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chromium-webrtc-autoroll a009b80d75 Roll chromium_revision 38388d3eed..0eb7e7f1f1 (1088290:1088490) 
Change log: 38388d3eed..0eb7e7f1f1
Full diff: 38388d3eed..0eb7e7f1f1

Changed dependencies
* src/base: 4067ea2b8c..eda41c82a3
* src/build: dcb82c6fe2..44b5138cf5
* src/ios: 365fc364d9..d4eaefe212
* src/testing: 65bb036271..c37f76a3df
* src/third_party: dedf3eb41e..1bf608489f
* src/third_party/androidx: fQDeJ1F444tz8xEl55yvq_R7cVqlC9-SsU1E9TD9NREC..Hjrr3kAG6HDx0Ht22sw-pZY5W_Vnltz4SdhqbBcEdw4C
* src/third_party/depot_tools: 5b0c934026..252b19866a
* src/tools: 084be5fc66..5275d1bd53
DEPS diff: 38388d3eed..0eb7e7f1f1/DEPS

No update to Clang.

BUG=None

Change-Id: Ib1f9dd230539d0dc1ec121fb0abcc10fc86a6aa4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290488
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@{#38990}
2023-01-04 00:34:57 +00:00
api
RtcEventLogImpl: Add mocked time test.
2023-01-03 14:08:16 +00:00
audio
Revert "Remove use of ReceiveStreamRtpConfig:transport_cc"
2023-01-03 16:18:08 +00:00
build_overrides
Use default values provided by PartitionAlloc instead of hard-coded ones
2022-12-07 09:11:35 +00:00
call
Revert "Remove use of ReceiveStreamRtpConfig:transport_cc"
2023-01-03 16:18:08 +00:00
common_audio
Make header files self contained.
2022-10-08 08:38:36 +00:00
common_video
Make header files self contained.
2022-10-08 08:38:36 +00:00
data
…
docs
Update webrtc build command instruction to autoninja.
2022-11-25 09:19:37 +00:00
examples
Split media_channel and its dependencies from the rtc_media_base target
2022-12-16 12:15:22 +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
Reland "Remove old Nexus perf bots"
2022-12-20 13:34:59 +00:00
logging
RtcEventLogImpl: Add mocked time test.
2023-01-03 14:08:16 +00:00
media
Revert "Remove use of ReceiveStreamRtpConfig:transport_cc"
2023-01-03 16:18:08 +00:00
modules
Add RtpPacket::SetRawExtension function.
2023-01-02 16:18:16 +00:00
net/dcsctp
Add conformance tests for integer unwrapping.
2022-12-15 13:22:18 +00:00
p2p
Don't log PW mismatch messages in production
2022-12-21 11:43:05 +00:00
pc
In preparation for track deletion, fix some more tests.
2023-01-03 12:10:40 +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
Use non recursive mutex in rtc::ThreadManager
2023-01-03 13:28:50 +00:00
rtc_tools
Generate packets of original packet length in video_replay.
2023-01-03 09:27:19 +00:00
sdk
Get RTCCameraVideoCapturerTests working again
2022-12-20 15:02:33 +00:00
stats
Add scalability mode to RTCOutboundRtpStreamStats stats
2022-12-08 11:46:06 +00:00
system_wrappers
APM: remove denormal disabler field trial
2022-12-21 11:27:02 +00:00
test
Revert "Remove use of ReceiveStreamRtpConfig:transport_cc"
2023-01-03 16:18:08 +00:00
tools_webrtc
Reland "Remove old Nexus perf bots"
2022-12-20 13:34:59 +00:00
video
Revert "Remove use of ReceiveStreamRtpConfig:transport_cc"
2023-01-03 16:18:08 +00:00
.clang-format
…
.git-blame-ignore-revs
…
.gitignore
…
.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
fixes crash caused by race derefing pixelbufferpool ivar while being destroyed and replaced by format change
2022-12-20 10:22:52 +00:00
BUILD.gn
Ensure task queues delete closures in task queue context.
2022-12-19 10:41:03 +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 38388d3eed..0eb7e7f1f1 (1088290:1088490)
2023-01-04 00:34:57 +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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