glaznev@webrtc.org 58202946a7 Cleaning up Android AppRTCDemo. ...

- Move signaling code from Activity to a separate class
and add interface for AppRTC signaling. For now
only pure GAE signaling implements this interface.
- Move peer connection, video source and peer connection
and SDP observer code from Activity to a separate class.
- Main Activity class will do only high level calls and
event handling for peer connection and signaling classes.
- Also add video renderer position update and use full
screen for local preview until the connection is established.

BUG=
R=braveyao@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7469 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-10-17 17:42:38 +00:00

chromium

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

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

data

Adjust speech probability in NS when echo

2014-10-13 20:48:05 +00:00

resources

Minor modifications to test::RtpFileReader

2014-09-30 11:08:44 +00:00

talk

Cleaning up Android AppRTCDemo.

2014-10-17 17:42:38 +00:00

third_party

GN: Enable libvpx, add link target and convert some test targets

2014-09-30 18:05:02 +00:00

tools

rtc_unittest: copied gtest excludes from libjingle_p2p_unittest since its tests have move to rtc_unittests.

2014-10-17 16:12:33 +00:00

webrtc

Moving creating TURN configration to the host machine instead of the bots - rtcBot

2014-10-17 16:43:50 +00:00

.clang-format

Style-option file for clang-format.

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

.gitignore

Add support for MSan

2014-10-01 08:03:19 +00:00

all.gyp

Add a variable for deciding when to use openmax_dl.

2014-10-08 18:01:27 +00:00

android-webrtc.mk

Changed a flag in ARM build for platform build.

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

AUTHORS

Unit tests for SSLAdapter

2014-09-23 05:56:44 +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 a root codereview.settings file.

2013-04-24 02:08:50 +00:00

COPYING

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

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

DEPS

Roll chromium_revision 2d714fa..de13cf4 (298667:299488)

2014-10-15 05:59:42 +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

Update root OWNERS file

2014-08-25 14:41:41 +00:00

PATENTS

Add references to src/ copies for LICENSE etc.

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

PRESUBMIT.py

Adds PRESUBMIT.py dispensation for depending on rtc_base.

2014-10-01 14:40:58 +00:00

setup_links.py

Add support for MSan

2014-10-01 08:03:19 +00:00

sync_chromium.py

Cleanup .gclient.bot_entries to avoid sync problems on bots.

2014-09-27 18:41:03 +00:00

WATCHLISTS

Add myself to common_audio and audio_processing watchlists

2014-09-11 10:11:43 +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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