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chromium-webrtc-autoroll 6fa743fbab Roll chromium_revision 336689d906..e1fb84c37d (1249985:1250109) 
Change log: 336689d906..e1fb84c37d
Full diff: 336689d906..e1fb84c37d

Changed dependencies
* src/base: baa20ba9ab..36ecc8e397
* src/buildtools: 17ce6d2f04..aadc2aa5f7
* src/buildtools/linux64: git_revision:b5adfe5f574d7110b80feb9aae6fae97c366840b..git_revision:f99e015ac35f689cfdbf46e4eb174e5d2da78d8e
* src/buildtools/mac: git_revision:b5adfe5f574d7110b80feb9aae6fae97c366840b..git_revision:f99e015ac35f689cfdbf46e4eb174e5d2da78d8e
* src/buildtools/win: git_revision:b5adfe5f574d7110b80feb9aae6fae97c366840b..git_revision:f99e015ac35f689cfdbf46e4eb174e5d2da78d8e
* src/ios: 1217b7e1ac..e18cc47f93
* src/testing: 922daad5d2..450bfd79ee
* src/third_party: 5b889f2713..692fab5c00
* src/third_party/perfetto: 28eadcf5ef..d6af17fef2
* src/tools: 2f3f0542ce..51d5368f22
DEPS diff: 336689d906..e1fb84c37d/DEPS

No update to Clang.

BUG=None

Change-Id: Idff6ad4241514ebfcaec85e444a0ed10483495b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335661
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@{#41594}
2024-01-22 12:55:26 +00:00
api
[Stats] Make RTCStatsMember<T> a type alias for absl::optional<T>.
2024-01-22 12:48:07 +00:00
audio
Record audio timestamps from iOS.
2024-01-19 12:35:53 +00:00
build_overrides
Roll chromium_revision 01dc2965ca..917876224a (1209117:1211391)
2023-10-18 15:15:07 +00:00
call
Update WebRTC code version (2024-01-21T04:12:31).
2024-01-21 05:58:41 +00:00
common_audio
Fix pointer overflow in neon implemenation of audio filters
2023-10-13 06:41:08 +00:00
common_video
Fix fuzzing issue for h265 bitstream parser
2024-01-19 16:34:15 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
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
Adds WebRTC-AV1-OverridePriorityBitrate to change bit rate allocation between audio and video
2024-01-20 07:15:38 +00:00
g3doc
Update TODO example in the style guide.
2023-11-21 23:11:09 +00:00
infra
Run video_codec_perf_tests using the quick mode on Android try bots.
2024-01-16 10:07:48 +00:00
logging
Set visibility for RTC event log events BUILD targets
2024-01-19 17:36:36 +00:00
media
Deprecate rtc::TaskQueue variant of AudioProcessing::CreateAndAttachAecDump
2024-01-15 13:36:35 +00:00
modules
Tighten som DCHECKs to CHECKs in VP9 packetization.
2024-01-19 16:58:09 +00:00
net/dcsctp
Fixing unspecified evaluation order of std:move(), to avoid future issues.
2024-01-16 08:53:28 +00:00
p2p
Use is_* getters when checking the Candidate type
2024-01-19 15:37:32 +00:00
pc
Revert^4 "Delete pc/peerconnection build target"
2024-01-21 22:59:48 +00:00
resources
Ignore .binarypb files.
2023-10-30 14:56:36 +00:00
rtc_base
Uninherit TaskQueueForTest from legacy rtc::TaskQueue
2024-01-19 15:28:49 +00:00
rtc_tools
Remove usage of rtc::TaskQueue in video replay tool
2024-01-18 13:02:16 +00:00
sdk
Revert^4 "Delete pc/peerconnection build target"
2024-01-21 22:59:48 +00:00
stats
[Stats] Make RTCStatsMember<T> a type alias for absl::optional<T>.
2024-01-22 12:48:07 +00:00
system_wrappers
Use //third_party/cpu_features directly
2023-06-02 07:17:36 +00:00
test
Use default H264 SDP parameters
2024-01-19 15:01:12 +00:00
tools_webrtc
Add disable_trace_events compilation
2024-01-08 11:01:30 +00:00
video
Remove usage of the rtc::TaskQueue in video/
2024-01-22 11:50:26 +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
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
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 336689d906..e1fb84c37d (1249985:1250109)
2024-01-22 12:55:26 +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
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
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