mirror of
https://github.com/ZLMediaKit/ZLMediaKit.git
synced 2024-11-22 10:40:05 +08:00
startSendRtp接口支持rtcp接收超时主动停止
This commit is contained in:
parent
b7ef766701
commit
9f0c15a4f0
@ -1 +1 @@
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Subproject commit 4e127f53a93dd1b7665f04a0ec7312d67a34ab6f
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Subproject commit 658271fdf4fb497b0665c06544627789844a003a
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@ -1562,6 +1562,12 @@
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"value": "1",
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"description": "rtp es方式打包时,是否只打包音频,该参数非必选参数",
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"disabled": true
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},
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{
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"key": "udp_rtcp_timeout",
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"value": "0",
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"description": "udp方式推流时,是否开启rtcp发送和rtcp接收超时判断,开启后(默认关闭),如果接收rr rtcp超时,将导致主动停止rtp发送",
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"disabled": true
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}
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]
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}
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@ -1147,7 +1147,8 @@ void installWebApi() {
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args.src_port = allArgs["src_port"];
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args.pt = allArgs["pt"].empty() ? 96 : allArgs["pt"].as<int>();
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args.use_ps = allArgs["use_ps"].empty() ? true : allArgs["use_ps"].as<bool>();
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args.only_audio = allArgs["only_audio"].empty() ? false : allArgs["only_audio"].as<bool>();
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args.only_audio = allArgs["only_audio"].as<bool>();
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args.udp_rtcp_timeout = allArgs["udp_rtcp_timeout"];
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TraceL << "startSendRtp, pt " << int(args.pt) << " ps " << args.use_ps << " audio " << args.only_audio;
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src->getOwnerPoller()->async([=]() mutable {
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@ -1178,7 +1179,7 @@ void installWebApi() {
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args.src_port = allArgs["src_port"];
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args.pt = allArgs["pt"].empty() ? 96 : allArgs["pt"].as<int>();
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args.use_ps = allArgs["use_ps"].empty() ? true : allArgs["use_ps"].as<bool>();
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args.only_audio = allArgs["only_audio"].empty() ? false : allArgs["only_audio"].as<bool>();
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args.only_audio = allArgs["only_audio"].as<bool>();
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TraceL << "startSendRtpPassive, pt " << int(args.pt) << " ps " << args.use_ps << " audio " << args.only_audio;
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src->getOwnerPoller()->async([=]() mutable {
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@ -111,6 +111,15 @@ public:
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uint16_t dst_port;
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// 发送目标主机地址,可以是ip或域名
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std::string dst_url;
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//udp发送时,是否开启rr rtcp接收超时判断
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bool udp_rtcp_timeout = false;
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//tcp被动发送服务器延时关闭事件,单位毫秒
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uint32_t tcp_passive_close_delay_ms = 5 * 1000;
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//udp 发送时,rr rtcp包接收超时时间,单位毫秒
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uint32_t rtcp_timeout_ms = 30 * 1000;
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//udp 发送时,发送sr rtcp包间隔,单位毫秒
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uint32_t rtcp_send_interval_ms = 5 * 1000;
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};
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// 开始发送ps-rtp
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@ -223,7 +223,7 @@ void MultiMediaSourceMuxer::startSendRtp(MediaSource &, const MediaSourceEvent::
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#if defined(ENABLE_RTPPROXY)
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auto rtp_sender = std::make_shared<RtpSender>();
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weak_ptr<MultiMediaSourceMuxer> weak_self = shared_from_this();
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rtp_sender->startSend(args, [args, weak_self, rtp_sender, cb](uint16_t local_port, const SockException &ex) {
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rtp_sender->startSend(args, [args, weak_self, rtp_sender, cb](uint16_t local_port, const SockException &ex) mutable {
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cb(local_port, ex);
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auto strong_self = weak_self.lock();
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if (!strong_self || ex) {
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@ -233,7 +233,15 @@ void MultiMediaSourceMuxer::startSendRtp(MediaSource &, const MediaSourceEvent::
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rtp_sender->addTrack(track);
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}
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rtp_sender->addTrackCompleted();
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strong_self->_rtp_sender[args.ssrc] = rtp_sender;
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auto ssrc = args.ssrc;
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rtp_sender->setOnClose([weak_self, ssrc]() {
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if (auto strong_self = weak_self.lock()) {
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strong_self->_rtp_sender.erase(ssrc);
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WarnL << "stream:" << strong_self->_get_origin_url() << " stop send rtp:" << ssrc;
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}
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});
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strong_self->_rtp_sender[args.ssrc] = std::move(rtp_sender);
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});
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#else
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cb(0, SockException(Err_other, "该功能未启用,编译时请打开ENABLE_RTPPROXY宏"));
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@ -22,7 +22,7 @@ namespace mediakit{
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RtpSender::RtpSender() {
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_poller = EventPollerPool::Instance().getPoller();
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_socket = Socket::createSocket(_poller, false);
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_socket_rtp = Socket::createSocket(_poller, false);
