| /* |
| * Copyright 2012 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 "examples/peerconnection/client/conductor.h" |
| |
| #include <stddef.h> |
| #include <stdint.h> |
| |
| #include <memory> |
| #include <utility> |
| #include <vector> |
| |
| #include "absl/memory/memory.h" |
| #include "absl/types/optional.h" |
| #include "api/audio/audio_device.h" |
| #include "api/audio/audio_mixer.h" |
| #include "api/audio/audio_processing.h" |
| #include "api/audio_codecs/audio_decoder_factory.h" |
| #include "api/audio_codecs/audio_encoder_factory.h" |
| #include "api/audio_codecs/builtin_audio_decoder_factory.h" |
| #include "api/audio_codecs/builtin_audio_encoder_factory.h" |
| #include "api/audio_options.h" |
| #include "api/create_peerconnection_factory.h" |
| #include "api/rtp_sender_interface.h" |
| #include "api/video_codecs/video_decoder_factory.h" |
| #include "api/video_codecs/video_decoder_factory_template.h" |
| #include "api/video_codecs/video_decoder_factory_template_dav1d_adapter.h" |
| #include "api/video_codecs/video_decoder_factory_template_libvpx_vp8_adapter.h" |
| #include "api/video_codecs/video_decoder_factory_template_libvpx_vp9_adapter.h" |
| #include "api/video_codecs/video_decoder_factory_template_open_h264_adapter.h" |
| #include "api/video_codecs/video_encoder_factory.h" |
| #include "api/video_codecs/video_encoder_factory_template.h" |
| #include "api/video_codecs/video_encoder_factory_template_libaom_av1_adapter.h" |
| #include "api/video_codecs/video_encoder_factory_template_libvpx_vp8_adapter.h" |
| #include "api/video_codecs/video_encoder_factory_template_libvpx_vp9_adapter.h" |
| #include "api/video_codecs/video_encoder_factory_template_open_h264_adapter.h" |
| #include "examples/peerconnection/client/defaults.h" |
| #include "modules/video_capture/video_capture.h" |
| #include "modules/video_capture/video_capture_factory.h" |
| #include "p2p/base/port_allocator.h" |
| #include "pc/video_track_source.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/logging.h" |
| #include "rtc_base/rtc_certificate_generator.h" |
| #include "rtc_base/strings/json.h" |
| #include "test/vcm_capturer.h" |
| |
| namespace { |
| // Names used for a IceCandidate JSON object. |
| const char kCandidateSdpMidName[] = "sdpMid"; |
| const char kCandidateSdpMlineIndexName[] = "sdpMLineIndex"; |
| const char kCandidateSdpName[] = "candidate"; |
| |
| // Names used for a SessionDescription JSON object. |
| const char kSessionDescriptionTypeName[] = "type"; |
| const char kSessionDescriptionSdpName[] = "sdp"; |
| |
| class DummySetSessionDescriptionObserver |
| : public webrtc::SetSessionDescriptionObserver { |
| public: |
| static rtc::scoped_refptr<DummySetSessionDescriptionObserver> Create() { |
| return rtc::make_ref_counted<DummySetSessionDescriptionObserver>(); |
| } |
| virtual void OnSuccess() { RTC_LOG(LS_INFO) << __FUNCTION__; } |
| virtual void OnFailure(webrtc::RTCError error) { |
| RTC_LOG(LS_INFO) << __FUNCTION__ << " " << ToString(error.type()) << ": " |
| << error.message(); |
| } |
| }; |
| |
| class CapturerTrackSource : public webrtc::VideoTrackSource { |
| public: |
| static rtc::scoped_refptr<CapturerTrackSource> Create() { |
| const size_t kWidth = 640; |
| const size_t kHeight = 480; |
| const size_t kFps = 30; |
| std::unique_ptr<webrtc::test::VcmCapturer> capturer; |
| std::unique_ptr<webrtc::VideoCaptureModule::DeviceInfo> info( |
