Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-11-16 Thread Themis

In any case, it would have been much better if in the digital audio
protocol the clock was explicitly indicated by the A/D (in the data) and
stored with it. This way, transport wouldn't have to add its own jitter.

Now, whatever is done, we can't go back.


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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-11-15 Thread bhaagensen

cliveb;483331 Wrote: 
 Naim's ring-buffer approach just averages out the jitter over a longer
 time base. In principle it does the same thing as a low frequency PLL.
 As long as the DAC receives data from a transport with its own
 free-running clock, surely it is in principle impossible to completely
 eliminate the incoming jitter. Doesn't matter how you try to average it
 out (ring buffer, PLL, ASRC) - ultimately the DAC is a slave to the
 transport's clock.

Thats what I thought too, but its not true. Naim uses the ring-buffer
to find an average for the incoming data rate and then chooses among 10
internal clocks for clocking the data out of the buffer. If none of the
10 match closely enough, they fallback to ASRC. 

I assume that within reasonable systems it is possible to obtain an
average incoming data rate stable enough for one of the internal clocks
to match. Given this assumption the total jitter at the chip in the dac
is totally independent of jitter added by the source.

How well the assumption holds up in practise I dont know. I guess we'll
see when it comes out (which it does these very days). 

In conclusion, its basically an asynchronous implementation over
s/pdif, if you will.

Bjørn


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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-11-15 Thread cliveb

bhaagensen;485610 Wrote: 
 Thats what I thought too, but its not true. Naim uses the ring-buffer to
 find an average for the incoming data rate and then chooses among 10
 internal clocks for clocking the data out of the buffer. If none of the
 10 match closely enough, they fallback to ASRC.
Thanks for the additional info. It's a cunning strategy, but of course
the chances that any one of their 10 internal clock rates is an exact
match for the incoming average rate seems vanishingly small. And the
consequence of this is that the ring buffer needs to be large enough
that it doesn't overflow or underflow within some defined time period
based on the clock rate mismatch.

Since you don't want to hang around waiting for the buffer to half-fill
before starting playback, presumably Naim always choose an internal
clock which is slower than the incoming one - that would guarantee
underflow cannot happen and you can start clocking out from the buffer
straight away. You just need a big enough buffer to avoid overflow.

In the bad old days of CDs, you could confidently set the time limit
within which overflow must not happen to about 80-90 mins. But with
streaming sources like Squeezeboxes, the time is potentially unlimited.
Clearly they can't include an infinitely large buffer, so the device
will *have* to resort to sample-dropping in order to resync in extreme
cases.


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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-11-15 Thread DCtoDaylight

bhaagensen;485610 Wrote: 
 Thats what I thought too, but its not true. Naim uses the ring-buffer to
 find an average for the incoming data rate and then chooses among 10
 internal clocks for clocking the data out of the buffer. If none of the
 10 match closely enough, they fallback to ASRC. 

If Naim's internal clock runs a little slower than the source clock,
then this should work fine, with the buffer slowly filling as the CD
plays.  Memory is cheap these day's, you can easily buffer an entire CD
in ram!  Ten years ago, when I first saw this approach used in a jitter
reduction box, the cost of the ram dominated the product price, but not
so anymore


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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-11-15 Thread htrd

DCtoDaylight;485651 Wrote: 
 Memory is cheap these day's

You still wouldnt need a huge buffer - just switch to the next slowest
clock when the buffer gets slightly full, after waiting for those two
clocks to be in phase.


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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-11-15 Thread bhaagensen

cliveb;485650 Wrote: 
 
 
 Since you don't want to hang around waiting for the buffer to half-fill
 before starting playback, presumably Naim always choose an internal
 clock which is slower than the incoming one - that would guarantee
 underflow cannot happen and you can start clocking out from the buffer
 straight away. You just need a big enough buffer to avoid overflow.
 

Exactly, but I don't know how they would (or maybe they don't) avoid
that the DAC keeps on playing after stopping the transport. I think I'll
post at Naim's forum asking some of the beta-testers to try.

How likely a match is of course depends on many setup-specific factors,
but I think I read that there is an indicator on the front-panel which
lights up when its running ASRC. Again a post for the Naim-forum
perhaps.

I would say though, that if it works as advertised, I'm surprised it
isn't implemented in all newer DAC's???


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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-11-15 Thread bhaagensen

htrd;485655 Wrote: 
 You still wouldnt need a huge buffer - just switch to the next slowest
 clock when the buffer gets slightly full, after waiting for those two
 clocks to be in phase.

But as I see it you cant just do this, due to the nature of s/pdif,
even if memory was free, as per my previous post.


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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-11-15 Thread cliveb

I have to say that all this ducking  diving by the likes of Naim to try
and immunise their DACs from SPDIF shortcomings seems like a heck of a
lot of effort when it would just be simpler to add a clock output on all
DACs and a clock input on all transports.

I wonder if Logitech might consider adding a word clock input to the SB
Touch's successor? If (as seems likely) the Transporter drops from the
product range, there will be no Squeezebox class device with a
state-of-the-art DAC, and the need to fully support external DACs
becomes more important.


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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-11-15 Thread Grahame

It has been argued that S/PDIF is ultimately a flawed protocol, and
engineering effort expended to fix it is just papering over the cracks.


Word clock from the DAC is one solution, but its uptake is not
universal.

Is can be argued that async USB support for external dacs is the
solution going forward. It appears that the touch will/may support
this.

The 'Aha' moment is that jitter free (or low jitter) playback requires
a pull clocked data feed from the DAC, but until recently only push
clock and data has been available from the transport. In the absence of
a standard that separated clock from data, we could expect little else.

Conventional CD transports by their nature are inherently push
based.
Computer based transports can be pull based given the right
implementation.

Raw S/PDIF, by its nature, is push based.


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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-11-15 Thread Kellen

All this talk about transmission problems with S/PDIF and USB . can
the Ethernet protocol (TCP/IP) not be used for the transmission of these
digital files in place of the current problematic ones?


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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-11-15 Thread seanadams

Kellen;485819 Wrote: 
 All this talk about transmission problems with S/PDIF and USB . can
 the Ethernet protocol (TCP/IP) not be used for the transmission of these
 digital files in place of the current problematic ones?

I think there are products that do that


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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-11-15 Thread Kellen

seanadams;485825 Wrote: 
 I think there are products that do that
I mean connecting an external DAC to a transport. 

Is this not feasible to do with TCP/IP? If so, is it not better than
SPDIF and USB?


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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-11-15 Thread seanadams

Kellen;485831 Wrote: 
 I mean connecting an external DAC to a transport. 

