platform-external-webrtc

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Haibo Huang 715411c11b Remove dependency on std::tr1

This change is to make webrtc compatible with new googletest.

Test: compile
Change-Id: I91fa5ce8a986e35209986bef5dc0b2202155623f

2018-07-11 05:04:46 +00:00

build_overrides

Leave only an empty top level OWNERS file.

2017-02-23 22:25:35 -08:00

chromium

Leave only an empty top level OWNERS file.

2017-02-23 22:25:35 -08:00

data

Enable AudioProcessing48kHzSupport by default

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

infra/config

Leave only an empty top level OWNERS file.

2017-02-23 22:25:35 -08:00

resources

Leave only an empty top level OWNERS file.

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talk

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third_party

Leave only an empty top level OWNERS file.

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tools

Leave only an empty top level OWNERS file.

2017-02-23 22:25:35 -08:00

webrtc

Remove dependency on std::tr1

2018-07-11 05:04:46 +00:00

.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.bp

Use -Werror in external/webrtc

2017-10-26 10:24:04 -07:00

Android.mk

Add empty Android.mk to prevent recursion.

2018-01-29 16:31:28 -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

METADATA

Add METADATA for webrtc

2018-07-10 15:11:57 -07:00

MODULE_LICENSE_BSD

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NOTICE

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OWNERS

Leave only an empty top level OWNERS file.

2017-02-23 22:25:35 -08:00

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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