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Digital high frequency distortion

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Anonymous
June 17, 2004 2:29:49 AM

Archived from groups: rec.audio.high-end (More info?)

I think what hasn't been stressed about the issue of digital
distortion is that it is not a traditional phase distortion we are
talking about, whereby the phase anomalies are related to frequency in
a linear way, such as would happen in a cheap equalizer.

digital high frequency distortion is more destructive to music because
it makes the waveform fit into a time grid.

on a 100 Hz sine wave this is not so much of a problem because there
are hundreds of samples of one complete wave cycle therefore the wave
is reproduced fairly accurately.

a 5 Khz waveform only has around 10 or so samples for the whole
waveform 10 Khz has half of that and 20 half of half etc.

raising the sample rate does raise bandwith, but the human ear cannot
hear above 20KHz (most people stop at 15-16Khz).

High sample rates are useful because they increase resolution for
high frequencies.

5khz at 192K sampling rate will have more samples to re-build the
waveform with.

reconstruction filters only fill in the holes. If something happens
inbetween the samples it simply isn't recorded in the first place.

Filters just 'connect the dots'


If it were possible to sample at an extremely high sample rate from
the beginning cost-wise (keeping within a consumer price range) it
would already have been done.


remember that 15 years ago digital recorders costed a fortune even
though they only went up to 48khz.

Now the price barrier isn't there anymore.
Anonymous
June 17, 2004 3:59:31 AM

Archived from groups: rec.audio.high-end (More info?)

In article <caqhkt02ugh@news2.newsguy.com>, maxdm
<maxdimario@aliceposta.it> wrote:

> I think what hasn't been stressed about the issue of digital
> distortion is that it is not a traditional phase distortion we are
> talking about, whereby the phase anomalies are related to frequency in
> a linear way, such as would happen in a cheap equalizer.
>
> digital high frequency distortion is more destructive to music because
> it makes the waveform fit into a time grid.
>
> on a 100 Hz sine wave this is not so much of a problem because there
> are hundreds of samples of one complete wave cycle therefore the wave
> is reproduced fairly accurately.
>
> a 5 Khz waveform only has around 10 or so samples for the whole
> waveform 10 Khz has half of that and 20 half of half etc.
>
> raising the sample rate does raise bandwith, but the human ear cannot
> hear above 20KHz (most people stop at 15-16Khz).
>
> High sample rates are useful because they increase resolution for
> high frequencies.
>
> 5khz at 192K sampling rate will have more samples to re-build the
> waveform with.
>
> reconstruction filters only fill in the holes. If something happens
> inbetween the samples it simply isn't recorded in the first place.

There isn't anything "between the samples" that is missed because the
input to the ADC MUST be band limited for the system to work correctly.
If the input to the ADC is correctly band limited, then the
reconstruction filter EXACTLY reproduces the input. Do not trust your
intuition on this. If you have enough of a math background, look into
the math behind discrete time sampling. If you do not have enough math
background to follow the math you will just have to take it on faith.
However, this is proven in the same way that 1+1=2 is proven. It's not
a "theory" - it's a theorem.

>
> Filters just 'connect the dots'
>
>
> If it were possible to sample at an extremely high sample rate from
> the beginning cost-wise (keeping within a consumer price range) it
> would already have been done.
>

If you can't hear above 20 kHz, there is no reason to sample at more
than 40 kHz plus a little extra for the filter transition. It won't
give you any more information below 20 kHz if you sample at 96 kHz. If
new systems are being introduced with higher sample rates it can only
be because 1) it doesn't cost much and it's good marketing fodder, 2)
someone doesn't understand how digital works, or 3) someone believes
that people can hear to 40 kHz. My money is on 1.

Marc Foster

>
> remember that 15 years ago digital recorders costed a fortune even
> though they only went up to 48khz.
>
> Now the price barrier isn't there anymore.
Anonymous
June 17, 2004 6:05:17 PM

Archived from groups: rec.audio.high-end (More info?)

inbetween the samples it simply isn't recorded in the first place.
>
> There isn't anything "between the samples" that is missed because the
> input to the ADC MUST be band limited for the system to work correctly.
> If the input to the ADC is correctly band limited, then the
> reconstruction filter EXACTLY reproduces the input. Do not trust your
> intuition on this. If you have enough of a math background, look into
> the math behind discrete time sampling. If you do not have enough math
> background to follow the math you will just have to take it on faith.
> However, this is proven in the same way that 1+1=2 is proven. It's not
> a "theory" - it's a theorem.
>

This is true if you only consider a waveform as a simple helmholtz
model.
although I do not have a math backround and it sounds like you do, the
issue at hand is a lot simpler:

I know it's hard to talk about something which is difficult to hear
for some people. If the distortion was more evident (in the mid
frequencies) then perhaps more 'complete' theorems to describe what
is happening would have gained popularity, I feel.

musical sound is not only made of harmonic content, or frequency
dependent information.
the best music reproduction systems can accurately reproduce
transients in waveforms in a natural way.

if you were to take an analog lc filter and limit bandwith at 6 KHz
the effect would be vastly different than lowering the sampling
frequency to 6Khz bandwidth on a digital system.

even if you filter the signal going into the adc so it has no harmonic
content over 6 KHz and then filter all the 'steps' created by the dac
out with another filter that limits the bandwith at 6KHz in output the
sound is grainy. Towards the upper end of the spectrum distortion
which is mathematically related to the sampling frequency occurs.

if the waveform manifests a small yet significant waveform
irregularity between samples, the best that a digital system can do is
take the output of the lowpass filter at the next sample and, since
the lowpass filter is supposed (theoretically) to filter out any
irrelevant information (above 6KHz) it is assumed that this is an
accurate representation of what is being recorded.

In practice this is not so. Information gets lost, and as we near the
bandwith limit the system becomes even more unstable because any error
due to jitter or less than perfect brickwall filters/converters.

It is more probable that a digital system operates within it's
theoretical ideal at a point well below it's sampling frequency.

I am sure that math apart, most people who have the capacity to listen
critically to sound will notice a difference in 192Khz audio compared
to 44.1 even if the speakers or headphones or amplifiers are frequency
limited to20 Khz.

Anyone out there with listening experience ?
Related resources
Anonymous
June 18, 2004 2:31:34 AM

Archived from groups: rec.audio.high-end (More info?)

"maxdm" <maxdimario@aliceposta.it> wrote in message
news:caqhkt02ugh@news2.newsguy.com...
> Filters just 'connect the dots'

This is sufficient for perfect reproduction.

An excellent analogy, used in a recently posted link is to imagine a circle.
Only three points on the circumference are required to fully define it, and
any more are an unnecessary waste of bandwidth.
Anonymous
June 18, 2004 2:37:37 AM

Archived from groups: rec.audio.high-end (More info?)

Marc Foster <mfoster2@cfl.rr.com> wrote:
> In article <caqhkt02ugh@news2.newsguy.com>, maxdm
> <maxdimario@aliceposta.it> wrote:
>> reconstruction filters only fill in the holes. If something happens
>> inbetween the samples it simply isn't recorded in the first place.

> There isn't anything "between the samples" that is missed because the
> input to the ADC MUST be band limited for the system to work correctly.
> If the input to the ADC is correctly band limited, then the
> reconstruction filter EXACTLY reproduces the input. Do not trust your
> intuition on this. If you have enough of a math background, look into
> the math behind discrete time sampling. If you do not have enough math
> background to follow the math you will just have to take it on faith.

Alternatively, there is a good graphical ilustration of this counterintuitive
property at:
http://www.lavryengineering.com/documents/Sampling_Theo...

See in pages 23-25 the example of a 17 KHz sine wave sampled at 44.1 KHz.

> However, this is proven in the same way that 1+1=2 is proven. It's not
> a "theory" - it's a theorem.


--
http://www.mat.uc.pt/~rps/

..pt is Portugal| `Whom the gods love die young'-Menander (342-292 BC)
Europe | Villeneuve 50-82, Toivonen 56-86, Senna 60-94
June 18, 2004 2:38:53 AM

Archived from groups: rec.audio.high-end (More info?)

maxdm wrote:
> inbetween the samples it simply isn't recorded in the first place.

You have to understand that if the signal is band-limited to 20KHz,
there is no additional information gained by sampling at higher than
40KHz. Sampling a 20 KHz band-limited signal at 192 KHz results in no
additional information over sampling at 44.1 KHz.

Please note that we are not talking about only sinewaves as input
signals. We are talking about music, speech, or whatever time-varying
waveforms that are band-limited. Music with transients are band-limited.
You have a problem with redbook CD sampling only if you can hear above
20 KHz.

>>
>> There isn't anything "between the samples" that is missed because the
>> input to the ADC MUST be band limited for the system to work correctly.
>> If the input to the ADC is correctly band limited, then the
>> reconstruction filter EXACTLY reproduces the input. Do not trust your
>> intuition on this. If you have enough of a math background, look into
>> the math behind discrete time sampling. If you do not have enough math
>> background to follow the math you will just have to take it on faith.
>> However, this is proven in the same way that 1+1=2 is proven. It's not
>> a "theory" - it's a theorem.
>>
>
> This is true if you only consider a waveform as a simple helmholtz
> model.

