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chromium-webrtc-autoroll 5f59dedd23 Roll chromium_revision 09356cf622..c97ba0b8b1 (1094626:1094776) 
Change log: 09356cf622..c97ba0b8b1
Full diff: 09356cf622..c97ba0b8b1

Changed dependencies
* src/base: b317709a6d..c205d5d5e2
* src/build: f87127474b..dac013f262
* src/ios: 6f952f13c9..48df4e6c61
* src/testing: 72f688213d..410234368a
* src/third_party: b4079b157f..05a37f2d47
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/1a41c2d59d..4d42668497
* src/third_party/depot_tools: bf7eb5292c..c8f23e26be
* src/third_party/perfetto: 0d57f4770c..53971a4039
* src/tools: 5622a601bb..c2192e9649
DEPS diff: 09356cf622..c97ba0b8b1/DEPS

No update to Clang.

BUG=None

Change-Id: I883e45c2ea57c169f93df3fb5fdf6a1c5788b03d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291184
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@{#39155}
2023-01-20 01:03:08 +00:00
api
Populate RTCInboundRtpStreamStats::playoutId when appropriate
2023-01-19 15:44:36 +00:00
audio
Revert "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp"
2023-01-19 11:41:42 +00:00
build_overrides
Use default values provided by PartitionAlloc instead of hard-coded ones
2022-12-07 09:11:35 +00:00
call
Ensure FakeNetwork propages arrival_time
2023-01-19 15:27:33 +00:00
common_audio
Make header files self contained.
2022-10-08 08:38:36 +00:00
common_video
Add 444 10 bits support for H264 and VP9
2023-01-17 12:32:26 +00:00
data
…
docs
Update WebRTC doc related to webrtc.org accounts.
2023-01-16 09:34:28 +00:00
examples
Remove all usage of //rtc_base target
2023-01-16 14:36:06 +00:00
experiments
Add tool for generating field trial registry header
2022-10-18 07:25:43 +00:00
g3doc
Update WebRTC style guide about GN.
2022-12-16 10:19:13 +00:00
infra
Add Fuchsia perf output and fix upload
2023-01-18 14:10:56 +00:00
logging
Remove all usage of //rtc_base target
2023-01-16 14:36:06 +00:00
media
Remove WebRTC-Dav1dDecoder kill switch.
2023-01-18 15:13:58 +00:00
modules
User Timestamp and TimeDelta instead of raw ints in RtpSenderEgress
2023-01-19 16:23:59 +00:00
net/dcsctp
Remove all usage of //rtc_base target
2023-01-16 14:36:06 +00:00
p2p
Log resolved & unresolved turn server address as sensitive string.
2023-01-19 18:26:50 +00:00
pc
Reland "Ensure RTCRtpSenders are always created with one encoding"
2023-01-19 15:49:04 +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
Add Fuchsia support to memory_usage.
2023-01-20 00:06:50 +00:00
rtc_tools
Ensure VideoRtpReplayer use new PacketReceiver::DeliverRtp packet.
2023-01-18 12:47:41 +00:00
sdk
Disable RTCCameraVideoCapturerTestsWithMockedCaptureSession.
2023-01-18 09:45:06 +00:00
stats
Populate RTCInboundRtpStreamStats::playoutId when appropriate
2023-01-19 15:44:36 +00:00
system_wrappers
[Unwrap] Migrate RtpToNtpEstimator to use RtpTimestampUnwrapper
2023-01-11 17:14:41 +00:00
test
Revert "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp"
2023-01-19 11:41:42 +00:00
tools_webrtc
Old iOS sim bots clean up.
2023-01-18 11:14:50 +00:00
video
Revert "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp"
2023-01-19 11:41:42 +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
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
Remove dimension check in SimulcastUtility::ValidSimulcastParameters
2023-01-11 13:41:55 +00:00
BUILD.gn
Add Fuchsia perf output and fix upload
2023-01-18 14:10:56 +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 09356cf622..c97ba0b8b1 (1094626:1094776)
2023-01-20 01:03:08 +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
Add infra owners file
2022-12-02 09:21:47 +00:00
OWNERS_INFRA
Add infra owners file
2022-12-02 09:21:47 +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 xooglers from WATCHLISTS and OWNERS
2022-11-30 15:33:25 +00:00
webrtc_lib_link_test.cc
Deprecate PeerConnectionFactory::CreatePeerConnection
2021-05-10 08:47:48 +00:00
webrtc.gni
Verify field trials looked up through field_trial::FindFullName
2022-10-20 10:46:01 +00:00
whitespace.txt
Trigger bots
2022-11-17 21:29:53 +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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