Replace scoped_ptr with unique_ptr in webrtc/media/

webrtc/media/engine/webrtcvoiceengine.cc

 /*
  *  Copyright (c) 2004 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 1318 matching lines @@
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
     RTC_DCHECK(stream_);
     return stream_->GetStats();
   }
 
   int channel() const {
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
     return config_.voe_channel_id;
   }
 
-  void SetRawAudioSink(rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) {
+  void SetRawAudioSink(std::unique_ptr<webrtc::AudioSinkInterface> sink) {
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-    stream_->SetSink(rtc::ScopedToUnique(std::move(sink)));
+    stream_->SetSink(std::move(sink));
   }
 
  private:
   void RecreateAudioReceiveStream(
       bool use_transport_cc,
       const std::vector<webrtc::RtpExtension>& extensions) {
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
     if (stream_) {
       call_->DestroyAudioReceiveStream(stream_);
       stream_ = nullptr;
@@ skipping 903 matching lines @@
     StreamParams sp;
     sp.ssrcs.push_back(ssrc);
     LOG(LS_INFO) << "Creating default receive stream for SSRC=" << ssrc << ".";
     if (!AddRecvStream(sp)) {
       LOG(LS_WARNING) << "Could not create default receive stream.";
       return;
     }
     default_recv_ssrc_ = ssrc;
     SetOutputVolume(default_recv_ssrc_, default_recv_volume_);
     if (default_sink_) {
-      rtc::scoped_ptr<webrtc::AudioSinkInterface> proxy_sink(
+      std::unique_ptr<webrtc::AudioSinkInterface> proxy_sink(
           new ProxySink(default_sink_.get()));
       SetRawAudioSink(default_recv_ssrc_, std::move(proxy_sink));
     }
   }
 
   // Forward packet to Call. If the SSRC is unknown we'll return after this.
   const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp,
                                               packet_time.not_before);
   webrtc::PacketReceiver::DeliveryStatus delivery_result =
       call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO,
@@ skipping 209 matching lines @@
     rinfo.decoding_plc_cng = stats.decoding_plc_cng;
     rinfo.capture_start_ntp_time_ms = stats.capture_start_ntp_time_ms;
     info->receivers.push_back(rinfo);
   }
 
   return true;
 }
 
 void WebRtcVoiceMediaChannel::SetRawAudioSink(
     uint32_t ssrc,
-    rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) {
+    std::unique_ptr<webrtc::AudioSinkInterface> sink) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::SetRawAudioSink: ssrc:" << ssrc
                   << " " << (sink ? "(ptr)" : "NULL");
   if (ssrc == 0) {
     if (default_recv_ssrc_ != -1) {
-      rtc::scoped_ptr<webrtc::AudioSinkInterface> proxy_sink(
+      std::unique_ptr<webrtc::AudioSinkInterface> proxy_sink(
           sink ? new ProxySink(sink.get()) : nullptr);
       SetRawAudioSink(default_recv_ssrc_, std::move(proxy_sink));
     }
     default_sink_ = std::move(sink);
     return;
   }
   const auto it = recv_streams_.find(ssrc);
   if (it == recv_streams_.end()) {
     LOG(LS_WARNING) << "SetRawAudioSink: no recv stream" << ssrc;
     return;
@@ skipping 34 matching lines @@
     }
   } else {
     LOG(LS_INFO) << "Stopping playout for channel #" << channel;
     engine()->voe()->base()->StopPlayout(channel);
   }
   return true;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE