pbos@webrtc.org
6e98ef4b35
Fix deadlock in RegisterPreDecodeImageCallback.
...
Fixes lock-order inversion between ViEChannel::callback_cs_ and
VideoReceiver::_receiveCritSect detected on DrMemory Full which
exhibited different timing behavior.
Also removes most of the suppressions on DrMemory Full as they're able
to run again without deadlocking.
BUG=3336,3375
TEST=Run DrMemory Full trybots.
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20499004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6228 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-23 09:41:07 +00:00
pbos@webrtc.org
bc524ae41a
Added mirror of gtest-parallel.
...
gtest-parallel is a Python script that runs gtest binaries in parallel.
R=andrew@webrtc.org , kjellander@webrtc.org , mflodman@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/11309004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6227 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-23 09:37:29 +00:00
stefan@webrtc.org
b60bfe4759
Suppress webrtc trace races detected by tsan.
...
BUG=3372
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6226 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-23 07:29:37 +00:00
wu@webrtc.org
10f871f29b
Remove the restriction to allow having both webrtc and talk changes in the same cl.
...
This restriction is no longer needed as the auto sync script can handle changes to both folder in same commit correctly.
BUG=
R=andrew@webrtc.org , henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15479006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6225 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-22 22:35:46 +00:00
tnakamura@webrtc.org
0720758f9f
Bump WebRTC version number to 3.54
...
TBR=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17619006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6222 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-22 17:51:18 +00:00
henrike@webrtc.org
1bb5da04fe
Adds missing include of assert header.
...
BUG=3380
R=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/14569008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6221 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-22 14:31:14 +00:00
braveyao@webrtc.org
21f7d6d2fe
WebRTCDemo: move the deletion of CritSect to end of the dtor to fix a crash in Android video renderer.
...
BUG=3368
TEST=Manual Test
Review URL: https://webrtc-codereview.appspot.com/21519005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6220 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-22 02:57:55 +00:00
mallinath@webrtc.org
8e755c1ad2
Connect SignalDestroyed in AllocationSequence after TURN ports are destroyed
...
when TURN ports are using shared socket with UDP port.
This is required as AllocationSequence maintains a map of turn ports. If the
ports are destroyed without the knowledge of AllocationSequence, sequence will
try to deliver packets to the destoyed ports.
R=jiayl@webrtc.org
BUG=https://code.google.com/p/chromium/issues/detail?id=368877
Review URL: https://webrtc-codereview.appspot.com/14569007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6219 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-21 23:00:46 +00:00
henrike@webrtc.org
88fbb2d86b
Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
...
Same as https://webrtc-codereview.appspot.com/19519004 . The issue in
http://chromegw.corp.google.com/i/internal.chromium.webrtc.fyi/builders/Linux ...
is solved by this change
http://src.chromium.org/viewvc/chrome/trunk/src/third_party/libjingle/libjing ...
(tested locally).
BUG=3380
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17619005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6218 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-21 21:18:46 +00:00
henrike@webrtc.org
99b4162ccf
Rebase webrtc/base 6163:6216 (svn diff -r 6163:6216 http://webrtc.googlecode.com/svn/trunk/talk/base , apply diff manually)
...
BUG=3379
TBR=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17619004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6217 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-21 20:42:17 +00:00
buildbot@webrtc.org
f9f1bfbdae
(Auto)update libjingle 67686255-> 67689476
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6216 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-21 17:02:15 +00:00
henrike@webrtc.org
a148704b4b
Rename webrtc/base's IS_ALIGNED macro to RTC_IS_ALIGNED to avoid conflict between webrtc/base/basictypes.h and third_party/.../vpx_codec.h.
...
BUG=3380
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17579005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6215 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-21 16:52:14 +00:00
buildbot@webrtc.org
ce4201df52
(Auto)update libjingle 67643194-> 67686255
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6214 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-21 16:22:51 +00:00
jiayl@webrtc.org
7ca277b574
Initializes WINDOWPLACEMENT::length in GetCroppedWindowRect.
...