| webrtc::VideoCaptureFactory::CreateDeviceInfo()); |
| if (!info) { |
| return nullptr; |
| } |
| int num_devices = info->NumberOfDevices(); |
| for (int i = 0; i < num_devices; ++i) { |
| capturer = absl::WrapUnique( |
| webrtc::test::VcmCapturer::Create(kWidth, kHeight, kFps, i)); |
| if (capturer) { |
| return rtc::make_ref_counted<CapturerTrackSource>(std::move(capturer)); |
| } |
| } |
| |
| return nullptr; |
| } |
| |
| protected: |
| explicit CapturerTrackSource( |
| std::unique_ptr<webrtc::test::VcmCapturer> capturer) |
| : VideoTrackSource(/*remote=*/false), capturer_(std::move(capturer)) {} |
| |
| private: |
| rtc::VideoSourceInterface<webrtc::VideoFrame>* source() override { |
| return capturer_.get(); |
| } |
| std::unique_ptr<webrtc::test::VcmCapturer> capturer_; |
| }; |
| |
| } // namespace |
| |
| Conductor::Conductor(PeerConnectionClient* client, MainWindow* main_wnd) |
| : peer_id_(-1), loopback_(false), client_(client), main_wnd_(main_wnd) { |
| client_->RegisterObserver(this); |
| main_wnd->RegisterObserver(this); |
| } |
| |
| Conductor::~Conductor() { |
| RTC_DCHECK(!peer_connection_); |
| } |
| |
| bool Conductor::connection_active() const { |
| return peer_connection_ != nullptr; |
| } |
| |
| void Conductor::Close() { |
| client_->SignOut(); |
| DeletePeerConnection(); |
| } |
| |
| bool Conductor::InitializePeerConnection() { |
| RTC_DCHECK(!peer_connection_factory_); |
| RTC_DCHECK(!peer_connection_); |
| |
| if (!signaling_thread_.get()) { |
| signaling_thread_ = rtc::Thread::CreateWithSocketServer(); |
| signaling_thread_->Start(); |
| } |
| peer_connection_factory_ = webrtc::CreatePeerConnectionFactory( |
| nullptr /* network_thread */, nullptr /* worker_thread */, |
| signaling_thread_.get(), nullptr /* default_adm */, |
| webrtc::CreateBuiltinAudioEncoderFactory(), |
| webrtc::CreateBuiltinAudioDecoderFactory(), |
| std::make_unique<webrtc::VideoEncoderFactoryTemplate< |
| webrtc::LibvpxVp8EncoderTemplateAdapter, |
| webrtc::LibvpxVp9EncoderTemplateAdapter, |
| webrtc::OpenH264EncoderTemplateAdapter, |
| webrtc::LibaomAv1EncoderTemplateAdapter>>(), |
| std::make_unique<webrtc::VideoDecoderFactoryTemplate< |
| webrtc::LibvpxVp8DecoderTemplateAdapter, |
| webrtc::LibvpxVp9DecoderTemplateAdapter, |
| webrtc::OpenH264DecoderTemplateAdapter, |
| webrtc::Dav1dDecoderTemplateAdapter>>(), |
| nullptr /* audio_mixer */, nullptr /* audio_processing */); |
| |
| if (!peer_connection_factory_) { |
| main_wnd_->MessageBox("Error", "Failed to initialize PeerConnectionFactory", |
| true); |
| DeletePeerConnection(); |
| return false; |
| } |
| |
| if (!CreatePeerConnection()) { |
| main_wnd_->MessageBox("Error", "CreatePeerConnection failed", true); |
| DeletePeerConnection(); |
| } |
| |
| AddTracks(); |
| |
| return peer_connection_ != nullptr; |
| } |
| |
| bool Conductor::ReinitializePeerConnectionForLoopback() { |
| loopback_ = true; |
| std::vector<rtc::scoped_refptr<webrtc::RtpSenderInterface>> senders = |
| peer_connection_->GetSenders(); |
| peer_connection_ = nullptr; |
| // Loopback is only possible if encryption is disabled. |
| webrtc::PeerConnectionFactoryInterface::Options options; |
| options.disable_encryption = true; |
| peer_connection_factory_->SetOptions(options); |
| if (CreatePeerConnection()) { |
| for (const auto& sender : senders) { |
| peer_connection_->AddTrack(sender->track(), sender->stream_ids()); |
