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chromium-webrtc-autoroll 0451baa9c2 Roll chromium_revision 13b44452d4..bb8855a075 (1157265:1157397) 
Change log: 13b44452d4..bb8855a075
Full diff: 13b44452d4..bb8855a075

Changed dependencies
* src/base: 544a69623f..2b4c08f00a
* src/build: 66f0829526..a838ac9f43
* src/buildtools/linux64: git_revision:1cd35c1b722472e714c30d12031af81443bb20ae..git_revision:4bd1a77e67958fb7f6739bd4542641646f264e5d
* src/buildtools/mac: git_revision:1cd35c1b722472e714c30d12031af81443bb20ae..git_revision:4bd1a77e67958fb7f6739bd4542641646f264e5d
* src/buildtools/win: git_revision:1cd35c1b722472e714c30d12031af81443bb20ae..git_revision:4bd1a77e67958fb7f6739bd4542641646f264e5d
* src/ios: 54d1b3c9cf..d6c4d25a9e
* src/testing: 6fbefed0a9..8cf7070d58
* src/third_party: 775177feef..1c94601439
* src/third_party/android_build_tools/manifest_merger: nGzWwB6kx8dBDTH7L1jUxm8y3INDX3qVm9co3ntGk7EC..UNXioFXYvz7k7UmE2WYAaXuYIK3Ky0aSQ0IuDEdS9soC
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/45275b64c5..ca32af6260
* src/third_party/libyuv: 2a5d7e2fbc..552571e8b2
* src/tools: d44da7e507..c1fc7563e9
DEPS diff: 13b44452d4..bb8855a075/DEPS

No update to Clang.

BUG=None

Change-Id: I70637bb8d786660994b61a3120be901da097481e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308660
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@{#40275}
2023-06-14 08:23:20 +00:00
api
Make SetRTPTimestamp pure virtual in TransformableAudioFrameInterface
2023-06-12 13:41:27 +00:00
audio
Ensure RtpSenderEgress run on worker queue
2023-06-09 13:40:35 +00:00
build_overrides
Define enable_safe_libcxx in build_overrides/build.gni.
2023-05-03 08:18:25 +00:00
call
Update WebRTC code version (2023-06-14T04:02:08).
2023-06-14 05:07:49 +00:00
common_audio
Format ^(api|call|common_audio|examples|media|net|p2p|pc)/
2023-05-03 11:09:26 +00:00
common_video
webrtc_libyuv: Add support for more video types for consistency
2023-04-24 19:06:25 +00:00
data
…
docs
docs: explain release note process
2023-05-24 14:09:54 +00:00
examples
Format ^(api|call|common_audio|examples|media|net|p2p|pc)/
2023-05-03 11:09:26 +00:00
experiments
Add tool for generating field trial registry header
2022-10-18 07:25:43 +00:00
g3doc
Add docs about adding a new test binary.
2023-03-07 11:12:33 +00:00
infra
[Analysis] Remove old threshold fields
2023-06-08 00:31:28 +00:00
logging
Use DD encoder/decoder in RTC event log encoder/parser.
2023-04-24 10:35:22 +00:00
media
Sort WebRtcAudio{Send,Receive}Channel implementation
2023-06-12 16:04:30 +00:00
modules
Change flexfec header writer to finalize header according to updated RFC
2023-06-13 14:30:09 +00:00
net/dcsctp
Format ^(api|call|common_audio|examples|media|net|p2p|pc)/
2023-05-03 11:09:26 +00:00
p2p
Format ^(api|call|common_audio|examples|media|net|p2p|pc)/
2023-05-03 11:09:26 +00:00
pc
Remove expired histograms WebRTC.PeerConnection.SrtpCryptoSuite
2023-06-09 14:48:38 +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
Remove unused Win32Window class
2023-05-25 10:48:40 +00:00
rtc_tools
[rct_tools/video_encoder] Add video encoder tool
2023-06-13 11:45:43 +00:00
sdk
Revert "Adopt EglThread in EglRenderer"
2023-06-13 11:27:27 +00:00
stats
Delete RTC[NonStandard/Restricted]StatsMember.
2023-05-25 08:39:48 +00:00
system_wrappers
Use //third_party/cpu_features directly
2023-06-02 07:17:36 +00:00
test
Implement DelayVariationCalculator for events analysis.
2023-06-08 17:42:53 +00:00
tools_webrtc
Roll chromium_revision a3756bb36c..dd02f6b781 (1154916:1156166)
2023-06-12 14:51:51 +00:00
video
Add H264Encoder::Create()
2023-06-02 17:40:26 +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
Set use_cxx to true.
2023-05-17 06:30:04 +00:00
.mailmap
Add .mailmap for git.
2022-02-20 14:22:13 +00:00
.style.yapf
…
.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
Expose setCodecPreferences/getCapabilities for android
2023-05-15 19:24:15 +00:00
BUILD.gn
Remove low_bandwidth_audio_test.
2023-06-01 07:20:38 +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 13b44452d4..bb8855a075 (1157265:1157397)
2023-06-14 08:23:20 +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
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
Remove low_bandwidth_audio_test.
2023-06-01 07:20:38 +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: 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
Replace BuiltinVideo{Encoder,Decoder}Factory with Video{Encoder,Decoder}FactoryTemplate.
2023-03-31 19:31:50 +00:00
webrtc.gni
Replace BuiltinVideo{Encoder,Decoder}Factory with Video{Encoder,Decoder}FactoryTemplate.
2023-03-31 19:31:50 +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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