Create //webrtc/api:libjingle_peerconnection_api + refactorings.

webrtc/api/peerconnection_unittest.cc

-/*
- *  Copyright 2012 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 <stdio.h>
-
-#include <algorithm>
-#include <list>
-#include <map>
-#include <memory>
-#include <utility>
-#include <vector>
-
-#include "webrtc/api/dtmfsender.h"
-#include "webrtc/api/fakemetricsobserver.h"
-#include "webrtc/api/localaudiosource.h"
-#include "webrtc/api/mediastreaminterface.h"
-#include "webrtc/api/peerconnection.h"
-#include "webrtc/api/peerconnectionfactory.h"
-#include "webrtc/api/peerconnectioninterface.h"
-#include "webrtc/api/test/fakeaudiocapturemodule.h"
-#include "webrtc/api/test/fakeconstraints.h"
-#include "webrtc/api/test/fakeperiodicvideocapturer.h"
-#include "webrtc/api/test/fakertccertificategenerator.h"
-#include "webrtc/api/test/fakevideotrackrenderer.h"
-#include "webrtc/api/test/mockpeerconnectionobservers.h"
-#include "webrtc/base/fakenetwork.h"
-#include "webrtc/base/gunit.h"
-#include "webrtc/base/helpers.h"
-#include "webrtc/base/physicalsocketserver.h"
-#include "webrtc/base/ssladapter.h"
-#include "webrtc/base/sslstreamadapter.h"
-#include "webrtc/base/thread.h"
-#include "webrtc/base/virtualsocketserver.h"
-#include "webrtc/media/engine/fakewebrtcvideoengine.h"
-#include "webrtc/p2p/base/p2pconstants.h"
-#include "webrtc/p2p/base/portinterface.h"
-#include "webrtc/p2p/base/sessiondescription.h"
-#include "webrtc/p2p/base/testturnserver.h"
-#include "webrtc/p2p/client/basicportallocator.h"
-#include "webrtc/pc/mediasession.h"
-
-#define MAYBE_SKIP_TEST(feature)                    \
-  if (!(feature())) {                               \
-    LOG(LS_INFO) << "Feature disabled... skipping"; \
-    return;                                         \
-  }
-
-using cricket::ContentInfo;
-using cricket::FakeWebRtcVideoDecoder;
-using cricket::FakeWebRtcVideoDecoderFactory;
-using cricket::FakeWebRtcVideoEncoder;
-using cricket::FakeWebRtcVideoEncoderFactory;
-using cricket::MediaContentDescription;
-using webrtc::DataBuffer;
-using webrtc::DataChannelInterface;
-using webrtc::DtmfSender;
-using webrtc::DtmfSenderInterface;
-using webrtc::DtmfSenderObserverInterface;
-using webrtc::FakeConstraints;
-using webrtc::MediaConstraintsInterface;
-using webrtc::MediaStreamInterface;
-using webrtc::MediaStreamTrackInterface;
-using webrtc::MockCreateSessionDescriptionObserver;
-using webrtc::MockDataChannelObserver;
-using webrtc::MockSetSessionDescriptionObserver;
-using webrtc::MockStatsObserver;
-using webrtc::ObserverInterface;
-using webrtc::PeerConnectionInterface;
-using webrtc::PeerConnectionFactory;
-using webrtc::SessionDescriptionInterface;
-using webrtc::StreamCollectionInterface;
-
-namespace {
-
-static const int kMaxWaitMs = 10000;
-// Disable for TSan v2, see
-// https://code.google.com/p/webrtc/issues/detail?id=1205 for details.
-// This declaration is also #ifdef'd as it causes uninitialized-variable
-// warnings.
-#if !defined(THREAD_SANITIZER)
-static const int kMaxWaitForStatsMs = 3000;
-#endif
-static const int kMaxWaitForActivationMs = 5000;
-static const int kMaxWaitForFramesMs = 10000;
-static const int kEndAudioFrameCount = 3;
-static const int kEndVideoFrameCount = 3;
-
-static const char kStreamLabelBase[] = "stream_label";
-static const char kVideoTrackLabelBase[] = "video_track";
-static const char kAudioTrackLabelBase[] = "audio_track";
-static const char kDataChannelLabel[] = "data_channel";
-
-// Disable for TSan v2, see
-// https://code.google.com/p/webrtc/issues/detail?id=1205 for details.
-// This declaration is also #ifdef'd as it causes unused-variable errors.
-#if !defined(THREAD_SANITIZER)
-// SRTP cipher name negotiated by the tests. This must be updated if the
-// default changes.
-static const int kDefaultSrtpCryptoSuite = rtc::SRTP_AES128_CM_SHA1_32;
-static const int kDefaultSrtpCryptoSuiteGcm = rtc::SRTP_AEAD_AES_256_GCM;
-#endif
-
-// Used to simulate signaling ICE/SDP between two PeerConnections.
-enum Message { MSG_SDP_MESSAGE, MSG_ICE_MESSAGE };
-
-struct SdpMessage {
-  std::string type;
-  std::string msg;
-};
-
-struct IceMessage {
-  std::string sdp_mid;
-  int sdp_mline_index;
-  std::string msg;
-};
-
-static void RemoveLinesFromSdp(const std::string& line_start,
-                               std::string* sdp) {
-  const char kSdpLineEnd[] = "\r\n";
-  size_t ssrc_pos = 0;
-  while ((ssrc_pos = sdp->find(line_start, ssrc_pos)) !=
-      std::string::npos) {
-    size_t end_ssrc = sdp->find(kSdpLineEnd, ssrc_pos);
-    sdp->erase(ssrc_pos, end_ssrc - ssrc_pos + strlen(kSdpLineEnd));
-  }
-}
-
-bool StreamsHaveAudioTrack(StreamCollectionInterface* streams) {
-  for (size_t idx = 0; idx < streams->count(); idx++) {
-    auto stream = streams->at(idx);
-    if (stream->GetAudioTracks().size() > 0) {
-      return true;
-    }
-  }
-  return false;
-}
-
-bool StreamsHaveVideoTrack(StreamCollectionInterface* streams) {
-  for (size_t idx = 0; idx < streams->count(); idx++) {
-    auto stream = streams->at(idx);
-    if (stream->GetVideoTracks().size() > 0) {
-      return true;
-    }
-  }
-  return false;
-}
-
-class SignalingMessageReceiver {
- public:
-  virtual void ReceiveSdpMessage(const std::string& type,
-                                 std::string& msg) = 0;
-  virtual void ReceiveIceMessage(const std::string& sdp_mid,
-                                 int sdp_mline_index,
-                                 const std::string& msg) = 0;
-
- protected:
-  SignalingMessageReceiver() {}
-  virtual ~SignalingMessageReceiver() {}
-};
-
-class MockRtpReceiverObserver : public webrtc::RtpReceiverObserverInterface {
- public:
-  MockRtpReceiverObserver(cricket::MediaType media_type)
-      : expected_media_type_(media_type) {}
-
-  void OnFirstPacketReceived(cricket::MediaType media_type) override {
-    ASSERT_EQ(expected_media_type_, media_type);
-    first_packet_received_ = true;
-  }
-
-  bool first_packet_received() { return first_packet_received_; }
-
-  virtual ~MockRtpReceiverObserver() {}
-
- private:
-  bool first_packet_received_ = false;
-  cricket::MediaType expected_media_type_;
-};
-
-class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
-                                 public SignalingMessageReceiver,
-                                 public ObserverInterface,
-                                 public rtc::MessageHandler {
- public:
-  // We need these using declarations because there are two versions of each of
-  // the below methods and we only override one of them.
-  // TODO(deadbeef): Remove once there's only one version of the methods.
-  using PeerConnectionObserver::OnAddStream;
-  using PeerConnectionObserver::OnRemoveStream;
-  using PeerConnectionObserver::OnDataChannel;
-
-  // If |config| is not provided, uses a default constructed RTCConfiguration.
-  static PeerConnectionTestClient* CreateClientWithDtlsIdentityStore(
-      const std::string& id,
-      const MediaConstraintsInterface* constraints,
-      const PeerConnectionFactory::Options* options,
-      const PeerConnectionInterface::RTCConfiguration* config,
-      std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
-      bool prefer_constraint_apis,
-      rtc::Thread* network_thread,
-      rtc::Thread* worker_thread) {
-    PeerConnectionTestClient* client(new PeerConnectionTestClient(id));
-    if (!client->Init(constraints, options, config, std::move(cert_generator),
-                      prefer_constraint_apis, network_thread, worker_thread)) {
-      delete client;
-      return nullptr;
-    }
-    return client;
-  }
-
-  static PeerConnectionTestClient* CreateClient(
-      const std::string& id,
-      const MediaConstraintsInterface* constraints,
-      const PeerConnectionFactory::Options* options,
-      const PeerConnectionInterface::RTCConfiguration* config,
-      rtc::Thread* network_thread,
-      rtc::Thread* worker_thread) {
-    std::unique_ptr<FakeRTCCertificateGenerator> cert_generator(
-        rtc::SSLStreamAdapter::HaveDtlsSrtp() ?
-            new FakeRTCCertificateGenerator() : nullptr);
-
-    return CreateClientWithDtlsIdentityStore(id, constraints, options, config,
-                                             std::move(cert_generator), true,
-                                             network_thread, worker_thread);
-  }
-
-  static PeerConnectionTestClient* CreateClientPreferNoConstraints(
-      const std::string& id,
-      const PeerConnectionFactory::Options* options,
-      rtc::Thread* network_thread,
-      rtc::Thread* worker_thread) {
-    std::unique_ptr<FakeRTCCertificateGenerator> cert_generator(
-        rtc::SSLStreamAdapter::HaveDtlsSrtp() ?
-            new FakeRTCCertificateGenerator() : nullptr);
-
-    return CreateClientWithDtlsIdentityStore(id, nullptr, options, nullptr,
-                                             std::move(cert_generator), false,
-                                             network_thread, worker_thread);
-  }
-
-  ~PeerConnectionTestClient() {
-  }
-
-  void Negotiate() { Negotiate(true, true); }
-
-  void Negotiate(bool audio, bool video) {
-    std::unique_ptr<SessionDescriptionInterface> offer;
-    ASSERT_TRUE(DoCreateOffer(&offer));
-
-    if (offer->description()->GetContentByName("audio")) {
-      offer->description()->GetContentByName("audio")->rejected = !audio;
-    }
-    if (offer->description()->GetContentByName("video")) {
-      offer->description()->GetContentByName("video")->rejected = !video;
-    }
-
-    std::string sdp;
-    EXPECT_TRUE(offer->ToString(&sdp));
-    EXPECT_TRUE(DoSetLocalDescription(offer.release()));
-    SendSdpMessage(webrtc::SessionDescriptionInterface::kOffer, sdp);
-  }
-
-  void SendSdpMessage(const std::string& type, std::string& msg) {
-    if (signaling_delay_ms_ == 0) {
-      if (signaling_message_receiver_) {
-        signaling_message_receiver_->ReceiveSdpMessage(type, msg);
-      }
-    } else {
-      rtc::Thread::Current()->PostDelayed(
-          RTC_FROM_HERE, signaling_delay_ms_, this, MSG_SDP_MESSAGE,
-          new rtc::TypedMessageData<SdpMessage>({type, msg}));
-    }
-  }
-
-  void SendIceMessage(const std::string& sdp_mid,
-                      int sdp_mline_index,
-                      const std::string& msg) {
-    if (signaling_delay_ms_ == 0) {
-      if (signaling_message_receiver_) {
-        signaling_message_receiver_->ReceiveIceMessage(sdp_mid, sdp_mline_index,
-                                                       msg);
-      }
-    } else {
-      rtc::Thread::Current()->PostDelayed(RTC_FROM_HERE, signaling_delay_ms_,
-                                          this, MSG_ICE_MESSAGE,
-                                          new rtc::TypedMessageData<IceMessage>(
-                                              {sdp_mid, sdp_mline_index, msg}));
-    }
-  }
-
-  // MessageHandler callback.
-  void OnMessage(rtc::Message* msg) override {
-    switch (msg->message_id) {
-      case MSG_SDP_MESSAGE: {
-        auto sdp_message =
-            static_cast<rtc::TypedMessageData<SdpMessage>*>(msg->pdata);
-        if (signaling_message_receiver_) {
-          signaling_message_receiver_->ReceiveSdpMessage(
-              sdp_message->data().type, sdp_message->data().msg);
-        }
-        delete sdp_message;
-        break;
-      }
-      case MSG_ICE_MESSAGE: {
-        auto ice_message =
-            static_cast<rtc::TypedMessageData<IceMessage>*>(msg->pdata);
-        if (signaling_message_receiver_) {
-          signaling_message_receiver_->ReceiveIceMessage(
-              ice_message->data().sdp_mid, ice_message->data().sdp_mline_index,
-              ice_message->data().msg);
-        }
-        delete ice_message;
-        break;
-      }
-      default:
-        RTC_CHECK(false);
-    }
-  }
-
-  // SignalingMessageReceiver callback.
-  void ReceiveSdpMessage(const std::string& type, std::string& msg) override {
-    FilterIncomingSdpMessage(&msg);
-    if (type == webrtc::SessionDescriptionInterface::kOffer) {
-      HandleIncomingOffer(msg);
-    } else {
-      HandleIncomingAnswer(msg);
-    }
-  }
-
-  // SignalingMessageReceiver callback.
-  void ReceiveIceMessage(const std::string& sdp_mid,
-                         int sdp_mline_index,
-                         const std::string& msg) override {
-    LOG(INFO) << id_ << "ReceiveIceMessage";
-    std::unique_ptr<webrtc::IceCandidateInterface> candidate(
-        webrtc::CreateIceCandidate(sdp_mid, sdp_mline_index, msg, nullptr));
-    EXPECT_TRUE(pc()->AddIceCandidate(candidate.get()));
-  }
-
-  // PeerConnectionObserver callbacks.
-  void OnSignalingChange(
-      webrtc::PeerConnectionInterface::SignalingState new_state) override {
-    EXPECT_EQ(pc()->signaling_state(), new_state);
-  }
-  void OnAddStream(
-      rtc::scoped_refptr<MediaStreamInterface> media_stream) override {
-    media_stream->RegisterObserver(this);
-    for (size_t i = 0; i < media_stream->GetVideoTracks().size(); ++i) {
-      const std::string id = media_stream->GetVideoTracks()[i]->id();
-      ASSERT_TRUE(fake_video_renderers_.find(id) ==
-                  fake_video_renderers_.end());
-      fake_video_renderers_[id].reset(new webrtc::FakeVideoTrackRenderer(
-          media_stream->GetVideoTracks()[i]));
-    }
-  }
-  void OnRemoveStream(
-      rtc::scoped_refptr<MediaStreamInterface> media_stream) override {}
-  void OnRenegotiationNeeded() override {}
-  void OnIceConnectionChange(
-      webrtc::PeerConnectionInterface::IceConnectionState new_state) override {
-    EXPECT_EQ(pc()->ice_connection_state(), new_state);
-  }
-  void OnIceGatheringChange(
-      webrtc::PeerConnectionInterface::IceGatheringState new_state) override {
-    EXPECT_EQ(pc()->ice_gathering_state(), new_state);
-  }
-  void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override {
-    LOG(INFO) << id_ << "OnIceCandidate";
-
-    std::string ice_sdp;
-    EXPECT_TRUE(candidate->ToString(&ice_sdp));
-    if (signaling_message_receiver_ == nullptr) {
-      // Remote party may be deleted.
-      return;
-    }
-    SendIceMessage(candidate->sdp_mid(), candidate->sdp_mline_index(), ice_sdp);
-  }
-
-  // MediaStreamInterface callback
-  void OnChanged() override {
-    // Track added or removed from MediaStream, so update our renderers.
