale(1,
+ st->codec.frame_rate_base*(int64_t)st->time_base.den,
+ st->codec.frame_rate *(int64_t)st->time_base.num);
+ }
+ if(st->last_IP_pts < pkt->pts){
+ pkt->dts= st->last_IP_pts;
+ st->last_IP_pts= pkt->pts;
+ }else
+ pkt->dts= pkt->pts;
+ }else
+ pkt->dts= pkt->pts;
+ }
+
+// av_log(s, AV_LOG_DEBUG, "av_write_frame: pts2:%lld dts2:%lld\n", pkt->pts, pkt->dts);
+ st->cur_dts= pkt->dts;
+ st->pts.val= pkt->dts;
+
+ pts_mask = (2LL << (st->pts_wrap_bits-1)) - 1;
+ pkt->pts &= pts_mask;
+ pkt->dts &= pts_mask;
+
+ ret = s->oformat->write_packet(s, pkt);
if (ret < 0)
return ret;
@@ -1864,11 +1939,11 @@ int av_write_frame(AVFormatContext *s, int stream_index, const uint8_t *buf,
/* update pts */
switch (st->codec.codec_type) {
case CODEC_TYPE_AUDIO:
- frame_size = get_audio_frame_size(&st->codec, size);
+ frame_size = get_audio_frame_size(&st->codec, pkt->size);
/* HACK/FIXME, we skip the initial 0-size packets as they are most likely equal to the encoder delay,
but it would be better if we had the real timestamps from the encoder */
- if (frame_size >= 0 && (size || st->pts.num!=st->pts.den>>1 || st->pts.val)) {
+ if (frame_size >= 0 && (pkt->size || st->pts.num!=st->pts.den>>1 || st->pts.val)) {
av_frac_add(&st->pts, (int64_t)st->time_base.den * frame_size);
}
break;