webrtc

History

leozwang@webrtc.org 6c08f26c4e Terminate version string

This cl doesn't directly solve b/6750185, but it's a potential bug
if string is not terminated correctly

BUG=
TEST=vie_auto_test
Review URL: https://webrtc-codereview.appspot.com/674009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2515 4adac7df-926f-26a2-2b94-8c16560cd09d

2012-07-13 22:00:16 +00:00

build

Re-enable WEBRTC_SVNREVISION script

2012-07-10 20:36:29 +00:00

common_audio

Move test to src/test.

2012-06-27 01:41:54 +00:00

common_video

1. Adding odd size support to LibYuv wrapper.

2012-07-10 20:48:48 +00:00

modules

Add the FEC mask type to FecProtectionParams and set the mask type in the VCM.

2012-07-13 16:27:51 +00:00

system_wrappers

Move test to src/test.

2012-06-27 01:41:54 +00:00

test

Improved fuzzer. It will now throw in additional refreshes, which is known to mess with lifetime assumptions.

2012-07-04 12:29:09 +00:00

video_engine

Terminate version string

2012-07-13 22:00:16 +00:00

voice_engine

Pass capture time (wallclock) to the RTP sender to compute transmission offset

2012-07-03 13:21:22 +00:00

common_types.h

Expose a set of options to the OveruseDetector supporting experiments

2012-06-26 10:47:04 +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

[MIPS] Adding support for MIPS architecture for WebRTC.

2012-06-27 22:24:43 +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.