webrtc_m130

History

kma@webrtc.org bfa7f96d1e Optimized WebRtcSpl_ComplexBitReverse() for general ARM platforms and generic C.

In ARMv5, the cycles were reduced by 88% (weight in VoE reduced from 3.554% to 0.432%). The tradeoff is a memory increase of 704 bytes.
Review URL: https://webrtc-codereview.appspot.com/388003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1757 4adac7df-926f-26a2-2b94-8c16560cd09d

2012-02-23 22:38:56 +00:00

build

Add android target

2012-02-22 16:05:19 +00:00

common_audio

Optimized WebRtcSpl_ComplexBitReverse() for general ARM platforms and generic C.

2012-02-23 22:38:56 +00:00

common_video

Reverting r1749.

2012-02-23 10:15:04 +00:00

modules

Change default packetization mode to an equal size mode.

2012-02-23 18:52:53 +00:00

system_wrappers

Fixes coverity warning that I missed in system wrappers.

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

video_engine

Remove deleted file from vie_auto_test.gypi.

2012-02-23 22:05:24 +00:00

voice_engine

Rewrote the call report test.

2012-02-20 08:55:04 +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.