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chromium-webrtc-autoroll 1ecac13149 Roll chromium_revision be3e47cd99..aae661725b (1148555:1148994) 
Change log: be3e47cd99..aae661725b
Full diff: be3e47cd99..aae661725b

Changed dependencies
* src/base: 26878e946c..1a5625efce
* src/build: 91fe56808c..40ec269a47
* src/buildtools: 5ba94a7ca3..edbefcee3d
* src/ios: 1df9c39f61..02f8084426
* src/testing: 21c061bfe0..6882fbb105
* src/third_party: 7bbd131784..622048e390
* src/third_party/androidx: Q3AZB5fRWzo_uiJt47UQ11_LRBtSFcurOt1Aofa9Wy4C..Ll2UQauMCmQu65qrrb_Z3qzG-sZhesVx9RnsetEDXPUC
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/b05d23a928..2ed88c5bbb
* src/third_party/depot_tools: 85c13de0be..456d085e75
* src/third_party/perfetto: 603af58208..face7c192d
* src/third_party/r8: h3mD8cigmlOPxMqeMRW8Z2Wb364bRLWUMdjhwWy_UPsC..dJL8gwEiDUoK5pirX6la4l7ofdBufwcc1HUaCv4CZ8cC
* src/tools: 55c101eb37..4aa651cbef
DEPS diff: be3e47cd99..aae661725b/DEPS

No update to Clang.

BUG=None

Change-Id: Iaf62f065a4479ce8506b2cbf78673e6e8d59e0a3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306540
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@{#40142}
2023-05-25 11:20:24 +00:00
api
Delete RTC[NonStandard/Restricted]StatsMember.
2023-05-25 08:39:48 +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
Fix video version of RTCInboundRtpStreamStats.jitterBufferDelay to obey spec.
2023-05-25 07:33:39 +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
docs: explain release note process
2023-05-24 14:09:54 +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
Fix video version of RTCInboundRtpStreamStats.jitterBufferDelay to obey spec.
2023-05-25 07:33:39 +00:00
modules
In RtcpTransciever refactor outgoing transport interface
2023-05-24 14:14:53 +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
Delete RTC[NonStandard/Restricted]StatsMember.
2023-05-25 08:39:48 +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
Remove unused Win32Window class
2023-05-25 10:48:40 +00:00
rtc_tools
Format the rest
2023-05-03 12:56:39 +00:00
sdk
Adopt EglThread in EglRenderer
2023-05-25 11:03:53 +00:00
stats
Delete RTC[NonStandard/Restricted]StatsMember.
2023-05-25 08:39:48 +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
Fix video version of RTCInboundRtpStreamStats.jitterBufferDelay to obey spec.
2023-05-25 07:33:39 +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 be3e47cd99..aae661725b (1148555:1148994)
2023-05-25 11:20:24 +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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