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  #1   Report Post  
tubesforall
 
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Default How good is CD Technology before is distorts?

Nyquist theorem states that a non-variant signal freqency can be reproduced
that is 1/2 the sample rate. Unfortunately, music that is invariant is not
terribly interesting. Thus, the common wisdom that 44.1KHz sampling can
reproduce 22 KHz music is not true.

A seminal paper from MIT shows that distortion related to sampling must
consider both the sample rate and the target word size. For today's
CDs--that is 16 bits. Thus, according to this paper, a minimum of 8X
frequency is required--10X is better. Working backwards, that means that CD
technology can only reproduce, at best, 5.5 KHz before distortion starts to
enter in. This is independant of the construction of filters and assumes a
boxcar filter (impossible in real life.)

Other solutions have worked hard to reduce this problem by oversampling,
adding bits, etc. All these solutions smooth the distortion created by the
original system, but they can not add information back in that is lost.
What they can do is create better sounding music by smoothing out the
jaggies in the distortion.


  #2   Report Post  
Phil Allison
 
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"tubesforall" = a deranged liar


Nyquist theorem states that a non-variant signal freqency can be
reproduced that is 1/2 the sample rate.



** WRONG !!!!

It says that ANY signal with frequency components not exceeding a certain
bandwidth can be *exactly* reproduced by sampling it at a rate of double
that bandwidth.


Unfortunately, music that is invariant is not terribly interesting. Thus,
the common wisdom that 44.1KHz sampling can reproduce 22 KHz music is not
true.



** A stupid false conclusion based on the stupid false stating of Nyquist
above.



A seminal paper from MIT shows that distortion related to sampling must
consider both the sample rate and the target word size.



** Correct - the Nyquist sampling theorem assumes accurate samples.


For today's CDs--that is 16 bits.



** Which is **highly accurate** sampling.


Thus, according to this paper, a minimum of 8X frequency is required--10X
is better.



** Pure horse manure.


Working backwards, that means that CD technology can only reproduce, at
best, 5.5 KHz before distortion starts to enter in.



** More asinine bull****.

CD players can reproduce 19 kHz and 20 kHz simultaneously with no IM at
all.

Proof of perfect high frequency linearity.



Other solutions have worked hard to reduce this problem by oversampling,
adding bits, etc.



** You are a miserable, bloody liar.

**** off !!!!



.............. Phil







  #3   Report Post  
audiodir
 
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Nyquist assumes that all you want is to reproduce is a sine wave. Music is
not all sine waves.


Stu


  #4   Report Post  
Phil Allison
 
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"audiodir"

Nyquist assumes that all you want is to reproduce is a sine wave. Music is
not all sine waves.



** BULL**** !!!!!!!!

The theorem is true for any possible wave form, or combinations of
waveforms and varying in any possible way.


Stupid






............. Phil


  #5   Report Post  
R
 
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"audiodir" wrote in news:mGRWd.73204$Dc.23025
@trnddc06:

Nyquist assumes that all you want is to reproduce is a sine wave. Music is
not all sine waves.


Stu



That depends on what you consider music. One could have a recording of
square waves as an effect but there are no naturally occuring square waves.
I believe there are some bass tracks on some pop CD;s that are square waves
but that only occurs electronically and in the studio.

Most square waves on CD's are low frequencies so they get reproduced quite
well. Conversly, the LP is quite incapable of producing square waves. The
ristime is so fast that the corners would be quickly stripped off.

I recall when this topic came up once before, someone threatened to write a
sonata for function generator and drum. I don't think it ever happened
though.

r


  #6   Report Post  
Stewart Pinkerton
 
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On Sun, 6 Mar 2005 20:31:24 -0800, "tubesforall"
wrote:

Nyquist theorem states that a non-variant signal freqency can be reproduced
that is 1/2 the sample rate.


No, it actually says *less* than half the sample rate.

Unfortunately, music that is invariant is not
terribly interesting. Thus, the common wisdom that 44.1KHz sampling can
reproduce 22 KHz music is not true.


Yes. it is.

A seminal paper from MIT shows that distortion related to sampling must
consider both the sample rate and the target word size. For today's
CDs--that is 16 bits. Thus, according to this paper, a minimum of 8X
frequency is required--10X is better. Working backwards, that means that CD
technology can only reproduce, at best, 5.5 KHz before distortion starts to
enter in. This is independant of the construction of filters and assumes a
boxcar filter (impossible in real life.)


Please cite the papar, as this is contrary to current theory - and
more importantly, to current measurements, which demonstrate that
44.1k sampling is adequate for *perfect* capture of any waveform
within a 22kHz bandwidth

Other solutions have worked hard to reduce this problem by oversampling,
adding bits, etc. All these solutions smooth the distortion created by the
original system, but they can not add information back in that is lost.
What they can do is create better sounding music by smoothing out the
jaggies in the distortion.


There is *no* distortion. Cite the paper, or cite *any* measurements
which can demonstrate such distortion. Otherwise go away, troll.
--

Stewart Pinkerton | Music is Art - Audio is Engineering
  #7   Report Post  
Stewart Pinkerton
 
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On Mon, 07 Mar 2005 05:30:58 GMT, "audiodir"
wrote:

Nyquist assumes that all you want is to reproduce is a sine wave. Music is
not all sine waves.


Fourier demonstrates that *any* waveform, including music, can be
represented as a series of superimposed sinewaves. Hence, Nyquist and
Shannon are correct in their postulations. While music may not
*appear* to be sinewaves, it can be so treated for the purposes of
reproduction. Bottom line of course is that digital audio works, and
reproduces music more accurately than any other system.
--

Stewart Pinkerton | Music is Art - Audio is Engineering
  #8   Report Post  
Gregg
 
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Behold, Stewart Pinkerton scribed on tube chassis:

On Mon, 07 Mar 2005 05:30:58 GMT, "audiodir"
wrote:

Nyquist assumes that all you want is to reproduce is a sine wave. Music
is not all sine waves.


