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Victor Boivie 767235dda3 media: sctp: Convert frequent logs to RTC_DLOG
In benchmarks, each log statement represent 2% of the CPU usage. RTC_LOG
is not very expensive, but not free either, and it's called for every
received and sent packet.

Bug: webrtc:12943
Change-Id: Id65baafb5e494091a3a7604687718fdd4f477d86
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231223
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34929}
2021-09-06 13:28:19 +00:00
api [PCLF] Remove temporary AddAndReturnPeer method 2021-09-03 23:44:37 +00:00
audio Revert "Wire up non-sender RTT for audio, and implement related standardized stats." 2021-09-01 17:32:00 +00:00
build_overrides Roll chromium + fix: blacklist -> ignorelist for sanitizers suppressions 2021-05-27 16:16:01 +00:00
call Revert "Reland "Enable WebRTC-Vp9DependencyDescriptor by default"" 2021-09-06 13:11:51 +00:00
common_audio Use backticks not vertical bars to denote variables in comments 2021-08-10 10:40:03 +00:00
common_video SimulcastEncoderAdapter: Use FramerateController instead of FramerateControllerDeprecated. 2021-08-30 10:20:55 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs Update Mac prerequisites 2021-08-23 19:52:17 +00:00
examples Delete BasicPacketSocketFactory default constructor 2021-09-03 10:46:29 +00:00
g3doc Use absl instead of self-made function for low-level bit counting 2021-08-26 08:56:37 +00:00
logging Use backticks not vertical bars to denote variables in comments 2021-08-10 10:40:03 +00:00
media media: sctp: Convert frequent logs to RTC_DLOG 2021-09-06 13:28:19 +00:00
modules Revert "Reland "Enable WebRTC-Vp9DependencyDescriptor by default"" 2021-09-06 13:11:51 +00:00
net/dcsctp dcsctp: Increase RTO limits 2021-09-06 10:24:06 +00:00
p2p Delete BasicPacketSocketFactory default constructor 2021-09-03 10:46:29 +00:00
pc Revert "Wire up non-sender RTT for audio, and implement related standardized stats." 2021-09-01 17:32:00 +00:00
resources Disable high-pass filtering of the AEC reference 2021-02-23 07:06:11 +00:00
rtc_base Delete unused variants of rtc::tokenize 2021-09-03 14:28:47 +00:00
rtc_tools Delete legacy rtp header parser as no longer used 2021-08-09 12:14:52 +00:00
sdk Fix race condition in RTCNetworkMonitor/objc_network_monitor. 2021-09-03 12:44:17 +00:00
stats Revert "Wire up non-sender RTT for audio, and implement related standardized stats." 2021-09-01 17:32:00 +00:00
system_wrappers Use GTEST_SKIP() instead of early return. 2021-08-12 15:24:13 +00:00
test Revert "Reland "Enable WebRTC-Vp9DependencyDescriptor by default"" 2021-09-06 13:11:51 +00:00
tools_webrtc Add TSAN suppression for usrsctp 2021-09-03 12:18:16 +00:00
video frame transformer: expose payload type 2021-08-25 08:33:20 +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 Increase iOS deployment target from 10 to 12. 2021-07-02 17:02:27 +00:00
.vpython Update links to point at main branch 2021-07-22 16:41:26 +00:00
AUTHORS Fix _hRecThread,_hPlayThread RTC_DCHECK reverse bug. 2021-08-18 11:36:46 +00:00
BUILD.gn Allow export of Obj-C symbols without C++ ones. 2021-07-30 22:54:59 +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 58d05cfd21..48501b3f18 (917624:917742) 2021-09-02 18:38:08 +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 Update template to follow chromium copyright style 2013-04-24 01:01:28 +00:00
native-api.md Reference "main" branches instead of "master" branches. 2021-07-15 11:07:44 +00:00
OWNERS Remove inactive owners. 2021-08-31 14:27:49 +00:00
PATENTS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
PRESUBMIT.py fix some typos 2021-08-12 18:37:10 +00:00
presubmit_test.py Add presubmit check to guard against assert() usage. 2021-07-22 17:08:26 +00:00
presubmit_test_mocks.py Add presubmit check to guard against assert() usage. 2021-07-22 17:08:26 +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 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 Revert "Reland "PipeWire capturer: implement proper DMA-BUFs support""" 2021-09-03 11:28:26 +00:00
webrtc_lib_link_test.cc Deprecate PeerConnectionFactory::CreatePeerConnection 2021-05-10 08:47:48 +00:00
whitespace.txt Trigger bots 2021-09-02 10:34:30 +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