-    rtc::scoped_refptr<StreamCollectionInterface> remote_streams =
-        pc()->remote_streams();
-    // Remove renderers for tracks that were removed.
-    for (auto it = fake_video_renderers_.begin();
-         it != fake_video_renderers_.end();) {
-      if (remote_streams->FindVideoTrack(it->first) == nullptr) {
-        auto to_remove = it++;
-        removed_fake_video_renderers_.push_back(std::move(to_remove->second));
-        fake_video_renderers_.erase(to_remove);
-      } else {
-        ++it;
-      }
-    }
-    // Create renderers for new video tracks.
-    for (size_t stream_index = 0; stream_index < remote_streams->count();
-         ++stream_index) {
-      MediaStreamInterface* remote_stream = remote_streams->at(stream_index);
-      for (size_t track_index = 0;
-           track_index < remote_stream->GetVideoTracks().size();
-           ++track_index) {
-        const std::string id =
-            remote_stream->GetVideoTracks()[track_index]->id();
-        if (fake_video_renderers_.find(id) != fake_video_renderers_.end()) {
-          continue;
-        }
-        fake_video_renderers_[id].reset(new webrtc::FakeVideoTrackRenderer(
-            remote_stream->GetVideoTracks()[track_index]));
-      }
-    }
-  }
-
-  void SetVideoConstraints(const webrtc::FakeConstraints& video_constraint) {
-    video_constraints_ = video_constraint;
-  }
-
-  void AddMediaStream(bool audio, bool video) {
-    std::string stream_label =
-        kStreamLabelBase +
-        rtc::ToString<int>(static_cast<int>(pc()->local_streams()->count()));
-    rtc::scoped_refptr<MediaStreamInterface> stream =
-        peer_connection_factory_->CreateLocalMediaStream(stream_label);
-
-    if (audio && can_receive_audio()) {
-      stream->AddTrack(CreateLocalAudioTrack(stream_label));
-    }
-    if (video && can_receive_video()) {
-      stream->AddTrack(CreateLocalVideoTrack(stream_label));
-    }
-
-    EXPECT_TRUE(pc()->AddStream(stream));
-  }
-
-  size_t NumberOfLocalMediaStreams() { return pc()->local_streams()->count(); }
-
-  bool SessionActive() {
-    return pc()->signaling_state() == webrtc::PeerConnectionInterface::kStable;
-  }
-
-  // Automatically add a stream when receiving an offer, if we don't have one.
-  // Defaults to true.
-  void set_auto_add_stream(bool auto_add_stream) {
-    auto_add_stream_ = auto_add_stream;
-  }
-
-  void set_signaling_message_receiver(
-      SignalingMessageReceiver* signaling_message_receiver) {
-    signaling_message_receiver_ = signaling_message_receiver;
-  }
-
-  void set_signaling_delay_ms(int delay_ms) { signaling_delay_ms_ = delay_ms; }
-
-  void EnableVideoDecoderFactory() {
-    video_decoder_factory_enabled_ = true;
-    fake_video_decoder_factory_->AddSupportedVideoCodecType(
-        webrtc::kVideoCodecVP8);
-  }
-
-  void IceRestart() {
-    offer_answer_constraints_.SetMandatoryIceRestart(true);
-    offer_answer_options_.ice_restart = true;
-    SetExpectIceRestart(true);
-  }
-
-  void SetExpectIceRestart(bool expect_restart) {
-    expect_ice_restart_ = expect_restart;
-  }
-
-  bool ExpectIceRestart() const { return expect_ice_restart_; }
-
-  void SetExpectIceRenomination(bool expect_renomination) {
-    expect_ice_renomination_ = expect_renomination;
-  }
-  void SetExpectRemoteIceRenomination(bool expect_renomination) {
-    expect_remote_ice_renomination_ = expect_renomination;
-  }
-  bool ExpectIceRenomination() { return expect_ice_renomination_; }
-  bool ExpectRemoteIceRenomination() { return expect_remote_ice_renomination_; }
-
-  // The below 3 methods assume streams will be offered.
-  // Thus they'll only set the "offer to receive" flag to true if it's
-  // currently false, not if it's just unset.
-  void SetReceiveAudioVideo(bool audio, bool video) {
-    SetReceiveAudio(audio);
-    SetReceiveVideo(video);
-    ASSERT_EQ(audio, can_receive_audio());
-    ASSERT_EQ(video, can_receive_video());
-  }
-
-  void SetReceiveAudio(bool audio) {
-    if (audio && can_receive_audio()) {
-      return;
-    }
-    offer_answer_constraints_.SetMandatoryReceiveAudio(audio);
-    offer_answer_options_.offer_to_receive_audio = audio ? 1 : 0;
-  }
-
-  void SetReceiveVideo(bool video) {
-    if (video && can_receive_video()) {
-      return;
-    }
-    offer_answer_constraints_.SetMandatoryReceiveVideo(video);
-    offer_answer_options_.offer_to_receive_video = video ? 1 : 0;
-  }
-
-  void SetOfferToReceiveAudioVideo(bool audio, bool video) {
-    offer_answer_constraints_.SetMandatoryReceiveAudio(audio);
-    offer_answer_options_.offer_to_receive_audio = audio ? 1 : 0;
-    offer_answer_constraints_.SetMandatoryReceiveVideo(video);
-    offer_answer_options_.offer_to_receive_video = video ? 1 : 0;
-  }
-
-  void RemoveMsidFromReceivedSdp(bool remove) { remove_msid_ = remove; }
-
-  void RemoveSdesCryptoFromReceivedSdp(bool remove) { remove_sdes_ = remove; }
-
-  void RemoveBundleFromReceivedSdp(bool remove) { remove_bundle_ = remove; }
-
-  void RemoveCvoFromReceivedSdp(bool remove) { remove_cvo_ = remove; }
-
-  bool can_receive_audio() {
-    bool value;
-    if (prefer_constraint_apis_) {
-      if (webrtc::FindConstraint(
-              &offer_answer_constraints_,
-              MediaConstraintsInterface::kOfferToReceiveAudio, &value,
-              nullptr)) {
-        return value;
-      }
-      return true;
-    }
-    return offer_answer_options_.offer_to_receive_audio > 0 ||
-           offer_answer_options_.offer_to_receive_audio ==
-               PeerConnectionInterface::RTCOfferAnswerOptions::kUndefined;
-  }
-
-  bool can_receive_video() {
-    bool value;
-    if (prefer_constraint_apis_) {
-      if (webrtc::FindConstraint(
-              &offer_answer_constraints_,
-              MediaConstraintsInterface::kOfferToReceiveVideo, &value,
-              nullptr)) {
-        return value;
-      }
-      return true;
-    }
-    return offer_answer_options_.offer_to_receive_video > 0 ||
-           offer_answer_options_.offer_to_receive_video ==
-               PeerConnectionInterface::RTCOfferAnswerOptions::kUndefined;
-  }
-
-  void OnDataChannel(
-      rtc::scoped_refptr<DataChannelInterface> data_channel) override {
-    LOG(INFO) << id_ << "OnDataChannel";
-    data_channel_ = data_channel;
-    data_observer_.reset(new MockDataChannelObserver(data_channel));
-  }
-
-  void CreateDataChannel() { CreateDataChannel(nullptr); }
-
-  void CreateDataChannel(const webrtc::DataChannelInit* init) {
-    data_channel_ = pc()->CreateDataChannel(kDataChannelLabel, init);
-    ASSERT_TRUE(data_channel_.get() != nullptr);
-    data_observer_.reset(new MockDataChannelObserver(data_channel_));
-  }
-
-  rtc::scoped_refptr<webrtc::AudioTrackInterface> CreateLocalAudioTrack(
-      const std::string& stream_label) {
-    FakeConstraints constraints;
-    // Disable highpass filter so that we can get all the test audio frames.
-    constraints.AddMandatory(MediaConstraintsInterface::kHighpassFilter, false);
-    rtc::scoped_refptr<webrtc::AudioSourceInterface> source =
-        peer_connection_factory_->CreateAudioSource(&constraints);
-    // TODO(perkj): Test audio source when it is implemented. Currently audio
-    // always use the default input.
-    std::string label = stream_label + kAudioTrackLabelBase;
-    return peer_connection_factory_->CreateAudioTrack(label, source);
-  }
-
-  rtc::scoped_refptr<webrtc::VideoTrackInterface> CreateLocalVideoTrack(
-      const std::string& stream_label) {
-    // Set max frame rate to 10fps to reduce the risk of the tests to be flaky.
-    FakeConstraints source_constraints = video_constraints_;
-    source_constraints.SetMandatoryMaxFrameRate(10);
-
-    cricket::FakeVideoCapturer* fake_capturer =
-        new webrtc::FakePeriodicVideoCapturer();
-    fake_capturer->SetRotation(capture_rotation_);
-    video_capturers_.push_back(fake_capturer);
-    rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> source =
-        peer_connection_factory_->CreateVideoSource(fake_capturer,
-                                                    &source_constraints);
-    std::string label = stream_label + kVideoTrackLabelBase;
-
-    rtc::scoped_refptr<webrtc::VideoTrackInterface> track(
-        peer_connection_factory_->CreateVideoTrack(label, source));
-    if (!local_video_renderer_) {
-      local_video_renderer_.reset(new webrtc::FakeVideoTrackRenderer(track));
-    }
-    return track;
-  }
-
-  DataChannelInterface* data_channel() { return data_channel_; }
-  const MockDataChannelObserver* data_observer() const {
-    return data_observer_.get();
-  }
-
-  webrtc::PeerConnectionInterface* pc() const { return peer_connection_.get(); }
-
-  void StopVideoCapturers() {
-    for (auto* capturer : video_capturers_) {
-      capturer->Stop();
-    }
-  }
-
-  void SetCaptureRotation(webrtc::VideoRotation rotation) {
-    ASSERT_TRUE(video_capturers_.empty());
-    capture_rotation_ = rotation;
-  }
-
-  bool AudioFramesReceivedCheck(int number_of_frames) const {
-    return number_of_frames <= fake_audio_capture_module_->frames_received();
-  }
-
-  int audio_frames_received() const {
-    return fake_audio_capture_module_->frames_received();
-  }
-
-  bool VideoFramesReceivedCheck(int number_of_frames) {
-    if (video_decoder_factory_enabled_) {
-      const std::vector<FakeWebRtcVideoDecoder*>& decoders
-          = fake_video_decoder_factory_->decoders();
-      if (decoders.empty()) {
-        return number_of_frames <= 0;
-      }
-      // Note - this checks that EACH decoder has the requisite number
-      // of frames. The video_frames_received() function sums them.
-      for (FakeWebRtcVideoDecoder* decoder : decoders) {
-        if (number_of_frames > decoder->GetNumFramesReceived()) {
-          return false;
-        }
-      }
-      return true;
-    } else {
-      if (fake_video_renderers_.empty()) {
-        return number_of_frames <= 0;
-      }
-
-      for (const auto& pair : fake_video_renderers_) {
-        if (number_of_frames > pair.second->num_rendered_frames()) {
-          return false;
-        }
-      }
-      return true;
-    }
-  }
-
-  int video_frames_received() const {
-    int total = 0;
-    if (video_decoder_factory_enabled_) {
-      const std::vector<FakeWebRtcVideoDecoder*>& decoders =
-          fake_video_decoder_factory_->decoders();
-      for (const FakeWebRtcVideoDecoder* decoder : decoders) {
-        total += decoder->GetNumFramesReceived();
-      }
-    } else {
-      for (const auto& pair : fake_video_renderers_) {
-        total += pair.second->num_rendered_frames();
-      }
-      for (const auto& renderer : removed_fake_video_renderers_) {
-        total += renderer->num_rendered_frames();
-      }
-    }
-    return total;
-  }
-
-  // Verify the CreateDtmfSender interface
-  void VerifyDtmf() {
-    std::unique_ptr<DummyDtmfObserver> observer(new DummyDtmfObserver());
-    rtc::scoped_refptr<DtmfSenderInterface> dtmf_sender;
-
-    // We can't create a DTMF sender with an invalid audio track or a non local
-    // track.
-    EXPECT_TRUE(peer_connection_->CreateDtmfSender(nullptr) == nullptr);
-    rtc::scoped_refptr<webrtc::AudioTrackInterface> non_localtrack(
-        peer_connection_factory_->CreateAudioTrack("dummy_track", nullptr));
-    EXPECT_TRUE(peer_connection_->CreateDtmfSender(non_localtrack) == nullptr);
-
-    // We should be able to create a DTMF sender from a local track.
-    webrtc::AudioTrackInterface* localtrack =
-        peer_connection_->local_streams()->at(0)->GetAudioTracks()[0];
-    dtmf_sender = peer_connection_->CreateDtmfSender(localtrack);
-    EXPECT_TRUE(dtmf_sender.get() != nullptr);
-    dtmf_sender->RegisterObserver(observer.get());
-
-    // Test the DtmfSender object just created.
-    EXPECT_TRUE(dtmf_sender->CanInsertDtmf());
-    EXPECT_TRUE(dtmf_sender->InsertDtmf("1a", 100, 50));
-
-    // We don't need to verify that the DTMF tones are actually sent out because
-    // that is already covered by the tests of the lower level components.
-
-    EXPECT_TRUE_WAIT(observer->completed(), kMaxWaitMs);
-    std::vector<std::string> tones;
-    tones.push_back("1");
-    tones.push_back("a");
-    tones.push_back("");
-    observer->Verify(tones);
-
-    dtmf_sender->UnregisterObserver();
-  }
-
-  // Verifies that the SessionDescription have rejected the appropriate media
-  // content.
