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chromium-webrtc-autoroll 713fdc0c67 Roll chromium_revision 513ae2fffc..b49d6c9ec1 (1102858:1103023) 
Change log: 513ae2fffc..b49d6c9ec1
Full diff: 513ae2fffc..b49d6c9ec1

Changed dependencies
* src/base: 5156eb889b..987b55e37c
* src/ios: a5010c85e1..4b4e61e03b
* src/testing: 9c8dc598e1..5ab3bb2eef
* src/third_party: 0c83a0d592..b1395e4f52
* src/third_party/depot_tools: 2ec2918216..cd2395991f
* src/third_party/freetype/src: 27b2cd4101..4c3916e901
* src/tools: 76adb287ba..ce86aaefb9
* src/tools/luci-go: git_revision:a8b84fba102daff5bf5e65975dcc0887da7ab62a..git_revision:f6b5518e872364f59bb17dd5a967270b38331b84
* src/tools/luci-go: git_revision:a8b84fba102daff5bf5e65975dcc0887da7ab62a..git_revision:f6b5518e872364f59bb17dd5a967270b38331b84
DEPS diff: 513ae2fffc..b49d6c9ec1/DEPS

No update to Clang.

BUG=None

Change-Id: I78846e6bdcfc1b3fc359668f57914805c783a0a8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/292763
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@{#39276}
2023-02-09 01:05:00 +00:00
api
Add capture time as identifier in webrtc::VideoFrame
2023-02-08 11:05:47 +00:00
audio
Create unit test for the population of capture_start_ntp_time
2023-02-06 14:00:39 +00:00
build_overrides
Use default values provided by PartitionAlloc instead of hard-coded ones
2022-12-07 09:11:35 +00:00
call
Allow RTX ssrc to be updated on receive streams
2023-02-01 12:54:46 +00:00
common_audio
Check FMA3 support before use it in SincResampler
2023-01-31 17:28:55 +00:00
common_video
Add 444 10 bits support for H264 and VP9
2023-01-17 12:32:26 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
docs
Update WebRTC doc related to webrtc.org accounts.
2023-01-16 09:34:28 +00:00
examples
Remove all usage of //rtc_base target
2023-01-16 14:36:06 +00:00
experiments
Add tool for generating field trial registry header
2022-10-18 07:25:43 +00:00
g3doc
Fix doc path
2023-01-31 10:14:47 +00:00
infra
Upgrade Linux MSan to Focal
2023-02-05 14:14:31 +00:00
logging
RtcEventLogImpl: Add test cases
2023-02-03 09:55:33 +00:00
media
Remove leftover function signatures.
2023-02-02 16:23:07 +00:00
modules
Fix Destruction inside WGC Callback
2023-02-08 23:19:22 +00:00
net/dcsctp
Remove all usage of //rtc_base target
2023-01-16 14:36:06 +00:00
p2p
Add an ICE switch reason for a switch requested by an application.
2023-02-06 16:19:49 +00:00
pc
Verify codec and scalability mode in simulcast test.
2023-02-08 14:20:33 +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
Add method to get FD for physical socket
2023-01-31 10:25:45 +00:00
rtc_tools
Ensure VideoRtpReplayer use new PacketReceiver::DeliverRtp packet.
2023-01-18 12:47:41 +00:00
sdk
Enable rotation tests marked as expected failures
2023-02-02 10:48:32 +00:00
stats
Reland "Migrate WebRTC documentation to new renderer"
2023-01-31 09:30:04 +00:00
system_wrappers
Check FMA3 support before use it in SincResampler
2023-01-31 17:28:55 +00:00
test
Move video video receiver transformable frame to modules/rtc_rtcp/source
2023-02-03 12:59:19 +00:00
tools_webrtc
Roll chromium_revision e182675fbb..2e99d26f79 (1098562:1102579)
2023-02-08 10:47:04 +00:00
video
Move video video receiver transformable frame to modules/rtc_rtcp/source
2023-02-03 12:59:19 +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
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
Delete unused Audio Bwe integration test.
2023-01-26 09:31:44 +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 513ae2fffc..b49d6c9ec1 (1102858:1103023)
2023-02-09 01:05:00 +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
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
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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