webrtc_m130

History

andresp@webrtc.org c3c29113d1 Expose setPayloadType on the rtp_sender. Thus allowing other users of this module

to set the payload type to be used without having to call SendOutgoingData.

BUG=3694
R=asapersson@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6988 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-08-27 09:39:43 +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

Expose setPayloadType on the rtp_sender. Thus allowing other users of this module

2014-08-27 09:39:43 +00:00

test

Remove the send-side cname getter APIs from voice and video engine.