webrtc

History

hclam@chromium.org 81cffd1f2c Port Chromium's atomicops to WebRTC

Porting Chromium's base/atomicops.h and friends into WebRTC.

Included the original unit test in Chromium.
Review URL: https://webrtc-codereview.appspot.com/964026

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3241 4adac7df-926f-26a2-2b94-8c16560cd09d

2012-12-05 22:51:01 +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

Port Chromium's atomicops to WebRTC

2012-12-05 22:51:01 +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.