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 "api/peer_connection_interface.h"
  
  #include <limits.h>
  #include <stdint.h>
  #include <string.h>
  
  #include <memory>
  #include <string>
  #include <utility>
  #include <vector>
  
  #include "absl/strings/str_replace.h"
  #include "absl/types/optional.h"
  #include "api/audio/audio_mixer.h"
  #include "api/audio_codecs/audio_decoder_factory.h"
  #include "api/audio_codecs/audio_encoder_factory.h"
  #include "api/audio_codecs/builtin_audio_decoder_factory.h"
  #include "api/audio_codecs/builtin_audio_encoder_factory.h"
  #include "api/call/call_factory_interface.h"
  #include "api/create_peerconnection_factory.h"
  #include "api/data_channel_interface.h"
  #include "api/jsep.h"
  #include "api/jsep_session_description.h"
  #include "api/media_stream_interface.h"
  #include "api/media_types.h"
  #include "api/rtc_error.h"
  #include "api/rtc_event_log/rtc_event_log.h"
  #include "api/rtc_event_log/rtc_event_log_factory.h"
  #include "api/rtc_event_log_output.h"
  #include "api/rtc_event_log_output_file.h"
  #include "api/rtp_receiver_interface.h"
  #include "api/rtp_sender_interface.h"
  #include "api/rtp_transceiver_interface.h"
  #include "api/scoped_refptr.h"
  #include "api/task_queue/default_task_queue_factory.h"
  #include "api/transport/field_trial_based_config.h"
  #include "api/video_codecs/builtin_video_decoder_factory.h"
  #include "api/video_codecs/builtin_video_encoder_factory.h"
  #include "api/video_codecs/video_decoder_factory.h"
  #include "api/video_codecs/video_encoder_factory.h"
  #include "media/base/codec.h"
  #include "media/base/media_config.h"
  #include "media/base/media_engine.h"
  #include "media/base/stream_params.h"
  #include "media/engine/webrtc_media_engine.h"
  #include "media/engine/webrtc_media_engine_defaults.h"
  #include "media/sctp/sctp_transport_internal.h"
  #include "modules/audio_device/include/audio_device.h"
  #include "modules/audio_processing/include/audio_processing.h"
  #include "p2p/base/fake_port_allocator.h"
  #include "p2p/base/p2p_constants.h"
  #include "p2p/base/port.h"
  #include "p2p/base/port_allocator.h"
  #include "p2p/base/transport_description.h"
  #include "p2p/base/transport_info.h"
  #include "pc/audio_track.h"
  #include "pc/media_session.h"
  #include "pc/media_stream.h"
  #include "pc/peer_connection.h"
  #include "pc/peer_connection_factory.h"
  #include "pc/rtc_stats_collector.h"
  #include "pc/rtp_sender.h"
  #include "pc/session_description.h"
  #include "pc/stream_collection.h"
  #include "pc/test/fake_audio_capture_module.h"
  #include "pc/test/fake_rtc_certificate_generator.h"
  #include "pc/test/fake_video_track_source.h"
  #include "pc/test/mock_peer_connection_observers.h"
  #include "pc/test/test_sdp_strings.h"
  #include "pc/video_track.h"
  #include "rtc_base/checks.h"
  #include "rtc_base/copy_on_write_buffer.h"
  #include "rtc_base/gunit.h"
  #include "rtc_base/ref_counted_object.h"
  #include "rtc_base/rtc_certificate_generator.h"
  #include "rtc_base/socket_address.h"
  #include "rtc_base/thread.h"
  #include "rtc_base/time_utils.h"
  #include "rtc_base/virtual_socket_server.h"
  #include "test/gmock.h"
  #include "test/gtest.h"
  #include "test/testsupport/file_utils.h"
  
  #ifdef WEBRTC_ANDROID
  #include "pc/test/android_test_initializer.h"
  #endif
  
  namespace webrtc {
  namespace {
  
  static const char kStreamId1[] = "local_stream_1";
  static const char kStreamId2[] = "local_stream_2";
  static const char kStreamId3[] = "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: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 kSdpStringWithStream1PlanB[] =
      "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";
  // Same string as above but with the MID changed to the Unified Plan default.
  // This is needed so that this SDP can be used as an answer for a Unified Plan
  // offer.
  static const char kSdpStringWithStream1UnifiedPlan[] =
      "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:0\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:1\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 kSdpStringWithStream1And2PlanB[] =
      "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";
  static const char kSdpStringWithStream1And2UnifiedPlan[] =
      "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:0\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"
      "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:1\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"
      "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:2\r\n"
      "a=sendrecv\r\n"
      "a=rtcp-mux\r\n"
      "a=rtpmap:103 ISAC/16000\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:3\r\n"
      "a=sendrecv\r\n"
      "a=rtcp-mux\r\n"
      "a=rtpmap:120 VP8/0\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:0 AES_CM_128_HMAC_SHA1_80 "
      "inline:NzB4d1BINUAvLEw6UzF3WSJ+PSdFcGdUJShpX1Zj|2^20|1:32 "
      "dummy_session_params\r\n";
  
  class RtcEventLogOutputNull final : public RtcEventLogOutput {
   public:
    bool IsActive() const override { return true; }
    bool Write(const std::string& output) override { return true; }
  };
  
  using ::cricket::StreamParams;
  using ::testing::Exactly;
  using ::testing::Values;
  
  using RTCConfiguration = PeerConnectionInterface::RTCConfiguration;
  using RTCOfferAnswerOptions = PeerConnectionInterface::RTCOfferAnswerOptions;
  
  // 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 =
        content_info->media_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_id,
                     const std::string& track_id) {
    for (const cricket::StreamParams& params : streams) {
      if (params.first_stream_id() == stream_id && 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(),
                                       rtc::Thread::Current()));
        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)->id() != s2->at(i)->id()) {
        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_METHOD(void, OnChanged, (), (override));
  
   private:
    NotifierInterface* notifier_;
  };
  
  // The PeerConnectionMediaConfig tests below verify that configuration and
  // constraints are propagated into the PeerConnection's MediaConfig. These
  // settings are intended for MediaChannel constructors, but that is not
  // exercised by these unittest.
  class PeerConnectionFactoryForTest : public webrtc::PeerConnectionFactory {
   public:
    static rtc::scoped_refptr<PeerConnectionFactoryForTest>
    CreatePeerConnectionFactoryForTest() {
      PeerConnectionFactoryDependencies dependencies;
      dependencies.worker_thread = rtc::Thread::Current();
      dependencies.network_thread = rtc::Thread::Current();
      dependencies.signaling_thread = rtc::Thread::Current();
      dependencies.task_queue_factory = CreateDefaultTaskQueueFactory();
      dependencies.trials = std::make_unique<FieldTrialBasedConfig>();
      cricket::MediaEngineDependencies media_deps;
      media_deps.task_queue_factory = dependencies.task_queue_factory.get();
      // Use fake audio device module since we're only testing the interface
      // level, and using a real one could make tests flaky when run in parallel.
      media_deps.adm = FakeAudioCaptureModule::Create();
      SetMediaEngineDefaults(&media_deps);
      media_deps.trials = dependencies.trials.get();
      dependencies.media_engine =
          cricket::CreateMediaEngine(std::move(media_deps));
      dependencies.call_factory = webrtc::CreateCallFactory();
      dependencies.event_log_factory = std::make_unique<RtcEventLogFactory>(
          dependencies.task_queue_factory.get());
  
      return new rtc::RefCountedObject<PeerConnectionFactoryForTest>(
          std::move(dependencies));
    }
  
    using PeerConnectionFactory::PeerConnectionFactory;
  
   private:
    rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_;
  };
  
  // TODO(steveanton): Convert to use the new PeerConnectionWrapper.
  class PeerConnectionInterfaceBaseTest : public ::testing::Test {
   protected:
    explicit PeerConnectionInterfaceBaseTest(SdpSemantics sdp_semantics)
        : vss_(new rtc::VirtualSocketServer()),
          main_(vss_.get()),
          sdp_semantics_(sdp_semantics) {
  #ifdef WEBRTC_ANDROID
      webrtc::InitializeAndroidObjects();
  #endif
    }
  
    void SetUp() override {
      // Use fake audio capture module since we're only testing the interface
      // level, and using a real one could make tests flaky when run in parallel.
      fake_audio_capture_module_ = FakeAudioCaptureModule::Create();
      pc_factory_ = webrtc::CreatePeerConnectionFactory(
          rtc::Thread::Current(), rtc::Thread::Current(), rtc::Thread::Current(),
          rtc::scoped_refptr<webrtc::AudioDeviceModule>(
              fake_audio_capture_module_),
          webrtc::CreateBuiltinAudioEncoderFactory(),
          webrtc::CreateBuiltinAudioDecoderFactory(),
          webrtc::CreateBuiltinVideoEncoderFactory(),
          webrtc::CreateBuiltinVideoDecoderFactory(), nullptr /* audio_mixer */,
          nullptr /* audio_processing */);
      ASSERT_TRUE(pc_factory_);
      pc_factory_for_test_ =
          PeerConnectionFactoryForTest::CreatePeerConnectionFactoryForTest();
    }
  
    void TearDown() override {
      if (pc_)
        pc_->Close();
    }
  
    void CreatePeerConnection() {
      CreatePeerConnection(PeerConnectionInterface::RTCConfiguration());
    }
  
    // DTLS does not work in a loopback call, so is disabled for most of the
    // tests in this file.
    void CreatePeerConnectionWithoutDtls() {
      RTCConfiguration config;
      config.enable_dtls_srtp = false;
  
      CreatePeerConnection(config);
    }
  
    void CreatePeerConnectionWithIceTransportsType(
        PeerConnectionInterface::IceTransportsType type) {
      PeerConnectionInterface::RTCConfiguration config;
      config.type = type;
      return CreatePeerConnection(config);
    }
  
    void CreatePeerConnectionWithIceServer(const std::string& uri,
                                           const std::string& username,
                                           const std::string& password) {
      PeerConnectionInterface::RTCConfiguration config;
      PeerConnectionInterface::IceServer server;
      server.uri = uri;
      server.username = username;
      server.password = password;
      config.servers.push_back(server);
      CreatePeerConnection(config);
    }
  
    void CreatePeerConnection(const RTCConfiguration& config) {
      if (pc_) {
        pc_->Close();
        pc_ = nullptr;
      }
      std::unique_ptr<cricket::FakePortAllocator> port_allocator(
          new cricket::FakePortAllocator(rtc::Thread::Current(), nullptr));
      port_allocator_ = port_allocator.get();
  
      // Create certificate generator unless DTLS constraint is explicitly set to
      // false.
      std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator;
  
      if (config.enable_dtls_srtp.value_or(true)) {
        fake_certificate_generator_ = new FakeRTCCertificateGenerator();
        cert_generator.reset(fake_certificate_generator_);
      }
      RTCConfiguration modified_config = config;
      modified_config.sdp_semantics = sdp_semantics_;
      pc_ = pc_factory_->CreatePeerConnection(
          modified_config, 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);
      config.sdp_semantics = sdp_semantics_;
      rtc::scoped_refptr<PeerConnectionInterface> pc =
          pc_factory_->CreatePeerConnection(config, nullptr, nullptr, &observer_);
      EXPECT_EQ(nullptr, pc);
    }
  
    void CreatePeerConnectionExpectFail(
        PeerConnectionInterface::RTCConfiguration config) {
      PeerConnectionInterface::IceServer server;
      server.uri = kTurnIceServerUri;
      server.password = kTurnPassword;
      config.servers.push_back(server);
      config.sdp_semantics = sdp_semantics_;
      rtc::scoped_refptr<PeerConnectionInterface> pc =
          pc_factory_->CreatePeerConnection(config, 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, kTurnUsername,
                                        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);
    }
  
    rtc::scoped_refptr<VideoTrackInterface> CreateVideoTrack(
        const std::string& label) {
      return pc_factory_->CreateVideoTrack(label, FakeVideoTrackSource::Create());
    }
  
    void AddVideoTrack(const std::string& track_label,
                       const std::vector<std::string>& stream_ids = {}) {
      auto sender_or_error =
          pc_->AddTrack(CreateVideoTrack(track_label), stream_ids);
      ASSERT_EQ(RTCErrorType::NONE, sender_or_error.error().type());
    }
  
    void AddVideoStream(const std::string& label) {
      rtc::scoped_refptr<MediaStreamInterface> stream(
          pc_factory_->CreateLocalMediaStream(label));
      stream->AddTrack(CreateVideoTrack(label + "v0"));
      ASSERT_TRUE(pc_->AddStream(stream));
    }
  
    rtc::scoped_refptr<AudioTrackInterface> CreateAudioTrack(
        const std::string& label) {
      return pc_factory_->CreateAudioTrack(label, nullptr);
    }
  
    void AddAudioTrack(const std::string& track_label,
                       const std::vector<std::string>& stream_ids = {}) {
      auto sender_or_error =
          pc_->AddTrack(CreateAudioTrack(track_label), stream_ids);
      ASSERT_EQ(RTCErrorType::NONE, sender_or_error.error().type());
    }
  
    void AddAudioStream(const std::string& label) {
      rtc::scoped_refptr<MediaStreamInterface> stream(
          pc_factory_->CreateLocalMediaStream(label));
      stream->AddTrack(CreateAudioTrack(label + "a0"));
      ASSERT_TRUE(pc_->AddStream(stream));
    }
  
    void AddAudioVideoStream(const std::string& stream_id,
                             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_id));
      stream->AddTrack(CreateAudioTrack(audio_track_label));
      stream->AddTrack(CreateVideoTrack(video_track_label));
      ASSERT_TRUE(pc_->AddStream(stream));
    }
  
