Logo
Explore Help
Register Sign In
admin/webrtc_m130
1
0
Fork 0
You've already forked webrtc_m130
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/webrtc/modules/rtp_rtcp
History
stefan@webrtc.org c27543d297 Fix STAP-A bug where we might overflow the packet buffer due to not accounting for the length of the length field.
BUG=3679
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6881 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-13 07:40:45 +00:00
..
interface
Fix issue where padding is sent before media with undefined timestamps if not abs-send-time is enabled.
2014-07-17 16:10:14 +00:00
mocks
Add H.264 packetization.
2014-07-31 14:59:24 +00:00
source
Fix STAP-A bug where we might overflow the packet buffer due to not accounting for the length of the length field.
2014-08-13 07:40:45 +00:00
test
Remove the send-side cname getter APIs from voice and video engine.
2014-07-11 09:55:30 +00:00
BUILD.gn
Fix mistake in rtp/rtcp/BUILD.gn introduced with r6804.
2014-07-31 15:07:59 +00:00
OWNERS
GN: Add BUILD.gn files + kjellander to OWNERS
2014-06-23 19:21:07 +00:00
Powered by Gitea Version: 1.23.5 Page: 123ms Template: 3ms
English
Bahasa Indonesia Deutsch English Español Français Gaeilge Italiano Latviešu Magyar nyelv Nederlands Polski Português de Portugal Português do Brasil Suomi Svenska Türkçe Čeština Ελληνικά Български Русский Українська فارسی മലയാളം 日本語 简体中文 繁體中文(台灣) 繁體中文(香港) 한국어
Licenses API