大幅提高rtsp服务器性能

b169f94c · xiongziliang · 1352e159 · 681be205 · b169f94c · b169f94c
Commit b169f94c authored Apr 07, 2020 by xiongziliang
--- a/ZLToolKit @ 681be205
+++ b/ZLToolKit @ 681be205
-Subproject commit 58a74f8c5ab802a0dd9fdcdcc0fe4c5a3d841964
+Subproject commit 681be205ef164db08effd83f925bb750eb1fe149
--- a/src/Rtsp/RtpMultiCaster.cpp
+++ b/src/Rtsp/RtpMultiCaster.cpp
@@ -109,13 +109,18 @@ RtpMultiCaster::RtpMultiCaster(const EventPoller::Ptr &poller,const string &strL
        _apUdpSock[i]->setSendPeerAddr((struct sockaddr *)&peerAddr);
    }
    _pReader = src->getRing()->attach(poller);
-    _pReader->setReadCB([this](const RtpPacket::Ptr &pkt){
+    _pReader->setReadCB([this](const RtspMediaSource::RingDataType &pkt){
-        int i = (int)(pkt->type);
+        int i = 0;
-        auto &pSock = _apUdpSock[i];
+        int size = pkt->size();
-        auto &peerAddr = _aPeerUdpAddr[i];
+        pkt->for_each([&](const RtpPacket::Ptr &rtp) {
-        BufferRtp::Ptr buffer(new BufferRtp(pkt,4));
+            int i = (int) (rtp->type);
-        pSock->send(buffer);
+            auto &pSock = _apUdpSock[i];
+            auto &peerAddr = _aPeerUdpAddr[i];
+            BufferRtp::Ptr buffer(new BufferRtp(rtp, 4));
+            pSock->send(buffer, nullptr, 0, ++i == size);
+        });
    });
    _pReader->setDetachCB([this](){
        unordered_map<void * , onDetach > _mapDetach_copy;
        {

--- a/src/Rtsp/RtspMediaSource.h
+++ b/src/Rtsp/RtspMediaSource.h
@@ -27,10 +27,95 @@
 #include "Thread/ThreadPool.h"
 using namespace std;
 using namespace toolkit;
+#define RTP_GOP_SIZE 512
+namespace mediakit {
-#define RTP_GOP_SIZE 2048
+class RtpVideoCache {
+public:
-namespace mediakit {
+    RtpVideoCache() {
+        _cache = std::make_shared<List<RtpPacket::Ptr> >();
+    }
+    virtual ~RtpVideoCache() = default;
+    void inputVideoRtp(const RtpPacket::Ptr &rtp, bool key_pos) {
+        if (_last_rtp_stamp != rtp->timeStamp) {
+            //时间戳发生变化了
+            flushAll();
+        } else if (_cache->size() > RTP_GOP_SIZE) {
+            //这个逻辑用于避免时间戳异常的流导致的内存暴增问题
+            flushAll();
+        }
+        //追加数据到最后
+        _cache->emplace_back(rtp);
+        _last_rtp_stamp = rtp->timeStamp;
+        if (key_pos) {
+            _key_pos = key_pos;
+        }
+    }
+    virtual void onFlushVideoRtp(std::shared_ptr<List<RtpPacket::Ptr> > &, bool key_pos) = 0;
+private:
+    void flushAll() {
+        if (_cache->empty()) {
+            return;
+        }
+        onFlushVideoRtp(_cache, _key_pos);
+        _cache = std::make_shared<List<RtpPacket::Ptr> >();
+        _key_pos = false;
+    }
+private:
+    std::shared_ptr<List<RtpPacket::Ptr> > _cache;
+    uint32_t _last_rtp_stamp = 0;
+    bool _key_pos = false;
+};
+class RtpAudioCache {
+public:
+    RtpAudioCache() {
+        _cache = std::make_shared<List<RtpPacket::Ptr> >();
+    }
+    virtual ~RtpAudioCache() = default;
+    void inputAudioRtp(const RtpPacket::Ptr &rtp) {
+        if (rtp->timeStamp > _last_rtp_stamp + 100) {
+            //累积了100ms的音频数据
+            flushAll();
+        } else if (_cache->size() > 10) {
+            //或者audio rtp缓存超过10个
+            flushAll();
+        }
+        //追加数据到最后
+        _cache->emplace_back(rtp);
+        _last_rtp_stamp = rtp->timeStamp;
+    }
+    virtual void onFlushAudioRtp(std::shared_ptr<List<RtpPacket::Ptr> > &) = 0;
+private:
+    void flushAll() {
+        if (_cache->empty()) {
+            return;
+        }
+        onFlushAudioRtp(_cache);
+        _cache = std::make_shared<List<RtpPacket::Ptr> >();
+    }
+private:
+    std::shared_ptr<List<RtpPacket::Ptr> > _cache;
+    uint32_t _last_rtp_stamp = 0;
+};
 /**
 * rtsp媒体源的数据抽象
@@ -38,11 +123,12 @@ namespace mediakit {
 * 只要生成了这两要素，那么要实现rtsp推流、rtsp服务器就很简单了
 * rtsp推拉流协议中，先传递sdp，然后再协商传输方式(tcp/udp/组播)，最后一直传递rtp
 */
-class RtspMediaSource : public MediaSource, public RingDelegate<RtpPacket::Ptr> {
+class RtspMediaSource : public MediaSource, public RingDelegate<RtpPacket::Ptr>, public RtpVideoCache, public RtpAudioCache {
 public:
    typedef ResourcePool<RtpPacket> PoolType;
    typedef std::shared_ptr<RtspMediaSource> Ptr;
-    typedef RingBuffer<RtpPacket::Ptr> RingType;
+    typedef std::shared_ptr<List<RtpPacket::Ptr> > RingDataType;
+    typedef RingBuffer<RingDataType> RingType;
    /**
     * 构造函数
@@ -173,10 +259,34 @@ public:
                regist();
            }
        }
-        //不存在视频，为了减少缓存延时，那么关闭GOP缓存
-        _ring->write(rtp, _have_video ? keyPos : true);
+        if(rtp->type == TrackVideo){
+            RtpVideoCache::inputVideoRtp(rtp, keyPos);
+        }else{
+            RtpAudioCache::inputAudioRtp(rtp);
+        }
    }
 private:
+    /**
+     * 批量flush时间戳相同的视频rtp包时触发该函数
+     * @param rtp_list 时间戳相同的rtp包列表
+     * @param key_pos 是否包含关键帧
+     */
+    void onFlushVideoRtp(std::shared_ptr<List<RtpPacket::Ptr> > &rtp_list, bool key_pos) override {
+        _ring->write(rtp_list, key_pos);
+    }
+    /**
+     * 批量flush一定数量的音频rtp包时触发该函数
+     * @param rtp_list rtp包列表
+     */
+    void onFlushAudioRtp(std::shared_ptr<List<RtpPacket::Ptr> > &rtp_list) override{
+        //只有音频的话，就不存在gop缓存的意义
+        _ring->write(rtp_list, !_have_video);
+    }
    /**
     * 每次增减消费者都会触发该函数
     */

--- a/src/Rtsp/RtspPusher.cpp
+++ b/src/Rtsp/RtspPusher.cpp
@@ -301,23 +301,36 @@ void RtspPusher::sendOptions() {
    sendRtspRequest("OPTIONS",_strContentBase);
 }
-inline void RtspPusher::sendRtpPacket(const RtpPacket::Ptr & pkt) {
+inline void RtspPusher::sendRtpPacket(const RtspMediaSource::RingDataType &pkt) {
    //InfoL<<(int)pkt.Interleaved;
    switch (_eType) {
        case Rtsp::RTP_TCP: {
-            BufferRtp::Ptr buffer(new BufferRtp(pkt));
+            int i = 0;
-            send(buffer);
+            int size = pkt->size();
+            setSendFlushFlag(false);
+            pkt->for_each([&](const RtpPacket::Ptr &rtp) {
+                if (++i == size) {
+                    setSendFlushFlag(true);
+                }
+                BufferRtp::Ptr buffer(new BufferRtp(rtp));
+                send(buffer);
+            });
        }
            break;
        case Rtsp::RTP_UDP: {
-            int iTrackIndex = getTrackIndexByTrackType(pkt->type);
+            int i = 0;
-            auto &pSock = _apUdpSock[iTrackIndex];
+            int size = pkt->size();
-            if (!pSock) {
+            pkt->for_each([&](const RtpPacket::Ptr &rtp) {
-                shutdown(SockException(Err_shutdown,"udp sock not opened yet"));
+                int iTrackIndex = getTrackIndexByTrackType(rtp->type);
-                return;
+                auto &pSock = _apUdpSock[iTrackIndex];
-            }
+                if (!pSock) {
-            BufferRtp::Ptr buffer(new BufferRtp(pkt,4));
+                    shutdown(SockException(Err_shutdown,"udp sock not opened yet"));
-            pSock->send(buffer);
+                    return;
+                }
+                BufferRtp::Ptr buffer(new BufferRtp(rtp,4));
+                pSock->send(buffer, nullptr, 0, ++i == size);
+            });
        }
            break;
        default:
@@ -337,7 +350,6 @@ inline int RtspPusher::getTrackIndexByTrackType(TrackType type) {
    return -1;
 }
 void RtspPusher::sendRecord() {
    _onHandshake = [this](const Parser& parser){
        auto src = _pMediaSrc.lock();
@@ -347,7 +359,7 @@ void RtspPusher::sendRecord() {
        _pRtspReader = src->getRing()->attach(getPoller());
        weak_ptr<RtspPusher> weakSelf = dynamic_pointer_cast<RtspPusher>(shared_from_this());
-        _pRtspReader->setReadCB([weakSelf](const RtpPacket::Ptr &pkt){
+        _pRtspReader->setReadCB([weakSelf](const RtspMediaSource::RingDataType &pkt){
            auto strongSelf = weakSelf.lock();
            if(!strongSelf) {
                return;

