> "Jerry Avins" <jya@ieee.org> wrote in message
> news:4152d4da$0$4027$61fed72c@news.rcn.com...
>
>>Fred Marshall wrote:
>>
>> ...
>>
>>
>>>By "increase the rotational sample rate", I mean that you generate the
>>>sampling clock with something that looks like a frequency synthesizer.
>>>Inside the synthesizer is a phase-locked loop. Instead of the input
>>>being a "time stable", lower frequency, oscillator, the input is the tach
>>>and the synthesizer generates higher frequencies that are locked to the
>>>tach. By setting the synthesizer parameters you can change the output
>>>clock rate and generate higher rate clocks.
>>
>>Beware of clocking a sampler with the output a PLL. Residual FM --
>>usually a sawtooth -- can inject artifacts. When the PLL follows a
>>varying input, the effect is worse. A tach with more lines is wanted.
>>
>>If the tach signal is derived from a two-track bidirectional encoder, a
>>simple circuit will give 4 pulses per line. If the analog signals
>>(before the Schmitt triggers) is available, you can double that to 8.
>>An 8X increase in sampling resolution may be enough.
>>
>
> Jerry,
>
> Yeah, I thought about that - but not as much as you have. In a perfect
> (i.e. very high bandwidth) PLL implementation the frequency output might
> "jump" to a new value each time a new axis crossing of the input occurred.
> Thus the sawtooth error. That obviously doesn't follow a smooth ramp in
> rotational velocity. But, if the rotational acceleration is low enough it
> wouldn't be noticed.
With an analog PLL, its actual frequency drifts toward its free-running
frequency between corrections, unless there's a perfect integrator in
it. Digital integrators are perfect, but that doesn't help if the actual
speed is changing. The PLL frequency can only be corrected in jumps when
a correction pulse comes along, so it becomes a staircase, or a sawtooth
riding on a ramp depending on how you choose to look at it.
> ("low enough" and "noticed" are subjective terms - have to be qualified for
> the specific situation)
>
>
>>Beware of clocking a sampler with the output a PLL. Residual FM --
... The output *of* a PLL....
>>usually a sawtooth -- can inject artifacts. When the PLL follows a
>>varying input, the effect is worse. A tach with more lines is wanted.
>>
>
>
> Do you mean a one pulse per revolution tach? Otherwise I don't know what
> you mean by "clocking a sampler with the output a PLL". And, I'm curious or
> I wouldn't mention it.
I don't understand your puzzlement.
> I would think that the number of lines is almost *always* one of the things
> to be dealt with. I did a dc motor tape drive controller with a 1000 line
> tach and PLL for low wow and flutter - that was pretty cool. With 1000
> lines and constant speed it was easy. Here the OP has variable speed - so
> it takes more thought as you've pointed out.
If your 1000 lines are distributed among two tracks in quadrature -- the
standard incremental bi-directional encoder layout -- there are 2000
distinguishable locations around the disk. If those tracks have close to
equal areas of clear and dark, and if the phase shift id close to
quadrature throughout, then each location will occupy the same angle as
all the others. The actual track sensor outputs will be approximately
sinusoidal, and by running the outputs of both tracks into a single
differential comparator (and again with suitable inversions), a new set
of outputs can be generated 45 degrees displaced from the first set. In
all, 4000 locations (or clock transitions if you use them that way) are
stably generated from 1000 lines. Quadrature encoders are usually
characterized by the number of lines per track, so we would say with my
example that we generates 4000 locations with a 500-line encoder. In my
collection, I have the 10,000-line BEI encoder that I used (without the
last doubling) to provide accurate position information in a machine to
locate and repair defects on RCA's video disk masters. The amazing part
was not distinguishing 1/40,000th of a circle, but getting the servo to
settle onto it.
Jerry
--
... they proceeded on the sound principle that the magnitude of a lie
always contains a certain factor of credibility, ... and that therefor
... they more easily fall victim to a big lie than to a little one ...
