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/*
* 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 <memory>
#include <utility>
#include "api/enable_media_with_defaults.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 "examples/androidnativeapi/generated_jni/CallClient_jni.h"
#include "media/engine/internal_decoder_factory.h"
#include "media/engine/internal_encoder_factory.h"
#include "media/engine/webrtc_media_engine.h"
#include "sdk/android/native_api/jni/java_types.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(
rtc::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::IceCandidateInterface* candidate) override;
private:
AndroidCallClient* const client_;
};
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
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_ = 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 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.task_queue_factory = webrtc::CreateDefaultTaskQueueFactory();
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();
rtc::scoped_refptr<webrtc::VideoTrackInterface> local_video_track =
pcf_->CreateVideoTrack(video_source_, "video");
local_video_track->AddOrUpdateSink(local_sink_.get(), rtc::VideoSinkWants());
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 AndroidCallClient::Connect() {
webrtc::MutexLock lock(&pc_mutex_);
pc_->CreateOffer(rtc::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(
rtc::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::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(
rtc::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),
rtc::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