Basic Signal Processing Question

<Venkat.Vijay.Kumar@gmail.com> wrote in message
news:1121910110.692753.63950@g43g2000cwa.googlegroups.com...
>
>
> Bhaskar Thiagarajan wrote:
> > "Mark" <makolber@yahoo.com> wrote in message
> > news:1121866582.325514.102090@z14g2000cwz.googlegroups.com...
> > > That is an interesting question,
> > >
> > > I thought that impulse sampling is desirable because it provides an
> > > accurate representation of the signal, but that natural sampling (what
> > > the OP describes as sampling with a square wave) also is possible and
> > > also provides a representation of the signal except that natural
> > > sampling  creates sinx/x attenuation to the higher frequencies.  I
> > > think that in practice one cannot achieve ideal impulse sampling
> > > because the aperture cannot be open for an infinitesimal time but must
> > > be open for some short but finite time.
> > >
> > > I thought this is completely analogous to what happens in
> > > reconstruction in a DAC.  You can use (in theory) impulse
> > > reconstruction which creates no sinx/x attenuation.  But in practice
> > > most designs use natural sampling reconstruction (the samples persist
> > > for the entire time duration of the sample) and this creates sin x/ x
> > > attenuation which is compensated for elsewhere in the system.
> >
> > Mark
> > You are confusing the DAC process from the ADC process...
> > The ADC *samples* a signal to create discrete signals, while the DAC
creates
> > a continuous signal from a discrete signal.
> > The 2 operations are quite a bit different and the zero order hold of a
DAC
> > is not what i'd call 'natural sampling'.
> >
> > The OP might also be a little confused with the term 'clock' and
sampling
> > since a clock tends to be a nice square wave in circuit terms...however
the
> > model for sampling involves impulses. Despite feeding a square wave as a
> > clock to an ADC, the ADC doesn't use square waves in it's actual
'sampling'
> > of the input analog signal.
> >
> > Cheers
> > Bhaskar
>
> Hi Bhaskar,
>         My confusion is what you mentioned in your reply, that is about
> the clock and sampling. I agree with the sampling with impulses would
> retain the input signal. (Same as Fin and not Fclk+/-Fin)
>        But how does it work with a clock (which is a square wave). What
> do you mean by "feeding a square wave as a clock to an ADC, the ADC
> doesn't use square waves in it's actual 'sampling' of the input analog
> signal."
>
>       How does the sampling in an ADC work. I am bit confused by your
> above statement. Can you eloberate??

I'd probably do a poor job of explaining the internal workings of an ADC -
google is your friend here and you should be able to find plenty of app
notes and articles that explain how ADCs work.
I'll try to give you something really simple to start with...but you should
understand that real ADCs have a lot more going on inside.
Most ADCs have 2 important blocks, the sample and hold block and the
quantization block.
The sample and hold block basically 'samples' and holds that instantaneous
value for the quantization block to convert the analog voltage into a
discrete value. For example, in an N bit converter, this block would compare
this voltage to one of 2^N levels and output a N bit word based on the
comparison.

Back to what confuses you...the sample and hold circuit typically works off
the edges of the clock that you supply to the chip. Say, it uses the rising
edge only, then effectively, the sample and hold block generates 'impulses'
off the incoming clock to sample the input signal that the instants when the
clock has a rising edge.
Jerry referred to 'aperture time' - this is the time (usually very small but
not zero) that actual ADCs need to 'sample' the input signal. What this
amounts to is that instead of ideal impulses, the actual sampling occurs
over a very small delta_t. One can still approximate this to impulses for
casual analysis and modeling but this cannot be ignored when looking at the
capabilities and performance of real ADCs.

Hope I kept it simple enough to illustrate what you need to know. There is a
lot more to ADCs and I'd refer to you app notes and papers depending on how
deep you'd want to get.

Cheers
Bhaskar


>
> Thanks for your reply..
>
>
>
>
> >
> >
> > >
> > > So I thought impulse vs natural sampling analogy applies for both the
> > > sampling process and the reconstruction process.   Another word for
the
> > > high frequency attenuation that occurs with natural sampling is
> > > aperture attenuation and it can occur during sampling (capturing the
> > > signal) or during reconstruction.
> > >
> > > Is this not true?
> > >
> > > thanks for the interesting question.
> > >
> > > Mark
> > >
>

Mark wrote:

> I think we need to separate two aspects of what an A/D does...

> 1) sample

> 2) quantize

> An analog signal can be sampled without being quantized.  FM stereo
> does this.

I agree that signals can be sampled without being quantized,
but I don't believe that FM stereo does that.

It is an AM suppressed carrier modulated signal, both the subcarrier
and the baseband must be band limited so they don't overlap,
but they aren't sampled.

-- glen

Venkat.Vijay.Kumar@gmail.com wrote:

   ...

>         My confusion is what you mentioned in your reply, that is about
> the clock and sampling. I agree with the sampling with impulses would
> retain the input signal. (Same as Fin and not Fclk+/-Fin)
>        But how does it work with a clock (which is a square wave). What
> do you mean by "feeding a square wave as a clock to an ADC, the ADC
> doesn't use square waves in it's actual 'sampling' of the input analog
> signal."
> 
>       How does the sampling in an ADC work. I am bit confused by your
> above statement. Can you eloberate??

Now you're getting into hardware design. Do you really want to go there? 
Working parts are not made of equations, but of bits of wire, silicon, 
foil, solder, and more. Circuits can be made that are insensitive to 
level, but respond to changes in level Such devices are called "edge 
triggered" In explaining an ADC, I'll ignore them; just know that they 
can simplify a design.

