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chromium-webrtc-autoroll 7ea5c1ff11 Roll chromium_revision 241ac98bfc..5e5d0f9ef8 (627089:627255)

Change log: 241ac98bfc..5e5d0f9ef8
Full diff: 241ac98bfc..5e5d0f9ef8

Changed dependencies
* src/base: b75c898994..052ddc5f6e
* src/build: 9dbdd5c2ae..078c5476d6
* src/ios: 035b561e5c..d8c1cb941c
* src/testing: ee6e9ab571..45190ccdd0
* src/third_party: f20a194062..64fa4ace3a
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/a08f0fce79..1a9772ff05
* src/tools: 723be0ba24..d5d6050c4a
DEPS diff: 241ac98bfc..5e5d0f9ef8/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: Ia15feb7620beb63c7fbd9ac26c182a8602d1babd
Reviewed-on: https://webrtc-review.googlesource.com/c/120500
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#26458}

2019-01-30 01:54:01 +00:00

api

Remove deprecated OnKeyFrame method from video sink interface in media transport

2019-01-29 21:27:52 +00:00

audio

Receive-side ready for multiple channels.

2019-01-29 12:43:23 +00:00

build_overrides

Disable hermetic toolchain when building on macOS 10.14.

2018-12-07 13:57:28 +00:00

call

Moves packet overhead from network nodes to simulation.

2019-01-29 16:55:04 +00:00

common_audio

(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries

2019-01-11 17:11:39 +00:00

common_video

Remove rtc_base/scoped_ref_ptr.h.

2019-01-25 20:29:58 +00:00

data

Remove old data files.

2018-10-05 14:40:21 +00:00

examples

Added VcmCapturer::Create loop to allow nonzero device index.

2019-01-29 08:06:22 +00:00

infra

Re-add MSVC debug bots to CQ

2018-12-05 12:20:56 +00:00

logging

Ignore duplicated incoming RTCP packets in RTC event log parser.

2019-01-28 20:38:38 +00:00

media

Implement the encoding RtpParameter scaleResolutionDownBy

2019-01-29 14:32:17 +00:00

modules

Add ability to set max probing bitrate via GoogCcNetworkController

2019-01-29 19:19:04 +00:00

p2p

Fix post submit build

2019-01-30 00:42:01 +00:00

Create visible fake_ice_transport target

2019-01-29 22:32:46 +00:00

resources

Reland of Opus multistream.

2019-01-29 12:16:19 +00:00

rtc_base

Add ability to set max probing bitrate via GoogCcNetworkController

2019-01-29 19:19:04 +00:00

rtc_tools

[clang-tidy] Apply performance-faster-string-find fixes.

2019-01-28 11:31:53 +00:00

sdk

Remove +api from internal DEPS files.

2019-01-28 11:17:00 +00:00

stats

Remove +api from internal DEPS files.

2019-01-28 11:17:00 +00:00

style-guide

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

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

system_wrappers

Replace the usage of RTC_HISTOGRAM_COMMON_BLOCK with

2019-01-21 10:44:35 +00:00

test

Moves packet overhead from network nodes to simulation.

2019-01-29 16:55:04 +00:00

tools_webrtc

mb: remove 'type': 'gn' because it's the default and doesn't mean anything

2019-01-29 09:29:41 +00:00

video

Move FrameCountObserver from RTPSender to RtpVideoSender

2019-01-29 09:31:11 +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

Add Visual Studio Code project folder to gitignore file.

2019-01-21 18:42:33 +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

Allow use of functions in absl/algorithms

2019-01-28 18:42:16 +00:00

AUTHORS

Add ',' between elements in RTCStatsReport::ToJson

2019-01-08 16:05:23 +00:00

BUILD.gn

Adds repeating task class.

2019-01-18 10:55:41 +00:00

CODE_OF_CONDUCT.md

Add code of conduct to WebRTC repo

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

codereview.settings

Don't add webrtc-reviews@ to CC, it can be added globally on Gerrit

2018-10-25 08:19:53 +00:00

common_types.h

Delete constant RTP_PAYLOAD_NAME_SIZE

2019-01-25 07:59:52 +00:00

DEPS

Roll chromium_revision 241ac98bfc..5e5d0f9ef8 (627089:627255)

2019-01-30 01:54:01 +00:00

ENG_REVIEW_OWNERS

Enforce LGTM from owners of depends-on paths in DEPS via presubmit.

2018-09-28 12:49:54 +00:00

LICENSE

Moving src/webrtc into src/.

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

license_template.txt

Update template to follow chromium copyright style

2013-04-24 01:01:28 +00:00

native-api.md

Delete unused I420 "codec"

2018-12-18 12:30:58 +00:00

OWNERS

Clean up root OWNERS.

2018-11-09 14:23:59 +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

Fixing snake_case files that were renamed in PRESUBMIT.py

2019-01-23 18:38:53 +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 style rule about not using const optional<T>& arguments

2018-11-08 11:57:35 +00:00

WATCHLISTS

Remove likely obsolete entries from WATCHLISTS

2018-10-30 07:46:29 +00:00

webrtc.gni

Override default manifest from Chromium in WebRTC.

2019-01-24 13:00:49 +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++ 90.3%

Java 2.9%

C 2.2%

Objective-C++ 2%

Python 1.3%

Other 1%