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chromium-webrtc-autoroll cad0bfa566 Roll chromium_revision 7be4bb3958..b294093253 (1203597:1203697) 
Change log: 7be4bb3958..b294093253
Full diff: 7be4bb3958..b294093253

Changed dependencies
* fuchsia_version: version:15.20230920.1.1..version:15.20230930.1.1
* src/base: db158725fa..4bd1370aa7
* src/buildtools: ece39eb658..50c348906c
* src/ios: cb8060928e..8326446e62
* src/testing: 9f6812c5d9..cf08af2a02
* src/third_party: 0fa2ea7cfa..f27b31bc35
* src/third_party/androidx: Q4oJmxPgl3vOnzMc9hPT4z8W4TDmrAHU8KW8-tYFB-YC..l1aQ3x4WHPee1WDaKNyiN3zopCdZq7O3SJL6AQdgleAC
* src/third_party/libc++/src: 8a241ea043..7cf98622ab
* src/tools: 2a019f0917..76baa86950
* src/tools/luci-go: git_revision:3e49653ac027a003d9ef6cf89068acbf28f5bd9e..git_revision:589d8654cfa7808816a6ecb4284ed2fd72c2f6d5
* src/tools/luci-go: git_revision:3e49653ac027a003d9ef6cf89068acbf28f5bd9e..git_revision:589d8654cfa7808816a6ecb4284ed2fd72c2f6d5
DEPS diff: 7be4bb3958..b294093253/DEPS

No update to Clang.

BUG=None

Change-Id: I0eac987bf9e5615551f695408c4b25779aef1e44
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322062
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@{#40849}
2023-09-30 18:24:26 +00:00
api
Revert "Enable SRTP GCM ciphers by default"
2023-09-28 14:42:18 +00:00
audio
Pass the correct abs_capture_timestamp while cloning audio frame
2023-09-26 15:57:52 +00:00
build_overrides
Roll chromium_revision 6b95b8aa08..80c35b98ae (1175249:1180373)
2023-08-10 07:50:10 +00:00
call
Update WebRTC code version (2023-09-30T04:13:02).
2023-09-30 05:51:28 +00:00
common_audio
Avoid integer overflow detected by the compiler.
2023-08-30 08:24:30 +00:00
common_video
Fix fuzzing issue reported by Chromium fuzzing test
2023-09-05 09:29:27 +00:00
data
…
docs
Update bug reporting and contributing docs
2023-07-24 13:04:29 +00:00
examples
Roll chromium_revision 6ac7929166..eef62e8a0c (1190797:1197906)
2023-09-19 12:06:33 +00:00
experiments
Set permissions on experiments/field_trials.py
2023-09-29 08:17:15 +00:00
g3doc
Relax the field trial policy to not require an open bug
2023-09-29 08:03:21 +00:00
infra
Revert "Remove device_status dimensions for perf bot"
2023-09-12 07:07:46 +00:00
logging
Add codec name H265 to support H265 in WebRTC
2023-09-20 09:25:32 +00:00
media
Reduce log verbosity in codec selection implementation
2023-09-28 07:57:06 +00:00
modules
Clear PacketBuffer on large negative jumps at the start of the video stream
2023-09-29 08:56:15 +00:00
net/dcsctp
dcsctp: Don't generate FORWARD-TSN across stream resets
2023-09-28 11:31:02 +00:00
p2p
Change PortInterface::Type to string_view and make type_ member const
2023-09-25 13:48:50 +00:00
pc
Remove more Codec-related templating in MediaSession
2023-09-29 10:52:27 +00:00
resources
…
rtc_base
Add field trial for enabling SSL client hello extension permutation
2023-09-28 13:34:38 +00:00
rtc_tools
Rename EncodedImage property Timetamp to RtpTimestamp
2023-09-24 20:06:48 +00:00
sdk
Adopt EglThread in EglRenderer once again.
2023-09-29 13:37:21 +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
Deprecate old names for EncodedImage::RtpTimestamp accessor and setter
2023-09-28 07:29:22 +00:00
tools_webrtc
Make CreateOffer/CreateAnswer return RTCErrorOr<SessionDescription>
2023-09-26 09:41:30 +00:00
video
Rename EncodedImage property Timetamp to RtpTimestamp
2023-09-24 20:06:48 +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
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
…
DEPS
Roll chromium_revision 7be4bb3958..b294093253 (1203597:1203697)
2023-09-30 18:24:26 +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
…
OWNERS_INFRA
…
PATENTS
…
presubmit_test_mocks.py
…
presubmit_test.py
…
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
…
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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