text: /*
   *  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 <memory>
  #include <sstream>
  #include <string>
  #include <utility>
  
  #include "webrtc/api/audiotrack.h"
  #include "webrtc/api/jsepsessiondescription.h"
  #include "webrtc/api/mediastream.h"
  #include "webrtc/api/mediastreaminterface.h"
  #include "webrtc/api/peerconnection.h"
  #include "webrtc/api/peerconnectioninterface.h"
  #include "webrtc/api/rtpreceiverinterface.h"
  #include "webrtc/api/rtpsenderinterface.h"
  #include "webrtc/api/streamcollection.h"
  #include "webrtc/api/test/fakeconstraints.h"
  #include "webrtc/api/test/fakertccertificategenerator.h"
  #include "webrtc/api/test/fakevideotracksource.h"
  #include "webrtc/api/test/mockpeerconnectionobservers.h"
  #include "webrtc/api/test/testsdpstrings.h"
  #include "webrtc/api/videocapturertracksource.h"
  #include "webrtc/api/videotrack.h"
  #include "webrtc/base/gunit.h"
  #include "webrtc/base/ssladapter.h"
  #include "webrtc/base/sslstreamadapter.h"
  #include "webrtc/base/stringutils.h"
  #include "webrtc/base/thread.h"
  #include "webrtc/media/base/fakevideocapturer.h"
  #include "webrtc/media/sctp/sctptransportinternal.h"
  #include "webrtc/p2p/base/fakeportallocator.h"
  #include "webrtc/p2p/base/faketransportcontroller.h"
  #include "webrtc/pc/mediasession.h"
  #include "webrtc/test/gmock.h"
  
  #ifdef WEBRTC_ANDROID
  #include "webrtc/api/test/androidtestinitializer.h"
  #endif
  
  static const char kStreamLabel1[] = "local_stream_1";
  static const char kStreamLabel2[] = "local_stream_2";
  static const char kStreamLabel3[] = "local_stream_3";
  static const int kDefaultStunPort = 3478;
  static const char kStunAddressOnly[] = "stun:address";
  static const char kStunInvalidPort[] = "stun:address:-1";
  static const char kStunAddressPortAndMore1[] = "stun:address:port:more";
  static const char kStunAddressPortAndMore2[] = "stun:address:port more";
  static const char kTurnIceServerUri[] = "turn:user@turn.example.org";
  static const char kTurnUsername[] = "user";
  static const char kTurnPassword[] = "password";
  static const char kTurnHostname[] = "turn.example.org";
  static const uint32_t kTimeout = 10000U;
  
  static const char kStreams[][8] = {"stream1", "stream2"};
  static const char kAudioTracks[][32] = {"audiotrack0", "audiotrack1"};
  static const char kVideoTracks[][32] = {"videotrack0", "videotrack1"};
  
  static const char kRecvonly[] = "recvonly";
  static const char kSendrecv[] = "sendrecv";
  
  // Reference SDP with a MediaStream with label "stream1" and audio track with
  // id "audio_1" and a video track with id "video_1;
  static const char kSdpStringWithStream1[] =
      "v=0\r\n"
      "o=- 0 0 IN IP4 127.0.0.1\r\n"
      "s=-\r\n"
      "t=0 0\r\n"
      "m=audio 1 RTP/AVPF 103\r\n"
      "a=ice-ufrag:e5785931\r\n"
      "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
      "a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
      "BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
      "a=mid:audio\r\n"
      "a=sendrecv\r\n"
      "a=rtcp-mux\r\n"
      "a=rtpmap:103 ISAC/16000\r\n"
      "a=ssrc:1 cname:stream1\r\n"
      "a=ssrc:1 mslabel:stream1\r\n"
      "a=ssrc:1 label:audiotrack0\r\n"
      "m=video 1 RTP/AVPF 120\r\n"
      "a=ice-ufrag:e5785931\r\n"
      "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
      "a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
      "BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
      "a=mid:video\r\n"
      "a=sendrecv\r\n"
      "a=rtcp-mux\r\n"
      "a=rtpmap:120 VP8/90000\r\n"
      "a=ssrc:2 cname:stream1\r\n"
      "a=ssrc:2 mslabel:stream1\r\n"
      "a=ssrc:2 label:videotrack0\r\n";
  
  // Reference SDP with a MediaStream with label "stream1" and audio track with
  // id "audio_1";
  static const char kSdpStringWithStream1AudioTrackOnly[] =
      "v=0\r\n"
      "o=- 0 0 IN IP4 127.0.0.1\r\n"
      "s=-\r\n"
      "t=0 0\r\n"
      "m=audio 1 RTP/AVPF 103\r\n"
      "a=ice-ufrag:e5785931\r\n"
      "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
      "a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
      "BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
      "a=mid:audio\r\n"
      "a=sendrecv\r\n"
      "a=rtpmap:103 ISAC/16000\r\n"
      "a=ssrc:1 cname:stream1\r\n"
      "a=ssrc:1 mslabel:stream1\r\n"
      "a=ssrc:1 label:audiotrack0\r\n"
      "a=rtcp-mux\r\n";
  
  // Reference SDP with two MediaStreams with label "stream1" and "stream2. Each
  // MediaStreams have one audio track and one video track.
  // This uses MSID.
  static const char kSdpStringWithStream1And2[] =
      "v=0\r\n"
      "o=- 0 0 IN IP4 127.0.0.1\r\n"
      "s=-\r\n"
      "t=0 0\r\n"
      "a=msid-semantic: WMS stream1 stream2\r\n"
      "m=audio 1 RTP/AVPF 103\r\n"
      "a=ice-ufrag:e5785931\r\n"
      "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
      "a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
      "BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
      "a=mid:audio\r\n"
      "a=sendrecv\r\n"
      "a=rtcp-mux\r\n"
      "a=rtpmap:103 ISAC/16000\r\n"
      "a=ssrc:1 cname:stream1\r\n"
      "a=ssrc:1 msid:stream1 audiotrack0\r\n"
      "a=ssrc:3 cname:stream2\r\n"
      "a=ssrc:3 msid:stream2 audiotrack1\r\n"
      "m=video 1 RTP/AVPF 120\r\n"
      "a=ice-ufrag:e5785931\r\n"
      "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
      "a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
      "BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
      "a=mid:video\r\n"
      "a=sendrecv\r\n"
      "a=rtcp-mux\r\n"
      "a=rtpmap:120 VP8/0\r\n"
      "a=ssrc:2 cname:stream1\r\n"
      "a=ssrc:2 msid:stream1 videotrack0\r\n"
      "a=ssrc:4 cname:stream2\r\n"
      "a=ssrc:4 msid:stream2 videotrack1\r\n";
  
  // Reference SDP without MediaStreams. Msid is not supported.
  static const char kSdpStringWithoutStreams[] =
      "v=0\r\n"
      "o=- 0 0 IN IP4 127.0.0.1\r\n"
      "s=-\r\n"
      "t=0 0\r\n"
      "m=audio 1 RTP/AVPF 103\r\n"
      "a=ice-ufrag:e5785931\r\n"
      "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
      "a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
      "BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
      "a=mid:audio\r\n"
      "a=sendrecv\r\n"
      "a=rtcp-mux\r\n"
      "a=rtpmap:103 ISAC/16000\r\n"
      "m=video 1 RTP/AVPF 120\r\n"
      "a=ice-ufrag:e5785931\r\n"
      "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
      "a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
      "BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
      "a=mid:video\r\n"
      "a=sendrecv\r\n"
      "a=rtcp-mux\r\n"
      "a=rtpmap:120 VP8/90000\r\n";
  
  // Reference SDP without MediaStreams. Msid is supported.
  static const char kSdpStringWithMsidWithoutStreams[] =
      "v=0\r\n"
      "o=- 0 0 IN IP4 127.0.0.1\r\n"
      "s=-\r\n"
      "t=0 0\r\n"
      "a=msid-semantic: WMS\r\n"
      "m=audio 1 RTP/AVPF 103\r\n"
      "a=ice-ufrag:e5785931\r\n"
      "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
      "a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
      "BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
      "a=mid:audio\r\n"
      "a=sendrecv\r\n"
      "a=rtcp-mux\r\n"
      "a=rtpmap:103 ISAC/16000\r\n"
      "m=video 1 RTP/AVPF 120\r\n"
      "a=ice-ufrag:e5785931\r\n"
      "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
      "a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
      "BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
      "a=mid:video\r\n"
      "a=sendrecv\r\n"
      "a=rtcp-mux\r\n"
      "a=rtpmap:120 VP8/90000\r\n";
  
  // Reference SDP without MediaStreams and audio only.
  static const char kSdpStringWithoutStreamsAudioOnly[] =
      "v=0\r\n"
      "o=- 0 0 IN IP4 127.0.0.1\r\n"
      "s=-\r\n"
      "t=0 0\r\n"
      "m=audio 1 RTP/AVPF 103\r\n"
      "a=ice-ufrag:e5785931\r\n"
      "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
      "a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
      "BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
      "a=mid:audio\r\n"
      "a=sendrecv\r\n"
      "a=rtcp-mux\r\n"
      "a=rtpmap:103 ISAC/16000\r\n";
  
  // Reference SENDONLY SDP without MediaStreams. Msid is not supported.
  static const char kSdpStringSendOnlyWithoutStreams[] =
      "v=0\r\n"
      "o=- 0 0 IN IP4 127.0.0.1\r\n"
      "s=-\r\n"
      "t=0 0\r\n"
      "m=audio 1 RTP/AVPF 103\r\n"
      "a=ice-ufrag:e5785931\r\n"
      "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
      "a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
      "BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
      "a=mid:audio\r\n"
      "a=sendrecv\r\n"
      "a=sendonly\r\n"
      "a=rtcp-mux\r\n"
      "a=rtpmap:103 ISAC/16000\r\n"
      "m=video 1 RTP/AVPF 120\r\n"
      "a=ice-ufrag:e5785931\r\n"
      "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
      "a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
      "BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
      "a=mid:video\r\n"
      "a=sendrecv\r\n"
      "a=sendonly\r\n"
      "a=rtcp-mux\r\n"
      "a=rtpmap:120 VP8/90000\r\n";
  
  static const char kSdpStringInit[] =
      "v=0\r\n"
      "o=- 0 0 IN IP4 127.0.0.1\r\n"
      "s=-\r\n"
      "t=0 0\r\n"
      "a=msid-semantic: WMS\r\n";
  
  static const char kSdpStringAudio[] =
      "m=audio 1 RTP/AVPF 103\r\n"
      "a=ice-ufrag:e5785931\r\n"
      "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
      "a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
      "BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
      "a=mid:audio\r\n"
      "a=sendrecv\r\n"
      "a=rtcp-mux\r\n"
      "a=rtpmap:103 ISAC/16000\r\n";
  
  static const char kSdpStringVideo[] =
      "m=video 1 RTP/AVPF 120\r\n"
      "a=ice-ufrag:e5785931\r\n"
      "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
      "a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
      "BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
      "a=mid:video\r\n"
      "a=sendrecv\r\n"
      "a=rtcp-mux\r\n"
      "a=rtpmap:120 VP8/90000\r\n";
  
  static const char kSdpStringMs1Audio0[] =
      "a=ssrc:1 cname:stream1\r\n"
      "a=ssrc:1 msid:stream1 audiotrack0\r\n";
  
  static const char kSdpStringMs1Video0[] =
      "a=ssrc:2 cname:stream1\r\n"
      "a=ssrc:2 msid:stream1 videotrack0\r\n";
  
  static const char kSdpStringMs1Audio1[] =
      "a=ssrc:3 cname:stream1\r\n"
      "a=ssrc:3 msid:stream1 audiotrack1\r\n";
  
  static const char kSdpStringMs1Video1[] =
      "a=ssrc:4 cname:stream1\r\n"
      "a=ssrc:4 msid:stream1 videotrack1\r\n";
  
  static const char kDtlsSdesFallbackSdp[] =
      "v=0\r\n"
      "o=xxxxxx 7 2 IN IP4 0.0.0.0\r\n"
      "s=-\r\n"
      "c=IN IP4 0.0.0.0\r\n"
      "t=0 0\r\n"
      "a=group:BUNDLE audio\r\n"
      "a=msid-semantic: WMS\r\n"
      "m=audio 1 RTP/SAVPF 0\r\n"
      "a=sendrecv\r\n"
      "a=rtcp-mux\r\n"
      "a=mid:audio\r\n"
      "a=ssrc:1 cname:stream1\r\n"
      "a=ssrc:1 mslabel:stream1\r\n"
      "a=ssrc:1 label:audiotrack0\r\n"
      "a=ice-ufrag:e5785931\r\n"
      "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
      "a=rtpmap:0 pcmu/8000\r\n"
      "a=fingerprint:sha-1 "
      "4A:AD:B9:B1:3F:82:18:3B:54:02:12:DF:3E:5D:49:6B:19:E5:7C:AB\r\n"
      "a=setup:actpass\r\n"
      "a=crypto:1 AES_CM_128_HMAC_SHA1_32 "
      "inline:NzB4d1BINUAvLEw6UzF3WSJ+PSdFcGdUJShpX1Zj|2^20|1:32 "
      "dummy_session_params\r\n";
  
  #define MAYBE_SKIP_TEST(feature)                    \
    if (!(feature())) {                               \
      LOG(LS_INFO) << "Feature disabled... skipping"; \
      return;                                         \
    }
  
  using ::testing::Exactly;
  using cricket::StreamParams;
  using webrtc::AudioSourceInterface;
  using webrtc::AudioTrack;
  using webrtc::AudioTrackInterface;
  using webrtc::DataBuffer;
  using webrtc::DataChannelInterface;
  using webrtc::FakeConstraints;
  using webrtc::IceCandidateInterface;
  using webrtc::JsepSessionDescription;
  using webrtc::MediaConstraintsInterface;
  using webrtc::MediaStream;
  using webrtc::MediaStreamInterface;
  using webrtc::MediaStreamTrackInterface;
  using webrtc::MockCreateSessionDescriptionObserver;
  using webrtc::MockDataChannelObserver;
  using webrtc::MockSetSessionDescriptionObserver;
  using webrtc::MockStatsObserver;
  using webrtc::NotifierInterface;
  using webrtc::ObserverInterface;
  using webrtc::PeerConnectionInterface;
  using webrtc::PeerConnectionObserver;
  using webrtc::RTCError;
  using webrtc::RTCErrorType;
  using webrtc::RtpReceiverInterface;
  using webrtc::RtpSenderInterface;
  using webrtc::SdpParseError;
  using webrtc::SessionDescriptionInterface;
  using webrtc::StreamCollection;
  using webrtc::StreamCollectionInterface;
  using webrtc::VideoTrackSourceInterface;
  using webrtc::VideoTrack;
  using webrtc::VideoTrackInterface;
  
  typedef PeerConnectionInterface::RTCOfferAnswerOptions RTCOfferAnswerOptions;
  
  namespace {
  
  // Gets the first ssrc of given content type from the ContentInfo.
  bool GetFirstSsrc(const cricket::ContentInfo* content_info, int* ssrc) {
    if (!content_info || !ssrc) {
      return false;
    }
    const cricket::MediaContentDescription* media_desc =
        static_cast<const cricket::MediaContentDescription*>(
            content_info->description);
    if (!media_desc || media_desc->streams().empty()) {
      return false;
    }
    *ssrc = media_desc->streams().begin()->first_ssrc();
    return true;
  }
  
  // Get the ufrags out of an SDP blob. Useful for testing ICE restart
  // behavior.
  std::vector<std::string> GetUfrags(
      const webrtc::SessionDescriptionInterface* desc) {
    std::vector<std::string> ufrags;
    for (const cricket::TransportInfo& info :
         desc->description()->transport_infos()) {
      ufrags.push_back(info.description.ice_ufrag);
    }
    return ufrags;
  }
  
  void SetSsrcToZero(std::string* sdp) {
    const char kSdpSsrcAtribute[] = "a=ssrc:";
    const char kSdpSsrcAtributeZero[] = "a=ssrc:0";
    size_t ssrc_pos = 0;
    while ((ssrc_pos = sdp->find(kSdpSsrcAtribute, ssrc_pos)) !=
        std::string::npos) {
      size_t end_ssrc = sdp->find(" ", ssrc_pos);
      sdp->replace(ssrc_pos, end_ssrc - ssrc_pos, kSdpSsrcAtributeZero);
      ssrc_pos = end_ssrc;
    }
  }
  
  // Check if |streams| contains the specified track.
  bool ContainsTrack(const std::vector<cricket::StreamParams>& streams,
                     const std::string& stream_label,
                     const std::string& track_id) {
    for (const cricket::StreamParams& params : streams) {
      if (params.sync_label == stream_label && params.id == track_id) {
        return true;
      }
    }
    return false;
  }
  
  // Check if |senders| contains the specified sender, by id.
  bool ContainsSender(
      const std::vector<rtc::scoped_refptr<RtpSenderInterface>>& senders,
      const std::string& id) {
    for (const auto& sender : senders) {
      if (sender->id() == id) {
        return true;
      }
    }
    return false;
  }
  
  // Check if |senders| contains the specified sender, by id and stream id.
  bool ContainsSender(
      const std::vector<rtc::scoped_refptr<RtpSenderInterface>>& senders,
      const std::string& id,
      const std::string& stream_id) {
    for (const auto& sender : senders) {
      if (sender->id() == id && sender->stream_ids()[0] == stream_id) {
        return true;
      }
    }
    return false;
  }
  
  // Create a collection of streams.
  // CreateStreamCollection(1) creates a collection that
  // correspond to kSdpStringWithStream1.
  // CreateStreamCollection(2) correspond to kSdpStringWithStream1And2.
  rtc::scoped_refptr<StreamCollection> CreateStreamCollection(
      int number_of_streams,
      int tracks_per_stream) {
    rtc::scoped_refptr<StreamCollection> local_collection(
        StreamCollection::Create());
  
    for (int i = 0; i < number_of_streams; ++i) {
      rtc::scoped_refptr<webrtc::MediaStreamInterface> stream(
          webrtc::MediaStream::Create(kStreams[i]));
  
      for (int j = 0; j < tracks_per_stream; ++j) {
        // Add a local audio track.
        rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
            webrtc::AudioTrack::Create(kAudioTracks[i * tracks_per_stream + j],
                                       nullptr));
        stream->AddTrack(audio_track);
  
