WebRTC进阶流媒体服务器开发（六）Mediasoup源码分析之Mediasoup主业务流程
一：主业务的创建
主要场景是对房间的管理，多方进行音视频互动。
Router代表房间，Transport代表一个传输，每个用户加入房间都会创建一个对应的连接。
Producer生产者，共享的音视频流中，每个音频、视频流都会产生一个生产者
Consumer消费者，对于每个加入房间的用户，都可以消费其他用户的音视频数据
1.首先调用CreateRouter，创建房间Router，然后加入worker的管理列表中。对于每个worker都会包含多个Router
2.创建Router之后，调用CreateTransport创建Transport，返回accept告诉Router创建transport成功，返回给Router
3.Router发送connect信令给Transport，连接创建成功
4.连接创建成功之后，调用CreateProducer创建生产者（可以创建多个）
5.如果需要获取其他用户的数据，则需要去CreateConsumer创建消费者
二：主业务源码分析
见worker.cpp中OnChannelRequest方法，处理请求之后，使用Request->Accept(data);返回确认消息
inline void Worker::OnChannelRequest(Channel::ChannelSocket* /*channel*/, Channel::ChannelRequest* request) { MS_TRACE(); MS_DEBUG_DEV( "Channel request received [method:%s, id:%" PRIu32 "]", request->method.c_str(), request->id); switch (request->methodId) { case Channel::ChannelRequest::MethodId::WORKER_CLOSE: { if (this->closed) return; MS_DEBUG_DEV("Worker close request, stopping"); Close(); break; } case Channel::ChannelRequest::MethodId::WORKER_DUMP: { json data = json::object(); FillJson(data); request->Accept(data); break; } case Channel::ChannelRequest::MethodId::WORKER_GET_RESOURCE_USAGE: { json data = json::object(); FillJsonResourceUsage(data); request->Accept(data); break; } case Channel::ChannelRequest::MethodId::WORKER_UPDATE_SETTINGS: { Settings::HandleRequest(request); break; } case Channel::ChannelRequest::MethodId::WORKER_CREATE_ROUTER: //创建房间 { std::string routerId; // This may throw. SetNewRouterIdFromInternal(request->internal, routerId); //根据request获取routerId（在应用层随机产生的） auto* router = new RTC::Router(routerId); //创建router this->mapRouters[routerId] = router; //存放在字典中 MS_DEBUG_DEV("Router created [routerId:%s]", routerId.c_str()); request->Accept(); //回复消息，创建完成 break; } case Channel::ChannelRequest::MethodId::ROUTER_CLOSE: { // This may throw. RTC::Router* router = GetRouterFromInternal(request->internal); // Remove it from the map and delete it. this->mapRouters.erase(router->id); delete router; MS_DEBUG_DEV("Router closed [id:%s]", router->id.c_str()); request->Accept(); break; } // Any other request must be delivered to the corresponding Router. default: //其他的信令处理，比如创建transport，accept没有在这里处理，在HandleRequest中调用！！！！ { // This may throw. RTC::Router* router = GetRouterFromInternal(request->internal); router->HandleRequest(request); break; } } }
（一）进入Router.cpp中----创建Transport
void Router::SetNewTransportIdFromInternal(json& internal, std::string& transportId) const { MS_TRACE(); auto jsonTransportIdIt = internal.find("transportId"); //查找transportId,在jsonRequest中的5元组中的transportId中
if (jsonTransportIdIt == internal.end() || !jsonTransportIdIt->is_string()) MS_THROW_ERROR("missing internal.transportId"); transportId.assign(jsonTransportIdIt->get<std::string>()); //赋值 if (this->mapTransports.find(transportId) != this->mapTransports.end()) MS_THROW_ERROR("a Transport with same transportId already exists"); }
void Router::HandleRequest(Channel::ChannelRequest* request) { MS_TRACE(); switch (request->methodId) { case Channel::ChannelRequest::MethodId::ROUTER_DUMP: { json data = json::object(); FillJson(data); request->Accept(data); break; } case Channel::ChannelRequest::MethodId::ROUTER_CREATE_WEBRTC_TRANSPORT: //多个transport，这里主要查看WebRtcTransport { std::string transportId; // This may throw. SetNewTransportIdFromInternal(request->internal, transportId); //从request中获取transportId，存放到变量transportId中 // This may throw. auto* webRtcTransport = new RTC::WebRtcTransport(transportId, this, request->data); //创建WebRtcTransport对象，传入transportId和request5元组中的data数据 // Insert into the map. this->mapTransports[transportId] = webRtcTransport; //放入到map中去 MS_DEBUG_DEV("WebRtcTransport created [transportId:%s]", transportId.c_str()); json data = json::object(); //声明一个JSON格式对象 webRtcTransport->FillJson(data); //开始将webRtcTransport中的响应，返回确认消息到data request->Accept(data); //将接受的数据返回给JS层 break; } case Channel::ChannelRequest::MethodId::ROUTER_CREATE_PLAIN_TRANSPORT:case Channel::ChannelRequest::MethodId::ROUTER_CREATE_PIPE_TRANSPORT:case Channel::ChannelRequest::MethodId::ROUTER_CREATE_DIRECT_TRANSPORT:case Channel::ChannelRequest::MethodId::ROUTER_CREATE_AUDIO_LEVEL_OBSERVER:case Channel::ChannelRequest::MethodId::TRANSPORT_CLOSE:case Channel::ChannelRequest::MethodId::RTP_OBSERVER_CLOSE:case Channel::ChannelRequest::MethodId::RTP_OBSERVER_PAUSE:case Channel::ChannelRequest::MethodId::RTP_OBSERVER_RESUME:case Channel::ChannelRequest::MethodId::RTP_OBSERVER_ADD_PRODUCER:case Channel::ChannelRequest::MethodId::RTP_OBSERVER_REMOVE_PRODUCER:// Any other request must be delivered to the corresponding Transport. default: { // This may throw. RTC::Transport* transport = GetTransportFromInternal(request->internal); transport->HandleRequest(request); break; } } }
