In alt.music.home-studio,rec.audio.tech,rec.audio.pro, "Phil Allison"
wrote:

** The lack of the facility to post sketches and diagrams on usenet is a
*real* drawback. When I need to explain stuff to non-technical folk ( and
some technical ones too) I often reach for my pen and paper !!!!

Draw the pics on paper, scan them in, put the files (reasonably
compressed .jpg, please) on a website, and post the URL's in your
message.

Then, on second thoughts, the sketches that might appear most often could
be kinda pornographic in nature ;-)

Use Geocities, they're slow to clean up smut on their server.




.......... Phil




-----
http://mindspring.com/~benbradley

Mark wrote:

Bob,

You seem to think the Doppler effect doesn't happen in speakers beacue
the air is moving with the speaker cone.

I think this is wrong. The Doppler effect happens anyway. The
Doppler effect depends only on the distnace between the Rx and Tx
changing.

But it does depend on the way that is changing. It does not
occur for the reasons believed and it does not occur for at
least one configuration that the vernacular theory says it will.


Doppler happens for radio and light waves as well and there is no
ether to move or not move. The Doppler effect is a function of the
changing distance between the Rx and Tx and has nothing to do with the
propogating medium.

It has all to do with it. One way of stating why it doesn't
occur in an infinite tube (or one terminated by an acoustic
resistance equal to the characteritic impedence of air) is
that in that case the radiation impedence seen by the piston
exactly equals the characteristic impedence of the air.
This simple fact guarantees perfect reproduction and no
mixing of frequencies that will produce new ones. The
vernacular theory about little HF waves being frequency
shifted by big LF ones totally fails in this case. That's
not at all what is happening.


Bob
--

"Things should be described as simply as possible, but no
simpler."

A. Einstein

Bob Cain wrote in message ...

Gary is one nasty, sociopathic SOB....


You mindless and stupid assertions regarding the existence of dynamic
Doppler distortion has been proven incorrect. Obviously the above
remark reflects a case of sour grapes. It is also a classic example
of the calling the kettle black.

Bob Cain wrote in message ...

Gary is one nasty, sociopathic SOB....


You mindless and stupid assertions regarding the existence of dynamic
Doppler distortion has been proven incorrect. Obviously the above
remark reflects a case of sour grapes. It is also a classic example
of the calling the kettle black.

Bob Cain wrote in message ...

Gary is one nasty, sociopathic SOB....


You mindless and stupid assertions regarding the existence of dynamic
Doppler distortion has been proven incorrect. Obviously the above
remark reflects a case of sour grapes. It is also a classic example
of the calling the kettle black.

Bob Cain wrote in message ...

Gary is one nasty, sociopathic SOB....


You mindless and stupid assertions regarding the existence of dynamic
Doppler distortion has been proven incorrect. Obviously the above
remark reflects a case of sour grapes. It is also a classic example
of the calling the kettle black.

Scott Dorsey wrote:

So I think folks should
continue investigating doppler distortion because it's an interesting problem
even if not a terribly important one.

Mostly agreed. The extent to which unbelievably small
effects are claimed to be audible on the ProAudio mailing
list, if they are other than imagination, does push any such
effects like we are talking about into an arena at least
worth discussing, if not important. It wouldn't take much
_at all_ to swamp the things they consider very important.


Bob
--

"Things should be described as simply as possible, but no
simpler."

A. Einstein

Bob,

What makes you think Doppler does not occur in an infinite terminated
tube.

This is a similar case to a sliding RF transmission line. A radio
wave in a transmission line that is changing length will experience
Doppler shift.
A radio wave transmitted by a perfectly matched antenna will
experience Doppler shift if the antenna is moving relative to the
receiving antenna. In fact, the receiving antenna can be the one
that is moving. If the train is standing still, and you drive past in
a car, the train whistle will still Doppler in your ear anyway.
Doppler has nothing to do with impedance matching or anything like
that.

Shift happens :-)

I think you are stuck on the idea of linearity. FM is inherently a
non-linear process in the mathematical sense. If you take a
perfectly linear VCO (in the sense that the frequency is exactly
proportional to the control voltage) , it will still create sidebands
around the carrier because FM itself is a non-linear process.

So in the speaker case, even if the cone and the air are perfectly
linear loaded and the 2 waves add perfectly in amplitude, the Doppler
that produces FM is still a non-linear process and produced new
sidebands. Non-linear does not have to imply a defect of some kind.
Perfect FM modulators are non-linear in the sense that the modulation
process creates new sidebands.

Now in a real speaker which also has amplitude non-linearities due to
changing spider tension etc etc. it is defiantly a debatable issue
as to weather or not the Doppler effect is SIGNIFICANT relative to all
the other non-linear effects going on. I have no idea.

Mark

Bob Cain writes:

Randy Yates wrote:

Bob Cain writes:

Randy Yates wrote:


So it's dropped.

Randy, if you are going to be so nasty, the least you could do is
refute what I've said. Let me get something out of it anyway.
How can I refute what you have refused to clearly restate?

All right, I'll accept that you haven't read the thread.

I've found a new way to state it anyway, that I think makes it clearer.

I'm saying that the instantaneous velocity of the piston is transfered
to the wave in the right position, since it is moving in step with it,
to propegate that velocity out as an acoustic wave. It is _in_ the
acoustic wave it is creating and it is at the right place at all times
to impart the correct velocity _because_ it is in it.

It doesn't matter what signals might have been mixed upstream to get
the signal that controls the velocity of that piston. It will be
moving in lock step with the wave defined by that signal and will
always be in the right place to deliver the right velocity to the
outgoing wave.