        // Add a local video track.
        rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track(
            webrtc::VideoTrack::Create(kVideoTracks[i * tracks_per_stream + j],
                                       webrtc::FakeVideoTrackSource::Create()));
        stream->AddTrack(video_track);
      }
  
      local_collection->AddStream(stream);
    }
    return local_collection;
  }
  
  // Check equality of StreamCollections.
  bool CompareStreamCollections(StreamCollectionInterface* s1,
                                StreamCollectionInterface* s2) {
    if (s1 == nullptr || s2 == nullptr || s1->count() != s2->count()) {
      return false;
    }
  
    for (size_t i = 0; i != s1->count(); ++i) {
      if (s1->at(i)->label() != s2->at(i)->label()) {
        return false;
      }
      webrtc::AudioTrackVector audio_tracks1 = s1->at(i)->GetAudioTracks();
      webrtc::AudioTrackVector audio_tracks2 = s2->at(i)->GetAudioTracks();
      webrtc::VideoTrackVector video_tracks1 = s1->at(i)->GetVideoTracks();
      webrtc::VideoTrackVector video_tracks2 = s2->at(i)->GetVideoTracks();
  
      if (audio_tracks1.size() != audio_tracks2.size()) {
        return false;
      }
      for (size_t j = 0; j != audio_tracks1.size(); ++j) {
        if (audio_tracks1[j]->id() != audio_tracks2[j]->id()) {
          return false;
        }
      }
      if (video_tracks1.size() != video_tracks2.size()) {
        return false;
      }
      for (size_t j = 0; j != video_tracks1.size(); ++j) {
        if (video_tracks1[j]->id() != video_tracks2[j]->id()) {
          return false;
        }
      }
    }
    return true;
  }
  
  // Helper class to test Observer.
  class MockTrackObserver : public ObserverInterface {
   public:
    explicit MockTrackObserver(NotifierInterface* notifier)
        : notifier_(notifier) {
      notifier_->RegisterObserver(this);
    }
  
    ~MockTrackObserver() { Unregister(); }
  
    void Unregister() {
      if (notifier_) {
        notifier_->UnregisterObserver(this);
        notifier_ = nullptr;
      }
    }
  
    MOCK_METHOD0(OnChanged, void());
  
   private:
    NotifierInterface* notifier_;
  };
  
  class MockPeerConnectionObserver : public PeerConnectionObserver {
   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;
  
    MockPeerConnectionObserver() : remote_streams_(StreamCollection::Create()) {}
    virtual ~MockPeerConnectionObserver() {
    }
    void SetPeerConnectionInterface(PeerConnectionInterface* pc) {
      pc_ = pc;
      if (pc) {
        state_ = pc_->signaling_state();
      }
    }
    void OnSignalingChange(
        PeerConnectionInterface::SignalingState new_state) override {
      EXPECT_EQ(pc_->signaling_state(), new_state);
      state_ = new_state;
    }
    // TODO(bemasc): Remove this once callers transition to OnIceGatheringChange.
    virtual void OnStateChange(StateType state_changed) {
      if (pc_.get() == NULL)
        return;
      switch (state_changed) {
        case kSignalingState:
          // OnSignalingChange and OnStateChange(kSignalingState) should always
          // be called approximately simultaneously.  To ease testing, we require
          // that they always be called in that order.  This check verifies
          // that OnSignalingChange has just been called.
          EXPECT_EQ(pc_->signaling_state(), state_);
          break;
        case kIceState:
          ADD_FAILURE();
          break;
        default:
          ADD_FAILURE();
          break;
      }
    }
  
    MediaStreamInterface* RemoteStream(const std::string& label) {
      return remote_streams_->find(label);
    }
    StreamCollectionInterface* remote_streams() const { return remote_streams_; }
    void OnAddStream(rtc::scoped_refptr<MediaStreamInterface> stream) override {
      last_added_stream_ = stream;
      remote_streams_->AddStream(stream);
    }
    void OnRemoveStream(
        rtc::scoped_refptr<MediaStreamInterface> stream) override {
      last_removed_stream_ = stream;
      remote_streams_->RemoveStream(stream);
    }
    void OnRenegotiationNeeded() override { renegotiation_needed_ = true; }
    void OnDataChannel(
        rtc::scoped_refptr<DataChannelInterface> data_channel) override {
      last_datachannel_ = data_channel;
    }
  
    void OnIceConnectionChange(
        PeerConnectionInterface::IceConnectionState new_state) override {
      EXPECT_EQ(pc_->ice_connection_state(), new_state);
      callback_triggered_ = true;
    }
    void OnIceGatheringChange(
        PeerConnectionInterface::IceGatheringState new_state) override {
      EXPECT_EQ(pc_->ice_gathering_state(), new_state);
      ice_complete_ = new_state == PeerConnectionInterface::kIceGatheringComplete;
      callback_triggered_ = true;
    }
    void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override {
      EXPECT_NE(PeerConnectionInterface::kIceGatheringNew,
                pc_->ice_gathering_state());
  
      std::string sdp;
      EXPECT_TRUE(candidate->ToString(&sdp));
      EXPECT_LT(0u, sdp.size());
      last_candidate_.reset(webrtc::CreateIceCandidate(candidate->sdp_mid(),
          candidate->sdp_mline_index(), sdp, NULL));
      EXPECT_TRUE(last_candidate_.get() != NULL);
      callback_triggered_ = true;
    }
  
    void OnIceCandidatesRemoved(
        const std::vector<cricket::Candidate>& candidates) override {
      callback_triggered_ = true;
    }
  
    void OnIceConnectionReceivingChange(bool receiving) override {
      callback_triggered_ = true;
    }
  
    void OnAddTrack(
        rtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver,
        const std::vector<rtc::scoped_refptr<webrtc::MediaStreamInterface>>&
            streams) override {
      EXPECT_TRUE(receiver != nullptr);
      num_added_tracks_++;
      last_added_track_label_ = receiver->id();
    }
  
    // Returns the label of the last added stream.
    // Empty string if no stream have been added.
    std::string GetLastAddedStreamLabel() {
      if (last_added_stream_.get())
        return last_added_stream_->label();
      return "";
    }
    std::string GetLastRemovedStreamLabel() {
      if (last_removed_stream_.get())
        return last_removed_stream_->label();
      return "";
    }
  
    rtc::scoped_refptr<PeerConnectionInterface> pc_;
    PeerConnectionInterface::SignalingState state_;
    std::unique_ptr<IceCandidateInterface> last_candidate_;
    rtc::scoped_refptr<DataChannelInterface> last_datachannel_;
    rtc::scoped_refptr<StreamCollection> remote_streams_;
    bool renegotiation_needed_ = false;
    bool ice_complete_ = false;
    bool callback_triggered_ = false;
    int num_added_tracks_ = 0;
    std::string last_added_track_label_;
  
   private:
    rtc::scoped_refptr<MediaStreamInterface> last_added_stream_;
    rtc::scoped_refptr<MediaStreamInterface> last_removed_stream_;
  };
  
  }  // namespace
  
  // The PeerConnectionMediaConfig tests below verify that configuration
  // and constraints are propagated into the MediaConfig passed to
  // CreateMediaController. These settings are intended for MediaChannel
  // constructors, but that is not exercised by these unittest.
  class PeerConnectionFactoryForTest : public webrtc::PeerConnectionFactory {
   public:
    webrtc::MediaControllerInterface* CreateMediaController(
        const cricket::MediaConfig& config,
        webrtc::RtcEventLog* event_log) const override {
      create_media_controller_called_ = true;
      create_media_controller_config_ = config;
  
      webrtc::MediaControllerInterface* mc =
          PeerConnectionFactory::CreateMediaController(config, event_log);
      EXPECT_TRUE(mc != nullptr);
      return mc;
    }
  
    cricket::TransportController* CreateTransportController(
        cricket::PortAllocator* port_allocator,
        bool redetermine_role_on_ice_restart) override {
      transport_controller = new cricket::TransportController(
          rtc::Thread::Current(), rtc::Thread::Current(), port_allocator,
          redetermine_role_on_ice_restart);
      return transport_controller;
    }
  
    cricket::TransportController* transport_controller;
    // Mutable, so they can be modified in the above const-declared method.
    mutable bool create_media_controller_called_ = false;
    mutable cricket::MediaConfig create_media_controller_config_;
  };
  
  class PeerConnectionInterfaceTest : public testing::Test {
   protected:
    PeerConnectionInterfaceTest() {
  #ifdef WEBRTC_ANDROID
      webrtc::InitializeAndroidObjects();
  #endif
    }
  
    virtual void SetUp() {
      pc_factory_ = webrtc::CreatePeerConnectionFactory(
          rtc::Thread::Current(), rtc::Thread::Current(), rtc::Thread::Current(),
          nullptr, nullptr, nullptr);
      ASSERT_TRUE(pc_factory_);
      pc_factory_for_test_ =
          new rtc::RefCountedObject<PeerConnectionFactoryForTest>();
      pc_factory_for_test_->Initialize();
    }
  
    void CreatePeerConnection() {
      CreatePeerConnection(PeerConnectionInterface::RTCConfiguration(), nullptr);
    }
  
    // DTLS does not work in a loopback call, so is disabled for most of the
    // tests in this file.
    void CreatePeerConnectionWithoutDtls() {
      FakeConstraints no_dtls_constraints;
      no_dtls_constraints.AddMandatory(
          webrtc::MediaConstraintsInterface::kEnableDtlsSrtp, false);
  
      CreatePeerConnection(PeerConnectionInterface::RTCConfiguration(),
                           &no_dtls_constraints);
    }
  
    void CreatePeerConnection(webrtc::MediaConstraintsInterface* constraints) {
      CreatePeerConnection(PeerConnectionInterface::RTCConfiguration(),
                           constraints);
    }
  
    void CreatePeerConnectionWithIceTransportsType(
        PeerConnectionInterface::IceTransportsType type) {
      PeerConnectionInterface::RTCConfiguration config;
      config.type = type;
      return CreatePeerConnection(config, nullptr);
    }
  
    void CreatePeerConnectionWithIceServer(const std::string& uri,
                                           const std::string& password) {
      PeerConnectionInterface::RTCConfiguration config;
      PeerConnectionInterface::IceServer server;
      server.uri = uri;
      server.password = password;
      config.servers.push_back(server);
      CreatePeerConnection(config, nullptr);
    }
  
    void CreatePeerConnection(PeerConnectionInterface::RTCConfiguration config,
                              webrtc::MediaConstraintsInterface* constraints) {
      std::unique_ptr<cricket::FakePortAllocator> port_allocator(
          new cricket::FakePortAllocator(rtc::Thread::Current(), nullptr));
      port_allocator_ = port_allocator.get();
  
      std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator;
      bool dtls;
      if (FindConstraint(constraints,
                         webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
                         &dtls,
                         nullptr) && dtls) {
        fake_certificate_generator_ = new FakeRTCCertificateGenerator();
        cert_generator.reset(fake_certificate_generator_);
      }
      pc_ = pc_factory_->CreatePeerConnection(
          config, constraints, std::move(port_allocator),
          std::move(cert_generator), &observer_);
      ASSERT_TRUE(pc_.get() != NULL);
      observer_.SetPeerConnectionInterface(pc_.get());
      EXPECT_EQ(PeerConnectionInterface::kStable, observer_.state_);
    }
  
    void CreatePeerConnectionExpectFail(const std::string& uri) {
      PeerConnectionInterface::RTCConfiguration config;
      PeerConnectionInterface::IceServer server;
      server.uri = uri;
      config.servers.push_back(server);
  
      rtc::scoped_refptr<PeerConnectionInterface> pc;
      pc = pc_factory_->CreatePeerConnection(config, nullptr, nullptr, nullptr,
                                             &observer_);
      EXPECT_EQ(nullptr, pc);
    }
  
    void CreatePeerConnectionWithDifferentConfigurations() {
      CreatePeerConnectionWithIceServer(kStunAddressOnly, "");
      EXPECT_EQ(1u, port_allocator_->stun_servers().size());
      EXPECT_EQ(0u, port_allocator_->turn_servers().size());
      EXPECT_EQ("address", port_allocator_->stun_servers().begin()->hostname());
      EXPECT_EQ(kDefaultStunPort,
                port_allocator_->stun_servers().begin()->port());
  
      CreatePeerConnectionExpectFail(kStunInvalidPort);
      CreatePeerConnectionExpectFail(kStunAddressPortAndMore1);
      CreatePeerConnectionExpectFail(kStunAddressPortAndMore2);
  
      CreatePeerConnectionWithIceServer(kTurnIceServerUri, kTurnPassword);
      EXPECT_EQ(0u, port_allocator_->stun_servers().size());
      EXPECT_EQ(1u, port_allocator_->turn_servers().size());
      EXPECT_EQ(kTurnUsername,
                port_allocator_->turn_servers()[0].credentials.username);
      EXPECT_EQ(kTurnPassword,
                port_allocator_->turn_servers()[0].credentials.password);
      EXPECT_EQ(kTurnHostname,
                port_allocator_->turn_servers()[0].ports[0].address.hostname());
    }
  
    void ReleasePeerConnection() {
      pc_ = NULL;
      observer_.SetPeerConnectionInterface(NULL);
    }
  
    void AddVideoStream(const std::string& label) {
      // Create a local stream.
      rtc::scoped_refptr<MediaStreamInterface> stream(
          pc_factory_->CreateLocalMediaStream(label));
      rtc::scoped_refptr<VideoTrackSourceInterface> video_source(
          pc_factory_->CreateVideoSource(new cricket::FakeVideoCapturer(), NULL));
      rtc::scoped_refptr<VideoTrackInterface> video_track(
          pc_factory_->CreateVideoTrack(label + "v0", video_source));
      stream->AddTrack(video_track.get());
      EXPECT_TRUE(pc_->AddStream(stream));
      EXPECT_TRUE_WAIT(observer_.renegotiation_needed_, kTimeout);
      observer_.renegotiation_needed_ = false;
    }
  
    void AddVoiceStream(const std::string& label) {
      // Create a local stream.
      rtc::scoped_refptr<MediaStreamInterface> stream(
          pc_factory_->CreateLocalMediaStream(label));
      rtc::scoped_refptr<AudioTrackInterface> audio_track(
          pc_factory_->CreateAudioTrack(label + "a0", NULL));
      stream->AddTrack(audio_track.get());
      EXPECT_TRUE(pc_->AddStream(stream));
      EXPECT_TRUE_WAIT(observer_.renegotiation_needed_, kTimeout);
      observer_.renegotiation_needed_ = false;
    }
  
    void AddAudioVideoStream(const std::string& stream_label,
                             const std::string& audio_track_label,
                             const std::string& video_track_label) {
      // Create a local stream.
      rtc::scoped_refptr<MediaStreamInterface> stream(
          pc_factory_->CreateLocalMediaStream(stream_label));
      rtc::scoped_refptr<AudioTrackInterface> audio_track(
          pc_factory_->CreateAudioTrack(
              audio_track_label, static_cast<AudioSourceInterface*>(NULL)));
      stream->AddTrack(audio_track.get());
      rtc::scoped_refptr<VideoTrackInterface> video_track(
          pc_factory_->CreateVideoTrack(
              video_track_label,
              pc_factory_->CreateVideoSource(new cricket::FakeVideoCapturer())));
      stream->AddTrack(video_track.get());
      EXPECT_TRUE(pc_->AddStream(stream));
      EXPECT_TRUE_WAIT(observer_.renegotiation_needed_, kTimeout);
      observer_.renegotiation_needed_ = false;
    }
  
    bool DoCreateOfferAnswer(std::unique_ptr<SessionDescriptionInterface>* desc,
                             bool offer,
                             MediaConstraintsInterface* constraints) {
      rtc::scoped_refptr<MockCreateSessionDescriptionObserver>
          observer(new rtc::RefCountedObject<
              MockCreateSessionDescriptionObserver>());
      if (offer) {
        pc_->CreateOffer(observer, constraints);
      } else {
        pc_->CreateAnswer(observer, constraints);
      }
      EXPECT_EQ_WAIT(true, observer->called(), kTimeout);
      desc->reset(observer->release_desc());
      return observer->result();
    }
  
    bool DoCreateOffer(std::unique_ptr<SessionDescriptionInterface>* desc,
                       MediaConstraintsInterface* constraints) {
      return DoCreateOfferAnswer(desc, true, constraints);
    }
  
    bool DoCreateAnswer(std::unique_ptr<SessionDescriptionInterface>* desc,
                        MediaConstraintsInterface* constraints) {
      return DoCreateOfferAnswer(desc, false, constraints);
    }
  
    bool DoSetSessionDescription(SessionDescriptionInterface* desc, bool local) {
      rtc::scoped_refptr<MockSetSessionDescriptionObserver>
          observer(new rtc::RefCountedObject<
              MockSetSessionDescriptionObserver>());
      if (local) {
        pc_->SetLocalDescription(observer, desc);
      } else {
        pc_->SetRemoteDescription(observer, desc);
      }
      if (pc_->signaling_state() != PeerConnectionInterface::kClosed) {
        EXPECT_EQ_WAIT(true, observer->called(), kTimeout);
      }
      return observer->result();
    }
  
    bool DoSetLocalDescription(SessionDescriptionInterface* desc) {
      return DoSetSessionDescription(desc, true);
    }
  
    bool DoSetRemoteDescription(SessionDescriptionInterface* desc) {
      return DoSetSessionDescription(desc, false);
    }
  
    // Calls PeerConnection::GetStats and check the return value.
    // It does not verify the values in the StatReports since a RTCP packet might
    // be required.
    bool DoGetStats(MediaStreamTrackInterface* track) {
      rtc::scoped_refptr<MockStatsObserver> observer(
          new rtc::RefCountedObject<MockStatsObserver>());
      if (!pc_->GetStats(
          observer, track, PeerConnectionInterface::kStatsOutputLevelStandard))
        return false;
      EXPECT_TRUE_WAIT(observer->called(), kTimeout);
      return observer->called();
    }
  
    void InitiateCall() {
      CreatePeerConnectionWithoutDtls();
      // Create a local stream with audio&video tracks.
      AddAudioVideoStream(kStreamLabel1, "audio_label", "video_label");
      CreateOfferReceiveAnswer();
    }
  
    // Verify that RTP Header extensions has been negotiated for audio and video.
    void VerifyRemoteRtpHeaderExtensions() {
      const cricket::MediaContentDescription* desc =
          cricket::GetFirstAudioContentDescription(
              pc_->remote_description()->description());
      ASSERT_TRUE(desc != NULL);
      EXPECT_GT(desc->rtp_header_extensions().size(), 0u);
  
      desc = cricket::GetFirstVideoContentDescription(
          pc_->remote_description()->description());
      ASSERT_TRUE(desc != NULL);
      EXPECT_GT(desc->rtp_header_extensions().size(), 0u);
    }
  
    void CreateOfferAsRemoteDescription() {
      std::unique_ptr<SessionDescriptionInterface> offer;
      ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
      std::string sdp;
      EXPECT_TRUE(offer->ToString(&sdp));
      SessionDescriptionInterface* remote_offer =
          webrtc::CreateSessionDescription(SessionDescriptionInterface::kOffer,
                                           sdp, NULL);
      EXPECT_TRUE(DoSetRemoteDescription(remote_offer));
      EXPECT_EQ(PeerConnectionInterface::kHaveRemoteOffer, observer_.state_);
    }
  
    void CreateAndSetRemoteOffer(const std::string& sdp) {
      SessionDescriptionInterface* remote_offer =
          webrtc::CreateSessionDescription(SessionDescriptionInterface::kOffer,
                                           sdp, nullptr);
      EXPECT_TRUE(DoSetRemoteDescription(remote_offer));
      EXPECT_EQ(PeerConnectionInterface::kHaveRemoteOffer, observer_.state_);
    }
  
    void CreateAnswerAsLocalDescription() {
      std::unique_ptr<SessionDescriptionInterface> answer;
      ASSERT_TRUE(DoCreateAnswer(&answer, nullptr));
  