| } |
| peer_connection_->CreateOffer( |
| this, webrtc::PeerConnectionInterface::RTCOfferAnswerOptions()); |
| } |
| options.disable_encryption = false; |
| peer_connection_factory_->SetOptions(options); |
| return peer_connection_ != nullptr; |
| } |
| |
| bool Conductor::CreatePeerConnection() { |
| RTC_DCHECK(peer_connection_factory_); |
| RTC_DCHECK(!peer_connection_); |
| |
| webrtc::PeerConnectionInterface::RTCConfiguration config; |
| config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan; |
| webrtc::PeerConnectionInterface::IceServer server; |
| server.uri = GetPeerConnectionString(); |
| config.servers.push_back(server); |
| |
| webrtc::PeerConnectionDependencies pc_dependencies(this); |
| auto error_or_peer_connection = |
| peer_connection_factory_->CreatePeerConnectionOrError( |
| config, std::move(pc_dependencies)); |
| if (error_or_peer_connection.ok()) { |
| peer_connection_ = std::move(error_or_peer_connection.value()); |
| } |
| return peer_connection_ != nullptr; |
| } |
| |
| void Conductor::DeletePeerConnection() { |
| main_wnd_->StopLocalRenderer(); |
| main_wnd_->StopRemoteRenderer(); |
| peer_connection_ = nullptr; |
| peer_connection_factory_ = nullptr; |
| peer_id_ = -1; |
| loopback_ = false; |
| } |
| |
| void Conductor::EnsureStreamingUI() { |
| RTC_DCHECK(peer_connection_); |
| if (main_wnd_->IsWindow()) { |
| if (main_wnd_->current_ui() != MainWindow::STREAMING) |
| main_wnd_->SwitchToStreamingUI(); |
| } |
| } |
| |
| // |
| // PeerConnectionObserver implementation. |
| // |
| |
| void Conductor::OnAddTrack( |
| rtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver, |
| const std::vector<rtc::scoped_refptr<webrtc::MediaStreamInterface>>& |
| streams) { |
| RTC_LOG(LS_INFO) << __FUNCTION__ << " " << receiver->id(); |
| main_wnd_->QueueUIThreadCallback(NEW_TRACK_ADDED, |
| receiver->track().release()); |
| } |
| |
| void Conductor::OnRemoveTrack( |
| rtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver) { |
| RTC_LOG(LS_INFO) << __FUNCTION__ << " " << receiver->id(); |
| main_wnd_->QueueUIThreadCallback(TRACK_REMOVED, receiver->track().release()); |
| } |
| |
| void Conductor::OnIceCandidate(const webrtc::IceCandidateInterface* candidate) { |
| RTC_LOG(LS_INFO) << __FUNCTION__ << " " << candidate->sdp_mline_index(); |
| // For loopback test. To save some connecting delay. |
| if (loopback_) { |
| if (!peer_connection_->AddIceCandidate(candidate)) { |
| RTC_LOG(LS_WARNING) << "Failed to apply the received candidate"; |
| } |
| return; |
| } |
| |
| Json::Value jmessage; |
| jmessage[kCandidateSdpMidName] = candidate->sdp_mid(); |
| jmessage[kCandidateSdpMlineIndexName] = candidate->sdp_mline_index(); |
| std::string sdp; |
| if (!candidate->ToString(&sdp)) { |
| RTC_LOG(LS_ERROR) << "Failed to serialize candidate"; |
| return; |
| } |
| jmessage[kCandidateSdpName] = sdp; |
| |
| Json::StreamWriterBuilder factory; |
| SendMessage(Json::writeString(factory, jmessage)); |
| } |
| |
| // |
| // PeerConnectionClientObserver implementation. |
| // |
| |
| void Conductor::OnSignedIn() { |
| RTC_LOG(LS_INFO) << __FUNCTION__; |
| main_wnd_->SwitchToPeerList(client_->peers()); |
| } |
| |
| void Conductor::OnDisconnected() { |
| RTC_LOG(LS_INFO) << __FUNCTION__; |
| |
| DeletePeerConnection(); |
| |
| if (main_wnd_->IsWindow()) |
| main_wnd_->SwitchToConnectUI(); |
| } |
| |