If, on the other hand the piston is moving with a constant velocity
superimposed on the signal velocity, it has no way to transfer that
constant velocity to the air because contant velocity doesn't create a
wave. At f=0 it runs out of punch. It ceases abruptly to transduce
at all. In that case, the piston will always be in the wrong position
to correctly impart the desired velocity signal and that error is
Doppler shift.

If the above is truly crazy, then I'll agree that I'm over the edge.
But I don't think so. What, exactly, is wrong with it?

I don't want to talk about trains and whistles.

Refutations:

a) Non-Causality: If the speaker cone and the wave are EXACTLY in
sync, then the wave must instanteously "know" which way the piston is
going. This is ludicrous.

b) Discontinuous Frequency Offsets: Doppler operating as you describe
would result in a sharp discontinuity in the perceived frequency as
the relative acceleration changed from zero to non-zero. Similarly
for the opposite case - i.e., while the acceleration is non-zero there
would be no Doppler shift but as soon as it reaches zero the Doppler
would discontinuously jump to the frequency predicted by the Doppler
equation. This is ludicrous.

c) Acoustic wave propagation in air is longitudinal and thus pressure-waves.
The velocity of the transducer is not related to the velocity of the wave
and imparts no information to the propagated signal.
--
% Randy Yates % "I met someone who looks alot like you,
%% Fuquay-Varina, NC % she does the things you do,
%%% 919-577-9882 % but she is an IBM."
%%%% % 'Yours Truly, 2095', *Time*, ELO
http://home.earthlink.net/~yatescr

Methinks The Ghost has a compulsive need to repeat everything three times.
Maybe he labors under the misapprehension that repeating a falsehood enough
will make it true.

"The Ghost" wrote in message
om...
Bob Cain wrote in message
...

Gary is one nasty, sociopathic SOB....


You mindless and stupid assertions regarding the existence of dynamic
Doppler distortion has been proven incorrect. Obviously the above
remark reflects a case of sour grapes. It is also a classic example
of the calling the kettle black.

If you want believable, try it for yourself and compare the results
across a wide range of LF and HF tones.
I suspect it's because we're dealing with digital waves, but there could be
other factors involved.

"Arny Krueger" wrote in message
news
"Porky" wrote in message

"Arny Krueger" wrote in message
...
"Porky" wrote in message


The experiment I suggested will give the results I gave, and if it
is right at under the circumstances I suggested, it should be right
under all circumstances with the same conditions, right? In other
words, if it applies with a LF of .1 Hz or 1 Hz, it will still apply
at LF's 20Hz or 50Hz, is that not correct?

right. However, its a lot harder to properly measure doppler when
the LF tone has a very low frequency. To measure it with a FFT you
must use a FFT size that covers at least one cycle, and hopefully
several cycles of the process. If the LF tone is 0.1 Hz, this means
an absolute minimum of 10 seconds of data, and ideally 30 or more.
At 44,100 Hz sampling, this would be a FFT composed of a minimum of
441,000 samples, and preferably several million samples.

Consider the original example - the LF tone was 50 Hz. It had an 882
sample
period. Note how much overkill there was when analyzed using a 65k
sample FFT, or as I used a one million point FFT.

One of the problems with FFT analysis that we've all overlooked is
that we aren't really dealing with analog waveforms in our
simulations, and we can get erroneous results when using high FFT
numbers because we start playing in the digital "cracks", so to
speak,

I'd like to see a believable fuirther explantion of that.

FFT and I are going on our 42nd year, and we've been pretty good friends
the
whole time.

Having done a bit of experimentation, I've found that I get
the most consistent results across the whole range be using an FFT
number of 16K or 32K, higher rates give false results, especially at
higher frequency HF tones.

I'd like to see a believable further explanation of that.

Alternatively, if your equipment will
handle it, try creating the wave models at 24 or 32/96, or even
32/192, you'll see a considerable difference in your results,
especially at higher HF tones and FFT numbers, and your results will
be more consistent across the entire range of LF and HF tones.

I'd also like to see a believable further explanation of that.

When dealing with analog, I would agree with you wholeheartedly, but since
any digital is only an approximation of the real thing, even with a very
close approximation there is an error factor that increases as the tone
frequency to sample frequency ratio decreases, (as the tone frequency gets
higher. Doing tests that were identical in every respect, except that one
test wave was 16/44.1 and the other test wave was 16/48, resulted in quite
different analyzer waveforms. This is what suggested to me that it might be
a digital issue.

"William Sommerwerck" wrote in message
...
One of the problems with FFT analysis that we've all overlooked is
that we aren't really dealing with analog waveforms in our
simulations, and we can get erroneous results when using high FFT
numbers because we start playing in the digital "cracks", so to
speak,

I have to agree with Arny on this. (I used to do FFT and waterfall
measurements
when I reviewed for Stereophile.) Higher sampling rates are almost always
better, other than their effect on measuring LF response. Regardless, the
higher
the rate, the _fewer_ the cracks.

Actually, using a "brick" analogy, digital may be bricks of varying widths,
but analog is one solid sheet of very smooth concrete, and even the very
best laid brickwork still has cracks.

"Arny Krueger" wrote in message
...
"William Sommerwerck" wrote in message

One of the problems with FFT analysis that we've all overlooked is
that we aren't really dealing with analog waveforms in our
simulations, and we can get erroneous results when using high FFT
numbers because we start playing in the digital "cracks", so to
speak,

I have to agree with Arny on this. (I used to do FFT and waterfall
measurements when I reviewed for Stereophile.) Higher sampling rates
are almost always better, other than their effect on measuring LF
response. Regardless, the higher the rate, the _fewer_ the cracks.

