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/*
* Copyright (c) 2017 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/unityplugin/simple_peer_connection.h"
#include <utility>
#include "absl/memory/memory.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/create_peerconnection_factory.h"
#include "media/engine/internal_decoder_factory.h"
#include "media/engine/internal_encoder_factory.h"
#include "media/engine/multiplex_codec_factory.h"
#include "modules/audio_device/include/audio_device.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "modules/video_capture/video_capture_factory.h"
#include "pc/video_track_source.h"
#include "test/vcm_capturer.h"
#if defined(WEBRTC_ANDROID)
#include "examples/unityplugin/class_reference_holder.h"
#include "modules/utility/include/helpers_android.h"
#include "sdk/android/src/jni/android_video_track_source.h"
#include "sdk/android/src/jni/jni_helpers.h"
#endif
// Names used for media stream ids.
const char kAudioLabel[] = "audio_label";
const char kVideoLabel[] = "video_label";
const char kStreamId[] = "stream_id";
namespace {
static int g_peer_count = 0;
static std::unique_ptr<rtc::Thread> g_worker_thread;
static std::unique_ptr<rtc::Thread> g_signaling_thread;
static rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface>
g_peer_connection_factory;
#if defined(WEBRTC_ANDROID)
// Android case: the video track does not own the capturer, and it
// relies on the app to dispose the capturer when the peerconnection
// shuts down.
static jobject g_camera = nullptr;
#else
class CapturerTrackSource : public webrtc::VideoTrackSource {
public:
static rtc::scoped_refptr<CapturerTrackSource> Create() {
const size_t kWidth = 640;
const size_t kHeight = 480;
const size_t kFps = 30;
const size_t kDeviceIndex = 0;
std::unique_ptr<webrtc::test::VcmCapturer> capturer = absl::WrapUnique(
webrtc::test::VcmCapturer::Create(kWidth, kHeight, kFps, kDeviceIndex));
if (!capturer) {
return nullptr;
}
return rtc::make_ref_counted<CapturerTrackSource>(std::move(capturer));
}
protected:
explicit CapturerTrackSource(
std::unique_ptr<webrtc::test::VcmCapturer> capturer)
: VideoTrackSource(/*remote=*/false), capturer_(std::move(capturer)) {}
private:
rtc::VideoSourceInterface<webrtc::VideoFrame>* source() override {
return capturer_.get();
}
std::unique_ptr<webrtc::test::VcmCapturer> capturer_;
};
#endif
std::string GetEnvVarOrDefault(const char* env_var_name,
const char* default_value) {
std::string value;
const char* env_var = getenv(env_var_name);
if (env_var)
value = env_var;
if (value.empty())
value = default_value;
return value;
}
std::string GetPeerConnectionString() {
return GetEnvVarOrDefault("WEBRTC_CONNECT", "stun:stun.l.google.com:19302");
}
class DummySetSessionDescriptionObserver
: public webrtc::SetSessionDescriptionObserver {
public:
static rtc::scoped_refptr<DummySetSessionDescriptionObserver> Create() {
return rtc::make_ref_counted<DummySetSessionDescriptionObserver>();
}
virtual void OnSuccess() { RTC_LOG(LS_INFO) << __FUNCTION__; }
virtual void OnFailure(webrtc::RTCError error) {
RTC_LOG(LS_INFO) << __FUNCTION__ << " " << ToString(error.type()) << ": "
<< error.message();
}
protected:
DummySetSessionDescriptionObserver() {}
~DummySetSessionDescriptionObserver() {}
};
} // namespace
