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    <syn:updatePeriod>hourly</syn:updatePeriod>
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  <item rdf:about="http://comments.gmane.org/gmane.comp.audio.jackit/27404">
    <title>HDSPe AIO and Jack</title>
    <link>http://comments.gmane.org/gmane.comp.audio.jackit/27404</link>
    <description>&lt;pre&gt;Hi :)

it seems to be that I've got a misconfigured jackd, since Flo claims the
following at alsa-user: "Of course sample rates &amp;gt;48kHz are supported" -
http://www.mail-archive.com/alsa-user&amp;lt; at &amp;gt;lists.sourceforge.net/msg29636.html

This is the setup [1] on Arch Linux 64-bit architecture.
As you can see, jackd does start at 48KHz [2]. As you can see [3] - [5]
jackd doesn't start at higher sample rates.

The card can be used with higher sample rates on Windows.

What am I doing wrong?

Regards,
Ralf

PS and OT: If somebody does use this card with all 8 ADAT channels, it
would be nice to send me a hdspmixer file off list, that does enable
this. I never get it working for more than the first 2 ADAT channels
when using TotalMix, aka hdspconf on Linux. I'm able to do it on
Windows. Note, I'm not a Windows user ;), I installed it some days ago
to check the card.

[1]
[rocketmouse&amp;lt; at &amp;gt;archlinux ~]$ service rtirq status

  PID CLS RTPRIO  NI PRI %CPU STAT COMMAND
   36 FF      90   - 130  0.0 S    irq/8-rtc0
  169 FF      8&lt;/pre&gt;</description>
    <dc:creator>Ralf Mardorf</dc:creator>
    <dc:date>2013-05-22T05:22:20</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.comp.audio.jackit/27402">
    <title>Jack Linux/Windows</title>
    <link>http://comments.gmane.org/gmane.comp.audio.jackit/27402</link>
    <description>&lt;pre&gt;Hello,

I need to connect two PCs via JACK. Master machine has linux and slave machine has Windows.

After too much head scratching, I found netjack to be of no use.
Anyone who has done this, plz help.

Best Regards, 
Harshit
 
_______________________________________________
Jack-Devel mailing list
Jack-Devel&amp;lt; at &amp;gt;lists.jackaudio.org
http://lists.jackaudio.org/listinfo.cgi/jack-devel-jackaudio.org
&lt;/pre&gt;</description>
    <dc:creator>Harshit Suri</dc:creator>
    <dc:date>2013-05-22T04:29:18</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.comp.audio.jackit/27396">
    <title>JACK_PROMISCUOUS_SERVER not working</title>
    <link>http://comments.gmane.org/gmane.comp.audio.jackit/27396</link>
    <description>&lt;pre&gt;Hi

Encouraged by the report of someone who ran jackd as his default sound
server for non-professional uses (without RT priority as he/she mentioned),
and saying that he (or she) liked the simplicity of jackd, I've tried the
same thing too. But I'm running apps from various user ids, which still
doesn't seem to be supported with jackd, except that it *is* meant to be
supported through the env var JACK_PROMISCUOUS_SERVER. Except that this
doesn't seem to work (anymore? [*]).

This is jackd from the Debian Wheezy package,
1.9.8~dfsg.4+20120529git007cdc37-5

I'm setting umask 0000 for both server and client,
JACK_PROMISCUOUS_SERVER="" (I've also tried other values), but the client
tries to open a different socket path; when I hard link the socket path to
what the client expects, it segfaults.

I'm considering debugging this but I'd appreciate some hints, like, do you
have an idea what broke it (assuming that it's a bug that broke it), or
whatever else I should know about it. Is there a reason I should check out&lt;/pre&gt;</description>
    <dc:creator>Christian Jaeger</dc:creator>
    <dc:date>2013-05-20T01:26:50</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.comp.audio.jackit/27390">
    <title>Too many servers already active</title>
    <link>http://comments.gmane.org/gmane.comp.audio.jackit/27390</link>
    <description>&lt;pre&gt;Hi,

At the radio station where I volunteer, we're using 100% open source 
software.  We have a playout system for audio files, where users log in 
one at a time and as part of their session jackd is started, along with 
the newly created audio player 4deckradio.  When they log out, all their 
running programs (including jackd) are killed.

