写在前面的话
写一个简单的播放器,通过RTP接收视频流,进行实时播放。最初,使用ffplay或者vlc接收按照SDP协议文件可以播放视频,但是视频中断后重启,不能正确的解包,时常会出现如下的错误信息。
[sdp @ 0x7ffb35034e00] RTP: dropping old packet received too late
Last message repeated 15 times
使用ffplay播放udp视频。
➜ view-show ffplay -protocol_whitelist "file,http,https,rtp,udp,tcp,tls" test.sdp
➜ view-show cat test.sdp
m=video 6665 RTP/AVP 96
a=rtpmap:96 H264/90000
c=IN IP4 0.0.0.0
分析ffmpeg原代码,在rtpdec.c代码文件中,这个错误的原因是ffplay接收RTP视频流时,如果前一个RTP包的流水号大于后一个帧时,会将当前RTP报文丢弃。
static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt, uint8_t **bufptr, int len)
{
...
if ((s->seq == 0 && !s->queue) || s->queue_size <= 1) {
/* First packet, or no reordering */
return rtp_parse_packet_internal(s, pkt, buf, len);
} else {
uint16_t seq = AV_RB16(buf + 2);
int16_t diff = seq - s->seq;
if (diff < 0) {
/* 注意看这里 Packet older than the previously emitted one, drop */
av_log(s->ic, AV_LOG_WARNING,
"RTP: dropping old packet received too late\n");
return -1;
} else if (diff <= 1) {
/* Correct packet */
rv = rtp_parse_packet_internal(s, pkt, buf, len);
return rv;
} else {
/* Still missing some packet, enqueue this one. */
rv = enqueue_packet(s, buf, len);
if (rv < 0)
return rv;
*bufptr = NULL;
/* Return the first enqueued packet if the queue is full,
* even if we're missing something */
if (s->queue_len >= s->queue_size) {
av_log(s->ic, AV_LOG_WARNING, "jitter buffer full\n");
return rtp_parse_queued_packet(s, pkt);
}
return -1;
}
}
}
但是,实际的业务场合中,对于一个大的视频文件,会按照MTU(以太网1500)拆分成很多个RTP报文(1400大小),多帧视频