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chromium-webrtc-autoroll d59c0c35c6 Roll chromium_revision 790425576f..73f1a09e0a (1103139:1103251) 
Change log: 790425576f..73f1a09e0a
Full diff: 790425576f..73f1a09e0a

Changed dependencies
* src/base: 56be631faa..d7094e484d
* src/build: ec73ae49c0..6abd6f5df3
* src/ios: 4b4e61e03b..0069f6e99c
* src/testing: 0e9e5db2f3..bbe6bfc5ca
* src/third_party: df834f553a..72f0570d19
* src/third_party/perfetto: 0374f0872c..5ca28c07b1
* src/tools: 3a60443d9f..7f479dc3f7
DEPS diff: 790425576f..73f1a09e0a/DEPS

No update to Clang.

BUG=None

Change-Id: I7d9c1767754abdfd0b2f95797dd8a4c2ef7238fd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/292786
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@{#39282}
2023-02-09 14:21:35 +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
…
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
[infra] Remove goma properties
2023-02-09 09:10:54 +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
Handling NetEqSetMinimumDelay events in neteq_rtpplay.
2023-02-09 09:39:29 +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
Test what happens when asking for simulcast VP9 (not yet supported).
2023-02-09 09:59:34 +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
Handling NetEqSetMinimumDelay events in neteq_rtpplay.
2023-02-09 09:39:29 +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
Handling NetEqSetMinimumDelay events in neteq_rtpplay.
2023-02-09 09:39:29 +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
…
.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
…
.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
…
DEPS
Roll chromium_revision 790425576f..73f1a09e0a (1103139:1103251)
2023-02-09 14:21:35 +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
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
…
WATCHLISTS
Remove xooglers from WATCHLISTS and OWNERS
2022-11-30 15:33:25 +00:00
webrtc_lib_link_test.cc
…
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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