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chromium-webrtc-autoroll 35962284b3 Roll chromium_revision eb21f8d0bf..dd58dcde17 (1083442:1083550) 
Change log: eb21f8d0bf..dd58dcde17
Full diff: eb21f8d0bf..dd58dcde17

Changed dependencies
* src/base: c08c8700b2..4e0905ca44
* src/build: b0e86d8d38..c91a4dbdb6
* src/testing: c3bc37b0b9..739e075aeb
* src/third_party: 0dfb3f668c..0d45c1f9b4
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/e28e9f182e..0d416fe68a
* src/third_party/depot_tools: 18359bb9a1..9ace9b4dc4
* src/third_party/perfetto: aa7e493dcf..4ffacbee2e
* src/tools: 9cbfb8bb67..0d9b91db1d
DEPS diff: eb21f8d0bf..dd58dcde17/DEPS

No update to Clang.

BUG=None

Change-Id: Iacf7aa2629ecee750ac65248426158ad23d2235f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288141
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@{#38894}
2022-12-15 04:39:51 +00:00
api
Make requested_resolution_alignment of webrtc::EncoderInfo as uint32_t.
2022-12-12 06:02:59 +00:00
audio
Remove pending packets from the pacer when an RTP module is removed.
2022-12-13 11:32:58 +00:00
build_overrides
Use default values provided by PartitionAlloc instead of hard-coded ones
2022-12-07 09:11:35 +00:00
call
Update WebRTC code version (2022-12-14T04:03:07).
2022-12-14 05:51:33 +00:00
common_audio
Make header files self contained.
2022-10-08 08:38:36 +00:00
common_video
Make header files self contained.
2022-10-08 08:38:36 +00:00
data
…
docs
Update webrtc build command instruction to autoninja.
2022-11-25 09:19:37 +00:00
examples
pc: Add asynchronous RtpSender::SetParameters() call
2022-11-15 15:31:40 +00:00
experiments
Add tool for generating field trial registry header
2022-10-18 07:25:43 +00:00
g3doc
Explicitly forbid dynamic_cast
2022-11-04 12:37:57 +00:00
infra
Let iOS bots run unittests with XCTest.
2022-12-09 10:44:23 +00:00
logging
Add missing dependencies.
2022-10-10 15:51:33 +00:00
media
Split audio and video channels into Send and Receive APIs.
2022-12-14 11:00:17 +00:00
modules
Unify AGC2 experiment field trials into one
2022-12-14 16:40:41 +00:00
net/dcsctp
Add missing dependencies.
2022-10-10 10:18:37 +00:00
p2p
Improve logging of DTLS transport handshake
2022-12-14 07:41:36 +00:00
pc
Split audio and video channels into Send and Receive APIs.
2022-12-14 11:00:17 +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
Make requested_resolution_alignment of webrtc::EncoderInfo as uint32_t.
2022-12-12 06:02:59 +00:00
rtc_tools
Delete RGB to I420 and YUV to IVF converters
2022-12-02 08:21:21 +00:00
sdk
AndroidNetworkMonitor - loosen assumptions even more
2022-12-05 10:01:01 +00:00
stats
Add scalability mode to RTCOutboundRtpStreamStats stats
2022-12-08 11:46:06 +00:00
system_wrappers
Rename test helper for registering field trial keys
2022-12-02 13:21:28 +00:00
test
Remove webrtc::webrtc_pc_e2e::GetCurrentTestName().
2022-12-11 09:36:52 +00:00
tools_webrtc
Only instrument changed lines on coverage bot
2022-12-12 10:13:49 +00:00
video
Call OnRenderedFrame after frame is passed to renderer
2022-12-13 18:18:44 +00:00
.clang-format
Add IncludeBlocks to clang-format.
2021-02-03 16:29:07 +00:00
.git-blame-ignore-revs
…
.gitignore
Add .cache to .gitignore.
2021-01-20 15:01:07 +00:00
.gn
Set Fuchsia Api level + update SDK version
2022-09-14 08:49:56 +00:00
.mailmap
Add .mailmap for git.
2022-02-20 14:22:13 +00:00
.style.yapf
Fix mb.py presubmit issues.
2021-12-08 08:53:00 +00:00
.vpython
Remove unused script webrtc_dashboard_upload.py
2022-03-21 12:54:42 +00:00
.vpython3
Add python grpc to .vpython3 for ios test runner
2022-09-16 12:26:48 +00:00
AUTHORS
Fix bugs in discarding stun address
2022-11-30 13:51:41 +00:00
BUILD.gn
Add fuchsia_perf_test to the default build group.
2022-12-05 10:09:00 +00:00
CODE_OF_CONDUCT.md
Reference "main" branches instead of "master" branches.
2021-07-15 11:07:44 +00:00
codereview.settings
…
DEPS
Roll chromium_revision eb21f8d0bf..dd58dcde17 (1083442:1083550)
2022-12-15 04:39:51 +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
g3doc.lua
Improve webrtc documentation infra. Preview at:
2021-03-30 10:29:30 +00:00
LICENSE
…
license_template.txt
…
native-api.md
Reference "main" branches instead of "master" branches.
2021-07-15 11:07:44 +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
Update portaudio to the latest
2022-05-13 09:01:34 +00:00
pylintrc
tools_webrtc dir converted to py3 + top level PRESUBMIT script
2022-02-08 14:42:26 +00:00
README.chromium
Add CPEPrefix.
2020-07-13 11:42:07 +00:00
README.md
doc: add g3doc sitemap to toplevel readme
2021-07-23 07:55:17 +00:00
WATCHLISTS
Remove xooglers from WATCHLISTS and OWNERS
2022-11-30 15:33:25 +00:00
webrtc_lib_link_test.cc
Deprecate PeerConnectionFactory::CreatePeerConnection
2021-05-10 08:47:48 +00:00
webrtc.gni
Verify field trials looked up through field_trial::FindFullName
2022-10-20 10:46:01 +00:00
whitespace.txt
Trigger bots
2022-11-17 21:29:53 +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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