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chromium-webrtc-autoroll 72c363c44a Roll chromium_revision bd812e5c8f..61baf6d3b3 (1037901:1038029) 
Change log: bd812e5c8f..61baf6d3b3
Full diff: bd812e5c8f..61baf6d3b3

Changed dependencies
* src/base: bb56c07514..6991cddfd4
* src/build: f1e943c044..eb4b420860
* src/ios: 63c6a9ae0d..ba28b8202b
* src/testing: fecf59bb27..800930ac53
* src/third_party: 727eb8af66..57d301ccb3
* src/third_party/androidx: nJRbqGtdXzotJnB8kFaAckjFBKzXW4O9BdApCCC7J8AC..SK5OSmpPVWJZugMDYiKQz2ERMeOTqQVSdK-sEGm96JcC
* src/third_party/fuchsia-sdk/sdk: version:9.20220822.1.1..version:9.20220822.3.1
* src/third_party/perfetto: 09cadaf285..4b9a2988e1
* src/third_party/turbine: RxaW8ZVJZ7rF3dbAGbL1OV6f9ZELjVeiiQbpkDsuxA4C..cMFT-KTiKSkVefC3nASLc0KQv9JrVmwmFUgJlo6jOosC
* src/tools: 1d5d5cca18..25ba08a546
DEPS diff: bd812e5c8f..61baf6d3b3/DEPS

No update to Clang.

BUG=None

Change-Id: I1f8945703d9258e12a7892950c27363536d207d0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272642
Bot-Commit: 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/main@{#37873}
2022-08-23 00:48:26 +00:00
api
[PCLF] Add ToString method to VideoDumpOptions
2022-08-22 12:47:37 +00:00
audio
Update rtc::Event::Wait call sites to use TimeDelta.
2022-08-19 10:07:28 +00:00
build_overrides
Roll chromium + fix: blacklist -> ignorelist for sanitizers suppressions
2021-05-27 16:16:01 +00:00
call
Update WebRTC code version (2022-08-22T04:02:37).
2022-08-22 05:12:29 +00:00
common_audio
Reenable WebRTC PushResampler format checks on Windows clang debug builds
2022-08-05 11:03:08 +00:00
common_video
Migrate remaining webrtc usage of TaskQueueBase to absl::AnyInvocable
2022-07-20 08:15:08 +00:00
data
…
docs
[ios] Remove the support for bitcode
2022-07-04 09:01:52 +00:00
examples
Reenable some iOS tests.
2022-08-19 13:10:17 +00:00
g3doc
Clarify how to reference WebRTC bugs in TODOs
2022-07-01 08:03:34 +00:00
infra
Add Weetbix realm ACL
2022-08-22 06:39:39 +00:00
logging
Encode remote link capacity estimates in legacy RTC event log format
2022-08-11 12:57:02 +00:00
media
Remove sigslot usage from SctpTransportInternal
2022-08-22 13:51:17 +00:00
modules
Add additional checks in case of early portal error
2022-08-22 21:30:59 +00:00
net/dcsctp
dcsctp: Add handover state for stream counts
2022-08-22 11:04:31 +00:00
p2p
Replace RTCCertificateGeneratorCallback interface with an AnyInvocable
2022-08-22 16:53:14 +00:00
pc
Replace RTCCertificateGeneratorCallback interface with an AnyInvocable
2022-08-22 16:53:14 +00:00
resources
Clarify and extend test support for certain sample rates in audio processing
2022-08-03 14:26:36 +00:00
rtc_base
Replace RTCCertificateGeneratorCallback interface with an AnyInvocable
2022-08-22 16:53:14 +00:00
rtc_tools
Update rtc::Event::Wait call sites to use TimeDelta.
2022-08-19 10:07:28 +00:00
sdk
rtc::Event: Finalize migration to TimeDelta.
2022-08-19 13:44:57 +00:00
stats
stats: implement outbound-rtp.active
2022-07-28 13:35:40 +00:00
system_wrappers
Delete nisse@webrtc.org from OWNERS files
2022-07-28 08:47:38 +00:00
test
Remove sigslot usage from SctpTransportInternal
2022-08-22 13:51:17 +00:00
tools_webrtc
Migrate mb script to python3.
2022-08-17 13:50:43 +00:00
video
Move decoder instance ownership to VideoReceiver2
2022-08-22 13:42:47 +00:00
.clang-format
Add IncludeBlocks to clang-format.
2021-02-03 16:29:07 +00:00
.git-blame-ignore-revs
…
.gitignore
Add .cache to .gitignore.
2021-01-20 15:01:07 +00:00
.gn
Prevent jsoncpp from hiding deprecated declarations in WebRTC
2022-04-11 12:33:47 +00:00
.mailmap
Add .mailmap for git.
2022-02-20 14:22:13 +00:00
.style.yapf
Fix mb.py presubmit issues.
2021-12-08 08:53:00 +00:00
.vpython
Remove unused script webrtc_dashboard_upload.py
2022-03-21 12:54:42 +00:00
.vpython3
Update protobuf-py2_py3 wheel.
2022-07-01 15:17:36 +00:00
AUTHORS
Enable Multithreaded H264 Encoding For OpenH264
2022-08-19 10:30:37 +00:00
BUILD.gn
Delete QueuedTask and ToQueuedTask as no longer needed
2022-08-09 11:11:26 +00:00
CODE_OF_CONDUCT.md
Reference "main" branches instead of "master" branches.
2021-07-15 11:07:44 +00:00
codereview.settings
…
DEPS
Roll chromium_revision bd812e5c8f..61baf6d3b3 (1037901:1038029)
2022-08-23 00:48:26 +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 phoglund from ENG_REVIEW_OWNERS
2021-10-08 08:29:42 +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
Reference "main" branches instead of "master" branches.
2021-07-15 11:07:44 +00:00
OWNERS
Fix add some eng prod owners to PRESUBMIT.py.
2022-03-18 13:19:07 +00:00
PATENTS
…
presubmit_test_mocks.py
tools_webrtc dir converted to py3 + top level PRESUBMIT script
2022-02-08 14:42:26 +00:00
presubmit_test.py
tools_webrtc dir converted to py3 + top level PRESUBMIT script
2022-02-08 14:42:26 +00:00
PRESUBMIT.py
Update portaudio to the latest
2022-05-13 09:01:34 +00:00
pylintrc
tools_webrtc dir converted to py3 + top level PRESUBMIT script
2022-02-08 14:42:26 +00:00
README.chromium
Add CPEPrefix.
2020-07-13 11:42:07 +00:00
README.md
doc: add g3doc sitemap to toplevel readme
2021-07-23 07:55:17 +00:00
WATCHLISTS
Remove fhernqvist from watchlists
2022-08-11 14:44:52 +00:00
webrtc_lib_link_test.cc
Deprecate PeerConnectionFactory::CreatePeerConnection
2021-05-10 08:47:48 +00:00
webrtc.gni
[Cast Convergence] Replace is_chromecast with new args
2022-06-16 00:50:08 +00:00
whitespace.txt
Trigger CI.
2022-08-12 11:03:03 +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

Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
Readme BSD-3-Clause 446 MiB
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