查看WebRtcTransport.cpp文件，构造函数
WebRtcTransport::WebRtcTransport(const std::string& id, RTC::Transport::Listener* listener, json& data) : RTC::Transport::Transport(id, listener, data) { //主要实现JSON数据的解析---request->data就是data bool enableUdp{ true }; auto jsonEnableUdpIt = data.find("enableUdp");bool enableTcp{ false }; auto jsonEnableTcpIt = data.find("enableTcp");bool preferUdp{ false }; auto jsonPreferUdpIt = data.find("preferUdp");bool preferTcp{ false }; auto jsonPreferTcpIt = data.find("preferTcp"); auto jsonListenIpsIt = data.find("listenIps"); std::vector<ListenIp> listenIps(jsonListenIpsIt->size()); for (size_t i{ 0 }; i < jsonListenIpsIt->size(); ++i) { auto& jsonListenIp = (*jsonListenIpsIt)[i]; auto& listenIp = listenIps[i]; if (!jsonListenIp.is_object()) MS_THROW_TYPE_ERROR("wrong listenIp (not an object)"); auto jsonIpIt = jsonListenIp.find("ip"); if (jsonIpIt == jsonListenIp.end()) MS_THROW_TYPE_ERROR("missing listenIp.ip"); else if (!jsonIpIt->is_string()) MS_THROW_TYPE_ERROR("wrong listenIp.ip (not an string"); listenIp.ip.assign(jsonIpIt->get<std::string>()); // This may throw. Utils::IP::NormalizeIp(listenIp.ip); auto jsonAnnouncedIpIt = jsonListenIp.find("announcedIp"); if (jsonAnnouncedIpIt != jsonListenIp.end()) { if (!jsonAnnouncedIpIt->is_string()) MS_THROW_TYPE_ERROR("wrong listenIp.announcedIp (not an string)"); listenIp.announcedIp.assign(jsonAnnouncedIpIt->get<std::string>()); } } try { uint16_t iceLocalPreferenceDecrement{ 0 }; if (enableUdp && enableTcp) this->iceCandidates.reserve(2 * jsonListenIpsIt->size()); else this->iceCandidates.reserve(jsonListenIpsIt->size()); for (auto& listenIp : listenIps) { if (enableUdp) { uint16_t iceLocalPreference = IceCandidateDefaultLocalPriority - iceLocalPreferenceDecrement; if (preferUdp) iceLocalPreference += 1000; uint32_t icePriority = generateIceCandidatePriority(iceLocalPreference); // This may throw. auto* udpSocket = new RTC::UdpSocket(this, listenIp.ip); this->udpSockets[udpSocket] = listenIp.announcedIp; if (listenIp.announcedIp.empty()) this->iceCandidates.emplace_back(udpSocket, icePriority); else this->iceCandidates.emplace_back(udpSocket, icePriority, listenIp.announcedIp); } if (enableTcp) { uint16_t iceLocalPreference = IceCandidateDefaultLocalPriority - iceLocalPreferenceDecrement; if (preferTcp) iceLocalPreference += 1000; uint32_t icePriority = generateIceCandidatePriority(iceLocalPreference); // This may throw. auto* tcpServer = new RTC::TcpServer(this, this, listenIp.ip); this->tcpServers[tcpServer] = listenIp.announcedIp; if (listenIp.announcedIp.empty()) this->iceCandidates.emplace_back(tcpServer, icePriority); else this->iceCandidates.emplace_back(tcpServer, icePriority, listenIp.announcedIp); } // Decrement initial ICE local preference for next IP. iceLocalPreferenceDecrement += 100; } // Create a ICE server. this->iceServer = new RTC::IceServer( this, Utils::Crypto::GetRandomString(16), Utils::Crypto::GetRandomString(32)); // Create a DTLS transport. this->dtlsTransport = new RTC::DtlsTransport(this); } catch (const MediaSoupError& error) { // Must delete everything since the destructor won't be called. } }
Mediasoup手册：https://mediasoup.org/documentation/v3/mediasoup/api/#WebRtcTransport
（二）进入Router.cpp中----创建connect连接
void Router::HandleRequest(Channel::ChannelRequest* request) { MS_TRACE(); switch (request->methodId) { //因为connect不属于Router，所以最后也是进入default中去！！！ // Any other request must be delivered to the corresponding Transport. default: { // This may throw. RTC::Transport* transport = GetTransportFromInternal(request->internal); //从request中获取transport对象 transport->HandleRequest(request); break; } } }
进入WebRtcTransport.cpp中去，查看对应的HandleRequest方法，创建connect连接！
