platform-external-webrtc

Files

phoglund@webrtc.org a22a9bd9ca Cleaned up the data path for payload data, made callbacks to rtp_receiver nonoptional.

The audio receiver is now completely independent of rtp_receiver: video will hopefully be too in the next patch.

BUG=
TEST=vie & voe_auto_test full runs

Review URL: https://webrtc-codereview.appspot.com/1014006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3372 4adac7df-926f-26a2-2b94-8c16560cd09d

2013-01-14 10:01:55 +00:00

build

Fix android clang build.

2013-01-12 01:52:07 +00:00

common_audio

Address a build issue with Android-Clang compiler:

2013-01-10 00:46:37 +00:00

common_video

Making I420VideoFrame ref-counted

2012-12-19 00:07:57 +00:00

modules

Cleaned up the data path for payload data, made callbacks to rtp_receiver nonoptional.

2013-01-14 10:01:55 +00:00

system_wrappers

Reformatted tick_util.

2013-01-03 09:50:17 +00:00

test

Generalized mechanism for excluding gtests on platforms, disabled broken tests on mac.

2013-01-09 16:53:42 +00:00

tools

Added possibility to repeat frames. Also added unittest for that feature.

2012-12-20 11:42:45 +00:00

video_engine

logical 'and' of mutually exclusive tests is always false in ViECodecImpl::CodecValid()

2013-01-14 01:52:26 +00:00

voice_engine

Fix android clang build.

2013-01-12 01:09:03 +00:00

.gitignore

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

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

common_types.h

Add number of inserted samples to NetEq statistics.

2012-12-14 00:06:18 +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.