webrtc_m130

History

andrew@webrtc.org 587c844741 Query the capture volume immediately on Win Core.

This allows the capture volume to be queried immediately at capture
startup, rather than waiting the usual one second interval.

Review URL: http://webrtc-codereview.appspot.com/297003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1064 4adac7df-926f-26a2-2b94-8c16560cd09d

2011-11-30 17:43:05 +00:00

build

Use clang by default on Mac.

2011-11-30 01:16:06 +00:00

common_audio

Added support for 96 kHz sampling frequency.

2011-11-23 13:50:41 +00:00

common_video

libyuv: Updating paths for test files

2011-11-29 17:50:07 +00:00

modules

Query the capture volume immediately on Win Core.

2011-11-30 17:43:05 +00:00

system_wrappers

Fixes broken unit test http://code.google.com/p/webrtc/issues/detail?id=154

2011-11-29 18:21:46 +00:00

video_engine

Since the CocoaRenderView is forward declared with @class instead of imported,

2011-11-29 23:55:19 +00:00

voice_engine

Removing deprecated NetEQ APIs

2011-11-30 16:21:22 +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

Removing statistics API from NetEQ

2011-11-23 09:36:23 +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.