openRtpServer接口新增only_audio参数，优化语音对讲场景

postman/ZLMediaKit.postman_collection.json

@@ -1404,6 +1404,12 @@
 							"value": "0",
 							"description": "是否指定收流的rtp ssrc, 十进制数字，不指定或指定0时则不过滤rtp，非必选参数",
 							"disabled": true
+						},
+						{
+							"key": "only_audio",
+							"value": "1",
+							"description": "是否为单音频track，用于语音对讲",
+							"disabled": true
 						}
 					]
 				}

server/WebApi.cpp

@@ -391,7 +391,7 @@ Value makeMediaSourceJson(MediaSource &media){
 }
 
 #if defined(ENABLE_RTPPROXY)
-uint16_t openRtpServer(uint16_t local_port, const string &stream_id, int tcp_mode, const string &local_ip, bool re_use_port, uint32_t ssrc) {
+uint16_t openRtpServer(uint16_t local_port, const string &stream_id, int tcp_mode, const string &local_ip, bool re_use_port, uint32_t ssrc, bool only_audio) {
     lock_guard<recursive_mutex> lck(s_rtpServerMapMtx);
     if (s_rtpServerMap.find(stream_id) != s_rtpServerMap.end()) {
         //为了防止RtpProcess所有权限混乱的问题，不允许重复添加相同的stream_id
@@ -399,7 +399,7 @@ uint16_t openRtpServer(uint16_t local_port, const string &stream_id, int tcp_mod
     }
 
     RtpServer::Ptr server = std::make_shared<RtpServer>();
-    server->start(local_port, stream_id, (RtpServer::TcpMode)tcp_mode, local_ip.c_str(), re_use_port, ssrc);
+    server->start(local_port, stream_id, (RtpServer::TcpMode)tcp_mode, local_ip.c_str(), re_use_port, ssrc, only_audio);
     server->setOnDetach([stream_id]() {
         //设置rtp超时移除事件
         lock_guard<recursive_mutex> lck(s_rtpServerMapMtx);
@@ -1140,7 +1140,7 @@ void installWebApi() {
             tcp_mode = 1;
         }
         auto port = openRtpServer(allArgs["port"], stream_id, tcp_mode, "::", allArgs["re_use_port"].as<bool>(),
-                                  allArgs["ssrc"].as<uint32_t>());
+                                  allArgs["ssrc"].as<uint32_t>(), allArgs["only_audio"].as<bool>());
         if (port == 0) {
             throw InvalidArgsException("该stream_id已存在");
         }

src/Common/MediaSink.cpp

@@ -17,16 +17,20 @@ using namespace std;
 namespace mediakit{
 
 bool MediaSink::addTrack(const Track::Ptr &track_in) {
+    if (_only_audio && track_in->getTrackType() != TrackAudio) {
+        InfoL << "Only audio enabled, track ignored: " << track_in->getCodecName();
+        return false;
+    }
     if (!_enable_audio) {
-        //关闭音频时，加快单视频流注册速度
+        // 关闭音频时，加快单视频流注册速度
-        _max_track_size = 1;
         if (track_in->getTrackType() == TrackAudio) {
-            //音频被全局忽略
+            // 音频被全局忽略
+            InfoL << "Audio disabled, audio track ignored";
             return false;
         }
     }
     if (_all_track_ready) {
-        WarnL << "all track is ready, add this track too late!";
+        WarnL << "All track is ready, add track too late: " << track_in->getCodecName();
         return false;
     }
     //克隆Track，只拷贝其数据，不拷贝其数据转发关系
@@ -48,7 +52,7 @@ bool MediaSink::addTrack(const Track::Ptr &track_in) {
         if (frame_unread.size() > kMaxUnreadyFrame) {
             //未就绪的的track，不能缓存太多的帧，否则可能内存溢出
             frame_unread.clear();
-            WarnL << "cached frame of unready track(" << frame->getCodecName() << ") is too much, now cleared";
+            WarnL << "Cached frame of unready track(" << frame->getCodecName() << ") is too much, now cleared";
         }
         //还有Track未就绪，先缓存之
         frame_unread.emplace_back(Frame::getCacheAbleFrame(frame));
@@ -124,8 +128,16 @@ void MediaSink::checkTrackIfReady(){
     }
 }
 
