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Tim Martin
Guest
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 Posted: Fri Jun 03, 2005 9:04 pm Post subject: Re: HiFi vs. Computer |
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<dpierce@cartchunk.org> wrote in message
news:1117820624.239670.28190@g47g2000cwa.googlegroups.com...
| Quote: |
Tim Martin wrote:
"Karl Uppiano" <karl.uppiano@verizon.net> wrote in message
news:SXTne.16106$qJ3.7554@trnddc05...
Nyquist's Theoerem is about representation of periodic signals;
most
sounds
are not periodic signals.
Any arbitrary sound can be decomposed into a sum of periodic signals.
OK; let's suppose the sound consists of silence, followed by one second
of
1kHz sine wave, followed by silence.
What sum of periodic signals can this be decomposed into?
It's exactly the same as a 1 kHz sine wave 100% modulated by a 0.5
Hz square wave ...
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Wouldn't that produce a repeating signal of one second of 1000Hz alternating
with one second of silence... or have I misunderstood what you're
suggesting?
Tim |
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Geoff Wood
Guest
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| Posted: Fri Jun 03, 2005 9:04 pm Post subject: Re: HiFi vs. Computer |
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"Tim Martin" <tim2718281@ntlworld.com> wrote in message news:RyWne.298
| Quote: | Take a waveform consisting of, say a 1000Hz sine wave that is repeatedly
switched on and off at random times. This is not a periodic waveform, and
cannot be represented exactly by a Fourier transform. It has infinite
bandwidth. (Conceptually at least. As you've previously remarked, there
are physical constraints imposed by the transmission medium.) S
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Duration is not bandwidth.
geoff |
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Tim Martin
Guest
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| Posted: Fri Jun 03, 2005 10:03 pm Post subject: Re: HiFi vs. Computer |
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"Stewart Pinkerton" <patent3@dircon.co.uk> wrote in message
news:dg41a15qfesbh6ib6fg84jj2nasgkh74jh@4ax.com...
Tim
| Quote: | If we are sampling 48000 times a second, and if the sine wave is long
enough
when on, our digital signal will, after some start-up sequence, consist
of a
repeating pattern of 24 ones followed by 24 zeroes. Once we're in this
part
of the signal, we can reproduce the original sine wave exactly (within
the
resolution limits, which are not the issue here.)
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Srewart
| Quote: | No, it won't. You really don't understand the basics of A/D
conversion, do you? The waveform will exactly replicate the original
sine wave, so there will be less ones than zeroes, in the appropriate
ratio (can't be arsed to work it out offhand, but it's around 10:14
ratio, rather than 24:24)
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That's not correct. Each digital value represents a range of signal levels.
I picked the case of a sine wave; if the amplitude is x, then one of the
digital values will represent signal levels from 0 to x, and the other will
represent values from 0 to -x. (Remember, I specified a signal whose
amplitude was such as to be represented by one bit.)
The ratio of the frequencies of the two digital values represents the
offset - that is, the average value of the sine wave signal. I simply took
the case where the average is zero, so there are equal numbers of above-zero
and below-zero values.
Of course one can devise other representation systems; however, the average
of the values represented by the digital signals will be the average value
of the signal being represented.
Tim |
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Codifus
Guest
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| Posted: Fri Jun 03, 2005 10:03 pm Post subject: Re: HiFi vs. Computer |
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Tim Martin wrote:
| Quote: | "calmar" <calmar@calmar.ws> wrote in message
news:slrnd9k81l.ei1.calmar@news.calmar.ws...
Maybe it's possible to say some general things about these ways to play
musik quality related?
Well, yes. With a computer-based system, you can play the audio files
directly, without first having to convert them to a hi-fi-friendly format
(eg CD), thus omitting the accompany audio degradation. Whether the
difference is detectable is debatable, of course.
Also, the computer-based system can use software processing. So you can,
for example, compensate for speaker driver impedance variations, without
having to build it into the speaker crossover or install a hardware EQ or
DSP system.
