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Paul Hallak b59e9045bf Add the ability to convert a timestamp to NTP time.
The timestamps returned by the clocks do not have an epoch. Each clock
should be able to convert a timestamp it returns to an NTP time.
The default implementation for querying for an NTP time is converting
the current timestamp.

This is favored over returning the offset between the relative and the
NTP time because there is a field trial that makes the real clock revert
to using system dependent methods for getting the NTP time.

Bug: webrtc:11327
Change-Id: Ia139b2744b407cae94420bf9112212ec577efb16
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219687
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Paul Hallak <phallak@google.com>
Cr-Commit-Position: refs/heads/master@{#34071}
2021-05-21 04:34:11 +00:00
api Make local to capturer clock offset a separate entry in PacketInfo. 2021-05-20 13:42:57 +00:00
audio Make local to capturer clock offset a separate entry in PacketInfo. 2021-05-20 13:42:57 +00:00
build_overrides Roll chromium_revision 34f3c82122..2dffe06711 (867171:871492) 2021-04-12 18:25:58 +00:00
call Update WebRTC code version (2021-05-20T04:01:58). 2021-05-20 05:46:18 +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 Replace legacy getStats with standard getStats in the iOS example 2021-05-20 08:40:11 +00:00
g3doc Move style guide and abseil-in-webrtc into g3doc subfolder 2021-05-13 14:43:10 +00:00
logging Start with a BeginLog event in event log encoder unittest 2021-05-20 17:44:30 +00:00
media sctp: Reduce logging level for common calls 2021-05-19 13:22:43 +00:00
modules Make AgcManagerDirect clipping parameters configurable 2021-05-20 16:53:59 +00:00
net/dcsctp dcsctp: Report duplicate TSNs 2021-05-19 12:57:03 +00:00
p2p Define cricket::DtlsTransportState in terms of webrtc::DtlsTransportState. 2021-05-12 07:06:38 +00:00
pc Add test for many non-dropped packets in unreliable mode. 2021-05-19 17:30:46 +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 Add the ability to convert a timestamp to NTP time. 2021-05-21 04:34:11 +00:00
test Add the ability to convert a timestamp to NTP time. 2021-05-21 04:34:11 +00:00
tools_webrtc crc32c: Point the licensing script to the LICENSE file 2021-05-03 16:46:30 +00:00
video Make local to capturer clock offset a separate entry in PacketInfo. 2021-05-20 13:42:57 +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 Switch from check_targets to no_check_targets in .gn 2021-05-20 10:42:21 +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 Switch from check_targets to no_check_targets in .gn 2021-05-20 10:42:21 +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 6291fe6f0e..e5dd2eb61e (884821:884954) 2021-05-20 11:06:22 +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

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