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chromium-webrtc-autoroll e947f5bb83 Roll chromium_revision 3f8fc6e359..ada88c87de (1025382:1025896) 
Change log: 3f8fc6e359..ada88c87de
Full diff: 3f8fc6e359..ada88c87de

Changed dependencies
* src/base: d28c12571a..cc9cb205fe
* src/build: 55fee0e565..9e6a9a9613
* src/buildtools: 5cee6a2f1c..a4506d5d63
* src/buildtools/third_party/libc++/trunk: b1269813ea..88bf407048
* src/ios: ec94f41124..4c00d7f74e
* src/testing: 4a2ba06460..06fcab2d92
* src/third_party: 2fa6453891..256411515b
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/1240e973d5..0c6b8d6722
* src/third_party/depot_tools: 8168f76753..90191b14c2
* src/third_party/fuchsia-sdk/sdk: version:8.20220713.3.1..version:9.20220719.2.1
* src/third_party/lss: https://chromium.googlesource.com/linux-syscall-support.git/+log/32a80cda3c..3bb6859795
* src/third_party/perfetto: 761031a428..ac8e40ad9e
* src/tools: 2cfda2e9bf..72c1516005
DEPS diff: 3f8fc6e359..ada88c87de/DEPS

No update to Clang.

BUG=None

Change-Id: I73a4af4e5235b550bb05c1192fd53b73aba812c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268883
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#37559}
2022-07-19 20:52:03 +00:00
api
[PCLF] Add ability to use fixed frame reate for video dump
2022-07-18 13:28:41 +00:00
audio
Delete rtp_sender_ check in ModuleRtpRtcpImpl2::SetSendingMediaStatus
2022-07-18 14:28:31 +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 (2022-07-18T04:02:25).
2022-07-18 05:31:21 +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
adopt absl::string_view in logging/
2022-07-15 16:08:39 +00:00
media
Video and flexfec receive stream config changes without recreate.
2022-07-14 20:51:08 +00:00
modules
AgcManagerDirect unit tests refactoring
2022-07-19 12:46:42 +00:00
net/dcsctp
Migrate net/dcsctp/ to absl::AnyInvocable based TaskQueueBase interface
2022-07-06 09:37:53 +00:00
p2p
[TCPConnection] Check for valid port_ in OnClose and OnConnect.
2022-07-19 15:56:43 +00:00
pc
peerconnection: measure invalid ice-chars in remote description
2022-07-14 15:29:47 +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
Migrate rtc_base and rtc_tools to absl::AnyInvocable based TaskQueueBase interface
2022-07-19 10:57:42 +00:00
rtc_tools
Add lower/upper link capacity to the outgoing bitrate graph.
2022-07-19 13:22:32 +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
In video replace non-owning pointer to rtc::TaskQueue with non-owning pointer to TaskQueueBase
2022-07-18 13:59:32 +00:00
tools_webrtc
Clobber win bots
2022-07-19 11:34:02 +00:00
video
In video replace non-owning pointer to rtc::TaskQueue with non-owning pointer to TaskQueueBase
2022-07-18 13:59:32 +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
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
Add missing header to fix build error when using linux system libraries
2022-07-19 12:25:42 +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
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 3f8fc6e359..ada88c87de (1025382:1025896)
2022-07-19 20:52:03 +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
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
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