The hard drive is the transport, Squeezebox/TP is the DAC.

 Is this not feasible to do with TCP/IP? If so, is it not better than
 SPDIF and USB?

Yes of course it's better, but I'm not sure what you're suggesting that
is different than what these products do. When you say transport do you
mean a CD player or what?


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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-11-15 Thread Kellen

Sorry for the confusion, Sean.

I'll try and re-word this better.

At the moment I use a TCP/IP protocol cable to connect a hard drive to
my Squeezebox in order that I can transmit a digital signal between
them. From what I understand there is no jitter issue resultant with
this type of connection. Now, if I wish to then send this digital signal
to an external DAC I have use a SPDIF cable to connect the Squeezebox to
the external DAC. Doing this causes unwanted jitter related issues as
talked about. Is it not possible to design a TCP/IP connection between
Squeezbox and external DAC instead of using SPDIF?


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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-11-15 Thread Pat Farrell
Kellen wrote:
 seanadams;485825 Wrote: 
 I think there are products that do that
 I mean connecting an external DAC to a transport. 
 
 Is this not feasible to do with TCP/IP? If so, is it not better than
 SPDIF and USB?

Why would you want to do that?

All the Transporter is in essence is a TCP/IP collector and a nice DAC.
If you want to use some other DAC, start with an SB3.


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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-11-15 Thread Kellen

pfarrell;485840 Wrote: 
 Kellen wrote:
  seanadams;485825 Wrote: 
  I think there are products that do that
  I mean connecting an external DAC to a transport. 
  
  Is this not feasible to do with TCP/IP? If so, is it not better than
  SPDIF and USB?
 
 Why would you want to do that?
 
 All the Transporter is in essence is a TCP/IP collector and a nice
 DAC.
 If you want to use some other DAC, start with an SB3.
 
 
 -- 
 Pat Farrell
 http://www.pfarrell.com/
I don't have a Transporter. I already have a SB3 which I am using as a
transport to feed an EAD DAC via SPDIF. 

With all of the talk about how poor SPDIF is wrt jitter, I was curious
if it is possible to use TCP/IP (which doesn't seem to be as prone to
jitter) in its place. So, hypothetically, instead of using a SPDIF cable
to connect my SB3 to my EAD, use TCP/IP cable instead.


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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-11-15 Thread Pat Farrell
Kellen wrote:
 I don't have a Transporter. I already have a SB3 which I am using as a
 transport to feed an EAD DAC via SPDIF. 
 
 With all of the talk about how poor SPDIF is wrt jitter, I was curious
 if it is possible to use TCP/IP (which doesn't seem to be as prone to
 jitter) in its place. So, hypothetically, instead of using a SPDIF cable
 to connect my SB3 to my EAD, use TCP/IP cable instead.

The SB3 does exactly that already. Its what it, and the SB2 and the SB1
and even the Slimp3 do. They talk  TCP/IP and output audio or SPDIF

All this talk about Jitter is just a way to separate audiophiles from
their money.

You are confusing what SPDIF/AES aim to do, and what TCP/IP does


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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-11-15 Thread Kellen

pfarrell;485860 Wrote: 
 Kellen wrote:
  I don't have a Transporter. I already have a SB3 which I am using as
 a
  transport to feed an EAD DAC via SPDIF. 
  
  With all of the talk about how poor SPDIF is wrt jitter, I was
 curious
  if it is possible to use TCP/IP (which doesn't seem to be as prone
 to
  jitter) in its place. So, hypothetically, instead of using a SPDIF
 cable
  to connect my SB3 to my EAD, use TCP/IP cable instead.
 
 The SB3 does exactly that already. Its what it, and the SB2 and the
 SB1
 and even the Slimp3 do. They talk  TCP/IP and output audio or SPDIF
 
 All this talk about Jitter is just a way to separate audiophiles from
 their money.
 
 You are confusing what SPDIF/AES aim to do, and what TCP/IP does
 
 
 -- 
 Pat Farrell
 http://www.pfarrell.com/
I must be confusing things then because, as things stand at the moment,
I am unable to use TCP/IP to connect the digi out of my SB3 to the digi
in on my EAD. I can only make such a connection with either SPDIF or
optical. Is this because it's not possible to have TCP/IP handle such a
connection?


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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-11-15 Thread Pat Farrell
We really, really need to start a separate thread, this is not at all
about a simple question for the guy who is now retired from
SlimDevices/Logitech

Kellen wrote:
 pfarrell;485860 Wrote: 
 The SB3 does exactly that already. 


 I must be confusing things then because, as things stand at the moment,
 I am unable to use TCP/IP to connect the digi out of my SB3 to the digi
 in on my EAD. I can only make such a connection with either SPDIF or
 optical. Is this because it's not possible to have TCP/IP handle such a
 connection?

It is not possible. You are confusing the types of things that the
connection can do.

SPDIF is a direct device connection, it sends out square waves and the
receiving device converts them to analogue music. it goes from one
device to a single target.

TCP/IP is a network connection protocol, it handles retransmission,
error correction, routing, etc.

When Sean said there are devices that use TCP/IP he meant that things
like the SB1/SB2/SB3/Receiver/Boom/Radio use TCP/IP to send music over
the ether.

You can't do what you are asking, and you do not want to do what you are
asking.

If you worry about jitter over 3 feet of SPDIF cable, you should not
even think about TCP/IP, it has no jitter because its not real time.

All of this talk about jitter is BS in my opinion. Just don't worry
about it. There are zero scientific studies, experiments, etc. that show
that any level of jitter is bad, or any is good. Without science, you
can't engineer.

People are entitled to have opinions that this or that component doesn't
sound right, and they can claim its due to jitter being too high or too
low. But its just opinion.

No one will make a device that uses TCP/IP in place of SPDIF. Never will.

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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-11-15 Thread Kellen

Thanks for the explanation, Pat.

As far as jitter goes, I'm just along for the ride of what others say
since I don't even know what to listen for or what jitter sounds like.


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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-11-15 Thread Pat Farrell
Kellen wrote:
 As far as jitter goes, I'm just along for the ride of what others say
 since I don't even know what to listen for or what jitter sounds like.

you and the rest of the audiophile world.

One thing that the thread subject did say/write long ago was that the
jitter tests that Stereophile magazine prints are worthless. He didn't
go into much depth on why they were worthless, or what would be more useful.

But I found that interesting, because Stereophile is one of the few
audiophile magazines that even attempts to measure and graph jitter and
the susceptibility of a given DAC or CD player to jitter errors. What
Stereophile does not attempt to do is indicate what the audible meaning
of the measurements are.

