[SlimDevices: Audiophiles] Re: Transporter review

2006-11-10 Thread Patrick Dixon

opaqueice;153738 Wrote: 
 
 And if the mod is digital only then I _really_ don't see the point
 without some convincing demonstration that it actually sounds better
 when connected to a good DAC.I'm not too sure what you're on about, but what 
 we do is to disconnect
ALL the SB2/3 output circuitry, and replace it with our own DAC,
analogue and digital output circuits together with a linear PSU and
plenty of local regulation.  I personally have always prefered the
analogue outputs of the SB+ to the SB+ fed to an external DAC, but the
customer has the choice, and in any event it sounds considerably better
that the stock SB2/3.

We do convincing demonstrations all the time - otherwise people
wouldn't buy them (and we wouldn't have it any other way).


-- 
Patrick Dixon

www.at-tunes.co.uk

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[SlimDevices: Audiophiles] Re: Better sound with digital out. Why?

2006-11-10 Thread cliveb

beerbunny;153750 Wrote: 
 Thanks for this suggestion. It's a great idea and helped me to (I think)
 understand what is going on. I set the NAD aside and ran 1) SB2 optical
 to Yamaha: good sound and bass; 2) SB2 RCA to Yamaha: poor sound, weak
 bass; 3) didn't have digital coax cable so no SB2 digital coax to
 Yamaha; 4) SB2 optical to Yamaha, Yamaha RCA to NAD: good sound and
 bass; 5) SB2 RCA to Yamaha, Yamaha RCA to NAD: poor sound, weak bass.
 After running these tests, my guess would be that the Yamaha's DAC is
 superior to the SB2's DAC. Do you agree?
I don't necessarily agree with your conclusion. My experience is that
the SB2 DAC is actually pretty high quality.

Do we know whether the Yamaha digitises its analogue inputs? If it
does, then there is bound to be a gain mismatch between the digital and
analogue inputs, and this could easily account for perceived quality
differences (louder is generally interpreted as better).

Here's the reason for the gain mismatch. When you use a digital input,
it can simply be passed straight to the DAC. When you use an analogue
input, it has to be passed through an A/D converter. Now, since the
Yamaha cannot possibly know what signal level you might feed it, it has
to operate the A/D converter at a level so that the largest conceivable
input won't cause digital clipping. Therefore more typical input levels
will be operating the A/D converter below its peak level capability.
Apart from getting a quieter digital signal (compared to feeding the SB
digital output into the Yamaha), you'll also be losing a bit of
resolution. But I'd be surprised if the Yamaha didn't have a decent 24
bit A/D converter, so the resolution issue shouldn't be a big deal - my
money is on the analogue input simply providing less volume.

(Of course if the Yamaha has a poor quality A/D converter, then that
could be a much more straightforward reason for the reduced quality via
its analogue inputs).

Incidentally, to test the SB digital coax, just use any RCA lead you
have lying around. As long as the length of cable is not too long,
you'll be fine.


-- 
cliveb

Performers - dozens of mixers and effects - clipped/hypercompressed
mastering - you think a few extra ps of jitter matters?

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[SlimDevices: Audiophiles] Re: Transporter review

2006-11-10 Thread P Floding

opaqueice;153785 Wrote: 
 I think this takes the prize for most incoherent post - congratulations!
 It's really very boring to argue with you, so I'll sign off here.
 
 ezkcdude, I'm still interested in your response - why do you say it's
 not true that buffering and re-clocking eliminates input jitter?

You don't argue. You just keep telling people that you are right, and
that is that. Useless guesses based on what you think the talents and
budgets are of various people. Fan-boyism, if there ever was any.


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[SlimDevices: Audiophiles] Re: Better sound with digital out. Why?

2006-11-10 Thread tommypeters

Is your analogue volume at full...?


-- 
tommypeters

SB3--Meridian G68--NuForce Ref8--Bc Acoustique ACT A3

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[SlimDevices: Audiophiles] Re: Transporter review

2006-11-10 Thread PhilNYC

opaqueice;153738 Wrote: 
 
 I don't know what's inside the Benchmark - but it's certainly NOT
 implausible to say a DAC can be jitter rejecting or even immune.  If
 you think it's hype, why don't you back that up with some facts rather
 than simply making rude assertions?