BUG=https://code.google.com/p/webrtc/issues/detail?id=3196
R=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/21529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6213 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-21 16:02:31 +00:00
henrike@webrtc.org
000658a138
Revert of 6211 as it was committed despite of PRESUBMIT.py warning. The commit breaks the sync bot.
...
BUG=N/A
TBR=mcasas@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21519006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6212 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-21 16:01:13 +00:00
mcasas@webrtc.org
3b7e282caa
Disabling systematically failing
...
WebRtcVideoMediaChannelTest.SendVp8HdAndReceiveAdaptedVp8Vga
TBR= pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14569006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6211 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-21 14:25:20 +00:00
mcasas@webrtc.org
2fa7f79094
Revert 6202 "Switch to using base/constructormagic.h and remove ..."
...
> Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
>
> BUG=N/A
> R=andrew@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/19519004
TBR=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14579007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6210 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-21 11:07:29 +00:00
mcasas@webrtc.org
c2213b6a0f
Revert 6208 "Patch from henrike@webrtc.org"
...
Wasn't enough. I'll have to revert the whole rev 6202.
> Patch from henrike@webrtc.org
> https://code.google.com/p/webrtc/source/detail?r=6202
> didn't work for at least one file and broke most of
> the compile steps in the FYI bots. The file is reverted
> here.
>
> TBR= henrike@webrtc.org , sergeyu@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/17609004
TBR=mcasas@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14579006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6209 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-21 10:03:09 +00:00
mcasas@webrtc.org
86df8acc92
Patch from henrike@webrtc.org
...
https://code.google.com/p/webrtc/source/detail?r=6202
didn't work for at least one file and broke most of
the compile steps in the FYI bots. The file is reverted
here.
TBR= henrike@webrtc.org , sergeyu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17609004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6208 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-21 08:40:56 +00:00
braveyao@webrtc.org
1a79bb8d30
WebRTCDemo: clean the error message due to API clean up and add ability to route the audio through all three outputs, headset/earpiece/loudspeaker
...
BUG=3366
TEST=Manual Test
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6207 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-21 03:37:45 +00:00
buildbot@webrtc.org
49a6a27bf0
(Auto)update libjingle 67555838-> 67643194
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6206 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-21 00:24:54 +00:00
wu@webrtc.org
82c4b8531c
Calculate capture ntp timestamp in local timebase for decoded audio frame.
...
BUG=3111
R=stefan@webrtc.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19449005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6205 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-20 22:55:01 +00:00
henrik.lundin@webrtc.org
48438c2c90
Enabling NetEq bit-exactness test for Win x64
...
A new reference file (neteq4_universal_ref_win_64.pcm) was generated and
uploaded.
Also removing the old hack to have different reference files
for different version of Visual Studio. The test is now only supporting
VS 2012 and later (_MSC_VER >= 1700). This makes the windows 32-bit
output identical to the generic reference file
(neteq4_universal_ref.pcm), so the specialized one
(neteq4_universal_ref_win_32.pcm) could have been removed. However,
since the resources sync mechanism does not include removing of old
files, a client could pick up the old reference and fail. Therefore,
this cl also updates neteq4_universal_ref_win_32.pcm to be identical to
neteq4_universal_ref.pcm.
BUG=1458
R=kjellander@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14569005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6204 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-20 16:07:43 +00:00
henrik.lundin@webrtc.org
aed31fe8ab
Modifying WATCHLISTS
...
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6203 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-20 16:05:47 +00:00
henrike@webrtc.org
125ffd709d
Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
...
BUG=N/A
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19519004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6202 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-20 15:20:44 +00:00
stefan@webrtc.org
4059c2f579
Disable CallPerfTest.CaptureNtpTimeWithNetworkDelay due to being flaky.
...
TBR=wu@webrtc.org
BUG=3374
Review URL: https://webrtc-codereview.appspot.com/14579005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6201 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-20 15:12:29 +00:00
stefan@webrtc.org
70bb2d5755
Revert r6198 "Expose the original packet length in in the RTP play tools."
...
TBR=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6200 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-20 13:25:48 +00:00
stefan@webrtc.org
83599cba77
Reenable WebRtcVideoEngineTestFake.SendReceiveBitratesStats under DrMemory.
