| /* |
| * Copyright 2018 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/objcnativeapi/objc/objccallclient.h" |
| |
| #include <utility> |
| |
| #import <WebRTC/RTCCameraPreviewView.h> |
| #import <WebRTC/RTCVideoCodecFactory.h> |
| #import <WebRTC/RTCVideoRenderer.h> |
| |
| #include "api/audio_codecs/builtin_audio_decoder_factory.h" |
| #include "api/audio_codecs/builtin_audio_encoder_factory.h" |
| #include "api/peerconnectioninterface.h" |
| #include "media/engine/webrtcmediaengine.h" |
| #include "modules/audio_processing/include/audio_processing.h" |
| #include "rtc_base/ptr_util.h" |
| #include "sdk/objc/Framework/Native/api/video_capturer.h" |
| #include "sdk/objc/Framework/Native/api/video_decoder_factory.h" |
| #include "sdk/objc/Framework/Native/api/video_encoder_factory.h" |
| #include "sdk/objc/Framework/Native/api/video_renderer.h" |
| |
| namespace webrtc_examples { |
| |
| namespace { |
| |
| class CreateOfferObserver : public webrtc::CreateSessionDescriptionObserver { |
| public: |
| explicit CreateOfferObserver(rtc::scoped_refptr<webrtc::PeerConnectionInterface> pc); |
| |
| void OnSuccess(webrtc::SessionDescriptionInterface* desc) override; |
| void OnFailure(webrtc::RTCError error) override; |
| |
| private: |
| const rtc::scoped_refptr<webrtc::PeerConnectionInterface> pc_; |
| }; |
| |
| class SetRemoteSessionDescriptionObserver : public webrtc::SetRemoteDescriptionObserverInterface { |
| public: |
| void OnSetRemoteDescriptionComplete(webrtc::RTCError error) override; |
| }; |
| |
| class SetLocalSessionDescriptionObserver : public webrtc::SetSessionDescriptionObserver { |
| public: |
| void OnSuccess() override; |
| void OnFailure(webrtc::RTCError error) override; |
| }; |
| |
| } // namespace |
| |
| ObjCCallClient::ObjCCallClient() |
| : call_started_(false), pc_observer_(rtc::MakeUnique<PCObserver>(this)) { |
| thread_checker_.DetachFromThread(); |
| CreatePeerConnectionFactory(); |
| } |
| |
| void ObjCCallClient::Call(RTCVideoCapturer* capturer, id<RTCVideoRenderer> remote_renderer) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| |
| rtc::CritScope lock(&pc_mutex_); |
| if (call_started_) { |
| RTC_LOG(LS_WARNING) << "Call already started."; |
| return; |
| } |
| call_started_ = true; |
| |
| remote_sink_ = webrtc::ObjCToNativeVideoRenderer(remote_renderer); |
| |
| video_source_ = |
| webrtc::ObjCToNativeVideoCapturer(capturer, signaling_thread_.get(), worker_thread_.get()); |
| |
| CreatePeerConnection(); |
| Connect(); |
| } |
| |
| void ObjCCallClient::Hangup() { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| |
| call_started_ = false; |
| |
| { |
| rtc::CritScope lock(&pc_mutex_); |
| if (pc_ != nullptr) { |
| pc_->Close(); |
| pc_ = nullptr; |
| } |
| } |
| |
| remote_sink_ = nullptr; |
| video_source_ = nullptr; |
| } |
| |
| void ObjCCallClient::CreatePeerConnectionFactory() { |
| network_thread_ = rtc::Thread::CreateWithSocketServer(); |
| network_thread_->SetName("network_thread", nullptr); |
| RTC_CHECK(network_thread_->Start()) << "Failed to start thread"; |
| |
| worker_thread_ = rtc::Thread::Create(); |
| worker_thread_->SetName("worker_thread", nullptr); |
| RTC_CHECK(worker_thread_->Start()) << "Failed to start thread"; |
| |
| signaling_thread_ = rtc::Thread::Create(); |
| signaling_thread_->SetName("signaling_thread", nullptr); |
| RTC_CHECK(signaling_thread_->Start()) << "Failed to start thread"; |
| |
| std::unique_ptr<webrtc::VideoDecoderFactory> videoDecoderFactory = |
| webrtc::ObjCToNativeVideoDecoderFactory([[RTCDefaultVideoDecoderFactory alloc] init]); |
| std::unique_ptr<webrtc::VideoEncoderFactory> videoEncoderFactory = |
| webrtc::ObjCToNativeVideoEncoderFactory([[RTCDefaultVideoEncoderFactory alloc] init]); |
| |
| std::unique_ptr<cricket::MediaEngineInterface> media_engine = |
| cricket::WebRtcMediaEngineFactory::Create(nullptr /* adm */, |
| webrtc::CreateBuiltinAudioEncoderFactory(), |
| webrtc::CreateBuiltinAudioDecoderFactory(), |
| std::move(videoEncoderFactory), |
| std::move(videoDecoderFactory), |
| nullptr /* audio_mixer */, |
| webrtc::AudioProcessingBuilder().Create()); |
| RTC_LOG(LS_INFO) << "Media engine created: " << media_engine.get(); |
| |
| pcf_ = webrtc::CreateModularPeerConnectionFactory(network_thread_.get(), |
| worker_thread_.get(), |
| signaling_thread_.get(), |
| std::move(media_engine), |
