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Author SHA1 Message Date
mallinath@webrtc.org
0d92ef67c4 Libjingle source code has some spelling mistakes and one of them is "renegotation", which should be "renegotiation". 
This CL is attempting to correct those.

BUG=2810
TBR=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5411 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-22 02:21:22 +00:00
mallinath@webrtc.org
68cbd01216 enabling disabled data channels tests on win32. The real culprit was that ice candidates not included in SDP when there were failure causing transport channels never becoming writable. 
BUG=2799
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5410 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-22 00:16:46 +00:00
andrew@webrtc.org
7de3bb9df9 Output logs to stderr from voe_cmd_test by default. 
Add a flag --log_file which produces the existing behaviour of dumping
logs of all severities to a file. By default, warnings and errors will
now be output to stderr. This is generally more useful for the testing
done with voe_cmd_test.

TESTED=logs output to stderr by default and to the usual file when the
flag is specified.

R=tnakamura@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6849005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5409 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-21 22:17:43 +00:00
henrike@webrtc.org
28da47c52f Android example apps: fixes issue where useful failure information was suppressed. 
BUG=2808
R=andrew@webrtc.org, fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5408 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-21 19:03:51 +00:00
fbarchard@google.com
1d2c034861 libyuv use extern c around jpeg includes. includes fixes to gyp build for intel/mips android, cros arm, ios, and pnacl. 
BUG=none
TESTED=try bots
R=andrew@webrtc.org, jzern@chromium.org

Review URL: https://webrtc-codereview.appspot.com/7179005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5407 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-21 18:27:05 +00:00
sprang@webrtc.org
7dba27c740 Potential dead lock in receive statistics 
A dead lock could occur if the following to code paths are called
concurrently:

ReceiveStatisticsImpl::IncomingPacket() ->
  StreamStatisticianImpl::IncomingPacket()

StreamStatisticianImpl::GetStatistics() ->
  ReceiveStatisticsImpl::StatisticsUpdated()

Solution is to release ReceiveStatisticsImpl lock after lookup/lazy-init of StreamStatisticianImpl. Don't need to hold it when doing the call to StreamStatisticianImpl::IncomingPacket().

BUG=2818
R=asapersson@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5406 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-21 16:33:37 +00:00
elham@webrtc.org
32c3247418 Fix for libtalkmobile build error 
bug=b/12549061

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7329004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5404 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-21 16:16:58 +00:00
henrike@webrtc.org
7ef7df57d8 Removes script for generating supplement.gypi also adds git ignore for tools/gn. 
BUG=N/A
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5403 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-21 15:54:56 +00:00
pbos@webrtc.org
e02d47515f Set up receiver RTX config using a std::map. 
This change removes video_payload_type from RtxConfig as it can be
inferred from the map key or config otherwise. Wiring up this config is
part of issue 2399.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5402 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-20 14:43:55 +00:00
asapersson@webrtc.org
efaeda0c76 Add configuration and test for extended RTCP reference time reports to new video api. 
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5401 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-20 08:34:49 +00:00
henrike@webrtc.org
32c26eb90b Android, OpenSlDemo: moved to webrtc/examples/android/opensl_loopback 
BUG=N/A
R=andrew@webrtc.org, fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5400 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-17 23:12:51 +00:00
jiayl@webrtc.org
4985927d36 Implement screen enumeration and individual screen capturing for Windows. 
BUG=2787
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/7239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5399 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-17 17:19:16 +00:00
henrike@webrtc.org
ead202b973 Android, OpenSlDemo: fixes issue where app would crash as soon as the application is started. 
BUG=2801
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7259005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5398 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-16 23:26:37 +00:00
henrike@webrtc.org
2ce9a64b75 Talk: Removes deprecated example apps and moves the server apps to trunk/talk/examples. 
BUG=12545067
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7159004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5397 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-16 16:49:53 +00:00
henrike@webrtc.org
0af1ffa84d Android, WebRTCDemo: fix issue where changing remote IP was not working properly. 
BUG=2783
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7209004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5396 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-16 16:45:15 +00:00
aluebs@webrtc.org
4ffd9c7423 Add full path to headers 
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5395 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-16 09:01:39 +00:00
bjornv@webrtc.org
6a94734d4d Adds back set_sample_rate_hz() when Init is called in recordings. 
Recordings that had a AnalyzeReverseStream() call prior to ProcessStream() where aborted due to sample rates being set upon call by ProcessStream(). That change was done in r5346.
Before we have a smarter handling on how to set sample rate automatically, this CL adds back that setting.