    rtc::scoped_refptr<RtpReceiverInterface> GetFirstReceiverOfType(
        cricket::MediaType media_type) {
      for (auto receiver : pc_->GetReceivers()) {
        if (receiver->media_type() == media_type) {
          return receiver;
        }
      }
      return nullptr;
    }
  
    bool DoCreateOfferAnswer(std::unique_ptr<SessionDescriptionInterface>* desc,
                             const RTCOfferAnswerOptions* options,
                             bool offer) {
      rtc::scoped_refptr<MockCreateSessionDescriptionObserver> observer(
          new rtc::RefCountedObject<MockCreateSessionDescriptionObserver>());
      if (offer) {
        pc_->CreateOffer(observer, options ? *options : RTCOfferAnswerOptions());
      } else {
        pc_->CreateAnswer(observer, options ? *options : RTCOfferAnswerOptions());
      }
      EXPECT_EQ_WAIT(true, observer->called(), kTimeout);
      *desc = observer->MoveDescription();
      return observer->result();
    }
  
    bool DoCreateOffer(std::unique_ptr<SessionDescriptionInterface>* desc,
                       const RTCOfferAnswerOptions* options) {
      return DoCreateOfferAnswer(desc, options, true);
    }
  
    bool DoCreateAnswer(std::unique_ptr<SessionDescriptionInterface>* desc,
                        const RTCOfferAnswerOptions* options) {
      return DoCreateOfferAnswer(desc, options, false);
    }
  
    bool DoSetSessionDescription(
        std::unique_ptr<SessionDescriptionInterface> desc,
        bool local) {
      rtc::scoped_refptr<MockSetSessionDescriptionObserver> observer(
          new rtc::RefCountedObject<MockSetSessionDescriptionObserver>());
      if (local) {
        pc_->SetLocalDescription(observer, desc.release());
      } else {
        pc_->SetRemoteDescription(observer, desc.release());
      }
      if (pc_->signaling_state() != PeerConnectionInterface::kClosed) {
        EXPECT_EQ_WAIT(true, observer->called(), kTimeout);
      }
      return observer->result();
    }
  
    bool DoSetLocalDescription(
        std::unique_ptr<SessionDescriptionInterface> desc) {
      return DoSetSessionDescription(std::move(desc), true);
    }
  
    bool DoSetRemoteDescription(
        std::unique_ptr<SessionDescriptionInterface> desc) {
      return DoSetSessionDescription(std::move(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();
    }
  
    // Call the standards-compliant GetStats function.
    bool DoGetRTCStats() {
      rtc::scoped_refptr<webrtc::MockRTCStatsCollectorCallback> callback(
          new rtc::RefCountedObject<webrtc::MockRTCStatsCollectorCallback>());
      pc_->GetStats(callback);
      EXPECT_TRUE_WAIT(callback->called(), kTimeout);
      return callback->called();
    }
  
    void InitiateCall() {
      CreatePeerConnectionWithoutDtls();
      // Create a local stream with audio&video tracks.
      if (sdp_semantics_ == SdpSemantics::kPlanB) {
        AddAudioVideoStream(kStreamId1, "audio_track", "video_track");
      } else {
        // Unified Plan does not support AddStream, so just add an audio and video
        // track.
        AddAudioTrack(kAudioTracks[0], {kStreamId1});
        AddVideoTrack(kVideoTracks[0], {kStreamId1});
      }
      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));
      std::unique_ptr<SessionDescriptionInterface> remote_offer(
          webrtc::CreateSessionDescription(SdpType::kOffer, sdp));
      EXPECT_TRUE(DoSetRemoteDescription(std::move(remote_offer)));
      EXPECT_EQ(PeerConnectionInterface::kHaveRemoteOffer, observer_.state_);
    }
  
    void CreateAndSetRemoteOffer(const std::string& sdp) {
      std::unique_ptr<SessionDescriptionInterface> remote_offer(
          webrtc::CreateSessionDescription(SdpType::kOffer, sdp));
      EXPECT_TRUE(DoSetRemoteDescription(std::move(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));
      std::unique_ptr<SessionDescriptionInterface> new_answer(
          webrtc::CreateSessionDescription(SdpType::kAnswer, sdp));
      EXPECT_TRUE(DoSetLocalDescription(std::move(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));
      std::unique_ptr<SessionDescriptionInterface> pr_answer(
          webrtc::CreateSessionDescription(SdpType::kPrAnswer, sdp));
      EXPECT_TRUE(DoSetLocalDescription(std::move(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));
      std::unique_ptr<SessionDescriptionInterface> new_offer(
          webrtc::CreateSessionDescription(SdpType::kOffer, sdp));
  
      EXPECT_TRUE(DoSetLocalDescription(std::move(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_gathering_complete_, kTimeout);
    }
  
    void CreateAnswerAsRemoteDescription(const std::string& sdp) {
      std::unique_ptr<SessionDescriptionInterface> answer(
          webrtc::CreateSessionDescription(SdpType::kAnswer, sdp));
      ASSERT_TRUE(answer);
      EXPECT_TRUE(DoSetRemoteDescription(std::move(answer)));
      EXPECT_EQ(PeerConnectionInterface::kStable, observer_.state_);
    }
  
    void CreatePrAnswerAndAnswerAsRemoteDescription(const std::string& sdp) {
      std::unique_ptr<SessionDescriptionInterface> pr_answer(
          webrtc::CreateSessionDescription(SdpType::kPrAnswer, sdp));
      ASSERT_TRUE(pr_answer);
      EXPECT_TRUE(DoSetRemoteDescription(std::move(pr_answer)));
      EXPECT_EQ(PeerConnectionInterface::kHaveRemotePrAnswer, observer_.state_);
      std::unique_ptr<SessionDescriptionInterface> answer(
          webrtc::CreateSessionDescription(SdpType::kAnswer, sdp));
      ASSERT_TRUE(answer);
      EXPECT_TRUE(DoSetRemoteDescription(std::move(answer)));
      EXPECT_EQ(PeerConnectionInterface::kStable, observer_.state_);
    }
  
    // Waits until a remote stream with the given id is signaled. This helper
    // function will verify both OnAddTrack and OnAddStream (Plan B only) are
    // called with the given stream id and expected number of tracks.
    void WaitAndVerifyOnAddStream(const std::string& stream_id,
                                  int expected_num_tracks) {
      // Verify that both OnAddStream and OnAddTrack are called.
      EXPECT_EQ_WAIT(stream_id, observer_.GetLastAddedStreamId(), kTimeout);
      EXPECT_EQ_WAIT(expected_num_tracks,
                     observer_.CountAddTrackEventsForStream(stream_id), 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_id = kStreams[0];
  
      rtc::scoped_refptr<webrtc::MediaStreamInterface> stream(
          webrtc::MediaStream::Create(mediastream_id));
      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(SdpType::kOffer, sdp_ms1));
    }
  
    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(),
                                     rtc::Thread::Current()));
      ASSERT_TRUE(stream->AddTrack(video_track));
    }
  
    std::unique_ptr<SessionDescriptionInterface> CreateOfferWithOneAudioTrack() {
      CreatePeerConnectionWithoutDtls();
      AddAudioTrack(kAudioTracks[0]);
      std::unique_ptr<SessionDescriptionInterface> offer;
      EXPECT_TRUE(DoCreateOffer(&offer, nullptr));
      return offer;
    }
  
    std::unique_ptr<SessionDescriptionInterface> CreateOfferWithOneAudioStream() {
      CreatePeerConnectionWithoutDtls();
      AddAudioStream(kStreamId1);
      std::unique_ptr<SessionDescriptionInterface> offer;
      EXPECT_TRUE(DoCreateOffer(&offer, nullptr));
      return offer;
    }
  
    std::unique_ptr<SessionDescriptionInterface> CreateAnswerWithOneAudioTrack() {
      EXPECT_TRUE(DoSetRemoteDescription(CreateOfferWithOneAudioTrack()));
      std::unique_ptr<SessionDescriptionInterface> answer;
      EXPECT_TRUE(DoCreateAnswer(&answer, nullptr));
      return answer;
    }
  
    std::unique_ptr<SessionDescriptionInterface>
    CreateAnswerWithOneAudioStream() {
      EXPECT_TRUE(DoSetRemoteDescription(CreateOfferWithOneAudioStream()));
      std::unique_ptr<SessionDescriptionInterface> answer;
      EXPECT_TRUE(DoCreateAnswer(&answer, nullptr));
      return answer;
    }
  
    const std::string& GetFirstAudioStreamCname(
        const SessionDescriptionInterface* desc) {
      const cricket::AudioContentDescription* audio_desc =
          cricket::GetFirstAudioContentDescription(desc->description());
      return audio_desc->streams()[0].cname;
    }
  
    std::unique_ptr<SessionDescriptionInterface> CreateOfferWithOptions(
        const RTCOfferAnswerOptions& offer_answer_options) {
      RTC_DCHECK(pc_);
      rtc::scoped_refptr<MockCreateSessionDescriptionObserver> observer(
          new rtc::RefCountedObject<MockCreateSessionDescriptionObserver>());
      pc_->CreateOffer(observer, offer_answer_options);
      EXPECT_EQ_WAIT(true, observer->called(), kTimeout);
      return observer->MoveDescription();
    }
  
    void CreateOfferWithOptionsAsRemoteDescription(
        std::unique_ptr<SessionDescriptionInterface>* desc,
        const RTCOfferAnswerOptions& offer_answer_options) {
      *desc = CreateOfferWithOptions(offer_answer_options);
      ASSERT_TRUE(desc != nullptr);
      std::string sdp;
      EXPECT_TRUE((*desc)->ToString(&sdp));
      std::unique_ptr<SessionDescriptionInterface> remote_offer(
          webrtc::CreateSessionDescription(SdpType::kOffer, sdp));
      EXPECT_TRUE(DoSetRemoteDescription(std::move(remote_offer)));
      EXPECT_EQ(PeerConnectionInterface::kHaveRemoteOffer, observer_.state_);
    }
  
    void CreateOfferWithOptionsAsLocalDescription(
        std::unique_ptr<SessionDescriptionInterface>* desc,
        const RTCOfferAnswerOptions& offer_answer_options) {
      *desc = CreateOfferWithOptions(offer_answer_options);
      ASSERT_TRUE(desc != nullptr);
      std::string sdp;
      EXPECT_TRUE((*desc)->ToString(&sdp));
      std::unique_ptr<SessionDescriptionInterface> new_offer(
          webrtc::CreateSessionDescription(SdpType::kOffer, sdp));
  
      EXPECT_TRUE(DoSetLocalDescription(std::move(new_offer)));
      EXPECT_EQ(PeerConnectionInterface::kHaveLocalOffer, observer_.state_);
    }
  
    bool HasCNCodecs(const cricket::ContentInfo* content) {
      RTC_DCHECK(content);
      RTC_DCHECK(content->media_description());
      for (const cricket::AudioCodec& codec :
           content->media_description()->as_audio()->codecs()) {
        if (codec.name == "CN") {
          return true;
        }
      }
      return false;
    }
  
    const char* GetSdpStringWithStream1() const {
      if (sdp_semantics_ == SdpSemantics::kPlanB) {
        return kSdpStringWithStream1PlanB;
      } else {
        return kSdpStringWithStream1UnifiedPlan;
      }
    }
  
    const char* GetSdpStringWithStream1And2() const {
      if (sdp_semantics_ == SdpSemantics::kPlanB) {
        return kSdpStringWithStream1And2PlanB;
      } else {
        return kSdpStringWithStream1And2UnifiedPlan;
      }
    }
  
    std::unique_ptr<rtc::VirtualSocketServer> vss_;
    rtc::AutoSocketServerThread main_;
    rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_;
    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_;
    const SdpSemantics sdp_semantics_;
  };
  
  class PeerConnectionInterfaceTest
      : public PeerConnectionInterfaceBaseTest,
        public ::testing::WithParamInterface<SdpSemantics> {
   protected:
    PeerConnectionInterfaceTest() : PeerConnectionInterfaceBaseTest(GetParam()) {}
  };
  
  class PeerConnectionInterfaceTestPlanB
      : public PeerConnectionInterfaceBaseTest {
   protected:
    PeerConnectionInterfaceTestPlanB()
        : PeerConnectionInterfaceBaseTest(SdpSemantics::kPlanB) {}
  };
  
  // 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_P(PeerConnectionInterfaceTest, CnameGenerationInOffer) {
    std::unique_ptr<SessionDescriptionInterface> offer1 =
        CreateOfferWithOneAudioTrack();
    std::unique_ptr<SessionDescriptionInterface> offer2 =
        CreateOfferWithOneAudioTrack();
    EXPECT_NE(GetFirstAudioStreamCname(offer1.get()),
              GetFirstAudioStreamCname(offer2.get()));
  }
  
  TEST_P(PeerConnectionInterfaceTest, CnameGenerationInAnswer) {
    std::unique_ptr<SessionDescriptionInterface> answer1 =
        CreateAnswerWithOneAudioTrack();
    std::unique_ptr<SessionDescriptionInterface> answer2 =
        CreateAnswerWithOneAudioTrack();
    EXPECT_NE(GetFirstAudioStreamCname(answer1.get()),
              GetFirstAudioStreamCname(answer2.get()));
  }
  
  TEST_P(PeerConnectionInterfaceTest,
         CreatePeerConnectionWithDifferentConfigurations) {
    CreatePeerConnectionWithDifferentConfigurations();
  }
  
  TEST_P(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_P(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);
  
