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marpan@google.com 191b780741 Added a correction factor to FEC overhead in media_opt_util. 
This is too handle cases of rate-mismatch (at low rates/low packet number) between estimate in mediaOpt and actual FEC generated in RTP.
Review URL: http://webrtc-codereview.appspot.com/93012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@284 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-08-01 19:59:57 +00:00
..
build
Revert an accidental change to prefer_fixed_point.
2011-07-30 01:33:22 +00:00
common_audio
Clean up ANDROID macro definitions [common_audio]
2011-07-27 18:58:33 +00:00
common_video
common_video: Removing compiler warnings
2011-07-29 20:23:01 +00:00
modules
Added a correction factor to FEC overhead in media_opt_util.
2011-08-01 19:59:57 +00:00
system_wrappers
Rename all files/classes shared by Linux and Mac to Posix.
2011-08-01 17:04:04 +00:00
video_engine
Correct parameters in WEBRTC_TRACE functions.
2011-07-29 22:55:52 +00:00
voice_engine
Replace ANDROID with WEBRTC_ANDROID
2011-08-01 17:36:09 +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
git-svn-id: http://webrtc.googlecode.com/svn/trunk@156 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-07-07 08:21:25 +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
Adding copies of license files to src/ so that Chromium will get those as well.
2011-07-14 08:00:33 +00:00
LICENSE_THIRD_PARTY
Adding copies of license files to src/ so that Chromium will get those as well.
2011-07-14 08:00:33 +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@156 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-07-07 08:21:25 +00:00
video_engine.gyp
git-svn-id: http://webrtc.googlecode.com/svn/trunk@156 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-07-07 08:21:25 +00:00
voice_engine.gyp
Ensures that all test files in VoE and ADM are read from
2011-07-07 14:10:34 +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.