BUG=
TESTED=trybots, modules_unittests
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7189004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5394 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-16 08:41:09 +00:00
andrew@webrtc.org
ea9392d5eb MIPS optimizations for NS audio processing module 
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4139006

Patch from Ljubomir Papuga <lpapuga@mips.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5393 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-16 07:22:01 +00:00
sergeyu@chromium.org
fb4e256d49 Fix crash in MouseCursor::CopyOf() 
This issue was causing test failures with the latest webrtc roll.

R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7249005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5392 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-16 04:45:35 +00:00
andrew@webrtc.org
8f35afab8c Exclude protoc objects from merge_libs.py. 
BUG=b/12567343
R=wjia@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5391 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-16 00:31:57 +00:00
sergeyu@chromium.org
4b26e2eee3 Update libjingle to 59676287 
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5390 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-15 23:15:54 +00:00
mallinath@webrtc.org
7a2ca7c621 Update needed to MockScreenCapturer after new methods addition to webrtc::ScreenCapturer. 
This change is also must for rolling webrtc in chrome.

R=jiayl@webrtc.org
TBR=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/7219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5389 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-15 19:00:13 +00:00
wu@webrtc.org
8f19cb9fbc Revert 5387 "Re-enable webrtcvoice/videoengine unittests." 
Missed the result from the last try bot.

> Re-enable webrtcvoice/videoengine unittests.
> 
> TEST=try bots
> BUG=
> R=mallinath@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/7149004

TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5388 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-14 22:31:11 +00:00
wu@webrtc.org
eda6823397 Re-enable webrtcvoice/videoengine unittests. 
TEST=try bots
BUG=
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5387 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-14 22:15:09 +00:00
jiayl@webrtc.org
017b619010 Extends the ScreenCapturer interface for individual display screen cast. 
Real implementations for each platform will be added in future CLs.

BUG=2787
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/6819005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5386 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-14 18:26:37 +00:00
wjia@webrtc.org
03cfde2d10 Roll Chromium 238260 -> 243863 
R=andrew@webrtc.org, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5385 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-14 17:48:34 +00:00
andresp@webrtc.org
8c5b27de9a Allow to skip turn by passing ts=false to apprtc. 
R=braveyao@webrtc.org, fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5384 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-14 17:00:23 +00:00
pbos@webrtc.org
39fcfd78ae Remove empty VideoCodecGeneric struct. 
Struct was added prematurely and triggers a warning with
-Wextern-c-compat in latest clang.

R=henrika@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/7119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5383 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-14 12:55:59 +00:00
henrik.lundin@webrtc.org
d9faa46d57 Changing to using factory methods for some classes in NetEq 
In this CL, the Expand, Accelerate and PreemptiveExpand objects are
created using factory methods. The factory methods are injected into
NetEqImpl on creation. This is a step towards implementing a no-decode
operation.

BUG=2776
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6999005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5382 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-14 10:18:45 +00:00
henrika@webrtc.org
aebb1ade9d pRevert 5371 "Revert 5367 "Update talk to 59410372."" 
> Revert 5367 "Update talk to 59410372."
> 
> > Update talk to 59410372.
> > 
> > R=jiayl@webrtc.org, wu@webrtc.org
> > 
> > Review URL: https://webrtc-codereview.appspot.com/6929004
> 
> TBR=mallinath@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/6999004

TBR=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5381 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-14 10:00:58 +00:00
aluebs@webrtc.org
4371d4650a Temporarily disabling some more audio processing tests. 
R=andrew@webrtc.org, bjornv@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7019005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5380 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-14 08:57:22 +00:00
sergeyu@chromium.org
eb31b45aaf Fix MouseCursorMonitorMac to return correct hotspot position. 
Previusly (0, 0) was always return as mouse cursor hotspot.

BUG=2779
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5379 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-13 23:25:17 +00:00
henrike@webrtc.org
3907c2e7e5 Removes the remaining uses of the list wrapper class and the list wrapper class. 
BUG=2164
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7019007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5378 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-13 22:41:34 +00:00
fischman@webrtc.org
dde7aee40f WebRTCDemo: fix out-of-bounds array read. 
Also removed the WebRtcCamera class, which has become an empty wrapper around
CameraInfo in the post-rewrite world.

First pointed out by Jeremy Mao <yujie.mao@webrtc.org> in
http://review.webrtc.org/6869004/

R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5377 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-13 22:15:38 +00:00
fischman@webrtc.org
d7568a08c3 PeerConnection(java): Add OnRenegotiationNeeded support 
Also:
- Make PeerConnectionObserver::OnRenegotiationNeeded() pure virtual to avoid
  this sort of mistake in the future.
- Sprinkle @Override annotations on some callback definitions that were missing
  them.
- Fix a JNI method-signature-lookup typo (s/(V)V/()V/) in PCOJava::OnError()
- Add an explicit ScopedLocalFrameRef to PCOJava::OnError() to match all other
  C++-fired callbacks, for consistency.