Fourier demonstrates that *any* waveform, including music, can be
represented as a series of superimposed sinewaves. Hence, Nyquist and
Shannon are correct in their postulations. While music may not *appear*
to be sinewaves, it can be so treated for the purposes of reproduction.
Bottom line of course is that digital audio works, and reproduces music
more accurately than any other system.


And after going through any compression, other than lossless (FLAC or
APE), makes the whoke kit-and-kaboodle math moot.

--
Gregg "t3h g33k"
http://geek.scorpiorising.ca
*Ratings are for transistors, tubes have guidelines*
  #9   Report Post  
audiodir
 
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I have no doubt as to the Fourier statements and theory, but that is also
based on unlimited data. It would seem that in the case where there is an
upper limit, then the data available becomes truncated as you approach that
limit, making recreation of a waveform much more difficult.
Perhaps the problem is not so much with the Nyquist theory as its
application. After all, a DAC has to set the parameters of what waveform it
seeks to recreate. I believe the Spectral DAC had an option to set different
algorithms to use in the decoding process. The waveforms generated are not
dissimilar to the differences seen in a cadcam system when asked to
interpolate a curve over various points.
While most music is a series of sine waves, there are a lot of impulses and
other unusual waveforms (think of the 'grundge' associated with rock
electric guitars and the inherent distortion those instruments can produce).
No wonder that the classical community was the first to embrace CD. I know
many rockers that even today claim that analog captures the guitar sound
more accurately.
I believe the Synclavier uses a sampling frequency of 100kHz. If it needs
that much to create a specific sound, how can a lowly 44.1 kHz sampling rate
reveal the subtleties that a programmer/musician may want to play.
At any rate, to continue this discussion is fruitless for me. There are
limitations, and whether one can hear it or not is a subjective thing.
Different people are sensitive to different things, but of course your own
personal sensitivities are all that counts.


Stu


  #10   Report Post  
Phil Allison
 
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"audiodir"

I have no doubt as to the Fourier statements and theory,




** You are a mentally defective ass with no comprehension of anything.



.............. Phil








  #11   Report Post  
Tim Williams
 
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"tubesforall" wrote in message
...
Working backwards, that means that CD
technology can only reproduce, at best, 5.5 KHz before distortion starts

to
enter in. This is independant of the construction of filters and assumes

a
boxcar filter (impossible in real life.)


Astounding theoretical work, sir! It really makes me question how a CD, let
alone a 128k MP3 can still sound good. Or maybe the theory itself is pot.
Whichever...

Tim

--
"California is the breakfast state: fruits, nuts and flakes."
Website: http://webpages.charter.net/dawill/tmoranwms


  #12   Report Post  
xrongor
 
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"Phil Allison" wrote in message
...

"tubesforall" = a deranged liar


Nyquist theorem states that a non-variant signal freqency can be
reproduced that is 1/2 the sample rate.



** WRONG !!!!

It says that ANY signal with frequency components not exceeding a certain
bandwidth can be *exactly* reproduced by sampling it at a rate of double
that bandwidth.


since you have gone on a tirade, you should be a bit more careful.

it says you need to sample it at MORE than double. double exactly is not
enough.

randy


  #13   Report Post  
Phil Allison
 
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"xrongor" = pedantic jerk off

"Phil Allison

"tubesforall" = a deranged liar



Nyquist theorem states that a non-variant signal freqency can be
reproduced that is 1/2 the sample rate.



** WRONG !!!!

It says that ANY signal with frequency components not exceeding a certain
bandwidth can be *exactly* reproduced by sampling it at a rate of double
that bandwidth.


since you have gone on a tirade,



** **** you - asshole.


you should be a bit more careful.



** So should have your parents, boy have they paid for their mistake.


it says you need to sample it at MORE than double.



** It must not be less than double the highest signal frequency - but can
be made arbitrarily close to double.

Your point is as worthless as you are.




.............. Phil




  #14   Report Post  
Patrick Turner
 
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audiodir wrote:

Nyquist assumes that all you want is to reproduce is a sine wave. Music is
not all sine waves.


But I thought is was a plethera of combined sine waves,
many with varying amplitude, phase, and frequency.
In fact music resembles noise, but music's content
has most of its frequencies related numerically...

Music by Heavy Metal is very little different to pink noise I use to test
equipment.

A Motzart concetto is quite a lot different to noise.

All have lotsa sine waves.


Patrick Turner.



Stu


  #15   Report Post  
Stewart Pinkerton
 
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On Mon, 07 Mar 2005 08:11:00 GMT, "audiodir"
wrote:

I have no doubt as to the Fourier statements and theory, but that is also
based on unlimited data. It would seem that in the case where there is an
upper limit, then the data available becomes truncated as you approach that
limit, making recreation of a waveform much more difficult.
Perhaps the problem is not so much with the Nyquist theory as its
application. After all, a DAC has to set the parameters of what waveform it
seeks to recreate. I believe the Spectral DAC had an option to set different
algorithms to use in the decoding process. The waveforms generated are not
dissimilar to the differences seen in a cadcam system when asked to
interpolate a curve over various points.


Different situation entirely. The whole point of Nyquist/Shannon is
that, if you have only two sampled points of reference, i.e. if the
samples are of a signal having a frequency between one-half and
one-third of the sampling rate, and if you *know* that the input
signal is bandlimited to less than half the sampling frequency, then
only *one* possible curve will fit the two points - it is a sine wave
of a specific frequency and amplitude.

