4966 Commits

Author SHA1 Message Date
andrew@webrtc.org
aa3d1c8169 Make unittest log printouts opt-in with a --logs flag.
TESTED=Using modules_unittests, no logs are printed by default.
Specifying --logs prints logs. gtest flags work correctly.

R=henrike@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2181004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4686 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-05 21:16:29 +00:00
alexeypa@chromium.org
bebf3995ce Pre-multiply images for MouseCursorShape.
BUG=chromium:267270
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/2173004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4685 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-05 19:32:46 +00:00
fischman@webrtc.org
31b4a5ac82 Recognize armv7 target_arch for ios support in webrtc common.gyp
BUG=2343
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2176004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4684 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-05 16:46:36 +00:00
braveyao@webrtc.org
be588f9a58 Apprtc Demo: calling createOffer/Answer without failureCallback is deprecated in FF
BUG=2313
Test=Manual test

R=dutton@google.com, juberti@google.com, vikasmarwaha@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2175004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4683 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-05 16:44:55 +00:00
andrew@webrtc.org
9080518a39 Restore severity precondition to logging.h.
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
2013-09-05 16:40:43 +00:00
pbos@webrtc.org
95e51f509c Remove send and receive streams when destroyed.
Fixes crash where packets were sent to a receive stream that had been
destroyed but not removed from the ssrc mapping from call to receiver.
Added a repro case that reliably crashed before the fix.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2161007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4681 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-05 12:38:54 +00:00
henrik.lundin@webrtc.org
164c4f71ba Add clockdrift to RtpGenerator
RtpGenerator is a help class for NetEq testing. This change
add the possibility to simulate clockdrift. If no clockdrift is
set, the default is 0 (i.e., no drift).

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2175005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4680 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-05 12:16:38 +00:00
pbos@webrtc.org
7e1bf318bf Allow unknown flags in test_main.cc.
Adds AllowCommandLineParsing to allow us to ignore "--no-sandbox" given
by new TSanV2 bots. Not ignoring this flag prevents the test from
running on this machine. Also removing unnecessary asserts that clutter
code.

BUG=
TEST=Locally running video_engine_tests with --no-sandbox.
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2178004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4679 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-05 10:27:46 +00:00
henrik.lundin@webrtc.org
36439bf906 NetEq4: Small change to reduce allocs in AudioMultiVector
This change reduced the allocation count by 20000 in the bit-exactness
test.

BUG=Issue 1363
TEST=out/Debug/modules_unittests --gtest_filter=NetEqDecodingTest.TestBitExactness
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2157004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4678 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-05 06:02:56 +00:00
mflodman@webrtc.org
e2d4da6586 Enable EngineTest.ReceivesPliAndRecoversWithNack and fix memcheck suppression filter.
BUG=2346
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4677 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-04 14:21:57 +00:00
mflodman@webrtc.org
be23b32727 Adding tsan suppression for BUG 2349.
TSAN found a read/write race for RTPSender::[packets_sent_/payload_bytes_sent)] between RTPSender::SendToNetwork and RTCPSender::SendRTCP.

BUG=2349
R=holmer@google.com

Review URL: https://webrtc-codereview.appspot.com/2168004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4676 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-04 13:36:52 +00:00
andresp@webrtc.org
77bf5c28c8 Clean capture timestamp code.
BUG=
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2134004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4675 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-04 11:35:43 +00:00
mflodman@webrtc.org
06f1f74331 Disable EngineTest.ReceivesPliAndRecoversWithNack.
The test times out on Linux memcheck bot at times.

