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chromium-webrtc-autoroll 81e13d3d6e Roll chromium_revision 2826799ea1..1b27d646a6 (885837:886225)

Change log: 2826799ea1..1b27d646a6
Full diff: 2826799ea1..1b27d646a6

Changed dependencies
* src/base: 0ee65b122a..d26844a7c8
* src/build: 1cf931db34..de91a1ebf8
* src/ios: 5ead6b825c..25276fd892
* src/testing: ada074d45e..0519f53328
* src/third_party: 2f7883d4d1..0746065687
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/cde766ee8a..53102de187
* src/third_party/depot_tools: 6097a92db0..7fc02e848a
* src/third_party/freetype/src: 2468e59af1..5f485339be
* src/third_party/perfetto: 84f640ce02..e2a0b3126e
* src/tools: 08f3a48d95..7a2715d3bd
* src/tools/luci-go: git_revision:ff0c02e960b17be5a8d02c13afc28373f236a860..git_revision:03ef3f6285cef3766c18a5c7b8cc8c0069eaa2ed
* src/tools/luci-go: git_revision:ff0c02e960b17be5a8d02c13afc28373f236a860..git_revision:03ef3f6285cef3766c18a5c7b8cc8c0069eaa2ed
* src/tools/luci-go: git_revision:ff0c02e960b17be5a8d02c13afc28373f236a860..git_revision:03ef3f6285cef3766c18a5c7b8cc8c0069eaa2ed
DEPS diff: 2826799ea1..1b27d646a6/DEPS

No update to Clang.

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

Change-Id: Ic8dadfaac07170c30e8346d7c69ea07d16e8addf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220120
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@{#34112}

2021-05-25 10:28:20 +00:00

api

Reland "Fix race between enabled() and set_enabled() in VideoTrack."

2021-05-24 16:17:35 +00:00

audio

Make local to capturer clock offset a separate entry in PacketInfo.

2021-05-20 13:42:57 +00:00

build_overrides

Roll chromium_revision 34f3c82122..2dffe06711 (867171:871492)

2021-04-12 18:25:58 +00:00

call

Update WebRTC code version (2021-05-25T04:03:57).

2021-05-25 05:06:51 +00:00

common_audio

Avoid undefined behavior in a division operation.

2021-04-23 07:49:24 +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

Replace legacy getStats with standard getStats in the iOS example

2021-05-20 08:40:11 +00:00

g3doc

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

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

logging

Revert "Deprecate microsecond timestamps in RTC event log."

2021-05-24 13:11:10 +00:00

media

Reland "Fix race between enabled() and set_enabled() in VideoTrack."

2021-05-24 16:17:35 +00:00

modules

Refactor and improve RtpSender packet history test.

2021-05-25 09:53:27 +00:00

net/dcsctp

dcsctp: Limit the size of generated SACK chunks

2021-05-25 07:56:56 +00:00

p2p

Remove cricket::DtlsTransportState.

2021-05-21 21:45:29 +00:00

Reland "Fix race between enabled() and set_enabled() in VideoTrack."

2021-05-24 16:17:35 +00:00

resources

Disable high-pass filtering of the AEC reference

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

rtc_base

Update BitBuffer methods to style guide

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

rtc_tools

Convert to NTP time using the real clock.

2021-05-21 19:29:32 +00:00

sdk

Set nativeObserver to 0 to avoid double release.

2021-05-17 19:53:43 +00:00

stats

Simplify reference counting implementation of PendingTaskSafetyFlag.

2021-04-21 07:04:01 +00:00

system_wrappers

Make Clock::ConvertTimestampToNtpTime pure virtual

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

test

Implement the mixer-to-client per CSRC audio level extension (RFC 6465).

2021-05-24 14:11:28 +00:00

tools_webrtc

crc32c: Point the licensing script to the LICENSE file

2021-05-03 16:46:30 +00:00

video

Make local to capturer clock offset a separate entry in PacketInfo.

2021-05-20 13:42:57 +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

Switch from check_targets to no_check_targets in .gn

2021-05-20 10:42:21 +00:00

.vpython

Update six library version

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

AUTHORS

Revert "Ensure method which updates UI is called in main thread"

2021-04-30 09:26:03 +00:00

BUILD.gn

Switch from check_targets to no_check_targets in .gn

2021-05-20 10:42:21 +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 2826799ea1..1b27d646a6 (885837:886225)

2021-05-25 10:28:20 +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

…

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

…

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

sctp: Rename SctpTransport to UsrSctpTransport

2021-04-12 10:40:34 +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

Turn on the RTC_ENABLE_WIN_WGC build flag.

2021-05-10 20:16:52 +00:00

whitespace.txt

Reland "Triggering CI."

2021-03-22 11:57:23 +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%