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andrew@webrtc.org 5f23d64cf2 Set the stream delay to zero if too low. 
- Return a stream warning instead of an error.
- Add a few delay offset tests.

BUG=
TEST=audioproc_unittest

Review URL: https://webrtc-codereview.appspot.com/607004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2316 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-29 21:14:06 +00:00
data
Move trunk/test/data -> trunk/data
2012-05-23 15:39:01 +00:00
peerconnection
* Add gold as deps.
2012-05-09 23:22:30 +00:00
src
Set the stream delay to zero if too low.
2012-05-29 21:14:06 +00:00
test
Move trunk/test/data -> trunk/data
2012-05-23 15:39:01 +00:00
third_party
Add google-gflags header files for linux/arm platform
2012-05-21 18:07:28 +00:00
third_party_mods
Removing PeerConnection sample client and libjingle from webrtc.
2012-02-09 19:23:17 +00:00
tools
Syncing tests on try master with build master.
2012-05-29 14:49:39 +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
Update DEPS comment and remove tabs from common.gypi.
2012-05-08 18:00:57 +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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