-  void VerifyRejectedMediaInSessionDescription() {
-    ASSERT_TRUE(peer_connection_->remote_description() != nullptr);
-    ASSERT_TRUE(peer_connection_->local_description() != nullptr);
-    const cricket::SessionDescription* remote_desc =
-        peer_connection_->remote_description()->description();
-    const cricket::SessionDescription* local_desc =
-        peer_connection_->local_description()->description();
-
-    const ContentInfo* remote_audio_content = GetFirstAudioContent(remote_desc);
-    if (remote_audio_content) {
-      const ContentInfo* audio_content =
-          GetFirstAudioContent(local_desc);
-      EXPECT_EQ(can_receive_audio(), !audio_content->rejected);
-    }
-
-    const ContentInfo* remote_video_content = GetFirstVideoContent(remote_desc);
-    if (remote_video_content) {
-      const ContentInfo* video_content =
-          GetFirstVideoContent(local_desc);
-      EXPECT_EQ(can_receive_video(), !video_content->rejected);
-    }
-  }
-
-  void VerifyLocalIceUfragAndPassword() {
-    ASSERT_TRUE(peer_connection_->local_description() != nullptr);
-    const cricket::SessionDescription* desc =
-        peer_connection_->local_description()->description();
-    const cricket::ContentInfos& contents = desc->contents();
-
-    for (size_t index = 0; index < contents.size(); ++index) {
-      if (contents[index].rejected)
-        continue;
-      const cricket::TransportDescription* transport_desc =
-          desc->GetTransportDescriptionByName(contents[index].name);
-
-      std::map<int, IceUfragPwdPair>::const_iterator ufragpair_it =
-          ice_ufrag_pwd_.find(static_cast<int>(index));
-      if (ufragpair_it == ice_ufrag_pwd_.end()) {
-        ASSERT_FALSE(ExpectIceRestart());
-        ice_ufrag_pwd_[static_cast<int>(index)] =
-            IceUfragPwdPair(transport_desc->ice_ufrag, transport_desc->ice_pwd);
-      } else if (ExpectIceRestart()) {
-        const IceUfragPwdPair& ufrag_pwd = ufragpair_it->second;
-        EXPECT_NE(ufrag_pwd.first, transport_desc->ice_ufrag);
-        EXPECT_NE(ufrag_pwd.second, transport_desc->ice_pwd);
-      } else {
-        const IceUfragPwdPair& ufrag_pwd = ufragpair_it->second;
-        EXPECT_EQ(ufrag_pwd.first, transport_desc->ice_ufrag);
-        EXPECT_EQ(ufrag_pwd.second, transport_desc->ice_pwd);
-      }
-    }
-  }
-
-  void VerifyLocalIceRenomination() {
-    ASSERT_TRUE(peer_connection_->local_description() != nullptr);
-    const cricket::SessionDescription* desc =
-        peer_connection_->local_description()->description();
-    const cricket::ContentInfos& contents = desc->contents();
-
-    for (auto content : contents) {
-      if (content.rejected)
-        continue;
-      const cricket::TransportDescription* transport_desc =
-          desc->GetTransportDescriptionByName(content.name);
-      const auto& options = transport_desc->transport_options;
-      auto iter = std::find(options.begin(), options.end(),
-                            cricket::ICE_RENOMINATION_STR);
-      EXPECT_EQ(ExpectIceRenomination(), iter != options.end());
-    }
-  }
-
-  void VerifyRemoteIceRenomination() {
-    ASSERT_TRUE(peer_connection_->remote_description() != nullptr);
-    const cricket::SessionDescription* desc =
-        peer_connection_->remote_description()->description();
-    const cricket::ContentInfos& contents = desc->contents();
-
-    for (auto content : contents) {
-      if (content.rejected)
-        continue;
-      const cricket::TransportDescription* transport_desc =
-          desc->GetTransportDescriptionByName(content.name);
-      const auto& options = transport_desc->transport_options;
-      auto iter = std::find(options.begin(), options.end(),
-                            cricket::ICE_RENOMINATION_STR);
-      EXPECT_EQ(ExpectRemoteIceRenomination(), iter != options.end());
-    }
-  }
-
-  int GetAudioOutputLevelStats(webrtc::MediaStreamTrackInterface* track) {
-    rtc::scoped_refptr<MockStatsObserver>
-        observer(new rtc::RefCountedObject<MockStatsObserver>());
-    EXPECT_TRUE(peer_connection_->GetStats(
-        observer, track, PeerConnectionInterface::kStatsOutputLevelStandard));
-    EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs);
-    EXPECT_NE(0, observer->timestamp());
-    return observer->AudioOutputLevel();
-  }
-
-  int GetAudioInputLevelStats() {
-    rtc::scoped_refptr<MockStatsObserver>
-        observer(new rtc::RefCountedObject<MockStatsObserver>());
-    EXPECT_TRUE(peer_connection_->GetStats(
-        observer, nullptr, PeerConnectionInterface::kStatsOutputLevelStandard));
-    EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs);
-    EXPECT_NE(0, observer->timestamp());
-    return observer->AudioInputLevel();
-  }
-
-  int GetBytesReceivedStats(webrtc::MediaStreamTrackInterface* track) {
-    rtc::scoped_refptr<MockStatsObserver>
-    observer(new rtc::RefCountedObject<MockStatsObserver>());
-    EXPECT_TRUE(peer_connection_->GetStats(
-        observer, track, PeerConnectionInterface::kStatsOutputLevelStandard));
-    EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs);
-    EXPECT_NE(0, observer->timestamp());
-    return observer->BytesReceived();
-  }
-
-  int GetBytesSentStats(webrtc::MediaStreamTrackInterface* track) {
-    rtc::scoped_refptr<MockStatsObserver>
-    observer(new rtc::RefCountedObject<MockStatsObserver>());
-    EXPECT_TRUE(peer_connection_->GetStats(
-        observer, track, PeerConnectionInterface::kStatsOutputLevelStandard));
-    EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs);
-    EXPECT_NE(0, observer->timestamp());
-    return observer->BytesSent();
-  }
-
-  int GetAvailableReceivedBandwidthStats() {
-    rtc::scoped_refptr<MockStatsObserver>
-        observer(new rtc::RefCountedObject<MockStatsObserver>());
-    EXPECT_TRUE(peer_connection_->GetStats(
-        observer, nullptr, PeerConnectionInterface::kStatsOutputLevelStandard));
-    EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs);
-    EXPECT_NE(0, observer->timestamp());
-    int bw = observer->AvailableReceiveBandwidth();
-    return bw;
-  }
-
-  std::string GetDtlsCipherStats() {
-    rtc::scoped_refptr<MockStatsObserver>
-        observer(new rtc::RefCountedObject<MockStatsObserver>());
-    EXPECT_TRUE(peer_connection_->GetStats(
-        observer, nullptr, PeerConnectionInterface::kStatsOutputLevelStandard));
-    EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs);
-    EXPECT_NE(0, observer->timestamp());
-    return observer->DtlsCipher();
-  }
-
-  std::string GetSrtpCipherStats() {
-    rtc::scoped_refptr<MockStatsObserver>
-        observer(new rtc::RefCountedObject<MockStatsObserver>());
-    EXPECT_TRUE(peer_connection_->GetStats(
-        observer, nullptr, PeerConnectionInterface::kStatsOutputLevelStandard));
-    EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs);
-    EXPECT_NE(0, observer->timestamp());
-    return observer->SrtpCipher();
-  }
-
-  int rendered_width() {
-    EXPECT_FALSE(fake_video_renderers_.empty());
-    return fake_video_renderers_.empty() ? 1 :
-        fake_video_renderers_.begin()->second->width();
-  }
-
-  int rendered_height() {
-    EXPECT_FALSE(fake_video_renderers_.empty());
-    return fake_video_renderers_.empty() ? 1 :
-        fake_video_renderers_.begin()->second->height();
-  }
-
-  webrtc::VideoRotation rendered_rotation() {
-    EXPECT_FALSE(fake_video_renderers_.empty());
-    return fake_video_renderers_.empty()
-               ? webrtc::kVideoRotation_0
-               : fake_video_renderers_.begin()->second->rotation();
-  }
-
-  int local_rendered_width() {
-    return local_video_renderer_ ? local_video_renderer_->width() : 1;
-  }
-
-  int local_rendered_height() {
-    return local_video_renderer_ ? local_video_renderer_->height() : 1;
-  }
-
-  size_t number_of_remote_streams() {
-    if (!pc())
-      return 0;
-    return pc()->remote_streams()->count();
-  }
-
-  StreamCollectionInterface* remote_streams() const {
-    if (!pc()) {
-      ADD_FAILURE();
-      return nullptr;
-    }
-    return pc()->remote_streams();
-  }
-
-  StreamCollectionInterface* local_streams() {
-    if (!pc()) {
-      ADD_FAILURE();
-      return nullptr;
-    }
-    return pc()->local_streams();
-  }
-
-  bool HasLocalAudioTrack() { return StreamsHaveAudioTrack(local_streams()); }
-
-  bool HasLocalVideoTrack() { return StreamsHaveVideoTrack(local_streams()); }
-
-  webrtc::PeerConnectionInterface::SignalingState signaling_state() {
-    return pc()->signaling_state();
-  }
-
-  webrtc::PeerConnectionInterface::IceConnectionState ice_connection_state() {
-    return pc()->ice_connection_state();
-  }
-
-  webrtc::PeerConnectionInterface::IceGatheringState ice_gathering_state() {
-    return pc()->ice_gathering_state();
-  }
-
-  std::vector<std::unique_ptr<MockRtpReceiverObserver>> const&
-  rtp_receiver_observers() {
-    return rtp_receiver_observers_;
-  }
-
-  void SetRtpReceiverObservers() {
-    rtp_receiver_observers_.clear();
-    for (auto receiver : pc()->GetReceivers()) {
-      std::unique_ptr<MockRtpReceiverObserver> observer(
-          new MockRtpReceiverObserver(receiver->media_type()));
-      receiver->SetObserver(observer.get());
-      rtp_receiver_observers_.push_back(std::move(observer));
-    }
-  }
-
- private:
-  class DummyDtmfObserver : public DtmfSenderObserverInterface {
-   public:
-    DummyDtmfObserver() : completed_(false) {}
-
-    // Implements DtmfSenderObserverInterface.
-    void OnToneChange(const std::string& tone) override {
-      tones_.push_back(tone);
-      if (tone.empty()) {
-        completed_ = true;
-      }
-    }
-
-    void Verify(const std::vector<std::string>& tones) const {
-      ASSERT_TRUE(tones_.size() == tones.size());
-      EXPECT_TRUE(std::equal(tones.begin(), tones.end(), tones_.begin()));
-    }
-
-    bool completed() const { return completed_; }
-
-   private:
-    bool completed_;
-    std::vector<std::string> tones_;
-  };
-
-  explicit PeerConnectionTestClient(const std::string& id) : id_(id) {}
-
-  bool Init(
-      const MediaConstraintsInterface* constraints,
-      const PeerConnectionFactory::Options* options,
-      const PeerConnectionInterface::RTCConfiguration* config,
-      std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
-      bool prefer_constraint_apis,
-      rtc::Thread* network_thread,
-      rtc::Thread* worker_thread) {
-    EXPECT_TRUE(!peer_connection_);
-    EXPECT_TRUE(!peer_connection_factory_);
-    if (!prefer_constraint_apis) {
-      EXPECT_TRUE(!constraints);
-    }
-    prefer_constraint_apis_ = prefer_constraint_apis;
-
-    fake_network_manager_.reset(new rtc::FakeNetworkManager());
-    fake_network_manager_->AddInterface(rtc::SocketAddress("192.168.1.1", 0));
-
-    std::unique_ptr<cricket::PortAllocator> port_allocator(
-        new cricket::BasicPortAllocator(fake_network_manager_.get()));
-    fake_audio_capture_module_ = FakeAudioCaptureModule::Create();
-
-    if (fake_audio_capture_module_ == nullptr) {
-      return false;
-    }
-    fake_video_decoder_factory_ = new FakeWebRtcVideoDecoderFactory();
-    fake_video_encoder_factory_ = new FakeWebRtcVideoEncoderFactory();
-    rtc::Thread* const signaling_thread = rtc::Thread::Current();
-    peer_connection_factory_ = webrtc::CreatePeerConnectionFactory(
-        network_thread, worker_thread, signaling_thread,
-        fake_audio_capture_module_, fake_video_encoder_factory_,
-        fake_video_decoder_factory_);
-    if (!peer_connection_factory_) {
-      return false;
-    }
-    if (options) {
-      peer_connection_factory_->SetOptions(*options);
-    }
-    peer_connection_ =
-        CreatePeerConnection(std::move(port_allocator), constraints, config,
-                             std::move(cert_generator));
-    return peer_connection_.get() != nullptr;
-  }
-
-  rtc::scoped_refptr<webrtc::PeerConnectionInterface> CreatePeerConnection(
-      std::unique_ptr<cricket::PortAllocator> port_allocator,
-      const MediaConstraintsInterface* constraints,
-      const PeerConnectionInterface::RTCConfiguration* config,
-      std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator) {
-    // CreatePeerConnection with RTCConfiguration.
-    PeerConnectionInterface::RTCConfiguration default_config;
-
-    if (!config) {
-      config = &default_config;
-    }
-
-    return peer_connection_factory_->CreatePeerConnection(
-        *config, constraints, std::move(port_allocator),
-        std::move(cert_generator), this);
-  }
-
-  void HandleIncomingOffer(const std::string& msg) {
-    LOG(INFO) << id_ << "HandleIncomingOffer ";
-    if (NumberOfLocalMediaStreams() == 0 && auto_add_stream_) {
-      // If we are not sending any streams ourselves it is time to add some.
-      AddMediaStream(true, true);
-    }
-    std::unique_ptr<SessionDescriptionInterface> desc(
-        webrtc::CreateSessionDescription("offer", msg, nullptr));
-    EXPECT_TRUE(DoSetRemoteDescription(desc.release()));
-    // Set the RtpReceiverObserver after receivers are created.
-    SetRtpReceiverObservers();
-    std::unique_ptr<SessionDescriptionInterface> answer;
-    EXPECT_TRUE(DoCreateAnswer(&answer));
-    std::string sdp;
-    EXPECT_TRUE(answer->ToString(&sdp));
-    EXPECT_TRUE(DoSetLocalDescription(answer.release()));
-    SendSdpMessage(webrtc::SessionDescriptionInterface::kAnswer, sdp);
-  }
-
-  void HandleIncomingAnswer(const std::string& msg) {
-    LOG(INFO) << id_ << "HandleIncomingAnswer";
-    std::unique_ptr<SessionDescriptionInterface> desc(
-        webrtc::CreateSessionDescription("answer", msg, nullptr));
-    EXPECT_TRUE(DoSetRemoteDescription(desc.release()));
-    // Set the RtpReceiverObserver after receivers are created.
-    SetRtpReceiverObservers();
-  }
-
-  bool DoCreateOfferAnswer(std::unique_ptr<SessionDescriptionInterface>* desc,
-                           bool offer) {
-    rtc::scoped_refptr<MockCreateSessionDescriptionObserver>
-        observer(new rtc::RefCountedObject<
-            MockCreateSessionDescriptionObserver>());
-    if (prefer_constraint_apis_) {
-      if (offer) {
-        pc()->CreateOffer(observer, &offer_answer_constraints_);
-      } else {
-        pc()->CreateAnswer(observer, &offer_answer_constraints_);
-      }
-    } else {
-      if (offer) {
-        pc()->CreateOffer(observer, offer_answer_options_);
-      } else {
-        pc()->CreateAnswer(observer, offer_answer_options_);
-      }
-    }
-    EXPECT_EQ_WAIT(true, observer->called(), kMaxWaitMs);
-    desc->reset(observer->release_desc());
-    if (observer->result() && ExpectIceRestart()) {
-      EXPECT_EQ(0u, (*desc)->candidates(0)->count());
-    }
-    return observer->result();
-  }
-
-  bool DoCreateOffer(std::unique_ptr<SessionDescriptionInterface>* desc) {
-    return DoCreateOfferAnswer(desc, true);
-  }
-
-  bool DoCreateAnswer(std::unique_ptr<SessionDescriptionInterface>* desc) {
-    return DoCreateOfferAnswer(desc, false);
-  }
-
-  bool DoSetLocalDescription(SessionDescriptionInterface* desc) {
-    rtc::scoped_refptr<MockSetSessionDescriptionObserver>
-            observer(new rtc::RefCountedObject<
-                MockSetSessionDescriptionObserver>());
-    LOG(INFO) << id_ << "SetLocalDescription ";
-    pc()->SetLocalDescription(observer, desc);
-    // Ignore the observer result. If we wait for the result with
-    // EXPECT_TRUE_WAIT, local ice candidates might be sent to the remote peer
-    // before the offer which is an error.
-    // The reason is that EXPECT_TRUE_WAIT uses
-    // rtc::Thread::Current()->ProcessMessages(1);
-    // ProcessMessages waits at least 1ms but processes all messages before
-    // returning. Since this test is synchronous and send messages to the remote
-    // peer whenever a callback is invoked, this can lead to messages being
-    // sent to the remote peer in the wrong order.
-    // TODO(perkj): Find a way to check the result without risking that the
-    // order of sent messages are changed. Ex- by posting all messages that are
-    // sent to the remote peer.
-    return true;
-  }
-
-  bool DoSetRemoteDescription(SessionDescriptionInterface* desc) {
-    rtc::scoped_refptr<MockSetSessionDescriptionObserver>
-        observer(new rtc::RefCountedObject<
-            MockSetSessionDescriptionObserver>());
-    LOG(INFO) << id_ << "SetRemoteDescription ";
-    pc()->SetRemoteDescription(observer, desc);
-    EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs);
-    return observer->result();
-  }
-
-  // This modifies all received SDP messages before they are processed.
