Replace scoped_ptr with unique_ptr in webrtc/audio/

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 9f19b32e592fcea2edf70ebec253539e4ac82268..69819f22887e61af7fc807b70bcfeae48d306af6 100644
--- a/webrtc/audio/audio_receive_stream.cc
+++ b/webrtc/audio/audio_receive_stream.cc
@@ -93,8 +93,7 @@ AudioReceiveStream::AudioReceiveStream(
   RTC_DCHECK(rtp_header_parser_);
 
   VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
-  channel_proxy_ =
-      rtc::UniqueToScoped(voe_impl->GetChannelProxy(config_.voe_channel_id));
+  channel_proxy_ = 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) {
@@ -229,9 +228,9 @@ webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {
   return stats;
 }
 
-void AudioReceiveStream::SetSink(rtc::scoped_ptr<AudioSinkInterface> sink) {
+void AudioReceiveStream::SetSink(std::unique_ptr<AudioSinkInterface> sink) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
-  channel_proxy_->SetSink(rtc::ScopedToUnique(std::move(sink)));
+  channel_proxy_->SetSink(std::move(sink));
 }
 
 const webrtc::AudioReceiveStream::Config& AudioReceiveStream::config() const {