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chromium-webrtc-autoroll edca73c884 Roll chromium_revision 11e681944b..e2601310e1 (1114399:1114554) 
Change log: 11e681944b..e2601310e1
Full diff: 11e681944b..e2601310e1

Changed dependencies
* fuchsia_vesion: version:11.20230307.3.1..version:12.20230308.1.1
* src/base: 6fb7e1ff0b..a0763fbd7f
* src/build: 984f82d89c..512b48958e
* src/ios: 128eeaf931..87c164ae2e
* src/testing: 37423855c5..d3de492699
* src/third_party: dfed239d19..18bae88bf7
* src/third_party/androidx: ZAoXSZsVgU3BsvsDCSCS47XxbfOLsIFrBr22Wo5yevsC..sTQ9mgPBKdbKHMEO0qOnTBp-ntlRVNVkVUd-IvXmK9sC
* src/third_party/freetype/src: d5c6b948ac..bd6208b712
* src/third_party/kotlin_stdlib: Mg7371mEUwDQH4_z29HdWqYWVlXN6t2dXX0kIutg_SwC..zeFlVAEGvnpaj3JJujWHzRlUiBEm4XeeaMQzVsdW6D4C
* src/third_party/perfetto: cb0e1eabbb..c43a9bc21a
* src/tools: b12d6e2b16..b9fe6bf682
DEPS diff: 11e681944b..e2601310e1/DEPS

No update to Clang.

BUG=None

Change-Id: Ida78bcea8459f0bc094f27934fc64f935d733b32
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296681
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@{#39506}
2023-03-08 16:56:50 +00:00
api
stats: rename RTCInboundRTPStreamStats and RTCOutboundRTPStreamStats
2023-03-07 14:27:47 +00:00
audio
Add a clone method to the audio frame transformer API.
2023-03-06 08:22:25 +00:00
build_overrides
Use default values provided by PartitionAlloc instead of hard-coded ones
2022-12-07 09:11:35 +00:00
call
Stop Posting tasks when we don't need to.
2023-03-06 15:13:39 +00:00
common_audio
Check FMA3 support before use it in SincResampler
2023-01-31 17:28:55 +00:00
common_video
Fix AVC PPS parser unit test.
2023-03-01 12:00:49 +00:00
data
…
docs
Update documentation links in docs/native-code/development/index.md
2023-02-20 16:35:23 +00:00
examples
Remove mentions of already deleted field trials
2023-03-01 15:53:37 +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
Add a new test binary to check if split-channel works
2023-03-07 11:14:31 +00:00
logging
Move RtcEventLogParseStatus and related macros to a separate file and buildtarget.
2023-03-08 15:18:28 +00:00
media
Reland "Use two MediaChannels for 2 directions."
2023-03-07 12:57:35 +00:00
modules
Skip calling rtcp callback on packets containing invalid rtcp message
2023-03-08 14:19:54 +00:00
net/dcsctp
Implement support for Chrome task origin tracing. #3.5/4
2023-03-01 11:11:37 +00:00
p2p
Replace use of test-only connections() with P2PTransportChannel member.
2023-02-27 16:49:05 +00:00
pc
Update SetHeaderExtensionsToNegotiate to match specification
2023-03-08 08:46:35 +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
Prevent warnings from timestamp aligner used in AudioDeviceBuffer
2023-03-06 15:47:51 +00:00
rtc_tools
Handling NetEqSetMinimumDelay events in neteq_rtpplay.
2023-02-09 09:39:29 +00:00
sdk
Remove resolution alignment requirement (part 2)
2023-03-07 17:18:55 +00:00
stats
stats: rename RTCInboundRTPStreamStats and RTCOutboundRTPStreamStats
2023-03-07 14:27:47 +00:00
system_wrappers
Add option to log a warning for unregistered field trials
2023-02-28 15:43:18 +00:00
test
stats: rename RTCInboundRTPStreamStats and RTCOutboundRTPStreamStats
2023-03-07 14:27:47 +00:00
tools_webrtc
Noop change to trigger bots
2023-02-13 10:30:38 +00:00
video
Use CodecTypeToPayloadString
2023-03-08 14:38:06 +00:00
.clang-format
Add IncludeBlocks to clang-format.
2021-02-03 16:29:07 +00:00
.git-blame-ignore-revs
…
.gitignore
…
.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
Changed OutputToDebug to create a CFString at compile-time, rather than runtime
2023-02-19 22:41:59 +00:00
BUILD.gn
Add a new test binary to check if split-channel works
2023-03-07 11:14:31 +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 11e681944b..e2601310e1 (1114399:1114554)
2023-03-08 16:56:50 +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
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
…
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
Add option to log a warning for unregistered field trials
2023-02-28 15:43:18 +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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