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..dfad79f9d76420460267f2fdd0e1265a928b8786 100644 |
--- a/webrtc/audio/audio_receive_stream.cc |
+++ b/webrtc/audio/audio_receive_stream.cc |
@@ -11,7 +11,9 @@ |
#include "webrtc/audio/audio_receive_stream.h" |
#include <string> |
+#include <utility> |
+#include "webrtc/audio/audio_sink.h" |
#include "webrtc/audio/audio_state.h" |
#include "webrtc/audio/conversion.h" |
#include "webrtc/base/checks.h" |
@@ -201,6 +203,11 @@ 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_; |