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diff --git a/third_party/libwebrtc/rtc_tools/data_channel_benchmark/grpc_signaling.cc b/third_party/libwebrtc/rtc_tools/data_channel_benchmark/grpc_signaling.cc
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+++ b/third_party/libwebrtc/rtc_tools/data_channel_benchmark/grpc_signaling.cc
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+/*
+ * 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