If the curve were *not* a sine wave, then it would *by definition*
contain harmonic content, and hence would *not* be band-limited to
less than half the sampling frequency. This causes an effect known as
aliasing, which is a *distortion* which does not exist in a properly
implemented sampling system.

It is true that the Spectral and several other DACs (also some Wadia,
Pioneer and Sony CD players), did indeed use reconstruction filters
which allowed out of band products to appear at the output. This false
imaging is a *distortion*, in other words it's a bug, not a feature.
Heck, some loonytunes 'high end' players from the like of YBA and
Audio Note, don't even *have* a reconstruction filter, they just let
*all* the rubbish out!

While most music is a series of sine waves,


Actually, *all* bandwidth-limited signals can be represented as a
series of sine waves.

there are a lot of impulses and
other unusual waveforms (think of the 'grundge' associated with rock
electric guitars and the inherent distortion those instruments can produce).


What *appear* to be impulses, certainly have finite leading edges, and
can threfore be represented by a series of sine waves - even if you
need to up the sampling rate to capture anything higher than 22kHz.
But why would you want to, unless you are a cat or a bat?

BTW, the 'grunge' associated with a heavily distorted electric guitar
is all below 10kHz, so no problems capturing it.

No wonder that the classical community was the first to embrace CD. I know
many rockers that even today claim that analog captures the guitar sound
more accurately.


People make all kinds of crazy claims - and would *you* take the word
of someone who's spent the last decade with his ears three feet from a
Marshall stack - and his nose in a snowdrift? :-)

I believe the Synclavier uses a sampling frequency of 100kHz. If it needs
that much to create a specific sound, how can a lowly 44.1 kHz sampling rate
reveal the subtleties that a programmer/musician may want to play.


Who says that the Synclavier *needs* a 100k sampling rate? With modern
kit, 24/96 sampling is trivially easy (and cheap) to do, but it's very
arguable that it's *necessary* for 'perfect sound'. You probably have
a 24/96 soundcard in your PC, but do you *need* 96k sampling?

At any rate, to continue this discussion is fruitless for me. There are
limitations, and whether one can hear it or not is a subjective thing.
Different people are sensitive to different things, but of course your own
personal sensitivities are all that counts.


And not one single person has yet been found who can reliably and
repeatably tell the difference between 44.1k and 96k sampling, when
they don't *know* which is playing.
--

Stewart Pinkerton | Music is Art - Audio is Engineering


  #16   Report Post  
Joseph Meditz
 
Posts: n/a
Default

A seminal paper from MIT shows that distortion related to sampling
must
consider both the sample rate and the target word size. For today's
CDs--that is 16 bits. Thus, according to this paper, a minimum of 8X


frequency is required--10X is better. Working backwards, that means

that CD
technology can only reproduce, at best, 5.5 KHz before distortion

starts to
enter in. This is independant of the construction of filters and

assumes a
boxcar filter (impossible in real life.)



Please cite the papar, as this is contrary to current theory - and
more importantly, to current measurements, which demonstrate that
44.1k sampling is adequate for *perfect* capture of any waveform
within a 22kHz bandwidth



Other solutions have worked hard to reduce this problem by

oversampling,
adding bits, etc. All these solutions smooth the distortion created

by the
original system, but they can not add information back in that is

lost.
What they can do is create better sounding music by smoothing out the


jaggies in the distortion.



There is *no* distortion. Cite the paper, or cite *any* measurements
which can demonstrate such distortion. Otherwise go away, troll.

--

Although the OP is tangled up in his own underwear, I think that he's
alluding to the relationship between sampling rate and quantization
noise.

Joe

  #17   Report Post  
John Byrns
 
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Default

In article .com,
"Joseph Meditz" wrote:

Although the OP is tangled up in his own underwear, I think that he's
alluding to the relationship between sampling rate and quantization
noise.


Can you explain the "relationship between sampling rate and quantization
noise"? I thought sampling and quantization were two independent effects,
sampling being an essentially analog effect creating no noise within the
signal bandwidth as long as the sampling rate is greater than two times
the signal bandwidth, while quantization is the conversion of sample
values to discrete digital values and does create noise?


Regards,

John Byrns


Surf my web pages at, http://users.rcn.com/jbyrns/
  #18   Report Post  
Joseph Meditz
 
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"Joseph Meditz" wrote:
Although the OP is tangled up in his own underwear, I think that

he's
alluding to the relationship between sampling rate and quantization
noise.


Can you explain the "relationship between sampling rate and

quantization
noise"? I thought sampling and quantization were two independent

effects,
sampling being an essentially analog effect creating no noise within

the
signal bandwidth as long as the sampling rate is greater than two

times
the signal bandwidth, while quantization is the conversion of sample
values to discrete digital values and does create noise?


The sampling theorem does not mention quantization noise. It assumes
that your samples are analog, i.e., infinite precision, snapshots of
the voltage waveform and that the reconstruction filter, which connects
the dots as it were, is also ideal. In a practical system samples are
quantized to some finite precision and the reconstruction filter is not
ideal.

If you had two identical systems each using 16 bit words but differing
only in sampling rate, one being Fs = 44.1kHz and the other with Fs =
88.2 kHz, and you sampled program material using both and then played
it on two systems that used Fs = 44.1 and 88.2 kHz respectively, then
the signal to quantization noise of the second would be 3 dB, or 1/2
bit, better than the first.