bool SimplePeerConnection::InitializePeerConnection(const char** turn_urls,
const int no_of_urls,
const char* username,
const char* credential,
bool is_receiver) {
RTC_DCHECK(peer_connection_.get() == nullptr);
if (g_peer_connection_factory == nullptr) {
g_worker_thread = rtc::Thread::Create();
g_worker_thread->Start();
g_signaling_thread = rtc::Thread::Create();
g_signaling_thread->Start();
g_peer_connection_factory = webrtc::CreatePeerConnectionFactory(
g_worker_thread.get(), g_worker_thread.get(), g_signaling_thread.get(),
nullptr, webrtc::CreateBuiltinAudioEncoderFactory(),
webrtc::CreateBuiltinAudioDecoderFactory(),
std::unique_ptr<webrtc::VideoEncoderFactory>(
new webrtc::MultiplexEncoderFactory(
std::make_unique<webrtc::InternalEncoderFactory>())),
std::unique_ptr<webrtc::VideoDecoderFactory>(
new webrtc::MultiplexDecoderFactory(
std::make_unique<webrtc::InternalDecoderFactory>())),
nullptr, nullptr);
}
if (!g_peer_connection_factory.get()) {
DeletePeerConnection();
return false;
}
g_peer_count++;
if (!CreatePeerConnection(turn_urls, no_of_urls, username, credential)) {
DeletePeerConnection();
return false;
}
mandatory_receive_ = is_receiver;
return peer_connection_.get() != nullptr;
}
bool SimplePeerConnection::CreatePeerConnection(const char** turn_urls,
const int no_of_urls,
const char* username,
const char* credential) {
RTC_DCHECK(g_peer_connection_factory.get() != nullptr);
RTC_DCHECK(peer_connection_.get() == nullptr);
local_video_observer_.reset(new VideoObserver());
remote_video_observer_.reset(new VideoObserver());
// Add the turn server.
if (turn_urls != nullptr) {
if (no_of_urls > 0) {
webrtc::PeerConnectionInterface::IceServer turn_server;
for (int i = 0; i < no_of_urls; i++) {
std::string url(turn_urls[i]);
if (url.length() > 0)
turn_server.urls.push_back(turn_urls[i]);
}
std::string user_name(username);
if (user_name.length() > 0)
turn_server.username = username;
std::string password(credential);
if (password.length() > 0)
turn_server.password = credential;
config_.servers.push_back(turn_server);
}
}
// Add the stun server.
webrtc::PeerConnectionInterface::IceServer stun_server;
stun_server.uri = GetPeerConnectionString();
config_.servers.push_back(stun_server);
auto result = g_peer_connection_factory->CreatePeerConnectionOrError(
config_, webrtc::PeerConnectionDependencies(this));
if (!result.ok()) {
peer_connection_ = nullptr;
return false;
}
peer_connection_ = result.MoveValue();
return true;
}
void SimplePeerConnection::DeletePeerConnection() {
g_peer_count--;
#if defined(WEBRTC_ANDROID)
if (g_camera) {
JNIEnv* env = webrtc::jni::GetEnv();
jclass pc_factory_class =
unity_plugin::FindClass(env, "org/webrtc/UnityUtility");
jmethodID stop_camera_method = webrtc::GetStaticMethodID(
env, pc_factory_class, "StopCamera", "(Lorg/webrtc/VideoCapturer;)V");
env->CallStaticVoidMethod(pc_factory_class, stop_camera_method, g_camera);
CHECK_EXCEPTION(env);
g_camera = nullptr;
}
#endif
CloseDataChannel();
peer_connection_ = nullptr;
active_streams_.clear();
if (g_peer_count == 0) {
g_peer_connection_factory = nullptr;
g_signaling_thread.reset();
g_worker_thread.reset();
}
}
bool SimplePeerConnection::CreateOffer() {
if (!peer_connection_.get())
return false;
webrtc::PeerConnectionInterface::RTCOfferAnswerOptions options;
if (mandatory_receive_) {
