完善webrtc相关对象生命周期管理逻辑

webrtc/WebRtcSession.cpp

@@ -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() {

webrtc/WebRtcTransport.cpp

@@ -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;
 

webrtc/WebRtcTransport.h

@@ -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;