A. H.
�����������������������������������������������������������������������
Reply by Fred Marshall●September 23, 20042004-09-23
"Jerry Avins" <jya@ieee.org> wrote in message
news:4152d4da$0$4027$61fed72c@news.rcn.com...
> Fred Marshall wrote:
>
> ...
>
>> By "increase the rotational sample rate", I mean that you generate the
>> sampling clock with something that looks like a frequency synthesizer.
>> Inside the synthesizer is a phase-locked loop. Instead of the input
>> being a "time stable", lower frequency, oscillator, the input is the tach
>> and the synthesizer generates higher frequencies that are locked to the
>> tach. By setting the synthesizer parameters you can change the output
>> clock rate and generate higher rate clocks.
>
> Beware of clocking a sampler with the output a PLL. Residual FM --
> usually a sawtooth -- can inject artifacts. When the PLL follows a
> varying input, the effect is worse. A tach with more lines is wanted.
>
> If the tach signal is derived from a two-track bidirectional encoder, a
> simple circuit will give 4 pulses per line. If the analog signals
> (before the Schmitt triggers) is available, you can double that to 8.
> An 8X increase in sampling resolution may be enough.
>
Jerry,
Yeah, I thought about that - but not as much as you have. In a perfect
(i.e. very high bandwidth) PLL implementation the frequency output might
"jump" to a new value each time a new axis crossing of the input occurred.
Thus the sawtooth error. That obviously doesn't follow a smooth ramp in
rotational velocity. But, if the rotational acceleration is low enough it
wouldn't be noticed.
("low enough" and "noticed" are subjective terms - have to be qualified for
the specific situation)
> Beware of clocking a sampler with the output a PLL. Residual FM --
> usually a sawtooth -- can inject artifacts. When the PLL follows a
> varying input, the effect is worse. A tach with more lines is wanted.
>
Do you mean a one pulse per revolution tach? Otherwise I don't know what
you mean by "clocking a sampler with the output a PLL". And, I'm curious or
I wouldn't mention it.
I would think that the number of lines is almost *always* one of the things
to be dealt with. I did a dc motor tape drive controller with a 1000 line
tach and PLL for low wow and flutter - that was pretty cool. With 1000
lines and constant speed it was easy. Here the OP has variable speed - so
it takes more thought as you've pointed out.
Fred
Reply by Jerry Avins●September 23, 20042004-09-23
Fred Marshall wrote:
...
> By "increase the rotational sample rate", I mean that you generate the
> sampling clock with something that looks like a frequency synthesizer.
> Inside the synthesizer is a phase-locked loop. Instead of the input being a
> "time stable", lower frequency, oscillator, the input is the tach and the
> synthesizer generates higher frequencies that are locked to the tach. By
> setting the synthesizer parameters you can change the output clock rate and
> generate higher rate clocks.
Beware of clocking a sampler with the output a PLL. Residual FM --
usually a sawtooth -- can inject artifacts. When the PLL follows a
varying input, the effect is worse. A tach with more lines is wanted.
If the tach signal is derived from a two-track bidirectional encoder, a
simple circuit will give 4 pulses per line. If the analog signals
(before the Schmitt triggers) is available, you can double that to 8.
An 8X increase in sampling resolution may be enough.
Jerry
--
... they proceeded on the sound principle that the magnitude of a lie
always contains a certain factor of credibility, ... and that therefor
... they more easily fall victim to a big lie than to a little one ...
A. H.
�����������������������������������������������������������������������
Reply by Jerry Avins●September 23, 20042004-09-23
glen herrmannsfeldt wrote:
> Fred Marshall wrote:
> (snip)
>
>> Understood - if the clock rate varies then the lowest sample rate sets
>> a lower Nyquist frequency. One wouldn't ignore noise in this situation.
>
>
>
> For non-uniform sampling, the theory tends toward the total number
> of samples over the total time. If you get too non-uniform
> there can be problems with noise, though.