Its about the music.

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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-11-13 Thread wayne325

Themis;479827 Wrote: 
 It means 17 picoseconds. The consensus is declaring the total jitter :
 http://en.wikipedia.org/wiki/Jitter

Jitter is specified as RMS jitter, not peak or peak-peak or any other
some
such defintion.

THere are machines
which fairly nicely spit out the number for you if you can get a probe
on
the signal being checked.  Usually any high-end digital scope nowadays
will have this feature.  (high end means you paid probably $50k)


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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-11-13 Thread wayne325

Phil Leigh;480300 Wrote: 
 There's nothing new in what Naim are doing...
 
 http://www.scalatech.co.uk/papers/aes93.pdf

Of course.  NAIM are using commercially available parts.  They cost $10
at
retail.  They haven't the money to do any of this themselves. All
they're
using is an asynchronous converter.

I made a DAC that has the same thing in it. The trick I pulled is my
DAC
sends a clock to my transporter, so my DAC's asynchronous converter
isn't
doing any sample additions or deletions, and a local clock clocks the
DAC chips directly so nearly no jitter, no interface problems such
as the papers in this thread talk about.  Beautiful.

The guys that were doing any real digital processing in the old days
were
Wadia.  And look what their stuff cost.  Now it looks like others are
playing with silicon a bit, but again box costs are $10k each, and
often you
need 3 boxes - including one that has an atomic clock.  (hahahahaah -
ya
now I'm sure THAT makes a big difference.)

If you want to play with this stuff, about the only way to go is with
an
FPGA.  Guys that design this stuff are kinda rare, and it's very
expensive
to develop.


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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-11-09 Thread cliveb

Andy8421;483292 Wrote: 
 Just one point I wanted to pick up on - the Naim DAC approach - where
 the data input data is written into a large ring buffer using the
 derived jittered clock, then read out using a stable local clock -
 wouldn't this  eliminate interface jitter for an asynchronous link?
Naim's ring-buffer approach just averages out the jitter over a longer
time base. In principle it does the same thing as a low frequency PLL.
As long as the DAC receives data from a transport with its own
free-running clock, surely it is in principle impossible to completely
eliminate the incoming jitter. Doesn't matter how you try to average it
out (ring buffer, PLL, ASRC) - ultimately the DAC is a slave to the
transport's clock.


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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-11-09 Thread Themis

JohnSwenson;483311 Wrote: 
 If you use a fixed frequency clock you have the problem that you need a
 fairly large buffer to handle an hours worth or so of music. That takes
 very creative buffer management when people want to skip songs, change
 playlists, fast forward etc. With a large enough buffer to handle
 significant frequency deviations from external to internal clock AND
 long playtimes before re-synching the lag in user interface could be
 very objectionable to many people.Is it possible to have a buffer (accurate 
 but without a very complicated
buffer management) for playback, and another (slave to the source clock)
for the other functions like skip, FF etc. ?


-- 
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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-11-08 Thread Andy8421

John,

Thank you for your thoughts. 

Just one point I wanted to pick up on - the Naim DAC approach - where
the data input data is written into a large ring buffer using the
derived jittered clock, then read out using a stable local clock -
wouldn't this  eliminate interface jitter for an asynchronous link? 

Running the local clock at the very long run average frequency of the
input jittered clock and having the capacity to hold many seconds worth
of data in RAM would allow link jitter to be effectively eliminated.  As
you point out, this doesn't stop the DAC introducing jitter of its own.

Andy.


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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-11-08 Thread JohnSwenson

There are some difficulties with the large buffer approach. You mention
using a local clock that is the long term average of the input data
rate, again this is some form of adaptive.

If you use a fixed frequency clock you have the problem that you need a
fairly large buffer to handle an hours worth or so of music. That takes
very creative buffer management when people want to skip songs, change
playlists, fast forward etc. With a large enough buffer to handle
significant frequency deviations from external to internal clock AND
long playtimes before re-synching the lag in user interface could be
very objectionable to many people.

Its certainly an approach that should be looked into as an option, but
it has not been done very often. I tried this several years ago and the
buffer management issues became so messy I gave up.

And even the companies that have implemented such an approach have
still had interface jitter affect the output sound. It seems that even
with what should be theoretically completely immune getting the details
perfect is not easy.

My take as a designer i sI'm not going to aim for perfection. I'm going
to do my best to implement a technically robust interface method and do
my best to implement it well and let the chips fall where they will. I
have never achieved a DAC that is completely immune to outside
conditions and I probably never will, so I'm not going to worry about
it. The best done DACs sound so good I'm not going ot worry about a $500
cable making it sound slightly different. I would like to design
something that is not the case, but I've given up on that as not really
necessary for achieving  excellent sound. 

John S.


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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-11-07 Thread JohnSwenson

I think I'll weigh in on this, I have a fair amount of experience with
the subject. For those that don't know I have been designing and
building DACs for many years (for fun, nothing I have done has made it
into a commercial product (yet, hopefully that will change soon)).

I think its important to distinguish between jitter at the DAC chip and
jitter of the interface. I am a firm believer that jitter at the DAC
chip DOES matter and can significantly affect the sound comming out of
the DAC. There have been statements in this thread that jitter does not
matter, if the poster was referring to jitter at the DAC chip then I
disagree strongly. 

If the reference was to interface jitter, then I partially agree.
Theoretically if the interface is done right jitter on the interface
should not affect sound quality. Unfortunately implementations that
achieve this are few and far between. Most of the interfaces out there
today are still susceptible to interface jitter, including ones whose
marketing literature say otherwise. 

I'm going to touch on the first point first on jitter at the DAC chip.
There are some people who believe certain AES papers that say that any
jitter less than 1ns is inaudible, so anybody trying to get less jitter
than that is a snake oil salesmen and just out to get your money. 

There is no way I believe this 1ns jitter threshold, its just plain
wrong. So where is it? Thats hard to say, partly because measuring very
low jitter is hard to do, and because jitter is not just a number its a
spectrum and the spectrum influences the sound significantly. Thus 50ps
of one type my be audible and 100ps of another may no be. This makes
doing threshold tests rather difficult. My take on this is that if you
hear someone say jitter below some number is inaudible and above that
number its audible, I would take that with a grain of salt. 

My experiments have shown that jitter in the 50ps range is definitely
audible.  I've done an experiment with an SB3 where I replace the stock
crystal with a low jitter oscillator which supposedly has 3ps of jitter.
I get significantly better sound out of the analog outs. (I don't know
exactly what the jitter was at the DAC chip since my test equipment can
only go down to about 15ps, it was 15ps or less). The stock setup comes
out to 50-60ps or so. 

