webrtc_m130

History

henrik.lundin@webrtc.org 5dedd0ee38 Handling of white-space in DataLog::Combine

The Combine method cannot handle white-space. Adding a comment to
the header file saying this, and modifying the unittests. Also,
adding a new unittest to test the method.

Review URL: http://webrtc-codereview.appspot.com/217001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@760 4adac7df-926f-26a2-2b94-8c16560cd09d

2011-10-18 05:45:08 +00:00

build

Move merge_libs targets to their own gyp.

2011-10-06 01:03:18 +00:00

common_audio

Removed the vad_const files and added the constants to the files where they are

2011-10-14 12:58:34 +00:00

common_video

vplib tests: Removing old and unused file and directories.

2011-10-06 22:57:06 +00:00

modules

When _audioConfigured, should not try to use the _video.

2011-10-17 21:40:32 +00:00

system_wrappers

Handling of white-space in DataLog::Combine

2011-10-18 05:45:08 +00:00

video_engine

vie_auto_test - moved custom call specific functions to be static, added video protect method to custom call

2011-10-17 21:57:08 +00:00

voice_engine

Added AEC delay logging metrics to VoE. Echo metrics and delay logging metrics are enabled simultaneously through the SetEcMetricsStatus(). Updated standard and extended VoE tests.

2011-10-17 08:49:23 +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

Adding copies of license files to src/ so that Chromium will get those as well.

2011-07-14 08:00:33 +00:00

LICENSE_THIRD_PARTY

Adding copies of license files to src/ so that Chromium will get those as well.

2011-07-14 08:00:33 +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 missing intrinsic casts for VS 2005.

2011-09-19 18:48:25 +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.