>
> -- glen
The sampling is strictly uniform in distance. Time needn't be
considered.
Jerry
--
... they proceeded on the sound principle that the magnitude of a lie
always contains a certain factor of credibility, ... and that therefor
... they more easily fall victim to a big lie than to a little one ...
A. H.
�����������������������������������������������������������������������
Reply by glen herrmannsfeldt●September 23, 20042004-09-23
Fred Marshall wrote:
(snip)
> Understood - if the clock rate varies then the lowest sample rate sets a
> lower Nyquist frequency. One wouldn't ignore noise in this situation.
For non-uniform sampling, the theory tends toward the total number
of samples over the total time. If you get too non-uniform
there can be problems with noise, though.
-- glen
Reply by Bernhard Holzmayer●September 23, 20042004-09-23
Fred Marshall wrote:
>
> Understood - if the clock rate varies then the lowest sample rate
> sets a
> lower Nyquist frequency. One wouldn't ignore noise in this
> situation. There are a number of observations I might make:
>
> 1) If you clock on the tachometer and the speed is varying, then
> the normal single-temporal frequency, sinusoidal additive noise
> may be spectrally spread by synching the sampling clock to the
> tach. NOTE: this means time-based frequency components - which
> might look like "noise" in the rotational domain.
Varying speed results in an effect similar to that of a
switched/slided filter at constant sample rate.
We're aware of it - but since we cannot remove the speed variation,
we'll have to live with that...
>
> 2) If you transform the thinking to the rotational position domain
> then the Nyquist criterion still applies - but now in radians per
> revolution instead
> of radians per second. You can have a sinusoid at 20k periods per
> revolution components that have to be sampled at higher than 40k
> samples per revolution.
Desirable. We achieve that for the spectrum of interest, but not for
the noise @ 3)
>
> 3) If you sample noise that's at a rotational frequency higher
> than 20k periods per revolution then you'll alias it - just as if
> you were using a
> time base. Sometimes it's acceptable to alias white noise and
> sometimes
> not. The problem is the same.
>
> So, in order to "properly" sample the data that's clocked with the
> tach you
> need to have an idea of the spectral character of the result. I
> guess one way to measure that character is to increase the
> rotational sample rate and FFT reasonable length sequences and
> find where the high frequency energy in
> the result goes away. Then the sample rate is probably high
> enough.
>
> By "increase the rotational sample rate", I mean that you generate
> the sampling clock with something that looks like a frequency
> synthesizer.
> Inside the synthesizer is a phase-locked loop. Instead of the
> input being a "time stable", lower frequency, oscillator, the
> input is the tach and the
> synthesizer generates higher frequencies that are locked to the
> tach. By setting the synthesizer parameters you can change the
> output clock rate and generate higher rate clocks.
I agree. That would help.
But at the stage before the sampling (or rate conversion), that
stage must allow the whole spectrum of interest, even at full
speed. Which means, that I have a static LPF at 20k periods per
revolution. Now when speed decreases, the Nyquist rate sinks below
that 20k, and a gap occurs, which is not covered by the LPF.
>
> Alternately or in combination, you might lowpass filter the
> (analog) sensor outputs to achieve the same results.
It's the same if we do this in analog or digital realm.
>
> We are stuck with fixed sample rates in 2-D (and 1-D) discrete
> array cameras all the time.
> The spatial sample rate is fixed and we simply accept the aliasing
> which
> shows up as Moire' patterns, etc. In 1-D signal processing we are
> often stuck with imperfect lowpapss filters or simply know that
> the signal bandwidth is "mostly" inside the Nyquist limit and
> accept the aliasing of
> lower energy components at higher frequencies. So, you're not
> alone in
> needing to sample a signal "blind". In the end it becomes a
> matter of judgement and a trade between what is theoretically
> necessary and what is practical to achieve.
>
> Fred
Thanks Fred, for the comfort.
Guess, I cannot help to sample the signal "blindly", and accept some
aliasing.
