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chromium-webrtc-autoroll cd42e347c1 Roll chromium_revision 74acb31550..4d448ca9a6 (1228767:1228869) 
Change log: 74acb31550..4d448ca9a6
Full diff: 74acb31550..4d448ca9a6

Changed dependencies
* src/base: 341add321e..48814fac0d
* src/ios: f6e22b9c75..9931cdfba2
* src/testing: 317c96423d..2586c4d821
* src/third_party: a676e4c7f6..6cae11a6f3
* src/third_party/androidx: 1BfjRMg9mztA2FAr_yQv_7XSWIAwDZ6HMPxdgKUeLfsC..gZjFQKzfZ-1D1L9TfvD_iyEjxjkR9w0kQgz4_bVCKLgC
* src/third_party/perfetto: eb9117dab7..196c2f82ed
* src/third_party/r8: Vjgh6DGZpPhxYqNjulYH9ogpNFhkWn31RDgrWeRQmPcC..Sy-kFTnYsH9Arx9MEiChAikFKSB4kGPTBMM3rgi1Jc0C
* src/tools: b7510e8017..178dab41a0
DEPS diff: 74acb31550..4d448ca9a6/DEPS

No update to Clang.

BUG=None

Change-Id: I678956c411937c1104ea8e83e8443ee044cdf401
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328489
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@{#41235}
2023-11-24 20:35:27 +00:00
api
VideoStreamEncoder: Clean up drop handling and update rects.
2023-11-23 17:19:33 +00:00
audio
Support shortcircuiting encoded transforms
2023-11-17 13:03:27 +00:00
build_overrides
Roll chromium_revision 01dc2965ca..917876224a (1209117:1211391)
2023-10-18 15:15:07 +00:00
call
Update WebRTC code version (2023-11-24T04:13:54).
2023-11-24 05:43:23 +00:00
common_audio
Fix pointer overflow in neon implemenation of audio filters
2023-10-13 06:41:08 +00:00
common_video
Add rtp packetizer for H265
2023-11-08 15:49:37 +00:00
data
…
docs
Add note about two-byte extension to VLA docs
2023-10-11 11:20:19 +00:00
examples
Use rtc::ReceivedPacket in Stun and TurnServer
2023-11-23 10:40:56 +00:00
experiments
Remove SSRCs from libSRTP when removing them from the rtp_demuxer
2023-11-08 10:24:10 +00:00
g3doc
Update TODO example in the style guide.
2023-11-21 23:11:09 +00:00
infra
Add xcode caches to dimensions.
2023-11-17 14:36:35 +00:00
logging
Try removing RTC_PUSH_IGNORING_WUNDEF() around proto includes
2023-11-01 08:21:05 +00:00
media
Make Codec::Matches also consider packetization
2023-11-14 08:14:14 +00:00
modules
Per default enable WebRTC-PaddingMode-RecentLargePacket
2023-11-22 17:43:43 +00:00
net/dcsctp
Move StrJoin to rtc_base/strings
2023-11-15 12:10:28 +00:00
p2p
Use AsyncPacketSocket::RegisterReceivedPacketCallback in StunProber
2023-11-24 09:37:27 +00:00
pc
Refactor MediaSession to unify audio/video codec handling
2023-11-23 14:27:54 +00:00
resources
Ignore .binarypb files.
2023-10-30 14:56:36 +00:00
rtc_base
Add CopyOnWriteBuffer::empty accessor
2023-11-24 18:37:34 +00:00
rtc_tools
Move some users to use webrtc::RefCountInterface
2023-11-02 14:45:57 +00:00
sdk
Trace render window state using a counter.
2023-11-23 14:56:31 +00:00
stats
stats: implement fecSsrc on inbound-rtp
2023-08-04 12:54:48 +00:00
system_wrappers
Use //third_party/cpu_features directly
2023-06-02 07:17:36 +00:00
test
fuzzer: add testonly to fuzzer tests
2023-11-24 11:33:33 +00:00
tools_webrtc
Remove not-needed webrtc:: prefixes in pc/
2023-11-13 13:23:04 +00:00
video
Fix parameter types inconsistency in FrameCadanceAdapter
2023-11-24 13:40:11 +00:00
.clang-format
…
.git-blame-ignore-revs
Add formatting CLs to .git-blame-ignore-revs
2023-05-07 09:27:47 +00:00
.gitignore
…
.gn
[Fuchsia] Remove fuchsia_target_api_level from .gn
2023-09-04 07:26:36 +00:00
.mailmap
Add .mailmap for git.
2022-02-20 14:22:13 +00:00
.style.yapf
Configure YAPF to follow PEP-8 altogether
2023-09-22 10:32:11 +00:00
.vpython
Remove unused script webrtc_dashboard_upload.py
2022-03-21 12:54:42 +00:00
.vpython3
Update vpython3 requests
2023-06-02 07:49:24 +00:00
AUTHORS
fix: Handle out-of-range device index after GetDevicesInfo
2023-09-19 12:13:39 +00:00
BUILD.gn
AsyncPacketSocket::RegisterReceivedPacketCallback
2023-11-15 11:17:53 +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 74acb31550..4d448ca9a6 (1228767:1228869)
2023-11-24 20:35:27 +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
Allow to keep old python style for existing files.
2023-10-17 13:52:56 +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
Add google-truth to WEBRTC_ONLY_DEPS to unblock Chromium roll.
2023-10-17 16:40:08 +00:00
pylintrc
Configure Pylint to follow PEP-8
2023-09-25 15:56:09 +00:00
pylintrc_old_style
Allow to keep old python style for existing files.
2023-10-17 13:52:56 +00:00
README.chromium
[ssci] Added Shipped field to READMEs
2023-07-12 07:31:06 +00:00
README.md
doc: Follow up link rename in I2dbe1ef0c74a0de8c5619b522fab39527e797d9c
2023-05-26 09:20:16 +00:00
WATCHLISTS
Remove xooglers from WATCHLISTS and OWNERS
2022-11-30 15:33:25 +00:00
webrtc_lib_link_test.cc
Migrate webrtc_link_test to include EnableMedia api
2023-11-02 22:45:40 +00:00
webrtc.gni
Make AEC3 json parsing code testonly
2023-10-26 12:03:02 +00:00
whitespace.txt
Trigger bots
2023-05-16 08:24:54 +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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