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chromium-webrtc-autoroll 8da0f3aecc Roll chromium_revision f6c2626500..2b544cbc99 (1092306:1092425) 
Change log: f6c2626500..2b544cbc99
Full diff: f6c2626500..2b544cbc99

Changed dependencies
* src/base: 0130614e06..8bc32913fa
* src/build: 4a795f5e05..e05402580b
* src/ios: 2f6bc869f3..de42da64d9
* src/testing: e55c98c4ee..0db2e3b770
* src/third_party: 8136a88260..c05ba65c33
* src/third_party/perfetto: c1a1facf09..db57f10ab3
* src/third_party/r8: haRbS4QoarHRjXQOZrl3EhIQinN95VFOrJhZT7cCQvsC..28aGNwW2oSdul7Vvstd4P8mSTJuSrv7cWe_s0RPmPIwC
* src/tools: 318451984e..dafdb3dabd
DEPS diff: f6c2626500..2b544cbc99/DEPS

No update to Clang.

BUG=None

Change-Id: I4dd5be672faa2d849277e3cf3e567c5e55651d55
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290909
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@{#39103}
2023-01-13 16:33:00 +00:00
api
Add GetRTPVideoHeaderCodecSpecifics() to metadata.
2023-01-13 11:33:40 +00:00
audio
[Unwrap] Use RtpTimestampUnwrapper in audio/channel_receive
2023-01-11 11:59:09 +00:00
build_overrides
Use default values provided by PartitionAlloc instead of hard-coded ones
2022-12-07 09:11:35 +00:00
call
Update WebRTC code version (2023-01-12T04:02:41).
2023-01-12 05:42:44 +00:00
common_audio
Make header files self contained.
2022-10-08 08:38:36 +00:00
common_video
Make header files self contained.
2022-10-08 08:38:36 +00:00
data
…
docs
Update webrtc build command instruction to autoninja.
2022-11-25 09:19:37 +00:00
examples
Split rtc_base into multiple targets
2023-01-09 12:21:25 +00:00
experiments
Add tool for generating field trial registry header
2022-10-18 07:25:43 +00:00
g3doc
Update WebRTC style guide about GN.
2022-12-16 10:19:13 +00:00
infra
Remove 'iOS Debug (simulator)' from LKGR.
2023-01-13 11:02:01 +00:00
logging
[Unwrap] Migrate RtcEventLog parser to use RtpSequenceNumberUnwrapper
2023-01-12 10:23:54 +00:00
media
Add SVC fallback.
2023-01-11 16:49:49 +00:00
modules
Add RTPVideoHeader::SetFromMetadata() and FromMetadata().
2023-01-13 13:38:42 +00:00
net/dcsctp
[Unwrap] Migrate dcsctp sequence numbers to SeqNumUnwrapper
2023-01-12 12:00:30 +00:00
p2p
Split rtc_base into multiple targets
2023-01-09 12:21:25 +00:00
pc
Use parsed packet from RtpTransport::DemuxPacket in engine and call
2023-01-10 15:06:50 +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
[Unwrap] Delete rtc::TimestampWrapAroundHandler
2023-01-12 14:21:03 +00:00
rtc_tools
Split rtc_base into multiple targets
2023-01-09 12:21:25 +00:00
sdk
Add dimensions to video settings in objc sdk camera backend.
2023-01-11 13:59:37 +00:00
stats
stats: deprecate timestamp_us constructor and method
2023-01-11 11:40:05 +00:00
system_wrappers
[Unwrap] Migrate RtpToNtpEstimator to use RtpTimestampUnwrapper
2023-01-11 17:14:41 +00:00
test
Use PacketReceiver::DeliverRtpPaket in scenario tests
2023-01-12 16:57:28 +00:00
tools_webrtc
Add "iOS Debug (simulator)" CI bot.
2023-01-13 08:16:10 +00:00
video
Use RtcEventLog::EncodingType::NewFormat in VideoQualityTest.
2023-01-13 10:47:38 +00:00
.clang-format
…
.git-blame-ignore-revs
…
.gitignore
…
.gn
Set Fuchsia Api level + update SDK version
2022-09-14 08:49:56 +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
Remove dimension check in SimulcastUtility::ValidSimulcastParameters
2023-01-11 13:41:55 +00:00
BUILD.gn
Only build fuchsia_perf_tests on fuchsia os.
2023-01-11 11:35:51 +00:00
CODE_OF_CONDUCT.md
Reference "main" branches instead of "master" branches.
2021-07-15 11:07:44 +00:00
codereview.settings
…
DEPS
Roll chromium_revision f6c2626500..2b544cbc99 (1092306:1092425)
2023-01-13 16:33:00 +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
g3doc.lua
Improve webrtc documentation infra. Preview at:
2021-03-30 10:29:30 +00:00
LICENSE
…
license_template.txt
…
native-api.md
Reference "main" branches instead of "master" branches.
2021-07-15 11:07:44 +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
Verify field trials looked up through field_trial::FindFullName
2022-10-20 10:46:01 +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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