andrew@webrtc.org 0594916a21 Add audio_e2e_test to LinuxLargeTests. ...

Ensure pulseaudio is running on the bot.

BUG=none
TEST=audio_e2e_test
Review URL: https://webrtc-codereview.appspot.com/646004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2392 4adac7df-926f-26a2-2b94-8c16560cd09d

2012-06-12 03:32:18 +00:00

data

Move trunk/test/data -> trunk/data

2012-05-23 15:39:01 +00:00

peerconnection

Update to chromium r139469.

2012-05-31 00:31:42 +00:00

src

Add VoiceEngine apm settings to test application

2012-06-11 21:23:16 +00:00

test

Removed the obsolete sanity check and added new test HTML files.

2012-06-04 10:06:52 +00:00

third_party

Disable multi_res_encoding in libvpx.

2012-06-08 22:53:55 +00:00

third_party_mods

Removing PeerConnection sample client and libjingle from webrtc.

2012-02-09 19:23:17 +00:00

tools

Add audio_e2e_test to LinuxLargeTests.

2012-06-12 03:32:18 +00:00

.gitignore

Refer to Chrome's DEPS to make rolling easier.

2012-02-23 21:32:37 +00:00

android-webrtc.mk

Enable apm in android.mk

2012-01-14 01:03:56 +00:00

Android.mk

Remove video from Android.mk

2012-03-13 17:22:44 +00:00

AUTHORS

Adding author

2011-11-02 12:19:22 +00:00

codereview.settings

Enabling try server in codereview.settings

2012-03-29 05:14:13 +00:00

DEPS

Roll Chromium 134666:140240.

2012-06-04 23:14:37 +00:00

libvpx.mk

Change android makefile to reflect recent file structure change

2011-07-13 17:49:49 +00:00

LICENSE

Add references to src/ copies for LICENSE etc.

2011-10-26 01:05:07 +00:00

license_template.txt

git-svn-id: http://webrtc.googlecode.com/svn/trunk@11 4adac7df-926f-26a2-2b94-8c16560cd09d

2011-05-30 11:51:34 +00:00

LICENSE_THIRD_PARTY

Add references to src/ copies for LICENSE etc.

2011-10-26 01:05:07 +00:00

OWNERS

Review URL: http://webrtc-codereview.appspot.com/137002

2011-08-25 16:52:34 +00:00

PATENTS

Add references to src/ copies for LICENSE etc.

2011-10-26 01:05:07 +00:00

PRESUBMIT.py

More PRESUBMIT checks.

2012-02-21 17:53:46 +00:00

WATCHLISTS

Change Watchlist

2012-04-19 18:09:23 +00:00

webrtc.gyp

Add a framework for audio end-to-end quality testing.

2012-05-10 18:45:11 +00:00

Description

The idea is to make CMake build for WebRTC m130 version - for audio processing module

BSD-3-Clause 446 MiB

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