Unless a digital system is using a master/slave clock architecture, it
needs to use a PLL to synchronize the transport signal to the DAC.  A
PLL can reduce jitter in the incoming signal, but has some of its own
inherent jitter if the clock in the transport has any difference from
the receiving clock.  A FIFO buffer can help reduce jitter in this kind
of design, but it cannot eliminate it.  Do a Yahoo/Google search on
asynchronous jitter and see what comes up...this topic has been
researched at length by the telecom industry, military, and yes, the
audio industry.  Pretty much all the credible research says that
re-clocking does not eliminate jitter, and in some cases it says that
the act of re-clocking can sometimes ADD jitter.

Here's an interesting one from said Yahoo search:

http://users.verat.net/~rogic2/1541/pdf/ASR_Measurements.pdf#search='asynchronous%20jitter'


-- 
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Sonic Spirits Inc.
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[SlimDevices: Audiophiles] Re: Trasporter display configuration - done much?

2006-11-10 Thread pc4ever1

I like it just the way it is by using the MusicInfoSCR plugin. It allows
me to display all the info I like to see at a distance ( Title, Artist,
Album, Progress, Playtime and File Extension). 
I know this could be done in one screen BUT at 10 feet or greater
distances it becomes unreadable. With two screens the same information
is divided and enlarged.

By the way I display File Extension so I can test myself of the sound
quality. Without looking at the File Extension on the display, I'll
listen to the song. Then I make a mental decision if it's either a Flac
or MP3 file ( these are the only types of files on my NAS). Then I check
on the display to see what it actually is. Overtime I have found that my
ear is getting better at distingishing between the two file types. I
only do this once per song, that way it doesn't become memory training.


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[SlimDevices: Audiophiles] Re: Transporter review

2006-11-10 Thread opaqueice

PhilNYC;153822 Wrote: 
 Unless a digital system is using a master/slave clock architecture, it
 needs to use a PLL to synchronize the transport signal to the DAC.  A
 PLL can reduce jitter in the incoming signal, but has some of its own
 inherent jitter if the clock in the transport has any difference from
 the receiving clock.  A FIFO buffer can help reduce jitter in this kind
 of design, but it cannot eliminate it.  Do a Yahoo/Google search on
 asynchronous jitter and see what comes up...this topic has been
 researched at length by the telecom industry, military, and yes, the
 audio industry.  Pretty much all the credible research says that
 re-clocking does not eliminate jitter, and in some cases it says that
 the act of re-clocking can sometimes ADD jitter.
 

I think we are talking about two different things.  If you read above
in this thread, there was a claim (by ezkcdude) that ASRC could
significantly reject jitter (or something, I'm not going to go back and
check his exact wording).  He gave an excellent reference written by a
chip designer which describes in great detail (it's about ten pages
long) how that works.  In a nutshell it is a PLL, but in the digital
domain and after upsampling by a huge factor, which allows it to have a
bandwidth of around 3 Hz.  Apparently that is much better than a more
standard approach where one uses a PLL to recover the clock from the
S/PDIF directly.  However it is also clear that it doesn't not entirely
elminate jitter (since it's still using the S/PDIF edges to construct
the clock).  If done according to his description it's not going to add
jitter, except possibly to a signal with very very low jitter to begin
with (and then it might increase it slightly).

However, there is another, entirely different approach possible, which
- as far as I can see - completely and totally eliminates the effects
of input jitter.  This does not use a PLL at all, because it does not
reconstruct the clock from the incoming data stream.  As I said above,
simply imagine having a huge buffer in your DAC.  Now run the audio
stream for, say, one hour (the length of a CD).  Record the entire
thing in the buffer.  You now have it stored as a digital sequence
which (barring bit errors) is identical to the sequence on the CD, and
has nothing at all to do with any jitter in the S/PDIF signal that
carried it.  Now, after waiting one hour, you get to listen to your
jitter-free music as the DAC plays out the data, using its own internal
crystal clock (which can be extremely clean).

Not very convenient, because you had to wait so long, but this totally
eliminates the effects of transport jitter (if not, I'm waiting for
someone to tell me why).  Now since this is rather inconvenient, you
can be more clever and reduce that initial pause to a nearly
imperceptible one, and that's what the Lavry does (according to their
white paper).  This does not use a PLL because it does not reconstruct
the clock from the incoming S/PDIF - it uses its own clock - and
therefore I fail to see how it can be affected by jitter.


