wu@webrtc.org 6c75c98964 Propagate capture ntp timestamp from rtp to renderer. ...

Mostly the interface changes, the real implementation of ntp timestamp will come in a follow up cl.

TEST=new tests and try bots
BUG=3111
R=niklas.enbom@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5911 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-04-15 17:46:33 +00:00

..

vie_base.h

Add configuration for ability to use the encode usage measure for triggering overuse/underuse.

2014-03-24 21:59:16 +00:00

vie_capture.h

Add SwapFrame() to VideoSendStreamInput.

2013-12-11 16:26:16 +00:00

vie_codec.h

Add API to query video engine for the send-side delay.

2013-12-05 14:05:07 +00:00

vie_errors.h

Remove ViE external encryption API.

2014-02-11 15:27:49 +00:00

vie_external_codec.h

Include files from webrtc/.. paths in video_engine/

2013-05-17 13:44:48 +00:00

vie_image_process.h

Propagate capture ntp timestamp from rtp to renderer.

2014-04-15 17:46:33 +00:00

vie_network.h

Adding API for setting bandwidth estimation configurations.

2014-03-25 10:37:31 +00:00

vie_render.h

Propagate capture ntp timestamp from rtp to renderer.

2014-04-15 17:46:33 +00:00

vie_rtp_rtcp.h

Add API to allow deducting bitrate from incoming estimates before the capacity is distributed among outgoing video streams. For example, this can be used to reserve space for audio streams.

2014-03-26 14:32:47 +00:00