webrtc_m130

History

marpan@webrtc.org 993494764d Added metrics test code for the FEC packet masks.

The test computes metrics (average residual loss) for each mask type and size, 
for a given set of loss models (random and bursty), and verifies various 
behaviour of the codes (including relation/comparison to RS code).
Review URL: https://webrtc-codereview.appspot.com/748008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3185 4adac7df-926f-26a2-2b94-8c16560cd09d

2012-11-28 19:43:58 +00:00

build

Add a logging_no_op.cc when enable_tracing==0.

2012-11-23 19:30:59 +00:00

common_audio

Porting ARM optimization from Android to ios.

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

common_video

Adding an aligned stride test to LibYuv

2012-11-20 23:52:50 +00:00

modules

Added metrics test code for the FEC packet masks.

2012-11-28 19:43:58 +00:00

system_wrappers

Add a logging_no_op.cc when enable_tracing==0.

2012-11-23 19:30:59 +00:00

test

Added conformance tests.

2012-11-28 13:03:17 +00:00

tools

Tool for editing of yuv-files. Specify a path to the clip that should be edited, the height and width of the clip, one set of frames that should be removed from the clip, and a path to where the result should be written. There is a executable created that make use of the library where the functionality is implemented. There is also a unittest added for the library.

2012-11-27 13:44:07 +00:00

video_engine

Remove ringtone from test app

2012-11-28 19:39:23 +00:00

voice_engine

Added last (?) suppressions for known issues.

2012-11-28 14:21:12 +00:00

common_types.h

Add a kTraceTerseInfo level for non-verbose logging.

2012-11-20 07:34:45 +00:00

engine_configurations.h

Move src/ -> webrtc/

2012-10-22 18:19:23 +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.