stefan@webrtc.org 976a7e61c1 Adding support for jointly estimating bandwidth using all streams from the same sending client.
- Broke out the bandwidth estimation from the RTP module.
- Added conversion between RTP and NTP time bases.
- Added unittests.

BUG=

Review URL: https://webrtc-codereview.appspot.com/784009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2798 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-09-21 13:20:21 +00:00
2012-04-19 18:09:23 +00:00
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
BSD-3-Clause 446 MiB
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