Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(82)

Unified Diff: webrtc/api/api.gyp

Issue 1788583004: Enable setting the maximum bitrate limit in RtpSender. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Removed support for bitrate limits for audio streams; corrected code review issues Created 4 years, 9 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 | « no previous file | webrtc/api/mediastreamprovider.h » ('j') | webrtc/media/engine/webrtcvideoengine2.h » ('J')
Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
Index: webrtc/api/api.gyp
diff --git a/webrtc/api/api.gyp b/webrtc/api/api.gyp
index d97c38b50c6fdc53ca4876ebc3495a0f2e1d09f0..7951089501e738bda7f33b9ce03b00b3f38b3558 100644
--- a/webrtc/api/api.gyp
+++ b/webrtc/api/api.gyp
@@ -294,6 +294,7 @@
'remoteaudiosource.h',
'remotevideocapturer.cc',
'remotevideocapturer.h',
+ 'rtpparameters.h',
'rtpreceiver.cc',
'rtpreceiver.h',
'rtpreceiverinterface.h',
« no previous file with comments | « no previous file | webrtc/api/mediastreamprovider.h » ('j') | webrtc/media/engine/webrtcvideoengine2.h » ('J')

Powered by Google App Engine
This is Rietveld 408576698