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完善webrtc相关对象生命周期管理逻辑

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This commit is contained in:
ziyue 2021-09-15 20:08:18 +08:00
parent 3034f19a66
commit 2b5cad97cf
3 changed files with 13 additions and 17 deletions
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2
webrtc/WebRtcSession.cpp
View File

@ -75,6 +75,8 @@ void WebRtcSession::onError(const SockException &err) {
//在udp链接迁移时，新的WebRtcSession对象将接管WebRtcTransport对象的生命周期
//本WebRtcSession对象将在超时后自动销毁
WarnP(this) << err.what();
//取消循环引用
_transport = nullptr;
}
void WebRtcSession::onManager() {

24
webrtc/WebRtcTransport.cpp
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@ -334,8 +334,7 @@ void WebRtcTransportImp::onDestory() {
WebRtcTransport::onDestory();
unregisterSelf();
auto session = _session.lock();
if (!session) {
if (!_session) {
return;
}
@ -350,7 +349,7 @@ void WebRtcTransportImp::onDestory() {
<< _media_info._streamid
<< ")结束播放,耗时(s):" << duration;
if (_bytes_usage >= iFlowThreshold * 1024) {
NoticeCenter::Instance().emitEvent(Broadcast::kBroadcastFlowReport, _media_info, _bytes_usage, duration, true, static_cast<SockInfo &>(*session));
NoticeCenter::Instance().emitEvent(Broadcast::kBroadcastFlowReport, _media_info, _bytes_usage, duration, true, static_cast<SockInfo &>(*_session));
}
}
@ -361,7 +360,7 @@ void WebRtcTransportImp::onDestory() {
<< _media_info._streamid
<< ")结束推流,耗时(s):" << duration;
if (_bytes_usage >= iFlowThreshold * 1024) {
NoticeCenter::Instance().emitEvent(Broadcast::kBroadcastFlowReport, _media_info, _bytes_usage, duration, false, static_cast<SockInfo &>(*session));
NoticeCenter::Instance().emitEvent(Broadcast::kBroadcastFlowReport, _media_info, _bytes_usage, duration, false, static_cast<SockInfo &>(*_session));
}
}
}
@ -377,14 +376,13 @@ void WebRtcTransportImp::attach(const RtspMediaSource::Ptr &src, const MediaInfo
}
void WebRtcTransportImp::onSendSockData(const char *buf, size_t len, struct sockaddr_in *dst, bool flush) {
auto session = _session.lock();
if (!session) {
if (!_session) {
WarnL << "send data failed:" << len;
return;
}
auto ptr = BufferRaw::create();
ptr->assign(buf, len);
session->send(std::move(ptr));
_session->send(std::move(ptr));
}
///////////////////////////////////////////////////////////////////
@ -956,11 +954,8 @@ void WebRtcTransportImp::onBeforeEncryptRtp(const char *buf, int &len, void *ctx
void WebRtcTransportImp::onShutdown(const SockException &ex){
WarnL << ex.what();
unrefSelf();
//触发发送dtls close通知
WebRtcTransport::onDestory();
auto session = _session.lock();
if (session) {
session->shutdown(ex);
if (_session) {
_session->shutdown(ex);
}
}
@ -999,14 +994,13 @@ string WebRtcTransportImp::getOriginUrl(MediaSource &sender) const {
}
std::shared_ptr<SockInfo> WebRtcTransportImp::getOriginSock(MediaSource &sender) const {
return static_pointer_cast<SockInfo>(_session.lock());
return static_pointer_cast<SockInfo>(_session);
}
void WebRtcTransportImp::setSession(weak_ptr<Session> session) {
void WebRtcTransportImp::setSession(Session::Ptr session) {
_session = std::move(session);
}
static mutex s_rtc_mtx;
static unordered_map<string, weak_ptr<WebRtcTransportImp> > s_rtc_map;

4
webrtc/WebRtcTransport.h
View File

@ -171,7 +171,7 @@ public:
static Ptr create(const EventPoller::Ptr &poller);
static Ptr getRtcTransport(const string &key, bool unref_self);
void setSession(weak_ptr<Session> session);
void setSession(Session::Ptr session);
/**
* 绑定rtsp媒体源
@ -237,7 +237,7 @@ private:
//pli rtcp计时器
Ticker _pli_ticker;
//udp session
weak_ptr<Session> _session;
Session::Ptr _session;
//推流的rtsp源
RtspMediaSource::Ptr _push_src;
unordered_map<string/*rid*/, RtspMediaSource::Ptr> _push_src_simulcast;