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chromium-webrtc-autoroll f2a92f9f64 Roll chromium_revision 97b144adc4..254e97f424 (1120470:1120661) 
Change log: 97b144adc4..254e97f424
Full diff: 97b144adc4..254e97f424

Changed dependencies
* fuchsia_vesion: version:12.20230321.3.1..version:12.20230322.1.1
* src/base: 22039bb79d..2a1b2eec25
* src/build: 70c1390779..c499da7bb0
* src/ios: ef842ef5d0..8d3d6b44be
* src/testing: a69814e2be..ac0cf0af86
* src/third_party: b78e410bad..3394794299
* src/third_party/androidx: 1b2_7gRLY5WVxj5lXGSEkTkPYO_QZTAIkqSyFCKt_yYC..ryrhsKJY8msYy8KMkwB25bMAu68qC6b3CKrKger0aywC
* src/third_party/perfetto: 1bb22965b0..a0ad54d66f
* src/tools: 456ed3095f..9f58dbf1e9
DEPS diff: 97b144adc4..254e97f424/DEPS

No update to Clang.

BUG=None

Change-Id: I1682e6f33ca62c0dc2d7fafc273b7a3dc24f4942
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298801
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@{#39645}
2023-03-22 18:35:16 +00:00
api
Add mock for NetworkControllerInterface
2023-03-22 15:49:32 +00:00
audio
stats: use uint64_t for RTCSentRtpStreamStats.packetsSent
2023-03-16 06:46:19 +00:00
build_overrides
Always check out google_benchmark, part 3.
2023-03-14 12:14:51 +00:00
call
Update WebRTC code version (2023-03-22T04:02:52).
2023-03-22 05:36:24 +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
Add webrtc_nonparallel_tests to Fuchsia bots.
2023-03-15 08:44:59 +00:00
logging
RtcEventLog: Separate LogToMemory from TaskQueue to current thread
2023-03-21 23:07:40 +00:00
media
Add thread guard for DcSctpTransport::ready_to_send_data_
2023-03-22 12:23:27 +00:00
modules
Adds WebRTC.DesktopCapture.Win.DirectXCursorEmbedded UMA
2023-03-22 17:19:06 +00:00
net/dcsctp
dcsctp: Make use of log_prefix consistent
2023-03-15 22:15:05 +00:00
p2p
Replace use of test-only connections() with P2PTransportChannel member.
2023-02-27 16:49:05 +00:00
pc
Split up GetSctpSslRole to include an _n variant.
2023-03-22 16:47:28 +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
Per default endable reading incoming packet timestamp from socket
2023-03-21 14:41:37 +00:00
rtc_tools
Handling NetEqSetMinimumDelay events in neteq_rtpplay.
2023-02-09 09:39:29 +00:00
sdk
New PeerConnectionFactory::CreateVideoTrack with refcounted source
2023-03-22 09:10:27 +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
Use an audio file when AudioConfig::input_file_name is specified.
2023-03-22 17:08:26 +00:00
tools_webrtc
Noop change to trigger bots
2023-02-13 10:30:38 +00:00
video
Always post RemovePacketsForSsrc
2023-03-22 14:24:50 +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 97b144adc4..254e97f424 (1120470:1120661)
2023-03-22 18:35:16 +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
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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