webrtc_m130

History

pbos@webrtc.org 72491b9a90 Count total bytes sent in RTPSender::Bytes().

Previously only media bytes were included, this adds header bytes and
padding bytes to the calculation.

BUG=
R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6654 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-07-10 16:24:54 +00:00

interface

Refactor registerable callbacks for FrameCountObserver from rtp_rtcp module into vie_channel.

2014-07-10 09:39:23 +00:00

mocks

Preserve RTP states for restarted VideoSendStreams.

2014-07-07 13:06:48 +00:00

source

Count total bytes sent in RTPSender::Bytes().

2014-07-10 16:24:54 +00:00

test

Some refactoring inside rtp_rtcp/.