Add suppressions for libjingle tests so they pass under TSan v2.
Disable the following tests for TSan v2 (only) since they're failing:
* StunServerTest.TestGood
* JsepPeerConnectionP2PTestClient.*
See build logs at:
http://build.chromium.org/p/client.webrtc.fyi/builders/Linux%20Tsan%20v2/
for more details.
BUG=1205,2078,2079,2080,2517
TEST=Ran a successful run of each test locally on Linux using:
GYP_DEFINES='tsan=1 linux_use_tcmalloc=0 release_extra_cflags="-gline-tables-only"' gclient runhooks
ninja -C out/Release
For each test, run standing in trunk/:
TSAN_OPTIONS="suppressions=tools/valgrind-webrtc/tsan_v2/suppressions.txt print_suppressions=1 report_signal_unsafe=0 report_thread_leaks=0 history_size=7" out/Release/[libjingle_testname]
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2411004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4977 4adac7df-926f-26a2-2b94-8c16560cd09d
This makes audio flow well bidirectionally to an iPod Touch (5th gen).
Also:
- Update to new turnserver JSON style:
- separate username field
- multiple URLs for the same server (e.g. both UDP & TCP)
- Added more explicit logging for ICE Connected since it's useful for debugging
- Give focus to the input field on app launch since that's the only useful
thing to have focus on, anyway.
- Fix minor typos
- Cleaned up trailing whitespace and hard tabs
BUG=2191
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2127004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4687 4adac7df-926f-26a2-2b94-8c16560cd09d
I mistakenly ommitted the checks when logging.h was ported from
libjingle to webrtc. This caused a significant CPU cost for logs which
were later filtered out anyway.
Verified with LS_VERBOSE logging in neteq4, running:
$ out/Release/modules_unittests \
--gtest_filter=NetEqDecodingTest.TestBitExactness \
--gtest_repeat=50 > time.txt
$ grep "case ran" time.txt | grep "[0-9]* ms" -o | sort
Results on a MacBook Retina, averaged over 5 runs:
Verbose logs disabled: 666 ms
Exisiting implementation, verbose logs enabled: 944 ms (1.42x)
New implementation, verbose logs enabled: 673 ms (1.01x)
BUG=2314
R=henrik.lundin@webrtc.org, henrike@webrtc.org, kjellander@webrtc.org, turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2160005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4682 4adac7df-926f-26a2-2b94-8c16560cd09d
- TCPPort::~TCPPort() was leaking incoming_ sockets; now they are deleted.
- PeerConnection::RemoveStream() now removes streams even if the
PeerConnection::IsClosed(). Previously such streams would never get removed.
- Gave MediaStreamTrackInterface a virtual destructor to ensure deletes on base
pointers are dispatched virtually.
- VideoTrack.dispose() delegates to super.dispose() (instead of leaking)
- PeerConnection.dispose() now removes streams before disposing of them.
- MediaStream.dispose() now removes tracks before disposing of them.
- VideoCapturer.dispose() only unowned VideoCapturers (mirroring C++ API)
- AppRTCDemo.disconnectAndExit() now correctly .dispose()s its
VideoSource and PeerConnectionFactory.
- CHECK that Release()d objects are deleted when expected to (i.e. no ref-cycles
or missing .dispose() calls) in the Java API.
- Create & Return webrtc::Traces at factory birth/death to be able to assert
that _all_ threads started during the test are collected by the end.
- Name threads attached to the JVM more informatively for debugging.
- Removed a bunch of unnecessary scoped_refptr instances in
peerconnection_jni.cc whose only job was messing with refcounts.
RISK=P2
TESTED=AppRTCDemo can be ended and restarted just fine instead of crashing on camera unavailability. No more post-app-exit logcat lines. PCTest.java now asserts that all threads are collected before exit.
BUG=2183
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2005004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4534 4adac7df-926f-26a2-2b94-8c16560cd09d
Things included in this CL:
- updated READMEs to provide an exact/reproable set of steps for getting the app
running.
- gyp changes to build the iOS AppRTCDemo sample app using gyp+ninja instead of
the hand-crafted Xcode project (which has never worked in its checked-in
form), including a gyp action to sign the sample app for deployment to an iOS
device (the app can also be used in the simulator)
- deleted the busted hand-crafted Xcode project for the sample app
- updated the sample app to match the PeerConnection API that ended up landing
(in a surprising twist of fate, the API landed quite a bit later than the
sample app and this is the first time the CR-time changes in the API are
reflected in the sample app)
- updated the sample app to reflect apprtc.appspot.com HTML/JS changes (equiv to
the AppRTCClient.java changes in http://s10/47299162)
- picked up the iossim DEPS to enable launching the sample app in the simulator
from the command-line.
- renamed some files to match capitalization of the classes they contain (Ice ->
ICE) per ObjC naming guidelines.
- ran the files involved in this CL through clang-format to deal with xcode
formatting craxy.
BUG=2106
RISK=P2
TESTED=unittest builds with ninja and passes on OS=mac; sample app builds with ninja and runs on simulator and device, though no audio flows from simulator/device (will fix in a follow-up CL)
R=andrew@webrtc.org, justincohen@google.com, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1874005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4466 4adac7df-926f-26a2-2b94-8c16560cd09d
