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chromium-webrtc-autoroll 531383c653 Roll chromium_revision bbf4ff3290..3d8a0c6a22 (1148035:1148203) 
Change log: bbf4ff3290..3d8a0c6a22
Full diff: bbf4ff3290..3d8a0c6a22

Changed dependencies
* src/base: 2593c032bb..d99b5d9669
* src/build: f8e391d8c4..9f567be8ad
* src/ios: e26bbe2094..d7d0d5171c
* src/testing: e1331a79b5..1c73d7d914
* src/third_party: 37e64a17d3..6f45715e2e
* src/third_party/android_build_tools/bundletool: Xt6znihJ60Ym06wn163_2B0HOt2tNsV5fVy-BpYJZcMC..LbB0aRQ3VKjRJZmc_PD0VTZ1O34_zD92rh71aOEOEcEC
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/4f43388d90..f8c80ae9df
* src/third_party/depot_tools: 63d20c330b..02838e6f4e
* src/third_party/perfetto: 86ba039eec..ab5c58d254
* src/tools: 00f92bf881..44949d05d9
DEPS diff: bbf4ff3290..3d8a0c6a22/DEPS

No update to Clang.

BUG=None

Change-Id: I0c3160c6db761aa9f15f5ae9f504adc76e2406f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306266
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@{#40125}
2023-05-23 22:40:01 +00:00
api
RequestedResolution - Bug fix
2023-05-22 13:58:50 +00:00
audio
Pass rtcp message to RtpTransportController through newer interface
2023-05-17 17:19:23 +00:00
build_overrides
Define enable_safe_libcxx in build_overrides/build.gni.
2023-05-03 08:18:25 +00:00
call
Update WebRTC code version (2023-05-23T04:03:48).
2023-05-23 05:36:31 +00:00
common_audio
Format ^(api|call|common_audio|examples|media|net|p2p|pc)/
2023-05-03 11:09:26 +00:00
common_video
webrtc_libyuv: Add support for more video types for consistency
2023-04-24 19:06:25 +00:00
data
…
docs
fix some more minor typos
2023-05-11 12:26:25 +00:00
examples
Format ^(api|call|common_audio|examples|media|net|p2p|pc)/
2023-05-03 11:09:26 +00:00
experiments
Add tool for generating field trial registry header
2022-10-18 07:25:43 +00:00
g3doc
Add docs about adding a new test binary.
2023-03-07 11:12:33 +00:00
infra
Add video_codec_perf_tests to desktop and android perf test suites
2023-05-23 12:13:29 +00:00
logging
Use DD encoder/decoder in RTC event log encoder/parser.
2023-04-24 10:35:22 +00:00
media
Unit tests for MediaChannel creation API
2023-05-23 13:24:46 +00:00
modules
Revert "pipewire capturer: Reduce the amount of copying"
2023-05-23 17:58:08 +00:00
net/dcsctp
Format ^(api|call|common_audio|examples|media|net|p2p|pc)/
2023-05-03 11:09:26 +00:00
p2p
Format ^(api|call|common_audio|examples|media|net|p2p|pc)/
2023-05-03 11:09:26 +00:00
pc
Add test for split-mode SSRC callback
2023-05-23 07:56:57 +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
Adds WebRTC-DisableRtxRateLimiter for enable/disable RTX rate limiter.
2023-05-23 09:59:11 +00:00
rtc_tools
Format the rest
2023-05-03 12:56:39 +00:00
sdk
Add EglThread class wrapping EglConnection and handler.
2023-05-23 14:02:21 +00:00
stats
stats: remove media_type which was an alias for kind
2023-05-09 11:46:52 +00:00
system_wrappers
Format the rest
2023-05-03 12:56:39 +00:00
test
Format the rest
2023-05-03 12:56:39 +00:00
tools_webrtc
Rewrite 'generate_sslroots' w/o OpenSSL.
2023-05-10 12:57:37 +00:00
video
OveruseFrameDetector: complete removal of mac rules kill switch.
2023-05-23 09:05:53 +00:00
.clang-format
Add IncludeBlocks to clang-format.
2021-02-03 16:29:07 +00:00
.git-blame-ignore-revs
Add formatting CLs to .git-blame-ignore-revs
2023-05-07 09:27:47 +00:00
.gitignore
Add .cache to .gitignore.
2021-01-20 15:01:07 +00:00
.gn
Set use_cxx to true.
2023-05-17 06:30:04 +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
Rewrite 'generate_sslroots' w/o OpenSSL.
2023-05-10 12:57:37 +00:00
AUTHORS
Expose setCodecPreferences/getCapabilities for android
2023-05-15 19:24:15 +00:00
BUILD.gn
Add video_codec_perf_tests to desktop and android perf test suites
2023-05-23 12:13:29 +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 bbf4ff3290..3d8a0c6a22 (1148035:1148203)
2023-05-23 22:40:01 +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
Replace BuiltinVideo{Encoder,Decoder}Factory with Video{Encoder,Decoder}FactoryTemplate.
2023-03-31 19:31:50 +00:00
webrtc.gni
Replace BuiltinVideo{Encoder,Decoder}Factory with Video{Encoder,Decoder}FactoryTemplate.
2023-03-31 19:31:50 +00:00
whitespace.txt
Trigger bots
2023-05-16 08:24:54 +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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