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[SlimDevices: Audiophiles] Re: Transporter review

2006-11-10 Thread opaqueice

Patrick Dixon;153790 Wrote: 
 I'm not too sure what you're on about, but what we do is to disconnect
 ALL the SB2/3 output circuitry, and replace it with our own DAC,
 analogue and digital output circuits together with a linear PSU and
 plenty of local regulation. I personally have always prefered the
 analogue outputs of the SB+ to the SB+ fed to an external DAC, but the
 customer has the choice, and in any event it sounds considerably better
 that the stock SB2/3.

I find it much more plausible that the analogue outs could be improved.
What I find especially hard to believe is that the digital outputs are
so significantly improved by your mod that you can hear the difference
played through a Benchmark.  I say that simply because I'd be surprised
if there's any perceptible difference between any two transports - say
between a $30 DVD player and the SB+ or a $5000 CD player or what have
you - when played through a good DAC.  I say this because it seems
clear that jitter can be entirely or almost entirely eliminated with a
good DAC (see my post above).

In any case, I'd be very interested to see evidence to the contrary,
but that must either be a measurement or a blind test, as we know (from
thousands of scientific studies) that non-blind tests are useless for
this.  Until I see that I wouldn't even consider buying such a mod. 
After seeing it I would.  There are quite a few people that think like
I do on this, so purely from a marketing point of view it would benefit
you to in fact conduct such a blind test.  Given the minimal effort
involved, it's interesting (to put it kindly) that this isn't done.


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[SlimDevices: Audiophiles] Re: Transporter review

2006-11-10 Thread cliveb

PhilNYC;153822 Wrote: 
 And here's a pretty good paper by dCS as an overview of jitter as it
 relates to audio:
 
 http://www.dcsltd.co.uk/technical_papers/jitter.pdf
An interesting point made in this paper is the claim that asynchronous
sample rate conversion (ASRC) embeds incoming jitter into the signal,
and that low jitter sources with short cable runs should be used when
the receiver employs ASRC.

Contrast this to Benchmark's claim (and published measurements) that
the DAC1 (which uses ASRC) is immune to incoming jitter - to the extent
that you can stick it on the end of a thousand feet of digital
interconnect with no ill effects.

There are precious few companies around whose literature I would be
inclined to accept on good faith, but dCS and Benchmark are two of
them. And yet they appear to have diametrically opposed views regarding
ASRC. Which leaves me in a bit of a quandry.

Anyone care to shed some light on this?


-- 
cliveb

Performers - dozens of mixers and effects - clipped/hypercompressed
mastering - you think a few extra ps of jitter matters?

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[SlimDevices: Audiophiles] Re: Transporter review

2006-11-10 Thread PhilNYC

opaqueice;153828 Wrote: 
 However, there is another, entirely different approach possible, which -
 as far as I can see - completely and totally eliminates the effects of
 input jitter.  This does not use a PLL at all, because it does not
 reconstruct the clock from the incoming data stream.  As I said above,
 simply imagine having a huge buffer in your DAC.  Now run the audio
 stream for, say, one hour (the length of a CD).  Record the entire
 thing in the buffer.  You now have it stored as a digital sequence
 which (barring bit errors) is identical to the sequence on the CD, and
 has nothing at all to do with any jitter in the S/PDIF signal that
 carried it.  Now, after waiting one hour, you get to listen to your
 jitter-free music as the DAC plays out the data, using its own internal
 crystal clock (which can be extremely clean).
 
 Not very convenient, because you had to wait so long, but this totally
 eliminates the effects of transport jitter (if not, I'm waiting for
 someone to tell me why).  Now since this is rather inconvenient, you
 can be more clever and reduce that initial wait from one hour to a
 nearly imperceptible pause, and that's what the Lavry does (according
 to their white paper).  This does not use a PLL because it does not
 reconstruct the clock from the incoming S/PDIF - it uses its own clock
 - and therefore I fail to see how it can be affected by jitter.

This is essentially a FIFO buffer (my Dodson DAC does the same thing,
albeit with a much bigger buffer than the Lavry), and in this
architecture, you need to deal with things like buffer
overflow/underflow, because the data is still streaming...and this
still requires some cooperation between the incoming clock and the
re-clocker.  This will still contain some inherent jitter.

The only way to truly do it without a master/slave architecture is to
completely load the audio data into solid state memory up front, so
there is no buffer management required.  There is a player that does
this called the Nova Physics Memory PLayer...


-- 
PhilNYC

Sonic Spirits Inc.
http://www.sonicspirits.com

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[SlimDevices: Audiophiles] Re: Will Transporter support video?

2006-11-10 Thread augustusflavius

Looking at the manual shows only limited audio format support, no
balanced outs, no AES/EBU, and no wordclock. The fact they call it
AudioPhile is just marketing imho.

http://www.neodigits.com/download/X5000_Manual.pdf


-- 
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Wadia 830 - Sophia Princess 845 Monoblocks - Rosinante Dulcineas

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[SlimDevices: Audiophiles] Re: Transporter review

2006-11-10 Thread Patrick Dixon

opaqueice;153831 Wrote: 
 I'd be surprised if there's any perceptible difference between any two
 transports - say between a $30 DVD player and the SB+ or a $5000 CD
 player or what have you - when played through a good DAC.  I say this
 because it seems clear that jitter can be entirely or almost entirely
 eliminated with a good DAC (see my post above).You seem to be following a 
 circular argument - (jitter rejecting) DACs
are transport agnostic therefore there is no perceptible difference
between transports.

Only problem is, in the real world, one can hear the difference.  Feel
free to try for yourself!


-- 
Patrick Dixon

www.at-tunes.co.uk

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[SlimDevices: Audiophiles] Re: Transporter review

2006-11-10 Thread opaqueice

ezkcdude;153858 Wrote: 
 If you're going to keep on with this, at least, you must remember that
 it is *input* jitter you're talking about. You have not addressed
 jitter generated in the D/A process (which is not inconsequential).

In almost every place (except the one you quoted) I was careful to
specify *input* jitter - clearly there will be some jitter in the
oscillator used to clock the DAC.  But that has nothing to do with the
transport, which was my point.  Also those clocks are extremely clean.

cliveb Wrote: 
 
 Doesn't this do exactly the same as a PLL with a bandwidth of 0.1Hz?

In some vague sense yes - but it's not a PLL, and its characteristics
are different.  In any case at least for me it's easier to simply think
of what happens - if we assume the DAC clock is perfect for a moment,
there will be one slightly jittered bit every ten seconds.  So one out
of every 441,000 sound samples will be very slightly wrong.  Precisely
which sample it is might depend very weakly on some characteristic of
the input jitter, depending on how the algorithm which decides when to
adjust the clock works.  Another question is whether a real crystal
oscillator already has more jitter in it than is caused by these
adjustments, in which case this is as good as you'll ever be able to
do.

PhilNYC Wrote: 
 
 Well, that might be a semantic difference, because jitter is measured
 as an average, not regarding each individual rising edge. As the dCS
 whitepaper describes, we're now talking about Signal-related timing
 error, not random noise-related jitter. The only reason why you'd
 have buffer overflow/underflow is because of a difference in the
 transport and DAC clocks, and this is essentially the same issue that
 is introduced by PLL's. The bottom line is if you have a system that
 uses two different clocks, there will be differences, and therefore
 jitter will be present. EDIT/ADDED: The remaining question is whether
 it is enough jitter to be audible.

Actually jitter has a spectrum - you can measure its RMS average if you
like, but that's only one number out of N necessary to fully specify it
(where N is the number of bits in the stream, or the times at which
each edge arrive).  I'm not sure what you mean by the issue that is
introduced by the PLL's - actually that's the issue which is *solved*
by the PLLs, albiet with some possibility of jitter contaminating the
output. 

Patrick Dixon Wrote: 
 
 You seem to be following a circular argument - (jitter rejecting) DACs
 are transport agnostic therefore there is no perceptible difference
 between transports.
 
 Only problem is, in the real world, one can hear the difference. Feel
 free to try for yourself!

I have, and there was no difference.  My original suggestion was that
you try this as well - evidently you haven't.  

I'm not sure where you see a circular argument - if DACs are transport
agnostic then it follows that transports played through them are
indistinguishable.  How is that circular?


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[SlimDevices: Audiophiles] Re: Better sound with digital out. Why?

2006-11-10 Thread adamslim

beerbunny;153750 Wrote: 
 Thanks for this suggestion. It's a great idea and helped me to (I think)
 understand what is going on. I set the NAD aside and ran 1) SB2 optical
 to Yamaha: good sound and bass; 2) SB2 RCA to Yamaha: poor sound, weak
 bass; 3) didn't have digital coax cable so no SB2 digital coax to
 Yamaha; 4) SB2 optical to Yamaha, Yamaha RCA to NAD: good sound and
 bass; 5) SB2 RCA to Yamaha, Yamaha RCA to NAD: poor sound, weak bass.
 After running these tests, my guess would be that the Yamaha's DAC is
 superior to the SB2's DAC. Do you agree?