    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 network-related RTCConfiguration members are applied to the
  // PortAllocator when CreatePeerConnection is called. Specifically:
  // - disable_ipv6_on_wifi
  // - max_ipv6_networks
  // - tcp_candidate_policy
  // - candidate_network_policy
  // - prune_turn_ports
  //
  // Note that the candidate filter (RTCConfiguration::type) is already tested
  // above.
  TEST_P(PeerConnectionInterfaceTest,
         CreatePeerConnectionAppliesNetworkConfigToPortAllocator) {
    // Create fake port allocator.
    std::unique_ptr<cricket::FakePortAllocator> port_allocator(
        new cricket::FakePortAllocator(rtc::Thread::Current(), nullptr));
    cricket::FakePortAllocator* raw_port_allocator = port_allocator.get();
  
    // Create RTCConfiguration with some network-related fields relevant to
    // PortAllocator populated.
    PeerConnectionInterface::RTCConfiguration config;
    config.disable_ipv6_on_wifi = true;
    config.max_ipv6_networks = 10;
    config.tcp_candidate_policy =
        PeerConnectionInterface::kTcpCandidatePolicyDisabled;
    config.candidate_network_policy =
        PeerConnectionInterface::kCandidateNetworkPolicyLowCost;
    config.prune_turn_ports = true;
  
    // Create the PC factory and PC with the above config.
    rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> pc_factory(
        webrtc::CreatePeerConnectionFactory(
            rtc::Thread::Current(), rtc::Thread::Current(),
            rtc::Thread::Current(), fake_audio_capture_module_,
            webrtc::CreateBuiltinAudioEncoderFactory(),
            webrtc::CreateBuiltinAudioDecoderFactory(),
            webrtc::CreateBuiltinVideoEncoderFactory(),
            webrtc::CreateBuiltinVideoDecoderFactory(), nullptr /* audio_mixer */,
            nullptr /* audio_processing */));
    rtc::scoped_refptr<PeerConnectionInterface> pc(
        pc_factory->CreatePeerConnection(config, std::move(port_allocator),
                                         nullptr, &observer_));
    EXPECT_TRUE(pc.get());
    observer_.SetPeerConnectionInterface(pc.get());
  
    // Now validate that the config fields set above were applied to the
    // PortAllocator, as flags or otherwise.
    EXPECT_FALSE(raw_port_allocator->flags() &
                 cricket::PORTALLOCATOR_ENABLE_IPV6_ON_WIFI);
    EXPECT_EQ(10, raw_port_allocator->max_ipv6_networks());
    EXPECT_TRUE(raw_port_allocator->flags() & cricket::PORTALLOCATOR_DISABLE_TCP);
    EXPECT_TRUE(raw_port_allocator->flags() &
                cricket::PORTALLOCATOR_DISABLE_COSTLY_NETWORKS);
    EXPECT_EQ(webrtc::PRUNE_BASED_ON_PRIORITY,
              raw_port_allocator->turn_port_prune_policy());
  }
  
  // Check that GetConfiguration returns the configuration the PeerConnection was
  // constructed with, before SetConfiguration is called.
  TEST_P(PeerConnectionInterfaceTest, GetConfigurationAfterCreatePeerConnection) {
    PeerConnectionInterface::RTCConfiguration config;
    config.type = PeerConnectionInterface::kRelay;
    CreatePeerConnection(config);
  
    PeerConnectionInterface::RTCConfiguration returned_config =
        pc_->GetConfiguration();
    EXPECT_EQ(PeerConnectionInterface::kRelay, returned_config.type);
  }
  
  // Check that GetConfiguration returns the last configuration passed into
  // SetConfiguration.
  TEST_P(PeerConnectionInterfaceTest, GetConfigurationAfterSetConfiguration) {
    PeerConnectionInterface::RTCConfiguration starting_config;
    starting_config.bundle_policy =
        webrtc::PeerConnection::kBundlePolicyMaxBundle;
    CreatePeerConnection(starting_config);
  
    PeerConnectionInterface::RTCConfiguration config = pc_->GetConfiguration();
    config.type = PeerConnectionInterface::kRelay;
    EXPECT_TRUE(pc_->SetConfiguration(config).ok());
  
    PeerConnectionInterface::RTCConfiguration returned_config =
        pc_->GetConfiguration();
    EXPECT_EQ(PeerConnectionInterface::kRelay, returned_config.type);
  }
  
  TEST_P(PeerConnectionInterfaceTest, SetConfigurationFailsAfterClose) {
    CreatePeerConnection();
  
    pc_->Close();
  
    EXPECT_FALSE(
        pc_->SetConfiguration(PeerConnectionInterface::RTCConfiguration()).ok());
  }
  
  TEST_F(PeerConnectionInterfaceTestPlanB, AddStreams) {
    CreatePeerConnectionWithoutDtls();
    AddVideoStream(kStreamId1);
    AddAudioStream(kStreamId2);
    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(kStreamId3));
    rtc::scoped_refptr<AudioTrackInterface> audio_track(
        pc_factory_->CreateAudioTrack(kStreamId3,
                                      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.
  // Don't run under Unified Plan since the stream API is not available.
  TEST_F(PeerConnectionInterfaceTestPlanB, AddedStreamsPresentInOffer) {
    CreatePeerConnectionWithoutDtls();
    AddAudioVideoStream(kStreamId1, "audio_track", "video_track");
    std::unique_ptr<SessionDescriptionInterface> offer;
    ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
  
    const cricket::AudioContentDescription* audio_desc =
        cricket::GetFirstAudioContentDescription(offer->description());
    EXPECT_TRUE(ContainsTrack(audio_desc->streams(), kStreamId1, "audio_track"));
  
    const cricket::VideoContentDescription* video_desc =
        cricket::GetFirstVideoContentDescription(offer->description());
    EXPECT_TRUE(ContainsTrack(video_desc->streams(), kStreamId1, "video_track"));
  
    // Add another stream and ensure the offer includes both the old and new
    // streams.
    AddAudioVideoStream(kStreamId2, "audio_track2", "video_track2");
    ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
  
    audio_desc = cricket::GetFirstAudioContentDescription(offer->description());
    EXPECT_TRUE(ContainsTrack(audio_desc->streams(), kStreamId1, "audio_track"));
    EXPECT_TRUE(ContainsTrack(audio_desc->streams(), kStreamId2, "audio_track2"));
  
    video_desc = cricket::GetFirstVideoContentDescription(offer->description());
    EXPECT_TRUE(ContainsTrack(video_desc->streams(), kStreamId1, "video_track"));
    EXPECT_TRUE(ContainsTrack(video_desc->streams(), kStreamId2, "video_track2"));
  }
  
  // Don't run under Unified Plan since the stream API is not available.
  TEST_F(PeerConnectionInterfaceTestPlanB, RemoveStream) {
    CreatePeerConnectionWithoutDtls();
    AddVideoStream(kStreamId1);
    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.
  // Only tested with Plan B since Unified Plan is covered in more detail by tests
  // in peerconnection_jsep_unittests.cc
  TEST_F(PeerConnectionInterfaceTestPlanB, AddTrackRemoveTrack) {
    CreatePeerConnectionWithoutDtls();
    rtc::scoped_refptr<AudioTrackInterface> audio_track(
        CreateAudioTrack("audio_track"));
    rtc::scoped_refptr<VideoTrackInterface> video_track(
        CreateVideoTrack("video_track"));
    auto audio_sender = pc_->AddTrack(audio_track, {kStreamId1}).MoveValue();
    auto video_sender = pc_->AddTrack(video_track, {kStreamId1}).MoveValue();
    EXPECT_EQ(1UL, audio_sender->stream_ids().size());
    EXPECT_EQ(kStreamId1, 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(kStreamId1, 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());
    EXPECT_TRUE(ContainsTrack(audio_content->media_description()->streams(),
                              kStreamId1, "audio_track"));
  
    const cricket::ContentInfo* video_content =
        cricket::GetFirstVideoContent(offer->description());
    EXPECT_TRUE(ContainsTrack(video_content->media_description()->streams(),
                              kStreamId1, "video_track"));
  
    EXPECT_TRUE(DoSetLocalDescription(std::move(offer)));
  
    // 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());
    EXPECT_FALSE(ContainsTrack(audio_content->media_description()->streams(),
                               kStreamId1, "audio_track"));
  
    video_content = cricket::GetFirstVideoContent(offer->description());
    EXPECT_FALSE(ContainsTrack(video_content->media_description()->streams(),
                               kStreamId1, "video_track"));
  
    EXPECT_TRUE(DoSetLocalDescription(std::move(offer)));
  
    // 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_P(PeerConnectionInterfaceTest, AddTrackWithoutStream) {
    CreatePeerConnectionWithoutDtls();
    rtc::scoped_refptr<AudioTrackInterface> audio_track(
        CreateAudioTrack("audio_track"));
    rtc::scoped_refptr<VideoTrackInterface> video_track(
        CreateVideoTrack("video_track"));
    auto audio_sender =
        pc_->AddTrack(audio_track, std::vector<std::string>()).MoveValue();
    auto video_sender =
        pc_->AddTrack(video_track, std::vector<std::string>()).MoveValue();
    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 (sdp_semantics_ == SdpSemantics::kPlanB) {
      // 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());
    } else {
      // We allows creating tracks without stream ids under Unified Plan
      // semantics.
      EXPECT_EQ(0u, video_sender->stream_ids().size());
      EXPECT_EQ(0u, audio_sender->stream_ids().size());
    }
  }
  
  // Test that we can call GetStats() after AddTrack but before connecting
  // the PeerConnection to a peer.
  TEST_P(PeerConnectionInterfaceTest, AddTrackBeforeConnecting) {
    CreatePeerConnectionWithoutDtls();
    rtc::scoped_refptr<AudioTrackInterface> audio_track(
        CreateAudioTrack("audio_track"));
    rtc::scoped_refptr<VideoTrackInterface> video_track(
        CreateVideoTrack("video_track"));
    auto audio_sender = pc_->AddTrack(audio_track, std::vector<std::string>());
    auto video_sender = pc_->AddTrack(video_track, std::vector<std::string>());
    EXPECT_TRUE(DoGetStats(nullptr));
  }
  
  TEST_P(PeerConnectionInterfaceTest, AttachmentIdIsSetOnAddTrack) {
    CreatePeerConnectionWithoutDtls();
    rtc::scoped_refptr<AudioTrackInterface> audio_track(
        CreateAudioTrack("audio_track"));
    rtc::scoped_refptr<VideoTrackInterface> video_track(
        CreateVideoTrack("video_track"));
    auto audio_sender = pc_->AddTrack(audio_track, std::vector<std::string>());
    ASSERT_TRUE(audio_sender.ok());
    auto* audio_sender_proxy =
        static_cast<RtpSenderProxyWithInternal<RtpSenderInternal>*>(
            audio_sender.value().get());
    EXPECT_NE(0, audio_sender_proxy->internal()->AttachmentId());
  
    auto video_sender = pc_->AddTrack(video_track, std::vector<std::string>());
    ASSERT_TRUE(video_sender.ok());
    auto* video_sender_proxy =
        static_cast<RtpSenderProxyWithInternal<RtpSenderInternal>*>(
            video_sender.value().get());
    EXPECT_NE(0, video_sender_proxy->internal()->AttachmentId());
  }
  
  // Don't run under Unified Plan since the stream API is not available.
  TEST_F(PeerConnectionInterfaceTestPlanB, AttachmentIdIsSetOnAddStream) {
    CreatePeerConnectionWithoutDtls();
    AddVideoStream(kStreamId1);
    auto senders = pc_->GetSenders();
    ASSERT_EQ(1u, senders.size());
    auto* sender_proxy =
        static_cast<RtpSenderProxyWithInternal<RtpSenderInternal>*>(
            senders[0].get());
    EXPECT_NE(0, sender_proxy->internal()->AttachmentId());
  }
  
  TEST_P(PeerConnectionInterfaceTest, CreateOfferReceiveAnswer) {
    InitiateCall();
    WaitAndVerifyOnAddStream(kStreamId1, 2);
    VerifyRemoteRtpHeaderExtensions();
  }
  
  TEST_P(PeerConnectionInterfaceTest, CreateOfferReceivePrAnswerAndAnswer) {
    CreatePeerConnectionWithoutDtls();
    AddVideoTrack(kVideoTracks[0], {kStreamId1});
    CreateOfferAsLocalDescription();
    std::string offer;
    EXPECT_TRUE(pc_->local_description()->ToString(&offer));
    CreatePrAnswerAndAnswerAsRemoteDescription(offer);
    WaitAndVerifyOnAddStream(kStreamId1, 1);
  }
  
  TEST_P(PeerConnectionInterfaceTest, ReceiveOfferCreateAnswer) {
    CreatePeerConnectionWithoutDtls();
    AddVideoTrack(kVideoTracks[0], {kStreamId1});
  
    CreateOfferAsRemoteDescription();
    CreateAnswerAsLocalDescription();
  
    WaitAndVerifyOnAddStream(kStreamId1, 1);
  }
  
  TEST_P(PeerConnectionInterfaceTest, ReceiveOfferCreatePrAnswerAndAnswer) {
    CreatePeerConnectionWithoutDtls();
    AddVideoTrack(kVideoTracks[0], {kStreamId1});
  
    CreateOfferAsRemoteDescription();
    CreatePrAnswerAsLocalDescription();
    CreateAnswerAsLocalDescription();
  
    WaitAndVerifyOnAddStream(kStreamId1, 1);
  }
  
  // Don't run under Unified Plan since the stream API is not available.
  TEST_F(PeerConnectionInterfaceTestPlanB, 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(kStreamId1);
    CreateOfferReceiveAnswer();
  }
  
