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andrew@webrtc.org eaefea665f Remove unused files from src/build. 
These files were intended for a standalone build without depending on
Chromium's build/. In some bright future we might use them, but for
the moment, they're just confusing.

BUG=
TEST=build on Linux

Review URL: https://webrtc-codereview.appspot.com/416002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1800 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-29 18:19:34 +00:00
..
build Remove unused files from src/build. 2012-02-29 18:19:34 +00:00
common_audio Optimized WebRtcSpl_ComplexBitReverse() for general ARM platforms and generic C. 2012-02-23 22:38:56 +00:00
common_video Change WebRtc_Word8 to char 2012-02-24 16:08:52 +00:00
modules Correct wrong usage of WebRtc_Word8 in rtp and udp module 2012-02-29 16:09:51 +00:00
system_wrappers Enable rw_lock_posix on andorid 2012-02-28 23:10:03 +00:00
video_engine Correct wrong usage of WebRtc_Word8 in rtp and udp module 2012-02-29 16:09:51 +00:00
voice_engine Only reset AudioProcessing if number of channels has changed. 2012-02-29 17:00:56 +00:00
common_types.h Initial revision of a ViE fuzz test. The idea is to inject randomized RTP packets and see what the video engine does. 2012-02-17 09:32:48 +00:00
engine_configurations.h remove vie file API to take away media_file and utility modules. 2012-02-08 10:38:12 +00:00
LICENSE Aligning license file with file header 2011-11-02 09:31:39 +00:00
LICENSE_THIRD_PARTY Introduced ARM version of WebRtcSpl_SqrtFloor(). Function cycles reduced by ~ 30% in a real time VOE test in an android device (Nexus-S, ARMv7a). 2012-02-07 17:15:15 +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.