Merge 7141759d78 into 8ccb4e95b3

postman/ZLMediaKit.postman_collection.json

@@ -2161,81 +2161,6 @@
 			},
 			"response": []
 		},
-		{
-			"name": "开始双向对讲(startSendRtpTalk)",
-			"request": {
-				"method": "GET",
-				"header": [],
-				"url": {
-					"raw": "{{ZLMediaKit_URL}}/index/api/startSendRtpTalk?secret={{ZLMediaKit_secret}}&vhost={{defaultVhost}}&app=live&stream=obs&ssrc=1&recv_stream_id=",
-					"host": [
-						"{{ZLMediaKit_URL}}"
-					],
-					"path": [
-						"index",
-						"api",
-						"startSendRtpTalk"
-					],
-					"query": [
-						{
-							"key": "secret",
-							"value": "{{ZLMediaKit_secret}}",
-							"description": "api操作密钥(配置文件配置)"
-						},
-						{
-							"key": "vhost",
-							"value": "{{defaultVhost}}",
-							"description": "虚拟主机，例如__defaultVhost__"
-						},
-						{
-							"key": "app",
-							"value": "rtp",
-							"description": "应用名，例如 rtp"
-						},
-						{
-							"key": "stream",
-							"value": "rtc",
-							"description": "流id，例如webrtc推流上来的流id"
-						},
-						{
-							"key": "ssrc",
-							"value": "1",
-							"description": "rtp推流出去的ssrc"
-						},
-						{
-							"key": "recv_stream_id",
-							"value": "",
-							"description": "对方rtp推流上来的流id，我们将通过这个链接回复他rtp流；请注意两个流的app和vhost需一致"
-						},
-						{
-							"key": "from_mp4",
-							"value": "0",
-							"description": "是否推送本地MP4录像，该参数非必选参数",
-							"disabled": true
-						},
-						{
-							"key": "type",
-							"value": "1",
-							"description": "0(ES流)、1(PS流)、2(TS流)，默认1(PS流)；该参数非必选参数",
-							"disabled": true
-						},
-						{
-							"key": "pt",
-							"value": "96",
-							"description": "rtp payload type，默认96；该参数非必选参数",
-							"disabled": true
-						},
-						{
-							"key": "only_audio",
-							"value": "1",
-							"description": "rtp es方式打包时，是否只打包音频；该参数非必选参数",
-							"disabled": true
-						}
-					]
-				}
-			},
-			"response": []
-		},
 		{
 			"name": "停止 发送rtp(stopSendRtp)",
 			"request": {

server/WebApi.cpp

@@ -1606,41 +1606,6 @@ void installWebApi() {
         start_send_rtp(true, API_ARGS_VALUE, invoker);
     });
 
-    api_regist("/index/api/startSendRtpTalk",[](API_ARGS_MAP_ASYNC){
-        CHECK_SECRET();
-        CHECK_ARGS("vhost", "app", "stream", "ssrc", "recv_stream_id");
-        auto src = MediaSource::find(allArgs["vhost"], allArgs["app"], allArgs["stream"], allArgs["from_mp4"].as<int>());
-        if (!src) {
-            throw ApiRetException("can not find the source stream", API::NotFound);
-        }
-        MediaSourceEvent::SendRtpArgs args;
-        args.con_type = mediakit::MediaSourceEvent::SendRtpArgs::kVoiceTalk;
-        args.ssrc = allArgs["ssrc"];
-        args.pt = allArgs["pt"].empty() ? 96 : allArgs["pt"].as<int>();
-        args.data_type = allArgs["type"].empty() ? MediaSourceEvent::SendRtpArgs::kRtpPS : (MediaSourceEvent::SendRtpArgs::DataType)(allArgs["type"].as<int>());
-        args.only_audio = allArgs["only_audio"].as<bool>();
-        args.recv_stream_id = allArgs["recv_stream_id"];
-        args.recv_stream_app = allArgs["app"];
-        args.recv_stream_vhost = allArgs["vhost"];
-
-        src->getOwnerPoller()->async([=]() mutable {
-            try {
-                src->startSendRtp(args, [val, headerOut, invoker](uint16_t local_port, const SockException &ex) mutable {
-                    if (ex) {
-                        val["code"] = API::OtherFailed;
-                        val["msg"] = ex.what();
-                    }
-                    val["local_port"] = local_port;
-                    invoker(200, headerOut, val.toStyledString());
-                });
-            } catch (std::exception &ex) {
-                val["code"] = API::Exception;
-                val["msg"] = ex.what();
-                invoker(200, headerOut, val.toStyledString());
-            }
-        });
-    });
-
     api_regist("/index/api/listRtpSender",[](API_ARGS_MAP_ASYNC){
         CHECK_SECRET();
         CHECK_ARGS("vhost", "app", "stream");