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}
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void RtpSender::startSend(const MediaSourceEvent::SendRtpArgs &args, const function<void(uint16_t local_port, const SockException &ex)> &cb){
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@ -55,7 +55,7 @@ void RtpSender::startSend(const MediaSourceEvent::SendRtpArgs &args, const funct
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makeSockPair(pr, "::", false, false);
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}
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// tcp服务器默认开启5秒
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auto delay_task = _poller->doDelayTask(5 * 1000, [tcp_listener, cb]() mutable {
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auto delay_task = _poller->doDelayTask(_args.tcp_passive_close_delay_ms, [tcp_listener, cb]() mutable {
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cb(0, SockException(Err_timeout, "wait tcp connection timeout"));
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tcp_listener = nullptr;
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return 0;
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@ -67,7 +67,7 @@ void RtpSender::startSend(const MediaSourceEvent::SendRtpArgs &args, const funct
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}
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//立即关闭tcp服务器
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delay_task->cancel();
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strong_self->_socket = sock;
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strong_self->_socket_rtp = sock;
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strong_self->onConnect();
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cb(sock->get_local_port(), SockException());
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InfoL << "accept connection from:" << sock->get_peer_ip() << ":" << sock->get_peer_port();
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@ -103,12 +103,12 @@ void RtpSender::startSend(const MediaSourceEvent::SendRtpArgs &args, const funct
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try {
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if (args.src_port) {
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//指定端口
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if (!strong_self->_socket->bindUdpSock(args.src_port, ifr_ip)) {
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if (!strong_self->_socket_rtp->bindUdpSock(args.src_port, ifr_ip)) {
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throw std::invalid_argument(StrPrinter << "bindUdpSock failed on port:" << args.src_port
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<< ", err:" << get_uv_errmsg(true));
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}
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} else {
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auto pr = std::make_pair(strong_self->_socket, Socket::createSocket(strong_self->_poller, false));
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auto pr = std::make_pair(strong_self->_socket_rtp, Socket::createSocket(strong_self->_poller, false));
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//从端口池获取随机端口
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makeSockPair(pr, ifr_ip, true);
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}
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@ -116,20 +116,20 @@ void RtpSender::startSend(const MediaSourceEvent::SendRtpArgs &args, const funct
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cb(0, SockException(Err_other, ex.what()));
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return;
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}
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strong_self->_socket->bindPeerAddr((struct sockaddr *)&addr);
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strong_self->_socket_rtp->bindPeerAddr((struct sockaddr *)&addr);
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strong_self->onConnect();
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cb(strong_self->_socket->get_local_port(), SockException());
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cb(strong_self->_socket_rtp->get_local_port(), SockException());
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});
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});
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} else {
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_socket->connect(args.dst_url, args.dst_port, [cb, weak_self](const SockException &err) {
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_socket_rtp->connect(args.dst_url, args.dst_port, [cb, weak_self](const SockException &err) {
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auto strong_self = weak_self.lock();
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if (strong_self) {
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if (!err) {
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//tcp连接成功
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strong_self->onConnect();
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}
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cb(strong_self->_socket->get_local_port(), err);
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cb(strong_self->_socket_rtp->get_local_port(), err);
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} else {
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cb(0, err);
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}
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@ -137,26 +137,73 @@ void RtpSender::startSend(const MediaSourceEvent::SendRtpArgs &args, const funct
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}
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}
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void RtpSender::createRtcpSocket() {
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if (_socket_rtcp) {
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return;
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}
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_socket_rtcp = Socket::createSocket(_socket_rtp->getPoller(), false);
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//rtcp端口使用户rtp端口+1
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if(!_socket_rtcp->bindUdpSock(_socket_rtp->get_local_port() + 1, _socket_rtp->get_local_ip(), false)){
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WarnL << "bind rtcp udp socket failed:" << get_uv_errmsg(true);
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_socket_rtcp = nullptr;
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return;
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}
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struct sockaddr_storage addr;
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//目标rtp端口
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SockUtil::get_sock_peer_addr(_socket_rtp->rawFD(), addr);
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//绑定目标rtcp端口(目标rtp端口 + 1)
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switch (addr.ss_family) {
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case AF_INET: ((sockaddr_in *)&addr)->sin_port = htons(ntohs(((sockaddr_in *)&addr)->sin_port) + 1); break;
