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chromium-webrtc-autoroll 533a97be2b Roll chromium_revision eb64addd52..2ce2f71c6c (1162930:1163494) 
Change log: eb64addd52..2ce2f71c6c
Full diff: eb64addd52..2ce2f71c6c

Changed dependencies
* fuchsia_version: version:13.20230626.3.1..version:13.20230628.0.1
* src/base: 781d6449f3..7af69c7d44
* src/build: d92611cbf8..8fefe21204
* src/buildtools: f089c59d7c..963bc09d28
* src/ios: 20f5c26a1e..8e8e8e8049
* src/testing: 24176a9633..11a9c1ad97
* src/third_party: 5f8845d6f1..bcdde60719
* src/third_party/android_build_tools/manifest_merger: pRHDE8UAgipcDQINCUsRz94lgm5BHQjiL-BLF6d6xC4C..MN3CF2GQ8xeB6obj4qf5J6l15-NoA43u4__RQTTe8I4C
* src/third_party/androidx: v8JL3E5KtNlFmho-mGQhRh5dg4wY33bqvymlJ0MKCdoC..BNS2DL_t6VtNjAQ9hEOz_rqTe-n-hN3VEmmXgu-c7jUC
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/1a0894f20f..e3d8143008
* src/third_party/depot_tools: f0fba1d307..70a4a17f44
* src/third_party/perfetto: 92e38d71f0..e20af3da17
* src/tools: 2718b35c0d..84a6c4b504
* src/tools/luci-go: git_revision:39f255d5875293d3e1d978888b819ac124a8b0cc..git_revision:58e1fcab6ced4d330cfd46287e00aa14fbd46dc6
* src/tools/luci-go: git_revision:39f255d5875293d3e1d978888b819ac124a8b0cc..git_revision:58e1fcab6ced4d330cfd46287e00aa14fbd46dc6
DEPS diff: eb64addd52..2ce2f71c6c/DEPS

No update to Clang.

BUG=None

Change-Id: I93164d04586dd21c8be2febf2d5833415c67a6c2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/310761
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@{#40367}
2023-06-28 12:46:14 +00:00
api
Deprecate encoded audio frame GetHeader
2023-06-27 13:13:34 +00:00
audio
Connected jitter_buffer_min_delay_ms to DelayManager's min_delay_ms by setting the neteq_config
2023-06-26 13:36:39 +00:00
build_overrides
Define enable_safe_libcxx in build_overrides/build.gni.
2023-05-03 08:18:25 +00:00
call
Update WebRTC code version (2023-06-28T04:03:42).
2023-06-28 05:48:34 +00:00
common_audio
Format ^(api|call|common_audio|examples|media|net|p2p|pc)/
2023-05-03 11:09:26 +00:00
common_video
webrtc_libyuv: Add support for more video types for consistency
2023-04-24 19:06:25 +00:00
data
…
docs
docs: explain release note process
2023-05-24 14:09:54 +00:00
examples
Format ^(api|call|common_audio|examples|media|net|p2p|pc)/
2023-05-03 11:09:26 +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
[Analysis] Remove old threshold fields
2023-06-08 00:31:28 +00:00
logging
Process events with the same timestamp in a defined order.
2023-06-20 14:21:17 +00:00
media
Split fake media channel classes
2023-06-28 11:40:47 +00:00
modules
rtp_rtcp/source: fix some minor typos
2023-06-27 21:32:46 +00:00
net/dcsctp
Format ^(api|call|common_audio|examples|media|net|p2p|pc)/
2023-05-03 11:09:26 +00:00
p2p
ICE: adjust priority of non-relay candidates
2023-06-20 08:02:27 +00:00
pc
Split fake media channel classes
2023-06-28 11:40:47 +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 TimeUTCMicros resolution on Windows
2023-06-27 14:15:09 +00:00
rtc_tools
[rtc_tools/video_encoder] Output ivf for all SVC decode targets
2023-06-27 23:57:08 +00:00
sdk
Fix candidate leak with initWithNativeCandidate.
2023-06-20 10:31:44 +00:00
stats
stats: implement flexfec fecBytesReceived stats for FlexFEC
2023-06-21 13:04:31 +00:00
system_wrappers
Use //third_party/cpu_features directly
2023-06-02 07:17:36 +00:00
test
Add a method to log AnalyzingVideoSink metrics.
2023-06-27 08:39:53 +00:00
tools_webrtc
Roll chromium_revision a3756bb36c..dd02f6b781 (1154916:1156166)
2023-06-12 14:51:51 +00:00
video
Pass and process capture time through SendPacketObserver with Timestamp type
2023-06-22 17:16:41 +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
Temporary set mac_deployment_target to 10.13
2023-06-26 07:57:18 +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
fix: do not use dispatch_queue_global_t directly
2023-06-20 10:23:18 +00:00
BUILD.gn
Add video_encoder to default build
2023-06-27 10:18:33 +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 eb64addd52..2ce2f71c6c (1162930:1163494)
2023-06-28 12:46:14 +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
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
tools_webrtc dir converted to py3 + top level PRESUBMIT script
2022-02-08 14:42:26 +00:00
README.chromium
…
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
Replace BuiltinVideo{Encoder,Decoder}Factory with Video{Encoder,Decoder}FactoryTemplate.
2023-03-31 19:31:50 +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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