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chromium-webrtc-autoroll 32632a80e3 Roll chromium_revision abe412afe7..0f7279573c (1026730:1026859) 
Change log: abe412afe7..0f7279573c
Full diff: abe412afe7..0f7279573c

Changed dependencies
* src/base: 468bdcb9c7..3d3a5fc325
* src/build: bed2979286..918b804ee4
* src/ios: 0b68b671ca..b6b463b851
* src/testing: fc01b2b000..aa10da843b
* src/third_party: 24f89f2000..0ec74d4211
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/abcd8ba23e..4c50669a2f
* src/third_party/depot_tools: 39e33e6eca..0ba2fd429d
* src/tools: f6d83033bb..d903290acb
DEPS diff: abe412afe7..0f7279573c/DEPS

No update to Clang.

BUG=None

Change-Id: I2326e2d0fbe39ade721d18a7a83f09b3b3973ae5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269224
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@{#37589}
2022-07-21 18:25:44 +00:00
api
Introduce method to simplify creation of ideal emulated network
2022-07-21 14:32:17 +00:00
audio
Implement RTCInboundRTPStreamStats.JitterBufferMinimumDelay
2022-07-20 09:14:03 +00:00
build_overrides
Roll chromium + fix: blacklist -> ignorelist for sanitizers suppressions
2021-05-27 16:16:01 +00:00
call
Update WebRTC code version (2022-07-21T04:04:55).
2022-07-21 05:58:35 +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
Migrate remaining webrtc usage of TaskQueueBase to absl::AnyInvocable
2022-07-20 08:15:08 +00:00
media
Adopt absl::string_view in modules/audio_coding/
2022-07-20 13:34:23 +00:00
modules
Avoid fraction_loss overflowing when packet loss is negative in send_side_bandwidth_estimation.cc.
2022-07-21 14:31:04 +00:00
net/dcsctp
Migrate remaining webrtc usage of TaskQueueBase to absl::AnyInvocable
2022-07-20 08:15:08 +00:00
p2p
[TCPConnection] Check for valid port_ in OnClose and OnConnect.
2022-07-19 15:56:43 +00:00
pc
Fix degradation_preference setting being ignored using RtpSender.SetParameters.
2022-07-20 13:48:27 +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 deprecated LogMessage ctor.
2022-07-21 10:33:15 +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
Implement RTCInboundRTPStreamStats.JitterBufferMinimumDelay
2022-07-20 09:14:03 +00:00
system_wrappers
Delete rtc_base/atomic_ops.h
2022-06-28 08:32:13 +00:00
test
Move common test flags to a separate library.
2022-07-21 09:52:54 +00:00
tools_webrtc
Clobber win bots
2022-07-19 11:34:02 +00:00
video
Delete tests with 4 temporal layers
2022-07-20 14:05:13 +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
Add missing header to fix build error when using linux system libraries
2022-07-19 12:25:42 +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 abe412afe7..0f7279573c (1026730:1026859)
2022-07-21 18:25:44 +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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