webrtc/examples/unityplugin/simple_peer_connection.cc

/*
 *  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 "webrtc/examples/unityplugin/simple_peer_connection.h"

#include <utility>

#include "webrtc/api/test/fakeconstraints.h"
#include "webrtc/media/engine/webrtcvideocapturerfactory.h"
#include "webrtc/modules/video_capture/video_capture_factory.h"
#include "webrtc/rtc_base/json.h"

// Names used for a IceCandidate JSON object.
const char kCandidateSdpMidName[] = "sdpMid";
const char kCandidateSdpMlineIndexName[] = "sdpMLineIndex";
const char kCandidateSdpName[] = "candidate";

// Names used for a SessionDescription JSON object.
const char kSessionDescriptionTypeName[] = "type";
const char kSessionDescriptionSdpName[] = "sdp";

// Names used for media stream labels.
const char kAudioLabel[] = "audio_label";
const char kVideoLabel[] = "video_label";
const char kStreamLabel[] = "stream_label";

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;

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 DummySetSessionDescriptionObserver* Create() {
    return new rtc::RefCountedObject<DummySetSessionDescriptionObserver>();
  }
  virtual void OnSuccess() { LOG(INFO) << __FUNCTION__; }
  virtual void OnFailure(const std::string& error) {
    LOG(INFO) << __FUNCTION__ << " " << error;
  }

 protected:
  DummySetSessionDescriptionObserver() {}
  ~DummySetSessionDescriptionObserver() {}
};

}  // namespace

bool SimplePeerConnection::InitializePeerConnection(bool is_receiver) {
  RTC_DCHECK(peer_connection_.get() == nullptr);

  if (g_peer_connection_factory == nullptr) {
    g_worker_thread.reset(new rtc::Thread());
    g_worker_thread->Start();
    g_signaling_thread.reset(new rtc::Thread());
    g_signaling_thread->Start();

    g_peer_connection_factory = webrtc::CreatePeerConnectionFactory(
        g_worker_thread.get(), g_worker_thread.get(), g_signaling_thread.get(),
        nullptr, nullptr, nullptr);
  }
  if (!g_peer_connection_factory.get()) {
    DeletePeerConnection();
    return false;
  }

  g_peer_count++;
  if (!CreatePeerConnection(is_receiver)) {
    DeletePeerConnection();
    return false;
  }
  return peer_connection_.get() != nullptr;
}

bool SimplePeerConnection::CreatePeerConnection(bool is_receiver) {
  RTC_DCHECK(g_peer_connection_factory.get() != nullptr);
  RTC_DCHECK(peer_connection_.get() == nullptr);

  webrtc::PeerConnectionInterface::RTCConfiguration config;
  webrtc::PeerConnectionInterface::IceServer server;
  server.uri = GetPeerConnectionString();
  config.servers.push_back(server);

  webrtc::FakeConstraints constraints;
  constraints.SetAllowDtlsSctpDataChannels();

  if (is_receiver) {
    constraints.SetMandatoryReceiveAudio(true);
    constraints.SetMandatoryReceiveVideo(true);
  }

  peer_connection_ = g_peer_connection_factory->CreatePeerConnection(
      config, &constraints, nullptr, nullptr, this);

  return peer_connection_.get() != nullptr;
}

void SimplePeerConnection::DeletePeerConnection() {
  g_peer_count--;

  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;

  peer_connection_->CreateOffer(this, nullptr);
  return true;
}

bool SimplePeerConnection::CreateAnswer() {
  if (!peer_connection_.get())
    return false;

  peer_connection_->CreateAnswer(this, nullptr);
  return true;
}

void SimplePeerConnection::OnSuccess(
    webrtc::SessionDescriptionInterface* desc) {
  peer_connection_->SetLocalDescription(
      DummySetSessionDescriptionObserver::Create(), desc);

  std::string sdp;
  desc->ToString(&sdp);

  Json::StyledWriter writer;
  Json::Value jmessage;
  jmessage[kSessionDescriptionTypeName] = desc->type();
  jmessage[kSessionDescriptionSdpName] = sdp;

  if (OnLocalSdpReady)
    OnLocalSdpReady(writer.write(jmessage).c_str());
}

void SimplePeerConnection::OnFailure(const std::string& error) {
  LOG(LERROR) << error;

  if (OnFailureMessage)
    OnFailureMessage(error.c_str());
}

void SimplePeerConnection::OnIceCandidate(
    const webrtc::IceCandidateInterface* candidate) {
  LOG(INFO) << __FUNCTION__ << " " << candidate->sdp_mline_index();

  Json::StyledWriter writer;
  Json::Value jmessage;

  jmessage[kCandidateSdpMidName] = candidate->sdp_mid();
  jmessage[kCandidateSdpMlineIndexName] = candidate->sdp_mline_index();
  std::string sdp;
  if (!candidate->ToString(&sdp)) {
    LOG(LS_ERROR) << "Failed to serialize candidate";
    return;
  }
  jmessage[kCandidateSdpName] = sdp;

  if (OnIceCandiateReady)
    OnIceCandiateReady(writer.write(jmessage).c_str());
}

void SimplePeerConnection::RegisterOnVideoFramReady(
    VIDEOFRAMEREADY_CALLBACK callback) {
  OnVideoFrameReady = 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::RegisterOnIceCandiateReadytoSend(
    ICECANDIDATEREADYTOSEND_CALLBACK callback) {
  OnIceCandiateReady = callback;
}

bool SimplePeerConnection::ReceivedSdp(const char* msg) {
  if (!peer_connection_)
    return false;

  std::string message(msg);