No. The key thing you have to understand is that the input waveform is
band-limited, i.e., all its energy are inside a frequency band. For CD,
it is approx. DC to 20 KHz.

Helmholtz models are irrelevant in this discussion.

> although I do not have a math backround and it sounds like you do, the
> issue at hand is a lot simpler:

Sampling theorem may not be intuitive. Your intuition in this case is
wrong. You don't need a strong math background, just basic calculus.

So far you seem to prefer to be led by your intuition instead of doing
the work to understand sampling.
>
> I know it's hard to talk about something which is difficult to hear
> for some people. If the distortion was more evident (in the mid
> frequencies) then perhaps more 'complete' theorems to describe what
> is happening would have gained popularity, I feel.

Sampling does not introduce any distortion, if the input signal is
band-limited. This may not be intuitive.

>
> musical sound is not only made of harmonic content, or frequency
> dependent information.
> the best music reproduction systems can accurately reproduce
> transients in waveforms in a natural way.

Music is band-limited. The best music production systems are also band
limited.

>
> if you were to take an analog lc filter and limit bandwith at 6 KHz
> the effect would be vastly different than lowering the sampling
> frequency to 6Khz bandwidth on a digital system.

This is highly irrelevant to the discussion. Sampling is not like analog
filtering.

>
> even if you filter the signal going into the adc so it has no harmonic
> content over 6 KHz and then filter all the 'steps' created by the dac
> out with another filter that limits the bandwith at 6KHz in output the
> sound is grainy. Towards the upper end of the spectrum distortion
> which is mathematically related to the sampling frequency occurs.
>
> if the waveform manifests a small yet significant waveform
> irregularity between samples,

It cannot, if the waveform is band-limited. There is only one possible
waveform that can be represented by those samples. This is a key to
understand sampling theorem. There cannot be an arbitrary number of
band-limited waveforms with the same samples. It is *not* like
connecting dots.

> the best that a digital system can do is
> take the output of the lowpass filter at the next sample and, since
> the lowpass filter is supposed (theoretically) to filter out any
> irrelevant information (above 6KHz) it is assumed that this is an
> accurate representation of what is being recorded.

Again, you do not understand sampling.
>
> In practice this is not so. Information gets lost, and as we near the
> bandwith limit the system becomes even more unstable because any error
> due to jitter or less than perfect brickwall filters/converters.
>
> It is more probable that a digital system operates within it's
> theoretical ideal at a point well below it's sampling frequency.
>
> I am sure that math apart, most people who have the capacity to listen
> critically to sound will notice a difference in 192Khz audio compared
> to 44.1 even if the speakers or headphones or amplifiers are frequency
> limited to20 Khz.

Any difference is due to implementation differences, and not a result of
the sampling rate difference, if the input signal is band-limited to 20 KHz.

>
> Anyone out there with listening experience ?
>
Anonymous
June 18, 2004 2:41:31 AM

Archived from groups: rec.audio.high-end (More info?)

maxdm <maxdimario@aliceposta.it> wrote:
> inbetween the samples it simply isn't recorded in the first place.
> >
> > There isn't anything "between the samples" that is missed because the
> > input to the ADC MUST be band limited for the system to work correctly.
> > If the input to the ADC is correctly band limited, then the
> > reconstruction filter EXACTLY reproduces the input. Do not trust your
> > intuition on this. If you have enough of a math background, look into
> > the math behind discrete time sampling. If you do not have enough math
> > background to follow the math you will just have to take it on faith.
> > However, this is proven in the same way that 1+1=2 is proven. It's not
> > a "theory" - it's a theorem.
> >

> This is true if you only consider a waveform as a simple helmholtz
> model.
> although I do not have a math backround and it sounds like you do, the
> issue at hand is a lot simpler:

> I know it's hard to talk about something which is difficult to hear
> for some people. If the distortion was more evident (in the mid
> frequencies) then perhaps more 'complete' theorems to describe what
> is happening would have gained popularity, I feel.

> musical sound is not only made of harmonic content, or frequency
> dependent information.
> the best music reproduction systems can accurately reproduce
> transients in waveforms in a natural way.

> if you were to take an analog lc filter and limit bandwith at 6 KHz
> the effect would be vastly different than lowering the sampling
> frequency to 6Khz bandwidth on a digital system.

> even if you filter the signal going into the adc so it has no harmonic
> content over 6 KHz and then filter all the 'steps' created by the dac
> out with another filter that limits the bandwith at 6KHz in output the
> sound is grainy. Towards the upper end of the spectrum distortion
> which is mathematically related to the sampling frequency occurs.

> if the waveform manifests a small yet significant waveform
> irregularity between samples, the best that a digital system can do is
> take the output of the lowpass filter at the next sample and, since
> the lowpass filter is supposed (theoretically) to filter out any
> irrelevant information (above 6KHz) it is assumed that this is an
> accurate representation of what is being recorded.

> In practice this is not so. Information gets lost, and as we near the
> bandwith limit the system becomes even more unstable because any error
> due to jitter or less than perfect brickwall filters/converters.

> It is more probable that a digital system operates within it's
> theoretical ideal at a point well below it's sampling frequency.

> I am sure that math apart, most people who have the capacity to listen
> critically to sound will notice a difference in 192Khz audio compared
> to 44.1 even if the speakers or headphones or amplifiers are frequency
> limited to20 Khz.

> Anyone out there with listening experience ?

How about yourself? You're *sure* that the difference is audible to
a critical listener. I must presume you count youself as one.
So, try a comparison under blind conditions. If you have the
capability, record some 196Khz audio to redbook. Surely you'll pass
with flying colors.

If you want other data points, you might also want to post
your questionn over on rec.audio.tech, and perhaps to
George Massenburg;s Mastering web board

http://webbd.nls.net:8080/~mastering/login


--

-S.
Why don't you just admit that you hate music and leave people alone. --
spiffy <thatsright@excite.co>
Anonymous
June 18, 2004 2:49:21 AM

Archived from groups: rec.audio.high-end (More info?)

In article <wyhAc.46425$2i5.22508@attbi_s52>, maxdm
<maxdimario@aliceposta.it> wrote:

> inbetween the samples it simply isn't recorded in the first place.
> >
> > There isn't anything "between the samples" that is missed because the
> > input to the ADC MUST be band limited for the system to work correctly.
> > If the input to the ADC is correctly band limited, then the
> > reconstruction filter EXACTLY reproduces the input. Do not trust your
> > intuition on this. If you have enough of a math background, look into
> > the math behind discrete time sampling. If you do not have enough math
> > background to follow the math you will just have to take it on faith.
> > However, this is proven in the same way that 1+1=2 is proven. It's not
> > a "theory" - it's a theorem.
> >
>
> This is true if you only consider a waveform as a simple helmholtz
> model.
> although I do not have a math backround and it sounds like you do, the
> issue at hand is a lot simpler:
>
> I know it's hard to talk about something which is difficult to hear
> for some people. If the distortion was more evident (in the mid
> frequencies) then perhaps more 'complete' theorems to describe what
> is happening would have gained popularity, I feel.
>
> musical sound is not only made of harmonic content, or frequency
> dependent information.
> the best music reproduction systems can accurately reproduce
> transients in waveforms in a natural way.
>
> if you were to take an analog lc filter and limit bandwith at 6 KHz
> the effect would be vastly different than lowering the sampling
> frequency to 6Khz bandwidth on a digital system.

No, it wouldn't. The digital system would have to be analog filtered to
6 kHz bandwidth before being sampled. At this point (the input to the
ADC) the signal is identical to the analog waveform. If it is sampled
at the proper rate and the correct reconstruction filter is
implemented, then the reconstructed signal at the output of the digital
system is IDENTICAL to the band limited analog waveform.

>
> even if you filter the signal going into the adc so it has no harmonic
> content over 6 KHz and then filter all the 'steps' created by the dac
> out with another filter that limits the bandwith at 6KHz in output the
> sound is grainy. Towards the upper end of the spectrum distortion
> which is mathematically related to the sampling frequency occurs.

No it doesn't. The use of dither (another REQUIRED part of a
functioning digital system) prevents signal or sample rate correlated
errors from appearing at the output.

>
> if the waveform manifests a small yet significant waveform
> irregularity between samples, the best that a digital system can do is
> take the output of the lowpass filter at the next sample and, since
> the lowpass filter is supposed (theoretically) to filter out any
> irrelevant information (above 6KHz) it is assumed that this is an
> accurate representation of what is being recorded.

If there is a "small yet significant waveform irregularity between
samples" then the signal has not been properly bandlimited prior to the
ADC. If the signal is properly bandlimited then there is no information
in the waveform that is not captured in the sampled data. Once again,
this is well established fact. It was old news when I took digital
signal processing classes in the 1970's.

