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Converting Arb Waveform to Data Points

Discussion in 'Electronic Design' started by [email protected], Apr 5, 2013.

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  1. Guest

    I have an MP3 file of an arbitrary waveform. From it I need to derive
    a set of data points with values from 0 to 256 (y) at 3 precise
    millisecond (x) intervals. What is the most straightforward way of
    accomplishing this?

    BTW I only have the analogue waveform, not access to an arb generator
    itself.

    John Farrell
     
  2. Tim Williams

    Tim Williams Guest

    Unless it's a really high bitrate MP3 (even then, I don't know), your
    waveform is already trash. MP3 is a lossy format with psychoacoustic
    adjustments, probably passable for frequency domain measurements, but any
    time domain waveforms will have phase shifts and humps you weren't
    expecting. Or something.

    So, you want to extract samples? Downsample? Quantize? Measure? What?
    I'd say try opening it in Audacity..

    Tim
     
  3. Jasen Betts

    Jasen Betts Guest

    "audacity"

    "mpg123" + "sox" if you need to script it,
     
  4. Guest

    Zeroth, I'm assuming you want to do this on a full desktop/laptop PC.
    If you need to implement it on some kind of microprocessor or embedded
    system, the steps are the same, but you get to write more of the
    software yourself.

    First, you need to get it out of MP3 and into something like WAVE or
    raw binary format. sox or Audacity will do this.

    At this point, it is optional to normalize it, such that the loudest
    sound (highest peak) in the waveform corresponds to the biggest number
    representable in the chosen format. This means finding the loudest
    peak in the data, figuring out the scale factor, and then multiplying
    all the samples by the same scale factor. Again, sox or Audacity will
    do this.

    Second, assuming that 0 to 255 is OK, you need to scale it to 8 bits.
    If you really, really, really need 0 to 256, you'll need to scale it
    to either 9 bits (which gives you 0 to 511) or 16 bits (0 to 65,535).
    Most all sound software will be happy to scale it to 8 or 16 bits but
    I don't know if many packages can do 9 bits. If you end up scaling it
    to 9 or 16 bits, you'll have to add a manual scaling step later to
    scale the 0-511 or 0-65535 range to 0-256.

    Third, if I understand your requirement correctly, you need a value
    every 3 milliseconds. You therefore probably need to resample to some
    rate that gives you samples on an integer multiple of 3 milliseconds.
    (You can resample to 333.33 Hz and use every sample, or resample to
    1 kHz and use every third sample, or resample to 10 kHz and use every
    30th sample, etc.) Note that resampling will limit the bandwidth of
    your data; if you resample to 333 Hz, no signal faster than about 167 Hz
    will make it through. sox or Audacity will do this.

    Fourth, you need to pick out whatever samples you have selected and
    read out their values. sox and Audacity can save audio as binary files
    with no header; you will need to either inspect the file on your own or
    write a program to scan the file and print out the results in human-
    readable form. This is probably where you would do the manual scaling
    if you need 0-256 and have 0-511 or 0-65535.

    sox is a command-line program and Audacity has a GUI. If you're doing
    this once, Audacity is easier. If you need to automate it, sox is
    easier. (Also, if you're automating it, record to an uncompressed
    format at first - *not* MP3.)

    It is probably a good idea to obtain an input file of something like a
    pure sine wave at 1 kHz or 10 kHz or whatever approximately matches your
    data, run it through all the steps, and make sure you see about what you
    expect. If you don't, then try an input file with a constant DC level
    in it.

    Audacity and sox should be installable packages on all major Linux
    distributions. There are Windows versions of both programs; see
    http://audacity.sourceforge.net/download/windows and
    http://sox.sourceforge.net/ .

    Matt Roberds
     
  5. Guest

    Except for the slightly nasty 11.025/22.05/44.1 kHz sampling rates,
    what is the problem ?

    With simple weighting (1st order interpolation), even those cases
    could be handled easily.
     
  6. Guest

    Thank your for your detailed reply.

    I have a copy of Adobe Audition which I assume will do about the same
    thing. It used to be called "CoolEdit" and you can still get it at
    www.oldapps.com

    Goldwave can implement math expressions, but I would need to brush up
    on these.

    John Farrell
     
  7. Guest

    You're welcome!
    CoolEdit/Audition should do most of the same things as Audacity, yes.
    I think Audacity's basic feature set and UI were partly modeled on
    CoolEdit, many years ago, but they have probably diverged since then.

    Depending on how old it is, Audition might not want to use an MP3 as
    input, even if it can save its output as MP3. In that case, there are
    lots of MP3-to-WAV converters around that you can use first.

    Audition may have some kind of scripting language in case you need to do
    this more than once. But get it working through the GUI first before
    you try to automate it.

    Matt Roberds
     
  8. Robert Macy

    Robert Macy Guest

    Great reply! Sorry for being a bit pedantic, Yes, the 'results' will
    be limited in frequency, but the 'content' will contain aliasing
    [energy] of every higher frequency folded into that spectrum. The
    result will not even represent very well the low frequency content of
    the original signal with all that folded in higher frequency content.
     
  9. Guest

    I seem to recall that Audacity will (or can) apply a low-pass filter
    before resampling to a lower rate. It's been a while since I've used
    that functionality, though.

    In general, you can *hear* the high-frequency aliasing in things that
    have been undersampled without a low-pass filter first. YouTube videos
    recorded at low bitrates from TV or radio tend to have it.

    Matt Roberds
     
  10. Robert Macy

    Robert Macy Guest

    Have a copy of Audacity, will try...some day.

    Interesting. Bet aliasing shows up more in speech than music, right?
    Interesting because *if* you can hear it, something incredible as the
    pattern recognition of the human brain could remove it. Well, maybe.
    Tantalizing.
     
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