  // Tests that after negotiating an audio only call, the respondent can perform a
  // renegotiation that removes the audio stream.
  TEST_F(PeerConnectionInterfaceTestPlanB, RenegotiateAudioOnly) {
    CreatePeerConnectionWithoutDtls();
    AddAudioStream(kStreamId1);
    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_P(PeerConnectionInterfaceTest, IceCandidates) {
    CreatePeerConnectionWithoutDtls();
  
    EXPECT_FALSE(pc_->AddIceCandidate(observer_.last_candidate()));
    // SetRemoteDescription takes ownership of offer.
    std::unique_ptr<SessionDescriptionInterface> offer;
    AddVideoTrack(kVideoTracks[0]);
    EXPECT_TRUE(DoCreateOffer(&offer, nullptr));
    EXPECT_TRUE(DoSetRemoteDescription(std::move(offer)));
  
    // SetLocalDescription takes ownership of answer.
    std::unique_ptr<SessionDescriptionInterface> answer;
    EXPECT_TRUE(DoCreateAnswer(&answer, nullptr));
    EXPECT_TRUE(DoSetLocalDescription(std::move(answer)));
  
    EXPECT_TRUE_WAIT(observer_.last_candidate() != nullptr, kTimeout);
    EXPECT_TRUE_WAIT(observer_.ice_gathering_complete_, kTimeout);
  
    EXPECT_TRUE(pc_->AddIceCandidate(observer_.last_candidate()));
  }
  
  // Test that CreateOffer and CreateAnswer will fail if the track labels are
  // not unique.
  TEST_F(PeerConnectionInterfaceTestPlanB, 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.
    AddAudioTrack("track_label", {kStreamId1});
    AddVideoTrack("track_label", {kStreamId1});
  
    // 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_P(PeerConnectionInterfaceTest, SsrcInOfferAnswer) {
    CreatePeerConnectionWithoutDtls();
    // Create a local stream with audio&video tracks having different labels.
    AddAudioTrack(kAudioTracks[0], {kStreamId1});
    AddVideoTrack(kVideoTracks[0], {kStreamId1});
  
    // 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(std::move(offer)));
    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.
  // Don't run under Unified Plan since the stream API is not available.
  TEST_F(PeerConnectionInterfaceTestPlanB, AddTrackAfterAddStream) {
    CreatePeerConnectionWithoutDtls();
    // Create audio stream and add to PeerConnection.
    AddAudioStream(kStreamId1);
    MediaStreamInterface* stream = pc_->local_streams()->at(0);
  
    // Add video track to the audio-only stream.
    rtc::scoped_refptr<VideoTrackInterface> video_track(
        CreateVideoTrack("video_label"));
    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.
  // Don't run under Unified Plan since the stream API is not available.
  TEST_F(PeerConnectionInterfaceTestPlanB, RemoveTrackAfterAddStream) {
    CreatePeerConnectionWithoutDtls();
    // Create audio/video stream and add to PeerConnection.
    AddAudioVideoStream(kStreamId1, "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.
  // Don't run under Unified Plan since the stream API is not available.
  TEST_F(PeerConnectionInterfaceTestPlanB, CreateSenderWithStream) {
    CreatePeerConnectionWithoutDtls();
    pc_->CreateSender("video", kStreamId1);
  
    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(kStreamId1, video_desc->streams()[0].first_stream_id());
  }
  
  // Test that we can specify a certain track that we want statistics about.
  TEST_P(PeerConnectionInterfaceTest, GetStatsForSpecificTrack) {
    InitiateCall();
    ASSERT_LT(0u, pc_->GetSenders().size());
    ASSERT_LT(0u, pc_->GetReceivers().size());
    rtc::scoped_refptr<MediaStreamTrackInterface> remote_audio =
        pc_->GetReceivers()[0]->track();
    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_->RemoveTrack(pc_->GetSenders()[0]);
    // Do a re-negotiation.
    CreateOfferReceiveAnswer();
  
    // 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_P(PeerConnectionInterfaceTest, GetStatsForVideoTrack) {
    InitiateCall();
    auto video_receiver = GetFirstReceiverOfType(cricket::MEDIA_TYPE_VIDEO);
    ASSERT_TRUE(video_receiver);
    EXPECT_TRUE(DoGetStats(video_receiver->track()));
  }
  
  // Test that we don't get statistics for an invalid track.
  TEST_P(PeerConnectionInterfaceTest, GetStatsForInvalidTrack) {
    InitiateCall();
    rtc::scoped_refptr<AudioTrackInterface> unknown_audio_track(
        pc_factory_->CreateAudioTrack("unknown track", NULL));
    EXPECT_FALSE(DoGetStats(unknown_audio_track));
  }
  
  TEST_P(PeerConnectionInterfaceTest, GetRTCStatsBeforeAndAfterCalling) {
    CreatePeerConnectionWithoutDtls();
    EXPECT_TRUE(DoGetRTCStats());
    // Clearing stats cache is needed now, but should be temporary.
    // https://bugs.chromium.org/p/webrtc/issues/detail?id=8693
    pc_->ClearStatsCache();
    AddAudioTrack(kAudioTracks[0], {kStreamId1});
    AddVideoTrack(kVideoTracks[0], {kStreamId1});
    EXPECT_TRUE(DoGetRTCStats());
    pc_->ClearStatsCache();
    CreateOfferReceiveAnswer();
    EXPECT_TRUE(DoGetRTCStats());
  }
  
  // This tests that a SCTP data channel is returned using different
  // DataChannelInit configurations.
  TEST_P(PeerConnectionInterfaceTest, CreateSctpDataChannel) {
    RTCConfiguration rtc_config;
    rtc_config.enable_dtls_srtp = true;
    CreatePeerConnection(rtc_config);
  
    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 = absl::nullopt;
    config.maxRetransmitTime = 0;
    channel = pc_->CreateDataChannel("4", &config);
    EXPECT_TRUE(channel != NULL);
    EXPECT_FALSE(channel->reliable());
    EXPECT_FALSE(observer_.renegotiation_needed_);
  }
  
  // For backwards compatibility, we want people who "unset" maxRetransmits
  // and maxRetransmitTime by setting them to -1 to get what they want.
  TEST_P(PeerConnectionInterfaceTest, CreateSctpDataChannelWithMinusOne) {
    RTCConfiguration rtc_config;
    rtc_config.enable_dtls_srtp = true;
    CreatePeerConnection(rtc_config);
  
    webrtc::DataChannelInit config;
    config.maxRetransmitTime = -1;
    config.maxRetransmits = -1;
    rtc::scoped_refptr<DataChannelInterface> channel =
        pc_->CreateDataChannel("1", &config);
    EXPECT_TRUE(channel != NULL);
  }
  
  // This tests that no data channel is returned if both maxRetransmits and
  // maxRetransmitTime are set for SCTP data channels.
  TEST_P(PeerConnectionInterfaceTest,
         CreateSctpDataChannelShouldFailForInvalidConfig) {
    RTCConfiguration rtc_config;
    rtc_config.enable_dtls_srtp = true;
    CreatePeerConnection(rtc_config);
  
    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_P(PeerConnectionInterfaceTest,
         CreateSctpDataChannelWithInvalidIdShouldFail) {
    RTCConfiguration rtc_config;
    rtc_config.enable_dtls_srtp = true;
    CreatePeerConnection(rtc_config);
  
    webrtc::DataChannelInit config;
    rtc::scoped_refptr<DataChannelInterface> channel;
  
    config.id = 1;
    config.negotiated = true;
    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;
    config.negotiated = true;
    channel = pc_->CreateDataChannel("max", &config);
    EXPECT_TRUE(channel != NULL);
    EXPECT_EQ(config.id, channel->id());
  
    config.id = cricket::kMaxSctpSid + 1;
    config.negotiated = true;
    channel = pc_->CreateDataChannel("x", &config);
    EXPECT_TRUE(channel == NULL);
  }
  
  // Verifies that duplicated label is allowed for SCTP data channel.
  TEST_P(PeerConnectionInterfaceTest, SctpDuplicatedLabelAllowed) {
    RTCConfiguration rtc_config;
    rtc_config.enable_dtls_srtp = true;
    CreatePeerConnection(rtc_config);
  
    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);
  }
  
  
  #ifdef WEBRTC_HAVE_SCTP
  // This tests that SCTP data channels can be rejected in an answer.
  TEST_P(PeerConnectionInterfaceTest, TestRejectSctpDataChannelInAnswer)
  #else
  TEST_P(PeerConnectionInterfaceTest, DISABLED_TestRejectSctpDataChannelInAnswer)
  #endif
  {
    RTCConfiguration rtc_config;
    CreatePeerConnection(rtc_config);
  
    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));
    std::unique_ptr<SessionDescriptionInterface> answer(
        webrtc::CreateSessionDescription(SdpType::kAnswer, sdp));
    ASSERT_TRUE(answer);
    cricket::ContentInfo* data_info =
        cricket::GetFirstDataContent(answer->description());
    data_info->rejected = true;
  
    DoSetRemoteDescription(std::move(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_P(PeerConnectionInterfaceTest, ReceiveFireFoxOffer) {
    RTCConfiguration rtc_config;
    rtc_config.enable_dtls_srtp = true;
    CreatePeerConnection(rtc_config);
    AddAudioTrack("audio_label");
    AddVideoTrack("video_label");
    std::unique_ptr<SessionDescriptionInterface> desc(
        webrtc::CreateSessionDescription(SdpType::kOffer,
                                         webrtc::kFireFoxSdpOffer, nullptr));
    EXPECT_TRUE(DoSetSessionDescription(std::move(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 WEBRTC_HAVE_SCTP
    content =
        cricket::GetFirstDataContent(pc_->local_description()->description());
    ASSERT_TRUE(content != NULL);
    EXPECT_FALSE(content->rejected);
  #endif
  }
  
  // Test that fallback from DTLS to SDES is not supported.
  // The fallback was previously supported but was removed to simplify the code
  // and because it's non-standard.
  TEST_P(PeerConnectionInterfaceTest, DtlsSdesFallbackNotSupported) {
    RTCConfiguration rtc_config;
    rtc_config.enable_dtls_srtp = true;
    CreatePeerConnection(rtc_config);
    // Wait for fake certificate to be generated. Previously, this is what caused
    // the "a=crypto" lines to be rejected.
    AddAudioTrack("audio_label");
    AddVideoTrack("video_label");
    ASSERT_NE(nullptr, fake_certificate_generator_);
    EXPECT_EQ_WAIT(1, fake_certificate_generator_->generated_certificates(),
                   kTimeout);
    std::unique_ptr<SessionDescriptionInterface> desc(
        webrtc::CreateSessionDescription(SdpType::kOffer, kDtlsSdesFallbackSdp,
                                         nullptr));
    EXPECT_FALSE(DoSetSessionDescription(std::move(desc), /*local=*/false));
  }
  
  // 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_P(PeerConnectionInterfaceTest, ReceiveUpdatedAudioOfferWithBadCodecs) {
    CreatePeerConnectionWithoutDtls();
    AddAudioTrack("audio_label");
    CreateOfferAsLocalDescription();
  
    const char* answer_sdp =
        (sdp_semantics_ == SdpSemantics::kPlanB ? webrtc::kAudioSdpPlanB
                                                : webrtc::kAudioSdpUnifiedPlan);
    std::unique_ptr<SessionDescriptionInterface> answer(
        webrtc::CreateSessionDescription(SdpType::kAnswer, answer_sdp, nullptr));
    EXPECT_TRUE(DoSetSessionDescription(std::move(answer), false));
  
    const char* reoffer_sdp =
        (sdp_semantics_ == SdpSemantics::kPlanB
             ? webrtc::kAudioSdpWithUnsupportedCodecsPlanB
             : webrtc::kAudioSdpWithUnsupportedCodecsUnifiedPlan);
    std::unique_ptr<SessionDescriptionInterface> updated_offer(
        webrtc::CreateSessionDescription(SdpType::kOffer, reoffer_sdp, nullptr));
    EXPECT_TRUE(DoSetSessionDescription(std::move(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_P(PeerConnectionInterfaceTest, CreateSubsequentRecvOnlyOffer) {
    RTCConfiguration rtc_config;
    rtc_config.enable_dtls_srtp = true;
    CreatePeerConnection(rtc_config);
    CreateAndSetRemoteOffer(GetSdpStringWithStream1());
    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());
    ASSERT_EQ(RtpTransceiverDirection::kRecvOnly,
              video_content->media_description()->direction());
  
    const cricket::ContentInfo* audio_content =
        cricket::GetFirstAudioContent(offer->description());
    ASSERT_EQ(RtpTransceiverDirection::kRecvOnly,
              audio_content->media_description()->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_P(PeerConnectionInterfaceTest, CreateSubsequentInactiveOffer) {
    RTCConfiguration rtc_config;
    rtc_config.enable_dtls_srtp = true;
    CreatePeerConnection(rtc_config);
    CreateAndSetRemoteOffer(GetSdpStringWithStream1());
    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;
    RTCOfferAnswerOptions options;
    options.offer_to_receive_audio = 0;
    options.offer_to_receive_video = 0;
    DoCreateOffer(&offer, &options);
  
    const cricket::ContentInfo* video_content =
        cricket::GetFirstVideoContent(offer->description());
    ASSERT_EQ(RtpTransceiverDirection::kInactive,
              video_content->media_description()->direction());
  
    const cricket::ContentInfo* audio_content =
        cricket::GetFirstAudioContent(offer->description());
    ASSERT_EQ(RtpTransceiverDirection::kInactive,
              audio_content->media_description()->direction());
  }
  