On that way, I'll nevertheless try the best to achieve at least a
minimum of anti-alias-filtering.
My best approach at time being is an averager with all 1s
coefficients, because it's the only one I can imagine if I don't
know the length of the filter (the sample ratio).
Bernhard
Reply by Fred Marshall●September 22, 20042004-09-22
"Bernhard Holzmayer" <holzmayer.bernhard@deadspam.com> wrote in message
news:2622338.VIXaoWmUPI@holzmayer.ifr.rt...
> Fred Marshall wrote:
>
>>
>> "Bernhard Holzmayer" <holzmayer.bernhard@deadspam.com> wrote in
>> message news:1176462.lnuKDRrfKf@holzmayer.ifr.rt...
>>> jim wrote:
>>>
...................................
>
> To show you, what concern bothers me, let's assume a signal with
> frequency content from DC to 20kHz.
> At a high positional clock rate (high speed), it will be sampled at
> some 40kS/s or more, then the mentioned IIR cuts away everything
> outside a band of 10...15kHz.
> Now let's assume that positional clock rate (speed) is lower, which
> might result in a momentary sampling rate of 20kS/s. The IIR filter
> will scale accordingly, and filter the signal at 5...7,5kHz.
> But Nyquist requires that the signal is bandlimited to 0...10kHz.
> If sampling rate were at 1kS/s, legal band would range from
> 0...500Hz.
> That's what the decimation filter usually deals with.
> And that's what I don't have and don't know how I should realize it.
> Since there are certainly frequency components (noise) which do not
> scale with speed, I cannot neglect this.
>
> If you remember that old trick, with machinery noise. How did they
> overcome the influence of noise which didn't scale with rotational
> speed? Just ignore the problem?
>
> Bernhard
>
Bernhard,
Understood - if the clock rate varies then the lowest sample rate sets a
lower Nyquist frequency. One wouldn't ignore noise in this situation.
There are a number of observations I might make:
1) If you clock on the tachometer and the speed is varying, then the normal
single-temporal frequency, sinusoidal additive noise may be spectrally
spread by synching the sampling clock to the tach. NOTE: this means
time-based frequency components - which might look like "noise" in the
rotational domain.
2) If you transform the thinking to the rotational position domain then the
Nyquist criterion still applies - but now in radians per revolution instead
of radians per second. You can have a sinusoid at 20k periods per
revolution components that have to be sampled at higher than 40k samples per
revolution.
3) If you sample noise that's at a rotational frequency higher than 20k
periods per revolution then you'll alias it - just as if you were using a
time base. Sometimes it's acceptable to alias white noise and sometimes
not. The problem is the same.
So, in order to "properly" sample the data that's clocked with the tach you
need to have an idea of the spectral character of the result. I guess one
way to measure that character is to increase the rotational sample rate and
FFT reasonable length sequences and find where the high frequency energy in
the result goes away. Then the sample rate is probably high enough.
By "increase the rotational sample rate", I mean that you generate the
sampling clock with something that looks like a frequency synthesizer.
Inside the synthesizer is a phase-locked loop. Instead of the input being a
"time stable", lower frequency, oscillator, the input is the tach and the
synthesizer generates higher frequencies that are locked to the tach. By
setting the synthesizer parameters you can change the output clock rate and
generate higher rate clocks.
Alternately or in combination, you might lowpass filter the (analog) sensor
outputs to achieve the same results.
We are stuck with fixed sample rates in 2-D (and 1-D) discrete array cameras
all the time.
The spatial sample rate is fixed and we simply accept the aliasing which
shows up as Moire' patterns, etc. In 1-D signal processing we are often
stuck with imperfect lowpapss filters or simply know that the signal
bandwidth is "mostly" inside the Nyquist limit and accept the aliasing of
lower energy components at higher frequencies. So, you're not alone in
needing to sample a signal "blind". In the end it becomes a matter of
judgement and a trade between what is theoretically necessary and what is
practical to achieve.