...
The uninitialized read has been fixed. Suppressing CL: https://code.google.com/p/webrtc/source/detail?r=6073
BUG=11288120
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14529005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6199 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-20 13:16:35 +00:00
stefan@webrtc.org
e208458643
Expose the original packet length in in the RTP play tools.
...
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6198 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-20 13:09:16 +00:00
stefan@webrtc.org
be4ab99a53
Disabling RealFFTTest.RealAndComplexMatch and AudioProcessingTest.Formats as they currently are broken with gcc 4.8.
...
BUG=3370
R=bjornv@webrtc.org , kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14569004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6197 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-20 12:42:01 +00:00
henrik.lundin@webrtc.org
a36db970bd
Suppress GMOCK printouts from TestVideoSenderWithVp8
...
Adding a missing EXPECT_CALL.
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20529005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6196 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-20 11:16:10 +00:00
bjornv@webrtc.org
f3e1341da7
VoEVolumeTest: Enabled Linux flaky tests
...
Fixed error checks only on Linux to be able to turn on flaky tests. The cause of flaky failures is due to late values in pulse audio.
Related (deleted) CLs:
https://webrtc-codereview.appspot.com/19469007/
https://webrtc-codereview.appspot.com/19469004/
BUG=367
TESTED=trybots, voe_auto_test repeated
R=henrikg@webrtc.org , tina.legrand@webrtc.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6195 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-20 10:43:42 +00:00
asapersson@webrtc.org
a826006132
Add NACK and RPSI packet types to RTCP packet builder.
...
Fixes bug found when parsing received RPSI packet.
BUG=2450
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6194 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-20 09:53:51 +00:00
minyue@webrtc.org
2db9f45038
Reduce flakiness of voe_auto_test MixingTest by checking dumped audio size
...
BUG=webrtc:2925
TEST=passed_all_trybots
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18479005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6193 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-20 08:33:30 +00:00
tkchin@webrtc.org
1732a591e7
Add a UIView for rendering a video track.
...
RTCEAGLVideoView provides functionality to render a supplied RTCVideoTrack using OpenGLES2.
R=fischman@webrtc.org
BUG=3188
Review URL: https://webrtc-codereview.appspot.com/12489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6192 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-19 23:26:01 +00:00
tkchin@webrtc.org
7ca1edb31d
Remove IOKit linkage from iOS builds.
...
IOKit has been removed in iOS7, so link fails. iOS build succeeds after removing this setting and the corresponding one in talk/libjingle.gyp. Presubmit script tells me that CLs aren't allowed to touch both talk/ and webrtc/ at the same time so doing this separately.
BUG=
R=fischman@webrtc.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20509005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6191 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-19 21:05:10 +00:00
fischman@webrtc.org
40bc7779aa
talk_base: remove lock inversion between MessageQueue and MessageQueueManager.
...
Removes the concept of a MessageQueue being "active" in favor of considering all
live MQ's to be active.
(previously a MQ was active starting from the first Post to it and stopped being
active in its dtor).
BUG=3230
R=sriniv@google.com
Review URL: https://webrtc-codereview.appspot.com/21489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6190 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-19 17:58:04 +00:00
wu@webrtc.org
cb711f77d2
Add interface to propagate audio capture timestamp to the renderer.
...
BUG=3111
R=andrew@webrtc.org , turaj@webrtc.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12239004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6189 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-19 17:39:11 +00:00
pbos@webrtc.org
ebb467fdc8
Avoid NACK-list flush error on keyframe packets.
...
Receiver code used to indicate a flush error even if the incoming packet
is a keyframe, forcing a request of a keyframe. Now it takes this
keyframe into account and doesn't error as the stream is decodable from
this point.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15549005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6188 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-19 15:28:02 +00:00
stefan@webrtc.org
64339a7069
Don't crash if a frame returned from the decoder is too old.
...
