bd6bdca57f
scoped_ptr.h: Renames function and change namespace scope to fix conflicts with Chromium not detected by the FYI bots.
...
BUG=N/A
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23259004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7546 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-28 18:06:42 +00:00
a296725d0e
audio_coding/codecs/isac/fix: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >>"
...
BUG=3348,3353
TESTED=locally on linux and trybots
R=henrik.lundin@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7544 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-28 13:05:43 +00:00
67ca26e087
common_audio: Removed trivial macro WEBRTC_SPL_UMUL_16_16
...
The macro made a cast to uint16_t before a plain multiplication. At the few places where it was used the variables were already uint16_t.
Affected components:
* isac/fix
BUG=3348,3353
TESTED=locally on linux and trybots
R=henrik.lundin@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29869004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7543 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-28 13:03:10 +00:00
ff8a98e352
Use neteq_unittest_tools in audio_decoder_unittests
...
With the recent move of RtpFileReader to the rtp_test_utils target
(in r7536), it is now possible to let audio_decoder_unittests depend
on neteq_unittest_tools without breaking the Android build.
BUG=2692
TBR=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7542 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-28 09:47:13 +00:00
820efd5b55
Fix double backslashes in incoming_video_stream.cc
...
Originally uploaded in https://codereview.appspot.com/149160043/ .
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27859004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7541 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-28 08:47:16 +00:00
aada86b261
Add a simple AudioConverter class.
...
This will be used to refactor AudioProcessing/AudioBuffer. We can
enable alternate downmixing schemes in AudioProcessing by pulling
the conversion logic out of AudioBuffer.
The unit test is largely stolen from voice_engine/utility_unittest.cc.
As commented, the voice_engine routines should be replaced with
AudioConverter.
BUG=chromium:405270
R=aluebs@webrtc.org , mgraczyk@chromium.org
TBR=kwiberg
Review URL: https://webrtc-codereview.appspot.com/30779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7538 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-27 18:18:17 +00:00
33a0e2d7ef
Only configure the SSL library in one place.
...
Build settings now use use_openssl in both Chromium and standalone builds. It
moves all the platform-specific SSL-related build checks to be conditioned on
this flag as appropriate.
This is to avoid colliding with Chromium's transition away from NSS.
This is a fixup of https://webrtc-codereview.appspot.com/29559004 to avoid
breaking use_legacy_ssl_defaults.
BUG=chromium:413497
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24109004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7537 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-27 18:13:40 +00:00
aca5803b19
Move (test) RtpFileReader to a lightweight target.
...
Moves RtpFileReader to rtp_packet_parser and renames it to
rtp_test_utils which is allowed to rely on rtp_rtcp.
R=andrew@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/24979004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7536 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-27 18:01:03 +00:00
b787f4c593
Move scoped_ptr "free" functions into the webrtc namespace.
...
Resolves a conflict with Chromium's scoped_ptr on the recently added
make_scoped_ptr().
TEST=local Chromium Linux build passes.
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7535 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-27 17:42:22 +00:00
df429882af
Upgrade our scoped_ptr copy to match Chromium's latest.
...
In particular add the move constructor and assignment operator.
Diff between our version and Chromium's:
https://paste.googleplex.com/4887047529562112
R=henrike@webrtc.org , kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31789004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7531 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-27 16:12:38 +00:00
a37f1dd6b8
Cleaning up audio_decoder_test.cc and adding ResampleInputAudioFile
...
This CL contains some cleaning up and refactoring of
audio_decoder_test.cc. A new class ResampleInputAudioFile is created
and used in the tests.
BUG=3926
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7528 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-27 12:58:18 +00:00
0552356fda
isacfix: Refactor big-endian reading and writing
...
Make subroutines for encoding and decoding arrays of 16-bit big-endian
integers, and in the process fix a bug: When decoding an odd number of
bytes from be16, the least significant byte of the last int16 in the
array was properly taken to be zero instead of actually being read
(since it's outside the array). However, when encoding an odd number
of bytes, the least significant byte of the last int16 in the array
was written to the output as-is instead of being taken to be zero;
thus, we encoded one byte more than we should. This was probably not
harmful, and the value was dropped at decoding anyway; nevertheless,
writing a constant zero is the safe thing to do, and this patch does
so.
R=aluebs@webrtc.org , bjornv@webrtc.org , henrik.lundin@webrtc.org , turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28569004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7527 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-27 11:25:37 +00:00
9fed099208
Increase max trace message size to 1024 characters.
...
A recent CL by pbos:
https://code.google.com/p/webrtc/source/detail?r=7518
added long log messages and triggered errors on the DrMemory bot due to
WEBRTC_TRACE. The trace mechanism _should_ truncate the log strings
but something appears to be going awry.