Every now and then, after maybe 15, 20 new logins, jackd refuses to 
start for all users.  The error message is that there are "too many 
servers already active".  However, there are no jackd processes still 
running when I do a "ps aux | grep jack".

A complete restart of the machine remedies the problem, temporarily at 
least.

My hunch is that jackd isn't able to clean up properly sometimes 
(perhaps because it is killed before it has the chance?) causing future 
servers to not start.  I'm at a loss however to figure out -what- 
exactly it doesn't clean up, if that is the case.

Could someone shed some light on what is preventing jackd from starting 
up?  If I know wha&lt;/pre&gt;</description>
    <dc:creator>Jan Middelkoop</dc:creator>
    <dc:date>2013-05-17T10:26:53</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.comp.audio.jackit/27374">
    <title>Netjack error: “slave is running with a different protocol”</title>
    <link>http://comments.gmane.org/gmane.comp.audio.jackit/27374</link>
    <description>&lt;pre&gt;First off, I've also posted this question to SO, so if anyone's able to
help me, helping
here&amp;lt;http://stackoverflow.com/questions/16553899/netjack-error-slave-is-running-with-a-different-protocol&amp;gt;
will
let other people benefit as well :)

 ------------------------------

I'm trying to run jack between two computers. I had this working on a
previous set of systems, but this time around... not so much.

System 1:

tobago$ jackd -R -d alsa -P hw:0 -C hw:1 -p 1024
tobago$ jack_load netadapter

System 2:

trinidad$ jackd -R -d alsa
trinidad$ jack_load netmanager

At this point, jackd starts printing on loop:

Waiting for a slave...
Error : slave is running with a different protocol tobago
Can't init new net master...

Unfortunately googling "slave is running with a different protocol" only
yields two results other than the source code, and neither is helpful. I
checked the version of jackd on each system; one is 1.9.9 with protocol 8,
the other is 1.9.9.5 with protocol 8. So I'm doubtful that's the issue.

Does an&lt;/pre&gt;</description>
    <dc:creator>Tom Williams</dc:creator>
    <dc:date>2013-05-15T16:34:09</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.comp.audio.jackit/27371">
    <title>Odp: Re:  JackTransport vs VST ppqPos</title>
    <link>http://comments.gmane.org/gmane.comp.audio.jackit/27371</link>
    <description>&lt;pre&gt;Dnia Niedziela, 12 Maja 2013 11:45 Stéphane Letz &amp;lt;letz&amp;lt; at &amp;gt;grame.fr&amp;gt; napisał(a) 

Thanks for answer Stephane.
Meantime I dig a little and I think that found a reason of this issue, If you look on this line from transport.c (jack_transport example client):
pos-&amp;gt;tick += nframes * pos-&amp;gt;ticks_per_beat * pos-&amp;gt;beats_per_minute / (pos-&amp;gt;frame_rate * 60);

IMHO it mean "how many ticks are in current period". If nframes is 256 and frame rate is 96k  and BPM is 5 (or less) you got 0.928798 ... but pos-&amp;gt;tick is uint32_t .. so you actually got 0 ;-). OFC this is related to your settings and that's way YMMV. Well ... in some cases jack_transport will not push BBT at all ;-)

The problem is in Transport design itself - because BBT master works always in current cycle space/context - I mean - it  doesn't look at the history (anyway it can't if we want to have tempo changes, right ?), and because ticks are uint32_t  we will have always some rounding.

The most simple way to fix this that I see is push tick to double - but this&lt;/pre&gt;</description>
    <dc:creator>Pawel</dc:creator>
    <dc:date>2013-05-15T04:31:20</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.comp.audio.jackit/27368">
    <title>Starting out with Jack and Asterisk</title>
    <link>http://comments.gmane.org/gmane.comp.audio.jackit/27368</link>
    <description>&lt;pre&gt;Hi,

New to the list so usual apols about addressing questions wrongly etc.

