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chromium-webrtc-autoroll 6fbd123d3f Roll chromium_revision c3cc09166c..58c59af8eb (1042192:1042333) 
Change log: c3cc09166c..58c59af8eb
Full diff: c3cc09166c..58c59af8eb

Changed dependencies
* src/base: ca4f7e8937..4740e2c0f5
* src/build: 127c45016e..d69d018432
* src/buildtools: c24a0d5e7d..734683a3d6
* src/buildtools/third_party/libc++/trunk: 42e738f0a1..84f06932d7
* src/ios: 13ddbd7e39..4c4846462b
* src/testing: f6cace01fd..21be381823
* src/third_party: 2dbcab74cc..6f7b715c31
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/10fef972e4..19009c51bf
* src/third_party/libaom/source/libaom: https://aomedia.googlesource.com/aom.git/+log/2fcb0760ea..ff7b753a63
* src/third_party/libvpx/source/libvpx: 8786aee582..9d6d0624d7
* src/third_party/perfetto: 13c35692e8..df569f9bea
* src/tools: 5c7568b09a..b2e03336e4
DEPS diff: c3cc09166c..58c59af8eb/DEPS

No update to Clang.

BUG=None

Change-Id: I63156f85957380eb2a0203d412e6358e83f85ce8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273841
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@{#37985}
2022-09-01 23:05:18 +00:00
api
peerconnection: invalidate stats cache during SLD/SRD
2022-09-01 15:18:27 +00:00
audio
Update rtc::Event::Wait call sites to use TimeDelta.
2022-08-19 10:07:28 +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 (2022-09-01T04:05:03).
2022-09-01 05:21:07 +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
doc: backfill M90-M105 release notes
2022-08-29 21:04:32 +00:00
examples
SocketServer: Migrate Wait/kForever to TimeDelta.
2022-08-25 13:01:34 +00:00
g3doc
Clarify how to reference WebRTC bugs in TODOs
2022-07-01 08:03:34 +00:00
infra
Remove py3 experiment since its now default
2022-08-31 07:54:53 +00:00
logging
Encode remote link capacity estimates in legacy RTC event log format
2022-08-11 12:57:02 +00:00
media
dcsctp: Track open channels accurately
2022-08-23 14:32:28 +00:00
modules
Conditionally include differ_vector_sse2.h only when on x86 platforms.
2022-09-01 17:22:47 +00:00
net/dcsctp
dcsctp: Expose negotiated stream counts
2022-08-23 08:51:38 +00:00
p2p
Improve IPv6 selection logic when gathering candidates.
2022-08-29 10:51:28 +00:00
pc
peerconnection: invalidate stats cache during SLD/SRD
2022-09-01 15:18:27 +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
Cleanup: Make AsyncResolveInterface::Start(addr,family) pure virtual
2022-08-30 10:09:32 +00:00
rtc_tools
Reland "Add TaskQueueStdlib experiment."
2022-08-29 10:48:42 +00:00
sdk
Reland "Reland "ObjC ADM: record/play implementation via RTCAudioDevice [3/3]""
2022-09-01 08:18:38 +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
[PCLF] Add ability to specifiy DegradationPreference
2022-08-30 09:45:41 +00:00
tools_webrtc
Add --use-remoteexec to build_ios_libs.py
2022-09-01 10:58:17 +00:00
video
FrameCadenceAdapter: survive layer updates for unconfigured layers.
2022-09-01 13:37:17 +00:00
.clang-format
…
.git-blame-ignore-revs
…
.gitignore
…
.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
Add -Wctad-maybe-unsupported.
2022-09-01 17:20:57 +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 c3cc09166c..58c59af8eb (1042192:1042333)
2022-09-01 23:05:18 +00:00
DIR_METADATA
…
ENG_REVIEW_OWNERS
Remove phoglund from ENG_REVIEW_OWNERS
2021-10-08 08:29:42 +00:00
g3doc.lua
…
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
…
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
Languages
C++ 90.3%
Java 2.9%
C 2.2%
Objective-C++ 2%
Python 1.3%
Other 1%