rtc stop push unregister mediasource

webrtc/WebRtcPusher.cpp

@@ -150,4 +150,15 @@ void WebRtcPusher::onRtcConfigure(RtcConfigure &configure) const {
 
 
 float WebRtcPusher::getLossRate(MediaSource &sender,mediakit::TrackType type){
 float WebRtcPusher::getLossRate(MediaSource &sender,mediakit::TrackType type){
     return WebRtcTransportImp::getLossRate(type);
     return WebRtcTransportImp::getLossRate(type);
+}
+
+void WebRtcPusher::OnDtlsTransportClosed(const RTC::DtlsTransport *dtlsTransport) {
+   //主动关闭推流，那么不等待重推
+    _push_src = nullptr;
+    WebRtcTransportImp::OnDtlsTransportClosed(dtlsTransport);
+}
+
+void WebRtcPusher::onRtcpBye(){
+    _push_src = nullptr;
+     WebRtcTransportImp::onRtcpBye();
 }
 }
\ No newline at end of file

webrtc/WebRtcPusher.h

@@ -26,6 +26,9 @@ protected:
     void onDestory() override;
     void onDestory() override;
     void onRtcConfigure(RtcConfigure &configure) const override;
     void onRtcConfigure(RtcConfigure &configure) const override;
     void onRecvRtp(MediaTrack &track, const std::string &rid, mediakit::RtpPacket::Ptr rtp) override;
     void onRecvRtp(MediaTrack &track, const std::string &rid, mediakit::RtpPacket::Ptr rtp) override;
+    void onRtcpBye() override;
+    ////  dtls相关的回调 ////
+   void OnDtlsTransportClosed(const RTC::DtlsTransport *dtlsTransport) override;
 
 
 protected:
 protected:
     ///////MediaSourceEvent override///////
     ///////MediaSourceEvent override///////

webrtc/WebRtcTransport.cpp

@@ -766,6 +766,7 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
                 }
                 }
                 _ssrc_to_track.erase(it);
                 _ssrc_to_track.erase(it);
             }
             }
+            onRtcpBye();
             onShutdown(SockException(Err_eof, "rtcp bye message received"));
             onShutdown(SockException(Err_eof, "rtcp bye message received"));
             break;
             break;
         }
         }
@@ -1051,6 +1052,8 @@ uint64_t WebRtcTransportImp::getDuration() const {
     return _alive_ticker.createdTime() / 1000;
     return _alive_ticker.createdTime() / 1000;
 }
 }
 
 
+void WebRtcTransportImp::onRtcpBye(){}
+
 /////////////////////////////////////////////////////////////////////////////////////////////
 /////////////////////////////////////////////////////////////////////////////////////////////
 
 
 void WebRtcTransportImp::registerSelf() {
 void WebRtcTransportImp::registerSelf() {

webrtc/WebRtcTransport.h

@@ -154,6 +154,7 @@ protected:
     virtual void onShutdown(const SockException &ex) = 0;
     virtual void onShutdown(const SockException &ex) = 0;
     virtual void onBeforeEncryptRtp(const char *buf, int &len, void *ctx) = 0;
     virtual void onBeforeEncryptRtp(const char *buf, int &len, void *ctx) = 0;
     virtual void onBeforeEncryptRtcp(const char *buf, int &len, void *ctx) = 0;
     virtual void onBeforeEncryptRtcp(const char *buf, int &len, void *ctx) = 0;
+    virtual void onRtcpBye() = 0;
 
 
 protected:
 protected:
     RTC::TransportTuple* getSelectedTuple() const;
     RTC::TransportTuple* getSelectedTuple() const;
@@ -264,6 +265,7 @@ protected:
     virtual void onRecvRtp(MediaTrack &track, const std::string &rid, mediakit::RtpPacket::Ptr rtp) = 0;
     virtual void onRecvRtp(MediaTrack &track, const std::string &rid, mediakit::RtpPacket::Ptr rtp) = 0;
     void updateTicker();
     void updateTicker();
     float getLossRate(mediakit::TrackType type);
     float getLossRate(mediakit::TrackType type);
+    void onRtcpBye() override;
 
 
 private:
 private:
     void onSortedRtp(MediaTrack &track, const std::string &rid, mediakit::RtpPacket::Ptr rtp);
     void onSortedRtp(MediaTrack &track, const std::string &rid, mediakit::RtpPacket::Ptr rtp);