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chromium-webrtc-autoroll 2405298a28 Roll chromium_revision 2fd3808a0e..860957b646 (1072560:1072671) 
Change log: 2fd3808a0e..860957b646
Full diff: 2fd3808a0e..860957b646

Changed dependencies
* src/base: 87ea13b25d..2d1d07fa07
* src/ios: c030f9b439..5c7df1d8aa
* src/testing: 404d10e36f..1f8966e6e2
* src/third_party: 0c347f5fad..c3da0f2e4e
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/db897bb206..79ba9244e5
* src/third_party/depot_tools: 25cf78395c..fadcbfdb27
* src/third_party/perfetto: 31d7f4d967..b25d8d9bb1
* src/tools: 09317a515a..3dcab21933
DEPS diff: 2fd3808a0e..860957b646/DEPS

No update to Clang.

BUG=None

Change-Id: Ifa4a88565e89742928451a47a47f8f6040147d77
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283788
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@{#38658}
2022-11-17 08:37:55 +00:00
api
Revert "[ACM] iSAC audio codec removed"
2022-11-16 20:40:52 +00:00
audio
pc: Add asynchronous RtpSender::SetParameters() call
2022-11-15 15:31:40 +00:00
build_overrides
Add stub for build_overrides/partition_alloc.gni
2022-08-29 12:17:02 +00:00
call
Refactor some config plumbing in call/.
2022-11-16 09:18:40 +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
doc,ios/index.md: fix Xcode project reference
2022-11-02 12:54:11 +00:00
examples
pc: Add asynchronous RtpSender::SetParameters() call
2022-11-15 15:31:40 +00:00
experiments
Add tool for generating field trial registry header
2022-10-18 07:25:43 +00:00
g3doc
Explicitly forbid dynamic_cast
2022-11-04 12:37:57 +00:00
infra
Ignore write-compiled-tests-json-to-writable-path flag.
2022-11-16 19:28:53 +00:00
logging
Add missing dependencies.
2022-10-10 15:51:33 +00:00
media
Revert "[ACM] iSAC audio codec removed"
2022-11-16 20:40:52 +00:00
modules
Unwrap the presentation timestamp before calling aom_codec_encode in LibaomAv1Encoder.
2022-11-17 08:32:18 +00:00
net/dcsctp
Add missing dependencies.
2022-10-10 10:18:37 +00:00
p2p
Convert StreamInterfaceChannel to new API
2022-11-16 10:01:28 +00:00
pc
Revert "[ACM] iSAC audio codec removed"
2022-11-16 20:40:52 +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
Revert "Remove backwards compatibility functions in openssl"
2022-11-16 14:24:11 +00:00
rtc_tools
Add --extend_run_time_duration flag to video_replay.
2022-10-19 06:16:12 +00:00
sdk
pc: Add asynchronous RtpSender::SetParameters() call
2022-11-15 15:31:40 +00:00
stats
Add power efficient stats to RTC stats
2022-11-08 08:35:47 +00:00
system_wrappers
Verify field trials looked up through field_trial::FindFullName
2022-10-20 10:46:01 +00:00
test
Revert "[ACM] iSAC audio codec removed"
2022-11-16 20:40:52 +00:00
tools_webrtc
Rename CI Android testers.
2022-11-16 15:38:21 +00:00
video
pc: Add asynchronous RtpSender::SetParameters() call
2022-11-15 15:31:40 +00:00
.clang-format
Add IncludeBlocks to clang-format.
2021-02-03 16:29:07 +00:00
.git-blame-ignore-revs
Let git-hyper-blame ignore new format cleanup.
2019-07-11 16:18:51 +00:00
.gitignore
Add .cache to .gitignore.
2021-01-20 15:01:07 +00:00
.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
Allow maxptime equal to minptime in the SDP.
2022-11-16 10:54:03 +00:00
BUILD.gn
Verify field trials looked up through field_trial::FindFullName
2022-10-20 10:46:01 +00:00
CODE_OF_CONDUCT.md
Reference "main" branches instead of "master" branches.
2021-07-15 11:07:44 +00:00
codereview.settings
Don't add webrtc-reviews@ to CC, it can be added globally on Gerrit
2018-10-25 08:19:53 +00:00
DEPS
Roll chromium_revision 2fd3808a0e..860957b646 (1072560:1072671)
2022-11-17 08:37:55 +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
Fix add some eng prod owners to PRESUBMIT.py.
2022-03-18 13:19:07 +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
Add CPEPrefix.
2020-07-13 11:42:07 +00:00
README.md
doc: add g3doc sitemap to toplevel readme
2021-07-23 07:55:17 +00:00
WATCHLISTS
Delete modules/video_processing
2022-09-30 13:50:49 +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
Triggering CI and LKGR.
2022-10-14 08:53:38 +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
Languages
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Java 2.9%
C 2.2%
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Python 1.3%
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