options.offer_to_receive_audio = true;
options.offer_to_receive_video = true;
}
peer_connection_->CreateOffer(this, options);
return true;
}
bool SimplePeerConnection::CreateAnswer() {
if (!peer_connection_.get())
return false;
webrtc::PeerConnectionInterface::RTCOfferAnswerOptions options;
if (mandatory_receive_) {
options.offer_to_receive_audio = true;
options.offer_to_receive_video = true;
}
peer_connection_->CreateAnswer(this, options);
return true;
}
void SimplePeerConnection::OnSuccess(
webrtc::SessionDescriptionInterface* desc) {
peer_connection_->SetLocalDescription(
DummySetSessionDescriptionObserver::Create().get(), desc);
std::string sdp;
desc->ToString(&sdp);
if (OnLocalSdpReady)
OnLocalSdpReady(desc->type().c_str(), sdp.c_str());
}
void SimplePeerConnection::OnFailure(webrtc::RTCError error) {
RTC_LOG(LS_ERROR) << ToString(error.type()) << ": " << error.message();
// TODO(hta): include error.type in the message
if (OnFailureMessage)
OnFailureMessage(error.message());
}
void SimplePeerConnection::OnIceCandidate(
const webrtc::IceCandidateInterface* candidate) {
RTC_LOG(LS_INFO) << __FUNCTION__ << " " << candidate->sdp_mline_index();
std::string sdp;
if (!candidate->ToString(&sdp)) {
RTC_LOG(LS_ERROR) << "Failed to serialize candidate";
return;
}
if (OnIceCandidateReady)
OnIceCandidateReady(sdp.c_str(), candidate->sdp_mline_index(),
candidate->sdp_mid().c_str());
}
void SimplePeerConnection::RegisterOnLocalI420FrameReady(
I420FRAMEREADY_CALLBACK callback) {
if (local_video_observer_)
local_video_observer_->SetVideoCallback(callback);
}
void SimplePeerConnection::RegisterOnRemoteI420FrameReady(
I420FRAMEREADY_CALLBACK callback) {
if (remote_video_observer_)
remote_video_observer_->SetVideoCallback(callback);
}
void SimplePeerConnection::RegisterOnLocalDataChannelReady(
LOCALDATACHANNELREADY_CALLBACK callback) {
OnLocalDataChannelReady = callback;
}
void SimplePeerConnection::RegisterOnDataFromDataChannelReady(
DATAFROMEDATECHANNELREADY_CALLBACK callback) {
OnDataFromDataChannelReady = callback;
}
void SimplePeerConnection::RegisterOnFailure(FAILURE_CALLBACK callback) {
OnFailureMessage = callback;
}
void SimplePeerConnection::RegisterOnAudioBusReady(
AUDIOBUSREADY_CALLBACK callback) {
OnAudioReady = callback;
}
void SimplePeerConnection::RegisterOnLocalSdpReadytoSend(
LOCALSDPREADYTOSEND_CALLBACK callback) {
OnLocalSdpReady = callback;
}
void SimplePeerConnection::RegisterOnIceCandidateReadytoSend(
ICECANDIDATEREADYTOSEND_CALLBACK callback) {
OnIceCandidateReady = callback;
}
bool SimplePeerConnection::SetRemoteDescription(const char* type,
const char* sdp) {
if (!peer_connection_)
return false;
std::string remote_desc(sdp);
std::string desc_type(type);
webrtc::SdpParseError error;
webrtc::SessionDescriptionInterface* session_description(
webrtc::CreateSessionDescription(desc_type, remote_desc, &error));
if (!session_description) {
RTC_LOG(LS_WARNING) << "Can't parse received session description message. "
"SdpParseError was: "
<< error.description;
return false;
}
RTC_LOG(LS_INFO) << " Received session description :" << remote_desc;
peer_connection_->SetRemoteDescription(
DummySetSessionDescriptionObserver::Create().get(), session_description);
return true;
}
bool SimplePeerConnection::AddIceCandidate(const char* candidate,