| void Conductor::OnPeerConnected(int id, const std::string& name) { |
| RTC_LOG(LS_INFO) << __FUNCTION__; |
| // Refresh the list if we're showing it. |
| if (main_wnd_->current_ui() == MainWindow::LIST_PEERS) |
| main_wnd_->SwitchToPeerList(client_->peers()); |
| } |
| |
| void Conductor::OnPeerDisconnected(int id) { |
| RTC_LOG(LS_INFO) << __FUNCTION__; |
| if (id == peer_id_) { |
| RTC_LOG(LS_INFO) << "Our peer disconnected"; |
| main_wnd_->QueueUIThreadCallback(PEER_CONNECTION_CLOSED, NULL); |
| } else { |
| // Refresh the list if we're showing it. |
| if (main_wnd_->current_ui() == MainWindow::LIST_PEERS) |
| main_wnd_->SwitchToPeerList(client_->peers()); |
| } |
| } |
| |
| void Conductor::OnMessageFromPeer(int peer_id, const std::string& message) { |
| RTC_DCHECK(peer_id_ == peer_id || peer_id_ == -1); |
| RTC_DCHECK(!message.empty()); |
| |
| if (!peer_connection_.get()) { |
| RTC_DCHECK(peer_id_ == -1); |
| peer_id_ = peer_id; |
| |
| if (!InitializePeerConnection()) { |
| RTC_LOG(LS_ERROR) << "Failed to initialize our PeerConnection instance"; |
| client_->SignOut(); |
| return; |
| } |
| } else if (peer_id != peer_id_) { |
| RTC_DCHECK(peer_id_ != -1); |
| RTC_LOG(LS_WARNING) |
| << "Received a message from unknown peer while already in a " |
| "conversation with a different peer."; |
| return; |
| } |
| |
| Json::CharReaderBuilder factory; |
| std::unique_ptr<Json::CharReader> reader = |
| absl::WrapUnique(factory.newCharReader()); |
| Json::Value jmessage; |
| if (!reader->parse(message.data(), message.data() + message.length(), |
| &jmessage, nullptr)) { |
| RTC_LOG(LS_WARNING) << "Received unknown message. " << message; |
| return; |
| } |
| std::string type_str; |
| std::string json_object; |
| |
| rtc::GetStringFromJsonObject(jmessage, kSessionDescriptionTypeName, |
| &type_str); |
| if (!type_str.empty()) { |
| if (type_str == "offer-loopback") { |
| // This is a loopback call. |
| // Recreate the peerconnection with DTLS disabled. |
| if (!ReinitializePeerConnectionForLoopback()) { |
| RTC_LOG(LS_ERROR) << "Failed to initialize our PeerConnection instance"; |
| DeletePeerConnection(); |
| client_->SignOut(); |
| } |
| return; |
| } |
| absl::optional<webrtc::SdpType> type_maybe = |
| webrtc::SdpTypeFromString(type_str); |
| if (!type_maybe) { |
| RTC_LOG(LS_ERROR) << "Unknown SDP type: " << type_str; |
| return; |
| } |
| webrtc::SdpType type = *type_maybe; |
| std::string sdp; |
| if (!rtc::GetStringFromJsonObject(jmessage, kSessionDescriptionSdpName, |
| &sdp)) { |
| RTC_LOG(LS_WARNING) |
| << "Can't parse received session description message."; |
| return; |
| } |
| webrtc::SdpParseError error; |
| std::unique_ptr<webrtc::SessionDescriptionInterface> session_description = |
| webrtc::CreateSessionDescription(type, sdp, &error); |
| if (!session_description) { |
| RTC_LOG(LS_WARNING) |
| << "Can't parse received session description message. " |
| "SdpParseError was: " |
| << error.description; |
| return; |
| } |
| RTC_LOG(LS_INFO) << " Received session description :" << message; |
| peer_connection_->SetRemoteDescription( |
| DummySetSessionDescriptionObserver::Create().get(), |
| session_description.release()); |
| if (type == webrtc::SdpType::kOffer) { |
| peer_connection_->CreateAnswer( |
| this, webrtc::PeerConnectionInterface::RTCOfferAnswerOptions()); |
| } |
| } else { |
| std::string sdp_mid; |
| int sdp_mlineindex = 0; |