Tim
The audio degradation going from digital-on-computer to |
digital-on-audio-CD is absolutely miniscule.
From what I've seen on your posts, I think you need to forget about the
computer for a second. Start with getting an understanding of how a
square wave is a gazzilion sine waves, all slightly out of phase from
each other, summed together. Then you'll be off to a good start.
CD |
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Tim Martin
Guest
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| Posted: Sat Jun 04, 2005 1:03 am Post subject: Re: HiFi vs. Computer |
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"Codifus" <codifus@optonline.net> wrote in message
news:d7qi5n$u8$1@news.interpublic.com...
| Quote: | From what I've seen on your posts, I think you need to forget about the
computer for a second. Start with getting an understanding of how a
square wave is a gazzilion sine waves, all slightly out of phase from
each other, summed together. Then you'll be off to a good start.
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They're not out of phase with each other, they are in phase..
Tim |
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Ben Bradley
Guest
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| Posted: Sat Jun 04, 2005 1:03 am Post subject: Re: HiFi vs. Computer |
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On Sat, 04 Jun 2005 00:20:23 GMT, "Tim Martin"
<tim2718281@ntlworld.com> wrote:
| Quote: |
"Codifus" <codifus@optonline.net> wrote in message
news:d7qi5n$u8$1@news.interpublic.com...
From what I've seen on your posts, I think you need to forget about the
computer for a second. Start with getting an understanding of how a
square wave is a gazzilion sine waves, all slightly out of phase from
each other, summed together. Then you'll be off to a good start.
They're not out of phase with each other, they are in phase..
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The phase doesn't even matter, except to that Gibbs guy...
-----
http://mindspring.com/~benbradley |
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Stewart Pinkerton
Guest
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| Posted: Sat Jun 04, 2005 6:57 am Post subject: Re: HiFi vs. Computer |
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On Wed, 1 Jun 2005 18:27:56 -0400, "mc" <mc_no_spam@uga.edu> wrote:
| Quote: | That is, for any method of storing an analog signal in x bits, it is
possible to devise a digital storage mechanism using >x bits which can be
used to reproduce a more accurate rendition of the original analog
signal.
What if the original x bits has more resolution than the original media ?
If it has sufficiently more, then the digital copy will be at least as
accurate as any analog copy.
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You don't understand how digital works, do you? Provided that there
are enough bits to encompass the dynamic range of the analoge signal,
then more bits will provide *zero* extra accuracy, and the *only*
inaccuracies will be those of implementation - in theory, the
recording *is* perfect.
--
Stewart Pinkerton | Music is Art - Audio is Engineering |
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Stewart Pinkerton
Guest
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| Posted: Sat Jun 04, 2005 6:57 am Post subject: Re: HiFi vs. Computer |
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On Fri, 03 Jun 2005 19:36:30 GMT, "Tim Martin"
<tim2718281@ntlworld.com> wrote:
| Quote: |
"Arny Krueger" <arnyk@hotpop.com> wrote in message
news:adadnbmtJeNhOT3fRVn-rg@comcast.com...
I thought this discussion was about audio, not how many
measurements can dance on the head of an imaginary pin! ;-)
Well, my initial reponse was to a comment about compression. Someone said
that he felt compression was undesirable; I've simply pointed that
compression is inherent in digital representation of signals.
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No, you've *claimed* that it is. You are wrong.
| Quote: | Having got rid of the compression bogey-man, one is free to assess any
storage scheme on its merits.
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Yup, and digital neither compresses the original analogue signal (if
sufficient sampling rate and bit depth are used), nor does it degrade
when copied.
--
Stewart Pinkerton | Music is Art - Audio is Engineering |
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Stewart Pinkerton
Guest
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| Posted: Sat Jun 04, 2005 6:57 am Post subject: Re: HiFi vs. Computer |
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On Wed, 1 Jun 2005 18:27:15 -0400, "mc" <mc_no_spam@uga.edu> wrote:
| Quote: | Any digital storage of an analog signal compresses it.