      // TODO(perkj): Currently SetLocalDescription fails if any parameters in an
      // audio codec change, even if the parameter has nothing to do with
      // receiving. Not all parameters are serialized to SDP.
      // Since CreatePrAnswerAsLocalDescription serialize/deserialize
      // the SessionDescription, it is necessary to do that here to in order to
      // get ReceiveOfferCreatePrAnswerAndAnswer and RenegotiateAudioOnly to pass.
      // https://code.google.com/p/webrtc/issues/detail?id=1356
      std::string sdp;
      EXPECT_TRUE(answer->ToString(&sdp));
      SessionDescriptionInterface* new_answer =
          webrtc::CreateSessionDescription(SessionDescriptionInterface::kAnswer,
                                           sdp, NULL);
      EXPECT_TRUE(DoSetLocalDescription(new_answer));
      EXPECT_EQ(PeerConnectionInterface::kStable, observer_.state_);
    }
  
    void CreatePrAnswerAsLocalDescription() {
      std::unique_ptr<SessionDescriptionInterface> answer;
      ASSERT_TRUE(DoCreateAnswer(&answer, nullptr));
  
      std::string sdp;
      EXPECT_TRUE(answer->ToString(&sdp));
      SessionDescriptionInterface* pr_answer =
          webrtc::CreateSessionDescription(SessionDescriptionInterface::kPrAnswer,
                                           sdp, NULL);
      EXPECT_TRUE(DoSetLocalDescription(pr_answer));
      EXPECT_EQ(PeerConnectionInterface::kHaveLocalPrAnswer, observer_.state_);
    }
  
    void CreateOfferReceiveAnswer() {
      CreateOfferAsLocalDescription();
      std::string sdp;
      EXPECT_TRUE(pc_->local_description()->ToString(&sdp));
      CreateAnswerAsRemoteDescription(sdp);
    }
  
    void CreateOfferAsLocalDescription() {
      std::unique_ptr<SessionDescriptionInterface> offer;
      ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
      // TODO(perkj): Currently SetLocalDescription fails if any parameters in an
      // audio codec change, even if the parameter has nothing to do with
      // receiving. Not all parameters are serialized to SDP.
      // Since CreatePrAnswerAsLocalDescription serialize/deserialize
      // the SessionDescription, it is necessary to do that here to in order to
      // get ReceiveOfferCreatePrAnswerAndAnswer and RenegotiateAudioOnly to pass.
      // https://code.google.com/p/webrtc/issues/detail?id=1356
      std::string sdp;
      EXPECT_TRUE(offer->ToString(&sdp));
      SessionDescriptionInterface* new_offer =
              webrtc::CreateSessionDescription(
                  SessionDescriptionInterface::kOffer,
                  sdp, NULL);
  
      EXPECT_TRUE(DoSetLocalDescription(new_offer));
      EXPECT_EQ(PeerConnectionInterface::kHaveLocalOffer, observer_.state_);
      // Wait for the ice_complete message, so that SDP will have candidates.
      EXPECT_TRUE_WAIT(observer_.ice_complete_, kTimeout);
    }
  
    void CreateAnswerAsRemoteDescription(const std::string& sdp) {
      webrtc::JsepSessionDescription* answer = new webrtc::JsepSessionDescription(
          SessionDescriptionInterface::kAnswer);
      EXPECT_TRUE(answer->Initialize(sdp, NULL));
      EXPECT_TRUE(DoSetRemoteDescription(answer));
      EXPECT_EQ(PeerConnectionInterface::kStable, observer_.state_);
    }
  
    void CreatePrAnswerAndAnswerAsRemoteDescription(const std::string& sdp) {
      webrtc::JsepSessionDescription* pr_answer =
          new webrtc::JsepSessionDescription(
              SessionDescriptionInterface::kPrAnswer);
      EXPECT_TRUE(pr_answer->Initialize(sdp, NULL));
      EXPECT_TRUE(DoSetRemoteDescription(pr_answer));
      EXPECT_EQ(PeerConnectionInterface::kHaveRemotePrAnswer, observer_.state_);
      webrtc::JsepSessionDescription* answer =
          new webrtc::JsepSessionDescription(
              SessionDescriptionInterface::kAnswer);
      EXPECT_TRUE(answer->Initialize(sdp, NULL));
      EXPECT_TRUE(DoSetRemoteDescription(answer));
      EXPECT_EQ(PeerConnectionInterface::kStable, observer_.state_);
    }
  
    // Help function used for waiting until a the last signaled remote stream has
    // the same label as |stream_label|. In a few of the tests in this file we
    // answer with the same session description as we offer and thus we can
    // check if OnAddStream have been called with the same stream as we offer to
    // send.
    void WaitAndVerifyOnAddStream(const std::string& stream_label) {
      EXPECT_EQ_WAIT(stream_label, observer_.GetLastAddedStreamLabel(), kTimeout);
    }
  
    // Creates an offer and applies it as a local session description.
    // Creates an answer with the same SDP an the offer but removes all lines
    // that start with a:ssrc"
    void CreateOfferReceiveAnswerWithoutSsrc() {
      CreateOfferAsLocalDescription();
      std::string sdp;
      EXPECT_TRUE(pc_->local_description()->ToString(&sdp));
      SetSsrcToZero(&sdp);
      CreateAnswerAsRemoteDescription(sdp);
    }
  
    // This function creates a MediaStream with label kStreams[0] and
    // |number_of_audio_tracks| and |number_of_video_tracks| tracks and the
    // corresponding SessionDescriptionInterface. The SessionDescriptionInterface
    // is returned and the MediaStream is stored in
    // |reference_collection_|
    std::unique_ptr<SessionDescriptionInterface>
    CreateSessionDescriptionAndReference(size_t number_of_audio_tracks,
                                         size_t number_of_video_tracks) {
      EXPECT_LE(number_of_audio_tracks, 2u);
      EXPECT_LE(number_of_video_tracks, 2u);
  
      reference_collection_ = StreamCollection::Create();
      std::string sdp_ms1 = std::string(kSdpStringInit);
  
      std::string mediastream_label = kStreams[0];
  
      rtc::scoped_refptr<webrtc::MediaStreamInterface> stream(
          webrtc::MediaStream::Create(mediastream_label));
      reference_collection_->AddStream(stream);
  
      if (number_of_audio_tracks > 0) {
        sdp_ms1 += std::string(kSdpStringAudio);
        sdp_ms1 += std::string(kSdpStringMs1Audio0);
        AddAudioTrack(kAudioTracks[0], stream);
      }
      if (number_of_audio_tracks > 1) {
        sdp_ms1 += kSdpStringMs1Audio1;
        AddAudioTrack(kAudioTracks[1], stream);
      }
  
      if (number_of_video_tracks > 0) {
        sdp_ms1 += std::string(kSdpStringVideo);
        sdp_ms1 += std::string(kSdpStringMs1Video0);
        AddVideoTrack(kVideoTracks[0], stream);
      }
      if (number_of_video_tracks > 1) {
        sdp_ms1 += kSdpStringMs1Video1;
        AddVideoTrack(kVideoTracks[1], stream);
      }
  
      return std::unique_ptr<SessionDescriptionInterface>(
          webrtc::CreateSessionDescription(SessionDescriptionInterface::kOffer,
                                           sdp_ms1, nullptr));
    }
  
    void AddAudioTrack(const std::string& track_id,
                       MediaStreamInterface* stream) {
      rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
          webrtc::AudioTrack::Create(track_id, nullptr));
      ASSERT_TRUE(stream->AddTrack(audio_track));
    }
  
    void AddVideoTrack(const std::string& track_id,
                       MediaStreamInterface* stream) {
      rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track(
          webrtc::VideoTrack::Create(track_id,
                                     webrtc::FakeVideoTrackSource::Create()));
      ASSERT_TRUE(stream->AddTrack(video_track));
    }
  
    std::unique_ptr<SessionDescriptionInterface> CreateOfferWithOneAudioStream() {
      CreatePeerConnectionWithoutDtls();
      AddVoiceStream(kStreamLabel1);
      std::unique_ptr<SessionDescriptionInterface> offer;
      EXPECT_TRUE(DoCreateOffer(&offer, nullptr));
      return offer;
    }
  
    std::unique_ptr<SessionDescriptionInterface>
    CreateAnswerWithOneAudioStream() {
      std::unique_ptr<SessionDescriptionInterface> offer =
          CreateOfferWithOneAudioStream();
      EXPECT_TRUE(DoSetRemoteDescription(offer.release()));
      std::unique_ptr<SessionDescriptionInterface> answer;
      EXPECT_TRUE(DoCreateAnswer(&answer, nullptr));
      return answer;
    }
  
    const std::string& GetFirstAudioStreamCname(
        const SessionDescriptionInterface* desc) {
      const cricket::ContentInfo* audio_content =
          cricket::GetFirstAudioContent(desc->description());
      const cricket::AudioContentDescription* audio_desc =
          static_cast<const cricket::AudioContentDescription*>(
              audio_content->description);
      return audio_desc->streams()[0].cname;
    }
  
    cricket::FakePortAllocator* port_allocator_ = nullptr;
    FakeRTCCertificateGenerator* fake_certificate_generator_ = nullptr;
    rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> pc_factory_;
    rtc::scoped_refptr<PeerConnectionFactoryForTest> pc_factory_for_test_;
    rtc::scoped_refptr<PeerConnectionInterface> pc_;
    MockPeerConnectionObserver observer_;
    rtc::scoped_refptr<StreamCollection> reference_collection_;
  };
  
  // Test that no callbacks on the PeerConnectionObserver are called after the
  // PeerConnection is closed.
  TEST_F(PeerConnectionInterfaceTest, CloseAndTestCallbackFunctions) {
    rtc::scoped_refptr<PeerConnectionInterface> pc(
        pc_factory_for_test_->CreatePeerConnection(
            PeerConnectionInterface::RTCConfiguration(), nullptr, nullptr,
            nullptr, &observer_));
    observer_.SetPeerConnectionInterface(pc.get());
    pc->Close();
  
    // No callbacks is expected to be called.
    observer_.callback_triggered_ = false;
    std::vector<cricket::Candidate> candidates;
    pc_factory_for_test_->transport_controller->SignalGatheringState(
        cricket::IceGatheringState{});
    pc_factory_for_test_->transport_controller->SignalCandidatesGathered(
        "", candidates);
    pc_factory_for_test_->transport_controller->SignalConnectionState(
        cricket::IceConnectionState{});
    pc_factory_for_test_->transport_controller->SignalCandidatesRemoved(
        candidates);
    pc_factory_for_test_->transport_controller->SignalReceiving(false);
    EXPECT_FALSE(observer_.callback_triggered_);
  }
  
  // Generate different CNAMEs when PeerConnections are created.
  // The CNAMEs are expected to be generated randomly. It is possible
  // that the test fails, though the possibility is very low.
  TEST_F(PeerConnectionInterfaceTest, CnameGenerationInOffer) {
    std::unique_ptr<SessionDescriptionInterface> offer1 =
        CreateOfferWithOneAudioStream();
    std::unique_ptr<SessionDescriptionInterface> offer2 =
        CreateOfferWithOneAudioStream();
    EXPECT_NE(GetFirstAudioStreamCname(offer1.get()),
              GetFirstAudioStreamCname(offer2.get()));
  }
  
  TEST_F(PeerConnectionInterfaceTest, CnameGenerationInAnswer) {
    std::unique_ptr<SessionDescriptionInterface> answer1 =
        CreateAnswerWithOneAudioStream();
    std::unique_ptr<SessionDescriptionInterface> answer2 =
        CreateAnswerWithOneAudioStream();
    EXPECT_NE(GetFirstAudioStreamCname(answer1.get()),
              GetFirstAudioStreamCname(answer2.get()));
  }
  
  TEST_F(PeerConnectionInterfaceTest,
         CreatePeerConnectionWithDifferentConfigurations) {
    CreatePeerConnectionWithDifferentConfigurations();
  }
  
  TEST_F(PeerConnectionInterfaceTest,
         CreatePeerConnectionWithDifferentIceTransportsTypes) {
    CreatePeerConnectionWithIceTransportsType(PeerConnectionInterface::kNone);
    EXPECT_EQ(cricket::CF_NONE, port_allocator_->candidate_filter());
    CreatePeerConnectionWithIceTransportsType(PeerConnectionInterface::kRelay);
    EXPECT_EQ(cricket::CF_RELAY, port_allocator_->candidate_filter());
    CreatePeerConnectionWithIceTransportsType(PeerConnectionInterface::kNoHost);
    EXPECT_EQ(cricket::CF_ALL & ~cricket::CF_HOST,
              port_allocator_->candidate_filter());
    CreatePeerConnectionWithIceTransportsType(PeerConnectionInterface::kAll);
    EXPECT_EQ(cricket::CF_ALL, port_allocator_->candidate_filter());
  }
  
  // Test that when a PeerConnection is created with a nonzero candidate pool
  // size, the pooled PortAllocatorSession is created with all the attributes
  // in the RTCConfiguration.
  TEST_F(PeerConnectionInterfaceTest, CreatePeerConnectionWithPooledCandidates) {
    PeerConnectionInterface::RTCConfiguration config;
    PeerConnectionInterface::IceServer server;
    server.uri = kStunAddressOnly;
    config.servers.push_back(server);
    config.type = PeerConnectionInterface::kRelay;
    config.disable_ipv6 = true;
    config.tcp_candidate_policy =
        PeerConnectionInterface::kTcpCandidatePolicyDisabled;
    config.candidate_network_policy =
        PeerConnectionInterface::kCandidateNetworkPolicyLowCost;
    config.ice_candidate_pool_size = 1;
    CreatePeerConnection(config, nullptr);
  
    const cricket::FakePortAllocatorSession* session =
        static_cast<const cricket::FakePortAllocatorSession*>(
            port_allocator_->GetPooledSession());
    ASSERT_NE(nullptr, session);
    EXPECT_EQ(1UL, session->stun_servers().size());
    EXPECT_EQ(0U, session->flags() & cricket::PORTALLOCATOR_ENABLE_IPV6);
    EXPECT_LT(0U, session->flags() & cricket::PORTALLOCATOR_DISABLE_TCP);
    EXPECT_LT(0U,
              session->flags() & cricket::PORTALLOCATOR_DISABLE_COSTLY_NETWORKS);
  }
  
  // Test that the PeerConnection initializes the port allocator passed into it,
  // and on the correct thread.
  TEST_F(PeerConnectionInterfaceTest,
         CreatePeerConnectionInitializesPortAllocator) {
    rtc::Thread network_thread;
    network_thread.Start();
    rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> pc_factory(
        webrtc::CreatePeerConnectionFactory(
            &network_thread, rtc::Thread::Current(), rtc::Thread::Current(),
            nullptr, nullptr, nullptr));
    std::unique_ptr<cricket::FakePortAllocator> port_allocator(
        new cricket::FakePortAllocator(&network_thread, nullptr));
    cricket::FakePortAllocator* raw_port_allocator = port_allocator.get();
    PeerConnectionInterface::RTCConfiguration config;
    rtc::scoped_refptr<PeerConnectionInterface> pc(
        pc_factory->CreatePeerConnection(
            config, nullptr, std::move(port_allocator), nullptr, &observer_));
    // FakePortAllocator RTC_CHECKs that it's initialized on the right thread,
    // so all we have to do here is check that it's initialized.
    EXPECT_TRUE(raw_port_allocator->initialized());
  }
  
  // Check that GetConfiguration returns the configuration the PeerConnection was
  // constructed with, before SetConfiguration is called.
  TEST_F(PeerConnectionInterfaceTest, GetConfigurationAfterCreatePeerConnection) {
    PeerConnectionInterface::RTCConfiguration config;
    config.type = PeerConnectionInterface::kRelay;
    CreatePeerConnection(config, nullptr);
  
    PeerConnectionInterface::RTCConfiguration returned_config =
        pc_->GetConfiguration();
    EXPECT_EQ(PeerConnectionInterface::kRelay, returned_config.type);
  }
  
  // Check that GetConfiguration returns the last configuration passed into
  // SetConfiguration.
  TEST_F(PeerConnectionInterfaceTest, GetConfigurationAfterSetConfiguration) {
    CreatePeerConnection();
  
    PeerConnectionInterface::RTCConfiguration config;
    config.type = PeerConnectionInterface::kRelay;
    EXPECT_TRUE(pc_->SetConfiguration(config));
  
    PeerConnectionInterface::RTCConfiguration returned_config =
        pc_->GetConfiguration();
    EXPECT_EQ(PeerConnectionInterface::kRelay, returned_config.type);
  }
  
  TEST_F(PeerConnectionInterfaceTest, AddStreams) {
    CreatePeerConnectionWithoutDtls();
    AddVideoStream(kStreamLabel1);
    AddVoiceStream(kStreamLabel2);
    ASSERT_EQ(2u, pc_->local_streams()->count());
  
    // Test we can add multiple local streams to one peerconnection.
    rtc::scoped_refptr<MediaStreamInterface> stream(
        pc_factory_->CreateLocalMediaStream(kStreamLabel3));
    rtc::scoped_refptr<AudioTrackInterface> audio_track(
        pc_factory_->CreateAudioTrack(kStreamLabel3,
                                      static_cast<AudioSourceInterface*>(NULL)));
    stream->AddTrack(audio_track.get());
    EXPECT_TRUE(pc_->AddStream(stream));
    EXPECT_EQ(3u, pc_->local_streams()->count());
  
    // Remove the third stream.
    pc_->RemoveStream(pc_->local_streams()->at(2));
    EXPECT_EQ(2u, pc_->local_streams()->count());
  
    // Remove the second stream.
    pc_->RemoveStream(pc_->local_streams()->at(1));
    EXPECT_EQ(1u, pc_->local_streams()->count());
  
    // Remove the first stream.
    pc_->RemoveStream(pc_->local_streams()->at(0));
    EXPECT_EQ(0u, pc_->local_streams()->count());
  }
  
  // Test that the created offer includes streams we added.
  TEST_F(PeerConnectionInterfaceTest, AddedStreamsPresentInOffer) {
    CreatePeerConnectionWithoutDtls();
    AddAudioVideoStream(kStreamLabel1, "audio_track", "video_track");
    std::unique_ptr<SessionDescriptionInterface> offer;
    ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
  
    const cricket::ContentInfo* audio_content =
        cricket::GetFirstAudioContent(offer->description());
    const cricket::AudioContentDescription* audio_desc =
        static_cast<const cricket::AudioContentDescription*>(
            audio_content->description);
    EXPECT_TRUE(
        ContainsTrack(audio_desc->streams(), kStreamLabel1, "audio_track"));
  
    const cricket::ContentInfo* video_content =
        cricket::GetFirstVideoContent(offer->description());
    const cricket::VideoContentDescription* video_desc =
        static_cast<const cricket::VideoContentDescription*>(
            video_content->description);
    EXPECT_TRUE(
        ContainsTrack(video_desc->streams(), kStreamLabel1, "video_track"));
  
