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Sampling Rate, Nyquist Frequency, and Aliasing

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Uploaded by on May 18, 2009

http://video_demos.colostate.edu

Demonstration of sampling rate, aliasing, Nyquist frequency, and amplitude fidelity.

Software used: "MAX MSP" from the company "Cycling 74"

Many more video demonstrations of engineering and physics principles and devices can be found at:

http://video_demos.colostate.edu

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Uploader Comments (DrDaveBilliards)

  • DrDaveBilliards does not understand sampling theory. The amount bad information presented in the comments section of this video is astounding. Please get your information elsewhere.

    (Sorry Dave. Nothing personal. I just don't like to see bad science being spread on the internet.)

  • @afreshcupofjoe

    Why don't you post a video showing how a reconstruction filter reproduces a signal (e.g., a simple sine wave) from various sample sets at different starting points on the wave, with a sampling rate just above Nyquist. That would be interesting and useful, and I would be happy to accept it as a video reply.

    Thanks,

    Dave

  • @DrDaveBilliards - cont.

    Obviously, if a "reconstruction filter" is not being used, you need to sample faster to get a better representation of the signal (e.g., if you just plot the raw sampled data). Most of my comments were dealing with simple plots of raw sampled data. I would certainly like to learn more about reconstruction filters

    Thanks again,

    Dave.

Top Comments

  • This just proves you don't understand the theory. You have to have a filter on the output if you want to avoid aliases above the Nyquist frequency distorting the output. Only then will inputs near the nyquist frequency pass though the system undistorted. Just how near you can go to the Nyquist frequency depends on how good your filter is.

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All Comments (43)

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  • short and sweet an very helpful . thnx

  • If you're looking to learn about the theory of the nyquist rate, disregard this comment, it is incorrect.

  • Now I see why it's 44.1 and not 44! hhh THNX!

  • @jonahansen; (continued)

    My understanding is that reconstruction of the signal is achieved by passing a train of impulse value spikes (from the sample points) through an f Hz filter (assuming the original sampling was done at 2f Hz).

    If anyone knows better, please correct the above.

    BTW, does anyone know if the unfiltered "join the sample values" waveform the video shows will give the same output as the equivalent train of sample-point impulses *after* both have been f Hz low-pass filtered?

  • @jonahansen; I'd prefer "arrogance and condescension" than the spread of misinformation.

    I'm no expert, but AFAIK Nyquist-Shannon does *not* state that the samples (or rather, the waveform recreated from them in the "join the dots" manner seen here) represents the final reconstructed output as implied.

    If that's not what the video is saying, then it needs to be clearer, because it's either very misleading or wrong.

  • Wow - the arrogance and condescension in these comments almost approaches the idiocy of comments in standard, non-technical youtube videos. Dr Dave said nothing wrong, really; there is a lot more to sampling, as many have touched upon, regarding reconstruction of the original waveform, etc. But he is not "flat-out wrong", he's quite correct as far as he goes, and it is a good demonstration of a basic principle of sampling. Dunning-Kruger effect anyone?

  • good job, some equations might help get the idea accrooss as well :)

    ty

  • @afreshcupofjoe

    Thank you for the document link. BTW, case seems to be important ("Sampling_Theory" works, but "sampling_theory" does not).

    I'll read through your document when I can find some time, and I will reply if I think it necessary or appropriate. The document looks like a good resource for people who want to explore the details more.

    BTW, how did you create the upside down text that fools YouTube's URL blocker?

    Thanks,

    Dave

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