-void MediaSink::addTrackCompleted(){
+void MediaSink::addTrackCompleted() {
-    _max_track_size = _track_map.size();
+    setMaxTrackCount(_track_map.size());
+}
+
+void MediaSink::setMaxTrackCount(size_t i) {
+    if (_all_track_ready) {
+        WarnL << "All track is ready, set max track count ignored";
+        return;
+    }
+    _max_track_size = MAX(MIN(i, 2), 1);
     checkTrackIfReady();
 }
 
@@ -134,14 +146,14 @@ void MediaSink::emitAllTrackReady() {
         return;
     }
 
-    DebugL << "all track ready use " << _ticker.elapsedTime() << "ms";
+    DebugL << "All track ready use " << _ticker.elapsedTime() << "ms";
     if (!_track_ready_callback.empty()) {
         //这是超时强制忽略未准备好的Track
         _track_ready_callback.clear();
         //移除未准备好的Track
         for (auto it = _track_map.begin(); it != _track_map.end();) {
             if (!it->second.second || !it->second.first->ready()) {
-                WarnL << "track not ready for a long time, ignored: " << it->second.first->getCodecName();
+                WarnL << "Track not ready for a long time, ignored: " << it->second.first->getCodecName();
                 it = _track_map.erase(it);
                 continue;
             }
@@ -256,7 +268,7 @@ bool MediaSink::addMuteAudioTrack() {
         return audio->inputFrame(frame);
     });
     onTrackReady(audio);
-    TraceL << "mute aac track added";
+    TraceL << "Mute aac track added";
     return true;
 }
 
@@ -266,6 +278,14 @@ bool MediaSink::isAllTrackReady() const {
 
 void MediaSink::enableAudio(bool flag) {
     _enable_audio = flag;
+    _max_track_size = flag ? 2 : 1;
+}
+
+void MediaSink::setOnlyAudio(){
+    _only_audio = true;
+    _enable_audio = true;
+    _add_mute_audio = false;
+    _max_track_size = 1;
 }
 
 void MediaSink::enableMuteAudio(bool flag) {

src/Common/MediaSink.h

@@ -94,6 +94,12 @@ public:
      */
     void addTrackCompleted() override;
 
+    /**
+     * 设置最大track数，取值范围1~2；该方法与addTrackCompleted类型；
+     * 在设置单track时，可以加快媒体注册速度
+     */
+    void setMaxTrackCount(size_t i);
+
     /**
      * 重置track
      */
@@ -115,6 +121,11 @@ public:
      */
     void enableAudio(bool flag);
 
+    /**
+     * 设置单音频
+     */
+    void setOnlyAudio();
+
     /**
      * 设置是否开启添加静音音频
      */
@@ -157,6 +168,7 @@ private:
 
 private:
     bool _enable_audio = true;
+    bool _only_audio = false;
     bool _add_mute_audio = true;
     bool _all_track_ready = false;
     size_t _max_track_size = 2;

src/Rtp/RtpProcess.cpp

@@ -187,6 +187,10 @@ void RtpProcess::setStopCheckRtp(bool is_check){
     }
 }
 
+void RtpProcess::setOnlyAudio(bool only_audio){
+    _only_audio = only_audio;
+}
+
 void RtpProcess::onDetach() {
     if (_on_detach) {
         _on_detach();
@@ -247,6 +251,9 @@ void RtpProcess::emitOnPublish() {
                                                                               strong_self->_media_info._app,
                                                                               strong_self->_media_info._streamid,0.0f,
                                                                               option);
+                if (strong_self->_only_audio) {
+                    strong_self->_muxer->setOnlyAudio();
+                }
                 strong_self->_muxer->setMediaListener(strong_self);
                 strong_self->doCachedFunc();
                 InfoP(strong_self) << "允许RTP推流";

src/Rtp/RtpProcess.h

@@ -57,6 +57,12 @@ public:
      */
     void setStopCheckRtp(bool is_check=false);
 
