signalapp/webrtc
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[BREAKGLASS] Append-only mirror of github.com/signalapp/webrtc
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Karl Wiberg bb19fcf3bd Add explicit cast to void to silence -Wcomma warning 
Bug: webrtc:9014
Change-Id: I390a8d722e40a101c29ca7a71c6429cba26c17ee
Reviewed-on: https://webrtc-review.googlesource.com/67560
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22787}
2018-04-09 10:00:09 +00:00
api Enable and fix chromium clang warnings in sdk/android targets. 2018-04-05 11:22:03 +00:00
audio This CL removes all usages of our custom ostream << overloads. 2018-04-03 12:51:00 +00:00
build_overrides Add phoglund@ to various OWNERS and remove kjellander@ 2017-10-19 09:21:12 +00:00
call Added SetClockOffset on FakeNetworkPipe. 2018-04-06 09:02:12 +00:00
common_audio Prepare for |is_posix| switch in the Fuchsia build 2018-04-05 17:25:39 +00:00
common_video Enable and fix chromium clang warnings in sdk/android targets. 2018-04-05 11:22:03 +00:00
data WebRTC: Replace ProjectRootPath by ResourcePath 2016-11-22 18:43:05 +00:00
examples Android: Add AudioDeviceModule interface and clean up implementation code 2018-04-06 10:13:02 +00:00
infra Reland "Add linux_internal_compile_lite trybot to CQ" 2018-03-26 10:12:59 +00:00
logging compare Optional<unsigned> only to unsigned integers 2018-04-07 10:07:47 +00:00
media Delete obsolete enable argument to SetVideoSend. 2018-04-09 08:45:29 +00:00
modules Add explicit cast to void to silence -Wcomma warning 2018-04-09 10:00:09 +00:00
ortc Use new TransportController implementation in PeerConnection. 2018-03-30 18:41:19 +00:00
p2p By default, don't use SRTP_AES128_CM_SHA1_32 protection profile. 2018-04-05 23:43:07 +00:00
pc Delete obsolete enable argument to SetVideoSend. 2018-04-09 08:45:29 +00:00
resources Adding FourPeople_1280x720_30.yuv. 2018-02-12 15:55:00 +00:00
rtc_base compare Optional<unsigned> only to unsigned integers 2018-04-07 10:07:47 +00:00
rtc_tools Fix path to AppRTC/collider on Windows 2018-04-04 14:47:40 +00:00
sdk AudioCodingModule::Create(): Require caller to supply an AudioDecoderFactory 2018-04-06 15:10:27 +00:00
stats PeerConnectionInterface::GetStats() with selector argument added. 2018-03-26 12:08:20 +00:00
style-guide Add style guide rule about paired .h and .cc files 2018-03-14 13:02:35 +00:00
system_wrappers Remove myself from some owners files. Fix order in those files. Replace myself in comment references. 2018-04-03 14:07:31 +00:00
test Revert "Floating-point exception observer for unit tests" 2018-04-06 12:46:33 +00:00
tools_webrtc Migrate autoroller from roll-dep-svn to gclient setdep 2018-04-04 11:27:30 +00:00
video Storing frame if encoder is paused. 2018-04-07 14:21:27 +00:00
.clang-format Tune ObjC clang-format configuration 2017-05-11 09:14:18 +00:00
.git-blame-ignore-revs Create .git-blame-ignore-revs and add Java format CL to it. 2016-10-20 09:20:39 +00:00
.gitignore Reland "Make it possible to run video_quality_loopback_test in swarming." 2018-01-23 13:03:17 +00:00
.gn Opt out of "Migrate the Android Support Lib to android_deps". 2018-04-05 13:40:53 +00:00
.vpython vpython: Specify dependency on pywin32 2018-02-14 13:56:39 +00:00
AUTHORS Fixed video capturing on Mac. 2018-04-03 16:23:01 +00:00
BUILD.gn Prepare for |is_posix| switch in the Fuchsia build 2018-04-05 17:25:39 +00:00
CODE_OF_CONDUCT.md Add code of conduct to WebRTC repo 2017-05-16 12:09:13 +00:00
codereview.settings Make Gerrit the default for WebRTC changes 2017-09-29 01:38:07 +00:00
common_types.cc Adding layering configurator and rate allocator for VP9 SVC. 2018-03-29 10:16:47 +00:00
common_types.h Adding layering configurator and rate allocator for VP9 SVC. 2018-03-29 10:16:47 +00:00
DEPS Roll chromium_revision d5c1e1eef5..971d5b4f79 (549017:549119) 2018-04-09 08:43:18 +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
LICENSE_THIRD_PARTY Remove custom MD5 / SHA-1 implementations. 2018-02-19 15:03:35 +00:00
native-api.md Remove legacy VoiceEngine. 2018-01-12 11:31:52 +00:00
OWNERS Add back OWNERS entris that went missing 2018-03-06 10:31:11 +00:00
PATENTS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
presubmit_test_mocks.py Using Change.BugsFromDescription to read CL bugs in PRESUBMIT checks. 2017-10-13 03:48:26 +00:00
presubmit_test.py Using Change.BugsFromDescription to read CL bugs in PRESUBMIT checks. 2017-10-13 03:48:26 +00:00
PRESUBMIT.py Add ToString() methods to classes with << operators, preparing for deprecations. 2018-04-03 11:21:30 +00:00
pylintrc Removing invalid-name from disabled pylint checks. 2017-10-11 08:06:49 +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 Add style guidance about forward declarations. 2018-03-28 20:58:27 +00:00
typedefs.h Move FALLTHROUGH macro to a separate header, and give it an RTC_ prefix 2018-02-05 11:24:59 +00:00
WATCHLISTS Move remaining traces of VoiceEngine 2018-01-17 13:27:47 +00:00
webrtc.gni Annotate rest of WebRTC with @Nullable. 2018-03-28 08:30:06 +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