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chromium-webrtc-autoroll 2b88704632 Roll chromium_revision e91245ed6d..b39fd8fad6 (1103355:1103515) 
Change log: e91245ed6d..b39fd8fad6
Full diff: e91245ed6d..b39fd8fad6

Changed dependencies
* fuchsia_vesion: version:11.20230208.3.1..version:11.20230209.2.1
* src/base: 9cf018c586..4c0de397db
* src/build: 1cc8cf072c..de801e9921
* src/ios: 5c0504eb28..0585530ca8
* src/testing: a3d0fd32f5..c304e79dbb
* src/third_party: 464caf8f6e..33fc1ea758
* src/third_party/depot_tools: d8fb7c9667..023ee12319
* src/third_party/perfetto: 5ca28c07b1..565d711bb6
* src/tools: 838a75986a..4b4825241b
DEPS diff: e91245ed6d..b39fd8fad6/DEPS

No update to Clang.

BUG=None

Change-Id: Ib3681e208abb22410fbf90cae50a9833d9eec520
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/292790
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@{#39290}
2023-02-09 22:28:12 +00:00
api
Consolidate TransformableVideoFrame mocks used inside webrtc
2023-02-09 16:06:29 +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
Change MediaChannel to have a Role parameter
2023-02-09 14:29:08 +00:00
modules
Consolidate TransformableVideoFrame mocks used inside webrtc
2023-02-09 16:06:29 +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
Change MediaChannel to have a Role parameter
2023-02-09 14:29:08 +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
Add IncludeBlocks to clang-format.
2021-02-03 16:29:07 +00:00
.git-blame-ignore-revs
…
.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
…
DEPS
Roll chromium_revision e91245ed6d..b39fd8fad6 (1103355:1103515)
2023-02-09 22:28:12 +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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