This sweeps the problem under the rug, but given that WEBRTC_TRACE
should die fairly soon, seems to be a reasonable tradeoff.
TEST=passing try on DrMemory.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27849004
Patch from Andrew MacDonald <andrew@webrtc.org >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7526 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-27 09:31:05 +00:00
c86ec3e3bc
Fix ::~LogMessage to print as a string.
...
R=andrew@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/26949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7525 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-27 09:22:03 +00:00
39b1743116
Adding the subtool rtcBot report visualizer
...
This tool for visualize the output reports of rtcBot by calculating
the average and max of a specific stats and plot the output.
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23169004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7519 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-24 09:26:16 +00:00
ad3b5a5c16
Move min transmit bitrate to VideoEncoderConfig.
...
min_transmit_bitrate_bps needs to be reconfigurable during a call (since
this is currently set only for screensharing through libjingle and can't
be set once and for all for the entire Call.
R=mflodman@webrtc.org , stefan@webrtc.org
BUG=1667
Review URL: https://webrtc-codereview.appspot.com/28779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7518 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-24 09:23:21 +00:00
7e19a11a71
Break out WebRtcNs_ComputeDdUpdate function in ns_core
...
This is done in order to make the code more readible and maintainable.
It generates bit-exact output.
BUG=webrtc:3811
R=bjornv@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31739004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7514 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-23 19:54:33 +00:00
f8ea0d5518
Break out WebRtcNs_UpdateNoise function in ns_core
...
This is done in order to make the code more readible and maintainable.
It generates bit-exact output.
BUG=webrtc:3811
R=bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25889004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7513 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-23 19:49:42 +00:00
799e88ae19
Break out FFT function in ns_core
...
This is done in order to make the code more readible and maintainable.
This introduces an error of only +1 and -1.
BUG=webrtc:3811
R=bjornv@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27749004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7512 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-23 19:36:42 +00:00
8454ad88ed
Break out ComputeSnr function in ns_core
...
This is done in order to make the code more readible and maintainable.
The output is bit-exact.
BUG=webrtc:3811
R=bjornv@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31729004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7511 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-23 19:34:14 +00:00
0d3e254c89
Adding three video conference bots test
...
A video conference between three bots, each bot creating two
peerConnections, and each peer connected to one of the other bots.
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24099004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7510 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-23 16:45:07 +00:00
0e19d0c2aa
Adding file from test.webrtc.org domain to be downloaded
...
This has been configured to allow cross domain to access this generated
file:
https://test.webrtc.org/test-download-file/9000KB.data
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25939004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7509 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-23 15:41:30 +00:00
580d367b14
Add macros and APIs for webrtc histograms.
...
BUG=crbug/419657
Code that links system_wrappers.gyp:system_wrappers should either:
- provide implementations for the APIs, or
- link with default implementations in system_wrappers.gyp:system_wrappers_default.
R=andresp@webrtc.org , kjellander@webrtc.org , mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7508 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-23 12:57:56 +00:00
82462aade0
Adds support for sending first set of packets at increasingly higher bitrates to probe the link and faster ramp up to a high bitrate.
...
Also wires up a finch experiment to control this.
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30639004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7505 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-23 11:57:05 +00:00
2192701135
Using the Unused turn configuration in two way test
...
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26909004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7504 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-23 08:40:53 +00:00
ad553a2731
Let video_loopback use internal VCM capturers.
...
R=stefan@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/27819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7503 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-23 08:24:02 +00:00
fce8f5d319
NOTE: This code review based on the running issue:
...
https://webrtc-codereview.appspot.com/24939004/
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7499 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-22 17:24:20 +00:00
3382059e55
Adding Two way video and audio streaming test to RtcBot
...
NOTE: This code review based on this running issue:
https://review.webrtc.org/24939004/
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25879004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7498 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-22 17:17:15 +00:00
e9b7d03db6
HTTPS Server used instead of HTTP for loading the bots to avoid the media permission pop-up clicks every time running the test.
...
This code review based on the running issue:
https://webrtc-codereview.appspot.com/24939004/
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27769004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7497 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-22 16:34:25 +00:00
32452b20b8
Make ReconfigureVideoEncoder use current bitrate.
...
Prevents bitrate drops when changing resolution etc.
R=stefan@webrtc.org
BUG=1667
Review URL: https://webrtc-codereview.appspot.com/24069004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7493 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-22 12:15:24 +00:00
3f8f5554a0
Disable TestVp8Impl.BaseUnitTest on MSan.
...
MemorySanitizer uses generic (non-optimized) libvpx which is not bit
exact. This may be a bug in upstream libvpx, but we're forced to disable
it now as it blocks launching the MSan bot.
