platform-external-webrtc

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Mirko Bonadei 4524c74171 Removing /wd4703 from rtc_base:rtc_base_generic.

This warning suppression flag is not needed.

Bug: webrtc:9251
Change-Id: I3319e57f737d44e991ab7283647fd5522bd4bd37
Reviewed-on: https://webrtc-review.googlesource.com/87581
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23886}

2018-07-09 14:32:06 +00:00

api

AEC3: Slower adaptation of main filter

2018-07-05 14:37:27 +00:00

audio

Use absl::make_unique and absl::WrapUnique directly

2018-07-05 10:59:49 +00:00

build_overrides

Add phoglund@ to various OWNERS and remove kjellander@

2017-10-19 09:21:12 +00:00

call

Use absl::make_unique and absl::WrapUnique directly

2018-07-05 10:59:49 +00:00

common_audio

Removing warning suppression flags in common_audio/.

2018-07-04 11:13:09 +00:00

common_video

Add bit depth information to PlanarYuvBuffer

2018-06-26 20:23:37 +00:00

data

WebRTC: Replace ProjectRootPath by ResourcePath

2016-11-22 18:43:05 +00:00

examples

Use absl::make_unique and absl::WrapUnique directly

2018-07-05 10:59:49 +00:00

infra

Flip luci.webrtc.try to production

2018-05-30 08:30:00 +00:00

logging

Use absl::make_unique and absl::WrapUnique directly

2018-07-05 10:59:49 +00:00

media

Lower TL0 bitrate to 40% for lowest stream when using short 3tl

2018-07-08 12:17:58 +00:00

modules

Change test VideoProcessor kMaxBufferedInputFrames from 10 to 20.

2018-07-09 11:56:43 +00:00

ortc

Use absl::make_unique and absl::WrapUnique directly

2018-07-05 10:59:49 +00:00

p2p

Use absl::make_unique and absl::WrapUnique directly

2018-07-05 10:59:49 +00:00

Use absl::make_unique and absl::WrapUnique directly

2018-07-05 10:59:49 +00:00

resources

AGC2 RNN VAD: Polishing.

2018-05-15 16:41:02 +00:00

rtc_base

Removing /wd4703 from rtc_base:rtc_base_generic.

2018-07-09 14:32:06 +00:00

rtc_tools

Break out Agc code from audio_processing.

2018-07-06 13:29:43 +00:00

sdk

Add unit test for JavaToNativeVideoCodecInfo.

2018-07-09 13:26:34 +00:00

stats

Adding "is_standardized" flag to RTCStatsMember.

2018-06-28 00:43:46 +00:00

style-guide

Add style guide rule about paired .h and .cc files

2018-03-14 13:02:35 +00:00

system_wrappers

Removing /wd4334 from system_wrappers.

2018-07-05 09:18:34 +00:00

test

Limit input size to iSAC and iLBC decoder fuzzers

2018-07-05 14:30:12 +00:00

tools_webrtc

roll_deps: Accept any prefix (like 'git_revision:'), not only 'version:' for CIPD

2018-07-03 09:41:53 +00:00

video

Stop using VideoCodec.targetBitrate for vp8 screenshare config

2018-07-06 15:13:18 +00:00

.clang-format

Tune ObjC clang-format configuration

2017-05-11 09:14:18 +00:00

.git-blame-ignore-revs

Let git-hyper-blame ignore format commit.

2018-06-20 09:26:44 +00:00

.gitignore

Roll back checking in the third_party directory

2018-06-27 13:04:08 +00:00

.gn

Set gtest_enable_absl_printers to true.

2018-06-29 09:36:17 +00:00

.vpython

Add vpython dependencies needed to run presubmit tests on LUCI

2018-05-18 08:10:25 +00:00

AUTHORS

Generalize SimulcastEncoderAdapter, use for H264 & VP8.

2018-06-21 15:57:43 +00:00

BUILD.gn

A new PeerConnection level perf test.

2018-06-27 23:19:05 +00:00

CODE_OF_CONDUCT.md

Add code of conduct to WebRTC repo

2017-05-16 12:09:13 +00:00

codereview.settings

Make Gerrit the default for WebRTC changes

2017-09-29 01:38:07 +00:00

common_types.h

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

DEPS

Roll chromium_revision 53ed5722bc..a9fbefaa78 (573145:573255)

2018-07-09 10:09:10 +00:00

LICENSE

Moving src/webrtc into src/.

2017-09-15 04:25:06 +00:00

license_template.txt

…

LICENSE_THIRD_PARTY

Remove third party dependecies that are not more in the source code

2018-06-21 11:33:41 +00:00

native-api.md

Remove legacy VoiceEngine.

2018-01-12 11:31:52 +00:00

OWNERS

Add mbonadei@ to build configs OWNERS.

2018-06-20 12:39:11 +00:00

PATENTS

Moving src/webrtc into src/.

2017-09-15 04:25:06 +00:00

presubmit_test_mocks.py

Reland: Add presubmit check for changes in 3pp

2018-05-22 13:11:18 +00:00

presubmit_test.py

Roll chromium_revision 95336cb92b..191d55580e (557816:557824)

2018-05-11 11:17:05 +00:00

PRESUBMIT.py

Adding CheckNoWarningSuppressionFlagsAreAdded.

2018-07-04 09:01:32 +00:00

pylintrc

Removing invalid-name from disabled pylint checks.

2017-10-11 08:06:49 +00:00

README.chromium

Moving src/webrtc into src/.

2017-09-15 04:25:06 +00:00

README.md

Tell users where they can find the native API headers

2017-11-14 10:36:46 +00:00

style-guide.md

Add style guidance about forward declarations.

2018-03-28 20:58:27 +00:00

typedefs.h

Remove typedefs.h from webrtc/ root (part 1)

2018-05-23 12:07:10 +00:00

WATCHLISTS

Adding mbonadei@ to build_files WATCHLIST.

2018-06-20 12:38:06 +00:00

webrtc.gni

Don't call deprecated FFmpeg API.

2018-06-26 13:57:35 +00:00

whitespace.txt

Whitespace change

2018-02-23 10:34:16 +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.

Development

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

Authoritative list of directories that contain the native API header files.

More info

Official web site: http://www.webrtc.org
Master source code repo: https://webrtc.googlesource.com/src
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
Coding style guide
Code of conduct

Languages

C++ 88.6%

C 3.3%

Java 3%

Objective-C++ 1.9%

Python 1.9%

Other 1%