-  void FilterIncomingSdpMessage(std::string* sdp) {
-    if (remove_msid_) {
-      const char kSdpSsrcAttribute[] = "a=ssrc:";
-      RemoveLinesFromSdp(kSdpSsrcAttribute, sdp);
-      const char kSdpMsidSupportedAttribute[] = "a=msid-semantic:";
-      RemoveLinesFromSdp(kSdpMsidSupportedAttribute, sdp);
-    }
-    if (remove_bundle_) {
-      const char kSdpBundleAttribute[] = "a=group:BUNDLE";
-      RemoveLinesFromSdp(kSdpBundleAttribute, sdp);
-    }
-    if (remove_sdes_) {
-      const char kSdpSdesCryptoAttribute[] = "a=crypto";
-      RemoveLinesFromSdp(kSdpSdesCryptoAttribute, sdp);
-    }
-    if (remove_cvo_) {
-      const char kSdpCvoExtenstion[] = "urn:3gpp:video-orientation";
-      RemoveLinesFromSdp(kSdpCvoExtenstion, sdp);
-    }
-  }
-
-  std::string id_;
-
-  std::unique_ptr<rtc::FakeNetworkManager> fake_network_manager_;
-
-  rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_;
-  rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface>
-      peer_connection_factory_;
-
-  bool prefer_constraint_apis_ = true;
-  bool auto_add_stream_ = true;
-
-  typedef std::pair<std::string, std::string> IceUfragPwdPair;
-  std::map<int, IceUfragPwdPair> ice_ufrag_pwd_;
-  bool expect_ice_restart_ = false;
-  bool expect_ice_renomination_ = false;
-  bool expect_remote_ice_renomination_ = false;
-
-  // Needed to keep track of number of frames sent.
-  rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_;
-  // Needed to keep track of number of frames received.
-  std::map<std::string, std::unique_ptr<webrtc::FakeVideoTrackRenderer>>
-      fake_video_renderers_;
-  // Needed to ensure frames aren't received for removed tracks.
-  std::vector<std::unique_ptr<webrtc::FakeVideoTrackRenderer>>
-      removed_fake_video_renderers_;
-  // Needed to keep track of number of frames received when external decoder
-  // used.
-  FakeWebRtcVideoDecoderFactory* fake_video_decoder_factory_ = nullptr;
-  FakeWebRtcVideoEncoderFactory* fake_video_encoder_factory_ = nullptr;
-  bool video_decoder_factory_enabled_ = false;
-  webrtc::FakeConstraints video_constraints_;
-
-  // For remote peer communication.
-  SignalingMessageReceiver* signaling_message_receiver_ = nullptr;
-  int signaling_delay_ms_ = 0;
-
-  // Store references to the video capturers we've created, so that we can stop
-  // them, if required.
-  std::vector<cricket::FakeVideoCapturer*> video_capturers_;
-  webrtc::VideoRotation capture_rotation_ = webrtc::kVideoRotation_0;
-  // |local_video_renderer_| attached to the first created local video track.
-  std::unique_ptr<webrtc::FakeVideoTrackRenderer> local_video_renderer_;
-
-  webrtc::FakeConstraints offer_answer_constraints_;
-  PeerConnectionInterface::RTCOfferAnswerOptions offer_answer_options_;
-  bool remove_msid_ = false;  // True if MSID should be removed in received SDP.
-  bool remove_bundle_ =
-      false;  // True if bundle should be removed in received SDP.
-  bool remove_sdes_ =
-      false;  // True if a=crypto should be removed in received SDP.
-  // |remove_cvo_| is true if extension urn:3gpp:video-orientation should be
-  // removed in the received SDP.
-  bool remove_cvo_ = false;
-
-  rtc::scoped_refptr<DataChannelInterface> data_channel_;
-  std::unique_ptr<MockDataChannelObserver> data_observer_;
-
-  std::vector<std::unique_ptr<MockRtpReceiverObserver>> rtp_receiver_observers_;
-};
-
-class P2PTestConductor : public testing::Test {
- public:
-  P2PTestConductor()
-      : pss_(new rtc::PhysicalSocketServer),
-        ss_(new rtc::VirtualSocketServer(pss_.get())),
-        network_thread_(new rtc::Thread(ss_.get())),
-        worker_thread_(rtc::Thread::Create()) {
-    RTC_CHECK(network_thread_->Start());
-    RTC_CHECK(worker_thread_->Start());
-  }
-
-  bool SessionActive() {
-    return initiating_client_->SessionActive() &&
-           receiving_client_->SessionActive();
-  }
-
-  // Return true if the number of frames provided have been received
-  // on the video and audio tracks provided.
-  bool FramesHaveArrived(int audio_frames_to_receive,
-                         int video_frames_to_receive) {
-    bool all_good = true;
-    if (initiating_client_->HasLocalAudioTrack() &&
-        receiving_client_->can_receive_audio()) {
-      all_good &=
-          receiving_client_->AudioFramesReceivedCheck(audio_frames_to_receive);
-    }
-    if (initiating_client_->HasLocalVideoTrack() &&
-        receiving_client_->can_receive_video()) {
-      all_good &=
-          receiving_client_->VideoFramesReceivedCheck(video_frames_to_receive);
-    }
-    if (receiving_client_->HasLocalAudioTrack() &&
-        initiating_client_->can_receive_audio()) {
-      all_good &=
-          initiating_client_->AudioFramesReceivedCheck(audio_frames_to_receive);
-    }
-    if (receiving_client_->HasLocalVideoTrack() &&
-        initiating_client_->can_receive_video()) {
-      all_good &=
-          initiating_client_->VideoFramesReceivedCheck(video_frames_to_receive);
-    }
-    return all_good;
-  }
-
-  void VerifyDtmf() {
-    initiating_client_->VerifyDtmf();
-    receiving_client_->VerifyDtmf();
-  }
-
-  void TestUpdateOfferWithRejectedContent() {
-    // Renegotiate, rejecting the video m-line.
-    initiating_client_->Negotiate(true, false);
-    ASSERT_TRUE_WAIT(SessionActive(), kMaxWaitForActivationMs);
-
-    int pc1_audio_received = initiating_client_->audio_frames_received();
-    int pc1_video_received = initiating_client_->video_frames_received();
-    int pc2_audio_received = receiving_client_->audio_frames_received();
-    int pc2_video_received = receiving_client_->video_frames_received();
-
-    // Wait for some additional audio frames to be received.
-    EXPECT_TRUE_WAIT(initiating_client_->AudioFramesReceivedCheck(
-                         pc1_audio_received + kEndAudioFrameCount) &&
-                         receiving_client_->AudioFramesReceivedCheck(
-                             pc2_audio_received + kEndAudioFrameCount),
-                     kMaxWaitForFramesMs);
-
-    // During this time, we shouldn't have received any additional video frames
-    // for the rejected video tracks.
-    EXPECT_EQ(pc1_video_received, initiating_client_->video_frames_received());
-    EXPECT_EQ(pc2_video_received, receiving_client_->video_frames_received());
-  }
-
-  void VerifyRenderedAspectRatio(int width, int height) {
-    VerifyRenderedAspectRatio(width, height, webrtc::kVideoRotation_0);
-  }
-
-  void VerifyRenderedAspectRatio(int width,
-                                 int height,
-                                 webrtc::VideoRotation rotation) {
-    double expected_aspect_ratio = static_cast<double>(width) / height;
-    double receiving_client_rendered_aspect_ratio =
-        static_cast<double>(receiving_client()->rendered_width()) /
-        receiving_client()->rendered_height();
-    double initializing_client_rendered_aspect_ratio =
-        static_cast<double>(initializing_client()->rendered_width()) /
-        initializing_client()->rendered_height();
-    double initializing_client_local_rendered_aspect_ratio =
-        static_cast<double>(initializing_client()->local_rendered_width()) /
-        initializing_client()->local_rendered_height();
-    // Verify end-to-end rendered aspect ratio.
-    EXPECT_EQ(expected_aspect_ratio, receiving_client_rendered_aspect_ratio);
-    EXPECT_EQ(expected_aspect_ratio, initializing_client_rendered_aspect_ratio);
-    // Verify aspect ratio of the local preview.
-    EXPECT_EQ(expected_aspect_ratio,
-              initializing_client_local_rendered_aspect_ratio);
-
-    // Verify rotation.
-    EXPECT_EQ(rotation, receiving_client()->rendered_rotation());
-    EXPECT_EQ(rotation, initializing_client()->rendered_rotation());
-  }
-
-  void VerifySessionDescriptions() {
-    initiating_client_->VerifyRejectedMediaInSessionDescription();
-    receiving_client_->VerifyRejectedMediaInSessionDescription();
-    initiating_client_->VerifyLocalIceUfragAndPassword();
-    receiving_client_->VerifyLocalIceUfragAndPassword();
-  }
-
-  ~P2PTestConductor() {
-    if (initiating_client_) {
-      initiating_client_->set_signaling_message_receiver(nullptr);
-    }
-    if (receiving_client_) {
-      receiving_client_->set_signaling_message_receiver(nullptr);
-    }
-  }
-
-  bool CreateTestClients() { return CreateTestClients(nullptr, nullptr); }
-
-  bool CreateTestClients(MediaConstraintsInterface* init_constraints,
-                         MediaConstraintsInterface* recv_constraints) {
-    return CreateTestClients(init_constraints, nullptr, nullptr,
-                             recv_constraints, nullptr, nullptr);
-  }
-
-  bool CreateTestClients(
-      const PeerConnectionInterface::RTCConfiguration& init_config,
-      const PeerConnectionInterface::RTCConfiguration& recv_config) {
-    return CreateTestClients(nullptr, nullptr, &init_config, nullptr, nullptr,
-                             &recv_config);
-  }
-
-  bool CreateTestClientsThatPreferNoConstraints() {
-    initiating_client_.reset(
-        PeerConnectionTestClient::CreateClientPreferNoConstraints(
-            "Caller: ", nullptr, network_thread_.get(), worker_thread_.get()));
-    receiving_client_.reset(
-        PeerConnectionTestClient::CreateClientPreferNoConstraints(
-            "Callee: ", nullptr, network_thread_.get(), worker_thread_.get()));
-    if (!initiating_client_ || !receiving_client_) {
-      return false;
-    }
-    // Remember the choice for possible later resets of the clients.
-    prefer_constraint_apis_ = false;
-    SetSignalingReceivers();
-    return true;
-  }
-
-  bool CreateTestClients(
-      MediaConstraintsInterface* init_constraints,
-      PeerConnectionFactory::Options* init_options,
-      const PeerConnectionInterface::RTCConfiguration* init_config,
-      MediaConstraintsInterface* recv_constraints,
-      PeerConnectionFactory::Options* recv_options,
-      const PeerConnectionInterface::RTCConfiguration* recv_config) {
-    initiating_client_.reset(PeerConnectionTestClient::CreateClient(
-        "Caller: ", init_constraints, init_options, init_config,
-        network_thread_.get(), worker_thread_.get()));
-    receiving_client_.reset(PeerConnectionTestClient::CreateClient(
-        "Callee: ", recv_constraints, recv_options, recv_config,
-        network_thread_.get(), worker_thread_.get()));
-    if (!initiating_client_ || !receiving_client_) {
-      return false;
-    }
-    SetSignalingReceivers();
-    return true;
-  }
-
-  void SetSignalingReceivers() {
-    initiating_client_->set_signaling_message_receiver(receiving_client_.get());
-    receiving_client_->set_signaling_message_receiver(initiating_client_.get());
-  }
-
-  void SetSignalingDelayMs(int delay_ms) {
-    initiating_client_->set_signaling_delay_ms(delay_ms);
-    receiving_client_->set_signaling_delay_ms(delay_ms);
-  }
-
-  void SetVideoConstraints(const webrtc::FakeConstraints& init_constraints,
-                           const webrtc::FakeConstraints& recv_constraints) {
-    initiating_client_->SetVideoConstraints(init_constraints);
-    receiving_client_->SetVideoConstraints(recv_constraints);
-  }
-
-  void SetCaptureRotation(webrtc::VideoRotation rotation) {
-    initiating_client_->SetCaptureRotation(rotation);
-    receiving_client_->SetCaptureRotation(rotation);
-  }
-
-  void EnableVideoDecoderFactory() {
-    initiating_client_->EnableVideoDecoderFactory();
-    receiving_client_->EnableVideoDecoderFactory();
-  }
-
-  // This test sets up a call between two parties. Both parties send static
-  // frames to each other. Once the test is finished the number of sent frames
-  // is compared to the number of received frames.
-  void LocalP2PTest() {
-    if (initiating_client_->NumberOfLocalMediaStreams() == 0) {
-      initiating_client_->AddMediaStream(true, true);
-    }
-    initiating_client_->Negotiate();
-    // Assert true is used here since next tests are guaranteed to fail and
-    // would eat up 5 seconds.
-    ASSERT_TRUE_WAIT(SessionActive(), kMaxWaitForActivationMs);
-    VerifySessionDescriptions();
-
-    int audio_frame_count = kEndAudioFrameCount;
-    int video_frame_count = kEndVideoFrameCount;
-    // TODO(ronghuawu): Add test to cover the case of sendonly and recvonly.
-
-    if ((!initiating_client_->can_receive_audio() &&
-         !initiating_client_->can_receive_video()) ||
-        (!receiving_client_->can_receive_audio() &&
-         !receiving_client_->can_receive_video())) {
-      // Neither audio nor video will flow, so connections won't be
-      // established. There's nothing more to check.
-      // TODO(hta): Check connection if there's a data channel.
-      return;
-    }
-
-    // Audio or video is expected to flow, so both clients should reach the
-    // Connected state, and the offerer (ICE controller) should proceed to
-    // Completed.
-    // Note: These tests have been observed to fail under heavy load at
-    // shorter timeouts, so they may be flaky.
-    EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
-                   initiating_client_->ice_connection_state(),
-                   kMaxWaitForFramesMs);
-    EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
-                   receiving_client_->ice_connection_state(),
-                   kMaxWaitForFramesMs);
-
-    // The ICE gathering state should end up in kIceGatheringComplete,
-    // but there's a bug that prevents this at the moment, and the state
-    // machine is being updated by the WEBRTC WG.
-    // TODO(hta): Update this check when spec revisions finish.
-    EXPECT_NE(webrtc::PeerConnectionInterface::kIceGatheringNew,
-              initiating_client_->ice_gathering_state());
-    EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceGatheringComplete,
-                   receiving_client_->ice_gathering_state(),
-                   kMaxWaitForFramesMs);
-
-    // Check that the expected number of frames have arrived.
-    EXPECT_TRUE_WAIT(FramesHaveArrived(audio_frame_count, video_frame_count),
-                     kMaxWaitForFramesMs);
-  }
-
-  void SetupAndVerifyDtlsCall() {
-    MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
-    FakeConstraints setup_constraints;
-    setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
-                                   true);
-    // Disable resolution adaptation, we don't want it interfering with the
-    // test results.
-    webrtc::PeerConnectionInterface::RTCConfiguration rtc_config;
-    rtc_config.set_cpu_adaptation(false);
-
-    ASSERT_TRUE(CreateTestClients(&setup_constraints, nullptr, &rtc_config,
-                                  &setup_constraints, nullptr, &rtc_config));
-    LocalP2PTest();
-    VerifyRenderedAspectRatio(640, 480);
-  }
-
-  PeerConnectionTestClient* CreateDtlsClientWithAlternateKey() {
-    FakeConstraints setup_constraints;
-    setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
-                                   true);
-    // Disable resolution adaptation, we don't want it interfering with the
-    // test results.
-    webrtc::PeerConnectionInterface::RTCConfiguration rtc_config;
-    rtc_config.set_cpu_adaptation(false);
-
-    std::unique_ptr<FakeRTCCertificateGenerator> cert_generator(
-        rtc::SSLStreamAdapter::HaveDtlsSrtp() ?
-            new FakeRTCCertificateGenerator() : nullptr);
-    cert_generator->use_alternate_key();
-
-    // Make sure the new client is using a different certificate.