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case AF_INET6: ((sockaddr_in6 *)&addr)->sin6_port = htons(ntohs(((sockaddr_in6 *)&addr)->sin6_port) + 1); break;
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default: assert(0); break;
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}
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_socket_rtcp->bindPeerAddr((struct sockaddr *)&addr);
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_rtcp_context = std::make_shared<RtcpContextForSend>();
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weak_ptr<RtpSender> weak_self = shared_from_this();
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_socket_rtcp->setOnRead([weak_self](const Buffer::Ptr &buf, struct sockaddr *, int) {
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//接收receive report rtcp
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auto strong_self = weak_self.lock();
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if (!strong_self) {
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return;
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}
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auto rtcp_arr = RtcpHeader::loadFromBytes(buf->data(), buf->size());
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for (auto &rtcp : rtcp_arr) {
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strong_self->onRecvRtcp(rtcp);
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}
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});
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InfoL << "open rtcp port success, start check rr rtcp timeout";
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}
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void RtpSender::onRecvRtcp(RtcpHeader *rtcp) {
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_rtcp_context->onRtcp(rtcp);
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_rtcp_recv_ticker.resetTime();
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}
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//连接建立成功事件
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void RtpSender::onConnect(){
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_is_connect = true;
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//加大发送缓存,防止udp丢包之类的问题
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SockUtil::setSendBuf(_socket->rawFD(), 4 * 1024 * 1024);
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SockUtil::setSendBuf(_socket_rtp->rawFD(), 4 * 1024 * 1024);
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if (!_args.is_udp) {
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//关闭tcp no_delay并开启MSG_MORE, 提高发送性能
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SockUtil::setNoDelay(_socket->rawFD(), false);
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_socket->setSendFlags(SOCKET_DEFAULE_FLAGS | FLAG_MORE);
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SockUtil::setNoDelay(_socket_rtp->rawFD(), false);
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_socket_rtp->setSendFlags(SOCKET_DEFAULE_FLAGS | FLAG_MORE);
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} else if (_args.udp_rtcp_timeout) {
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createRtcpSocket();
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}
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//连接建立成功事件
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weak_ptr<RtpSender> weak_self = shared_from_this();
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_socket->setOnErr([weak_self](const SockException &err) {
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_socket_rtp->setOnErr([weak_self](const SockException &err) {
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auto strong_self = weak_self.lock();
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if (strong_self) {
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strong_self->onErr(err);
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}
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});
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//获取本地端口,断开重连后确保端口不变
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_args.src_port = _socket->get_local_port();
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InfoL << "开始发送 rtp:" << _socket->get_peer_ip() << ":" << _socket->get_peer_port() << ", 是否为udp方式:" << _args.is_udp;
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_args.src_port = _socket_rtp->get_local_port();
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InfoL << "开始发送 rtp:" << _socket_rtp->get_peer_ip() << ":" << _socket_rtp->get_peer_port() << ", 是否为udp方式:" << _args.is_udp;
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}
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bool RtpSender::addTrack(const Track::Ptr &track){
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@ -177,6 +224,41 @@ bool RtpSender::inputFrame(const Frame::Ptr &frame) {
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return _is_connect ? _interface->inputFrame(frame) : false;
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}
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void RtpSender::onSendRtpUdp(const toolkit::Buffer::Ptr &buf, bool check) {
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if (!_socket_rtcp) {
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return;
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}
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auto rtp = static_pointer_cast<RtpPacket>(buf);
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_rtcp_context->onRtp(rtp->getSeq(), rtp->getStamp(), rtp->getStampMS(), 90000 /*not used*/, rtp->size());
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if (!check) {
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//减少判断次数
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return;
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}
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//每5秒发送一次rtcp
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if (_rtcp_send_ticker.elapsedTime() > _args.rtcp_send_interval_ms) {
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_rtcp_send_ticker.resetTime();
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//rtcp ssrc为rtp ssrc + 1
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auto sr = _rtcp_context->createRtcpSR(atoi(_args.ssrc.data()) + 1);
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//send sender report rtcp
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_socket_rtcp->send(sr);
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}
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if (_rtcp_recv_ticker.elapsedTime() > _args.rtcp_timeout_ms) {
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//接收rr rtcp超时
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WarnL << "recv rr rtcp timeout";
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_rtcp_recv_ticker.resetTime();
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onClose();
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}
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}
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void RtpSender::onClose() {
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auto cb = _on_close;
