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6416 DSK "Force Aliasing"

dspRookie
Intellectual 430 points

I am an instructor in a DSP design course. I want to show the students what aliasing looks like, but cannot do it using a function generator. I think the anti-aliasing filter on the input is squashing the input when its frequency is about Nyquist. Is there a way to disable the anti-aliasing filter temporarily?

Dennis Briggs

  • 0
    Intellectual 430 points

    Sorry -- I meant "above Nyquist", not "about Nyquist".

  • 0 in reply to dspRookie
    TI__Expert 8055 points

    Dennis,

    From what I can tell, there is no way to do this.  This board implements an AIC23B Codec for the Audio input/output.  (Documentation is here: http://www-s.ti.com/sc/ds/tlv320aic23b.pdf)  I believe these filters are part of the codec.  And while you can adjust the codec's sampling frequency, the anti-aliasing filter is automatically adjusted to that range.

    You could "simulate" aliasing by changing the settings on the DMA.  The DMA sorts the input into 2 buffers, a left and right channel.  If you change it to sort into 4 buffers, a left even, right even, left odd, and right odd, but then only use the even samples, you'll effectively be reducing the sampling rate in half, while the anti-aliasing filter passes everything up to the original Nyquist.

    Regards,

    Dan

     

     

  • 0 in reply to DanRinkes
    Intellectual 430 points

    Thanks for the quick response, Dan. I'll have a look at your suggestion. 

    Regards,

    Dennis

  • 0 in reply to dspRookie
    TI__Guru* 84110 points

    Be sure to tell your Dean how happy you are with the helpful people at TI's E2E forum! And let him/her know how impressed TI is with your dedication to your students and their learning experience.

    With any good sine wave coming in from your function generator, below the anti-aliasing filter's threashold, pull samples out of the input buffer at the 2x Nyquist rate. If the ADC's sample rate is 8KHz and you put in a 1KHz signal, pull out every 4th sample for a 2x rate.

    The reconstructed waveform from those samples will probably give you the right frequency output, but the amplitude will only be right if you hit the input signal right at the peaks.

    Now try pulling out every 12th sample. The effect is the same as increasing the input frequency. Your reconstructed output will be at a lower frequency because of aliasing.

    Regards,
    RandyP

  • 0 in reply to RandyP
    Intellectual 430 points

    This is a shout out to Randy and Dan for their responses to my question. Dan's answer turned me away from trying to alter the LPF, and Randy's answer confirmed the solution that I found.

    I have figured out how to demonstrate aliasing, but not by using a function generator for input. Instead, I create a circular buffer and draw from it, bypassing the ADC and the anti-aliasing filter.

    If the circular buffer is 48 long, and populated with 30 cycles of a sinusoid, and if the sample rate is 48 KHz, then running through the buffer generates a 30 KHz sinusoid. The DAC puts out an aliased signal consisting of two frequencies: 18 KHz and 6 KHz. Perfect!

    Thanks, you guys, for the answers you gave. 

    DSProokie