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Sergey Silkin 058972f84e Make LAYER_DROP and max_consec_drop=2 to be default settings 
Based on the results of the experiment (b/335129329).

Bug: webrtc:15827, b/320629637, b/335129329, chromium:329396373
Change-Id: I1599f4c1be79ee3385aac1ff345168982c8278f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360960
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Auto-Submit: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42895}
2024-08-30 10:10:09 +00:00
api Remove DegradedCall - To be submitted after 2024-07-01 2024-08-30 08:08:39 +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 Remove DegradedCall - To be submitted after 2024-07-01 2024-08-30 08:08:39 +00:00
common_audio Fix license metadata for spl_sqrt_floor, portaudio, sigslot 2024-08-29 19:11:29 +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 Make LAYER_DROP and max_consec_drop=2 to be default settings 2024-08-30 10:10:09 +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 Make LAYER_DROP and max_consec_drop=2 to be default settings 2024-08-30 10:10:09 +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 Remove DegradedCall - To be submitted after 2024-07-01 2024-08-30 08:08:39 +00:00
resources Delete unused YUV files 2024-07-11 20:26:16 +00:00
rtc_base Fix license metadata for spl_sqrt_floor, portaudio, sigslot 2024-08-29 19:11:29 +00:00
rtc_tools Inject field trials in NetEqTest instead of setting global. 2024-08-30 09:11:50 +00:00
sdk Only mute microphone while audio_unit is started. 2024-08-29 16:45:00 +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 Remove implicit this captures 2024-08-29 19:30:52 +00:00
tools_webrtc Fix linux_more_configs mb config. 2024-08-28 12:30:24 +00:00
video Propagate Environment into RtpVideoStreamReceiver2 2024-08-29 20:10:45 +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 10ff7fa1e3..b975bdde27 2024-08-30 08:12:16 +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