>
> In practice this is not so. Information gets lost, and as we near the
> bandwith limit the system becomes even more unstable because any error
> due to jitter or less than perfect brickwall filters/converters.
>
> It is more probable that a digital system operates within it's
> theoretical ideal at a point well below it's sampling frequency.
>
> I am sure that math apart, most people who have the capacity to listen
> critically to sound will notice a difference in 192Khz audio compared
> to 44.1 even if the speakers or headphones or amplifiers are frequency
> limited to20 Khz.

If the only difference between the two recordings is the sample rate
then one of the two systems must be broken to produce a real
difference.

>
> Anyone out there with listening experience ?
>
Anonymous
June 18, 2004 7:17:51 AM

Archived from groups: rec.audio.high-end (More info?)

Christopher Key wrote:

> "maxdm" <maxdimario@aliceposta.it> wrote in message
> news:caqhkt02ugh@news2.newsguy.com...
> > Filters just 'connect the dots'
>
> This is sufficient for perfect reproduction.
>
> An excellent analogy, used in a recently posted link is to imagine a circle.
> Only three points on the circumference are required to fully define it, and
> any more are an unnecessary waste of bandwidth.

True enough, but you also must know it's a circle. The
information content of that knowledge is nontrivial, and
without it, an infinity of shapes could be defined by three
points.

So maybe that's not such a great analogy to discribe
reproduction of an audio waveform.

Mike Prager
North Carolina, USA
Anonymous
June 18, 2004 6:36:39 PM

Archived from groups: rec.audio.high-end (More info?)

"Mike Prager" <hifi@ec.rr.com> wrote in message
news:z9tAc.66738$HG.63024@attbi_s53...
> Christopher Key wrote:
>
> > "maxdm" <maxdimario@aliceposta.it> wrote in message
> > news:caqhkt02ugh@news2.newsguy.com...
> > > Filters just 'connect the dots'
> >
> > This is sufficient for perfect reproduction.
> >
> > An excellent analogy, used in a recently posted link is to imagine a
circle.
> > Only three points on the circumference are required to fully define it,
and
> > any more are an unnecessary waste of bandwidth.
>
> True enough, but you also must know it's a circle. The
> information content of that knowledge is nontrivial, and
> without it, an infinity of shapes could be defined by three
> points.

Good point, although the fact that the signal is band limited means that you
do know a certain amount about the signal, ie that it can be expressed as
the sum of a set of sinusoids up to a given frequency.

> So maybe that's not such a great analogy to discribe
> reproduction of an audio waveform.
>

I do accept that it isn't a precise analogy, but thought nonetheless that it
was pretty good for showing people of a non mathematical background
(specifically the OP) roughly what was going on.

Chris Key
Anonymous
June 18, 2004 6:41:31 PM

Archived from groups: rec.audio.high-end (More info?)

Mike Prager <hifi@ec.rr.com> wrote in message news:<z9tAc.66738$HG.63024@attbi_s53>...
> Christopher Key wrote:
>
> > "maxdm" <maxdimario@aliceposta.it> wrote in message
> > news:caqhkt02ugh@news2.newsguy.com...
> > > Filters just 'connect the dots'
> >
> > This is sufficient for perfect reproduction.
> >
> > An excellent analogy, used in a recently posted link is to imagine a circle.
> > Only three points on the circumference are required to fully define it, and
> > any more are an unnecessary waste of bandwidth.
>
> True enough, but you also must know it's a circle.

In exactly the same way that in a properly implemented sampled
system, you KNOW, a priori, that the bandwidth is limited to
less than 1/2 the sample rate.

> The
> information content of that knowledge is nontrivial, and
> without it, an infinity of shapes could be defined by three
> points.

You don't need to know it's a cricle, you need to know simply
that it is a closed continuous function of the form ax + by = c
or something similar.

Everyone, absolutely EVERYONE is simply ignoring the basic tenets
of sampling when making all these handwaving objections about why
it can't work without having the slightest understanding of how
it does.

> So maybe that's not such a great analogy to discribe
> reproduction of an audio waveform.

It's a perfetcly good analogy, FAR more accurate than nonsense about
"connecting the dots" and "stuff between the samples" and the rest
of the high-end hooey being spouted.
Anonymous
June 18, 2004 9:49:59 PM

Archived from groups: rec.audio.high-end (More info?)

Hello Mike,

Mike Prager <hifi@ec.rr.com> wrote in
news:z9tAc.66738$HG.63024@attbi_s53:
....
> True enough, but you also must know it's a circle. The
....
> So maybe that's not such a great analogy to discribe
> reproduction of an audio waveform.

isn't this pretty much the same as the need to know that the original
signal is bandlimited? I think that from this point of view the analogy can
be considered a pretty good one.

Bye,

--
Denis Sbragion
InfoTecna
Tel: +39 0362 805396, Fax: +39 0362 805404
URL: http://www.infotecna.it
Anonymous
June 20, 2004 8:46:21 PM

Archived from groups: rec.audio.high-end (More info?)

Dick Pierce wrote:

> You don't need to know it's a cricle, you need to know simply
> that it is a closed continuous function of the form ax + by = c
> or something similar.

Seems to me that more than one ellipse will fit through the
same three points.

> Everyone, absolutely EVERYONE is simply ignoring the basic tenets
> of sampling when making all these handwaving objections about why
> it can't work without having the slightest understanding of how
> it does.

Not everyone.

> > So maybe that's not such a great analogy to discribe
> > reproduction of an audio waveform.
>
> It's a perfetcly good analogy,

Still disagree with that, but I never disputed sampling
theory.

> FAR more accurate than nonsense about
> "connecting the dots" and "stuff between the samples" and the rest
> of the high-end hooey being spouted.

There's plenty of hooey, of different flavors, to go around.

Mike Prager
North Carolina, USA
Anonymous
June 20, 2004 8:47:20 PM

Archived from groups: rec.audio.high-end (More info?)

dpierce@cartchunk.org (Dick Pierce) wrote in
news:vaDAc.133992$Ly.97088@attbi_s01:

> Mike Prager <hifi@ec.rr.com> wrote in message
> news:<z9tAc.66738$HG.63024@attbi_s53>...
>> Christopher Key wrote:
>>
>> > "maxdm" <maxdimario@aliceposta.it> wrote in message
>> > news:caqhkt02ugh@news2.newsguy.com...
>> > > Filters just 'connect the dots'
>> >
>> > This is sufficient for perfect reproduction.
>> >
>> > An excellent analogy, used in a recently posted link is to imagine a
>> > circle. Only three points on the circumference are required to fully
>> > define it, and any more are an unnecessary waste of bandwidth.
>>
>> True enough, but you also must know it's a circle.
>
> In exactly the same way that in a properly implemented sampled
> system, you KNOW, a priori, that the bandwidth is limited to
> less than 1/2 the sample rate.
>
>> The
>> information content of that knowledge is nontrivial, and
>> without it, an infinity of shapes could be defined by three
>> points.
>
> You don't need to know it's a cricle, you need to know simply
> that it is a closed continuous function of the form ax + by = c
> or something similar.
>
> Everyone, absolutely EVERYONE is simply ignoring the basic tenets
> of sampling when making all these handwaving objections about why
> it can't work without having the slightest understanding of how
> it does.
>
>> So maybe that's not such a great analogy to discribe
>> reproduction of an audio waveform.
>
> It's a perfetcly good analogy, FAR more accurate than nonsense about
> "connecting the dots" and "stuff between the samples" and the rest
> of the high-end hooey being spouted.
>

Dick,

Putting aside such malarky like "stuff between the samples" etc., I have
heard a definite difference between CD players. I call it digital grunge.
Is the better sound experienced with well engineered, pricier products
due to better D-As, better DSP processing or ????

What happened between 1990 and 2002 as far as CDPs are concerned? Things
are better AFAIC.

thanks

r

--
Nothing beats the bandwidth of a station wagon filled with DLT tapes.
Anonymous
June 21, 2004 7:07:26 AM

Archived from groups: rec.audio.high-end (More info?)