R=stefan@webrtc.org
TBR=marpan@webrtc.org
BUG=3904
Review URL: https://webrtc-codereview.appspot.com/24089004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7491 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-22 10:30:30 +00:00
76960d5f74
For FIR packet, payload length is zero, so SendToNetwork function is failing.
...
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23059004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7490 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-22 09:47:14 +00:00
67cf1d742b
Break out WebRtcNs_Windowing function in ns_core
...
This is done in order to make the code more readible and maintainable.
This introduces only +1 and -1 errors.
BUG=webrtc:3811
R=bjornv@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7488 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-21 22:35:40 +00:00
0e7099244c
Break out WebRtcNs_Energy function in ns_core
...
This is done in order to make the code more readible and maintainable.
This generates bit-exact output.
BUG=webrtc:3811
R=bjornv@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23099004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7487 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-21 22:14:10 +00:00
7634c09406
Break out WebRtcNs_IFFT function in ns_core
...
This is done in order to make the code more readible and maintainable.
This creates bit-exact output.
BUG=webrtc:3811
R=bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24039004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7486 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-21 21:27:00 +00:00
333e2556ed
Break out WebRtcNs_UpdateBuffer function in ns_core
...
This is done in order to make the code more readible and maintainable.
It generates bit-exact output.
BUG=webrtc:3811
R=bjornv@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7483 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-21 20:33:09 +00:00
def1e97ed2
Implement AudioEncoderPcmU/A classes and convert AudioDecoder tests
...
BUG=3926
R=kjellander@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7481 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-21 12:48:29 +00:00
78ea06dd34
audio_coding/codecs/ilbc: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >>
...
Removed usage of trivial macro.
BUG=3348,3353
TESTED=locally on linux and trybots
R=henrik.lundin@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29789004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7480 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-21 07:17:24 +00:00
913f7b8d5e
Fix for glitches in ACM when switching desired output sample rate
...
The problem was that if the output sample rate is changed such from one
where no resampling is needed to a rate that requires resampling, the
first output from the resampler will contain an onset period. The
solution provided in this CL is to keep a copy of the last output frame
in ACM, and if the resampler is engaged, it will be primed with this
old frame before resampling the current frame.
BUG=3919
R=bjornv@webrtc.org , turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27729004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7479 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-21 06:54:23 +00:00
b69ea9a35a
common_audio: Replaced invalid operand in min_max_operations_neon.S"
...
Vector Move immediate can not load #0x7FFF. Changed to us vdup from already loaded register.
BUG=N/A
TESTED=ios and android trybots
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26879004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7477 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-20 14:08:35 +00:00
b35b136480
Make avg_{psnr,ssim}_threshold_ const.
...
Triggered warning on next clang version being rolled as these variables
are annotated to be protected by crit_.
R=stefan@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/24949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7475 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-20 09:14:38 +00:00
2abebe7baf
audio_coding/codecs/isac/main: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >>
...
BUG=3348,3353
TESTED=locally on linux and trybots
R=henrik.lundin@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30739004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7474 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-20 08:26:41 +00:00
a5ce7bbe17
audio_coding/neteq: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >>
...
BUG=3348,3353
TESTED=locally on linux and trybots
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24009004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7473 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-20 08:24:54 +00:00
28100cb388
Reverts r7459 "Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p."
...
BUG=N/A
TBR=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7472 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-17 22:03:39 +00:00
b1dac33cac
Revert cls (original cl + fixes) 7422-7424 "Add VP9 codec to VCM..."
...
BUG=3932
R=marpan@google.com
Review URL: https://webrtc-codereview.appspot.com/27779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7470 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-17 18:54:46 +00:00
0371a37f85
Moving creating TURN configration to the host machine instead of the bots - rtcBot
...
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24939004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7468 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-17 16:43:50 +00:00
f7030d4ed7
Query Android device orientation on every camera frame received.
...
Remove orientation listener from Android camera, since device
orientation change events are not well synchronized with actual
device display orientation. Plus these event may not be delivered
at all if device is in stationary position causing initial camera
frames appear rotated.
BUG=
R=braveyao@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23009004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7467 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-17 16:25:06 +00:00
c221db6165
Test names changed from e.g) testOneWayVideo/chrome=>chrome to testOneWayVideo/chrome-chrome.
...
Because the symbol ">" is interpreted as special command for output to file in bash commands.
TBR= andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24929004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7465 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-17 09:13:43 +00:00
264e66f7a5
Add encoded_timestamp to AudioEncoder base class
...
BUG=3926
TBR=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24029004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7464 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-16 21:16:07 +00:00
9ea6f8a84d
New interface class AudioEncoder
...
This class will be the base for new C++ wrapper classes for all
encoders.
BUG=3926
TBR=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7463 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-16 11:26:24 +00:00
