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chromium-webrtc-autoroll aab4a4c753 Roll chromium_revision bbaf0f70e5..3a910ae6c9 (1239639:1239781) 
Change log: bbaf0f70e5..3a910ae6c9
Full diff: bbaf0f70e5..3a910ae6c9

Changed dependencies
* src/base: 5fac0dca46..81f767e332
* src/ios: 875cd9d321..a732c369ca
* src/testing: 9fc5a09b08..39a0e2448a
* src/third_party: 6993b56ab7..15d8817a40
* src/third_party/androidx: aX7aKt_tQj6cTy5WGXiiPYsjab9D2DZwSQc5xPUW0KUC..7tjLAjwGeXH-gSAYbOjpY5eVR6l6ndabiITTsM44N-AC
* src/third_party/libunwind/src: 3cafb40647..2602aecdf4
* src/third_party/perfetto: 7a68d15808..e3e11385e2
* src/tools: a00ea07eaf..ef31376f4f
DEPS diff: bbaf0f70e5..3a910ae6c9/DEPS

No update to Clang.

BUG=None

Change-Id: I68ff154a9a119c95877d4e517b0ed70d1a873f39
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/332160
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@{#41424}
2023-12-20 20:38:56 +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
Use Environment in RtpTransportyControllerSend
2023-12-20 14:47:51 +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
OpenH264: report error on unsupported pixel format
2023-12-20 08:24:24 +00:00
net/dcsctp
Add SendMany method to dcsctp socket
2023-12-15 21:35:14 +00:00
p2p
Revert^2 "Remove deprecated ByteBufferReader and ByteBufferWriter functions"
2023-12-20 07:48:54 +00:00
pc
Do no return media-playout stats unless there is an audio receiver
2023-12-20 16:14:05 +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
Deprecate RtcEventLogFactory constructor taking unused parameter
2023-12-07 21:46:56 +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
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
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 bbaf0f70e5..3a910ae6c9 (1239639:1239781)
2023-12-20 20:38:56 +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
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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