Joe

  #19   Report Post  
robert casey
 
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tubesforall wrote:
Nyquist theorem states that a non-variant signal freqency can be reproduced
that is 1/2 the sample rate. Unfortunately, music that is invariant is not
terribly interesting. Thus, the common wisdom that 44.1KHz sampling can
reproduce 22 KHz music is not true.


The problem is the phasing of the signal vs the sampling clock.
If the phase of the 22.05KHz tone happens to be in a phase that
creates samples of 0, max +, 0, max -, 0, etc, and then you shift
the phase by 45 degrees you'll get samples 0.707max -, 0.707 max +,
0.707max +, 0.707 max -, 0.707 max -, etc. The amplitude, if
looked at on a scope, will look lower for the second case vs the
first. However, a good reconstruction filter should fix this.
A filter designed to "ring" to "anticipate" the missing peaks
of the 22KHz wave. True, it would get a 22.05KHz square wave
wrong, but square waves have harmonics that cannot pass thru
the Nyquest theory at the ADC used to make the CD in the first
place. And human ears can't hear them anyway, so there's no
point to keep them. A lower frequency square wave does come
out as a square wave, and the reconstruction filter does not
make it into a sine wave. The reconstruction filter can be
done in the digital domain at some oversampled clock rate,
like 8X. And use deeper words to avoid truncation errors.
Then run the output of that thru a DAC running
at 8X clock rate, and use a simple low pass to get rid of
clock crud, and the audio signal will look to be reconstructed.
Distortion products from Nyquest fall outside of human
hearing, and thus not an issue.

At the ADC connected to a mic in the recording studio,
you need a brick wall low pass filter at 22KHz. Or
else that violin will violate Nyquest as its audio spectrum
most likely goes from audible to supersonic. The supersonic
stuff will alias ("fold down") into audible crud when the
CD is played back. Oversampling at the ADC and then LPF
the signal in the digital domain works well, assuming you have
a really good and deep ADC, say 24 bits at 8X.
Remastering analog master tapes also requires care to avoid
aliasing supersonic tape noise and such.



Other solutions have worked hard to reduce this problem by oversampling,
adding bits, etc. All these solutions smooth the distortion created by the
original system, but they can not add information back in that is lost.


But nobody will miss that info, so you need not preserve it.

What they can do is create better sounding music by smoothing out the
jaggies in the distortion.

The jaggies are supersonic anyway, so they are filtered out
(to avoid intermod problems in the user's audio amp system).

  #20   Report Post  
robert casey
 
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Fourier demonstrates that *any* waveform, including music, can be
represented as a series of superimposed sinewaves. Hence, Nyquist and
Shannon are correct in their postulations. While music may not *appear*
to be sinewaves, it can be so treated for the purposes of reproduction.
Bottom line of course is that digital audio works, and reproduces music
more accurately than any other system.



And after going through any compression, other than lossless (FLAC or
APE), makes the whoke kit-and-kaboodle math moot.

CDs don't use compression (mp3 sort or the sort of thing
done by radio stations). The sample frequency and bit depth
was chosen such that the errors fall outside normal human
hearing ability.


  #21   Report Post  
tubesforall
 
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I feel good.

I got dissed by Phil and called a troll by Stewart Pinkerton in the same
thread.
Stewart has some good points worth discussing. It appears Phil only has a
10 word vocabulary.

Clearly this subject matter excites the masses.

The paper I refer to was reprinted in the industry electronic magazine
ChipCenter. This was a mag closely related to Circuit Cellar for those who
remember those days. I have a hard copy print out that I have used over the
years as a basis for determining sampling requirements in the design of
medical instrumentation.

Author: Steve Hendrix
Title: "The Nyquist Frequency Fable"

I also refer the interested to Audio Engineering Society.
http://www.aes.org/journal/online/index.cfm

There are many techical articles there covering this issue in detail. I
refer you in particular to a good synopsis by:
Marek Roland-Mieszkowski in an article titled: "Consequenses of Nyquist
Theorem for Aucoustic Signals Stored in Digital Format"

One of the misconceptions maintained by some of the post responders is that
Nyquist f/2 criteria for accurate reproduction applies to time variant
signals that are sampled over a finite length of time. It doesn't. Read
the papers, or pull out your old sampling theorem books.

An excellent related article in AES Volume 52/11 is: "Dithered Noise Shapers
and Recursive DIgital Filters" by Stanley P. Lip****z.

Another one in the same publication is "Importance and Representation of
Phase in the Sinusoidal Model" by Tue Haste Andersen. This is interesting
because distortion can also be percieved as phase change.

--------------------
It is my "opinion" (not edict) that CD technology is a very poor medium for
critical listening. I am an (ex) classical concert violinist and hyper
sensitive to distortion in the upper registers. After years of trying to
buy CD players that could reproduce violins faithfully I gave up and went
back to vinyl. The best I thought I heard were Cary tube CD players--which
colored the sound and chopped of high frequencies, but were at least
pleasant to the ear.

jaggies in the distortion.



  #22   Report Post  
Mister
 
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On Sun, 6 Mar 2005 20:31:24 -0800, "tubesforall"
wrote:

Nyquist theorem states that a non-variant signal freqency can be reproduced
that is 1/2 the sample rate. Unfortunately, music that is invariant is not
terribly interesting.


wot the fuc does that mean? invariant? like disco? a pure sine wave? do you have
a clue?

Thus, the common wisdom that 44.1KHz sampling can
reproduce 22 KHz music is not true.


yes it is. happens all the time. actually, the spec is for 20khz, the clock was
made a wee bit higher, laddy, as a fudge factor.

