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pass AVPacket into av_write_frame()

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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
This commit is contained in:
Michael Niedermayer
2004-05-29 02:06:32 +00:00
parent a7b2871cd1
commit e928649b0b
33 changed files with 328 additions and 245 deletions
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113
libavformat/utils.c
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@@ -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;