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chromium-webrtc-autoroll b6d69708fe Roll chromium_revision 231d950fb5..3b13b74f8b (1225584:1225957) 
Change log: 231d950fb5..3b13b74f8b
Full diff: 231d950fb5..3b13b74f8b

Changed dependencies
* src/base: 2819ffc7eb..f4de6e3114
* src/build: 2e760d5c6c..23f94fa6ef
* src/ios: e76a595293..ee2c68c354
* src/testing: 7c59019a3b..9740b867f3
* src/third_party: 90035c7057..c34d2a98ad
* src/third_party/android_build_tools/manifest_merger: kyXImK5GOP19ROuy-2wf_I_zzcVr2RVsPvB0jhLyrCEC..SdNR04V227YL22FMmKoS4AdLYwv6MJe8HBAZKNhXoCsC
* src/third_party/androidx: nQrPFhDB1aX4Jlbq9boM5vvI9bsiAa736ZacDoJQp8oC..lVVaxxGWEoKfi-XMLQyERTC8Smt04PrfwkbezmNSAVoC
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/a96e7377fc..1b7fdbd910
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/b46041dad0..bb1266a081
* src/third_party/depot_tools: df3e577855..06c4261af7
* src/third_party/ffmpeg: e1ca3f06ad..c1d0dc4135
* src/third_party/freetype/src: 050a320764..e969289f88
* src/third_party/libc++abi/src: 0226cb1cdf..d529bb588f
* src/third_party/libunwind/src: 7b9fd7a061..19b1234bfb
* src/third_party/perfetto: 8e4410079d..6ba75cc5d9
* src/third_party/re2/src: 974f44c8d4..7e0c1a9e24
* src/tools: 070647af7d..5254b703bd
DEPS diff: 231d950fb5..3b13b74f8b/DEPS

No update to Clang.

BUG=None

Change-Id: I40f92643adfa88ef49ae168d5cd4bfbf507f1b3a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327625
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@{#41178}
2023-11-17 09:42:32 +00:00
api
Revert "Reland: Remove unsupported configuration value, allow_codec_switching"
2023-11-15 08:10:28 +00:00
audio
Expose audio mimeType for insertable streams
2023-11-03 09:49:12 +00:00
build_overrides
Roll chromium_revision 01dc2965ca..917876224a (1209117:1211391)
2023-10-18 15:15:07 +00:00
call
Update WebRTC code version (2023-11-17T04:02:07).
2023-11-17 05:40:22 +00:00
common_audio
Fix pointer overflow in neon implemenation of audio filters
2023-10-13 06:41:08 +00:00
common_video
Add rtp packetizer for H265
2023-11-08 15:49:37 +00:00
data
…
docs
Add note about two-byte extension to VLA docs
2023-10-11 11:20:19 +00:00
examples
Migrage objc sdk and examples to EnableMedia api
2023-11-02 13:55:02 +00:00
experiments
Remove SSRCs from libSRTP when removing them from the rtp_demuxer
2023-11-08 10:24:10 +00:00
g3doc
Move StrJoin to rtc_base/strings
2023-11-15 12:10:28 +00:00
infra
Remove deprecated fields for luci-analysis
2023-10-25 07:18:24 +00:00
logging
Try removing RTC_PUSH_IGNORING_WUNDEF() around proto includes
2023-11-01 08:21:05 +00:00
media
Make Codec::Matches also consider packetization
2023-11-14 08:14:14 +00:00
modules
Reverse the kbits logic according to RFC
2023-11-16 18:33:13 +00:00
net/dcsctp
Move StrJoin to rtc_base/strings
2023-11-15 12:10:28 +00:00
p2p
Allow list the places which send STUN_REQUEST w/o password
2023-11-15 11:31:45 +00:00
pc
Revert "Reland: Remove unsupported configuration value, allow_codec_switching"
2023-11-15 08:10:28 +00:00
resources
Ignore .binarypb files.
2023-10-30 14:56:36 +00:00
rtc_base
Use uint8_t as inner type for rtc::ByteBufferReader
2023-11-17 06:27:12 +00:00
rtc_tools
Move some users to use webrtc::RefCountInterface
2023-11-02 14:45:57 +00:00
sdk
Revert "Reland: Remove unsupported configuration value, allow_codec_switching"
2023-11-15 08:10:28 +00:00
stats
stats: implement fecSsrc on inbound-rtp
2023-08-04 12:54:48 +00:00
system_wrappers
Use //third_party/cpu_features directly
2023-06-02 07:17:36 +00:00
test
Move codecs handling from test to tester
2023-11-13 16:48:49 +00:00
tools_webrtc
Remove not-needed webrtc:: prefixes in pc/
2023-11-13 13:23:04 +00:00
video
Revert "Measure wall clock time of capture and encode processing."
2023-11-11 18:03:40 +00:00
.clang-format
…
.git-blame-ignore-revs
Add formatting CLs to .git-blame-ignore-revs
2023-05-07 09:27:47 +00:00
.gitignore
…
.gn
[Fuchsia] Remove fuchsia_target_api_level from .gn
2023-09-04 07:26:36 +00:00
.mailmap
…
.style.yapf
Configure YAPF to follow PEP-8 altogether
2023-09-22 10:32:11 +00:00
.vpython
…
.vpython3
Update vpython3 requests
2023-06-02 07:49:24 +00:00
AUTHORS
fix: Handle out-of-range device index after GetDevicesInfo
2023-09-19 12:13:39 +00:00
BUILD.gn
AsyncPacketSocket::RegisterReceivedPacketCallback
2023-11-15 11:17:53 +00:00
CODE_OF_CONDUCT.md
Reland "Migrate WebRTC documentation to new renderer"
2023-01-31 09:30:04 +00:00
codereview.settings
…
DEPS
Roll chromium_revision 231d950fb5..3b13b74f8b (1225584:1225957)
2023-11-17 09:42:32 +00:00
DIR_METADATA
…
ENG_REVIEW_OWNERS
…
LICENSE
…
license_template.txt
…
native-api.md
Reland "Migrate WebRTC documentation to new renderer"
2023-01-31 09:30:04 +00:00
OWNERS
Add infra owners file
2022-12-02 09:21:47 +00:00
OWNERS_INFRA
Allow to keep old python style for existing files.
2023-10-17 13:52:56 +00:00
PATENTS
…
presubmit_test_mocks.py
…
presubmit_test.py
…
PRESUBMIT.py
Add google-truth to WEBRTC_ONLY_DEPS to unblock Chromium roll.
2023-10-17 16:40:08 +00:00
pylintrc
Configure Pylint to follow PEP-8
2023-09-25 15:56:09 +00:00
pylintrc_old_style
Allow to keep old python style for existing files.
2023-10-17 13:52:56 +00:00
README.chromium
[ssci] Added Shipped field to READMEs
2023-07-12 07:31:06 +00:00
README.md
doc: Follow up link rename in I2dbe1ef0c74a0de8c5619b522fab39527e797d9c
2023-05-26 09:20:16 +00:00
WATCHLISTS
…
webrtc_lib_link_test.cc
Migrate webrtc_link_test to include EnableMedia api
2023-11-02 22:45:40 +00:00
webrtc.gni
Make AEC3 json parsing code testonly
2023-10-26 12:03:02 +00:00
whitespace.txt
Trigger bots
2023-05-16 08:24:54 +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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