No, it doesn't. This is utter nonsense, but a persistent urban myth.
No storage of any analog signal is perfect, whether you digitize it or not.
There are errors with digital storage but there are also errors with analog
storage.
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However, the errors with digital are those of implementation, *not* of
inherent limitation.
| Quote: | With digital, once you've encoded the signal, you can store and retrieve it
with no further change. Not with analog.
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Quite so, and perhaps the most important in terms of getting the
signal from the microphione to the speakers with minimum intervening
degradation. Compare and contrast with the *minimum* seven stages of
degradation between microphone and speakers in a vinyl-based system
playing an analogue recording.
| Quote: | Getting back to the question of whether there is such a thing as lossless
compression of digital signals: Yes, certainly. The simplest is run-length
encoding. Whenever a value recurs (e.g., a long series of zeroes for a
period of silence), instead of repeating the value over and over, you
precede it with a code that means "repeat this value such-and-such number of
times."
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Anf of course, being lossless, this has no bearing on the final output
signal.
--
Stewart Pinkerton | Music is Art - Audio is Engineering |
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Stewart Pinkerton
Guest
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| Posted: Sat Jun 04, 2005 6:59 am Post subject: Re: HiFi vs. Computer |
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On Fri, 03 Jun 2005 21:12:09 GMT, "Tim Martin"
<tim2718281@ntlworld.com> wrote:
| Quote: |
"Stewart Pinkerton" <patent3@dircon.co.uk> wrote in message
news:dg41a15qfesbh6ib6fg84jj2nasgkh74jh@4ax.com...
Tim
If we are sampling 48000 times a second, and if the sine wave is long
enough
when on, our digital signal will, after some start-up sequence, consist
of a
repeating pattern of 24 ones followed by 24 zeroes. Once we're in this
part
of the signal, we can reproduce the original sine wave exactly (within
the
resolution limits, which are not the issue here.)
Srewart
No, it won't. You really don't understand the basics of A/D
conversion, do you? The waveform will exactly replicate the original
sine wave, so there will be less ones than zeroes, in the appropriate
ratio (can't be arsed to work it out offhand, but it's around 10:14
ratio, rather than 24:24)
That's not correct. Each digital value represents a range of signal levels.
I picked the case of a sine wave; if the amplitude is x, then one of the
digital values will represent signal levels from 0 to x, and the other will
represent values from 0 to -x. (Remember, I specified a signal whose
amplitude was such as to be represented by one bit.)
The ratio of the frequencies of the two digital values represents the
offset - that is, the average value of the sine wave signal. I simply took
the case where the average is zero, so there are equal numbers of above-zero
and below-zero values.
Of course one can devise other representation systems; however, the average
of the values represented by the digital signals will be the average value
of the signal being represented.
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As noted, you don't understand digital audio. The word for today is
'dither', go look it up.
--
Stewart Pinkerton | Music is Art - Audio is Engineering |
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Karl Uppiano
Guest
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| Posted: Sat Jun 04, 2005 8:01 am Post subject: Re: HiFi vs. Computer |
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<dpierce@cartchunk.org> wrote in message
news:1117820624.239670.28190@g47g2000cwa.googlegroups.com...
| Quote: |
Tim Martin wrote:
"Karl Uppiano" <karl.uppiano@verizon.net> wrote in message
news:SXTne.16106$qJ3.7554@trnddc05...
Nyquist's Theoerem is about representation of periodic signals; most
sounds
are not periodic signals.
Any arbitrary sound can be decomposed into a sum of periodic signals.
OK; let's suppose the sound consists of silence, followed by one second
of
1kHz sine wave, followed by silence.
What sum of periodic signals can this be decomposed into?
It's exactly the same as a 1 kHz sine wave 100% modulated by a 0.5
Hz square wave, and such decomposes into a series of sine components
spaced 1 Hz apart spaced symmetrically about the 1 kHz component
offset from it by 0.5 Hz, (iow 1000.5, 1001.5, 1002.5, 999.5, 998.5,
997.5, ...) with amplitudes decreasing as we move away from 1 kHz by
a simple 1/n, n = 1, 3, 5, ... and so forth, all components in phase.