BUG=2348

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2159007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4674 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-04 11:00:07 +00:00
mflodman@webrtc.org
b21e528c60 Protecting Bitrate to avoid data race found by tsan.
TEST=try and vie_auto_test with tsan.
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2163004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4673 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-04 08:42:44 +00:00
mflodman@webrtc.org
65abb6b1ed Revert 4671 "Enable SetInitialPlayoutDelay on Android."
Tests enabled in r4671 failed:
build.chromium.org/p/client.webrtc/builders/Android%20Tests/builds/31/steps/slave_steps/logs/stdio

> Enable SetInitialPlayoutDelay on Android.
> 
> Background:
> In Chrome mirroring which uses 500ms buffering mode,
> audio video mismatch happens in the begining because of the lack of the api.
> 
> BUG=b/10538425
> TEST=pass 'git try' except tests which is aleady broken in the bot. pass 'build/android/test_runner.py gtest -s modules_tests --verbose --release  -f *InitialPlayoutDelayTest*'
> R=henrika@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/2144004

TBR=dwkang@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2160006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4672 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-04 07:47:39 +00:00
dwkang@webrtc.org
310ac91d2a Enable SetInitialPlayoutDelay on Android.
Background:
In Chrome mirroring which uses 500ms buffering mode,
audio video mismatch happens in the begining because of the lack of the api.

BUG=b/10538425
TEST=pass 'git try' except tests which is aleady broken in the bot. pass 'build/android/test_runner.py gtest -s modules_tests --verbose --release  -f *InitialPlayoutDelayTest*'
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2144004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4671 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-04 01:19:12 +00:00
mikhal@webrtc.org
3abb82d8df Suppress video engine test
BUG=2346
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2161005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4670 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-03 22:19:31 +00:00
mikhal@webrtc.org
3c5a9242fe Don't force cont' when enabling kWithErrors
BUG=
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2047004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4669 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-03 20:45:36 +00:00
mikhal@webrtc.org
635b2b88e4 Removing some TODO's from libyuv
BUG=1996
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2146004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4668 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-03 20:06:18 +00:00
mikhal@webrtc.org
2b810bf77b Removing non decodable count from session info: This value isn't used, and therfore can be removed. This is a step towards the refactor of the session info to use maps.
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2143004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4667 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-03 19:09:49 +00:00
fischman@webrtc.org
ccf8b56670 AppRTCDemo(android): prefer ISAC for audio codec.
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2126004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4666 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-03 19:02:11 +00:00
fischman@webrtc.org
8788167b9b PeerConnection Java: explicitly cast DataChannel* to jlong for Java.
Otherwise on 32-bit ARM Android the nativeDataChannel param the Java ctor sees
is a 64-bit value whose low 32 bits are the pointer, and whose high 32-bits are
garbage.

BUG=2302
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2114004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4665 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-03 18:58:12 +00:00
kjellander@webrtc.org
c8c32638be Remove JpegEncoder suppression as jpeg is now removed.
See https://code.google.com/p/webrtc/source/detail?r=4646

BUG=2322
TEST=Ran common_video_unittests with the suppression removed
TBR=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2164004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4664 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-03 18:38:52 +00:00
mflodman@webrtc.org
f5f5da0df1 Adding TSAN suppression for test posix udp transport.
This is race for reading a bool in the WebRTC test UDP transport and
not in any production code.

R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2159006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4663 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-03 14:15:21 +00:00
kjellander@webrtc.org
3a6ff41e8f Document the source of test scenarios for Dummynet wrapper script.
I just wanted to put this in here since I got the question
from an external user.

TEST=none
BUG=none
TBR=phoglund

Review URL: https://webrtc-codereview.appspot.com/2162004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4662 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-03 13:01:31 +00:00
mflodman@webrtc.org
cac7325b84 Adding critsect for child_modules_ in ModuleRtpRtcpImpl::Process() to avoid race with ModuleRtpRtcp::RegisterChildModule.
Found with tsan.

TEST=try job and tsan
R=holmer@google.com

Review URL: https://webrtc-codereview.appspot.com/2156004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4661 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-03 12:11:12 +00:00
pbos@webrtc.org
cb5118c14c Add FakeEncoder to VideoSendStream tests.
Breaks out config part of FakeEncoder from VideoSendStream tests to
FakeEncoder. Also sets FakeEncoder as encoder for VideoSendStream tests.
Anticipated speedup didn't happen as VP8 is still initialized by default
when creating channels in the old API. This will be sped up when moving
off the old API as VP8 won't be enabled by default.

