henrika@webrtc.org 45db7eefa2 Use Java based audio as default for WebRTC. ...

The work landed in 4034 (use of HW AEC in AppRTC) is currently not
active by default since we build for Open SL. I missed that when I
did my initial change (since I always disabled OpenSL by GYP_DEFINES).

This CL ensures that Java based audio is used as default in WebRTC.
It would be great if we could shift over to Open SL (to cut latency)
but that would (today) mean that we can't support the HW AEC.
Hence, we are not ready to do so yet.

BUG=4034
R=kjellander@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8040 4adac7df-926f-26a2-2b94-8c16560cd09d

2015-01-12 14:27:23 +00:00

chromium

Roll chromium_revision 6455c69..deaf2f7 (293954:295079)

2014-09-27 18:10:30 +00:00

data

Revert "Revert part of r7561, "Refactor audio conversion functions.""

2014-10-31 04:58:14 +00:00

resources

resources/audio_processing: Removed unused test files

2014-12-15 15:57:11 +00:00

talk

Use proxy macro for PeerConnectionFactory instead of sending messages internally in PeerConnectionFactory.

2015-01-12 08:30:16 +00:00

third_party

GN: Fix Windows build.

2015-01-08 21:22:01 +00:00

tools

Suppress AsyncHttpRequestTest.TestCancel leak for LSan

2015-01-08 10:45:59 +00:00

webrtc

Use Java based audio as default for WebRTC.

2015-01-12 14:27:23 +00:00

.clang-format

Style-option file for clang-format.

2013-11-25 15:30:37 +00:00

.gitignore

Roll chromium_revision 8e72e1d..271c6cc (307131:309333)

2015-01-06 07:24:27 +00:00

all.gyp

Revert r7886:7887.

2014-12-15 07:03:04 +00:00

android-webrtc.mk

Changed a flag in ARM build for platform build.

2012-09-08 00:32:59 +00:00

AUTHORS

Added support for an Origin header in STUN messages.

2015-01-10 00:47:02 +00:00

BUILD.gn

Initial GN work for WebRTC

2014-06-17 08:54:03 +00:00

check_root_dir.py

Remove root_dir variable from DEPS + enforce rename.

2014-10-09 09:11:27 +00:00

codereview.settings

Add FORCE_HTTPS_COMMIT_URL to codereview.settings.

2014-12-01 09:09:07 +00:00

COPYING

Libjingle in webrtc needs updated AUTHORS, COPYING, LICENSE_THIRD_PARTY AND README.

2013-07-16 18:04:56 +00:00

DEPS

GN: Fix Windows build.

2015-01-08 21:22:01 +00:00

drover.properties

Add a drover.properties file for reference.

2013-05-30 18:15:54 +00:00

libvpx.mk

Change android makefile to reflect recent file structure change

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

LICENSE

Updated license path in LICENSE and LICENSE_THIRD_PARTY.

2012-11-22 12:06:21 +00:00

license_template.txt

Update template to follow chromium copyright style

2013-04-24 01:01:28 +00:00

LICENSE_THIRD_PARTY

Libjingle in webrtc needs updated AUTHORS, COPYING, LICENSE_THIRD_PARTY AND README.

2013-07-16 18:04:56 +00:00

OWNERS

Add kjellander as PRESUBMIT.py OWNER

2014-11-04 17:06:31 +00:00

PATENTS

Add references to src/ copies for LICENSE etc.

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

PRESUBMIT.py

PRESUBMIT: Add GN trybots for Windows and Mac.

2015-01-09 15:38:29 +00:00

setup_links.py

Roll chromium_revision 8e72e1d..271c6cc (307131:309333)

2015-01-06 07:24:27 +00:00

sync_chromium.py

Roll chromium_revision d8c9041..309cf65

2014-11-27 10:41:04 +00:00

WATCHLISTS

Removing giles@mozilla.com from WebRTC watchlist.

2014-12-22 12:49:14 +00:00

Description

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

BSD-3-Clause 446 MiB

Languages

C++ 90.3%

Java 2.9%

C 2.2%

Objective-C++ 2%

Python 1.3%

Other 1%