-    return PeerConnectionTestClient::CreateClientWithDtlsIdentityStore(
-        "New Peer: ", &setup_constraints, nullptr, &rtc_config,
-        std::move(cert_generator), prefer_constraint_apis_,
-        network_thread_.get(), worker_thread_.get());
-  }
-
-  void SendRtpData(webrtc::DataChannelInterface* dc, const std::string& data) {
-    // Messages may get lost on the unreliable DataChannel, so we send multiple
-    // times to avoid test flakiness.
-    static const size_t kSendAttempts = 5;
-
-    for (size_t i = 0; i < kSendAttempts; ++i) {
-      dc->Send(DataBuffer(data));
-    }
-  }
-
-  rtc::Thread* network_thread() { return network_thread_.get(); }
-
-  rtc::VirtualSocketServer* virtual_socket_server() { return ss_.get(); }
-
-  PeerConnectionTestClient* initializing_client() {
-    return initiating_client_.get();
-  }
-
-  // Set the |initiating_client_| to the |client| passed in and return the
-  // original |initiating_client_|.
-  PeerConnectionTestClient* set_initializing_client(
-      PeerConnectionTestClient* client) {
-    PeerConnectionTestClient* old = initiating_client_.release();
-    initiating_client_.reset(client);
-    return old;
-  }
-
-  PeerConnectionTestClient* receiving_client() {
-    return receiving_client_.get();
-  }
-
-  // Set the |receiving_client_| to the |client| passed in and return the
-  // original |receiving_client_|.
-  PeerConnectionTestClient* set_receiving_client(
-      PeerConnectionTestClient* client) {
-    PeerConnectionTestClient* old = receiving_client_.release();
-    receiving_client_.reset(client);
-    return old;
-  }
-
-  bool AllObserversReceived(
-      const std::vector<std::unique_ptr<MockRtpReceiverObserver>>& observers) {
-    for (auto& observer : observers) {
-      if (!observer->first_packet_received()) {
-        return false;
-      }
-    }
-    return true;
-  }
-
-  void TestGcmNegotiation(bool local_gcm_enabled, bool remote_gcm_enabled,
-      int expected_cipher_suite) {
-    PeerConnectionFactory::Options init_options;
-    init_options.crypto_options.enable_gcm_crypto_suites = local_gcm_enabled;
-    PeerConnectionFactory::Options recv_options;
-    recv_options.crypto_options.enable_gcm_crypto_suites = remote_gcm_enabled;
-    ASSERT_TRUE(CreateTestClients(nullptr, &init_options, nullptr, nullptr,
-                                  &recv_options, nullptr));
-    rtc::scoped_refptr<webrtc::FakeMetricsObserver>
-        init_observer =
-            new rtc::RefCountedObject<webrtc::FakeMetricsObserver>();
-    initializing_client()->pc()->RegisterUMAObserver(init_observer);
-    LocalP2PTest();
-
-    EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(expected_cipher_suite),
-                   initializing_client()->GetSrtpCipherStats(),
-                   kMaxWaitMs);
-    EXPECT_EQ(1,
-              init_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher,
-                                            expected_cipher_suite));
-  }
-
- private:
-  // |ss_| is used by |network_thread_| so it must be destroyed later.
-  std::unique_ptr<rtc::PhysicalSocketServer> pss_;
-  std::unique_ptr<rtc::VirtualSocketServer> ss_;
-  // |network_thread_| and |worker_thread_| are used by both
-  // |initiating_client_| and |receiving_client_| so they must be destroyed
-  // later.
-  std::unique_ptr<rtc::Thread> network_thread_;
-  std::unique_ptr<rtc::Thread> worker_thread_;
-  std::unique_ptr<PeerConnectionTestClient> initiating_client_;
-  std::unique_ptr<PeerConnectionTestClient> receiving_client_;
-  bool prefer_constraint_apis_ = true;
-};
-
-// Disable for TSan v2, see
-// https://code.google.com/p/webrtc/issues/detail?id=1205 for details.
-#if !defined(THREAD_SANITIZER)
-
-TEST_F(P2PTestConductor, TestRtpReceiverObserverCallbackFunction) {
-  ASSERT_TRUE(CreateTestClients());
-  LocalP2PTest();
-  EXPECT_TRUE_WAIT(
-      AllObserversReceived(initializing_client()->rtp_receiver_observers()),
-      kMaxWaitForFramesMs);
-  EXPECT_TRUE_WAIT(
-      AllObserversReceived(receiving_client()->rtp_receiver_observers()),
-      kMaxWaitForFramesMs);
-}
-
-// The observers are expected to fire the signal even if they are set after the
-// first packet is received.
-TEST_F(P2PTestConductor, TestSetRtpReceiverObserverAfterFirstPacketIsReceived) {
-  ASSERT_TRUE(CreateTestClients());
-  LocalP2PTest();
-  // Reset the RtpReceiverObservers.
-  initializing_client()->SetRtpReceiverObservers();
-  receiving_client()->SetRtpReceiverObservers();
-  EXPECT_TRUE_WAIT(
-      AllObserversReceived(initializing_client()->rtp_receiver_observers()),
-      kMaxWaitForFramesMs);
-  EXPECT_TRUE_WAIT(
-      AllObserversReceived(receiving_client()->rtp_receiver_observers()),
-      kMaxWaitForFramesMs);
-}
-
-// This test sets up a Jsep call between two parties and test Dtmf.
-// TODO(holmer): Disabled due to sometimes crashing on buildbots.
-// See issue webrtc/2378.
-TEST_F(P2PTestConductor, DISABLED_LocalP2PTestDtmf) {
-  ASSERT_TRUE(CreateTestClients());
-  LocalP2PTest();
-  VerifyDtmf();
-}
-
-// This test sets up a Jsep call between two parties and test that we can get a
-// video aspect ratio of 16:9.
-TEST_F(P2PTestConductor, LocalP2PTest16To9) {
-  ASSERT_TRUE(CreateTestClients());
-  FakeConstraints constraint;
-  double requested_ratio = 640.0/360;
-  constraint.SetMandatoryMinAspectRatio(requested_ratio);
-  SetVideoConstraints(constraint, constraint);
-  LocalP2PTest();
-
-  ASSERT_LE(0, initializing_client()->rendered_height());
-  double initiating_video_ratio =
-      static_cast<double>(initializing_client()->rendered_width()) /
-      initializing_client()->rendered_height();
-  EXPECT_LE(requested_ratio, initiating_video_ratio);
-
-  ASSERT_LE(0, receiving_client()->rendered_height());
-  double receiving_video_ratio =
-      static_cast<double>(receiving_client()->rendered_width()) /
-      receiving_client()->rendered_height();
-  EXPECT_LE(requested_ratio, receiving_video_ratio);
-}
-
-// This test sets up a Jsep call between two parties and test that the
-// received video has a resolution of 1280*720.
-// TODO(mallinath): Enable when
-// http://code.google.com/p/webrtc/issues/detail?id=981 is fixed.
-TEST_F(P2PTestConductor, DISABLED_LocalP2PTest1280By720) {
-  ASSERT_TRUE(CreateTestClients());
-  FakeConstraints constraint;
-  constraint.SetMandatoryMinWidth(1280);
-  constraint.SetMandatoryMinHeight(720);
-  SetVideoConstraints(constraint, constraint);
-  LocalP2PTest();
-  VerifyRenderedAspectRatio(1280, 720);
-}
-
-// This test sets up a call between two endpoints that are configured to use
-// DTLS key agreement. As a result, DTLS is negotiated and used for transport.
-TEST_F(P2PTestConductor, LocalP2PTestDtls) {
-  SetupAndVerifyDtlsCall();
-}
-
-// This test sets up an one-way call, with media only from initiator to
-// responder.
-TEST_F(P2PTestConductor, OneWayMediaCall) {
-  ASSERT_TRUE(CreateTestClients());
-  receiving_client()->set_auto_add_stream(false);
-  LocalP2PTest();
-}
-
-TEST_F(P2PTestConductor, OneWayMediaCallWithoutConstraints) {
-  ASSERT_TRUE(CreateTestClientsThatPreferNoConstraints());
-  receiving_client()->set_auto_add_stream(false);
-  LocalP2PTest();
-}
-
-// This test sets up a audio call initially and then upgrades to audio/video,
-// using DTLS.
-TEST_F(P2PTestConductor, LocalP2PTestDtlsRenegotiate) {
-  MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
-  FakeConstraints setup_constraints;
-  setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
-                                 true);
-  ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
-  receiving_client()->SetReceiveAudioVideo(true, false);
-  LocalP2PTest();
-  receiving_client()->SetReceiveAudioVideo(true, true);
-  receiving_client()->Negotiate();
-}
-
-// This test sets up a call transfer to a new caller with a different DTLS
-// fingerprint.
-TEST_F(P2PTestConductor, LocalP2PTestDtlsTransferCallee) {
-  MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
-  SetupAndVerifyDtlsCall();
-
-  // Keeping the original peer around which will still send packets to the
-  // receiving client. These SRTP packets will be dropped.
-  std::unique_ptr<PeerConnectionTestClient> original_peer(
-      set_initializing_client(CreateDtlsClientWithAlternateKey()));
-  original_peer->pc()->Close();
-
-  SetSignalingReceivers();
-  receiving_client()->SetExpectIceRestart(true);
-  LocalP2PTest();
-  VerifyRenderedAspectRatio(640, 480);
-}
-
-// This test sets up a non-bundle call and apply bundle during ICE restart. When
-// bundle is in effect in the restart, the channel can successfully reset its
-// DTLS-SRTP context.
-TEST_F(P2PTestConductor, LocalP2PTestDtlsBundleInIceRestart) {
-  MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
-  FakeConstraints setup_constraints;
-  setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
-                                 true);
-  ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
-  receiving_client()->RemoveBundleFromReceivedSdp(true);
-  LocalP2PTest();
-  VerifyRenderedAspectRatio(640, 480);
-
-  initializing_client()->IceRestart();
-  receiving_client()->SetExpectIceRestart(true);
-  receiving_client()->RemoveBundleFromReceivedSdp(false);
-  LocalP2PTest();
-  VerifyRenderedAspectRatio(640, 480);
-}
-
-// This test sets up a call transfer to a new callee with a different DTLS
-// fingerprint.
-TEST_F(P2PTestConductor, LocalP2PTestDtlsTransferCaller) {
-  MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
-  SetupAndVerifyDtlsCall();
-
-  // Keeping the original peer around which will still send packets to the
-  // receiving client. These SRTP packets will be dropped.
-  std::unique_ptr<PeerConnectionTestClient> original_peer(
-      set_receiving_client(CreateDtlsClientWithAlternateKey()));
-  original_peer->pc()->Close();
-
-  SetSignalingReceivers();
-  initializing_client()->IceRestart();
-  LocalP2PTest();
-  VerifyRenderedAspectRatio(640, 480);
-}
-
-TEST_F(P2PTestConductor, LocalP2PTestCVO) {
-  ASSERT_TRUE(CreateTestClients());
-  SetCaptureRotation(webrtc::kVideoRotation_90);
-  LocalP2PTest();
-  VerifyRenderedAspectRatio(640, 480, webrtc::kVideoRotation_90);
-}
-
-TEST_F(P2PTestConductor, LocalP2PTestReceiverDoesntSupportCVO) {
-  ASSERT_TRUE(CreateTestClients());
-  SetCaptureRotation(webrtc::kVideoRotation_90);
-  receiving_client()->RemoveCvoFromReceivedSdp(true);
-  LocalP2PTest();
-  VerifyRenderedAspectRatio(480, 640, webrtc::kVideoRotation_0);
-}
-
-// This test sets up a call between two endpoints that are configured to use
-// DTLS key agreement. The offerer don't support SDES. As a result, DTLS is
-// negotiated and used for transport.
-TEST_F(P2PTestConductor, LocalP2PTestOfferDtlsButNotSdes) {
-  MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
-  FakeConstraints setup_constraints;
-  setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
-                                 true);
-  ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
-  receiving_client()->RemoveSdesCryptoFromReceivedSdp(true);
-  LocalP2PTest();
-  VerifyRenderedAspectRatio(640, 480);
-}
-
-#ifdef HAVE_SCTP
-// This test verifies that the negotiation will succeed with data channel only
-// in max-bundle mode.
-TEST_F(P2PTestConductor, LocalP2PTestOfferDataChannelOnly) {
-  webrtc::PeerConnectionInterface::RTCConfiguration rtc_config;
-  rtc_config.bundle_policy =
-      webrtc::PeerConnectionInterface::kBundlePolicyMaxBundle;
-  ASSERT_TRUE(CreateTestClients(rtc_config, rtc_config));
-  initializing_client()->CreateDataChannel();
-  initializing_client()->Negotiate();
-}
-#endif
-
-// This test sets up a Jsep call between two parties, and the callee only
-// accept to receive video.
-TEST_F(P2PTestConductor, LocalP2PTestAnswerVideo) {
-  ASSERT_TRUE(CreateTestClients());
-  receiving_client()->SetReceiveAudioVideo(false, true);
-  LocalP2PTest();
-}
-
-// This test sets up a Jsep call between two parties, and the callee only
-// accept to receive audio.
-TEST_F(P2PTestConductor, LocalP2PTestAnswerAudio) {
-  ASSERT_TRUE(CreateTestClients());
-  receiving_client()->SetReceiveAudioVideo(true, false);
-  LocalP2PTest();
-}
-
-// This test sets up a Jsep call between two parties, and the callee reject both
-// audio and video.
-TEST_F(P2PTestConductor, LocalP2PTestAnswerNone) {
-  ASSERT_TRUE(CreateTestClients());
-  receiving_client()->SetReceiveAudioVideo(false, false);
-  LocalP2PTest();
-}
-
-// This test sets up an audio and video call between two parties. After the call
-// runs for a while (10 frames), the caller sends an update offer with video
-// being rejected. Once the re-negotiation is done, the video flow should stop
-// and the audio flow should continue.
-TEST_F(P2PTestConductor, UpdateOfferWithRejectedContent) {
-  ASSERT_TRUE(CreateTestClients());
-  LocalP2PTest();
-  TestUpdateOfferWithRejectedContent();
-}
-
-// This test sets up a Jsep call between two parties. The MSID is removed from
-// the SDP strings from the caller.
-TEST_F(P2PTestConductor, LocalP2PTestWithoutMsid) {
-  ASSERT_TRUE(CreateTestClients());
-  receiving_client()->RemoveMsidFromReceivedSdp(true);
-  // TODO(perkj): Currently there is a bug that cause audio to stop playing if
-  // audio and video is muxed when MSID is disabled. Remove
-  // SetRemoveBundleFromSdp once
-  // https://code.google.com/p/webrtc/issues/detail?id=1193 is fixed.
-  receiving_client()->RemoveBundleFromReceivedSdp(true);
-  LocalP2PTest();
-}
-
-TEST_F(P2PTestConductor, LocalP2PTestTwoStreams) {
-  ASSERT_TRUE(CreateTestClients());
-  // Set optional video constraint to max 320pixels to decrease CPU usage.
-  FakeConstraints constraint;
-  constraint.SetOptionalMaxWidth(320);
-  SetVideoConstraints(constraint, constraint);
-  initializing_client()->AddMediaStream(true, true);
-  initializing_client()->AddMediaStream(false, true);
-  ASSERT_EQ(2u, initializing_client()->NumberOfLocalMediaStreams());
-  LocalP2PTest();
-  EXPECT_EQ(2u, receiving_client()->number_of_remote_streams());
-}
-
-// Test that we can receive the audio output level from a remote audio track.
