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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."

162 comments

  1. So.... by PeeOnYou2 · · Score: 1, Insightful

    Even better sound than what we now know as almost perfect? Great! Makes you wonder how much better it will get in the future, even when we have perfect sound....

    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. Re:So.... by Anonymous Coward · · Score: 0

      Yet the title says "More accurate digital tunes"

    3. Re:So.... by JimButton · · Score: 1

      Please read again, the classical sampling techniques required the signal to be band-limited, this new one claims to be able to also handle non-limited signals.

    4. Re:So.... by Anonymous Coward · · Score: 0

      It is applicable to music. Current techniques can do a good job of capturing the scalar component of sound, but not the vector component. Think non-uniform spatial sampling, rather than non-uniform time or amplitude sampling.

    5. Re:So.... by jejones · · Score: 2

      The reason the usual techniques are band limited is the problem of aliasing (as we all can remember from watching the wagon wheels go backwards in old Westerns). The limitation of our ears makes uniform sampling techniques feasible for digital audio, but that doesn't mean that the new theory isn't applicable to digital audio.

    6. Re:So.... by mateub · · Score: 1
      It seems to me that you could treat music as not-band-limited. CD's, for instance, start with the "static" assumption that the musical signal will be between 0 Hz and 22050 Hz (44.1 KHz sampling rate, right?). But most music isn't going to hit frequencies near 22KHz, so a lot of that sampling is going wasted. (Most people can't hear below 20-25Hz, but it's not worth the effort to avoid sampling that.)

      As an alternative, I could say that the music will usually be between 0 Hz and 10KHz. Now my sampling rate is cut in half, and if I'm wrong, I can iteratively adjust upward to recover the higher frequency information (if I'm understanding the basics of the paper correctly). This seems attractive to me, at least. Obviously it's too late to change the CD or DVD standards, but maybe some new music format for 3G cell phones, for instance?

      Or am I misunderstanding something here?

      adéu,
      Mateu

      --
      "And we're happy here, but we live in fear, we've seen a lot of temples crumble..." - Concrete Blonde
    7. Re:So.... by Tony-A · · Score: 2

      Actually, the bandwidth used by music is not limited. What humans can hear is limited. What audiophiles think they can hear is not so limited.
      A low-frequency note is shaped by high-frequency components. If a difference in shape of the lower-frequency can somehow be detected, then inaudible frequences still make a difference.
      Normal telephone IIRC cuts off about 3.5kHz.

    8. Re:So.... by Wumpus · · Score: 1

      Oh boy...

      If it wasn't in the original signal you sampled, you can't hear it. It's that simple. If your low frequency note was in fact shaped by some ultrasound signals that only your dog can hear, then filtering and sampling just the audible range will give you a perfect reproduction - as good as permitted by your sampling rate and the number of bits you can throw at a sample. Filtering, of course, is necessary to eliminate aliasing of high frequency components into low frequency noise.

      This is more obvious if you look at the human ear, and realize that it actually performs a transform to the frequency domain, and doesn't sample the actual wave form.

      Now, it seems from your wording that you already know that. I'm just pointing this out to people who might get confused.

    9. Re:So.... by MuMart · · Score: 1

      Not quite. Conventionally sampled audio has to be bandlimited to half the sampling frequency for the conventional Shannon interpolation technique to work. This technique promises to reconstruct audio that has been sampled without bandlimiting, which has lots of applications in audio, especially restoration of poorly sampled sound. Mart

    10. Re:So.... by Anonymous Coward · · Score: 0

      Imagine how much better their research paper will become when any missing details are filled in by their own software. Images, research papers...they're all just data.

      "There is no spoon" -- running software to fill in the details in the Matrix might alter reality. Processing details incorrectly in reality does not alter reality.

    11. Re:So.... by mateub · · Score: 1

      Oh, definitely. I didn't want to get into that topic because I thought it would confuse my main point. In the same way, we can feel "sound" below 20Hz in our bodies, so encoding this and being able to reproduce it could likewise be argued to have value, rather than just removing data below 20Hz in hopes of better compression.

      adéu,
      Mateu

      --
      "And we're happy here, but we live in fear, we've seen a lot of temples crumble..." - Concrete Blonde
    12. Re:So.... by Tony-A · · Score: 1

      The transform is actually into a frequency cross time domain. A,B,C,D fire at about the same time. The exact sequence and timing are affected by inaudible frequency components. Whether this is real or only in the imagination of audiophiles, I don't know.
      You can see a telephone line, which is smaller than the eye can resolve. With "perfect reproduction", the telephone line would vanish.

    13. Re:So.... by Wumpus · · Score: 1

      The exact sequence and timing are affected by inaudible frequency components.

      How are the affected? Can you point to a paper discussing this? It sounds interesting.

      You can see a telephone line, which is smaller than the eye can resolve. With "perfect reproduction", the telephone line would vanish.

      I'm not sure you can draw any conclusions on the auditory system from our vision system.

      Besides, the retina doesn't sample at idealized points, since photoreceptors have a surface area. That, and the wiring of phtoreceptors to ganglion cells, are different enough from the assumptions made by the sampling theorem to make me suspicious of attempts to draw conclusions from sampling theory to the vision system.

      ...which is exactly what I was doing in my previous post, only with the auditory system. Your point is well taken.

    14. Re:So.... by Tony-A · · Score: 1

      No idea as to any papers, but...
      Low note on an organ pipe (that produces no overtones). After a while you are aware that it has been there.
      Take a square wave. Lots of high-freq stuff there. Run it through a lo-pass filter. Now the starting edge is harder to pin down exactly when it happened.
      Switch from off to a 1v p-p 110Hz signal. Do same through a low-pass filter.
      If (and it's a big if) the human ear plus nerve cells has the ability to tell which overtone fired first then it's possible to discern the difference of the inaudible frequencies.
      The point with the telephone is that the wire is actually invisible in each of the photoreceptors. When something repeats, it may be possible to pull information from the total that doesn't exist in the pieces.
      Suspicious is the right attitude. We're actually pretty much in agreement.

    15. Re:So.... by Wumpus · · Score: 1

      The point with the telephone is that the wire is actually invisible in each of the photoreceptors.

      Not really. That's why I pointed out that photoreceptors don't sample at a single point. Instead, they average a small area, which makes a big difference.

      Let's look at some hypothetical telephone wires. Think of a very thin, very distant, and very bright phone wire. Photons emitted by the wire hit your eye, and get focused on your retina. The photons hit a very thin line on your retina - much thinner than a single photoreceptor. The number of photons hitting the photoreceptors will still be large enough to make each one of them fire, making your brain believe that what it's seeing is a telephone wire, as thick as the line of photoreceptors on the retina, projected back through the lens and cornea, all the way back to where the telephone wire really is. You won't know how thin the wire is, but you'll know it's there.

      The less extreme case, with an ordinary telephone wire, works just the same, except that the number of photons hitting your retina is much smaller, making them fire less frequently. If lighting conditions are bad enough, you will no longer see the wire, simply because the difference between the firing rate of photoreceptors that have the wire's image on them, and the firing rate of those that don't, is so close that it disappears into the noise.

  2. Non-uniform? by Anonymous Coward · · Score: 0
    non-uniform sampling theory

    Well, if you sample only part of the space, of course you're going to get an "improvement".

    Silly boffins!

  3. Brain scans? by Score0,+Overrated · · Score: 1


    Okay ... I don't mind them using sampling (i.e. guessing) for my CDs and movies ... but please try and be a bit more accurate with my brainscans!

    1. Re:Brain scans? by Anonymous Coward · · Score: 0
      I'm drunk but I just realised I could be the next music producer of this century!

      I'll be bringing you a lot of goth versions of NSYNC!

    2. 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.

      --
      "It take 9 months to bear a child, no matter how many women you assign to the job."
    3. Re:Brain scans? by perky · · Score: 2, Informative
      sampling is not guessing. The Nyquist sampling theorem shows that a signal that is sampled at twice the frequency of the highest frequency component in the signal can be reconstructed perfectly. With music this doesn't matter though, because humans have bandlimited hearing, so all we have to do is sample at twice the maximum frequency we can hear.

