Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(595)

Unified Diff: talk/media/webrtc/fakewebrtccall.h

Issue 1551813002: Storing raw audio sink for default audio track. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Removing obsolete "RefCountedObject" and adding an RTC_DCHECK. Created 4 years, 11 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
« no previous file with comments | « talk/app/webrtc/peerconnectioninterface_unittest.cc ('k') | talk/media/webrtc/webrtcvoiceengine.h » ('j') | no next file with comments »
Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
Index: talk/media/webrtc/fakewebrtccall.h
diff --git a/talk/media/webrtc/fakewebrtccall.h b/talk/media/webrtc/fakewebrtccall.h
index 3528c7a7b1e1e12a2290e71a5e9f5660bd5707f9..ab1e3b6a5202af4265b17eef48d263c25fd7eaa4 100644
--- a/talk/media/webrtc/fakewebrtccall.h
+++ b/talk/media/webrtc/fakewebrtccall.h
@@ -89,6 +89,7 @@ class FakeAudioReceiveStream final : public webrtc::AudioReceiveStream {
void SetStats(const webrtc::AudioReceiveStream::Stats& stats);
int received_packets() const { return received_packets_; }
void IncrementReceivedPackets();
+ const webrtc::AudioSinkInterface* sink() const { return sink_.get(); }
private:
// webrtc::ReceiveStream implementation.
« no previous file with comments | « talk/app/webrtc/peerconnectioninterface_unittest.cc ('k') | talk/media/webrtc/webrtcvoiceengine.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698