BUG=2312
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2155004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4659 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-03 09:10:37 +00:00
henrik.lundin@webrtc.org
8fb89533af Correcting two nits in InputAudioFile
First, the fread function returns number of elements read, not
necessarily the number of bytes. In this case, it is the number
of samples. Second, a spelling mistake was corrected.

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2161004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4658 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-03 08:43:28 +00:00
mflodman@webrtc.org
8d32066073 Changed method name.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4657 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-02 12:45:30 +00:00
mflodman@webrtc.org
814d5e9133 Renamed method.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4656 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-02 12:45:18 +00:00
mflodman@webrtc.org
d51bcffc1e Function name change.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4655 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-02 12:45:09 +00:00
mflodman@webrtc.org
dfbf52baac Fixing capture frame race in ViECapturer.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4654 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-02 12:44:57 +00:00
kjellander@webrtc.org
5aedb295d5 Add TSan and Dr Memory suppressions for Windows
This works enables us to add add more memory tools bots to the
WebRTC FYI waterfall at
http://build.chromium.org/p/client.webrtc.fyi/waterfall
These suppressions will be needed to get the bots green initially.

This CL also updates the PRESUBMIT.py scripts for the previous
memcheck and TSan suppression directories with the trybots we
currently have. It also adds a PRESUBMIT.py script for the
Dr Memory suppressions.

BUG=1938,2319,2321,2322,2323,2324,2328,2329,2330,2333
TEST=Local execution of the tests passes when these suppressions
are used.

R=niklas.enbom@webrtc.org, phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4653 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-02 11:50:46 +00:00
henrik.lundin@webrtc.org
b3e905cd91 Disable all LS_VERBOSE logging in NetEq4
This reduces exectution time of NetEqDecodingTest.TestBitExactness
with almost 30% and reduces the allocation count (from valgrind)
with almost 50% for the same test.

An issue has been created to re-enable logs when logging performance
is improved; see https://code.google.com/p/webrtc/issues/detail?id=2317.

BUG=1363
TEST=out/Release/modules_unittests --gtest_filter=NetEqDecodingTest.TestBitExactness
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2136004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4652 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-02 09:41:06 +00:00
henrik.lundin@webrtc.org
c487c6abb0 NetEq4: Make the algorithm buffer a member variable
This reduces the alloc count by more than 100,000 for
NetEqDecodingTest.TestBitExactness.

BUG=1363
TEST=out/Release/modules_unittests --gtest_filter=NetEqDecodingTest.TestBitExactness
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4651 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-02 07:59:30 +00:00
wu@webrtc.org
cadf9040cb Update talk to 51664136.
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2148004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4649 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-30 21:24:16 +00:00
pbos@webrtc.org
a957570d62 Overuse detection based on capture-input jitter.
This is believed to be more reliable in real-world cases. The camera seems to fall behind sooner than the encoder starts taking too long time encoding, so this is believed to be an earlier trigger.

BUG=2325
R=asapersson@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2140004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4648 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-30 17:16:32 +00:00
mikhal@webrtc.org
0b960cf7a1 Libjpeg is needed for Libyuv
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2147004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4647 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-30 16:30:53 +00:00
mikhal@webrtc.org
cf61bee5a2 Removing JPEG as it is not used.
BUG= 2322
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2138005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4646 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-30 16:00:08 +00:00
turaj@webrtc.org
45d2840623 Zero comfort noise for stereo insted of assertion.
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2084004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4645 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-30 15:37:08 +00:00
turaj@webrtc.org
3170b5750f Reorder and add critical section to the public method NetEqImpl::PacketBufferStatistics().
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2107005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4644 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-30 15:36:53 +00:00
sergeyu@chromium.org
9ded07e3a4 Fix typo in InvertedDesktopFrame
BUG=279334
R=wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/2141004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4643 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-30 01:05:14 +00:00
kjellander@webrtc.org
bfde359b15 Revert accidental checkin of DEPS
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4642 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-29 13:11:38 +00:00
kjellander@webrtc.org
c520fc9d97 Add svn:ignore on dirs that shouldn't be wiped during gclient revert
This will speed up our buildbots and trybots since it also enables
incremental builds again (currently out/ was wiped for each build).