A seminal paper from MIT shows that distortion related to sampling must
consider both the sample rate and the target word size. For today's
CDs--that is 16 bits. Thus, according to this paper, a minimum of 8X
frequency is required--10X is better. Working backwards, that means that CD
technology can only reproduce, at best, 5.5 KHz before distortion starts to
enter in.


that may be true... but harmonics above 5khz with .001% distortion are of no
conseguence. please note that all distortion of original digitized material
above 10khz is in-audible by being outside the bandwidrh.

This is independant of the construction of filters and assumes a
boxcar filter (impossible in real life.)


i guess you never swept a motorola 23 pole modem/voice filter...

Other solutions have worked hard to reduce this problem by oversampling,
adding bits, etc. All these solutions smooth the distortion created by the
original system, but they can not add information back in that is lost.


distortion of a signal doesen't lose anything, it only gains. frequency responce
anomalies can lesson some data.

What they can do is create better sounding music by smoothing out the
jaggies in the distortion.


now I guess you're ready to move on to french fry school...

  #23   Report Post  
Choky
 
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"R" wrote in message
...
| "audiodir" wrote in news:mGRWd.73204$Dc.23025
| @trnddc06:
|
| Nyquist assumes that all you want is to reproduce is a sine wave. Music
is
| not all sine waves.
|
|
| Stu
|
|
|
| That depends on what you consider music. One could have a recording of
| square waves as an effect but there are no naturally occuring square
waves.
| I believe there are some bass tracks on some pop CD;s that are square
waves
| but that only occurs electronically and in the studio.
|
| Most square waves on CD's are low frequencies so they get reproduced quite
| well. Conversly, the LP is quite incapable of producing square waves.
The
| ristime is so fast that the corners would be quickly stripped off.
|
| I recall when this topic came up once before, someone threatened to write
a
| sonata for function generator and drum. I don't think it ever happened
| though.
|
| r

luckily-those square wave aren't good cutted in vinyl ;
when you have veeeery pricey TT with almost cutter lathe quality ,
reproducing error will be of same magnitude as in cutting process,but with
exactly opposite meaning.........
ha-everything is in clever coding-encoding technology !

awesome TT is always better than awesome CD
lucky for me -I have awesome TT ,and I'm free off all this ****ty blah blah
CD against LP

btw-I also have awesome CD


--
--
.................................................. ........................
Choky
Prodanovic Aleksandar
YU

"don't use force, "don't use force,
use a larger hammer" use a larger tube
- Choky and IST"
- ZM
.................................................. ...........................


  #24   Report Post  
Mister
 
Posts: n/a
Default

On Mon, 7 Mar 2005 07:30:51 +0000 (UTC), Stewart Pinkerton
wrote:

On Mon, 07 Mar 2005 05:30:58 GMT, "audiodir"
wrote:

Nyquist assumes that all you want is to reproduce is a sine wave. Music is
not all sine waves.


Fourier demonstrates that *any* waveform, including music, can be
represented as a series of superimposed sinewaves. Hence, Nyquist and
Shannon are correct in their postulations. While music may not
*appear* to be sinewaves, it can be so treated for the purposes of
reproduction. Bottom line of course is that digital audio works, and
reproduces music more accurately than any other system.


no. digital is a way to create (perfect) storage and reproduction thereof due to
the fact you only need an acurate list of numbers.

but music and sound is analog, and an analog system with no D/A or A/D
converters is more accurate by rule of simplicity.

don't mistake added noise with quality of reproduction, as in the case of
scratchy vinyl!

an analog system with greater resolution will sound better then a digital
system, assuming you can hear the distortion products.

the last studio reel to reel machines that were made had a S/N of up to 120db,
far better than CD.

as for other systems, high freq. FM modulation tape systems also beat out CD.

all this techno and people run around listening to MP3s!

  #25   Report Post  
Phil Allison
 
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"tubesforall" =

I feel good.



** A dose of rat poison will fix that.


I got dissed by Phil and called a troll by Stewart Pinkerton in the same
thread.



** **** off - troll.


Stewart has some good points worth discussing. It appears Phil only has a
10 word vocabulary.



** **** off you stinking troll.



The paper I refer to was reprinted in the industry electronic magazine
ChipCenter. This was a mag closely related to Circuit Cellar for those
who remember those days.



** ROTFLMAO - a pile of worthless excreta indeed.


I also refer the interested to Audio Engineering Society.
http://www.aes.org/journal/online/index.cfm



** More ****e comes from that bunch of fruit loops than a down a sewer.


There are many techical articles there covering this issue in detail.



** What issue ??

You have not even vaguely described an issue !!!!



One of the misconceptions maintained by some of the post responders is
that Nyquist f/2 criteria for accurate reproduction applies to time
variant signals that are sampled over a finite length of time.



** All signals vary in time - or else they are just DC voltages. All
lengths of time are finite.

A /D converters use sample and hold circuits so the sample it taken at
particular instant.

You post no case since you ****ing do not have one.


It doesn't.



** **** off you asinine troll.


It is my "opinion" (not edict) that CD technology is a very poor medium
for critical listening. I am an (ex) classical concert violinist and
hyper sensitive to distortion in the upper registers. After years of
trying to buy CD players that could reproduce violins faithfully I gave up
and went back to vinyl. The best I thought I heard were Cary tube CD
players--which colored the sound and chopped of high frequencies, but were
at least pleasant to the ear.




** Now you reveal what this troll is all about - fetishist bloody vinyl
lunacy !!!


**** OFF YOU ASININE ****ING VINYL BIGOT !!!!






................... Phil







  #26   Report Post  
Phil Allison
 
Posts: n/a
Default


"Mister" = horse and cart lover

no. digital is a way to create (perfect) storage and reproduction thereof
due to
the fact you only need an acurate list of numbers.

but music and sound is analog, and an analog system with no D/A or A/D
converters is more accurate by rule of simplicity.



