platform-external-webrtc

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Autoroller bdc115c23d Roll chromium_revision e3170c4fc2..2338e7ccdb (590991:591117)

Change log: e3170c4fc2..2338e7ccdb
Full diff: e3170c4fc2..2338e7ccdb

Changed dependencies:
* src/base: 36a8382885..9b4f76b9ad
* src/build: 8e97ee738f..faf0511f93
* src/ios: 6e6e8b9e68..f90e1bf8a1
* src/testing: c058a65ed9..24c0ae4b1e
* src/third_party: 1cb9948eb6..b028e8c8a2
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/3e071665b9..5f6da8a57d
* src/tools: 9cd07469bb..2fe1aca913
DEPS diff: e3170c4fc2..2338e7ccdb/DEPS

No update to Clang.

TBR=buildbot@webrtc.org,
BUG=None
CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal

Change-Id: Ibd567ae00ee276bc54fd92154b6cd1f010f0db4b
Reviewed-on: https://webrtc-review.googlesource.com/100321
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24735}

2018-09-13 20:24:06 +00:00

api

Reland "Add initial support for RtpEncodingParameters max_framerate."

2018-09-13 10:06:33 +00:00

audio

Delete always true member voe::Channel::pacing_enabled_

2018-09-11 11:40:11 +00:00

build_overrides

Add phoglund@ to various OWNERS and remove kjellander@

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

call

Reland "Add initial support for RtpEncodingParameters max_framerate."

2018-09-13 10:06:33 +00:00

common_audio

Remove simple stringstream usages.

2018-09-06 12:53:19 +00:00

common_video

Make the rtp timestamp member of EncodedImage private

2018-09-12 13:44:36 +00:00

data

WebRTC: Replace ProjectRootPath by ResourcePath

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

examples

Revert "Compile frame_analyzer instead of using a prebuilt version."

2018-09-13 10:29:26 +00:00

infra

Remove MSVC debug bots from CQ.

2018-09-01 09:20:03 +00:00

logging

Remove all remaining non-test uses of std::stringstream.

2018-09-13 08:52:05 +00:00

media

Calculate NtpCaptureStartMs using clock, not the frame rtp timestamp

2018-09-13 13:49:47 +00:00

modules

Fixes issue where WebRTC.Audio.RecordSampleRateOffsetInPercent can report 100%

2018-09-13 16:04:31 +00:00

ortc

Remove all remaining non-test uses of std::stringstream.

2018-09-13 08:52:05 +00:00

p2p

Remove all remaining non-test uses of std::stringstream.

2018-09-13 08:52:05 +00:00

Reland "Add initial support for RtpEncodingParameters max_framerate."

2018-09-13 10:06:33 +00:00

resources

Reland "Add Y4mFileReader"

2018-09-05 09:30:08 +00:00

rtc_base

Remove all remaining non-test uses of std::stringstream.

2018-09-13 08:52:05 +00:00

rtc_tools

Allow printing graphs as protobuf in event_log_visualizer.

2018-09-11 09:53:12 +00:00

sdk

Removes flaky thread checker in AudioDeviceBuffer.

2018-09-13 11:41:52 +00:00

stats

getStats: add kind alias for mediaType

2018-09-03 11:49:30 +00:00

style-guide

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

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

system_wrappers

Refactor initialization of GetRealTimeClock singleton.

2018-09-10 18:00:44 +00:00

test

Print filename on open failure in FrameGenerator.

2018-09-13 15:20:07 +00:00

tools_webrtc

Revert "Compile frame_analyzer instead of using a prebuilt version."

2018-09-13 10:29:26 +00:00

video

Don't pad to min bitrate for streams with alr probing enabled

2018-09-13 16:15:21 +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

Revert "Compile frame_analyzer instead of using a prebuilt version."

2018-09-13 10:29:26 +00:00

.gn

Re-enable gtest absl pretty printers.

2018-08-13 13:54:05 +00:00

.vpython

Add vpython dependencies needed to run presubmit tests on LUCI

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

abseil-in-webrtc.md

Add a list of allowed and disallowed Abseil things

2018-08-27 08:37:20 +00:00

AUTHORS

Remove unused var in device info bits from video capture module for Linux

2018-09-12 10:36:33 +00:00

BUILD.gn

Enable -Wexit-time-destructors and -Wglobal-constructors.

2018-09-06 12:43:20 +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

Remove double declaration of 2 conversion functions.

2018-09-11 06:00:05 +00:00

DEPS

Roll chromium_revision e3170c4fc2..2338e7ccdb (590991:591117)

2018-09-13 20:24:06 +00:00

LICENSE

Moving src/webrtc into src/.

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

license_template.txt

…

native-api.md

Add documentation for WEBRTC_EXCLUDE_BUILT_IN_SSL_ROOT_CERTS.

2018-08-21 13:29:36 +00:00

OWNERS

Add a list of allowed and disallowed Abseil things

2018-08-27 08:37:20 +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

Fixing py lint errors

2018-07-23 15:28:48 +00:00

PRESUBMIT.py

Upload WebRTC CLs from Chromium.

2018-09-04 11:14:07 +00:00

pylintrc

Fixing py lint errors

2018-07-23 15:28:48 +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 a list of allowed and disallowed Abseil things

2018-08-27 08:37:20 +00:00

WATCHLISTS

Reland "Remove steveanton from api/ watchlist"

2018-09-05 11:45:33 +00:00

webrtc.gni

Fix paths for macOS framework’s copy actions.

2018-09-06 14:13:11 +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%