turaj@webrtc.org
78f0db4710
Fix the break caused by r5579.
...
TBR=tlegrand@google.com
BUG=
Review URL: https://webrtc-codereview.appspot.com/8939004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5581 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-19 23:07:31 +00:00
henrike@webrtc.org
571df2dca9
Update libjingle 61759961->61834300
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5580 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-19 23:04:26 +00:00
turaj@webrtc.org
c2d69d3229
Resolves memcheck issue in AudioCodingModuleTest. The issue is coditional jumnp based on uninitialized variable.
...
BUG=2944
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5579 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-19 20:31:17 +00:00
jiayl@webrtc.org
97e7a640d8
Make WindowCapturerLinux handling window resize events.
...
We need to re-initialize the XServerPixelBuffer to the new size
when a window resize event is received.
BUG=https://code.google.com/p/chromium/issues/detail?id=339953
R=sergeyu@chromium.org , wez@chromium.org
Review URL: https://webrtc-codereview.appspot.com/8679004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5578 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-19 17:28:41 +00:00
andresp@webrtc.org
242102517d
Added architecture for compiling under chrome NaCl.
...
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5577 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-19 13:55:02 +00:00
tina.legrand@webrtc.org
056287eee0
This CL separate all ACM tests with new and old implementation of ACM and NetEq. The reason is to debug an issue with failure on Android try bots. We need to see if the error only occurs with the new ACM/NetEq, or if it is a flakiness that affects both.
...
BUG=issue2874
R=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5576 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-19 13:45:54 +00:00
asapersson@webrtc.org
8098e07478
Add RTCP packet type counter (for getting statistics such as sent/received NACK and FIR).
...
Add counter to RTCP sender and RTCP receiver.
Add video api GetRtcpPacketTypes().
BUG=2638
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8179004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5575 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-19 11:59:02 +00:00
henrika@webrtc.org
b7a91fa95a
Removes VoERTP_RTCP::InsertExtraRTPPacket.
...
Reasons for removing:
- Feels like a complete hack IMHO.
- Not used by any client.
- Unclear functionality regarding time stamp, marker bit etc.
- Causes several issues in tests due to a bad design which mainly depends on the fact that this API "breaks" an ongoing data/packet flow and it complicates the threading model and creates risks for deadlock and memory corruption. Not worth trying to fix given the very unclear benefit of maintaining the API. Better to remove the API instead.
- We also see lots of TSan races and memcheck errors related to this API.
BUG=2296,2240
R=mflodman@webrtc.org , niklas.enbom@webrtc.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5574 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-19 08:58:08 +00:00
sergeyu@chromium.org
e384104166
Fix DesktopAndCursorComposer not to crash
...
DesktopAndCursorComposer was crashing when screen/window
capturer returns a NULL frame due to an error.
BUG=crbug.com/344093
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8769004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5573 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-18 23:26:34 +00:00
henrike@webrtc.org
5cf3e8f0f0
(Auto)update libjingle $LAST_P10_REVISION-> $NEW_P10_REVISION
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5572 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-18 22:28:52 +00:00
andrew@webrtc.org
27c6980239
Move the volume quantization workaround from VoE to AGC.
...
Voice engine shouldn't really have to manage this. Instead, have AGC
keep track of the last input volume, so that it can avoid getting stuck
under coarsely quantized conditions.
Add a test to verify the behavior.
TESTED=unittests, and observed that AGC didn't get stuck on a MacBook
where this problem can actually occur.
R=bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8729004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5571 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-18 20:24:56 +00:00
solenberg@webrtc.org
00844d7bef
Remove obsolete voe_unit_test.
...
BUG=
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8839004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5570 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-18 18:50:50 +00:00
fischman@webrtc.org
358e3367a3
PeerConnection(java): enable HW encoder on N5 for standalone build.
...
Now that bug 2899 is fixed (r5562) packet-loss is recoverable. Yay.
BUG=2575
R=noahric@google.com
Review URL: https://webrtc-codereview.appspot.com/8869004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5568 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-18 17:29:37 +00:00
fischman@webrtc.org
c2d75e0708
PeerConnection(java): account for thread shutdown vagaries.
...
Android's JVM requires threads to detach before they exit, but ONLY if
they needed to AttachCurrentThread. Conversly, threads that were
attached by the JVM (e.g. the result of making a native call from Java)
must NOT be detached by the application. This is bug 2441.
The fix for the above is to only pthread_setspecific() for threads that
Attach(), not for already-attached threads. To ensure that we only
detach Attached threads, added a GetEnv() call to ThreadDestructor(),
which revealed that Oracle's JVM can overly-eagerly clear TLS accounting
data, effectively detaching threads without their consent at shutdown.
Work around this with a specific check.
To guard against (some) regression, added a variant of PeerConnectionTest
that runs on a non-main thread. This revealed a bug in LinuxDeviceManager
which implicitly assumes its talk_base::Thread has already been
initialized. Fixed that here too.
BUG=2441
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8759004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5567 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-18 16:57:36 +00:00
mflodman@webrtc.org
c320027d6a
Don't print a warning if RTPPacketHistory::SetStorePacketStatus is called
...
twice with the same settings.
