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chromium-webrtc-autoroll c2c057480e Roll chromium_revision 5a75056b63..936fdee09e (1132724:1132874) 
Change log: 5a75056b63..936fdee09e
Full diff: 5a75056b63..936fdee09e

Changed dependencies
* fuchsia_vesion: version:12.20230419.1.1..version:12.20230419.3.1
* src/base: 2b96884a37..181ecd1f50
* src/buildtools/third_party/libunwind/trunk: 46b7561cc2..665c2e5429
* src/ios: 004e1cafbd..61e07bbba2
* src/testing: d58ec188b8..9ab3d6e6a9
* src/third_party: 74e8addb2b..3262d59a8f
* src/third_party/androidx: 1qQqPQtccCINLyFYfGH-DA8MpMP6I8TmS30ClgQyJB0C..0OknrfP6ltiomSIE9pUWuIHTCnf-02UsbPRcY8KuU5AC
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/c3a1432bec..843ad30cc3
* src/third_party/depot_tools: 20ca4c37b7..0d6cb8fa70
* src/third_party/freetype/src: ea9fca0add..8154d8e2be
* src/third_party/perfetto: d32d008dfa..4da2dec19e
* src/tools: 800fe064b9..2e892d60b7
DEPS diff: 5a75056b63..936fdee09e/DEPS

No update to Clang.

BUG=None

Change-Id: Ia097e1d0642c6321cc82c413a02097680726741a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302080
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@{#39902}
2023-04-20 02:33:10 +00:00
api
[DVQA] Add a GetSenderPeerName method.
2023-04-17 13:05:33 +00:00
audio
Use AudioDeviceModule instead of TestAudioDeviceModule.
2023-04-13 12:31:34 +00:00
build_overrides
Always check out google_benchmark, part 3.
2023-03-14 12:14:51 +00:00
call
Migrate PostTask+Wait to BlockingCall to avoid deadlock in DegradedCall.
2023-04-19 09:40:33 +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
Comment out libwebrtc_unity
2023-04-18 11:37:39 +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
Re-enable Win64 bots on CQ.
2023-04-13 18:21:16 +00:00
logging
Delete clamping cumulative loss in ReportBlocks on receiving side
2023-04-18 13:43:27 +00:00
media
Cleanup ReportBlockData class: use Timestamp and TimeDelta
2023-04-13 08:51:12 +00:00
modules
Format /modules
2023-04-20 02:02:45 +00:00
net/dcsctp
Revert "dcsctp: Support zero checksum packets"
2023-04-17 19:29:55 +00:00
p2p
Default enable WebRTC-IPv6NetworkResolutionFixes
2023-04-17 20:28:53 +00:00
pc
Reduce log spam
2023-04-18 12:31:49 +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
Set AV1 default bitrate limits.
2023-04-19 18:45:33 +00:00
rtc_tools
Use SendAsync in data channel benchmark.
2023-04-18 13:33:51 +00:00
sdk
Account for stride when calculating buffer size
2023-04-19 10:10:32 +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
[DVQA] Add a GetSenderPeerName method.
2023-04-17 13:05:33 +00:00
tools_webrtc
Noop change to trigger bots
2023-02-13 10:30:38 +00:00
video
Cleanup ReportBlockData class: use Timestamp and TimeDelta
2023-04-13 08:51:12 +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
Replace BuiltinVideo{Encoder,Decoder}Factory with Video{Encoder,Decoder}FactoryTemplate.
2023-03-31 19:31:50 +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 5a75056b63..936fdee09e (1132724:1132874)
2023-04-20 02:33:10 +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
Replace BuiltinVideo{Encoder,Decoder}Factory with Video{Encoder,Decoder}FactoryTemplate.
2023-03-31 19:31:50 +00:00
webrtc.gni
Replace BuiltinVideo{Encoder,Decoder}Factory with Video{Encoder,Decoder}FactoryTemplate.
2023-03-31 19:31:50 +00:00
whitespace.txt
Trigger builds
2023-04-12 07:09:41 +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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Java 2.9%
C 2.2%
Objective-C++ 2%
Python 1.3%
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