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修复webrtc播放不能秒开的问题

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This commit is contained in:
ziyue 2021-09-16 10:47:49 +08:00
parent dfa9001ed1
commit 0514278ce6
1 changed files with 15 additions and 15 deletions
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30
webrtc/WebRtcTransport.cpp
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@ -470,6 +470,21 @@ void WebRtcTransportImp::onStartWebRTC() {
_simulcast = getSdp(SdpType::answer).supportSimulcast();
}
if (canSendRtp()) {
RtcSession rtsp_send_sdp;
rtsp_send_sdp.loadFrom(_play_src->getSdp(), false);
for (auto &m : getSdp(SdpType::answer).media) {
if (m.type == TrackApplication) {
continue;
}
auto rtsp_media = rtsp_send_sdp.getMedia(m.type);
if (rtsp_media && getCodecId(rtsp_media->plan[0].codec) == getCodecId(m.plan[0].codec)) {
auto it = _pt_to_track.find(m.plan[0].pt);
CHECK(it != _pt_to_track.end());
//记录发送rtp时约定的信息，届时发送rtp时需要修改pt和ssrc
_type_to_track[m.type] = it->second.second;
}
}
_reader = _play_src->getRing()->attach(getPoller(), true);
weak_ptr<WebRtcTransportImp> weak_self = static_pointer_cast<WebRtcTransportImp>(shared_from_this());
_reader->setReadCB([weak_self](const RtspMediaSource::RingDataType &pkt) {
@ -489,21 +504,6 @@ void WebRtcTransportImp::onStartWebRTC() {
}
strongSelf->onShutdown(SockException(Err_shutdown, "rtsp ring buffer detached"));
});
RtcSession rtsp_send_sdp;
rtsp_send_sdp.loadFrom(_play_src->getSdp(), false);
for (auto &m : getSdp(SdpType::answer).media) {
if (m.type == TrackApplication) {
continue;
}
auto rtsp_media = rtsp_send_sdp.getMedia(m.type);
if (rtsp_media && getCodecId(rtsp_media->plan[0].codec) == getCodecId(m.plan[0].codec)) {
auto it = _pt_to_track.find(m.plan[0].pt);
CHECK(it != _pt_to_track.end());
//记录发送rtp时约定的信息，届时发送rtp时需要修改pt和ssrc
_type_to_track[m.type] = it->second.second;
}
}
}
//使用完毕后，释放强引用，这样确保推流器断开后能及时注销媒体
_play_src = nullptr;