** Just like a horse and cart is a better mode of transport !!!!!!!!

What a colossal ****wit !!!!



don't mistake added noise with quality of reproduction, as in the case of
scratchy vinyl!



** Audible noise is bad reproduction per se.



an analog system with greater resolution will sound better then a digital
system, assuming you can hear the distortion products.



** A cart with enough horses is better that a car ??


the last studio reel to reel machines that were made had a S/N of up to
120db,
far better than CD.



** Massive, stupid lie.


as for other systems, high freq. FM modulation tape systems also beat out
CD.



** Second massive, stupid lie.

Another demented vinyl bigot for sure.




................. Phil





  #28   Report Post  
Gregg
 
Posts: n/a
Default

Behold, robert casey scribed on tube chassis:



Fourier demonstrates that *any* waveform, including music, can be
represented as a series of superimposed sinewaves. Hence, Nyquist and
Shannon are correct in their postulations. While music may not *appear*
to be sinewaves, it can be so treated for the purposes of reproduction.
Bottom line of course is that digital audio works, and reproduces music
more accurately than any other system.



And after going through any compression, other than lossless (FLAC or
APE), makes the whoke kit-and-kaboodle math moot.

CDs don't use compression (mp3 sort or the sort of thing done by radio
stations). The sample frequency and bit depth was chosen such that the
errors fall outside normal human hearing ability.


I'm aware of that.

Even if you had the perfect CD (ie - a vinyl record), how many people
store that music as .wav?

That's what I meant.

--
Gregg "t3h g33k"
http://geek.scorpiorising.ca
*Ratings are for transistors, tubes have guidelines*
  #29   Report Post  
Stewart Pinkerton
 
Posts: n/a
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On 7 Mar 2005 10:56:21 -0800, "Joseph Meditz"
wrote:

Although the OP is tangled up in his own underwear, I think that he's
alluding to the relationship between sampling rate and quantization
noise.


That would still imply significant tangling, since there exists no
such relationship. Quantisation noise as a signal-correlated artifact
is completely removed by the correct use of around 1/2 LSB of dither.
This has nothing to do with sample rate.
--

Stewart Pinkerton | Music is Art - Audio is Engineering
  #30   Report Post  
Stewart Pinkerton
 
Posts: n/a
Default

On 7 Mar 2005 12:48:24 -0800, "Joseph Meditz"
wrote:

"Joseph Meditz" wrote:
Although the OP is tangled up in his own underwear, I think that

he's
alluding to the relationship between sampling rate and quantization
noise.


Can you explain the "relationship between sampling rate and

quantization
noise"? I thought sampling and quantization were two independent

effects,
sampling being an essentially analog effect creating no noise within

the
signal bandwidth as long as the sampling rate is greater than two

times
the signal bandwidth, while quantization is the conversion of sample
values to discrete digital values and does create noise?


The sampling theorem does not mention quantization noise. It assumes
that your samples are analog, i.e., infinite precision, snapshots of
the voltage waveform and that the reconstruction filter, which connects
the dots as it were, is also ideal. In a practical system samples are
quantized to some finite precision and the reconstruction filter is not
ideal.

If you had two identical systems each using 16 bit words but differing
only in sampling rate, one being Fs = 44.1kHz and the other with Fs =
88.2 kHz, and you sampled program material using both and then played
it on two systems that used Fs = 44.1 and 88.2 kHz respectively, then
the signal to quantization noise of the second would be 3 dB, or 1/2
bit, better than the first.


Ah, I see what you're getting at. This is true, but occurs at such a
low level as to be sonically insignificant. Certainly, no one has
demonstrated an ability to hear this effect.
--

Stewart Pinkerton | Music is Art - Audio is Engineering


  #31   Report Post  
Stewart Pinkerton
 
Posts: n/a
Default

On Tue, 08 Mar 2005 02:55:13 GMT, Chris Hornbeck
wrote:

On Mon, 07 Mar 2005 13:32:36 -0600, (John Byrns) wrote:

In article .com,
"Joseph Meditz" wrote:

Although the OP is tangled up in his own underwear, I think that he's
alluding to the relationship between sampling rate and quantization
noise.


Can you explain the "relationship between sampling rate and quantization
noise"? I thought sampling and quantization were two independent effects,
sampling being an essentially analog effect creating no noise within the
signal bandwidth as long as the sampling rate is greater than two times
the signal bandwidth, while quantization is the conversion of sample
values to discrete digital values and does create noise?


Another way to say what Joseph means is that finite quantization
introduces what are effectively timing errors ("jitter") in the
complete A/D/A conversion.


While this is true, it's happening at more than 90dB below peak level.
I'm not aware of anyone having demonstrated an ability to hear the
difference among various sample rates, given a common signal
band-limited to the requirements of the lowest sampling rate, i.e. the
20kHz which is commonly taken to be the limit of human hearing.

In the A/D/A worlds, noise and distortion are *not* different things.
And neither are amplitude and frequency modulation distortions. (Or
course, they weren't in the old analog world either; we just didn't
talk about it that way).

Digital storage is theoretically perfect after being bandwidth
limited, dynamic range limited, and quantized-and-back monotonically.
Discussion really ought to be targeted at the limitations, IMO.


Indeed, and these limits are *way* below the limits of any analogue
system, indeed they're below the noise floor of most tube amps!
--

Stewart Pinkerton | Music is Art - Audio is Engineering
  #32   Report Post  
Stewart Pinkerton
 
Posts: n/a
Default

On Tue, 08 Mar 2005 06:11:04 GMT, Gregg wrote:

Behold, robert casey scribed on tube chassis:

Fourier demonstrates that *any* waveform, including music, can be
represented as a series of superimposed sinewaves. Hence, Nyquist and
Shannon are correct in their postulations. While music may not *appear*
to be sinewaves, it can be so treated for the purposes of reproduction.
Bottom line of course is that digital audio works, and reproduces music
more accurately than any other system.


