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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
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@ -282,6 +282,7 @@ RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *r
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s->st = st;
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s->st = st;
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s->rtp_payload_data = rtp_payload_data;
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s->rtp_payload_data = rtp_payload_data;
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rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
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rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
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av_set_pts_info(s->st, 32, 1, 90000);
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if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
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if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
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s->ts = mpegts_parse_open(s->ic);
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s->ts = mpegts_parse_open(s->ic);
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if (s->ts == NULL) {
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if (s->ts == NULL) {
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@ -299,6 +300,9 @@ RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *r
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st->need_parsing = AVSTREAM_PARSE_FULL;
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st->need_parsing = AVSTREAM_PARSE_FULL;
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break;
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break;
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default:
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default:
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if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
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av_set_pts_info(st, 32, 1, st->codec->sample_rate);
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}
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break;
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break;
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}
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}
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}
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}
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@ -361,32 +365,16 @@ static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf)
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*/
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*/
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static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
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static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
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{
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{
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switch(s->st->codec->codec_id) {
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case CODEC_ID_MP2:
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case CODEC_ID_MPEG1VIDEO:
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case CODEC_ID_MPEG2VIDEO:
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if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
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if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
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int64_t addend;
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int64_t addend;
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int delta_timestamp;
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int delta_timestamp;
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/* XXX: is it really necessary to unify the timestamp base ? */
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/* compute pts from timestamp with received ntp_time */
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/* compute pts from timestamp with received ntp_time */
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delta_timestamp = timestamp - s->last_rtcp_timestamp;
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delta_timestamp = timestamp - s->last_rtcp_timestamp;
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/* convert to 90 kHz without overflow */
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/* convert to the PTS timebase */
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addend = (s->last_rtcp_ntp_time - s->first_rtcp_ntp_time) >> 14;
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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);
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addend = (addend * 5625) >> 14;
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pkt->pts = addend + delta_timestamp;
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pkt->pts = addend + delta_timestamp;
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}
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}
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break;
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case CODEC_ID_AAC:
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case CODEC_ID_H264:
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case CODEC_ID_MPEG4:
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pkt->pts = timestamp;
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break;
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default:
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/* no timestamp info yet */
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break;
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}
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pkt->stream_index = s->st->index;
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pkt->stream_index = s->st->index;
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}
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}
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