Without this change, setting up a call with the new video API will
print a trace warning.
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8859004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5566 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-18 14:51:00 +00:00
turaj@webrtc.org
2086e0fbf3
Remove unnecessary warnings.
...
BUG=
TEST=try job
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8719005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5565 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-18 14:22:20 +00:00
solenberg@webrtc.org
a07923339b
Remove external encryption API for VoE.
...
BUG=
R=henrika@webrtc.org , henrikg@webrtc.org , phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5564 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-18 11:27:22 +00:00
kjellander@webrtc.org
0a9d822812
Change mime type to text/html for multiple-relay.html
...
R=hta@chromium.org
Review URL: https://webrtc-codereview.appspot.com/8809005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5563 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-18 08:45:13 +00:00
sprang@webrtc.org
346094cb01
Incorrect overhead calculation when using FEC + RTP extension headers.
...
When frames are fragmented inte multiple RTP packets in order to not
exceed a maximum packet size, the header overhead calculation must
take into account that FEC redundancy packets may use more than the
12 bytes of the basic RTP header. For example, a csrc list or extension
headers may be present.
BUG=2899
R=phoglund@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8769005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5562 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-18 08:40:33 +00:00
asapersson@webrtc.org
b60346e951
Reset estimate if no frame has been seen for a certain time (to avoid large jitter if stop sending).
...
Add delay before start processing after a reset.
BUG=1577
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8699006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5561 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-17 19:02:15 +00:00
mallinath@webrtc.org
92fdfebedd
Update talk to 61699344.
...
TBR=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5560 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-17 18:49:41 +00:00
mflodman@webrtc.org
e3842897e2
Adding tsan suppression for error introduced in r5555, causing libjingle_unittest to fail on TSan bot.
...
BUG=2931
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8779005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5559 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-17 15:09:39 +00:00
henrik.lundin@webrtc.org
340746aa13
Misc small nits in NetEq
...
Fixing a few small things found recently. This is mostly cosmetics.
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8749005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5558 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-17 11:37:16 +00:00
hta@webrtc.org
1009798b31
Demo of multi-pass encode - used for testing limits.
...
This demo creates a sequence of PeerConnections, and passes
a videostream through all of them.
This allows one to test how many PeerConnections and how
many encodes/decodes the implementation will support before
breaking down or slowing down enough to be unusable.
BUG=
R=fischman@webrtc.org , hta@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5557 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-15 06:13:41 +00:00
andrew@webrtc.org
f92aaff104
AudioProcessing is not a Module.
...
Remove Module as the base class of AudioProcessing. The inherited
methods were all no-ops.
TBR=bjornv
Review URL: https://webrtc-codereview.appspot.com/8779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5556 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-15 04:22:49 +00:00
henrike@webrtc.org
b8c254abd6
(Auto)update libjingle 61549749-> 61608469
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5555 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-14 23:38:45 +00:00
bjornv@webrtc.org
e2fc13e42f
Refactoring common_audio/signal_processing: Removed two macros used by isac only.
...
Removed a macro for malloc() and one for free(). They are only used by the audio codec isac, where I replaced the macro with its implementation.
Further, the includes were updated with full paths and put in alphabetical order.
BUG=N/A
TESTED=trybots,module_tests,module_unittests
R=turaj@webrtc.org , turajs@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8449004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5554 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-14 23:12:34 +00:00
fischman@webrtc.org
c5d506a106
AppRTCDemo(android): clarified README on how to launch app using adb.
...
TBR=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8689005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5553 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-14 17:55:13 +00:00
stefan@webrtc.org
505f2a0348
Disabling WebRtcSessionTest.TestIceStatesBundle under memcheck.
...
BUG=2924
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8699005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5552 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-14 12:38:06 +00:00
stefan@webrtc.org
9075d519a2
Adding a critical section missing in r5543.
...
This fixes a race caught by the linux tsan bot.
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8739004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5551 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-14 09:45:58 +00:00
fischman@webrtc.org
a3708ecdfe
PeerConnectionTest(java): unbreak following 61460797-p10
...
BUG=1414
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5550 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-14 01:51:33 +00:00
mallinath@webrtc.org
385857dfd4
Update talk to 61549749.
...
TBR=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8709004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5549 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-14 00:56:12 +00:00
wu@webrtc.org
b9a088b920
Update talk to 61538839.
...
TBR=mallinath
Review URL: https://webrtc-codereview.appspot.com/8669005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5548 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-13 23:18:49 +00:00
wu@webrtc.org
0de29504ab
Revert 5545 "Update libjingle to 61514460"
...
> Update libjingle to 61514460
>
> TBR=tommi@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/8649004
TBR=xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5547 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-13 19:54:28 +00:00
andrew@webrtc.org
38bf249049
Initialize output_will_be_muted_.
...
TBR=aluebs
Review URL: https://webrtc-codereview.appspot.com/8659004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5546 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-13 17:43:44 +00:00
xians@webrtc.org
e749c9ebdb
Update libjingle to 61514460
...
