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New Sampling Techniques Make Up For Lost Data

An unnamed reader writes: "Professors at Vanderbilt and the University of Connneticut have published a non-uniform sampling theory that could yield better quality digital signals than the standard Uniform sampling techniques pioneered by Shannon at Bell Labs. The Vanderbilt press release and link to the published paper can be found here."

19 of 162 comments (clear)

  1. Re:So.... by pmc · · Score: 5, Informative

    No.

    As the abstract says

    "The new theory, however, handles situations where the sampling is non-uniform and the signal is not band-limited."

    So it isn't applicable to digital music (as this is band-limited by our hearing, and we can pick the sampling interval) but other signals that cannot be sampled well by regular sampling (either in time or in space). Examples given are seismic surveys and MRI scans. But you knew this as you'd have taken the time to read the linked article first, wouldn't you?

  2. Some useful niche applications by michaelmalak · · Score: 5, Insightful
    Think about computer displays. Would you ever want to have to deal with non-square pixels? Sometimes, yes, like in the CGA days where the goal was to display 80 columns while keeping memory and bandwidth costs down. In general, it's a PITA. Now multiply that pain by not only having non-square pixels but where the pixels also come in various sizes.

    What's the practicality of this? Well, spiral MRIs, for example, where for mechanical reasons you don't want to have to stop-and-start the very heavy "scanner", wasting time and jarring sensitive equipment. As I said, niche applications.

    As for compressing audio, there are already plenty of other psychoacoustic compression schemes -- whether non-uniform sampling is better or worse will likely depend on the application.

    1. Re:Some useful niche applications by ncc74656 · · Score: 3, Informative
      And, I'm still trying to figure out by what you mean by non-square pixels. Are you trying to say the physical size on the screen, or how they are stored in memory on the graphics adaptor?
      The pixels that make up a CGA image aren't square...they were drawn on a 640x200 grid. The pixels on a VGA display at most resolutions are square (1280x1024 is the most common exception)...for instance, (1024/4)/(768/3)=1. With CGA, (640/4)/(200/3)=2.4, which means it's stretched vertically.
      --
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  3. Nyquist, not Shannon by s20451 · · Score: 5, Informative

    It was at Bell Labs ... but the guy who developed the Uniform Sampling Theorem was Nyquist, not Shannon.

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  4. Anybody understand what's new? by KjetilK · · Score: 3, Interesting
    The article was really short on details, I think, so I found it very hard to understand what was new about this. Some time ago, Prof Jaan Pelt (who is also going to be the referee of my thesis), gave a really mind-blowing lecture about non-uniform sampling. Shortly thereafter, I posted a message to the Vorbis-dev mailing list about this stuff.

    In fact, you're not limited by the Nyquist frequency when you are sampling non-uniformly, so it has some strengths in that respect. However, it has to be more to it than this for it to be news. Can anybody who understands this better than I provide any insights?

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  5. medical imaging and compression? by bokmann · · Score: 4, Interesting

    About 7 years ago, I was involved in a research project, trying to use video teleconferencing and doctors for remote diagnosis of patients.

    We found that jpeg compression of images made medical diagnosis unreliable. Hairline fractures in x-rays are exactly the kind of small details that tend to get washed away in 'lossy' compression, and the banding caused can lead to false assumptions as well.

    The article suggests that this is still a lossy compression with small amounts of data loss. I know Doctors that would take that admission as a condemnation of the technique.

  6. new 1000:1 compression scheme by Anonymous Coward · · Score: 3, Funny


    Hereby I donate the following algorithm to the public. It's called GNU-squat.

    Step 1:
    Non-uniformly sample your favorite music using just 1 bit. This will ofcourse take up at least 8 bits on your harddisk but let's not nitpick. The good part is you don't even need special hardware to sample the music, just enter if the music is loud (1) or soft (0).

    Step 2:
    Use the Vanderbilt mathematical routines to extrapolate the rest of the data, and presto: the complete song re-appears from just one bit of data.

  7. ah, there is the problem by markj02 · · Score: 5, Funny

    Doctor to patient, after looking at the reconstructed images: "Ah there is the problem. The cause of your headaches is that you have a bunch of inch-long bony spikes sticking out of your neck, plus a bunch of holes in your skull."

  8. Some folks seem to be missing the point on the MRI by fatboy-fitz · · Score: 5, Informative

    example. It was not provided to show a compression mechanism in which the original image could be compressed. It was intended to show that if you sample randomly, then their algorithm can come up with a highly accurate representation of the original. The implication here is that given current capability to sample, if you apply the new technique, you can get a better image/audio recording using their technique, than you can using the current fixed sampling interval technique, making the image more vivid, or the musical recording more lifelike than current sampling provides.

