webrtc_m130

History

marpan@webrtc.org 95b48c3551 Update to codec unit test:

enable frame dropper for rate control test.
Review URL: https://webrtc-codereview.appspot.com/1099014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3536 4adac7df-926f-26a2-2b94-8c16560cd09d

2013-02-19 20:02:32 +00:00

build

Fixing/disabling Windows x64 warnings

2013-02-04 10:07:17 +00:00

common_audio

Modify SincResampler to build in webrtc.

2013-02-15 03:54:22 +00:00

common_video

Fixing/disabling Windows x64 warnings

2013-02-04 10:07:17 +00:00

modules

Update to codec unit test:

2013-02-19 20:02:32 +00:00

system_wrappers

Modify SincResampler to build in webrtc.

2013-02-15 03:54:22 +00:00

test

Fix Windows x64 errors in video_codecs_test_framework

2013-02-13 09:35:12 +00:00

tools

Fixing/disabling Windows x64 warnings

2013-02-04 10:07:17 +00:00

video_engine

fixing nack list size calculation

2013-02-19 19:42:35 +00:00

voice_engine

Make VoiceEngineImpl inherit from VoiceEngine.

2013-02-15 15:07:32 +00:00

.gitignore

.gitignore: Add *.mk, created as part of ChromiumOS build

2013-01-04 21:25:42 +00:00

common_types.h

Refactoring temporal layers implementation and adding VideoCodecMode for easier control of codec settings.

2013-02-18 14:40:18 +00:00

engine_configurations.h

First pass of MediaCodecDecoder which uses Android MediaCodec API.

2012-12-04 06:38:19 +00:00

LICENSE

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

LICENSE_THIRD_PARTY

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

PATENTS

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

README.chromium

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

typedefs.h

Move src/ -> webrtc/

2012-10-22 18:19:23 +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.