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chromium-webrtc-autoroll 2cbf35ffc4 Roll chromium_revision bac0ff1a2f..b5895b16b1 (1020023:1020162) 
Change log: bac0ff1a2f..b5895b16b1
Full diff: bac0ff1a2f..b5895b16b1

Changed dependencies
* src/base: 5650bc6a10..ba94246c92
* src/build: b09cdd74ca..473484fe92
* src/buildtools/clang_format/script: e435ad79c1..8b525d2747
* src/buildtools/linux64: git_revision:29accf5ac2eadfc53e687081583b7bc1592a8839..git_revision:ecaaf4b9e58a312a1610a37999eeccf58f73e264
* src/buildtools/mac: git_revision:29accf5ac2eadfc53e687081583b7bc1592a8839..git_revision:ecaaf4b9e58a312a1610a37999eeccf58f73e264
* src/buildtools/win: git_revision:29accf5ac2eadfc53e687081583b7bc1592a8839..git_revision:ecaaf4b9e58a312a1610a37999eeccf58f73e264
* src/ios: 90479fbd3f..e87365a15a
* src/testing: cc7d6973e1..b97d5810aa
* src/third_party: 38968a908c..227535f25a
* src/third_party/androidx: lYZhX1u-VWgsLITQXwIxYzbrN3URNzSwPTP2iuDhPBQC..qYbZhGFI6Byx-h1-gMAwav_sOAyRgupup2LcOewkUwYC
* src/third_party/freetype/src: 2db58e061e..bec4ef415e
* src/third_party/fuchsia-sdk/sdk: version:8.20220701.0.1..version:8.20220701.2.1
* src/third_party/perfetto: f413bdf835..63cdebc57f
* src/tools: 0b0dd91d7b..efc278b7d5
DEPS diff: bac0ff1a2f..b5895b16b1/DEPS

No update to Clang.

BUG=None

Change-Id: I8c1662f24d95d6af4ceb004e69df7d804b84b1d9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267441
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#37405}
2022-07-01 18:35:40 +00:00
api
Add support for scalability modes L2T3 and S2T3
2022-07-01 08:17:04 +00:00
audio
Move to_queued_task.h and pending_task_safety_flag.h into public API
2022-06-17 09:20:39 +00:00
build_overrides
Roll chromium + fix: blacklist -> ignorelist for sanitizers suppressions
2021-05-27 16:16:01 +00:00
call
video_replay: add flexfec support
2022-07-01 09:42:24 +00:00
common_audio
Adopt absl::string_view in common_audio/
2022-05-13 15:00:14 +00:00
common_video
Add 420 and 422 10 bit h264 decoding.
2022-06-17 11:12:10 +00:00
data
…
docs
Add contributing.md
2021-11-03 14:59:46 +00:00
examples
Adopt absl::string_view in p2p/
2022-06-30 13:19:18 +00:00
g3doc
Clarify how to reference WebRTC bugs in TODOs
2022-07-01 08:03:34 +00:00
infra
Wait for frames to arrive in WgcCapturer instead of returning nothing.
2022-07-01 17:42:20 +00:00
logging
Change RTCEventLogFactory to have a const Create function
2022-06-28 23:48:37 +00:00
media
When VP9 SVC is used, use SvcConfig to set max bitrate for the stream.
2022-06-29 14:23:08 +00:00
modules
Wait for frames to arrive in WgcCapturer instead of returning nothing.
2022-07-01 17:42:20 +00:00
net/dcsctp
dcsctp: Refactor OutstandingData
2022-07-01 16:44:40 +00:00
p2p
Adopt absl::string_view in p2p/
2022-06-30 13:19:18 +00:00
pc
Do not allow simulcast to be turned off using SDP munging
2022-07-01 09:06:44 +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
Use generate_stubs without //base dependency
2022-07-01 10:50:54 +00:00
rtc_tools
video_replay: add flexfec support
2022-07-01 09:42:24 +00:00
sdk
Switch from junit_binary to robolectric_binary.
2022-06-30 08:02:18 +00:00
stats
Implement Outbound/InboundRtpStreamStats.mid.
2022-06-17 08:44:09 +00:00
system_wrappers
Delete rtc_base/atomic_ops.h
2022-06-28 08:32:13 +00:00
test
Adopt absl::string_view in p2p/
2022-06-30 13:19:18 +00:00
tools_webrtc
Make WebRTC use third_party/libevent rather than base/third_party/libevent
2022-06-30 07:43:49 +00:00
video
Delete rtc_base/atomic_ops.h
2022-06-28 08:32:13 +00:00
.clang-format
Add IncludeBlocks to clang-format.
2021-02-03 16:29:07 +00:00
.git-blame-ignore-revs
…
.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
GCC: Avoid symbol clash in RenderBuffer
2022-05-24 10:47:56 +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
…
DEPS
Roll chromium_revision bac0ff1a2f..b5895b16b1 (1020023:1020162)
2022-07-01 18:35:40 +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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