| webrtc::CreateCallFactory(), |
| webrtc::CreateRtcEventLogFactory()); |
| RTC_LOG(LS_INFO) << "PeerConnectionFactory created: " << pcf_; |
| } |
| |
| void ObjCCallClient::CreatePeerConnection() { |
| rtc::CritScope lock(&pc_mutex_); |
| webrtc::PeerConnectionInterface::RTCConfiguration config; |
| config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan; |
| // DTLS SRTP has to be disabled for loopback to work. |
| config.enable_dtls_srtp = false; |
| pc_ = pcf_->CreatePeerConnection( |
| config, nullptr /* port_allocator */, nullptr /* cert_generator */, pc_observer_.get()); |
| RTC_LOG(LS_INFO) << "PeerConnection created: " << pc_; |
| |
| rtc::scoped_refptr<webrtc::VideoTrackInterface> local_video_track = |
| pcf_->CreateVideoTrack("video", video_source_); |
| pc_->AddTransceiver(local_video_track); |
| RTC_LOG(LS_INFO) << "Local video sink set up: " << local_video_track; |
| |
| for (const rtc::scoped_refptr<webrtc::RtpTransceiverInterface>& tranceiver : |
| pc_->GetTransceivers()) { |
| rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> track = tranceiver->receiver()->track(); |
| if (track && track->kind() == webrtc::MediaStreamTrackInterface::kVideoKind) { |
| static_cast<webrtc::VideoTrackInterface*>(track.get()) |
| ->AddOrUpdateSink(remote_sink_.get(), rtc::VideoSinkWants()); |
| RTC_LOG(LS_INFO) << "Remote video sink set up: " << track; |
| break; |
| } |
| } |
| } |
| |
| void ObjCCallClient::Connect() { |
| rtc::CritScope lock(&pc_mutex_); |
| pc_->CreateOffer(new rtc::RefCountedObject<CreateOfferObserver>(pc_), |
| webrtc::PeerConnectionInterface::RTCOfferAnswerOptions()); |
| } |
| |
| ObjCCallClient::PCObserver::PCObserver(ObjCCallClient* client) : client_(client) {} |
| |
| void ObjCCallClient::PCObserver::OnSignalingChange( |
| webrtc::PeerConnectionInterface::SignalingState new_state) { |
| RTC_LOG(LS_INFO) << "OnSignalingChange: " << new_state; |
| } |
| |
| void ObjCCallClient::PCObserver::OnDataChannel( |
| rtc::scoped_refptr<webrtc::DataChannelInterface> data_channel) { |
| RTC_LOG(LS_INFO) << "OnDataChannel"; |
| } |
| |
| void ObjCCallClient::PCObserver::OnRenegotiationNeeded() { |
| RTC_LOG(LS_INFO) << "OnRenegotiationNeeded"; |
| } |
| |
| void ObjCCallClient::PCObserver::OnIceConnectionChange( |
| webrtc::PeerConnectionInterface::IceConnectionState new_state) { |
| RTC_LOG(LS_INFO) << "OnIceConnectionChange: " << new_state; |
| } |
| |
| void ObjCCallClient::PCObserver::OnIceGatheringChange( |
| webrtc::PeerConnectionInterface::IceGatheringState new_state) { |
| RTC_LOG(LS_INFO) << "OnIceGatheringChange: " << new_state; |
| } |
| |
| void ObjCCallClient::PCObserver::OnIceCandidate(const webrtc::IceCandidateInterface* candidate) { |
| RTC_LOG(LS_INFO) << "OnIceCandidate: " << candidate->server_url(); |
| rtc::CritScope lock(&client_->pc_mutex_); |
| RTC_DCHECK(client_->pc_ != nullptr); |
| client_->pc_->AddIceCandidate(candidate); |
| } |
| |
| CreateOfferObserver::CreateOfferObserver(rtc::scoped_refptr<webrtc::PeerConnectionInterface> pc) |
| : pc_(pc) {} |
| |
| void CreateOfferObserver::OnSuccess(webrtc::SessionDescriptionInterface* desc) { |
| std::string sdp; |
| desc->ToString(&sdp); |
| RTC_LOG(LS_INFO) << "Created offer: " << sdp; |
| |
| // Ownership of desc was transferred to us, now we transfer it forward. |
| pc_->SetLocalDescription(new rtc::RefCountedObject<SetLocalSessionDescriptionObserver>(), desc); |
| |
| // Generate a fake answer. |
| std::unique_ptr<webrtc::SessionDescriptionInterface> answer( |
| webrtc::CreateSessionDescription(webrtc::SdpType::kAnswer, sdp)); |
| pc_->SetRemoteDescription(std::move(answer), |
| new rtc::RefCountedObject<SetRemoteSessionDescriptionObserver>()); |
| } |
| |
| void CreateOfferObserver::OnFailure(webrtc::RTCError error) { |
| RTC_LOG(LS_INFO) << "Failed to create offer: " << error.message(); |
| } |
| |
| void SetRemoteSessionDescriptionObserver::OnSetRemoteDescriptionComplete(webrtc::RTCError error) { |
| RTC_LOG(LS_INFO) << "Set remote description: " << error.message(); |
| } |
| |
| void SetLocalSessionDescriptionObserver::OnSuccess() { |
| RTC_LOG(LS_INFO) << "Set local description success!"; |
| } |
| |
| void SetLocalSessionDescriptionObserver::OnFailure(webrtc::RTCError error) { |
| RTC_LOG(LS_INFO) << "Set local description failure: " << error.message(); |
| } |
| |
| } // namespace webrtc_examples |