Fred
Reply by Bernhard Holzmayer●September 22, 20042004-09-22
Bernhard Holzmayer wrote:
> Trying to get your perspective (leaning back in my chair, feet on
> the desk...), I come to this premature conclusion:
Ooh: 'preliminary conclusion' was, what I intended to say.
But the other expression isn't that bad...
Bernhard
Reply by Bernhard Holzmayer●September 22, 20042004-09-22
Jerry Avins wrote:
> Bernhard Holzmayer wrote:
>
>> Jerry Avins wrote:
>
>
> ...
>
>>>The content is independent of speed?
>>
>> The part which I'm interested in: no.
>> Noise and other dirt effects, maybe yes.
>
> That part not dependent on speed may be coming out of the
> transducer, but it probably doesn't represent anything about the
> process you're measuring. Is it possibly noise you might somehow
> reject?
>
Yes.
>
>>>The Nyquist limit on the data rate relative to the sample rate is
>>>just like a highway's speed limit. You are not required to go
>>>that fast. Do I not understand?
>>
>> I agree. And I know. But the noise on my signal doesn't.
>> It's just present on my samples without taking notice of the
>> speed limit!
>
> That's all the more reason to believe (from this safe distance!)
> that your transducer not only measures eddy currents in the web,
> but also picks up noise induced from another source. Does
> shielding it or changing its orientation alter the noise? Could
> the noise be bucked by a differential design?
We do all that already.
Measurement is sometimes not far from noise floor, S/N ratio being
around 10dB.
Since there was no reason up to now to differentiate between noise
which comes from transducer and other sources, I really don't know.
And guessing is not so much a reliable method...
>
>> And since the noise is present, let's say at 14kHz, this would
>> mean, that I'm not allowed to drive the line that slow. I may
>> decrease speed only downto something above 28kS/s.
>> (line speed is a process parameter, which I can measure, but not
>> influence)
>
> If you can identify the noise source, maybe you can increase its
> frequency to the point that a single low-pass filter that doesn't
> disturb what you need to measure can block most of it.
I agree. That came to my mind already. I'll probably end up with a
mixture: one filter before the decimation process, another after.
The first must deal with the speed independent components, the other
with the effects which scale.
>
>>>>If sampling rate were at 1kS/s, legal band would range from
>>>>0...500Hz.
>>>>That's what the decimation filter usually deals with.
>>>
>>>You don't need to remove what isn't there. I'm clearly missing
>>>something important. Please try to explain.
>>
>>
>> Imagine very slow sampling at a rate of 200 samples/s (which is
>> not far from reality in my applications).
>> Let's assume a signal band from 0...100Hz.
>> The content will be a mixture between wanted signal components
>> like the ones caused by material defects (these will scale with
>> speed). Other components are unwanted but unavoidable. There are
>> two sorts of them:
>> - noise from the machinery (bearings, motors,...) which scale
>> with speed
>
> If you truly sense eddy currents, how are these picked up?
There's a system of coils with essentially one transmitter and
another single or differential receiver coil, such arranged, that
the currents from transmitter to receiver must pass the material
under test.
Unwanted "noise" is either from material effects which are of no
interest, or of the processing machinery (vibrations or distance
variations coming from straighteners, rolls, punches etc. on this
or a neighbour line)
>
>> - environmental noise (50Hz or 60Hz from mains), speed
>> independent vibrations,...
>
> Ah: Hum! Do you use an instrumentation amplifier referenced to the
> converter's AGND? IIRC, you have a carrier-based system. A
> transformer might replace the I-amp. I can understand hum added to
> the carrier, but not modulating it. A band-pass filter at the
> carrier frequency might clean up the signal.
It was a guess, hopefully I need not worry about this.
>
>> The problem are those sources which don't scale with speed.
>> The coupling from mains will certainly not scale with speed.
>> Now, assume that sampling rate decreases with speed to 40
>> samples/s.
>>
>> 50Hz/60Hz noise persists, but now rides way too fast on that
>> highway...
