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Author SHA1 Message Date
pbos@webrtc.org
26b0d77baf Suppress RTPSender race regardless of codec. 
New test uses SendGeneric instead of SendVP8.

BUG=2349
TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2194004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4705 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-09 15:34:36 +00:00
pbos@webrtc.org
841c8a44bb Rename VideoCall to Call. 
Call should encompass more than video, there's no point in calling it
VideoCall.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2191005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4704 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-09 15:04:25 +00:00
solenberg@webrtc.org
86136a0e8f Re-enable tests for Remote Bitrate Estimator 
BUG=
R=stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4703 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-09 13:06:52 +00:00
pbos@webrtc.org
0181b5f8dd ExternalVideoDecoder for new VideoEngine API. 
Implements the ExternalVideoDecoder interface for VideoReceiveStream.
Also adds a FakeDecoder used in tests, removing the overhead of running
the EngineTest tests with VP8 under Memcheck/TSan, allowing us to enable
them under Memcheck/TSan as well.

BUG=2346,2312
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2172004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4702 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-09 08:26:30 +00:00
pbos@webrtc.org
30e055c4dd Handle empty RTP video packets agnostic to codec. 
Sending empty RTP packets caused a crash when using a generic codec
instead of VP8. This fix moves handling of empty RTP packets out of
ReceiveVp8Codec and into ParseRtpPacket.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2185004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4701 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-08 11:15:00 +00:00
mallinath@webrtc.org
1b476d9a56 Disabling channelmanager unittest. This test is causing 
TSAN error. The problem could be in thread Invoke method.

TBR=wu@webrtc.org
BUG=https://code.google.com/p/webrtc/issues/detail?id=2355

Review URL: https://webrtc-codereview.appspot.com/2190004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4700 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-07 18:59:12 +00:00
mallinath@webrtc.org
ab5a0912a3 Fixing the build error on Windows. 
Problem is in coversion from size_t to int.

TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2189004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4698 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-07 00:12:57 +00:00
mallinath@webrtc.org
1b15f4226f Update talk to 51960985. 
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2188004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4696 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-06 22:56:28 +00:00
andrew@webrtc.org
b159c2e3dd Reduce cost of PushSincResampler::Resample(). 
Ideally, PushSincResampler would have very little overhead on
SincResampler. This gets closer to that ideal.

Replace std::min/max and floor with inline functions. Add a benchmark
test to verify the improvement.

On a MacBook Retina, this results in PushSincResampler::Resample()
accounting for ~1% of CPU usage on voe_cmd_test vs the earlier ~2%
(with ISAC16 and 48 kHz audio devices).

Using the new benchmark, this results in a performance improvement of:
16 -> 44.1 : 1.7x
16 -> 48   : 1.9x
32 -> 44.1 : 1.6x
32 -> 48   : 1.7x
44.1 -> 16 : 1.5x
44.1 -> 32 : 1.7x
44.1 -> 48 : 1.7x
48 -> 16   : 1.5x
48 -> 32   : 1.5x
48 -> 44.1 : 1.8x

R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2157005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4695 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-06 21:15:55 +00:00
fischman@webrtc.org
c7f708679d Clamp camera id to legal values. 
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2184004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4694 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-06 18:17:45 +00:00
stefan@webrtc.org
b2c8a952a7 Improving padding rules and breaking out bw allocation to ViEEncoder. 
BUG=1837
TESTS=vie_auto_test --automated, trybots
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2170004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4693 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-06 13:58:01 +00:00
stefan@webrtc.org
7bb8f02274 Adds support for combining RTX and FEC/RED. 
This is accomplished by breaking out RTX and FEC/RED functionality from the RTP module and keeping track of the base payload type, that is the payload type received when not receiving RTX.

Enables retransmissions over RTX by default in the loopback test.

BUG=1811
TESTS=voe/vie_auto_test --automated and trybots.
R=mflodman@webrtc.org, pbos@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2154004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4692 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-06 13:40:11 +00:00
andresp@webrtc.org
5500d93fe5 Add temporal layer factory. 
R=marpan@google.com, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2180004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4691 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-06 11:26:15 +00:00
fischman@webrtc.org
016eec0983 Unbreak build by adding new mandatory ICE username param. 
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2182004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4689 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-05 23:11:55 +00:00
mikhal@webrtc.org
f1e807c0e5 Removing FrameForStorage 
R=pwestin@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2142004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4688 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-05 22:34:41 +00:00
fischman@webrtc.org
c31d4d0324 AppRTCDemo(iOS): prefer ISAC as audio codec 
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
 2013-09-05 21:49:58 +00:00
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