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Markus Handell 3d46d0b200 Proxy: solve event tracing with compile time strings.
This change creates trace events with a single parameter
composed of ClassName::Method.

The change additionally causes the duration of the proxy call to be
traced, not only the occurrence.

Fixed: webrtc:12787
Change-Id: I1689862318d4c6fc1dcef343c3ccf3ae9f7e17df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219788
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34149}
2021-05-28 06:53:29 +00:00
api Move proxies into pc/. 2021-05-27 09:56:42 +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 + fix: blacklist -> ignorelist for sanitizers suppressions 2021-05-27 16:16:01 +00:00
call Update WebRTC code version (2021-05-28T04:03:27). 2021-05-28 05:20:39 +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 Introduce CreateDataChannelOrError 2021-05-26 09:43:29 +00:00
g3doc add srtp docs 2021-05-26 09:39:19 +00:00
logging Revert "Deprecate microsecond timestamps in RTC event log." 2021-05-24 13:11:10 +00:00
media dcsctp: Add public API for BufferedAmountLow 2021-05-27 15:27:27 +00:00
modules PipeWire capturer: request mouse cursor to be part of the stream 2021-05-26 19:08:17 +00:00
net/dcsctp dcsctp: Add public API for BufferedAmountLow 2021-05-27 15:27:27 +00:00
p2p Remove cricket::DtlsTransportState. 2021-05-21 21:45:29 +00:00
pc Proxy: solve event tracing with compile time strings. 2021-05-28 06:53:29 +00:00
resources Disable high-pass filtering of the AEC reference 2021-02-23 07:06:11 +00:00
rtc_base Proxy: solve event tracing with compile time strings. 2021-05-28 06:53:29 +00:00
rtc_tools Convert to NTP time using the real clock. 2021-05-21 19:29:32 +00:00
sdk Move proxies into pc/. 2021-05-27 09:56:42 +00:00
stats Support for map of string keys to uint64_t / double values in RTCStats 2021-05-25 20:38:08 +00:00
system_wrappers Make Clock::ConvertTimestampToNtpTime pure virtual 2021-05-21 09:55:14 +00:00
test RtpFrameReferenceFinder return frames directly instead of via callback. 2021-05-26 15:47:03 +00:00
tools_webrtc Roll chromium + fix: blacklist -> ignorelist for sanitizers suppressions 2021-05-27 16:16:01 +00:00
video RtpFrameReferenceFinder return frames directly instead of via callback. 2021-05-26 15:47:03 +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 Add code of conduct to WebRTC repo 2017-05-16 12:09:13 +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 f713d4fb04..4a5a62a362 (887229:887362) 2021-05-27 22:45:17 +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.py sctp: Rename SctpTransport to UsrSctpTransport 2021-04-12 10:40:34 +00:00
presubmit_test.py Reformat python files checked by pylint (part 1/2). 2020-10-30 10:13:11 +00:00
presubmit_test_mocks.py Reformat python files checked by pylint (part 1/2). 2020-10-30 10:13:11 +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.gni Turn on the RTC_ENABLE_WIN_WGC build flag. 2021-05-10 20:16:52 +00:00
webrtc_lib_link_test.cc Deprecate PeerConnectionFactory::CreatePeerConnection 2021-05-10 08:47:48 +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