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Danil Chapovalov 2bc41bc980 Detach RemoteBitrateEstimator interface from Module 
Bug: webrtc:7219
Change-Id: I8302c5044582d73b0918013a0df89b9390788728
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267140
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37393}
2022-07-01 10:17:40 +00:00
api Add support for scalability modes L2T3 and S2T3 2022-07-01 08:17:04 +00:00
audio Move to_queued_task.h and pending_task_safety_flag.h into public API 2022-06-17 09:20:39 +00:00
build_overrides Roll chromium + fix: blacklist -> ignorelist for sanitizers suppressions 2021-05-27 16:16:01 +00:00
call video_replay: add flexfec support 2022-07-01 09:42:24 +00:00
common_audio Adopt absl::string_view in common_audio/ 2022-05-13 15:00:14 +00:00
common_video Add 420 and 422 10 bit h264 decoding. 2022-06-17 11:12:10 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs Add contributing.md 2021-11-03 14:59:46 +00:00
examples Adopt absl::string_view in p2p/ 2022-06-30 13:19:18 +00:00
g3doc Clarify how to reference WebRTC bugs in TODOs 2022-07-01 08:03:34 +00:00
infra Revert "Temporarily disable video_capture_tests on linux" 2022-06-29 13:35:28 +00:00
logging Change RTCEventLogFactory to have a const Create function 2022-06-28 23:48:37 +00:00
media When VP9 SVC is used, use SvcConfig to set max bitrate for the stream. 2022-06-29 14:23:08 +00:00
modules Detach RemoteBitrateEstimator interface from Module 2022-07-01 10:17:40 +00:00
net/dcsctp dcsctp: Add metric for using message interleaving 2022-07-01 08:12:44 +00:00
p2p Adopt absl::string_view in p2p/ 2022-06-30 13:19:18 +00:00
pc Do not allow simulcast to be turned off using SDP munging 2022-07-01 09:06:44 +00:00
resources AEC3: Changing the default for the use_conservative_tail_frequency_response flag. 2021-12-21 17:35:26 +00:00
rtc_base Refactor RepeatingTaskHandle to use absl::AnyInvocable 2022-06-30 12:22:17 +00:00
rtc_tools video_replay: add flexfec support 2022-07-01 09:42:24 +00:00
sdk Switch from junit_binary to robolectric_binary. 2022-06-30 08:02:18 +00:00
stats Implement Outbound/InboundRtpStreamStats.mid. 2022-06-17 08:44:09 +00:00
system_wrappers Delete rtc_base/atomic_ops.h 2022-06-28 08:32:13 +00:00
test Adopt absl::string_view in p2p/ 2022-06-30 13:19:18 +00:00
tools_webrtc Make WebRTC use third_party/libevent rather than base/third_party/libevent 2022-06-30 07:43:49 +00:00
video Delete rtc_base/atomic_ops.h 2022-06-28 08:32:13 +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 Prevent jsoncpp from hiding deprecated declarations in WebRTC 2022-04-11 12:33:47 +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 Remove unused script webrtc_dashboard_upload.py 2022-03-21 12:54:42 +00:00
AUTHORS GCC: Avoid symbol clash in RenderBuffer 2022-05-24 10:47:56 +00:00
BUILD.gn SVC: Add end to end tests for VP8 and VP9 2022-06-22 11:07:01 +00:00
CODE_OF_CONDUCT.md Reference "main" branches instead of "master" branches. 2021-07-15 11:07:44 +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 4b737679fa..c4edcb406f (1019786:1019918) 2022-07-01 02:50:34 +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 Fix add some eng prod owners to PRESUBMIT.py. 2022-03-18 13:19:07 +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 Update WATCHLISTS 2021-08-23 13:37:55 +00:00
webrtc.gni [Cast Convergence] Replace is_chromecast with new args 2022-06-16 00:50:08 +00:00
webrtc_lib_link_test.cc Deprecate PeerConnectionFactory::CreatePeerConnection 2021-05-10 08:47:48 +00:00
whitespace.txt Trigger CI bots 2021-12-16 17:45:31 +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