| /* | |
| * 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/objc_call_client.h" | |
| #include <memory> | |
| #include <utility> | |
| #import "sdk/objc/base/RTCVideoRenderer.h" | |
| #import "sdk/objc/components/video_codec/RTCDefaultVideoDecoderFactory.h" | |
| #import "sdk/objc/components/video_codec/RTCDefaultVideoEncoderFactory.h" | |
| #import "sdk/objc/helpers/RTCCameraPreviewView.h" | |
| #include "api/audio/audio_processing.h" | |
| #include "api/audio/builtin_audio_processing_builder.h" | |
| #include "api/audio_codecs/builtin_audio_decoder_factory.h" | |
| #include "api/audio_codecs/builtin_audio_encoder_factory.h" | |
| #include "api/create_modular_peer_connection_factory.h" | |
| #include "api/enable_media.h" | |
| #include "api/environment/environment_factory.h" | |
| #include "api/peer_connection_interface.h" | |
| #include "api/rtc_event_log/rtc_event_log_factory.h" | |
| #include "sdk/objc/native/api/video_capturer.h" | |
| #include "sdk/objc/native/api/video_decoder_factory.h" | |
| #include "sdk/objc/native/api/video_encoder_factory.h" | |
| #include "sdk/objc/native/api/video_renderer.h" | |
| namespace webrtc_examples { | |
| namespace { | |
| class CreateOfferObserver : public webrtc::CreateSessionDescriptionObserver { | |
| public: | |
| explicit CreateOfferObserver( | |
| webrtc::scoped_refptr<webrtc::PeerConnectionInterface> pc); | |
| void OnSuccess(webrtc::SessionDescriptionInterface* desc) override; | |
| void OnFailure(webrtc::RTCError error) override; | |
| private: | |
| const webrtc::scoped_refptr<webrtc::PeerConnectionInterface> pc_; | |
| }; | |
| class SetRemoteSessionDescriptionObserver | |
| : public webrtc::SetRemoteDescriptionObserverInterface { | |
| public: | |
| void OnSetRemoteDescriptionComplete(webrtc::RTCError error) override; | |
| }; | |
| class SetLocalSessionDescriptionObserver | |
| : public webrtc::SetLocalDescriptionObserverInterface { | |
| public: | |
| void OnSetLocalDescriptionComplete(webrtc::RTCError error) override; | |
| }; | |
| } // namespace | |
| ObjCCallClient::ObjCCallClient() | |
| : env_(webrtc::CreateEnvironment()), | |
| call_started_(false), | |
| pc_observer_(std::make_unique<PCObserver>(this)) { | |
| thread_checker_.Detach(); | |
| CreatePeerConnectionFactory(); | |
| } | |
| void ObjCCallClient::Call(RTC_OBJC_TYPE(RTCVideoCapturer) * capturer, | |
| id<RTC_OBJC_TYPE(RTCVideoRenderer)> remote_renderer) { | |
| RTC_DCHECK_RUN_ON(&thread_checker_); | |
| webrtc::MutexLock 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, env_, signaling_thread_.get(), worker_thread_.get()); | |
| CreatePeerConnection(); | |
| Connect(); | |
| } | |
| void ObjCCallClient::Hangup() { | |
| RTC_DCHECK_RUN_ON(&thread_checker_); | |
| call_started_ = false; | |
| { | |
| webrtc::MutexLock lock(&pc_mutex_); | |
| if (pc_ != nullptr) { | |
| pc_->Close(); | |
| pc_ = nullptr; | |
| } | |
| } | |
| remote_sink_ = nullptr; | |
| video_source_ = nullptr; | |
| } | |
| void ObjCCallClient::CreatePeerConnectionFactory() { | |
| network_thread_ = webrtc::Thread::CreateWithSocketServer(); | |
| network_thread_->SetName("network_thread", nullptr); | |
| RTC_CHECK(network_thread_->Start()) << "Failed to start thread"; | |
| worker_thread_ = webrtc::Thread::Create(); | |
| worker_thread_->SetName("worker_thread", nullptr); | |
| RTC_CHECK(worker_thread_->Start()) << "Failed to start thread"; | |
| signaling_thread_ = webrtc::Thread::Create(); | |
| signaling_thread_->SetName("signaling_thread", nullptr); | |
| RTC_CHECK(signaling_thread_->Start()) << "Failed to start thread"; | |
| webrtc::PeerConnectionFactoryDependencies dependencies; | |