I did some blind tests with this and it was audible by a number of
people. (No I did not run a full controlled publication quality refereed
test, that wasn't my purpose, If someone really wants to do so I'll be
glad to help set it up, but I don't have the time or money to do it all
myself)

I've done many other such tests with different equipment where I have
tried to just change the jitter and I can definitely hear a difference.
For me this is enough evidence that jitter difference down in the less
than 50ps range IS audible in many circumstances, hence it is important
to design systems to control it. 

Now on to interface jitter. I THINK this is what most of the people on
this thread are talking about. There are several different classes of
interfaces, one grouping is adaptive and asynchronous. In adaptive the
clocking which goes to the DAC chip has to adapt to the data rate coming
from the source (examples are S/PDIF, adaptive USB, adaptive firwire).
In asynchronous there is a fixed frequency local clock and the source is
somehow slaved to this local clock. 

So far no one has been able to design an adaptive clock system that is
immune to interface jitter, some are much better than others, but they
all can still be affected by the interface. The asynchrounous interface
can theoretically be designed so it does not depend on the interface
jitter at all. Unfortunately there are not that many of these out there.
The squeezebox interface IS one of those, and thats one of the things to
drew me to it in the first place. 

There is sort of a weird hybrid system which uses an adaptive
interface, then feeds it into an asynchronous sample rate converter
(ASRC) driven by a local clock. On the surface this seems like a good
solution but in reality interface jitter can affect the sample rate
tracking which can actually change the data such that sound is slightly
distorted even though the actual clock going to the DAC chip is very low
jitter. 

Even some asynchronous interfaces are not implemented perfectly and are
STILL affected by interface jitter. Even some very well implemented ones
can still be slightly affected by interface jitter. This state of
affairs has caused some people to then state that the whole concept of
asynchronous interface was a bunch of marketing snake oil in the first
case. Its still a valid way of reducing jitter, its just that achieving
absolute perfection at anything is difficult! 

The result is that as of today there are a number of products where
interface jitter has a very small impact on sound quality, yet there are
still a large number of products being sold and a large number of
existing products that ARE 

Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-11-06 Thread Andy8421

Phil and Themis,

Thanks for posting both papers. I dont't have Pat's degree in maths,
but my degree in EE allowed me to struggle through.

Very interesting stuff.  The most interesting point for me was that the
S/PDIF encoding process (biphase mark) itself can introduce jitter in a
band limited link. All links are band limited to some degree, so even
with a zero jitter source, the recovered clock will exhibit jitter - to
make matters worse this is not random, but is a function of the encoded
audio signal.

As to what level this becomes audiable, the paper has a stab at
answering.  There were suggestions of 10ns to 200ps for a 16bit signal
down to 8ps for a 20 bit signal.  This is of course total jitter over
the entire link, not just jitter at the source.

I guess the real test of whether this is audiable is to be able to
switch jitter into an S/PDIF signal, do an A/B comparison with the
unjittered signal and see where it was noticable. The neat thing from
the paper is that this could be done with a simple RC low pass filter in
the biphase mark bitstream. I would give it a try, but I do not have
access to test equipment to allow me to measure the resulting clock
jitter.

The other neat thing (for me, perhaps I need to get out more) was that
the audio material could clearly be heard on the PLL loop control
voltage - implying that the PLL clock was being modulated by phase
errors on the input datastream correlated to the encoded audio signal.
Neat, huh? 

Is there any recent published material that performs A/B listening
tests for varying levels of jitter?  The test mentioned in the paper was
a while back and equipment has moved on.

None of this has answered Pat's question whether levels of jitter
exhibited by the current generation of equipment matter, but I can at
least see the mechanism know how it might.


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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-11-03 Thread Themis

Phil Leigh;480806 Wrote: 
 Pat - Did you read that AES paper I posted? The Hawksford/Dunn one?
 http://www.scalatech.co.uk/papers/aes93.pdf
You have also an easier to understand explanation by Dunn here :
http://www.stereophile.com/features/396bits/


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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-11-03 Thread cliveb

Phil Leigh;480806 Wrote: 
 Pat - Did you read that AES paper I posted? The Hawksford/Dunn one?
 http://www.scalatech.co.uk/papers/aes93.pdf
That's a lengthy paper with lots of complicated maths. I don't pretend
to understand all of it. It certainly demonstrates that the SPDIF
interface has shortcomings, but we all knew that anyway.

As far as its relevance to the question in hand - whether real-world
levels of jitter are actually audible, it says little. The authors
construct a theoretical model of what might conceivably be audible in
extreme pathological cases. They conclude that 180ps of random jitter
might just about be audible at 20kHz, with higher levels of jitter
required to be audible as the frequency reduces, and this when using a
receiver with no PLL filtering. They go on to explain that using a PLL
with a low cutoff frequency (which pretty much all modern half-decent
DACs use) alleviates the problems.

At no point are any actual listening tests used to confirm or refute
these speculations. (I'm not criticising the paper for that - as far as
I can see the purpose of the paper isn't to work out what *is* audible,
but what *might* be audible, in order to inform design engineers of
techniques thay can use to avoid the problem).

So, what can we conclude from this paper? That a DAC with a low
frequency PLL recovering the clock, jitter levels well above 200ps are
almost certainly inaudble, even under idealised listening conditions
unlikely to ever be encountered in the real world. That is not a million
miles away from the conclusions drawn by Benjamin and Gannon, based on
controlled listening tests, that jitter below about 10ns is undetectable
on actual music signals.

Meanwhile we have people obsessing over sub-20ps jitter, worrying that
anything greater will ruin their listening experience. Doesn't make
sense to me.


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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-11-03 Thread Themis

No-one really cares for sub-20ps jitter.
Since no jitter metrics are indicated nor analyzed by the industry
(which prefers the THD @ 1kHz ^^) we'll continue to listen and make
subjective comparisons. :D


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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-11-03 Thread Pat Farrell
cliveb wrote:
 Phil Leigh;480806 Wrote: 
 Pat - Did you read that AES paper I posted? The Hawksford/Dunn one?
 http://www.scalatech.co.uk/papers/aes93.pdf
 That's a lengthy paper with lots of complicated maths. I don't pretend
 to understand all of it. It certainly demonstrates that the SPDIF
 interface has shortcomings, but we all knew that anyway.
 
 As far as its relevance to the question in hand - whether real-world
 levels of jitter are actually audible, it says little. 

I read it fairly carefully. Perhaps this is one case where having a
degree in Math is helpful.

Its a nearly 20 year old paper with one claim, that some jitter is audible.

It does say that the solution is to have your Transporter provide clock
if you are feeding it from a SPDIF input, but everyone knows that now.

I found no serious support for the claim that jitter is an issue today
for moderate audiophile components.


 Meanwhile we have people obsessing over sub-20ps jitter, worrying that
 anything greater will ruin their listening experience. Doesn't make
 sense to me.

Right, in all audio engineering, the key is to know what measurements
are important and which are not. And in some areas, we don't know what
the important metrics are. This brought the 80s karma that all amps
measure the same, so they sound the same. Not the right conclusion, the
proper one is that the engineers were not measuring the right thing.

I remain skeptical that jitter is meaningful.


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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-11-03 Thread Themis

pfarrell;481175 Wrote: 
 
 I remain skeptical that jitter is meaningful.
 
I second Pat on this (although I don't have his background) : at one
point I was hoping the industry would bring some clear view on jitter,
although it seemed a secondary point. Instead, as time passes and clocks
and resampling get better, the problem (if any) tends to disappear even
on low-end dacs.
I think that in 5-10 years it will be forgotten, unless you use an
outdated dac.


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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-11-03 Thread Peter314

waiting for Sean-o :)


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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-11-02 Thread Patrick Dixon

pfarrell;480376 Wrote: 
 
 
 Seriously, just saying that its important and telling me to listen more
 carefully is not engineering.
 
No, but pyschoacoustics is where you need to look.  Engineering will
tell you how it makes a difference and what (numerical) levels that
might be at, but engineering can't tell you whether that's significant
when listening to music.

Unfortunately spending 10 minutes doing AB testing probably won't
either.


-- 
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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-11-02 Thread Patrick Dixon

Themis;480374 Wrote: 
 do you insinuate that the technical specification given by Slimdevices
 for the TP are intentionally  false, incomplete or misleading ? :)
No I'm just saying the number on it's own is completely meaningless as
you have to understand the complete context for where it came to make
any kind of sense of it.  These sort of raw numbers are for marketing
purposes only, and they're why you can't make meaningful comparisons of
audio kit on the internet.

PS. Yes of course you're right - the real answer is 42.


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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-11-02 Thread Themis

Patrick Dixon;480617 Wrote: 
 No I'm just saying the number on it's own is completely meaningless as
 you have to understand the complete context from where it came to make
 any kind of sense of it.  These sort of raw numbers are for marketing
 purposes only, and they're why you can't make meaningful comparisons of
 audio kit on the internet.
Listening for a few days at home is the best solution, I agree.
Sure, some experience of the people (designers) involved may help, more
than the numbers.

Patrick Dixon;480617 Wrote: 
 PS. Yes of course you're right - the real answer is 42.:D


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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-11-02 Thread Pat Farrell
Patrick Dixon wrote:
 pfarrell;480376 Wrote: 
 Seriously, just saying that its important and telling me to listen more
 carefully is not engineering.

 No, but pyschoacoustics is where you need to look.  

You are not making your case there.

I'm willing to do a little work on it, but since I mostly think its
marketing hype, to make a case, you need to point directly to the backup
arguments.

And not on some opinion forums, real papers, real science and engineering.

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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-11-02 Thread Phil Leigh

pfarrell;480660 Wrote: 
 Patrick Dixon wrote:
  pfarrell;480376 Wrote: 
  Seriously, just saying that its important and telling me to listen
 more
  carefully is not engineering.
 
  No, but pyschoacoustics is where you need to look.  
 
 You are not making your case there.
 
 I'm willing to do a little work on it, but since I mostly think its
 marketing hype, to make a case, you need to point directly to the
 backup
 arguments.
 
 And not on some opinion forums, real papers, real science and
 engineering.
 
 -- 
 Pat Farrell
 http://www.pfarrell.com/

Pat - Did you read that AES paper I posted? The Hawksford/Dunn one?
http://www.scalatech.co.uk/papers/aes93.pdf


-- 
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You want to see the signal path BEFORE it gets onto a CD/vinyl...it
ain't what you'd call minimal...
SB Touch Beta (wired) - TACT 2.2X (Linear PSU) + Good Vibrations S/W -
MF Triplethreat(Audiocom full mods) - Linn 5103 - Aktiv 5.1 system (6x
LK140's, ESPEK/TRIKAN/KATAN/SEIZMIK 10.5), Townsend Supertweeters, Blue
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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-11-01 Thread Andy8421

Pat,

I am not sure I agree on jitter.  My simplistic view of a digital audio
stream is that if plotted on a graph, the 'y' coordinates are provided
by the digital amplitude of the sample, the 'x' coordinates are implied
by the (assumed) regular sample period.  So to plot a point on the graph
accurately, you need to have both x and y coordinates correct. Having
the sample at the wrong time is just as bad as having a sample that is
the wrong size.

I suppose the real question is whether the level of jitter experienced
in the clock derived from S/PDIF is in any way significant from an audio
perspective. If my maths was up to it, I could probably take a jitter
distribution and convert it into either an implied noise or a distortion
figure in the derived audio signal. I would then be able to see how many
picoseconds of jitter mattered or whether this is negligable. Any link
to analysis on this would be welcome.

If the NAIM ring buffer approach catches on then I guess this all
becomes irrelevant. In the scheme of tricky hardware, reading in a
jittered signal into a ring buffer and reclocking out at a stable rate
with a local accurate clock is pretty basic. It feels like something
that could be very easily and cheaply implemented in all external DACs.

To put this in perspective, and again at the risk of showing my age, I
have seen timebase correctors that performed a very similar function
de-jittering a video signal built using delay lines that dated from the
60s. For an even older example, I have recently visited the telegraph
museum at Porthcurno in Cornwall where they have on display a reclocking
system that takes a jittered high speed morse signal from a
transatlantic cable, derives a clock from the incoming data, resamples
and de-jitters - and this is pre second world war and built with gears,
motors and switches.

To be fair, neither approach could cope with much jitter, and neither
had the luxury of large ring buffer storage, but it goes to show that
there is nothing new under the sun.


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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-11-01 Thread Pat Farrell
Andy8421 wrote:
 I am not sure I agree on jitter.  My simplistic view of a digital audio
 stream is that if plotted on a graph, the 'y' coordinates are provided
 by the digital amplitude of the sample, the 'x' coordinates are implied
 by the (assumed) regular sample period.  So to plot a point on the graph
 accurately, you need to have both x and y coordinates correct. Having
 the sample at the wrong time is just as bad as having a sample that is
 the wrong size.