    // Add another stream and ensure the offer includes both the old and new
    // streams.
    AddAudioVideoStream(kStreamLabel2, "audio_track2", "video_track2");
    ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
  
    audio_content = cricket::GetFirstAudioContent(offer->description());
    audio_desc = static_cast<const cricket::AudioContentDescription*>(
        audio_content->description);
    EXPECT_TRUE(
        ContainsTrack(audio_desc->streams(), kStreamLabel1, "audio_track"));
    EXPECT_TRUE(
        ContainsTrack(audio_desc->streams(), kStreamLabel2, "audio_track2"));
  
    video_content = cricket::GetFirstVideoContent(offer->description());
    video_desc = static_cast<const cricket::VideoContentDescription*>(
        video_content->description);
    EXPECT_TRUE(
        ContainsTrack(video_desc->streams(), kStreamLabel1, "video_track"));
    EXPECT_TRUE(
        ContainsTrack(video_desc->streams(), kStreamLabel2, "video_track2"));
  }
  
  TEST_F(PeerConnectionInterfaceTest, RemoveStream) {
    CreatePeerConnectionWithoutDtls();
    AddVideoStream(kStreamLabel1);
    ASSERT_EQ(1u, pc_->local_streams()->count());
    pc_->RemoveStream(pc_->local_streams()->at(0));
    EXPECT_EQ(0u, pc_->local_streams()->count());
  }
  
  // Test for AddTrack and RemoveTrack methods.
  // Tests that the created offer includes tracks we added,
  // and that the RtpSenders are created correctly.
  // Also tests that RemoveTrack removes the tracks from subsequent offers.
  TEST_F(PeerConnectionInterfaceTest, AddTrackRemoveTrack) {
    CreatePeerConnectionWithoutDtls();
    // Create a dummy stream, so tracks share a stream label.
    rtc::scoped_refptr<MediaStreamInterface> stream(
        pc_factory_->CreateLocalMediaStream(kStreamLabel1));
    std::vector<MediaStreamInterface*> stream_list;
    stream_list.push_back(stream.get());
    rtc::scoped_refptr<AudioTrackInterface> audio_track(
        pc_factory_->CreateAudioTrack("audio_track", nullptr));
    rtc::scoped_refptr<VideoTrackInterface> video_track(
        pc_factory_->CreateVideoTrack(
            "video_track",
            pc_factory_->CreateVideoSource(new cricket::FakeVideoCapturer())));
    auto audio_sender = pc_->AddTrack(audio_track, stream_list);
    auto video_sender = pc_->AddTrack(video_track, stream_list);
    EXPECT_EQ(1UL, audio_sender->stream_ids().size());
    EXPECT_EQ(kStreamLabel1, audio_sender->stream_ids()[0]);
    EXPECT_EQ("audio_track", audio_sender->id());
    EXPECT_EQ(audio_track, audio_sender->track());
    EXPECT_EQ(1UL, video_sender->stream_ids().size());
    EXPECT_EQ(kStreamLabel1, video_sender->stream_ids()[0]);
    EXPECT_EQ("video_track", video_sender->id());
    EXPECT_EQ(video_track, video_sender->track());
  
    // Now create an offer and check for the senders.
    std::unique_ptr<SessionDescriptionInterface> offer;
    ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
  
    const cricket::ContentInfo* audio_content =
        cricket::GetFirstAudioContent(offer->description());
    const cricket::AudioContentDescription* audio_desc =
        static_cast<const cricket::AudioContentDescription*>(
            audio_content->description);
    EXPECT_TRUE(
        ContainsTrack(audio_desc->streams(), kStreamLabel1, "audio_track"));
  
    const cricket::ContentInfo* video_content =
        cricket::GetFirstVideoContent(offer->description());
    const cricket::VideoContentDescription* video_desc =
        static_cast<const cricket::VideoContentDescription*>(
            video_content->description);
    EXPECT_TRUE(
        ContainsTrack(video_desc->streams(), kStreamLabel1, "video_track"));
  
    EXPECT_TRUE(DoSetLocalDescription(offer.release()));
  
    // Now try removing the tracks.
    EXPECT_TRUE(pc_->RemoveTrack(audio_sender));
    EXPECT_TRUE(pc_->RemoveTrack(video_sender));
  
    // Create a new offer and ensure it doesn't contain the removed senders.
    ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
  
    audio_content = cricket::GetFirstAudioContent(offer->description());
    audio_desc = static_cast<const cricket::AudioContentDescription*>(
        audio_content->description);
    EXPECT_FALSE(
        ContainsTrack(audio_desc->streams(), kStreamLabel1, "audio_track"));
  
    video_content = cricket::GetFirstVideoContent(offer->description());
    video_desc = static_cast<const cricket::VideoContentDescription*>(
        video_content->description);
    EXPECT_FALSE(
        ContainsTrack(video_desc->streams(), kStreamLabel1, "video_track"));
  
    EXPECT_TRUE(DoSetLocalDescription(offer.release()));
  
    // Calling RemoveTrack on a sender no longer attached to a PeerConnection
    // should return false.
    EXPECT_FALSE(pc_->RemoveTrack(audio_sender));
    EXPECT_FALSE(pc_->RemoveTrack(video_sender));
  }
  
  // Test creating senders without a stream specified,
  // expecting a random stream ID to be generated.
  TEST_F(PeerConnectionInterfaceTest, AddTrackWithoutStream) {
    CreatePeerConnectionWithoutDtls();
    // Create a dummy stream, so tracks share a stream label.
    rtc::scoped_refptr<AudioTrackInterface> audio_track(
        pc_factory_->CreateAudioTrack("audio_track", nullptr));
    rtc::scoped_refptr<VideoTrackInterface> video_track(
        pc_factory_->CreateVideoTrack(
            "video_track",
            pc_factory_->CreateVideoSource(new cricket::FakeVideoCapturer())));
    auto audio_sender =
        pc_->AddTrack(audio_track, std::vector<MediaStreamInterface*>());
    auto video_sender =
        pc_->AddTrack(video_track, std::vector<MediaStreamInterface*>());
    EXPECT_EQ("audio_track", audio_sender->id());
    EXPECT_EQ(audio_track, audio_sender->track());
    EXPECT_EQ("video_track", video_sender->id());
    EXPECT_EQ(video_track, video_sender->track());
    // If the ID is truly a random GUID, it should be infinitely unlikely they
    // will be the same.
    EXPECT_NE(video_sender->stream_ids(), audio_sender->stream_ids());
  }
  
  TEST_F(PeerConnectionInterfaceTest, CreateOfferReceiveAnswer) {
    InitiateCall();
    WaitAndVerifyOnAddStream(kStreamLabel1);
    VerifyRemoteRtpHeaderExtensions();
  }
  
  TEST_F(PeerConnectionInterfaceTest, CreateOfferReceivePrAnswerAndAnswer) {
    CreatePeerConnectionWithoutDtls();
    AddVideoStream(kStreamLabel1);
    CreateOfferAsLocalDescription();
    std::string offer;
    EXPECT_TRUE(pc_->local_description()->ToString(&offer));
    CreatePrAnswerAndAnswerAsRemoteDescription(offer);
    WaitAndVerifyOnAddStream(kStreamLabel1);
  }
  
  TEST_F(PeerConnectionInterfaceTest, ReceiveOfferCreateAnswer) {
    CreatePeerConnectionWithoutDtls();
    AddVideoStream(kStreamLabel1);
  
    CreateOfferAsRemoteDescription();
    CreateAnswerAsLocalDescription();
  
    WaitAndVerifyOnAddStream(kStreamLabel1);
  }
  
  TEST_F(PeerConnectionInterfaceTest, ReceiveOfferCreatePrAnswerAndAnswer) {
    CreatePeerConnectionWithoutDtls();
    AddVideoStream(kStreamLabel1);
  
    CreateOfferAsRemoteDescription();
    CreatePrAnswerAsLocalDescription();
    CreateAnswerAsLocalDescription();
  
    WaitAndVerifyOnAddStream(kStreamLabel1);
  }
  
  TEST_F(PeerConnectionInterfaceTest, Renegotiate) {
    InitiateCall();
    ASSERT_EQ(1u, pc_->remote_streams()->count());
    pc_->RemoveStream(pc_->local_streams()->at(0));
    CreateOfferReceiveAnswer();
    EXPECT_EQ(0u, pc_->remote_streams()->count());
    AddVideoStream(kStreamLabel1);
    CreateOfferReceiveAnswer();
  }
  
  // Tests that after negotiating an audio only call, the respondent can perform a
  // renegotiation that removes the audio stream.
  TEST_F(PeerConnectionInterfaceTest, RenegotiateAudioOnly) {
    CreatePeerConnectionWithoutDtls();
    AddVoiceStream(kStreamLabel1);
    CreateOfferAsRemoteDescription();
    CreateAnswerAsLocalDescription();
  
    ASSERT_EQ(1u, pc_->remote_streams()->count());
    pc_->RemoveStream(pc_->local_streams()->at(0));
    CreateOfferReceiveAnswer();
    EXPECT_EQ(0u, pc_->remote_streams()->count());
  }
  
  // Test that candidates are generated and that we can parse our own candidates.
  TEST_F(PeerConnectionInterfaceTest, IceCandidates) {
    CreatePeerConnectionWithoutDtls();
  
    EXPECT_FALSE(pc_->AddIceCandidate(observer_.last_candidate_.get()));
    // SetRemoteDescription takes ownership of offer.
    std::unique_ptr<SessionDescriptionInterface> offer;
    AddVideoStream(kStreamLabel1);
    EXPECT_TRUE(DoCreateOffer(&offer, nullptr));
    EXPECT_TRUE(DoSetRemoteDescription(offer.release()));
  
    // SetLocalDescription takes ownership of answer.
    std::unique_ptr<SessionDescriptionInterface> answer;
    EXPECT_TRUE(DoCreateAnswer(&answer, nullptr));
    EXPECT_TRUE(DoSetLocalDescription(answer.release()));
  
    EXPECT_TRUE_WAIT(observer_.last_candidate_.get() != NULL, kTimeout);
    EXPECT_TRUE_WAIT(observer_.ice_complete_, kTimeout);
  
    EXPECT_TRUE(pc_->AddIceCandidate(observer_.last_candidate_.get()));
  }
  
  // Test that CreateOffer and CreateAnswer will fail if the track labels are
  // not unique.
  TEST_F(PeerConnectionInterfaceTest, CreateOfferAnswerWithInvalidStream) {
    CreatePeerConnectionWithoutDtls();
    // Create a regular offer for the CreateAnswer test later.
    std::unique_ptr<SessionDescriptionInterface> offer;
    EXPECT_TRUE(DoCreateOffer(&offer, nullptr));
    EXPECT_TRUE(offer);
    offer.reset();
  
    // Create a local stream with audio&video tracks having same label.
    AddAudioVideoStream(kStreamLabel1, "track_label", "track_label");
  
    // Test CreateOffer
    EXPECT_FALSE(DoCreateOffer(&offer, nullptr));
  
    // Test CreateAnswer
    std::unique_ptr<SessionDescriptionInterface> answer;
    EXPECT_FALSE(DoCreateAnswer(&answer, nullptr));
  }
  
  // Test that we will get different SSRCs for each tracks in the offer and answer
  // we created.
  TEST_F(PeerConnectionInterfaceTest, SsrcInOfferAnswer) {
    CreatePeerConnectionWithoutDtls();
    // Create a local stream with audio&video tracks having different labels.
    AddAudioVideoStream(kStreamLabel1, "audio_label", "video_label");
  
    // Test CreateOffer
    std::unique_ptr<SessionDescriptionInterface> offer;
    ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
    int audio_ssrc = 0;
    int video_ssrc = 0;
    EXPECT_TRUE(GetFirstSsrc(GetFirstAudioContent(offer->description()),
                             &audio_ssrc));
    EXPECT_TRUE(GetFirstSsrc(GetFirstVideoContent(offer->description()),
                             &video_ssrc));
    EXPECT_NE(audio_ssrc, video_ssrc);
  
    // Test CreateAnswer
    EXPECT_TRUE(DoSetRemoteDescription(offer.release()));
    std::unique_ptr<SessionDescriptionInterface> answer;
    ASSERT_TRUE(DoCreateAnswer(&answer, nullptr));
    audio_ssrc = 0;
    video_ssrc = 0;
    EXPECT_TRUE(GetFirstSsrc(GetFirstAudioContent(answer->description()),
                             &audio_ssrc));
    EXPECT_TRUE(GetFirstSsrc(GetFirstVideoContent(answer->description()),
                             &video_ssrc));
    EXPECT_NE(audio_ssrc, video_ssrc);
  }
  
  // Test that it's possible to call AddTrack on a MediaStream after adding
  // the stream to a PeerConnection.
  // TODO(deadbeef): Remove this test once this behavior is no longer supported.
  TEST_F(PeerConnectionInterfaceTest, AddTrackAfterAddStream) {
    CreatePeerConnectionWithoutDtls();
    // Create audio stream and add to PeerConnection.
    AddVoiceStream(kStreamLabel1);
    MediaStreamInterface* stream = pc_->local_streams()->at(0);
  
    // Add video track to the audio-only stream.
    rtc::scoped_refptr<VideoTrackInterface> video_track(
        pc_factory_->CreateVideoTrack(
            "video_label",
            pc_factory_->CreateVideoSource(new cricket::FakeVideoCapturer())));
    stream->AddTrack(video_track.get());
  
    std::unique_ptr<SessionDescriptionInterface> offer;
    ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
  
    const cricket::MediaContentDescription* video_desc =
        cricket::GetFirstVideoContentDescription(offer->description());
    EXPECT_TRUE(video_desc != nullptr);
  }
  
  // Test that it's possible to call RemoveTrack on a MediaStream after adding
  // the stream to a PeerConnection.
  // TODO(deadbeef): Remove this test once this behavior is no longer supported.
  TEST_F(PeerConnectionInterfaceTest, RemoveTrackAfterAddStream) {
    CreatePeerConnectionWithoutDtls();
    // Create audio/video stream and add to PeerConnection.
    AddAudioVideoStream(kStreamLabel1, "audio_label", "video_label");
    MediaStreamInterface* stream = pc_->local_streams()->at(0);
  
    // Remove the video track.
    stream->RemoveTrack(stream->GetVideoTracks()[0]);
  
    std::unique_ptr<SessionDescriptionInterface> offer;
    ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
  
    const cricket::MediaContentDescription* video_desc =
        cricket::GetFirstVideoContentDescription(offer->description());
    EXPECT_TRUE(video_desc == nullptr);
  }
  
  // Test creating a sender with a stream ID, and ensure the ID is populated
  // in the offer.
  TEST_F(PeerConnectionInterfaceTest, CreateSenderWithStream) {
    CreatePeerConnectionWithoutDtls();
    pc_->CreateSender("video", kStreamLabel1);
  
    std::unique_ptr<SessionDescriptionInterface> offer;
    ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
  
    const cricket::MediaContentDescription* video_desc =
        cricket::GetFirstVideoContentDescription(offer->description());
    ASSERT_TRUE(video_desc != nullptr);
    ASSERT_EQ(1u, video_desc->streams().size());
    EXPECT_EQ(kStreamLabel1, video_desc->streams()[0].sync_label);
  }
  
  // Test that we can specify a certain track that we want statistics about.
  TEST_F(PeerConnectionInterfaceTest, GetStatsForSpecificTrack) {
    InitiateCall();
    ASSERT_LT(0u, pc_->remote_streams()->count());
    ASSERT_LT(0u, pc_->remote_streams()->at(0)->GetAudioTracks().size());
    rtc::scoped_refptr<MediaStreamTrackInterface> remote_audio =
        pc_->remote_streams()->at(0)->GetAudioTracks()[0];
    EXPECT_TRUE(DoGetStats(remote_audio));
  
    // Remove the stream. Since we are sending to our selves the local
    // and the remote stream is the same.
    pc_->RemoveStream(pc_->local_streams()->at(0));
    // Do a re-negotiation.
    CreateOfferReceiveAnswer();
  
    ASSERT_EQ(0u, pc_->remote_streams()->count());
  
    // Test that we still can get statistics for the old track. Even if it is not
    // sent any longer.
    EXPECT_TRUE(DoGetStats(remote_audio));
  }
  
  // Test that we can get stats on a video track.
  TEST_F(PeerConnectionInterfaceTest, GetStatsForVideoTrack) {
    InitiateCall();
    ASSERT_LT(0u, pc_->remote_streams()->count());
    ASSERT_LT(0u, pc_->remote_streams()->at(0)->GetVideoTracks().size());
    rtc::scoped_refptr<MediaStreamTrackInterface> remote_video =
        pc_->remote_streams()->at(0)->GetVideoTracks()[0];
    EXPECT_TRUE(DoGetStats(remote_video));
  }
  
  // Test that we don't get statistics for an invalid track.
  TEST_F(PeerConnectionInterfaceTest, GetStatsForInvalidTrack) {
    InitiateCall();
    rtc::scoped_refptr<AudioTrackInterface> unknown_audio_track(
        pc_factory_->CreateAudioTrack("unknown track", NULL));
    EXPECT_FALSE(DoGetStats(unknown_audio_track));
  }
  
  // This test setup two RTP data channels in loop back.
  TEST_F(PeerConnectionInterfaceTest, TestDataChannel) {
    FakeConstraints constraints;
    constraints.SetAllowRtpDataChannels();
    CreatePeerConnection(&constraints);
    rtc::scoped_refptr<DataChannelInterface> data1 =
        pc_->CreateDataChannel("test1", NULL);
    rtc::scoped_refptr<DataChannelInterface> data2 =
        pc_->CreateDataChannel("test2", NULL);
    ASSERT_TRUE(data1 != NULL);
    std::unique_ptr<MockDataChannelObserver> observer1(
        new MockDataChannelObserver(data1));
    std::unique_ptr<MockDataChannelObserver> observer2(
        new MockDataChannelObserver(data2));
  
    EXPECT_EQ(DataChannelInterface::kConnecting, data1->state());
    EXPECT_EQ(DataChannelInterface::kConnecting, data2->state());
    std::string data_to_send1 = "testing testing";
    std::string data_to_send2 = "testing something else";
    EXPECT_FALSE(data1->Send(DataBuffer(data_to_send1)));
  
    CreateOfferReceiveAnswer();
    EXPECT_TRUE_WAIT(observer1->IsOpen(), kTimeout);
    EXPECT_TRUE_WAIT(observer2->IsOpen(), kTimeout);
  
    EXPECT_EQ(DataChannelInterface::kOpen, data1->state());
    EXPECT_EQ(DataChannelInterface::kOpen, data2->state());
    EXPECT_TRUE(data1->Send(DataBuffer(data_to_send1)));
    EXPECT_TRUE(data2->Send(DataBuffer(data_to_send2)));
  
