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chromium-webrtc-autoroll cddcbccb23 Roll chromium_revision 245f67a353..4b479b78e7 (1247088:1247241) 
Change log: 245f67a353..4b479b78e7
Full diff: 245f67a353..4b479b78e7

Changed dependencies
* src/base: a192831e25..5083db544c
* src/build: 5215f51451..8424be9497
* src/buildtools/linux64: git_revision:5fd939de8a66c915778c14a2788a857be858ef6c..git_revision:b8562a4abd9528372f274b1155aec65c57ca84fa
* src/buildtools/mac: git_revision:5fd939de8a66c915778c14a2788a857be858ef6c..git_revision:b8562a4abd9528372f274b1155aec65c57ca84fa
* src/buildtools/win: git_revision:5fd939de8a66c915778c14a2788a857be858ef6c..git_revision:b8562a4abd9528372f274b1155aec65c57ca84fa
* src/ios: a152495063..579b4ba6b4
* src/testing: 46baa9db02..23de4d7889
* src/third_party: ec4e03eef3..6d16fe523d
* src/third_party/androidx: m-I30LLs_nNvsNtlBBaiWNEZJLLhBTlJvpCS5bRLXmAC..UQGTnEcqPqYSOIY-7jp2iDnYhSSg3usWw6fiExBaZq8C
* src/third_party/libc++abi/src: a1171ad549..aa4b8671c4
* src/third_party/perfetto: 22522e4de2..113f941058
* src/third_party/r8: eMgxt4IbanIdGLJr4iyWKbvZXmKzyr-QClR3CvzS4_QC..BxPeo4f5KOOBKB_IUg_cMRUUhX_5U8clQUoXPFHDeEsC
* src/tools: f8f406ec80..e70dd78970
DEPS diff: 245f67a353..4b479b78e7/DEPS

No update to Clang.

BUG=None

Change-Id: I6b4c7d85adcb4daf43e536f35dca8d8413ac50f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334501
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@{#41529}
2024-01-15 18:48:10 +00:00
api
[Stats] Migrate integration test to Attributes().
2024-01-12 11:30:06 +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 (2024-01-15T04:02:23).
2024-01-15 05:36:26 +00:00
common_audio
Fix pointer overflow in neon implemenation of audio filters
2023-10-13 06:41:08 +00:00
common_video
Expose WebRTC H.264 SPS parser APIs.
2024-01-03 10:48:51 +00:00
data
…
docs
Fix links
2024-01-12 16:39:54 +00:00
examples
Refactor Android Voip client to use rtc::ReceivedPackets
2023-12-14 11:57:17 +00:00
experiments
Remove unused field trial DisablePacerEmergencyStop
2024-01-09 15:03:34 +00:00
g3doc
Update TODO example in the style guide.
2023-11-21 23:11:09 +00:00
infra
Upgrade iOS simulator configuration.
2024-01-08 11:00:27 +00:00
logging
Remove RtcEventLogImpl dependency on rtc::TaskQueue
2024-01-08 12:12:10 +00:00
media
Deprecate rtc::TaskQueue variant of AudioProcessing::CreateAndAttachAecDump
2024-01-15 13:36:35 +00:00
modules
Deprecate rtc::TaskQueue variant of AudioProcessing::CreateAndAttachAecDump
2024-01-15 13:36:35 +00:00
net/dcsctp
dcsctp: Relax thread sequence checker
2024-01-09 11:50:44 +00:00
p2p
[CodeHealth][1/2] Remove expired histogram: WebRTC.PeerConnection.IceRegatheringReason
2023-12-22 09:51:27 +00:00
pc
[Stats] Migrate integration test to Attributes().
2024-01-12 11:30:06 +00:00
resources
Ignore .binarypb files.
2023-10-30 14:56:36 +00:00
rtc_base
Export rtc::Socket for Chromium component builds
2024-01-11 10:24:03 +00:00
rtc_tools
Refactor NetworkTester to use rtc::ReceivedPackets
2023-12-13 07:52:23 +00:00
sdk
[Stats] Migrate Java to Attributes().
2024-01-12 10:33:31 +00:00
stats
[Stats] Migrate RTCStats/Test to Attributes.
2024-01-12 10:27:14 +00:00
system_wrappers
Use //third_party/cpu_features directly
2023-06-02 07:17:36 +00:00
test
Deprecate rtc::TaskQueue variant of AudioProcessing::CreateAndAttachAecDump
2024-01-15 13:36:35 +00:00
tools_webrtc
Add disable_trace_events compilation
2024-01-08 11:01:30 +00:00
video
FrameCadenceAdapter: now sets queue_overload based on encoder load
2024-01-09 14:29:04 +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
Revert "Fix 'Image will be cropped if WindowCapturerWinGdi used'"
2024-01-12 10:16:26 +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 245f67a353..4b479b78e7 (1247088:1247241)
2024-01-15 18:48:10 +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
Add H264 packetizer fuzzer
2024-01-09 13:32:42 +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
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