Our ADC consists of a voltage-controlled switch (you can think of it as 
a relay), a sample holding capacitor, a comparator (greater-than, 
less-than) a set of latches, and a DAC. There's also a control circuit, 
but for now, that's magic.)

When the clock is high, the switch connects the analog signal to the 
capacitor, which follows the signal. When the clock goes low, the 
capacitor is disconnected from the signal and its voltage stays 
substantially constant. The MSB of the ADC is turned on by setting the 
corresponding latch and the comparator is consulted. If the capacitor 
voltage is less than the DAC voltage, the latch is cleared; otherwise, 
it is left on. Then the same is done with the next-highest bit, and so 
on until the LSB is tested. When the process is done, the state of the 
latches (and sometimes the comparator output) are output as a number 
that represents the voltage on the capacitor. When the clock goes high 
again, it clears the latches and reconnects the capacitor to the analog 
signal.

Let's be clear: there is no magic. To make one of these work, you need 
to actually build the control circuit. Moreover, there are many other 
ways to build an ADC, some of them with more delay than the 
successive-approximation type I describes, some of them slower. The 
output can come serially or all at once. I omitted the control signals 
that make the information transfers possible and that control timing.

We are often warned not to mistake the map for the territory. Similarly, 
we need to beware of mistaking the equations for the device. Also, don't 
assume how a circuit works and draw conclusions from the assumptions. If 
you really want to know, look at a schematic. It defeats all of us when 
we believe that something must be a certain way because we can't 
conceive of any other way. "There are more things in heaven and earth, 
Horatio, than are dreamt of ..." There's _always_ another way.

Jerry
-- 
Engineering is the art of making what you want from things you can get.
�����������������������������������������������������������������������

Mark wrote:
> I think we need to separate two aspects of what an A/D does...
> 
> 1) sample
> 
> 2) quantize
> 
> 
> An analog signal can be sampled without being quantized.  FM stereo
> does this.

So does a S/H

> Sampling can be impulse sampling or natural sampling (full duration) or
> anything in between.  True impulse sampling is impossible because a
> true impulse is infinitesimal in duration.

What duration? a full cycle of the sampled signal? How is that useful?

> Quantization converts the sample to a number.    Since the sample is
> not infinitesimal in time, the quantization process has to average or
> otherwise take a non-infinitesimal duration analog signal and convert
> it to a single number.

What happens to a "natural sample", whatever it is?

Jerry
-- 
Engineering is the art of making what you want from things you can get.
�����������������������������������������������������������������������

The L and R are time division multiplexed by sampling each at 38 kHz.
The L and R samples are interleaved and FM modulated onto the main RF
carrier.  This is spectrally equivalent to what you described.  This is
two different descriptions of the same thing, one description looking
at the frequency domain, and one description looking at the time
domain.   Actually there are two different design approaches to an FM
MPX encoder, the TDM approach and the L-R matrix approach, both create
the same output signal and are fully compatable.



Mark

>What duration? a full cycle of the sampled signal? How is that useful?

No a full cycle of the samplING signal which is at least 2x the samplED
signal.

Mark

Mark wrote:
> The L and R are time division multiplexed by sampling each at 38 kHz.
> The L and R samples are interleaved and FM modulated onto the main RF
> carrier.  This is spectrally equivalent to what you described.  This is
> two different descriptions of the same thing, one description looking
> at the frequency domain, and one description looking at the time
> domain.   Actually there are two different design approaches to an FM
> MPX encoder, the TDM approach and the L-R matrix approach, both create
> the same output signal and are fully compatable.

There is no sampling. 38 KHz is the frequency of the subcarrier that 
carries the difference signal. The sum signal is transmitted as in a 
mono signal, so that a mono signal is received as such and a mono 
receiver receives L+R. That's what's meant by compatibility. After the 
difference signal is recovered from by demodulating the subcarrier (with 
the aid of the 19 KHz pilot tone), the two signals are matrixed to 
create L and R. If the pilot tone is too weak, there is no matrixing.

Which standard have you been reading?

Jerry
-- 
Engineering is the art of making what you want from things you can get.
�����������������������������������������������������������������������

Mark wrote:
>>What duration? a full cycle of the sampled signal? How is that useful?
> 
> 
> No a full cycle of the samplING signal which is at least 2x the samplED
> signal.

When it's over and done with, what is the analog value of the analog 
sample on the hold capacitor? Another question: what's natural about it?

Jerry
-- 
Engineering is the art of making what you want from things you can get.
�����������������������������������������������������������������������

I did not invent the term "Natural sampling"

see:

http://doctord.dyndns.org:8000/courses/tutorials/Sampling.htm

http://uregina.ca/~douglatr/ENEL%20813/enel_813.htm

http://www.electronics.dit.ie/staff/ptobin/chapt06a.pdf


You can Google ("natural samping" + impulse)   for more examples


Mark

Mark wrote:
> I did not invent the term "Natural sampling"
> 
> see:
> 
> http://doctord.dyndns.org:8000/courses/tutorials/Sampling.htm
> 
> http://uregina.ca/~douglatr/ENEL%20813/enel_813.htm
> 
> http://www.electronics.dit.ie/staff/ptobin/chapt06a.pdf

I did not suggest that you did. I asked for an explanation; thanks for 
the links.

Jerry
-- 
Engineering is the art of making what you want from things you can get.
�����������������������������������������������������������������������