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chromium-webrtc-autoroll 348741b8b2 Roll chromium_revision f21fda010d..9cb220fd9f (1110889:1111467) 
Change log: f21fda010d..9cb220fd9f
Full diff: f21fda010d..9cb220fd9f

Changed dependencies
* src/base: a438ab5b41..71fa44c956
* src/build: 0c4c8e9d71..bb2ff97c7a
* src/buildtools: 121e823614..3d5742efd7
* src/ios: bad2228975..d84f2d6bab
* src/testing: b52f35f0f7..c7e279e566
* src/third_party: 95e6581764..0fa08ec507
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/dc83184d4c..a589e9031f
* src/third_party/dav1d/libdav1d: ed63a74593..0d6a31f275
* src/third_party/depot_tools: f8cab106a4..424814829a
* src/third_party/freetype/src: 2257f9abf6..638df2fe32
* src/third_party/grpc/src: a017e9b7f2..e022a3dfa9
* src/third_party/perfetto: 7f315a3f97..cecf0090dd
* src/tools: 77aff8f176..18df9bf9ef
DEPS diff: f21fda010d..9cb220fd9f/DEPS

No update to Clang.

BUG=None

Change-Id: I05e73ed7269272477c4823df4b60a877040dd7d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295473
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@{#39428}
2023-03-01 08:48:50 +00:00
api
Aggregate and log video codec metrics
2023-03-01 08:27:54 +00:00
audio
Delete RtpRtcpInterface::RemoteNtp as redundant to GetSenderReportStats
2023-02-28 13:55:27 +00:00
build_overrides
Use default values provided by PartitionAlloc instead of hard-coded ones
2022-12-07 09:11:35 +00:00
call
Update WebRTC code version (2023-02-27T04:10:52).
2023-02-27 06:07:28 +00:00
common_audio
Check FMA3 support before use it in SincResampler
2023-01-31 17:28:55 +00:00
common_video
webrtc_libyuv: Raise warnings for unhandled types at compile time
2023-02-27 22:11:02 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
docs
Update documentation links in docs/native-code/development/index.md
2023-02-20 16:35:23 +00:00
examples
Remove all usage of //rtc_base target
2023-01-16 14:36:06 +00:00
experiments
Add tool for generating field trial registry header
2022-10-18 07:25:43 +00:00
g3doc
Fix doc path
2023-01-31 10:14:47 +00:00
infra
Remove use of refs/heads/master mirror in WebRTC infra.
2023-02-21 09:24:41 +00:00
logging
RtcEventLogImpl: Add test cases
2023-02-03 09:55:33 +00:00
media
Reland "Remove ISAC media constant and payload type mapping"
2023-02-28 15:45:23 +00:00
modules
Aggregate and log video codec metrics
2023-03-01 08:27:54 +00:00
net/dcsctp
Remove all usage of //rtc_base target
2023-01-16 14:36:06 +00:00
p2p
Replace use of test-only connections() with P2PTransportChannel member.
2023-02-27 16:49:05 +00:00
pc
Add simulcast test coverage for H264.
2023-02-28 17:17:50 +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
Add option to configure AV1 EncoderInfo resolution_bitrate_limits.
2023-02-28 20:48:33 +00:00
rtc_tools
Handling NetEqSetMinimumDelay events in neteq_rtpplay.
2023-02-09 09:39:29 +00:00
sdk
Reland "Remove ISAC media constant and payload type mapping"
2023-02-28 15:45:23 +00:00
stats
stats: Deprecate RTCStatsReport(int64 timestamp_us)
2023-02-22 12:32:02 +00:00
system_wrappers
Add option to log a warning for unregistered field trials
2023-02-28 15:43:18 +00:00
test
Launch WebRTC-SendPacketsOnWorkerThread
2023-02-28 18:03:59 +00:00
tools_webrtc
Noop change to trigger bots
2023-02-13 10:30:38 +00:00
video
Launch WebRTC-SendPacketsOnWorkerThread
2023-02-28 18:03:59 +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
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
Changed OutputToDebug to create a CFString at compile-time, rather than runtime
2023-02-19 22:41:59 +00:00
BUILD.gn
Add option to log a warning for unregistered field trials
2023-02-28 15:43:18 +00:00
CODE_OF_CONDUCT.md
Reland "Migrate WebRTC documentation to new renderer"
2023-01-31 09:30:04 +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 f21fda010d..9cb220fd9f (1110889:1111467)
2023-03-01 08:48:50 +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
LICENSE
…
license_template.txt
…
native-api.md
Reland "Migrate WebRTC documentation to new renderer"
2023-01-31 09:30:04 +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
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 xooglers from WATCHLISTS and OWNERS
2022-11-30 15:33:25 +00:00
webrtc_lib_link_test.cc
Deprecate PeerConnectionFactory::CreatePeerConnection
2021-05-10 08:47:48 +00:00
webrtc.gni
Add option to log a warning for unregistered field trials
2023-02-28 15:43:18 +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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