    --
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  9. Fractal compression is pure math by Leeji · · Score: 4, Informative

    Along your point, there's actually a technique that uses the self similarity of images to help you compress themselves. For example, you might have seen the "Sierpinsky Triangle." You can generate this image with a few very simple recursive move/resize/draw operations.

    Fractal compression uses this technique on abstract images. It aims to find a set of operations (sometimes very large) to generate any given input picture. It's very cool, and you can get more information (including example pictures) at this page.

    The "state of the art" of fractal compression beats JPEG compression at some compression ratios, but looses at others. It's also interesting that a fractally-compressed image has no implicit size (ie: 640x460), so it enlarges MUCH better than simple image enlargement.

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  10. Time vs. Frequency by PingXao · · Score: 5, Informative
    Classical techniques also require that the original signal be "band limited" - a technical term meaning that the signal must stay within certain, defined limits.

    This is not quite accurate. The original signal is not "required" to be band-limited. Rather, it is accepted that frequencies outside of your design bandwidth will not be captured. The signal can stray outside of the "defined limits", but should it do so that information will be lost. Furthermore, Fourier's math tells us that a signal that is limited in time is unlimited in frequency, and a signal that is limited in frequency is unlimited in time. This has important ramifications. The biggest - and most obvious - is that all man-made signals are limited in time and therefore unlimited in frequency. Ergo there will ALWAYS be information lost no matter what bandwidth you design for.

    Now to read the rest of the article - it sounds intriguing...
    1. Re:Time vs. Frequency by madsatod · · Score: 3, Informative

      You're right about the Fourier-stuff.
      But I think you misunderstood the "band limited" thing.
      When you sample you have to the filter out frequencies above the Nyquist-freq., if you want to avoid aliasing-problems.
      Aliasing comes from the mirroring of the spectrum around n*Fsample. So if you don't want your original signal to get distorted when sampling, one have to use an anti-aliasing filter, that "band-limits" the signal to below Fsample/2.
      Does this new technique mean, you can skip anti-aliasing filters?

  11. Re:Brain scans? by Hal-9001 · · Score: 5, Insightful

    Any medical imaging technique can only be so accurate, due to either machine or physical limitations. This defines a maximal meaningful sampling rate or resolution for that imaging modality. For example, positron emission tomography (PET) has a physical resolution limit of 10mm because positrons can propagate up to 10mm from where they are generated before they decay into gamma radiation that can be detected by the machine. With this technique, a doctor can get an image with better than 10mm resolution, something that the machine by itself could never do.

    BTW, sampling doesn't mean that you're guessing. The sampled data points are the actual measured values of the signal at specified points in time or space. You have to sample because there is no way that you could collect all values for the signal for all points in time or space, and there is usually a sampling rate at which point you're collecting more data than you need to accurately represent the signal.

    --
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  12. Some Clarifications by dh003i · · Score: 3, Interesting

    From what I've read, some people seem to be thinking this is some kind of "magic bullet". For example, one comment, which emanated stupidity, was titled something like, "Infinite Zooming" and the implication of the post was that it might be possible with this method to "zoom in" on an image and accurately reconstruct the image. In other words, the idea is you could zoom in on a tiny head on a photograph and accurately reconstruct all of the details.

    This, my friends, is complete nonsense. You cannot zoom in on an image and accurately reconstruct further details. To imply that this is possible is to imply that you can add accurately representative data where there was none before.

    As for "zooming technology" it is possible to better reconstruct a zoomed-in image, though not any more accurately. For example, when I go into MS Paint and zoom in, it simply blows up all the pixels as larger blocks. This clearly is not good. You could create some kind of algorithm to determine the "shapes" of sharp edges, as well as where gradients where, and scale those up when zooming in...for example, small a circle can be composed of four pixels -- such a technology would scale this up, not as four very large blocks, but as a circle.

    But this involves assumptions about what the original pattern was representative of? Was it representative of a circle, or of four large blocks seen from a distance? So you're not really adding data, but just attempting to "zoom in" on an image "better" based on a set of good assumptions which generally work.

    Such a thing could be accomplished. Indeed, it already has been accomplished -- in us. When we look at a small photograph and want to draw a poster from it, we don't draw a large, blocky, pixelated image. We are able to tell what things -- such as frecles -- are details to be scaled up in our drawing; what things are gradients -- such as a dark to light gradient going from the near to the far side of a forehead -- to be scaled up and gradiated; and what are sharp borders, to kept sharp -- such as the sides of one's face.

    However, even this amazing system we have of reconstructing larger images from smaller one's cannot add detail where there is none. If a woman is freckled with tiny freckles, they won't be visible from 10 feet away; a picture taken from that distance won't show them, and if we wanted to make a portrait of her head based on that picture, we wouldn't know to add freckles.

