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chromium-webrtc-autoroll 662863ec93 Roll chromium_revision b7561cbdd7..3f014a9b1e (1240168:1240385) 
Change log: b7561cbdd7..3f014a9b1e
Full diff: b7561cbdd7..3f014a9b1e

Changed dependencies
* src/base: 4171020c05..15e76c77e2
* src/build: b7ac9b6900..ed970e6fb9
* src/ios: f0f33aa228..328f293c18
* src/testing: a018774366..e4cdb8e44f
* src/third_party: 999797279b..ca2aa8253d
* src/third_party/androidx: umiv0_Rv1JhkZJ9t-Onpz6nxwK83CDxtP6vrxt4XfTEC..-DNp3W6PysyC-ORTgGZEPS1cSd-wEW_KogHTJPL8jFgC
* src/third_party/kotlinc/current: TW-qEGSqERSeeEdITH9I24qeAFvaEkvFIzRTD0n5RIUC..qIzbRHHektfb27pu4daejk9dRfw48IcsmoUX4Xjt-JQC
* src/third_party/libaom/source/libaom: https://aomedia.googlesource.com/aom.git/+log/ebe946034a..f30a3ed4e8
* src/tools: f0ebc8e80c..205f9662a7
DEPS diff: b7561cbdd7..3f014a9b1e/DEPS

No update to Clang.

BUG=None

Change-Id: I7334b3929f06c1243f0fa63d51e20101d8005723
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/332302
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@{#41434}
2023-12-22 00:59:29 +00:00
api
Revert^2 "Remove deprecated ByteBufferReader and ByteBufferWriter functions"
2023-12-20 07:48:54 +00:00
audio
Use Environment in RtpTransportyControllerSend
2023-12-20 14:47:51 +00:00
build_overrides
Roll chromium_revision 01dc2965ca..917876224a (1209117:1211391)
2023-10-18 15:15:07 +00:00
call
Update WebRTC code version (2023-12-21T04:13:14).
2023-12-21 06:16:17 +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
Refactor Android Voip client to use rtc::ReceivedPackets
2023-12-14 11:57:17 +00:00
experiments
Remove WebRTC-Stats-RtxReceiveStats killswitch
2023-11-29 08:12:17 +00:00
g3doc
Update TODO example in the style guide.
2023-11-21 23:11:09 +00:00
infra
Setup tests for win11 tryservers
2023-12-13 16:35:55 +00:00
logging
Delete legacy RtcEventLogFactory Create functions
2023-12-03 11:10:35 +00:00
media
Revert "Unify access to SDP codec parameters"
2023-12-13 16:28:44 +00:00
modules
Ensure that sequence numbers are initialized in DelayBasedBwe unittests
2023-12-21 14:51:11 +00:00
net/dcsctp
Add SendMany method to dcsctp socket
2023-12-15 21:35:14 +00:00
p2p
[CodeHealth][1/2] Remove expired histogram: WebRTC.PeerConnection.IceRestartState
2023-12-21 12:17:19 +00:00
pc
Update LegacyStatsCollector to conform with Wc++11-narrowing
2023-12-21 14:58:48 +00:00
resources
Ignore .binarypb files.
2023-10-30 14:56:36 +00:00
rtc_base
Revert^2 "Remove deprecated ByteBufferReader and ByteBufferWriter functions"
2023-12-20 07:48:54 +00:00
rtc_tools
Refactor NetworkTester to use rtc::ReceivedPackets
2023-12-13 07:52:23 +00:00
sdk
Revert "Delete pc/peerconnection build target"
2023-12-21 12:40:44 +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
Revert "Unify access to SDP codec parameters"
2023-12-13 16:28:44 +00:00
tools_webrtc
Fix typo
2023-12-13 19:00:35 +00:00
video
Convert ByteBufferWriter to be type uint8_t
2023-12-15 12:27:50 +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
…
.style.yapf
Configure YAPF to follow PEP-8 altogether
2023-09-22 10:32:11 +00:00
.vpython
…
.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
Add environment_construction poison
2023-11-27 11:44:50 +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 b7561cbdd7..3f014a9b1e (1240168:1240385)
2023-12-22 00:59:29 +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
…
presubmit_test.py
…
PRESUBMIT.py
Disable cpplint readability/todo.
2023-11-30 08:03:01 +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
Deprecate RtcEventLogFactory constructor taking unused parameter
2023-12-07 21:46:56 +00:00
webrtc.gni
Add environment_construction poison
2023-11-27 11:44:50 +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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