  // Test that we can use SetConfiguration to change the ICE servers of the
  // PortAllocator.
  TEST_P(PeerConnectionInterfaceTest, SetConfigurationChangesIceServers) {
    CreatePeerConnection();
  
    PeerConnectionInterface::RTCConfiguration config = pc_->GetConfiguration();
    PeerConnectionInterface::IceServer server;
    server.uri = "stun:test_hostname";
    config.servers.push_back(server);
    EXPECT_TRUE(pc_->SetConfiguration(config).ok());
  
    EXPECT_EQ(1u, port_allocator_->stun_servers().size());
    EXPECT_EQ("test_hostname",
              port_allocator_->stun_servers().begin()->hostname());
  }
  
  TEST_P(PeerConnectionInterfaceTest, SetConfigurationChangesCandidateFilter) {
    CreatePeerConnection();
    PeerConnectionInterface::RTCConfiguration config = pc_->GetConfiguration();
    config.type = PeerConnectionInterface::kRelay;
    EXPECT_TRUE(pc_->SetConfiguration(config).ok());
    EXPECT_EQ(cricket::CF_RELAY, port_allocator_->candidate_filter());
  }
  
  TEST_P(PeerConnectionInterfaceTest, SetConfigurationChangesPruneTurnPortsFlag) {
    PeerConnectionInterface::RTCConfiguration config;
    config.prune_turn_ports = false;
    CreatePeerConnection(config);
    config = pc_->GetConfiguration();
    EXPECT_EQ(webrtc::NO_PRUNE, port_allocator_->turn_port_prune_policy());
  
    config.prune_turn_ports = true;
    EXPECT_TRUE(pc_->SetConfiguration(config).ok());
    EXPECT_EQ(webrtc::PRUNE_BASED_ON_PRIORITY,
              port_allocator_->turn_port_prune_policy());
  }
  
  // Test that the ice check interval can be changed. This does not verify that
  // the setting makes it all the way to P2PTransportChannel, as that would
  // require a very complex set of mocks.
  TEST_P(PeerConnectionInterfaceTest, SetConfigurationChangesIceCheckInterval) {
    PeerConnectionInterface::RTCConfiguration config;
    config.ice_check_min_interval = absl::nullopt;
    CreatePeerConnection(config);
    config = pc_->GetConfiguration();
    config.ice_check_min_interval = 100;
    EXPECT_TRUE(pc_->SetConfiguration(config).ok());
    config = pc_->GetConfiguration();
    EXPECT_EQ(config.ice_check_min_interval, 100);
  }
  
  TEST_P(PeerConnectionInterfaceTest,
         SetConfigurationChangesSurfaceIceCandidatesOnIceTransportTypeChanged) {
    PeerConnectionInterface::RTCConfiguration config;
    config.surface_ice_candidates_on_ice_transport_type_changed = false;
    CreatePeerConnection(config);
    config = pc_->GetConfiguration();
    EXPECT_FALSE(config.surface_ice_candidates_on_ice_transport_type_changed);
  
    config.surface_ice_candidates_on_ice_transport_type_changed = true;
    EXPECT_TRUE(pc_->SetConfiguration(config).ok());
    config = pc_->GetConfiguration();
    EXPECT_TRUE(config.surface_ice_candidates_on_ice_transport_type_changed);
  }
  
  // Test that when SetConfiguration changes both the pool size and other
  // attributes, the pooled session is created with the updated attributes.
  TEST_P(PeerConnectionInterfaceTest,
         SetConfigurationCreatesPooledSessionCorrectly) {
    CreatePeerConnection();
    PeerConnectionInterface::RTCConfiguration config = pc_->GetConfiguration();
    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).ok());
  
    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_P(PeerConnectionInterfaceTest,
         CantChangePoolSizeAfterSetLocalDescription) {
    CreatePeerConnection();
    // Start by setting a size of 1.
    PeerConnectionInterface::RTCConfiguration config = pc_->GetConfiguration();
    config.ice_candidate_pool_size = 1;
    EXPECT_TRUE(pc_->SetConfiguration(config).ok());
  
    // Set remote offer; can still change pool size at this point.
    CreateOfferAsRemoteDescription();
    config.ice_candidate_pool_size = 2;
    EXPECT_TRUE(pc_->SetConfiguration(config).ok());
  
    // Set local answer; now it's too late.
    CreateAnswerAsLocalDescription();
    config.ice_candidate_pool_size = 3;
    RTCError error = pc_->SetConfiguration(config);
    EXPECT_EQ(RTCErrorType::INVALID_MODIFICATION, error.type());
  }
  
  // Test that after setting an answer, extra pooled sessions are discarded. The
  // ICE candidate pool is only intended to be used for the first offer/answer.
  TEST_P(PeerConnectionInterfaceTest,
         ExtraPooledSessionsDiscardedAfterApplyingAnswer) {
    CreatePeerConnection();
  
    // Set a larger-than-necessary size.
    PeerConnectionInterface::RTCConfiguration config = pc_->GetConfiguration();
    config.ice_candidate_pool_size = 4;
    EXPECT_TRUE(pc_->SetConfiguration(config).ok());
  
    // Do offer/answer.
    CreateOfferAsRemoteDescription();
    CreateAnswerAsLocalDescription();
  
    // Expect no pooled sessions to be left.
    const cricket::PortAllocatorSession* session =
        port_allocator_->GetPooledSession();
    EXPECT_EQ(nullptr, session);
  }
  
  // After Close is called, pooled candidates should be discarded so as to not
  // waste network resources.
  TEST_P(PeerConnectionInterfaceTest, PooledSessionsDiscardedAfterClose) {
    CreatePeerConnection();
  
    PeerConnectionInterface::RTCConfiguration config = pc_->GetConfiguration();
    config.ice_candidate_pool_size = 3;
    EXPECT_TRUE(pc_->SetConfiguration(config).ok());
    pc_->Close();
  
    // Expect no pooled sessions to be left.
    const cricket::PortAllocatorSession* session =
        port_allocator_->GetPooledSession();
    EXPECT_EQ(nullptr, session);
  }
  
  // Test that SetConfiguration returns an invalid modification error if
  // modifying a field in the configuration that isn't allowed to be modified.
  TEST_P(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);
  
    PeerConnectionInterface::RTCConfiguration modified_config =
        pc_->GetConfiguration();
    modified_config.bundle_policy =
        PeerConnectionInterface::kBundlePolicyMaxBundle;
    RTCError error = pc_->SetConfiguration(modified_config);
    EXPECT_EQ(RTCErrorType::INVALID_MODIFICATION, error.type());
  
    modified_config = pc_->GetConfiguration();
    modified_config.rtcp_mux_policy =
        PeerConnectionInterface::kRtcpMuxPolicyRequire;
    error = pc_->SetConfiguration(modified_config);
    EXPECT_EQ(RTCErrorType::INVALID_MODIFICATION, error.type());
  
    modified_config = pc_->GetConfiguration();
    modified_config.continual_gathering_policy =
        PeerConnectionInterface::GATHER_CONTINUALLY;
    error = pc_->SetConfiguration(modified_config);
    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_P(PeerConnectionInterfaceTest,
         SetConfigurationReturnsRangeErrorForBadCandidatePoolSize) {
    PeerConnectionInterface::RTCConfiguration config;
    CreatePeerConnection(config);
    config = pc_->GetConfiguration();
  
    config.ice_candidate_pool_size = -1;
    RTCError error = pc_->SetConfiguration(config);
    EXPECT_EQ(RTCErrorType::INVALID_RANGE, error.type());
  
    config.ice_candidate_pool_size = INT_MAX;
    error = pc_->SetConfiguration(config);
    EXPECT_EQ(RTCErrorType::INVALID_RANGE, error.type());
  }
  
  // Test that SetConfiguration returns a syntax error if parsing an ICE server
  // URL failed.
  TEST_P(PeerConnectionInterfaceTest,
         SetConfigurationReturnsSyntaxErrorFromBadIceUrls) {
    PeerConnectionInterface::RTCConfiguration config;
    CreatePeerConnection(config);
    config = pc_->GetConfiguration();
  
    PeerConnectionInterface::IceServer bad_server;
    bad_server.uri = "stunn:www.example.com";
    config.servers.push_back(bad_server);
    RTCError error = pc_->SetConfiguration(config);
    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_P(PeerConnectionInterfaceTest,
         SetConfigurationReturnsInvalidParameterIfCredentialsMissing) {
    PeerConnectionInterface::RTCConfiguration config;
    CreatePeerConnection(config);
    config = pc_->GetConfiguration();
  
    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_EQ(pc_->SetConfiguration(config).type(),
              RTCErrorType::INVALID_PARAMETER);
  }
  
  // Test that PeerConnection::Close changes the states to closed and all remote
  // tracks change state to ended.
  TEST_P(PeerConnectionInterfaceTest, CloseAndTestStreamsAndStates) {
    // Initialize a PeerConnection and negotiate local and remote session
    // description.
    InitiateCall();
  
    // With Plan B, verify the stream count. The analog with Unified Plan is the
    // RtpTransceiver count.
    if (sdp_semantics_ == SdpSemantics::kPlanB) {
      ASSERT_EQ(1u, pc_->local_streams()->count());
      ASSERT_EQ(1u, pc_->remote_streams()->count());
    } else {
      ASSERT_EQ(2u, pc_->GetTransceivers().size());
    }
  
    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());
  
    if (sdp_semantics_ == SdpSemantics::kPlanB) {
      EXPECT_EQ(1u, pc_->local_streams()->count());
      EXPECT_EQ(1u, pc_->remote_streams()->count());
    } else {
      // Verify that the RtpTransceivers are still returned.
      EXPECT_EQ(2u, pc_->GetTransceivers().size());
    }
  
    auto audio_receiver = GetFirstReceiverOfType(cricket::MEDIA_TYPE_AUDIO);
    auto video_receiver = GetFirstReceiverOfType(cricket::MEDIA_TYPE_VIDEO);
    if (sdp_semantics_ == SdpSemantics::kPlanB) {
      ASSERT_TRUE(audio_receiver);
      ASSERT_TRUE(video_receiver);
      // Track state may be updated asynchronously.
      EXPECT_EQ_WAIT(MediaStreamTrackInterface::kEnded,
                     audio_receiver->track()->state(), kTimeout);
      EXPECT_EQ_WAIT(MediaStreamTrackInterface::kEnded,
                     video_receiver->track()->state(), kTimeout);
    } else {
      ASSERT_FALSE(audio_receiver);
      ASSERT_FALSE(video_receiver);
    }
  }
  
  // Test that PeerConnection methods fails gracefully after
  // PeerConnection::Close has been called.
  // Don't run under Unified Plan since the stream API is not available.
  TEST_F(PeerConnectionInterfaceTestPlanB, CloseAndTestMethods) {
    CreatePeerConnectionWithoutDtls();
    AddAudioVideoStream(kStreamId1, "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));
  
    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_FALSE(DoCreateOffer(&offer, nullptr));
    std::unique_ptr<SessionDescriptionInterface> answer;
    EXPECT_FALSE(DoCreateAnswer(&answer, nullptr));
  
    std::string sdp;
    ASSERT_TRUE(pc_->remote_description()->ToString(&sdp));
    std::unique_ptr<SessionDescriptionInterface> remote_offer(
        webrtc::CreateSessionDescription(SdpType::kOffer, sdp));
    EXPECT_FALSE(DoSetRemoteDescription(std::move(remote_offer)));
  
    ASSERT_TRUE(pc_->local_description()->ToString(&sdp));
    std::unique_ptr<SessionDescriptionInterface> local_offer(
        webrtc::CreateSessionDescription(SdpType::kOffer, sdp));
    EXPECT_FALSE(DoSetLocalDescription(std::move(local_offer)));
  }
  
  // Test that GetStats can still be called after PeerConnection::Close.
  TEST_P(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_P(PeerConnectionInterfaceTest, UpdateRemoteStreams) {
    RTCConfiguration config;
    config.enable_dtls_srtp = true;
    CreatePeerConnection(config);
    CreateAndSetRemoteOffer(GetSdpStringWithStream1());
  
    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(GetSdpStringWithStream1And2());
  
    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.
  // Don't run under Unified Plan since this test uses Plan B SDP to test Plan B
  // specific behavior.
  TEST_F(PeerConnectionInterfaceTestPlanB,
         AddRemoveTrackFromExistingRemoteMediaStream) {
    RTCConfiguration config;
    config.enable_dtls_srtp = true;
    CreatePeerConnection(config);
    std::unique_ptr<SessionDescriptionInterface> desc_ms1 =
        CreateSessionDescriptionAndReference(1, 1);
    EXPECT_TRUE(DoSetRemoteDescription(std::move(desc_ms1)));
    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(std::move(desc_ms1_two_tracks)));
    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(std::move(desc_ms2)));
    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_P(PeerConnectionInterfaceTest, RejectMediaContent) {
    RTCConfiguration config;
    config.enable_dtls_srtp = true;
    CreatePeerConnection(config);
    // First create and set a remote offer, then reject its video content in our
    // answer.
    CreateAndSetRemoteOffer(kSdpStringWithStream1PlanB);
    auto audio_receiver = GetFirstReceiverOfType(cricket::MEDIA_TYPE_AUDIO);
    ASSERT_TRUE(audio_receiver);
    auto video_receiver = GetFirstReceiverOfType(cricket::MEDIA_TYPE_VIDEO);
    ASSERT_TRUE(video_receiver);
  
    rtc::scoped_refptr<MediaStreamTrackInterface> remote_audio =
        audio_receiver->track();
    EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, remote_audio->state());
    rtc::scoped_refptr<MediaStreamTrackInterface> remote_video =
        video_receiver->track();
    EXPECT_EQ(MediaStreamTrackInterface::kLive, remote_video->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(std::move(local_answer)));
    EXPECT_EQ(MediaStreamTrackInterface::kEnded, remote_video->state());
    EXPECT_EQ(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(std::move(local_offer)));
    // Track state may be updated asynchronously.
    EXPECT_EQ_WAIT(MediaStreamTrackInterface::kEnded, remote_audio->state(),
                   kTimeout);
    EXPECT_EQ_WAIT(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.
  // Don't run under Unified Plan since the stream API is not available.
  TEST_F(PeerConnectionInterfaceTestPlanB, RemoveTrackThenRejectMediaContent) {
    RTCConfiguration config;
    config.enable_dtls_srtp = true;
    CreatePeerConnection(config);
    CreateAndSetRemoteOffer(GetSdpStringWithStream1());
    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(SdpType::kAnswer,
                                         GetSdpStringWithStream1(), 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(std::move(local_answer)));
  