I want to play around with passing Asterisk audio through Jack on Linux. I've successfully installed and configured Jack and Asterisk to talk to eachother, but the documentation for command-line style manipulation is quite light on the ground.

Currently, connecting my call to the JACK() app sends 5 seconds of silence. For my first proof of concept I'd like to get it to send back some sort of sound. This is a server system so there is no GUI etc. Even if it's streaming a sound file that would be fine.

After that I would like to manipulate audio with something like libfaust - wonder if anyone here has experience with that? A simple script to change the pitch of audio would be amazing to see.

Thanks!
Leo
&lt;/pre&gt;</description>
    <dc:creator>Leo Brown</dc:creator>
    <dc:date>2013-05-11T13:56:42</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.comp.audio.jackit/27365">
    <title>JackTransport vs VST ppqPos</title>
    <link>http://comments.gmane.org/gmane.comp.audio.jackit/27365</link>
    <description>&lt;pre&gt;Hi,

I discover strange problem while improving JackTransport support in FSTHost.

Most VST plugins while trying to synchronize with host usually ask for such things:
1) tempo ( == jack_pos.beats_per_minute)

2) ppqPos - which I understand like "current quarter note" == current jack_pos.beat.

3) barStartPos .. ok, this one is not a subject in this case ;-)

4) samplePos == jack_pos.frame

5) sampleRate == jack_pos.frame_rate

The problem is which pppPos. There are two ideas how to compute this:

1) Using BBT (which should be a preferred option here, right ?)
double ppqBar = (jp.bar - 1) * jp.beats_per_bar;
double ppqBeat = jp.beat - 1;
double ppqTick = (double) jp.tick / jp.ticks_per_beat;

ppqPos = ppqBar + ppqBeat + ppqTick;

2) Using sample rate / current frame / tempo - this is a standard way in VST world. We count how much samples is in quarter note (beat) ..

double ppq = jp.frame_rate * 60 / jp.beats_per_minute;
ppqPos = jp.frame / ppq;

The problem is that .. this second counter "hurry" (sorry can't&lt;/pre&gt;</description>
    <dc:creator>Pawel</dc:creator>
    <dc:date>2013-05-09T01:22:53</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.comp.audio.jackit/27358">
    <title>Problems with variable latency</title>
    <link>http://comments.gmane.org/gmane.comp.audio.jackit/27358</link>
    <description>&lt;pre&gt;Hello,

I am using jack in an application where I need sample accurate
synchronization between capture and playback. I am using jack_delay and
jack_iodelay to precisely measure the roundtrip latency in frames. On my
better machine (core i7, maudio delta 66, 64-bit Debian, jackdmp 1.9.9) the
latency in frames reported by jack_delay or jack_iodelay increases slowly
over time by up to 12 frames. For example, it will start at 3132.66 frames
but after ~5 minutes, it will jump up to 3133.66 frames. This will continue
until about 3144.66 frames is reached and then it may return back to
3132.66 or 3.133.66 frames latency. It may take hours for it to go through
this cycle. If I restart jack_delay, it does not go back to 3132.66 frames,
but continues with whatever the last delay I saw was. There are seldom
xruns or other errors produced during a jump. It happens at various
frames/period and periods/buffer settings. It happens at whatever CPU
governor setting I use (performance, on demand). It happens whether or not
I &lt;/pre&gt;</description>
    <dc:creator>Steve Haynal</dc:creator>
    <dc:date>2013-05-07T05:16:24</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.comp.audio.jackit/27357">
    <title>relating to the Android sticks... strange request</title>
    <link>http://comments.gmane.org/gmane.comp.audio.jackit/27357</link>
    <description>&lt;pre&gt;Hi,

This is indirectly related to the discussion on jack an Picuntu / Android
stick discussion...