Strictly speaking there are no cracks, its just that the bricks are wider.

"Randy Yates" wrote in message
...
"Porky" writes:

"Arny Krueger" wrote in message
...
"Porky" wrote in message


The experiment I suggested will give the results I gave, and if it
is
right at under the circumstances I suggested, it should be right
under all circumstances with the same conditions, right? In other
words, if it applies with a LF of .1 Hz or 1 Hz, it will still apply
at LF's 20Hz or 50Hz, is that not correct?

right. However, its a lot harder to properly measure doppler when the
LF
tone has a very low frequency. To measure it with a FFT you must use a
FFT
size that covers at least one cycle, and hopefully several cycles of
the
process. If the LF tone is 0.1 Hz, this means an absolute minimum of
10
seconds of data, and ideally 30 or more. At 44,100 Hz sampling, this
would
be a FFT composed of a minimum of 441,000 samples, and preferably
several
million samples.

Consider the original example - the LF tone was 50 Hz. It had an 882
sample
period. Note how much overkill there was when analyzed using a 65k
sample
FFT, or as I used a one million point FFT.

One of the problems with FFT analysis that we've all overlooked is that
we
aren't really dealing with analog waveforms in our simulations, and we
can
get erroneous results when using high FFT numbers because we start
playing
in the digital "cracks", so to speak,

The waveform being analog or digital makes no difference as long as
sufficient
bandwidth and dynamic range has been supplied by the A/D conversion.
Rather, the
problem you are ignorantly referring to is that an FFT implicitly assumes
the
input is periodic. If it isn't, you can get yourself befuddled. There is
also
the problem with using the FFT to estimate the spectrum of a random
signal -
it can be shown that there will be variance in the frequency estimates no
matter how many points are used in the FFT (see, for example, "Signal
Processing:
Discrete Spectral Analysis, Detection, and Estimation," Mischa Schwartz
and
Leonard Shaw).
--

Then why would the analysis show differing results with identical waveforms
whose only difference was that one was 16/44.1 and the other was 16/48? Note
that the difference was considerable. Since all the wave simulations were
done with Cool Edit, could it be something in the way Cool Edit generates
the sine waves?

Randy Yates wrote:


Refutations:

a) Non-Causality: If the speaker cone and the wave are EXACTLY in
sync, then the wave must instanteously "know" which way the piston is
going. This is ludicrous.

No, total locality. The air is in contact with the piston.
It knows it intimately and passes the information on down
the line. In the tube, the radiation impedence seen by the
piston is identically the characteristic impedence of the
air. Consider the implications of that.

I no longer believe that Doppler distortion is non-existant,
simply that the vernacular justification, the one reiterated
by Seigfried, is totally wrong. It is much more complicated
than that simple principle. It will only exist in the far
field where the transfer function from source to reciever
varies with frequency and to the degree that it does. In
the tube that transfer function is a constant, real function
of frequency for as far as you want to go from the piston,
i.e. an acoustic resistance.

More on all that to come. It's taken me a while to figure
out what's really going on and as of today I have the
assistance of Art Ludwig who has not considered this
phenomenon before. So far, at least, we seem in agreement.
I hope with his help to arrive at the full expression of
it, at least on the axis of a piston in a baffle, that I've
been looking for. FWIW, the full expression of it is
entirely dependant on the physical configuration of the
speaker and how it is thus coupled to the air as well as the
position in space from which the phenomenon is measured.
'Taint simple ay'tall.


b) Discontinuous Frequency Offsets: Doppler operating as you describe
would result in a sharp discontinuity in the perceived frequency as
the relative acceleration changed from zero to non-zero.

Why?

Similarly
for the opposite case - i.e., while the acceleration is non-zero there
would be no Doppler shift but as soon as it reaches zero the Doppler
would discontinuously jump to the frequency predicted by the Doppler
equation. This is ludicrous.

I'm not following this at all. Where is there any
discontinuity here? The signals we have been discussing are
infinitely differentiable.


c) Acoustic wave propagation in air is longitudinal and thus pressure-waves.
The velocity of the transducer is not related to the velocity of the wave
and imparts no information to the propagated signal.

It's not related to the rate of propegation but it is very
related to the particle velocity that propegates within the
wave.

Randy, I really do appreciate the specificity of your
argument. It's been really hard in this discussion to get
ahold of anything with the generalities that have greeted
everything I've tried to pin down. The understanding I'm
arriving at would have come _much_ sooner with attack such
as yours, correct or incorrect.


Thanks,

Bob
--

"Things should be described as simply as possible, but no
simpler."

A. Einstein

"Porky" wrote in message

Actually, using a "brick" analogy, digital may be bricks of varying
widths, but analog is one solid sheet of very smooth concrete, and
even the very best laid brickwork still has cracks.

Digital does not have cracks. There are no cracks betwen samples and there
are no cracks between the lines in a FFT spectrum.

"Porky" wrote in message


When dealing with analog, I would agree with you wholeheartedly, but
since any digital is only an approximation of the real thing,

So is analog an approximation of the real thing.

"Bob Cain" wrote in message

Scott Dorsey wrote:

So I think folks should
continue investigating doppler distortion because it's an
interesting problem even if not a terribly important one.

