Index: webrtc/audio/audio_receive_stream.cc |
diff --git a/webrtc/audio/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc |
index f64d3051e63c3d66eb1ffc09389f3f107c6e2f98..3777174411788d0aeb5de497c1177973b4f73866 100644 |
--- a/webrtc/audio/audio_receive_stream.cc |
+++ b/webrtc/audio/audio_receive_stream.cc |
@@ -219,6 +219,12 @@ void AudioReceiveStream::SetGain(float gain) { |
channel_proxy_->SetChannelOutputVolumeScaling(gain); |
} |
+const std::vector<RtpContributingSource>& |
+AudioReceiveStream::GetContributingSources() { |
+ RTC_DCHECK_RUN_ON(&worker_thread_checker_); |
+ return channel_proxy_->GetContributingSources(); |
+} |
+ |
AudioMixer::Source::AudioFrameInfo AudioReceiveStream::GetAudioFrameWithInfo( |
int sample_rate_hz, |
AudioFrame* audio_frame) { |