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Author SHA1 Message Date
mallinath@webrtc.org
82a846f0cb Adding Ami to the video renderer and capturer modules. 
TBR=fischman@webrtc.org,wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2202006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4735 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-12 17:43:17 +00:00
fischman@webrtc.org
36cf4d2309 The video render module for iOS. 
BUG=2105, 2028
R=fischman@webrtc.org, mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2064004

Patch from SeungJae Lee <sjlee@webrtc.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4734 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-12 17:39:53 +00:00
minyue@webrtc.org
e509f943ed This issue is related to 
https://chromereviews.googleplex.com/9908014/

I was thinking about shipping ACM2 from the signal repository. There seems to be too many changes in one CL.

BUG=
R=andrew@webrtc.org, turaj@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2171004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4733 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-12 17:03:00 +00:00
andrew@webrtc.org
8fa03a15ab Make PCM16 available in Chromium builds. 
PCM16 can be useful for unit tests in Chromium. In particular Mikhal
would like to use it for ChromeCast.

This currently (r222592) has no impact on Chrome binary size, presumably
because PCM16 is unused and the linker strips the symbols.

To measure the potential impact, I looked at the size (bytes) of
out/Release/vie_auto_test on Linux with various codecs removed:
r4724    : 4567384
No PCM16 : 4565936
No ILBC  : 4500424
No G722  : 4555800
No RED   : 4565880

Giving the following size increases of adding each codec:
PCM16 :  1.4 kB (0.03%)
ILBC  : 70.0 kB (1.49%)
G722  : 11.6 kB (0.25%)
RED   :  1.5 kB (0.03%)

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

Review URL: https://webrtc-codereview.appspot.com/2195005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4732 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-12 01:30:30 +00:00
andrew@webrtc.org
89df092807 Make the destructor of AudioCodingModule public. 
This allows the type to be used with a scoped_ptr. Remove all calls to
the deprecated Destroy() from tests.

R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2200006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4731 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-12 01:27:43 +00:00
andrew@webrtc.org
5eb997a2fd Fix unsigned/signed comparison error due to r4729. 
Fix it for real by switching to ints rather than casting.

TBR=xians

Review URL: https://webrtc-codereview.appspot.com/2191009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4730 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-12 01:01:42 +00:00
andrew@webrtc.org
8f94013651 Reduce frequency of high audio delay warning logs. 
This will log the warning every 5 seconds instead of every 10 ms.

BUG=b/10674993
TESTED=Ran voe_cmd_test with hard-coded high delay. Observed a log
every 5 seconds.

R=noahric@chromium.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2184009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4729 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-11 22:35:00 +00:00
henrike@webrtc.org
256b83146c Removes function that is not used anywhere but somehow still causing library load issues on Android Release build. 
BUG=2364
R=andrew@webrtc.org, fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2190008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4728 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-11 20:43:13 +00:00
pbos@webrtc.org
5c678eabd9 Implement 'abs-send-time' extension in VideoSendStream. 
BUG=2229
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2184010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4727 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-11 19:00:39 +00:00
henrike@webrtc.org
6138c5cfa4 OpenSl: fixes crashes externally reported in issue 2361 and 2362. 
BUG=2361,2362
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2196008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4726 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-11 18:50:06 +00:00
turaj@webrtc.org
036b7436df Adding APIs. These APIs are not implemented yet, they are to help developement of ACM. 
Un-implemented APIs.

TBR=henrik.lundin@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/2191008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4725 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-11 18:45:02 +00:00
braveyao@webrtc.org
a80ee74f69 AppRTC: using a footer element instead of div#footer in CSS. 
R=dutton@google.com

Review URL: https://webrtc-codereview.appspot.com/2200004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4724 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-11 16:24:07 +00:00
mikhal@webrtc.org
d4d59ac871 Remove FrameForStorage:Follow up on r4688 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2201004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4723 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-11 15:18:15 +00:00
pbos@webrtc.org
2902328cce Implement 'toffset' extension in VideoSendStream. 
BUG=2229
R=holmer@google.com, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4722 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-11 10:14:56 +00:00
stefan@webrtc.org
554d158ce6 Reset jitter buffer and timing if frames are getting too much delay. 
BUG=chromium/263867
TEST=trybots
R=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2177005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4721 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-11 08:45:26 +00:00
andrew@webrtc.org
835ef67d14 Remove repeated conditions key. 
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2196007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4720 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-11 00:16:00 +00:00
henrike@webrtc.org
82f014aa0b OpenSL (not default): Enables low latency audio on Android. 
BUG=1669
R=andrew@webrtc.org, fischman@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2032004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4719 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-10 18:24:07 +00:00
braveyao@webrtc.org
641340944b Show the signaling state and ice connection state in AppRTC by hooking up the peerconnections .onsignalingstatechange and .oniceconnectionstatechange events. 
Hopefully this will increase the quality of the "it does not work" reports from users by giving them more information about what is going on under the hood.

R=juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/2174004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4718 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-10 17:37:16 +00:00
pbos@webrtc.org
319c98d663 Fix format string in video_quality_analysis.cc. 
Fixes compilation errors on Android and Linux32 targets.

TBR=kjellander@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/2196005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4717 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-10 15:23:50 +00:00
pbos@webrtc.org
182d025d94 Remove include_dirs from voice_engine.gyp. 
BUG=1662
R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2193004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4716 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-10 15:01:26 +00:00
pbos@webrtc.org
df531a2eee Test that VideoSendStream responds to NACK. 
BUG=2228
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2194006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4715 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-10 14:56:33 +00:00
kjellander@webrtc.org
f880f863dd Convert printing in video quality tests to Chromium's perf format. 
Add support for --label flag to the frame_analyzer, that
decides what label shall be used for the perf output.

BUG=none
TEST=
Make sure to have zxing and ffmpeg in the PATH.
Create a captured video (from running vie_auto_test custom call)
webrtc/tools/compare_videos.py --ref_video=reference_video.yuv --test_video=captured_output.yuv --frame_analyzer=out/Release/frame_analyzer --label=TEST_VGA
And then inspecting the output that is prefixed with RESULT.

R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2190005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4714 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-10 12:10:01 +00:00
pbos@webrtc.org
e07049f19f Lock RTPSender statistics. 
Suppressing these errors in TSan has become tedious. It's better to just
lock them.

BUG=2349
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2197004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4713 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-10 11:29:17 +00:00
pbos@webrtc.org
744fbc7fe4 Split up EngineTests and RampupTests. 
This allows having one group of tests per file, the test files are
long enough as they are.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2196004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4712 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-10 09:26:25 +00:00
andrew@webrtc.org
eda189be14 Remove redundant STR_CASE_CMP macro definitions. 
R=minyue@webrtc.org, turaj@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2187005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4711 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-09 17:50:10 +00:00
elham@webrtc.org
a19c9f4173 Updated WebRTC version to 3.41 
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2190007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4709 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-09 17:23:44 +00:00
pbos@webrtc.org
021c42bfa8 Lock use of _packetRequestCallback in VCM. 
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2190006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4708 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-09 17:18:31 +00:00
pbos@webrtc.org
7ebf0e7f44 Remove include_dirs from video_engine_core.gypi. 
BUG=1662
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2181005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4707 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-09 16:56:31 +00:00
pbos@webrtc.org
59f20bb735 Break out RTCPSender dependency on ModuleRtpRtcpImpl. 
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2191004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4706 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-09 16:02:19 +00:00
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