新增语音对讲接口(startSendRtpTalk)

src/Rtp/RtpSender.cpp

@@ -190,6 +190,25 @@ 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");
     }
@@ -249,48 +268,51 @@ 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);
 
-        _socket_rtp->setOnRead([weak_self](const Buffer::Ptr &buf, struct sockaddr *addr, int addr_len) {
+    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) {
             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);
+            if (strong_self) {
+                strong_self->onErr(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;
 }
 
@@ -378,28 +400,51 @@ void RtpSender::onFlushRtpList(shared_ptr<List<Buffer::Ptr>> rtp_list) {
         return;
     }
 
-    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;
+    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);
             }
-            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;
+        });
+    };
+    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);
             }
-            default: CHECK(0);
-        }
-    });
+        });
+    } else {
+        send_func(rtp_list);
+    }
 }
 
 void RtpSender::onErr(const SockException &ex) {