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Henrik Boström e8b00a1a74 Fix OperationsChainTest.OnChainEmptyCallback flake.
This test was added in
https://webrtc-review.googlesource.com/c/src/+/180620. On my machine it
passes about 98% of the time.

The test is meant to count the number of times the callback that the
operations chain is empty has been invoked, and does this by ensuring
the last operation to make the chain empty has completed before
expecting that the counter has increased.

The race happns when the operation has completed but the callback that
the chain is empty has not happened yet. This CL fixes that by using
EXPECT_EQ_WAIT instead.

TBR=hta@webrtc.org

Bug: chromium:1060083
Change-Id: I2ebfac3e635ef895d6602f7360e5ec6006fc1d0a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182541
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31992}
2020-08-25 16:11:22 +00:00
api [Perfect Negotiation] Fire onnegotiationneeded when chain is empty. 2020-08-25 09:56:39 +00:00
audio DTMF Event Sub-API on VoIP API 2020-08-20 17:10:02 +00:00
build_overrides set perfetto flag to default value of false 2020-07-22 10:14:53 +00:00
call [Adaptation] Remove resource adaptation queue 2020-08-25 12:16:20 +00:00
common_audio Support AVX2/FMA intrinsics in audio FIR filter 2020-08-19 10:21:31 +00:00
common_video Delete unneeded dependencies on deprecated build target webrtc_common 2020-08-25 07:33:12 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs Updated IOS documentation with correct build_ios_libs.py path 2020-08-17 09:37:59 +00:00
examples Moved Asynchronicity From Java to C++ for AndroidVoip Demo App 2020-08-18 17:54:56 +00:00
logging Ensure RtcEventLogEncoderNewFormat::EncodeRemoteEstimate handles infite 2020-08-25 09:22:49 +00:00
media Removing call to deprecated SetExtraOptions method 2020-08-20 16:13:12 +00:00
modules Delete unneeded dependencies on deprecated build target webrtc_common 2020-08-25 07:33:12 +00:00
p2p p2p_transport_channel: Add estimated disconnected time to CandidatePairChangeEvent 2020-08-20 11:40:01 +00:00
pc [Perfect Negotiation] Fire onnegotiationneeded when chain is empty. 2020-08-25 09:56:39 +00:00
resources iSAC API wrapper unit test fix 2020-02-27 14:27:23 +00:00
rtc_base Fix OperationsChainTest.OnChainEmptyCallback flake. 2020-08-25 16:11:22 +00:00
rtc_tools Use non-deprecated EncodedImageCallback::OnEncodedImage in rtc_tools 2020-08-12 09:17:15 +00:00
sdk RTC_OBJC_TYPE RTCWrappedNativeVideo{Decoder,Encoder}. 2020-08-25 08:58:29 +00:00
stats Reland "Implement packets_(sent | received) for RTCTransportStats" 2020-07-10 11:50:59 +00:00
style-guide Add style guide rule about paired .h and .cc files 2018-03-14 13:02:35 +00:00
system_wrappers Reland "Support AVX2/FMA intrinsics in Audio Resampler module" 2020-08-17 10:40:44 +00:00
test Reland "Call OnReceivedOverhead after audio network adaptor is created." 2020-08-20 16:07:41 +00:00
tools_webrtc Fix clang revision regexp in chromium autoroller to match new format 2020-08-07 09:22:03 +00:00
video [Adaptation] Remove resource adaptation queue 2020-08-25 12:16:20 +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 Add .clangd to .gitignore 2019-10-28 12:27:50 +00:00
.gn Reenable libaom decoder by default 2020-03-18 18:04:41 +00:00
.vpython Add source-side perf upload script for WebRTC. 2019-11-18 14:37:01 +00:00
abseil-in-webrtc.md Use absl_deps in order to preapre to the Abseil component build release. 2020-06-08 12:59:40 +00:00
AUTHORS Changed AndroidVideoDecoder to also handle IllegalArgumentException and IllegalStateException during the init of the decoder and fallback to a software decoder 2020-08-05 09:41:49 +00:00
BUILD.gn Delete unneeded dependencies on deprecated build target webrtc_common 2020-08-25 07:33:12 +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 Move definition of SpatialLayer to api/video_codecs/spatial_layer.h 2020-08-17 09:45:19 +00:00
DEPS Roll chromium_revision ba69b98d32..0d713c9318 (801212:801321) 2020-08-25 10:39:51 +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 Update template to follow chromium copyright style 2013-04-24 01:01:28 +00:00
native-api.md Make transient suppression optionally excludable via defines 2020-04-02 11:44:07 +00:00
OWNERS Remove phoglund as root owner. 2020-03-30 12:15:56 +00:00
PATENTS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
PRESUBMIT.py Inclusive language in PRESUBMIT.py. 2020-07-22 10:01:23 +00:00
presubmit_test.py Use source_sets in component builds and static_library in release builds. 2019-10-17 21:17:18 +00:00
presubmit_test_mocks.py Inclusive language in PRESUBMIT.py. 2020-07-22 10:01:23 +00:00
pylintrc Fixing py lint errors 2018-07-23 15:28:48 +00:00
README.chromium Add CPEPrefix. 2020-07-13 11:42:07 +00:00
README.md Fix link in documentation. (take 2) 2020-04-16 11:08:43 +00:00
style-guide.md C++ style: We don't allow designated initializers 2020-06-03 09:11:09 +00:00
WATCHLISTS Remove benwright@webrtc.org from WATCHLISTS 2020-01-31 18:46:52 +00:00
webrtc.gni Reland "Support AVX2/FMA intrinsics in Audio Resampler module" 2020-08-17 10:40:44 +00:00
webrtc_lib_link_test.cc Rewrite the lib link test to just be a binary. 2019-10-18 07:42:20 +00:00
whitespace.txt Trigger CI bots. 2020-07-15 17:50:55 +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