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chromium-webrtc-autoroll 9629ca2c98 Roll chromium_revision 13440ad881..7a0c6115a5 (1080293:1080586) 
Change log: 13440ad881..7a0c6115a5
Full diff: 13440ad881..7a0c6115a5

Changed dependencies
* src/base: b6b2f94893..898029302c
* src/build: 73b80fa285..e3833ca15b
* src/ios: 7e6d74ba10..a353359f5f
* src/testing: 4d0bc6a224..21ed928654
* src/third_party: ea81a1d159..00aeb04924
* src/third_party/androidx: OsLJqVdZEwRSmYkbx3YR-AlSoa8_tTlhUa4E0p3O0ykC..9Ik3i0VlKSVthDuv4NoO-uty6zlMOUyV3u2K1kZz6SwC
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/11955a66e9..13419d2560
* src/third_party/depot_tools: a2658217d2..5175d18f8a
* src/third_party/freetype/src: 34c1bb66e4..b405fc5c1d
* src/third_party/perfetto: d581ead87c..df18f44408
* src/third_party/r8: smgiQDhBHEfz8KVdqBXas7z0ALaukjW3Yh-JVGPsFksC..5Ckp25uPxCBEH3n_3JC6k2Jf1STCD2VMtCxHeUDjz6oC
* src/tools: cf3756fe71..681b3f83d5
DEPS diff: 13440ad881..7a0c6115a5/DEPS

No update to Clang.

BUG=None

Change-Id: I70b96fef3c6cf4d7e54c73b5d7fb161237ad8ef6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/286920
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@{#38842}
2022-12-07 23:08:35 +00:00
api
generateKeyframe: pass frame_types in bypass mode
2022-12-06 10:54:59 +00:00
audio
Remove default enabled field trial WebRTC-SendSideBwe-WithOverhead
2022-11-30 20:19:36 +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 (2022-12-07T04:11:49).
2022-12-07 05:44:33 +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
pc: Add asynchronous RtpSender::SetParameters() call
2022-11-15 15:31:40 +00:00
experiments
Add tool for generating field trial registry header
2022-10-18 07:25:43 +00:00
g3doc
Explicitly forbid dynamic_cast
2022-11-04 12:37:57 +00:00
infra
[infra] Add Reclient shadow builders as experimental
2022-12-07 08:35:37 +00:00
logging
Add missing dependencies.
2022-10-10 15:51:33 +00:00
media
generateKeyframe: pass frame_types in bypass mode
2022-12-06 10:54:59 +00:00
modules
APM AGC2: consolidate GainController2
2022-12-07 21:47:45 +00:00
net/dcsctp
Add missing dependencies.
2022-10-10 10:18:37 +00:00
p2p
Resolve TURN hostname specific to network family behind field trial.
2022-12-05 13:34:57 +00:00
pc
sdp: ignore duplicate stream ids in msid parsing
2022-12-07 18:03:59 +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
Fix unaligned memory access detected by UBSAN
2022-12-05 09:56:00 +00:00
rtc_tools
Delete RGB to I420 and YUV to IVF converters
2022-12-02 08:21:21 +00:00
sdk
AndroidNetworkMonitor - loosen assumptions even more
2022-12-05 10:01:01 +00:00
stats
Add power efficient stats to RTC stats
2022-11-08 08:35:47 +00:00
system_wrappers
Rename test helper for registering field trial keys
2022-12-02 13:21:28 +00:00
test
Refactor YUV frame reader
2022-12-06 16:23:48 +00:00
tools_webrtc
Add instrumented_libraries_release arg to MSAN bots
2022-12-07 12:10:49 +00:00
video
Reland "Remame VideoSendStream::UpdateActiveSimulcastLayers to StartPerRtpStream"
2022-12-02 12:03:25 +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
Fix bugs in discarding stun address
2022-11-30 13:51:41 +00:00
BUILD.gn
Add fuchsia_perf_test to the default build group.
2022-12-05 10:09:00 +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 13440ad881..7a0c6115a5 (1080293:1080586)
2022-12-07 23:08:35 +00:00
DIR_METADATA
…
ENG_REVIEW_OWNERS
Remove phoglund from ENG_REVIEW_OWNERS
2021-10-08 08:29:42 +00:00
g3doc.lua
…
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
…
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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