Don't call the Pass methods of rtc::Buffer, rtc::scoped_ptr, and rtc::ScopedVector

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 678 matching lines @@
     transport_stats.transport_name = "audio";
     transport_stats.channel_stats.push_back(channel_stats);
 
     SessionStats session_stats;
     session_stats.transport_stats[transport_stats.transport_name] =
         transport_stats;
 
     // Fake certificate to report
     rtc::scoped_refptr<rtc::RTCCertificate> local_certificate(
         rtc::RTCCertificate::Create(rtc::scoped_ptr<rtc::FakeSSLIdentity>(
-                                        new rtc::FakeSSLIdentity(local_cert))
-                                        .Pass()));
+            new rtc::FakeSSLIdentity(local_cert))));
 
     // Configure MockWebRtcSession
     EXPECT_CALL(session_,
                 GetLocalCertificate(transport_stats.transport_name, _))
         .WillOnce(DoAll(SetArgPointee<1>(local_certificate), Return(true)));
     EXPECT_CALL(session_,
                 GetRemoteSSLCertificate(transport_stats.transport_name, _))
         .WillOnce(
             DoAll(SetArgPointee<1>(remote_cert.GetReference()), Return(true)));
     EXPECT_CALL(session_, GetTransportStats(_))
@@ skipping 1029 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