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chromium-webrtc-autoroll 047f656993 Roll chromium_revision e9f81a7ec9..28429d43ee (1190277:1190580) 
Change log: e9f81a7ec9..28429d43ee
Full diff: e9f81a7ec9..28429d43ee

Changed dependencies
* src/base: 0836b98f17..acfba9339c
* src/build: ef1e241143..334b8dfc9c
* src/ios: eb7e37edf9..cfab145357
* src/testing: 45e1922e9c..5b3c37f8da
* src/third_party: 093ca359d9..c11a2151af
* src/third_party/androidx: d3SZeP-0r67zTi3JXOzM68IUhwn1ub7W3a7J1ZHqSjIC..nPnktPVZnrvtO_GCDttGFN5jJfRhVxruWs81oS-DSZgC
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/47ca282426..b8c4f2d99a
* src/third_party/depot_tools: abf4847b04..427f0f43ad
* src/third_party/icu: de4ce0071e..985b9a6f70
* src/third_party/perfetto: e8f8016ad8..3a2627c87c
* src/tools: 1b9e3248e1..e62138ed7e
DEPS diff: e9f81a7ec9..28429d43ee/DEPS

No update to Clang.

BUG=None

Change-Id: Ib0fca5e339fb227e39224c99c34c1151612e102b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318383
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@{#40669}
2023-08-31 08:40:36 +00:00
api
Remove public GenerateKeyFrame(list-of-rids) API from RtpSender
2023-08-30 14:54:17 +00:00
audio
Move RTPTimestamp offset handling out of encoded transform delegate
2023-08-29 12:28:41 +00:00
build_overrides
Roll chromium_revision 6b95b8aa08..80c35b98ae (1175249:1180373)
2023-08-10 07:50:10 +00:00
call
Update WebRTC code version (2023-08-29T04:11:51).
2023-08-29 05:57:37 +00:00
common_audio
Avoid integer overflow detected by the compiler.
2023-08-30 08:24:30 +00:00
common_video
Fix fuzzing test issues reported by Chromium fuzzing test
2023-08-22 13:08:39 +00:00
data
…
docs
Update bug reporting and contributing docs
2023-07-24 13:04:29 +00:00
examples
Rland "Revert "Reland "Reland "Delete old Android ADM.""""
2023-06-30 13:10:12 +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
Drop email recipients from lkgr finder.
2023-08-15 08:39:43 +00:00
logging
Process events with the same timestamp in a defined order.
2023-06-20 14:21:17 +00:00
media
Remove EncodedFrame::MissingFrame and start removing Decode() param
2023-08-30 10:38:35 +00:00
modules
Add Opus FEC options to rtp_encode tool
2023-08-31 06:11:46 +00:00
net/dcsctp
Format ^(api|call|common_audio|examples|media|net|p2p|pc)/
2023-05-03 11:09:26 +00:00
p2p
More unused sigslot includes
2023-08-29 12:20:44 +00:00
pc
Remove public GenerateKeyFrame(list-of-rids) API from RtpSender
2023-08-30 14:54:17 +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
More unused sigslot includes
2023-08-29 12:20:44 +00:00
rtc_tools
Remove EncodedFrame::MissingFrame and start removing Decode() param
2023-08-30 10:38:35 +00:00
sdk
Remove EncodedFrame::MissingFrame and start removing Decode() param
2023-08-30 10:38:35 +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
Remove EncodedFrame::MissingFrame and start removing Decode() param
2023-08-30 10:38:35 +00:00
tools_webrtc
Added HEVC parser for WebRTC
2023-08-18 17:48:11 +00:00
video
Remove EncodedFrame::MissingFrame and start removing Decode() param
2023-08-30 10:38:35 +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
Roll chromium_revision 6b95b8aa08..80c35b98ae (1175249:1180373)
2023-08-10 07:50:10 +00:00
.mailmap
Add .mailmap for git.
2022-02-20 14:22:13 +00:00
.style.yapf
Fix mb.py presubmit issues.
2021-12-08 08:53:00 +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
Add Opus FEC options to rtp_encode tool
2023-08-31 06:11:46 +00:00
BUILD.gn
Add an async DNS resolver suitable for defaulting.
2023-08-23 10:08:52 +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 e9f81a7ec9..28429d43ee (1190277:1190580)
2023-08-31 08:40:36 +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
[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
Replace BuiltinVideo{Encoder,Decoder}Factory with Video{Encoder,Decoder}FactoryTemplate.
2023-03-31 19:31:50 +00:00
webrtc.gni
Add rtc_common_public_deps
2023-08-22 11:32:06 +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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