And after going through any compression, other than lossless (FLAC or
APE), makes the whoke kit-and-kaboodle math moot.

CDs don't use compression (mp3 sort or the sort of thing done by radio
stations). The sample frequency and bit depth was chosen such that the
errors fall outside normal human hearing ability.


I'm aware of that.

Even if you had the perfect CD (ie - a vinyl record),


Um, a vinyl record is *very* far from perfect! This is easily
demonstrated by recording vinyl onto CD-R - it sounds just like the
original vinyl. Now try cutting vinyl from a CD, and see what you
get................

Any professional recording engineer can tell you that CD is *much*
closer to the master tape than is vinyl.

how many people
store that music as .wav?


Me, for one. A single 120GB disk can hold about 200 CDs.

--

Stewart Pinkerton | Music is Art - Audio is Engineering
  #33   Report Post  
Stewart Pinkerton
 
Posts: n/a
Default

"tubesforall" wrote:

It is my "opinion" (not edict) that CD technology is a very poor medium
for critical listening.


That is definitely a minority opinion, and it's a plain fact that it's
massively superior top vinyl, so that doiesn't leave you with much
option in prerecorded music.............

I am an (ex) classical concert violinist and
hyper sensitive to distortion in the upper registers. After years of
trying to buy CD players that could reproduce violins faithfully I gave up
and went back to vinyl. The best I thought I heard were Cary tube CD
players--which colored the sound and chopped of high frequencies, but were
at least pleasant to the ear.


You may *prefer* the distinctive sound of vinyl, but it's certainly
not accurate. I share your sensitivity to harsh treble, but I have
found CD to be exceptionally good in the upper reaches of the
spectrum, whereas vinyl becomes very 'splashy', especially in the
inner grooves. Perhaps you need better speakers? :-)
--

Stewart Pinkerton | Music is Art - Audio is Engineering
  #34   Report Post  
Gregg
 
Posts: n/a
Default

Behold, Stewart Pinkerton scribed on tube chassis:

On Tue, 08 Mar 2005 06:11:04 GMT, Gregg wrote:

Behold, robert casey scribed on tube chassis:

Fourier demonstrates that *any* waveform, including music, can be
represented as a series of superimposed sinewaves. Hence, Nyquist and
Shannon are correct in their postulations. While music may not
*appear* to be sinewaves, it can be so treated for the purposes of
reproduction. Bottom line of course is that digital audio works, and
reproduces music more accurately than any other system.


And after going through any compression, other than lossless (FLAC or
APE), makes the whoke kit-and-kaboodle math moot.

CDs don't use compression (mp3 sort or the sort of thing done by radio
stations). The sample frequency and bit depth was chosen such that
the errors fall outside normal human hearing ability.


I'm aware of that.

Even if you had the perfect CD (ie - a vinyl record),


Um, a vinyl record is *very* far from perfect! This is easily
demonstrated by recording vinyl onto CD-R - it sounds just like the
original vinyl. Now try cutting vinyl from a CD, and see what you
get................

Any professional recording engineer can tell you that CD is *much*
closer to the master tape than is vinyl.


That was sorta tongue-in-cheek ;-)

There are cases where vinyl is more accurate. Example is Rhino Records
embelleshed an "oldies sound" to the CD's, where the original recordings
sounded quite nice.

how many people
store that music as .wav?


Me, for one. A single 120GB disk can hold about 200 CDs.


Awesome!

I prefer FLAC, just to make more room and it's less CPU intensive in my
player, XMMS.

Lossless rocks :-)))))))

--
Gregg "t3h g33k"
http://geek.scorpiorising.ca
*Ratings are for transistors, tubes have guidelines*
  #35   Report Post  
Ruud Broens
 
Posts: n/a
Default


"Stewart Pinkerton" wrote in message
...
: "tubesforall" wrote:
:
: It is my "opinion" (not edict) that CD technology is a very poor medium
: for critical listening.
:
: That is definitely a minority opinion, and it's a plain fact that it's
: massively superior top vinyl, so that doiesn't leave you with much
: option in prerecorded music.............

Not an argument at all. Rather the point should be, why stick with the
at-the-time best solution nowadays, that is, there is no need to limit the
format
to 16 bits 44.1 KHz sampling - polycarbonate costs the same, whatever you
sandwitch in between. With some 9 GB available on DVD, provision for
200 GB on the Blu Ray format disks, the CD format should become a relic !
Rudy

:
: I am an (ex) classical concert violinist and
: hyper sensitive to distortion in the upper registers. After years of
: trying to buy CD players that could reproduce violins faithfully I gave up
: and went back to vinyl. The best I thought I heard were Cary tube CD
: players--which colored the sound and chopped of high frequencies, but were
: at least pleasant to the ear.
:
: You may *prefer* the distinctive sound of vinyl, but it's certainly
: not accurate. I share your sensitivity to harsh treble, but I have
: found CD to be exceptionally good in the upper reaches of the
: spectrum, whereas vinyl becomes very 'splashy', especially in the
: inner grooves. Perhaps you need better speakers? :-)
: --
:
: Stewart Pinkerton | Music is Art - Audio is Engineering




  #36   Report Post  
Patrick Turner
 
Posts: n/a
Default



Stewart Pinkerton wrote:

"tubesforall" wrote:

It is my "opinion" (not edict) that CD technology is a very poor medium
for critical listening.