The consistency of your answers and my knowledge of the SB's sound
leads me to one of three conclusions:

1 - Your Yamaha is somehow artificially boosting the bass, and this
sounds better in your system to your ears, or

2 - You have a problem in your setup; possibly a contact failing or
really bad mains interference, or

3 - Your SB3 is faulty in that its analogue output is not working
correctly.

The SB3 is only slightly lighter sounding in the bass than my CD
player, and that's a nice CD!  I can't believe that your Yamaha is
going to be as good, period.

I do know that AV amps tend to add a lot of thump to the bass; in a
balanced system this is not ideal, but if you have tiny speakers, it
could sound better...?


-- 
adamslim

SB3 and Shanling CDT-100, Rotel RT-990BX, Esoteric Audio Research 859,
Living Voice Auditorium IIs, Nordost cables

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[SlimDevices: Audiophiles] Rave: New setup + SB2 = great sound!

2006-11-10 Thread ob_kook

I have been listening to music for several hours straight now and could
not contain myself any longer - I needed to rave about my new system!

I've been using SB2 for a long time now, but due to convenience, had it
plugged into a Bose Wave radio, and sometimes a Denon AVR-4800 with
Klipsch KG4's. Howver, my true-love speakers were in my other apartment
(Dynaudio Contour 1.3 MKII). They were being driven by my old college
amp (Luxman L-430) which finally decided to give up the right channel.

OK, time to ship those puppies home and set them up in my living room
to create a proper system! With a dead amp, I was able to convince the
wife that I had to buy a new amp. And my goal was a minimalist system
which just sounded good. I finally chose the following:

Amp - after a lot of research, I decided that I couldn't go wrong
trying out the Portal Panache with its 60 day return policy. It got
great reviews, seemed like it would love my Dyns, and the designer was
a minimalist who bleieved in spending the momey on the parts and not
on_any_fluff (no remote even). I called Joe at Portal to she what he
thought of the match, and he told me that he happened to have a pair of
the monoblocks in stock from a magazine review that he could let me have
at considerable discount. Hmm...overbudget, but another great excuse to
upgrade. Sold! Now I have these brawny yet subtle Portal Paladin
Monoblocks, but needed a preamp to control the volume. 

Preamp - I read every thread on this site about using the SB2 to
control the volume and finally Sean posted to a thread I started saying
please don't. Ok, I guess that sealed it. I looked into a lot of
passive options (Endler Attenuators etc.) and finally decided on the
Axiom passive pre (with cassock upgrade) made by Luminous Audio. 

Speakers arrived this morning and I put it all together and
oh...my...god... I just didn't know this could all sound so good. 

And with the ease I can cue up any song I feel like listening too, I
have not moved from the chair with my laptop browser controlling SS and
a big stupid grin on my face.

Just wanted to say that if anyone is interested in good value and high
quality with no BS and people who are a pleasure to work with, check
into the guys above (Portal and Luminous).

http://portalaudio.com/paladin.html
http://www.luminousaudio.com/axiom.html


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[SlimDevices: Audiophiles] Re: Better sound with digital out. Why?

2006-11-10 Thread beerbunny

cliveb;153795 Wrote: 
 I don't necessarily agree with your conclusion. My experience is that
 the SB2 DAC is actually pretty high quality.
 
 Do we know whether the Yamaha digitises its analogue inputs? If it
 does, then there is bound to be a gain mismatch between the digital and
 analogue inputs, and this could easily account for perceived quality
 differences (louder is generally interpreted as better).
 
 Here's the reason for the gain mismatch. When you use a digital input,
 it can simply be passed straight to the DAC. When you use an analogue
 input, it has to be passed through an A/D converter. Now, since the
 Yamaha cannot possibly know what signal level you might feed it, it has
 to operate the A/D converter at a level so that the largest conceivable
 input won't cause digital clipping. Therefore more typical input levels
 will be operating the A/D converter below its peak level capability.
 Apart from getting a quieter digital signal (compared to feeding the SB
 digital output into the Yamaha), you'll also be losing a bit of
 resolution. But I'd be surprised if the Yamaha didn't have a decent 24
 bit A/D converter, so the resolution issue shouldn't be a big deal - my
 money is on the analogue input simply providing less volume.
 
 (Of course if the Yamaha has a poor quality A/D converter, then that
 could be a much more straightforward reason for the reduced quality via
 its analogue inputs).
 