The Chromium toolchain somehow has changed behavior, since the out/
dir didn't use to be wiped. This change should get faster builds
back again.



git-svn-id: http://webrtc.googlecode.com/svn/trunk@4641 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-29 13:10:21 +00:00
kjellander@webrtc.org
de499669ef Fix fileutils.cc for tests running under Win memory tools.
On Windows, when tests are run under memory tools like
DrMemory and TSan, slashes occur in the path as directory
separators. This makes the fileutils.cc fail to figure out
the project root, making paths to resource files invalid.

By replacing slashes in the path with backslashes for Windows
this can be fixed.

BUG=2318
TEST=running the same command as the bot locally, succeeding after this patch is applied.
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2137004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4640 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-29 11:26:41 +00:00
kjellander@webrtc.org
f8c16b8bf4 Disabling CondVarTest for TSan v2 (take 2)
This test goes into an eternal loop sometimes when run under
TSan v2, therefore I'm disabling it until this is resolved.
Doing that, the TSan v2 bot in our new WebRTC FYI waterfall at
http://build.chromium.org/p/client.webrtc.fyi/waterfall
will hopefully go green.

The define used only exists for TSan v2, so v1 should be
unaffected.

The suppression added in http://review.webrtc.org/2126005/
turns out actually has no effect, so I'm removing it again.
It's probably useless since TSan v2 obviously alters the code's
behavior, causing these errors. Since the errors are not TSan v2
failures, the suppression has no effect.

BUG=2259
TEST=test passing locally when run as described in 2259 (since
no tests are run).

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2115005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4639 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-29 09:47:18 +00:00
dwkang@webrtc.org
b295a3f592 Update SSRC in RtpRtcp for audio channel so that it can have NTP values for further AV sync.
Background:
Since we had http://review.webrtc.org/2048004, the SSRC value in
RtpRtcp for audio hasn't been updated. Because this prevents NTP update in RtpRtcp, the sync logic in ViESyncModule::Process() does not work.

BUG=b/10484087
TEST= pass 'git try' except tests already broken in http://build.chromium.org/p/tryserver.webrtc/console
R=henrika@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2131004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4638 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-29 07:34:12 +00:00
minyue@webrtc.org
d7301775f5 update neteq 4 to facilitate NACK
BUG=
R=turaj@webrtc.org, turajs@google.com

Review URL: https://webrtc-codereview.appspot.com/2008004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4637 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-29 00:58:14 +00:00
kjellander@webrtc.org
8ae641edfb Add suppressions file for Leak Sanitizer.
Add a suppressions file that must exist in order for the Leak Sanitizer
(LSan) bot to be able to run with the default configuration.
This bot is located in the new WebRTC FYI waterfall at:
http://build.chromium.org/p/client.webrtc.fyi/waterfall

More info about LSan on http://www.chromium.org/developers/testing/leaksanitizer

BUG=none
TEST=Running a local master+slave passes fine with this file added locally.
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2135004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4636 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-28 20:29:35 +00:00
kjellander@webrtc.org
5f8d05add2 Disabling CondVarTest for TSan v2.
This test goes into an eternal loop sometimes when run under
TSan v2, therefore I'm disabling it until this is resolved.
Doing that, our new WebRTC FYI waterfall with a TSan v2 bot will hopefully go green when deployed (see https://codereview.chromium.org/23231004)

The define used only exists for TSan v2, so v1 should be
unaffected.

BUG=2259
TEST=test passing locally when run as described in 2259 (since
no tests are run).

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2126005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4635 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-28 19:54:33 +00:00