AND, if you insist on truning on and off the sine wave INSTANTANEOUSLY,
these sine components extend to +-infinite frequency.
Such a signal, as I am sure you will agree, could never be PRODUCED
perfectly in ANY system existing for finite time or limited to finite
energy.
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Which is why, for any practical A/D converter, anti-aliasing filters are
built in. Because it is quite possible, even likely, that some signals
entering the system would otherwise violate the Nyquist criterion for any
given sample rate. They must be removed. But the signals that *are* recorded
can be accurately reproduced within the bandwidth specified by the design. |
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Eiron
Guest
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| Posted: Sat Jun 04, 2005 8:01 am Post subject: Re: HiFi vs. Computer |
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Ben Bradley wrote:
| Quote: | From what I've seen on your posts, I think you need to forget about the
computer for a second. Start with getting an understanding of how a
square wave is a gazzilion sine waves, all slightly out of phase from
each other, summed together. Then you'll be off to a good start.
They're not out of phase with each other, they are in phase..
The phase doesn't even matter, except to that Gibbs guy...
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And it's not a gazillion sine waves. You would be surprised at how few
harmonics are required to make it indistinguishable (to the ear) from a
real square wave.
--
Eiron. |
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Karl Uppiano
Guest
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| Posted: Sat Jun 04, 2005 9:02 am Post subject: Re: HiFi vs. Computer |
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"Tim Martin" <tim2718281@ntlworld.com> wrote in message
news:_FUne.122$K5.62@newsfe4-win.ntli.net...
| Quote: |
"Karl Uppiano" <karl.uppiano@verizon.net> wrote in message
news:SXTne.16106$qJ3.7554@trnddc05...
Nyquist's Theoerem is about representation of periodic signals; most
sounds
are not periodic signals.
Any arbitrary sound can be decomposed into a sum of periodic signals.
OK; let's suppose the sound consists of silence, followed by one second
of
1kHz sine wave, followed by silence.
What sum of periodic signals can this be decomposed into?
Tim
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Why not pick something really tough, like silence since the beginning of
time (t = -infinity), followed by a single, infinitely short impulse of
amplitude = 1 when t = 0, followed by silence until the end of time (t =
+infinity). What sum of periodic signals can this be decomposed into?
Answer: The sum of all periodic signals from DC to infinity. This
interesting signal (or a realizable approximation of it) can be used to
determine the frequency response of a system literally with one click. When
you take the Fourier transform of the impulse response of a system, you get
the frequency response (and phase response!). |
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Stewart Pinkerton
Guest
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| Posted: Sat Jun 04, 2005 11:56 am Post subject: Re: HiFi vs. Computer |
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On Sat, 04 Jun 2005 12:04:25 +0100, Laurence Payne
<lp@laurenceNOSPAMpayne.freeserve.co.uk> wrote:
| Quote: | On Wed, 01 Jun 2005 20:59:25 -0400, Joe Kesselman
keshlam-nospam@comcast.net> wrote:
Well, yes, analog is _theoretically_ infinite precision.
Only if your theory allows zero noise and infinite signal amplitude.
That's VERY theoretical :-)
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And could only happen at absolute zero, which wouldn't be good for
your vinyl................ :-)
--
Stewart Pinkerton | Music is Art - Audio is Engineering |
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Laurence Payne
Guest
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| Posted: Sat Jun 04, 2005 12:06 pm Post subject: Re: HiFi vs. Computer |
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On Wed, 01 Jun 2005 20:59:25 -0400, Joe Kesselman
<keshlam-nospam@comcast.net> wrote:
| Quote: | Well, yes, analog is _theoretically_ infinite precision.
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Only if your theory allows zero noise and infinite signal amplitude.
That's VERY theoretical :-) |
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