webrtc

Files

andrew@webrtc.org 828af1b4b9 Add lookahead to the delay estimator.

TEST=audioproc_unittest

Review URL: http://webrtc-codereview.appspot.com/279014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@992 4adac7df-926f-26a2-2b94-8c16560cd09d

2011-11-22 22:40:27 +00:00

build

Enable WEBRTC_NO_TRACE for Chromium builds.

2011-11-14 09:39:31 +00:00

common_audio

Restructuring and adding unit test targets on project level instead of in common_audio.

2011-11-17 08:35:47 +00:00

common_video

Fix win compile error for interpolator_test

2011-11-09 08:02:16 +00:00

modules

Add lookahead to the delay estimator.

2011-11-22 22:40:27 +00:00

system_wrappers

Fixes linux build error introduced in r980.

2011-11-19 00:14:37 +00:00

video_engine

Move ViE main/test/AutoTest to test/auto_test.

2011-11-22 14:03:15 +00:00

voice_engine

Addressed review comments from http://webrtc-codereview.appspot.com/256004/

2011-11-18 19:59:32 +00:00

common_settings.gypi

git-svn-id: http://webrtc.googlecode.com/svn/trunk@156 4adac7df-926f-26a2-2b94-8c16560cd09d

2011-07-07 08:21:25 +00:00

common_types.h

First implementation of simulcast, adds VP8 simulcast to video engine.

2011-10-13 15:19:55 +00:00

engine_configurations.h

git-svn-id: http://webrtc.googlecode.com/svn/trunk@156 4adac7df-926f-26a2-2b94-8c16560cd09d

2011-07-07 08:21:25 +00:00

LICENSE

Aligning license file with file header

2011-11-02 09:31:39 +00:00

LICENSE_THIRD_PARTY

Add references to src/ copies for LICENSE etc.

2011-10-26 01:05:07 +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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@785 4adac7df-926f-26a2-2b94-8c16560cd09d

2011-10-20 12:30:35 +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.