This is an odd request. Can any leaders in the jack developer team have a
Skype or gotomeeting with me?

Please email privately.

To establish some minimal cred, Google Perry Kivolowitz

Thank you
_______________________________________________
Jack-Devel mailing list
Jack-Devel&amp;lt; at &amp;gt;lists.jackaudio.org
http://lists.jackaudio.org/listinfo.cgi/jack-devel-jackaudio.org
&lt;/pre&gt;</description>
    <dc:creator>Perry Kivolowitz</dc:creator>
    <dc:date>2013-05-07T03:28:27</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.comp.audio.jackit/27343">
    <title>Jack on arm hf</title>
    <link>http://comments.gmane.org/gmane.comp.audio.jackit/27343</link>
    <description>&lt;pre&gt;Hi list,
 
the guy with the UG802 tv stick again. I've tried to get Jack to run on this rk3066 arm hf computer without success for a while now. I posted some of the weird error messages on this list but nothing got solved. So, I'd like to ask a more general question:
 
Developers, is anyone working on or even interested in getting Jack to run on an arm cortex a9 platform? What are possible problems?
 
In my opinion, these cheap little sticks and tablets could be a wonderful platform for creating audio fx boxes, sound installations etc. They are so much more powerful than, e.g., the raspberry pi which is not much cheaper and has a much weaker cpu and ram.
 
I don't have much money but I would pay anyone who manages to get jack to run with an usb interface on a UG802 or similar computer. I'm sure there are others interested in this as well.
 
Looking forward to answers,
Axel


---
Alle Postfächer an einem Ort. Jetzt wechseln und E-Mail-Adresse mitnehmen! Rundum glücklich mit freenetMail_________________&lt;/pre&gt;</description>
    <dc:creator>axl99&lt; at &gt;freenet.de</dc:creator>
    <dc:date>2013-05-05T22:05:38</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.comp.audio.jackit/27338">
    <title>Jack thread cancellation</title>
    <link>http://comments.gmane.org/gmane.comp.audio.jackit/27338</link>
    <description>&lt;pre&gt;Hi list,

I have an embedded application that makes use of jack (Version 1.9.8).
I have started seeing crashes from the clients in the JackError() call.

The stack trace shows that the thread executing (Which is handling the
JackSocketChannel::Execute call) receives is being killed
(sigcancel_handler is called) and since the thread is in
CANCEL_ASYNCHRONOUS mode, it crashes since an unsafe function is
called (vnsprintf).

The client I am using is actually unaware of using Jack at all, it is
using the alsa jack plugin, and it opens/closes the alsa devices many
times during its lifetime.

Now, I have a few doubts:
- Why are threads created with cancellation type PTHREAD_CANCEL_ASYNCHRONOUS?
  Is this because the realtime threads need this or is there some other reason?
- Would it be possible to mark the threads used by the JackMessageBuffer and
  JackSocketClientChannel as PTHREAD_CANCEL_DELAYED instead?
- Why is fThread.Kill() used in JackSocketClientChannel::Stop instead of fThread.Stop()?
  The latter shoul&lt;/pre&gt;</description>
    <dc:creator>MONTANARO Luciano (MM</dc:creator>
    <dc:date>2013-05-03T15:11:16</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.comp.audio.jackit/27325">
    <title>JD-Rack (JACK Done Right Audio Connection Kit)</title>
    <link>http://comments.gmane.org/gmane.comp.audio.jackit/27325</link>
    <description>&lt;pre&gt;http://pydaw.org/wp/2013/04/29/pydaw-os-now-available-and-some-details-of-the-tentatively-named-jd-rack/

Just FYI, I haven't used PyDAW myself and am in no way affiliated.