const int sdp_mlineindex,
const char* sdp_mid) {
if (!peer_connection_)
return false;
webrtc::SdpParseError error;
std::unique_ptr<webrtc::IceCandidateInterface> ice_candidate(
webrtc::CreateIceCandidate(sdp_mid, sdp_mlineindex, candidate, &error));
if (!ice_candidate.get()) {
RTC_LOG(LS_WARNING) << "Can't parse received candidate message. "
"SdpParseError was: "
<< error.description;
return false;
}
if (!peer_connection_->AddIceCandidate(ice_candidate.get())) {
RTC_LOG(LS_WARNING) << "Failed to apply the received candidate";
return false;
}
RTC_LOG(LS_INFO) << " Received candidate :" << candidate;
return true;
}
void SimplePeerConnection::SetAudioControl(bool is_mute, bool is_record) {
is_mute_audio_ = is_mute;
is_record_audio_ = is_record;
SetAudioControl();
}
void SimplePeerConnection::SetAudioControl() {
if (!remote_stream_)
return;
webrtc::AudioTrackVector tracks = remote_stream_->GetAudioTracks();
if (tracks.empty())
return;
rtc::scoped_refptr<webrtc::AudioTrackInterface>& audio_track = tracks[0];
if (is_record_audio_)
audio_track->AddSink(this);
else
audio_track->RemoveSink(this);
for (auto& track : tracks) {
if (is_mute_audio_)
track->set_enabled(false);
else
track->set_enabled(true);
}
}
void SimplePeerConnection::OnAddStream(
rtc::scoped_refptr<webrtc::MediaStreamInterface> stream) {
RTC_LOG(LS_INFO) << __FUNCTION__ << " " << stream->id();
remote_stream_ = stream;
if (remote_video_observer_ && !remote_stream_->GetVideoTracks().empty()) {
remote_stream_->GetVideoTracks()[0]->AddOrUpdateSink(
remote_video_observer_.get(), rtc::VideoSinkWants());
}
SetAudioControl();
}
void SimplePeerConnection::AddStreams(bool audio_only) {
if (active_streams_.find(kStreamId) != active_streams_.end())
return; // Already added.
rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
g_peer_connection_factory->CreateLocalMediaStream(kStreamId);
rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
g_peer_connection_factory->CreateAudioTrack(
kAudioLabel,
g_peer_connection_factory->CreateAudioSource(cricket::AudioOptions())
.get()));
stream->AddTrack(audio_track);
if (!audio_only) {
#if defined(WEBRTC_ANDROID)
JNIEnv* env = webrtc::jni::GetEnv();
jclass pc_factory_class =
unity_plugin::FindClass(env, "org/webrtc/UnityUtility");
jmethodID load_texture_helper_method = webrtc::GetStaticMethodID(
env, pc_factory_class, "LoadSurfaceTextureHelper",
"()Lorg/webrtc/SurfaceTextureHelper;");
jobject texture_helper = env->CallStaticObjectMethod(
pc_factory_class, load_texture_helper_method);
CHECK_EXCEPTION(env);
RTC_DCHECK(texture_helper != nullptr)
<< "Cannot get the Surface Texture Helper.";
auto source = rtc::make_ref_counted<webrtc::jni::AndroidVideoTrackSource>(
g_signaling_thread.get(), env, /*is_screencast=*/false,
/*align_timestamps=*/true);
// link with VideoCapturer (Camera);
jmethodID link_camera_method = webrtc::GetStaticMethodID(
env, pc_factory_class, "LinkCamera",
"(JLorg/webrtc/SurfaceTextureHelper;)Lorg/webrtc/VideoCapturer;");
jobject camera_tmp =
env->CallStaticObjectMethod(pc_factory_class, link_camera_method,
(jlong)source.get(), texture_helper);
CHECK_EXCEPTION(env);
g_camera = (jobject)env->NewGlobalRef(camera_tmp);
rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track(
g_peer_connection_factory->CreateVideoTrack(kVideoLabel,
