webrtc

History

henrike@webrtc.org f3760dc8e9 Fixes coverity warning that I missed in system wrappers.

BUG=Coverity
TEST=N/A

Review URL: https://webrtc-codereview.appspot.com/395005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1717 4adac7df-926f-26a2-2b94-8c16560cd09d

2012-02-17 16:27:25 +00:00

build

Removed warnings on Windows and enabled warnings-as-errors on Windows.

2012-02-07 14:56:45 +00:00

common_audio

Optimized function WebRtcSpl_DownsampleFast for ARM-NEON platform.

2012-02-07 18:03:11 +00:00

common_video

Remove common_settings.gypi.

2012-02-16 01:16:28 +00:00

modules

Disabled RTPModule VP8 packetizer assert.

2012-02-17 13:07:01 +00:00

system_wrappers

Fixes coverity warning that I missed in system wrappers.

2012-02-17 16:27:25 +00:00

video_engine

Added a retry mechanism to vie_auto_test's verifying tests to make them less flaky.

2012-02-17 13:08:57 +00:00

voice_engine

Update Android.mk and add test app

2012-02-17 01:16:43 +00:00

common_types.h

Initial revision of a ViE fuzz test. The idea is to inject randomized RTP packets and see what the video engine does.

2012-02-17 09:32:48 +00:00

engine_configurations.h

remove vie file API to take away media_file and utility modules.

2012-02-08 10:38:12 +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

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.