+    /**
+     * 设置为单track，单音频时可以加快媒体注册速度
+     * 请在inputRtp前调用此方法，否则可能会是空操作
+     */
+    void setOnlyAudio(bool only_audio);
+
     /**
      * flush输出缓存
      */
@@ -87,6 +93,7 @@ private:
     void doCachedFunc();
 
 private:
+    bool _only_audio = false;
     uint64_t _dts = 0;
     uint64_t _total_bytes = 0;
     std::unique_ptr<sockaddr_storage> _addr;

src/Rtp/RtpServer.cpp

@@ -42,11 +42,12 @@ public:
         }
     }
 
-    void setRtpServerInfo(uint16_t local_port,RtpServer::TcpMode mode,bool re_use_port,uint32_t ssrc){
+    void setRtpServerInfo(uint16_t local_port,RtpServer::TcpMode mode,bool re_use_port,uint32_t ssrc, bool only_audio) {
         _local_port = local_port;
         _tcp_mode = mode;
         _re_use_port = re_use_port;
         _ssrc = ssrc;
+        _only_audio = only_audio;
     }
 
     void setOnDetach(function<void()> cb) {
@@ -60,6 +61,7 @@ public:
     void onRecvRtp(const Socket::Ptr &sock, const Buffer::Ptr &buf, struct sockaddr *addr) {
         if (!_process) {
             _process = RtpSelector::Instance().getProcess(_stream_id, true);
+            _process->setOnlyAudio(_only_audio);
             _process->setOnDetach(std::move(_on_detach));
             cancelDelayTask();
         }
@@ -137,6 +139,7 @@ private:
 
 private:
     bool _re_use_port = false;
+    bool _only_audio = false;
     uint16_t _local_port = 0;
     uint32_t _ssrc = 0;
     RtpServer::TcpMode _tcp_mode = RtpServer::NONE;
@@ -150,7 +153,7 @@ private:
     EventPoller::DelayTask::Ptr _delay_task;
 };
 
-void RtpServer::start(uint16_t local_port, const string &stream_id, TcpMode tcp_mode, const char *local_ip, bool re_use_port, uint32_t ssrc) {
+void RtpServer::start(uint16_t local_port, const string &stream_id, TcpMode tcp_mode, const char *local_ip, bool re_use_port, uint32_t ssrc, bool only_audio) {
     //创建udp服务器
     Socket::Ptr rtp_socket = Socket::createSocket(nullptr, true);
     Socket::Ptr rtcp_socket = Socket::createSocket(nullptr, true);
@@ -176,6 +179,7 @@ void RtpServer::start(uint16_t local_port, const string &stream_id, TcpMode tcp_
         tcp_server = std::make_shared<TcpServer>(rtp_socket->getPoller());
         (*tcp_server)[RtpSession::kStreamID] = stream_id;
         (*tcp_server)[RtpSession::kSSRC] = ssrc;
+        (*tcp_server)[RtpSession::kOnlyAudio] = only_audio;
         if (tcp_mode == PASSIVE) {
             tcp_server->start<RtpSession>(rtp_socket->get_local_port(), local_ip);
         } else if (stream_id.empty()) {
@@ -191,7 +195,7 @@ void RtpServer::start(uint16_t local_port, const string &stream_id, TcpMode tcp_
         //指定了流id，那么一个端口一个流(不管是否包含多个ssrc的多个流，绑定rtp源后，会筛选掉ip端口不匹配的流)
         helper = std::make_shared<RtcpHelper>(std::move(rtcp_socket), stream_id);
         helper->startRtcp();
-        helper->setRtpServerInfo(local_port,tcp_mode,re_use_port,ssrc);
+        helper->setRtpServerInfo(local_port, tcp_mode, re_use_port, ssrc, only_audio);
         bool bind_peer_addr = false;
         rtp_socket->setOnRead([rtp_socket, helper, ssrc, bind_peer_addr](const Buffer::Ptr &buf, struct sockaddr *addr, int addr_len) mutable {
             RtpHeader *header = (RtpHeader *)buf->data();
@@ -211,6 +215,7 @@ void RtpServer::start(uint16_t local_port, const string &stream_id, TcpMode tcp_
 #if 1
         //单端口多线程接收多个流，根据ssrc区分流
         udp_server = std::make_shared<UdpServer>(rtp_socket->getPoller());
+        (*udp_server)[RtpSession::kOnlyAudio] = only_audio;
         udp_server->start<RtpSession>(rtp_socket->get_local_port(), local_ip);
         rtp_socket = nullptr;
 #else