source.release()));
stream->AddTrack(video_track);
#else
rtc::scoped_refptr<CapturerTrackSource> video_device =
CapturerTrackSource::Create();
if (video_device) {
rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track(
g_peer_connection_factory->CreateVideoTrack(kVideoLabel,
video_device.get()));
stream->AddTrack(video_track);
}
#endif
if (local_video_observer_ && !stream->GetVideoTracks().empty()) {
stream->GetVideoTracks()[0]->AddOrUpdateSink(local_video_observer_.get(),
rtc::VideoSinkWants());
}
}
if (!peer_connection_->AddStream(stream.get())) {
RTC_LOG(LS_ERROR) << "Adding stream to PeerConnection failed";
}
typedef std::pair<std::string,
rtc::scoped_refptr<webrtc::MediaStreamInterface>>
MediaStreamPair;
active_streams_.insert(MediaStreamPair(stream->id(), stream));
}
bool SimplePeerConnection::CreateDataChannel() {
struct webrtc::DataChannelInit init;
init.ordered = true;
init.reliable = true;
auto result = peer_connection_->CreateDataChannelOrError("Hello", &init);
if (result.ok()) {
data_channel_ = result.MoveValue();
data_channel_->RegisterObserver(this);
RTC_LOG(LS_INFO) << "Succeeds to create data channel";
return true;
} else {
RTC_LOG(LS_INFO) << "Fails to create data channel";
return false;
}
}
void SimplePeerConnection::CloseDataChannel() {
if (data_channel_.get()) {
data_channel_->UnregisterObserver();
data_channel_->Close();
}
data_channel_ = nullptr;
}
bool SimplePeerConnection::SendDataViaDataChannel(const std::string& data) {
if (!data_channel_.get()) {
RTC_LOG(LS_INFO) << "Data channel is not established";
return false;
}
webrtc::DataBuffer buffer(data);
data_channel_->Send(buffer);
return true;
}
// Peerconnection observer
void SimplePeerConnection::OnDataChannel(
rtc::scoped_refptr<webrtc::DataChannelInterface> channel) {
channel->RegisterObserver(this);
}
void SimplePeerConnection::OnStateChange() {
if (data_channel_) {
webrtc::DataChannelInterface::DataState state = data_channel_->state();
if (state == webrtc::DataChannelInterface::kOpen) {
if (OnLocalDataChannelReady)
OnLocalDataChannelReady();
RTC_LOG(LS_INFO) << "Data channel is open";
}
}
}
// A data buffer was successfully received.
void SimplePeerConnection::OnMessage(const webrtc::DataBuffer& buffer) {
size_t size = buffer.data.size();
char* msg = new char[size + 1];
memcpy(msg, buffer.data.data(), size);
msg[size] = 0;
if (OnDataFromDataChannelReady)
OnDataFromDataChannelReady(msg);
delete[] msg;
}
// AudioTrackSinkInterface implementation.
void SimplePeerConnection::OnData(const void* audio_data,
int bits_per_sample,
int sample_rate,
size_t number_of_channels,
size_t number_of_frames) {
if (OnAudioReady)
OnAudioReady(audio_data, bits_per_sample, sample_rate,
static_cast<int>(number_of_channels),
static_cast<int>(number_of_frames));
}
std::vector<uint32_t> SimplePeerConnection::GetRemoteAudioTrackSsrcs() {
std::vector<rtc::scoped_refptr<webrtc::RtpReceiverInterface>> receivers =
peer_connection_->GetReceivers();
std::vector<uint32_t> ssrcs;
for (const auto& receiver : receivers) {
if (receiver->media_type() != cricket::MEDIA_TYPE_AUDIO)
continue;
std::vector<webrtc::RtpEncodingParameters> params =
receiver->GetParameters().encodings;
for (const auto& param : params) {
uint32_t ssrc = param.ssrc.value_or(0);
if (ssrc > 0)
ssrcs.push_back(ssrc);
}
}
return ssrcs;
}