On Sun, 20 Jun 2004 16:47:20 GMT, "Rich.Andrews"

..... some stuff deleted.......
>Dick,
>
>Putting aside such malarky like "stuff between the samples" etc., I have
>heard a definite difference between CD players. I call it digital grunge.
> Is the better sound experienced with well engineered, pricier products
>due to better D-As, better DSP processing or ????
>
>What happened between 1990 and 2002 as far as CDPs are concerned? Things
>are better AFAIC.
>
My experience is quite limited, but I was able to compare two CD
players for about a month. One was an old NAD player, from about 1985,
and the other is a Linn Numerik/Karik combo, with the new power
supplies etc. One sold for about $300, the other about $4000. I set
them both up so that playback levels were within 0.1 db, put the same
CD's into each (starting at the same time), and listened for hours,
switched back and forth etc. I even had a random switcher so I
couldn't tellwhich was playing, and had to figure it out myself, and
then compare with the (hidden) indicator. My guesses were essentially
random, I could not determine which was playing (50-100 tests,
whenever I wanted, whatever music, for whatever time interval).
I then used my equipment to see what the differences were,
especially for things like jitter. The differences were beyond the
ability of my HP 3581A wave analyzer (90 db dynamic range), and only
by doing some special circuitry that would reach down to -100 to -110
db was I able to see much difference.
The extra "stuff" that was different was very close to the tested
frequencies, so audio masking would have made it impossible for me to
discern.
Before I did any serious testing I DID hear differences. Afterwards
I DIDN'T! My expectations changed my perceptions of sound.
I have noticed that the audio quality (subjective) and overall
effect of listening to a CD changes dramatically from one listening
session to another. This is probably heresy for this newsgroup, but it
is my opinion that my physical, psychological and spiritual state have
many orders of magnitude (that's POWERS of 10) more effect on sound
than what you could possible hear between REASONABLY designed CD
players. On the other hand, things like phono cartridges, and speakers
have a very big difference. Although you might think I have tin ears,
I am very picky about those electromechanical thingies like
cartridges.
Because of the influence of my mental state, I tend to trust the
equipment more than my ears, especially when I know what kind of
measured "junk" causes my ears grief.
So my question to you is how do you know (RELIABLY) that your
preference for one type of equipment over another is not the result of
mental state, expectations, hype, magic (any high technology is
indistinguishable from magic), coffee, stress, the cool look of the
equipment, etc. Could you get more than random guesses if you set up
your comparison as above? It's damned hard to make a good testing
setup! It has to be very well executed, or you'll believe the test
setup has confounded the test itself.

-Paul

................................................................
Paul Guy
Somewhere in the Nova Scotia fog
Anonymous
June 21, 2004 6:46:52 PM

Archived from groups: rec.audio.high-end (More info?)

Paul Guy <paulguy@eastlink.ca> wrote in news:cb5jde0ucj@news4.newsguy.com:

> On Sun, 20 Jun 2004 16:47:20 GMT, "Rich.Andrews"
>
> ..... some stuff deleted.......
>>Dick,
>>
>>Putting aside such malarky like "stuff between the samples" etc., I have
>>heard a definite difference between CD players. I call it digital
>>grunge.
>> Is the better sound experienced with well engineered, pricier products
>>due to better D-As, better DSP processing or ????
>>
>>What happened between 1990 and 2002 as far as CDPs are concerned?
>>Things are better AFAIC.
>>
> My experience is quite limited, but I was able to compare two CD
> players for about a month. One was an old NAD player, from about 1985,
> and the other is a Linn Numerik/Karik combo, with the new power
> supplies etc. One sold for about $300, the other about $4000. I set
> them both up so that playback levels were within 0.1 db, put the same
> CD's into each (starting at the same time), and listened for hours,
> switched back and forth etc. I even had a random switcher so I
> couldn't tellwhich was playing, and had to figure it out myself, and
> then compare with the (hidden) indicator. My guesses were essentially
> random, I could not determine which was playing (50-100 tests,
> whenever I wanted, whatever music, for whatever time interval).
> I then used my equipment to see what the differences were,
> especially for things like jitter. The differences were beyond the
> ability of my HP 3581A wave analyzer (90 db dynamic range), and only
> by doing some special circuitry that would reach down to -100 to -110
> db was I able to see much difference.
> The extra "stuff" that was different was very close to the tested
> frequencies, so audio masking would have made it impossible for me to
> discern.
> Before I did any serious testing I DID hear differences. Afterwards
> I DIDN'T! My expectations changed my perceptions of sound.
> I have noticed that the audio quality (subjective) and overall
> effect of listening to a CD changes dramatically from one listening
> session to another. This is probably heresy for this newsgroup, but it
> is my opinion that my physical, psychological and spiritual state have
> many orders of magnitude (that's POWERS of 10) more effect on sound
> than what you could possible hear between REASONABLY designed CD
> players. On the other hand, things like phono cartridges, and speakers
> have a very big difference. Although you might think I have tin ears,
> I am very picky about those electromechanical thingies like
> cartridges.
> Because of the influence of my mental state, I tend to trust the
> equipment more than my ears, especially when I know what kind of
> measured "junk" causes my ears grief.
> So my question to you is how do you know (RELIABLY) that your
> preference for one type of equipment over another is not the result of
> mental state, expectations, hype, magic (any high technology is
> indistinguishable from magic), coffee, stress, the cool look of the
> equipment, etc. Could you get more than random guesses if you set up
> your comparison as above? It's damned hard to make a good testing
> setup! It has to be very well executed, or you'll believe the test
> setup has confounded the test itself.
>
> -Paul
>
> ...............................................................
> Paul Guy
> Somewhere in the Nova Scotia fog
>

Paul,

I own two Denon DCD1520 players and a McIntosh MCD7008. When I bought the
MCD7008, I wasn't expecting anything. I was interested in the changer.
The audible differences between the Denon and the McIntosh are not subtle.
A friend of mine compared a cheap 1 year old Sony to his MCD7007 player
and heard significant differences. He then purchased a new McIntosh CDP
to replace the Sony.

The audible differences are probably most pronounced with good orchestral
program material generally found on labels like Telarc, EMI, and DG.

r

--
Nothing beats the bandwidth of a station wagon filled with DLT tapes.
Anonymous
June 23, 2004 2:52:13 AM

Archived from groups: rec.audio.high-end (More info?)

Well, don't we know it's a sine wave? Anything else has a higher frequency
component.


"Christopher Key" <cjk32@cam.ac.uk> wrote in message
news:X5DAc.50592$Hg2.42246@attbi_s04...
> "Mike Prager" <hifi@ec.rr.com> wrote in message
> news:z9tAc.66738$HG.63024@attbi_s53...
> > Christopher Key wrote:
> >
> > > "maxdm" <maxdimario@aliceposta.it> wrote in message
> > > news:caqhkt02ugh@news2.newsguy.com...
> > > > Filters just 'connect the dots'
> > >
> > > This is sufficient for perfect reproduction.
> > >
> > > An excellent analogy, used in a recently posted link is to imagine a
> circle.
> > > Only three points on the circumference are required to fully define
it,
> and
> > > any more are an unnecessary waste of bandwidth.
> >
> > True enough, but you also must know it's a circle. The
> > information content of that knowledge is nontrivial, and
> > without it, an infinity of shapes could be defined by three
> > points.
>
> Good point, although the fact that the signal is band limited means that
you
> do know a certain amount about the signal, ie that it can be expressed as
> the sum of a set of sinusoids up to a given frequency.
>
> > So maybe that's not such a great analogy to discribe
> > reproduction of an audio waveform.
> >
>
> I do accept that it isn't a precise analogy, but thought nonetheless that
it
> was pretty good for showing people of a non mathematical background
> (specifically the OP) roughly what was going on.
>
> Chris Key
>
Anonymous
June 23, 2004 2:53:37 AM

Archived from groups: rec.audio.high-end (More info?)

On 6/20/04 11:07 PM, in article cb5jde0ucj@news4.newsguy.com, "Paul Guy"
<paulguy@eastlink.ca> wrote:

> So my question to you is how do you know (RELIABLY) that your
> preference for one type of equipment over another is not the result of
> mental state, expectations, hype, magic (any high technology is
> indistinguishable from magic), coffee, stress, the cool look of the
> equipment, etc. Could you get more than random guesses if you set up
> your comparison as above? It's damned hard to make a good testing
> setup! It has to be very well executed, or you'll believe the test
> setup has confounded the test itself.

This is why the only valid testing is long term testing. If I have had a
stressful day, it takes awhile before music - no matter what playback method
or if it is a live concert - will move me. Though if in the right mood, it
might move me to become teary.

My "subjective" method is that there are some recordings that will move me
emotionally if I play them - and if the playback stack doesn't mangle them
too badly. The ones that help with this, stay.

I lost faith in simple ABX testing because it didn't seem to match how I
would actually USE the equipment.
Anonymous
June 23, 2004 9:28:46 AM

Archived from groups: rec.audio.high-end (More info?)

In article <KGXBc.160040$Ly.55785@attbi_s01>,
"Midlant" <sherman1125@cox.net> wrote:

> "Dick Pierce" <dpierce@cartchunk.org> >
> > > > But, this is old stuff: I don't know of a single 44.1 kHz
> product
> > > > on the market that does NOT use oversampling.
> > >
>
> Richard, I am not arguing with you, simply looking for a better
> understanding, like so many others.
> Can you please explain this/these then:
> John
> http://www.sakurasystems.com/articles/Kusunoki.html
> http://www.audionote.co.jp/digital/essay.htm
>

You can add Audio Research to the list.

http://www.audioresearch.com/CD3.htm

"While using the latest 24/192-capable Crystal DAC, the CD3 does not
upsample, because our empirical research shows sonic compromise is
unavoidable due to sample rate manipulation and approximating errors."
Anonymous
June 23, 2004 10:29:01 AM

Archived from groups: rec.audio.high-end (More info?)