Mostly agreed. The extent to which unbelievably small
effects are claimed to be audible on the ProAudio mailing
list, if they are other than imagination, does push any such
effects like we are talking about into an arena at least
worth discussing, if not important.

The point is well taken. Gosh, I even brought it up about a week ago and our
resident opamp wine tasters completely missed it.

Go figure!

It wouldn't take much
_at all_ to swamp the things they consider very important.

Speaker Doppler as insignificant as it is, is positively huge compared to
the errors that a common nasty old 5532 or TL072 makes in most audio
circuits.

Mark wrote:

Bob,

What makes you think Doppler does not occur in an infinite terminated
tube.

That the radiation impedence seen by the piston is
identically that of the characteristic impedence of the gas.
Consider the implcations of that.


This is a similar case to a sliding RF transmission line. A radio
wave in a transmission line that is changing length will experience
Doppler shift.

Well, yes if the reciever is in motion, other things begin
to happen. I'm only, at this point, considering the
situation where the transmitter and reciever are at a
constant distance from each other. If the vernacular
argument for Doppler distortion predicts the phenomenon for
that case then it is incorrect, and it does.

If you read my recent post to Randy you'll see that my
thinking has changed signifigantly about Doppler distortion
but not at all about the vernacular argument that started me
questioning it. Based on the death of the "HF waves riding
on LF ones" argument, which can be disproved by analysis of
a piston in a tube, I threw out the whole idea of Doppler
distortion but there is a mechanism that is a whole lot more
complex than that which is so widely reprinted.

I think you are stuck on the idea of linearity.

Not any more. Something is clearly wrong with my belief
that a system is linear if all complex exponentials are
passed changed only by a complex multiplier. I'm certain
that is an "only if" condition. I started to think about
that last night but haven't come back to it. My provisional
thought on it is that the accepted eigenfunction/eigenvalue
definition applies to linear _time invariant_ systems and
that we are somehow looking at a time varying system here
(in terms of impedences.) Not sure that is right or not.
Sorta shooting in the dark at this one so far. Anybody?
What is it that allows a system to look linear when
presented with sinusoids but not if presented with more
complex signals? Please don't say "Doppler distortion".
There has to be a more general systems principle involved.

So in the speaker case, even if the cone and the air are perfectly
linear loaded and the 2 waves add perfectly in amplitude, the Doppler
that produces FM is still a non-linear process and produced new
sidebands. Non-linear does not have to imply a defect of some kind.
Perfect FM modulators are non-linear in the sense that the modulation
process creates new sidebands.

I agree, it is a mathematically defined constraint, not a
defect.

Now in a real speaker which also has amplitude non-linearities due to
changing spider tension etc etc. it is defiantly a debatable issue
as to weather or not the Doppler effect is SIGNIFICANT relative to all
the other non-linear effects going on. I have no idea.

Me either yet.

I hope it is becoming clearer to anyone still with this
discussion that I could give a rat's ass about being "right"
on this, I just want to emerge correct. My ego isn't on the
line, thank you very much, my understanding is.

From a lot of the personal crap that has been slung around
it is obvious that a lot of people don't understand the
nature of scientific debate. This is it, folks.

Not that there isn't a time and place for slinging crap in
almost any scientific debate. :-)


Thanks,

Bob
--

"Things should be described as simply as possible, but no
simpler."

A. Einstein

Porky wrote:

If you want believable, try it for yourself and compare the results
across a wide range of LF and HF tones.
I suspect it's because we're dealing with digital waves, but there could be
other factors involved.

I suspect you are wrong on this one, Mike.


Bob
--

"Things should be described as simply as possible, but no
simpler."

A. Einstein

"Bob Cain" wrote in message
...


Mark Simonetti wrote:

Of course it will. And its instanteious velocity is added to the
instantaneous velocity due to the upper note.


Okay, but it seems like people are saying that this isn't the case, and
is the reason doppler distortion would NOT occur, which seems plain
bizzare, IMHO.

In some circumstances it does and in others it doesn't but
that isn't the reason, at any rate, for either result.


My position is that the speaker's motion does NOT directly produce the
sound, so there is no tone riding on any other tone coming from the speaker.
The complex motion of the speaker cone corresponds exactly with the
compression and rarefaction of the air in front of it, and any instantaneous
position of the cone is directly related to the corresponding level of
compression or rarefaction of the complex acoustic wave. Adding a DC
component doesn't matter, because the excursion levels will still be exactly
the same as long as the linear excursion factor of the speaker is not
exceeded, the center position will just be moved a bit. Slowly varying DC
doesn't affect it either, since the variation becomes part of the acoustic
wave being produced, whether we hear it or not. The flaw there was assuming
that because we couldn't hear it, wasn't a sound wave. The only thing that
can cause Doppler shift in a speaker is physically moving the speaker
relative to the listener with some force other than that producing the
complex wave driving the speaker motor.
Here's another way of looking at it, putting a sound wave through a hole
in the wall can't produce Doppler shift, no matter how many tones are in the
waveform, and a speaker is effectively an artificial hole in the wall, in
that the effective sound source isn't the speaker any more than it is the
hole in the wall. Does anyone here think that if you stretched a thin
diaphragm over a hole in a soundproof wall and had a band playing behind it,
the diaphragm would cause Doppler Distortion? The speaker, provided it isn't
exceeding its linear limits, is effectively exactly the same thing for all
practical purposes. Instead of being driven by the sound source in the other
room, it's driven by the electrical equivalent of the sound source in the
other room. Can any of you provide an explanation of how an acoustic wave
driving a diaphragm and passing the soundwave
through it is in any way different than the diaphragm being driven by a
motor being supplied with the exact electrical analog of that acoustic wave?
And by that I mean that the difference will be such that the electrically
driven one will produce Doppler distortion while the acoustically driven one
doesn't.