      --
      "The new wave is not value-added; it's garbage-subtracted" - Esther Dyson, Dec 1994
  4. Better Compression by sketerpot · · Score: 0, Offtopic
    This could make compression for a lot of things much better, when you don't mind the compression being lossy. Think of movie and picture files even more compressed than they are now. Then think of the reaction of the [MP,RI]AA!

    Happy downloading!

    1. Re:Better Compression by Anonymous Coward · · Score: 1, Interesting

      Actually this paper won't really make a whole big difference in compression any time soon. It's all about solving the problems that come up when you have a bunch of samples that aren't evenly spaced (that whole 'non-uniform' thing in the title), which is not really an issue for digital images, audio, or video, since the conditions for sampling those things are pretty easy to control. It does have some potential to improve algorithms for error correcting noisy signals or filling in dropped packets from a streaming signal.

      Could we use non-uniform sampling techniques for these forms of media in the first place? Could be interesting. Jittered sampling tends to mask visual artifacts (anti-aliasing); same could be true for audio. Their techniques are supported by wavelet transforms, which can get some great compression anyways. Maybe Creative will bring us the SBLiveNU, with on-the-fly variable sample rates from 1-96 khz?

    2. Re:Better Compression by SagSaw · · Score: 1

      Actually this paper won't really make a whole big difference in compression any time soon. It's all about solving the problems that come up when you have a bunch of samples that aren't evenly spaced (that whole 'non-uniform' thing in the title), which is not really an issue for digital images, audio, or video, since the conditions for sampling those things are pretty easy to control

      There are lots of comression applications for this research. (Although who knows how they would compare to current methods). Let's say I have an image I want to compress. I look at this image and notice that there are areas with high detail and areas with low detail. To the best of my understanding, the transforms used in JPEG compression require me to sample uniformly, so I have to throw out the same amount of information throughout the image. Since the transforms presented in the paper allow for non-uniform samples, I can pick and choose how many samples I throw out, and from where. This might allow me to keep most of the samples from the high-detail areas of the image, and throw out most of the samples from the low-detail areas of the image. Whether this would improve size/quality compared to current methods is unclear.

      --
      Come test your mettle in the world of Alter Aeon!
    3. Re:Better Compression by Anonymous Coward · · Score: 0

      Before WMF, Sony had better schemes and codecs - they are sitting on them - and they leave mp3 for dead

  5. Re:Good shag by Anonymous Coward · · Score: 0

    King's Cross.

  6. First ... by hey · · Score: 1

    You fill in the missing data

  7. New Sampling Techniques? by Anonymous Coward · · Score: 0

    Funny, I saw no mention of DJ Shadow in this article.

  8. Not that ground-breaking... by Anonymous Coward · · Score: 1, Interesting

    The paper does not seem that new. It basically is using more modern methods of wavelets and an adaptive filter to deconstruct digital samples. This does not differ too much from current JPEG encoding or MP3 encoding. Such techniques have been used in control systems for a while. For that matter, non-uniform sampling has been in use for a while, for example the telephone system (which the article implied used uniform sampling). The telephone system samples using a uLaw algorithm, though it does occur at regular sample intervals.

    1. Re:Not that ground-breaking... by SpinyNorman · · Score: 1

      You're confused. uLaw uses uniform sampling but non-uniform quantising, which makes sense given the human perceptual system. It's similar to using Mel coefficients as the basis of speech recognition or compression).

  9. old news! by Monkelectric · · Score: 1

    I tried a similar technique on a Statistics paper I had to write, and got an F for plagarism!

    --

    Religion is a gateway psychosis. -- Dave Foley

  10. New data compression or is it too inaccurate? by napa1m · · Score: 1

    Could this lead to new data compression schemes for non-detail critical images and other files? A sort of JPG with half detail and half math? It wouldnt be high quality but it would be a fraction of the size, perhaps yet another low-bandwidth video codec?

    Well.. one thing's for sure, if I ever have a doctor reading my brain MRI I sure as hell don't want half of it removed (neither my brain, nor the scan).

    1. Re:New data compression or is it too inaccurate? by chhamilton · · Score: 1

      Could this lead to new data compression schemes for non-detail critical images and other files? A sort of JPG with half detail and half math? It wouldnt be high quality but it would be a fraction of the size, perhaps yet another low-bandwidth video codec?

      Are you even familiar with how JPEG works? It's already half-detail, half-math... JPEG images involve a significant amount of math and statistical trickery (in throwing away data). However, all the math used in standard signal processing (image and audio compression fall in this category) make the basic assumption that the signal is sampled at a uniform rate... a lot of the current techniques wouldn't apply without tweaking...

  11. Who gets rights to the technology by kenneth_martens · · Score: 0, Offtopic

    If this is indeed a breakthrough, I hope the National Science Foundation (who is funding the research) decides to make the information free for anyone to use. The last thing we need is for them to kill the technology by attempting to retain control of it through copyrights, patents, and controlled licensing. Research like this should be published and given freely to the world community, not licensed to corporations to try to make a buck.

    1. Re:Who gets rights to the technology by Anonymous Coward · · Score: 1, Funny

      Research like this should be published and given freely to the world community, not licensed to corporations to try to make a buck.

      Maybe a compromise is possible: "We hereby publish our research with half of the data removed randomly, see if you can recover what's missing."

    2. Re:Who gets rights to the technology by Anonymous Coward · · Score: 0

      Thanks. Now I have a template post that I can cut-and-paste into any Slashdot story about any new technology.

    3. Re:Who gets rights to the technology by Anonymous Coward · · Score: 0

      You're welcome.

    4. Re:Who gets rights to the technology by Anonymous Coward · · Score: 0

      keep dreaming you socialist fool. Researchers and investors should be able to garner the profit in this intellectual property for the amount of risk, time and effort used to derive this breakthrough.

  12. Re:Better Compression-Video ogg. by Anonymous Coward · · Score: 0

    Are there any patents already taken out? If so then we could have before everyone else the video equivilant of ogg.

  13. See? by Anonymous Coward · · Score: 0

    Bible-thumping doctors are always saying how "perfect" the human DNA is, how it's the "world's most perfect code," which I, as a Hacker, should understand.
    Yeah, bible-thumpers,
    WHERE'S THE ECC?

  14. Decompression? by dzym · · Score: 0, Offtopic

    Is this how ZeroSync's decompression algorithm works?

    1. Re:Decompression? by Anonymous Coward · · Score: 0

      no

    2. Re:Decompression? by ethandoesntknowmuch · · Score: 1

      ZeroSync is not real. It's a lie.

    3. Re:Decompression? by Anonymous Coward · · Score: 0

      what would you know? Do you even know anything about the multidimensional mathematical entities they developed?

  15. 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 hrieke · · Score: 2, Informative

      Accually the width value of 80 for CGA display goes back to the punchcard days, not as you state in trying to keep memory and bandwidth costs down.

      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?

      If these guys have the ability to return useful data from non reporting areas I can see a whole range of non niche applications - and real word applications where data recovery would be useful.

      --
      III.IIVIVIXIIVIVIIIVVIIIIXVIIIXIIIIIIIIVIIIIVVIIIV IIVIIIIIIVIII...
    2. Re:Some useful niche applications by normandr · · Score: 1

      MRI (magnetic resonance imaging) involves no moving parts. I think yon meant "Spiral CT"

    3. Re:Some useful niche applications by michaelmalak · · Score: 2
      MRI (magnetic resonance imaging) involves no moving parts. I think yon meant "Spiral CT"

      Hmm... you're right. The "spiral" in Spiral MRI evidently refers to frequency/amplitude space rather than physical space.

    4. 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.
      --
      20 January 2017: the End of an Error.
    5. Re:Some useful niche applications by Gordonjcp · · Score: 2

      Actually, digital video uses non-square pixels.

