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chromium-webrtc-autoroll 09196e44af Roll chromium_revision 0f2fa982fe..ef902d1658 (1023827:1023960) 
Change log: 0f2fa982fe..ef902d1658
Full diff: 0f2fa982fe..ef902d1658

Changed dependencies
* src/base: eea0d5913b..abf6ba1ea2
* src/build: a900a40416..dc2ab03404
* src/ios: 1b5251bfea..feccebf89f
* src/testing: 7871ca2750..54a2b17279
* src/third_party: 3b5cd729ed..19e3565577
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/8e958343e5..c49cba8235
* src/third_party/fuchsia-sdk/sdk: version:8.20220713.1.1..version:8.20220713.2.1
* src/third_party/perfetto: 00a6e13924..29a81e33e6
* src/tools: 7746775e79..3c30b667e7
DEPS diff: 0f2fa982fe..ef902d1658/DEPS

No update to Clang.

BUG=None

Change-Id: Ic90d10f1e5d3c197d7a88ddb7dddf739666f1cf3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268500
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@{#37516}
2022-07-13 22:39:30 +00:00
api
Compare only SdpVideoFormat::name and SdpVideoFormat::parameters in the VideoEncoderFactoryTemplate.
2022-07-13 10:54:00 +00:00
audio
Update old TODO comments
2022-07-05 09:59:33 +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 (2022-07-13T04:05:07).
2022-07-13 05:47:31 +00:00
common_audio
Adopt absl::string_view in common_audio/
2022-05-13 15:00:14 +00:00
common_video
Delete useless test fixture H264SpsParserTest
2022-07-05 08:10:43 +00:00
data
…
docs
[ios] Remove the support for bitcode
2022-07-04 09:01:52 +00:00
examples
Delete ProcessThread and related Module interface
2022-07-04 10:20:35 +00:00
g3doc
Clarify how to reference WebRTC bugs in TODOs
2022-07-01 08:03:34 +00:00
infra
Reland "Wait for frames to arrive in WgcCapturer instead of returning nothing."
2022-07-06 20:28:26 +00:00
logging
Change RTCEventLogFactory to have a const Create function
2022-06-28 23:48:37 +00:00
media
Make dav1d the default AV1 decoder.
2022-07-07 14:04:46 +00:00
modules
Revert "AgcManagerDirect: stop enforcing min mic level override with 0 level"
2022-07-13 20:58:07 +00:00
net/dcsctp
Migrate net/dcsctp/ to absl::AnyInvocable based TaskQueueBase interface
2022-07-06 09:37:53 +00:00
p2p
Migrate p2p/ to absl::AnyInvocable based TaskQueueBase interface
2022-07-07 13:03:44 +00:00
pc
Disallow invalid arguments in RestoreEncodingLayers.
2022-07-13 10:55:03 +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
GCC: make UnitBase::operator*(double) constexpr
2022-07-08 13:21:51 +00:00
rtc_tools
Add default conversational speech file to the .rodata section.
2022-07-06 07:30:43 +00:00
sdk
Update old TODO comments
2022-07-05 09:09:44 +00:00
stats
Implement RTCInboundRTPStreamStats.JitterBufferTargetDelay
2022-07-05 11:34:53 +00:00
system_wrappers
Delete rtc_base/atomic_ops.h
2022-06-28 08:32:13 +00:00
test
Remove unnecessary overload in RtcEventLogOutput
2022-07-12 22:09:36 +00:00
tools_webrtc
Remove unused dependencies
2022-07-07 07:13:24 +00:00
video
Remove NackSender argument from RtpVideoStreamReceiver2.
2022-07-08 13:30:44 +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
Update protobuf-py2_py3 wheel.
2022-07-01 15:17:36 +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 0f2fa982fe..ef902d1658 (1023827:1023960)
2022-07-13 22:39:30 +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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C 2.2%
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