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chromium-webrtc-autoroll 0b06b2785b Roll chromium_revision 289bd8077e..5e8209d152 (1126466:1126606) 
Change log: 289bd8077e..5e8209d152
Full diff: 289bd8077e..5e8209d152

Changed dependencies
* src/base: 2d7d6f873a..d353f626e4
* src/buildtools/linux64: git_revision:41fef642de70ecdcaaa26be96d56a0398f95abd4..git_revision:28b7b6c507eb808567e3aea446cd259f7691fddc
* src/buildtools/mac: git_revision:41fef642de70ecdcaaa26be96d56a0398f95abd4..git_revision:28b7b6c507eb808567e3aea446cd259f7691fddc
* src/buildtools/win: git_revision:41fef642de70ecdcaaa26be96d56a0398f95abd4..git_revision:28b7b6c507eb808567e3aea446cd259f7691fddc
* src/ios: 2b65b27911..28f8887fce
* src/testing: 3ae4ef5aad..2deab66531
* src/third_party: cfe34887de..2a370beed3
* src/third_party/androidx: fu51N_Sm0ZHkNdj7nyq4KgmGSU5RdSHivUs__whZyqUC..n5nwXemAGKab8EX4F56bXn1fuTFxXH9u4ezd5WJzztkC
* src/third_party/perfetto: 7aa553c67e..df804984ff
* src/tools: c617061e91..acef22adf3
DEPS diff: 289bd8077e..5e8209d152/DEPS

No update to Clang.

BUG=None

Change-Id: I853fff6e1f0d52ba9f625bfc18a15719f06ab0dc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300382
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@{#39769}
2023-04-05 16:32:38 +00:00
api
Fuzzy match the SdpVideoFormat in VideoEncoderFactoryTemplate::CreateVideoEncoder.
2023-04-05 11:48:26 +00:00
audio
Remove AudioConfig::Mode.
2023-04-04 08:44:23 +00:00
build_overrides
Always check out google_benchmark, part 3.
2023-03-14 12:14:51 +00:00
call
Update WebRTC code version (2023-04-03T04:03:14).
2023-04-03 05:23:52 +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
Remove dependency to android_support_test_runner in webrtc
2023-04-05 08:40:19 +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
Unnest VideoEncoderFactoryTemplate in webrtc_video_engine_unittest.cc
2023-04-05 15:13:36 +00:00
modules
Remove IsSinglecastOrAllNonFirstLayersInactive() helper.
2023-04-04 13:59:07 +00:00
net/dcsctp
dcsctp: Support zero checksum packets
2023-04-02 21:38:00 +00:00
p2p
Print discovered network interfaces for debugging ICE_NEVER_CONNECTED endcause.
2023-03-29 11:58:22 +00:00
pc
Close PC before test ends to reduce risk of flake.
2023-04-05 15:28:37 +00:00
resources
…
rtc_base
Print discovered network interfaces for debugging ICE_NEVER_CONNECTED endcause.
2023-03-29 11:58:22 +00:00
rtc_tools
In rtc event log visualizer use modern api of the ReceiveSideCongestionController
2023-04-04 10:30:44 +00:00
sdk
Remove dependency to android_support_test_runner in webrtc
2023-04-05 08:40:19 +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
Remove AudioConfig::Mode.
2023-04-04 08:44:23 +00:00
tools_webrtc
Noop change to trigger bots
2023-02-13 10:30:38 +00:00
video
Remove IsSinglecastOrAllNonFirstLayersInactive() helper.
2023-04-04 13:59:07 +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
…
.style.yapf
…
.vpython
…
.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
…
DEPS
Roll chromium_revision 289bd8077e..5e8209d152 (1126466:1126606)
2023-04-05 16:32:38 +00:00
DIR_METADATA
…
ENG_REVIEW_OWNERS
…
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
…
presubmit_test.py
…
PRESUBMIT.py
…
pylintrc
…
README.chromium
…
README.md
…
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 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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