 Incidentally, to test the SB digital coax, just use any RCA lead you
 have lying around. As long as the length of cable is not too long,
 you'll be fine.

I bought a digital coax cable and tested it with the SB2 to Yamaha and
it was indistinguishable in sound and bass quality from the digital
optical. Since I can't run the digital through anything but the Yamaha,
it is very difficult to say what the difference is. I do appreciate your
thoughts about volume, although I don't think that's it. When I play a
bass heavy song like Burning Down the House at volume with the
digital output connections (and tone controls neutral), it punches you
in the stomach and shakes the filament of a nearby lightbulb. When I
run the same song through the RCA analog at volume, it has no punch and
sounds too bright, almost noisy.

I should say that the reason for my original question is that I am
setting up my SB2 with my NAD S200 separately from my home theater (HT)
for space and imaging reasons so the Yamaha is staying with the HT. I
plan to first buy the S100 preamp that was designed to mate with the
S200. Because it doesn't have digital inputs of any kind (all RCA), I
will be testing it with the SB2's RCA analog output. This might solve
my problem. Do you think so? If it doesn't, I would be inclined to buy
a decent DAC that accepts a digital coax input and run SB2 to DAC to
NAD S100 to S200.


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[SlimDevices: Audiophiles] Re: Better sound with digital out. Why?

2006-11-10 Thread beerbunny

To answer the other questions on page 2: I think my interconnects are
fine. On volume, when using the Yamaha, I keep the Slimserver software
volume at full and use the Yamaha volume to regulate speaker sound. I
do not have a subwoofer connected as the Paradigm Studio 100's produce
plenty of bass.


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[SlimDevices: Audiophiles] Re: Trasporter display configuration - done much?

2006-11-10 Thread ModelCitizen

pc4ever1;153823 Wrote: 
 I like it just the way it is by using the MusicInfoSCR plugin.
I love the plugin too. It's just that overall display configuration is
very hard unless you work with computers! It needs simplifying, whilst
retaining the same configurability.
MC


-- 
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Now what?

Transporter  Naim NAP 250  PMC OB1s.
Music catalog: http://modelcitizen.mine.nu/music.txt

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[SlimDevices: Audiophiles] Re: Transporter review

2006-11-10 Thread opaqueice

cliveb;153833 Wrote: 
 An interesting point made in this paper is the claim that asynchronous
 sample rate conversion (ASRC) embeds incoming jitter into the signal,
 and that low jitter sources with short cable runs should be used when
 the receiver employs ASRC.
 
 Contrast this to Benchmark's claim (and published measurements) that
 the DAC1 (which uses ASRC) is immune to incoming jitter - to the extent
 that you can stick it on the end of a thousand feet of digital
 interconnect with no ill effects.
 
 There are precious few companies around whose literature I would be
 inclined to accept on good faith, but dCS and Benchmark are two of
 them. And yet they appear to have diametrically opposed views regarding
 ASRC. Which leaves me in a bit of a quandry.
 
 Anyone care to shed some light on this?

Hi Clive,

I understand what he is saying there.  He says:

 
 DATA JITTER AND ASYNCHRONOUS SAMPLE RATE CONVERTERS
 Asynchronous sample rate converters can respond to data jitter and
 process it into the signal, 
 irrevocably.  For this reason they should be used with care, and if
 they have to be used, 
 should be used with low data jitter sources with short cable runs. 
 

The point is that ASRC embeds some jitter into the digital sequence
itself.  So you start with something which, when represented as a
string of digits, is identical to the track on the CD.  Jitter
manifests itself only as small variations in the time at which those
bits arrive.  But after ASRC you have embedded those timing variations,
to some small degree, into the digital sequence representing the sound
samples themselves. 

It's like doing D-A-D', so the jitter in D gets into D' and can never
be removed.

Of course if you then immediately play out to a DAC that's not a bad
thing, but if you stored the digital sequence for later use, or did
multiple such conversions, you might run into problems.


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[SlimDevices: Audiophiles] Re: Trasporter display configuration - done much?

2006-11-10 Thread ceejay

I also like the way that MusicInfoScr works, and agree with MC that the
overall structure isn't right.  I've been exercising a few brain cells
over this and so far have come up with some requirements and a few
thoughts towards a spec.

