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Anti-aliasing ADC samples

Discussion in 'Electronic Design' started by eliben, May 8, 2006.

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  1. eliben

    eliben Guest

    Hello all,

    I have an ADC that samples some analog signal at 1 Msps. The samples
    are then processed by a digital logic / software. To avoid aliasing, I
    must make sure that signals entering the ADC have no harmonics at > 500
    KHz. Is the only way to do it by an analog lowpass filter (i.e. RC)
    before the ADC ? I assume that there is no way whatsoever to filter out
    the unwanted frequencies digitally after the ADC, since the ADC
    sampling itself already got distorted because of aliasing, is this
    right ?

    And yet, in some places I've seen mentioned digital lowpass filters for
    anti-aliasing. How is this possible ?

    Thanks
     
  2. John_H

    John_H Guest

    If you want to analyze frequency content to 500 kHz, the design of a
    "brick wall" low pass filter makes the filter design extremely
    difficult. If you sampled at 10 MHz, your low pass filter just needs to
    supply the proper rejection at nearly 20x your frequency of interest
    making the analog filter very simple. The digital filtering would be
    applied to the high sample rate to deliver the necessary "brick wall"
    filter and deliver your signals of interest up to 500 kHz without
    significant aliasing or other distortions that an analog filter would
    typically produce.

    It's the oversampling that allows digital filters to deliver such great
    performance for antialiasing. The analog filter is still needed, it's
    just so much simpler.
     
  3. Jon

    Jon Guest

    There is a way to digitally filter a signal without suffering from
    aliasing. It is called the bi-linear z transform method. You design
    an analog filter, and then transform it to a digital filter by
    substituting (2/T)[(1-z^-1)/(1+z^-1)] for s, where T is the sampling
    interval. Actually, the (2/T) factor is not necessary, but most books
    on DSP include it. The resultant filter will have better attentuation
    performance than the corresponding analog filter prototype. The
    transformed filter suffers from "frequency warping". You compensate for
    this by pre-warping the critical frequencies before you design the
    analog prototype filter. Neither the phase vs frequency curve nor the
    step response of the digital filter will match the correspondimg phase
    and step response characteristic of the analog prototype filter. Any
    book that covers digital filtering will have a section on the bilinear
    z-transform. See, for example, "Therory and Application of Digital
    Signal Processing" by Rabiner and Gold.
     
  4. John_H

    John_H Guest

    But aliasing still needs to be addressed at the analog level. You can only
    get a digital filter to work on frequencies below Nyquist. I think the
    original question was whether digital filters could get rid of the aliazing
    problems without use of any external analog filters. I'm happy to be
    mistaken.
     
  5. PN2222A

    PN2222A Guest

    Man, this is revolutionary!

    Do you mean I can sample a baseband signal with frequencies extending to f0
    at a sampling frequency of f0, and have no aliasing, just by using a
    bilinear filter?

    That is _so cool!_

    Maybe I'm missing something in your explanation?

    regards
    PN2222A
     
  6. Mac

    Mac Guest

    I have never seen anybody claim that you can get around aliasing the way
    you describe.

    The only thing I have heard of is "digital down-converting" where you
    bandpass filter a signal so that all that is left is that portion of the
    signal from, say, Fs to 1.5 * Fs. Then the signal is aliased, but can
    still be reconstructed correctly since the number of times it has
    "wrapped" is known.

    It could also be 1.5 * Fs to 2 * Fs. The only constraint is that the
    signal must still occupy a bandwidth of less than Fs/2 when it is actually
    sampled.

    And of course, if the signal is known to have no frequency content outside
    of the aforementioned band, then no filter is necessary.

    --Mac
     
  7. Ban

    Ban Guest

    First you have to understand what aliasing means. IF you sample at 1MHz but
    only want to use audio frequencies up to 20kHz, aliasing doesn't disturb
    unless it is folded back into 0 - 20kHz. This means you only need to care
    that at 980kHz and above no signal goes into the ADC which is above
    say -90dB or whatever noise floor you are digitizing.
    All those aliased frequencies between 500k and 980k can be eliminated
    digitally when downsampling, but anything in the bandwidth of interest can
    not be eliminated later.
     
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