webrtc_m130

History

mflodman@webrtc.org 2f225cadde Add logs when no RTCP RR has been received for three regular RTCP intervals.

BUG=1267
TEST=Unittest added.

Review URL: https://webrtc-codereview.appspot.com/1019006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3346 4adac7df-926f-26a2-2b94-8c16560cd09d

2013-01-09 13:54:43 +00:00

build

Replace the last occurrence of .s with .h

2012-12-05 07:12:15 +00:00

common_audio

Porting ARM optimization from Android to ios.

2012-11-17 00:22:46 +00:00

common_video

Making I420VideoFrame ref-counted

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

modules

Add logs when no RTCP RR has been received for three regular RTCP intervals.

2013-01-09 13:54:43 +00:00

system_wrappers

Reformatted tick_util.

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

test

Using Convert in lieu of ExtractBuffer: Less error prone (as we don't need to compute buffer sizes etc.). This cl is first in a series (doing all of WebRtc would make it quite a big cl). While at it, fixing a few headers.

2013-01-08 19:19:59 +00:00

tools

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

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

video_engine

Disabled GQoS since it breaks ViE auto test.

2013-01-09 09:13:00 +00:00

voice_engine

Change Sleep() comment in test fixture.

2013-01-05 03:30:11 +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.