src/Common/MediaSource.h

@@ -124,8 +124,7 @@ public:
             kTcpActive = 0, // tcp主动模式，tcp客户端主动连接对方并发送rtp
             kUdpActive = 1, // udp主动模式，主动发送数据给对方
             kTcpPassive = 2, // tcp被动模式，tcp服务器，等待对方连接并回复rtp
-            kUdpPassive = 3, // udp被动方式，等待对方发送nat打洞包，然后回复rtp至打洞包源地址
-            kVoiceTalk = 4,  // 语音对讲模式，对方必须想推流上来，通过他的推流链路再回复rtp数据
+            kUdpPassive = 3 // udp被动方式，等待对方发送nat打洞包，然后回复rtp至打洞包源地址
         };
 
         // rtp类型  [AUTO-TRANSLATED:acca40ab]

src/Record/MP4Recorder.cpp

@@ -123,15 +123,11 @@ bool MP4Recorder::inputFrame(const Frame::Ptr &frame) {
     if (!(_have_video && frame->getTrackType() == TrackAudio)) {
         // 如果有视频且输入的是音频，那么应该忽略切片逻辑  [AUTO-TRANSLATED:fbb15d93]
         // If there is video and the input is audio, then the slice logic should be ignored
-        if (_last_dts == 0) {
-            // first frame assign dts
-            _last_dts = frame->dts();
-        } else if (_last_dts > frame->dts()) {
+        if (_last_dts == 0 || _last_dts > frame->dts()) {
             // b帧情况下dts时间戳可能回退  [AUTO-TRANSLATED:1de38f77]
             // In the case of b-frames, the dts timestamp may regress
             _last_dts = MIN(frame->dts(), _last_dts);
         }
-        
         auto duration = 5u; // 默认至少一帧5ms
         if (frame->dts() > 0 && frame->dts() > _last_dts) {
             duration = MAX(duration, frame->dts() - _last_dts);

src/Rtp/RtpProcess.cpp

@@ -346,9 +346,5 @@ float RtpProcess::getLossRate(MediaSource &sender, TrackType type) {
     return getLostInterval() * 100 / expected;
 }
 
-const toolkit::Socket::Ptr& RtpProcess::getSock() const {
-    return _sock;
-}
-
 }//namespace mediakit
 #endif//defined(ENABLE_RTPPROXY)

src/Rtp/RtpProcess.h

@@ -102,8 +102,6 @@ public:
     uint16_t get_peer_port() override;
     std::string getIdentifier() const override;
 
-    const toolkit::Socket::Ptr& getSock() const;
-
 protected:
     bool inputFrame(const Frame::Ptr &frame) override;
     bool addTrack(const Track::Ptr & track) override;

src/Rtp/RtpSender.cpp

@@ -190,25 +190,6 @@ void RtpSender::startSend(const MediaSourceEvent::SendRtpArgs &args, const funct
             }
         }, delay_ms / 1000.0, "::", args.src_port);
         InfoL << "start tcp active send rtp to: " << args.dst_url << ":" << args.dst_port;
-    } else if (args.con_type == MediaSourceEvent::SendRtpArgs::kVoiceTalk) {
-        auto src = MediaSource::find(args.recv_stream_vhost, args.recv_stream_app, args.recv_stream_id);
-        if (!src) {
-            cb(0, SockException(Err_other, "can not find the target stream"));
-            return;
-        }
-        auto processor = src->getRtpProcess();
-        if (!processor) {
-            cb(0, SockException(Err_other, "get rtp processor from target stream failed"));
-            return;
-        }
-        auto sock = processor->getSock();
-        if (!sock) {
-            cb(0, SockException(Err_other, "get sock from rtp processor failed"));
-            return;
-        }
-        _socket_rtp = std::move(sock);
-        onConnect();
-        cb(_socket_rtp->get_local_port(), SockException());
     } else {
         CHECK(0, "invalid con type");
     }
@@ -268,51 +249,48 @@ void RtpSender::onConnect() {
     // 加大发送缓存,防止udp丢包之类的问题  [AUTO-TRANSLATED:6e1cb40a]
     // Increase the send buffer to prevent problems such as UDP packet loss
     SockUtil::setSendBuf(_socket_rtp->rawFD(), 4 * 1024 * 1024);
+    if (_args.con_type == MediaSourceEvent::SendRtpArgs::kTcpActive || _args.con_type == MediaSourceEvent::SendRtpArgs::kTcpPassive) {
+        // 关闭tcp no_delay并开启MSG_MORE, 提高发送性能  [AUTO-TRANSLATED:c0f4e378]
+        // Close TCP no_delay and enable MSG_MORE to improve sending performance
+        SockUtil::setNoDelay(_socket_rtp->rawFD(), false);
+        _socket_rtp->setSendFlags(SOCKET_DEFAULE_FLAGS | FLAG_MORE);
+    } else if (_args.udp_rtcp_timeout) {
+        createRtcpSocket();
+    }
+    // 连接建立成功事件  [AUTO-TRANSLATED:ac279c86]
+    // Connection established successfully event
+    weak_ptr<RtpSender> weak_self = shared_from_this();
+    if (!_args.recv_stream_id.empty()) {
+        mINI ini;
+        ini[RtpSession::kStreamID] = _args.recv_stream_id;
+        // 强制同步接收流和发送流的app和vhost  [AUTO-TRANSLATED:134c9663]
+        // Force synchronization of the app and vhost of the receive stream and send stream
+        ini[RtpSession::kApp] = _args.recv_stream_app;
+        ini[RtpSession::kVhost] = _args.recv_stream_vhost;
+        _rtp_session = std::make_shared<RtpSession>(_socket_rtp);
+        _rtp_session->setParams(ini);
 
