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authorLuca Abeni <lucabe72@email.it>2008-07-02 10:23:27 +0000
committerLuca Abeni <lucabe72@email.it>2008-07-02 10:23:27 +0000
commitd6b9e57af2f45fe3f1c76c3ebd4059a3239c3e17 (patch)
tree5480bc5dbdc62b33ec626a0c84af66f0aa461398
parent49de8d1e9cc159985ff996c9f512714adc4ddf97 (diff)
downloadffmpeg-d6b9e57af2f45fe3f1c76c3ebd4059a3239c3e17.tar.gz
Fix A/V synch for RTP streams that do not contain MPEG1 or 2
(correctly compute the presentation times based on the RTP timestamps and the RTCP SR packets) Originally committed as revision 14045 to svn://svn.ffmpeg.org/ffmpeg/trunk
-rw-r--r--libavformat/rtpdec.c24
1 files changed, 6 insertions, 18 deletions
diff --git a/libavformat/rtpdec.c b/libavformat/rtpdec.c
index dc9e044227..3fa1a4afb3 100644
--- a/libavformat/rtpdec.c
+++ b/libavformat/rtpdec.c
@@ -282,6 +282,7 @@ RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *r
s->st = st;
s->rtp_payload_data = rtp_payload_data;
rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
+ av_set_pts_info(s->st, 32, 1, 90000);
if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
s->ts = mpegts_parse_open(s->ic);
if (s->ts == NULL) {
@@ -299,6 +300,9 @@ RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *r
st->need_parsing = AVSTREAM_PARSE_FULL;
break;
default:
+ if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
+ av_set_pts_info(st, 32, 1, st->codec->sample_rate);
+ }
break;
}
}
@@ -361,32 +365,16 @@ static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf)
*/
static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
{
- switch(s->st->codec->codec_id) {
- case CODEC_ID_MP2:
- case CODEC_ID_MPEG1VIDEO:
- case CODEC_ID_MPEG2VIDEO:
if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
int64_t addend;
int delta_timestamp;
- /* XXX: is it really necessary to unify the timestamp base ? */
/* compute pts from timestamp with received ntp_time */
delta_timestamp = timestamp - s->last_rtcp_timestamp;
- /* convert to 90 kHz without overflow */
- addend = (s->last_rtcp_ntp_time - s->first_rtcp_ntp_time) >> 14;
- addend = (addend * 5625) >> 14;
+ /* convert to the PTS timebase */
+ addend = av_rescale(s->last_rtcp_ntp_time - s->first_rtcp_ntp_time, s->st->time_base.den, (uint64_t)s->st->time_base.num << 32);
pkt->pts = addend + delta_timestamp;
}
- break;
- case CODEC_ID_AAC:
- case CODEC_ID_H264:
- case CODEC_ID_MPEG4:
- pkt->pts = timestamp;
- break;
- default:
- /* no timestamp info yet */
- break;
- }
pkt->stream_index = s->st->index;
}