First, the requirements:
- want to be able to define display config as MusicInfoScr does today
(this isn't broken, lets keep it!)
- want full flexibility on when different information layouts, and
visualisers, appear on one or both screens
- must allow for plugins doing weird things with the display at will
(eg Snow screensaver!)
- must be able to offer good out of the box default layouts while
allowing fancy customisation for the keen
- make the use of the various configuration options well structured and
logical

Second, some thoughts inching towards a spec...

I think that there are several layers of definition here, some of which
are already in use:

1 - define the formats for information strings that you want to be able
to use (Server wide)(today: Server settings/Formatting/Title Format)

2 - Select which of those format strings you want to use for a
particular player for a given purpose (today: player settings/basic
settings/title format) - there might be a need for other comparable
strings at this level but I can't offhand think of one maybe Album
name format?

3 - define one or more different Display Views.  Give them a name so
you can refer to them later.  MusicInfoScr does a great job of this, as
far as text layout is concerned.  I'd like to see the use of visualisers
integrated here, though, and perhaps the various scroll settings. (I
think this set of Display Views would be Server wide).

4 - Apply the different Display Views to the display (or displays if
you are lucky enough to have two!) for a variety of modes (see the
table in my earlier post). Per Player.

Putting all of these together, you'd be able to define your own display
layout (as per MusicInfoScr today) for each display - or use the
visualisers - where and when you want. You'd be able to have
visualisers on either side. You'd be able to have a display look like a
visualiser when playing, but have a specific text layout when you press
Now Playing (say).  It would clean up the currently confused
relationship between the core slimserver functions and the MusicInfoScr
plugin.

OK, so it would allow me to do what I want with the displays (which I'm
fairly sure I can't at the moment), but I think it offers a lot of
options for what others might want too.

Just one other thing - if anything like this happens, I'd also request
a save display configuration to file and load from... option - it
would allow users/support to post configurations around as well as make
it easy to copy from one player to another.

Any thoughts?
Ceejay


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[SlimDevices: Audiophiles] low cost dac recommendations?

2006-11-10 Thread pvadbx

While I've got the Transporter for my main system, I'd like some
suggestions for a low cost DAC to get audio out of my computer.  The
Mac has optical out, and I'd like an upgrade from the analog out I'm
currently using in my office.  Any suggestions that use an optical,
USB, or firewire connection in the range of $100 new or used?  Or is
that too low?


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[SlimDevices: Audiophiles] Re: Rave: New setup + SB2 = great sound!

2006-11-10 Thread stevo

Fantastic! The Axiom passive preamps look like a really nice product. 

I also went through a similar process, retiring my old kit speakers and
amp to the kitchen and investing in some new gear. Keeping things simple
and uncluttered is one of the beautiful things about SqueezeBox. No cd's
cluttering the lounge room floor, no need to search for that one cd you
really want to listen to, only to find your daughter has stolen it.

I run an SB3 direct into my Perreaux 200ip amp using Rothwell
attenuators. The preamp in the integrated Perreaux is very good but I
have chosen to bypass it. The sound then runs to some magnificent VAF
Research I93's speakers I bought after heading down to Adelaide to look
at their much cheaper DC-X speakers. I spent 3 hours listening to music
in their showroom and chatting with the owner of the business -
fantastic service. So the only true import to the Southern Continent is
the SB3 (Australia and New Zealand really should be one country there
are too many borders in the world). 

I am yet to set up Inguz room correction on the new speakers but that
will come next after I have fully sorted out the speaker positions
etc.

There is so much good gear about - a Transporter should turn up soon as
well. I kept eying off DAC's but the Transporter seems to have it all at
a reasonable price compared to the DAC's I was checking out. 

I average at least 3 to 4 cd purchases a month - probably more. So much
good music so little time. I didn't realise how good cd's could sound.
It's wonderful.

Stevo
www.vaf.com.au
http://www.perreaux.com/


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[SlimDevices: Audiophiles] Re: low cost dac recommendations?

2006-11-10 Thread NewBuyer

stevo;153944 Wrote: 
 TDA1545 TDA1545A Non-OS DAC -- USB Plus Constantine DAC's look nice. You
 will find them on Ebay. I haven't tried them.
 
 Stevo

I second this. I have tried the Mhdt Labs DAC units and they are a true
bargain, wonderful sound.


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[SlimDevices: Audiophiles] Re: Rave: New setup + SB2 = great sound!

2006-11-10 Thread ob_kook

Hey Stevo - I just checked out that Perreaux amp and it looks like it
was built with very similar objectives - lot's of raw yet refined
power! Nice looking piece of kit. I went round and round about
attenuators, but finally having a knob that was easy to turn up and
down won out. And actually the unit looks even better in real life. For
anyone considering passive attenuation, I wouldn't hesitate to recommend
it.

