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rtsp支持指定播放单一track (#1937)

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This commit is contained in:
custompal
2022-09-13 16:27:00 +08:00
committed by GitHub
parent d04a6d4355
commit 08789454c3
7 changed files with 64 additions and 26 deletions
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60
src/Rtsp/RtspSession.cpp
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@@ -801,13 +801,14 @@ void RtspSession::handleReq_Play(const Parser &parser) {
InfoP(this) << "rtsp seekTo(ms):" << iStartTime;
}
vector<TrackType> inited_tracks;
_StrPrinter rtp_info;
for (auto &track : _sdp_track) {
if (track->_inited == false) {
//还有track没有setup
shutdown(SockException(Err_shutdown, "track not setuped"));
return;
//为支持播放器播放单一track, 不校验没有发setup的track
continue;
}
inited_tracks.emplace_back(track->_type);
track->_ssrc = play_src->getSsrc(track->_type);
track->_seq = play_src->getSeqence(track->_type);
track->_time_stamp = play_src->getTimeStamp(track->_type);
@@ -824,6 +825,12 @@ void RtspSession::handleReq_Play(const Parser &parser) {
res_header.emplace("Range", StrPrinter << "npt=" << setiosflags(ios::fixed) << setprecision(2) << play_src->getTimeStamp(TrackInvalid) / 1000.0);
sendRtspResponse("200 OK", res_header);
//设置播放track
if (inited_tracks.size() == 1) {
_target_play_track = inited_tracks[0];
InfoP(this) << "指定播放track:" << _target_play_track;
}
//在回复rtsp信令后再恢复播放
play_src->pause(false);
@@ -956,13 +963,17 @@ void RtspSession::onRcvPeerUdpData(int interleaved, const Buffer::Ptr &buf, cons
} else if (!_udp_connected_flags.count(interleaved)) {
//这是rtsp播放器的rtp打洞包
_udp_connected_flags.emplace(interleaved);
_rtp_socks[interleaved / 2]->bindPeerAddr((struct sockaddr *)&addr);
if (_rtp_socks[interleaved / 2]) {
_rtp_socks[interleaved / 2]->bindPeerAddr((struct sockaddr *)&addr);
}
}
} else {
//rtcp包
if (!_udp_connected_flags.count(interleaved)) {
_udp_connected_flags.emplace(interleaved);
_rtcp_socks[(interleaved - 1) / 2]->bindPeerAddr((struct sockaddr *)&addr);
if (_rtcp_socks[(interleaved - 1) / 2]) {
_rtcp_socks[(interleaved - 1) / 2]->bindPeerAddr((struct sockaddr *)&addr);
}
}
onRtcpPacket((interleaved - 1) / 2, _sdp_track[(interleaved - 1) / 2], buf->data(), buf->size());
}
@@ -1190,32 +1201,39 @@ void RtspSession::updateRtcpContext(const RtpPacket::Ptr &rtp){
void RtspSession::sendRtpPacket(const RtspMediaSource::RingDataType &pkt) {
switch (_rtp_type) {
case Rtsp::RTP_TCP: {
size_t i = 0;
auto size = pkt->size();
setSendFlushFlag(false);
pkt->for_each([&](const RtpPacket::Ptr &rtp) {
updateRtcpContext(rtp);
if (++i == size) {
setSendFlushFlag(true);
if (_target_play_track == TrackInvalid || _target_play_track == rtp->type) {
updateRtcpContext(rtp);
send(rtp);
}
send(rtp);
});
flushAll();
setSendFlushFlag(true);
}
break;
case Rtsp::RTP_UDP: {
size_t i = 0;
auto size = pkt->size();
//下标0表示视频，1表示音频
Socket::Ptr rtp_socks[2];
rtp_socks[TrackVideo] = _rtp_socks[getTrackIndexByTrackType(TrackVideo)];
rtp_socks[TrackAudio] = _rtp_socks[getTrackIndexByTrackType(TrackAudio)];
pkt->for_each([&](const RtpPacket::Ptr &rtp) {
updateRtcpContext(rtp);
int track_index = getTrackIndexByTrackType(rtp->type);
auto &pSock = _rtp_socks[track_index];
if (!pSock) {
shutdown(SockException(Err_shutdown, "udp sock not opened yet"));
return;
if (_target_play_track == TrackInvalid || _target_play_track == rtp->type) {
updateRtcpContext(rtp);
auto &sock = rtp_socks[rtp->type];
if (!sock) {
shutdown(SockException(Err_shutdown, "udp sock not opened yet"));
return;
}
_bytes_usage += rtp->size() - RtpPacket::kRtpTcpHeaderSize;
sock->send(std::make_shared<BufferRtp>(rtp, RtpPacket::kRtpTcpHeaderSize), nullptr, 0, false);
}
_bytes_usage += rtp->size() - RtpPacket::kRtpTcpHeaderSize;
pSock->send(std::make_shared<BufferRtp>(rtp, RtpPacket::kRtpTcpHeaderSize), nullptr, 0, ++i == size);
});
for (auto &sock : rtp_socks) {
if (sock) {
sock->flushAll();
}
}
}
break;
default: