优化rtp性能

src/Rtp/GB28181Process.cpp

@@ -34,7 +34,7 @@ bool GB28181Process::inputRtp(bool, const char *data, size_t data_len) {
     return handleOneRtp(0, TrackVideo, 90000, (unsigned char *) data, data_len);
 }
 
-void GB28181Process::onRtpSorted(const RtpPacket::Ptr &rtp, int) {
+void GB28181Process::onRtpSorted(RtpPacket::Ptr rtp, int) {
     auto pt = rtp->getHeader()->pt;
     if (!_rtp_decoder) {
         switch (pt) {

src/Rtp/GB28181Process.h

@@ -36,7 +36,7 @@ public:
     bool inputRtp(bool, const char *data, size_t data_len) override;
 
 protected:
-    void onRtpSorted(const RtpPacket::Ptr &rtp, int track_index) override ;
+    void onRtpSorted(RtpPacket::Ptr rtp, int track_index) override ;
     const char *onSearchPacketTail(const char *data,size_t len) override;
     ssize_t onRecvHeader(const char *data,size_t len) override { return 0; };
 

src/Rtsp/RtpCodec.cpp

@@ -14,7 +14,7 @@ namespace mediakit{
 
 RtpPacket::Ptr RtpInfo::makeRtp(TrackType type, const void* data, size_t len, bool mark, uint32_t stamp) {
     uint16_t payload_len = (uint16_t) (len + RtpPacket::kRtpHeaderSize);
-    auto rtp = ResourcePoolHelper<RtpPacket>::obtainObj();
+    auto rtp = RtpPacket::create();
     rtp->setCapacity(payload_len + RtpPacket::kRtpTcpHeaderSize);
     rtp->setSize(payload_len + RtpPacket::kRtpTcpHeaderSize);
     rtp->sample_rate = _sample_rate;

src/Rtsp/RtpCodec.h

@@ -58,9 +58,9 @@ protected:
     RingType::Ptr _rtpRing;
 };
 
-class RtpInfo : public ResourcePoolHelper<RtpPacket>{
+class RtpInfo{
 public:
-    typedef std::shared_ptr<RtpInfo> Ptr;
+    using Ptr = std::shared_ptr<RtpInfo>;
 
     RtpInfo(uint32_t ssrc, size_t mtu_size, uint32_t sample_rate, uint8_t pt, uint8_t interleaved) {
         if (ssrc == 0) {
@@ -73,7 +73,7 @@ public:
         _interleaved = interleaved;
     }
 
-    ~RtpInfo() override {}
+    virtual ~RtpInfo() {}
 
     //返回rtp负载最大长度
     size_t getMaxSize() const {

src/Rtsp/RtpReceiver.cpp

@@ -19,7 +19,7 @@ RtpReceiver::RtpReceiver() {
     int index = 0;
     for (auto &sortor : _rtp_sortor) {
         sortor.setOnSort([this, index](uint16_t seq, RtpPacket::Ptr &packet) {
-            onRtpSorted(packet, index);
+            onRtpSorted(std::move(packet), index);
         });
         ++index;
     }
@@ -61,7 +61,7 @@ bool RtpReceiver::handleOneRtp(int index, TrackType type, int sample_rate, uint8
         return false;
     }
 
-    auto rtp = _rtp_pool.obtain();
+    auto rtp = RtpPacket::create();
     //需要添加4个字节的rtp over tcp头
     rtp->setCapacity(RtpPacket::kRtpTcpHeaderSize + len);
     rtp->setSize(RtpPacket::kRtpTcpHeaderSize + len);
@@ -90,10 +90,6 @@ void RtpReceiver::clear() {
     }
 }
 
-void RtpReceiver::setPoolSize(size_t size) {
-    _rtp_pool.setSize(size);
-}
-
 size_t RtpReceiver::getJitterSize(int index) const{
     return _rtp_sortor[index].getJitterSize();
 }

src/Rtsp/RtpReceiver.h

@@ -182,7 +182,7 @@ protected:
      * @param rtp rtp数据包
      * @param track_index track索引
      */
-    virtual void onRtpSorted(const RtpPacket::Ptr &rtp, int track_index) {}
+    virtual void onRtpSorted(RtpPacket::Ptr rtp, int track_index) {}
 
     /**
      * 解析出rtp但还未排序
@@ -192,7 +192,6 @@ protected:
     virtual void onBeforeRtpSorted(const RtpPacket::Ptr &rtp, int track_index) {}
 
     void clear();
-    void setPoolSize(size_t size);
     size_t getJitterSize(int track_index) const;
     size_t getCycleCount(int track_index) const;
     uint32_t getSSRC(int track_index) const;
@@ -201,8 +200,6 @@ private:
     uint32_t _ssrc[2] = {0, 0};
     //rtp排序缓存，根据seq排序
     PacketSortor<RtpPacket::Ptr> _rtp_sortor[2];
-    //rtp循环池
-    RtspMediaSource::PoolType _rtp_pool;
 };
 
 }//namespace mediakit

src/Rtsp/Rtsp.cpp

@@ -531,4 +531,18 @@ size_t RtpPacket::getPayloadSize(){
     return getHeader()->getPayloadSize(size() - kRtpTcpHeaderSize);
 }
 
+RtpPacket::Ptr RtpPacket::create(){
+#if 0
+    static ResourcePool<RtpPacket> packet_pool;
+    static onceToken token([]() {
+        packet_pool.setSize(1024);
+    });
+    auto ret = packet_pool.obtain();
+    ret->setSize(0);
+    return ret;
+#else
+    return Ptr(new RtpPacket);
+#endif
+}
+
 }//namespace mediakit
\ No newline at end of file

src/Rtsp/Rtsp.h

@@ -166,6 +166,12 @@ public:
     TrackType type;
     //音频为采样率，视频一般为90000
     uint32_t sample_rate;
+
+    static Ptr create();
+
+private:
+    friend class ResourcePool_l<RtpPacket>;
+    RtpPacket() = default;
 };
 
 class RtpPayload {

src/Rtsp/RtspPlayer.cpp

@@ -494,10 +494,10 @@ void RtspPlayer::onRtcpPacket(int track_idx, SdpTrack::Ptr &track, uint8_t *data
     }
 }
 
-void RtspPlayer::onRtpSorted(const RtpPacket::Ptr &rtppt, int trackidx){
+void RtspPlayer::onRtpSorted(RtpPacket::Ptr rtppt, int trackidx){
     _stamp[trackidx] = rtppt->getStampMS();
     _rtp_recv_ticker.resetTime();
-    onRecvRTP(rtppt, _sdp_track[trackidx]);
+    onRecvRTP(std::move(rtppt), _sdp_track[trackidx]);
 }
 
 float RtspPlayer::getPacketLossRate(TrackType type) const{

src/Rtsp/RtspPlayer.h

@@ -33,7 +33,7 @@ using namespace toolkit;
 namespace mediakit {
 
 //实现了rtsp播放器协议部分的功能，及数据接收功能
-class RtspPlayer: public PlayerBase,public TcpClient, public RtspSplitter, public RtpReceiver {
+class RtspPlayer: public PlayerBase, public TcpClient, public RtspSplitter, public RtpReceiver {
 public:
     typedef std::shared_ptr<RtspPlayer> Ptr;
 
@@ -48,7 +48,7 @@ public:
 protected:
     //派生类回调函数
     virtual bool onCheckSDP(const string &sdp) = 0;
-    virtual void onRecvRTP(const RtpPacket::Ptr &rtp, const SdpTrack::Ptr &track) = 0;
+    virtual void onRecvRTP(RtpPacket::Ptr rtp, const SdpTrack::Ptr &track) = 0;
     uint32_t getProgressMilliSecond() const;
     void seekToMilliSecond(uint32_t ms);
 
@@ -70,7 +70,7 @@ protected:
      * @param rtp rtp数据包
      * @param track_idx track索引
      */
-    void onRtpSorted(const RtpPacket::Ptr &rtp, int track_idx) override;
+    void onRtpSorted(RtpPacket::Ptr rtp, int track_idx) override;
 
     /**
      * 解析出rtp但还未排序

src/Rtsp/RtspPlayerImp.h

@@ -59,18 +59,18 @@ private:
         return true;
     }
 
-    void onRecvRTP(const RtpPacket::Ptr &rtp, const SdpTrack::Ptr &track) override {
-        if (_rtsp_media_src) {
-            // rtsp直接代理是无法判断该rtp是否是I帧，所以GOP缓存基本是无效的
-            // 为了减少内存使用，那么我们设置为一直关键帧以便清空GOP缓存
-            _rtsp_media_src->onWrite(rtp, true);
-        }
+    void onRecvRTP(RtpPacket::Ptr rtp, const SdpTrack::Ptr &track) override {
         _delegate->inputRtp(rtp);
-
         if (_max_analysis_ms && _delegate->isInited(_max_analysis_ms)) {
             PlayerImp<RtspPlayer, RtspDemuxer>::onPlayResult(SockException(Err_success, "play rtsp success"));
             _max_analysis_ms = 0;
         }
+
+        if (_rtsp_media_src) {
+            // rtsp直接代理是无法判断该rtp是否是I帧，所以GOP缓存基本是无效的
+            // 为了减少内存使用，那么我们设置为一直关键帧以便清空GOP缓存
+            _rtsp_media_src->onWrite(std::move(rtp), true);
+        }
     }
 
     //在RtspPlayer中触发onPlayResult事件只是代表收到play回复了，

src/Rtsp/RtspSession.cpp

@@ -913,8 +913,8 @@ void RtspSession::send_NotAcceptable() {
     sendRtspResponse("406 Not Acceptable",{"Connection","Close"});
 }
 
-void RtspSession::onRtpSorted(const RtpPacket::Ptr &rtp, int track_idx) {
-    _push_src->onWrite(rtp, false);
+void RtspSession::onRtpSorted(RtpPacket::Ptr rtp, int track_idx) {
+    _push_src->onWrite(std::move(rtp), false);
 }
 
 void RtspSession::onRcvPeerUdpData(int interleaved, const Buffer::Ptr &buf, const struct sockaddr &addr) {

src/Rtsp/RtspSession.h

@@ -79,7 +79,7 @@ protected:
     ssize_t getContentLength(Parser &parser) override;
 
     ////RtpReceiver override////
-    void onRtpSorted(const RtpPacket::Ptr &rtp, int track_idx) override;
+    void onRtpSorted(RtpPacket::Ptr rtp, int track_idx) override;
     void onBeforeRtpSorted(const RtpPacket::Ptr &rtp, int track_index) override;
 
     ///////MediaSourceEvent override///////