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chromium-webrtc-autoroll 47efd3310c Roll chromium_revision 5d2a61c45f..2244d2e5fc (1007024:1007134) 
Change log: 5d2a61c45f..2244d2e5fc
Full diff: 5d2a61c45f..2244d2e5fc

Changed dependencies
* src/base: fc17a58ad0..669c47a1c5
* src/build: 2d7e5fe1f6..568535bfc2
* src/ios: 230aed28f8..6514efa955
* src/testing: 7606027843..a844265bb1
* src/third_party: c8d013e64d..e14d72aa44
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/f2b13af941..01d3c4cd99
* src/third_party/depot_tools: 26a1d24c04..d67468faa0
* src/third_party/perfetto: d099a600a3..d1cb81f2aa
* src/tools: 8982d8925b..dab8ebf863
DEPS diff: 5d2a61c45f..2244d2e5fc/DEPS

No update to Clang.

BUG=None

Change-Id: I7498b4a7f8a02f21992f209af16761ae95610e86
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263560
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@{#36992}
2022-05-25 01:03:00 +00:00
api
Remove legacy (unused) config param: jitter_buffer_enable_rtx_handling
2022-05-21 23:06:21 +00:00
audio
Rename AudioReceiveStream to AudioReceiveStreamInterface
2022-05-23 08:44:26 +00:00
build_overrides
…
call
Remove ReceiveStream definition.
2022-05-24 14:53:24 +00:00
common_audio
Adopt absl::string_view in common_audio/
2022-05-13 15:00:14 +00:00
common_video
Remove dependency on rtc_base_approved from most targets
2022-04-25 12:15:30 +00:00
data
…
docs
Add contributing.md
2021-11-03 14:59:46 +00:00
examples
Delete useless code.
2022-05-24 12:07:21 +00:00
g3doc
Remove mention of MSVC support from docs
2022-05-02 09:26:32 +00:00
infra
Use ios_perf_test for M1 arm64 perf tester
2022-05-24 14:59:12 +00:00
logging
Improve RTC event log parser.
2022-05-18 09:14:40 +00:00
media
Reland "Replace InternalVideoEncoderFactory implementation with VideoEncoderFactoryTemplate."
2022-05-24 12:40:57 +00:00
modules
Update VP9Encoder::SupportsScalabilityMode
2022-05-24 15:51:53 +00:00
net/dcsctp
dcsctp: Cleanup Metrics
2022-05-13 15:11:34 +00:00
p2p
Remove 'new_connection' attribute from ConnectionInfo
2022-05-23 12:58:21 +00:00
pc
Reland "Don't create channel_manager++ when media_engine is not set"
2022-05-23 15:51:21 +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
Include all trace categories for internal usage
2022-05-23 09:10:36 +00:00
rtc_tools
Rename VideoReceiveStream to VideoReceiveStreamInterface
2022-05-22 10:54:38 +00:00
sdk
Delete useless code.
2022-05-24 12:07:21 +00:00
stats
stats: add frame assembly time stats
2022-05-18 09:16:10 +00:00
system_wrappers
Adopt absl::string_view in system_wrappers/
2022-05-17 08:23:24 +00:00
test
Rename AudioReceiveStream to AudioReceiveStreamInterface
2022-05-23 08:44:26 +00:00
tools_webrtc
Remove crosscompile for Perf Mac M1 Arm64 12
2022-05-23 09:54:48 +00:00
video
Log when conflicting recv video playout delays are set
2022-05-23 16:37:22 +00:00
.clang-format
…
.git-blame-ignore-revs
…
.gitignore
…
.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
Remove unused script webrtc_dashboard_upload.py
2022-03-21 12:54:42 +00:00
AUTHORS
GCC: Avoid symbol clash in RenderBuffer
2022-05-24 10:47:56 +00:00
BUILD.gn
Remove objc tests from rtc_unittests and move tests to sdk_unittests
2022-05-13 11:59:06 +00:00
CODE_OF_CONDUCT.md
…
codereview.settings
…
DEPS
Roll chromium_revision 5d2a61c45f..2244d2e5fc (1007024:1007134)
2022-05-25 01:03:00 +00:00
DIR_METADATA
…
ENG_REVIEW_OWNERS
Remove phoglund from ENG_REVIEW_OWNERS
2021-10-08 08:29:42 +00:00
g3doc.lua
…
LICENSE
…
license_template.txt
…
native-api.md
…
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
…
README.md
…
WATCHLISTS
Update WATCHLISTS
2021-08-23 13:37:55 +00:00
webrtc_lib_link_test.cc
…
webrtc.gni
Fork test/ios/google_test_runner.mm from Chromium's //base
2022-05-04 11:26:07 +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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