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chromium-webrtc-autoroll 1620db743e Roll chromium_revision acfa5a0113..fed88f41ed (830473:830583)

Change log: acfa5a0113..fed88f41ed
Full diff: acfa5a0113..fed88f41ed

Changed dependencies
* src/base: 67efd05620..df7d1bcfe2
* src/build: 356ef25e24..0b7bec4032
* src/ios: feae45c075..c22e99448e
* src/third_party: 9b8961da57..1f137efbc1
* src/third_party/depot_tools: 260eb0f662..b35dd76cd5
* src/tools: d28bf829ec..880674d483
* src/tools/luci-go: git_revision:6cbe3f56e9f00b8f65eae21f01838a8b58191a47..git_revision:1a022d3a4c50be4207ee93451255d71896416596
* src/tools/luci-go: git_revision:6cbe3f56e9f00b8f65eae21f01838a8b58191a47..git_revision:1a022d3a4c50be4207ee93451255d71896416596
* src/tools/luci-go: git_revision:6cbe3f56e9f00b8f65eae21f01838a8b58191a47..git_revision:1a022d3a4c50be4207ee93451255d71896416596
DEPS diff: acfa5a0113..fed88f41ed/DEPS

No update to Clang.

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

Change-Id: I251b329a2677ea78a165ee3dd03949de83f090e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195240
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@{#32687}

2020-11-24 16:44:48 +00:00

api

Defining API result types on VoIP API

2020-11-20 18:02:05 +00:00

audio

Default enable sending transport sequence numbers on audio packets.

2020-11-24 09:19:54 +00:00

build_overrides

set perfetto flag to default value of false

2020-07-22 10:14:53 +00:00

call

Default enable sending transport sequence numbers on audio packets.

2020-11-24 09:19:54 +00:00

common_audio

Introduce RTC_CHECK_NOTREACHED(), an always-checking RTC_NOTREACHED()

2020-11-09 10:47:55 +00:00

common_video

Reland "Rename FATAL() into RTC_FATAL()."

2020-11-18 20:49:08 +00:00

data

Remove old data files.

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

docs

Document purpose and wire format of extension http://www.webrtc.org/experiments/rtp-hdrext/video-layers-allocation00

2020-11-23 14:55:49 +00:00

examples

Defining API result types on VoIP API

2020-11-20 18:02:05 +00:00

logging

Reland "Introduce RTC_NO_UNIQUE_ADDRESS."

2020-11-23 11:29:36 +00:00

media

fixing build for when HAVE_WEBRTC_VIDEO is not defined

2020-11-24 11:14:26 +00:00

modules

Removed unused H264 specific code from RtpFrameReferenceFinder.

2020-11-24 13:51:36 +00:00

p2p

p2p: Fix bug causing old candidates on ice restart

2020-11-23 14:58:39 +00:00

Default enable sending transport sequence numbers on audio packets.

2020-11-24 09:19:54 +00:00

resources

Increased high frequency transparency

2020-11-02 10:42:10 +00:00

rtc_base

p2p: Fix bug causing old candidates on ice restart

2020-11-23 14:58:39 +00:00

rtc_tools

Reland "Introduce RTC_NO_UNIQUE_ADDRESS."

2020-11-23 11:29:36 +00:00

sdk

Delete objc RTCRtpFragmentationHeader

2020-11-19 14:05:48 +00:00

stats

stats: add transportId to codec stats

2020-11-17 12:34:39 +00:00

style-guide

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

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

system_wrappers

Delete use of RWLockWrapper from SimulatedClock

2020-11-04 08:01:08 +00:00

test

Default enable sending transport sequence numbers on audio packets.

2020-11-24 09:19:54 +00:00

tools_webrtc

Fix timout message while uploading to the dashboard

2020-11-24 15:56:08 +00:00

video

Default enable sending transport sequence numbers on audio packets.

2020-11-24 09:19:54 +00:00

.clang-format

…

.git-blame-ignore-revs

Let git-hyper-blame ignore new format cleanup.

2019-07-11 16:18:51 +00:00

.gitignore

Add .clangd to .gitignore

2019-10-28 12:27:50 +00:00

.gn

Rename PlayoutDelay --> VideoPlayoutDelay, move to api/video/video_timing.h

2020-09-07 08:37:14 +00:00

.vpython

Reland "Add protobuf-py2_py3 3.13.0 to .vpython."

2020-11-20 07:52:26 +00:00

abseil-in-webrtc.md

Polish the "Using Abseil in WebRTC" docs

2020-10-16 13:42:00 +00:00

AUTHORS

Fix "control reaches end of non-void function" warnings

2020-10-27 10:22:23 +00:00

BUILD.gn

Rename RoboCaller to CallbackList.

2020-10-23 15:14:22 +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

DEPS

Roll chromium_revision acfa5a0113..fed88f41ed (830473:830583)

2020-11-24 16:44:48 +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

…

native-api.md

modules/bitrate_controller/include is no longer a legacy api directory

2020-11-23 10:43:16 +00:00

OWNERS

Remove phoglund as root owner.

2020-03-30 12:15:56 +00:00

PATENTS

Moving src/webrtc into src/.

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

presubmit_test_mocks.py

Reformat python files checked by pylint (part 1/2).

2020-10-30 10:13:11 +00:00

presubmit_test.py

Reformat python files checked by pylint (part 1/2).

2020-10-30 10:13:11 +00:00

PRESUBMIT.py

Add transition define RTC_OBJC_HAVE_LEGACY_RTC_RTP_FRAGMENTATION_HEADER

2020-11-17 13:51:29 +00:00

pylintrc

Undo enforcing of PEP-8 pylint changes for method and function names.

2020-11-10 18:26:25 +00:00

README.chromium

Add CPEPrefix.

2020-07-13 11:42:07 +00:00

README.md

doc: move bug reporting instructions to the repository

2020-10-21 14:47:49 +00:00

style-guide.md

C++ style: We don't allow designated initializers

2020-06-03 09:11:09 +00:00

WATCHLISTS

Remove benwright@webrtc.org from WATCHLISTS

2020-01-31 18:46:52 +00:00

webrtc_lib_link_test.cc

Rewrite the lib link test to just be a binary.

2019-10-18 07:42:20 +00:00

webrtc.gni

Reland "Remove placeholder Obj-C headers and use angle-bracketed headers."

2020-10-22 11:29:48 +00:00

whitespace.txt

Trigger bots for FYI waterfall.

2020-11-20 09:35:54 +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 here 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
Reporting bugs

Languages

C++ 90.3%

Java 2.9%

C 2.2%

Objective-C++ 2%

Python 1.3%

Other 1%