| /* | |
| * 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/androidnativeapi/jni/android_call_client.h" | |
| #include <jni.h> | |
| #include <memory> | |
| #include <string> | |
| #include <utility> | |
| #include "api/data_channel_interface.h" | |
| #include "api/enable_media_with_defaults.h" | |
| #include "api/jsep.h" | |
| #include "api/make_ref_counted.h" | |
| #include "api/media_stream_interface.h" | |
| #include "api/peer_connection_interface.h" | |
| #include "api/rtc_error.h" | |
| #include "api/rtc_event_log/rtc_event_log_factory.h" | |
| #include "api/rtp_transceiver_interface.h" | |
| #include "api/scoped_refptr.h" | |
| #include "api/sequence_checker.h" | |
| #include "api/set_remote_description_observer_interface.h" | |
| #include "api/video/video_source_interface.h" | |
| #include "examples/androidnativeapi/generated_jni/CallClient_jni.h" | |
| #include "media/engine/internal_decoder_factory.h" | |
| #include "media/engine/internal_encoder_factory.h" | |
| #include "rtc_base/checks.h" | |
| #include "rtc_base/logging.h" | |
| #include "rtc_base/synchronization/mutex.h" | |
| #include "rtc_base/thread.h" | |
| #include "sdk/android/native_api/jni/java_types.h" | |
| #include "sdk/android/native_api/jni/scoped_java_ref.h" | |
| #include "sdk/android/native_api/video/video_source.h" | |
| #include "sdk/android/native_api/video/wrapper.h" | |
| namespace webrtc_examples { | |
| class AndroidCallClient::PCObserver : public webrtc::PeerConnectionObserver { | |
| public: | |
| explicit PCObserver(AndroidCallClient* client); | |
| void OnSignalingChange( | |
| webrtc::PeerConnectionInterface::SignalingState new_state) override; | |
| void OnDataChannel(webrtc::scoped_refptr<webrtc::DataChannelInterface> | |
| data_channel) override; | |
| void OnRenegotiationNeeded() override; | |
| void OnIceConnectionChange( | |
| webrtc::PeerConnectionInterface::IceConnectionState new_state) override; | |
| void OnIceGatheringChange( | |
| webrtc::PeerConnectionInterface::IceGatheringState new_state) override; | |
| void OnIceCandidate(const webrtc::IceCandidate* candidate) override; | |
| private: | |
| AndroidCallClient* const client_; | |
| }; | |
| 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::SetSessionDescriptionObserver { | |
| public: | |
| void OnSuccess() override; | |
| void OnFailure(webrtc::RTCError error) override; | |
| }; | |
| } // namespace | |
| AndroidCallClient::AndroidCallClient() | |
| : call_started_(false), pc_observer_(std::make_unique<PCObserver>(this)) { | |
| thread_checker_.Detach(); | |
| CreatePeerConnectionFactory(); | |
| } | |
| AndroidCallClient::~AndroidCallClient() = default; | |
| void AndroidCallClient::Call(JNIEnv* env, | |
| const webrtc::JavaRef<jobject>& local_sink, | |
| const webrtc::JavaRef<jobject>& remote_sink) { | |
| 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; | |
| local_sink_ = webrtc::JavaToNativeVideoSink(env, local_sink.obj()); | |
| remote_sink_ = webrtc::JavaToNativeVideoSink(env, remote_sink.obj()); | |
| video_source_ = webrtc::CreateJavaVideoSource(env, signaling_thread_.get(), | |
| /* is_screencast= */ false, | |
| /* align_timestamps= */ true); | |
| CreatePeerConnection(); | |
| Connect(); | |
| } | |
| void AndroidCallClient::Hangup(JNIEnv* env) { | |
| RTC_DCHECK_RUN_ON(&thread_checker_); | |
| call_started_ = false; | |
| { | |
| webrtc::MutexLock lock(&pc_mutex_); | |
| if (pc_ != nullptr) { | |
| pc_->Close(); | |
| pc_ = nullptr; | |
| } | |
| } | |
| local_sink_ = nullptr; | |
