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chromium-webrtc-autoroll 84b557c5db Roll chromium_revision 39175ee90d..1d9d85466a (1053842:1053948) 
Change log: 39175ee90d..1d9d85466a
Full diff: 39175ee90d..1d9d85466a

Changed dependencies
* src/base: 5f70b3d2ba..9927712cc5
* src/build: 4743ee2b80..a77d268e20
* src/ios: 999971b8fa..c0ac06469e
* src/testing: 74f9541a22..6ff813a5ed
* src/third_party: 520771ff5a..11caf13762
* src/third_party/androidx: S60UtEBNkQs2k4ttPmH_fhp_yR5pIbvmfxIpC6VBSgcC..aq6lceEJ4fkTCId1QIp9CN9Ds-A-7vRq9Kwd3u5OENsC
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/5eb0675d9c..16eb727884
* src/third_party/depot_tools: 1b8211ff13..a724859f7a
* src/third_party/freetype/src: 905270925d..de8f14a8e4
* src/third_party/perfetto: 43b695bca3..c7e6da9718
* src/tools: 48dd2c8210..7dc87138e3
* src/tools/luci-go: git_revision:78063b01b53dd33a541938207b785cc86d34be37..git_revision:20c50aa39686d91330c2daceccaa4ef1a0a72ee4
* src/tools/luci-go: git_revision:78063b01b53dd33a541938207b785cc86d34be37..git_revision:20c50aa39686d91330c2daceccaa4ef1a0a72ee4
DEPS diff: 39175ee90d..1d9d85466a/DEPS

No update to Clang.

BUG=None

Change-Id: I29e23986eeea8e9a4699416b9dd597a1505c2089
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277616
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@{#38266}
2022-10-01 22:58:25 +00:00
api
Revise video owners
2022-09-30 08:44:30 +00:00
audio
Audio stack traces
2022-09-27 15:05:51 +00:00
build_overrides
Add stub for build_overrides/partition_alloc.gni
2022-08-29 12:17:02 +00:00
call
Update WebRTC code version (2022-09-30T04:11:14).
2022-09-30 06:51:07 +00:00
common_audio
Fix math involving enums in C++20
2022-09-27 06:55:31 +00:00
common_video
Move IncomingVideoStream closer to its only user.
2022-09-30 10:37:37 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
docs
doc: backfill M90-M105 release notes
2022-08-29 21:04:32 +00:00
examples
Replace Thread::Invoke with Thread::BlockingCall
2022-09-09 10:44:17 +00:00
g3doc
Clarify how to reference WebRTC bugs in TODOs
2022-07-01 08:03:34 +00:00
infra
Add reclient properties to perf builders
2022-09-09 09:04:35 +00:00
logging
Encode remote link capacity estimates in legacy RTC event log format
2022-08-11 12:57:02 +00:00
media
Populate ScalabilityMode of subcodecs in SimulcastEncoderAdapter.
2022-10-01 17:59:54 +00:00
modules
Delete modules/video_processing
2022-09-30 13:50:49 +00:00
net/dcsctp
Fix several UBsan issues with memcpy
2022-09-14 09:35:39 +00:00
p2p
Add test coverage for P2PTransportChannel refactor (#8/8)
2022-09-21 16:49:02 +00:00
pc
Fix race in RTCStatsCollector's cache.
2022-09-30 18:21:20 +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
Implement RTC_LOG_IF and RTC_DLOG_IF
2022-09-30 12:01:28 +00:00
rtc_tools
In video_replay, destroy call on worker thread even if input validation fail.
2022-09-30 08:43:28 +00:00
sdk
Remove the libaom av1 decoder.
2022-09-30 08:42:25 +00:00
stats
[Stats] Avoid DCHECK crashing if SSRCs are not unique.
2022-09-26 10:28:01 +00:00
system_wrappers
Delete nisse@webrtc.org from OWNERS files
2022-07-28 08:47:38 +00:00
test
Add flag to export in new MetricSet proto format
2022-09-30 13:14:09 +00:00
tools_webrtc
Do not use gtest_parallell for fuchsia
2022-09-28 12:31:14 +00:00
video
Delete modules/video_processing
2022-09-30 13:50:49 +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
Set Fuchsia Api level + update SDK version
2022-09-14 08:49:56 +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
Add python grpc to .vpython3 for ios test runner
2022-09-16 12:26:48 +00:00
AUTHORS
loongarch: Add loongarch architecture support.
2022-09-13 10:44:59 +00:00
BUILD.gn
Introduce MetricsExporter API with stdout implementation
2022-09-17 00:56:29 +00:00
CODE_OF_CONDUCT.md
Reference "main" branches instead of "master" branches.
2021-07-15 11:07:44 +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 39175ee90d..1d9d85466a (1053842:1053948)
2022-10-01 22:58:25 +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
Delete modules/video_processing
2022-09-30 13:50:49 +00:00
webrtc_lib_link_test.cc
Deprecate PeerConnectionFactory::CreatePeerConnection
2021-05-10 08:47:48 +00:00
webrtc.gni
Add Fuchsia platform deps
2022-09-23 07:49:47 +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
Languages
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C 2.2%
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Python 1.3%
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