      PAL DV has pixels slightly taller than square, and NTSC DV has pixels slightly shorter than square. It makes editing on a square-pixel PC monitor a bit wierd, because the images look stretched or squashed.

      I had to recode an NTSC DV tape for PAL once, which was a total PITA. Different frame rate, different resolution, horrible smeary colour... Never again.

  16. Seems pretty similar to... by cliffy2000 · · Score: 0, Redundant

    MP3 (and Ogg Vorbis, etc.) technology. I mean, mathematical non-lossless compression technology... different algorithm, same purpose. (Side note: It seems ironic that as storage space grows, this becomes less and less necessary.) In any case, it seems to indicate a trend: the ability to get the maximum from a minimal amount of data.

  17. 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.

    --
    Toronto-area transit rider? Rate your ride.
    1. Re:Nyquist, not Shannon by td · · Score: 2, Redundant

      Nyquist formulated the sampling theorem in 1928, but never proved it. Shannon provided the first proof in 1949.

      --
      -Tom Duff
  18. As a CpE (or ECE, or Comp. Eng.) this screws up... by Hollinger · · Score: 1

    As a Computer Engineering Major, this tosses a significant portion of my degree out the window, but, I suppose, it's a good thing. Aliasing (Java) and Folding (no link) were always a pain.

  19. 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?

    --
    Employee of Inrupt, Project Release Manager and Community Manager for Solid
    1. Re:Anybody understand what's new? by Anonymous Coward · · Score: 0

      It's all about the reconstruction algorithms and theory, dude.

    2. Re:Anybody understand what's new? by matrix29 · · Score: 1

      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 [xiph.org] about this stuff.

      So this is a classical expansion on the variable bit rate sampling as is done on MP3 files now. The only difference is that this is done on bitmap files in place of sound files.

      --
      "Face it, a nation that maintains a 72% approval rating on George W. Bush is a nation with a very loose grip on reality.
    3. Re:Anybody understand what's new? by Alsee · · Score: 2

      Can anybody who understands this better than I provide any insights?

      We have a lot of powerful tools (such as Fourier transforms) for analizing precisely sampled data. For example sound is often sampled at a precise frequency - about 44khz.

      The problem is that sometimes the data available isn't spaced regularly. This makes most analysis techniques throw fits. He's come up with tools to ues here that do a good job of taking irregular data and returning a very good estimation of the values everywhere.

      If you're familiar with Fourier transforms, this is a more generalized version.

      -

      --
      - - You can't take something off the Internet! That's like trying to take pee out of a swimming pool.
    4. Re:Anybody understand what's new? by KjetilK · · Score: 1
      Hm, not really, you see, irregulary sampled data isn't a problem, it's a Good Thing[tm], and it has been realized long ago that it is a Good Thing[tm]. Obviously, Fourier transforms is an important part of this.

      So, really, what is new about this is still kinda fuzzy... :-)

      --
      Employee of Inrupt, Project Release Manager and Community Manager for Solid
    5. Re:Anybody understand what's new? by Alsee · · Score: 2

      Hm, not really, you see, irregulary sampled data isn't a problem

      I'm not an expert in the field, so I just did a google search, and as far as I can tell, all of the first 30 links state uniform sample spacing and/or specificlly say that irregular sample spacing is a problem.

      -

      --
      - - You can't take something off the Internet! That's like trying to take pee out of a swimming pool.
    6. Re:Anybody understand what's new? by KjetilK · · Score: 2
      I'm not an expert either! BTW, the expert I mentioned in my post was aware of the Slashdot article!

      Anyway, try a better search.

      But of course, it depends on what you mean by "problem".

      --
      Employee of Inrupt, Project Release Manager and Community Manager for Solid
    7. Re:Anybody understand what's new? by KjetilK · · Score: 2
      No, no, no. It has nothing to do with it.

      Variable bit-rate, if I have understood it correctly, is about say that you have a period in the sound file that is very quiet, then you don't need many bits to represent it well. Therefore, you don't use many bits per sample, and you save some space.

      You still sample regularily, for example, if you sample with a 44.1 kHz sampling frequency, then you take a sample every 0.023 milliseconds, exactly.

      This stuff is different. Instead of taking a sample with exactly the same interval, you sample at random, or you sample every now and then. The number of bits you have for each sample is a completely different matter, that may or may not be variable.

      The funny thing is that you can actually use this to reconstruct the signal much better in many cases, which is pretty counterintuitive when you think about it! (until you've thought much about it, because then it makes a lot of sense... :-) )

      --
      Employee of Inrupt, Project Release Manager and Community Manager for Solid
    8. Re:Anybody understand what's new? by RockyJSquirel · · Score: 1

      The article mentioned algorithms. That's a red flag that someone did some nearly obvious math and applied for a patent.

      Or maybe there really IS a good new algorithm there. I doubt it, one of the example's looked awful. I'm sure I could do THAT well.

      Rocky J. Squirrel

    9. Re:Anybody understand what's new? by guygee · · Score: 1

      A good example of a situation where non-uniform sampling is a "good thing" comes up often in computer graphics. Imagine trying to model the terrain surface in the vicinity of Denver, including the nearby Rocky mountains. It the only criteria of a good model were accuracy of the model and efficiency of the representation, then obviously non-uniform sampling (e.g.triangulated irregular network) is a "good thing, since fewer sample (triangles) are required to accurately model the gentle rolling plains then are required to accurately model the ragged height field of the Rockies. Thus, using non-uniform sampling, we can use fewer total samples to obtain the same or better overall accuracy of our terrain surface representation.

  20. 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.

    1. Re:medical imaging and compression? by Anonymous Coward · · Score: 0

      Dude, maybe the idea is to get a better zoom. Like, you take the dots which normally end up being just a porridge of pixels, and then you run it through this and get a hugely improved zoom image. Also of interest I think would be military uses, spy satellites for example. Also, for astronomy. If it can indeed be used to artificially improve zooming.

    2. Re:medical imaging and compression? by SagSaw · · Score: 1

      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.

      From what I read, the paper does not represent a compression technique, but a better way to fill in the missing data between samples, especially when the samples are nonuniform, or samples are missing. This would allow you to remove data for storage/transport and recover a similar image later, or as it would probably be used with medical imaging, to recover data lost during the imaging process due to sampling and quantizing error. In the second case, the fracture shouldn't be lost if done correctly.

      --
      Come test your mettle in the world of Alter Aeon!
    3. Re:medical imaging and compression? by Alsee · · Score: 2

      The article suggests that this is still a lossy compression with small amounts of data loss.

      Nope. Re-read the article. It's not a compression scheme. If anything, it's the reverse, and expansion scheme. It takes all the available data and does a good job of filling in the gaps. It even works when the available data isn't arranged nice and neatly.

      Used in the right context it would make things like hairline fractures MORE visible. You wouldn't usually use it in video teleconferencing though.

      -

      --
      - - You can't take something off the Internet! That's like trying to take pee out of a swimming pool.
  21. 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.

    1. Re:new 1000:1 compression scheme by ethandoesntknowmuch · · Score: 2, Funny

      Do you work for ZeroSync?

  22. 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."

  23. 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.

    --
    I'm better, because I'm bigger
  24. near-perfect zooming by ShadeARG · · Score: 1, Interesting

    If it is possible to use these mathematics techniques to replace all of the unknown parts of an image, then why not resize the image a few times larger than the original and save the random parts of it? This would allow the algorithms to fill in even more detail to each image relative pixel. Upon resizing the image to 2x the original size, you would find much better clarity and precision than just resizing the image without.

    On a side note, you could apply random color-relative noise on to the entire zoomed image before you save the random parts, then it might pick up the slack of the algorithm placing the same bordering colors over the unknown pixels.

    If they consider digital music captured with this set of algorithms near-perfect, then near-perfect zooming is just around the corner.

