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chromium-webrtc-autoroll 572c6efd23 Roll chromium_revision 6dae2061c5..996bf8ce67 (1187756:1187868) 
Change log: 6dae2061c5..996bf8ce67
Full diff: 6dae2061c5..996bf8ce67

Changed dependencies
* src/base: 63abbb39b0..2b087a18c3
* src/build: e04720d46f..83d97d340b
* src/ios: 13366e4c8b..4abe59906e
* src/testing: af59c8f3d6..451fc92885
* src/third_party: 05398d5a46..229a946cd3
* src/third_party/androidx: 65wGBQFZsIrAsUTcGfaFnGVkiya8p7FU_Y7oHI2XobkC..RHDvRgz7XWDd6x69fnjjf4shtOVB_SSNJ9qh3AbgOhEC
* src/third_party/perfetto: 81fa260ab3..c9dd257f3c
* src/third_party/r8: 5fmK4ltjLHNRPbUQ34VE7o4hXrmeRbwpUTtUPadhj0wC..yG6aCoW-nKG7aqWMZkE1TsvMCxvP7KxJ9PsaRbytuyEC
* src/tools: 4f6049d87e..14dc972c3b
DEPS diff: 6dae2061c5..996bf8ce67/DEPS

No update to Clang.

BUG=None

Change-Id: I37d1c1f8b9fe1fef3185d340802780fdd8b3e637
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317509
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@{#40619}
2023-08-24 16:59:45 +00:00
api
Make PendingTaskSafetyFlag compatible with component builds
2023-08-24 14:24:49 +00:00
audio
Support receiving cloned encoded audio frames
2023-08-21 09:43:52 +00:00
build_overrides
Roll chromium_revision 6b95b8aa08..80c35b98ae (1175249:1180373)
2023-08-10 07:50:10 +00:00
call
Update WebRTC code version (2023-08-23T04:04:00).
2023-08-23 05:20:09 +00:00
common_audio
[ssci] Added Shipped field to READMEs
2023-07-12 07:31:06 +00:00
common_video
Fix fuzzing test issues reported by Chromium fuzzing test
2023-08-22 13:08:39 +00:00
data
…
docs
Update bug reporting and contributing docs
2023-07-24 13:04:29 +00:00
examples
Rland "Revert "Reland "Reland "Delete old Android ADM.""""
2023-06-30 13:10:12 +00:00
experiments
Add tool for generating field trial registry header
2022-10-18 07:25:43 +00:00
g3doc
Add docs about adding a new test binary.
2023-03-07 11:12:33 +00:00
infra
Drop email recipients from lkgr finder.
2023-08-15 08:39:43 +00:00
logging
Process events with the same timestamp in a defined order.
2023-06-20 14:21:17 +00:00
media
Implement codec selection api
2023-08-24 13:18:04 +00:00
modules
In VideoPlayoutDelay delete access to integer representation of min/max values
2023-08-23 16:14:26 +00:00
net/dcsctp
Format ^(api|call|common_audio|examples|media|net|p2p|pc)/
2023-05-03 11:09:26 +00:00
p2p
Use the AsyncDnsResolver in PeerConnection defaults
2023-08-23 20:29:55 +00:00
pc
Implement codec selection api
2023-08-24 13:18:04 +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
Fix TSAN conflict in AsyncDnsResolver
2023-08-24 11:36:02 +00:00
rtc_tools
Remove internal overrides using old SendRtp and SendRtcp interfaces.
2023-08-15 13:20:21 +00:00
sdk
Adopt EglThread in EglRenderer
2023-08-18 21:05:57 +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
Add rtc_common_public_deps
2023-08-22 11:32:06 +00:00
tools_webrtc
Added HEVC parser for WebRTC
2023-08-18 17:48:11 +00:00
video
In VideoPlayoutDelay delete access to integer representation of min/max values
2023-08-23 16:14:26 +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
Roll chromium_revision 6b95b8aa08..80c35b98ae (1175249:1180373)
2023-08-10 07:50:10 +00:00
.mailmap
Add .mailmap for git.
2022-02-20 14:22:13 +00:00
.style.yapf
…
.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
Exposed Adapter types for Android
2023-08-11 08:29:14 +00:00
BUILD.gn
Add an async DNS resolver suitable for defaulting.
2023-08-23 10:08:52 +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 6dae2061c5..996bf8ce67 (1187756:1187868)
2023-08-24 16:59:45 +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
Add infra owners file
2022-12-02 09:21:47 +00:00
PATENTS
…
presubmit_test_mocks.py
…
presubmit_test.py
…
PRESUBMIT.py
Remove low_bandwidth_audio_test.
2023-06-01 07:20:38 +00:00
pylintrc
…
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
Languages
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