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chromium-webrtc-autoroll 175d77ce77 Roll chromium_revision 37771fc5bc..4fff7d553e (1030366:1030494) 
Change log: 37771fc5bc..4fff7d553e
Full diff: 37771fc5bc..4fff7d553e

Changed dependencies
* src/base: e2855e8b9d..feb65a503a
* src/build: b1d3dcea6a..4527ab500e
* src/ios: 8b01b97a44..de78d0ec39
* src/testing: 0260b894e7..03d22b4428
* src/third_party: fd5911272d..9ac8c9aef2
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/682d9b1d88..81c03c3de9
* src/third_party/fuchsia-sdk/sdk: version:9.20220801.1.1..version:9.20220801.2.1
* src/tools: a75f817bb1..d06fb78650
DEPS diff: 37771fc5bc..4fff7d553e/DEPS

No update to Clang.

BUG=None

Change-Id: Id8597da1ae0f65e36deb1a87f79daddb8d9a59cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270362
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@{#37663}
2022-08-02 12:36:44 +00:00
api
stats: implement outbound-rtp.active
2022-07-28 13:35:40 +00:00
audio
Allow recursive check for RTC_DCHECK_RUN_ON macro
2022-07-26 09:27:23 +00:00
build_overrides
Roll chromium + fix: blacklist -> ignorelist for sanitizers suppressions
2021-05-27 16:16:01 +00:00
call
[WebRtcVideoReceiveStream] Add ability to config flexfec post init.
2022-08-02 11:37:26 +00:00
common_audio
Adopt absl::string_view in common_audio/
2022-05-13 15:00:14 +00:00
common_video
Migrate remaining webrtc usage of TaskQueueBase to absl::AnyInvocable
2022-07-20 08:15:08 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
docs
[ios] Remove the support for bitcode
2022-07-04 09:01:52 +00:00
examples
Delete ProcessThread and related Module interface
2022-07-04 10:20:35 +00:00
g3doc
Clarify how to reference WebRTC bugs in TODOs
2022-07-01 08:03:34 +00:00
infra
Reland "Wait for frames to arrive in WgcCapturer instead of returning nothing."
2022-07-06 20:28:26 +00:00
logging
Delete inter arrival jitter rtcp packet as unused
2022-07-27 14:53:05 +00:00
media
[WebRtcVideoReceiveStream] Add ability to config flexfec post init.
2022-08-02 11:37:26 +00:00
modules
Move TaskQueuePacedSender::EnqueuePackets tracing onto the task queue.
2022-08-02 12:26:34 +00:00
net/dcsctp
Migrate remaining webrtc usage of TaskQueueBase to absl::AnyInvocable
2022-07-20 08:15:08 +00:00
p2p
Modify GetBestIP to prefer global IPv6 addresses to the link local IPv6 addresses.
2022-08-02 10:16:35 +00:00
pc
stats: implement outbound-rtp.active
2022-07-28 13:35:40 +00:00
resources
AEC3: Changing the default for the use_conservative_tail_frequency_response flag.
2021-12-21 17:35:26 +00:00
rtc_base
Remove unused sigslot::repeater
2022-08-02 10:17:38 +00:00
rtc_tools
Add lower/upper link capacity to the outgoing bitrate graph.
2022-07-19 13:22:32 +00:00
sdk
Migrate remaining webrtc usage of TaskQueueBase to absl::AnyInvocable
2022-07-20 08:15:08 +00:00
stats
stats: implement outbound-rtp.active
2022-07-28 13:35:40 +00:00
system_wrappers
Delete nisse@webrtc.org from OWNERS files
2022-07-28 08:47:38 +00:00
test
Speed up vp9 encoder reference fuzzer
2022-08-02 07:57:43 +00:00
tools_webrtc
Add new webrtc-only dependency to auto roller
2022-07-29 12:43:10 +00:00
video
[WebRtcVideoReceiveStream] Add ability to config flexfec post init.
2022-08-02 11:37:26 +00:00
.clang-format
Add IncludeBlocks to clang-format.
2021-02-03 16:29:07 +00:00
.git-blame-ignore-revs
Let git-hyper-blame ignore new format cleanup.
2019-07-11 16:18:51 +00:00
.gitignore
Add .cache to .gitignore.
2021-01-20 15:01:07 +00:00
.gn
Prevent jsoncpp from hiding deprecated declarations in WebRTC
2022-04-11 12:33:47 +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 protobuf-py2_py3 wheel.
2022-07-01 15:17:36 +00:00
AUTHORS
Fixed mistake in DelayBasedBwe. Use audio_inter_arrival_delta_ when separate audio is enabled.
2022-07-28 09:31:49 +00:00
BUILD.gn
SVC: Add end to end tests for VP8 and VP9
2022-06-22 11:07:01 +00:00
CODE_OF_CONDUCT.md
Reference "main" branches instead of "master" branches.
2021-07-15 11:07:44 +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 37771fc5bc..4fff7d553e (1030366:1030494)
2022-08-02 12:36:44 +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
g3doc.lua
Improve webrtc documentation infra. Preview at:
2021-03-30 10:29:30 +00:00
LICENSE
…
license_template.txt
…
native-api.md
Reference "main" branches instead of "master" branches.
2021-07-15 11:07:44 +00:00
OWNERS
Fix add some eng prod owners to PRESUBMIT.py.
2022-03-18 13:19:07 +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
Update portaudio to the latest
2022-05-13 09:01:34 +00:00
pylintrc
tools_webrtc dir converted to py3 + top level PRESUBMIT script
2022-02-08 14:42:26 +00:00
README.chromium
Add CPEPrefix.
2020-07-13 11:42:07 +00:00
README.md
doc: add g3doc sitemap to toplevel readme
2021-07-23 07:55:17 +00:00
WATCHLISTS
Update WATCHLISTS
2021-08-23 13:37:55 +00:00
webrtc_lib_link_test.cc
Deprecate PeerConnectionFactory::CreatePeerConnection
2021-05-10 08:47:48 +00:00
webrtc.gni
[Cast Convergence] Replace is_chromecast with new args
2022-06-16 00:50:08 +00:00
whitespace.txt
Trigger CI bots
2021-12-16 17:45:31 +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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