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Harald Alvestrand 1f206b841e Use ArrayView in the IncomingRtcpPacket function.
The lowest level and some of the highest levels of this function are
already using ArrayView. Make this consistent throughout.
Use deprecation for the old API rather than deleting it, since upstream
may be using it.

Bug: webrtc:14870
Change-Id: If5e1a6e9802ecf7e8e3ec27befb5167ca9985517
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291706
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39241}
2023-02-01 12:19:03 +00:00
api Add possibility to set MetricsSet metadata. 2023-01-31 12:41:47 +00:00
audio Use ArrayView in the IncomingRtcpPacket function. 2023-02-01 12:19:03 +00:00
build_overrides Use default values provided by PartitionAlloc instead of hard-coded ones 2022-12-07 09:11:35 +00:00
call Use ArrayView in the IncomingRtcpPacket function. 2023-02-01 12:19:03 +00:00
common_audio Check FMA3 support before use it in SincResampler 2023-01-31 17:28:55 +00:00
common_video Add 444 10 bits support for H264 and VP9 2023-01-17 12:32:26 +00:00
data
docs Update WebRTC doc related to webrtc.org accounts. 2023-01-16 09:34:28 +00:00
examples Remove all usage of //rtc_base target 2023-01-16 14:36:06 +00:00
experiments Add tool for generating field trial registry header 2022-10-18 07:25:43 +00:00
g3doc Fix doc path 2023-01-31 10:14:47 +00:00
infra Fix gtest-output and resultdb for fuchsia 2023-01-25 14:27:38 +00:00
logging Reland "Migrate WebRTC documentation to new renderer" 2023-01-31 09:30:04 +00:00
media Remove rtp header extension from config of Call audio and video receivers 2023-01-31 11:58:43 +00:00
modules Use ArrayView in the IncomingRtcpPacket function. 2023-02-01 12:19:03 +00:00
net/dcsctp Remove all usage of //rtc_base target 2023-01-16 14:36:06 +00:00
p2p Reland "Migrate WebRTC documentation to new renderer" 2023-01-31 09:30:04 +00:00
pc Reland "Migrate WebRTC documentation to new renderer" 2023-01-31 09:30:04 +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 Add method to get FD for physical socket 2023-01-31 10:25:45 +00:00
rtc_tools Ensure VideoRtpReplayer use new PacketReceiver::DeliverRtp packet. 2023-01-18 12:47:41 +00:00
sdk Disable RTCCameraVideoCapturerTestsWithMockedCaptureSession. 2023-01-18 09:45:06 +00:00
stats Reland "Migrate WebRTC documentation to new renderer" 2023-01-31 09:30:04 +00:00
system_wrappers Check FMA3 support before use it in SincResampler 2023-01-31 17:28:55 +00:00
test Use ArrayView in the IncomingRtcpPacket function. 2023-02-01 12:19:03 +00:00
tools_webrtc [infra] Clean up mb_config.pyl after reclient migration 2023-01-30 07:47:12 +00:00
video Use ArrayView in the IncomingRtcpPacket function. 2023-02-01 12:19:03 +00:00
.clang-format Add IncludeBlocks to clang-format. 2021-02-03 16:29:07 +00:00
.git-blame-ignore-revs Let git-hyper-blame ignore new format cleanup. 2019-07-11 16:18:51 +00:00
.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 Remove dimension check in SimulcastUtility::ValidSimulcastParameters 2023-01-11 13:41:55 +00:00
BUILD.gn Delete unused Audio Bwe integration test. 2023-01-26 09:31:44 +00:00
CODE_OF_CONDUCT.md Reland "Migrate WebRTC documentation to new renderer" 2023-01-31 09:30:04 +00:00
codereview.settings
DEPS Add android-tiramisuprivacysandbox to DEPS. 2023-01-31 09:19:08 +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.py Update portaudio to the latest 2022-05-13 09:01:34 +00:00
presubmit_test.py tools_webrtc dir converted to py3 + top level PRESUBMIT script 2022-02-08 14:42:26 +00:00
presubmit_test_mocks.py tools_webrtc dir converted to py3 + top level PRESUBMIT script 2022-02-08 14:42:26 +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.gni Verify field trials looked up through field_trial::FindFullName 2022-10-20 10:46:01 +00:00
webrtc_lib_link_test.cc Deprecate PeerConnectionFactory::CreatePeerConnection 2021-05-10 08:47:48 +00:00
whitespace.txt Trigger bots 2022-11-17 21:29:53 +00:00

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