webrtc

History

henrik.lundin@webrtc.org b6e58eb5a1 Fix formatting of rtp_format_vp8*

Sorting out all lint issues and fixing indentation.

Review URL: http://webrtc-codereview.appspot.com/301011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1111 4adac7df-926f-26a2-2b94-8c16560cd09d

2011-12-06 15:56:18 +00:00

build

Make protobuf use optional.

2011-12-03 00:03:31 +00:00

common_audio

Made tables static.

2011-12-01 15:40:50 +00:00

common_video

libyuv unit test: adding check for fread return value

2011-12-06 00:44:16 +00:00

modules

Fix formatting of rtp_format_vp8*

2011-12-06 15:56:18 +00:00

system_wrappers

For Android ARMv7 platforms, added a feature of dynamically detecting the existence of Neon,

2011-12-03 18:34:50 +00:00

video_engine

Fixed vie_auto_test shutdown race conditions.

2011-12-06 10:42:05 +00:00

voice_engine

Fixes two minor issues reported by the Coverty Integration Manager.

2011-12-05 07:02:22 +00:00

common_settings.gypi

git-svn-id: http://webrtc.googlecode.com/svn/trunk@156 4adac7df-926f-26a2-2b94-8c16560cd09d

2011-07-07 08:21:25 +00:00

common_types.h

Fixes to temporal layers, Henrika please review src/common_types.h

2011-12-02 11:31:08 +00:00

engine_configurations.h

To be able to get webrtc into chrome, we need to reduce the size of the binary and the usage of memory.

2011-11-30 15:35:44 +00:00

LICENSE

Aligning license file with file header

2011-11-02 09:31:39 +00:00

LICENSE_THIRD_PARTY

Add references to src/ copies for LICENSE etc.

2011-10-26 01:05:07 +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.