void WebRtcTransport::HandleRequest(Channel::ChannelRequest* request) { MS_TRACE(); switch (request->methodId) { case Channel::ChannelRequest::MethodId::TRANSPORT_CONNECT: //涉及到很多协议！ { // Ensure this method is not called twice. if (this->connectCalled) MS_THROW_ERROR("connect() already called"); RTC::DtlsTransport::Fingerprint dtlsRemoteFingerprint; RTC::DtlsTransport::Role dtlsRemoteRole; auto jsonDtlsParametersIt = request->data.find("dtlsParameters"); if (jsonDtlsParametersIt == request->data.end() || !jsonDtlsParametersIt->is_object()) MS_THROW_TYPE_ERROR("missing dtlsParameters"); auto jsonFingerprintsIt = jsonDtlsParametersIt->find("fingerprints"); if (jsonFingerprintsIt == jsonDtlsParametersIt->end() || !jsonFingerprintsIt->is_array()) { MS_THROW_TYPE_ERROR("missing dtlsParameters.fingerprints"); } else if (jsonFingerprintsIt->empty()) { MS_THROW_TYPE_ERROR("empty dtlsParameters.fingerprints array"); } // NOTE: Just take the first fingerprint. for (auto& jsonFingerprint : *jsonFingerprintsIt) { if (!jsonFingerprint.is_object()) MS_THROW_TYPE_ERROR("wrong entry in dtlsParameters.fingerprints (not an object)"); auto jsonAlgorithmIt = jsonFingerprint.find("algorithm"); if (jsonAlgorithmIt == jsonFingerprint.end()) MS_THROW_TYPE_ERROR("missing fingerprint.algorithm"); else if (!jsonAlgorithmIt->is_string()) MS_THROW_TYPE_ERROR("wrong fingerprint.algorithm (not a string)"); dtlsRemoteFingerprint.algorithm = RTC::DtlsTransport::GetFingerprintAlgorithm(jsonAlgorithmIt->get<std::string>()); if (dtlsRemoteFingerprint.algorithm == RTC::DtlsTransport::FingerprintAlgorithm::NONE) { MS_THROW_TYPE_ERROR("invalid fingerprint.algorithm value"); } auto jsonValueIt = jsonFingerprint.find("value"); if (jsonValueIt == jsonFingerprint.end()) MS_THROW_TYPE_ERROR("missing fingerprint.value"); else if (!jsonValueIt->is_string()) MS_THROW_TYPE_ERROR("wrong fingerprint.value (not a string)"); dtlsRemoteFingerprint.value = jsonValueIt->get<std::string>(); // Just use the first fingerprint. break; } auto jsonRoleIt = jsonDtlsParametersIt->find("role"); if (jsonRoleIt != jsonDtlsParametersIt->end()) { if (!jsonRoleIt->is_string()) MS_THROW_TYPE_ERROR("wrong dtlsParameters.role (not a string)"); dtlsRemoteRole = RTC::DtlsTransport::StringToRole(jsonRoleIt->get<std::string>()); if (dtlsRemoteRole == RTC::DtlsTransport::Role::NONE) MS_THROW_TYPE_ERROR("invalid dtlsParameters.role value"); } else { dtlsRemoteRole = RTC::DtlsTransport::Role::AUTO; } // Set local DTLS role. switch (dtlsRemoteRole) { case RTC::DtlsTransport::Role::CLIENT: { this->dtlsRole = RTC::DtlsTransport::Role::SERVER; break; } // If the peer has role "auto" we become "client" since we are ICE controlled. case RTC::DtlsTransport::Role::SERVER: case RTC::DtlsTransport::Role::AUTO: { this->dtlsRole = RTC::DtlsTransport::Role::CLIENT; break; } case RTC::DtlsTransport::Role::NONE: { MS_THROW_TYPE_ERROR("invalid remote DTLS role"); } } this->connectCalled = true; // Pass the remote fingerprint to the DTLS transport. if (this->dtlsTransport->SetRemoteFingerprint(dtlsRemoteFingerprint)) { // If everything is fine, we may run the DTLS transport if ready. MayRunDtlsTransport(); } // Tell the caller about the selected local DTLS role. json data = json::object(); switch (this->dtlsRole) { case RTC::DtlsTransport::Role::CLIENT: data["dtlsLocalRole"] = "client"; break; case RTC::DtlsTransport::Role::SERVER: data["dtlsLocalRole"] = "server"; break; default: MS_ABORT("invalid local DTLS role"); } request->Accept(data); break; } case Channel::ChannelRequest::MethodId::TRANSPORT_RESTART_ICE: { std::string usernameFragment = Utils::Crypto::GetRandomString(16); std::string password = Utils::Crypto::GetRandomString(32); this->iceServer->SetUsernameFragment(usernameFragment); this->iceServer->SetPassword(password); MS_DEBUG_DEV( "WebRtcTransport ICE usernameFragment and password changed [id:%s]", this->id.c_str()); // Reply with the updated ICE local parameters. json data = json::object(); data["iceParameters"] = json::object(); auto jsonIceParametersIt = data.find("iceParameters"); (*jsonIceParametersIt)["usernameFragment"] = this->iceServer->GetUsernameFragment(); (*jsonIceParametersIt)["password"] = this->iceServer->GetPassword(); (*jsonIceParametersIt)["iceLite"] = true; request->Accept(data); break; } default: { // Pass it to the parent class. RTC::Transport::HandleRequest(request); } } }