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if (cb) {
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_poller->async([cb]() { cb(); }, false);
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}
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}
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//此函数在其他线程执行
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void RtpSender::onFlushRtpList(shared_ptr<List<Buffer::Ptr> > rtp_list) {
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if(!_is_connect){
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@ -184,18 +266,22 @@ void RtpSender::onFlushRtpList(shared_ptr<List<Buffer::Ptr> > rtp_list) {
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return;
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}
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auto is_udp = _args.is_udp;
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auto socket = _socket;
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_poller->async([rtp_list, is_udp, socket]() {
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weak_ptr<RtpSender> weak_self = shared_from_this();
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_poller->async([rtp_list, weak_self]() {
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auto strong_self = weak_self.lock();
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if (!strong_self) {
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return;
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}
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size_t i = 0;
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auto size = rtp_list->size();
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rtp_list->for_each([&](Buffer::Ptr &packet) {
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if (is_udp) {
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if (strong_self->_args.is_udp) {
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strong_self->onSendRtpUdp(packet, i == 0);
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//udp模式,rtp over tcp前4个字节可以忽略
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socket->send(std::make_shared<BufferRtp>(std::move(packet), 4), nullptr, 0, ++i == size);
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strong_self->_socket_rtp->send(std::make_shared<BufferRtp>(std::move(packet), RtpPacket::kRtpTcpHeaderSize), nullptr, 0, ++i == size);
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} else {
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//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
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socket->send(std::make_shared<BufferRtp>(std::move(packet), 2), nullptr, 0, ++i == size);
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strong_self->_socket_rtp->send(std::make_shared<BufferRtp>(std::move(packet), 2), nullptr, 0, ++i == size);
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}
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});
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});
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@ -206,6 +292,8 @@ void RtpSender::onErr(const SockException &ex, bool is_connect) {
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if (_args.passive) {
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WarnL << "tcp passive connection lost: " << ex.what();
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//tcp被动模式,如果对方断开连接,应该停止发送rtp
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onClose();
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} else {
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//监听socket断开事件,方便重连
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if (is_connect) {
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@ -232,5 +320,9 @@ void RtpSender::onErr(const SockException &ex, bool is_connect) {
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}, _poller);
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}
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void RtpSender::setOnClose(std::function<void()> on_close){
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_on_close = std::move(on_close);
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}
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}//namespace mediakit
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#endif// defined(ENABLE_RTPPROXY)
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#if defined(ENABLE_RTPPROXY)
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#include "PSEncoder.h"
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#include "Extension/CommonRtp.h"
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#include "Rtcp/RtcpContext.h"
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namespace mediakit{
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@ -53,6 +54,8 @@ public:
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*/
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virtual void resetTracks() override;
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void setOnClose(std::function<void()> on_close);
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private:
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//合并写输出
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void onFlushRtpList(std::shared_ptr<toolkit::List<toolkit::Buffer::Ptr> > rtp_list);
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@ -60,14 +63,23 @@ private:
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void onConnect();
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//异常断开socket事件
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void onErr(const toolkit::SockException &ex, bool is_connect = false);
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void createRtcpSocket();
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void onRecvRtcp(RtcpHeader *rtcp);
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void onSendRtpUdp(const toolkit::Buffer::Ptr &buf, bool check);
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void onClose();
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private:
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bool _is_connect = false;
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MediaSourceEvent::SendRtpArgs _args;
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toolkit::Socket::Ptr _socket;
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toolkit::Socket::Ptr _socket_rtp;
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toolkit::Socket::Ptr _socket_rtcp;
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toolkit::EventPoller::Ptr _poller;
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toolkit::Timer::Ptr _connect_timer;
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MediaSinkInterface::Ptr _interface;
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std::shared_ptr<RtcpContext> _rtcp_context;
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toolkit::Ticker _rtcp_send_ticker;
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toolkit::Ticker _rtcp_recv_ticker;
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std::function<void()> _on_close;
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};
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}//namespace mediakit
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