Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(944)

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

Issue 1551813002: Storing raw audio sink for default audio track. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Adding back "if (default_sink_)" when creating default stream. 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
Index: talk/media/webrtc/webrtcvoiceengine.h
diff --git a/talk/media/webrtc/webrtcvoiceengine.h b/talk/media/webrtc/webrtcvoiceengine.h
index 0f2f59e4928f0ea619adb5e7088fc3b524388bf5..88115e102573708982fb5ef66093ea8df91546c4 100644
--- a/talk/media/webrtc/webrtcvoiceengine.h
+++ b/talk/media/webrtc/webrtcvoiceengine.h
@@ -269,6 +269,8 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
int64_t default_recv_ssrc_ = -1;
// Volume for unsignalled stream, which may be set before the stream exists.
double default_recv_volume_ = 1.0;
+ // Sink for unsignalled stream, which may be set before the stream exists.
+ rtc::scoped_ptr<webrtc::AudioSinkInterface> default_sink_;
// Default SSRC to use for RTCP receiver reports in case of no signaled
// send streams. See: https://code.google.com/p/webrtc/issues/detail?id=4740
// and https://code.google.com/p/chromium/issues/detail?id=547661

Powered by Google App Engine
This is Rietveld 408576698