"Randy Yates" wrote in message
...
Bob Cain writes:

Randy Yates wrote:

Bob Cain writes:

Randy Yates wrote:


So it's dropped.

Randy, if you are going to be so nasty, the least you could do is
refute what I've said. Let me get something out of it anyway.
How can I refute what you have refused to clearly restate?

All right, I'll accept that you haven't read the thread.

I've found a new way to state it anyway, that I think makes it clearer.

I'm saying that the instantaneous velocity of the piston is transfered
to the wave in the right position, since it is moving in step with it,
to propegate that velocity out as an acoustic wave. It is _in_ the
acoustic wave it is creating and it is at the right place at all times
to impart the correct velocity _because_ it is in it.

It doesn't matter what signals might have been mixed upstream to get
the signal that controls the velocity of that piston. It will be
moving in lock step with the wave defined by that signal and will
always be in the right place to deliver the right velocity to the
outgoing wave.

If, on the other hand the piston is moving with a constant velocity
superimposed on the signal velocity, it has no way to transfer that
constant velocity to the air because contant velocity doesn't create a
wave. At f=0 it runs out of punch. It ceases abruptly to transduce
at all. In that case, the piston will always be in the wrong position
to correctly impart the desired velocity signal and that error is
Doppler shift.

If the above is truly crazy, then I'll agree that I'm over the edge.
But I don't think so. What, exactly, is wrong with it?

I don't want to talk about trains and whistles.

Refutations:

a) Non-Causality: If the speaker cone and the wave are EXACTLY in
sync, then the wave must instanteously "know" which way the piston is
going. This is ludicrous.

b) Discontinuous Frequency Offsets: Doppler operating as you describe
would result in a sharp discontinuity in the perceived frequency as
the relative acceleration changed from zero to non-zero. Similarly
for the opposite case - i.e., while the acceleration is non-zero there
would be no Doppler shift but as soon as it reaches zero the Doppler
would discontinuously jump to the frequency predicted by the Doppler
equation. This is ludicrous.

c) Acoustic wave propagation in air is longitudinal and thus
pressure-waves.
The velocity of the transducer is not related to the velocity of the wave
and imparts no information to the propagated signal.

Gee, couldn't one to construe (c) to mean that Doppler distortion doesn't
exist in a speaker, since it is a transducer?

"Arny Krueger" wrote in message
...
"Porky" wrote in message


When dealing with analog, I would agree with you wholeheartedly, but
since any digital is only an approximation of the real thing,

So is analog an approximation of the real thing.

Nope, under perfect condidtions, analog is an exact duplication of the
real thing, and that's something that cannot be said of digital, not matter
how high the digital resolution, or how perfect the conditions, unless you
can show me how if is possible to produce a digital signal with infinite bit
width and infinitely high sampling rate. Wait, that would BE analog!!!:-)

"Bob Cain" wrote in message
...


Porky wrote:

If you want believable, try it for yourself and compare the results
across a wide range of LF and HF tones.
I suspect it's because we're dealing with digital waves, but there
could be
other factors involved.

I suspect you are wrong on this one, Mike.


I could very well be wrong, I'm basically just speculating on possible
causes for the variations I see with different sampling frequencies.:-)

Bob Cain wrote:

Randy, I really do appreciate the specificity of your argument. It's
been really hard in this discussion to get ahold of anything with the
generalities that have greeted everything I've tried to pin down. The
understanding I'm arriving at would have come _much_ sooner with attack
such as yours, correct or incorrect.

It sounds like I'm saying that no one else ever did that and
that's not what I meant at all. There was just an awful lot
to sort through to get to the stuff that made me revise my
thinking. I was just especially appreciative at the moment
I wrote that. Too tired to be writing.


Bob
--

"Things should be described as simply as possible, but no
simpler."

A. Einstein

Porky wrote:

Then why would the analysis show differing results with identical waveforms
whose only difference was that one was 16/44.1 and the other was 16/48?

Damn good question. I dunno. Sure seems worth finding an
answer to though.

Note
that the difference was considerable. Since all the wave simulations were
done with Cool Edit, could it be something in the way Cool Edit generates
the sine waves?

Ya, know. At one point when I was playing with a possible
model of the vernacular argument in Matlab I did get a
nearly perfectly flat top. Not keeping notes, however, and
not curious enough at that point because I was looking for
something unrelated, I can't remember what expression I
used. Dammit.


Bob
--

"Things should be described as simply as possible, but no
simpler."

A. Einstein

Arny Krueger wrote:

Speaker Doppler as insignificant as it is, is positively huge compared to
the errors that a common nasty old 5532 or TL072 makes in most audio
circuits.

What's that say about the known and much larger non-linear
effects in speakers that we all know and love? I think the
guys that claim to hear these miniscule phase and dither
differences played back through the best of the available
speakers are blowing smoke...

They argue about what they hear as a function of a couple of
degrees of phase shift at Nyquist and the kind of dither
applied to get to 24 bits!

I notice that the one thing they are too totally polite to
ever do is question each other's "golden" ears despite the
differences among themselves in what they hear. Feet of
clay all around, perhaps.

It was my annoyance and disbelief in all of this that
motivated me to look hard at Doppler distortion and find a
way to quantify it. The rest, as they say, is history. :-)

Actually, I am quite happy to find that it exists, if not
for the usual reasons given, because of this original
motivation.


