完善ssrc相关处理

webrtc/WebRtcTransport.h

@@ -338,25 +338,26 @@ private:
     SdpAttrCandidate::Ptr getIceCandidate() const;
     bool canSendRtp() const;
     bool canRecvRtp() const;
-    const RtcSession& getSdpWithSSRC() const;
 
     class RtpPayloadInfo {
     public:
         using Ptr = std::shared_ptr<RtpPayloadInfo>;
-
-        bool is_common_rtp;
         const RtcCodecPlan *plan_rtp;
         const RtcCodecPlan *plan_rtx;
+        uint32_t offer_ssrc_rtp = 0;
+        uint32_t offer_ssrc_rtx = 0;
+        uint32_t answer_ssrc_rtp = 0;
+        uint32_t answer_ssrc_rtx = 0;
         const RtcMedia *media;
-        std::shared_ptr<RtpReceiverImp> receiver;
-        RtcpContext::Ptr rtcp_context_recv;
-        RtcpContext::Ptr rtcp_context_send;
         NackList nack_list;
         NackContext nack_ctx;
+        RtcpContext::Ptr rtcp_context_recv;
+        RtcpContext::Ptr rtcp_context_send;
+        std::shared_ptr<RtpReceiverImp> receiver;
     };
 
     void onSortedRtp(RtpPayloadInfo &info, RtpPacket::Ptr rtp);
-    void onNack(RtpPayloadInfo &info, const FCI_NACK &nack);
+    void onSendNack(RtpPayloadInfo &info, const FCI_NACK &nack);
 
 private:
     //用掉的总流量
@@ -371,8 +372,6 @@ private:
     Ticker _alive_ticker;
     //pli rtcp计时器
     Ticker _pli_ticker;
-    //记录协商的发送rtp的pt和ssrc
-    RtpPayloadInfo::Ptr _send_rtp_info[2];
     //复合udp端口，接收一切rtp与rtcp
     Socket::Ptr _socket;
     //推流的rtsp源
@@ -381,30 +380,14 @@ private:
     RtspMediaSource::Ptr _play_src;
     //播放rtsp源的reader对象
     RtspMediaSource::RingType::RingReader::Ptr _reader;
-    //根据rtp的pt获取相关信息
-    unordered_map<uint8_t/*pt*/, RtpPayloadInfo::Ptr> _rtp_info_pt;
+    //根据发送rtp的track类型获取相关信息
+    RtpPayloadInfo::Ptr _send_rtp_info[2];
+    //根据接收rtp的pt获取相关信息
+    unordered_map<uint8_t/*pt*/, std::pair<bool/*is rtx*/,RtpPayloadInfo::Ptr> > _rtp_info_pt;
     //根据rtcp的ssrc获取相关信息
-    unordered_map<uint32_t/*ssrc*/, RtpPayloadInfo::Ptr> _rtp_info_ssrc;
+    unordered_map<uint32_t/*ssrc*/, std::pair<bool/*is rtx*/,RtpPayloadInfo::Ptr> > _rtp_info_ssrc;
     //发送rtp时需要修改rtp ext id
     map<RtpExtType, uint8_t> _rtp_ext_type_to_id;
     //接收rtp时需要修改rtp ext id
     unordered_map<uint8_t, RtpExtType> _rtp_ext_id_to_type;
-};
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+};
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