-TEST_F(P2PTestConductor, GetAudioOutputLevelStats) {
-  ASSERT_TRUE(CreateTestClients());
-  LocalP2PTest();
-
-  StreamCollectionInterface* remote_streams =
-      initializing_client()->remote_streams();
-  ASSERT_GT(remote_streams->count(), 0u);
-  ASSERT_GT(remote_streams->at(0)->GetAudioTracks().size(), 0u);
-  MediaStreamTrackInterface* remote_audio_track =
-      remote_streams->at(0)->GetAudioTracks()[0];
-
-  // Get the audio output level stats. Note that the level is not available
-  // until a RTCP packet has been received.
-  EXPECT_TRUE_WAIT(
-      initializing_client()->GetAudioOutputLevelStats(remote_audio_track) > 0,
-      kMaxWaitForStatsMs);
-}
-
-// Test that an audio input level is reported.
-TEST_F(P2PTestConductor, GetAudioInputLevelStats) {
-  ASSERT_TRUE(CreateTestClients());
-  LocalP2PTest();
-
-  // Get the audio input level stats.  The level should be available very
-  // soon after the test starts.
-  EXPECT_TRUE_WAIT(initializing_client()->GetAudioInputLevelStats() > 0,
-      kMaxWaitForStatsMs);
-}
-
-// Test that we can get incoming byte counts from both audio and video tracks.
-TEST_F(P2PTestConductor, GetBytesReceivedStats) {
-  ASSERT_TRUE(CreateTestClients());
-  LocalP2PTest();
-
-  StreamCollectionInterface* remote_streams =
-      initializing_client()->remote_streams();
-  ASSERT_GT(remote_streams->count(), 0u);
-  ASSERT_GT(remote_streams->at(0)->GetAudioTracks().size(), 0u);
-  MediaStreamTrackInterface* remote_audio_track =
-      remote_streams->at(0)->GetAudioTracks()[0];
-  EXPECT_TRUE_WAIT(
-      initializing_client()->GetBytesReceivedStats(remote_audio_track) > 0,
-      kMaxWaitForStatsMs);
-
-  MediaStreamTrackInterface* remote_video_track =
-      remote_streams->at(0)->GetVideoTracks()[0];
-  EXPECT_TRUE_WAIT(
-      initializing_client()->GetBytesReceivedStats(remote_video_track) > 0,
-      kMaxWaitForStatsMs);
-}
-
-// Test that we can get outgoing byte counts from both audio and video tracks.
-TEST_F(P2PTestConductor, GetBytesSentStats) {
-  ASSERT_TRUE(CreateTestClients());
-  LocalP2PTest();
-
-  StreamCollectionInterface* local_streams =
-      initializing_client()->local_streams();
-  ASSERT_GT(local_streams->count(), 0u);
-  ASSERT_GT(local_streams->at(0)->GetAudioTracks().size(), 0u);
-  MediaStreamTrackInterface* local_audio_track =
-      local_streams->at(0)->GetAudioTracks()[0];
-  EXPECT_TRUE_WAIT(
-      initializing_client()->GetBytesSentStats(local_audio_track) > 0,
-      kMaxWaitForStatsMs);
-
-  MediaStreamTrackInterface* local_video_track =
-      local_streams->at(0)->GetVideoTracks()[0];
-  EXPECT_TRUE_WAIT(
-      initializing_client()->GetBytesSentStats(local_video_track) > 0,
-      kMaxWaitForStatsMs);
-}
-
-// Test that DTLS 1.0 is used if both sides only support DTLS 1.0.
-TEST_F(P2PTestConductor, GetDtls12None) {
-  PeerConnectionFactory::Options init_options;
-  init_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
-  PeerConnectionFactory::Options recv_options;
-  recv_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
-  ASSERT_TRUE(CreateTestClients(nullptr, &init_options, nullptr, nullptr,
-                                &recv_options, nullptr));
-  rtc::scoped_refptr<webrtc::FakeMetricsObserver>
-      init_observer = new rtc::RefCountedObject<webrtc::FakeMetricsObserver>();
-  initializing_client()->pc()->RegisterUMAObserver(init_observer);
-  LocalP2PTest();
-
-  EXPECT_TRUE_WAIT(
-      rtc::SSLStreamAdapter::IsAcceptableCipher(
-          initializing_client()->GetDtlsCipherStats(), rtc::KT_DEFAULT),
-      kMaxWaitForStatsMs);
-  EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite),
-                 initializing_client()->GetSrtpCipherStats(),
-                 kMaxWaitForStatsMs);
-  EXPECT_EQ(1,
-            init_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher,
-                                          kDefaultSrtpCryptoSuite));
-}
-
-// Test that DTLS 1.2 is used if both ends support it.
-TEST_F(P2PTestConductor, GetDtls12Both) {
-  PeerConnectionFactory::Options init_options;
-  init_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
-  PeerConnectionFactory::Options recv_options;
-  recv_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
-  ASSERT_TRUE(CreateTestClients(nullptr, &init_options, nullptr, nullptr,
-                                &recv_options, nullptr));
-  rtc::scoped_refptr<webrtc::FakeMetricsObserver>
-      init_observer = new rtc::RefCountedObject<webrtc::FakeMetricsObserver>();
-  initializing_client()->pc()->RegisterUMAObserver(init_observer);
-  LocalP2PTest();
-
-  EXPECT_TRUE_WAIT(
-      rtc::SSLStreamAdapter::IsAcceptableCipher(
-          initializing_client()->GetDtlsCipherStats(), rtc::KT_DEFAULT),
-      kMaxWaitForStatsMs);
-  EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite),
-                 initializing_client()->GetSrtpCipherStats(),
-                 kMaxWaitForStatsMs);
-  EXPECT_EQ(1,
-            init_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher,
-                                          kDefaultSrtpCryptoSuite));
-}
-
-// Test that DTLS 1.0 is used if the initator supports DTLS 1.2 and the
-// received supports 1.0.
-TEST_F(P2PTestConductor, GetDtls12Init) {
-  PeerConnectionFactory::Options init_options;
-  init_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
-  PeerConnectionFactory::Options recv_options;
-  recv_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
-  ASSERT_TRUE(CreateTestClients(nullptr, &init_options, nullptr, nullptr,
-                                &recv_options, nullptr));
-  rtc::scoped_refptr<webrtc::FakeMetricsObserver>
-      init_observer = new rtc::RefCountedObject<webrtc::FakeMetricsObserver>();
-  initializing_client()->pc()->RegisterUMAObserver(init_observer);
-  LocalP2PTest();
-
-  EXPECT_TRUE_WAIT(
-      rtc::SSLStreamAdapter::IsAcceptableCipher(
-          initializing_client()->GetDtlsCipherStats(), rtc::KT_DEFAULT),
-      kMaxWaitForStatsMs);
-  EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite),
-                 initializing_client()->GetSrtpCipherStats(),
-                 kMaxWaitForStatsMs);
-  EXPECT_EQ(1,
-            init_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher,
-                                          kDefaultSrtpCryptoSuite));
-}
-
-// Test that DTLS 1.0 is used if the initator supports DTLS 1.0 and the
-// received supports 1.2.
-TEST_F(P2PTestConductor, GetDtls12Recv) {
-  PeerConnectionFactory::Options init_options;
-  init_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
-  PeerConnectionFactory::Options recv_options;
-  recv_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
-  ASSERT_TRUE(CreateTestClients(nullptr, &init_options, nullptr, nullptr,
-                                &recv_options, nullptr));
-  rtc::scoped_refptr<webrtc::FakeMetricsObserver>
-      init_observer = new rtc::RefCountedObject<webrtc::FakeMetricsObserver>();
-  initializing_client()->pc()->RegisterUMAObserver(init_observer);
-  LocalP2PTest();
-
-  EXPECT_TRUE_WAIT(
-      rtc::SSLStreamAdapter::IsAcceptableCipher(
-          initializing_client()->GetDtlsCipherStats(), rtc::KT_DEFAULT),
-      kMaxWaitForStatsMs);
-  EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite),
-                 initializing_client()->GetSrtpCipherStats(),
-                 kMaxWaitForStatsMs);
-  EXPECT_EQ(1,
-            init_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher,
-                                          kDefaultSrtpCryptoSuite));
-}
-
-// Test that a non-GCM cipher is used if both sides only support non-GCM.
-TEST_F(P2PTestConductor, GetGcmNone) {
-  TestGcmNegotiation(false, false, kDefaultSrtpCryptoSuite);
-}
-
-// Test that a GCM cipher is used if both ends support it.
-TEST_F(P2PTestConductor, GetGcmBoth) {
-  TestGcmNegotiation(true, true, kDefaultSrtpCryptoSuiteGcm);
-}
-
-// Test that GCM isn't used if only the initiator supports it.
-TEST_F(P2PTestConductor, GetGcmInit) {
-  TestGcmNegotiation(true, false, kDefaultSrtpCryptoSuite);
-}
-
-// Test that GCM isn't used if only the receiver supports it.
-TEST_F(P2PTestConductor, GetGcmRecv) {
-  TestGcmNegotiation(false, true, kDefaultSrtpCryptoSuite);
-}
-
-// This test sets up a call between two parties with audio, video and an RTP
-// data channel.
-TEST_F(P2PTestConductor, LocalP2PTestRtpDataChannel) {
-  FakeConstraints setup_constraints;
-  setup_constraints.SetAllowRtpDataChannels();
-  ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
-  initializing_client()->CreateDataChannel();
-  LocalP2PTest();
-  ASSERT_TRUE(initializing_client()->data_channel() != nullptr);
-  ASSERT_TRUE(receiving_client()->data_channel() != nullptr);
-  EXPECT_TRUE_WAIT(initializing_client()->data_observer()->IsOpen(),
-                   kMaxWaitMs);
-  EXPECT_TRUE_WAIT(receiving_client()->data_observer()->IsOpen(),
-                   kMaxWaitMs);
-
-  std::string data = "hello world";
-
-  SendRtpData(initializing_client()->data_channel(), data);
-  EXPECT_EQ_WAIT(data, receiving_client()->data_observer()->last_message(),
-                 kMaxWaitMs);
-
-  SendRtpData(receiving_client()->data_channel(), data);
-  EXPECT_EQ_WAIT(data, initializing_client()->data_observer()->last_message(),
-                 kMaxWaitMs);
-
-  receiving_client()->data_channel()->Close();
-  // Send new offer and answer.
-  receiving_client()->Negotiate();
-  EXPECT_FALSE(initializing_client()->data_observer()->IsOpen());
-  EXPECT_FALSE(receiving_client()->data_observer()->IsOpen());
-}
-
-#ifdef HAVE_SCTP
-// This test sets up a call between two parties with audio, video and an SCTP
-// data channel.
-TEST_F(P2PTestConductor, LocalP2PTestSctpDataChannel) {
-  ASSERT_TRUE(CreateTestClients());
-  initializing_client()->CreateDataChannel();
-  LocalP2PTest();
-  ASSERT_TRUE(initializing_client()->data_channel() != nullptr);
-  EXPECT_TRUE_WAIT(receiving_client()->data_channel() != nullptr, kMaxWaitMs);
-  EXPECT_TRUE_WAIT(initializing_client()->data_observer()->IsOpen(),
-                   kMaxWaitMs);
-  EXPECT_TRUE_WAIT(receiving_client()->data_observer()->IsOpen(), kMaxWaitMs);
-
-  std::string data = "hello world";
-
-  initializing_client()->data_channel()->Send(DataBuffer(data));
-  EXPECT_EQ_WAIT(data, receiving_client()->data_observer()->last_message(),
-                 kMaxWaitMs);
-
-  receiving_client()->data_channel()->Send(DataBuffer(data));
-  EXPECT_EQ_WAIT(data, initializing_client()->data_observer()->last_message(),
-                 kMaxWaitMs);
-
-  receiving_client()->data_channel()->Close();
-  EXPECT_TRUE_WAIT(!initializing_client()->data_observer()->IsOpen(),
-                   kMaxWaitMs);
-  EXPECT_TRUE_WAIT(!receiving_client()->data_observer()->IsOpen(), kMaxWaitMs);
-}
-
-TEST_F(P2PTestConductor, UnorderedSctpDataChannel) {
-  ASSERT_TRUE(CreateTestClients());
-  webrtc::DataChannelInit init;
-  init.ordered = false;
-  initializing_client()->CreateDataChannel(&init);
-
-  // Introduce random network delays.
-  // Otherwise it's not a true "unordered" test.
-  virtual_socket_server()->set_delay_mean(20);
-  virtual_socket_server()->set_delay_stddev(5);
-  virtual_socket_server()->UpdateDelayDistribution();
-
-  initializing_client()->Negotiate();
-  ASSERT_TRUE(initializing_client()->data_channel() != nullptr);
-  EXPECT_TRUE_WAIT(receiving_client()->data_channel() != nullptr, kMaxWaitMs);
-  EXPECT_TRUE_WAIT(initializing_client()->data_observer()->IsOpen(),
-                   kMaxWaitMs);
-  EXPECT_TRUE_WAIT(receiving_client()->data_observer()->IsOpen(), kMaxWaitMs);
-
-  static constexpr int kNumMessages = 100;
-  // Deliberately chosen to be larger than the MTU so messages get fragmented.
-  static constexpr size_t kMaxMessageSize = 4096;
-  // Create and send random messages.
-  std::vector<std::string> sent_messages;
-  for (int i = 0; i < kNumMessages; ++i) {
-    size_t length = (rand() % kMaxMessageSize) + 1;
-    std::string message;
-    ASSERT_TRUE(rtc::CreateRandomString(length, &message));
-    initializing_client()->data_channel()->Send(DataBuffer(message));
-    receiving_client()->data_channel()->Send(DataBuffer(message));
-    sent_messages.push_back(message);
-  }
-
-  EXPECT_EQ_WAIT(
-      kNumMessages,
-      initializing_client()->data_observer()->received_message_count(),
-      kMaxWaitMs);
-  EXPECT_EQ_WAIT(kNumMessages,
-                 receiving_client()->data_observer()->received_message_count(),
-                 kMaxWaitMs);
-
-  // Sort and compare to make sure none of the messages were corrupted.
-  std::vector<std::string> initializing_client_received_messages =
-      initializing_client()->data_observer()->messages();
-  std::vector<std::string> receiving_client_received_messages =
-      receiving_client()->data_observer()->messages();
-  std::sort(sent_messages.begin(), sent_messages.end());
-  std::sort(initializing_client_received_messages.begin(),
-            initializing_client_received_messages.end());
-  std::sort(receiving_client_received_messages.begin(),
-            receiving_client_received_messages.end());
-  EXPECT_EQ(sent_messages, initializing_client_received_messages);
-  EXPECT_EQ(sent_messages, receiving_client_received_messages);
-
-  receiving_client()->data_channel()->Close();
-  EXPECT_TRUE_WAIT(!initializing_client()->data_observer()->IsOpen(),
-                   kMaxWaitMs);
-  EXPECT_TRUE_WAIT(!receiving_client()->data_observer()->IsOpen(), kMaxWaitMs);
-}
-#endif  // HAVE_SCTP
-
-// This test sets up a call between two parties and creates a data channel.
-// The test tests that received data is buffered unless an observer has been
-// registered.
-// Rtp data channels can receive data before the underlying
-// transport has detected that a channel is writable and thus data can be
-// received before the data channel state changes to open. That is hard to test
-// but the same buffering is used in that case.
-TEST_F(P2PTestConductor, RegisterDataChannelObserver) {
-  FakeConstraints setup_constraints;
-  setup_constraints.SetAllowRtpDataChannels();
-  ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
-  initializing_client()->CreateDataChannel();
-  initializing_client()->Negotiate();
-
-  ASSERT_TRUE(initializing_client()->data_channel() != nullptr);
-  ASSERT_TRUE(receiving_client()->data_channel() != nullptr);
-  EXPECT_TRUE_WAIT(initializing_client()->data_observer()->IsOpen(),
-                   kMaxWaitMs);
-  EXPECT_EQ_WAIT(DataChannelInterface::kOpen,
-                 receiving_client()->data_channel()->state(), kMaxWaitMs);
-
-  // Unregister the existing observer.