Bob
--

"Things should be described as simply as possible, but no
simpler."

A. Einstein

"Porky" wrote in message

"Arny Krueger" wrote in message
...
"Porky" wrote in message


When dealing with analog, I would agree with you wholeheartedly, but
since any digital is only an approximation of the real thing,

So is analog an approximation of the real thing.

Nope, under perfect condidtions, analog is an exact duplication of
the real thing,

Irrelevant since it has never been observed, nor can it ever be observed in
the real world.

and that's something that cannot be said of digital,
not matter how high the digital resolution, or how perfect the
conditions, unless you can show me how if is possible to produce a
digital signal with infinite bit width and infinitely high sampling
rate.

All real world analog signals have finite dynamic range and frequency
response. Dynamic range in the analog domain is practically limited by
thermal noise and the size of the largest signal that can be made with a
good noise figure.

In contrast, the dynamic range of a digital signal is limited only by our
ability to implement and manipulate large numbers of bits. This just about
unlimited.

Wait, that would BE analog!!!:-)

Wrong. In audio we know that flat frequency reponse over the audible range
is more valuable and useful than rough or skewed frequency response from DC
to light. We also know that good frequency response in the audible range is
very hard to preserve in the analog domain as the processing, recording, and
transmission steps build up. In contrast digital signals by default have
perfectly flat response over any reasonble defined range of frequencies,
right up into the microwave region, and down in to the frequency realm of
siesmic events.

"Bob Cain" wrote in message

Arny Krueger wrote:

Speaker Doppler as insignificant as it is, is positively huge
compared to the errors that a common nasty old 5532 or TL072 makes
in most audio circuits.

What's that say about the known and much larger non-linear
effects in speakers that we all know and love?

They are there, and distract our ears from the itty-bitty ones.

I think the guys that claim to hear these miniscule phase and dither
differences played back through the best of the available
speakers are blowing smoke...

The rubber hits the road in a blind, level-matched, time-synched,
bias-controlled listening test. They routinely lose out.

They argue about what they hear as a function of a couple of
degrees of phase shift at Nyquist and the kind of dither
applied to get to 24 bits!

Obviously they don't have a lot of good listening experiments under their
belt. Cut your chops on good listening tests and you start singing a
different song.

I notice that the one thing they are too totally polite to
ever do is question each other's "golden" ears despite the
differences among themselves in what they hear. Feet of
clay all around, perhaps.

As long as they stay away from ABX and the like, their belief structures are
preserved.

It was my annoyance and disbelief in all of this that
motivated me to look hard at Doppler distortion and find a
way to quantify it. The rest, as they say, is history. :-)

It's there!

Actually, I am quite happy to find that it exists, if not
for the usual reasons given, because of this original
motivation.

The point is that AM distortion, which dominates and is relatively large in
speakers, and often pretty audible, is only present in good electronics in
far smaller quantities. Masking rules, we hear distortion in speakers far
more so than in non-clipping, non-noisy electronics.

Arny Krueger wrote:
Speaker Doppler as insignificant as it is, is positively huge compared to
the errors that a common nasty old 5532 or TL072 makes in most audio
circuits.

Please excuse my ignorance, but what is a 5532 and a TL072 ? What
problems are they causing ? I know I could look it up (and I just have)
but others may be curious too. I find all this interesting.

You'd be surprised (or perhaps not) how many people really aren't
bothered about hearing nice clear audio through a decent system, never
mind the (small?) distortions we are discussing in this NG. Sometimes I
think its because people don't realise what they are missing. Even with
my modest set up, I've bought a smile to the face of some people who've
sat and listened to my relatively good system, compared to your average
low end extremely coloured systems (complete with "3D" sound
"enhancement" and all that)..

I think others who refer to us, who do care, as anal about it perhaps do
not appreciate music in the same way. To me there is nothing better
than hearing the subtle beautiful tones of a well recorded well played
piano or acoustic guitar; these are just plain invisible or distorted on
a lot of todays cheap crap, and people are getting used to it.

Disclaimer: IMHO, YMMV, etc.

Cheers.

--
Mark Simonetti.
Freelance Software Engineer.

Mark Simonetti wrote:
I think others who refer to us, who do care, as anal about it perhaps do
not appreciate music in the same way. To me there is nothing better
than hearing the subtle beautiful tones of a well recorded well played
piano or acoustic guitar; these are just plain invisible or distorted on
a lot of todays cheap crap, and people are getting used to it.

Disclaimer: IMHO, YMMV, etc.

Cheers.


Or maybe I'm just having a bad day and being cynical ;-)

--
Mark Simonetti.
Freelance Software Engineer.

"Mark Simonetti" wrote in message

Arny Krueger wrote:

Speaker Doppler as insignificant as it is, is positively huge
compared to the errors that a common nasty old 5532 or TL072 makes
in most audio circuits.

Please excuse my ignorance, but what is a 5532 and a TL072 ?

Common, err mature audio op amp chips, still widely used particularly the
former.

What problems are they causing ?

In correctly-designed circuits, that would be a good quesiton.

I know I could look it up (and I just
have) but others may be curious too. I find all this interesting.

You'd be surprised (or perhaps not) how many people really aren't
bothered about hearing nice clear audio through a decent system, never
mind the (small?) distortions we are discussing in this NG.

I'm sorry, but these sentence's structure and length makes me unclear about
what it means.

Sometimes I think its because people don't realise what they are
missing.

Are you suggesting that I'm missing something, or making these comments in a
state of total ignorance?