| std::string sdp; |
| if (!rtc::GetStringFromJsonObject(jmessage, kCandidateSdpMidName, |
| &sdp_mid) || |
| !rtc::GetIntFromJsonObject(jmessage, kCandidateSdpMlineIndexName, |
| &sdp_mlineindex) || |
| !rtc::GetStringFromJsonObject(jmessage, kCandidateSdpName, &sdp)) { |
| RTC_LOG(LS_WARNING) << "Can't parse received message."; |
| return; |
| } |
| webrtc::SdpParseError error; |
| std::unique_ptr<webrtc::IceCandidateInterface> candidate( |
| webrtc::CreateIceCandidate(sdp_mid, sdp_mlineindex, sdp, &error)); |
| if (!candidate.get()) { |
| RTC_LOG(LS_WARNING) << "Can't parse received candidate message. " |
| "SdpParseError was: " |
| << error.description; |
| return; |
| } |
| if (!peer_connection_->AddIceCandidate(candidate.get())) { |
| RTC_LOG(LS_WARNING) << "Failed to apply the received candidate"; |
| return; |
| } |
| RTC_LOG(LS_INFO) << " Received candidate :" << message; |
| } |
| } |
| |
| void Conductor::OnMessageSent(int err) { |
| // Process the next pending message if any. |
| main_wnd_->QueueUIThreadCallback(SEND_MESSAGE_TO_PEER, NULL); |
| } |
| |
| void Conductor::OnServerConnectionFailure() { |
| main_wnd_->MessageBox("Error", ("Failed to connect to " + server_).c_str(), |
| true); |
| } |
| |
| // |
| // MainWndCallback implementation. |
| // |
| |
| void Conductor::StartLogin(const std::string& server, int port) { |
| if (client_->is_connected()) |
| return; |
| server_ = server; |
| client_->Connect(server, port, GetPeerName()); |
| } |
| |
| void Conductor::DisconnectFromServer() { |
| if (client_->is_connected()) |
| client_->SignOut(); |
| } |
| |
| void Conductor::ConnectToPeer(int peer_id) { |
| RTC_DCHECK(peer_id_ == -1); |
| RTC_DCHECK(peer_id != -1); |
| |
| if (peer_connection_.get()) { |
| main_wnd_->MessageBox( |
| "Error", "We only support connecting to one peer at a time", true); |
| return; |
| } |
| |
| if (InitializePeerConnection()) { |
| peer_id_ = peer_id; |
| peer_connection_->CreateOffer( |
| this, webrtc::PeerConnectionInterface::RTCOfferAnswerOptions()); |
| } else { |
| main_wnd_->MessageBox("Error", "Failed to initialize PeerConnection", true); |
| } |
| } |
| |
| void Conductor::AddTracks() { |
| if (!peer_connection_->GetSenders().empty()) { |
| return; // Already added tracks. |
| } |
| |
| rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track( |
| peer_connection_factory_->CreateAudioTrack( |
| kAudioLabel, |
| peer_connection_factory_->CreateAudioSource(cricket::AudioOptions()) |
| .get())); |
| auto result_or_error = peer_connection_->AddTrack(audio_track, {kStreamId}); |
| if (!result_or_error.ok()) { |
| RTC_LOG(LS_ERROR) << "Failed to add audio track to PeerConnection: " |
| << result_or_error.error().message(); |
| } |
| |
| rtc::scoped_refptr<CapturerTrackSource> video_device = |
| CapturerTrackSource::Create(); |
| if (video_device) { |
| rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track_( |
| peer_connection_factory_->CreateVideoTrack(video_device, kVideoLabel)); |
| main_wnd_->StartLocalRenderer(video_track_.get()); |
| |
| result_or_error = peer_connection_->AddTrack(video_track_, {kStreamId}); |
| if (!result_or_error.ok()) { |
| RTC_LOG(LS_ERROR) << "Failed to add video track to PeerConnection: " |
| << result_or_error.error().message(); |
| } |
| } else { |
| RTC_LOG(LS_ERROR) << "OpenVideoCaptureDevice failed"; |
| } |
| |
| main_wnd_->SwitchToStreamingUI(); |
| } |
| |
