Index: webrtc/audio/audio_receive_stream.cc |
diff --git a/webrtc/audio/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc |
index 87cb215bae6fa4473fd7588b85acfad4c124ae49..a2fe181173e5e8c2dca82aa139d40a37d069e09a 100644 |
--- a/webrtc/audio/audio_receive_stream.cc |
+++ b/webrtc/audio/audio_receive_stream.cc |
@@ -11,6 +11,7 @@ |
#include "webrtc/audio/audio_receive_stream.h" |
#include <string> |
+#include <utility> |
#include "webrtc/audio/audio_state.h" |
#include "webrtc/audio/conversion.h" |
@@ -201,6 +202,11 @@ webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const { |
return stats; |
} |
+void AudioReceiveStream::SetSink(rtc::scoped_ptr<AudioSinkInterface> sink) { |
+ RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
+ channel_proxy_->SetSink(std::move(sink)); |
+} |
+ |
const webrtc::AudioReceiveStream::Config& AudioReceiveStream::config() const { |
RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
return config_; |