--- a/src/Rtsp/RtspPusher.h
+++ b/src/Rtsp/RtspPusher.h
@@ -67,7 +67,7 @@ private:
    inline int getTrackIndexByTrackType(TrackType type);
-    void sendRtpPacket(const RtpPacket::Ptr & pkt) ;
+    void sendRtpPacket(const RtspMediaSource::RingDataType & pkt) ;
    void sendRtspRequest(const string &cmd, const string &url ,const StrCaseMap &header = StrCaseMap(),const string &sdp = "" );
    void sendRtspRequest(const string &cmd, const string &url ,const std::initializer_list<string> &header,const string &sdp = "");

--- a/src/Rtsp/RtspSession.cpp
+++ b/src/Rtsp/RtspSession.cpp
@@ -796,7 +796,7 @@ void RtspSession::handleReq_Play(const Parser &parser) {
                }
                strongSelf->shutdown(SockException(Err_shutdown,"rtsp ring buffer detached"));
            });
-            _pRtpReader->setReadCB([weakSelf](const RtpPacket::Ptr &pack) {
+            _pRtpReader->setReadCB([weakSelf](const RtspMediaSource::RingDataType &pack) {
                auto strongSelf = weakSelf.lock();
                if(!strongSelf) {
                    return;
@@ -1123,23 +1123,36 @@ int RtspSession::totalReaderCount(MediaSource &sender) {
    return _pushSrc ? _pushSrc->totalReaderCount() : sender.readerCount();
 }
-void RtspSession::sendRtpPacket(const RtpPacket::Ptr & pkt) {
+void RtspSession::sendRtpPacket(const RtspMediaSource::RingDataType &pkt) {
    //InfoP(this) <<(int)pkt.Interleaved;
    switch (_rtpType) {
        case Rtsp::RTP_TCP: {
-            send(pkt);
+            int i = 0;
+            int size = pkt->size();
+            setSendFlushFlag(false);
+            pkt->for_each([&](const RtpPacket::Ptr &rtp) {
+                if (++i == size) {
+                    setSendFlushFlag(true);
+                }
+                send(rtp);
+            });
        }
            break;
        case Rtsp::RTP_UDP: {
-            int iTrackIndex = getTrackIndexByTrackType(pkt->type);
+            int i = 0;
-            auto &pSock = _apRtpSock[iTrackIndex];
+            int size = pkt->size();
-            if (!pSock) {
+            pkt->for_each([&](const RtpPacket::Ptr &rtp) {
-                shutdown(SockException(Err_shutdown,"udp sock not opened yet"));
+                int iTrackIndex = getTrackIndexByTrackType(rtp->type);
-                return;
+                auto &pSock = _apRtpSock[iTrackIndex];
-            }
+                if (!pSock) {
-            BufferRtp::Ptr buffer(new BufferRtp(pkt,4));
+                    shutdown(SockException(Err_shutdown, "udp sock not opened yet"));
-            _ui64TotalBytes += buffer->size();
+                    return;
-            pSock->send(buffer);
+                }
+                BufferRtp::Ptr buffer(new BufferRtp(rtp, 4));
+                _ui64TotalBytes += buffer->size();
+                pSock->send(buffer, nullptr, 0, ++i == size);
+            });
        }
            break;
        default:

--- a/src/Rtsp/RtspSession.h
+++ b/src/Rtsp/RtspSession.h
@@ -162,7 +162,7 @@ private:
    void onAuthDigest(const string &realm,const string &strMd5);
    //发送rtp给客户端
-    void sendRtpPacket(const RtpPacket::Ptr &pkt);
+    void sendRtpPacket(const RtspMediaSource::RingDataType &pkt);
    //回复客户端
    bool sendRtspResponse(const string &res_code,const std::initializer_list<string> &header, const string &sdp = "" , const char *protocol = "RTSP/1.0");
    bool sendRtspResponse(const string &res_code,const StrCaseMap &header = StrCaseMap(), const string &sdp = "",const char *protocol = "RTSP/1.0");
@@ -186,7 +186,7 @@ private:
    //rtsp播放器绑定的直播源
    std::weak_ptr<RtspMediaSource> _pMediaSrc;
    //直播源读取器
-    RingBuffer<RtpPacket::Ptr>::RingReader::Ptr _pRtpReader;
+    RtspMediaSource::RingType::RingReader::Ptr _pRtpReader;
    //推流或拉流客户端采用的rtp传输方式
    Rtsp::eRtpType _rtpType = Rtsp::RTP_Invalid;
    //sdp里面有效的track,包含音频或视频