webrtc_m130

History

kma@webrtc.org 0af0d3d3f4 Address a build issue with Android-Clang compiler:

error: the value is truncated when put into register, use a modifier to specify the size [-Werror,-Wasm-operand-widths]
  __asm __volatile ("ssat %0, #16, %1" : "=r"(out16) : "r"(value32));
Review URL: https://webrtc-codereview.appspot.com/1029006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3352 4adac7df-926f-26a2-2b94-8c16560cd09d

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

build

Replace the last occurrence of .s with .h

2012-12-05 07:12:15 +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

Rounding error fix in media_opt_util.

2013-01-09 22:13:19 +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

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

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

voice_engine

Use %d for signed value in trace.

2013-01-09 17:11:50 +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.