BUG=crbug/371805
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6187 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-19 13:31:35 +00:00
michaelbai@google.com
725e582461
Use the new gyp_var_prefix local variable set by gyp instead of the
...
global GYP_VAR_PREFIX set by the makefiles, since the latter is not
guaranteed to still be the same value at the time the command is
executed. Also, use abspath instead of realpath to convert paths to
absolute, since realpath expands to the empty string if the target file
doesn't exist, complicating build debugging.
BUG=
R=andrew@webrtc.org , torne@chromium.org
Review URL: https://webrtc-codereview.appspot.com/12559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6186 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-16 17:56:10 +00:00
henrike@webrtc.org
14abcc7322
libvpx's UNUSED macro conflicts with webrtc/base's. Added missing include of assert.h. Globally defined function "Unused" in talk/base and its copy (webrtc/base) is causing a conflict.
...
libvpx macro (UNUSED) can be found here:
http://src.chromium.org/viewvc/chrome/trunk/deps/third_party/libvpx/source/libvpx/vpx/vpx_codec.h
BUG=N/A
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6185 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-16 16:54:44 +00:00
bjornv@webrtc.org
a3b5673879
common_audio/signal_processing: Removes macro WEBRTC_SPL_UMUL_RSFT16
...
This macro was only used on two lines in iSACfix and I replaced those with the operations the macro performed.
BUG=3348
TESTED=trybots, manual unittests
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6184 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-16 12:11:20 +00:00
pbos@webrtc.org
1e019d10b8
Fix delivery error-checking missed in r6151.
...
Gets rid of quite a bit of false-warning logging in WebRtcVideoEngine2.
BUG=3228
R=perkj@webrtc.org
TBR=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6183 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-16 11:38:45 +00:00
solenberg@webrtc.org
57e060251a
Fix flaky test SendRtpRtcpHeaderExtensionsTest.SentPackets*.
...
Flakiness was caused by a race condition between two atomic integers shared by two threads. Fixed by counting bad packets (those not containing the expected extension) instead of the good packets.
The CL also eliminates another possible flake by introducing a test fixture which doesn't automatically start sending audio packets when constructed.
BUG=3340,3356
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14499004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6182 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-16 11:27:09 +00:00
andresp@webrtc.org
60015d27ae
Wire up --force_fieldtrials for vie_auto_test and for test targets linking with test/test.gyp:{test_main|test_support_main}
...
This allows use of webrtc field trials and opens up the possibility to try the different code paths when running the unit tests by wiring them up to a --force_fieldtrials.
Tested: running a test target that links with the above with a flag --force_fieldtrials=invalid leads the test to crash.
BUG=crbug/367114
R=mflodman@webrtc.org , phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6181 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-16 09:39:51 +00:00
bjornv@webrtc.org
1b21a57902
common_audio/signal_processing: Removed macro WEBRTC_SPL_SUB_SAT_W16
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Macro was only mapping a function used in one place.
BUG=3348
TESTED=trybots, unittests
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17499004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6180 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-16 06:40:31 +00:00
bjornv@webrtc.org
d83d607271
common_audio/signal_processing: Removed macro WEBRTC_SPL_MAX_SEED_USED
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* Moved the macro to randomization_functions and made it static const.
* Made WebRtc_IncreaseSeed() static, since it is not used outside this function.
* Style guide changes.
BUG=3348,3353
TESTED=trybots, common_audio_unittests, modules_unittests, modules_tests
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6179 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-16 06:38:47 +00:00
wu@webrtc.org
75718cf80a
* Implement WindowsRealTimeClock::CurrentTimeVal with GetSystemTimeAsFileTime as it supposes to return a POSIX gettimeofday, so that later it can be converted to NTP timee correctly.
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* Updated to ClockTest.NtpTime to verify the returned NTP is at least larger than kNtpJan1970.
Current implementation uses timeGetTime, which returns the time since windows is started, which can't be converted to NTP time.
BUG=3325
R=pwestin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6178 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-15 23:54:14 +00:00
henrike@webrtc.org
bf58a75dd9
removed webrtc_base_tests_utils from merge libs as it was breaking some builds.
...
BUG=N/A
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6177 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-15 21:45:09 +00:00