  13. Varying audio sample rates by dstone · · Score: 4, Interesting

    I have a question/theory about nonuniform sampling rates. Okay, sticking with a 44kHz sample rate, will you hear the differeces between 8, 16, and 24 bit samples? Yes, of course. It's common in digital audio to use 16 bit samples to save space, not because it's the ultimate sample size. (While it's arguable the 44kHz rate side of the equation is pretty darn good.) It's subjective and some ears don't need any "more" audio information to be happy, but I see the choice of sample size as more of a variable than the "provable" sufficient rate for 20kHz audio cutoff behing 44kHz. All I'm saying is that there is potentially audible information below 20kHz that isn't getting encoded and recreated not because of sample rate, but because of sample size. For example, if my source material didn't "need" 44kHz througout a song, could the sample rate be trimmed back in places while the sample size was increased? In the end, it's all just a stream of x samples per second, y bits deep. So if a new sampling technique allows us to reproportion (optimize) those two dimensionons in the same amount of overall space, it's possible that better audio will result. Thoughts?

    1. Re:Varying audio sample rates by SpinyNorman · · Score: 3, Informative

      Nope - the sampling accuracy and quantization is only going to affect the accuracy to which you can reconstruct the component frequencies. Whether or not your sampling is capturing given frequency components is a matter of the sampling rate (or more generally - as is applicable here in the case on non-uniform sampling - the minimum inter-sample delays). Higher sampling rate will only gain you higher frequency components; the lower frequency components are already going to be there unless you deliberately chose to lose them via a high pass filter.

      Regarding 16 bit vs 24 bit "samples", note that there's a difference between sampling accuracy and the number of bits to store your quantized samples. The two are only the same if you're using linear quantization and thus, for example, storing your 24-bit accuracy sample "itself" (i.e. linearly quantized into 2**24 discrete steps). Linear quantization is rather wasteful as the human hearing system does not have equal discrimination at all volume levels, so you might want to quantize more roughly at higher volume levels something like this:

      (0) (1) (2) .. (10 11) (12 13) ... (20 21 22) (23 24 25) etc

      So you could sample at 24 bits to capture additional detail at low volume and yet non-linearly quantize to store your samples in 16 bits wihtout losing that detail.

    2. Re:Varying audio sample rates by pclminion · · Score: 3, Insightful
      First of all, sampling rate implies sampling size. A "sample" is meant to represent the value of a signal over a period of time, not at an instant in time. Consider the following situation. A 44100-th of a second segment of waveform enters an ADC chip. Imagine that the signal has a very high value over this entire duration, except for a brief instant in the middle. It is at this point that the ADC takes a sample. What results is a sample which is not a very good representative of that portion of wave.

      This is why ADCs do not just sample the incoming voltage -- they integrate over a period of time, to "boil down" the voltage over that time period to an average value, that best represents what the signal was doing during that sampling period.

      Now, moving on to your point, which is to vary the sampling rate according to the characteristics of the source; this is somewhat a wasted effort, since in order to determine the source characteristics, you must perform some type of frequency analysis, or autoregression. This is intensive computation, and you would be better off spending that time doing some real compression, such as spectral quantization, or perceptual coding.

      Varying the sampling rate from sample-to-sample would be the ultimate, if it were possible to gain anything from it. Unfortunately, if you vary the sampling rate at each sample, then in order to transmit the sampled stream you must transmit not only the samples, but the duration between samples as well. In the worst case you have doubled your data rate, not compressed it.

      However, as you say, this could work wonders for the fidelity of the sampled signal. Instead of sampling at regular time intervals, we could build a predictive ADC that samples only when the predicted signal value becomes different from the actual by some predetermined amount. Then, send two values: the sample itself, and the duration since the last sample. This works because the DAC which converts the signal also does interpolation. It would be possible to keep the error arbitrarily small, no matter what the characteristics of the signal, up to the limits of the ADC chip itself.

  14. Both are right: by volpe · · Score: 5, Informative
    From Engineering Fundamentals


    The sampling theorem is considered to have been articulated by Nyquist in 1928 and mathematically proven by Shannon in 1949. Some books use the term "Nyquist Sampling Theorem", and others use "Shannon Sampling Theorem". They are in fact the same sampling theorem.
  15. Re:Short on any real details... boo! by pclminion · · Score: 3, Insightful
    Nyquist was talking about aliasing of the input signal. If you sample a 220 Hz sinusoidal wave at 440 Hz, then output it through a linearly interpolating DAC, you will hear a triangular wave. In other words, there is aliasing of the output signal.

    If you are sampling audio at 44100 Hz, then an 8000 Hz tone will only be sampled at about 5 spots in its cycle. Although the frequency information of that 8000 Hz tone is retained, the actual waveform is lost. Exactly what the reconstructed waveform will look like is up to the DAC.

    Whether the human ear can hear the difference at higher sampling rates is another question, however.