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5545 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-13 15:09:40 +00:00
asapersson@webrtc.org
8f690bc222
Increase overuse and normal use thresholds for Mac.
...
BUG=1577
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8629004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5544 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-13 14:43:18 +00:00
stefan@webrtc.org
ae2563ae2f
Fixes a race when writing to send_padding_.
...
TEST=trybots
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8619004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5543 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-13 13:48:38 +00:00
kjellander@webrtc.org
12cb88cab9
Add check to verify tree is open to PRESUBMIT.py.
...
This will disallow commits when our tree is closed.
BUG=chromium:342743
TEST=ran git cl presubmit with an open tree (no error). Then I closed the tree at http://webrtc-status.appspot.com and ran it again, got this message:
Tree state is: closed
***************
Tree is temporarily closed (testing presubmit hook real quick)
http://webrtc-status.appspot.com/current?format=json
***************
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8609004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5542 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-13 11:53:43 +00:00
henrik.lundin@webrtc.org
fcfc6a990e
Small refactoring of NetEq unittest for CNG with clock drift
...
Converting the test to a method within the test fixture, and setting
up two tests that call this method. One for positive and one for
negative clock drift. The one with positive clock drift is disabled
for now since it does not pass, but will be re-enabled shortly.
This change is only made for NetEq4.
R=tlegrand@google.com
Review URL: https://webrtc-codereview.appspot.com/8599004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5541 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-13 11:42:28 +00:00
fischman@webrtc.org
3eda643a91
PeerConnection(java): added MediaConstraints support to AudioSource, now fed to AudioTrack.
...
BUG=2912
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5540 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-13 04:01:04 +00:00
fischman@webrtc.org
540acde5b3
PeerConnection(java): use MediaCodec for HW-accelerated video encode where available.
...
Still disabled by default until https://code.google.com/p/webrtc/issues/detail?id=2899 is resolved.
Also (because I needed them during development):
- make AppRTCDemo "debuggable" for extra JNI checks
- honor audio constraints served by apprtc.appspot.com
- don't "restart" video when it hasn't been stopped (affects running with the
screen off)
BUG=2575
R=noahric@google.com
Review URL: https://webrtc-codereview.appspot.com/8269004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5539 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-13 03:56:14 +00:00
andrew@webrtc.org
17342e5092
Add a method to inform AudioProcessing that its output will be muted.
...
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5538 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-12 22:28:31 +00:00
jiayl@webrtc.org
de782180b0
Change the type of propagation delta from int64 to int.
...
The delta value never exceeds the range of int. Changing it to int will save memory and copying cost.
BUG=2910
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8549004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5537 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-12 19:19:23 +00:00
andrew@webrtc.org
07b5950c12
Initialize key_pressed_.
...
Was resulting in an error on Mac Asan:
[ RUN ] ApmTest.DebugDump
[libprotobuf FATAL ../../third_party/protobuf/src/google/protobuf/message_lite.cc:224] CHECK failed: !coded_out.HadError():
TBR=aluebs
Review URL: https://webrtc-codereview.appspot.com/8539004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5536 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-12 16:41:13 +00:00
andrew@webrtc.org
ce8e077cf0
Add a keypress field to the audioproc debug proto.
...
Log the value in AudioProcessing, and unpack it to a new file in the
unpacking tool.
TESTED=
- The new tool can unpack old dumps.
- The old tool can unpack new dumps (without keypress.bool).
- Unpacking a new dump from voe_cmd_test produces a keypress.bool that
appears correct when examined.
R=aluebs@webrtc.org , bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8509005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5535 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-12 15:28:30 +00:00
pbos@webrtc.org
8118f1861f
Set pacing bitrates in SetEncoder.
...
Before the change no padding was allowed before the first remote bitrate
estimation was received. This bitrate estimation is based on what's
actually sent. In tests I set codec.startBitrate to 300 instead of
325, which incidentally means that only the first layer gets encoded.
As we only send ~150kbps instead of 300, the first REMB will
significantly pull down the remote bitrate estimate instead of keeping
the existing rate, even though there's no problem with the link.
This was detected in RampUpTest.PacingWithRtx,
(send_config.codec.startBitrate=300), which caused the tests to become a
lot slower, and flake out more. By allowing padding initially we're able
to keep our initial bitrate estimate.
R=stefan@webrtc.org
TEST=Running RampUpTest.WithPacingAndRtx with startBandwidth=300.
BUG=
Review URL: https://webrtc-codereview.appspot.com/8529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5534 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-12 14:50:29 +00:00
solenberg@webrtc.org
67e70442b5
Remove unused and not working voe_extended_test.
...
BUG=2913
R=henrikg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5533 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-12 09:58:49 +00:00
pbos@webrtc.org
5591046ab1
.gitignore: + /third_party/{clang_format,usrcsctp}
...
clang_format and usrcsctp are both synced in through gclient and should
be suppressed.
BUG=
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8159004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5532 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-12 09:33:22 +00:00
jiayl@webrtc.org
14d80793a8
PeerConnectionClient needs to initialize SSL.
...
BUG=2911
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8499004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5531 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-12 00:41:59 +00:00