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Unified Diff: talk/app/webrtc/webrtcsession.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: talk/app/webrtc/webrtcsession.cc
diff --git a/talk/app/webrtc/webrtcsession.cc b/talk/app/webrtc/webrtcsession.cc
index 29a4f3394a32c8674d272aa7b007bc0053066c52..8d4cd9ae557287b65f66dc9d0f95d598db677931 100644
--- a/talk/app/webrtc/webrtcsession.cc
+++ b/talk/app/webrtc/webrtcsession.cc
@@ -1320,13 +1320,14 @@ void WebRtcSession::SetAudioPlayoutVolume(uint32_t ssrc, double volume) {
}
}
-void WebRtcSession::SetRawAudioSink(uint32_t ssrc,
- rtc::scoped_ptr<AudioSinkInterface> sink) {
+void WebRtcSession::SetRawAudioSink(
+ uint32_t ssrc,
+ rtc::scoped_refptr<AudioSinkInterface> sink) {
ASSERT(signaling_thread()->IsCurrent());
if (!voice_channel_)
return;
- voice_channel_->SetRawAudioSink(ssrc, std::move(sink));
+ voice_channel_->SetRawAudioSink(ssrc, sink);
}
bool WebRtcSession::SetCaptureDevice(uint32_t ssrc,

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