src/Rtp/RtpServer.h

@@ -44,7 +44,7 @@ public:
      * @param ssrc 指定的ssrc
      */
     void start(uint16_t local_port, const std::string &stream_id = "", TcpMode tcp_mode = PASSIVE,
-               const char *local_ip = "::", bool re_use_port = true, uint32_t ssrc = 0);
+               const char *local_ip = "::", bool re_use_port = true, uint32_t ssrc = 0, bool only_audio = false);
 
     /**
      * 连接到tcp服务(tcp主动模式)
@@ -75,6 +75,7 @@ protected:
     std::shared_ptr<RtcpHelper> _rtcp_helper;
     std::function<void()> _on_cleanup;
 
+    bool _only_audio = false;
     //用于tcp主动模式
     TcpMode _tcp_mode = NONE;
 };

src/Rtp/RtpSession.cpp

@@ -22,6 +22,7 @@ namespace mediakit{
 
 const string RtpSession::kStreamID = "stream_id";
 const string RtpSession::kSSRC = "ssrc";
+const string RtpSession::kOnlyAudio = "only_audio";
 
 void RtpSession::attachServer(const Server &server) {
     setParams(const_cast<Server &>(server));
@@ -30,6 +31,7 @@ void RtpSession::attachServer(const Server &server) {
 void RtpSession::setParams(mINI &ini) {
     _stream_id = ini[kStreamID];
     _ssrc = ini[kSSRC];
+    _only_audio = ini[kOnlyAudio];
 }
 
 RtpSession::RtpSession(const Socket::Ptr &sock) : Session(sock) {
@@ -101,6 +103,7 @@ void RtpSession::onRtpPacket(const char *data, size_t len) {
         }
         //tcp情况下，一个tcp链接只可能是一路流，不需要通过多个ssrc来区分，所以不需要频繁getProcess
         _process = RtpSelector::Instance().getProcess(_stream_id, true);
+        _process->setOnlyAudio(_only_audio);
         _process->setDelegate(dynamic_pointer_cast<RtpSession>(shared_from_this()));
     }
     try {

src/Rtp/RtpSession.h

@@ -24,6 +24,7 @@ class RtpSession : public toolkit::Session, public RtpSplitter, public MediaSour
 public:
     static const std::string kStreamID;
     static const std::string kSSRC;
+    static const std::string kOnlyAudio;
 
     RtpSession(const toolkit::Socket::Ptr &sock);
     ~RtpSession() override;
@@ -45,6 +46,7 @@ private:
     bool _is_udp = false;
     bool _search_rtp = false;
     bool _search_rtp_finished = false;
+    bool _only_audio = false;
     uint32_t _ssrc = 0;
     toolkit::Ticker _ticker;
     std::string _stream_id;