Even with my modest set up, I've bought a smile to the face
of some people who've sat and listened to my relatively good system,
compared to your average low end extremely coloured systems (complete
with "3D" sound "enhancement" and all that)..

That's not the scale of what we are talking about here. We're about a dozen
cuts above at the baseline.

I think others who refer to us, who do care, as anal about it perhaps
do not appreciate music in the same way.

I don't think you know who you are talking to, but frankly you should
reserve such comments for your dogs, cats and other house pets.

To me there is nothing
better than hearing the subtle beautiful tones of a well recorded
well played piano or acoustic guitar; these are just plain invisible
or distorted on a lot of todays cheap crap, and people are getting
used to it.

Why do you presume that somehow this makes you different from me?

Disclaimer: IMHO, YMMV, etc.

A wise man checks out the lay of the land before he rushes spouting all
sorts of self-serving insults.

"Arny Krueger" wrote in message
...
"Porky" wrote in message

"Arny Krueger" wrote in message
...
"Porky" wrote in message


When dealing with analog, I would agree with you wholeheartedly, but
since any digital is only an approximation of the real thing,

So is analog an approximation of the real thing.

Nope, under perfect condidtions, analog is an exact duplication of
the real thing,

Irrelevant since it has never been observed, nor can it ever be observed
in
the real world.

and that's something that cannot be said of digital,
not matter how high the digital resolution, or how perfect the
conditions, unless you can show me how if is possible to produce a
digital signal with infinite bit width and infinitely high sampling
rate.

All real world analog signals have finite dynamic range and frequency
response. Dynamic range in the analog domain is practically limited by
thermal noise and the size of the largest signal that can be made with a
good noise figure.

In contrast, the dynamic range of a digital signal is limited only by our
ability to implement and manipulate large numbers of bits. This just about
unlimited.

Wait, that would BE analog!!!:-)

Wrong. In audio we know that flat frequency reponse over the audible range
is more valuable and useful than rough or skewed frequency response from
DC
to light. We also know that good frequency response in the audible range
is
very hard to preserve in the analog domain as the processing, recording,
and
transmission steps build up. In contrast digital signals by default have
perfectly flat response over any reasonble defined range of frequencies,
right up into the microwave region, and down in to the frequency realm of
siesmic events.

Well, assuming that we can't reach infinity in digital bit depth, perfect
analog is one bit depth better than perfect digital :-)

I think perhaps I didn't write that very well, you've midunderstood me,
sorry.

I'm basically stating that in general people, as in the general public,
do not appreciate quality audio, which is a sad state of affairs.

Mark.
--

Arny Krueger wrote:
"Mark Simonetti" wrote in message


Arny Krueger wrote:


Speaker Doppler as insignificant as it is, is positively huge
compared to the errors that a common nasty old 5532 or TL072 makes
in most audio circuits.


Please excuse my ignorance, but what is a 5532 and a TL072 ?


Common, err mature audio op amp chips, still widely used particularly the
former.


What problems are they causing ?


In correctly-designed circuits, that would be a good quesiton.


I know I could look it up (and I just
have) but others may be curious too. I find all this interesting.


You'd be surprised (or perhaps not) how many people really aren't
bothered about hearing nice clear audio through a decent system, never
mind the (small?) distortions we are discussing in this NG.


I'm sorry, but these sentence's structure and length makes me unclear about
what it means.


Sometimes I think its because people don't realise what they are
missing.


Are you suggesting that I'm missing something, or making these comments in a
state of total ignorance?


Even with my modest set up, I've bought a smile to the face
of some people who've sat and listened to my relatively good system,
compared to your average low end extremely coloured systems (complete
with "3D" sound "enhancement" and all that)..


That's not the scale of what we are talking about here. We're about a dozen
cuts above at the baseline.


I think others who refer to us, who do care, as anal about it perhaps
do not appreciate music in the same way.


I don't think you know who you are talking to, but frankly you should
reserve such comments for your dogs, cats and other house pets.


To me there is nothing
better than hearing the subtle beautiful tones of a well recorded
well played piano or acoustic guitar; these are just plain invisible
or distorted on a lot of todays cheap crap, and people are getting
used to it.


Why do you presume that somehow this makes you different from me?


Disclaimer: IMHO, YMMV, etc.


A wise man checks out the lay of the land before he rushes spouting all
sorts of self-serving insults.




--
Mark Simonetti.
Freelance Software Engineer.

"Mark Simonetti" wrote in message


I think perhaps I didn't write that very well, you've midunderstood
me, sorry.

You did more than express yourself poorly, you cast aspersons on other
people.

I'm basically stating that in general people, as in the general
public, do not appreciate quality audio, which is a sad state of
affairs.

Why state the obvious in this context?

I think perhaps I didn't write that very well, you've midunderstood
me, sorry.

You did more than express yourself poorly, you cast aspersons
on other people.

No, Mark, your point of view (though perhaps not geramaine to the discussion)
was perfectly clear. Arny deliberately "misunderstood" you. He does this all the
time. It's his snide way of attacking people he disagrees with, rather than
directly confronting the issue. (He has a high IQ, but little insight.) You're
not the only person who's been on the receiving end.

