[BREAKGLASS] Append-only mirror of github.com/signalapp/webrtc
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Sebastian Jansson f70bc5eeff Removes pause check from RoundRobinPacketQueue.
This CL removes a check in RoundRobinPacketQueue::FinalizePop. This
check will trigger if a the pause state is changed in PacedSender while
a packet is sent. This is a rare occurrence but would yield flaky
behavior. The check should not be required for the code to function
since the paused state is not read in FinalizePop other than for this
check.

Bug: webrtc:9586
Change-Id: Ib9476168eb637dc2f9710d0592bed92c4b03dacb
Reviewed-on: https://webrtc-review.googlesource.com/92090
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24197}
2018-08-06 15:22:41 +00:00
api Add AsyncResolverFactory interface and basic implementation. 2018-08-02 21:20:15 +00:00
audio Delete telephone-event handling from RTPReceiverAudio. 2018-08-02 12:55:40 +00:00
build_overrides Add phoglund@ to various OWNERS and remove kjellander@ 2017-10-19 09:21:12 +00:00
call Reland "Remove RTPVideoHeader::h264() accessors." 2018-08-03 09:16:50 +00:00
common_audio Explicitly add -mfpu=neon to all targets that use NEON 2018-08-01 13:15:42 +00:00
common_video Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
data WebRTC: Replace ProjectRootPath by ResourcePath 2016-11-22 18:43:05 +00:00
examples Remove unnecessary //base:base_java dependencies. 2018-08-06 12:04:35 +00:00
infra Remove linux_internal_compile_lite from CQ. 2018-08-06 14:59:31 +00:00
logging Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
media Convert webrtcvideoengine CVO tests away from cricket::VideoCapturer. 2018-07-30 14:41:23 +00:00
modules Removes pause check from RoundRobinPacketQueue. 2018-08-06 15:22:41 +00:00
ortc Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
p2p Remove definition of FEATURE_ENABLE_VOICEMAIL, FEATURE_ENABLE_PSTN. 2018-08-03 10:08:21 +00:00
pc Add AsyncResolverFactory interface and basic implementation. 2018-08-02 21:20:15 +00:00
resources Fix gitignore pattern to never exclude sha1 files 2018-08-03 12:45:45 +00:00
rtc_base Don't allocate an array of size 0, it causes a UBSan failure 2018-08-03 07:49:46 +00:00
rtc_tools Remove unnecessary //base:base_java dependencies. 2018-08-06 12:04:35 +00:00
sdk Remove unnecessary //base:base_java dependencies. 2018-08-06 12:04:35 +00:00
stats Removing clang:find_bad_constructs from stats/. 2018-07-16 14:27:07 +00:00
style-guide Add style guide rule about paired .h and .cc files 2018-03-14 13:02:35 +00:00
system_wrappers Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
test Adds BBR field trial to CallTest. 2018-08-03 18:07:51 +00:00
tools_webrtc Fix remaining target name for AppRTCMobile_stubbed_video_io_test_apk. 2018-08-03 14:17:55 +00:00
video Reland "Remove RTPVideoHeader::h264() accessors." 2018-08-03 09:16:50 +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 format commit. 2018-06-20 09:26:44 +00:00
.gitignore Roll back checking in the third_party directory 2018-06-27 13:04:08 +00:00
.gn Follow "Migrate the Android Support Lib to android_deps" from Chromium 2018-07-19 12:46:02 +00:00
.vpython Add vpython dependencies needed to run presubmit tests on LUCI 2018-05-18 08:10:25 +00:00
AUTHORS Fixed issue with BGRA RTCCVPixelBuffer scale and crop 2018-07-24 08:23:26 +00:00
BUILD.gn Rename rtc_instrumentation_test_apk targets to end with _test_apk. 2018-08-03 12:44:40 +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.h Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
DEPS Roll chromium_revision d6a13562c8..39d45f08f5 (580628:580730) 2018-08-04 03:26:10 +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 Move fft4g to proper third_party directory 2018-07-25 15:44:53 +00:00
native-api.md Remove legacy VoiceEngine. 2018-01-12 11:31:52 +00:00
OWNERS Add mbonadei@ to build configs OWNERS. 2018-06-20 12:39:11 +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 Adding CheckNoWarningSuppressionFlagsAreAdded. 2018-07-04 09:01:32 +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 Add style guidance about forward declarations. 2018-03-28 20:58:27 +00:00
WATCHLISTS Adding mbonadei@ to build_files WATCHLIST. 2018-06-20 12:38:06 +00:00
webrtc.gni Don't call deprecated FFmpeg API. 2018-06-26 13:57:35 +00:00
whitespace.txt Whitespace change 2018-02-23 10:34:16 +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 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