webrtc_m130

History

leozwang@webrtc.org 430e31c2c0 Change VQE settings

Change some VQE settings and make iSAC (item 0 in the list) to be the
default

BUG=None
TEST=local
Review URL: https://webrtc-codereview.appspot.com/790005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2754 4adac7df-926f-26a2-2b94-8c16560cd09d

2012-09-12 03:47:08 +00:00

build

Change MAC_IPHONE to WEBRTC_IOS.

2012-09-11 17:25:46 +00:00

common_audio

This CL is WebRTC VoiceEngine for iOS and is from CL713004.

2012-09-10 17:58:21 +00:00

common_video

Consolidate common_video targets to improve gyp run time.

2012-09-11 01:34:21 +00:00

modules

Added calling WebRtcSpl_Init() in fixed point NS, and removed it from floating point NS.

2012-09-12 00:23:40 +00:00

system_wrappers

Change MAC_IPHONE to WEBRTC_IOS.

2012-09-11 17:25:46 +00:00

test

Consolidate common_video targets to improve gyp run time.

2012-09-11 01:34:21 +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

Change VQE settings

2012-09-12 03:47:08 +00:00

voice_engine

Change MAC_IPHONE to WEBRTC_IOS.

2012-09-11 17:25:46 +00:00

common_types.h

Tune for faster ramp-up.

2012-09-11 14:11:58 +00:00

engine_configurations.h

Change MAC_IPHONE to WEBRTC_IOS.

2012-09-11 17:25:46 +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.