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chromium-webrtc-autoroll a102d8ed2b Roll chromium_revision 0d58dd3c85..17c6abea92 (899016:899632)

Change log: 0d58dd3c85..17c6abea92
Full diff: 0d58dd3c85..17c6abea92

Changed dependencies
* src/base: c1e65992d9..4a5a81e82b
* src/build: e16e08c1c3..fff5048571
* src/buildtools: fd3f3c1998..2500c1d8f3
* src/buildtools/third_party/libc++/trunk: 8fa8794677..79a2e924d9
* src/buildtools/third_party/libc++abi/trunk: d87a06daa9..cb34896ebd
* src/ios: 8e4c05bd2c..6d09c985f3
* src/testing: 57a831388b..d8cbec3370
* src/third_party: 1b79d8525d..89bb511d77
* src/third_party/androidx: PTOkBlPq_HcuCNU_wN2ZymkGWNszZRV4RCn5jnaVp7YC..29574JKqBbhq5FiO3D4ydclUDICPzLTJGfyNc4k4ldYC
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/096f6b42b5..2fff900ff7
* src/third_party/libunwindstack: 8c06e391ab..b34a0059a6
* src/third_party/perfetto: e989e5e45a..566975367c
* src/tools: a8205b76d0..3aa2ead994
DEPS diff: 0d58dd3c85..17c6abea92/DEPS

Clang version changed llvmorg-13-init-14634-gf814cd74:llvmorg-13-init-14732-g8a7b5ebf
Details: 0d58dd3c85..17c6abea92/tools/clang/scripts/update.py

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

Change-Id: I727c6a8aadf758595f1f0354a13b91e74332e8b6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225560
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@{#34439}

2021-07-08 19:52:06 +00:00

api

Create name->value text map for frame and video statistics

2021-07-08 08:38:50 +00:00

audio

Implement nack_count metric for inbound audio rtp streams.

2021-07-01 10:38:44 +00:00

build_overrides

Roll chromium + fix: blacklist -> ignorelist for sanitizers suppressions

2021-05-27 16:16:01 +00:00

call

Use flat_map/flat_set in RtpDemuxer

2021-07-07 08:22:55 +00:00

common_audio

Fix -Wunreachable-code-aggressive.

2021-06-30 11:14:37 +00:00

common_video

Update BitBuffer methods to style guide

2021-05-18 11:10:27 +00:00

data

Remove old data files.

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

docs

Create a VideoFrameTrackingId RTP header extension.

2021-03-25 17:25:18 +00:00

examples

iOS: Get WebRTC building for Mac Catalyst

2021-07-06 21:07:59 +00:00

g3doc

Move supported_platforms.md under g3doc/.

2021-06-28 12:07:45 +00:00

logging

Replace PacketView by vector of pointers in a wrapper class.

2021-07-02 08:17:32 +00:00

media

Unify helpers IsRtpPacket and IsRtcpPacket

2021-07-06 10:39:00 +00:00

modules

Support header only parsing by RtpPacket

2021-07-08 14:43:28 +00:00

net/dcsctp

dcsctp: Only reset paused streams when peer acks

2021-07-08 10:49:11 +00:00

p2p

Reland "Port: migrate to TaskQueue."

2021-06-21 22:21:04 +00:00

Implement nack_count metric for inbound audio rtp streams.

2021-07-01 10:38:44 +00:00

resources

Disable high-pass filtering of the AEC reference

2021-02-23 07:06:11 +00:00

rtc_base

Explicitly expose EraseIf in flat_map/flat_set

2021-07-07 12:34:49 +00:00

rtc_tools

Use default NetEq config for simulation in event log visualizer.

2021-07-01 14:06:20 +00:00

sdk

iOS: Get WebRTC building for Mac Catalyst

2021-07-06 21:07:59 +00:00

stats

Move WebRTC to non deprecated jsoncpp APIs.

2021-06-28 12:09:14 +00:00

system_wrappers

Make Clock::ConvertTimestampToNtpTime pure virtual

2021-05-21 09:55:14 +00:00

test

Destroy threads and TaskQueue at the end of tests.

2021-07-08 12:00:01 +00:00

tools_webrtc

Add ability to build XCFramework for iOS

2021-07-06 11:23:00 +00:00

video

Unify helpers IsRtpPacket and IsRtcpPacket

2021-07-06 10:39:00 +00:00

.clang-format

Add IncludeBlocks to clang-format.

2021-02-03 16:29:07 +00:00

.git-blame-ignore-revs

Let git-hyper-blame ignore new format cleanup.

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

.gitignore

Add .cache to .gitignore.

2021-01-20 15:01:07 +00:00

.gn

Increase iOS deployment target from 10 to 12.

2021-07-02 17:02:27 +00:00

.vpython

Update six library version

2021-04-26 16:39:07 +00:00

AUTHORS

Added PeerConnectionObserverJni::OnRemoveTrack()

2021-06-03 19:24:55 +00:00

BUILD.gn

Removing RTC_SUPPORTS_METAL compilation flag. This flag is a holdover from before either macOS or the iOS Simulator supported Metal rendering.

2021-06-23 09:56:20 +00:00

CODE_OF_CONDUCT.md

…

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 0d58dd3c85..17c6abea92 (899016:899632)

2021-07-08 19:52:06 +00:00

DIR_METADATA

Move metadata in OWNERS files to DIR_METADATA files.

2021-02-08 19:09:33 +00:00

ENG_REVIEW_OWNERS

Remove kwiberg@webrtc.org from OWNERS files

2020-12-04 15:11:26 +00:00

g3doc.lua

Improve webrtc documentation infra. Preview at:

2021-03-30 10:29:30 +00:00

LICENSE

Moving src/webrtc into src/.

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

license_template.txt

…

native-api.md

Make the remote_bitrate_estimator build target private

2020-11-26 12:21:22 +00:00

OWNERS

Move style guide and abseil-in-webrtc into g3doc subfolder

2021-05-13 14:43:10 +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

Run pylint on presubmit only for modified python files.

2021-07-02 15:06:56 +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

Move style guide and abseil-in-webrtc into g3doc subfolder

2021-05-13 14:43:10 +00:00

WATCHLISTS

Add hta@ to rtc_base/ and api/ WATCHLISTS.

2021-01-06 09:43:34 +00:00

webrtc_lib_link_test.cc

Deprecate PeerConnectionFactory::CreatePeerConnection

2021-05-10 08:47:48 +00:00

webrtc.gni

iOS: Get WebRTC building for Mac Catalyst

2021-07-06 21:07:59 +00:00

whitespace.txt

Revert "Reland "Trigger postsubmit tests.""

2021-06-28 19:44:42 +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: https://ci.chromium.org/p/webrtc/g/ci/console
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%