| void Conductor::DisconnectFromCurrentPeer() { |
| RTC_LOG(LS_INFO) << __FUNCTION__; |
| if (peer_connection_.get()) { |
| client_->SendHangUp(peer_id_); |
| DeletePeerConnection(); |
| } |
| |
| if (main_wnd_->IsWindow()) |
| main_wnd_->SwitchToPeerList(client_->peers()); |
| } |
| |
| void Conductor::UIThreadCallback(int msg_id, void* data) { |
| switch (msg_id) { |
| case PEER_CONNECTION_CLOSED: |
| RTC_LOG(LS_INFO) << "PEER_CONNECTION_CLOSED"; |
| DeletePeerConnection(); |
| |
| if (main_wnd_->IsWindow()) { |
| if (client_->is_connected()) { |
| main_wnd_->SwitchToPeerList(client_->peers()); |
| } else { |
| main_wnd_->SwitchToConnectUI(); |
| } |
| } else { |
| DisconnectFromServer(); |
| } |
| break; |
| |
| case SEND_MESSAGE_TO_PEER: { |
| RTC_LOG(LS_INFO) << "SEND_MESSAGE_TO_PEER"; |
| std::string* msg = reinterpret_cast<std::string*>(data); |
| if (msg) { |
| // For convenience, we always run the message through the queue. |
| // This way we can be sure that messages are sent to the server |
| // in the same order they were signaled without much hassle. |
| pending_messages_.push_back(msg); |
| } |
| |
| if (!pending_messages_.empty() && !client_->IsSendingMessage()) { |
| msg = pending_messages_.front(); |
| pending_messages_.pop_front(); |
| |
| if (!client_->SendToPeer(peer_id_, *msg) && peer_id_ != -1) { |
| RTC_LOG(LS_ERROR) << "SendToPeer failed"; |
| DisconnectFromServer(); |
| } |
| delete msg; |
| } |
| |
| if (!peer_connection_.get()) |
| peer_id_ = -1; |
| |
| break; |
| } |
| |
| case NEW_TRACK_ADDED: { |
| auto* track = reinterpret_cast<webrtc::MediaStreamTrackInterface*>(data); |
| if (track->kind() == webrtc::MediaStreamTrackInterface::kVideoKind) { |
| auto* video_track = static_cast<webrtc::VideoTrackInterface*>(track); |
| main_wnd_->StartRemoteRenderer(video_track); |
| } |
| track->Release(); |
| break; |
| } |
| |
| case TRACK_REMOVED: { |
| // Remote peer stopped sending a track. |
| auto* track = reinterpret_cast<webrtc::MediaStreamTrackInterface*>(data); |
| track->Release(); |
| break; |
| } |
| |
| default: |
| RTC_DCHECK_NOTREACHED(); |
| break; |
| } |
| } |
| |
| void Conductor::OnSuccess(webrtc::SessionDescriptionInterface* desc) { |
| peer_connection_->SetLocalDescription( |
| DummySetSessionDescriptionObserver::Create().get(), desc); |
| |
| std::string sdp; |
| desc->ToString(&sdp); |
| |
| // For loopback test. To save some connecting delay. |
| if (loopback_) { |
| // Replace message type from "offer" to "answer" |
| std::unique_ptr<webrtc::SessionDescriptionInterface> session_description = |
| webrtc::CreateSessionDescription(webrtc::SdpType::kAnswer, sdp); |
| peer_connection_->SetRemoteDescription( |
| DummySetSessionDescriptionObserver::Create().get(), |
| session_description.release()); |
| return; |
| } |
| |
| Json::Value jmessage; |
| jmessage[kSessionDescriptionTypeName] = |
| webrtc::SdpTypeToString(desc->GetType()); |
| jmessage[kSessionDescriptionSdpName] = sdp; |
| |
| Json::StreamWriterBuilder factory; |
| SendMessage(Json::writeString(factory, jmessage)); |
| } |
| |
| void Conductor::OnFailure(webrtc::RTCError error) { |
| RTC_LOG(LS_ERROR) << ToString(error.type()) << ": " << error.message(); |
| } |
| |
| void Conductor::SendMessage(const std::string& json_object) { |
| std::string* msg = new std::string(json_object); |
| main_wnd_->QueueUIThreadCallback(SEND_MESSAGE_TO_PEER, msg); |
| } |