Walter and Gianni--
Might also be interested in this real time analyzer. Also has
harmonics overlay.
Free 15 day demo
http://www.hairersoft.com/Amadeus.html
Rob D.
= = = =
>Gianni Pavan wrote:
>>
>> At 13.07 21/06/2002 -0400, you wrote:
>> >When using sonograms be sure and be aware that the math involved in
>> >producing sonograms produces artifacts in the display. In particular a
>> >lot of what looks like harmonics off the louder parts of the call is not
>> >there.
>> >
>> >Walt
>> >
>>
>> Walter,
>> if harmonics appear on a spectrogram I think it is because
>> something is in the signal, maybe due to distorsions (overloads) in the
>> input stages of many small DATs (the "brickwalling" effect of the small
>> SONY DATs is well known...) or in a poor sound card. I can't believe
>> harmonics are generated by the math involved in computing digital
>> spectrograms. Of course FFT computation should be done with the right
>> algorithm and with large dynamic (I mean using float math instead of
>> integer math and arranging scaling to avoid overflows). Although, it is
>> true that the time-frequency uncertainty principle may cause confusion when
>> analyzing pulse trains: they may appear as continuous tones with lots
>> of harmonics when the pulse interval is shorter than the analysis window.
>> In such a case waveform inspection and a correct tuning of analysis
>> parameters (fft size, window size, window shape, scanning step, dynamic
>> range of the display) may help in revealing or at least in clarifying a bit
>> more the real structure of a signal.
>
>I'm certainly aware of lots of ways harmonics creep into recordings
>through failings of the equipment. And those are a whole different subject.
>
>Nor am I contending that all harmonics in sonogram displays are a result
>of the computations. After all, some are real.
>
>The trouble is in real life most of us are using programs that don't
>have all that much adjustment. And even if we did we may not be
>mathematically savvy enough to make intelligent choices. And the other
>nasty part seems to be as you adjust parameters, at the same time you
>don't get this strange stuff your display also loses it's resolution and
>clarity.
>
>In my current "best" sonogram producing program, Spark XL, I have four
>choices for FFT size from 512 through 4096 and usually choose 4096. I
>also have several steps in scrolling rate (timescale) and have almost
>infinite control on the frequency range (in linear or log scale) and the
>correspondence of color to dB level. And there is a quality adjustment,
>for which I'm not sure what math is involved, but set it better quality
>and the sonogram will show a little more detail. The sonogram window is
>of fixed size. I can, of course, filter the sound, or resample it. This
>program shows such "harmonics" off the most intense part of the call for
>some calls when otherwise adjusted to optimally create a sonogram.
>
>In any case there's a quick way to see which it is, real or math. Pass
>the file through a good cut filter to slice off all the frequencies from
>just above the primary. Or lacking such a filter resample the file at a
>rate that will cut off at the right point, then resample back up to the
>original rate and do the sonogram. If that does not change the
>"harmonics" showing up on the display, then you can be fairly sure it's
>the math and not something introduced earlier in the recording or
>transfer process. And by math, I don't mean the formulas we think we are
>calculating but what the computer is really doing. Add, subtract and
>fakes multiply and divide, computers don't really do exotic math.
>
>I've seen this sort of thing with every sonogram program I've used. Long
>ago when I first started working with computers writing computer models
>I learned that what we thought went on and what does are not always the
>same with computers. Which is probably why the word algorithm is stuck
>in there when talking about what a computer does. The sonograms seem to
>be no different in this regard. Still, computer generated sonograms beat
>the old mechanically generated ones by a wide margin. And provide a
>wealth of information.
>
>One last thing, what I'm talking about is most likely if the highest
>intensity parts are close to the 0dB clipping point. (note I'm talking
>about sounds that were not clipped) Like within 10-15 dB. So they can be
>minimized by keeping the sound from that area. Readjusting the gain of a
>digital file can often stop them.
>
>> Just to complete my thought, I should say that the sampling rate often is
>> not high enough to reveal the high frequency structure of a sound: a pulse
>> train with 1/10000s pulse interval appears like a perfect sinusoidal tone
>> at 10 kHz if sampled at 32kHz (with good anti-aliasing filters, of course);
>> a 48 kHz sampling reveals a 2nd harmonic... only a very high sampling rate
>> reveals its "pulsed" and not continuous nature.
>
>Beyond what most nature recordists have for equipment. Off into the
>realm of Bioacoustics gear. And to truly find that it's pulsed, all the
>input gear has to be up to it. My thought is that probably the gear used
>by nearly all members of this group will feed a sinusoidal signal into
>the recorder from the pulse.
>
>But the thought is something to pay attention to. A frequency of half
>the sampling rate is mangled as to waveform and thus sound quality. If
>your frequencies of interest fall into the top third of the frequencies
>recorded with digital a lot of the time, consider going to a higher
>sampling rate. That is not justification for the current fad to jump to
>higher sampling rates than the 44k CD rate. The upper third of the
>resulting 22k range is not occupied by very much of what we are trying
>to listen to, which is lucky because our ears are also very poor at
>picking up sound in this range.
>
>Walt
>
>
>
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Rob Danielson
Film Department
University of Wisconsin-Milwaukee
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