Index: webrtc/media/engine/webrtcvoiceengine.cc |
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc |
index 70f7f3a4b4c5a79e93a3dde73ab1e9858701c054..d2db76a7bdba63347740116c18984f428e073d28 100644 |
--- a/webrtc/media/engine/webrtcvoiceengine.cc |
+++ b/webrtc/media/engine/webrtcvoiceengine.cc |
@@ -1336,9 +1336,9 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream { |
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: |
@@ -2262,7 +2262,7 @@ void WebRtcVoiceMediaChannel::OnPacketReceived( |
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)); |
} |
@@ -2492,13 +2492,13 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) { |
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)); |
} |