| 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_;
|
|
|