-    if (_args.con_type != MediaSourceEvent::SendRtpArgs::kVoiceTalk) {
-        if (_args.con_type == MediaSourceEvent::SendRtpArgs::kTcpActive || _args.con_type == MediaSourceEvent::SendRtpArgs::kTcpPassive) {
-            // 关闭tcp no_delay并开启MSG_MORE, 提高发送性能  [AUTO-TRANSLATED:c0f4e378]
-            // Close TCP no_delay and enable MSG_MORE to improve sending performance
-            SockUtil::setNoDelay(_socket_rtp->rawFD(), false);
-            _socket_rtp->setSendFlags(SOCKET_DEFAULE_FLAGS | FLAG_MORE);
-        } else if (_args.udp_rtcp_timeout) {
-            createRtcpSocket();
-        }
-        // 连接建立成功事件  [AUTO-TRANSLATED:ac279c86]
-        // Connection established successfully event
-        weak_ptr<RtpSender> weak_self = shared_from_this();
-        if (!_args.recv_stream_id.empty()) {
-            mINI ini;
-            ini[RtpSession::kStreamID] = _args.recv_stream_id;
-            // 强制同步接收流和发送流的app和vhost  [AUTO-TRANSLATED:134c9663]
-            // Force synchronization of the app and vhost of the receive stream and send stream
-            ini[RtpSession::kApp] = _args.recv_stream_app;
-            ini[RtpSession::kVhost] = _args.recv_stream_vhost;
-            _rtp_session = std::make_shared<RtpSession>(_socket_rtp);
-            _rtp_session->setParams(ini);
-
-            _socket_rtp->setOnRead([weak_self](const Buffer::Ptr &buf, struct sockaddr *addr, int addr_len) {
-                auto strong_self = weak_self.lock();
-                if (!strong_self) {
-                    return;
-                }
-                try {
-                    strong_self->_rtp_session->onRecv(buf);
-                } catch (std::exception &ex) {
-                    SockException err(toolkit::Err_shutdown, ex.what());
-                    strong_self->_rtp_session->shutdown(err);
-                }
-            });
-        } else {
-            _socket_rtp->setOnRead(nullptr);
-        }
-        _socket_rtp->setOnErr([weak_self](const SockException &err) {
+        _socket_rtp->setOnRead([weak_self](const Buffer::Ptr &buf, struct sockaddr *addr, int addr_len) {
             auto strong_self = weak_self.lock();
-            if (strong_self) {
-                strong_self->onErr(err);
+            if (!strong_self) {
+                return;
+            }
+            try {
+                strong_self->_rtp_session->onRecv(buf);
+            } catch (std::exception &ex) {
+                SockException err(toolkit::Err_shutdown, ex.what());
+                strong_self->_rtp_session->shutdown(err);
             }
         });
+    } else {
+        _socket_rtp->setOnRead(nullptr);
     }
+    _socket_rtp->setOnErr([weak_self](const SockException &err) {
+        auto strong_self = weak_self.lock();
+        if (strong_self) {
+            strong_self->onErr(err);
+        }
+    });
     InfoL << "startSend rtp success: " << _socket_rtp->get_peer_ip() << ":" << _socket_rtp->get_peer_port() << ", data_type: " << _args.data_type << ", con_type: " << _args.con_type;
 }
 
@@ -400,51 +378,28 @@ void RtpSender::onFlushRtpList(shared_ptr<List<Buffer::Ptr>> rtp_list) {
         return;
     }
 