    EXPECT_EQ_WAIT(data_to_send1, observer1->last_message(), kTimeout);
    EXPECT_EQ_WAIT(data_to_send2, observer2->last_message(), kTimeout);
  
    data1->Close();
    EXPECT_EQ(DataChannelInterface::kClosing, data1->state());
    CreateOfferReceiveAnswer();
    EXPECT_FALSE(observer1->IsOpen());
    EXPECT_EQ(DataChannelInterface::kClosed, data1->state());
    EXPECT_TRUE(observer2->IsOpen());
  
    data_to_send2 = "testing something else again";
    EXPECT_TRUE(data2->Send(DataBuffer(data_to_send2)));
  
    EXPECT_EQ_WAIT(data_to_send2, observer2->last_message(), kTimeout);
  }
  
  // This test verifies that sendnig binary data over RTP data channels should
  // fail.
  TEST_F(PeerConnectionInterfaceTest, TestSendBinaryOnRtpDataChannel) {
    FakeConstraints constraints;
    constraints.SetAllowRtpDataChannels();
    CreatePeerConnection(&constraints);
    rtc::scoped_refptr<DataChannelInterface> data1 =
        pc_->CreateDataChannel("test1", NULL);
    rtc::scoped_refptr<DataChannelInterface> data2 =
        pc_->CreateDataChannel("test2", NULL);
    ASSERT_TRUE(data1 != NULL);
    std::unique_ptr<MockDataChannelObserver> observer1(
        new MockDataChannelObserver(data1));
    std::unique_ptr<MockDataChannelObserver> observer2(
        new MockDataChannelObserver(data2));
  
    EXPECT_EQ(DataChannelInterface::kConnecting, data1->state());
    EXPECT_EQ(DataChannelInterface::kConnecting, data2->state());
  
    CreateOfferReceiveAnswer();
    EXPECT_TRUE_WAIT(observer1->IsOpen(), kTimeout);
    EXPECT_TRUE_WAIT(observer2->IsOpen(), kTimeout);
  
    EXPECT_EQ(DataChannelInterface::kOpen, data1->state());
    EXPECT_EQ(DataChannelInterface::kOpen, data2->state());
  
    rtc::CopyOnWriteBuffer buffer("test", 4);
    EXPECT_FALSE(data1->Send(DataBuffer(buffer, true)));
  }
  
  // This test setup a RTP data channels in loop back and test that a channel is
  // opened even if the remote end answer with a zero SSRC.
  TEST_F(PeerConnectionInterfaceTest, TestSendOnlyDataChannel) {
    FakeConstraints constraints;
    constraints.SetAllowRtpDataChannels();
    CreatePeerConnection(&constraints);
    rtc::scoped_refptr<DataChannelInterface> data1 =
        pc_->CreateDataChannel("test1", NULL);
    std::unique_ptr<MockDataChannelObserver> observer1(
        new MockDataChannelObserver(data1));
  
    CreateOfferReceiveAnswerWithoutSsrc();
  
    EXPECT_TRUE_WAIT(observer1->IsOpen(), kTimeout);
  
    data1->Close();
    EXPECT_EQ(DataChannelInterface::kClosing, data1->state());
    CreateOfferReceiveAnswerWithoutSsrc();
    EXPECT_EQ(DataChannelInterface::kClosed, data1->state());
    EXPECT_FALSE(observer1->IsOpen());
  }
  
  // This test that if a data channel is added in an answer a receive only channel
  // channel is created.
  TEST_F(PeerConnectionInterfaceTest, TestReceiveOnlyDataChannel) {
    FakeConstraints constraints;
    constraints.SetAllowRtpDataChannels();
    CreatePeerConnection(&constraints);
  
    std::string offer_label = "offer_channel";
    rtc::scoped_refptr<DataChannelInterface> offer_channel =
        pc_->CreateDataChannel(offer_label, NULL);
  
    CreateOfferAsLocalDescription();
  
    // Replace the data channel label in the offer and apply it as an answer.
    std::string receive_label = "answer_channel";
    std::string sdp;
    EXPECT_TRUE(pc_->local_description()->ToString(&sdp));
    rtc::replace_substrs(offer_label.c_str(), offer_label.length(),
                               receive_label.c_str(), receive_label.length(),
                               &sdp);
    CreateAnswerAsRemoteDescription(sdp);
  
    // Verify that a new incoming data channel has been created and that
    // it is open but can't we written to.
    ASSERT_TRUE(observer_.last_datachannel_ != NULL);
    DataChannelInterface* received_channel = observer_.last_datachannel_;
    EXPECT_EQ(DataChannelInterface::kConnecting, received_channel->state());
    EXPECT_EQ(receive_label, received_channel->label());
    EXPECT_FALSE(received_channel->Send(DataBuffer("something")));
  
    // Verify that the channel we initially offered has been rejected.
    EXPECT_EQ(DataChannelInterface::kClosed, offer_channel->state());
  
    // Do another offer / answer exchange and verify that the data channel is
    // opened.
    CreateOfferReceiveAnswer();
    EXPECT_EQ_WAIT(DataChannelInterface::kOpen, received_channel->state(),
                   kTimeout);
  }
  
  // This test that no data channel is returned if a reliable channel is
  // requested.
  // TODO(perkj): Remove this test once reliable channels are implemented.
  TEST_F(PeerConnectionInterfaceTest, CreateReliableRtpDataChannelShouldFail) {
    FakeConstraints constraints;
    constraints.SetAllowRtpDataChannels();
    CreatePeerConnection(&constraints);
  
    std::string label = "test";
    webrtc::DataChannelInit config;
    config.reliable = true;
    rtc::scoped_refptr<DataChannelInterface> channel =
        pc_->CreateDataChannel(label, &config);
    EXPECT_TRUE(channel == NULL);
  }
  
  // Verifies that duplicated label is not allowed for RTP data channel.
  TEST_F(PeerConnectionInterfaceTest, RtpDuplicatedLabelNotAllowed) {
    FakeConstraints constraints;
    constraints.SetAllowRtpDataChannels();
    CreatePeerConnection(&constraints);
  
    std::string label = "test";
    rtc::scoped_refptr<DataChannelInterface> channel =
        pc_->CreateDataChannel(label, nullptr);
    EXPECT_NE(channel, nullptr);
  
    rtc::scoped_refptr<DataChannelInterface> dup_channel =
        pc_->CreateDataChannel(label, nullptr);
    EXPECT_EQ(dup_channel, nullptr);
  }
  
  // This tests that a SCTP data channel is returned using different
  // DataChannelInit configurations.
  TEST_F(PeerConnectionInterfaceTest, CreateSctpDataChannel) {
    FakeConstraints constraints;
    constraints.SetAllowDtlsSctpDataChannels();
    CreatePeerConnection(&constraints);
  
    webrtc::DataChannelInit config;
  
    rtc::scoped_refptr<DataChannelInterface> channel =
        pc_->CreateDataChannel("1", &config);
    EXPECT_TRUE(channel != NULL);
    EXPECT_TRUE(channel->reliable());
    EXPECT_TRUE(observer_.renegotiation_needed_);
    observer_.renegotiation_needed_ = false;
  
    config.ordered = false;
    channel = pc_->CreateDataChannel("2", &config);
    EXPECT_TRUE(channel != NULL);
    EXPECT_TRUE(channel->reliable());
    EXPECT_FALSE(observer_.renegotiation_needed_);
  
    config.ordered = true;
    config.maxRetransmits = 0;
    channel = pc_->CreateDataChannel("3", &config);
    EXPECT_TRUE(channel != NULL);
    EXPECT_FALSE(channel->reliable());
    EXPECT_FALSE(observer_.renegotiation_needed_);
  
    config.maxRetransmits = -1;
    config.maxRetransmitTime = 0;
    channel = pc_->CreateDataChannel("4", &config);
    EXPECT_TRUE(channel != NULL);
    EXPECT_FALSE(channel->reliable());
    EXPECT_FALSE(observer_.renegotiation_needed_);
  }
  
  // This tests that no data channel is returned if both maxRetransmits and
  // maxRetransmitTime are set for SCTP data channels.
  TEST_F(PeerConnectionInterfaceTest,
         CreateSctpDataChannelShouldFailForInvalidConfig) {
    FakeConstraints constraints;
    constraints.SetAllowDtlsSctpDataChannels();
    CreatePeerConnection(&constraints);
  
    std::string label = "test";
    webrtc::DataChannelInit config;
    config.maxRetransmits = 0;
    config.maxRetransmitTime = 0;
  
    rtc::scoped_refptr<DataChannelInterface> channel =
        pc_->CreateDataChannel(label, &config);
    EXPECT_TRUE(channel == NULL);
  }
  
  // The test verifies that creating a SCTP data channel with an id already in use
  // or out of range should fail.
  TEST_F(PeerConnectionInterfaceTest,
         CreateSctpDataChannelWithInvalidIdShouldFail) {
    FakeConstraints constraints;
    constraints.SetAllowDtlsSctpDataChannels();
    CreatePeerConnection(&constraints);
  
    webrtc::DataChannelInit config;
    rtc::scoped_refptr<DataChannelInterface> channel;
  
    config.id = 1;
    channel = pc_->CreateDataChannel("1", &config);
    EXPECT_TRUE(channel != NULL);
    EXPECT_EQ(1, channel->id());
  
    channel = pc_->CreateDataChannel("x", &config);
    EXPECT_TRUE(channel == NULL);
  
    config.id = cricket::kMaxSctpSid;
    channel = pc_->CreateDataChannel("max", &config);
    EXPECT_TRUE(channel != NULL);
    EXPECT_EQ(config.id, channel->id());
  
    config.id = cricket::kMaxSctpSid + 1;
    channel = pc_->CreateDataChannel("x", &config);
    EXPECT_TRUE(channel == NULL);
  }
  
  // Verifies that duplicated label is allowed for SCTP data channel.
  TEST_F(PeerConnectionInterfaceTest, SctpDuplicatedLabelAllowed) {
    FakeConstraints constraints;
    constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
                             true);
    CreatePeerConnection(&constraints);
  
    std::string label = "test";
    rtc::scoped_refptr<DataChannelInterface> channel =
        pc_->CreateDataChannel(label, nullptr);
    EXPECT_NE(channel, nullptr);
  
    rtc::scoped_refptr<DataChannelInterface> dup_channel =
        pc_->CreateDataChannel(label, nullptr);
    EXPECT_NE(dup_channel, nullptr);
  }
  
  // This test verifies that OnRenegotiationNeeded is fired for every new RTP
  // DataChannel.
  TEST_F(PeerConnectionInterfaceTest, RenegotiationNeededForNewRtpDataChannel) {
    FakeConstraints constraints;
    constraints.SetAllowRtpDataChannels();
    CreatePeerConnection(&constraints);
  
    rtc::scoped_refptr<DataChannelInterface> dc1 =
        pc_->CreateDataChannel("test1", NULL);
    EXPECT_TRUE(observer_.renegotiation_needed_);
    observer_.renegotiation_needed_ = false;
  
    rtc::scoped_refptr<DataChannelInterface> dc2 =
        pc_->CreateDataChannel("test2", NULL);
    EXPECT_TRUE(observer_.renegotiation_needed_);
  }
  
  // This test that a data channel closes when a PeerConnection is deleted/closed.
  TEST_F(PeerConnectionInterfaceTest, DataChannelCloseWhenPeerConnectionClose) {
    FakeConstraints constraints;
    constraints.SetAllowRtpDataChannels();
    CreatePeerConnection(&constraints);
  
    rtc::scoped_refptr<DataChannelInterface> data1 =
        pc_->CreateDataChannel("test1", NULL);
    rtc::scoped_refptr<DataChannelInterface> data2 =
        pc_->CreateDataChannel("test2", NULL);
    ASSERT_TRUE(data1 != NULL);
    std::unique_ptr<MockDataChannelObserver> observer1(
        new MockDataChannelObserver(data1));
    std::unique_ptr<MockDataChannelObserver> observer2(
        new MockDataChannelObserver(data2));
  
    CreateOfferReceiveAnswer();
    EXPECT_TRUE_WAIT(observer1->IsOpen(), kTimeout);
    EXPECT_TRUE_WAIT(observer2->IsOpen(), kTimeout);
  
    ReleasePeerConnection();
    EXPECT_EQ(DataChannelInterface::kClosed, data1->state());
    EXPECT_EQ(DataChannelInterface::kClosed, data2->state());
  }
  
  // This test that data channels can be rejected in an answer.
  TEST_F(PeerConnectionInterfaceTest, TestRejectDataChannelInAnswer) {
    FakeConstraints constraints;
    constraints.SetAllowRtpDataChannels();
    CreatePeerConnection(&constraints);
  
    rtc::scoped_refptr<DataChannelInterface> offer_channel(
        pc_->CreateDataChannel("offer_channel", NULL));
  
    CreateOfferAsLocalDescription();
  
    // Create an answer where the m-line for data channels are rejected.
    std::string sdp;
    EXPECT_TRUE(pc_->local_description()->ToString(&sdp));
    webrtc::JsepSessionDescription* answer = new webrtc::JsepSessionDescription(
        SessionDescriptionInterface::kAnswer);
    EXPECT_TRUE(answer->Initialize(sdp, NULL));
    cricket::ContentInfo* data_info =
        answer->description()->GetContentByName("data");
    data_info->rejected = true;
  
    DoSetRemoteDescription(answer);
    EXPECT_EQ(DataChannelInterface::kClosed, offer_channel->state());
  }
  
  // Test that we can create a session description from an SDP string from
  // FireFox, use it as a remote session description, generate an answer and use
  // the answer as a local description.
  TEST_F(PeerConnectionInterfaceTest, ReceiveFireFoxOffer) {
    MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
    FakeConstraints constraints;
    constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
                             true);
    CreatePeerConnection(&constraints);
    AddAudioVideoStream(kStreamLabel1, "audio_label", "video_label");
    SessionDescriptionInterface* desc =
        webrtc::CreateSessionDescription(SessionDescriptionInterface::kOffer,
                                         webrtc::kFireFoxSdpOffer, nullptr);
    EXPECT_TRUE(DoSetSessionDescription(desc, false));
    CreateAnswerAsLocalDescription();
    ASSERT_TRUE(pc_->local_description() != NULL);
    ASSERT_TRUE(pc_->remote_description() != NULL);
  
    const cricket::ContentInfo* content =
        cricket::GetFirstAudioContent(pc_->local_description()->description());
    ASSERT_TRUE(content != NULL);
    EXPECT_FALSE(content->rejected);
  
    content =
        cricket::GetFirstVideoContent(pc_->local_description()->description());
    ASSERT_TRUE(content != NULL);
    EXPECT_FALSE(content->rejected);
  #ifdef HAVE_SCTP
    content =
        cricket::GetFirstDataContent(pc_->local_description()->description());
    ASSERT_TRUE(content != NULL);
    EXPECT_TRUE(content->rejected);
  #endif
  }
  
  // Test that an offer can be received which offers DTLS with SDES fallback.
  // Regression test for issue:
  // https://bugs.chromium.org/p/webrtc/issues/detail?id=6972
  TEST_F(PeerConnectionInterfaceTest, ReceiveDtlsSdesFallbackOffer) {
    FakeConstraints constraints;
    constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
                             true);
    CreatePeerConnection(&constraints);
    // Wait for fake certificate to be generated. Previously, this is what caused
    // the "a=crypto" lines to be rejected.
    AddAudioVideoStream(kStreamLabel1, "audio_label", "video_label");
    ASSERT_NE(nullptr, fake_certificate_generator_);
    EXPECT_EQ_WAIT(1, fake_certificate_generator_->generated_certificates(),
                   kTimeout);
    SessionDescriptionInterface* desc = webrtc::CreateSessionDescription(
        SessionDescriptionInterface::kOffer, kDtlsSdesFallbackSdp, nullptr);
    EXPECT_TRUE(DoSetSessionDescription(desc, false));
    CreateAnswerAsLocalDescription();
  }
  
  // Test that we can create an audio only offer and receive an answer with a
  // limited set of audio codecs and receive an updated offer with more audio
  // codecs, where the added codecs are not supported.
  TEST_F(PeerConnectionInterfaceTest, ReceiveUpdatedAudioOfferWithBadCodecs) {
    CreatePeerConnectionWithoutDtls();
    AddVoiceStream("audio_label");
    CreateOfferAsLocalDescription();
  
    SessionDescriptionInterface* answer =
        webrtc::CreateSessionDescription(SessionDescriptionInterface::kAnswer,
                                         webrtc::kAudioSdp, nullptr);
    EXPECT_TRUE(DoSetSessionDescription(answer, false));
  
    SessionDescriptionInterface* updated_offer =
        webrtc::CreateSessionDescription(SessionDescriptionInterface::kOffer,
                                         webrtc::kAudioSdpWithUnsupportedCodecs,
                                         nullptr);
    EXPECT_TRUE(DoSetSessionDescription(updated_offer, false));
    CreateAnswerAsLocalDescription();
  }
  
  // Test that if we're receiving (but not sending) a track, subsequent offers
  // will have m-lines with a=recvonly.
  TEST_F(PeerConnectionInterfaceTest, CreateSubsequentRecvOnlyOffer) {
    FakeConstraints constraints;
    constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
                             true);
    CreatePeerConnection(&constraints);
    CreateAndSetRemoteOffer(kSdpStringWithStream1);
    CreateAnswerAsLocalDescription();
  
    // At this point we should be receiving stream 1, but not sending anything.
    // A new offer should be recvonly.
    std::unique_ptr<SessionDescriptionInterface> offer;
    DoCreateOffer(&offer, nullptr);
  
    const cricket::ContentInfo* video_content =
        cricket::GetFirstVideoContent(offer->description());
    const cricket::VideoContentDescription* video_desc =
        static_cast<const cricket::VideoContentDescription*>(
            video_content->description);
    ASSERT_EQ(cricket::MD_RECVONLY, video_desc->direction());
  
    const cricket::ContentInfo* audio_content =
        cricket::GetFirstAudioContent(offer->description());
    const cricket::AudioContentDescription* audio_desc =
        static_cast<const cricket::AudioContentDescription*>(
            audio_content->description);
    ASSERT_EQ(cricket::MD_RECVONLY, audio_desc->direction());
  }
  
  // Test that if we're receiving (but not sending) a track, and the
  // offerToReceiveVideo/offerToReceiveAudio constraints are explicitly set to
  // false, the generated m-lines will be a=inactive.
  TEST_F(PeerConnectionInterfaceTest, CreateSubsequentInactiveOffer) {
    FakeConstraints constraints;
    constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
                             true);
    CreatePeerConnection(&constraints);
    CreateAndSetRemoteOffer(kSdpStringWithStream1);
    CreateAnswerAsLocalDescription();
  
