Replace scoped_ptr with unique_ptr in webrtc/voice_engine/

webrtc/voice_engine/channel.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.
  */
 
@@ skipping 1053 matching lines @@
   _voiceEngineObserverPtr = voiceEngineObserver;
   _callbackCritSectPtr = callbackCritSect;
   return 0;
 }
 
 int32_t Channel::UpdateLocalTimeStamp() {
   _timeStamp += static_cast<uint32_t>(_audioFrame.samples_per_channel_);
   return 0;
 }
 
-void Channel::SetSink(rtc::scoped_ptr<AudioSinkInterface> sink) {
+void Channel::SetSink(std::unique_ptr<AudioSinkInterface> sink) {
   rtc::CritScope cs(&_callbackCritSect);
   audio_sink_ = std::move(sink);
 }
 
 int32_t Channel::StartPlayout() {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::StartPlayout()");
   if (channel_state_.Get().playing) {
     return 0;
   }
@@ skipping 2173 matching lines @@
 int Channel::GetRtpRtcp(RtpRtcp** rtpRtcpModule,
                         RtpReceiver** rtp_receiver) const {
   *rtpRtcpModule = _rtpRtcpModule.get();
   *rtp_receiver = rtp_receiver_.get();
   return 0;
 }
 
 // TODO(andrew): refactor Mix functions here and in transmit_mixer.cc to use
 // a shared helper.
 int32_t Channel::MixOrReplaceAudioWithFile(int mixingFrequency) {
-  rtc::scoped_ptr<int16_t[]> fileBuffer(new int16_t[640]);
+  std::unique_ptr<int16_t[]> fileBuffer(new int16_t[640]);
   size_t fileSamples(0);
 
   {
     rtc::CritScope cs(&_fileCritSect);
 
     if (_inputFilePlayerPtr == NULL) {
       WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
                    "Channel::MixOrReplaceAudioWithFile() fileplayer"
                    " doesnt exist");
       return -1;
@@ skipping 27 matching lines @@
     _audioFrame.UpdateFrame(
         _channelId, 0xFFFFFFFF, fileBuffer.get(), fileSamples, mixingFrequency,
         AudioFrame::kNormalSpeech, AudioFrame::kVadUnknown, 1);
   }
   return 0;
 }
 
 int32_t Channel::MixAudioWithFile(AudioFrame& audioFrame, int mixingFrequency) {
   assert(mixingFrequency <= 48000);
 
-  rtc::scoped_ptr<int16_t[]> fileBuffer(new int16_t[960]);
+  std::unique_ptr<int16_t[]> fileBuffer(new int16_t[960]);
   size_t fileSamples(0);
 
   {
     rtc::CritScope cs(&_fileCritSect);
 
     if (_outputFilePlayerPtr == NULL) {
       WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
                    "Channel::MixAudioWithFile() file mixing failed");
       return -1;
     }
@@ skipping 323 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
       0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc