优化rtp解析代码

src/Rtp/RtpProcess.cpp

@@ -21,12 +21,6 @@ static string printAddress(const struct sockaddr *addr){
 }
 
 RtpProcess::RtpProcess(const string &stream_id) {
-    _track = std::make_shared<SdpTrack>();
-    _track->_interleaved = 0;
-    _track->_samplerate = 90000;
-    _track->_type = TrackVideo;
-    _track->_ssrc = 0;
-
     _media_info._schema = RTP_APP_NAME;
     _media_info._vhost = DEFAULT_VHOST;
     _media_info._app = RTP_APP_NAME;
@@ -104,7 +98,7 @@ bool RtpProcess::inputRtp(const Socket::Ptr &sock, const char *data, int data_le
     }
 
     _total_bytes += data_len;
-    bool ret = handleOneRtp(0,_track,(unsigned char *)data,data_len);
+    bool ret = handleOneRtp(0, TrackVideo, 90000, (unsigned char *) data, data_len);
     if(dts_out){
         *dts_out = _dts;
     }

src/Rtp/RtpProcess.h

@@ -78,7 +78,6 @@ private:
     std::shared_ptr<FILE> _save_file_rtp;
     std::shared_ptr<FILE> _save_file_ps;
     std::shared_ptr<FILE> _save_file_video;
-    SdpTrack::Ptr _track;
     struct sockaddr *_addr = nullptr;
     uint16_t _sequence = 0;
     MultiMediaSourceMuxer::Ptr _muxer;

src/Rtsp/RtpReceiver.cpp

@@ -27,7 +27,7 @@ namespace mediakit {
 RtpReceiver::RtpReceiver() {}
 RtpReceiver::~RtpReceiver() {}
 
-bool RtpReceiver::handleOneRtp(int track_index,SdpTrack::Ptr &track, unsigned char *rtp_raw_ptr, unsigned int rtp_raw_len) {
+bool RtpReceiver::handleOneRtp(int track_index, TrackType type, int samplerate, unsigned char *rtp_raw_ptr, unsigned int rtp_raw_len) {
     if(rtp_raw_len < 12){
         WarnL << "rtp包太小:" << rtp_raw_len;
         return false;
@@ -46,8 +46,8 @@ bool RtpReceiver::handleOneRtp(int track_index,SdpTrack::Ptr &track, unsigned ch
     auto rtp_ptr = _rtp_pool.obtain();
     auto &rtp = *rtp_ptr;
 
-    rtp.type = track->_type;
+    rtp.type = type;
-    rtp.interleaved = 2 * track->_type;
+    rtp.interleaved = 2 * type;
     rtp.mark = rtp_raw_ptr[1] >> 7;
     rtp.PT = rtp_raw_ptr[1] & 0x7F;
 
@@ -59,29 +59,29 @@ bool RtpReceiver::handleOneRtp(int track_index,SdpTrack::Ptr &track, unsigned ch
     memcpy(&rtp.timeStamp, rtp_raw_ptr + 4, 4);
     rtp.timeStamp = ntohl(rtp.timeStamp);
 
-    if(!track->_samplerate){
+    if(!samplerate){
         //无法把时间戳转换成毫秒
         return false;
     }
     //时间戳转换成毫秒
-    rtp.timeStamp = rtp.timeStamp * 1000LL / track->_samplerate;
+    rtp.timeStamp = rtp.timeStamp * 1000LL / samplerate;
 
     //ssrc,内存对齐
     memcpy(&rtp.ssrc, rtp_raw_ptr + 8, 4);
     rtp.ssrc = ntohl(rtp.ssrc);
 
-    if (track->_ssrc != rtp.ssrc) {
+    if (_ssrc[track_index] != rtp.ssrc) {
-        if (track->_ssrc == 0) {
+        if (_ssrc[track_index] == 0) {
             //保存SSRC至track对象
-            track->_ssrc = rtp.ssrc;
+            _ssrc[track_index] = rtp.ssrc;
         }else{
             //ssrc错误
-            WarnL << "ssrc错误:" << rtp.ssrc << " != " << track->_ssrc;
+            WarnL << "ssrc错误:" << rtp.ssrc << " != " << _ssrc[track_index];
             if (_ssrc_err_count[track_index]++ > 10) {
                 //ssrc切换后清除老数据
-                WarnL << "ssrc更换:" << track->_ssrc << " -> " << rtp.ssrc;
+                WarnL << "ssrc更换:" << _ssrc[track_index] << " -> " << rtp.ssrc;
                 _rtp_sort_cache_map[track_index].clear();
-                track->_ssrc = rtp.ssrc;
+                _ssrc[track_index] = rtp.ssrc;
             }
             return false;
         }
@@ -168,11 +168,12 @@ void RtpReceiver::sortRtp(const RtpPacket::Ptr &rtp,int track_index){
 }
 
 void RtpReceiver::clear() {
-    CLEAR_ARR(_last_seq)
+    CLEAR_ARR(_last_seq);
-    CLEAR_ARR(_ssrc_err_count)
+    CLEAR_ARR(_ssrc);
-    CLEAR_ARR(_seq_ok_count)
+    CLEAR_ARR(_ssrc_err_count);
-    CLEAR_ARR(_sort_started)
+    CLEAR_ARR(_seq_ok_count);
-    CLEAR_ARR(_seq_cycle_count)
+    CLEAR_ARR(_sort_started);
+    CLEAR_ARR(_seq_cycle_count);
 
