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stefan@webrtc.org 94a8c03141 Slightly increased bandwidth adaptation at both receive- and send-side. 
The send-side increase factor is increased to better follow the pace
of the receive-side estimate, while the receive-side factor is
increased to speed up adaptation.

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/297002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1030 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-25 14:09:37 +00:00
..
build
Enable WEBRTC_NO_TRACE for Chromium builds.
2011-11-14 09:39:31 +00:00
common_audio
Added support for 96 kHz sampling frequency.
2011-11-23 13:50:41 +00:00
common_video
Fix win compile error for interpolator_test
2011-11-09 08:02:16 +00:00
modules
Slightly increased bandwidth adaptation at both receive- and send-side.
2011-11-25 14:09:37 +00:00
system_wrappers
Fixes linux build error introduced in r980.
2011-11-19 00:14:37 +00:00
video_engine
Refactored ViESharedData.
2011-11-24 15:16:00 +00:00
voice_engine
fixing the racing conditions
2011-11-25 10:58:15 +00:00
common_settings.gypi
git-svn-id: http://webrtc.googlecode.com/svn/trunk@156 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-07-07 08:21:25 +00:00
common_types.h
Removing statistics API from NetEQ
2011-11-23 09:36:23 +00:00
engine_configurations.h
git-svn-id: http://webrtc.googlecode.com/svn/trunk@156 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-07-07 08:21:25 +00:00
LICENSE
Aligning license file with file header
2011-11-02 09:31:39 +00:00
LICENSE_THIRD_PARTY
Add references to src/ copies for LICENSE etc.
2011-10-26 01:05:07 +00:00
PATENTS
Adding copies of license files to src/ so that Chromium will get those as well.
2011-07-14 08:00:33 +00:00
README.chromium
git-svn-id: http://webrtc.googlecode.com/svn/trunk@156 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-07-07 08:21:25 +00:00
typedefs.h
git-svn-id: http://webrtc.googlecode.com/svn/trunk@785 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-10-20 12:30:35 +00:00

README.chromium 

Name: WebRTC
URL: http://www.webrtc.org
Version: 90
License: BSD
License File: LICENSE

Description:
WebRTC provides real time voice and video processing
functionality to enable the implementation of 
PeerConnection/MediaStream.

Third party code used in this project is described 
in the file LICENSE_THIRD_PARTY.