"Porky" writes:

"Randy Yates" wrote in message
...
"Porky" writes:

"Arny Krueger" wrote in message
...
"Porky" wrote in message


The experiment I suggested will give the results I gave, and if it
is
right at under the circumstances I suggested, it should be right
under all circumstances with the same conditions, right? In other
words, if it applies with a LF of .1 Hz or 1 Hz, it will still apply
at LF's 20Hz or 50Hz, is that not correct?

right. However, its a lot harder to properly measure doppler when the
LF
tone has a very low frequency. To measure it with a FFT you must use a
FFT
size that covers at least one cycle, and hopefully several cycles of
the
process. If the LF tone is 0.1 Hz, this means an absolute minimum of
10
seconds of data, and ideally 30 or more. At 44,100 Hz sampling, this
would
be a FFT composed of a minimum of 441,000 samples, and preferably
several
million samples.

Consider the original example - the LF tone was 50 Hz. It had an 882
sample
period. Note how much overkill there was when analyzed using a 65k
sample
FFT, or as I used a one million point FFT.

One of the problems with FFT analysis that we've all overlooked is that
we
aren't really dealing with analog waveforms in our simulations, and we
can
get erroneous results when using high FFT numbers because we start
playing
in the digital "cracks", so to speak,

The waveform being analog or digital makes no difference as long as
sufficient
bandwidth and dynamic range has been supplied by the A/D conversion.
Rather, the
problem you are ignorantly referring to is that an FFT implicitly assumes
the
input is periodic. If it isn't, you can get yourself befuddled. There is
also
the problem with using the FFT to estimate the spectrum of a random
signal -
it can be shown that there will be variance in the frequency estimates no
matter how many points are used in the FFT (see, for example, "Signal
Processing:
Discrete Spectral Analysis, Detection, and Estimation," Mischa Schwartz
and
Leonard Shaw).
--

Then why would the analysis show differing results with identical waveforms
whose only difference was that one was 16/44.1 and the other was 16/48? Note
that the difference was considerable.

Hi Porky,

Yes, I believe you. The reason is because, for the same number of
input samples to the FFT (let's just use an example length of 1024,
i.e., we're doing a 1024-point FFT), a different sample rate
corresponds to a different period of time, and those 1024 points are
considered periodic with period 1024*Ts (Ts = 1/Fs, Fs the sample
rate).

Here's a simple example. Let's say your sample rate is 1024 samples/second,
and you're generating a 1 Hz sine wave. Then a 1024-point FFT will gather
exactly one cycle of your 1 Hz sine wave and you will see a single, beautiful
peak in bin 1 (counting starting with 0) of the FFT. That is because the
1024-point FFT implicitly assumes those 1024 samples are one period of
a periodic waveform with period of 1024 * 1/Fs = 1024 * 1/1024 = 1 second.

OK, now change the sample rate to 2048 samples/second. Now, 1024
samples corresponds to 1/2 cycle of your 1 kHz sine wave. When you
perform the 1024-point FFT you will see energy all over the
spectrum. Again, that is because the 1024-point FFT implicitly assumes
those 1024 samples are one period of a periodic waveform with period
of 1024 * 1/Fs = 1024 * 1/2048 = 0.5 second. You're thus seeing the
spectrum of a half-sine (a rectified sine wave) at 1 Hz.

Since all the wave simulations were
done with Cool Edit, could it be something in the way Cool Edit generates
the sine waves?

As outlined above, it's not Cool Edit but rather the fundamental nature of
the FFT.

As a final note, we could replace "FFT" with "DFT" in all the above and
the statements would still be true. The FFT is just an efficient way
to compute the DFT.
--
Randy Yates
Sony Ericsson Mobile Communications
Research Triangle Park, NC, USA
, 919-472-1124

"Mark Simonetti" wrote in message
...
I think perhaps I didn't write that very well, you've midunderstood me,
sorry.

I'm basically stating that in general people, as in the general public,
do not appreciate quality audio, which is a sad state of affairs.

I agree wholeheartedly with that statement, however, what the traind and
experienced discerning listener, and what the self-proclaimed "golden-eared
audiophile" claim to hear are often totally different things. The difference
between the two is that what the trained and experinced discerning listener
claims to hear can be verified by scientific double-blind tests, and what
the self-proclaimed "golden-eared audiophile" claims to hear often cannot.
That isn't to say that all audiophiles are bogus, that isn't true, I know of
audiophiles who are exceptionally accurate and discerning listeners. However
the "audiophile" who buys $400 a foot speaker cable and claims to hear "a
big difference" over good quality regular speaker cable, or the "audiophile"
who claims to hear a phase difference of 2 degrees at far field in a live
room, is full of it!:-)

Randy Yates writes:
[...]
OK, now change the sample rate to 2048 samples/second. Now, 1024
samples corresponds to 1/2 cycle of your 1 kHz sine wave.

Correction: 1 Hz sine wave, not 1 kHz.
--
Randy Yates
Sony Ericsson Mobile Communications
Research Triangle Park, NC, USA
, 919-472-1124

Mark Simonetti wrote:
I'm basically stating that in general people, as in the general
public, do not appreciate quality audio, which is a sad state of
affairs.

Mark.
--

Further;

I don't mean through any fault of there own, I meant because of the
mainstream audio systems that are out there that are often sub-standard,
and people get used to how that sounds.

Often I've found people aren't actually too worried about this, until
they sit down and really listen closely to a high quality system, and
then realise what they are missing.

The distortions you are talking about in this thread are comparitively
small, please correct me if I'm wrong. That being the case, I was
basically stating that you often find people are not even bothered about
the larger distortions from sub standard systems, and that I felt it is
unfortunate. This is often the case because they've become so used to
hearing car stereos and very cheap sub-standard music centers and do not
know any different.

It was certainly not a personal attack against you or anyone.
Appologies for not being clear.

Cheers,

Mark.
--