|  | /* | 
|  | *  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/peer_connection_interface.h" | 
|  | #include "api/rtc_event_log/rtc_event_log_factory.h" | 
|  | #include "api/task_queue/default_task_queue_factory.h" | 
|  | #include "media/engine/webrtc_media_engine.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(); | 
|  | cricket::MediaEngineDependencies media_deps; | 
|  | media_deps.task_queue_factory = dependencies.task_queue_factory.get(); | 
|  | media_deps.audio_encoder_factory = webrtc::CreateBuiltinAudioEncoderFactory(); | 
|  | media_deps.audio_decoder_factory = webrtc::CreateBuiltinAudioDecoderFactory(); | 
|  | media_deps.video_encoder_factory = webrtc::ObjCToNativeVideoEncoderFactory( | 
|  | [[RTC_OBJC_TYPE(RTCDefaultVideoEncoderFactory) alloc] init]); | 
|  | media_deps.video_decoder_factory = webrtc::ObjCToNativeVideoDecoderFactory( | 
|  | [[RTC_OBJC_TYPE(RTCDefaultVideoDecoderFactory) alloc] init]); | 
|  | media_deps.audio_processing = webrtc::AudioProcessingBuilder().Create(); | 
|  | dependencies.media_engine = cricket::CreateMediaEngine(std::move(media_deps)); | 
|  | RTC_LOG(LS_INFO) << "Media engine created: " << dependencies.media_engine.get(); | 
|  | dependencies.call_factory = webrtc::CreateCallFactory(); | 
|  | dependencies.event_log_factory = | 
|  | std::make_unique<webrtc::RtcEventLogFactory>(dependencies.task_queue_factory.get()); | 
|  | 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 |