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chromium-webrtc-autoroll 09eccc7261 Roll chromium_revision b0c0d15a35..6668472a87 (1115644:1116317) 
Change log: b0c0d15a35..6668472a87
Full diff: b0c0d15a35..6668472a87

Changed dependencies
* fuchsia_vesion: version:12.20230308.3.1..version:12.20230312.3.1
* src/base: 236aa48f90..e81408704e
* src/build: 9bfa5a9d2e..168bfaa8bf
* src/buildtools: 1e2d30f5e4..e1fc35a07d
* src/ios: d1ea4374ef..fcf70e62e9
* src/testing: 6ec188dc0e..92ae58d1f6
* src/third_party: 1c0b8f9074..ff9d1d5645
* src/third_party/androidx: X6mj51w5gvqwPAm19oEQHgV33JA7r-iIrlA0xm6O7PAC..fAwky11GHWi_G3y48BIl4JpIfPC3X6W0ikMvZjAeIjsC
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/ca1690e221..082e953a13
* src/third_party/depot_tools: a9a7eecf37..d9e2d47985
* src/third_party/ffmpeg: 5d5c6592d2..a51c75b09b
* src/third_party/grpc/src: e022a3dfa9..822dab21d9
* src/third_party/libaom/source/libaom: https://aomedia.googlesource.com/aom.git/+log/080db61c3b..35410dede1
* src/third_party/libvpx/source/libvpx: 79b1347a51..f7ca33c46c
* src/third_party/r8: QcJGU2P6jjudE2LELurmeujLPwQhvk7OD5AWGlLIzrYC..M9qqyShmnvqDcIIsdbwvO7LJ9WFLu552c6c29zYKCdIC
* src/tools: aba8dbf531..ed1a1be638
* src/tools/luci-go: git_revision:e260f2e6d3531f534378dd1017e140374ba8df48..git_revision:320bf3ed60cd4d24549d0ea9ee3a94394f2665ce
* src/tools/luci-go: git_revision:e260f2e6d3531f534378dd1017e140374ba8df48..git_revision:320bf3ed60cd4d24549d0ea9ee3a94394f2665ce
DEPS diff: b0c0d15a35..6668472a87/DEPS

No update to Clang.

BUG=None

Change-Id: I34cd58c161580ed299da70daac33a30b02b2ee01
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/297381
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Owners-Override: Christoffer Jansson <jansson@webrtc.org>
Commit-Queue: Christoffer Jansson <jansson@webrtc.org>
Reviewed-by: Christoffer Jansson <jansson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39546}
2023-03-13 13:22:40 +00:00
api
Reland "stats: remove RTCRtpInboundRTPStream and RTCRtpoutboundRTPStream aliases"
2023-03-10 15:22:01 +00:00
audio
De-flake NonSenderRttStats and make it faster to run on average.
2023-03-10 13:25:34 +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 (2023-03-12T04:12:04).
2023-03-12 05:21:51 +00:00
common_audio
Check FMA3 support before use it in SincResampler
2023-01-31 17:28:55 +00:00
common_video
Fix AVC PPS parser unit test.
2023-03-01 12:00:49 +00:00
data
…
docs
doc: rename index.md to README.md
2023-03-13 13:16:22 +00:00
examples
Remove mentions of already deleted field trials
2023-03-01 15:53:37 +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 mac_chromium_compile on 12 cores machines.
2023-03-10 10:57:54 +00:00
logging
Remove deprecated logged_events.h
2023-03-09 19:42:57 +00:00
media
EncoderSimulcastProxy: Respect "supports_simulcast" info.
2023-03-13 10:55:16 +00:00
modules
Adds 0Hz support to WGC behind a flag
2023-03-13 11:59:53 +00:00
net/dcsctp
Implement support for Chrome task origin tracing. #3.5/4
2023-03-01 11:11:37 +00:00
p2p
Replace use of test-only connections() with P2PTransportChannel member.
2023-02-27 16:49:05 +00:00
pc
Remove the SctpDataChannel::config_ member variable.
2023-03-12 17:28:14 +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 missing include of pthread.h
2023-03-13 13:06:56 +00:00
rtc_tools
Handling NetEqSetMinimumDelay events in neteq_rtpplay.
2023-02-09 09:39:29 +00:00
sdk
Remove resolution alignment requirement (part 2)
2023-03-07 17:18:55 +00:00
stats
stats: rename RTCInboundRTPStreamStats and RTCOutboundRTPStreamStats
2023-03-07 14:27:47 +00:00
system_wrappers
Add option to log a warning for unregistered field trials
2023-02-28 15:43:18 +00:00
test
stats: rename RTCInboundRTPStreamStats and RTCOutboundRTPStreamStats
2023-03-07 14:27:47 +00:00
tools_webrtc
Noop change to trigger bots
2023-02-13 10:30:38 +00:00
video
Disable flaky PictureIdTests.
2023-03-10 11:52:23 +00:00
.clang-format
Add IncludeBlocks to clang-format.
2021-02-03 16:29:07 +00:00
.git-blame-ignore-revs
…
.gitignore
…
.gn
Increase android32_ndk_api_level to 21.
2023-03-13 12:37:57 +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
Changed OutputToDebug to create a CFString at compile-time, rather than runtime
2023-02-19 22:41:59 +00:00
BUILD.gn
Add a new test binary to check if split-channel works
2023-03-07 11:14:31 +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 b0c0d15a35..6668472a87 (1115644:1116317)
2023-03-13 13:22:40 +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
…
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
Add option to log a warning for unregistered field trials
2023-02-28 15:43:18 +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
Languages
C++ 90.3%
Java 2.9%
C 2.2%
Objective-C++ 2%
Python 1.3%
Other 1%