-    auto send_func = [this](const shared_ptr<List<Buffer::Ptr>> &rtp_list) {
-        size_t i = 0;
-        auto size = rtp_list->size();
-        rtp_list->for_each([&](Buffer::Ptr &packet) {
-            switch (_args.con_type) {
-                case MediaSourceEvent::SendRtpArgs::kUdpActive:
-                case MediaSourceEvent::SendRtpArgs::kUdpPassive: {
-                    onSendRtpUdp(packet, i == 0);
-                    // udp模式，rtp over tcp前4个字节可以忽略  [AUTO-TRANSLATED:5d648f4b]
-                    // UDP mode, the first 4 bytes of rtp over tcp can be ignored
-                    _socket_rtp->send(std::make_shared<BufferRtp>(std::move(packet), RtpPacket::kRtpTcpHeaderSize), nullptr, 0, ++i == size);
-                    break;
-                }
-                case MediaSourceEvent::SendRtpArgs::kTcpActive:
-                case MediaSourceEvent::SendRtpArgs::kTcpPassive: {
-                    // tcp模式, rtp over tcp前2个字节可以忽略,只保留后续rtp长度的2个字节  [AUTO-TRANSLATED:a3bc338a]
-                    // TCP mode, the first 2 bytes of rtp over tcp can be ignored, only the subsequent 2 bytes of rtp length are retained
-                    _socket_rtp->send(std::make_shared<BufferRtp>(std::move(packet), 2), nullptr, 0, ++i == size);
-                    break;
-                }
-                case MediaSourceEvent::SendRtpArgs::kVoiceTalk: {
-                    auto type = _socket_rtp->alive() ? _socket_rtp->sockType() : SockNum::Sock_Invalid;
-                    if (type == SockNum::Sock_UDP) {
-                        _socket_rtp->send(std::make_shared<BufferRtp>(std::move(packet), RtpPacket::kRtpTcpHeaderSize), nullptr, 0, ++i == size);
-                    } else if (type == SockNum::Sock_TCP) {
-                        _socket_rtp->send(std::make_shared<BufferRtp>(std::move(packet), 2), nullptr, 0, ++i == size);
-                    } else {
-                        onErr(SockException(Err_other, "dst socket disconnected"));
-                    }
-                    break;
-                }
-                default: CHECK(0);
+    size_t i = 0;
+    auto size = rtp_list->size();
+    rtp_list->for_each([&](Buffer::Ptr &packet) {
+        switch (_args.con_type) {
+            case MediaSourceEvent::SendRtpArgs::kUdpActive:
+            case MediaSourceEvent::SendRtpArgs::kUdpPassive: {
+                onSendRtpUdp(packet, i == 0);
+                // udp模式，rtp over tcp前4个字节可以忽略  [AUTO-TRANSLATED:5d648f4b]
+                // UDP mode, the first 4 bytes of rtp over tcp can be ignored
+                _socket_rtp->send(std::make_shared<BufferRtp>(std::move(packet), RtpPacket::kRtpTcpHeaderSize), nullptr, 0, ++i == size);
+                break;
             }
-        });
-    };
-    if (_args.con_type != MediaSourceEvent::SendRtpArgs::kVoiceTalk) {
-        weak_ptr<RtpSender> weak_self = shared_from_this();
-        _socket_rtp->getPoller()->async([weak_self, rtp_list, send_func]() {
-            if (auto strong_self = weak_self.lock()) {
-                send_func(rtp_list);
+            case MediaSourceEvent::SendRtpArgs::kTcpActive:
+            case MediaSourceEvent::SendRtpArgs::kTcpPassive: {
+                // tcp模式, rtp over tcp前2个字节可以忽略,只保留后续rtp长度的2个字节  [AUTO-TRANSLATED:a3bc338a]
+                // TCP mode, the first 2 bytes of rtp over tcp can be ignored, only the subsequent 2 bytes of rtp length are retained
+                _socket_rtp->send(std::make_shared<BufferRtp>(std::move(packet), 2), nullptr, 0, ++i == size);
+                break;
             }
-        });
-    } else {
-        send_func(rtp_list);
-    }
+            default: CHECK(0);
+        }
+    });
 }
 
 void RtpSender::onErr(const SockException &ex) {

src/Rtsp/RtspSession.cpp

@@ -1136,7 +1136,7 @@ int RtspSession::getTrackIndexByTrackType(TrackType type) {
 
 int RtspSession::getTrackIndexByControlUrl(const string &control_url) {
     for (size_t i = 0; i < _sdp_track.size(); ++i) {
-        if (control_url.find(_sdp_track[i]->getControlUrl(_content_base)) == 0) {
+        if (control_url == _sdp_track[i]->getControlUrl(_content_base)) {
             return i;
         }
     }