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Danil Chapovalov 300a230f16 Delete inter arrival jitter rtcp packet as unused
WebRTC doesn't produces such packet and ignores it when receive.

Bug: None
Change-Id: I4af8cb3308cb2422808bdfc420a85fa175085bfb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269181
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37627}
2022-07-27 14:53:05 +00:00
api Allow recursive check for RTC_DCHECK_RUN_ON macro 2022-07-26 09:27:23 +00:00
audio Allow recursive check for RTC_DCHECK_RUN_ON macro 2022-07-26 09:27:23 +00:00
build_overrides Roll chromium + fix: blacklist -> ignorelist for sanitizers suppressions 2021-05-27 16:16:01 +00:00
call Remove legacy WebRTCFakeNetwork field trials. 2022-07-27 07:33:15 +00:00
common_audio Adopt absl::string_view in common_audio/ 2022-05-13 15:00:14 +00:00
common_video Migrate remaining webrtc usage of TaskQueueBase to absl::AnyInvocable 2022-07-20 08:15:08 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs [ios] Remove the support for bitcode 2022-07-04 09:01:52 +00:00
examples Delete ProcessThread and related Module interface 2022-07-04 10:20:35 +00:00
g3doc Clarify how to reference WebRTC bugs in TODOs 2022-07-01 08:03:34 +00:00
infra Reland "Wait for frames to arrive in WgcCapturer instead of returning nothing." 2022-07-06 20:28:26 +00:00
logging Delete inter arrival jitter rtcp packet as unused 2022-07-27 14:53:05 +00:00
media Allow recursive check for RTC_DCHECK_RUN_ON macro 2022-07-26 09:27:23 +00:00
modules Delete inter arrival jitter rtcp packet as unused 2022-07-27 14:53:05 +00:00
net/dcsctp Migrate remaining webrtc usage of TaskQueueBase to absl::AnyInvocable 2022-07-20 08:15:08 +00:00
p2p Allow recursive check for RTC_DCHECK_RUN_ON macro 2022-07-26 09:27:23 +00:00
pc TrackMediaInfoMap: Use rtc::ArrayView in Initialize. 2022-07-27 11:28:25 +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 Remove unused field trial WebRTC-JitterUpperBound 2022-07-26 10:21:04 +00:00
rtc_tools Add lower/upper link capacity to the outgoing bitrate graph. 2022-07-19 13:22:32 +00:00
sdk Migrate remaining webrtc usage of TaskQueueBase to absl::AnyInvocable 2022-07-20 08:15:08 +00:00
stats Implement RTCInboundRTPStreamStats.JitterBufferMinimumDelay 2022-07-20 09:14:03 +00:00
system_wrappers Delete rtc_base/atomic_ops.h 2022-06-28 08:32:13 +00:00
test Delete inter arrival jitter rtcp packet as unused 2022-07-27 14:53:05 +00:00
tools_webrtc Clobber win bots 2022-07-19 11:34:02 +00:00
video Allow recursive check for RTC_DCHECK_RUN_ON macro 2022-07-26 09:27:23 +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 Update protobuf-py2_py3 wheel. 2022-07-01 15:17:36 +00:00
AUTHORS Add missing header to fix build error when using linux system libraries 2022-07-19 12:25:42 +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 4382d0306b..0296d79495 (1028104:1028209) 2022-07-26 13:22:19 +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 Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
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 Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
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

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