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chromium-webrtc-autoroll 3a71e3a8f0 Roll chromium_revision a6f31c2d02..95e63e07f8 (1117547:1117698) 
Change log: a6f31c2d02..95e63e07f8
Full diff: a6f31c2d02..95e63e07f8

Changed dependencies
* src/base: 9c7ab304e4..3b194faf5f
* src/build: f0103dc237..5dab994f42
* src/ios: e72326d87c..cc8ff548d5
* src/testing: f383a6ec90..1bd2a641b5
* src/third_party: 5408c0e9dc..4d656b540d
* src/third_party/perfetto: 376827e2c4..722a85f82b
* src/tools: aca3baa8e2..bfc14efe0e
Added dependency
* src/third_party/android_deps/libs/org_mockito_mockito_subclass
DEPS diff: a6f31c2d02..95e63e07f8/DEPS

No update to Clang.

BUG=None

Change-Id: I4cd696ad2bc7a8c9a266897d4b2bfae673bb8e6f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/297841
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@{#39569}
2023-03-15 21:12:48 +00:00
api
[DVQA] Add support for DVQA to pause/resume receiving of stream by peer
2023-03-15 18:16:49 +00:00
audio
De-flake NonSenderRttStats and make it faster to run on average.
2023-03-10 13:25:34 +00:00
build_overrides
Always check out google_benchmark, part 3.
2023-03-14 12:14:51 +00:00
call
Update WebRTC code version (2023-03-15T04:11:36).
2023-03-15 05:56:15 +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
Remove old data files.
2018-10-05 14:40:21 +00:00
docs
doc: rename index.md to README.md
2023-03-13 13:16:22 +00:00
examples
Remove mentions of already deleted field trials
2023-03-01 15:53:37 +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
Add webrtc_nonparallel_tests to Fuchsia bots.
2023-03-15 08:44:59 +00:00
logging
Remove deprecated logged_events.h
2023-03-09 19:42:57 +00:00
media
Remove cricket::ReceiveDataParams
2023-03-15 13:27:55 +00:00
modules
x_server_pixel_buffer: Use CopyPixelsFrom instead of memcpy
2023-03-15 17:15:54 +00:00
net/dcsctp
Implement support for Chrome task origin tracing. #3.5/4
2023-03-01 11:11:37 +00:00
p2p
Replace use of test-only connections() with P2PTransportChannel member.
2023-02-27 16:49:05 +00:00
pc
Remove SctpDataChannelControllerInterface::DisconnectDataChannel
2023-03-15 15:32:31 +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 WaitPoll for Fuchsia
2023-03-15 14:35:08 +00:00
rtc_tools
Handling NetEqSetMinimumDelay events in neteq_rtpplay.
2023-02-09 09:39:29 +00:00
sdk
Remove resolution alignment requirement (part 2)
2023-03-07 17:18:55 +00:00
stats
stats: rename RTCInboundRTPStreamStats and RTCOutboundRTPStreamStats
2023-03-07 14:27:47 +00:00
system_wrappers
Add option to log a warning for unregistered field trials
2023-02-28 15:43:18 +00:00
test
[DVQA] Add support for DVQA to pause/resume receiving of stream by peer
2023-03-15 18:16:49 +00:00
tools_webrtc
Noop change to trigger bots
2023-02-13 10:30:38 +00:00
video
Add a flag to control legacy vs spec-compliant scalability mode.
2023-03-14 12:05:24 +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
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
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 a6f31c2d02..95e63e07f8 (1117547:1117698)
2023-03-15 21:12:48 +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
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
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