  Json::Reader reader;
  Json::Value jmessage;
  if (!reader.parse(message, jmessage)) {
    LOG(WARNING) << "Received unknown message. " << message;
    return false;
  }
  std::string type;
  std::string json_object;

  rtc::GetStringFromJsonObject(jmessage, kSessionDescriptionTypeName, &type);
  if (type.empty())
    return false;

  std::string sdp;
  if (!rtc::GetStringFromJsonObject(jmessage, kSessionDescriptionSdpName,
                                    &sdp)) {
    LOG(WARNING) << "Can't parse received session description message.";
    return false;
  }
  webrtc::SdpParseError error;
  webrtc::SessionDescriptionInterface* session_description(
      webrtc::CreateSessionDescription(type, sdp, &error));
  if (!session_description) {
    LOG(WARNING) << "Can't parse received session description message. "
                 << "SdpParseError was: " << error.description;
    return false;
  }
  LOG(INFO) << " Received session description :" << message;
  peer_connection_->SetRemoteDescription(
      DummySetSessionDescriptionObserver::Create(), session_description);

  return true;
}

bool SimplePeerConnection::ReceivedIceCandidate(const char* ice_candidate) {
  if (!peer_connection_)
    return false;

  std::string message(ice_candidate);

  Json::Reader reader;
  Json::Value jmessage;
  if (!reader.parse(message, jmessage)) {
    LOG(WARNING) << "Received unknown message. " << message;
    return false;
  }
  std::string type;
  std::string json_object;

  rtc::GetStringFromJsonObject(jmessage, kSessionDescriptionTypeName, &type);
  if (!type.empty())
    return false;

  std::string sdp_mid;
  int sdp_mlineindex = 0;
  std::string sdp;
  if (!rtc::GetStringFromJsonObject(jmessage, kCandidateSdpMidName, &sdp_mid) ||
      !rtc::GetIntFromJsonObject(jmessage, kCandidateSdpMlineIndexName,
                                 &sdp_mlineindex) ||
      !rtc::GetStringFromJsonObject(jmessage, kCandidateSdpName, &sdp)) {
    LOG(WARNING) << "Can't parse received message.";
    return false;
  }
  webrtc::SdpParseError error;
  std::unique_ptr<webrtc::IceCandidateInterface> candidate(
      webrtc::CreateIceCandidate(sdp_mid, sdp_mlineindex, sdp, &error));
  if (!candidate.get()) {
    LOG(WARNING) << "Can't parse received candidate message. "
                 << "SdpParseError was: " << error.description;
    return false;
  }
  if (!peer_connection_->AddIceCandidate(candidate.get())) {
    LOG(WARNING) << "Failed to apply the received candidate";
    return false;
  }
  LOG(INFO) << " Received candidate :" << message;
  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;

  webrtc::AudioTrackInterface* audio_track = tracks[0];
  std::string id = audio_track->id();
  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) {
  LOG(INFO) << __FUNCTION__ << " " << stream->label();
  remote_stream_ = stream;

  SetAudioControl();
}

std::unique_ptr<cricket::VideoCapturer>
SimplePeerConnection::OpenVideoCaptureDevice() {
  std::vector<std::string> device_names;
  {
    std::unique_ptr<webrtc::VideoCaptureModule::DeviceInfo> info(
        webrtc::VideoCaptureFactory::CreateDeviceInfo());
    if (!info) {
      return nullptr;
    }
    int num_devices = info->NumberOfDevices();
    for (int i = 0; i < num_devices; ++i) {
      const uint32_t kSize = 256;
      char name[kSize] = {0};
      char id[kSize] = {0};
      if (info->GetDeviceName(i, name, kSize, id, kSize) != -1) {
        device_names.push_back(name);
      }
    }
  }

  cricket::WebRtcVideoDeviceCapturerFactory factory;
  std::unique_ptr<cricket::VideoCapturer> capturer;
  for (const auto& name : device_names) {
    capturer = factory.Create(cricket::Device(name, 0));
    if (capturer) {
      break;
    }
  }
  return capturer;
}

void SimplePeerConnection::AddStreams(bool audio_only) {
  if (active_streams_.find(kStreamLabel) != active_streams_.end())
    return;  // Already added.

  rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
      g_peer_connection_factory->CreateLocalMediaStream(kStreamLabel);

  rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
      g_peer_connection_factory->CreateAudioTrack(
          kAudioLabel, g_peer_connection_factory->CreateAudioSource(nullptr)));
  std::string id = audio_track->id();
  stream->AddTrack(audio_track);

  if (!audio_only) {
    std::unique_ptr<cricket::VideoCapturer> capture = OpenVideoCaptureDevice();
    if (capture) {
      rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track(
          g_peer_connection_factory->CreateVideoTrack(
              kVideoLabel, g_peer_connection_factory->CreateVideoSource(
                               OpenVideoCaptureDevice(), nullptr)));

      stream->AddTrack(video_track);
    }
  }

  if (!peer_connection_->AddStream(stream)) {
    LOG(LS_ERROR) << "Adding stream to PeerConnection failed";
  }

  typedef std::pair<std::string,
                    rtc::scoped_refptr<webrtc::MediaStreamInterface>>
      MediaStreamPair;
  active_streams_.insert(MediaStreamPair(stream->label(), stream));
}

bool SimplePeerConnection::CreateDataChannel() {
  struct webrtc::DataChannelInit init;
  init.ordered = true;
  init.reliable = true;
  data_channel_ = peer_connection_->CreateDataChannel("Hello", &init);
  if (data_channel_.get()) {
    data_channel_->RegisterObserver(this);
    LOG(LS_INFO) << "Succeeds to create data channel";
    return true;
  } else {
    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()) {
    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();
      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;
}