Right. I understand how jitter could be important in theory.

I just don't believe that it actually is in practice for real audiophile
gear or for 99% of humans listening.

Gross clocking errors would be bad. I just don't see evidence that we
are getting them.

Rather, I see it as a marketing phrase that can be used to claim that
one box is better than another. When you have perfect sound forever
its hard to justify spending $10,000 for a transport over the $20
transport in your PC.

Now if the CD was designed today, instead of way back before
microprocessors were common, it could have been made better, and the
possibility of jitter designed out. But that didn't happen.

This is personal belief, but its based on the fact that I have never
seen any engineering evidence that jitter is a real problem.

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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-11-01 Thread Phil Leigh

There's nothing new in what Naim are doing...

http://www.scalatech.co.uk/papers/aes93.pdf


-- 
Phil Leigh

You want to see the signal path BEFORE it gets onto a CD/vinyl...it
ain't what you'd call minimal...
SB Touch Beta (wired) - TACT 2.2X (Linear PSU) + Good Vibrations S/W -
MF Triplethreat(Audiocom full mods) - Linn 5103 - Aktiv 5.1 system (6x
LK140's, ESPEK/TRIKAN/KATAN/SEIZMIK 10.5), Townsend Supertweeters, Blue
Jeans Digital,Kimber Speaker  Chord Interconnect cables
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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-11-01 Thread loserica

Phil Leigh;479949 Wrote: 
 I could be wrong, but I'm not sure Sean has commented on the Touch in
 this respect. I must say that I find the Touch definitely superior to
 the SB3 in this regard. I'm not convinced that a TP is really noticeably
 better than the Touch when used as a digital transport.

Thanks. If the Touch is definitely superior to the SB3, maybe $1300 for
one of the audiogon.com Transporters could be a good buy, especially
taking into account the XLR digital output and the word clock in of the
TP.

This are Sean's comments :

http://forums.slimdevices.com/showpost.php?p=454830postcount=23

Now, I would say that the SB3 has a good digital output, and I am
hoping that the TP would have an extremely good one, as Sean said :)


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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-11-01 Thread Patrick Dixon

Themis;479827 Wrote: 
 It means 17 picoseconds. The consensus is declaring the total jitter :
 http://en.wikipedia.org/wiki/Jitter

I am pretty sure 27 is the answer to everything.

Pat, if you don't think jitter is relevant, then you haven't been
listening hard enough.


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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-11-01 Thread Pat Farrell
Patrick Dixon wrote:
 Pat, if you don't think jitter is relevant, then you haven't been
 listening hard enough.

I do serious listening through my Transporter.

I clock all of the ADC and DAC to a common clock in my studio.

I don't care about serious listening with my Boom, Duet or ancient SB1.

Seriously, just saying that its important and telling me to listen more
carefully is not engineering.

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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-11-01 Thread Themis

Patrick Dixon;480362 Wrote: 
 I am pretty sure 27 is the answer to everything.
If it is a joke, then the answer should be 42. ;)

If it is not, you need to be more explicit : do you insinuate that the
technical specification given by Slimdevices for the TP are
intentionally  false, incomplete or misleading ? :)


-- 
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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-10-31 Thread Themis

Patrick Dixon;479752 Wrote: 
 Yes, but what does 17ps mean?
It means 17 picoseconds.


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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-10-31 Thread cliveb

loserica;479729 Wrote: 
 Well, in this scenario, let's suppose I want just a different flavor,
 so my question would still make sense :)
So you're saying you'd like to use a non-neutral DAC to voice the
playback. Nothing wrong with that. But if the DAC you use imparts its
own characteristic sound, my guess would be that the (debatable) effect
that differing amounts of source jitter might have would most likely be
swamped by the deliberate voicing in the DAC. In other words, it still
doesn't make any sense to use a TP instead of a regular SB purely as a
digital transport.


-- 
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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-10-31 Thread loserica

cliveb;479853 Wrote: 
 So you're saying you'd like to use a non-neutral DAC to voice the
 playback. Nothing wrong with that. But if the DAC you use imparts its
 own characteristic sound, my guess would be that the (debatable) effect
 that differing amounts of source jitter might have would most likely be
 swamped by the deliberate voicing in the DAC. In other words, it still
 doesn't make any sense to use a TP instead of a regular SB purely as a
 digital transport.

My comment meant to be a half-joke, in response to this affirmation.
The DAC in the transporter is better than any DAC you may have.. I am
sure that the DAC in the transporter is very good, but why shouldn't I
have the possibility to try for myself some external DAC's (one example
would be EMM DCC2SE that one of my friends owns) and have my own
conclusions.

Really, my problem is that I found out that the digital output of the
Duet is not as good as it gets (and that using a moderately priced DAC
such as the Lavry DA-10). I have tried three different transports, and
all were audibily better than the SB3 digital out.

So, what I am really asking is : can the TP digital out match the
quality of any really good transport, because clearly the Duet digital
out CAN NOT.


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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-10-31 Thread Themis

loserica;479873 Wrote: 
 So, what I am really asking is : can the TP digital out match the
 quality of any really good transport, because clearly the Duet digital
 out CAN NOT (I am considering to use a really high end DAC, and not the
 Lavry, since obviously the TP + DA-10 may be of the same quality with
 the TP analog outs)
If you can hear the differences between digital transports, then imho
you will find TP a good improvement, especially through the XLR
connector.


-- 
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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-10-31 Thread cliveb

loserica;479873 Wrote: 
 My comment was meant to be a half-joke, in response to this affirmation
 : The DAC in the transporter is better than any DAC you may have.
Fair enough. It is obvious that there could be better DACs available.

loserica;479873 Wrote: 
 Really, my problem is that I found out that the digital output of the
 Duet is not as good as it gets (and that using a moderately priced DAC
 such as the Lavry DA-10). I have tried three different transports, and
 all were audibily better than the SB3 digital out.
Interesting. Given that the Lavry DA10 is an example of a DAC that
claims to be completely immune to input jitter, perhaps the differences
between transports that you hear might possibly be nothing to do with
jitter (in which case focussing on the TP's jitter spec may not tell you
what you need to know). There are other plausible explanations, such as
EMI fields from the transports that impinge differently on the DA10's
analogue circuitry, or (if you're not using an optical connection) noise
on the ground plane of the transport being injected into the DAC. (If
you were doing the listening tests sighted, it could simply be
psychological factors at work).

My feeling is that the only way you will find out is to borrow a TP and
see how it performs in your system using your DAC of choice. You clearly
don't want to buy one on the off-chance it will sound better than an SB3
as a transport.


