platform-external-webrtc

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Chih-hung Hsieh 4805e8260a Merge "Suppress unused-parameter warnings."

am: 14a01a403a

* commit '14a01a403a3a03350623687a5a5ca1e64a985224':
  Suppress unused-parameter warnings.

2016-02-12 02:56:10 +00:00

build_overrides

GN: Remove build_overrides/v8.gni

2015-10-22 08:50:43 +00:00

chromium

Updated chromium/.gclient and sync_chromium.py to not ignore third_party/ffmpeg.

2016-01-12 14:05:39 +00:00

data

Enable AudioProcessing48kHzSupport by default

2015-05-22 18:59:59 +00:00

infra

Remove libfuzzer trybot from default trybot set.

2016-01-14 10:03:31 +00:00

resources

Cleaning neteq_unittest resource files.

2016-01-08 19:30:18 +00:00

talk

Eliminate defines in talk/

2016-01-14 19:01:25 +00:00

third_party

Roll chromium_revision d66326c..4df108a (367167:367307)

2016-01-04 18:16:01 +00:00

tools

Re-enable tests that failed under Linux_Msan.

2016-01-13 16:35:51 +00:00

webrtc

Suppress unused-parameter warnings.

2016-02-11 16:11:43 -08:00

.clang-format

Style-option file for clang-format.

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

.gitignore

Update .gitignore

2016-01-12 07:17:49 +00:00

.gn

H.264: Default flags and pulling in openh264 and ffmpeg.

2016-01-11 18:19:06 +00:00

all.gyp

Remove webrtc/examples/android/media_demo.

2015-11-11 12:11:21 +00:00

android-webrtc.mk

WIP: Changes after merge commit 'cb3f9bd'

2015-11-25 11:43:05 -08:00

Android.mk

WIP: Changes after merge commit 'cb3f9bd'

2015-11-25 11:43:05 -08:00

AUTHORS

Fixing integer underflow in FileAudioDevice (webrtc issue 4554)

2016-01-07 20:38:36 +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

Move to Chromium's Rietveld at codereview.webrtc.org

2015-06-04 07:07:42 +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 346fea9..099be58 (369082:369139)

2016-01-13 15:52:44 +00:00

LICENSE

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license_template.txt

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LICENSE_THIRD_PARTY

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

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

MODULE_LICENSE_BSD

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NOTICE

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OWNERS

So long and thanks for all the code reviews!

2015-11-18 11:07:45 +00:00

PATENTS

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PRESUBMIT.py

Lint fix for webrtc/modules/video_coding PART 3!

2015-12-21 16:23:29 +00:00

pylintrc

PRESUBMIT: Improve PyLint check and add GN format check.

2015-05-25 10:55:50 +00:00

README.md

Add README.md to root directory

2015-09-11 09:04:09 +00:00

setup_links.py

H.264: Default flags and pulling in openh264 and ffmpeg.

2016-01-11 18:19:06 +00:00

sync_chromium.py

Updated chromium/.gclient and sync_chromium.py to not ignore third_party/ffmpeg.

2016-01-12 14:05:39 +00:00

TODO

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WATCHLISTS

Add webrtc/audio and webrtc/call to WATCHLISTS.

2015-12-21 13:30:01 +00:00

README.md

WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.

Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.

The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others. This page is maintained by the Google Chrome team.

Development

See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.

More info

Official web site: http://www.webrtc.org
Master source code repo: https://chromium.googlesource.com/external/webrtc
Samples and reference apps: https://github.com/webrtc
Mailing list: http://groups.google.com/group/discuss-webrtc
Continuous build: http://build.chromium.org/p/client.webrtc

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