    // 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_P(PeerConnectionInterfaceTest, RecvonlyDescriptionDoesntCreateStream) {
    RTCConfiguration config;
    config.enable_dtls_srtp = true;
    CreatePeerConnection(config);
  
    std::string recvonly_offer = GetSdpStringWithStream1();
    absl::StrReplaceAll({{kSendrecv, 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.
  // Don't run under Unified Plan since this behavior is Plan B specific.
  TEST_F(PeerConnectionInterfaceTestPlanB, SdpWithoutMsidCreatesDefaultStream) {
    RTCConfiguration config;
    config.enable_dtls_srtp = true;
    CreatePeerConnection(config);
    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->id());
  
    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.
  // Don't run under Unified Plan since this behavior is Plan B specific.
  TEST_F(PeerConnectionInterfaceTestPlanB,
         SendOnlySdpWithoutMsidCreatesDefaultStream) {
    RTCConfiguration config;
    config.enable_dtls_srtp = true;
    CreatePeerConnection(config);
    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->id());
  }
  
  // This tests that it won't crash when PeerConnection tries to remove
  // a remote track that as already been removed from the MediaStream.
  // Don't run under Unified Plan since this behavior is Plan B specific.
  TEST_F(PeerConnectionInterfaceTestPlanB, RemoveAlreadyGoneRemoteStream) {
    RTCConfiguration config;
    config.enable_dtls_srtp = true;
    CreatePeerConnection(config);
    CreateAndSetRemoteOffer(GetSdpStringWithStream1());
    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.
  // Don't run under Unified Plan since this behavior is Plan B specific.
  TEST_F(PeerConnectionInterfaceTestPlanB,
         SdpWithoutMsidAndStreamsCreatesDefaultStream) {
    RTCConfiguration config;
    config.enable_dtls_srtp = true;
    CreatePeerConnection(config);
    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.
  // Don't run under Unified Plan since this behavior is Plan B specific.
  TEST_F(PeerConnectionInterfaceTestPlanB, SdpWithMsidDontCreatesDefaultStream) {
    RTCConfiguration config;
    config.enable_dtls_srtp = true;
    CreatePeerConnection(config);
    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
  // Don't run under Unified Plan since this behavior is Plan B specific.
  TEST_F(PeerConnectionInterfaceTestPlanB,
         DefaultTracksNotDestroyedAndRecreated) {
    RTCConfiguration config;
    config.enable_dtls_srtp = true;
    CreatePeerConnection(config);
    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.
  // Don't run under Unified Plan since this behavior is Plan B specific.
  TEST_F(PeerConnectionInterfaceTestPlanB, VerifyDefaultStreamIsNotCreated) {
    RTCConfiguration config;
    config.enable_dtls_srtp = true;
    CreatePeerConnection(config);
    CreateAndSetRemoteOffer(GetSdpStringWithStream1());
    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 a default MediaStream is created if a remote SDP comes from
  // an endpoint that doesn't signal SSRCs, but signals media stream IDs.
  TEST_F(PeerConnectionInterfaceTestPlanB,
         SdpWithMsidWithoutSsrcCreatesDefaultStream) {
    RTCConfiguration config;
    config.enable_dtls_srtp = true;
    CreatePeerConnection(config);
    std::string sdp_string = kSdpStringWithoutStreamsAudioOnly;
    // Add a=msid lines to simulate a Unified Plan endpoint that only
    // signals stream IDs with a=msid lines.
    sdp_string.append("a=msid:audio_stream_id audio_track_id\n");
  
    CreateAndSetRemoteOffer(sdp_string);
  
    ASSERT_EQ(1u, observer_.remote_streams()->count());
    MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
    EXPECT_EQ("default", remote_stream->id());
    ASSERT_EQ(1u, remote_stream->GetAudioTracks().size());
  }
  
  // This tests that when a Plan B endpoint receives an SDP that signals no media
  // stream IDs indicated by the special character "-" in the a=msid line, that
  // a default stream ID will be used for the MediaStream ID. This can occur
  // when a Unified Plan endpoint signals no media stream IDs, but signals both
  // a=ssrc msid and a=msid lines for interop signaling with Plan B.
  TEST_F(PeerConnectionInterfaceTestPlanB,
         SdpWithEmptyMsidAndSsrcCreatesDefaultStreamId) {
    RTCConfiguration config;
    config.enable_dtls_srtp = true;
    CreatePeerConnection(config);
    // Add a a=msid line to the SDP. This is prioritized when parsing the SDP, so
    // the sender's stream ID will be interpreted as no stream IDs.
    std::string sdp_string = kSdpStringWithStream1AudioTrackOnly;
    sdp_string.append("a=msid:- audiotrack0\n");
  
    CreateAndSetRemoteOffer(sdp_string);
  
    ASSERT_EQ(1u, observer_.remote_streams()->count());
    // Because SSRCs are signaled the track ID will be what was signaled in the
    // a=msid line.
    EXPECT_EQ("audiotrack0", observer_.last_added_track_label_);
    MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
    EXPECT_EQ("default", remote_stream->id());
    ASSERT_EQ(1u, remote_stream->GetAudioTracks().size());
  
    // Previously a bug ocurred when setting the remote description a second time.
    // This is because we checked equality of the remote StreamParams stream ID
    // (empty), and the previously set stream ID for the remote sender
    // ("default"). This cause a track to be removed, then added, when really
    // nothing should occur because it is the same track.
    CreateAndSetRemoteOffer(sdp_string);
    EXPECT_EQ(0u, observer_.remove_track_events_.size());
    EXPECT_EQ(1u, observer_.add_track_events_.size());
    EXPECT_EQ("audiotrack0", observer_.last_added_track_label_);
    remote_stream = observer_.remote_streams()->at(0);
    EXPECT_EQ("default", remote_stream->id());
    ASSERT_EQ(1u, remote_stream->GetAudioTracks().size());
  }
  
  // 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.
  // Don't run under Unified Plan since this behavior is Plan B specific.
  TEST_F(PeerConnectionInterfaceTestPlanB, LocalDescriptionChanged) {
    RTCConfiguration config;
    config.enable_dtls_srtp = true;
    CreatePeerConnection(config);
  
    // 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(std::move(offer)));
  
    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(std::move(offer)));
  
    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.
  // Don't run under Unified Plan since this behavior is Plan B specific.
  TEST_F(PeerConnectionInterfaceTestPlanB,
         AddLocalStreamAfterLocalDescriptionChanged) {
    RTCConfiguration config;
    config.enable_dtls_srtp = true;
    CreatePeerConnection(config);
  
    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(std::move(offer)));
    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_P(PeerConnectionInterfaceTest,
         ChangeSsrcOnTrackInLocalSessionDescription) {
    RTCConfiguration config;
    config.enable_dtls_srtp = true;
    CreatePeerConnection(config);
  
    AddAudioTrack(kAudioTracks[0]);
    AddVideoTrack(kVideoTracks[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<SessionDescriptionInterface> modified_offer =
        webrtc::CreateSessionDescription(
            webrtc::SdpType::kOffer, offer->session_id(),
            offer->session_version(), offer->description()->Clone());
    EXPECT_TRUE(DoSetLocalDescription(std::move(offer)));
  
    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(std::move(modified_offer)));
    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.
  // Don't run under Unified Plan since the stream API is not available.
  TEST_F(PeerConnectionInterfaceTestPlanB,
         SignalSameTracksInSeparateMediaStream) {
    RTCConfiguration config;
    config.enable_dtls_srtp = true;
    CreatePeerConnection(config);
  
    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(std::move(offer)));
  
    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(std::move(offer)));
  
    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_P(PeerConnectionInterfaceTest, OnAddTrackCallback) {
    RTCConfiguration config;
    config.enable_dtls_srtp = true;
    CreatePeerConnection(config);
    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(kSdpStringWithStream1PlanB);
    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_P(PeerConnectionInterfaceTest, SetConfigurationCausingIceRestart) {
    PeerConnectionInterface::RTCConfiguration config;
    config.type = PeerConnectionInterface::kRelay;
    CreatePeerConnection(config);
    config = pc_->GetConfiguration();
    AddAudioTrack(kAudioTracks[0], {kStreamId1});
    AddVideoTrack(kVideoTracks[0], {kStreamId1});
  
    // Do initial offer/answer so there's something to restart.
    CreateOfferAsLocalDescription();
    CreateAnswerAsRemoteDescription(GetSdpStringWithStream1());
  
    // 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).ok());
    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_P(PeerConnectionInterfaceTest, SetConfigurationNotCausingIceRestart) {
    PeerConnectionInterface::RTCConfiguration config;
    config.type = PeerConnectionInterface::kRelay;
    CreatePeerConnection(config);
    config = pc_->GetConfiguration();
    AddAudioTrack(kAudioTracks[0]);
    AddVideoTrack(kVideoTracks[0]);
  
    // Do initial offer/answer so there's something to restart.
    CreateOfferAsLocalDescription();
    CreateAnswerAsRemoteDescription(GetSdpStringWithStream1());
  
    // 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).ok());
    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_P(PeerConnectionInterfaceTest, SetConfigurationCausingPartialIceRestart) {
    PeerConnectionInterface::RTCConfiguration config;
    config.type = PeerConnectionInterface::kRelay;
    CreatePeerConnection(config);
    config = pc_->GetConfiguration();
    AddAudioTrack(kAudioTracks[0], {kStreamId1});
    AddVideoTrack(kVideoTracks[0], {kStreamId1});
  
    // Do initial offer/answer so there's something to restart.
    CreateOfferAsLocalDescription();
    CreateAnswerAsRemoteDescription(GetSdpStringWithStream1());
  
    // Change ICE policy, which should set the "needs-ice-restart" flag.
    config.type = PeerConnectionInterface::kAll;
    EXPECT_TRUE(pc_->SetConfiguration(config).ok());
  
    // Do ICE restart for the first m= section, initiated by remote peer.
    std::unique_ptr<webrtc::SessionDescriptionInterface> remote_offer(
        webrtc::CreateSessionDescription(SdpType::kOffer,
                                         GetSdpStringWithStream1(), nullptr));
    ASSERT_TRUE(remote_offer);
    remote_offer->description()->transport_infos()[0].description.ice_ufrag =
        "modified";
    EXPECT_TRUE(DoSetRemoteDescription(std::move(remote_offer)));
    CreateAnswerAsLocalDescription();
  
    // Grab the ufrags.
    std::vector<std::string> initial_ufrags = GetUfrags(pc_->local_description());
    ASSERT_EQ(2U, initial_ufrags.size());
  
    // Create offer and grab the new ufrags.
    CreateOfferAsLocalDescription();
    std::vector<std::string> subsequent_ufrags =
        GetUfrags(pc_->local_description());
    ASSERT_EQ(2U, 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_P(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> local_offer;
    ASSERT_TRUE(DoCreateOffer(&local_offer, nullptr));
    std::string sdp;
    EXPECT_TRUE(local_offer->ToString(&sdp));
  
    // Set local offer.
    SessionDescriptionInterface* local_offer_ptr = local_offer.get();
    EXPECT_TRUE(DoSetLocalDescription(std::move(local_offer)));
    EXPECT_EQ(local_offer_ptr, 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.
    std::unique_ptr<SessionDescriptionInterface> remote_pranswer(
        webrtc::CreateSessionDescription(SdpType::kPrAnswer, sdp));
    SessionDescriptionInterface* remote_pranswer_ptr = remote_pranswer.get();
    EXPECT_TRUE(DoSetRemoteDescription(std::move(remote_pranswer)));
    EXPECT_EQ(local_offer_ptr, pc_->pending_local_description());
    EXPECT_EQ(remote_pranswer_ptr, pc_->pending_remote_description());
    EXPECT_EQ(nullptr, pc_->current_local_description());
    EXPECT_EQ(nullptr, pc_->current_remote_description());
  
    // Set remote answer.
    std::unique_ptr<SessionDescriptionInterface> remote_answer(
        webrtc::CreateSessionDescription(SdpType::kAnswer, sdp));
    SessionDescriptionInterface* remote_answer_ptr = remote_answer.get();
    EXPECT_TRUE(DoSetRemoteDescription(std::move(remote_answer)));
    EXPECT_EQ(nullptr, pc_->pending_local_description());
    EXPECT_EQ(nullptr, pc_->pending_remote_description());
    EXPECT_EQ(local_offer_ptr, pc_->current_local_description());
    EXPECT_EQ(remote_answer_ptr, pc_->current_remote_description());
  
