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chromium-webrtc-autoroll a551bd1dc5 Roll chromium_revision b32d012bf3..1d6fad0b34 (923877:923999) 
Change log: b32d012bf3..1d6fad0b34
Full diff: b32d012bf3..1d6fad0b34

Changed dependencies
* src/base: c8e4ec9844..c6d7f9b58b
* src/ios: d109952515..a58dcba71e
* src/testing: eaf1a7c60c..91cb32e466
* src/third_party: 81ff5d2e24..116e04602c
* src/third_party/androidx: zzVIaEA3uTWNbAiFgL34RFj52xan_NtxkXTFIZjsrroC..esnQTh6IB0KM-t2F2AbAmiTNYdxM_WOyT5lLzYKt188C
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/25814bb53d..c62bb0f159
* src/third_party/depot_tools: eff810e93d..b504ca40f3
* src/third_party/perfetto: 8f0fabf54b..09e840c057
* src/third_party/turbine: n2quuVAecebwosy5EPRjDGSO5EQU94aFRBg_EdrYtpsC..EIJ2nS3aoQlV-cKpmxr4hNR4E1Qs82brcidjG4A1I5QC
* src/tools: 111c4792db..385e7fde51
DEPS diff: b32d012bf3..1d6fad0b34/DEPS

No update to Clang.

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

Change-Id: I5484057140f83698610e9df8e1340246b4baca2e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232723
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@{#35070}
2021-09-22 20:32:06 +00:00
api
Mark toI420 as Nullable
2021-09-21 10:05:09 +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 (2021-09-22T04:04:23).
2021-09-22 05:58:49 +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
sctp: dcsctp: Manage lifecycle explicitly
2021-09-22 07:17:29 +00:00
modules
rtp_rtcp: use webrtc::flat_map for remote_senders
2021-09-21 13:20:16 +00:00
net/dcsctp
dcsctp: Allow specifying minimum RTT variance
2021-09-22 20:22:54 +00:00
p2p
Delete BasicPacketSocketFactory constructor with thread argument
2021-09-22 12:15:06 +00:00
pc
Delete BasicPacketSocketFactory constructor with thread argument
2021-09-22 12:15:06 +00:00
resources
Disable high-pass filtering of the AEC reference
2021-02-23 07:06:11 +00:00
rtc_base
Delete BitBuffer
2021-09-21 16:28:38 +00:00
rtc_tools
Delete BasicPacketSocketFactory constructor with thread argument
2021-09-22 12:15:06 +00:00
sdk
[ios] Fix two rtc_unittests that fail when using lld as linker
2021-09-22 19:45:44 +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
In vp9 encoder fuzzer exclude testing unsupported bitrate configurations
2021-09-22 16:20:56 +00:00
tools_webrtc
Revert "Fix iOS sim bot to for the new Chromium roll"
2021-09-22 20:31:04 +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 b32d012bf3..1d6fad0b34 (923877:923999)
2021-09-22 20:32:06 +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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Java 2.9%
C 2.2%
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Python 1.3%
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