webrtc_m130

History

andrew@webrtc.org b2b628d5cd Further relax thresholds in mixing test.

TBR=mikhal

Review URL: https://webrtc-codereview.appspot.com/1019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3327 4adac7df-926f-26a2-2b94-8c16560cd09d

2013-01-02 18:50:13 +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

Making I420VideoFrame ref-counted

2012-12-19 00:07:57 +00:00

modules

Replace voice engine utility functions with system wrapper variants.

2013-01-02 16:06:39 +00:00

system_wrappers

Reformatted thread and static_instance.

2013-01-02 08:45:03 +00:00

test

Will now only require near-perfect PSNR and SSIM.

2012-12-13 10:15:06 +00:00

tools

Added possibility to repeat frames. Also added unittest for that feature.

2012-12-20 11:42:45 +00:00

video_engine

Changed assert to log.

2012-12-20 09:26:17 +00:00

voice_engine

Further relax thresholds in mixing test.

2013-01-02 18:50:13 +00:00

common_types.h

Add number of inserted samples to NetEq statistics.

2012-12-14 00:06:18 +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.