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chromium-webrtc-autoroll f767fc0fb1 Roll chromium_revision 64efe0fec7..552e886b83 (1123151:1123510) 
Change log: 64efe0fec7..552e886b83
Full diff: 64efe0fec7..552e886b83

Changed dependencies
* src/base: 13af57d61b..1cde5aaaa0
* src/build: 625573ad8b..850cf64582
* src/buildtools: c97a3b8902..d2c292117b
* src/buildtools/third_party/libc++/trunk: 26ace673c4..e0dca93272
* src/ios: 7061eccc0e..9c9596c969
* src/testing: 903a10b7df..2a62e47eb9
* src/third_party: 161fc2d0b6..9045376e26
* src/third_party/androidx: zvyRaq_hY1gg3vn5ByQj77DzD8tJ1oEHIci-xNgOCVoC..q-4EGT6m1jFTmhGIBcb9baxf_vf5jExDN7FYNlNp5DwC
* src/third_party/depot_tools: f724f296b7..4a7343007c
* src/third_party/perfetto: c44cab035c..ad418a3f70
* src/third_party/r8: BTs9W6pUBDK8YTePjjF3Q0-ZQo1snO7GBCT6frWMIj8C..0AAkHl3ztCss8QvTqGZbCU0Wy2W-0CIoWwCYZnYg7j0C
* src/tools: d7ef7a908e..1254fa5044
Removed dependency
* src/third_party/android_deps/libs/com_google_android_gms_play_services_fido
DEPS diff: 64efe0fec7..552e886b83/DEPS

No update to Clang.

BUG=None

Change-Id: I06868b3ed9848a3c0ef688a51e5948666fc09e2a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299403
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@{#39714}
2023-03-29 11:06:41 +00:00
api
Remove deprecated TransformableVideoFrameInterface::GetMetadata
2023-03-29 08:33:46 +00:00
audio
Use SequenceChecker(SequenceChecker::kDetached) in a few places.
2023-03-24 07:44:18 +00:00
build_overrides
Always check out google_benchmark, part 3.
2023-03-14 12:14:51 +00:00
call
Allow injecting packets of type Any to Call::DeliverRtpPacket
2023-03-29 06:36:17 +00:00
common_audio
Check FMA3 support before use it in SincResampler
2023-01-31 17:28:55 +00:00
common_video
Changed sps parser and sps parser unit test case for h264, and it is working
2023-03-14 12:15:54 +00:00
data
…
docs
doc: rename index.md to README.md
2023-03-13 13:16:22 +00:00
examples
New PeerConnectionFactory::CreateVideoTrack with refcounted source
2023-03-22 09:10:27 +00:00
experiments
Add tool for generating field trial registry header
2022-10-18 07:25:43 +00:00
g3doc
Add docs about adding a new test binary.
2023-03-07 11:12:33 +00:00
infra
Run Fuchsia Perf Testers on stronger machines.
2023-03-27 09:06:45 +00:00
logging
Ensure correct destruction order in RtcEventLogImplTest
2023-03-23 11:50:41 +00:00
media
Add RTP header extension API regression tests
2023-03-28 11:06:24 +00:00
modules
Remove deprecated TransformableVideoFrameInterface::GetMetadata
2023-03-29 08:33:46 +00:00
net/dcsctp
dcsctp: add zero checksum acceptable chunk parameter
2023-03-28 12:15:39 +00:00
p2p
Replace use of test-only connections() with P2PTransportChannel member.
2023-02-27 16:49:05 +00:00
pc
Remove SctpDataChannel::writable_
2023-03-28 12:13:40 +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
Use SequenceChecker(SequenceChecker::kDetached) in a few places.
2023-03-24 07:44:18 +00:00
rtc_tools
Revert "Unifying the handling of the events in NetEqInput."
2023-03-29 08:01:46 +00:00
sdk
Add support for JavaToNativeFloatArray
2023-03-29 11:03:30 +00:00
stats
Remove deprecated RTCStatsReport(int64) and timestamp_us
2023-03-22 08:00:53 +00:00
system_wrappers
Add option to log a warning for unregistered field trials
2023-02-28 15:43:18 +00:00
test
Revert "Unifying the handling of the events in NetEqInput."
2023-03-29 08:01:46 +00:00
tools_webrtc
Noop change to trigger bots
2023-02-13 10:30:38 +00:00
video
Fix VP9 {active,inactive,inactive} bitrate issue causing spatial drop.
2023-03-28 08:28:41 +00:00
.clang-format
…
.git-blame-ignore-revs
…
.gitignore
…
.gn
Increase android32_ndk_api_level to 21.
2023-03-13 12:37:57 +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
Always check out google_benchmark, part 5.
2023-03-15 07:52:04 +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 64efe0fec7..552e886b83 (1123151:1123510)
2023-03-29 11:06:41 +00:00
DIR_METADATA
…
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
…
README.md
…
WATCHLISTS
Remove xooglers from WATCHLISTS and OWNERS
2022-11-30 15:33:25 +00:00
webrtc_lib_link_test.cc
…
webrtc.gni
Always check out google_benchmark, part 3.
2023-03-14 12:14:51 +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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C 2.2%
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Python 1.3%
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