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chromium-webrtc-autoroll 50decc6bd0 Roll chromium_revision c4e2cd0641..3416746b88 (923181:923281) 
Change log: c4e2cd0641..3416746b88
Full diff: c4e2cd0641..3416746b88

Changed dependencies
* src/build: 103712808f..b4be3eb78f
* src/ios: 0ff547a875..d11c980c8a
* src/testing: cb7a980fa7..2d497d93e4
* src/third_party: d4e588bc54..b82e1dc3f4
* src/third_party/androidx: jagpxzoq8SusfDjtewx558yIOnyvOSm2eGzxQc7-n7EC..-p7HUEhWbHNpHzWMFxgf5JMct4Dp0DHUeY6Ybtni3IwC
* src/third_party/depot_tools: c1fd44b01a..9a76798c9e
* src/tools: 3102a8e728..b92dee658b
DEPS diff: c4e2cd0641..3416746b88/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: I0eb0ad7ddf80521ad7f4185024bd20a74045a6d5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232586
Reviewed-by: 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@{#35048}
2021-09-21 08:52:19 +00:00
api
New H264PacketBuffer consolidating a bunch of H264 specific hacks into one class.
2021-09-15 09:57:29 +00:00
audio
Deduplicate set of the rtp header extension uri constants
2021-09-14 13:38:44 +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 (2021-09-21T04:02:15).
2021-09-21 05:22:29 +00:00
common_audio
Use backticks not vertical bars to denote variables in comments
2021-08-10 10:40:03 +00:00
common_video
Fix integer overflow in h264 pps parser
2021-09-20 11:28:36 +00:00
data
…
docs
Update Mac prerequisites
2021-08-23 19:52:17 +00:00
examples
Replace AV1X with AV1
2021-09-14 08:29:02 +00:00
g3doc
Update missed link to chromium C++11 styleguide
2021-09-20 21:05:39 +00:00
logging
Migrate rtc event log from rtc::BitBuffer to BitstreamReader
2021-09-07 14:03:27 +00:00
media
APM: remove webrtc::Config
2021-09-17 11:21:32 +00:00
modules
Default the safetyped calculation of packet interarrival times.
2021-09-17 13:58:55 +00:00
net/dcsctp
dcsctp: move HandoverReadinessStatus::ToString definition to build target public:socket
2021-09-20 12:25:15 +00:00
p2p
Delete unused listen-mode of AsyncStunTCPSocket
2021-09-21 08:01:15 +00:00
pc
Return proxied object in OnTransceiver
2021-09-16 09:40:52 +00:00
resources
Disable high-pass filtering of the AEC reference
2021-02-23 07:06:11 +00:00
rtc_base
Fix integer underflow in BitstreamReader::ConsumeBits
2021-09-20 19:37:49 +00:00
rtc_tools
Improve points visualization in metrics_plotter.
2021-09-10 10:59:58 +00:00
sdk
Verify that locks are handled correctly on RTCAudioSession
2021-09-17 01:41:42 +00:00
stats
Reland "Wire up non-sender RTT for audio, and implement related standardized stats."
2021-09-06 14:26:55 +00:00
system_wrappers
Use GTEST_SKIP() instead of early return.
2021-08-12 15:24:13 +00:00
test
Fix integer underflow in BitstreamReader::ConsumeBits
2021-09-20 19:37:49 +00:00
tools_webrtc
rename use_x11 to ozone_platform_x11
2021-09-20 15:13:55 +00:00
video
New H264PacketBuffer consolidating a bunch of H264 specific hacks into one class.
2021-09-15 09:57:29 +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
Increase iOS deployment target from 10 to 12.
2021-07-02 17:02:27 +00:00
.vpython
Update links to point at main branch
2021-07-22 16:41:26 +00:00
.vpython3
Add .vpython3 file to webrtc
2021-09-15 16:56:30 +00:00
AUTHORS
rename use_x11 to ozone_platform_x11
2021-09-20 15:13:55 +00:00
BUILD.gn
Allow export of Obj-C symbols without C++ ones.
2021-07-30 22:54:59 +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 c4e2cd0641..3416746b88 (923181:923281)
2021-09-21 08:52:19 +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 kwiberg@webrtc.org from OWNERS files
2020-12-04 15:11:26 +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
Add .vpython3 file to webrtc
2021-09-15 16:56:30 +00:00
PATENTS
…
presubmit_test_mocks.py
Add presubmit check to guard against assert() usage.
2021-07-22 17:08:26 +00:00
presubmit_test.py
Add presubmit check to guard against assert() usage.
2021-07-22 17:08:26 +00:00
PRESUBMIT.py
fix some typos
2021-08-12 18:37:10 +00:00
pylintrc
Undo enforcing of PEP-8 pylint changes for method and function names.
2020-11-10 18:26:25 +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
rename use_x11 to ozone_platform_x11
2021-09-20 15:13:55 +00:00
whitespace.txt
Trigger bots
2021-09-08 18:30:03 +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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Python 1.3%
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