  25. Re:Anybody understand what's new?-PDF by Anonymous Coward · · Score: 0

    http://citeseer.nj.nec.com/rd/72823020%2C302362%2C 1%2C0.25%2CDownload/http%253A%252F%252Fciteseer.nj .nec.com/cache/papers/cs/14699/http%253AzSzzSzatla s.math.vanderbilt.eduzSz%257EaldroubizSzSIAMRV.pdf /aldroubi00nonuniform.pdf

    Go read the PDF then get back with us.

  26. gives you an answer, not necessarily the right one by markj02 · · Score: 2, Interesting
    When you reconstruct a function from sampled data, there are an infinite number of possible reconstructions. That issue is resolved by making certain assumptions about the functions you are reconstructing. An assumption of band-limited data is useful because it approximates what happens in many communications systems and (perhaps more importantly) because it leads to simple and efficient algorithms (some comment about only having hammers and everything looking like nails is in order).

    There are already many other methods for reconstructing functions from sparse, non-uniformly sampled data, so this paper doesn't solve an unsolved problem. Rather, it provides one more solution under a set of assumptions that are mathematically a bit more like those of the original sampling theorem.

    Will it be useful? That's hard to tell at this point. I think it will take a lot more work to figure out whether this method is any better than existing methods on real-world problems, whether its application can be justified in real problems, and whether it leads to algorithms that are practical. It may also turn out that the method is closely related to methods already in use in other fields; for example, the kinds of function spaces they study have received some attention in neural networks, but the authors cite no papers from that work and may not be aware of it.

  27. 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.

    --
    It all goes downhill from first post ...
    1. Re:Fractal compression is pure math by maetenloch · · Score: 2, Interesting

      Fractal compression is a cool idea, and it can achieve incredible compression rates. Unfortunately it hasn't quite panned out in the real world.

      One main problem has been that no one has found an efficient way to create the PIFS functions for an arbitrary image. So fractal compression can take a long time and is non-deterministic (i.e. you can't tell ahead of time how long it will take).
      Another problem is that Barnsley et al. hold patents on many of the techniques used. Until its performance makes it a clear winner, why pay royalties.

      It's been a couple of years since I paid close attention fo fractal compression, but I haven't heard of anything that changes the above problems.

  28. 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 Fourier · · Score: 2

      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.

      Well, that's not entirely accurate either. The presence of frequencies outside of the design bandwidth will lead to aliasing. The reconstructed signal will have additional low-frequency energy that should not be there.

      In practice, we often use analog "anti-aliasing" lowpass filters to band-limit the signal before sampling.

    2. 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?

    3. Re:Time vs. Frequency by pslam · · Score: 2, Interesting
      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.

      Two of the other replies point out that this isn't quite right - the frequencies outside of nyquist just alias. However, this can actually be used to your advantage if you know that a signal lies within a narrow band of frequencies centered around a high frequency.

      For example, you can perfectly sample a signal confined to 1.0-1.1MHz using a sampling rate of just 200kHz, instead of 2.2MHz. What's even more interesting is that you can play this 200kHz sample back and get the same signal in the 1.0-1.1MHz band you had originally, but along with aliases all over the rest of the spectrum. In this case, you need bandpass anti-aliasing filters and not lowpass bandlimiting ones.

    4. Re:Time vs. Frequency by madsatod · · Score: 1

      Nice...
      Didn't know that.
      So your signal BW just have to be confined to Nyquist-interval, for this to work!?

      /Mads

    5. Re:Time vs. Frequency by twalton · · Score: 1

      "In this case, you need bandpass anti-aliasing filters and not lowpass bandlimiting ones."

      You also need A/D converters specified for the top of the actual "carrier" frequency band, i.e., with sample-hold times and aperture-jitter specs suitable for - in your example - the 1.1 MHz top-end frequency of the original signal. This is where this apparent miracle meets the real world - and why there are still analog mixers in the front ends of digital radios

    6. Re:Time vs. Frequency by markmoss · · Score: 2

      The article being extremely light on technical details, I think what was meant is that by non-uniform sampling intervals (deliberately jittering your sample time, but _knowing_ the actual time each sample was taken), you can dodge the aliasing problem. That is, although your average sample rate is (say) 20K/sec, so a 12 KHz sine wave would alias to 2KHz, you have samples taken at other intervals that reveal that a 2KHz wave won't fit.

      I'm not sure if this is new at all. Some digital scopes will attempt to hit higher effective bandwidths by shifting the time of starting sampling at each sweep, so as to fill in between the dots of the first sweep. This only works if the signal you are measuring is absolutely regular, and the triggering (detection of the start point in the signal) is perfect...

  29. Re:Some folks seem to be missing the point on the by markov_chain · · Score: 1

    If their point is that they can better reconstruct the original image from non-uniform samples, then it would have been more interesting to see a comparison of reconstructions: in particular, take one MRI images with random pixels missing, and the second MRI images with the same number of missing pixels but arranged in a regular pattern, such as a grid.

    However, I suspect their point is that they can reconstruct the original at all with non-uniform sampling. This is useful in cases when it is not feasible to obtain fixed samples.

    --
    Tsunami -- You can't bring a good wave down!
  30. Nothing new here. by gewalker · · Score: 2, Insightful

    I was making up missing data for lab reports twenty years ago. It filled in the gaps well enough to fool the teachers :)

    News article was, as usual, totally lacking in technical details. But they did link to technical articles at the bottom of the story.

    NON-UNIFORM SAMPLING AND RECONSTRUCTION IN SHIFT-INVARIANT SPACES.

    I skimmed the technical article (heavy math alert), and the results seem to be along the lines that: given an irregular (and possibly noisy) sample of data, reconstruct a
    function that gives smoothed (continuous, not discrete) approximation for entire data set.

    There is some nice mathematics that make it suitable for such purposes. The algorithms are selected to limit number of terms and guarantee convergance, and are computationally efficient. If you think of it as fancy interpolation, you are not far off the mark from what I saw.

    This is not to disparage the efforts here (it looks to be quite useful in several domains), but it is a technique for generate a smooth, continuous function to represent a set of non-uniform samples. It cannot magically find missing results not were not evident in the limited sample data.

    The author

  31. Ah, good! by kcbrown · · Score: 1

    I was hoping someone would finally come up with a good method to use in lunzip (see this for more details. In short, it's a superior compression utility, at least for certain jobs, like prepping your computer for the FBI).

    --
    Use 'slashdot stuff' in the subject line in any email you send me if you want to get past the spam filter.
  32. 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.

  33. Maybe this should be combined... by jeti · · Score: 1

    Look at the sample image. Most of the details are reconstructed correctly. But some errors like the spikes are obvious.
    If you use a classic technique like interpolation through splines, diff the images and remove the gross errors created by this new method, the result might be quite convincing.

  34. Short on any real details... boo! by tomstdenis · · Score: 1

    For starters, if human threshold of hearing tapers off at around 20khz (its actually closer to 18khz where at 20khz most audio is fully attenuated... but anyways)...

    How will a "new and improved" method of sampling help me hear audio I can't hear anyways?

    Nyquist proved that with uniform sampling at 2/T you will lose no spectral information between DC and 1/T.

    Somehow I think this is more "Magic Ph.D" material than real science.

    Tom

    --
    Someday, I'll have a real sig.
    1. Re:Short on any real details... boo! by Anonymous Coward · · Score: 0

      Current PCM techniques are good for capturing the sound AT A SINGLE POINT IN SPACE.

      Non-uniform sampling might be useful in improving the capture and reconstruction of the sound field of a room. We need to capture with a finite number of microphones, and reconstruct with a finite number of speakers. And the spacial distribution differs between microphones and speakers, and neither is uniform.

    2. Re:Short on any real details... boo! by tomstdenis · · Score: 1

      Ah ok that makes more sense. The article [with the funny MRI picture] is very misleading. They showed magical improvement within a single sample set [the picture].

      Despite what they think if your source is of low quality no amount of math will increase the accuracy/resolution. You can only make it more visually appealing.

      Well are there any papers to read on the subject online? Anyone have citeceer links they want to lend me?

