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Sampling without PAM, PWM, carrier signals, or modulation -- Analog Electronic Chip

Discussion in 'Electronic Basics' started by Radium, Mar 20, 2007.

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  1. Radium

    Radium Guest

    Okay so sampling is a *must*. But can this sampling be done without
    using PAM, PWM, or any carrier waves? Can't the charges be sampled in
    the capacitors without adding any sort of modulation to them? If not,
    they why?
     
  2. Radium

    Radium Guest

    Sorry I'd didn't understand what you just wrote.
     
  3. Bob Myers

    Bob Myers Guest

    No, per the Einstein-Feldburg equations, sampled charges
    must be modulated through a transfinite hyperconductive
    infidebulum matrix, with the resulting para-stable phase
    incoherency reduced through application of the usual methods,
    including negative beta-state feedback or loosely-coupled
    ferro-optical heterodyne coils. The more astute reader will
    note the latter can, at least potentially, produce undesirable
    endochronic states and in extreme cases distortion of the
    transconductive slew-rate limited noise figures, but in practice
    these are of low enough vector magnitude in the frequency
    domain so as to be ignorable.

    Unless, of course, you've got a cold solder joint in there
    somewhere.

    Bob M.
     
  4. Bob Masta

    Bob Masta Guest

    Huh? Whatever are you going on about *now*? PAM, PWM, and
    carrier waves have *NOTHING WHATSOEVER* to do with sampling.


    Bob Masta

    D A Q A R T A
    Data AcQuisition And Real-Time Analysis
    www.daqarta.com
    Scope, Spectrum, Spectrogram, Signal Generator
    Science with your sound card!
     
  5. John Fields

    John Fields Guest

    ---
    As Bob Masta said: "Huh? Whatever are you going on about *now*?"

    Here's what sampling is: Let's say that you have a 1000Hz 1VPP sine
    wave of which you want to store one cycle and that you've got 360
    capacitors on hand.

    Then you might do something like this:

    A B
    ACIN>--O--> | <---------------------->OUT
    | S2 1
    S1 O---O--->O----------+
    C C |
    2 O------+ [C1]
    . | |
    . [C2] |
    . | |
    360 O--+ | |
    | | |
    [C360] | |
    | | |
    GND>--------------------+---+---+---->GND

    S1 is a SPDT center-off analog switch of some kind and S2 is a 360
    channel break-before-make analog switch. In order to store the
    waveform what you do is start with S2 in position 1, very briefly
    connect S1C to S1A, then return S1 to the OFF position. This will
    charge C1 (Which is a tiny sample-and-hold cap) up to the voltage
    of the input signal, more or less.

    Once that's done, S2C is connected to S2-2 and the input signal
    sampled again.

    After 360 samples have been taken, S2 is is placed in the '1'
    position, S2 is placed in the 'B' position, and S1 made to traverse
    S1-1 through S1-360 at the same rate as when the samples were taken.

    When that's done, the voltages each capacitor has been charged to
    will be presented at "OUT" sequentially, recreating the input
    waveform with only very light low pass filtering required between
    the different output samples.

    What I've shown you is _very_ rudimentary, and there needs to be a
    mechanism employed to discharge the caps, which I haven't shown, but
    which I'm sure you can find if you Google "sample and hold"
     
  6. Bob Myers

    Bob Myers Guest

    Perhaps you should do some more research in this field,
    then.

    Bob M.
     
  7. Radium

    Radium Guest

    Thanks for clearing that up.
     
  8. Radium

    Radium Guest

    Thanks for the illustration.

    Why is sampling required before digitizing a signal?
     
  9. John Fields

    John Fields Guest

     
  10. Bob Myers

    Bob Myers Guest

    Strictly speaking, though, it isn't really mandatory that the
    sampling process come first. One could easily devise a "continuous"
    sort of A/D (just a string of resistors and some comparators
    should do the trick) from which the digital values could then
    be sampled at regular intervals. There are, of course, practical
    reasons for putting a sample-and-hold in there first, but in
    theory the process works just the same either way.

    What IS required, of course, is bandlimiting of the signal
    prior to sampling...

    Bob M.
     
  11. Don Bowey

    Don Bowey Guest

    How does one digitize a non- sample?
    A sample is a sample regardless of the method. John did not say it had to
    use a sample-and-hold process.
    It's not mandatory if the sample rate is high enough.
     
  12. Bob Myers

    Bob Myers Guest

    The situation described would have N bits of output
    from the continuous A -> D function, which would be
    changing without any particular relationship to one
    another in time. Admittedly, you would have to sample
    (e.g., capture the value via a clocked register) to make
    much sense of it (particularly if it was changing quickly
    enough! :)).
    There's still no sampling going on in the above example PRIOR
    to the output's state at a given instant being captured in a register.
    Quantization, yes, but not sampling.
    Unless the sampling rate is infinite - which has proven to
    be difficult to achieve in practical designs - the bandwidth
    of the input signal MUST be strictly limited to less than 1/2
    the sample rate in order to avoid aliasing, per the Gospel
    According to St. Nyquist.

    Bob M.
     
  13. Don Bowey

    Don Bowey Guest

    As I inferred in my comment, the sampling rate can be much greater than the
    minimum required to meet the Nyquist requirement.
     
  14. Bob Myers

    Bob Myers Guest

    Assuming you mean "implied" - of course. But there's
    still a requirement that the input be bandlimited. If you
    can count on the input being well below 1/2 the sampling
    rate, which is what I INFER you mean, then that
    requirement is already met, right?

    Bob M.
     
  15. Bob Masta

    Bob Masta Guest

    The band-limiting Bob Meyers is referring to is a function
    of sample rate... you need to limit the band to less than
    half the sample rate.

    The string-of-comparators A/D is a typical "flash" converter
    approach, used where speed is more important than resolution.
    (You almost never see these with more than 8 bits, since that
    requires 255 comparators.) Their conversion rate is limited by
    the settling time of all the comparators, but they often go into
    the 10s or 100s of MHz. In principle, they would be "continuous"
    if they had infinitely-fast comparators. This is a separate issue
    from the sampling process, which has to do with when and how often
    you *read* those comparators.

    There is also a clever (but not terribly useful) continuous conversion
    design that only has one stage per bit, and can be cascaded for more
    bits. It involves determining whether the incoming signal is above
    or below half of full-scale, and subtracting off that half if so,
    before passing the signal to the next stage. But this is even harder
    to do quickly, and hard to make accurate, so it's not used as a
    stand-alone converter. But the basic concept is sometimes used
    as a front end, to handle a few bits of pre-conversion ranging, etc.

    Best regards,




    Bob Masta

    D A Q A R T A
    Data AcQuisition And Real-Time Analysis
    www.daqarta.com
    Scope, Spectrum, Spectrogram, Signal Generator
    Science with your sound card!
     
  16. Don Bowey

    Don Bowey Guest

    You infer correctly.

    Don
     
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