（三）进入WebRtcTransport.cpp中----使用CreateProducer创建生产者（消费者）
default: //如前面的WebRtcTransport中的是我itch所示，进入default中去，调用父类方法 { // Pass it to the parent class. RTC::Transport::HandleRequest(request); }
进入Transport.cpp文件中去！，创建生产者和消费者（两个方式一致）
void Transport::HandleRequest(Channel::ChannelRequest* request) { MS_TRACE(); switch (request->methodId) { case Channel::ChannelRequest::MethodId::TRANSPORT_PRODUCE: //创建生产者 { std::string producerId; // This may throw. SetNewProducerIdFromInternal(request->internal, producerId); //获取producerId // This may throw. auto* producer = new RTC::Producer(producerId, this, request->data); //根据上面的Id创建producer生产者 // Insert the Producer into the RtpListener. // This may throw. If so, delete the Producer and throw. try { this->rtpListener.AddProducer(producer); //加入数据 } catch (const MediaSoupError& error) { delete producer; throw; } // Notify the listener. // This may throw if a Producer with same id already exists. try { this->listener->OnTransportNewProducer(this, producer); //router中加入生产者 } catch (const MediaSoupError& error) { this->rtpListener.RemoveProducer(producer); delete producer; throw; } // Insert into the map. this->mapProducers[producerId] = producer; MS_DEBUG_DEV("Producer created [producerId:%s]", producerId.c_str()); // Take the transport related RTP header extensions of the Producer and // add them to the Transport. // NOTE: Producer::GetRtpHeaderExtensionIds() returns the original // header extension ids of the Producer (and not their mapped values). const auto& producerRtpHeaderExtensionIds = producer->GetRtpHeaderExtensionIds(); if (producerRtpHeaderExtensionIds.mid != 0u) { this->recvRtpHeaderExtensionIds.mid = producerRtpHeaderExtensionIds.mid; } if (producerRtpHeaderExtensionIds.rid != 0u) { this->recvRtpHeaderExtensionIds.rid = producerRtpHeaderExtensionIds.rid; } if (producerRtpHeaderExtensionIds.rrid != 0u) { this->recvRtpHeaderExtensionIds.rrid = producerRtpHeaderExtensionIds.rrid; } if (producerRtpHeaderExtensionIds.absSendTime != 0u) { this->recvRtpHeaderExtensionIds.absSendTime = producerRtpHeaderExtensionIds.absSendTime; } if (producerRtpHeaderExtensionIds.transportWideCc01 != 0u) { this->recvRtpHeaderExtensionIds.transportWideCc01 = producerRtpHeaderExtensionIds.transportWideCc01; } // Create status response. json data = json::object(); data["type"] = RTC::RtpParameters::GetTypeString(producer->GetType()); request->Accept(data); // Check if TransportCongestionControlServer or REMB server must be // created. const auto& rtpHeaderExtensionIds = producer->GetRtpHeaderExtensionIds(); const auto& codecs = producer->GetRtpParameters().codecs; // Set TransportCongestionControlServer. if (!this->tccServer) { bool createTccServer{ false }; RTC::BweType bweType; // Use transport-cc if: // - there is transport-wide-cc-01 RTP header extension, and // - there is "transport-cc" in codecs RTCP feedback. // // clang-format off if ( rtpHeaderExtensionIds.transportWideCc01 != 0u && std::any_of( codecs.begin(), codecs.end(), [](const RTC::RtpCodecParameters& codec) { return std::any_of( codec.rtcpFeedback.begin(), codec.rtcpFeedback.end(), [](const RTC::RtcpFeedback& fb) { return fb.type == "transport-cc"; }); }) ) // clang-format on { MS_DEBUG_TAG(bwe, "enabling TransportCongestionControlServer with transport-cc"); createTccServer = true; bweType = RTC::BweType::TRANSPORT_CC; } // Use REMB if: // - there is abs-send-time RTP header extension, and // - there is "remb" in codecs RTCP feedback. // // clang-format off else if ( rtpHeaderExtensionIds.absSendTime != 0u && std::any_of( codecs.begin(), codecs.end(), [](const RTC::RtpCodecParameters& codec) { return std::any_of( codec.rtcpFeedback.begin(), codec.rtcpFeedback.end(), [](const RTC::RtcpFeedback& fb) { return fb.type == "goog-remb"; }); }) ) // clang-format on { MS_DEBUG_TAG(bwe, "enabling TransportCongestionControlServer with REMB"); createTccServer = true; bweType = RTC::BweType::REMB; } if (createTccServer) { this->tccServer = new RTC::TransportCongestionControlServer(this, bweType, RTC::MtuSize); if (this->maxIncomingBitrate != 0u) this->tccServer->SetMaxIncomingBitrate(this->maxIncomingBitrate); if (IsConnected()) this->tccServer->TransportConnected(); } } break; }