-  receiving_client()->data_channel()->UnregisterObserver();
-
-  std::string data = "hello world";
-  SendRtpData(initializing_client()->data_channel(), data);
-
-  // Wait a while to allow the sent data to arrive before an observer is
-  // registered..
-  rtc::Thread::Current()->ProcessMessages(100);
-
-  MockDataChannelObserver new_observer(receiving_client()->data_channel());
-  EXPECT_EQ_WAIT(data, new_observer.last_message(), kMaxWaitMs);
-}
-
-// This test sets up a call between two parties with audio, video and but only
-// the initiating client support data.
-TEST_F(P2PTestConductor, LocalP2PTestReceiverDoesntSupportData) {
-  FakeConstraints setup_constraints_1;
-  setup_constraints_1.SetAllowRtpDataChannels();
-  // Must disable DTLS to make negotiation succeed.
-  setup_constraints_1.SetMandatory(
-      MediaConstraintsInterface::kEnableDtlsSrtp, false);
-  FakeConstraints setup_constraints_2;
-  setup_constraints_2.SetMandatory(
-      MediaConstraintsInterface::kEnableDtlsSrtp, false);
-  ASSERT_TRUE(CreateTestClients(&setup_constraints_1, &setup_constraints_2));
-  initializing_client()->CreateDataChannel();
-  LocalP2PTest();
-  EXPECT_TRUE(initializing_client()->data_channel() != nullptr);
-  EXPECT_FALSE(receiving_client()->data_channel());
-  EXPECT_FALSE(initializing_client()->data_observer()->IsOpen());
-}
-
-// This test sets up a call between two parties with audio, video. When audio
-// and video is setup and flowing and data channel is negotiated.
-TEST_F(P2PTestConductor, AddDataChannelAfterRenegotiation) {
-  FakeConstraints setup_constraints;
-  setup_constraints.SetAllowRtpDataChannels();
-  ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
-  LocalP2PTest();
-  initializing_client()->CreateDataChannel();
-  // Send new offer and answer.
-  initializing_client()->Negotiate();
-  ASSERT_TRUE(initializing_client()->data_channel() != nullptr);
-  ASSERT_TRUE(receiving_client()->data_channel() != nullptr);
-  EXPECT_TRUE_WAIT(initializing_client()->data_observer()->IsOpen(),
-                   kMaxWaitMs);
-  EXPECT_TRUE_WAIT(receiving_client()->data_observer()->IsOpen(),
-                   kMaxWaitMs);
-}
-
-// This test sets up a Jsep call with SCTP DataChannel and verifies the
-// negotiation is completed without error.
-#ifdef HAVE_SCTP
-TEST_F(P2PTestConductor, CreateOfferWithSctpDataChannel) {
-  MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
-  FakeConstraints constraints;
-  constraints.SetMandatory(
-      MediaConstraintsInterface::kEnableDtlsSrtp, true);
-  ASSERT_TRUE(CreateTestClients(&constraints, &constraints));
-  initializing_client()->CreateDataChannel();
-  initializing_client()->Negotiate(false, false);
-}
-#endif
-
-// This test sets up a call between two parties with audio, and video.
-// During the call, the initializing side restart ice and the test verifies that
-// new ice candidates are generated and audio and video still can flow.
-TEST_F(P2PTestConductor, IceRestart) {
-  ASSERT_TRUE(CreateTestClients());
-
-  // Negotiate and wait for ice completion and make sure audio and video plays.
-  LocalP2PTest();
-
-  // Create a SDP string of the first audio candidate for both clients.
-  const webrtc::IceCandidateCollection* audio_candidates_initiator =
-      initializing_client()->pc()->local_description()->candidates(0);
-  const webrtc::IceCandidateCollection* audio_candidates_receiver =
-      receiving_client()->pc()->local_description()->candidates(0);
-  ASSERT_GT(audio_candidates_initiator->count(), 0u);
-  ASSERT_GT(audio_candidates_receiver->count(), 0u);
-  std::string initiator_candidate;
-  EXPECT_TRUE(
-      audio_candidates_initiator->at(0)->ToString(&initiator_candidate));
-  std::string receiver_candidate;
-  EXPECT_TRUE(audio_candidates_receiver->at(0)->ToString(&receiver_candidate));
-
-  // Restart ice on the initializing client.
-  receiving_client()->SetExpectIceRestart(true);
-  initializing_client()->IceRestart();
-
-  // Negotiate and wait for ice completion again and make sure audio and video
-  // plays.
-  LocalP2PTest();
-
-  // Create a SDP string of the first audio candidate for both clients again.
-  const webrtc::IceCandidateCollection* audio_candidates_initiator_restart =
-      initializing_client()->pc()->local_description()->candidates(0);
-  const webrtc::IceCandidateCollection* audio_candidates_reciever_restart =
-      receiving_client()->pc()->local_description()->candidates(0);
-  ASSERT_GT(audio_candidates_initiator_restart->count(), 0u);
-  ASSERT_GT(audio_candidates_reciever_restart->count(), 0u);
-  std::string initiator_candidate_restart;
-  EXPECT_TRUE(audio_candidates_initiator_restart->at(0)->ToString(
-      &initiator_candidate_restart));
-  std::string receiver_candidate_restart;
-  EXPECT_TRUE(audio_candidates_reciever_restart->at(0)->ToString(
-      &receiver_candidate_restart));
-
-  // Verify that the first candidates in the local session descriptions has
-  // changed.
-  EXPECT_NE(initiator_candidate, initiator_candidate_restart);
-  EXPECT_NE(receiver_candidate, receiver_candidate_restart);
-}
-
-TEST_F(P2PTestConductor, IceRenominationDisabled) {
-  PeerConnectionInterface::RTCConfiguration config;
-  config.enable_ice_renomination = false;
-  ASSERT_TRUE(CreateTestClients(config, config));
-  LocalP2PTest();
-
-  initializing_client()->VerifyLocalIceRenomination();
-  receiving_client()->VerifyLocalIceRenomination();
-  initializing_client()->VerifyRemoteIceRenomination();
-  receiving_client()->VerifyRemoteIceRenomination();
-}
-
-TEST_F(P2PTestConductor, IceRenominationEnabled) {
-  PeerConnectionInterface::RTCConfiguration config;
-  config.enable_ice_renomination = true;
-  ASSERT_TRUE(CreateTestClients(config, config));
-  initializing_client()->SetExpectIceRenomination(true);
-  initializing_client()->SetExpectRemoteIceRenomination(true);
-  receiving_client()->SetExpectIceRenomination(true);
-  receiving_client()->SetExpectRemoteIceRenomination(true);
-  LocalP2PTest();
-
-  initializing_client()->VerifyLocalIceRenomination();
-  receiving_client()->VerifyLocalIceRenomination();
-  initializing_client()->VerifyRemoteIceRenomination();
-  receiving_client()->VerifyRemoteIceRenomination();
-}
-
-// This test sets up a call between two parties with audio, and video.
-// It then renegotiates setting the video m-line to "port 0", then later
-// renegotiates again, enabling video.
-TEST_F(P2PTestConductor, LocalP2PTestVideoDisableEnable) {
-  ASSERT_TRUE(CreateTestClients());
-
-  // Do initial negotiation. Will result in video and audio sendonly m-lines.
-  receiving_client()->set_auto_add_stream(false);
-  initializing_client()->AddMediaStream(true, true);
-  initializing_client()->Negotiate();
-
-  // Negotiate again, disabling the video m-line (receiving client will
-  // set port to 0 due to mandatory "OfferToReceiveVideo: false" constraint).
-  receiving_client()->SetReceiveVideo(false);
-  initializing_client()->Negotiate();
-
-  // Enable video and do negotiation again, making sure video is received
-  // end-to-end.
-  receiving_client()->SetReceiveVideo(true);
-  receiving_client()->AddMediaStream(true, true);
-  LocalP2PTest();
-}
-
-// This test sets up a Jsep call between two parties with external
-// VideoDecoderFactory.
-// TODO(holmer): Disabled due to sometimes crashing on buildbots.
-// See issue webrtc/2378.
-TEST_F(P2PTestConductor, DISABLED_LocalP2PTestWithVideoDecoderFactory) {
-  ASSERT_TRUE(CreateTestClients());
-  EnableVideoDecoderFactory();
-  LocalP2PTest();
-}
-
-// This tests that if we negotiate after calling CreateSender but before we
-// have a track, then set a track later, frames from the newly-set track are
-// received end-to-end.
-TEST_F(P2PTestConductor, EarlyWarmupTest) {
-  ASSERT_TRUE(CreateTestClients());
-  auto audio_sender =
-      initializing_client()->pc()->CreateSender("audio", "stream_id");
-  auto video_sender =
-      initializing_client()->pc()->CreateSender("video", "stream_id");
-  initializing_client()->Negotiate();
-  // Wait for ICE connection to complete, without any tracks.
-  // Note that the receiving client WILL (in HandleIncomingOffer) create
-  // tracks, so it's only the initiator here that's doing early warmup.
-  ASSERT_TRUE_WAIT(SessionActive(), kMaxWaitForActivationMs);
-  VerifySessionDescriptions();
-  EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
-                 initializing_client()->ice_connection_state(),
-                 kMaxWaitForFramesMs);
-  EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
-                 receiving_client()->ice_connection_state(),
-                 kMaxWaitForFramesMs);
-  // Now set the tracks, and expect frames to immediately start flowing.
-  EXPECT_TRUE(
-      audio_sender->SetTrack(initializing_client()->CreateLocalAudioTrack("")));
-  EXPECT_TRUE(
-      video_sender->SetTrack(initializing_client()->CreateLocalVideoTrack("")));
-  EXPECT_TRUE_WAIT(FramesHaveArrived(kEndAudioFrameCount, kEndVideoFrameCount),
-                   kMaxWaitForFramesMs);
-}
-
-#ifdef HAVE_QUIC
-// This test sets up a call between two parties using QUIC instead of DTLS for
-// audio and video, and a QUIC data channel.
-TEST_F(P2PTestConductor, LocalP2PTestQuicDataChannel) {
-  PeerConnectionInterface::RTCConfiguration quic_config;
-  quic_config.enable_quic = true;
-  ASSERT_TRUE(CreateTestClients(quic_config, quic_config));
-  webrtc::DataChannelInit init;
-  init.ordered = false;
-  init.reliable = true;
-  init.id = 1;
-  initializing_client()->CreateDataChannel(&init);
-  receiving_client()->CreateDataChannel(&init);
-  LocalP2PTest();
-  ASSERT_NE(nullptr, initializing_client()->data_channel());
-  ASSERT_NE(nullptr, receiving_client()->data_channel());
-  EXPECT_TRUE_WAIT(initializing_client()->data_observer()->IsOpen(),
-                   kMaxWaitMs);
-  EXPECT_TRUE_WAIT(receiving_client()->data_observer()->IsOpen(), kMaxWaitMs);
-
-  std::string data = "hello world";
-
-  initializing_client()->data_channel()->Send(DataBuffer(data));
-  EXPECT_EQ_WAIT(data, receiving_client()->data_observer()->last_message(),
-                 kMaxWaitMs);
-
-  receiving_client()->data_channel()->Send(DataBuffer(data));
-  EXPECT_EQ_WAIT(data, initializing_client()->data_observer()->last_message(),
-                 kMaxWaitMs);
-}
-
-// Tests that negotiation of QUIC data channels is completed without error.
-TEST_F(P2PTestConductor, NegotiateQuicDataChannel) {
-  PeerConnectionInterface::RTCConfiguration quic_config;
-  quic_config.enable_quic = true;
-  ASSERT_TRUE(CreateTestClients(quic_config, quic_config));
-  FakeConstraints constraints;
-  constraints.SetMandatory(MediaConstraintsInterface::kEnableDtlsSrtp, true);
-  ASSERT_TRUE(CreateTestClients(&constraints, &constraints));
-  webrtc::DataChannelInit init;
-  init.ordered = false;
-  init.reliable = true;
-  init.id = 1;
-  initializing_client()->CreateDataChannel(&init);
-  initializing_client()->Negotiate(false, false);
-}
-
-// This test sets up a JSEP call using QUIC. The callee only receives video.
-TEST_F(P2PTestConductor, LocalP2PTestVideoOnlyWithQuic) {
-  PeerConnectionInterface::RTCConfiguration quic_config;
-  quic_config.enable_quic = true;
-  ASSERT_TRUE(CreateTestClients(quic_config, quic_config));
-  receiving_client()->SetReceiveAudioVideo(false, true);
-  LocalP2PTest();
-}
-
-// This test sets up a JSEP call using QUIC. The callee only receives audio.
-TEST_F(P2PTestConductor, LocalP2PTestAudioOnlyWithQuic) {
-  PeerConnectionInterface::RTCConfiguration quic_config;
-  quic_config.enable_quic = true;
-  ASSERT_TRUE(CreateTestClients(quic_config, quic_config));
-  receiving_client()->SetReceiveAudioVideo(true, false);
-  LocalP2PTest();
-}
-
-// This test sets up a JSEP call using QUIC. The callee rejects both audio and
-// video.
-TEST_F(P2PTestConductor, LocalP2PTestNoVideoAudioWithQuic) {
-  PeerConnectionInterface::RTCConfiguration quic_config;
-  quic_config.enable_quic = true;
-  ASSERT_TRUE(CreateTestClients(quic_config, quic_config));
-  receiving_client()->SetReceiveAudioVideo(false, false);
-  LocalP2PTest();
-}
-
-#endif  // HAVE_QUIC
-
-TEST_F(P2PTestConductor, ForwardVideoOnlyStream) {
-  ASSERT_TRUE(CreateTestClients());
-  // One-way stream
-  receiving_client()->set_auto_add_stream(false);
-  // Video only, audio forwarding not expected to work.
-  initializing_client()->AddMediaStream(false, true);
-  initializing_client()->Negotiate();
-
-  ASSERT_TRUE_WAIT(SessionActive(), kMaxWaitForActivationMs);
-  VerifySessionDescriptions();
-
-  ASSERT_TRUE(initializing_client()->can_receive_video());
-  ASSERT_TRUE(receiving_client()->can_receive_video());
-
-  EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
-                 initializing_client()->ice_connection_state(),
-                 kMaxWaitForFramesMs);
-  EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
-                 receiving_client()->ice_connection_state(),
-                 kMaxWaitForFramesMs);
-
-  ASSERT_TRUE(receiving_client()->remote_streams()->count() == 1);
-
-  // Echo the stream back.
-  receiving_client()->pc()->AddStream(
-      receiving_client()->remote_streams()->at(0));
-  receiving_client()->Negotiate();
-
-  EXPECT_TRUE_WAIT(
-      initializing_client()->VideoFramesReceivedCheck(kEndVideoFrameCount),
-      kMaxWaitForFramesMs);
-}
-
-// Test that we achieve the expected end-to-end connection time, using a
-// fake clock and simulated latency on the media and signaling paths.
-// We use a TURN<->TURN connection because this is usually the quickest to
-// set up initially, especially when we're confident the connection will work
-// and can start sending media before we get a STUN response.
-//
-// With various optimizations enabled, here are the network delays we expect to
-// be on the critical path:
-// 1. 2 signaling trips: Signaling offer and offerer's TURN candidate, then
-//                       signaling answer (with DTLS fingerprint).
-// 2. 9 media hops: Rest of the DTLS handshake. 3 hops in each direction when
-//                  using TURN<->TURN pair, and DTLS exchange is 4 packets,
-//                  the first of which should have arrived before the answer.
-TEST_F(P2PTestConductor, EndToEndConnectionTimeWithTurnTurnPair) {
-  rtc::ScopedFakeClock fake_clock;
-  // Some things use a time of "0" as a special value, so we need to start out
-  // the fake clock at a nonzero time.