-- 
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Transporter - ATC SCM100A

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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-10-31 Thread loserica

cliveb;479898 Wrote: 
 
 Interesting. Given that the Lavry DA10 is an example of a DAC that
 claims to be completely immune to input jitter, perhaps the differences
 between transports that you hear might possibly be nothing to do with
 jitter
 

Well, I thought it was immune to input jitter, but it seems it really
isn't. We used both a coaxial and an optical connection, and all the
four persons present at the test including me heard the difference.
Lavry DA-10 is a good product, just that it is not jitter free (though
Lavry Gold might be)

cliveb;479898 Wrote: 
 
 My feeling is that the only way you will find out is to borrow a TP and
 see how it performs in your system using your DAC of choice. You clearly
 don't want to buy one on the off-chance it will sound better than an SB3
 as a transport.

Of course that this would be the logical thing to do. But I live in
Romania, Europe, and I don't know of anyone who has a Transporter, and I
cannot just buy one and return it, as this policy is not valid here. Of
course that I don't want to buy it just to be disappointed, that is why
I was asking.

Thanks


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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-10-31 Thread Stratmangler

YOu might be better off (cost wise) with a Touch with an external DAC.

Here is a review of a Transporter with an external DAC
http://theartofsound.net/forum/showpost.php?p=62300postcount=245

Chris:)


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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-10-31 Thread loserica

Stratmangler;479934 Wrote: 
 YOu might be better off (cost wise) with a Touch with an external DAC.
 
 Here is a review of a Transporter with an external DAC
 http://theartofsound.net/forum/showpost.php?p=62300postcount=245
 
 Chris:)

I am not looking for the best solution performance wise, not cost wise.
I am willing to buy the Transporter to get a bit of improvement over the
Touch, if this is the case.

I think that Sean once said that the digital out of the Transporter
should be better than the SB3 and the Touchbut I needed a
confirmation :)


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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-10-31 Thread JezA

NAIM Audio have announced an interesting new DAC, which should hit the
shops shortly. It features a large RAM buffer and no PLL clocks, so
should be largely immune from the effects of jitter in the source. If it
does what NAIM claim it does, it may well be possible to drive it
adequately from a squeezebox. There is a SYNC light on the front which
indicates whether the source is within acceptable tolerance - if it
lights, data will be buffered and reclocked so jitter would not be an
issue. So, I would wait until this DAC comes out, and see which sources
will sync with it.

You can read about the NAIM DAC here: (the white-paper is worth reading
too)

http://www.naim-audio.com/products/dac.html


-- 
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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-10-31 Thread Phil Leigh

loserica;479941 Wrote: 
 I am not looking for the best solution performance wise, not cost wise.
 I am willing to buy the Transporter to get a bit of improvement over the
 Touch, if this is the case.
 
 I think that Sean once said that the digital out of the Transporter
 should be better than the SB3 and the Touchbut I needed a
 confirmation :)

I could be wrong, but I'm not sure Sean has commented on the Touch in
this respect. I must say that I find the Touch definitely superior to
the SB3 in this regard. I'm not convinced that a TP is really noticeably
better than the Touch when used as a digital transport.


-- 
Phil Leigh

You want to see the signal path BEFORE it gets onto a CD/vinyl...it
ain't what you'd call minimal...
SB Touch Beta (wired) - TACT 2.2X (Linear PSU) + Good Vibrations S/W -
MF Triplethreat(Audiocom full mods) - Linn 5103 - Aktiv 5.1 system (6x
LK140's, ESPEK/TRIKAN/KATAN/SEIZMIK 10.5), Townsend Supertweeters, Blue
Jeans Digital,Kimber Speaker  Chord Interconnect cables
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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-10-31 Thread Rodney_Gold

I have compared the TP and SB3 as well as some higher end dedicated
transports (ML and Theta) as pure transports into various equipment ,
there was no audible difference - I also compared the TP via spdiff ,
toslink and AES/EBU and found no reliable  audible difference between em
- I used premium cables in all cases.
I do however not believe that the TP's dac is the best out there as
others I have tried seem to have better bass and more air or low level
resolution. Im currently auditioning a PS audio perfect wave dac vs the
Tp's. 
The one thing I have found in the digital domain , especially using a
few DSP units in the effects loop is that various flavours of dither and
noise shaping have a fairly large effect on the sonics. For example I
have recently inserted a Behringer deq2496 in the loop for some
headphone eq and merely changing it's dither from off to 24 bit has a
huge impact on sonics.
To cut a long story short - if you need a transport with spdif out only
, there is no point in spending almost 10x the price of a SB3/classic
for a TP.


-- 
Rodney_Gold

Sb3/Z-sys RDP1/meridian DSP5500's
TP/X-cans v3/Senns 650's
TP/SCM 50a's
SB3/Meridian DSP5000's
TP/PS audio perfectwave DAC/woo audio Wa2 Headphone amp/Sehneisser
Hd800's

The nicest thing about smacking your head against the wall is...the
feeling you get when you stop

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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-10-31 Thread Pat Farrell
Rodney_Gold wrote:
 The one thing I have found in the digital domain , especially using a
 few DSP units in the effects loop is that various flavours of dither and
 noise shaping have a fairly large effect on the sonics.


Unlike jitter, which I believe to be mostly audiophile hype, dither and
noise shaping are real. It does not take much to tell a proper dither
from a naive truncation after you apply some DSP to the signal.

This is why a lot of post-production is done using 24 bit processing,
even if the source is just 16 bit. By keeping it 24 bits and then doing
proper dithering as you output the 16 bits for RedBook, it makes a big
and easily audible difference.

There are also serious arguments about which dithering algorithm to use,
but the short answer is that any real dithering algorithm when going
from 24 bits to 16 is way better than just chopping off the low order bits.

-- 
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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-10-31 Thread Rodney_Gold

Yeh , I used to use a Meridian 518 to do dither downwards in the old
days ...
I just wondered why , without any eq , the DEQ had such a pronounced
difference between 24 bit and off ..off had deeper bass and a lot more
ambience than 24 bit which sounded more toe tapping and had a more
staccato type of bass and far less ambeince ...obviously something to
do with the 2496's manipulation of the signal as it comes out the
digital portion of the TP (using AES). I was listening thru the phones
and it really was an instantly noticeable difference.
In fact , I have found just inserting a digital box even when used in
bypass mode , has a sonic signature in a lot of digital equipment , for
example just using the meridian 518 and merely passing thru a 16/44.1
signal from a transport had a sonic effect - not always a desireable one
at that.


