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Evan Shrubsole d8b6b06e70 [Unwrap] Delete rtc::TimestampWrapAroundHandler 
Bug: webrtc:13982
Change-Id: Ia2999e952a55d97dbd69ff19cf12c8f712b1a62f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290882
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39087}
2023-01-12 14:21:03 +00:00
api stats: deprecate timestamp_us constructor and method 2023-01-11 11:40:05 +00:00
audio [Unwrap] Use RtpTimestampUnwrapper in audio/channel_receive 2023-01-11 11:59:09 +00:00
build_overrides Use default values provided by PartitionAlloc instead of hard-coded ones 2022-12-07 09:11:35 +00:00
call Update WebRTC code version (2023-01-12T04:02:41). 2023-01-12 05:42:44 +00:00
common_audio Make header files self contained. 2022-10-08 08:38:36 +00:00
common_video Make header files self contained. 2022-10-08 08:38:36 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs Update webrtc build command instruction to autoninja. 2022-11-25 09:19:37 +00:00
examples Split rtc_base into multiple targets 2023-01-09 12:21:25 +00:00
experiments Add tool for generating field trial registry header 2022-10-18 07:25:43 +00:00
g3doc Update WebRTC style guide about GN. 2022-12-16 10:19:13 +00:00
infra Add a ios_x64_dbg_simulator try bot. 2023-01-12 14:07:37 +00:00
logging [Unwrap] Migrate RtcEventLog parser to use RtpSequenceNumberUnwrapper 2023-01-12 10:23:54 +00:00
media Add SVC fallback. 2023-01-11 16:49:49 +00:00
modules Skip trimming packet arrival history at the beginning 2023-01-12 11:59:27 +00:00
net/dcsctp [Unwrap] Migrate dcsctp sequence numbers to SeqNumUnwrapper 2023-01-12 12:00:30 +00:00
p2p Split rtc_base into multiple targets 2023-01-09 12:21:25 +00:00
pc Use parsed packet from RtpTransport::DemuxPacket in engine and call 2023-01-10 15:06:50 +00:00
resources Clarify and extend test support for certain sample rates in audio processing 2022-08-03 14:26:36 +00:00
rtc_base [Unwrap] Delete rtc::TimestampWrapAroundHandler 2023-01-12 14:21:03 +00:00
rtc_tools Split rtc_base into multiple targets 2023-01-09 12:21:25 +00:00
sdk Add dimensions to video settings in objc sdk camera backend. 2023-01-11 13:59:37 +00:00
stats stats: deprecate timestamp_us constructor and method 2023-01-11 11:40:05 +00:00
system_wrappers [Unwrap] Migrate RtpToNtpEstimator to use RtpTimestampUnwrapper 2023-01-11 17:14:41 +00:00
test [DVQA] Remove default value for report_infra_metrics in VideoQualityAnalyzerInjectionHelper 2023-01-10 13:07:48 +00:00
tools_webrtc Add a ios_x64_dbg_simulator try bot. 2023-01-12 14:07:37 +00:00
video [Unwrap] Migrate rtp_rtcp_tests to RtpSequenceNumberUnwrapper 2023-01-12 10:55:15 +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 Set Fuchsia Api level + update SDK version 2022-09-14 08:49:56 +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 Add python grpc to .vpython3 for ios test runner 2022-09-16 12:26:48 +00:00
AUTHORS Remove dimension check in SimulcastUtility::ValidSimulcastParameters 2023-01-11 13:41:55 +00:00
BUILD.gn Only build fuchsia_perf_tests on fuchsia os. 2023-01-11 11:35:51 +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 40afafa78c..b924865c52 (1091586:1091696) 2023-01-12 06:22:31 +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 Add infra owners file 2022-12-02 09:21:47 +00:00
OWNERS_INFRA Add infra owners file 2022-12-02 09:21:47 +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 Remove xooglers from WATCHLISTS and OWNERS 2022-11-30 15:33:25 +00:00
webrtc.gni Verify field trials looked up through field_trial::FindFullName 2022-10-20 10:46:01 +00:00
webrtc_lib_link_test.cc Deprecate PeerConnectionFactory::CreatePeerConnection 2021-05-10 08:47:48 +00:00
whitespace.txt Trigger bots 2022-11-17 21:29:53 +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