修复WebRTC播放特殊情况下推流器断开无法触发媒体注销的问题

This commit is contained in:
TalusL 2023-02-20 15:36:15 +08:00
parent 2f8dab66c5
commit 2e066f2b77
2 changed files with 14 additions and 9 deletions

View File

@ -33,15 +33,19 @@ WebRtcPlayer::WebRtcPlayer(const EventPoller::Ptr &poller,
bool preferred_tcp) : WebRtcTransportImp(poller,preferred_tcp) {
_media_info = info;
_play_src = src;
CHECK(_play_src);
CHECK(src);
}
void WebRtcPlayer::onStartWebRTC() {
CHECK(_play_src);
auto playSrc = _play_src.lock();
if(!playSrc){
onShutdown(SockException(Err_shutdown, "rtsp media source was shutdown"));
return ;
}
WebRtcTransportImp::onStartWebRTC();
if (canSendRtp()) {
_play_src->pause(false);
_reader = _play_src->getRing()->attach(getPoller(), true);
playSrc->pause(false);
_reader = playSrc->getRing()->attach(getPoller(), true);
weak_ptr<WebRtcPlayer> weak_self = static_pointer_cast<WebRtcPlayer>(shared_from_this());
weak_ptr<Session> weak_session = getSession();
_reader->setGetInfoCB([weak_session]() { return weak_session.lock(); });
@ -64,8 +68,6 @@ void WebRtcPlayer::onStartWebRTC() {
strong_self->onShutdown(SockException(Err_shutdown, "rtsp ring buffer detached"));
});
}
//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;
}
void WebRtcPlayer::onDestory() {
WebRtcTransportImp::onDestory();
@ -86,11 +88,14 @@ void WebRtcPlayer::onDestory() {
}
void WebRtcPlayer::onRtcConfigure(RtcConfigure &configure) const {
CHECK(_play_src);
auto playSrc = _play_src.lock();
if(!playSrc){
return ;
}
WebRtcTransportImp::onRtcConfigure(configure);
//这是播放
configure.audio.direction = configure.video.direction = RtpDirection::sendonly;
configure.setPlayRtspInfo(_play_src->getSdp());
configure.setPlayRtspInfo(playSrc->getSdp());
}
}// namespace mediakit

View File

@ -36,7 +36,7 @@ private:
//媒体相关元数据
MediaInfo _media_info;
//播放的rtsp源
RtspMediaSource::Ptr _play_src;
std::weak_ptr<RtspMediaSource> _play_src;
//播放rtsp源的reader对象
RtspMediaSource::RingType::RingReader::Ptr _reader;
};