      Tom

      --
      Someday, I'll have a real sig.
    3. Re:Short on any real details... boo! by Darlington · · Score: 1

      You make a good point, but there's actually a considerable amount of debate in the recording world as to the acceptability of 44.1 KHz / 16-bit audio (aka CD-quality). My own hardware records up to 48 KHz / 24-bit, and there's gear out there that will go up to like 96 KHz, for making DVD-audio or some freakin' thing.

      Now, 44.1 KHz / 16-bit is just fine for me, but I can at least consider the idea that there are things happening in the frequencies above 22.05 KHz (the top frequency 44.1 can record) that have some affect on us even if we can't consciously hear them. Well, fine, but I'm not going to record everything at 96 KHz and increase all my audio file sizes to 218% of their current size just so that SuperAudioFileMan can hear the dog whistle in the background. But if I can get a variable sampling scheme that will grab some extra frequencies when the source material's spectral content warrants it, and maybe even sample below 44.1 when the tympani solo comes along, that works for me and is at least an improvement for the hypertreble freaks.

    4. Re:Short on any real details... boo! by decefett · · Score: 2, Interesting

      I was reading somhere (can't remember where) that although we can't hear above 20Khz, sounds that are above that range will lower in frequency when they bounce around the room and fall into some peoples hearing range.

      CD's sampled at 44khz miss some of these sounds and that is what audiophiles complain about when they say digital audio sounds flat.

      --
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    5. Re:Short on any real details... boo! by Squiffy · · Score: 1

      Don't forget that there's more to digital audio than the faithful representation of a mono signal. Differences between the signals of two or more channels contribute to the overall spatial image of the recording. Sometimes this phenomenon takes the form of a phase difference between two channels in the range of a few microseconds, which is easier to reproduce at higher sampling rates. More importantly, problems with A-D and D-A conversion are more easily solved at higher sampling rates.

    6. 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.

    7. Re:Short on any real details... boo! by mclinc · · Score: 1

      Errr thats why there's a reconstruction filter with a sinc function!!

      --
      "Oh no, not again"
    8. Re:Short on any real details... boo! by pclminion · · Score: 2
      So? Reverse the situation. Suppose the input signal was triangular and is sampled at the Nyquist rate. Then the reconstructed signal will be sinusoidal.

      The point is, the Nyquist rate tells you the highest spectral component that can be sampled without aliasing. But a triangular wave has frequency components higher than this threshold. These components will be lost in the sampling, and the waveform will not be preserved, although its spectrum will be -- up to the Nyquist rate.

    9. Re:Short on any real details... boo! by heinzkeinz · · Score: 1

      The quality of the input signal is not a result of the DAC, but of the filtering afterwards.

      If you have a perfect filter (they don't really exist, obviously), one can sample a 220 HZ signal at 440 HZ and recover the signal perfectly with an ideal low pass filter with a cutoff of 220 HZ.

      Practically, a samplign rate of 3 or 4 times the bandwidth (maximum frequency) is adequate to use real filters to recover the signal perfectly.

      When the signal exits the D/A it is stepped, and the low pass filter interpolates those steps into the actual, perfect sinusoid.

      No one uses linear interpolating D/A's for this reason. All D/A conversion is followed by subsequent low-pass filtering. Even your speakers act as low pass filters, and in many cases suffice.

    10. Re:Short on any real details... boo! by Anonymous Coward · · Score: 0

      What do you mean by saying that the waveform will not be preserved but the spectrum will be? They are just two equivalent representations of the same signal, so if one changes, so does the other. When sampling a triangular signal, its frequency components above the Nyquist rate are aliased and appear in the spectrum of the sampled signal.

  35. Idiot by EggplantMan · · Score: 1
    All sorts of ECC's, CRC's and so on are designed to stop a certain percentage of errors overall, and also to prevent certain types of errors the most. This system is analogous to the system we have in our bodies.

    We have enzymes called nucleases whose job is to repair specific types of DNA damage. We have nucleases that repair uv damage (in the form of thymine dymers) to our DNA for example. Anyhow, before you complain about human biology, I suggest you RTFM and take a Bio course beyond highschool level.

    --

    ?-|||-----x<*))))><
  36. Why compression research continues by yerricde · · Score: 2

    (Side note: It seems ironic that as storage space grows, this becomes less and less necessary.)

    Compression research continues because in the domain where latency is less than one minute (that is, not FedEx), data communication throughput does not increase nearly as quickly as storage space. Sure, you have 100 GB to store uncompressed images and audio, but how are you going to transfer the information to another computer?

    --
    Will I retire or break 10K?
  37. Nyquist conjectured it; Shannon proved it by yerricde · · Score: 1

    the guy who developed the Uniform Sampling Theorem [wolfram.com] was Nyquist, not Shannon.

    Nyquist conjectured it in 1928; Shannon proved it in 1949. Many texts split the credit, calling it the "Nyquist-Shannon sampling theorem."

    --
    Will I retire or break 10K?
  38. 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 donglekey · · Score: 1

      Very keen and in fact techniques like that are used in Vorbis sound compression. It has lower frequencies separated from higher, and the higher are recreated using a number to set the sensitivity and a number to set the sample. At least that is how I took it from the people in #vorbis.

    2. Re:Varying audio sample rates by perky · · Score: 1
      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?


      interesting idea. The reason that we use 44kHz as the standard sampling rate is that most people's hearing ferequency cutoff is at about 20kHz, and hence the Nyquist sampling theorem shows that we need to sample at 40kHz. add a little bit to account for the fact that anti-aliasing filters aren't infinitely steep and we get 44kHz. So the real question is, in music are there blocks in which the highest frequency is below 20kHz? Then ask whether the reduced quantization brought by using higher sample sizes audible?


      and audiophiles care to comment?

      --
      "The new wave is not value-added; it's garbage-subtracted" - Esther Dyson, Dec 1994
    3. Re:Varying audio sample rates by Anonymous Coward · · Score: 0

      Well, like I wrote in another comment these 44kHz and 16bit sampling are not really adapted to sampling "sounds"- they are more like a brute force method. We increased the accuracy and frequency until we don't hear a difference any more- that doesn't really sound scientific.

      Of course we have the theorem that with 44kHz sampling rate the highest frequency that can be represented is 22kHz and this should be enough for most of us, but still it's not like how we "listen"- our hearing works different- we don't just discretize the signal in time.

      Then for the bits: This is disretization in voltage, also a brute force method. Any discretization will introduce round-off errors that will result in random noise. We just use such high accuracies (16bit) such that the signal-noise ratio is high enough.

      But does that make sense? Not really. Do we have a solution- eh- well, that's why ogg and mp3 work so well... These are more reasonable representations of sound and represent sound non in "equidistant sampling points" but in a frequency domain!

      Thus variation of sampling rates won't help you, and anyway ogg and mp3 are far more advanced (!!) than that (so what's the news at all??). What we really need is a hardware implementation of ogg and mp3 that (as the hardware still needs to discretizes in time and voltage) use far higher accuracy at input (to absolutely minimize roundoff-error) but then give suitable representations of sound to the software.

      *This* is where we should get to and what's far more advanced, at least speaking of audio!

    4. 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.

    5. Re:Varying audio sample rates by Anonymous Coward · · Score: 0

      "and audiophiles care to comment?"

      It would be better to get comments from people who actually know what they're talking about. Comments from people who think that using a marker on the outside edge of a CD increases sound quality are worse than worthless.

    6. Re:Varying audio sample rates by Anonymous Coward · · Score: 1, Interesting

      Interesting post. If you push this line of reasoning, you come up with high-level compression methods like MP3. Think about it-- all MP3 says is, instead of representing the next one second of audio as 44100 16-bit numbers, represent it as some function of time f(t), which (if you're lucky) takes much fewer bytes to store. There is no reason that you couldn't encode very high frequency sounds this way, say even higher than 96kHz. However, I think the real problem is the lack of such quality on input.

    7. 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.

