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webrtc_m130/webrtc/modules/rtp_rtcp/interface
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stefan@webrtc.org 4ef438e2de Remove the send-side cname getter APIs from voice and video engine.
These APIs aren't being used, and introduces deadlocks when using GetStats() in the new Call api. Having getters for cname at the send-side is pointless, as it's always the user who sets the cname.

R=henrika@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6659 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-11 09:55:30 +00:00
..
fec_receiver.h
Convert logs in rtp rtcp module from WEBRTC_TRACE into LOG.
2014-04-08 11:06:12 +00:00
receive_statistics.h
Add callbacks for receive channel RTP statistics
2014-01-23 10:00:39 +00:00
remote_ntp_time_estimator.h
Move the capture ntp computing code to ntp_calculator so that later it can be shared with voe.
2014-05-14 16:53:51 +00:00
rtp_header_parser.h
Some refactoring inside rtp_rtcp/.
2014-07-08 12:10:51 +00:00
rtp_payload_registry.h
Some refactoring inside rtp_rtcp/.
2014-07-08 12:10:51 +00:00
rtp_receiver.h
Fix for making sure that the packet in order checks are done prior to updating the last received packet state.
2013-11-08 15:18:52 +00:00
rtp_rtcp_defines.h
Preserve RTP states for restarted VideoSendStreams.
2014-07-07 13:06:48 +00:00
rtp_rtcp.h
Remove the send-side cname getter APIs from voice and video engine.
2014-07-11 09:55:30 +00:00
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