| /* |
| * Copyright (c) 2021 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| #include "rtc_tools/data_channel_benchmark/grpc_signaling.h" |
| |
| #include <grpc/support/log.h> |
| #include <grpcpp/grpcpp.h> |
| |
| #include <string> |
| #include <utility> |
| |
| #include "api/jsep.h" |
| #include "api/jsep_ice_candidate.h" |
| #include "rtc_base/thread.h" |
| #include "rtc_tools/data_channel_benchmark/peer_connection_signaling.grpc.pb.h" |
| |
| namespace webrtc { |
| namespace { |
| |
| using GrpcSignaling::IceCandidate; |
| using GrpcSignaling::PeerConnectionSignaling; |
| using GrpcSignaling::SessionDescription; |
| using GrpcSignaling::SignalingMessage; |
| |
| template <class T> |
| class SessionData : public webrtc::SignalingInterface { |
| public: |
| SessionData() {} |
| explicit SessionData(T* stream) : stream_(stream) {} |
| void SetStream(T* stream) { stream_ = stream; } |
| |
| void SendIceCandidate(const IceCandidateInterface* candidate) override { |
| RTC_LOG(LS_INFO) << "SendIceCandidate"; |
| std::string serialized_candidate; |
| if (!candidate->ToString(&serialized_candidate)) { |
| RTC_LOG(LS_ERROR) << "Failed to serialize ICE candidate"; |
| return; |
| } |
| |
| SignalingMessage message; |
| IceCandidate* proto_candidate = message.mutable_candidate(); |
| proto_candidate->set_description(serialized_candidate); |
| proto_candidate->set_mid(candidate->sdp_mid()); |
| proto_candidate->set_mline_index(candidate->sdp_mline_index()); |
| |
| stream_->Write(message); |
| } |
| |
| void SendDescription(const SessionDescriptionInterface* sdp) override { |
| RTC_LOG(LS_INFO) << "SendDescription"; |
| |
| std::string serialized_sdp; |
| sdp->ToString(&serialized_sdp); |
| |
| SignalingMessage message; |
| if (sdp->GetType() == SdpType::kOffer) |
| message.mutable_description()->set_type(SessionDescription::OFFER); |
| else if (sdp->GetType() == SdpType::kAnswer) |
| message.mutable_description()->set_type(SessionDescription::ANSWER); |
| message.mutable_description()->set_content(serialized_sdp); |
| |
| stream_->Write(message); |
| } |
| |
| void OnRemoteDescription( |
| std::function<void(std::unique_ptr<SessionDescriptionInterface> sdp)> |
| callback) override { |
| RTC_LOG(LS_INFO) << "OnRemoteDescription"; |
| remote_description_callback_ = callback; |
| } |
| |
| void OnIceCandidate( |
| std::function<void(std::unique_ptr<IceCandidateInterface> candidate)> |
| callback) override { |
| RTC_LOG(LS_INFO) << "OnIceCandidate"; |
| ice_candidate_callback_ = callback; |
| } |
| |
| T* stream_; |
| |
| std::function<void(std::unique_ptr<webrtc::IceCandidateInterface>)> |
| ice_candidate_callback_; |
| std::function<void(std::unique_ptr<webrtc::SessionDescriptionInterface>)> |
| remote_description_callback_; |
| }; |
| |
| using ServerSessionData = |
| SessionData<grpc::ServerReaderWriter<SignalingMessage, SignalingMessage>>; |
| using ClientSessionData = |
| SessionData<grpc::ClientReaderWriter<SignalingMessage, SignalingMessage>>; |
| |
| template <class MessageType, class StreamReader, class SessionData> |
| void ProcessMessages(StreamReader* stream, SessionData* session) { |
| MessageType message; |
| |
| while (stream->Read(&message)) { |
| switch (message.Content_case()) { |
| case SignalingMessage::ContentCase::kCandidate: { |
| webrtc::SdpParseError error; |
| auto jsep_candidate = std::make_unique<webrtc::JsepIceCandidate>( |
| message.candidate().mid(), message.candidate().mline_index()); |
| if (!jsep_candidate->Initialize(message.candidate().description(), |
| &error)) { |
| RTC_LOG(LS_ERROR) << "Failed to deserialize ICE candidate '" |
| << message.candidate().description() << "'"; |
| RTC_LOG(LS_ERROR) |
| << "Error at line " << error.line << ":" << error.description; |
| continue; |
| } |
| |
| session->ice_candidate_callback_(std::move(jsep_candidate)); |
| break; |
| } |
| case SignalingMessage::ContentCase::kDescription: { |
| auto& description = message.description(); |
| auto content = description.content(); |
| |
| auto sdp = webrtc::CreateSessionDescription( |
| description.type() == SessionDescription::OFFER |
