webrtc

History

bjornv@webrtc.org cb0a86e913 VAD refactoring: Added function for repeated code.

Added WeightedAverage() to calculate global mean. This removes hard coded Gaussian model size and repeated code.

Tested with vad_unittests, audioproc_unittest and trybots.

BUG=None
TEST=None

Review URL: https://webrtc-codereview.appspot.com/571006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2275 4adac7df-926f-26a2-2b94-8c16560cd09d

2012-05-23 07:56:51 +00:00

build

Changed merge-name.

2012-05-22 08:32:20 +00:00

common_audio

VAD refactoring: Added function for repeated code.

2012-05-23 07:56:51 +00:00

common_video

Reduce PSNR because I420ToARGB888 return lowers number on windows

2012-05-04 21:31:22 +00:00

modules

FEC: For multi-frame FEC, allow for the size of the code to be increased,

2012-05-22 16:50:00 +00:00

system_wrappers

Fixing memory leak error in test.

2012-05-22 16:55:16 +00:00

test/libtest

Added a bit flip fuzz test to the voice engine.

2012-05-18 08:02:37 +00:00

video_engine

Cleanup code.

2012-05-22 14:23:25 +00:00

voice_engine

Avoid flakiness by skipping more output verification.

2012-05-23 00:45:00 +00:00

common_types.h

Upgrade libvpx to 6b66c01 and enabling temporal denoising.

2012-03-21 16:52:03 +00:00

engine_configurations.h

Enable iSAC_FIX on android

2012-04-18 16:52:00 +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

Typedef WebRtc_word8 to int8_t

2012-03-12 16:52:28 +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.