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chromium-webrtc-autoroll 83f62cbfbc Roll chromium_revision d5197a5f61..bf69d8bee3 (1032616:1032795) 
Change log: d5197a5f61..bf69d8bee3
Full diff: d5197a5f61..bf69d8bee3

Changed dependencies
* src/base: 0672b38bfa..f75d20d419
* src/build: b1868b3db9..4ad3bc949e
* src/buildtools/linux64: git_revision:c8c63300ac8ecb66d8126af5407257209ae59044..git_revision:3d773bba0927e67eae8fdaee5e28b0f6203d3bee
* src/buildtools/mac: git_revision:c8c63300ac8ecb66d8126af5407257209ae59044..git_revision:3d773bba0927e67eae8fdaee5e28b0f6203d3bee
* src/buildtools/third_party/libunwind/trunk: 47974e9376..c38cbd4028
* src/buildtools/win: git_revision:c8c63300ac8ecb66d8126af5407257209ae59044..git_revision:3d773bba0927e67eae8fdaee5e28b0f6203d3bee
* src/ios: 82bf060677..de5514c6d3
* src/testing: c85b2cfda0..22430aac05
* src/third_party: 0bfe4df4ba..0d8b1d8aa1
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/1c7fc8f4bd..dd7b9e9e26
* src/third_party/depot_tools: 2cd48234f1..647cfe6bf1
* src/third_party/freetype/src: 90e1e39f50..7cd3f19f21
* src/third_party/libaom/source/libaom: https://aomedia.googlesource.com/aom.git/+log/60e4e24a41..121f385145
* src/tools: 3840892f3d..23fb3d4a31
DEPS diff: d5197a5f61..bf69d8bee3/DEPS

No update to Clang.

BUG=None

Change-Id: I66e14d049625a2d5a22965b4662f1cdf1364db1d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270945
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@{#37714}
2022-08-09 00:50:37 +00:00
api
Inline assert in RTC_DCHECK_RUN_ON macro
2022-08-03 13:28:28 +00:00
audio
Clarify and extend test support for certain sample rates in audio processing
2022-08-03 14:26:36 +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 (2022-08-08T04:02:58).
2022-08-08 06:42:45 +00:00
common_audio
Reenable WebRTC PushResampler format checks on Windows clang debug builds
2022-08-05 11:03:08 +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
Make nack history configurable.
2022-08-05 22:58:43 +00:00
modules
Wayland screencast: fix crash on fd ownership violation
2022-08-08 17:19:46 +00:00
net/dcsctp
Migrate remaining webrtc usage of TaskQueueBase to absl::AnyInvocable
2022-07-20 08:15:08 +00:00
p2p
clean up misc TimeDelta use
2022-08-02 13:52:36 +00:00
pc
Remove sigslot usage from DtmfProviderInterface
2022-08-03 14:16:35 +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
Fix LogLineRef for when logging is disabled.
2022-08-03 15:16:54 +00:00
rtc_tools
Add lower/upper link capacity to the outgoing bitrate graph.
2022-07-19 13:22:32 +00:00
sdk
Map the stopped transceiver direction on Android
2022-08-05 07:03:16 +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
In Vp9 encoder wrapper force key frame on svc controller request
2022-08-03 19:15:45 +00:00
tools_webrtc
Option to pack unstripped .so into libwebrtc.aar
2022-08-04 13:31:45 +00:00
video
Relax expectations for a few VideoStreamEncoder tests.
2022-08-08 16:29:45 +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 d5197a5f61..bf69d8bee3 (1032616:1032795)
2022-08-09 00:50:37 +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
Languages
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C 2.2%
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