| remote_sink_ = nullptr; | |
| video_source_ = nullptr; | |
| } | |
| void AndroidCallClient::Delete(JNIEnv* env) { | |
| RTC_DCHECK_RUN_ON(&thread_checker_); | |
| delete this; | |
| } | |
| webrtc::ScopedJavaLocalRef<jobject> | |
| AndroidCallClient::GetJavaVideoCapturerObserver(JNIEnv* env) { | |
| RTC_DCHECK_RUN_ON(&thread_checker_); | |
| return video_source_->GetJavaVideoCapturerObserver(env); | |
| } | |
| void AndroidCallClient::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 pcf_deps; | |
| pcf_deps.network_thread = network_thread_.get(); | |
| pcf_deps.worker_thread = worker_thread_.get(); | |
| pcf_deps.signaling_thread = signaling_thread_.get(); | |
| pcf_deps.event_log_factory = std::make_unique<webrtc::RtcEventLogFactory>(); | |
| pcf_deps.video_encoder_factory = | |
| std::make_unique<webrtc::InternalEncoderFactory>(); | |
| pcf_deps.video_decoder_factory = | |
| std::make_unique<webrtc::InternalDecoderFactory>(); | |
| EnableMediaWithDefaults(pcf_deps); | |
| pcf_ = CreateModularPeerConnectionFactory(std::move(pcf_deps)); | |
| RTC_LOG(LS_INFO) << "PeerConnectionFactory created: " << pcf_.get(); | |
| } | |
| void AndroidCallClient::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 deps(pc_observer_.get()); | |
| pc_ = pcf_->CreatePeerConnectionOrError(config, std::move(deps)).MoveValue(); | |
| RTC_LOG(LS_INFO) << "PeerConnection created: " << pc_.get(); | |
| webrtc::scoped_refptr<webrtc::VideoTrackInterface> local_video_track = | |
| pcf_->CreateVideoTrack(video_source_, "video"); | |
| local_video_track->AddOrUpdateSink(local_sink_.get(), | |
| webrtc::VideoSinkWants()); | |
| 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 AndroidCallClient::Connect() { | |
| webrtc::MutexLock lock(&pc_mutex_); | |
| pc_->CreateOffer(webrtc::make_ref_counted<CreateOfferObserver>(pc_).get(), | |
| webrtc::PeerConnectionInterface::RTCOfferAnswerOptions()); | |
| } | |
| AndroidCallClient::PCObserver::PCObserver(AndroidCallClient* client) | |
| : client_(client) {} | |
| void AndroidCallClient::PCObserver::OnSignalingChange( | |
| webrtc::PeerConnectionInterface::SignalingState new_state) { | |
| RTC_LOG(LS_INFO) << "OnSignalingChange: " << new_state; | |
| } | |
| void AndroidCallClient::PCObserver::OnDataChannel( | |
| webrtc::scoped_refptr<webrtc::DataChannelInterface> data_channel) { | |
| RTC_LOG(LS_INFO) << "OnDataChannel"; | |
| } | |
| void AndroidCallClient::PCObserver::OnRenegotiationNeeded() { | |
| RTC_LOG(LS_INFO) << "OnRenegotiationNeeded"; | |
| } | |
| void AndroidCallClient::PCObserver::OnIceConnectionChange( | |
| webrtc::PeerConnectionInterface::IceConnectionState new_state) { | |
| RTC_LOG(LS_INFO) << "OnIceConnectionChange: " << new_state; | |
| } | |
| void AndroidCallClient::PCObserver::OnIceGatheringChange( | |
| webrtc::PeerConnectionInterface::IceGatheringState new_state) { | |
| RTC_LOG(LS_INFO) << "OnIceGatheringChange: " << new_state; | |
| } | |
| void AndroidCallClient::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( | |
| webrtc::make_ref_counted<SetLocalSessionDescriptionObserver>().get(), | |
| desc); | |
| // 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: " << ToString(error.type()) | |
| << ": " << 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: " | |
| << ToString(error.type()) << ": " << error.message(); | |
| } | |
| static jlong JNI_CallClient_CreateClient(JNIEnv* env) { | |
| return webrtc::NativeToJavaPointer(new webrtc_examples::AndroidCallClient()); | |
| } | |
| } // namespace webrtc_examples |