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chromium-webrtc-autoroll a1475c2210 Roll chromium_revision f7fb707ebb..116647225e (1201480:1202523) 
Change log: f7fb707ebb..116647225e
Full diff: f7fb707ebb..116647225e

Changed dependencies
* src/base: 7d979505e8..226ff96839
* src/build: 5f393261bc..ddf84d52d6
* src/buildtools: 9d9c064e40..2c6e20e2ff
* src/ios: 74621ea7bd..d3aa14fe6f
* src/testing: 3c3be88d84..a151ef21fd
* src/third_party: 5005138049..bebeb00015
* src/third_party/androidx: a9CGOell056bhQij2JhATC4nOCgZpeDXqqwWjr1_ewAC..3HbvNkC0xzB9gU74utNAxJTRJC5xKCGjqIwNIxkoOoUC
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/a1843d660b..d24a38200f
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/0a5cbdd220..328888ea45
* src/third_party/depot_tools: d95084e851..34e0ecf20e
* src/third_party/libaom/source/libaom: https://aomedia.googlesource.com/aom.git/+log/82018abee6..96983fe072
* src/third_party/libc++/src: 26cbb86c20..055d494c5c
* src/third_party/libc++abi/src: c28dc42f0d..e8e4eb8f1c
* src/third_party/libunwind/src: f2ca661b1d..724cd40fb6
* src/third_party/libvpx/source/libvpx: 6da1bd01d6..38a707faef
* src/third_party/perfetto: f92196dfb6..d5143d26fa
* src/third_party/r8: w5NK9teKNDuEhU_7rQE8WYwuBzdXA8lVkJwo6VOgWv8C..ZQH8yoO1Ol5rCFRL2hzp4_mcdq23Fk19Z_gVHn5pzk8C
* src/tools: 295b391656..afa4d85c5a
DEPS diff: f7fb707ebb..116647225e/DEPS

No update to Clang.

BUG=None

Change-Id: I5e4154992191e077e7467202bae39cbf1b05aaa5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321784
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@{#40830}
2023-09-28 13:21:39 +00:00
api
Enable SRTP GCM ciphers by default
2023-09-28 10:18:56 +00:00
audio
Pass the correct abs_capture_timestamp while cloning audio frame
2023-09-26 15:57:52 +00:00
build_overrides
Roll chromium_revision 6b95b8aa08..80c35b98ae (1175249:1180373)
2023-08-10 07:50:10 +00:00
call
Update WebRTC code version (2023-09-26T04:02:19).
2023-09-26 06:51:20 +00:00
common_audio
Avoid integer overflow detected by the compiler.
2023-08-30 08:24:30 +00:00
common_video
Fix fuzzing issue reported by Chromium fuzzing test
2023-09-05 09:29:27 +00:00
data
…
docs
Update bug reporting and contributing docs
2023-07-24 13:04:29 +00:00
examples
Roll chromium_revision 6ac7929166..eef62e8a0c (1190797:1197906)
2023-09-19 12:06:33 +00:00
experiments
Populate field trial registry
2023-09-27 07:35:38 +00:00
g3doc
Add instructions for adding and removing field trials
2023-09-27 07:41:45 +00:00
infra
Revert "Remove device_status dimensions for perf bot"
2023-09-12 07:07:46 +00:00
logging
Add codec name H265 to support H265 in WebRTC
2023-09-20 09:25:32 +00:00
media
Reduce log verbosity in codec selection implementation
2023-09-28 07:57:06 +00:00
modules
Subtract an additional 5kbps of the bitrate when backing off.
2023-09-28 09:36:36 +00:00
net/dcsctp
dcsctp: Don't generate FORWARD-TSN across stream resets
2023-09-28 11:31:02 +00:00
p2p
Change PortInterface::Type to string_view and make type_ member const
2023-09-25 13:48:50 +00:00
pc
Enable SRTP GCM ciphers by default
2023-09-28 10:18:56 +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
Add AV1 singlecast bitrate limits
2023-09-27 22:19:55 +00:00
rtc_tools
Rename EncodedImage property Timetamp to RtpTimestamp
2023-09-24 20:06:48 +00:00
sdk
Deprecate old names for EncodedImage::RtpTimestamp accessor and setter
2023-09-28 07:29:22 +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
Deprecate old names for EncodedImage::RtpTimestamp accessor and setter
2023-09-28 07:29:22 +00:00
tools_webrtc
Make CreateOffer/CreateAnswer return RTCErrorOr<SessionDescription>
2023-09-26 09:41:30 +00:00
video
Rename EncodedImage property Timetamp to RtpTimestamp
2023-09-24 20:06:48 +00:00
.clang-format
Add IncludeBlocks to clang-format.
2021-02-03 16:29:07 +00:00
.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
Make WEBRTC_UNSAFE_FUZZER_MODE dependent only on use_fuzzing_engine
2023-09-20 16:04:17 +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 f7fb707ebb..116647225e (1201480:1202523)
2023-09-28 13:21:39 +00:00
DIR_METADATA
Move metadata in OWNERS files to DIR_METADATA files.
2021-02-08 19:09:33 +00:00
ENG_REVIEW_OWNERS
Remove phoglund from ENG_REVIEW_OWNERS
2021-10-08 08:29:42 +00:00
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
Configure Pylint to follow PEP-8
2023-09-25 15:56:09 +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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