Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(361)

Unified Diff: webrtc/modules/rtp_rtcp/include/rtp_rtcp.h

Issue 2061423003: Refactor NACK bitrate allocation (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Fixed data race Created 4 years, 6 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: webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
diff --git a/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h b/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
index 7c72e5917c8ab2feb8acdac1c119cd3e25b8b5c6..94ab763a26b1f8e00c999b61987f5e5bc35c0e70 100644
--- a/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
+++ b/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
@@ -286,7 +286,7 @@ class RtpRtcp : public Module {
virtual void BitrateSent(uint32_t* totalRate,
uint32_t* videoRate,
uint32_t* fecRate,
- uint32_t* nackRate) const = 0;
danilchap 2016/06/20 16:53:46 this would break downstream code.
sprang_webrtc 2016/06/23 07:54:29 Acknowledged.
+ uint32_t* nackRate) = 0;
/*
* Used by the codec module to deliver a video or audio frame for

Powered by Google App Engine
This is Rietveld 408576698