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chromium-webrtc-autoroll d2c15aa550 Roll chromium_revision 92aea4500e..e0c02c1406 (1095006:1095203) 
Change log: 92aea4500e..e0c02c1406
Full diff: 92aea4500e..e0c02c1406

Changed dependencies
* src/base: 50eaeff67e..718df78ad3
* src/build: dac013f262..0c3dd023b8
* src/ios: 2ad8e2ea06..accdd4e600
* src/testing: a5512ff5ec..3892949280
* src/third_party: 35350960af..3d573eed4a
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/3251ca1f63..45b8d7bbd7
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/e4385f1ba0..5eeb59b9e9
* src/third_party/depot_tools: aec39c38f8..7c6ebe7fd1
* src/third_party/perfetto: 53971a4039..61f6f28a07
* src/third_party/r8: P1d7YJypbKC2Tv6UCo5QRdo1UwIyECszSGO583AzYigC..-zULvtoUlL3ZlX-wYBRJuptakgqclqk9dpsQizt8gysC
* src/tools: 50ee6c1a8c..b6ce0136e2
DEPS diff: 92aea4500e..e0c02c1406/DEPS

No update to Clang.

BUG=None

Change-Id: I674fa11be6093820fe34c6e80734a7f3c691ce8e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291264
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#39166}
2023-01-20 23:12:33 +00:00
api
mark RTPHeader struct RTC_EXPORT
2023-01-20 15:46:01 +00:00
audio
Reland "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp"
2023-01-20 06:32:29 +00:00
build_overrides
Use default values provided by PartitionAlloc instead of hard-coded ones
2022-12-07 09:11:35 +00:00
call
Reland "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp"
2023-01-20 06:32:29 +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
Remove old data files.
2018-10-05 14:40:21 +00:00
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
Retune AGC2 input volume controller speech ratio threshold
2023-01-20 14:03:58 +00:00
net/dcsctp
Remove all usage of //rtc_base target
2023-01-16 14:36:06 +00:00
p2p
Add some RTC_EXPORT needed by Chromium.
2023-01-20 18:29:08 +00:00
pc
Handle the case of missing certificates.
2023-01-20 13:31:07 +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 some RTC_EXPORT needed by Chromium.
2023-01-20 18:29:08 +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
Reland "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp"
2023-01-20 06:32:29 +00:00
tools_webrtc
Old iOS sim bots clean up.
2023-01-18 11:14:50 +00:00
video
Avoid log spamming when the dependency descriptor fail to parse.
2023-01-20 17:29:38 +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
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
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 92aea4500e..e0c02c1406 (1095006:1095203)
2023-01-20 23:12:33 +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
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 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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