BUG=2771
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5376 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-13 22:04:12 +00:00
elham@webrtc.org
ad1863de74 Updated Webrtc version to 3.49 
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5374 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-13 17:49:49 +00:00
henrike@webrtc.org
79cf3acc79 Removes usage of ListWrapper from several files. 
BUG=2164
R=andrew@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5373 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-13 15:21:30 +00:00
andresp@webrtc.org
d0b436a935 Revert "Activate ACM test for Android in modules_tests." (rev5364). 
TBR=turaj@webrtc.org,tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6999006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5372 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-13 13:15:59 +00:00
henrika@webrtc.org
44461fa5cb Revert 5367 "Update talk to 59410372." 
> Update talk to 59410372.
> 
> R=jiayl@webrtc.org, wu@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/6929004

TBR=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5371 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-13 09:35:02 +00:00
aluebs@webrtc.org
8bc4fcfeb6 Temporarily disabling audio processing tests. 
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6889005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5370 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-13 09:14:47 +00:00
henrik.lundin@webrtc.org
2c03bf1641 Increasing simulation time for NetEqPerformanceTest 
This is to get better "signal-to-noise ratio" in the performance bots.
The neteq4-runtime metric is expected to increase by a factor of 10.

BUG=2397
TBR=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6989005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5369 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-13 09:04:23 +00:00
bjornv@webrtc.org
bbd47fc5b5 Enables robust delay validation in AEC delay logging. 
* Explicitly disabled robust validation in AECM.
* Updated audio_processing_unittests for using robust delay validation in AEC.
* Updated output_data_float.pb (not needed for Android nor fixed point, since AECM is untouched).

BUG=None
TESTED=modules_unittests, trybots
R=aluebs@webrtc.org, andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5368 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-13 08:54:34 +00:00
mallinath@webrtc.org
0f3356e20b Update talk to 59410372. 
R=jiayl@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5367 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-11 01:26:23 +00:00
andrew@webrtc.org
023cc5abc7 Minor voice engine improvements around AGC. 
- Remove one unneeded lock in CaptureLevel(), as the call to this
method should always come on the same thread as PrepareDemux().
- Remove check on analog AGC before doing volume calculations. Saves a
bit of code. Instead check if the incoming volume is set to zero, which
is a potentially common occurrence as it indicates no volume is
available.

R=aluebs@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5366 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-11 01:25:53 +00:00
henrike@webrtc.org
573a1b45b5 Android: Fixes crash when exiting WebRTCDemo. 
BUG=2738
R=fischman@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5365 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-10 22:58:06 +00:00
turaj@webrtc.org
7cc64b3747 Activate ACM test for Android in modules_tests. 
TEST=local on Nexus 7.
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5364 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-10 22:35:09 +00:00
pbos@webrtc.org
f777cf2547 Permitting double start/stopping of streams. 
It doesn't make too much sense to hard enforce that the user keeps track
of which streams are started and which are not.

BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5363 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-10 18:47:32 +00:00
henrik.lundin@webrtc.org
a366e810a9 Adding NetEq performance test to webrtc_perf_tests 
The performance test is based on the neteq4_speed_test application. The
bulk of the test code is extracted into a test class, and included into
the neteq_unittest_tools target. The actual gtest that runs the
performance test is implemented in neteq_performance_unittest.cc, and
built as a part of webrtc_perf_tests.

The old stand-alone test application is now made dependent on the new
test class, to avoid code duplication.

BUG=2397
R=andrew@webrtc.org, kjellander@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5362 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-10 08:24:04 +00:00
bjornv@webrtc.org
fa8d534e09 Delay Estimator: Adds unittests for robust validation. 
In addition to unittests a cast losing constness was corrected.
The tests added are:
1. Adjusting allowed_offset when robust validation is disabled should have no impact.
2. For noise free signals there should be no difference between robust validation or not.
3. Robust validation acts faster during startup.

BUG=None
TESTED=modules_unittests, trybots
R=aluebs@webrtc.org, andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5361 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-10 07:42:07 +00:00
sergeyu@chromium.org
4625df3e3e Fix NaCl compilation 
nethelpers.cc was using LOG() but didn't include logging.h

R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6829005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5360 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-09 21:26:50 +00:00