platform-external-webrtc

Files

wjia@webrtc.org df94329a4b Use different cpufeatures library when building with chrome.

There are more than one target when building with chrome. They have different build setup.
This patch just puts content of build/android/cpufeatures.gypi inside system_wrappers.gyp.
In the future, if more modules will use cpufeatures lib, we can move the code into a gypi file.
Review URL: https://webrtc-codereview.appspot.com/939030

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3242 4adac7df-926f-26a2-2b94-8c16560cd09d

2012-12-05 22:54:52 +00:00

build

Replace the last occurrence of .s with .h

2012-12-05 07:12:15 +00:00

common_audio

Porting ARM optimization from Android to ios.

2012-11-17 00:22:46 +00:00

common_video

Fixed standard PSNR/SSIM test.

2012-11-29 10:08:16 +00:00

modules

Dual-stream implementation, not including VoE APIs.

2012-12-03 22:13:31 +00:00

system_wrappers

Use different cpufeatures library when building with chrome.

2012-12-05 22:54:52 +00:00

test

Removed stale version of fuzzer; it's now internal.

2012-12-03 14:28:40 +00:00

tools

Fixed bug that caused frame_cutter_unittest to fail when built with MVS2008.

2012-11-30 12:37:14 +00:00

video_engine

Added auto-call feature to WebRTCDemo.

2012-12-04 16:53:43 +00:00

voice_engine

Expose Set and Get Recording/Playout sample rate apis

2012-12-04 19:11:55 +00:00

common_types.h

Add a kTraceTerseInfo level for non-verbose logging.

2012-11-20 07:34:45 +00:00

engine_configurations.h

First pass of MediaCodecDecoder which uses Android MediaCodec API.

2012-12-04 06:38:19 +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

Move src/ -> webrtc/

2012-10-22 18:19:23 +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.