| dependencies.env = env_; | |
| dependencies.network_thread = network_thread_.get(); | |
| dependencies.worker_thread = worker_thread_.get(); | |
| dependencies.signaling_thread = signaling_thread_.get(); | |
| dependencies.audio_encoder_factory = | |
| webrtc::CreateBuiltinAudioEncoderFactory(); | |
| dependencies.audio_decoder_factory = | |
| webrtc::CreateBuiltinAudioDecoderFactory(); | |
| dependencies.video_encoder_factory = webrtc::ObjCToNativeVideoEncoderFactory( | |
| [[RTC_OBJC_TYPE(RTCDefaultVideoEncoderFactory) alloc] init]); | |
| dependencies.video_decoder_factory = webrtc::ObjCToNativeVideoDecoderFactory( | |
| [[RTC_OBJC_TYPE(RTCDefaultVideoDecoderFactory) alloc] init]); | |
| dependencies.audio_processing_builder = | |
| std::make_unique<webrtc::BuiltinAudioProcessingBuilder>(); | |
| webrtc::EnableMedia(dependencies); | |
| dependencies.event_log_factory = | |
| std::make_unique<webrtc::RtcEventLogFactory>(); | |
| pcf_ = webrtc::CreateModularPeerConnectionFactory(std::move(dependencies)); | |
| RTC_LOG(LS_INFO) << "PeerConnectionFactory created: " << pcf_.get(); | |
| } | |
| void ObjCCallClient::CreatePeerConnection() { | |
| webrtc::MutexLock lock(&pc_mutex_); | |
| webrtc::PeerConnectionInterface::RTCConfiguration config; | |
| config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan; | |
| // Encryption has to be disabled for loopback to work. | |
| webrtc::PeerConnectionFactoryInterface::Options options; | |
| options.disable_encryption = true; | |
| pcf_->SetOptions(options); | |
| webrtc::PeerConnectionDependencies pc_dependencies(pc_observer_.get()); | |
| pc_ = pcf_->CreatePeerConnectionOrError(config, std::move(pc_dependencies)) | |
| .MoveValue(); | |
| RTC_LOG(LS_INFO) << "PeerConnection created: " << pc_.get(); | |
| webrtc::scoped_refptr<webrtc::VideoTrackInterface> local_video_track = | |
| pcf_->CreateVideoTrack(video_source_, "video"); | |
| pc_->AddTransceiver(local_video_track); | |
| RTC_LOG(LS_INFO) << "Local video sink set up: " << local_video_track.get(); | |
| for (const webrtc::scoped_refptr<webrtc::RtpTransceiverInterface>& | |
| tranceiver : pc_->GetTransceivers()) { | |
| webrtc::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(), webrtc::VideoSinkWants()); | |
| RTC_LOG(LS_INFO) << "Remote video sink set up: " << track.get(); | |
| break; | |
| } | |
| } | |
| } | |
| void ObjCCallClient::Connect() { | |
| webrtc::MutexLock lock(&pc_mutex_); | |
| pc_->CreateOffer(webrtc::make_ref_counted<CreateOfferObserver>(pc_).get(), | |
| 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( | |
| webrtc::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::IceCandidate* candidate) { | |
| RTC_LOG(LS_INFO) << "OnIceCandidate: " << candidate->server_url(); | |
| webrtc::MutexLock lock(&client_->pc_mutex_); | |
| RTC_DCHECK(client_->pc_ != nullptr); | |
| client_->pc_->AddIceCandidate(candidate); | |
| } | |
| CreateOfferObserver::CreateOfferObserver( | |
| webrtc::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( | |
| absl::WrapUnique(desc), | |
| webrtc::make_ref_counted<SetLocalSessionDescriptionObserver>()); | |
| // Generate a fake answer. | |
| std::unique_ptr<webrtc::SessionDescriptionInterface> answer( | |
| webrtc::CreateSessionDescription(webrtc::SdpType::kAnswer, sdp)); | |
| pc_->SetRemoteDescription( | |
| std::move(answer), | |
| webrtc::make_ref_counted<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::OnSetLocalDescriptionComplete( | |
| webrtc::RTCError error) { | |
| RTC_LOG(LS_INFO) << "Set local description: " << error.message(); | |
| } | |
| } // namespace webrtc_examples |