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chromium-webrtc-autoroll 19a04c641f Roll chromium_revision 74ff8663a2..ddbda5d5fb (1110600:1110778) 
Change log: 74ff8663a2..ddbda5d5fb
Full diff: 74ff8663a2..ddbda5d5fb

Changed dependencies
* src/base: ad657c8fc9..ea7258d7fb
* src/buildtools: bdd20ec72c..121e823614
* src/ios: 39e0c45e42..d4aaedb860
* src/testing: 668569d3d7..7bb027b853
* src/third_party: b99154c7a8..72d03875bd
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/8a26fb906e..dc83184d4c
* src/third_party/depot_tools: 03b4ba83f3..7bf7e06945
* src/third_party/libaom/source/libaom: https://aomedia.googlesource.com/aom.git/+log/70b12695e1..ba3388ace3
* src/tools: 1ee21deacc..08977baeed
* src/tools/luci-go: git_revision:8a8b4f2ea65c7ff5fde8a0c522008aed78d42d9d..git_revision:67696c7e93f9dbf03f7bcf7f8fafe9c5728e1ebe
* src/tools/luci-go: git_revision:8a8b4f2ea65c7ff5fde8a0c522008aed78d42d9d..git_revision:67696c7e93f9dbf03f7bcf7f8fafe9c5728e1ebe
DEPS diff: 74ff8663a2..ddbda5d5fb/DEPS

No update to Clang.

BUG=None

Change-Id: Ie28e7b1423bedfe18fc622eed4a37851fd58ce8a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295421
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@{#39410}
2023-02-28 06:33:12 +00:00
api
Make FrameGeneratorInterface::fps() pure virtual.
2023-02-27 17:39:27 +00:00
audio
Create unit test for the population of capture_start_ntp_time
2023-02-06 14:00:39 +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
…
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 interim .h files from rtc_media_base"
2023-02-27 15:41:38 +00:00
modules
Add TransformableVideoFrameInterface::Metadata()
2023-02-27 15:38:32 +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 plumbing for video NACK to be coupled between channels.
2023-02-22 14:54:38 +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
Implement support for Chrome task origin tracing. #2/4
2023-02-27 13:17:38 +00:00
rtc_tools
Handling NetEqSetMinimumDelay events in neteq_rtpplay.
2023-02-09 09:39:29 +00:00
sdk
Revert "Remove ISAC media constant and payload type mapping"
2023-02-23 15:00:38 +00:00
stats
stats: Deprecate RTCStatsReport(int64 timestamp_us)
2023-02-22 12:32:02 +00:00
system_wrappers
Check FMA3 support before use it in SincResampler
2023-01-31 17:28:55 +00:00
test
Make Y4mFrameGenerator read FPS from file format.
2023-02-27 14:19:04 +00:00
tools_webrtc
Noop change to trigger bots
2023-02-13 10:30:38 +00:00
video
Introduce capture_time_identifier in webrtc::EncodedImage
2023-02-22 17:08:53 +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
Changed OutputToDebug to create a CFString at compile-time, rather than runtime
2023-02-19 22:41:59 +00:00
BUILD.gn
Delete unused Audio Bwe integration test.
2023-01-26 09:31:44 +00:00
CODE_OF_CONDUCT.md
Reland "Migrate WebRTC documentation to new renderer"
2023-01-31 09:30:04 +00:00
codereview.settings
…
DEPS
Roll chromium_revision 74ff8663a2..ddbda5d5fb (1110600:1110778)
2023-02-28 06:33:12 +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
Verify field trials looked up through field_trial::FindFullName
2022-10-20 10:46:01 +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
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