进入Producer.cpp中去创建Producer生产者：https://mediasoup.org/documentation/v3/mediasoup/api/#Producer
Producer::Producer(const std::string& id, RTC::Producer::Listener* listener, json& data) : id(id), listener(listener) { MS_TRACE(); auto jsonKindIt = data.find("kind"); //对JSON数据进行解析！---kind包含音视频---下面层层解析！！！ if (jsonKindIt == data.end() || !jsonKindIt->is_string()) { MS_THROW_TYPE_ERROR("missing kind"); } // This may throw. this->kind = RTC::Media::GetKind(jsonKindIt->get<std::string>()); if (this->kind == RTC::Media::Kind::ALL) { MS_THROW_TYPE_ERROR("invalid empty kind"); } auto jsonRtpParametersIt = data.find("rtpParameters"); if (jsonRtpParametersIt == data.end() || !jsonRtpParametersIt->is_object()) { MS_THROW_TYPE_ERROR("missing rtpParameters"); } // This may throw. this->rtpParameters = RTC::RtpParameters(*jsonRtpParametersIt); // Evaluate type. this->type = RTC::RtpParameters::GetType(this->rtpParameters); // Reserve a slot in rtpStreamByEncodingIdx and rtpStreamsScores vectors // for each RTP stream. this->rtpStreamByEncodingIdx.resize(this->rtpParameters.encodings.size(), nullptr); this->rtpStreamScores.resize(this->rtpParameters.encodings.size(), 0u); auto& encoding = this->rtpParameters.encodings[0]; auto* mediaCodec = this->rtpParameters.GetCodecForEncoding(encoding); if (!RTC::Codecs::Tools::IsValidTypeForCodec(this->type, mediaCodec->mimeType)) { MS_THROW_TYPE_ERROR( "%s codec not supported for %s", mediaCodec->mimeType.ToString().c_str(), RTC::RtpParameters::GetTypeString(this->type).c_str()); } auto jsonRtpMappingIt = data.find("rtpMapping"); if (jsonRtpMappingIt == data.end() || !jsonRtpMappingIt->is_object()) { MS_THROW_TYPE_ERROR("missing rtpMapping"); } auto jsonCodecsIt = jsonRtpMappingIt->find("codecs"); //获取编码器 if (jsonCodecsIt == jsonRtpMappingIt->end() || !jsonCodecsIt->is_array()) { MS_THROW_TYPE_ERROR("missing rtpMapping.codecs"); } for (auto& codec : *jsonCodecsIt) { if (!codec.is_object()) { MS_THROW_TYPE_ERROR("wrong entry in rtpMapping.codecs (not an object)"); } auto jsonPayloadTypeIt = codec.find("payloadType"); // clang-format off if ( jsonPayloadTypeIt == codec.end() || !Utils::Json::IsPositiveInteger(*jsonPayloadTypeIt) ) // clang-format on { MS_THROW_TYPE_ERROR("wrong entry in rtpMapping.codecs (missing payloadType)"); } auto jsonMappedPayloadTypeIt = codec.find("mappedPayloadType"); // clang-format off if ( jsonMappedPayloadTypeIt == codec.end() || !Utils::Json::IsPositiveInteger(*jsonMappedPayloadTypeIt) ) // clang-format on { MS_THROW_TYPE_ERROR("wrong entry in rtpMapping.codecs (missing mappedPayloadType)"); } this->rtpMapping.codecs[jsonPayloadTypeIt->get<uint8_t>()] = jsonMappedPayloadTypeIt->get<uint8_t>(); } auto jsonEncodingsIt = jsonRtpMappingIt->find("encodings"); if (jsonEncodingsIt == jsonRtpMappingIt->end() || !jsonEncodingsIt->is_array()) { MS_THROW_TYPE_ERROR("missing rtpMapping.encodings"); } this->rtpMapping.encodings.reserve(jsonEncodingsIt->size()); for (auto& encoding : *jsonEncodingsIt) { if (!encoding.is_object()) { MS_THROW_TYPE_ERROR("wrong entry in rtpMapping.encodings"); } this->rtpMapping.encodings.emplace_back(); auto& encodingMapping = this->rtpMapping.encodings.back(); // ssrc is optional. auto jsonSsrcIt = encoding.find("ssrc"); // clang-format off if ( jsonSsrcIt != encoding.end() && Utils::Json::IsPositiveInteger(*jsonSsrcIt) ) // clang-format on { encodingMapping.ssrc = jsonSsrcIt->get<uint32_t>(); } // rid is optional. auto jsonRidIt = encoding.find("rid"); if (jsonRidIt != encoding.end() && jsonRidIt->is_string()) { encodingMapping.rid = jsonRidIt->get<std::string>(); } // However ssrc or rid must be present (if more than 1 encoding). // clang-format off if ( jsonEncodingsIt->size() > 1 && jsonSsrcIt == encoding.end() && jsonRidIt == encoding.end() ) // clang-format on { MS_THROW_TYPE_ERROR("wrong entry in rtpMapping.encodings (missing ssrc or rid)"); } // If there is no mid and a single encoding, ssrc or rid must be present. // clang-format off if ( this->rtpParameters.mid.empty() && jsonEncodingsIt->size() == 1 && jsonSsrcIt == encoding.end() && jsonRidIt == encoding.end() ) // clang-format on { MS_THROW_TYPE_ERROR( "wrong entry in rtpMapping.encodings (missing ssrc or rid, or rtpParameters.mid)"); } // mappedSsrc is mandatory. auto jsonMappedSsrcIt = encoding.find("mappedSsrc"); // clang-format off if ( jsonMappedSsrcIt == encoding.end() || !Utils::Json::IsPositiveInteger(*jsonMappedSsrcIt) ) // clang-format on { MS_THROW_TYPE_ERROR("wrong entry in rtpMapping.encodings (missing mappedSsrc)"); } encodingMapping.mappedSsrc = jsonMappedSsrcIt->get<uint32_t>(); } auto jsonPausedIt = data.find("paused"); if (jsonPausedIt != data.end() && jsonPausedIt->is_boolean()) { this->paused = jsonPausedIt->get<bool>(); } // The number of encodings in rtpParameters must match the number of encodings // in rtpMapping. if (this->rtpParameters.encodings.size() != this->rtpMapping.encodings.size()) { MS_THROW_TYPE_ERROR("rtpParameters.encodings size does not match rtpMapping.encodings size"); } // Fill RTP header extension ids. // This may throw. for (auto& exten : this->rtpParameters.headerExtensions) { if (exten.id == 0u) { MS_THROW_TYPE_ERROR("RTP extension id cannot be 0"); } if (this->rtpHeaderExtensionIds.mid == 0u && exten.type == RTC::RtpHeaderExtensionUri::Type::MID) { this->rtpHeaderExtensionIds.mid = exten.id; } if (this->rtpHeaderExtensionIds.rid == 0u && exten.type == RTC::RtpHeaderExtensionUri::Type::RTP_STREAM_ID) { this->rtpHeaderExtensionIds.rid = exten.id; } if (this->rtpHeaderExtensionIds.rrid == 0u && exten.type == RTC::RtpHeaderExtensionUri::Type::REPAIRED_RTP_STREAM_ID) { this->rtpHeaderExtensionIds.rrid = exten.id; } if (this->rtpHeaderExtensionIds.absSendTime == 0u && exten.type == RTC::RtpHeaderExtensionUri::Type::ABS_SEND_TIME) { this->rtpHeaderExtensionIds.absSendTime = exten.id; } if (this->rtpHeaderExtensionIds.transportWideCc01 == 0u && exten.type == RTC::RtpHeaderExtensionUri::Type::TRANSPORT_WIDE_CC_01) { this->rtpHeaderExtensionIds.transportWideCc01 = exten.id; } // NOTE: Remove this once framemarking draft becomes RFC. if (this->rtpHeaderExtensionIds.frameMarking07 == 0u && exten.type == RTC::RtpHeaderExtensionUri::Type::FRAME_MARKING_07) { this->rtpHeaderExtensionIds.frameMarking07 = exten.id; } if (this->rtpHeaderExtensionIds.frameMarking == 0u && exten.type == RTC::RtpHeaderExtensionUri::Type::FRAME_MARKING) { this->rtpHeaderExtensionIds.frameMarking = exten.id; } if (this->rtpHeaderExtensionIds.ssrcAudioLevel == 0u && exten.type == RTC::RtpHeaderExtensionUri::Type::SSRC_AUDIO_LEVEL) { this->rtpHeaderExtensionIds.ssrcAudioLevel = exten.id; } if (this->rtpHeaderExtensionIds.videoOrientation == 0u && exten.type == RTC::RtpHeaderExtensionUri::Type::VIDEO_ORIENTATION) { this->rtpHeaderExtensionIds.videoOrientation = exten.id; } if (this->rtpHeaderExtensionIds.toffset == 0u && exten.type == RTC::RtpHeaderExtensionUri::Type::TOFFSET) { this->rtpHeaderExtensionIds.toffset = exten.id; } } // Set the RTCP report generation interval. if (this->kind == RTC::Media::Kind::AUDIO) this->maxRtcpInterval = RTC::RTCP::MaxAudioIntervalMs; else this->maxRtcpInterval = RTC::RTCP::MaxVideoIntervalMs; // Create a KeyFrameRequestManager. if (this->kind == RTC::Media::Kind::VIDEO) { auto jsonKeyFrameRequestDelayIt = data.find("keyFrameRequestDelay"); uint32_t keyFrameRequestDelay = 0u; // clang-format off if ( jsonKeyFrameRequestDelayIt != data.end() && jsonKeyFrameRequestDelayIt->is_number_integer() ) // clang-format on { keyFrameRequestDelay = jsonKeyFrameRequestDelayIt->get<uint32_t>(); } this->keyFrameRequestManager = new RTC::KeyFrameRequestManager(this, keyFrameRequestDelay); } }
进入router.cpp中查看添加生产者方法
inline void Router::OnTransportNewProducer(RTC::Transport* /*transport*/, RTC::Producer* producer) { MS_TRACE(); MS_ASSERT( this->mapProducerConsumers.find(producer) == this->mapProducerConsumers.end(), "Producer already present in mapProducerConsumers"); if (this->mapProducers.find(producer->id) != this->mapProducers.end()) { MS_THROW_ERROR("Producer already present in mapProducers [producerId:%s]", producer->id.c_str()); } // Insert the Producer in the maps. this->mapProducers[producer->id] = producer; //添加到map中去！ this->mapProducerConsumers[producer]; this->mapProducerRtpObservers[producer]; }
（四）进入Transport.cpp中----查看TRANSPORT_CONSUME创建消费者
