Replacing SSLIdentity* with scoped_refptr<RTCCertificate> in TransportChannel layer

talk/app/webrtc/statscollector_unittest.cc

 /*
  * libjingle
  * Copyright 2014 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 655 matching lines @@
     rtc::scoped_ptr<cricket::FakeTransport> transport(
         new cricket::FakeTransport(
             session_.signaling_thread(),
             session_.worker_thread(),
             transport_stats.content_name));
     transport->SetCertificate(local_certificate);
     cricket::FakeTransportChannel* channel =
         static_cast<cricket::FakeTransportChannel*>(
             transport->CreateChannel(channel_stats.component));
     EXPECT_FALSE(channel == NULL);
-    channel->SetRemoteCertificate(remote_cert_copy.get());
+    channel->SetRemoteSSLCertificate(remote_cert_copy.get());
 
     // Configure MockWebRtcSession
     EXPECT_CALL(session_, GetTransport(transport_stats.content_name))
       .WillRepeatedly(Return(transport.get()));
     EXPECT_CALL(session_, GetTransportStats(_))
       .WillOnce(DoAll(SetArgPointee<0>(session_stats),
                       Return(true)));
     EXPECT_CALL(session_, video_channel()).WillRepeatedly(ReturnNull());
     EXPECT_CALL(session_, voice_channel()).WillRepeatedly(ReturnNull());
 
@@ skipping 1017 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