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

webrtc/modules/video_coding/jitter_buffer.cc

 /*
  *  Copyright (c) 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 "webrtc/modules/video_coding/jitter_buffer.h"
@@ skipping 201 matching lines @@
       frame_it.second->SetGofInfo(ss_it->second, gof_idx);
     }
   }
 }
 
 VCMJitterBuffer::VCMJitterBuffer(Clock* clock,
                                  rtc::scoped_ptr<EventWrapper> event)
     : clock_(clock),
       running_(false),
       crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
-      frame_event_(event.Pass()),
+      frame_event_(std::move(event)),
       max_number_of_frames_(kStartNumberOfFrames),
       free_frames_(),
       decodable_frames_(),
       incomplete_frames_(),
       last_decoded_state_(),
       first_packet_since_reset_(true),
       stats_callback_(NULL),
       incoming_frame_rate_(0),
       incoming_frame_count_(0),
       time_last_incoming_frame_count_(0),
@@ skipping 1101 matching lines @@
   }
   // Evaluate if the RTT is higher than |high_rtt_nack_threshold_ms_|, and in
   // that case we don't wait for retransmissions.
   if (high_rtt_nack_threshold_ms_ >= 0 &&
       rtt_ms_ >= high_rtt_nack_threshold_ms_) {
     return false;
   }
   return true;
 }
 }  // namespace webrtc