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Unified Diff: webrtc/audio/audio_receive_stream.cc

Issue 1551813002: Storing raw audio sink for default audio track. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Adding unit tests for SetRawAudioSink. Created 4 years, 11 months ago
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Index: webrtc/audio/audio_receive_stream.cc
diff --git a/webrtc/audio/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc
index dfad79f9d76420460267f2fdd0e1265a928b8786..7fe65ffbd7bc63a248b766d8172b0361fc9e5af3 100644
--- a/webrtc/audio/audio_receive_stream.cc
+++ b/webrtc/audio/audio_receive_stream.cc
@@ -203,9 +203,9 @@ webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {
return stats;
}
-void AudioReceiveStream::SetSink(rtc::scoped_ptr<AudioSinkInterface> sink) {
+void AudioReceiveStream::SetSink(rtc::scoped_refptr<AudioSinkInterface> sink) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
- channel_proxy_->SetSink(std::move(sink));
+ channel_proxy_->SetSink(sink);
}
const webrtc::AudioReceiveStream::Config& AudioReceiveStream::config() const {

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