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Danil Chapovalov a99bf7fa84 Delete deprecated AudioDecoderOpus::MakeAudioDecoder 
Bug: webrtc:356878416
Change-Id: I2dc830c46fb5eece3b93a0354fd1e8a323a5e2ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360841
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42880}
2024-08-29 08:55:27 +00:00
api Delete deprecated AudioDecoderOpus::MakeAudioDecoder 2024-08-29 08:55:27 +00:00
audio Fix AudioSendStream reconfigure - stop processing during unconfigured state 2024-08-20 16:22:04 +00:00
build_overrides build: add options to configure libsrtp for boringssl or other libraries 2024-08-27 07:17:52 +00:00
call Update WebRTC code version (2024-08-29T04:04:15). 2024-08-29 06:11:38 +00:00
common_audio Remove PushResampler<T>::InitializeIfNeeded 2024-07-04 10:33:21 +00:00
common_video Add message container for the corruption detection extension 2024-08-14 12:48:49 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs Fix formatting for corruption detection header explainer. 2024-08-27 15:18:27 +00:00
examples Refactor WebRTC self assignments in if clauses 2024-08-26 15:56:43 +00:00
experiments Dont signal ReadyToSend in RtpTransport::SendPacket 2024-08-27 14:16:53 +00:00
g3doc Add doc on how to handle python presubmit failures. 2024-07-29 14:17:35 +00:00
infra Re-enable iOS simulator from CQ and LKGR. 2024-08-27 06:05:24 +00:00
logging Remove more sstream deps 2024-07-09 10:30:26 +00:00
media Support standard simulcast with requested_resolution. 2024-08-21 09:35:52 +00:00
modules Use NetEq::GetCurrentDecoderFormat in AcmReceiver. 2024-08-28 17:33:36 +00:00
net/dcsctp dcsctp: Re-add lost validating in test case 2024-08-26 09:22:13 +00:00
p2p Dont signal ReadyToSend in RtpTransport::SendPacket 2024-08-27 14:16:53 +00:00
pc Dont signal ReadyToSend in RtpTransport::SendPacket 2024-08-27 14:16:53 +00:00
resources Delete unused YUV files 2024-07-11 20:26:16 +00:00
rtc_base build: add options to configure libsrtp for boringssl or other libraries 2024-08-27 07:17:52 +00:00
rtc_tools video_replay: ignore non-rtp packets 2024-07-16 16:24:04 +00:00
sdk Extend objc RTCVideoCodecInfo to include scalability modes 2024-08-27 09:40:36 +00:00
stats Mass removal of absl_deps in all BUILD.gn files 2024-05-23 15:09:46 +00:00
system_wrappers Mass removal of absl_deps in all BUILD.gn files 2024-05-23 15:09:46 +00:00
test Add the corruption detection extension to RTPExtensionType 2024-08-27 08:27:20 +00:00
tools_webrtc Fix linux_more_configs mb config. 2024-08-28 12:30:24 +00:00
video Remove redundant mapping. 2024-08-28 11:44:43 +00:00
.clang-format Add IncludeBlocks to clang-format. 2021-02-03 16:29:07 +00:00
.git-blame-ignore-revs Add formatting CLs to .git-blame-ignore-revs 2023-05-07 09:27:47 +00:00
.gitignore Add .cache to .gitignore. 2021-01-20 15:01:07 +00:00
.gn Roll chromium_revision ba1ae79f58..6f9b3224db (1319128:1338914) 2024-08-08 09:20:02 +00:00
.mailmap Add .mailmap for git. 2022-02-20 14:22:13 +00:00
.style.yapf Configure YAPF to follow PEP-8 altogether 2023-09-22 10:32:11 +00:00
.vpython3 Update to vpython 3.11 and remove .vpython (v2.x) 2024-01-25 11:12:20 +00:00
AUTHORS remove deprecated <codecvt> 2024-08-22 10:37:00 +00:00
BUILD.gn build: add options to configure libsrtp for boringssl or other libraries 2024-08-27 07:17:52 +00:00
CODE_OF_CONDUCT.md Reland "Migrate WebRTC documentation to new renderer" 2023-01-31 09:30:04 +00:00
codereview.settings Don't add webrtc-reviews@ to CC, it can be added globally on Gerrit 2018-10-25 08:19:53 +00:00
DEPS Roll chromium_revision c3a359139e..10ff7fa1e3 (1348059:1348232) 2024-08-28 21:30:56 +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 AudioProcessingImpl: Remove the use of transient suppressor 2024-08-05 12:38:37 +00:00
OWNERS Add infra owners file 2022-12-02 09:21:47 +00:00
OWNERS_INFRA Allow to keep old python style for existing files. 2023-10-17 13:52:56 +00:00
PATENTS
PRESUBMIT.py Add 'SkipNextFrame' to the FrameGeneratorInterface. 2024-08-01 14:38:52 +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 Configure Pylint to follow PEP-8 2023-09-25 15:56:09 +00:00
pylintrc_old_style Allow to keep old python style for existing files. 2023-10-17 13:52:56 +00:00
README.chromium Add Security Critical field to README.chromium. 2024-08-27 07:38:26 +00:00
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.gni AudioProcessingImpl: Remove the use of transient suppressor 2024-08-05 12:38:37 +00:00
webrtc_lib_link_test.cc Move webrtc::AudioDeviceModule include to api/ folder 2024-04-22 08:56:31 +00:00
whitespace.txt Test CQ 2024-05-27 12:46:57 +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