>
> ...
>
> It seems to me, leaning back in my armchair with my feet up on the
> ottoman, that your signal is contaminated with possibly removable
> dreck. If it were my problem, I would do everything possible to
> get a clean analog signal before sampling it. Sometimes, little or
> nothing is possible, but I would try to be sure before exploring
> other approaches.
>
> Jerry
In the past, we were doing all that in analog realm.
Most of it with filters.
These filters, which I try to replace by digital filters, and
especially by filters which scale with speed.
Trying to get your perspective (leaning back in my chair, feet on
the desk...), I come to this premature conclusion:
I need the static (outer bandpass) filters to remove all the
non-scaling dreck effects.
I can remove all the scaling effects with the speed-variant filter
(inner bandpass).
There stays a danger, that unwanted components pass the outer
bandpass. Of these, we must accept the facts
- that they are present outside the inner filter.
- that aliasing effects make these components fold into the region
of interest.
The last point leaves me unhappy/challenged.
Bernhard
Reply by Jerry Avins●September 21, 20042004-09-21
Bernhard Holzmayer wrote:
> Jerry Avins wrote:
...
>>The content is independent of speed?
>
> The part which I'm interested in: no.
> Noise and other dirt effects, maybe yes.
That part not dependent on speed may be coming out of the transducer,
but it probably doesn't represent anything about the process you're
measuring. Is it possibly noise you might somehow reject?
...
>>The Nyquist limit on the data rate relative to the sample rate is
>>just like a highway's speed limit. You are not required to go that
>>fast. Do I not understand?
>
> I agree. And I know. But the noise on my signal doesn't.
> It's just present on my samples without taking notice of the speed
> limit!
That's all the more reason to believe (from this safe distance!) that
your transducer not only measures eddy currents in the web, but also
picks up noise induced from another source. Does shielding it or
changing its orientation alter the noise? Could the noise be bucked by a
differential design?
> And since the noise is present, let's say at 14kHz, this would mean,
> that I'm not allowed to drive the line that slow. I may decrease
> speed only downto something above 28kS/s.
> (line speed is a process parameter, which I can measure, but not
> influence)
If you can identify the noise source, maybe you can increase its
frequency to the point that a single low-pass filter that doesn't
disturb what you need to measure can block most of it.
>>>If sampling rate were at 1kS/s, legal band would range from
>>>0...500Hz.
>>>That's what the decimation filter usually deals with.
>>
>>You don't need to remove what isn't there. I'm clearly missing
>>something important. Please try to explain.
>
>
> Imagine very slow sampling at a rate of 200 samples/s (which is not
> far from reality in my applications).
> Let's assume a signal band from 0...100Hz.
> The content will be a mixture between wanted signal components like
> the ones caused by material defects (these will scale with speed).
> Other components are unwanted but unavoidable. There are two sorts
> of them:
> - noise from the machinery (bearings, motors,...) which scale with
> speed
If you truly sense eddy currents, how are these picked up?
> - environmental noise (50Hz or 60Hz from mains), speed independent
> vibrations,...
Ah: Hum! Do you use an instrumentation amplifier referenced to the
converter's AGND? IIRC, you have a carrier-based system. A transformer
might replace the I-amp. I can understand hum added to the carrier, but
not modulating it. A band-pass filter at the carrier frequency might
clean up the signal.
> The problem are those sources which don't scale with speed.
> The coupling from mains will certainly not scale with speed.
> Now, assume that sampling rate decreases with speed to 40 samples/s.
>
> 50Hz/60Hz noise persists, but now rides way too fast on that
> highway...
...
It seems to me, leaning back in my armchair with my feet up on the
ottoman, that your signal is contaminated with possibly removable dreck.
If it were my problem, I would do everything possible to get a clean
analog signal before sampling it. Sometimes, little or nothing is
possible, but I would try to be sure before exploring other approaches.
Jerry
--
Engineering is the art of making what you want from things you can get.
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