    // Set remote offer.
    std::unique_ptr<SessionDescriptionInterface> remote_offer(
        webrtc::CreateSessionDescription(SdpType::kOffer, sdp));
    SessionDescriptionInterface* remote_offer_ptr = remote_offer.get();
    EXPECT_TRUE(DoSetRemoteDescription(std::move(remote_offer)));
    EXPECT_EQ(remote_offer_ptr, pc_->pending_remote_description());
    EXPECT_EQ(nullptr, pc_->pending_local_description());
    EXPECT_EQ(local_offer_ptr, pc_->current_local_description());
    EXPECT_EQ(remote_answer_ptr, pc_->current_remote_description());
  
    // Set local pranswer.
    std::unique_ptr<SessionDescriptionInterface> local_pranswer(
        webrtc::CreateSessionDescription(SdpType::kPrAnswer, sdp));
    SessionDescriptionInterface* local_pranswer_ptr = local_pranswer.get();
    EXPECT_TRUE(DoSetLocalDescription(std::move(local_pranswer)));
    EXPECT_EQ(remote_offer_ptr, pc_->pending_remote_description());
    EXPECT_EQ(local_pranswer_ptr, pc_->pending_local_description());
    EXPECT_EQ(local_offer_ptr, pc_->current_local_description());
    EXPECT_EQ(remote_answer_ptr, pc_->current_remote_description());
  
    // Set local answer.
    std::unique_ptr<SessionDescriptionInterface> local_answer(
        webrtc::CreateSessionDescription(SdpType::kAnswer, sdp));
    SessionDescriptionInterface* local_answer_ptr = local_answer.get();
    EXPECT_TRUE(DoSetLocalDescription(std::move(local_answer)));
    EXPECT_EQ(nullptr, pc_->pending_remote_description());
    EXPECT_EQ(nullptr, pc_->pending_local_description());
    EXPECT_EQ(remote_offer_ptr, pc_->current_remote_description());
    EXPECT_EQ(local_answer_ptr, pc_->current_local_description());
  }
  
  // Tests that it won't crash when calling StartRtcEventLog or StopRtcEventLog
  // after the PeerConnection is closed.
  // This version tests the StartRtcEventLog version that receives an object
  // of type |RtcEventLogOutput|.
  TEST_P(PeerConnectionInterfaceTest,
         StartAndStopLoggingToOutputAfterPeerConnectionClosed) {
    CreatePeerConnection();
    // The RtcEventLog will be reset when the PeerConnection is closed.
    pc_->Close();
  
    EXPECT_FALSE(
        pc_->StartRtcEventLog(std::make_unique<webrtc::RtcEventLogOutputNull>(),
                              webrtc::RtcEventLog::kImmediateOutput));
    pc_->StopRtcEventLog();
  }
  
  // Test that generated offers/answers include "ice-option:trickle".
  TEST_P(PeerConnectionInterfaceTest, OffersAndAnswersHaveTrickleIceOption) {
    CreatePeerConnection();
  
    // First, create an offer with audio/video.
    RTCOfferAnswerOptions options;
    options.offer_to_receive_audio = 1;
    options.offer_to_receive_video = 1;
    std::unique_ptr<SessionDescriptionInterface> offer;
    ASSERT_TRUE(DoCreateOffer(&offer, &options));
    cricket::SessionDescription* desc = offer->description();
    ASSERT_EQ(2u, desc->transport_infos().size());
    EXPECT_TRUE(desc->transport_infos()[0].description.HasOption("trickle"));
    EXPECT_TRUE(desc->transport_infos()[1].description.HasOption("trickle"));
  
    // Apply the offer as a remote description, then create an answer.
    EXPECT_FALSE(pc_->can_trickle_ice_candidates());
    EXPECT_TRUE(DoSetRemoteDescription(std::move(offer)));
    ASSERT_TRUE(pc_->can_trickle_ice_candidates());
    EXPECT_TRUE(*(pc_->can_trickle_ice_candidates()));
    std::unique_ptr<SessionDescriptionInterface> answer;
    ASSERT_TRUE(DoCreateAnswer(&answer, &options));
    desc = answer->description();
    ASSERT_EQ(2u, desc->transport_infos().size());
    EXPECT_TRUE(desc->transport_infos()[0].description.HasOption("trickle"));
    EXPECT_TRUE(desc->transport_infos()[1].description.HasOption("trickle"));
  }
  
  // Test that ICE renomination isn't offered if it's not enabled in the PC's
  // RTCConfiguration.
  TEST_P(PeerConnectionInterfaceTest, IceRenominationNotOffered) {
    PeerConnectionInterface::RTCConfiguration config;
    config.enable_ice_renomination = false;
    CreatePeerConnection(config);
    AddAudioTrack("foo");
  
    std::unique_ptr<SessionDescriptionInterface> offer;
    ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
    cricket::SessionDescription* desc = offer->description();
    EXPECT_EQ(1u, desc->transport_infos().size());
    EXPECT_FALSE(
        desc->transport_infos()[0].description.GetIceParameters().renomination);
  }
  
  // Test that the ICE renomination option is present in generated offers/answers
  // if it's enabled in the PC's RTCConfiguration.
  TEST_P(PeerConnectionInterfaceTest, IceRenominationOptionInOfferAndAnswer) {
    PeerConnectionInterface::RTCConfiguration config;
    config.enable_ice_renomination = true;
    CreatePeerConnection(config);
    AddAudioTrack("foo");
  
    std::unique_ptr<SessionDescriptionInterface> offer;
    ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
    cricket::SessionDescription* desc = offer->description();
    EXPECT_EQ(1u, desc->transport_infos().size());
    EXPECT_TRUE(
        desc->transport_infos()[0].description.GetIceParameters().renomination);
  
    // Set the offer as a remote description, then create an answer and ensure it
    // has the renomination flag too.
    EXPECT_TRUE(DoSetRemoteDescription(std::move(offer)));
    std::unique_ptr<SessionDescriptionInterface> answer;
    ASSERT_TRUE(DoCreateAnswer(&answer, nullptr));
    desc = answer->description();
    EXPECT_EQ(1u, desc->transport_infos().size());
    EXPECT_TRUE(
        desc->transport_infos()[0].description.GetIceParameters().renomination);
  }
  
  // Test that if CreateOffer is called with the deprecated "offer to receive
  // audio/video" constraints, they're processed and result in an offer with
  // audio/video sections just as if RTCOfferAnswerOptions had been used.
  TEST_P(PeerConnectionInterfaceTest, CreateOfferWithOfferToReceiveConstraints) {
    CreatePeerConnection();
  
    RTCOfferAnswerOptions options;
    options.offer_to_receive_audio = 1;
    options.offer_to_receive_video = 1;
    std::unique_ptr<SessionDescriptionInterface> offer;
    ASSERT_TRUE(DoCreateOffer(&offer, &options));
  
    cricket::SessionDescription* desc = offer->description();
    const cricket::ContentInfo* audio = cricket::GetFirstAudioContent(desc);
    const cricket::ContentInfo* video = cricket::GetFirstVideoContent(desc);
    ASSERT_NE(nullptr, audio);
    ASSERT_NE(nullptr, video);
    EXPECT_FALSE(audio->rejected);
    EXPECT_FALSE(video->rejected);
  }
  
  // Test that if CreateAnswer is called with the deprecated "offer to receive
  // audio/video" constraints, they're processed and can be used to reject an
  // offered m= section just as can be done with RTCOfferAnswerOptions;
  // Don't run under Unified Plan since this behavior is not supported.
  TEST_F(PeerConnectionInterfaceTestPlanB,
         CreateAnswerWithOfferToReceiveConstraints) {
    CreatePeerConnection();
  
    // First, create an offer with audio/video and apply it as a remote
    // description.
    RTCOfferAnswerOptions options;
    options.offer_to_receive_audio = 1;
    options.offer_to_receive_video = 1;
    std::unique_ptr<SessionDescriptionInterface> offer;
    ASSERT_TRUE(DoCreateOffer(&offer, &options));
    EXPECT_TRUE(DoSetRemoteDescription(std::move(offer)));
  
    // Now create answer that rejects audio/video.
    options.offer_to_receive_audio = 0;
    options.offer_to_receive_video = 0;
    std::unique_ptr<SessionDescriptionInterface> answer;
    ASSERT_TRUE(DoCreateAnswer(&answer, &options));
  
    cricket::SessionDescription* desc = answer->description();
    const cricket::ContentInfo* audio = cricket::GetFirstAudioContent(desc);
    const cricket::ContentInfo* video = cricket::GetFirstVideoContent(desc);
    ASSERT_NE(nullptr, audio);
    ASSERT_NE(nullptr, video);
    EXPECT_TRUE(audio->rejected);
    EXPECT_TRUE(video->rejected);
  }
  
  // Test that negotiation can succeed with a data channel only, and with the max
  // bundle policy. Previously there was a bug that prevented this.
  #ifdef WEBRTC_HAVE_SCTP
  TEST_P(PeerConnectionInterfaceTest, DataChannelOnlyOfferWithMaxBundlePolicy) {
  #else
  TEST_P(PeerConnectionInterfaceTest,
         DISABLED_DataChannelOnlyOfferWithMaxBundlePolicy) {
  #endif  // WEBRTC_HAVE_SCTP
    PeerConnectionInterface::RTCConfiguration config;
    config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
    CreatePeerConnection(config);
  
    // First, create an offer with only a data channel and apply it as a remote
    // description.
    pc_->CreateDataChannel("test", nullptr);
    std::unique_ptr<SessionDescriptionInterface> offer;
    ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
    EXPECT_TRUE(DoSetRemoteDescription(std::move(offer)));
  
    // Create and set answer as well.
    std::unique_ptr<SessionDescriptionInterface> answer;
    ASSERT_TRUE(DoCreateAnswer(&answer, nullptr));
    EXPECT_TRUE(DoSetLocalDescription(std::move(answer)));
  }
  
  TEST_P(PeerConnectionInterfaceTest, SetBitrateWithoutMinSucceeds) {
    CreatePeerConnection();
    BitrateSettings bitrate;
    bitrate.start_bitrate_bps = 100000;
    EXPECT_TRUE(pc_->SetBitrate(bitrate).ok());
  }
  
  TEST_P(PeerConnectionInterfaceTest, SetBitrateNegativeMinFails) {
    CreatePeerConnection();
    BitrateSettings bitrate;
    bitrate.min_bitrate_bps = -1;
    EXPECT_FALSE(pc_->SetBitrate(bitrate).ok());
  }
  
  TEST_P(PeerConnectionInterfaceTest, SetBitrateCurrentLessThanMinFails) {
    CreatePeerConnection();
    BitrateSettings bitrate;
    bitrate.min_bitrate_bps = 5;
    bitrate.start_bitrate_bps = 3;
    EXPECT_FALSE(pc_->SetBitrate(bitrate).ok());
  }
  
  TEST_P(PeerConnectionInterfaceTest, SetBitrateCurrentNegativeFails) {
    CreatePeerConnection();
    BitrateSettings bitrate;
    bitrate.start_bitrate_bps = -1;
    EXPECT_FALSE(pc_->SetBitrate(bitrate).ok());
  }
  
  TEST_P(PeerConnectionInterfaceTest, SetBitrateMaxLessThanCurrentFails) {
    CreatePeerConnection();
    BitrateSettings bitrate;
    bitrate.start_bitrate_bps = 10;
    bitrate.max_bitrate_bps = 8;
    EXPECT_FALSE(pc_->SetBitrate(bitrate).ok());
  }
  
  TEST_P(PeerConnectionInterfaceTest, SetBitrateMaxLessThanMinFails) {
    CreatePeerConnection();
    BitrateSettings bitrate;
    bitrate.min_bitrate_bps = 10;
    bitrate.max_bitrate_bps = 8;
    EXPECT_FALSE(pc_->SetBitrate(bitrate).ok());
  }
  
  TEST_P(PeerConnectionInterfaceTest, SetBitrateMaxNegativeFails) {
    CreatePeerConnection();
    BitrateSettings bitrate;
    bitrate.max_bitrate_bps = -1;
    EXPECT_FALSE(pc_->SetBitrate(bitrate).ok());
  }
  
  // The current bitrate from BitrateSettings is currently clamped
  // by Call's BitrateConstraints, which comes from the SDP or a default value.
  // This test checks that a call to SetBitrate with a current bitrate that will
  // be clamped succeeds.
  TEST_P(PeerConnectionInterfaceTest, SetBitrateCurrentLessThanImplicitMin) {
    CreatePeerConnection();
    BitrateSettings bitrate;
    bitrate.start_bitrate_bps = 1;
    EXPECT_TRUE(pc_->SetBitrate(bitrate).ok());
  }
  
  // The following tests verify that the offer can be created correctly.
  TEST_P(PeerConnectionInterfaceTest,
         CreateOfferFailsWithInvalidOfferToReceiveAudio) {
    RTCOfferAnswerOptions rtc_options;
  
    // Setting offer_to_receive_audio to a value lower than kUndefined or greater
    // than kMaxOfferToReceiveMedia should be treated as invalid.
    rtc_options.offer_to_receive_audio = RTCOfferAnswerOptions::kUndefined - 1;
    CreatePeerConnection();
    EXPECT_FALSE(CreateOfferWithOptions(rtc_options));
  
    rtc_options.offer_to_receive_audio =
        RTCOfferAnswerOptions::kMaxOfferToReceiveMedia + 1;
    EXPECT_FALSE(CreateOfferWithOptions(rtc_options));
  }
  
  TEST_P(PeerConnectionInterfaceTest,
         CreateOfferFailsWithInvalidOfferToReceiveVideo) {
    RTCOfferAnswerOptions rtc_options;
  
