signalapp/webrtc
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[BREAKGLASS] Append-only mirror of github.com/signalapp/webrtc
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C++ 91.4%
Java 2.8%
Objective-C++ 2%
C 1.4%
Python 1.2%
Other 1%
ce9da1636a
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Kuang-che Wu ce9da1636a Use FakeRenderer when fuzzing 
Do not fuzz with real renderer because it is merely frame copying and
doesn't exercise different control flows. This CL also improved fuzzing
performance and fixed a memory leak.

Bug: chromium:952606, chromium:1009077, chromium:1009073
Change-Id: I77c6f2581db82bfd95edb18e5f0e541a94c78208
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156620
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29522}
2019-10-17 18:44:03 +00:00
api AEC3: Add support for logging warnings on delay buffer changes 2019-10-15 12:16:51 +00:00
audio Avoid running NullAudioPoller without receiving streams 2019-10-17 10:58:44 +00:00
build_overrides Remove crbug.com/904400 workaround. 2019-03-15 18:36:23 +00:00
call Delete unused method PacedSender::QueueSizePackets 2019-10-17 08:35:38 +00:00
common_audio Avoid reading outside of memory in WebRtcVad_FindMinimum 2019-10-17 10:59:49 +00:00
common_video Always pass arguments to INSTANTIATE_TEST_SUITE_P. 2019-09-30 12:52:07 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs Fixing some typos. 2019-09-10 10:03:50 +00:00
examples New build target api:media_interface 2019-09-19 09:32:27 +00:00
logging Cleanup includes in modules/include/module_common_types.h 2019-10-07 16:06:26 +00:00
media Delete deprecated bytes_sent/bytes_rcvd stat values 2019-10-17 06:41:38 +00:00
modules AEC3: Add multichannel support in the ERLE estimation 2019-10-17 17:34:18 +00:00
p2p Add a turn port prune policy to keep the first ready turn port. 2019-10-14 19:08:23 +00:00
pc Update the header extensions capabilities with mid, rid and rrid 2019-10-15 14:45:58 +00:00
resources Delete voice_detection() pointer to submodule 2019-10-07 13:06:05 +00:00
rtc_base Increase timeout in test-only helper SendTask back to infinity 2019-10-16 18:22:30 +00:00
rtc_tools Plot delay feedback in RTCP arrival order. 2019-10-11 13:52:10 +00:00
sdk Add a turn port prune policy to keep the first ready turn port. 2019-10-14 19:08:23 +00:00
stats Reland "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" 2019-10-15 10:43:59 +00:00
style-guide Add style guide rule about paired .h and .cc files 2018-03-14 13:02:35 +00:00
system_wrappers Move code related to VideoCodingModule to its own build target 2019-09-10 12:34:38 +00:00
test Use FakeRenderer when fuzzing 2019-10-17 18:44:03 +00:00
tools_webrtc Use symbol_level=1 also for MSVC dbg bots. 2019-10-17 12:02:50 +00:00
video Pass full RtpPacket to RtpVideoStreamReceiver::OnReceivedPayload 2019-10-17 14:48:32 +00:00
.clang-format Tune ObjC clang-format configuration 2017-05-11 09:14:18 +00:00
.git-blame-ignore-revs Let git-hyper-blame ignore new format cleanup. 2019-07-11 16:18:51 +00:00
.gitignore Change apprtc_webrtc_browsertest resource dir to avoid MAX_PATH. 2019-09-04 18:49:28 +00:00
.gn Switch to compiling WebRTC -std=c++14 by default 2019-09-09 19:24:16 +00:00
.vpython Add vpython dependencies needed to run presubmit tests on LUCI 2018-05-18 08:10:25 +00:00
abseil-in-webrtc.md Update style guide for absl::make_unique. 2019-09-18 06:10:58 +00:00
AUTHORS Add Ramprakash Jelari to AUTHORS. 2019-10-15 08:07:26 +00:00
BUILD.gn Add a test which breaks if libwebrtc.a don't pull in the right symbols. 2019-10-17 15:03:06 +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
common_types.h Format almost everything. 2019-07-08 13:45:15 +00:00
DEPS Add a test which breaks if libwebrtc.a don't pull in the right symbols. 2019-10-17 15:03:06 +00:00
ENG_REVIEW_OWNERS Enforce LGTM from owners of depends-on paths in DEPS via presubmit. 2018-09-28 12:49:54 +00:00
LICENSE Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
license_template.txt
native-api.md Delete unused I420 "codec" 2018-12-18 12:30:58 +00:00
OWNERS Add #COMPONENT to WebRTC. 2019-10-08 12:20:39 +00:00
PATENTS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
presubmit_test_mocks.py Reland: Add presubmit check for changes in 3pp 2018-05-22 13:11:18 +00:00
presubmit_test.py Fixing py lint errors 2018-07-23 15:28:48 +00:00
PRESUBMIT.py absl::make_unique presubmit check. 2019-09-17 17:47:31 +00:00
pylintrc Fixing py lint errors 2018-07-23 15:28:48 +00:00
README.chromium Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
README.md Tell users where they can find the native API headers 2017-11-14 10:36:46 +00:00
style-guide.md Update WebRTC's C++ style guide to reflect the switch to C++14. 2019-09-16 11:45:35 +00:00
WATCHLISTS Add saza to audio watchlists 2019-09-03 14:55:43 +00:00
webrtc_lib_link_test.cc Add a test which breaks if libwebrtc.a don't pull in the right symbols. 2019-10-17 15:03:06 +00:00
webrtc.gni Remove rtc_use_lto GN arg. 2019-08-20 14:00:49 +00:00
whitespace.txt Whitespace change 2018-02-23 10:34:16 +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 http://www.webrtc.org/native-code/development 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