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chromium-webrtc-autoroll 3725398e2a Roll chromium_revision 6a1112fe10..1a86984556 (1204286:1204416) 
Change log: 6a1112fe10..1a86984556
Full diff: 6a1112fe10..1a86984556

Changed dependencies
* src/base: 0e9e349c40..19c600fca5
* src/build: 690dc92b42..8a44cdc629
* src/buildtools: 50c348906c..c8bee3e69d
* src/ios: 50cbb98de4..533f961c05
* src/testing: 29a09864e6..86a2370993
* src/third_party: 921a7946ae..55c4796b77
* src/third_party/androidx: IrpbN40qmrgJhRlrreSznX0-bvcK-SQ73XYHV4x-Y0gC..tfNYaYQPGwu8upYGYo38upn04cSQFYR49Ai89gPC0PAC
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/4f81c1e295..b018f901e8
* src/third_party/depot_tools: 90a30a5b53..f90b3b035c
* src/third_party/libc++/src: 7cf98622ab..a094e2535c
* src/third_party/libunwind/src: 43e5a34c5b..7608093d21
* src/third_party/perfetto: a3d4c1de9b..d44dbe4ad2
* src/tools: 6ed8cc384a..8a2e7818c8
DEPS diff: 6a1112fe10..1a86984556/DEPS

No update to Clang.

BUG=None

Change-Id: Ifb09f2eba9559a11d4711b387405c287613038a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322240
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@{#40858}
2023-10-03 02:53:15 +00:00
api
Revert "Enable SRTP GCM ciphers by default"
2023-09-28 14:42:18 +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 (2023-10-02T04:11:24).
2023-10-02 06:03:52 +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
Remove old data files.
2018-10-05 14:40:21 +00:00
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
Remove field trial WebRTC-Turn-AllowSystemPorts
2023-10-02 15:09:40 +00:00
g3doc
Relax the field trial policy to not require an open bug
2023-09-29 08:03:21 +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
Clear PacketBuffer on large negative jumps at the start of the video stream
2023-09-29 08:56:15 +00:00
net/dcsctp
dcsctp: Only process meaningful FORWARD-TSN
2023-10-02 16:05:11 +00:00
p2p
Remove field trial WebRTC-Turn-AllowSystemPorts
2023-10-02 15:09:40 +00:00
pc
Remove more Codec-related templating in MediaSession
2023-09-29 10:52:27 +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 field trial for enabling SSL client hello extension permutation
2023-09-28 13:34:38 +00:00
rtc_tools
Rename EncodedImage property Timetamp to RtpTimestamp
2023-09-24 20:06:48 +00:00
sdk
Adopt EglThread in EglRenderer once again.
2023-09-29 13:37:21 +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
Adds support for tracking OnFrame PostTask delta times
2023-10-02 12:03:36 +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
Add .cache to .gitignore.
2021-01-20 15:01:07 +00:00
.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
Don't add webrtc-reviews@ to CC, it can be added globally on Gerrit
2018-10-25 08:19:53 +00:00
DEPS
Roll chromium_revision 6a1112fe10..1a86984556 (1204286:1204416)
2023-10-03 02:53:15 +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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