    // At this point we should be receiving stream 1, but not sending anything.
    // A new offer would be recvonly, but we'll set the "no receive" constraints
    // to make it inactive.
    std::unique_ptr<SessionDescriptionInterface> offer;
    FakeConstraints offer_constraints;
    offer_constraints.AddMandatory(
        webrtc::MediaConstraintsInterface::kOfferToReceiveVideo, false);
    offer_constraints.AddMandatory(
        webrtc::MediaConstraintsInterface::kOfferToReceiveAudio, false);
    DoCreateOffer(&offer, &offer_constraints);
  
    const cricket::ContentInfo* video_content =
        cricket::GetFirstVideoContent(offer->description());
    const cricket::VideoContentDescription* video_desc =
        static_cast<const cricket::VideoContentDescription*>(
            video_content->description);
    ASSERT_EQ(cricket::MD_INACTIVE, video_desc->direction());
  
    const cricket::ContentInfo* audio_content =
        cricket::GetFirstAudioContent(offer->description());
    const cricket::AudioContentDescription* audio_desc =
        static_cast<const cricket::AudioContentDescription*>(
            audio_content->description);
    ASSERT_EQ(cricket::MD_INACTIVE, audio_desc->direction());
  }
  
  // Test that we can use SetConfiguration to change the ICE servers of the
  // PortAllocator.
  TEST_F(PeerConnectionInterfaceTest, SetConfigurationChangesIceServers) {
    CreatePeerConnection();
  
    PeerConnectionInterface::RTCConfiguration config;
    PeerConnectionInterface::IceServer server;
    server.uri = "stun:test_hostname";
    config.servers.push_back(server);
    EXPECT_TRUE(pc_->SetConfiguration(config));
  
    EXPECT_EQ(1u, port_allocator_->stun_servers().size());
    EXPECT_EQ("test_hostname",
              port_allocator_->stun_servers().begin()->hostname());
  }
  
  TEST_F(PeerConnectionInterfaceTest, SetConfigurationChangesCandidateFilter) {
    CreatePeerConnection();
    PeerConnectionInterface::RTCConfiguration config;
    config.type = PeerConnectionInterface::kRelay;
    EXPECT_TRUE(pc_->SetConfiguration(config));
    EXPECT_EQ(cricket::CF_RELAY, port_allocator_->candidate_filter());
  }
  
  TEST_F(PeerConnectionInterfaceTest, SetConfigurationChangesPruneTurnPortsFlag) {
    PeerConnectionInterface::RTCConfiguration config;
    config.prune_turn_ports = false;
    CreatePeerConnection(config, nullptr);
    EXPECT_FALSE(port_allocator_->prune_turn_ports());
  
    config.prune_turn_ports = true;
    EXPECT_TRUE(pc_->SetConfiguration(config));
    EXPECT_TRUE(port_allocator_->prune_turn_ports());
  }
  
  // Test that when SetConfiguration changes both the pool size and other
  // attributes, the pooled session is created with the updated attributes.
  TEST_F(PeerConnectionInterfaceTest,
         SetConfigurationCreatesPooledSessionCorrectly) {
    CreatePeerConnection();
    PeerConnectionInterface::RTCConfiguration config;
    config.ice_candidate_pool_size = 1;
    PeerConnectionInterface::IceServer server;
    server.uri = kStunAddressOnly;
    config.servers.push_back(server);
    config.type = PeerConnectionInterface::kRelay;
    EXPECT_TRUE(pc_->SetConfiguration(config));
  
    const cricket::FakePortAllocatorSession* session =
        static_cast<const cricket::FakePortAllocatorSession*>(
            port_allocator_->GetPooledSession());
    ASSERT_NE(nullptr, session);
    EXPECT_EQ(1UL, session->stun_servers().size());
  }
  
  // Test that after SetLocalDescription, changing the pool size is not allowed,
  // and an invalid modification error is returned.
  TEST_F(PeerConnectionInterfaceTest,
         CantChangePoolSizeAfterSetLocalDescription) {
    CreatePeerConnection();
    // Start by setting a size of 1.
    PeerConnectionInterface::RTCConfiguration config;
    config.ice_candidate_pool_size = 1;
    EXPECT_TRUE(pc_->SetConfiguration(config));
  
    // Set remote offer; can still change pool size at this point.
    CreateOfferAsRemoteDescription();
    config.ice_candidate_pool_size = 2;
    EXPECT_TRUE(pc_->SetConfiguration(config));
  
    // Set local answer; now it's too late.
    CreateAnswerAsLocalDescription();
    config.ice_candidate_pool_size = 3;
    RTCError error;
    EXPECT_FALSE(pc_->SetConfiguration(config, &error));
    EXPECT_EQ(RTCErrorType::INVALID_MODIFICATION, error.type());
  }
  
  // Test that SetConfiguration returns an invalid modification error if
  // modifying a field in the configuration that isn't allowed to be modified.
  TEST_F(PeerConnectionInterfaceTest,
         SetConfigurationReturnsInvalidModificationError) {
    PeerConnectionInterface::RTCConfiguration config;
    config.bundle_policy = PeerConnectionInterface::kBundlePolicyBalanced;
    config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyNegotiate;
    config.continual_gathering_policy = PeerConnectionInterface::GATHER_ONCE;
    CreatePeerConnection(config, nullptr);
  
    PeerConnectionInterface::RTCConfiguration modified_config = config;
    modified_config.bundle_policy =
        PeerConnectionInterface::kBundlePolicyMaxBundle;
    RTCError error;
    EXPECT_FALSE(pc_->SetConfiguration(modified_config, &error));
    EXPECT_EQ(RTCErrorType::INVALID_MODIFICATION, error.type());
  
    modified_config = config;
    modified_config.rtcp_mux_policy =
        PeerConnectionInterface::kRtcpMuxPolicyRequire;
    error.set_type(RTCErrorType::NONE);
    EXPECT_FALSE(pc_->SetConfiguration(modified_config, &error));
    EXPECT_EQ(RTCErrorType::INVALID_MODIFICATION, error.type());
  
    modified_config = config;
    modified_config.continual_gathering_policy =
        PeerConnectionInterface::GATHER_CONTINUALLY;
    error.set_type(RTCErrorType::NONE);
    EXPECT_FALSE(pc_->SetConfiguration(modified_config, &error));
    EXPECT_EQ(RTCErrorType::INVALID_MODIFICATION, error.type());
  }
  
  // Test that SetConfiguration returns a range error if the candidate pool size
  // is negative or larger than allowed by the spec.
  TEST_F(PeerConnectionInterfaceTest,
         SetConfigurationReturnsRangeErrorForBadCandidatePoolSize) {
    PeerConnectionInterface::RTCConfiguration config;
    CreatePeerConnection(config, nullptr);
  
    config.ice_candidate_pool_size = -1;
    RTCError error;
    EXPECT_FALSE(pc_->SetConfiguration(config, &error));
    EXPECT_EQ(RTCErrorType::INVALID_RANGE, error.type());
  
    config.ice_candidate_pool_size = INT_MAX;
    error.set_type(RTCErrorType::NONE);
    EXPECT_FALSE(pc_->SetConfiguration(config, &error));
    EXPECT_EQ(RTCErrorType::INVALID_RANGE, error.type());
  }
  
  // Test that SetConfiguration returns a syntax error if parsing an ICE server
  // URL failed.
  TEST_F(PeerConnectionInterfaceTest,
         SetConfigurationReturnsSyntaxErrorFromBadIceUrls) {
    PeerConnectionInterface::RTCConfiguration config;
    CreatePeerConnection(config, nullptr);
  
    PeerConnectionInterface::IceServer bad_server;
    bad_server.uri = "stunn:www.example.com";
    config.servers.push_back(bad_server);
    RTCError error;
    EXPECT_FALSE(pc_->SetConfiguration(config, &error));
    EXPECT_EQ(RTCErrorType::SYNTAX_ERROR, error.type());
  }
  
  // Test that SetConfiguration returns an invalid parameter error if a TURN
  // IceServer is missing a username or password.
  TEST_F(PeerConnectionInterfaceTest,
         SetConfigurationReturnsInvalidParameterIfCredentialsMissing) {
    PeerConnectionInterface::RTCConfiguration config;
    CreatePeerConnection(config, nullptr);
  
    PeerConnectionInterface::IceServer bad_server;
    bad_server.uri = "turn:www.example.com";
    // Missing password.
    bad_server.username = "foo";
    config.servers.push_back(bad_server);
    RTCError error;
    EXPECT_FALSE(pc_->SetConfiguration(config, &error));
    EXPECT_EQ(RTCErrorType::INVALID_PARAMETER, error.type());
  }
  
  // Test that PeerConnection::Close changes the states to closed and all remote
  // tracks change state to ended.
  TEST_F(PeerConnectionInterfaceTest, CloseAndTestStreamsAndStates) {
    // Initialize a PeerConnection and negotiate local and remote session
    // description.
    InitiateCall();
    ASSERT_EQ(1u, pc_->local_streams()->count());
    ASSERT_EQ(1u, pc_->remote_streams()->count());
  
    pc_->Close();
  
    EXPECT_EQ(PeerConnectionInterface::kClosed, pc_->signaling_state());
    EXPECT_EQ(PeerConnectionInterface::kIceConnectionClosed,
              pc_->ice_connection_state());
    EXPECT_EQ(PeerConnectionInterface::kIceGatheringComplete,
              pc_->ice_gathering_state());
  
    EXPECT_EQ(1u, pc_->local_streams()->count());
    EXPECT_EQ(1u, pc_->remote_streams()->count());
  
    rtc::scoped_refptr<MediaStreamInterface> remote_stream =
        pc_->remote_streams()->at(0);
    // Track state may be updated asynchronously.
    EXPECT_EQ_WAIT(MediaStreamTrackInterface::kEnded,
                   remote_stream->GetAudioTracks()[0]->state(), kTimeout);
    EXPECT_EQ_WAIT(MediaStreamTrackInterface::kEnded,
                   remote_stream->GetVideoTracks()[0]->state(), kTimeout);
  }
  
  // Test that PeerConnection methods fails gracefully after
  // PeerConnection::Close has been called.
  TEST_F(PeerConnectionInterfaceTest, CloseAndTestMethods) {
    CreatePeerConnectionWithoutDtls();
    AddAudioVideoStream(kStreamLabel1, "audio_label", "video_label");
    CreateOfferAsRemoteDescription();
    CreateAnswerAsLocalDescription();
  
    ASSERT_EQ(1u, pc_->local_streams()->count());
    rtc::scoped_refptr<MediaStreamInterface> local_stream =
        pc_->local_streams()->at(0);
  
    pc_->Close();
  
    pc_->RemoveStream(local_stream);
    EXPECT_FALSE(pc_->AddStream(local_stream));
  
    ASSERT_FALSE(local_stream->GetAudioTracks().empty());
    rtc::scoped_refptr<webrtc::DtmfSenderInterface> dtmf_sender(
        pc_->CreateDtmfSender(local_stream->GetAudioTracks()[0]));
    EXPECT_TRUE(NULL == dtmf_sender);  // local stream has been removed.
  
    EXPECT_TRUE(pc_->CreateDataChannel("test", NULL) == NULL);
  
    EXPECT_TRUE(pc_->local_description() != NULL);
    EXPECT_TRUE(pc_->remote_description() != NULL);
  
    std::unique_ptr<SessionDescriptionInterface> offer;
    EXPECT_TRUE(DoCreateOffer(&offer, nullptr));
    std::unique_ptr<SessionDescriptionInterface> answer;
    EXPECT_TRUE(DoCreateAnswer(&answer, nullptr));
  
    std::string sdp;
    ASSERT_TRUE(pc_->remote_description()->ToString(&sdp));
    SessionDescriptionInterface* remote_offer =
        webrtc::CreateSessionDescription(SessionDescriptionInterface::kOffer,
                                         sdp, NULL);
    EXPECT_FALSE(DoSetRemoteDescription(remote_offer));
  
    ASSERT_TRUE(pc_->local_description()->ToString(&sdp));
    SessionDescriptionInterface* local_offer =
          webrtc::CreateSessionDescription(SessionDescriptionInterface::kOffer,
                                           sdp, NULL);
    EXPECT_FALSE(DoSetLocalDescription(local_offer));
  }
  
  // Test that GetStats can still be called after PeerConnection::Close.
  TEST_F(PeerConnectionInterfaceTest, CloseAndGetStats) {
    InitiateCall();
    pc_->Close();
    DoGetStats(NULL);
  }
  
  // NOTE: The series of tests below come from what used to be
  // mediastreamsignaling_unittest.cc, and are mostly aimed at testing that
  // setting a remote or local description has the expected effects.
  
  // This test verifies that the remote MediaStreams corresponding to a received
  // SDP string is created. In this test the two separate MediaStreams are
  // signaled.
  TEST_F(PeerConnectionInterfaceTest, UpdateRemoteStreams) {
    FakeConstraints constraints;
    constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
                             true);
    CreatePeerConnection(&constraints);
    CreateAndSetRemoteOffer(kSdpStringWithStream1);
  
    rtc::scoped_refptr<StreamCollection> reference(CreateStreamCollection(1, 1));
    EXPECT_TRUE(
        CompareStreamCollections(observer_.remote_streams(), reference.get()));
    MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
    EXPECT_TRUE(remote_stream->GetVideoTracks()[0]->GetSource() != nullptr);
  
    // Create a session description based on another SDP with another
    // MediaStream.
    CreateAndSetRemoteOffer(kSdpStringWithStream1And2);
  
    rtc::scoped_refptr<StreamCollection> reference2(CreateStreamCollection(2, 1));
    EXPECT_TRUE(
        CompareStreamCollections(observer_.remote_streams(), reference2.get()));
  }
  
  // This test verifies that when remote tracks are added/removed from SDP, the
  // created remote streams are updated appropriately.
  TEST_F(PeerConnectionInterfaceTest,
         AddRemoveTrackFromExistingRemoteMediaStream) {
    FakeConstraints constraints;
    constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
                             true);
    CreatePeerConnection(&constraints);
    std::unique_ptr<SessionDescriptionInterface> desc_ms1 =
        CreateSessionDescriptionAndReference(1, 1);
    EXPECT_TRUE(DoSetRemoteDescription(desc_ms1.release()));
    EXPECT_TRUE(CompareStreamCollections(observer_.remote_streams(),
                                         reference_collection_));
  
    // Add extra audio and video tracks to the same MediaStream.
    std::unique_ptr<SessionDescriptionInterface> desc_ms1_two_tracks =
        CreateSessionDescriptionAndReference(2, 2);
    EXPECT_TRUE(DoSetRemoteDescription(desc_ms1_two_tracks.release()));
    EXPECT_TRUE(CompareStreamCollections(observer_.remote_streams(),
                                         reference_collection_));
    rtc::scoped_refptr<AudioTrackInterface> audio_track2 =
        observer_.remote_streams()->at(0)->GetAudioTracks()[1];
    EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, audio_track2->state());
    rtc::scoped_refptr<VideoTrackInterface> video_track2 =
        observer_.remote_streams()->at(0)->GetVideoTracks()[1];
    EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, video_track2->state());
  
    // Remove the extra audio and video tracks.
    std::unique_ptr<SessionDescriptionInterface> desc_ms2 =
        CreateSessionDescriptionAndReference(1, 1);
    MockTrackObserver audio_track_observer(audio_track2);
    MockTrackObserver video_track_observer(video_track2);
  
    EXPECT_CALL(audio_track_observer, OnChanged()).Times(Exactly(1));
    EXPECT_CALL(video_track_observer, OnChanged()).Times(Exactly(1));
    EXPECT_TRUE(DoSetRemoteDescription(desc_ms2.release()));
    EXPECT_TRUE(CompareStreamCollections(observer_.remote_streams(),
                                         reference_collection_));
    // Track state may be updated asynchronously.
    EXPECT_EQ_WAIT(webrtc::MediaStreamTrackInterface::kEnded,
                   audio_track2->state(), kTimeout);
    EXPECT_EQ_WAIT(webrtc::MediaStreamTrackInterface::kEnded,
                   video_track2->state(), kTimeout);
  }
  
  // This tests that remote tracks are ended if a local session description is set
  // that rejects the media content type.
  TEST_F(PeerConnectionInterfaceTest, RejectMediaContent) {
    FakeConstraints constraints;
    constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
                             true);
    CreatePeerConnection(&constraints);
    // First create and set a remote offer, then reject its video content in our
    // answer.
    CreateAndSetRemoteOffer(kSdpStringWithStream1);
    ASSERT_EQ(1u, observer_.remote_streams()->count());
    MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
    ASSERT_EQ(1u, remote_stream->GetVideoTracks().size());
    ASSERT_EQ(1u, remote_stream->GetAudioTracks().size());
  
    rtc::scoped_refptr<webrtc::VideoTrackInterface> remote_video =
        remote_stream->GetVideoTracks()[0];
    EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, remote_video->state());
    rtc::scoped_refptr<webrtc::AudioTrackInterface> remote_audio =
        remote_stream->GetAudioTracks()[0];
    EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, remote_audio->state());
  
    std::unique_ptr<SessionDescriptionInterface> local_answer;
    EXPECT_TRUE(DoCreateAnswer(&local_answer, nullptr));
    cricket::ContentInfo* video_info =
        local_answer->description()->GetContentByName("video");
    video_info->rejected = true;
    EXPECT_TRUE(DoSetLocalDescription(local_answer.release()));
    EXPECT_EQ(webrtc::MediaStreamTrackInterface::kEnded, remote_video->state());
    EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, remote_audio->state());
  
    // Now create an offer where we reject both video and audio.
    std::unique_ptr<SessionDescriptionInterface> local_offer;
    EXPECT_TRUE(DoCreateOffer(&local_offer, nullptr));
    video_info = local_offer->description()->GetContentByName("video");
    ASSERT_TRUE(video_info != nullptr);
    video_info->rejected = true;
    cricket::ContentInfo* audio_info =
        local_offer->description()->GetContentByName("audio");
    ASSERT_TRUE(audio_info != nullptr);
    audio_info->rejected = true;
    EXPECT_TRUE(DoSetLocalDescription(local_offer.release()));
    // Track state may be updated asynchronously.
    EXPECT_EQ_WAIT(webrtc::MediaStreamTrackInterface::kEnded,
                   remote_audio->state(), kTimeout);
    EXPECT_EQ_WAIT(webrtc::MediaStreamTrackInterface::kEnded,
                   remote_video->state(), kTimeout);
  }
  
  // This tests that we won't crash if the remote track has been removed outside
  // of PeerConnection and then PeerConnection tries to reject the track.
  TEST_F(PeerConnectionInterfaceTest, RemoveTrackThenRejectMediaContent) {
    FakeConstraints constraints;
    constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
                             true);
    CreatePeerConnection(&constraints);
    CreateAndSetRemoteOffer(kSdpStringWithStream1);
    MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
    remote_stream->RemoveTrack(remote_stream->GetVideoTracks()[0]);
    remote_stream->RemoveTrack(remote_stream->GetAudioTracks()[0]);
  
    std::unique_ptr<SessionDescriptionInterface> local_answer(
        webrtc::CreateSessionDescription(SessionDescriptionInterface::kAnswer,
                                         kSdpStringWithStream1, nullptr));
    cricket::ContentInfo* video_info =
        local_answer->description()->GetContentByName("video");
    video_info->rejected = true;
    cricket::ContentInfo* audio_info =
        local_answer->description()->GetContentByName("audio");
    audio_info->rejected = true;
    EXPECT_TRUE(DoSetLocalDescription(local_answer.release()));
  