Regards,

Jeremy
&lt;/pre&gt;</description>
    <dc:creator>Jeremy Jongepier</dc:creator>
    <dc:date>2013-05-01T09:31:38</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.comp.audio.jackit/27322">
    <title>raspberry pi simple in-out (raw)</title>
    <link>http://comments.gmane.org/gmane.comp.audio.jackit/27322</link>
    <description>&lt;pre&gt;All:

As I RTFM all that jackaudio has, my objective is to sample an A/D channel (10 bit) raw input and route each sample to the audio jack at an arbitrarily high rate, like, 50K samples per second; the period is never precisely the same and I do not need fidelity or repeatability.  I'm a C coder searching for libraries to get me there from here.  I do not need to store any sampled data for later replay.

Has this wheel been invented?  I reward with beer, as I am creating labs for a new Raspberry Pi developers course at UCSD.

fun, fun, fun,


- - -
Gene W. Anderson
Escondido, CA
gwa&amp;lt; at &amp;gt;cox.net
- - -

_______________________________________________
Jack-Devel mailing list
Jack-Devel&amp;lt; at &amp;gt;lists.jackaudio.org
http://lists.jackaudio.org/listinfo.cgi/jack-devel-jackaudio.org
&lt;/pre&gt;</description>
    <dc:creator>Gene Anderson</dc:creator>
    <dc:date>2013-04-25T04:28:10</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.comp.audio.jackit/27321">
    <title>m-audio FastTrack Ultra 8R no playback with hw device</title>
    <link>http://comments.gmane.org/gmane.comp.audio.jackit/27321</link>
    <description>&lt;pre&gt;Hi everyone,

I've been using the M-Audio FastTrack Ultra 8R with jack for a while
now and everything works fine with the plughw alsa device, but playback
does not work with the hw interface.

This is what I get when starting jack:

$ jackd -d alsa -d hw:2  -r 44100
Copyright 2001-2005 Paul Davis and others.
Copyright 2004-2013 Grame.
jackdmp comes with ABSOLUTELY NO WARRANTY
This is free software, and you are welcome to redistribute it
under certain conditions; see the file COPYING for details
JACK server starting in realtime mode with priority 10
creating alsa driver ... hw:2|hw:2|1024|2|44100|0|0|nomon|swmeter|-|32bit
configuring for 44100Hz, period = 1024 frames (23.2 ms), buffer = 2 periods
ALSA: final selected sample format for capture: 24bit little-endian
ALSA: use 2 periods for capture
ALSA: final selected sample format for playback: 24bit little-endian
ALSA: use 2 periods for playback


Nothing suspicious here.

$ file test.wav
test.wav: RIFF (little-endian) data, WAVE audio, mono 44100 Hz

$ mplaye&lt;/pre&gt;</description>
    <dc:creator>Niko Efthymiou</dc:creator>
    <dc:date>2013-04-22T18:21:03</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.comp.audio.jackit/27313">
    <title>Cleanup of clients in libjack2 causes crash</title>
    <link>http://comments.gmane.org/gmane.comp.audio.jackit/27313</link>
    <description>&lt;pre&gt;Hello,

I've written a program that can create multiple jack clients and connect 
them to jackd.
I'm using a callback passed to jack_on_info_shutdown() to know when a 
jack client has exited,
when that happens I deactivate and close the client, and I try to make a 
new one every second.
This way I can restart jackd without having to restart my program.

So far so good, until my program started to segfault whenever I closed 
jackd.
After some digging I found this code in JackLibGlobals.h:

if (!JackGlobals::fServerRunning &amp;amp;&amp;amp; fClientCount &amp;gt; 0) {

     // Cleanup remaining clients
     jack_error("Jack server was closed but clients are still allocated, 
cleanup...");
     for (int i = 0; i &amp;lt; CLIENT_NUM; i++) {
         JackClient* client = JackGlobals::fClientTable[i];
         if (client) {
             jack_error("Cleanup client ref = %d", i);
             client-&amp;gt;Close();
             delete client;
             JackGlobals::fClientTable[i] = NULL;
         }
     }

So what happens here is, jackd exits, I d&lt;/pre&gt;</description>
    <dc:creator>Bob van Loosen</dc:creator>
    <dc:date>2013-04-16T18:56:23</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.comp.audio.jackit/27305">
    <title>Jack and Pulse - how outdated is this?</title>
    <link>http://comments.gmane.org/gmane.comp.audio.jackit/27305</link>
    <description>&lt;pre&gt;In my quest to get skype and google-talk working with jack running, i came
across this: http://www.harald-hoyer.de/linux/pulseaudio-and-jackd

A poster said that following this got it working for them. However, it's
from 2011.