-  // TODO(deadbeef): Fix this.
-  fake_clock.AdvanceTime(rtc::TimeDelta::FromSeconds(1));
-
-  static constexpr int media_hop_delay_ms = 50;
-  static constexpr int signaling_trip_delay_ms = 500;
-  // For explanation of these values, see comment above.
-  static constexpr int required_media_hops = 9;
-  static constexpr int required_signaling_trips = 2;
-  // For internal delays (such as posting an event asychronously).
-  static constexpr int allowed_internal_delay_ms = 20;
-  static constexpr int total_connection_time_ms =
-      media_hop_delay_ms * required_media_hops +
-      signaling_trip_delay_ms * required_signaling_trips +
-      allowed_internal_delay_ms;
-
-  static const rtc::SocketAddress turn_server_1_internal_address{"88.88.88.0",
-                                                                 3478};
-  static const rtc::SocketAddress turn_server_1_external_address{"88.88.88.1",
-                                                                 0};
-  static const rtc::SocketAddress turn_server_2_internal_address{"99.99.99.0",
-                                                                 3478};
-  static const rtc::SocketAddress turn_server_2_external_address{"99.99.99.1",
-                                                                 0};
-  cricket::TestTurnServer turn_server_1(network_thread(),
-                                        turn_server_1_internal_address,
-                                        turn_server_1_external_address);
-  cricket::TestTurnServer turn_server_2(network_thread(),
-                                        turn_server_2_internal_address,
-                                        turn_server_2_external_address);
-  // Bypass permission check on received packets so media can be sent before
-  // the candidate is signaled.
-  turn_server_1.set_enable_permission_checks(false);
-  turn_server_2.set_enable_permission_checks(false);
-
-  PeerConnectionInterface::RTCConfiguration client_1_config;
-  webrtc::PeerConnectionInterface::IceServer ice_server_1;
-  ice_server_1.urls.push_back("turn:88.88.88.0:3478");
-  ice_server_1.username = "test";
-  ice_server_1.password = "test";
-  client_1_config.servers.push_back(ice_server_1);
-  client_1_config.type = webrtc::PeerConnectionInterface::kRelay;
-  client_1_config.presume_writable_when_fully_relayed = true;
-
-  PeerConnectionInterface::RTCConfiguration client_2_config;
-  webrtc::PeerConnectionInterface::IceServer ice_server_2;
-  ice_server_2.urls.push_back("turn:99.99.99.0:3478");
-  ice_server_2.username = "test";
-  ice_server_2.password = "test";
-  client_2_config.servers.push_back(ice_server_2);
-  client_2_config.type = webrtc::PeerConnectionInterface::kRelay;
-  client_2_config.presume_writable_when_fully_relayed = true;
-
-  ASSERT_TRUE(CreateTestClients(client_1_config, client_2_config));
-  // Set up the simulated delays.
-  SetSignalingDelayMs(signaling_trip_delay_ms);
-  virtual_socket_server()->set_delay_mean(media_hop_delay_ms);
-  virtual_socket_server()->UpdateDelayDistribution();
-
-  initializing_client()->SetOfferToReceiveAudioVideo(true, true);
-  initializing_client()->Negotiate();
-  // TODO(deadbeef): kIceConnectionConnected currently means both ICE and DTLS
-  // are connected. This is an important distinction. Once we have separate ICE
-  // and DTLS state, this check needs to use the DTLS state.
-  EXPECT_TRUE_SIMULATED_WAIT(
-      (receiving_client()->ice_connection_state() ==
-           webrtc::PeerConnectionInterface::kIceConnectionConnected ||
-       receiving_client()->ice_connection_state() ==
-           webrtc::PeerConnectionInterface::kIceConnectionCompleted) &&
-          (initializing_client()->ice_connection_state() ==
-               webrtc::PeerConnectionInterface::kIceConnectionConnected ||
-           initializing_client()->ice_connection_state() ==
-               webrtc::PeerConnectionInterface::kIceConnectionCompleted),
-      total_connection_time_ms, fake_clock);
-  // Need to free the clients here since they're using things we created on
-  // the stack.
-  delete set_initializing_client(nullptr);
-  delete set_receiving_client(nullptr);
-}
-
-class IceServerParsingTest : public testing::Test {
- public:
-  // Convenience for parsing a single URL.
-  bool ParseUrl(const std::string& url) {
-    return ParseUrl(url, std::string(), std::string());
-  }
-
-  bool ParseTurnUrl(const std::string& url) {
-    return ParseUrl(url, "username", "password");
-  }
-
-  bool ParseUrl(const std::string& url,
-                const std::string& username,
-                const std::string& password) {
-    return ParseUrl(
-        url, username, password,
-        PeerConnectionInterface::TlsCertPolicy::kTlsCertPolicySecure);
-  }
-
-  bool ParseUrl(const std::string& url,
-                const std::string& username,
-                const std::string& password,
-                PeerConnectionInterface::TlsCertPolicy tls_certificate_policy) {
-    PeerConnectionInterface::IceServers servers;
-    PeerConnectionInterface::IceServer server;
-    server.urls.push_back(url);
-    server.username = username;
-    server.password = password;
-    server.tls_cert_policy = tls_certificate_policy;
-    servers.push_back(server);
-    return webrtc::ParseIceServers(servers, &stun_servers_, &turn_servers_) ==
-           webrtc::RTCErrorType::NONE;
-  }
-
- protected:
-  cricket::ServerAddresses stun_servers_;
-  std::vector<cricket::RelayServerConfig> turn_servers_;
-};
-
-// Make sure all STUN/TURN prefixes are parsed correctly.
-TEST_F(IceServerParsingTest, ParseStunPrefixes) {
-  EXPECT_TRUE(ParseUrl("stun:hostname"));
-  EXPECT_EQ(1U, stun_servers_.size());
-  EXPECT_EQ(0U, turn_servers_.size());
-  stun_servers_.clear();
-
-  EXPECT_TRUE(ParseUrl("stuns:hostname"));
-  EXPECT_EQ(1U, stun_servers_.size());
-  EXPECT_EQ(0U, turn_servers_.size());
-  stun_servers_.clear();
-
-  EXPECT_TRUE(ParseTurnUrl("turn:hostname"));
-  EXPECT_EQ(0U, stun_servers_.size());
-  EXPECT_EQ(1U, turn_servers_.size());
-  EXPECT_EQ(cricket::PROTO_UDP, turn_servers_[0].ports[0].proto);
-  turn_servers_.clear();
-
-  EXPECT_TRUE(ParseTurnUrl("turns:hostname"));
-  EXPECT_EQ(0U, stun_servers_.size());
-  EXPECT_EQ(1U, turn_servers_.size());
-  EXPECT_EQ(cricket::PROTO_TLS, turn_servers_[0].ports[0].proto);
-  EXPECT_TRUE(turn_servers_[0].tls_cert_policy ==
-              cricket::TlsCertPolicy::TLS_CERT_POLICY_SECURE);
-  turn_servers_.clear();
-
-  EXPECT_TRUE(ParseUrl(
-      "turns:hostname", "username", "password",
-      PeerConnectionInterface::TlsCertPolicy::kTlsCertPolicyInsecureNoCheck));
-  EXPECT_EQ(0U, stun_servers_.size());
-  EXPECT_EQ(1U, turn_servers_.size());
-  EXPECT_TRUE(turn_servers_[0].tls_cert_policy ==
-              cricket::TlsCertPolicy::TLS_CERT_POLICY_INSECURE_NO_CHECK);
-  EXPECT_EQ(cricket::PROTO_TLS, turn_servers_[0].ports[0].proto);
-  turn_servers_.clear();
-
-  // invalid prefixes
-  EXPECT_FALSE(ParseUrl("stunn:hostname"));
-  EXPECT_FALSE(ParseUrl(":hostname"));
-  EXPECT_FALSE(ParseUrl(":"));
-  EXPECT_FALSE(ParseUrl(""));
-}
-
-TEST_F(IceServerParsingTest, VerifyDefaults) {
-  // TURNS defaults
-  EXPECT_TRUE(ParseTurnUrl("turns:hostname"));
-  EXPECT_EQ(1U, turn_servers_.size());
-  EXPECT_EQ(5349, turn_servers_[0].ports[0].address.port());
-  EXPECT_EQ(cricket::PROTO_TLS, turn_servers_[0].ports[0].proto);
-  turn_servers_.clear();
-
-  // TURN defaults
-  EXPECT_TRUE(ParseTurnUrl("turn:hostname"));
-  EXPECT_EQ(1U, turn_servers_.size());
-  EXPECT_EQ(3478, turn_servers_[0].ports[0].address.port());
-  EXPECT_EQ(cricket::PROTO_UDP, turn_servers_[0].ports[0].proto);
-  turn_servers_.clear();
-
-  // STUN defaults
-  EXPECT_TRUE(ParseUrl("stun:hostname"));
-  EXPECT_EQ(1U, stun_servers_.size());
-  EXPECT_EQ(3478, stun_servers_.begin()->port());
-  stun_servers_.clear();
-}
-
-// Check that the 6 combinations of IPv4/IPv6/hostname and with/without port
-// can be parsed correctly.
-TEST_F(IceServerParsingTest, ParseHostnameAndPort) {
-  EXPECT_TRUE(ParseUrl("stun:1.2.3.4:1234"));
-  EXPECT_EQ(1U, stun_servers_.size());
-  EXPECT_EQ("1.2.3.4", stun_servers_.begin()->hostname());
-  EXPECT_EQ(1234, stun_servers_.begin()->port());
-  stun_servers_.clear();
-
-  EXPECT_TRUE(ParseUrl("stun:[1:2:3:4:5:6:7:8]:4321"));
-  EXPECT_EQ(1U, stun_servers_.size());
-  EXPECT_EQ("1:2:3:4:5:6:7:8", stun_servers_.begin()->hostname());
-  EXPECT_EQ(4321, stun_servers_.begin()->port());
-  stun_servers_.clear();
-
-  EXPECT_TRUE(ParseUrl("stun:hostname:9999"));
-  EXPECT_EQ(1U, stun_servers_.size());
-  EXPECT_EQ("hostname", stun_servers_.begin()->hostname());
-  EXPECT_EQ(9999, stun_servers_.begin()->port());
-  stun_servers_.clear();
-
-  EXPECT_TRUE(ParseUrl("stun:1.2.3.4"));
-  EXPECT_EQ(1U, stun_servers_.size());
-  EXPECT_EQ("1.2.3.4", stun_servers_.begin()->hostname());
-  EXPECT_EQ(3478, stun_servers_.begin()->port());
-  stun_servers_.clear();
-
-  EXPECT_TRUE(ParseUrl("stun:[1:2:3:4:5:6:7:8]"));
-  EXPECT_EQ(1U, stun_servers_.size());
-  EXPECT_EQ("1:2:3:4:5:6:7:8", stun_servers_.begin()->hostname());
-  EXPECT_EQ(3478, stun_servers_.begin()->port());
-  stun_servers_.clear();
-
-  EXPECT_TRUE(ParseUrl("stun:hostname"));
-  EXPECT_EQ(1U, stun_servers_.size());
-  EXPECT_EQ("hostname", stun_servers_.begin()->hostname());
-  EXPECT_EQ(3478, stun_servers_.begin()->port());
-  stun_servers_.clear();
-
-  // Try some invalid hostname:port strings.
-  EXPECT_FALSE(ParseUrl("stun:hostname:99a99"));
-  EXPECT_FALSE(ParseUrl("stun:hostname:-1"));
-  EXPECT_FALSE(ParseUrl("stun:hostname:port:more"));
-  EXPECT_FALSE(ParseUrl("stun:hostname:port more"));
-  EXPECT_FALSE(ParseUrl("stun:hostname:"));
-  EXPECT_FALSE(ParseUrl("stun:[1:2:3:4:5:6:7:8]junk:1000"));
-  EXPECT_FALSE(ParseUrl("stun::5555"));
-  EXPECT_FALSE(ParseUrl("stun:"));
-}
-
-// Test parsing the "?transport=xxx" part of the URL.
-TEST_F(IceServerParsingTest, ParseTransport) {
-  EXPECT_TRUE(ParseTurnUrl("turn:hostname:1234?transport=tcp"));
-  EXPECT_EQ(1U, turn_servers_.size());
-  EXPECT_EQ(cricket::PROTO_TCP, turn_servers_[0].ports[0].proto);
-  turn_servers_.clear();
-
-  EXPECT_TRUE(ParseTurnUrl("turn:hostname?transport=udp"));
-  EXPECT_EQ(1U, turn_servers_.size());
-  EXPECT_EQ(cricket::PROTO_UDP, turn_servers_[0].ports[0].proto);
-  turn_servers_.clear();
-
-  EXPECT_FALSE(ParseTurnUrl("turn:hostname?transport=invalid"));
-  EXPECT_FALSE(ParseTurnUrl("turn:hostname?transport="));
-  EXPECT_FALSE(ParseTurnUrl("turn:hostname?="));
-  EXPECT_FALSE(ParseTurnUrl("?"));
-}
-
-// Test parsing ICE username contained in URL.
-TEST_F(IceServerParsingTest, ParseUsername) {
-  EXPECT_TRUE(ParseTurnUrl("turn:user@hostname"));
-  EXPECT_EQ(1U, turn_servers_.size());
-  EXPECT_EQ("user", turn_servers_[0].credentials.username);
-  turn_servers_.clear();
-
-  EXPECT_FALSE(ParseTurnUrl("turn:@hostname"));
-  EXPECT_FALSE(ParseTurnUrl("turn:username@"));
-  EXPECT_FALSE(ParseTurnUrl("turn:@"));
-  EXPECT_FALSE(ParseTurnUrl("turn:user@name@hostname"));
-}
-
-// Test that username and password from IceServer is copied into the resulting
-// RelayServerConfig.
-TEST_F(IceServerParsingTest, CopyUsernameAndPasswordFromIceServer) {
-  EXPECT_TRUE(ParseUrl("turn:hostname", "username", "password"));
-  EXPECT_EQ(1U, turn_servers_.size());
-  EXPECT_EQ("username", turn_servers_[0].credentials.username);
-  EXPECT_EQ("password", turn_servers_[0].credentials.password);
-}
-
-// Ensure that if a server has multiple URLs, each one is parsed.
-TEST_F(IceServerParsingTest, ParseMultipleUrls) {
-  PeerConnectionInterface::IceServers servers;
-  PeerConnectionInterface::IceServer server;
-  server.urls.push_back("stun:hostname");
-  server.urls.push_back("turn:hostname");
-  server.username = "foo";
-  server.password = "bar";
-  servers.push_back(server);
-  EXPECT_EQ(webrtc::RTCErrorType::NONE,
-            webrtc::ParseIceServers(servers, &stun_servers_, &turn_servers_));
-  EXPECT_EQ(1U, stun_servers_.size());
-  EXPECT_EQ(1U, turn_servers_.size());
-}
-
-// Ensure that TURN servers are given unique priorities,
-// so that their resulting candidates have unique priorities.
-TEST_F(IceServerParsingTest, TurnServerPrioritiesUnique) {
-  PeerConnectionInterface::IceServers servers;
-  PeerConnectionInterface::IceServer server;
-  server.urls.push_back("turn:hostname");
-  server.urls.push_back("turn:hostname2");
-  server.username = "foo";
-  server.password = "bar";
-  servers.push_back(server);
-  EXPECT_EQ(webrtc::RTCErrorType::NONE,
-            webrtc::ParseIceServers(servers, &stun_servers_, &turn_servers_));
-  EXPECT_EQ(2U, turn_servers_.size());
-  EXPECT_NE(turn_servers_[0].priority, turn_servers_[1].priority);
-}
-
-#endif // if !defined(THREAD_SANITIZER)
-
-}  // namespace