修复webrtc rtx包计入receiver report统计导致simulcast推流失效的问题

src/Rtp/GB28181Process.cpp

@@ -36,7 +36,7 @@ public:
     ~RtpReceiverImp() override = default;
 
     bool inputRtp(TrackType type, uint8_t *ptr, size_t len){
-        return RtpTrack::inputRtp(type, _sample_rate, ptr, len);
+        return RtpTrack::inputRtp(type, _sample_rate, ptr, len).operator bool();
     }
 
 private:

src/Rtsp/RtpReceiver.cpp

@@ -31,18 +31,18 @@ void RtpTrack::clear() {
     PacketSortor<RtpPacket::Ptr>::clear();
 }
 
-bool RtpTrack::inputRtp(TrackType type, int sample_rate, uint8_t *ptr, size_t len) {
+RtpPacket::Ptr RtpTrack::inputRtp(TrackType type, int sample_rate, uint8_t *ptr, size_t len) {
     if (len < RtpPacket::kRtpHeaderSize) {
         WarnL << "rtp包太小:" << len;
-        return false;
+        return nullptr;
     }
     if (len > RTP_MAX_SIZE) {
         WarnL << "超大的rtp包:" << len << " > " << RTP_MAX_SIZE;
-        return false;
+        return nullptr;
     }
     if (!sample_rate) {
         //无法把时间戳转换成毫秒
-        return false;
+        return nullptr;
     }
     RtpHeader *header = (RtpHeader *) ptr;
     if (header->version != RtpPacket::kRtpVersion) {
@@ -50,7 +50,7 @@ bool RtpTrack::inputRtp(TrackType type, int sample_rate, uint8_t *ptr, size_t le
     }
     if (!header->getPayloadSize(len)) {
         //无有效负载的rtp包
-        return false;
+        return nullptr;
     }
 
     //比对缓存ssrc
@@ -68,7 +68,7 @@ bool RtpTrack::inputRtp(TrackType type, int sample_rate, uint8_t *ptr, size_t le
         if (_ssrc_alive.elapsedTime() < 3 * 1000) {
             //接受正确ssrc的rtp在10秒内，那么我们认为存在多路rtp,忽略掉ssrc不匹配的rtp
             WarnL << "ssrc不匹配,rtp已丢弃:" << ssrc << " != " << _ssrc;
-            return false;
+            return nullptr;
         }
         InfoL << "rtp流ssrc切换:" << _ssrc << " -> " << ssrc;
         _ssrc = ssrc;
@@ -90,14 +90,11 @@ bool RtpTrack::inputRtp(TrackType type, int sample_rate, uint8_t *ptr, size_t le
     data[3] = len & 0xFF;
     //拷贝rtp
     memcpy(&data[4], ptr, len);
-
     //设置ntp时间戳
     rtp->ntp_stamp = _ntp_stamp.getNtpStamp(ntohl(rtp->getHeader()->stamp), sample_rate);
-
     onBeforeRtpSorted(rtp);
-    auto seq = rtp->getSeq();
+    sortPacket(rtp->getSeq(), rtp);
-    sortPacket(seq, std::move(rtp));
+    return rtp;
-    return true;
 }
 
 void RtpTrack::setNtpStamp(uint32_t rtp_stamp, uint32_t sample_rate, uint64_t ntp_stamp_ms){

src/Rtsp/RtpReceiver.h

@@ -175,7 +175,7 @@ public:
 
     void clear();
     uint32_t getSSRC() const;
-    bool inputRtp(TrackType type, int sample_rate, uint8_t *ptr, size_t len);
+    RtpPacket::Ptr inputRtp(TrackType type, int sample_rate, uint8_t *ptr, size_t len);
     void setNtpStamp(uint32_t rtp_stamp, uint32_t sample_rate, uint64_t ntp_stamp_ms);
 
 protected:
@@ -236,7 +236,7 @@ public:
      * @return 解析成功返回true
      */
     bool handleOneRtp(int index, TrackType type, int sample_rate, uint8_t *ptr, size_t len){
-        return _track[index].inputRtp(type, sample_rate, ptr, len);
+        return _track[index].inputRtp(type, sample_rate, ptr, len).operator bool();
     }
 
     /**

webrtc/WebRtcTransport.cpp

@@ -594,14 +594,15 @@ public:
 
     ~RtpChannel() override = default;
 
-    bool inputRtp(TrackType type, int sample_rate, uint8_t *ptr, size_t len, bool is_rtx){
+    RtpPacket::Ptr inputRtp(TrackType type, int sample_rate, uint8_t *ptr, size_t len, bool is_rtx) {
-        if (!is_rtx) {
+        auto rtp = RtpTrack::inputRtp(type, sample_rate, ptr, len);
-            RtpHeader *rtp = (RtpHeader *) ptr;
+        if (!is_rtx && rtp) {
-            auto seq = ntohs(rtp->seq);
             //统计rtp接受情况，便于生成nack rtcp包
+            auto seq = rtp->getSeq();
             _nack_ctx.received(seq);
+            _rtcp_context.onRtp(seq, rtp->getStamp(), rtp->ntp_stamp, sample_rate, len);
         }
-        return RtpTrack::inputRtp(type, sample_rate, ptr, len);
+        return rtp;
     }
 
     Buffer::Ptr createRtcpRR(RtcpHeader *sr, uint32_t ssrc) {
@@ -609,14 +610,6 @@ public:
         return _rtcp_context.createRtcpRR(ssrc, getSSRC());
     }
 
-protected:
-    void onBeforeRtpSorted(const RtpPacket::Ptr &rtp) override {
-        //统计rtp收到的情况，好做rr汇报
-        _rtcp_context.onRtp(rtp->getSeq(), rtp->getStamp(), rtp->ntp_stamp, rtp->sample_rate,
-                            rtp->size() - RtpPacket::kRtpTcpHeaderSize);
-        RtpTrackImp::onBeforeRtpSorted(rtp);
-    }
-
 private:
     NackContext _nack_ctx;
     RtcpContext _rtcp_context{true};