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chromium-webrtc-autoroll 063f68a612 Roll chromium_revision a0f406407b..1a7287fea3 (1107323:1107431) 
Change log: a0f406407b..1a7287fea3
Full diff: a0f406407b..1a7287fea3

Changed dependencies
* fuchsia_vesion: version:11.20230219.2.1..version:11.20230220.1.1
* src/base: 17550eadaf..e94131d14d
* src/ios: eabc1ecd0b..29b8b80521
* src/testing: 9686404aed..c58cac70c7
* src/third_party: b3ece5d0a3..fb714f117d
* src/third_party/androidx: MZ9_PEpIzKK0mJj-ofqMi2FNv_6yzhVurgpyh5pHkK8C..5XJ66D6bMy33f2PGbWb1DSMYbCeqjKLqPB5CJXuhL9wC
* src/third_party/r8: EzD88AUfNs2DWG8ytv86PmRrgkNmeLLaiQOrNNLUWwAC..zPOmDMk4QXwcUPUy6oi9KUfzwEIlk0aIB0fl0KQWCGAC
* src/tools: 23ee8d2022..d9c579e7ee
DEPS diff: a0f406407b..1a7287fea3/DEPS

No update to Clang.

BUG=None

Change-Id: Ie38e40e95ce420a405753878009503afaf01b7c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294301
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@{#39351}
2023-02-20 16:20:36 +00:00
api
Introduce WebRTC-VideoEncoderSettings/encoder_thread_limit:X.
2023-02-20 14:01:32 +00:00
audio
Create unit test for the population of capture_start_ntp_time
2023-02-06 14:00:39 +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 "Make SimulcastIndex() and SpatialIndex() distinct (remove fallback)."
2023-02-20 12:47:37 +00:00
common_audio
Check FMA3 support before use it in SincResampler
2023-01-31 17:28:55 +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
Fix doc path
2023-01-31 10:14:47 +00:00
infra
Add a new webrtc_linux_chromium bot.
2023-02-16 07:47:01 +00:00
logging
RtcEventLogImpl: Add test cases
2023-02-03 09:55:33 +00:00
media
Change FakeMediaEngine to use send/receive channels
2023-02-20 11:27:33 +00:00
modules
Build video capture implementation for chromium
2023-02-20 15:29:43 +00:00
net/dcsctp
Remove all usage of //rtc_base target
2023-01-16 14:36:06 +00:00
p2p
Add an ICE switch reason for a switch requested by an application.
2023-02-06 16:19:49 +00:00
pc
Change FakeMediaEngine to use send/receive channels
2023-02-20 11:27:33 +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
Changed OutputToDebug to create a CFString at compile-time, rather than runtime
2023-02-19 22:41:59 +00:00
rtc_tools
Handling NetEqSetMinimumDelay events in neteq_rtpplay.
2023-02-09 09:39:29 +00:00
sdk
Handle frame_types=null in VideoEncoderWrapper::Encode()
2023-02-16 11:09:51 +00:00
stats
stats: remove deprecated timestamp_us constructor variant
2023-02-20 12:18:42 +00:00
system_wrappers
Check FMA3 support before use it in SincResampler
2023-01-31 17:28:55 +00:00
test
Update documentation links in test/pc/e2e/g3doc/default_video_quality_analyzer.md
2023-02-20 13:09:45 +00:00
tools_webrtc
Noop change to trigger bots
2023-02-13 10:30:38 +00:00
video
Introduce WebRTC-VideoEncoderSettings/encoder_thread_limit:X.
2023-02-20 14:01:32 +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
Changed OutputToDebug to create a CFString at compile-time, rather than runtime
2023-02-19 22:41:59 +00:00
BUILD.gn
Delete unused Audio Bwe integration test.
2023-01-26 09:31:44 +00:00
CODE_OF_CONDUCT.md
Reland "Migrate WebRTC documentation to new renderer"
2023-01-31 09:30:04 +00:00
codereview.settings
…
DEPS
Roll chromium_revision a0f406407b..1a7287fea3 (1107323:1107431)
2023-02-20 16:20:36 +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
LICENSE
…
license_template.txt
…
native-api.md
Reland "Migrate WebRTC documentation to new renderer"
2023-01-31 09:30:04 +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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