    8. Re:Varying audio sample rates by nusuth · · Score: 2
      All current high compression ratio audio comressors work in frequency space, and to some extend, perception "space." In frequency space if some frequencies are non existant, they are not encoded. It doesn't really matter what your original sampling frequency was.

      Also 16 bits is quite enough, although not very well used. Nowadays most CDs are published with high average volume to have sort of an upper hand in broadcasts (check classic music titles for comparison), a better approach is maintaining only the peaks of music near to highest representable number, the high-average volume approach severly limits dynamic range. 65535 different volume steps is quite enough for human ear, you are not supposed to hear any difference beyond that for processed music (for raw recordings, it is better to have higher resolution.)

      --

      Gentlemen, you can't fight in here, this is the War Room!

    9. Re:Varying audio sample rates by Andux · · Score: 1
      Actually, "half the sampling rate" is a bit too optimistic. You could, theoretically, record a 22050Hz wave as 44100Hz, if the peaks and troughs of the wave were exactly in sync with your sampling. Realistically, you start getting diminishing returns at around (IIRC) 1/4 the sampling rate.

      While decreasing the sample rate would give you some savings, if you tried to get a smaller file size than an MP3, your maximum frequency response would probably be less than 3000Hz.

      --
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    10. Re:Varying audio sample rates by perky · · Score: 2, Informative

      Realistically, you start getting diminishing returns at around (IIRC) 1/4 the sampling rate.

      That's not true. The whole point is that if a signal is sampled at frequency f, then it can be reconstructed perfectly if its bandwidth is less than f/2. Go learn the maths instead of making vague statements that you think must be right intuitively, but which you actually don't know about.

      --
      "The new wave is not value-added; it's garbage-subtracted" - Esther Dyson, Dec 1994
    11. Re:Varying audio sample rates by jmv · · Score: 2

      A "sample" is meant to represent the value of a signal over a period of time, not at an instant in time

      No, no, no. A sample is an instantanuous value, not an integration. The reason why sampling is not sensitive to very short (compared to sampling frequency) is that there is normally a (anti-aliasing) low-pass filter before the sampling operation.
      As for transmitting the value and "time it stays the same", I'd suggest you first get more familiar with the sampling theory before innovating...

    12. Re:Varying audio sample rates by Andux · · Score: 1
      If you know a method for turning the data points 0, 0.7, -1, 0.7, 0 -0.7, 1, -0.7, 0, 0.7, -1... into a perfect analog reconstruction of a 16537Hz sine wave, I'd love to hear about it.

      And my point about resampling vs. MP3 still stands, regardless.

      --
      (Do not sign anything.) -- Fell, Planescape: Torment
    13. Re:Varying audio sample rates by Mr.+Asdf · · Score: 1

      good idea. I don't claim to be an expert in this, but I believe in order for this to work you would have to sample the signal multiple times simultaneously with the different bandwidths (or have some sort of master signal which you could resample). For example, imagine 3 A/D converters at 0-44kHz, 0-22kHz and 22KHz-44KHz. If both the higher and lower frequencies contain data, then use the 0-44kHz data. If the 22KHz-44kHz is empty, than a flag would be inserted into the "data" to use the 0-22kHz data for this time period. BUT, in order to get 16 bits out of the lower 0-22kHz data you would need to have 16 bit accuracy in the range, or put another way, 32 bit converters throughout, which begs the question of simply storing the data as 32 bit in the first place, and then using lossless compression for the parts without the higher frequencies. So, in effect, I believe your idea to be a good one, but implementing it for realistic applications might better be served by oversampling, and then dithering or compressing to the desired storage size. IMHO. As far as whether or not we or audiophiles could hear the difference, this is an on-going debate. I would say that there are very few people in the world who, played music recorded at 16 bit and the exact same music at 32 bit (or 1000 bit for that matter), could distinguish them. Many people claim to be able to hear the difference, but put them in a room with high quality audio components and play one or the other at random 50 times, I'd bet that most them couldn't get it right much more than half the time. 16 bit 44.1kHz audio is pretty damn good.

    14. Re:Varying audio sample rates by jovlinger · · Score: 2

      That's actually not true. The problem is phase. Imagine a sine wave f of frequency 1, so that f(0)=0, f(0.25)=1, f(0.5)=0, f(0.75)=-1, f(1)=0.

      now if you sample it at frequency 2, you will get a great reconstruction if you sample at time 0.25 and 0.75. However, you will get a much worse reconstruction if you sample at time 0 and 0.5. The phase interaction between the samples and the signal become more noticiable the closer the signal is to the nyquist frequency.

      Now I think you owe the previous poster an apology. A little humilty wouldn't be out of place.

    15. Re:Varying audio sample rates by pclminion · · Score: 2
      I think we are just using different terminology, not talking about different concepts. I just meant that the signal is smoothed out over the sampling period, and the sample value is representative of the entire period, not just a momentary voltage. I certainly did not mean "integrate" in the sense of an integrating filter.

      I wasn't talking about "the time the signal stays the same," I was talking about the time period over which the prediction error reaches some threshold value. For example if I am using a first-order linear predictor for digital-to-analog conversion, and the signal changes linearly from 0 to 1 over a period of 1 second, then I only have to send two samples during that one second period in order to completely describe the behavior of the signal.

    16. Re:Varying audio sample rates by jmv · · Score: 2

      First, the anti-aliasing filter doesn't just "smooth the signak over the sampling period", it has a much wider effect. Theoratically, it would be a perfect low-pass filter: "sin(t)/t", which has an _infinite_ response (Of course, practically we approximate infinity with something smaller ;-) ).

      As for using a "first-order linear predictor for digital-to-analog conversion", the cost/complexity of building a good analog linear predictor would far exceed any gain you'd otherwise have...

    17. Re:Varying audio sample rates by pclminion · · Score: 2

      Thanks for enlightening me. I'm fairly familiar with the mathematical side of DSP but I don't know much about what's actually happening in the guts of an ADC/DAC. So are you saying that the input signal is filtered by a perfect low-pass filter over the entire time domain, instead of on a per-sample basis? I suppose this makes sense, doesn't it :)

    18. Re:Varying audio sample rates by BlaisePascal · · Score: 1
      It seems to me that if you have a sine wave with a frequency (or bandwidth) of 1, and you sample it at a sampling rate of 2, then you are expecting more than the sampling theorem will give you. What's the waveform that samples at f(0)=0, f(pi) = 0, f(2*pi) = 0, f(3*pi) = 0, etc, and no frequency greater than one wave every 2*pi? The only choice is c*sin(k*x), where c is 0 or k is 1. What's the waveform that under the same condition samples f(0)=1, f(pi)=-1, ... f(n*pi)=-1^n? The only choice is cos(x).


      That's what Nyquist gives you.

    19. Re:Varying audio sample rates by markmoss · · Score: 2

      It can be reconstructed perfectly -- if it continues repeating itself exactly forever at a rate less than 1/2 the sample rate. Because if the repetition rate is f/sec and the sample rate is (2f+1)/sec, then eventually the samples will cover every part of the waveform. But real music doesn't work this way (except maybe for Yoko Ono and bad church choirs), it's continually changing, and anything too close to 22KHz may not be adequately rendered in the number of samples taken before the sound ends or changes.

      For an extreme example, consider a 21.9KHz tone that only goes for 1 cycle. The sample received may be to points at the top and bottom, in which case reconstruction will be pretty close. Or it may be two points at (almost) the zero crossing, so it appears that there is almost no sound.

    20. Re:Varying audio sample rates by jmv · · Score: 2

      The exact LP filter is implementation-dependent, but it should cut everything that's above half the sampling frequency. Then, "instantanuous" sampling is made.

      A sampled signal is represented by equally-spaces impulses (delta) of various amplitudes, which is the same as multiplying the low-passed signal by an impulse train.