    // No crash is a pass.
  }
  
  // This tests that if a recvonly remote description is set, no remote streams
  // will be created, even if the description contains SSRCs/MSIDs.
  // See: https://code.google.com/p/webrtc/issues/detail?id=5054
  TEST_F(PeerConnectionInterfaceTest, RecvonlyDescriptionDoesntCreateStream) {
    FakeConstraints constraints;
    constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
                             true);
    CreatePeerConnection(&constraints);
  
    std::string recvonly_offer = kSdpStringWithStream1;
    rtc::replace_substrs(kSendrecv, strlen(kSendrecv), kRecvonly,
                         strlen(kRecvonly), &recvonly_offer);
    CreateAndSetRemoteOffer(recvonly_offer);
  
    EXPECT_EQ(0u, observer_.remote_streams()->count());
  }
  
  // This tests that a default MediaStream is created if a remote session
  // description doesn't contain any streams and no MSID support.
  // It also tests that the default stream is updated if a video m-line is added
  // in a subsequent session description.
  TEST_F(PeerConnectionInterfaceTest, SdpWithoutMsidCreatesDefaultStream) {
    FakeConstraints constraints;
    constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
                             true);
    CreatePeerConnection(&constraints);
    CreateAndSetRemoteOffer(kSdpStringWithoutStreamsAudioOnly);
  
    ASSERT_EQ(1u, observer_.remote_streams()->count());
    MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
  
    EXPECT_EQ(1u, remote_stream->GetAudioTracks().size());
    EXPECT_EQ(0u, remote_stream->GetVideoTracks().size());
    EXPECT_EQ("default", remote_stream->label());
  
    CreateAndSetRemoteOffer(kSdpStringWithoutStreams);
    ASSERT_EQ(1u, observer_.remote_streams()->count());
    ASSERT_EQ(1u, remote_stream->GetAudioTracks().size());
    EXPECT_EQ("defaulta0", remote_stream->GetAudioTracks()[0]->id());
    EXPECT_EQ(MediaStreamTrackInterface::kLive,
              remote_stream->GetAudioTracks()[0]->state());
    ASSERT_EQ(1u, remote_stream->GetVideoTracks().size());
    EXPECT_EQ("defaultv0", remote_stream->GetVideoTracks()[0]->id());
    EXPECT_EQ(MediaStreamTrackInterface::kLive,
              remote_stream->GetVideoTracks()[0]->state());
  }
  
  // This tests that a default MediaStream is created if a remote session
  // description doesn't contain any streams and media direction is send only.
  TEST_F(PeerConnectionInterfaceTest,
         SendOnlySdpWithoutMsidCreatesDefaultStream) {
    FakeConstraints constraints;
    constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
                             true);
    CreatePeerConnection(&constraints);
    CreateAndSetRemoteOffer(kSdpStringSendOnlyWithoutStreams);
  
    ASSERT_EQ(1u, observer_.remote_streams()->count());
    MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
  
    EXPECT_EQ(1u, remote_stream->GetAudioTracks().size());
    EXPECT_EQ(1u, remote_stream->GetVideoTracks().size());
    EXPECT_EQ("default", remote_stream->label());
  }
  
  // This tests that it won't crash when PeerConnection tries to remove
  // a remote track that as already been removed from the MediaStream.
  TEST_F(PeerConnectionInterfaceTest, RemoveAlreadyGoneRemoteStream) {
    FakeConstraints constraints;
    constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
                             true);
    CreatePeerConnection(&constraints);
    CreateAndSetRemoteOffer(kSdpStringWithStream1);
    MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
    remote_stream->RemoveTrack(remote_stream->GetAudioTracks()[0]);
    remote_stream->RemoveTrack(remote_stream->GetVideoTracks()[0]);
  
    CreateAndSetRemoteOffer(kSdpStringWithoutStreams);
  
    // No crash is a pass.
  }
  
  // This tests that a default MediaStream is created if the remote session
  // description doesn't contain any streams and don't contain an indication if
  // MSID is supported.
  TEST_F(PeerConnectionInterfaceTest,
         SdpWithoutMsidAndStreamsCreatesDefaultStream) {
    FakeConstraints constraints;
    constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
                             true);
    CreatePeerConnection(&constraints);
    CreateAndSetRemoteOffer(kSdpStringWithoutStreams);
  
    ASSERT_EQ(1u, observer_.remote_streams()->count());
    MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
    EXPECT_EQ(1u, remote_stream->GetAudioTracks().size());
    EXPECT_EQ(1u, remote_stream->GetVideoTracks().size());
  }
  
  // This tests that a default MediaStream is not created if the remote session
  // description doesn't contain any streams but does support MSID.
  TEST_F(PeerConnectionInterfaceTest, SdpWithMsidDontCreatesDefaultStream) {
    FakeConstraints constraints;
    constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
                             true);
    CreatePeerConnection(&constraints);
    CreateAndSetRemoteOffer(kSdpStringWithMsidWithoutStreams);
    EXPECT_EQ(0u, observer_.remote_streams()->count());
  }
  
  // This tests that when setting a new description, the old default tracks are
  // not destroyed and recreated.
  // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=5250
  TEST_F(PeerConnectionInterfaceTest,
         DefaultTracksNotDestroyedAndRecreated) {
    FakeConstraints constraints;
    constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
                             true);
    CreatePeerConnection(&constraints);
    CreateAndSetRemoteOffer(kSdpStringWithoutStreamsAudioOnly);
  
    ASSERT_EQ(1u, observer_.remote_streams()->count());
    MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
    ASSERT_EQ(1u, remote_stream->GetAudioTracks().size());
  
    // Set the track to "disabled", then set a new description and ensure the
    // track is still disabled, which ensures it hasn't been recreated.
    remote_stream->GetAudioTracks()[0]->set_enabled(false);
    CreateAndSetRemoteOffer(kSdpStringWithoutStreamsAudioOnly);
    ASSERT_EQ(1u, remote_stream->GetAudioTracks().size());
    EXPECT_FALSE(remote_stream->GetAudioTracks()[0]->enabled());
  }
  
  // This tests that a default MediaStream is not created if a remote session
  // description is updated to not have any MediaStreams.
  TEST_F(PeerConnectionInterfaceTest, VerifyDefaultStreamIsNotCreated) {
    FakeConstraints constraints;
    constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
                             true);
    CreatePeerConnection(&constraints);
    CreateAndSetRemoteOffer(kSdpStringWithStream1);
    rtc::scoped_refptr<StreamCollection> reference(CreateStreamCollection(1, 1));
    EXPECT_TRUE(
        CompareStreamCollections(observer_.remote_streams(), reference.get()));
  
    CreateAndSetRemoteOffer(kSdpStringWithoutStreams);
    EXPECT_EQ(0u, observer_.remote_streams()->count());
  }
  
  // This tests that an RtpSender is created when the local description is set
  // after adding a local stream.
  // TODO(deadbeef): This test and the one below it need to be updated when
  // an RtpSender's lifetime isn't determined by when a local description is set.
  TEST_F(PeerConnectionInterfaceTest, LocalDescriptionChanged) {
    FakeConstraints constraints;
    constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
                             true);
    CreatePeerConnection(&constraints);
  
    // Create an offer with 1 stream with 2 tracks of each type.
    rtc::scoped_refptr<StreamCollection> stream_collection =
        CreateStreamCollection(1, 2);
    pc_->AddStream(stream_collection->at(0));
    std::unique_ptr<SessionDescriptionInterface> offer;
    ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
    EXPECT_TRUE(DoSetLocalDescription(offer.release()));
  
    auto senders = pc_->GetSenders();
    EXPECT_EQ(4u, senders.size());
    EXPECT_TRUE(ContainsSender(senders, kAudioTracks[0]));
    EXPECT_TRUE(ContainsSender(senders, kVideoTracks[0]));
    EXPECT_TRUE(ContainsSender(senders, kAudioTracks[1]));
    EXPECT_TRUE(ContainsSender(senders, kVideoTracks[1]));
  
    // Remove an audio and video track.
    pc_->RemoveStream(stream_collection->at(0));
    stream_collection = CreateStreamCollection(1, 1);
    pc_->AddStream(stream_collection->at(0));
    ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
    EXPECT_TRUE(DoSetLocalDescription(offer.release()));
  
    senders = pc_->GetSenders();
    EXPECT_EQ(2u, senders.size());
    EXPECT_TRUE(ContainsSender(senders, kAudioTracks[0]));
    EXPECT_TRUE(ContainsSender(senders, kVideoTracks[0]));
    EXPECT_FALSE(ContainsSender(senders, kAudioTracks[1]));
    EXPECT_FALSE(ContainsSender(senders, kVideoTracks[1]));
  }
  
  // This tests that an RtpSender is created when the local description is set
  // before adding a local stream.
  TEST_F(PeerConnectionInterfaceTest,
         AddLocalStreamAfterLocalDescriptionChanged) {
    FakeConstraints constraints;
    constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
                             true);
    CreatePeerConnection(&constraints);
  
    rtc::scoped_refptr<StreamCollection> stream_collection =
        CreateStreamCollection(1, 2);
    // Add a stream to create the offer, but remove it afterwards.
    pc_->AddStream(stream_collection->at(0));
    std::unique_ptr<SessionDescriptionInterface> offer;
    ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
    pc_->RemoveStream(stream_collection->at(0));
  
    EXPECT_TRUE(DoSetLocalDescription(offer.release()));
    auto senders = pc_->GetSenders();
    EXPECT_EQ(0u, senders.size());
  
    pc_->AddStream(stream_collection->at(0));
    senders = pc_->GetSenders();
    EXPECT_EQ(4u, senders.size());
    EXPECT_TRUE(ContainsSender(senders, kAudioTracks[0]));
    EXPECT_TRUE(ContainsSender(senders, kVideoTracks[0]));
    EXPECT_TRUE(ContainsSender(senders, kAudioTracks[1]));
    EXPECT_TRUE(ContainsSender(senders, kVideoTracks[1]));
  }
  
  // This tests that the expected behavior occurs if the SSRC on a local track is
  // changed when SetLocalDescription is called.
  TEST_F(PeerConnectionInterfaceTest,
         ChangeSsrcOnTrackInLocalSessionDescription) {
    FakeConstraints constraints;
    constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
                             true);
    CreatePeerConnection(&constraints);
  
    rtc::scoped_refptr<StreamCollection> stream_collection =
        CreateStreamCollection(2, 1);
    pc_->AddStream(stream_collection->at(0));
    std::unique_ptr<SessionDescriptionInterface> offer;
    ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
    // Grab a copy of the offer before it gets passed into the PC.
    std::unique_ptr<JsepSessionDescription> modified_offer(
        new JsepSessionDescription(JsepSessionDescription::kOffer));
    modified_offer->Initialize(offer->description()->Copy(), offer->session_id(),
                               offer->session_version());
    EXPECT_TRUE(DoSetLocalDescription(offer.release()));
  
    auto senders = pc_->GetSenders();
    EXPECT_EQ(2u, senders.size());
    EXPECT_TRUE(ContainsSender(senders, kAudioTracks[0]));
    EXPECT_TRUE(ContainsSender(senders, kVideoTracks[0]));
  
    // Change the ssrc of the audio and video track.
    cricket::MediaContentDescription* desc =
        cricket::GetFirstAudioContentDescription(modified_offer->description());
    ASSERT_TRUE(desc != NULL);
    for (StreamParams& stream : desc->mutable_streams()) {
      for (unsigned int& ssrc : stream.ssrcs) {
        ++ssrc;
      }
    }
  
    desc =
        cricket::GetFirstVideoContentDescription(modified_offer->description());
    ASSERT_TRUE(desc != NULL);
    for (StreamParams& stream : desc->mutable_streams()) {
      for (unsigned int& ssrc : stream.ssrcs) {
        ++ssrc;
      }
    }
  
    EXPECT_TRUE(DoSetLocalDescription(modified_offer.release()));
    senders = pc_->GetSenders();
    EXPECT_EQ(2u, senders.size());
    EXPECT_TRUE(ContainsSender(senders, kAudioTracks[0]));
    EXPECT_TRUE(ContainsSender(senders, kVideoTracks[0]));
    // TODO(deadbeef): Once RtpSenders expose parameters, check that the SSRC
    // changed.
  }
  
  // This tests that the expected behavior occurs if a new session description is
  // set with the same tracks, but on a different MediaStream.
  TEST_F(PeerConnectionInterfaceTest,
         SignalSameTracksInSeparateMediaStream) {
    FakeConstraints constraints;
    constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
                             true);
    CreatePeerConnection(&constraints);
  
    rtc::scoped_refptr<StreamCollection> stream_collection =
        CreateStreamCollection(2, 1);
    pc_->AddStream(stream_collection->at(0));
    std::unique_ptr<SessionDescriptionInterface> offer;
    ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
    EXPECT_TRUE(DoSetLocalDescription(offer.release()));
  
    auto senders = pc_->GetSenders();
    EXPECT_EQ(2u, senders.size());
    EXPECT_TRUE(ContainsSender(senders, kAudioTracks[0], kStreams[0]));
    EXPECT_TRUE(ContainsSender(senders, kVideoTracks[0], kStreams[0]));
  
    // Add a new MediaStream but with the same tracks as in the first stream.
    rtc::scoped_refptr<webrtc::MediaStreamInterface> stream_1(
        webrtc::MediaStream::Create(kStreams[1]));
    stream_1->AddTrack(stream_collection->at(0)->GetVideoTracks()[0]);
    stream_1->AddTrack(stream_collection->at(0)->GetAudioTracks()[0]);
    pc_->AddStream(stream_1);
  
    ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
    EXPECT_TRUE(DoSetLocalDescription(offer.release()));
  
    auto new_senders = pc_->GetSenders();
    // Should be the same senders as before, but with updated stream id.
    // Note that this behavior is subject to change in the future.
    // We may decide the PC should ignore existing tracks in AddStream.
    EXPECT_EQ(senders, new_senders);
    EXPECT_TRUE(ContainsSender(new_senders, kAudioTracks[0], kStreams[1]));
    EXPECT_TRUE(ContainsSender(new_senders, kVideoTracks[0], kStreams[1]));
  }
  
  // This tests that PeerConnectionObserver::OnAddTrack is correctly called.
  TEST_F(PeerConnectionInterfaceTest, OnAddTrackCallback) {
    FakeConstraints constraints;
    constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
                             true);
    CreatePeerConnection(&constraints);
    CreateAndSetRemoteOffer(kSdpStringWithStream1AudioTrackOnly);
    EXPECT_EQ(observer_.num_added_tracks_, 1);
    EXPECT_EQ(observer_.last_added_track_label_, kAudioTracks[0]);
  
    // Create and set the updated remote SDP.
    CreateAndSetRemoteOffer(kSdpStringWithStream1);
    EXPECT_EQ(observer_.num_added_tracks_, 2);
    EXPECT_EQ(observer_.last_added_track_label_, kVideoTracks[0]);
  }
  
  // Test that when SetConfiguration is called and the configuration is
  // changing, the next offer causes an ICE restart.
  TEST_F(PeerConnectionInterfaceTest, SetConfigurationCausingIceRetart) {
    PeerConnectionInterface::RTCConfiguration config;
    config.type = PeerConnectionInterface::kRelay;
    // Need to pass default constraints to prevent disabling of DTLS...
    FakeConstraints default_constraints;
    CreatePeerConnection(config, &default_constraints);
    AddAudioVideoStream(kStreamLabel1, "audio_label", "video_label");
  
    // Do initial offer/answer so there's something to restart.
    CreateOfferAsLocalDescription();
    CreateAnswerAsRemoteDescription(kSdpStringWithStream1);
  
    // Grab the ufrags.
    std::vector<std::string> initial_ufrags = GetUfrags(pc_->local_description());
  
    // Change ICE policy, which should trigger an ICE restart on the next offer.
    config.type = PeerConnectionInterface::kAll;
    EXPECT_TRUE(pc_->SetConfiguration(config));
    CreateOfferAsLocalDescription();
  
    // Grab the new ufrags.
    std::vector<std::string> subsequent_ufrags =
        GetUfrags(pc_->local_description());
  
    // Sanity check.
    EXPECT_EQ(initial_ufrags.size(), subsequent_ufrags.size());
    // Check that each ufrag is different.
    for (int i = 0; i < static_cast<int>(initial_ufrags.size()); ++i) {
      EXPECT_NE(initial_ufrags[i], subsequent_ufrags[i]);
    }
  }
  
  // Test that when SetConfiguration is called and the configuration *isn't*
  // changing, the next offer does *not* cause an ICE restart.
  TEST_F(PeerConnectionInterfaceTest, SetConfigurationNotCausingIceRetart) {
    PeerConnectionInterface::RTCConfiguration config;
    config.type = PeerConnectionInterface::kRelay;
    // Need to pass default constraints to prevent disabling of DTLS...
    FakeConstraints default_constraints;
    CreatePeerConnection(config, &default_constraints);
    AddAudioVideoStream(kStreamLabel1, "audio_label", "video_label");
  
    // Do initial offer/answer so there's something to restart.
    CreateOfferAsLocalDescription();
    CreateAnswerAsRemoteDescription(kSdpStringWithStream1);
  
    // Grab the ufrags.
    std::vector<std::string> initial_ufrags = GetUfrags(pc_->local_description());
  
    // Call SetConfiguration with a config identical to what the PC was
    // constructed with.
    EXPECT_TRUE(pc_->SetConfiguration(config));
    CreateOfferAsLocalDescription();
  
    // Grab the new ufrags.
    std::vector<std::string> subsequent_ufrags =
        GetUfrags(pc_->local_description());
  
    EXPECT_EQ(initial_ufrags, subsequent_ufrags);
  }
  
  // Test for a weird corner case scenario:
  // 1. Audio/video session established.
  // 2. SetConfiguration changes ICE config; ICE restart needed.
  // 3. ICE restart initiated by remote peer, but only for one m= section.
  // 4. Next createOffer should initiate an ICE restart, but only for the other
  //    m= section; it would be pointless to do an ICE restart for the m= section
  //    that was already restarted.
  TEST_F(PeerConnectionInterfaceTest, SetConfigurationCausingPartialIceRestart) {
    PeerConnectionInterface::RTCConfiguration config;
    config.type = PeerConnectionInterface::kRelay;
    // Need to pass default constraints to prevent disabling of DTLS...
    FakeConstraints default_constraints;
    CreatePeerConnection(config, &default_constraints);
    AddAudioVideoStream(kStreamLabel1, "audio_label", "video_label");
  
    // Do initial offer/answer so there's something to restart.
    CreateOfferAsLocalDescription();
    CreateAnswerAsRemoteDescription(kSdpStringWithStream1);
  
    // Change ICE policy, which should set the "needs-ice-restart" flag.
    config.type = PeerConnectionInterface::kAll;
    EXPECT_TRUE(pc_->SetConfiguration(config));
  