     _rtp_sort_cache_map[0].clear();
     _rtp_sort_cache_map[1].clear();

src/Rtsp/RtpReceiver.h

@@ -32,12 +32,13 @@ protected:
     /**
      * 输入数据指针生成并排序rtp包
      * @param track_index track下标索引
-     * @param track  sdp track相关信息
+     * @param type track类型
+     * @param samplerate rtp时间戳基准时钟，视频为90000，音频为采样率
      * @param rtp_raw_ptr rtp数据指针
      * @param rtp_raw_len rtp数据指针长度
      * @return 解析成功返回true
      */
-    bool handleOneRtp(int track_index,SdpTrack::Ptr &track, unsigned char *rtp_raw_ptr, unsigned int rtp_raw_len);
+    bool handleOneRtp(int track_index, TrackType type, int samplerate, unsigned char *rtp_raw_ptr, unsigned int rtp_raw_len);
 
     /**
      * rtp数据包排序后输出
@@ -49,9 +50,12 @@ protected:
     void setPoolSize(int size);
     int getJitterSize(int track_index);
     int getCycleCount(int track_index);
+
 private:
     void sortRtp(const RtpPacket::Ptr &rtp , int track_index);
+
 private:
+    uint32_t _ssrc[2] = { 0, 0 };
     //ssrc不匹配计数
     uint32_t _ssrc_err_count[2] = { 0, 0 };
     //上次seq

src/Rtsp/RtspPlayer.cpp

@@ -319,7 +319,8 @@ void RtspPlayer::handleResSETUP(const Parser &parser, unsigned int uiTrackIndex)
                 WarnL << "收到其他地址的rtp数据:" << SockUtil::inet_ntoa(((struct sockaddr_in *) addr)->sin_addr);
                 return;
             }
-            strongSelf->handleOneRtp(uiTrackIndex, strongSelf->_sdp_track[uiTrackIndex], (unsigned char *) buf->data(), buf->size());
+            strongSelf->handleOneRtp(uiTrackIndex, strongSelf->_sdp_track[uiTrackIndex]->_type,
+                                     strongSelf->_sdp_track[uiTrackIndex]->_samplerate, (unsigned char *) buf->data(), buf->size());
         });
 
         if(pRtcpSockRef) {
@@ -464,7 +465,7 @@ void RtspPlayer::onRtpPacket(const char *data, uint64_t len) {
     uint8_t interleaved = data[1];
     if(interleaved %2 == 0){
         trackIdx = getTrackIndexByInterleaved(interleaved);
-        handleOneRtp(trackIdx, _sdp_track[trackIdx], (unsigned char *)data + 4, len - 4);
+        handleOneRtp(trackIdx, _sdp_track[trackIdx]->_type, _sdp_track[trackIdx]->_samplerate, (unsigned char *)data + 4, len - 4);
     }else{
         trackIdx = getTrackIndexByInterleaved(interleaved - 1);
         onRtcpPacket(trackIdx, _sdp_track[trackIdx], (unsigned char *) data + 4, len - 4);

src/Rtsp/RtspSession.cpp

@@ -189,7 +189,7 @@ void RtspSession::onRtpPacket(const char *data, uint64_t len) {
     uint8_t interleaved = data[1];
     if(interleaved %2 == 0){
         trackIdx = getTrackIndexByInterleaved(interleaved);
-        handleOneRtp(trackIdx,_aTrackInfo[trackIdx],(unsigned char *)data + 4, len - 4);
+        handleOneRtp(trackIdx, _aTrackInfo[trackIdx]->_type, _aTrackInfo[trackIdx]->_samplerate, (unsigned char *) data + 4, len - 4);
     }else{
         trackIdx = getTrackIndexByInterleaved(interleaved - 1);
         onRtcpPacket(trackIdx, _aTrackInfo[trackIdx], (unsigned char *) data + 4, len - 4);
@@ -922,7 +922,8 @@ inline void RtspSession::onRcvPeerUdpData(int intervaled, const Buffer::Ptr &pBu
     if(intervaled % 2 == 0){
         if(_pushSrc){
             //这是rtsp推流上来的rtp包
-            handleOneRtp(intervaled / 2,_aTrackInfo[intervaled / 2],( unsigned char *)pBuf->data(),pBuf->size());
+            auto &ref = _aTrackInfo[intervaled / 2];
+            handleOneRtp(intervaled / 2, ref->_type, ref->_samplerate, (unsigned char *) pBuf->data(), pBuf->size());
         }else if(!_udpSockConnected.count(intervaled)){
             //这是rtsp播放器的rtp打洞包
             _udpSockConnected.emplace(intervaled);