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chromium-webrtc-autoroll b7531ef7ef Roll chromium_revision 5f70135093..ea8a94b309 (1047943:1048110) 
Change log: 5f70135093..ea8a94b309
Full diff: 5f70135093..ea8a94b309

Changed dependencies
* src/base: b9c7aac9b1..ca4eeacb2c
* src/build: 809ce4b9ec..48abf311f6
* src/buildtools/linux64: git_revision:e70d8c3d5620bc0ddcbad23a36b1b26f815ca90a..git_revision:cc28efe62ef0c2fb32455f414a29c4a55bb7fbc4
* src/buildtools/mac: git_revision:e70d8c3d5620bc0ddcbad23a36b1b26f815ca90a..git_revision:cc28efe62ef0c2fb32455f414a29c4a55bb7fbc4
* src/buildtools/third_party/libunwind/trunk: 60a480ee18..77b82eb53f
* src/buildtools/win: git_revision:e70d8c3d5620bc0ddcbad23a36b1b26f815ca90a..git_revision:cc28efe62ef0c2fb32455f414a29c4a55bb7fbc4
* src/ios: 2bbe76d6e2..a39bf788d4
* src/testing: 6235452f8c..7b1540d785
* src/third_party: 8819efcddb..7fef8181f7
* src/third_party/androidx: JhE-AIohCaCjpsq8dv30ZYeZmDQ4qebqPEvXPcbNsScC..Is8uteQpDOZkyPAYfFDv2xzhwkI22rad5XI774XLUqMC
* src/third_party/perfetto: deca583af1..de4b5d49c1
* src/third_party/r8/d8: 4DSZMtRXZITDo6YY90ljp92vzRT0eY52akTTR-2Jh7kC..9D5mlKXcvEeq8EeDZf3NXnhO5BuGy0eDy5q1hLou4WwC
* src/tools: 1b73b1f06a..bdc9e5f900
DEPS diff: 5f70135093..ea8a94b309/DEPS

No update to Clang.

BUG=None

Change-Id: Id60ba8b56b0a83b8a946523ae6271a3730b851b7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275745
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@{#38105}
2022-09-16 18:47:28 +00:00
api
RtpSource: remove deprecated ctor, use designated initializers
2022-09-15 07:42:27 +00:00
audio
Replace ChannelReceive::GetRTT() with ModuleRtpRtcpImpl2::RTT()
2022-09-09 13:24:38 +00:00
build_overrides
Add stub for build_overrides/partition_alloc.gni
2022-08-29 12:17:02 +00:00
call
Revert "rtp sender: don't send BYE on deactivating streams"
2022-09-16 09:40:18 +00:00
common_audio
Reenable WebRTC PushResampler format checks on Windows clang debug builds
2022-08-05 11:03:08 +00:00
common_video
Migrate remaining webrtc usage of TaskQueueBase to absl::AnyInvocable
2022-07-20 08:15:08 +00:00
data
…
docs
doc: backfill M90-M105 release notes
2022-08-29 21:04:32 +00:00
examples
Replace Thread::Invoke with Thread::BlockingCall
2022-09-09 10:44:17 +00:00
g3doc
Clarify how to reference WebRTC bugs in TODOs
2022-07-01 08:03:34 +00:00
infra
Add reclient properties to perf builders
2022-09-09 09:04:35 +00:00
logging
Encode remote link capacity estimates in legacy RTC event log format
2022-08-11 12:57:02 +00:00
media
Add AV1 profile-1 video decode support to WebRTC
2022-09-06 16:31:40 +00:00
modules
Remove forward declares
2022-09-16 13:33:08 +00:00
net/dcsctp
Fix several UBsan issues with memcpy
2022-09-14 09:35:39 +00:00
p2p
Fix P2PTransportChannel unit tests to not rely on connections ordering
2022-09-15 08:15:47 +00:00
pc
Delete UMA histograms relating to Plan B vs Unified Plan.
2022-09-13 14:19:29 +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
Unbind VirtualSocket from rtc::MessageHandler
2022-09-16 14:13:41 +00:00
rtc_tools
AEC dump Stream::level renamed
2022-09-09 14:39:35 +00:00
sdk
Renamed methods.
2022-09-15 14:08:22 +00:00
stats
stats: replace new with std::make_unique
2022-09-07 11:06:19 +00:00
system_wrappers
Delete nisse@webrtc.org from OWNERS files
2022-07-28 08:47:38 +00:00
test
Replace Thread::Invoke with Thread::BlockingCall
2022-09-09 10:44:17 +00:00
tools_webrtc
Reset cache on Windows bots.
2022-09-14 07:07:30 +00:00
video
VideoStreamEncoder: set at target quality based on codec.
2022-09-15 12:15:17 +00:00
.clang-format
Add IncludeBlocks to clang-format.
2021-02-03 16:29:07 +00:00
.git-blame-ignore-revs
…
.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
loongarch: Add loongarch architecture support.
2022-09-13 10:44:59 +00:00
BUILD.gn
Move TimestampExtrapolator closer to its single user
2022-09-15 13:49:18 +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 5f70135093..ea8a94b309 (1047943:1048110)
2022-09-16 18:47:28 +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
Remove fhernqvist from watchlists
2022-08-11 14:44:52 +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.
2022-08-12 11:03:03 +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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