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[BREAKGLASS] Append-only mirror of github.com/signalapp/webrtc
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Danil Chapovalov cf0ec283d2 Delete RtcpStatistic struct as no longer used 
Bug: webrtc:10678
Change-Id: Ic99910817f8b3044124a212627f0a754a54b69e7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219284
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34046}
2021-05-19 08:07:56 +00:00
api VideoRtpReceiver & AudioRtpReceiver threading fixes. 2021-05-17 14:37:55 +00:00
audio Use Timestamp to represent packet receive timestamps 2021-05-04 13:16:54 +00:00
build_overrides Roll chromium_revision 34f3c82122..2dffe06711 (867171:871492) 2021-04-12 18:25:58 +00:00
call Update WebRTC code version (2021-05-19T04:03:01). 2021-05-19 05:37:13 +00:00
common_audio Avoid undefined behavior in a division operation. 2021-04-23 07:49:24 +00:00
common_video Update BitBuffer methods to style guide 2021-05-18 11:10:27 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs Create a VideoFrameTrackingId RTP header extension. 2021-03-25 17:25:18 +00:00
examples Deprecate PeerConnectionFactory::CreatePeerConnection 2021-05-10 08:47:48 +00:00
g3doc Move style guide and abseil-in-webrtc into g3doc subfolder 2021-05-13 14:43:10 +00:00
logging Update BitBuffer methods to style guide 2021-05-18 11:10:27 +00:00
media Remove redundant VideoSendStream::rtcp_stats field 2021-05-18 13:37:51 +00:00
modules Delete RtcpStatistic struct as no longer used 2021-05-19 08:07:56 +00:00
net/dcsctp dcsctp: Restrict fuzzing input length 2021-05-18 09:22:37 +00:00
p2p Define cricket::DtlsTransportState in terms of webrtc::DtlsTransportState. 2021-05-12 07:06:38 +00:00
pc peerconnection: add test for createOffer({voiceActivityDetection: true}) 2021-05-18 09:10:27 +00:00
resources Disable high-pass filtering of the AEC reference 2021-02-23 07:06:11 +00:00
rtc_base Update BitBuffer methods to style guide 2021-05-18 11:10:27 +00:00
rtc_tools Remove REMB throttling funcionality from PacketRouter 2021-05-12 11:24:58 +00:00
sdk Set nativeObserver to 0 to avoid double release. 2021-05-17 19:53:43 +00:00
stats Simplify reference counting implementation of PendingTaskSafetyFlag. 2021-04-21 07:04:01 +00:00
system_wrappers Consolidate the different NTP clocks into one. 2021-04-08 13:54:04 +00:00
test dcsctp: Add socket fuzzer 2021-05-14 06:41:10 +00:00
tools_webrtc crc32c: Point the licensing script to the LICENSE file 2021-05-03 16:46:30 +00:00
video Remove redundant VideoSendStream::rtcp_stats field 2021-05-18 13:37:51 +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 Enable GN check on //net. 2021-05-03 14:23:09 +00:00
.vpython Update six library version 2021-04-26 16:39:07 +00:00
AUTHORS Revert "Ensure method which updates UI is called in main thread" 2021-04-30 09:26:03 +00:00
BUILD.gn Disable range-loop-analysis when use_xcode_clang=true. 2021-05-07 15:24:28 +00:00
CODE_OF_CONDUCT.md
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 19bcae5aaa..108d11241e (884108:884294) 2021-05-19 04:40:23 +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 kwiberg@webrtc.org from OWNERS files 2020-12-04 15:11:26 +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 Make the remote_bitrate_estimator build target private 2020-11-26 12:21:22 +00:00
OWNERS Move style guide and abseil-in-webrtc into g3doc subfolder 2021-05-13 14:43:10 +00:00
PATENTS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
presubmit_test_mocks.py Reformat python files checked by pylint (part 1/2). 2020-10-30 10:13:11 +00:00
presubmit_test.py Reformat python files checked by pylint (part 1/2). 2020-10-30 10:13:11 +00:00
PRESUBMIT.py sctp: Rename SctpTransport to UsrSctpTransport 2021-04-12 10:40:34 +00:00
pylintrc Undo enforcing of PEP-8 pylint changes for method and function names. 2020-11-10 18:26:25 +00:00
README.chromium Add CPEPrefix. 2020-07-13 11:42:07 +00:00
README.md Move style guide and abseil-in-webrtc into g3doc subfolder 2021-05-13 14:43:10 +00:00
WATCHLISTS Add hta@ to rtc_base/ and api/ WATCHLISTS. 2021-01-06 09:43:34 +00:00
webrtc_lib_link_test.cc Deprecate PeerConnectionFactory::CreatePeerConnection 2021-05-10 08:47:48 +00:00
webrtc.gni Turn on the RTC_ENABLE_WIN_WGC build flag. 2021-05-10 20:16:52 +00:00
whitespace.txt Reland "Triggering CI." 2021-03-22 11:57:23 +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