platform-external-webrtc

Files

bjornv@webrtc.org 3e10249f20 Added delay estimation test to audio processing unit tests.

The test verifies that we get proper delay metrics when inserting delayed versions of the same file to far-end and near-end.
Failure of the test has been verified through a missmatch between AEC delay buffer size and test buffer size.
Also added a missing file rewind to another test and removed some lint warnings.

TEST=audioproc_unittest, trybots
BUG=None

Review URL: https://webrtc-codereview.appspot.com/1100004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3514 4adac7df-926f-26a2-2b94-8c16560cd09d

2013-02-14 15:29:09 +00:00

build

Fixing/disabling Windows x64 warnings

2013-02-04 10:07:17 +00:00

common_audio

Import SincResampler from Chromium.

2013-02-13 23:00:49 +00:00

common_video

Fixing/disabling Windows x64 warnings

2013-02-04 10:07:17 +00:00

modules

Added delay estimation test to audio processing unit tests.

2013-02-14 15:29:09 +00:00

system_wrappers

Import stringize_macros from Chromium.

2013-02-13 23:37:14 +00:00

test

Fix Windows x64 errors in video_codecs_test_framework

2013-02-13 09:35:12 +00:00

tools

Fixing/disabling Windows x64 warnings

2013-02-04 10:07:17 +00:00

video_engine

Remove MultiStreamMode from test.

2013-02-14 11:35:20 +00:00

voice_engine

Implement initial delay. This CL allows clients of VoE to set an initial delay. Playout of audio is delayed and the extra playout delay is maintained during the call. While packets are buffered (in NetEq) to acheive the desired delay. ACM will playout silence (zeros). Initial delay has to be set before any packet is pushed into ACM.

2013-02-12 21:42:18 +00:00

.gitignore

.gitignore: Add *.mk, created as part of ChromiumOS build

2013-01-04 21:25:42 +00:00

common_types.h

VP8: Making key frame interval a tunnable parameter

2013-01-31 16:37:13 +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.