On 6/22/04 10:50 AM, in article KGXBc.160040$Ly.55785@attbi_s01, "Midlant"
<sherman1125@cox.net> wrote:

> "Dick Pierce" <dpierce@cartchunk.org> >
>>>> But, this is old stuff: I don't know of a single 44.1 kHz
> product
>>>> on the market that does NOT use oversampling.
>>>
>
> Richard, I am not arguing with you, simply looking for a better
> understanding, like so many others.
> Can you please explain this/these then:
> John
> http://www.sakurasystems.com/articles/Kusunoki.html
> http://www.audionote.co.jp/digital/essay.htm
>

Has anyone used this stuff? It seems to spring out of the Japanese
"minimalist" esthetic - apparently some really like it - and since one is
not supposed to be able to hear above 22.05kHz anyway - why even filter it,
eh? :-)
Anonymous
June 23, 2004 10:30:07 AM

Archived from groups: rec.audio.high-end (More info?)

On Thu, 17 Jun 2004 14:05:17 GMT, maxdimario@aliceposta.it (maxdm)
wrote:

>I am sure that math apart, most people who have the capacity to listen
>critically to sound will notice a difference in 192Khz audio compared
>to 44.1 even if the speakers or headphones or amplifiers are frequency
>limited to20 Khz.
>
>Anyone out there with listening experience ?

Yup, loads of us - and there is as yet *zero* evidence that 24/192
sounds different from 16/44 - when that is the *only* difference.

The real test is to compare 24/192 to 16/48, since that is a very
simple mathematical transform, and will introduce no peculiar
artifacts. As noted, there is as yet *zero* evidence that these sound
different.
--

Stewart Pinkerton | Music is Art - Audio is Engineering
Anonymous
June 23, 2004 6:45:25 PM

Archived from groups: rec.audio.high-end (More info?)

Bromo <bromo@ix.netcom.com> wrote in message news:<Nq9Cc.91780$eu.10945@attbi_s02>...
> On 6/22/04 10:50 AM, in article KGXBc.160040$Ly.55785@attbi_s01, "Midlant"
> <sherman1125@cox.net> wrote:
>
> > "Dick Pierce" <dpierce@cartchunk.org> >
> >>>> But, this is old stuff: I don't know of a single 44.1 kHz
> product
> >>>> on the market that does NOT use oversampling.
> >>>

I should have satated: there are no competent products...

> > Richard, I am not arguing with you, simply looking for a better
> > understanding, like so many others.
> > Can you please explain this/these then:
> > John
> > http://www.sakurasystems.com/articles/Kusunoki.html
> > http://www.audionote.co.jp/digital/essay.htm
> >
>
> Has anyone used this stuff? It seems to spring out of the Japanese
> "minimalist" esthetic - apparently some really like it - and since one is
> not supposed to be able to hear above 22.05kHz anyway - why even filter it,
> eh? :-)

Because it's a real bad idea?

One of the overarching principles of sampled data is that the data
stream contains the baseband audio and EVERY image of that baseband
out to infinity, and the entire point to anti-imaging filtering is
to remove all the images of the original, and leave only the original.

The problem with NOT filtering them out is that every image contains
the same total energy as the baseband original. That means that if you
can imagine the spectrum of the music contained in the 0-20kHz band,
there is a mirror image of that in the 24-44 kHz band, another non-
mirror image in the 44-64 kHz band, a second mirror image in the 68-88
kHz band, a second non-mirro4ed image in the 88-108 kHz band, and so on
all the way up to infinity (ignoring practical bandwidth limitations).
Every one of those 20 kHz bands has exactly the same total energy as
the original 20 kHz band.

(now, before someone comes running in waving there hands claiming
I said these things have infinite power output because the images
go out to infinity, look at what I said: their bandwidth is
ultimately limited by practical bandwidth limitations.
Mathematically, the images DO go out to infinity, and if the
equipment had infinite bandwidth but finite power output, then
each image, and the original, would have infinitesimal power)

Now, imagine all your downstream equipment being fed all this energy,
much of which is up in the RF reqion. That excess energy is wasting
power, interfering with the proper operation of the electronics,
combining with other sources of energy and getting modulated down
wherte it can be heard, heating tweeter voice coils and a whole
lot more.

And, because you DO have all the images present, which you WILL get
if you omit the antiimaging reconstruction filter, only then will
you see the "classic" stair-step output that audiophilic digiphobes
around the globe have been pointing as the source of all digital
evil.

So, in effect, what this equipment says is "if you don't like your
preconceptions of digital, try this, which is precisely your worst
nightmare."

Only in high end audio to you see this sort of insanity.
Anonymous
June 23, 2004 6:46:57 PM

Archived from groups: rec.audio.high-end (More info?)

In article <iy8Cc.78427$Hg2.37239@attbi_s04>,
MINe 109 <smcatut@mail.utexas.edu> wrote:

>> "Dick Pierce" <dpierce@cartchunk.org> >
>> > > > But, this is old stuff: I don't know of a single 44.1 kHz
>> product
>> > > > on the market that does NOT use oversampling.
>> > >
>>
>> Richard, I am not arguing with you, simply looking for a better
>> understanding, like so many others.
>> Can you please explain this/these then:
>> John
>> http://www.sakurasystems.com/articles/Kusunoki.html
>> http://www.audionote.co.jp/digital/essay.htm
>>
>
>You can add Audio Research to the list.
>
>http://www.audioresearch.com/CD3.htm
>
>"While using the latest 24/192-capable Crystal DAC, the CD3 does not
>upsample, because our empirical research shows sonic compromise is
>unavoidable due to sample rate manipulation and approximating errors."

The Crystal Semiconductor DACs are all (as best as I can tell) of the
delta-sigma design. This type of DAC architecture generates output
(analog) samples at a much higher rate than the incoming digital
samples.

Audio Research's statement indicates that they do not implement their
own oversampling/upsampling logic _before_ the signal is fed into the
actual DAC chip. They don't need to, as the DAC chip used in the CD3
does the oversampling/upsampling internally, as part of the
delta-sigma modulation process.

--
Dave Platt <dplatt@radagast.org> AE6EO
Hosting the Jade Warrior home page: http://www.radagast.org/jade-warrior
I do _not_ wish to receive unsolicited commercial email, and I will
boycott any company which has the gall to send me such ads!
Anonymous
June 24, 2004 3:09:46 AM

Archived from groups: rec.audio.high-end (More info?)

Bromo bromo@ix.netcom.com wrote:



>On 6/20/04 11:07 PM, in article cb5jde0ucj@news4.newsguy.com, "Paul Guy"
><paulguy@eastlink.ca> wrote:
>
>> So my question to you is how do you know (RELIABLY) that your
>> preference for one type of equipment over another is not the result of
>> mental state, expectations, hype, magic (any high technology is
>> indistinguishable from magic), coffee, stress, the cool look of the
>> equipment, etc. Could you get more than random guesses if you set up
>> your comparison as above? It's damned hard to make a good testing
>> setup! It has to be very well executed, or you'll believe the test
>> setup has confounded the test itself.
>
>This is why the only valid testing is long term testing. If I have had a
>stressful day, it takes awhile before music - no matter what playback method
>or if it is a live concert - will move me. Though if in the right mood, it
>might move me to become teary.
>
>My "subjective" method is that there are some recordings that will move me
>emotionally if I play them - and if the playback stack doesn't mangle them
>too badly. The ones that help with this, stay.
>
>I lost faith in simple ABX testing because it didn't seem to match how I
>would actually USE the equipment.

You mean you don't plan on listening to it? That's the beauty of any type of
bias controlled listening (ABX or otherwise); it requires that you "listen"
with only your ears leaving out all the personal, social and written middlemen.
June 24, 2004 6:57:36 AM

Archived from groups: rec.audio.high-end (More info?)

Bromo wrote:

> On 6/20/04 11:07 PM, in article cb5jde0ucj@news4.newsguy.com, "Paul Guy"
> <paulguy@eastlink.ca> wrote:
>
>> So my question to you is how do you know (RELIABLY) that your
>> preference for one type of equipment over another is not the result of
>> mental state, expectations, hype, magic (any high technology is
>> indistinguishable from magic), coffee, stress, the cool look of the
>> equipment, etc. Could you get more than random guesses if you set up
>> your comparison as above? It's damned hard to make a good testing
>> setup! It has to be very well executed, or you'll believe the test
>> setup has confounded the test itself.
>
> This is why the only valid testing is long term testing. If I have had a
> stressful day, it takes awhile before music - no matter what playback method
> or if it is a live concert - will move me. Though if in the right mood, it
> might move me to become teary.
>
> My "subjective" method is that there are some recordings that will move me
> emotionally if I play them - and if the playback stack doesn't mangle them
> too badly. The ones that help with this, stay.
>
> I lost faith in simple ABX testing because it didn't seem to match how I
> would actually USE the equipment.