-- 
Rodney_Gold

Sb3/Z-sys RDP1/meridian DSP5500's
TP/X-cans v3/Senns 650's
TP/SCM 50a's
SB3/Meridian DSP5000's
TP/PS audio perfectwave DAC/woo audio Wa2 Headphone amp/Sehneisser
Hd800's

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[SlimDevices: Audiophiles] Query for Sean Adams

2009-10-30 Thread gregeas

Sean,

I have a quick non-technical question for you about the Transporter.
Can you email me at gw...@yahoo.com?

Thanks,

Greg


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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-10-30 Thread loserica

Excuse me if I am hijacking your thread, but I also have a quick
question for Mr. Adams, and if you don't mind, maybe he will be so kind
to respond to me on this thread.

Do you think that purchasing a Transporter in order to use just its
digital outputs makes sense ? (I already have both a SB3 and a Duet). In
other words, what do you think about the jitter of the S/PDIF outputs of
SB3 vs Transporter (and maybe add Touch to this equation).

Sorry again for intruding,
Thanks


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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-10-30 Thread radish

gregeas;479628 Wrote: 
 Sean,
 
 I have a quick non-technical question for you about the Transporter.
 Can you email me at gw...@yahoo.com?
 
 Thanks,
 
 Greg

Why wouldn't you just ask your question here?


-- 
radish

'HELP ME RAISE MONEY FOR CHILDREN'S CANCER RESEARCH!'
(HTTP://WWW.ADAMREEVE.COM/24IN24/)
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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-10-30 Thread Pat Farrell
radish wrote:
 gregeas;479628 Wrote: 
 I have a quick non-technical question for you about the Transporter.
 Can you email me at gw...@yahoo.com?
 
 Why wouldn't you just ask your question here?

Yeah, and @gregeas, you do realize that Sean has retired, and only
occasionally reads the forums out of the goodness of his heart.

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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-10-30 Thread Phil Leigh

radish;479646 Wrote: 
 Why wouldn't you just ask your question here?

I was going to ask that, but I decided it might be construed as rude...


-- 
Phil Leigh

You want to see the signal path BEFORE it gets onto a CD/vinyl...it
ain't what you'd call minimal...
SB Touch Beta (wired) - TACT 2.2X (Linear PSU) + Good Vibrations S/W -
MF Triplethreat(Audiocom full mods) - Linn 5103 - Aktiv 5.1 system (6x
LK140's, ESPEK/TRIKAN/KATAN/SEIZMIK 10.5), Townsend Supertweeters, Blue
Jeans Digital,Kimber Speaker  Chord Interconnect cables
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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-10-30 Thread Robin Bowes
On 30/10/09 18:51, Phil Leigh wrote:
 
 radish;479646 Wrote: 
 Why wouldn't you just ask your question here?
 
 I was going to ask that, but I decided it might be construed as rude...

Ha!!!

R.
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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-10-30 Thread Phil Leigh

Robin Bowes;479669 Wrote: 
 On 30/10/09 18:51, Phil Leigh wrote:
  
  radish;479646 Wrote: 
  Why wouldn't you just ask your question here?
  
  I was going to ask that, but I decided it might be construed as
 rude...
 
 Ha!!!
 
 R.
19 exclamation marks is profligate. You have used up your entire supply
for the year..


-- 
Phil Leigh

You want to see the signal path BEFORE it gets onto a CD/vinyl...it
ain't what you'd call minimal...
SB Touch Beta (wired) - TACT 2.2X (Linear PSU) + Good Vibrations S/W -
MF Triplethreat(Audiocom full mods) - Linn 5103 - Aktiv 5.1 system (6x
LK140's, ESPEK/TRIKAN/KATAN/SEIZMIK 10.5), Townsend Supertweeters, Blue
Jeans Digital,Kimber Speaker  Chord Interconnect cables
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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-10-30 Thread SuperQ

loserica;479637 Wrote: 
 Do you think that purchasing a Transporter in order to use just its
 digital outputs makes sense ? (I already have both a SB3 and a Duet). In
 other words, what do you think about the jitter of the S/PDIF outputs of
 SB3 vs Transporter (and maybe those of the Touch).

No, doesn't make sense to me.  To paraphrase a quote from Sean.. The
DAC in the transporter is better than *any* DAC you may have.


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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-10-30 Thread loserica

SuperQ;479700 Wrote: 
 No, doesn't make sense to me.  To paraphrase a quote from Sean.. The
 DAC in the transporter is better than *any* DAC you may have.

Well, in this scenario, let's suppose I want just a different flavor,
so my question would still make sense :)


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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-10-30 Thread Themis

loserica;479637 Wrote: 
 Excuse me if I am hijacking your thread, but I also have a quick
 question for Mr. Adams, and if you don't mind, maybe he will be so kind
 to respond to me on this thread.
 
 Do you think that purchasing a Transporter in order to use just its
 digital outputs makes sense ? (I already have both a SB3 and a Duet). In
 other words, what do you think about the jitter of the S/PDIF outputs of
 SB3 vs Transporter (and maybe those of the Touch).
 
 Sorry again for intruding,
 Thanks
A TP has very (very) carefully designed digital outputs.
It is specified 17ps at DAC output and 35ps at S/PDIF receiver. If you
find much better (or even a little bit better), please let me know what
it is. ;)


-- 
Themis

SB3 - North Star dac 192 - Cyrus 8xp - Sonus Faber Grand Piano Domus

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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-10-30 Thread Patrick Dixon

Themis;479741 Wrote: 
 A TP has very (very) carefully designed digital outputs.
 It is specified 17ps at DAC output and 35ps at S/PDIF receiver. If you
 find much better (or even a little bit better), please let me know what
 it is. ;)

Yes, but what does 17ps mean?


-- 
Patrick Dixon

www.at-tunes.co.uk

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Re: [SlimDevices: Audiophiles] Query for Sean Adams

2009-10-30 Thread Andy8421

Patrick,

OK, I will bite.

Assuming that this wasn't a rhetorical question, it is most likely to
be the standard deviation of the phase difference of the jittered signal
relative to an 'ideal clock' measured in picoseconds.

There are many ways of measuring jitter, so you really need to specify
what you are measuring. Could be max deviation, RMS deviation (which if
I recall my EE maths correctly should equal the 1 SD number if the
jitter is normal in distribution) or a host of other things. There may
be a generally accepted measurement that gets used when referring to
S/PDIF, but if there is, I couldn't find it.

When I was a lad, we measured jitter as a phase angle, but that was
when T1 was considered state of the art.

Andy.


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