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chromium-webrtc-autoroll 25ffa806e8 Roll chromium_revision 8496a82e1d..5c3b57c4c8 (1093621:1093721) 
Change log: 8496a82e1d..5c3b57c4c8
Full diff: 8496a82e1d..5c3b57c4c8

Changed dependencies
* src/base: 1f7bc02188..5e68eb8f03
* src/build: 14b0a21a0a..e831815137
* src/buildtools: a1adda97a8..d843e69371
* src/ios: 0971740636..ae4ca26382
* src/testing: baa990ec41..2bc38e6394
* src/third_party: b085d968b0..6aa94f676f
* src/third_party/android_build_tools/manifest_merger: 7ZPeHZjITxCcJzrEuxb5yznF7h65-RTQrbhzILJz4_gC..Oe3FpLcNFdPYOQQYUNnC4ajNSBfgmsFHDUaAimk7m6MC
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/a4c817894b..7a311fe439
* src/third_party/depot_tools: 4f50adb332..86cfa62b07
* src/tools: 9eb8161069..38d6645ded
DEPS diff: 8496a82e1d..5c3b57c4c8/DEPS

No update to Clang.

BUG=None

Change-Id: Ib756b93a2316ee1a0331439a63c868ab28bfa86a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291068
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@{#39129}
2023-01-18 04:43:23 +00:00
api
Add 444 10 bits support for H264 and VP9
2023-01-17 12:32:26 +00:00
audio
Avoid waking up encoder thread when audio send stream is stopped.
2023-01-17 15:52:45 +00:00
build_overrides
Use default values provided by PartitionAlloc instead of hard-coded ones
2022-12-07 09:11:35 +00:00
call
Update WebRTC code version (2023-01-17T04:11:09).
2023-01-17 05:57:42 +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
Remove 'iOS Debug (simulator)' from LKGR.
2023-01-13 11:02:01 +00:00
logging
Remove all usage of //rtc_base target
2023-01-16 14:36:06 +00:00
media
Only fill send/recv stats if there are send/receive streams
2023-01-17 11:44:32 +00:00
modules
Add 444 10 bits support for H264 and VP9
2023-01-17 12:32:26 +00:00
net/dcsctp
Remove all usage of //rtc_base target
2023-01-16 14:36:06 +00:00
p2p
Remove all usage of //rtc_base target
2023-01-16 14:36:06 +00:00
pc
Ensure RTCRtpSenders are always created with one encoding
2023-01-17 18:05:49 +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 time functions for Fuchsia.
2023-01-16 23:06:21 +00:00
rtc_tools
Remove all usage of //rtc_base target
2023-01-16 14:36:06 +00:00
sdk
Remove all usage of //rtc_base target
2023-01-16 14:36:06 +00:00
stats
Add RTCAudioPlayoutStats to GetStats().
2023-01-16 13:19:45 +00:00
system_wrappers
[Unwrap] Migrate RtpToNtpEstimator to use RtpTimestampUnwrapper
2023-01-11 17:14:41 +00:00
test
Remove all usage of //rtc_base target
2023-01-16 14:36:06 +00:00
tools_webrtc
Add "iOS Debug (simulator)" CI bot.
2023-01-13 08:16:10 +00:00
video
Remove all usage of //rtc_base target
2023-01-16 14:36:06 +00:00
.clang-format
Add IncludeBlocks to clang-format.
2021-02-03 16:29:07 +00:00
.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
Remove dimension check in SimulcastUtility::ValidSimulcastParameters
2023-01-11 13:41:55 +00:00
BUILD.gn
Remove all usage of //rtc_base target
2023-01-16 14:36:06 +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 8496a82e1d..5c3b57c4c8 (1093621:1093721)
2023-01-18 04:43:23 +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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