  39. Hey! I know that guy... by DaedalusLogic · · Score: 1

    On a totally unrelated note, I took differential equations from Akram Aldroubi. On the first day of class when we all sat down he said, "Welcome to French 101." Scared the hell out of a class full of engineers that haven't seen anything to do with humanities/arts courses in a while. If you ever have to take math at VU this guy is really good at explaining it.

  40. 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.
    1. Re:Both are right: by s20451 · · Score: 1

      Thanks. Who would have thought that Slashdot would be educational?

      --
      Toronto-area transit rider? Rate your ride.
  41. Choosing the perfect sampling method by Anonymous Coward · · Score: 2, Interesting

    There are quite some examples in math how non equidistant sampling methods can vastly improve the order of accuracy, let's think about quadratures (numerical Integration):

    Integrating a function f(x) from a to b means measuring the area below the graph. So the first estimation would be to split the interval from a to b into equidistant parts and sum up the area of the rectangles below or over the graph (that would be about f(x_n)*h, where h is the width). This method is called Riemann-Sums or iterated Trapezodial-Rule.

    But you could also try to plot piece-wise polynomials through these equidistant points and calculate the areas below. This would yield better (order) results; these methods are then called iterated simpsons or millne rules. But if you go higher than polynomials of 4th degree, you will get to methods that could compute negative integrals of positive functions, which does not make sense. The reason is that high order polynomials tend to "oszillate" or "run out of bonds" at the end of the intervals. Thus these "Newton-Cotes" methods of equidistant sampling points are of limited capabilites...

    But if you drop the assumption that you need to take equidistant (uniform) sampling points, you will get to far better methods: With Gaussian Quadratures the sampling points are far more dense at start and end of the intervals and thus the interpolating polynomials yield far better order results!

    Thus if you know what you are going to use your data for, then you can always find better sampling methods to optimize for your needs- IMO it really doesn't make sense to simply sample the voltage of the signal at equidistant time frames when trying to digitally represent sound! Where as "lossy compressions" like ogg or mp3 drop information that is less interesting, this equidistant 44kHz sampling just drops anything that does not fit into this sampling; it's kind of a "brute-force" method. And if you then compress to ogg or mp3 it's the same problem like why you should never convert mp3s to ogg... It can (and will) only get worse.

    If you are interested in that quadrature methods then read "Numerical Analysis" by "Kendall E. Atkinson" Chapter 5.

    1. Re:Choosing the perfect sampling method by perky · · Score: 1

      equidistant 44kHz sampling just drops anything that does not fit into this sampling

      but unless you first oversample and then selectively reduce the sample rate you do not know which are the "detailed" parts, which warrant a higher sampling frequency and which aren't. Secondly, since you refer to music, human hearing is bandlimited anyhow, so there is no point reconstructing freequencies outside of our perceptual range.

      --
      "The new wave is not value-added; it's garbage-subtracted" - Esther Dyson, Dec 1994
    2. Re:Choosing the perfect sampling method by Anonymous Coward · · Score: 0

      You are assuming that every piece of music is completely unique. While this may be true for every piece of music, it is certainly not true of a CD by Britney Spears.

      A technique such as this would be really useful in a situation such as MRI or a picture of a galaxy. In both of those situations, you know where there is more detail in the signal. You could use this algorithm to reconstruct a complete image.

    3. Re:Choosing the perfect sampling method by mclinc · · Score: 1

      Phones don't use uniform sampling they use u-law or a-law. Which samples in a way that compliments our hearing The reasion this isn't done with hi-fi is down to practical problems implienting the required DAC/ADCs at the required number of bits. (without making the quanization noise swamp any advantage).

      Also see KL transform (AKA principle component anayisys) for removing statisticly redundent data. Not very practical for everyday use, but it has been demonstrated for storing faces on the 58 byte magnetic strip of a credit card. This works because faces span a very small subspace of the original images so the number of dimentions can be reduced.

      --
      "Oh no, not again"
    4. Re:Choosing the perfect sampling method by perky · · Score: 1

      mu-law and a-law are companding schemes, and have nothing to do with the sampling rate.

      --
      "The new wave is not value-added; it's garbage-subtracted" - Esther Dyson, Dec 1994
  42. Not like Variable bit-rate by rhh · · Score: 2, Informative

    The variable bit-rate in MP3 compression does not
    alter the amount of time between each sample. In
    terms of sampling frequency MP3, even VBR is still
    uniform, uniform as in time. VBR changes how many
    bits are in a sample, not the time between samples.

    1. Re:Not like Variable bit-rate by Anonymous Coward · · Score: 0

      Small correction:

      The bitrate referred to by VBR is the number of bits in the mp3 file itself (compressed), per second of audio. The number of bits per sample (uncompressed) is still going to be constant.

  43. Re:Some folks seem to be missing the point on the by pslam · · Score: 1
    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.

    48kHz 24 bit is all you need to generate a perfect reproduction of audio as far as the human ear is concerned. These days, audio in the pre-amplified stage is about as good as it's going to get, because it's already as good as the human ear.

    Non-uniform sampling, if it really improves matters (which I doubt in the case of audio), can't improve on what's already perfect.

    Just to emphasise: by perfect I mean the theory says that none of the distortions generated are even close to what a human ear can hear. This is also true in practice.

  44. Re:old news? -who cares? by Anonymous Coward · · Score: 0

    Who cares- 98.43% of all quoted statistics are made up.

    Lucius Sour

  45. D'oh! by Anonymous Coward · · Score: 0

    s/as/at/

  46. Is it just me... by justinstreufert · · Score: 2, Insightful

    Take a look at the triad of MRI images in this article. If you look at the image on the left, it appears to have been scaled up about 2-3x from the original size. If you zoom in on it, you can see that the smallest represented detail in the picture is about 3 pixels across. It looks like they just imported the MRI into Photoshop and did a Bicubic scale to 300%!

    They then remove 50% of the data in the second picture, and proceed to mathematically reconstruct it in the third. In my mind, this would be a great feat, except for two things:

    - More than 50% of the data was unnecessary to present the data in the first place. The original is quite obviously scaled up from its native size.

    - The mathematical reconstruction introduces artifacts that were not even present in the random image, such as huge horizontal pixel smears.

    Can someone point to a better demo of this set of algorithms?

    Justin

    --
    "Why would God give us a waist if we wasn't supposed to rest our pants on it?" - Rev. Roy McDaniels
  47. nothing new... by ChadN · · Score: 2

    I looked through the paper quickly, and it is a survey of existing techniques. The benefits of non-uniform sampling have long been known. Current low-end graphics hardware uses non-uniform sub-sampling grids to give better anti-aliasing results.

    It was shown in the 70's or early 80's by A. Ahumada that the human eye uses a non-uniform distribution of rods and cones (outside the fovea) because it can give better frequency response than a uniform grid (given the same number of cones over a given area).

    In short, while this paper makes good reading, don't think that it represent a breakthrough in the field.

    --
    "It's overkill, of course. But you can never have too much overkill." - Anonymous Slashdot Coward
    1. Re:nothing new... by Anonymous Coward · · Score: 1

      Thanks for your post. I was beginning to think that
      I dreamt some college lectures (=sad). I would have sworn that nonlinear sampling was part of some courses at college.

      Luckily, I can now rest in peace knowing that I was just a regular student who sometimes was not entirely sober enough to remember all details...

  48. PhotoCD/ FlashPix does that by purduephotog · · Score: 2

    Formats supported by Eastman Kodak Company-

    PhotoCD works with a differential 'error' image that was created by comparing the resampled to the original, and then that was compressed. Effect? Take a small image, blow it up by a factor of 2x, apply this itty bitty 'error' transform, and you have a nearly perfect 'fixed' image for the cost of some small change on disk space

    Then there is the 'much better clarity' etc statement- there's 'inverse point transform' for getting out defects.. they used that on the Hubble Telescope. Looked pretty good for being wildly out of focus.

    Everything you've mentioned is already available... the technique looks interesting but it's all data dependent ... given enough training data you can make a GA to give 'guesses' into any dataset.