    // Do ICE restart for the first m= section, initiated by remote peer.
    webrtc::JsepSessionDescription* remote_offer =
        new webrtc::JsepSessionDescription(SessionDescriptionInterface::kOffer);
    EXPECT_TRUE(remote_offer->Initialize(kSdpStringWithStream1, nullptr));
    remote_offer->description()->transport_infos()[0].description.ice_ufrag =
        "modified";
    EXPECT_TRUE(DoSetRemoteDescription(remote_offer));
    CreateAnswerAsLocalDescription();
  
    // Grab the ufrags.
    std::vector<std::string> initial_ufrags = GetUfrags(pc_->local_description());
    ASSERT_EQ(2, initial_ufrags.size());
  
    // Create offer and grab the new ufrags.
    CreateOfferAsLocalDescription();
    std::vector<std::string> subsequent_ufrags =
        GetUfrags(pc_->local_description());
    ASSERT_EQ(2, subsequent_ufrags.size());
  
    // Ensure that only the ufrag for the second m= section changed.
    EXPECT_EQ(initial_ufrags[0], subsequent_ufrags[0]);
    EXPECT_NE(initial_ufrags[1], subsequent_ufrags[1]);
  }
  
  // Tests that the methods to return current/pending descriptions work as
  // expected at different points in the offer/answer exchange. This test does
  // one offer/answer exchange as the offerer, then another as the answerer.
  TEST_F(PeerConnectionInterfaceTest, CurrentAndPendingDescriptions) {
    // This disables DTLS so we can apply an answer to ourselves.
    CreatePeerConnection();
  
    // Create initial local offer and get SDP (which will also be used as
    // answer/pranswer);
    std::unique_ptr<SessionDescriptionInterface> offer;
    ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
    std::string sdp;
    EXPECT_TRUE(offer->ToString(&sdp));
  
    // Set local offer.
    SessionDescriptionInterface* local_offer = offer.release();
    EXPECT_TRUE(DoSetLocalDescription(local_offer));
    EXPECT_EQ(local_offer, pc_->pending_local_description());
    EXPECT_EQ(nullptr, pc_->pending_remote_description());
    EXPECT_EQ(nullptr, pc_->current_local_description());
    EXPECT_EQ(nullptr, pc_->current_remote_description());
  
    // Set remote pranswer.
    SessionDescriptionInterface* remote_pranswer =
        webrtc::CreateSessionDescription(SessionDescriptionInterface::kPrAnswer,
                                         sdp, nullptr);
    EXPECT_TRUE(DoSetRemoteDescription(remote_pranswer));
    EXPECT_EQ(local_offer, pc_->pending_local_description());
    EXPECT_EQ(remote_pranswer, pc_->pending_remote_description());
    EXPECT_EQ(nullptr, pc_->current_local_description());
    EXPECT_EQ(nullptr, pc_->current_remote_description());
  
    // Set remote answer.
    SessionDescriptionInterface* remote_answer = webrtc::CreateSessionDescription(
        SessionDescriptionInterface::kAnswer, sdp, nullptr);
    EXPECT_TRUE(DoSetRemoteDescription(remote_answer));
    EXPECT_EQ(nullptr, pc_->pending_local_description());
    EXPECT_EQ(nullptr, pc_->pending_remote_description());
    EXPECT_EQ(local_offer, pc_->current_local_description());
    EXPECT_EQ(remote_answer, pc_->current_remote_description());
  
    // Set remote offer.
    SessionDescriptionInterface* remote_offer = webrtc::CreateSessionDescription(
        SessionDescriptionInterface::kOffer, sdp, nullptr);
    EXPECT_TRUE(DoSetRemoteDescription(remote_offer));
    EXPECT_EQ(remote_offer, pc_->pending_remote_description());
    EXPECT_EQ(nullptr, pc_->pending_local_description());
    EXPECT_EQ(local_offer, pc_->current_local_description());
    EXPECT_EQ(remote_answer, pc_->current_remote_description());
  
    // Set local pranswer.
    SessionDescriptionInterface* local_pranswer =
        webrtc::CreateSessionDescription(SessionDescriptionInterface::kPrAnswer,
                                         sdp, nullptr);
    EXPECT_TRUE(DoSetLocalDescription(local_pranswer));
    EXPECT_EQ(remote_offer, pc_->pending_remote_description());
    EXPECT_EQ(local_pranswer, pc_->pending_local_description());
    EXPECT_EQ(local_offer, pc_->current_local_description());
    EXPECT_EQ(remote_answer, pc_->current_remote_description());
  
    // Set local answer.
    SessionDescriptionInterface* local_answer = webrtc::CreateSessionDescription(
        SessionDescriptionInterface::kAnswer, sdp, nullptr);
    EXPECT_TRUE(DoSetLocalDescription(local_answer));
    EXPECT_EQ(nullptr, pc_->pending_remote_description());
    EXPECT_EQ(nullptr, pc_->pending_local_description());
    EXPECT_EQ(remote_offer, pc_->current_remote_description());
    EXPECT_EQ(local_answer, pc_->current_local_description());
  }
  
  class PeerConnectionMediaConfigTest : public testing::Test {
   protected:
    void SetUp() override {
      pcf_ = new rtc::RefCountedObject<PeerConnectionFactoryForTest>();
      pcf_->Initialize();
    }
    const cricket::MediaConfig& TestCreatePeerConnection(
        const PeerConnectionInterface::RTCConfiguration& config,
        const MediaConstraintsInterface *constraints) {
      pcf_->create_media_controller_called_ = false;
  
      rtc::scoped_refptr<PeerConnectionInterface> pc(pcf_->CreatePeerConnection(
          config, constraints, nullptr, nullptr, &observer_));
      EXPECT_TRUE(pc.get());
      EXPECT_TRUE(pcf_->create_media_controller_called_);
      return pcf_->create_media_controller_config_;
    }
  
    rtc::scoped_refptr<PeerConnectionFactoryForTest> pcf_;
    MockPeerConnectionObserver observer_;
  };
  
  // This test verifies the default behaviour with no constraints and a
  // default RTCConfiguration.
  TEST_F(PeerConnectionMediaConfigTest, TestDefaults) {
    PeerConnectionInterface::RTCConfiguration config;
    FakeConstraints constraints;
  
    const cricket::MediaConfig& media_config =
        TestCreatePeerConnection(config, &constraints);
  
    EXPECT_FALSE(media_config.enable_dscp);
    EXPECT_TRUE(media_config.video.enable_cpu_overuse_detection);
    EXPECT_FALSE(media_config.video.disable_prerenderer_smoothing);
    EXPECT_FALSE(media_config.video.suspend_below_min_bitrate);
  }
  
  // This test verifies the DSCP constraint is recognized and passed to
  // the CreateMediaController call.
  TEST_F(PeerConnectionMediaConfigTest, TestDscpConstraintTrue) {
    PeerConnectionInterface::RTCConfiguration config;
    FakeConstraints constraints;
  
    constraints.AddOptional(webrtc::MediaConstraintsInterface::kEnableDscp, true);
    const cricket::MediaConfig& media_config =
        TestCreatePeerConnection(config, &constraints);
  
    EXPECT_TRUE(media_config.enable_dscp);
  }
  
  // This test verifies the cpu overuse detection constraint is
  // recognized and passed to the CreateMediaController call.
  TEST_F(PeerConnectionMediaConfigTest, TestCpuOveruseConstraintFalse) {
    PeerConnectionInterface::RTCConfiguration config;
    FakeConstraints constraints;
  
    constraints.AddOptional(
        webrtc::MediaConstraintsInterface::kCpuOveruseDetection, false);
    const cricket::MediaConfig media_config =
        TestCreatePeerConnection(config, &constraints);
  
    EXPECT_FALSE(media_config.video.enable_cpu_overuse_detection);
  }
  
  // This test verifies that the disable_prerenderer_smoothing flag is
  // propagated from RTCConfiguration to the CreateMediaController call.
  TEST_F(PeerConnectionMediaConfigTest, TestDisablePrerendererSmoothingTrue) {
    PeerConnectionInterface::RTCConfiguration config;
    FakeConstraints constraints;
  
    config.set_prerenderer_smoothing(false);
    const cricket::MediaConfig& media_config =
        TestCreatePeerConnection(config, &constraints);
  
    EXPECT_TRUE(media_config.video.disable_prerenderer_smoothing);
  }
  
  // This test verifies the suspend below min bitrate constraint is
  // recognized and passed to the CreateMediaController call.
  TEST_F(PeerConnectionMediaConfigTest,
         TestSuspendBelowMinBitrateConstraintTrue) {
    PeerConnectionInterface::RTCConfiguration config;
    FakeConstraints constraints;
  
    constraints.AddOptional(
        webrtc::MediaConstraintsInterface::kEnableVideoSuspendBelowMinBitrate,
        true);
    const cricket::MediaConfig media_config =
        TestCreatePeerConnection(config, &constraints);
  
    EXPECT_TRUE(media_config.video.suspend_below_min_bitrate);
  }
  
  // The following tests verify that session options are created correctly.
  // TODO(deadbeef): Convert these tests to be more end-to-end. Instead of
  // "verify options are converted correctly", should be "pass options into
  // CreateOffer and verify the correct offer is produced."
  
  TEST(CreateSessionOptionsTest, GetOptionsForOfferWithInvalidAudioOption) {
    RTCOfferAnswerOptions rtc_options;
    rtc_options.offer_to_receive_audio = RTCOfferAnswerOptions::kUndefined - 1;
  
    cricket::MediaSessionOptions options;
    EXPECT_FALSE(ExtractMediaSessionOptions(rtc_options, true, &options));
  
    rtc_options.offer_to_receive_audio =
        RTCOfferAnswerOptions::kMaxOfferToReceiveMedia + 1;
    EXPECT_FALSE(ExtractMediaSessionOptions(rtc_options, true, &options));
  }
  
  TEST(CreateSessionOptionsTest, GetOptionsForOfferWithInvalidVideoOption) {
    RTCOfferAnswerOptions rtc_options;
    rtc_options.offer_to_receive_video = RTCOfferAnswerOptions::kUndefined - 1;
  
    cricket::MediaSessionOptions options;
    EXPECT_FALSE(ExtractMediaSessionOptions(rtc_options, true, &options));
  
    rtc_options.offer_to_receive_video =
        RTCOfferAnswerOptions::kMaxOfferToReceiveMedia + 1;
    EXPECT_FALSE(ExtractMediaSessionOptions(rtc_options, true, &options));
  }
  
  // Test that a MediaSessionOptions is created for an offer if
  // OfferToReceiveAudio and OfferToReceiveVideo options are set.
  TEST(CreateSessionOptionsTest, GetMediaSessionOptionsForOfferWithAudioVideo) {
    RTCOfferAnswerOptions rtc_options;
    rtc_options.offer_to_receive_audio = 1;
    rtc_options.offer_to_receive_video = 1;
  
    cricket::MediaSessionOptions options;
    EXPECT_TRUE(ExtractMediaSessionOptions(rtc_options, true, &options));
    EXPECT_TRUE(options.has_audio());
    EXPECT_TRUE(options.has_video());
    EXPECT_TRUE(options.bundle_enabled);
  }
  
  // Test that a correct MediaSessionOptions is created for an offer if
  // OfferToReceiveAudio is set.
  TEST(CreateSessionOptionsTest, GetMediaSessionOptionsForOfferWithAudio) {
    RTCOfferAnswerOptions rtc_options;
    rtc_options.offer_to_receive_audio = 1;
  
    cricket::MediaSessionOptions options;
    EXPECT_TRUE(ExtractMediaSessionOptions(rtc_options, true, &options));
    EXPECT_TRUE(options.has_audio());
    EXPECT_FALSE(options.has_video());
    EXPECT_TRUE(options.bundle_enabled);
  }
  
  // Test that a correct MediaSessionOptions is created for an offer if
  // the default OfferOptions are used.
  TEST(CreateSessionOptionsTest, GetDefaultMediaSessionOptionsForOffer) {
    RTCOfferAnswerOptions rtc_options;
  
    cricket::MediaSessionOptions options;
    options.transport_options["audio"] = cricket::TransportOptions();
    options.transport_options["video"] = cricket::TransportOptions();
    EXPECT_TRUE(ExtractMediaSessionOptions(rtc_options, true, &options));
    EXPECT_TRUE(options.has_audio());
    EXPECT_FALSE(options.has_video());
    EXPECT_TRUE(options.bundle_enabled);
    EXPECT_TRUE(options.vad_enabled);
    EXPECT_FALSE(options.transport_options["audio"].ice_restart);
    EXPECT_FALSE(options.transport_options["video"].ice_restart);
  }
  
  // Test that a correct MediaSessionOptions is created for an offer if
  // OfferToReceiveVideo is set.
  TEST(CreateSessionOptionsTest, GetMediaSessionOptionsForOfferWithVideo) {
    RTCOfferAnswerOptions rtc_options;
    rtc_options.offer_to_receive_audio = 0;
    rtc_options.offer_to_receive_video = 1;
  
    cricket::MediaSessionOptions options;
    EXPECT_TRUE(ExtractMediaSessionOptions(rtc_options, true, &options));
    EXPECT_FALSE(options.has_audio());
    EXPECT_TRUE(options.has_video());
    EXPECT_TRUE(options.bundle_enabled);
  }
  
  // Test that a correct MediaSessionOptions is created for an offer if
  // UseRtpMux is set to false.
  TEST(CreateSessionOptionsTest,
       GetMediaSessionOptionsForOfferWithBundleDisabled) {
    RTCOfferAnswerOptions rtc_options;
    rtc_options.offer_to_receive_audio = 1;
    rtc_options.offer_to_receive_video = 1;
    rtc_options.use_rtp_mux = false;
  
    cricket::MediaSessionOptions options;
    EXPECT_TRUE(ExtractMediaSessionOptions(rtc_options, true, &options));
    EXPECT_TRUE(options.has_audio());
    EXPECT_TRUE(options.has_video());
    EXPECT_FALSE(options.bundle_enabled);
  }
  
  // Test that a correct MediaSessionOptions is created to restart ice if
  // IceRestart is set. It also tests that subsequent MediaSessionOptions don't
  // have |audio_transport_options.ice_restart| etc. set.
  TEST(CreateSessionOptionsTest, GetMediaSessionOptionsForOfferWithIceRestart) {
    RTCOfferAnswerOptions rtc_options;
    rtc_options.ice_restart = true;
  
    cricket::MediaSessionOptions options;
    options.transport_options["audio"] = cricket::TransportOptions();
    options.transport_options["video"] = cricket::TransportOptions();
    EXPECT_TRUE(ExtractMediaSessionOptions(rtc_options, true, &options));
    EXPECT_TRUE(options.transport_options["audio"].ice_restart);
    EXPECT_TRUE(options.transport_options["video"].ice_restart);
  
    rtc_options = RTCOfferAnswerOptions();
    EXPECT_TRUE(ExtractMediaSessionOptions(rtc_options, true, &options));
    EXPECT_FALSE(options.transport_options["audio"].ice_restart);
    EXPECT_FALSE(options.transport_options["video"].ice_restart);
  }
  
  // Test that the MediaConstraints in an answer don't affect if audio and video
  // is offered in an offer but that if kOfferToReceiveAudio or
  // kOfferToReceiveVideo constraints are true in an offer, the media type will be
  // included in subsequent answers.
  TEST(CreateSessionOptionsTest, MediaConstraintsInAnswer) {
    FakeConstraints answer_c;
    answer_c.SetMandatoryReceiveAudio(true);
    answer_c.SetMandatoryReceiveVideo(true);
  
    cricket::MediaSessionOptions answer_options;
    EXPECT_TRUE(ParseConstraintsForAnswer(&answer_c, &answer_options));
    EXPECT_TRUE(answer_options.has_audio());
    EXPECT_TRUE(answer_options.has_video());
  
    RTCOfferAnswerOptions rtc_offer_options;
  
    cricket::MediaSessionOptions offer_options;
    EXPECT_TRUE(
        ExtractMediaSessionOptions(rtc_offer_options, false, &offer_options));
    EXPECT_TRUE(offer_options.has_audio());
    EXPECT_TRUE(offer_options.has_video());
  
    RTCOfferAnswerOptions updated_rtc_offer_options;
    updated_rtc_offer_options.offer_to_receive_audio = 1;
    updated_rtc_offer_options.offer_to_receive_video = 1;
  
    cricket::MediaSessionOptions updated_offer_options;
    EXPECT_TRUE(ExtractMediaSessionOptions(updated_rtc_offer_options, false,
                                           &updated_offer_options));
    EXPECT_TRUE(updated_offer_options.has_audio());
    EXPECT_TRUE(updated_offer_options.has_video());
  
    // Since an offer has been created with both audio and video, subsequent
    // offers and answers should contain both audio and video.
    // Answers will only contain the media types that exist in the offer
    // regardless of the value of |updated_answer_options.has_audio| and
    // |updated_answer_options.has_video|.
    FakeConstraints updated_answer_c;
    answer_c.SetMandatoryReceiveAudio(false);
    answer_c.SetMandatoryReceiveVideo(false);
  
    cricket::MediaSessionOptions updated_answer_options;
    EXPECT_TRUE(
        ParseConstraintsForAnswer(&updated_answer_c, &updated_answer_options));
    EXPECT_TRUE(updated_answer_options.has_audio());
    EXPECT_TRUE(updated_answer_options.has_video());
  }
  
  TEST(RTCErrorTypeTest, OstreamOperator) {
    std::ostringstream oss;
    oss << webrtc::RTCErrorType::NONE << ' '
        << webrtc::RTCErrorType::INVALID_PARAMETER << ' '
        << webrtc::RTCErrorType::INTERNAL_ERROR;
    EXPECT_EQ("NONE INVALID_PARAMETER INTERNAL_ERROR", oss.str());
  }
  
  // Tests a few random fields being different.
  TEST(RTCConfigurationTest, ComparisonOperators) {
    PeerConnectionInterface::RTCConfiguration a;
    PeerConnectionInterface::RTCConfiguration b;
    EXPECT_EQ(a, b);
  
    PeerConnectionInterface::RTCConfiguration c;
    c.servers.push_back(PeerConnectionInterface::IceServer());
    EXPECT_NE(a, c);
  
    PeerConnectionInterface::RTCConfiguration d;
    d.type = PeerConnectionInterface::kRelay;
    EXPECT_NE(a, d);
  
    PeerConnectionInterface::RTCConfiguration e;
    e.audio_jitter_buffer_max_packets = 5;
    EXPECT_NE(a, e);
  
    PeerConnectionInterface::RTCConfiguration f;
    f.ice_connection_receiving_timeout = 1337;
    EXPECT_NE(a, f);
  
    PeerConnectionInterface::RTCConfiguration g;
    g.disable_ipv6 = true;
    EXPECT_NE(a, g);
  
    PeerConnectionInterface::RTCConfiguration h(
        PeerConnectionInterface::RTCConfigurationType::kAggressive);
    EXPECT_NE(a, h);
  }
  