Replace scoped_ptr with unique_ptr in webrtc/voice_engine/

webrtc/audio/audio_receive_stream.cc

Index: webrtc/audio/audio_receive_stream.cc
diff --git a/webrtc/audio/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc
index e8bc0f1857280b0c8463b172609cb3c5790033d4..9f19b32e592fcea2edf70ebec253539e4ac82268 100644
--- a/webrtc/audio/audio_receive_stream.cc
+++ b/webrtc/audio/audio_receive_stream.cc
@@ -93,7 +93,8 @@ AudioReceiveStream::AudioReceiveStream(
   RTC_DCHECK(rtp_header_parser_);
 
   VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
-  channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id);
+  channel_proxy_ =
+      rtc::UniqueToScoped(voe_impl->GetChannelProxy(config_.voe_channel_id));
   channel_proxy_->SetLocalSSRC(config.rtp.local_ssrc);
   for (const auto& extension : config.rtp.extensions) {
     if (extension.name == RtpExtension::kAudioLevel) {
@@ -230,7 +231,7 @@ webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {
 
 void AudioReceiveStream::SetSink(rtc::scoped_ptr<AudioSinkInterface> sink) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
-  channel_proxy_->SetSink(std::move(sink));
+  channel_proxy_->SetSink(rtc::ScopedToUnique(std::move(sink)));
 }
 
 const webrtc::AudioReceiveStream::Config& AudioReceiveStream::config() const {