| ? webrtc::SdpType::kOffer |
| : webrtc::SdpType::kAnswer, |
| description.content()); |
| session->remote_description_callback_(std::move(sdp)); |
| break; |
| } |
| default: |
| RTC_DCHECK_NOTREACHED(); |
| } |
| } |
| } |
| |
| class GrpcNegotiationServer : public GrpcSignalingServerInterface, |
| public PeerConnectionSignaling::Service { |
| public: |
| GrpcNegotiationServer( |
| std::function<void(webrtc::SignalingInterface*)> callback, |
| int port, |
| bool oneshot) |
| : connect_callback_(std::move(callback)), |
| requested_port_(port), |
| oneshot_(oneshot) {} |
| ~GrpcNegotiationServer() override { |
| Stop(); |
| if (server_stop_thread_) |
| server_stop_thread_->Stop(); |
| } |
| |
| void Start() override { |
| std::string server_address = "[::]"; |
| |
| grpc::ServerBuilder builder; |
| builder.AddListeningPort( |
| server_address + ":" + std::to_string(requested_port_), |
| grpc::InsecureServerCredentials(), &selected_port_); |
| builder.RegisterService(this); |
| server_ = builder.BuildAndStart(); |
| } |
| |
| void Wait() override { server_->Wait(); } |
| |
| void Stop() override { server_->Shutdown(); } |
| |
| int SelectedPort() override { return selected_port_; } |
| |
| grpc::Status Connect( |
| grpc::ServerContext* context, |
| grpc::ServerReaderWriter<SignalingMessage, SignalingMessage>* stream) |
| override { |
| if (oneshot_) { |
| // Request the termination of the server early so we don't serve another |
| // client in parallel. |
| server_stop_thread_ = rtc::Thread::Create(); |
| server_stop_thread_->Start(); |
| server_stop_thread_->PostTask([this] { Stop(); }); |
| } |
| |
| ServerSessionData session(stream); |
| |
| auto reading_thread = rtc::Thread::Create(); |
| reading_thread->Start(); |
| reading_thread->PostTask([&session, &stream] { |
| ProcessMessages<SignalingMessage>(stream, &session); |
| }); |
| |
| connect_callback_(&session); |
| |
| reading_thread->Stop(); |
| |
| return grpc::Status::OK; |
| } |
| |
| private: |
| std::function<void(webrtc::SignalingInterface*)> connect_callback_; |
| int requested_port_; |
| int selected_port_; |
| bool oneshot_; |
| |
| std::unique_ptr<grpc::Server> server_; |
| std::unique_ptr<rtc::Thread> server_stop_thread_; |
| }; |
| |
| class GrpcNegotiationClient : public GrpcSignalingClientInterface { |
| public: |
| explicit GrpcNegotiationClient(const std::string& server) { |
| channel_ = grpc::CreateChannel(server, grpc::InsecureChannelCredentials()); |
| stub_ = PeerConnectionSignaling::NewStub(channel_); |
| } |
| |
| ~GrpcNegotiationClient() override { |
| context_.TryCancel(); |
| if (reading_thread_) |
| reading_thread_->Stop(); |
| } |
| |
| bool Start() override { |
| if (!channel_->WaitForConnected( |
| absl::ToChronoTime(absl::Now() + absl::Seconds(3)))) { |
| return false; |
| } |
| |
| stream_ = stub_->Connect(&context_); |
| session_.SetStream(stream_.get()); |
| |
| reading_thread_ = rtc::Thread::Create(); |
| reading_thread_->Start(); |
| reading_thread_->PostTask([this] { |
| ProcessMessages<SignalingMessage>(stream_.get(), &session_); |
| }); |
| |
| return true; |
| } |
| |
| webrtc::SignalingInterface* signaling_client() override { return &session_; } |
| |
| private: |
| std::shared_ptr<grpc::Channel> channel_; |
| std::unique_ptr<PeerConnectionSignaling::Stub> stub_; |
| std::unique_ptr<rtc::Thread> reading_thread_; |
| grpc::ClientContext context_; |
| std::unique_ptr< |
| ::grpc::ClientReaderWriter<SignalingMessage, SignalingMessage>> |
| stream_; |
| ClientSessionData session_; |
| }; |
| } // namespace |
| |
| std::unique_ptr<GrpcSignalingServerInterface> |
| GrpcSignalingServerInterface::Create( |
| std::function<void(webrtc::SignalingInterface*)> callback, |
| int port, |
| bool oneshot) { |
| return std::make_unique<GrpcNegotiationServer>(std::move(callback), port, |
| oneshot); |
| } |
| |
| std::unique_ptr<GrpcSignalingClientInterface> |
| GrpcSignalingClientInterface::Create(const std::string& server) { |
| return std::make_unique<GrpcNegotiationClient>(server); |
| } |
| |
| } // namespace webrtc |