How outdated is this document and what should i alter to take advantage of
the latest pulse/jack integration capabilities?

Thanks

&lt;/pre&gt;</description>
    <dc:creator>Bearcat M. Şándor</dc:creator>
    <dc:date>2013-04-14T22:24:31</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.comp.audio.jackit/27284">
    <title>Getting started with jack</title>
    <link>http://comments.gmane.org/gmane.comp.audio.jackit/27284</link>
    <description>&lt;pre&gt;Hi ! I'm trying to get started with Overtone &amp;lt;http://overtone.github.io/&amp;gt;.
I need to make jack properly run on my computer, but I have some trouble.
Would you be nice and help me?
I'm using Arch GNU/Linux:

I've installed jack:

but I'm unable to start it properly :-(, as shown above:

Even if I follow advice for alsa &amp;amp; jack given on that site (
http://jackaudio.org/pulseaudio_and_jack), it still doesn't work:

Trust me, it puzzles me, I don't know how to handle that trouble. Could
anyone help me? :)

Caocoa
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&lt;/pre&gt;</description>
    <dc:creator>Caocoa Caocoa</dc:creator>
    <dc:date>2013-04-12T12:51:01</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.comp.audio.jackit/27263">
    <title>alsa_out problem</title>
    <link>http://comments.gmane.org/gmane.comp.audio.jackit/27263</link>
    <description>&lt;pre&gt;Dear All,

I am using jackd2 1.9.8 with netjack1 for sending sound over ethernet. I am 
also using alsa_in and alsa_out as my sound cards. Unfortunately, "alsa_out" is 
not working properly. Often, a broken sound is heard at the output with a huge 
number of "delay = -1088" prints in the "alsa_out" terminal. The frequency of 
this problem is approx. 1 time out of 7 instances. What may be the cause for 
this? Also how can I solve this?

We are using Red Hat Enterprise Linux 6.0 with alsa 1.0.23.
We downloaded JACK from jackaudio.org.

Thanks &amp;amp; Regards! 
----------------- 
Harshit
&lt;/pre&gt;</description>
    <dc:creator>Harshit Suri</dc:creator>
    <dc:date>2013-04-06T11:43:25</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.comp.audio.jackit/27262">
    <title>Merge jackd1 and jackd2 code bases</title>
    <link>http://comments.gmane.org/gmane.comp.audio.jackit/27262</link>
    <description>&lt;pre&gt;Hi!

Please have a look at the following situation:

   http://trac.jackaudio.org/ticket/227#comment:1

An utterly trivial ticket that hasn't been dealt with for two years.
Worse, the same fix required three individual commits, because common
code between jackd1 and jackd2 isn't properly shared but replicated.

Can we please align the two projects and share things like tools,
example-clients and man pages? Given the lack of manpower, doing
everything twice doesn't exactly help.



Cheers
&lt;/pre&gt;</description>
    <dc:creator>Adrian Knoth</dc:creator>
    <dc:date>2013-04-06T11:44:15</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.comp.audio.jackit/27260">
    <title>Is it possible to get the system sound level?</title>
    <link>http://comments.gmane.org/gmane.comp.audio.jackit/27260</link>
    <description>&lt;pre&gt; 

Hi,

 

I need to get the system sound level, does JACK support this?

 

Or any suggestion?

 

My target platform is windows and mac.

 

Thank you,

 

Cid

 

 

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&lt;/pre&gt;</description>
    <dc:creator>Cid_Huang[黃懋燁]</dc:creator>
    <dc:date>2013-04-06T08:16:46</dc:date>
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