webrtc

Files

andrew@webrtc.org c732ca697b Return audio_processing_tests.gypi to module.gyp.

TBR=xians

Review URL: https://webrtc-codereview.appspot.com/781005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2728 4adac7df-926f-26a2-2b94-8c16560cd09d

2012-09-10 16:33:43 +00:00

build

Remove WEBRTC_MAC_INTEL.

2012-09-06 18:17:00 +00:00

common_audio

Changed a flag in ARM build for platform build.

2012-09-08 00:32:59 +00:00

common_video

Removing RawImage.

2012-08-02 22:18:47 +00:00

modules

Return audio_processing_tests.gypi to module.gyp.

2012-09-10 16:33:43 +00:00

system_wrappers

Remove WEBRTC_MAC_INTEL.

2012-09-06 18:17:00 +00:00

test

Moving frame_analyzer and rgba_to_i420_converter to src/tools.

2012-08-22 08:12:00 +00:00

tools

A small tool to run PSNR and SSIM between two I420 videos.

2012-08-27 14:57:19 +00:00

video_engine

Adding test for relaying all simulcast streams to different receive channels.

2012-09-10 13:27:47 +00:00

voice_engine

Added run time ARM-Neon detection feature in SPL functions.

2012-09-08 00:09:26 +00:00

common_types.h

Makes it possible to disable frame dropping in the VP8 codec.

2012-08-24 23:01:38 +00:00

engine_configurations.h

Change preprocessor define WEBRTC_ANDROID into WEBRTC_ARCH_ARM, for ARM platform depended code.

2012-08-28 04:09:50 +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

Add WEBRTC_POSIX and remove unneeded defines.

2012-08-31 23:17:53 +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.