| /* |
| * 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_codecs/builtin_audio_decoder_factory.h" |
| #include "api/audio_codecs/builtin_audio_encoder_factory.h" |
| #include "api/enable_media.h" |
| #include "api/peer_connection_interface.h" |
| #include "api/rtc_event_log/rtc_event_log_factory.h" |
| #include "api/task_queue/default_task_queue_factory.h" |
| #include "modules/audio_processing/include/audio_processing.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(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::SetLocalDescriptionObserverInterface { |
| public: |
| void OnSetLocalDescriptionComplete(webrtc::RTCError error) override; |
| }; |
| |
| } // namespace |
| |
| ObjCCallClient::ObjCCallClient() |
| : 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, 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_ = 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"; |
| |
| webrtc::PeerConnectionFactoryDependencies dependencies; |
| dependencies.network_thread = network_thread_.get(); |
| dependencies.worker_thread = worker_thread_.get(); |
| dependencies.signaling_thread = signaling_thread_.get(); |
| dependencies.task_queue_factory = webrtc::CreateDefaultTaskQueueFactory(); |
| 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 = webrtc::AudioProcessingBuilder().Create(); |
| 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(); |
| |
| rtc::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 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.get(); |
| break; |
| } |
| } |
| } |
| |
| void ObjCCallClient::Connect() { |
| webrtc::MutexLock lock(&pc_mutex_); |
| pc_->CreateOffer(rtc::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( |
| 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(); |
| webrtc::MutexLock 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(absl::WrapUnique(desc), |
| rtc::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), |
| rtc::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 |