    // Setting offer_to_receive_video to a value lower than kUndefined or greater
    // than kMaxOfferToReceiveMedia should be treated as invalid.
    rtc_options.offer_to_receive_video = RTCOfferAnswerOptions::kUndefined - 1;
    CreatePeerConnection();
    EXPECT_FALSE(CreateOfferWithOptions(rtc_options));
  
    rtc_options.offer_to_receive_video =
        RTCOfferAnswerOptions::kMaxOfferToReceiveMedia + 1;
    EXPECT_FALSE(CreateOfferWithOptions(rtc_options));
  }
  
  // Test that the audio and video content will be added to an offer if both
  // |offer_to_receive_audio| and |offer_to_receive_video| options are 1.
  TEST_P(PeerConnectionInterfaceTest, CreateOfferWithAudioVideoOptions) {
    RTCOfferAnswerOptions rtc_options;
    rtc_options.offer_to_receive_audio = 1;
    rtc_options.offer_to_receive_video = 1;
  
    std::unique_ptr<SessionDescriptionInterface> offer;
    CreatePeerConnection();
    offer = CreateOfferWithOptions(rtc_options);
    ASSERT_TRUE(offer);
    EXPECT_NE(nullptr, GetFirstAudioContent(offer->description()));
    EXPECT_NE(nullptr, GetFirstVideoContent(offer->description()));
  }
  
  // Test that only audio content will be added to the offer if only
  // |offer_to_receive_audio| options is 1.
  TEST_P(PeerConnectionInterfaceTest, CreateOfferWithAudioOnlyOptions) {
    RTCOfferAnswerOptions rtc_options;
    rtc_options.offer_to_receive_audio = 1;
    rtc_options.offer_to_receive_video = 0;
  
    std::unique_ptr<SessionDescriptionInterface> offer;
    CreatePeerConnection();
    offer = CreateOfferWithOptions(rtc_options);
    ASSERT_TRUE(offer);
    EXPECT_NE(nullptr, GetFirstAudioContent(offer->description()));
    EXPECT_EQ(nullptr, GetFirstVideoContent(offer->description()));
  }
  
  // Test that only video content will be added if only |offer_to_receive_video|
  // options is 1.
  TEST_P(PeerConnectionInterfaceTest, CreateOfferWithVideoOnlyOptions) {
    RTCOfferAnswerOptions rtc_options;
    rtc_options.offer_to_receive_audio = 0;
    rtc_options.offer_to_receive_video = 1;
  
    std::unique_ptr<SessionDescriptionInterface> offer;
    CreatePeerConnection();
    offer = CreateOfferWithOptions(rtc_options);
    ASSERT_TRUE(offer);
    EXPECT_EQ(nullptr, GetFirstAudioContent(offer->description()));
    EXPECT_NE(nullptr, GetFirstVideoContent(offer->description()));
  }
  
  // Test that no media content will be added to the offer if using default
  // RTCOfferAnswerOptions.
  TEST_P(PeerConnectionInterfaceTest, CreateOfferWithDefaultOfferAnswerOptions) {
    RTCOfferAnswerOptions rtc_options;
  
    std::unique_ptr<SessionDescriptionInterface> offer;
    CreatePeerConnection();
    offer = CreateOfferWithOptions(rtc_options);
    ASSERT_TRUE(offer);
    EXPECT_EQ(nullptr, GetFirstAudioContent(offer->description()));
    EXPECT_EQ(nullptr, GetFirstVideoContent(offer->description()));
  }
  
  // Test that if |ice_restart| is true, the ufrag/pwd will change, otherwise
  // ufrag/pwd will be the same in the new offer.
  TEST_P(PeerConnectionInterfaceTest, CreateOfferWithIceRestart) {
    CreatePeerConnection();
  
    RTCOfferAnswerOptions rtc_options;
    rtc_options.ice_restart = false;
    rtc_options.offer_to_receive_audio = 1;
  
    std::unique_ptr<SessionDescriptionInterface> offer;
    CreateOfferWithOptionsAsLocalDescription(&offer, rtc_options);
    std::string mid = cricket::GetFirstAudioContent(offer->description())->name;
    auto ufrag1 =
        offer->description()->GetTransportInfoByName(mid)->description.ice_ufrag;
    auto pwd1 =
        offer->description()->GetTransportInfoByName(mid)->description.ice_pwd;
  
    // |ice_restart| is false, the ufrag/pwd shouldn't change.
    CreateOfferWithOptionsAsLocalDescription(&offer, rtc_options);
    auto ufrag2 =
        offer->description()->GetTransportInfoByName(mid)->description.ice_ufrag;
    auto pwd2 =
        offer->description()->GetTransportInfoByName(mid)->description.ice_pwd;
  
    // |ice_restart| is true, the ufrag/pwd should change.
    rtc_options.ice_restart = true;
    CreateOfferWithOptionsAsLocalDescription(&offer, rtc_options);
    auto ufrag3 =
        offer->description()->GetTransportInfoByName(mid)->description.ice_ufrag;
    auto pwd3 =
        offer->description()->GetTransportInfoByName(mid)->description.ice_pwd;
  
    EXPECT_EQ(ufrag1, ufrag2);
    EXPECT_EQ(pwd1, pwd2);
    EXPECT_NE(ufrag2, ufrag3);
    EXPECT_NE(pwd2, pwd3);
  }
  
  // Test that if |use_rtp_mux| is true, the bundling will be enabled in the
  // offer; if it is false, there won't be any bundle group in the offer.
  TEST_P(PeerConnectionInterfaceTest, CreateOfferWithRtpMux) {
    RTCOfferAnswerOptions rtc_options;
    rtc_options.offer_to_receive_audio = 1;
    rtc_options.offer_to_receive_video = 1;
  
    std::unique_ptr<SessionDescriptionInterface> offer;
    CreatePeerConnection();
  
    rtc_options.use_rtp_mux = true;
    offer = CreateOfferWithOptions(rtc_options);
    ASSERT_TRUE(offer);
    EXPECT_NE(nullptr, GetFirstAudioContent(offer->description()));
    EXPECT_NE(nullptr, GetFirstVideoContent(offer->description()));
    EXPECT_TRUE(offer->description()->HasGroup(cricket::GROUP_TYPE_BUNDLE));
  
    rtc_options.use_rtp_mux = false;
    offer = CreateOfferWithOptions(rtc_options);
    ASSERT_TRUE(offer);
    EXPECT_NE(nullptr, GetFirstAudioContent(offer->description()));
    EXPECT_NE(nullptr, GetFirstVideoContent(offer->description()));
    EXPECT_FALSE(offer->description()->HasGroup(cricket::GROUP_TYPE_BUNDLE));
  }
  
  // This test ensures OnRenegotiationNeeded is called when we add track with
  // MediaStream -> AddTrack in the same way it is called when we add track with
  // PeerConnection -> AddTrack.
  // The test can be removed once addStream is rewritten in terms of addTrack
  // https://bugs.chromium.org/p/webrtc/issues/detail?id=7815
  // Don't run under Unified Plan since the stream API is not available.
  TEST_F(PeerConnectionInterfaceTestPlanB,
         MediaStreamAddTrackRemoveTrackRenegotiate) {
    CreatePeerConnectionWithoutDtls();
    rtc::scoped_refptr<MediaStreamInterface> stream(
        pc_factory_->CreateLocalMediaStream(kStreamId1));
    pc_->AddStream(stream);
    rtc::scoped_refptr<AudioTrackInterface> audio_track(
        CreateAudioTrack("audio_track"));
    rtc::scoped_refptr<VideoTrackInterface> video_track(
        CreateVideoTrack("video_track"));
    stream->AddTrack(audio_track);
    EXPECT_TRUE_WAIT(observer_.renegotiation_needed_, kTimeout);
    observer_.renegotiation_needed_ = false;
  
    CreateOfferReceiveAnswer();
    stream->AddTrack(video_track);
    EXPECT_TRUE_WAIT(observer_.renegotiation_needed_, kTimeout);
    observer_.renegotiation_needed_ = false;
  
    CreateOfferReceiveAnswer();
    stream->RemoveTrack(audio_track);
    EXPECT_TRUE_WAIT(observer_.renegotiation_needed_, kTimeout);
    observer_.renegotiation_needed_ = false;
  
    CreateOfferReceiveAnswer();
    stream->RemoveTrack(video_track);
    EXPECT_TRUE_WAIT(observer_.renegotiation_needed_, kTimeout);
    observer_.renegotiation_needed_ = false;
  }
  
  // Tests that an error is returned if a description is applied that has fewer
  // media sections than the existing description.
  TEST_P(PeerConnectionInterfaceTest,
         MediaSectionCountEnforcedForSubsequentOffer) {
    CreatePeerConnection();
    AddAudioTrack("audio_label");
    AddVideoTrack("video_label");
  
    std::unique_ptr<SessionDescriptionInterface> offer;
    ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
    EXPECT_TRUE(DoSetRemoteDescription(std::move(offer)));
  
    // A remote offer with fewer media sections should be rejected.
    ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
    offer->description()->contents().pop_back();
    offer->description()->contents().pop_back();
    ASSERT_TRUE(offer->description()->contents().empty());
    EXPECT_FALSE(DoSetRemoteDescription(std::move(offer)));
  
    std::unique_ptr<SessionDescriptionInterface> answer;
    ASSERT_TRUE(DoCreateAnswer(&answer, nullptr));
    EXPECT_TRUE(DoSetLocalDescription(std::move(answer)));
  
    // A subsequent local offer with fewer media sections should be rejected.
    ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
    offer->description()->contents().pop_back();
    offer->description()->contents().pop_back();
    ASSERT_TRUE(offer->description()->contents().empty());
    EXPECT_FALSE(DoSetLocalDescription(std::move(offer)));
  }
  
  TEST_P(PeerConnectionInterfaceTest, ExtmapAllowMixedIsConfigurable) {
    RTCConfiguration config;
    // Default behavior is true.
    CreatePeerConnection(config);
    std::unique_ptr<SessionDescriptionInterface> offer;
    ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
    EXPECT_TRUE(offer->description()->extmap_allow_mixed());
    // Possible to set to false.
    config.offer_extmap_allow_mixed = false;
    CreatePeerConnection(config);
    offer = nullptr;
    ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
    EXPECT_FALSE(offer->description()->extmap_allow_mixed());
  }
  
  INSTANTIATE_TEST_SUITE_P(PeerConnectionInterfaceTest,
                           PeerConnectionInterfaceTest,
                           Values(SdpSemantics::kPlanB,
                                  SdpSemantics::kUnifiedPlan));
  
  class PeerConnectionMediaConfigTest : public ::testing::Test {
   protected:
    void SetUp() override {
      pcf_ = PeerConnectionFactoryForTest::CreatePeerConnectionFactoryForTest();
    }
    const cricket::MediaConfig TestCreatePeerConnection(
        const RTCConfiguration& config) {
      rtc::scoped_refptr<PeerConnectionInterface> pc(
          pcf_->CreatePeerConnection(config, nullptr, nullptr, &observer_));
      EXPECT_TRUE(pc.get());
      observer_.SetPeerConnectionInterface(pc.get());
      return pc->GetConfiguration().media_config;
    }
  
    rtc::scoped_refptr<PeerConnectionFactoryForTest> pcf_;
    MockPeerConnectionObserver observer_;
  };
  
  // This sanity check validates the test infrastructure itself.
  TEST_F(PeerConnectionMediaConfigTest, TestCreateAndClose) {
    PeerConnectionInterface::RTCConfiguration config;
    rtc::scoped_refptr<PeerConnectionInterface> pc(
        pcf_->CreatePeerConnection(config, nullptr, nullptr, &observer_));
    EXPECT_TRUE(pc.get());
    observer_.SetPeerConnectionInterface(pc.get());  // Required.
    pc->Close();                                     // No abort -> ok.
    SUCCEED();
  }
  
  // This test verifies the default behaviour with no constraints and a
  // default RTCConfiguration.
  TEST_F(PeerConnectionMediaConfigTest, TestDefaults) {
    PeerConnectionInterface::RTCConfiguration config;
  
    const cricket::MediaConfig& media_config = TestCreatePeerConnection(config);
  
    EXPECT_FALSE(media_config.enable_dscp);
    EXPECT_TRUE(media_config.video.enable_cpu_adaptation);
    EXPECT_TRUE(media_config.video.enable_prerenderer_smoothing);
    EXPECT_FALSE(media_config.video.suspend_below_min_bitrate);
    EXPECT_FALSE(media_config.video.experiment_cpu_load_estimator);
  }
  
  // This test verifies that the enable_prerenderer_smoothing flag is
  // propagated from RTCConfiguration to the PeerConnection.
  TEST_F(PeerConnectionMediaConfigTest, TestDisablePrerendererSmoothingTrue) {
    PeerConnectionInterface::RTCConfiguration config;
  
    config.set_prerenderer_smoothing(false);
    const cricket::MediaConfig& media_config = TestCreatePeerConnection(config);
  
    EXPECT_FALSE(media_config.video.enable_prerenderer_smoothing);
  }
  
  // This test verifies that the experiment_cpu_load_estimator flag is
  // propagated from RTCConfiguration to the PeerConnection.
  TEST_F(PeerConnectionMediaConfigTest, TestEnableExperimentCpuLoadEstimator) {
    PeerConnectionInterface::RTCConfiguration config;
  
    config.set_experiment_cpu_load_estimator(true);
    const cricket::MediaConfig& media_config = TestCreatePeerConnection(config);
  
    EXPECT_TRUE(media_config.video.experiment_cpu_load_estimator);
  }
  
  // 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);
  }
  
  }  // namespace
  }  // namespace webrtc
  