  49. rofl!!! by Anonymous Coward · · Score: 0

    I've been doing this for years. Nothing new here. Move on.

  50. I noticed that too. Potential downsides? by Tablizer · · Score: 1

    Some of the "filling in the dots" indeed produced long spikes that are *not* in the orginal. In the sample chosen, that was not a problem because we know that is not how the (normal) skull goes.

    However, in some other image settings this might not be the case. For example, where there are a lot of linear-dimensioned information that tends to go by the same grain as the pixels.

    They might be pulling some wool over our eyes by picking samples that minimize the downsides of their algorithms.

    Perhaps they should focus on esthetics improvement, such as music and clipart and not on domains where you can get your ess sued off if somebody dies from a misleading image.

    (troll mode on)

    This kind of reminds me of the OOP books which tend to show change patterns that OOP seems to benefit, but completely ignores change patterns which tend to get messy under OOP.

    (troll mode off)

  51. Re:Some folks seem to be missing the point on the by Tony-A · · Score: 2

    Hmmm, this might be good for:
    Non-skipping CD players.
    De-scratching old LP records.
    Reconstructing old photographs.

  52. I love articles about advanced math... by Sgs-Cruz · · Score: 1

    ...where the object is to be excited about the new theorem/method/sampling technique without mentioning any details about how it works... I always get a laugh when I see things like "Our theory - which is based on a lot of beautiful new mathematics". I can just imagine the reporter: "tell us about your new mathematical theorem without mentioning anything at all about how it works!"

    --

    Karma: pi (Mostly due to circular reasoning in posts).

  53. New Techniques? by Anonymous Coward · · Score: 0

    check out chapter 13-8:

    http://www.ulib.org/webRoot/Books/Numerical_Reci pe s/bookc.html

    (btw: that book is a decade old)

  54. Issues of jitter by Anonymous Coward · · Score: 1, Insightful

    I don't see that as being true - although you may be able to create a better image of the sound through the non-uniform technique, you still have to have a highly controlled time resolution so that you at least know where your samples are. If your technology is fast enough to give you that time resolution with a low jitter, you're still better off sampling uniformly and getting rid of data later than randomly deleting data at the source.

  55. Re:Some folks seem to be missing the point on the by Anonymous Coward · · Score: 0

    I don't think the reconstruction they showed was anywhere near a "highly accurate representation of the original". Furthermore, I strongly suspect that simple median filtering would have resulted in at least a visually better result. And there are certainly lots of methods around already that will do better on this problem.

  56. Bayesian Analysis ? by nqp · · Score: 1

    non-uniform sampling is not that new.... and data recovery techniques are not dependant on uniform or non-uniform sampling... maxaimum entropy methods, or bayesian analysis are very powerful at this - I wonder if a compaison has been made ?

  57. Impressive, but... by mrjb · · Score: 2, Insightful

    Looking at the 'restored' pic I see only 'horizontal' distortion, imagine how well the picture would have been restored if they would have applied their maths in *two* dimensions...

    --
    Visit http://ringbreak.dnd.utwente.nl/~mrjb/growingbettersoftware to download your free copy of the book
  58. don't tell the republicans by envelope · · Score: 1

    So how's this sampling going to affect the census??

    --

    appended to the end of comments you post, 120 chars
  59. A Pixel Is Not A Little Square by Peter+Lake · · Score: 2, Informative

    It's essentially a POINT - it has no dimensions. When you see those little squares you actually see a poor (and fast) representation of pixels - pixels themselves are not square or non-square. Pixels won't come in various sizes, they'll still be regular 0-sized points.

    Here's a good paper on why it's important to keep in mind the true nature of pixels (by Alvy Ray Smith):

    A Pixel Is Not A Little Square, A Pixel Is Not A Little Square, A Pixel Is Not A Little Square! (And a Voxel is Not a Little Cube)

    --

    All Rights Reversed.
  60. Not non-uniform "sampling", but reconstructing by RatOmeter · · Score: 1

    Err, I read the article. I'm left a little confused. It seems they're talking less about non-uniform sampling than how to best recreate the function of an existing dataset.

    For down-to-earth, "non-uniform" sampling methods, I've used variations of two in the past:

    1) Relatively low sample rate until a known significant event in the sample stream is detected.
    Then shift to the appropriate sample rate (or even variable, based upon a timing profile).
    This has been especially useful when decoding (radio) "bursty" datastreams where there's long periods of "silence"
    interrupted infrequently by data transmissions. No sense in wasting buffer space or overhead until you know you've got data to look at.

    2) A variation I've used more often: operate at full (Nyquist or better) spec sample rate, storing to a fixed-size buffer whose contents are discarded/ignored if, upon inspection, it contains no interesting data. If data of interest shows up, say, halfway into the buffer, that becomes the beginning of the buffer (can you say "ring buffer"?) and you continue to sample til the buffer is used up. Best example of this would be the way a DSO (Digital Sampling O-scope) works.

    I've implemented one system which used 4 different sample rates within a single data set. The data is a pressure waveform from a pump/valve dynamic response test. In it are regions of interest which range in sampling priority from to "nothing's happened" to "gotta know what happened up to 4.7 mS after this event".

    That's non-uniform sampling in my book. No need to "reconstruct missing data".

  61. Guessing the algorthm by looking at the brainscan by jovhl · · Score: 1

    When I first looked at the brainscan images, my first thought was that this must either be a joke or a hoax. All the horizontal stripes makes it quite obvious that no vertical interpolation is done whatsoever, and all the filled-in pixels seems to be just copies of their closest horizontal (valid) neighbour. Surely you would get a better interpolation by using vertical data as well, or even do an average of the other valid datapoints weighted by, say, the inverse cube of the distance or something.
    But lets think a bit harder. What if the algorith only handles one-dimensional data? That would explain why there are only horizontal stripes, but why the constant intensity in the stripes? It does give a human viewer the ability to "see" the reconstructed pixels, while at the same time doing a reasonable job of reconstructing a close resemblance to the original image. Sounds OK, but that's not mentioned in the article, and seems to be at odds with the claim in the headline. But back to the closest horizontal neighbour algorithm. If this is the case, where does the line in the middle of the eye come from? There are no datapoints of those intensities there? My best bet would be that the images we see are scaled-down images of the ones the algorithm was processing, and that the two pixels that seems to have generated that line was lost in the downscaling. Seeing as there is no intensity-smoothing on the lines sticking out, I'd say it was scaled purely by picking every n'th pixel, which fits my argument well.

    So, is this for real or what?

  62. scaling example ... by RockyJSquirel · · Score: 1

    The only way the scaling example makes sense is if the smaller picture was made from non-uniform samples.

    My first thought was that this was a cheat because every pixel has more data than just it's value, it also has it's (non-uniform) x and y coordinate. More information in = more information out.

    Then I remembered that non-uniform sampling doesn't have to be at random intervals, it just needs to have a pattern that doesn't coorelate with the source data. So, if you always use the same pattern, you don't have to store the x and y info.

    Anyway I remember the stuff in "numerical recipies" about making a maximum entropy estimatimation of a spectrum from non-uniform samples.

    Does anyone know if that relates to this guy's method?

    Rocky J. Squirrel

  63. A better article by markmoss · · Score: 2

    I finally found a better explanation of the new sampling theory. It has to take repeated passes at the same analogue data. First pass is sampled at regular intervals, as usual. This data is analyzed, then on the second pass areas where the data changed fast are sampled at a higher rate. Repeat if needed...

    This will usually give results similar to scanning at the maximum sample rate, then "compressing" by throwing out data points where the values are not changing much -- you need less RAM, but the maximum digitizer speed is the same, and you have to replay the analog data somehow. For instance, in an MRI, the multiple scans might mean holding the patient in the machine longer. That's not good, and enough RAM to hold everything isn't going to add much to the cost of the machine. Also, there is one condition where the results could be different -- if a detail such as a hairline fracture is so fine that it might be entirely missed between the points on the first coarse scan. If you scan at maximum resolution first, you won't miss that.