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chromium-webrtc-autoroll dc8f81652b Roll chromium_revision 12a1bdba49..05f3934693 (1208649:1208764) 
Change log: 12a1bdba49..05f3934693
Full diff: 12a1bdba49..05f3934693

Changed dependencies
* src/base: 27ce87dd23..3841cd2118
* src/ios: a3b4f5efda..274966acfb
* src/testing: 569bfa6df5..25d5ed3d10
* src/third_party: 1ed2e049ff..c8b96fdfa6
* src/third_party/androidx: n-49vMsfeUdbL-4uiGFCKiXPdIOzcg6-aFmLL64c2foC..rmHTIvq0ZcpO2SLGZ62OzCmZLi_-iLI0USquMgUITeoC
* src/third_party/depot_tools: 69e597f8e5..0b943400a4
* src/third_party/perfetto: fdbd437367..747df2371b
* src/tools: d7757d197a..4ad32b9ce9
DEPS diff: 12a1bdba49..05f3934693/DEPS

No update to Clang.

BUG=None

Change-Id: I6a35da83dac48c0a17f177c1ff8604befc5282d0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323046
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@{#40918}
2023-10-12 12:39:13 +00:00
api
Added AsString() function for color space for easier debugging
2023-10-10 09:08:28 +00:00
audio
Move field trial check WebRTC-DisableRtxRateLimiter
2023-10-11 10:34:17 +00:00
build_overrides
Roll chromium_revision 1a86984556..de49bd6f22 (1204416:1205695)
2023-10-05 10:33:16 +00:00
call
Move field trial check WebRTC-DisableRtxRateLimiter
2023-10-11 10:34:17 +00:00
common_audio
Avoid integer overflow detected by the compiler.
2023-08-30 08:24:30 +00:00
common_video
Fix use-of-uninitialized-value and integer-overflow issues reported by chromium fuzz testing
2023-10-03 14:17:06 +00:00
data
…
docs
Add note about two-byte extension to VLA docs
2023-10-11 11:20:19 +00:00
examples
Prepare to deprecate AsyncResolver class
2023-10-10 12:16:33 +00:00
experiments
Replace WebRTC-QuickPerfTest field trial with similar named dedicated test flag
2023-10-10 12:20:33 +00:00
g3doc
Relax the field trial policy to not require an open bug
2023-09-29 08:03:21 +00:00
infra
Replace WebRTC-QuickPerfTest field trial with similar named dedicated test flag
2023-10-10 12:20:33 +00:00
logging
Add codec name H265 to support H265 in WebRTC
2023-09-20 09:25:32 +00:00
media
Allow streams to be sent without |source_| being initially set
2023-10-12 10:08:26 +00:00
modules
Upper limit pacer send bursts to about 63Kbyte
2023-10-11 14:22:44 +00:00
net/dcsctp
dcsctp: Reset synchronously with incoming request
2023-10-09 09:47:57 +00:00
p2p
Add RTC_DCHECK that port_ is always valid in Connection.
2023-10-11 10:16:15 +00:00
pc
Generalize ssrc-group check to apply to groups other than SIM
2023-10-09 05:59:48 +00:00
resources
Clarify and extend test support for certain sample rates in audio processing
2022-08-03 14:26:36 +00:00
rtc_base
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 19/inf
2023-10-11 11:09:35 +00:00
rtc_tools
Change the type of PacedPacketInfo::send_bitrate_bps from int to strongly-typed DataRate.
2023-10-04 14:20:22 +00:00
sdk
Avoid excessive eglMakeCurrent calls
2023-10-06 14:20:28 +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
Update test_flags visibility.
2023-10-12 09:59:33 +00:00
tools_webrtc
Set optimize_for_size for iOS perf tests.
2023-10-12 08:00:24 +00:00
video
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 19/inf
2023-10-11 11:09:35 +00:00
.clang-format
Add IncludeBlocks to clang-format.
2021-02-03 16:29:07 +00:00
.git-blame-ignore-revs
Add formatting CLs to .git-blame-ignore-revs
2023-05-07 09:27:47 +00:00
.gitignore
Add .cache to .gitignore.
2021-01-20 15:01:07 +00:00
.gn
[Fuchsia] Remove fuchsia_target_api_level from .gn
2023-09-04 07:26:36 +00:00
.mailmap
Add .mailmap for git.
2022-02-20 14:22:13 +00:00
.style.yapf
Configure YAPF to follow PEP-8 altogether
2023-09-22 10:32:11 +00:00
.vpython
Remove unused script webrtc_dashboard_upload.py
2022-03-21 12:54:42 +00:00
.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
Make WEBRTC_UNSAFE_FUZZER_MODE dependent only on use_fuzzing_engine
2023-09-20 16:04:17 +00:00
CODE_OF_CONDUCT.md
Reland "Migrate WebRTC documentation to new renderer"
2023-01-31 09:30:04 +00:00
codereview.settings
Don't add webrtc-reviews@ to CC, it can be added globally on Gerrit
2018-10-25 08:19:53 +00:00
DEPS
Roll chromium_revision 12a1bdba49..05f3934693 (1208649:1208764)
2023-10-12 12:39:13 +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
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
Add infra owners file
2022-12-02 09:21:47 +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
Remove low_bandwidth_audio_test.
2023-06-01 07:20:38 +00:00
pylintrc
Configure Pylint to follow PEP-8
2023-09-25 15:56:09 +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
Remove xooglers from WATCHLISTS and OWNERS
2022-11-30 15:33:25 +00:00
webrtc_lib_link_test.cc
Replace BuiltinVideo{Encoder,Decoder}Factory with Video{Encoder,Decoder}FactoryTemplate.
2023-03-31 19:31:50 +00:00
webrtc.gni
Add rtc_common_public_deps
2023-08-22 11:32:06 +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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