Use std::move() instead of scoped_ptr<>::Pass().

remoting/protocol/webrtc_connection_to_client.cc

 // Copyright 2015 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "remoting/protocol/webrtc_connection_to_client.h"
 
 #include "base/bind.h"
 #include "base/location.h"
 #include "net/base/io_buffer.h"
 #include "remoting/codec/video_encoder.h"
@@ skipping 56 matching lines @@
   event_handler_->OnInputEventReceived(this, timestamp);
 }
 
 scoped_ptr<VideoStream> WebrtcConnectionToClient::StartVideoStream(
     scoped_ptr<webrtc::DesktopCapturer> desktop_capturer) {
   // TODO(sergeyu): Reconsider Transport interface and how it's used here.
   WebrtcTransport* transport = session_->GetTransport()->AsWebrtcTransport();
   CHECK(transport);
 
   scoped_ptr<WebrtcVideoCapturerAdapter> video_capturer_adapter(
-      new WebrtcVideoCapturerAdapter(desktop_capturer.Pass()));
+      new WebrtcVideoCapturerAdapter(std::move(desktop_capturer)));
 
   // Set video stream constraints.
   webrtc::FakeConstraints video_constraints;
   video_constraints.AddMandatory(
       webrtc::MediaConstraintsInterface::kMinFrameRate, 5);
 
   rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track =
       transport->peer_connection_factory()->CreateVideoTrack(
           kVideoLabel,
           transport->peer_connection_factory()->CreateVideoSource(
               video_capturer_adapter.release(), &video_constraints));
 
   rtc::scoped_refptr<webrtc::MediaStreamInterface> video_stream =
       transport->peer_connection_factory()->CreateLocalMediaStream(
           kStreamLabel);
 
   if (!video_stream->AddTrack(video_track) ||
       !transport->peer_connection()->AddStream(video_stream)) {
     return nullptr;
   }
 
-  scoped_ptr<VideoStream> result(
+  return make_scoped_ptr(
       new WebrtcVideoStream(transport->peer_connection(), video_stream));
-  return result.Pass();
 }
 
 AudioStub* WebrtcConnectionToClient::audio_stub() {
   DCHECK(thread_checker_.CalledOnValidThread());
   return nullptr;
 }
 
 // Return pointer to ClientStub.
 ClientStub* WebrtcConnectionToClient::client_stub() {
   DCHECK(thread_checker_.CalledOnValidThread());
@@ skipping 77 matching lines @@
     ErrorCode error) {
   DCHECK(thread_checker_.CalledOnValidThread());
 
   LOG(ERROR) << "Failed to connect channel "
              << channel_dispatcher->channel_name();
   session_->Close(CHANNEL_CONNECTION_ERROR);
 }
 
 }  // namespace protocol
 }  // namespace remoting