webrtc_m130

History

mflodman@webrtc.org ad6083f414 Added condition for which jpeg lib to use.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/638004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2357 4adac7df-926f-26a2-2b94-8c16560cd09d

2012-06-04 19:10:43 +00:00

build

Refactored ARM specific code in Noise Suppression. Bit exact.

2012-05-26 01:05:27 +00:00

common_audio

VAD Refactoring: Replaced pointer operation with array index

2012-05-28 12:50:05 +00:00

common_video

Added condition for which jpeg lib to use.

2012-06-04 19:10:43 +00:00

modules

ACM PCM16B, fixing a copy-and-paste error.

2012-06-04 11:47:49 +00:00

system_wrappers

Added gyp variable to include/exclude all tests.

2012-05-24 13:23:35 +00:00

test/libtest

Added a bit flip fuzz test to the voice engine.

2012-05-18 08:02:37 +00:00

video_engine

Fix a bug where a RAII object was created for just one line instead of a block.

2012-05-28 18:56:20 +00:00

voice_engine

In the past we support calling StartPlayingFileLocally() before StartPlayout(), then when playout is started, the file would be played out immediately.

2012-06-04 03:26:39 +00:00

common_types.h

Add option to configure error concealment and disable by default.

2012-05-30 11:25:00 +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

Typedef WebRtc_word8 to int8_t

2012-03-12 16:52:28 +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.