Add PeerConnection IsClosed check.

webrtc/api/statscollector_unittest.cc

 /*
  *  Copyright 2014 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.
  */
 
@@ skipping 58 matching lines @@
  public:
   // TODO(nisse): Valid overrides commented out, because the gmock
   // methods don't use any override declarations, and we want to avoid
   // warnings from -Winconsistent-missing-override. See
   // http://crbug.com/428099.
   explicit MockWebRtcSession(webrtc::MediaControllerInterface* media_controller)
       : WebRtcSession(media_controller,
                       rtc::Thread::Current(),
                       rtc::Thread::Current(),
                       rtc::Thread::Current(),
-                      nullptr) {}
+                      nullptr,
+                      new cricket::TransportController(rtc::Thread::Current(),
+                                                       rtc::Thread::Current(),
+                                                       nullptr)) {}
   MOCK_METHOD0(voice_channel, cricket::VoiceChannel*());
   MOCK_METHOD0(video_channel, cricket::VideoChannel*());
   // Libjingle uses "local" for a outgoing track, and "remote" for a incoming
   // track.
   MOCK_METHOD2(GetLocalTrackIdBySsrc, bool(uint32_t, std::string*));
   MOCK_METHOD2(GetRemoteTrackIdBySsrc, bool(uint32_t, std::string*));
   MOCK_METHOD1(GetTransportStats, bool(SessionStats*));
   MOCK_METHOD2(GetLocalCertificate,
                bool(const std::string& transport_name,
                     rtc::scoped_refptr<rtc::RTCCertificate>* certificate));
@@ skipping 1848 matching lines @@
   cricket::VoiceSenderInfo new_voice_sender_info;
   InitVoiceSenderInfo(&new_voice_sender_info);
   cricket::VoiceMediaInfo new_stats_read;
   reports.clear();
   SetupAndVerifyAudioTrackStats(
       new_audio_track.get(), stream_.get(), &stats, &voice_channel, kVcName,
       media_channel, &new_voice_sender_info, NULL, &new_stats_read, &reports);
 }
 
 }  // namespace webrtc