So, as an RF engineer, have you lost faith in all measuring equipment
because that was not how you would use the devices under test? For
example, you would lose faith in a distortion analyzer because no one
listens to sine waves? Or you would lose faith in sqaure wave responses
because these devices do not recieve square waves in real life?

I find this statement by an alleged engineer utterly astonishing.
Anonymous
June 26, 2004 6:10:57 AM

Archived from groups: rec.audio.high-end (More info?)

On 6/23/04 10:57 PM, in article AqrCc.97729$Sw.3328@attbi_s51, "chung"
<chunglau@covad.net> wrote:

>> I lost faith in simple ABX testing because it didn't seem to match how I
>> would actually USE the equipment.
>
> So, as an RF engineer, have you lost faith in all measuring equipment
> because that was not how you would use the devices under test? For
> example, you would lose faith in a distortion analyzer because no one
> listens to sine waves? Or you would lose faith in sqaure wave responses
> because these devices do not recieve square waves in real life?
>
> I find this statement by an alleged engineer utterly astonishing.

No, sir, but we as a group are forgetting that simulation (whether that be
on a computer or using sine waves to test a piece of equipment that is
designed to reproduce music) is NOT how the piece of equipment is *used.*

I have not lost faith in measuring equipment, but I do know that putting
full faith in tests that are not fully representative of actual practice, is
a mistake. It may indicate something - but it may not be accurate.

For instance in the 1920's there were ABX demonstrations where an orchestra
played music, stopped and a 78 was played - and the audience didn't notice
in the rapid ABX fashion.

Does this mean that 78's sounded the same as live music? Of course not, but
it may reveal a weakness in a test.

IN this manner, I will take the data from a test, but think about what it
does mean, and what it does not.

To do less would be to be sloppy.
Anonymous
June 26, 2004 6:11:06 AM

Archived from groups: rec.audio.high-end (More info?)

On 6/23/04 7:09 PM, in article cbd2jq01k2n@news1.newsguy.com, "Nousaine"
<nousaine@aol.com> wrote:

>> I lost faith in simple ABX testing because it didn't seem to match how I
>> would actually USE the equipment.
>
> You mean you don't plan on listening to it? That's the beauty of any type of
> bias controlled listening (ABX or otherwise); it requires that you "listen"
> with only your ears leaving out all the personal, social and written
> middlemen.

I will be listening to the player for HOURS not for a couple of seconds
(though I suspect you understood that, but wanted to interpret it in the way
you did).

I found a rather expensive CD player made by a high end company (Mark
Levinson) sounded much worse than my humble NAD C541i player, but in a rapid
blind ABX I preferred the Mark Levinson. So, no, if I had bought due to an
ABX test I would have set fire to a whole pile of cash.
Anonymous
June 26, 2004 8:08:38 AM

Archived from groups: rec.audio.high-end (More info?)

Bromo <bromo@ix.netcom.com> wrote:

> I found a rather expensive CD player made by a high end company (Mark
> Levinson) sounded much worse than my humble NAD C541i player, but in a rapid
> blind ABX I preferred the Mark Levinson. So, no, if I had bought due to an
> ABX test I would have set fire to a whole pile of cash.

Did you ever consider that if you use the wrong test for a particular goal the
results won't make any sense?
Anonymous
June 26, 2004 6:30:10 PM

Archived from groups: rec.audio.high-end (More info?)

On 6/23/04 2:30 AM, in article Pr9Cc.162900$Ly.31887@attbi_s01, "Stewart
Pinkerton" <patent3@dircon.co.uk> wrote:

>
> Yup, loads of us - and there is as yet *zero* evidence that 24/192
> sounds different from 16/44 - when that is the *only* difference.

Keep in mind that the CD medium has only 15 information bits, 1 bit is error
correction.
June 26, 2004 6:33:01 PM

Archived from groups: rec.audio.high-end (More info?)

Bromo wrote:

> On 6/23/04 7:09 PM, in article cbd2jq01k2n@news1.newsguy.com, "Nousaine"
> <nousaine@aol.com> wrote:
>
>>> I lost faith in simple ABX testing because it didn't seem to match how I
>>> would actually USE the equipment.
>>
>> You mean you don't plan on listening to it? That's the beauty of any type of
>> bias controlled listening (ABX or otherwise); it requires that you "listen"
>> with only your ears leaving out all the personal, social and written
>> middlemen.
>
> I will be listening to the player for HOURS not for a couple of seconds
> (though I suspect you understood that, but wanted to interpret it in the way
> you did).

I suspect that you do understand the ABX principles. You can take as
long as you like to listen to each device. You do not have to listen for
only a couple of seconds.
>
> I found a rather expensive CD player made by a high end company (Mark
> Levinson) sounded much worse than my humble NAD C541i player, but in a rapid
> blind ABX I preferred the Mark Levinson.

What in the world is a rapid blind ABX? The ABX test is used primarily
to detect differences, not preferences. Can you provide more detail
about how the test was done?

> So, no, if I had bought due to an
> ABX test I would have set fire to a whole pile of cash.
>

How would you know that you wouldn't have liked the Mark Levinson more?

Given that you totally misunderstood ABX or DBT, perhaps you should
reconsider your position.
Anonymous
June 26, 2004 8:24:18 PM

Archived from groups: rec.audio.high-end (More info?)

Bromo <bromo@ix.netcom.com> writes:

> On 6/23/04 2:30 AM, in article Pr9Cc.162900$Ly.31887@attbi_s01, "Stewart
> Pinkerton" <patent3@dircon.co.uk> wrote:
>
> >
> > Yup, loads of us - and there is as yet *zero* evidence that 24/192
> > sounds different from 16/44 - when that is the *only* difference.
>
> Keep in mind that the CD medium has only 15 information bits, 1 bit is error
> correction.

No, it's 16 bits of information.

---Ketil
Anonymous
June 26, 2004 8:46:13 PM

Archived from groups: rec.audio.high-end (More info?)

On 6/26/04 12:08 AM, in article aF6Dc.95776$2i5.25561@attbi_s52,
"jjnunes@sonic.net" <jjnunes@sonic.net> wrote:

> Bromo <bromo@ix.netcom.com> wrote:
>
>> I found a rather expensive CD player made by a high end company (Mark
>> Levinson) sounded much worse than my humble NAD C541i player, but in a rapid
>> blind ABX I preferred the Mark Levinson. So, no, if I had bought due to an
>> ABX test I would have set fire to a whole pile of cash.
>
> Did you ever consider that if you use the wrong test for a particular goal the
> results won't make any sense?
>

That is an excellent point! Perhaps for choosing equipment for long term
use, a quick, blind ABX test may be a good start, but not the final arbiter.

I would suspect that the ABX test, given its history, may be only good for
proving some narrow points, and *may* not be so good for the uses to which
it is put.
June 26, 2004 8:46:31 PM

Archived from groups: rec.audio.high-end (More info?)

Bromo wrote:
> On 6/23/04 10:57 PM, in article AqrCc.97729$Sw.3328@attbi_s51, "chung"
> <chunglau@covad.net> wrote:
>
>>> I lost faith in simple ABX testing because it didn't seem to match how I
>>> would actually USE the equipment.
>>
>> So, as an RF engineer, have you lost faith in all measuring equipment
>> because that was not how you would use the devices under test? For
>> example, you would lose faith in a distortion analyzer because no one
>> listens to sine waves? Or you would lose faith in sqaure wave responses
>> because these devices do not recieve square waves in real life?
>>
>> I find this statement by an alleged engineer utterly astonishing.
>
> No, sir, but we as a group are forgetting that simulation (whether that be
> on a computer or using sine waves to test a piece of equipment that is
> designed to reproduce music) is NOT how the piece of equipment is *used.*
>
> I have not lost faith in measuring equipment, but I do know that putting
> full faith in tests that are not fully representative of actual practice, is
> a mistake. It may indicate something - but it may not be accurate.

Then why did you lose faith on ABX? Doesn't ABX at least indicate
something? Have you taken an ABX test yourself, and what were the
results, if so?

>
> For instance in the 1920's there were ABX demonstrations where an orchestra
> played music, stopped and a 78 was played - and the audience didn't notice
> in the rapid ABX fashion.

If you based your distrust of ABX on that incident, then you are clearly
mistaken. The earliest mention I can find of ABX is in the mid-70's. So
I don't know what you were talking about.

>
> Does this mean that 78's sounded the same as live music? Of course not, but
> it may reveal a weakness in a test.

Obviously that was not an ABX test. Or even a DBT test. So you are
figuratively barking up a wrong tree.

>
> IN this manner, I will take the data from a test, but think about what it
> does mean, and what it does not.
>
> To do less would be to be sloppy.
>

So how come no one has proven that DBT or ABX's are not effective?
Anonymous
June 27, 2004 1:52:22 AM

Archived from groups: rec.audio.high-end (More info?)