case Channel::ChannelRequest::MethodId::TRANSPORT_CONSUME: { auto jsonProducerIdIt = request->internal.find("producerId"); //获取生产者id if (jsonProducerIdIt == request->internal.end() || !jsonProducerIdIt->is_string()) MS_THROW_ERROR("missing internal.producerId"); std::string producerId = jsonProducerIdIt->get<std::string>(); std::string consumerId; // This may throw. SetNewConsumerIdFromInternal(request->internal, consumerId); //获取消费者id // Get type. auto jsonTypeIt = request->data.find("type"); //获取消费者类型 if (jsonTypeIt == request->data.end() || !jsonTypeIt->is_string()) MS_THROW_TYPE_ERROR("missing type"); // This may throw. auto type = RTC::RtpParameters::GetType(jsonTypeIt->get<std::string>()); RTC::Consumer* consumer{ nullptr }; switch (type) //多种consumer { case RTC::RtpParameters::Type::NONE: { MS_THROW_TYPE_ERROR("invalid type 'none'"); break; } case RTC::RtpParameters::Type::SIMPLE: //普通音视频 { consumer = new RTC::SimpleConsumer(consumerId, producerId, this, request->data); break; } case RTC::RtpParameters::Type::SIMULCAST: consumer = new RTC::SimulcastConsumer(consumerId, producerId, this, request->data); break; case RTC::RtpParameters::Type::SVC: consumer = new RTC::SvcConsumer(consumerId, producerId, this, request->data); break; case RTC::RtpParameters::Type::PIPE: consumer = new RTC::PipeConsumer(consumerId, producerId, this, request->data); break; } // Notify the listener. // This may throw if no Producer is found. this->listener->OnTransportNewConsumer(this, consumer, producerId); // Insert into the maps. this->mapConsumers[consumerId] = consumer; for (auto ssrc : consumer->GetMediaSsrcs()) { this->mapSsrcConsumer[ssrc] = consumer; } for (auto ssrc : consumer->GetRtxSsrcs()) { this->mapRtxSsrcConsumer[ssrc] = consumer; } // Create status response. json data = json::object(); data["paused"] = consumer->IsPaused(); data["producerPaused"] = consumer->IsProducerPaused(); consumer->FillJsonScore(data["score"]); auto preferredLayers = consumer->GetPreferredLayers(); if (preferredLayers.spatial > -1 && preferredLayers.temporal > -1) { data["preferredLayers"]["spatialLayer"] = preferredLayers.spatial; data["preferredLayers"]["temporalLayer"] = preferredLayers.temporal; } request->Accept(data); // Check if Transport Congestion Control client must be created. const auto& rtpHeaderExtensionIds = consumer->GetRtpHeaderExtensionIds(); const auto& codecs = consumer->GetRtpParameters().codecs; // Set TransportCongestionControlClient. if (!this->tccClient) { bool createTccClient{ false }; RTC::BweType bweType; // Use transport-cc if: // - it's a video Consumer, and // - there is transport-wide-cc-01 RTP header extension, and // - there is "transport-cc" in codecs RTCP feedback. // // clang-format off if ( consumer->GetKind() == RTC::Media::Kind::VIDEO && rtpHeaderExtensionIds.transportWideCc01 != 0u && std::any_of( codecs.begin(), codecs.end(), [](const RTC::RtpCodecParameters& codec) { return std::any_of( codec.rtcpFeedback.begin(), codec.rtcpFeedback.end(), [](const RTC::RtcpFeedback& fb) { return fb.type == "transport-cc"; }); }) ) // clang-format on { MS_DEBUG_TAG(bwe, "enabling TransportCongestionControlClient with transport-cc"); createTccClient = true; bweType = RTC::BweType::TRANSPORT_CC; } // Use REMB if: // - it's a video Consumer, and // - there is abs-send-time RTP header extension, and // - there is "remb" in codecs RTCP feedback. // // clang-format off else if ( consumer->GetKind() == RTC::Media::Kind::VIDEO && rtpHeaderExtensionIds.absSendTime != 0u && std::any_of( codecs.begin(), codecs.end(), [](const RTC::RtpCodecParameters& codec) { return std::any_of( codec.rtcpFeedback.begin(), codec.rtcpFeedback.end(), [](const RTC::RtcpFeedback& fb) { return fb.type == "goog-remb"; }); }) ) // clang-format on { MS_DEBUG_TAG(bwe, "enabling TransportCongestionControlClient with REMB"); createTccClient = true; bweType = RTC::BweType::REMB; } if (createTccClient) { // Tell all the Consumers that we are gonna manage their bitrate. for (auto& kv : this->mapConsumers) { auto* consumer = kv.second; consumer->SetExternallyManagedBitrate(); }; this->tccClient = new RTC::TransportCongestionControlClient( this, bweType, this->initialAvailableOutgoingBitrate, this->maxOutgoingBitrate); if (IsConnected()) this->tccClient->TransportConnected(); } } // If applicable, tell the new Consumer that we are gonna manage its // bitrate. if (this->tccClient) consumer->SetExternallyManagedBitrate(); #ifdef ENABLE_RTC_SENDER_BANDWIDTH_ESTIMATOR // Create SenderBandwidthEstimator if: // - not already created, // - it's a video Consumer, and // - there is transport-wide-cc-01 RTP header extension, and // - there is "transport-cc" in codecs RTCP feedback. // // clang-format off if ( !this->senderBwe && consumer->GetKind() == RTC::Media::Kind::VIDEO && rtpHeaderExtensionIds.transportWideCc01 != 0u && std::any_of( codecs.begin(), codecs.end(), [](const RTC::RtpCodecParameters& codec) { return std::any_of( codec.rtcpFeedback.begin(), codec.rtcpFeedback.end(), [](const RTC::RtcpFeedback& fb) { return fb.type == "transport-cc"; }); }) ) // clang-format on { MS_DEBUG_TAG(bwe, "enabling SenderBandwidthEstimator"); // Tell all the Consumers that we are gonna manage their bitrate. for (auto& kv : this->mapConsumers) { auto* consumer = kv.second; consumer->SetExternallyManagedBitrate(); }; this->senderBwe = new RTC::SenderBandwidthEstimator(this, this->initialAvailableOutgoingBitrate); if (IsConnected()) this->senderBwe->TransportConnected(); } // If applicable, tell the new Consumer that we are gonna manage its // bitrate. if (this->senderBwe) consumer->SetExternallyManagedBitrate(); #endif if (IsConnected()) consumer->TransportConnected(); break; }
进入SimpleConsumer.cpp文件中去，查看创建SimpleConsumer构造函数
SimpleConsumer::SimpleConsumer( const std::string& id, const std::string& producerId, RTC::Consumer::Listener* listener, json& data) : RTC::Consumer::Consumer(id, producerId, listener, data, RTC::RtpParameters::Type::SIMPLE) { MS_TRACE(); // Ensure there is a single encoding. if (this->consumableRtpEncodings.size() != 1u) MS_THROW_TYPE_ERROR("invalid consumableRtpEncodings with size != 1"); auto& encoding = this->rtpParameters.encodings[0]; const auto* mediaCodec = this->rtpParameters.GetCodecForEncoding(encoding); this->keyFrameSupported = RTC::Codecs::Tools::CanBeKeyFrame(mediaCodec->mimeType); // Create RtpStreamSend instance for sending a single stream to the remote. CreateRtpStream(); //创建RTP流 }
void SimpleConsumer::CreateRtpStream() { MS_TRACE(); auto& encoding = this->rtpParameters.encodings[0]; const auto* mediaCodec = this->rtpParameters.GetCodecForEncoding(encoding); MS_DEBUG_TAG( rtp, "[ssrc:%" PRIu32 ", payloadType:%" PRIu8 "]", encoding.ssrc, mediaCodec->payloadType); // Set stream params. RTC::RtpStream::Params params; params.ssrc = encoding.ssrc; params.payloadType = mediaCodec->payloadType; params.mimeType = mediaCodec->mimeType; params.clockRate = mediaCodec->clockRate; params.cname = this->rtpParameters.rtcp.cname; // Check in band FEC in codec parameters. if (mediaCodec->parameters.HasInteger("useinbandfec") && mediaCodec->parameters.GetInteger("useinbandfec") == 1) { MS_DEBUG_TAG(rtp, "in band FEC enabled"); params.useInBandFec = true; } // Check DTX in codec parameters. if (mediaCodec->parameters.HasInteger("usedtx") && mediaCodec->parameters.GetInteger("usedtx") == 1) { MS_DEBUG_TAG(rtp, "DTX enabled"); params.useDtx = true; } // Check DTX in the encoding. if (encoding.dtx) { MS_DEBUG_TAG(rtp, "DTX enabled"); params.useDtx = true; } for (const auto& fb : mediaCodec->rtcpFeedback) { if (!params.useNack && fb.type == "nack" && fb.parameter.empty()) { MS_DEBUG_2TAGS(rtp, rtcp, "NACK supported"); params.useNack = true; } else if (!params.usePli && fb.type == "nack" && fb.parameter == "pli") { MS_DEBUG_2TAGS(rtp, rtcp, "PLI supported"); params.usePli = true; } else if (!params.useFir && fb.type == "ccm" && fb.parameter == "fir") { MS_DEBUG_2TAGS(rtp, rtcp, "FIR supported"); params.useFir = true; } } // Create a RtpStreamSend for sending a single media stream. size_t bufferSize = params.useNack ? 600u : 0u; this->rtpStream = new RTC::RtpStreamSend(this, params, bufferSize); //创建RTPStreamSend发送给远端客户端 this->rtpStreams.push_back(this->rtpStream); //保存在数据结构中去！ // If the Consumer is paused, tell the RtpStreamSend. if (IsPaused() || IsProducerPaused()) this->rtpStream->Pause(); const auto* rtxCodec = this->rtpParameters.GetRtxCodecForEncoding(encoding); if (rtxCodec && encoding.hasRtx) this->rtpStream->SetRtx(rtxCodec->payloadType, encoding.rtx.ssrc); }
创建Consumer时，会创建一个RtpStreamSend对象，为后面向客户端发送数据做准备。
当共享者的数据传输上来，服务端通过ssrc找到对应的生产者，通过生产者获取消费者，最后通过RtpStreamSend对象发送给最终的消费者！！