You are right about the gear that is out there these days - having
secured a budget of $2500 put me in an area that was overwhelming with
choices! 

I have also been following the Inguz post with great interest. My
listening room is a tad hard around the edges and there will be no foam
finding their way onto the walls, and it sounds like DRC might be the
way to go anyway. Intriguing...


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[SlimDevices: Audiophiles] Re: Any fellows SB3er using Icepower class D w/ BW speakers ?

2006-11-10 Thread nelamvr6

There's a LOT of info on calss D in this month's Stereophile.


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[SlimDevices: Audiophiles] Re: SB3 into TacT - how good can it get?

2006-11-10 Thread NewBuyer

SoftwireEngineer;143467 Wrote: 
 Try this one from partsexpress. It should be better, I think. 
 
 http://www.partsexpress.com/pe/showdetl.cfm?DID=7Partnumber=180-952
 
 I have the sound professionals one and I think both are same because
 they have the same number of fibres - 65.
 
 http://www.soundprofessionals.com/cgi-bin/gold/item/SP-TOC-99HQ
 
 The audioasylum review is linked in the sound professionals site. 
 
 I think these toslinks will easily beat your electrical digital cable
 (it betters my Zu Ash)...

Softwire (and others with glass optical cables), I often hear about
these types of cables being very fragile because of the glass material
used. I'm wondering, just how fragile are we talking about here? Does
the user need to be worried everytime the cable starts to bend? Or are
they fragile in a different way, etc? Would really appreciate any
comments on this, before buying one of these nice Toslink cables...


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[SlimDevices: Audiophiles] Total BS (Logitech) Sellout

2006-11-10 Thread Walleyefisher

Does anyone else feel slighted by the Logitech buyout crap.  Granted I'm
sure they got paidbut all the Slimdevices credibility just went out
the freaking window.  

Now all of us device owners will be religated to the bureaucratic
steaming excrement of logitech.  

Guess what open source -- bye bye

Hello license fees and content restriction.

God I hope it was worth it.  I wish I was within my 30 day return
policy for my Transport.

This sucks


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Re: [SlimDevices: Audiophiles] Total BS (Logitech) Sellout

2006-11-10 Thread Pat Farrell
Walleyefisher wrote:
 Does anyone else feel slighted by the Logitech buyout crap.  Granted I'm
 sure they got paidbut all the Slimdevices credibility just went out
 the freaking window.  

Troll?

Its not news, and its not bad.
But if you hate it, sell your unit on eBay.

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[SlimDevices: Audiophiles] Re: Any fellows SB3er using Icepower class D w/ BW speakers ?

2006-11-10 Thread Sleestack

i shed all my SS amps (Parasound and Meridian) for 4 Bel canto Ref1000s
and TACT BOZ 216/2200s.  Couldn't be happier.  The BelCantos never
blink.


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*headphone:* singlepower sds-xlr at, classe sacd2, hd650  
*2 channel:* tact rcs 2.2.xp w/ full aberdeen mods, bel canto oneref.
1000 monoblocks x 4,teac esoteric p-03/d-03, epiphany 12-12s (waiting
for my 20-21s), tact w210 corner load subs
*5.1 channel:* tact tcs mkii w/ aberdeen power supply, tact boz
216/2200 (x5) w/ aberdeen power supply, tact adc6 w/ full aberdeen
mods, denon 5910, bel canto pl-1a, eggleston andra ii (x5), velodyne
dd-15, pioneer elite pro 1130hd

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[SlimDevices: Audiophiles] Re: Total BS (Logitech) Sellout

2006-11-10 Thread Skunk

Walleyefisher;153978 Wrote: 
 
 God I hope it was worth it.  I wish I was within my 30 day return
 policy for my Transport.

I would advise keeping it if the box and badge read Transport.
Misprints often become valuable, so you may have a collectors item.
Best box it up to keep the condition though, which shouldn't matter
since the sound degraded so badly after the announcement.


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[SlimDevices: Audiophiles] Re: Total BS (Logitech) Sellout

2006-11-10 Thread TiredLegs

Skunk;153984 Wrote: 
 Best box it up to keep the condition though, which shouldn't matter
 since the sound degraded so badly after the announcement.Not sure if the 
 sound degraded, but the noise part of the
signal-to-noise ratio certainly increased.


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