Bromo bromo@ix.netcom.com wrote:

>On 6/23/04 7:09 PM, in article cbd2jq01k2n@news1.newsguy.com, "Nousaine"
><nousaine@aol.com> wrote:
>
>>> I lost faith in simple ABX testing because it didn't seem to match how I
>>> would actually USE the equipment.
>>
>> You mean you don't plan on listening to it? That's the beauty of any type
>of
>> bias controlled listening (ABX or otherwise); it requires that you "listen"
>> with only your ears leaving out all the personal, social and written
>> middlemen.
>
>I will be listening to the player for HOURS not for a couple of seconds
>(though I suspect you understood that, but wanted to interpret it in the way
>you did).

There is no limit on the time of programs used for comparison with the ABX
protocol. But it has been shown that selected short segments are more sensitive
to difference, which is why most people like to use them.

You seem to be willing to ignore that ABX and other bias controlled protocols
require that the listener use ONLY his hearing mechanism for decision making.

>
>I found a rather expensive CD player made by a high end company (Mark
>Levinson) sounded much worse than my humble NAD C541i player, but in a rapid
>blind ABX I preferred the Mark Levinson. So, no, if I had bought due to an
>ABX test I would have set fire to a whole pile of cash.

From your description of events it seems likely that you have never done an
"ABX" comparison. This protocol has a specific meaning and it appears you are
not familiar with the tool.
Anonymous
June 27, 2004 9:50:06 AM

Archived from groups: rec.audio.high-end (More info?)

On 26 Jun 2004 14:30:10 GMT, Bromo <bromo@ix.netcom.com> wrote:

>On 6/23/04 2:30 AM, in article Pr9Cc.162900$Ly.31887@attbi_s01, "Stewart
>Pinkerton" <patent3@dircon.co.uk> wrote:
>
>>
>> Yup, loads of us - and there is as yet *zero* evidence that 24/192
>> sounds different from 16/44 - when that is the *only* difference.
>
>Keep in mind that the CD medium has only 15 information bits, 1 bit is error
>correction.

Wrong - and also irrelevant.
--

Stewart Pinkerton | Music is Art - Audio is Engineering
Anonymous
June 27, 2004 5:53:21 PM

Archived from groups: rec.audio.high-end (More info?)

Bromo bromo@ix.netcom.com wrote:

>On 6/23/04 10:57 PM, in article AqrCc.97729$Sw.3328@attbi_s51, "chung"
><chunglau@covad.net> wrote:
>
>>> I lost faith in simple ABX testing because it didn't seem to match how I
>>> would actually USE the equipment.
>>
>> So, as an RF engineer, have you lost faith in all measuring equipment
>> because that was not how you would use the devices under test? For
>> example, you would lose faith in a distortion analyzer because no one
>> listens to sine waves? Or you would lose faith in sqaure wave responses
>> because these devices do not recieve square waves in real life?
>>
>> I find this statement by an alleged engineer utterly astonishing.
>
>No, sir, but we as a group are forgetting that simulation (whether that be
>on a computer or using sine waves to test a piece of equipment that is
>designed to reproduce music) is NOT how the piece of equipment is *used.*

>I have not lost faith in measuring equipment, but I do know that putting
>full faith in tests that are not fully representative of actual practice, is
>a mistake. It may indicate something - but it may not be accurate.
>
>For instance in the 1920's there were ABX demonstrations where an orchestra
>played music, stopped and a 78 was played - and the audience didn't notice
>in the rapid ABX fashion.

There have been reported 'live' vs 'recording' comparisons but the ABX protocol
wasn't present in the 20s. But what is the difference between any comparison of
sound between such a test other than the elapsed time between successive
presentations?

>
>Does this mean that 78's sounded the same as live music? Of course not, but
>it may reveal a weakness in a test.

In what test? A live vs recorded comparison? How could ANY other comparative
technique be MORE sensitive? Sure there may have been some other confounding
factor in any given experiment but a side-side comparison allows maximal
sensitivity.

>IN this manner, I will take the data from a test, but think about what it
>does mean, and what it does not.
>
>To do less would be to be sloppy.

Sure but you must also interpret results with the knowledge that even
successive identical presentations will be interpreted as different with
technqies that do not control bias.
Anonymous
June 27, 2004 5:53:52 PM

Archived from groups: rec.audio.high-end (More info?)

Bromo <bromo@ix.netcom.com> wrote:
> On 6/26/04 12:08 AM, in article aF6Dc.95776$2i5.25561@attbi_s52,
> "jjnunes@sonic.net" <jjnunes@sonic.net> wrote:

>> Bromo <bromo@ix.netcom.com> wrote:
>>
>>> I found a rather expensive CD player made by a high end company (Mark
>>> Levinson) sounded much worse than my humble NAD C541i player, but in a rapid
>>> blind ABX I preferred the Mark Levinson. So, no, if I had bought due to an
>>> ABX test I would have set fire to a whole pile of cash.
>>
>> Did you ever consider that if you use the wrong test for a particular goal the
>> results won't make any sense?
>>

> That is an excellent point! Perhaps for choosing equipment for long term
> use, a quick, blind ABX test may be a good start, but not the final arbiter.

> I would suspect that the ABX test, given its history, may be only good for
> proving some narrow points, and *may* not be so good for the uses to which
> it is put.

ABX is used for determining if a subtle acoustical difference exists or not.
It is practically useless for preference.

If you want to do a blind test for preference, it's ABC/HR. (hidden
reference)

The most precise testing for sound quality <alone> involves both of the above.


If you just want to find out what you like or prefer in general, do anything
that you desire. It is sensible to to add that one should be aware that
judgements of sound quality are unreliable when doing this.
Anonymous
June 27, 2004 5:56:28 PM

Archived from groups: rec.audio.high-end (More info?)

On 6/26/04 12:46 PM, in article HLhDc.97731$2i5.41119@attbi_s52, "chung"
<chunglau@covad.net> wrote:

>> IN this manner, I will take the data from a test, but think about what it
>> does mean, and what it does not.
>>
>> To do less would be to be sloppy.
>>
>
> So how come no one has proven that DBT or ABX's are not effective?

I have no idea nor do you.

Main thing is to think about what a test will prove and what it wont prove.
Keeping in mind of the range that the test covers. You also have to note
the implications and what you are concluding and extrapolating from the
data.

Please give me links or a resource in which I can further educate myself
about the tests in question.
Anonymous
June 27, 2004 5:58:11 PM

Archived from groups: rec.audio.high-end (More info?)

On 6/26/04 12:24 PM, in article SqhDc.120757$Sw.39384@attbi_s51, "Ketil
Kirkerud Elgethun" <ketil+news@elgethun.no> wrote:

> Bromo <bromo@ix.netcom.com> writes:
>
>> On 6/23/04 2:30 AM, in article Pr9Cc.162900$Ly.31887@attbi_s01, "Stewart
>> Pinkerton" <patent3@dircon.co.uk> wrote:
>>
>>>
>>> Yup, loads of us - and there is as yet *zero* evidence that 24/192
>>> sounds different from 16/44 - when that is the *only* difference.
>>
>> Keep in mind that the CD medium has only 15 information bits, 1 bit is error
>> correction.
>
> No, it's 16 bits of information.

15 data bits, 1 error correction bit?
Anonymous
June 27, 2004 6:08:00 PM

Archived from groups: rec.audio.high-end (More info?)

On 6/27/04 1:50 AM, in article ietDc.160075$3x.68214@attbi_s54, "Stewart
Pinkerton" <patent3@dircon.co.uk> wrote:

> On 26 Jun 2004 14:30:10 GMT, Bromo <bromo@ix.netcom.com> wrote:
>
>> On 6/23/04 2:30 AM, in article Pr9Cc.162900$Ly.31887@attbi_s01, "Stewart
>> Pinkerton" <patent3@dircon.co.uk> wrote:
>>
>>>
>>> Yup, loads of us - and there is as yet *zero* evidence that 24/192
>>> sounds different from 16/44 - when that is the *only* difference.
>>
>> Keep in mind that the CD medium has only 15 information bits, 1 bit is error
>> correction.
>
> Wrong - and also irrelevant.

Would be very relevant -

16 bits would mean that the available signal peak is 96dB above the
quantization noise floor, and 15 bits means it would be 90dB above it.

But, yes. Looked up the IEC908 standard this AM - and the error correction
is in addition to the 16 bit PCM.

I found it rather interesting how everyone relished pointing out my error,
but no one seemed interested in setting the record straight.
Anonymous
June 27, 2004 8:39:22 PM

Archived from groups: rec.audio.high-end (More info?)

On 6/27/04 9:53 AM, in article cbmjgh0ck0@news1.newsguy.com, "Nousaine"
<nousaine@aol.com> wrote:

>> Does this mean that 78's sounded the same as live music? Of course not, but
>> it may reveal a weakness in a test.
>
> In what test? A live vs recorded comparison? How could ANY other comparative
> technique be MORE sensitive? Sure there may have been some other confounding
> factor in any given experiment but a side-side comparison allows maximal
> sensitivity.

