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c891fee7aba4e6bcc33f6e03ec9e7f3a2940e03c
platform-external-webrtc/webrtc/modules/rtp_rtcp
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fbarchard@google.com c891fee7ab Make a int64 constant use ULL suffix so it wont get truncated.
BUG=3690
TESTED=try bots
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15139004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6878 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-12 22:39:06 +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
Make a int64 constant use ULL suffix so it wont get truncated.
2014-08-12 22:39:06 +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
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