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chromium-webrtc-autoroll b3c8327850 Roll chromium_revision 8f07eaff87..d59bd71380 (1086489:1086619) 
Change log: 8f07eaff87..d59bd71380
Full diff: 8f07eaff87..d59bd71380

Changed dependencies
* src/build: f341382131..ff6be8b34d
* src/buildtools/third_party/libc++abi/trunk: df3cc8ecee..dc82f3042d
* src/ios: ce3e7578ff..9e6837ab1c
* src/third_party: 77cd61d592..aacc97fce4
* src/third_party/androidx: XNQCcCxsa8Vznu546BS4QiWwlsCp1_1rV1J_5rSghKwC..Q0RddCfn0BxFtUdiySojmUsDxQcuYrX4W1JRRHXk-30C
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/b10b1305ce..ebbb83f192
* src/third_party/depot_tools: cf31045b34..0b96058844
* src/tools: 6ff35895f0..220a380655
DEPS diff: 8f07eaff87..d59bd71380/DEPS

No update to Clang.

BUG=None

Change-Id: I511a8f31d4de6294558f6ca7e821b82f910dda31
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/289022
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@{#38948}
2022-12-23 02:34:09 +00:00
api
Don't use low complexity Opus on all ARM devices.
2022-12-20 10:30:30 +00:00
audio
Revert "Enforce stream id uniqueness in RtpSender::set_stream_ids"
2022-12-21 13:56:05 +00:00
build_overrides
Use default values provided by PartitionAlloc instead of hard-coded ones
2022-12-07 09:11:35 +00:00
call
Change RecoveredPacket::OnRecoveredPacket to produce webrtc::RtpPacketReceived
2022-12-22 14:04:21 +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
Update webrtc build command instruction to autoninja.
2022-11-25 09:19:37 +00:00
examples
Split media_channel and its dependencies from the rtc_media_base target
2022-12-16 12:15:22 +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
Reland "Remove old Nexus perf bots"
2022-12-20 13:34:59 +00:00
logging
Add missing dependencies.
2022-10-10 15:51:33 +00:00
media
[Unwrap] Remove unused TimestampWrapAroundHandler from VideoChannel
2022-12-21 14:01:50 +00:00
modules
Free memory allocated by GetStreamCaps
2022-12-22 14:46:08 +00:00
net/dcsctp
Add conformance tests for integer unwrapping.
2022-12-15 13:22:18 +00:00
p2p
Don't log PW mismatch messages in production
2022-12-21 11:43:05 +00:00
pc
Enforce stream id uniqueness in RtpSender::set_stream_ids
2022-12-22 10:05:02 +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
Extend DivideRoundToNearest to support negative numbers and use for UnitBase
2022-12-22 11:02:29 +00:00
rtc_tools
Split media_channel and its dependencies from the rtc_media_base target
2022-12-16 12:15:22 +00:00
sdk
Get RTCCameraVideoCapturerTests working again
2022-12-20 15:02:33 +00:00
stats
Add scalability mode to RTCOutboundRtpStreamStats stats
2022-12-08 11:46:06 +00:00
system_wrappers
APM: remove denormal disabler field trial
2022-12-21 11:27:02 +00:00
test
Change RecoveredPacket::OnRecoveredPacket to produce webrtc::RtpPacketReceived
2022-12-22 14:04:21 +00:00
tools_webrtc
Reland "Remove old Nexus perf bots"
2022-12-20 13:34:59 +00:00
video
Change RecoveredPacket::OnRecoveredPacket to produce webrtc::RtpPacketReceived
2022-12-22 14:04:21 +00:00
.clang-format
…
.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
fixes crash caused by race derefing pixelbufferpool ivar while being destroyed and replaced by format change
2022-12-20 10:22:52 +00:00
BUILD.gn
Ensure task queues delete closures in task queue context.
2022-12-19 10:41:03 +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 8f07eaff87..d59bd71380 (1086489:1086619)
2022-12-23 02:34:09 +00:00
DIR_METADATA
…
ENG_REVIEW_OWNERS
Remove phoglund from ENG_REVIEW_OWNERS
2021-10-08 08:29:42 +00:00
g3doc.lua
…
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
…
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
Languages
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