Well, if that would be the case as you say, from the simple tests in the
1920's, by your own logic the whole endeavor of sound reproduction has been
a colossal waste of resources and time since it had already reached
perfection with the 78 record c. 1927.
June 27, 2004 8:42:44 PM

Archived from groups: rec.audio.high-end (More info?)

Bromo wrote:
> On 6/27/04 1:50 AM, in article ietDc.160075$3x.68214@attbi_s54, "Stewart
> Pinkerton" <patent3@dircon.co.uk> wrote:
>
>> On 26 Jun 2004 14:30:10 GMT, Bromo <bromo@ix.netcom.com> wrote:
>>
>>> On 6/23/04 2:30 AM, in article Pr9Cc.162900$Ly.31887@attbi_s01, "Stewart
>>> Pinkerton" <patent3@dircon.co.uk> wrote:
>>>
>>>>
>>>> Yup, loads of us - and there is as yet *zero* evidence that 24/192
>>>> sounds different from 16/44 - when that is the *only* difference.
>>>
>>> Keep in mind that the CD medium has only 15 information bits, 1 bit is error
>>> correction.
>>
>> Wrong - and also irrelevant.
>
> Would be very relevant -
>
> 16 bits would mean that the available signal peak is 96dB above the
> quantization noise floor, and 15 bits means it would be 90dB above it.
>
> But, yes. Looked up the IEC908 standard this AM - and the error correction
> is in addition to the 16 bit PCM.
>
> I found it rather interesting how everyone relished pointing out my error,
> but no one seemed interested in setting the record straight.

The very first response to you from Ketil Kirkerud Elgethen set the
record straight for you.

I would think that CD has 16 information bits is very common knowledge
for anyone interested in audio, especially engineers. How did you ever
come up with 15? Have you ever read a CD player spec sheet? Or a DAC
spec sheet?

And how can anyone do error *correction* with just 1 bit? (No, you would
have to supply the answer on this one by yourself.) There is huge money
and fame if you can do this.
Anonymous
June 27, 2004 9:08:08 PM

Archived from groups: rec.audio.high-end (More info?)

On 6/27/04 9:53 AM, in article cbmjgh0ck0@news1.newsguy.com, "Nousaine"
<nousaine@aol.com> wrote:

>> To do less would be to be sloppy.
>
> Sure but you must also interpret results with the knowledge that even
> successive identical presentations will be interpreted as different with
> technqies that do not control bias.

Granted, but you also have to analyze tests very carefully to be absolutely
sure what you are proving and what you are not.

Additionally, you have to make you're you are removing the right biases.
For instance in wine tasting, there are reds that taste awful without food.
Are they bad wines? No, but in a "blind" test they will not do as well as
those that don't need food. Removing that bias can remove it from the
"range" of use. I do agree that the only merit for judging hifi equipment
is sound - but you have to design and perform tests properly and clearly
spell out what you will be proving and what you will not be proving.
June 27, 2004 9:12:27 PM

Archived from groups: rec.audio.high-end (More info?)

Bromo wrote:

> On 6/26/04 12:46 PM, in article HLhDc.97731$2i5.41119@attbi_s52, "chung"
> <chunglau@covad.net> wrote:
>
>>> IN this manner, I will take the data from a test, but think about what it
>>> does mean, and what it does not.
>>>
>>> To do less would be to be sloppy.
>>>
>>
>> So how come no one has proven that DBT or ABX's are not effective?
>
> I have no idea nor do you.
>
> Main thing is to think about what a test will prove and what it wont prove.
> Keeping in mind of the range that the test covers. You also have to note
> the implications and what you are concluding and extrapolating from the
> data.
>
> Please give me links or a resource in which I can further educate myself
> about the tests in question.

http://www.pcavtech.com/abx/abx_peri.htm
Anonymous
June 28, 2004 1:47:57 AM

Archived from groups: rec.audio.high-end (More info?)

Bromo bromo@ix.netcom.com wrote:



>
>On 6/27/04 9:53 AM, in article cbmjgh0ck0@news1.newsguy.com, "Nousaine"
><nousaine@aol.com> wrote:
>
>>> Does this mean that 78's sounded the same as live music? Of course not,
>but
>>> it may reveal a weakness in a test.
>>
>> In what test? A live vs recorded comparison? How could ANY other
>comparative
>> technique be MORE sensitive? Sure there may have been some other
>confounding
>> factor in any given experiment but a side-side comparison allows maximal
>> sensitivity.
>
>Well, if that would be the case as you say, from the simple tests in the
>1920's, by your own logic the whole endeavor of sound reproduction has been
>a colossal waste of resources and time since it had already reached
>perfection with the 78 record c. 1927.

If I recall correctly you said that ABX tests had been conducted in the 20s and
that they had short time intervals. Yet it was clear that they were not ABX
tests are you dropping the time complaint?

Anyway. exactly which tests are you referencing? What were the results, other
than a guess?
Anonymous
June 28, 2004 3:15:56 AM

Archived from groups: rec.audio.high-end (More info?)

Bromo <bromo@ix.netcom.com> writes:

> On 6/26/04 12:24 PM, in article SqhDc.120757$Sw.39384@attbi_s51, "Ketil
> Kirkerud Elgethun" <ketil+news@elgethun.no> wrote:
>
> > Bromo <bromo@ix.netcom.com> writes:
> >
> >> On 6/23/04 2:30 AM, in article Pr9Cc.162900$Ly.31887@attbi_s01, "Stewart
> >> Pinkerton" <patent3@dircon.co.uk> wrote:
> >>
> >>>
> >>> Yup, loads of us - and there is as yet *zero* evidence that 24/192
> >>> sounds different from 16/44 - when that is the *only* difference.
> >>
> >> Keep in mind that the CD medium has only 15 information bits, 1 bit is error
> >> correction.
> >
> > No, it's 16 bits of information.
>
> 15 data bits, 1 error correction bit?

No, 16 bits of information, various bits of error correction
(depending on where in the signal chain you look). The _resolution_
of the signal is 16 bits. Error correction/CRC/... is not counted in
resolution.

---Ketil
Anonymous
June 28, 2004 8:03:47 AM

Archived from groups: rec.audio.high-end (More info?)

On 6/27/04 5:47 PM, in article cbnfad0cli@news4.newsguy.com, "Nousaine"
<nousaine@aol.com> wrote:

>> Well, if that would be the case as you say, from the simple tests in the
>> 1920's, by your own logic the whole endeavor of sound reproduction has been
>> a colossal waste of resources and time since it had already reached
>> perfection with the 78 record c. 1927.
>
> If I recall correctly you said that ABX tests had been conducted in the 20s
> and
> that they had short time intervals. Yet it was clear that they were not ABX
> tests are you dropping the time complaint?

I corrected myself, and meant to say that it was ABX like - since I
described the experiment you were able to draw your own conclusions.

> Anyway. exactly which tests are you referencing? What were the results, other
> than a guess?

If you look at the previous post, I did place the reference I used in it.
Anonymous
June 28, 2004 8:52:10 PM

Archived from groups: rec.audio.high-end (More info?)

On 27 Jun 2004 14:08:00 GMT, Bromo <bromo@ix.netcom.com> wrote:

>On 6/27/04 1:50 AM, in article ietDc.160075$3x.68214@attbi_s54, "Stewart
>Pinkerton" <patent3@dircon.co.uk> wrote:
>
>> On 26 Jun 2004 14:30:10 GMT, Bromo <bromo@ix.netcom.com> wrote:
>>
>>> On 6/23/04 2:30 AM, in article Pr9Cc.162900$Ly.31887@attbi_s01, "Stewart
>>> Pinkerton" <patent3@dircon.co.uk> wrote:
>>>>
>>>> Yup, loads of us - and there is as yet *zero* evidence that 24/192
>>>> sounds different from 16/44 - when that is the *only* difference.
>>>
>>> Keep in mind that the CD medium has only 15 information bits, 1 bit is error
>>> correction.
>>
>> Wrong - and also irrelevant.
>
>Would be very relevant -
>
>16 bits would mean that the available signal peak is 96dB above the
>quantization noise floor, and 15 bits means it would be 90dB above it.

But neither would be relevant to *sound* quality, given that there are
no musical master tapes with a dynamic range grteater than 80dB. BTW,
you're still wrong here, as a properly dithered CD has about 93dB
dynamic range, not 96.

>But, yes. Looked up the IEC908 standard this AM - and the error correction
>is in addition to the 16 bit PCM.
>
>I found it rather interesting how everyone relished pointing out my error,
>but no one seemed interested in setting the record straight.

What? Everyone except you already *knows* that Red Book CD is 16 bits
of information, so what's this 'setting the record straight'?

--

Stewart Pinkerton | Music is Art - Audio is Engineering
!