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chromium-webrtc-autoroll 45a933c27f Roll chromium_revision f431ec7c65..4b737679fa (1018991:1019786) 
Change log: f431ec7c65..4b737679fa
Full diff: f431ec7c65..4b737679fa

Changed dependencies
* src/base: 2f3d3763f3..e58fcd7314
* src/build: 1df60f6d93..8a45f558f0
* src/ios: 38581d5247..62e3807647
* src/testing: 7663e0cfff..cd963d3a73
* src/third_party: 4b0c114187..e6a90d267b
* src/third_party/android_build_tools/bundletool: AqsPZpWJh-ZyGraHKlbH8XgjRnmyDmolX4HhwPEo9XUC..qLkNwA6wjoqznVqaa151GelgGBP4X495n0z-jluACPcC
* src/third_party/androidx: e9shudpT-TaQTtFHEtiVZxDjU8W9VMdjLdoi0JkyALQC..JSSu4Vtq9pLuRxzIKExbKSC1XkN6OCd7mLkeX5nWB7EC
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/c76626ec30..bd235b08d6
* src/third_party/depot_tools: a83ac9395c..0e5881fb4e
* src/third_party/fuchsia-sdk/sdk: version:8.20220627.3.1..version:8.20220630.0.1
* src/third_party/libyuv: fe4a50df8e..6900494d90
* src/third_party/perfetto: d8081faeb0..2c4a70d951
* src/tools: 6adbb78ead..be263a53b0
DEPS diff: f431ec7c65..4b737679fa/DEPS

No update to Clang.

BUG=None

Change-Id: I4b5cdea789f968f370e5dd8999e8aa8a2b1294b6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267320
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@{#37383}
2022-06-30 20:44:33 +00:00
api
Change RTCEventLogFactory to have a const Create function
2022-06-28 23:48:37 +00:00
audio
Move to_queued_task.h and pending_task_safety_flag.h into public API
2022-06-17 09:20:39 +00:00
build_overrides
Roll chromium + fix: blacklist -> ignorelist for sanitizers suppressions
2021-05-27 16:16:01 +00:00
call
Update WebRTC code version (2022-06-30T04:06:24).
2022-06-30 06:08:26 +00:00
common_audio
Adopt absl::string_view in common_audio/
2022-05-13 15:00:14 +00:00
common_video
Add 420 and 422 10 bit h264 decoding.
2022-06-17 11:12:10 +00:00
data
…
docs
Add contributing.md
2021-11-03 14:59:46 +00:00
examples
Adopt absl::string_view in p2p/
2022-06-30 13:19:18 +00:00
g3doc
Refactor RepeatingTaskHandle to use absl::AnyInvocable
2022-06-30 12:22:17 +00:00
infra
Revert "Temporarily disable video_capture_tests on linux"
2022-06-29 13:35:28 +00:00
logging
Change RTCEventLogFactory to have a const Create function
2022-06-28 23:48:37 +00:00
media
When VP9 SVC is used, use SvcConfig to set max bitrate for the stream.
2022-06-29 14:23:08 +00:00
modules
in RtpRtcp configuration delete unused remote bitrate estimator
2022-06-30 14:07:49 +00:00
net/dcsctp
dcsctp: Enable message interleaving
2022-06-28 12:05:23 +00:00
p2p
Adopt absl::string_view in p2p/
2022-06-30 13:19:18 +00:00
pc
Change RTCEventLogFactory to have a const Create function
2022-06-28 23:48:37 +00:00
resources
AEC3: Changing the default for the use_conservative_tail_frequency_response flag.
2021-12-21 17:35:26 +00:00
rtc_base
Refactor RepeatingTaskHandle to use absl::AnyInvocable
2022-06-30 12:22:17 +00:00
rtc_tools
Remove //test:field_trials from RTCEventLogs viewer.
2022-06-30 12:01:48 +00:00
sdk
Switch from junit_binary to robolectric_binary.
2022-06-30 08:02:18 +00:00
stats
Implement Outbound/InboundRtpStreamStats.mid.
2022-06-17 08:44:09 +00:00
system_wrappers
Delete rtc_base/atomic_ops.h
2022-06-28 08:32:13 +00:00
test
Adopt absl::string_view in p2p/
2022-06-30 13:19:18 +00:00
tools_webrtc
Make WebRTC use third_party/libevent rather than base/third_party/libevent
2022-06-30 07:43:49 +00:00
video
Delete rtc_base/atomic_ops.h
2022-06-28 08:32:13 +00:00
.clang-format
Add IncludeBlocks to clang-format.
2021-02-03 16:29:07 +00:00
.git-blame-ignore-revs
…
.gitignore
…
.gn
Prevent jsoncpp from hiding deprecated declarations in WebRTC
2022-04-11 12:33:47 +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
Remove unused script webrtc_dashboard_upload.py
2022-03-21 12:54:42 +00:00
AUTHORS
GCC: Avoid symbol clash in RenderBuffer
2022-05-24 10:47:56 +00:00
BUILD.gn
SVC: Add end to end tests for VP8 and VP9
2022-06-22 11:07:01 +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 f431ec7c65..4b737679fa (1018991:1019786)
2022-06-30 20:44:33 +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
…
README.md
doc: add g3doc sitemap to toplevel readme
2021-07-23 07:55:17 +00:00
WATCHLISTS
Update WATCHLISTS
2021-08-23 13:37:55 +00:00
webrtc_lib_link_test.cc
Deprecate PeerConnectionFactory::CreatePeerConnection
2021-05-10 08:47:48 +00:00
webrtc.gni
[Cast Convergence] Replace is_chromecast with new args
2022-06-16 00:50:08 +00:00
whitespace.txt
Trigger CI bots
2021-12-16 17:45:31 +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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