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platform-external-webrtc/webrtc
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henrike@webrtc.org cfc07c943f Revert of r3747. 
TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1277005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3752 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-02 14:55:44 +00:00
..
build
Use ninja with merge_libs.py on Windows and clean up.
2013-02-26 02:47:50 +00:00
common_audio
Refactored inline assembly code in complex_fft.c, by combining the individual __asm lines into a single block, to avoid potential register usage problems when building with different tools.
2013-03-01 23:01:14 +00:00
common_video
Fixing/disabling Windows x64 warnings
2013-02-04 10:07:17 +00:00
modules
Revert of r3747.
2013-04-02 14:55:44 +00:00
system_wrappers
Fixes build break in previous cl (https://code.google.com/p/webrtc/source/detail?r=3739) found by Android bots.
2013-03-28 16:31:51 +00:00
test
Add some VoE and AudioProcessing mocks.
2013-03-25 21:20:38 +00:00
tools
Fix frame_editing_unittest reference file handling.
2013-03-11 15:30:33 +00:00
video_engine
Updated Webrtc version to 3.28
2013-04-01 16:17:26 +00:00
voice_engine
Removed CPU APIs from VoEHardware. Code is now only used by test applications.
2013-03-28 15:58:49 +00:00
.gitignore
.gitignore: Add *.mk, created as part of ChromiumOS build
2013-01-04 21:25:42 +00:00
common_types.h
Removed all code enclosed in WEBRTC_SRTP #ifdefs, and the unsupported VoE SRTP APIs. Test stubs are left in place as we still have the (De)RegisterExternalEncryption() APIs, although they are currently untested.
2013-03-28 09:14:36 +00:00
engine_configurations.h
Removed all code enclosed in WEBRTC_SRTP #ifdefs, and the unsupported VoE SRTP APIs. Test stubs are left in place as we still have the (De)RegisterExternalEncryption() APIs, although they are currently untested.
2013-03-28 09:14:36 +00:00
LICENSE
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
LICENSE_THIRD_PARTY
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
PATENTS
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
README.chromium
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
typedefs.h
Add some VoE and AudioProcessing mocks.
2013-03-25 21:20:38 +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.