HiFi vs. Computer

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"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..

Tim

 

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On Sat, 04 Jun 2005 00:20:23 GMT, "Tim Martin"
<tim2718281@ntlworld.com> wrote:

>
>"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..

The phase doesn't even matter, except to that Gibbs guy...

>
>Tim
>

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On Wed, 1 Jun 2005 18:27:56 -0400, "mc" <mc_no_spam@uga.edu> wrote:

>>> 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.

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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On Fri, 03 Jun 2005 19:36:30 GMT, "Tim Martin"
<tim2718281@ntlworld.com> wrote:

>
>"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.

No, you've *claimed* that it is. You are wrong.

>Having got rid of the compression bogey-man, one is free to assess any
>storage scheme on its merits.

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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On Wed, 1 Jun 2005 18:27:15 -0400, "mc" <mc_no_spam@uga.edu> wrote:

>>>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.

However, the errors with digital are those of implementation, *not* of
inherent limitation.

>With digital, once you've encoded the signal, you can store and retrieve it
>with no further change. Not with analog.

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.

>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."

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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On Fri, 03 Jun 2005 21:12:09 GMT, "Tim Martin"
<tim2718281@ntlworld.com> wrote:

>
>"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.

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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<dpierce@cartchunk.org> wrote in message
news:1117820624.239670.28190@g47g2000cwa.googlegroups.com...
>
>
> 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.
>

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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"Tim Martin" <tim2718281@ntlworld.com> wrote in message
news:_FUne.122$K5.62@newsfe4-win.ntli.net...
>
> "Karl Uppiano" <karl.uppiano@verizon.net> wrote in message
> news:SXTne.16106$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

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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Ben Bradley wrote:

>>> 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...

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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"Tim Martin" <tim2718281@ntlworld.com> wrote in message news:RyWne.298

> 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

Duration is not bandwidth.

geoff

 

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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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On Sat, 04 Jun 2005 12:04:25 +0100, Laurence Payne
<lp@laurenceNOSPAMpayne.freeserve.co.uk> wrote:

>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

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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On Fri, 03 Jun 2005 02:07:30 GMT, "Tim Martin"
<tim2718281@ntlworld.com> wrote:

>What have tapes and microphones got to do with it? I was talking about
>analog signals, not recorded approximations of analog signals.

And this analogue signal was captured from where?

 

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On Fri, 03 Jun 2005 02:19:12 GMT, "Tim Martin"
<tim2718281@ntlworld.com> wrote:

>An analog signal, such as a bird singing in the woods, has infinite
>bandwidth.

No it isn't. It is constrained by all sorts of physical limitations,
both in the bird and in the environment.

 

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Stewart Pinkerton <patent3@dircon.co.uk> wrote:
> On Sat, 04 Jun 2005 12:04:25 +0100, Laurence Payne
> <lp@laurenceNOSPAMpayne.freeserve.co.uk> wrote:

>>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

> And could only happen at absolute zero, which wouldn't be good for
> your vinyl................

Or the bird.

 

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On Sat, 4 Jun 2005 11:56:25 +0000 (UTC), Stewart Pinkerton
<patent3@dircon.co.uk> wrote:

>On Sat, 04 Jun 2005 12:04:25 +0100, Laurence Payne
><lp@laurenceNOSPAMpayne.freeserve.co.uk> wrote:
>
>>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
>
>And could only happen at absolute zero, which wouldn't be good for
>your vinyl................

Not even at absolute zero. No escaping the old zero point energy I'm
afraid - Heisenberg will not be argued with.

d

Pearce Consulting
http/www.pearce.uk.com

 

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"Richard Crowley" <richard.7.crowley@intel.com> wrote in message
news:d7lqpp$okb$1@news01.intel.com...
> Do you have some magic method of creating additional data where
> there was none before?

Any computer can do that if suitably programmed.
You can add any sort of additional data you want. It can even have something
to do with the original data if you like

>Don't even answer this message, rush down
> and patent it and become an instant millionaire.

You could probably patent a novel way of generating random numbers. Whether
you will become a millionaire is another matter.

MrT.

 

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"Stewart Pinkerton" <patent3@dircon.co.uk> wrote in message
news:7m53a1hqvjcmqec6g4ad5cpmafe3vjnti2@4ax.com...

>Only if your theory allows zero noise and infinite signal amplitude.
> >That's VERY theoretical
>
> And could only happen at absolute zero, which wouldn't be good for
> your vinyl................

Suppose thereare two birds silent, and both start singing.

What do you think is the minimum detectable time difference between the
start of the birds' songs? (That is, if the difference in time is greater
than this interval, we can determine which bird started singing first,
otherwise we are unable to determine which started singing first.)

Tim

 

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"Tim Martin" <tim2718281@ntlworld.com> wrote in message
news:iaqoe.6548$hT6.2151@newsfe3-gui.ntli.net...
>
> "Stewart Pinkerton" <patent3@dircon.co.uk> wrote in message
> news:7m53a1hqvjcmqec6g4ad5cpmafe3vjnti2@4ax.com...
>
> >Only if your theory allows zero noise and infinite signal amplitude.
>> >That's VERY theoretical
>>
>> And could only happen at absolute zero, which wouldn't be good for
>> your vinyl................
>
> Suppose thereare two birds silent, and both start singing.
>
> What do you think is the minimum detectable time difference between
> the
> start of the birds' songs? (That is, if the difference in time is
> greater
> than this interval, we can determine which bird started singing first,
> otherwise we are unable to determine which started singing first.)

Picoseconds or even femtoseconds. But likely not reliable at that
resolution in a real forrest with noise and air movement.

 

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On Sat, 04 Jun 2005 22:45:34 GMT, "Tim Martin"
<tim2718281@ntlworld.com> wrote:

>
>"Stewart Pinkerton" <patent3@dircon.co.uk> wrote in message
>news:7m53a1hqvjcmqec6g4ad5cpmafe3vjnti2@4ax.com...
>
> >Only if your theory allows zero noise and infinite signal amplitude.
>> >That's VERY theoretical
>>
>> And could only happen at absolute zero, which wouldn't be good for
>> your vinyl................
>
>Suppose thereare two birds silent, and both start singing.
>
>What do you think is the minimum detectable time difference between the
>start of the birds' songs? (That is, if the difference in time is greater
>than this interval, we can determine which bird started singing first,
>otherwise we are unable to determine which started singing first.)

What do you mean? Are these unladen birds? And using what tools?
Your ears and a stopwatch? An analog or digital stopwatch?

>Tim

-----
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