That is definitely a minority opinion, and it's a plain fact that it's
massively superior top vinyl, so that doiesn't leave you with much
option in prerecorded music.............

I am an (ex) classical concert violinist and
hyper sensitive to distortion in the upper registers. After years of
trying to buy CD players that could reproduce violins faithfully I gave up
and went back to vinyl. The best I thought I heard were Cary tube CD
players--which colored the sound and chopped of high frequencies, but were
at least pleasant to the ear.


You may *prefer* the distinctive sound of vinyl, but it's certainly
not accurate. I share your sensitivity to harsh treble, but I have
found CD to be exceptionally good in the upper reaches of the
spectrum, whereas vinyl becomes very 'splashy', especially in the
inner grooves. Perhaps you need better speakers? :-)


I know several people with the very latest
in digital replay equipment.
They also have vinyl replay gear worth many times the value of the digital
replay gear.
I have also been to their residences while they played digital discs
and vinyl discs from the same master tapes at the same time,
and switched between the two, using the same speakers and amps,
although the vinyl did need the extra preamp.

In nearly all cases, the vinyl was percieved to be more accurate,
threw up a better sound stage, and gave a more emotionally involving
experience which seemed to be more real than the one conveyed digitally.

Vinyl refuses to go entirely away.

When its really good, and by no means is it ever always good,
its bleedin fabulous, and the same might be said about digital,
but I don't know many who'd say that digital will always be better than the best
from vinyl..

Now pinky, your stance about
percieved accuracy or real accuracy is of little merit in the minds of the ppl I
know
who don't have a clue anout technical issues; all that matters is the sound.

No need to throw away your TT yet folks.....

Just enjoy what is enjoyable.....

Patrick Turner.


--

Stewart Pinkerton | Music is Art - Audio is Engineering


  #37   Report Post  
Sander deWaal
 
Posts: n/a
Default

Mister said:

the last studio reel to reel machines that were made had a S/N of up to 120db,
far better than CD.



??????????????????

--
Sander de Waal
" SOA of a KT88? Sufficient. "
  #38   Report Post  
Stewart Pinkerton
 
Posts: n/a
Default

On Wed, 09 Mar 2005 02:36:17 +1100, Patrick Turner
wrote:

Vinyl refuses to go entirely away.


Gettin' darned close these days..... :-)

When its really good, and by no means is it ever always good,
its bleedin fabulous, and the same might be said about digital,
but I don't know many who'd say that digital will always be better than the best
from vinyl..


The best digital always will be.

Now pinky, your stance about
percieved accuracy or real accuracy is of little merit in the minds of the ppl I
know
who don't have a clue anout technical issues; all that matters is the sound.


They probably prefer toobs, too................. :-)

No need to throw away your TT yet folks.....


Not until you've archived all your vinyl, at least! I do find it
hilarious that you can easily capture all that 'vinyl magic' by
recording it to CD-R, and yet people *still* have this weird belief
that CD somehow magically 'loses' something that vinyl
retains.........

Just enjoy what is enjoyable.....


Indeed - it's the *performance* that counts, even on an old table
radio.

--

Stewart Pinkerton | Music is Art - Audio is Engineering
  #39   Report Post  
Don Pearce
 
Posts: n/a
Default

On Tue, 8 Mar 2005 15:41:23 +0100, "Ruud Broens"
wrote:

Not an argument at all. Rather the point should be, why stick with the
at-the-time best solution nowadays, that is, there is no need to limit the
format
to 16 bits 44.1 KHz sampling - polycarbonate costs the same, whatever you
sandwitch in between. With some 9 GB available on DVD, provision for
200 GB on the Blu Ray format disks, the CD format should become a relic !
Rudy


You still have to buy the stuff, and I'm sure I wouldn't want to buy
the kind of music package that needed that kind of storage capacity. I
wouldn't listen to most of it.

I think the CD is around for a while yet.

d

Pearce Consulting
http://www.pearce.uk.com
  #40   Report Post  
cowboy
 
Posts: n/a
Default


"Stewart Pinkerton" wrote in message
.. .
On Wed, 09 Mar 2005 02:36:17 +1100, Patrick Turner
wrote:

Vinyl refuses to go entirely away.


Gettin' darned close these days..... :-)

When its really good, and by no means is it ever always good,
its bleedin fabulous, and the same might be said about digital,
but I don't know many who'd say that digital will always be better than
the best
from vinyl..


The best digital always will be.


Pinkerton:

we could actually take you much more seriously on the subject of digital vs.
vinyl, if it weren't for the fact that your ears are too tone-deaf to hear
the day & night superiority of the musicality of tubes over the relatively
ugly harsh sound of even the best solid state gear. I am not saying that
vinyl is superior to digital, but rather just saying that Pinky is
ill-equipped to hear the differences, he has no credibility if he can't even
hear that tubes sound better than transistors.

This group would not even exist, if not for the fact that a lot of people in
the world have the ears to hear the huge difference between tubes and solid
state.

It is sad that you must have used a circular saw too much in your youth, and
can nowadays not hear the harshness of even the best solid state equipment.
If transistors and negative feedback were good things, no one in their right
mind would put up with tubes, tube reliability, tube efficiency, tube
testers or any of this mess.

It hurts my heart that you haven't discovered the magic of tubes, reminds me
of a friend who has never been fortunate enough to make love to a beautiful
woman, I can describe it to him, but he will never understand what all the
fuss is about until he experiences it and the magic will have taken him in.
Great food is another analogy, one can say great food is over-rated, only
until they experience it and it clicks in their brain that they have been
missing a huge something all these years.


cheers!

cowboy


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