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Best way to measure precise harmonics?

Discussion in 'Electronic Design' started by eromlignod, Oct 18, 2007.

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  1. Martin Brown

    Martin Brown Guest

    That goes with the territory. To obtain a +/- 0.001 Hz frequency
    resolution you would have to measure the signal for ~1000 seconds (you
    might get away with 250s using some mathematical symmetry tricks if
    you can control the initial conditions well enough).

    For an upper frequency limit of A at 440Hz and oversampled by x2 that
    is roughly 1,000,000 samples and should be doable with relative ease
    in software on a PC. Your required precision seems to be severe
    overkill, but let that pass. To go beyond the edge of human hearing at
    20kHZ and oversampled x2 it is 40,000,000 samples which is still
    doable but a fair bit slower - and you might want to go for 40kHz for
    added headroom.

    BTW You probably want to measure your signal for a fixed number of
    cycles of the fundamental.
    FFT will do it if you can supply enough data in the time domain
    (choose a 2^N FFT). Practical implementations will require potentially
    anti-alias regridding and a few other tweaks to sort out boundary

    Some hardware FFT based analysers can zoom in on a region of interest,
    but non of them can get around the uncertainty principle. A short
    burst of pure tone decaying in amplitude always contains a range of
    frequencies around the fundamental.

    Martin Brown
  2. Question: How good are todays A to D Converters?
    Will the conversion introduce serious artifacts?

    I'm thinking that once the wave form is in digital
    form it's just software to compare that to a perfect
    sine wave at various frequencies. I'm sure that's
    be done.
  3. Martin Brown

    Martin Brown Guest

    Good enough for studio quality digital audio to have taken over from
    Shouldn't do if it is done correctly. The most important thing is to
    have an analogue brick wall filter to ensure that no out of band
    frequencies reach the input to the ADC. Any timing phase jitter in the
    converter will also hurt.
    Just a SMOP... But for these volumes of data it requires some skill to
    obtain the optimum results for a high dynamic range spectrum
    containing a fundamental and a bunch of its near harmonics.

    FFT is just a quick way to decompose a signal into its frequency
    components. Classical slow DFT would be glacially slow on large
    datasets unless you were only looking at a handful of likely

    Martin Brown
  4. Tom Bruhns

    Tom Bruhns Guest

    We've been building FFT analyzers for many years; I can assure you
    that ADCs that are very linear do a great job in these analyzers. The
    advent of delta-sigma converters made life a lot easier, and of course
    for audio they are pretty much universal now. Obviously, if you are
    looking for overtones in the spectrum of an excited string, and those
    overtones are very close to the frequency of harmonics of the
    fundamental, you'll want to know just how much harmonic distortion is
    being introduced in the signal path. It can come from the transducer
    that goes from acoustic to electrical, and in the amplifiers ahead of
    the ADC, and in the ADC itself. It's quite possible to get distortion
    in the electrical path lower than -100dBc in the audio range, but it's
    also pretty difficult (in my experience) to find hard specs on the
    distortion introduced by the acoustic to electrical transducer:
    microphone or other pickup. If the overtone and fundamental are both
    pure enough tones, and if the harmonics are enough different in
    frequency from the overtones, the analyzer can resolve them.

    Given a stream of samples from a good audio "card" (or external USB
    audio port or whatever), the processor in a modern PC should have no
    trouble at all keeping up doing "zooming" and decimating. That makes
    the display of the results somewhat easier and the FFT processing can
    be done real-time, since you're doing FFTs on relatively small blocks
    of data at a slow data rate (after decimation). Does anyone sell
    software that actually does all this (nearly) real-time? We used to
    do it for audio-range analyzers using a custom ASIC chip set, but
    these days, there's certainly plenty of processing power in a typical
    PC. We still do it with an ASIC, but now much, much faster.

    With respect to determining frequencies, _IF_ I know a priori that I'm
    dealing with a pure tone (and therefore stable in phase and
    amplitude), and the signal-to-noise ratio is good, I can determine the
    frequency to within 0.001Hz with well under 100 seconds of data. The
    reason is that I know exactly the response of each FFT point to any
    frequency, and the response of a set of FFT points to the waveform can
    only have happened with a particular input. It's equivalent, I guess,
    to fitting a sinusoid to the digitized points; if I am quite sure the
    input is a sinusoid with unknown frequency, phase, amplitude and
    perhaps DC offset, I don't need very many samples to nail down those
    four unknowns. Of course, the difficulty is that I almost never can
    be really SURE that my input is a pure sinusoid. I must also have
    enough data points to sufficiently average out whatever noise there
    is; thus, a really good SNR allows fewer points to determine the

    You mentioned filtering to avoid aliasing. That's something else that
    has been aided a whole lot by the delta sigma converters, since the
    sample rate is much higher than the highest input frequency of
    interest. The analog filter can be relatively gentle, and the
    filtering becomes mainly a digital process; it can be linear phase FIR
    filters, which makes corrections somewhat easier, too.

    Even when you don't know what the input waveform really is, or when
    you know it contains harmonics and overtones and the like, maybe even
    multiple "fundamentals," an FFT analyzer can give you a very good
    picture of what your signal looks like, spectrally. You do need to
    understand things like "windowing" and what happens if your input
    frequencies are not integer multiples of 1/(time record length)

  5. Interesting, I'll pull this quote,
    " hard specs on the distortion introduced
    by the acoustic to electrical transducer:"

    Just so you guys know I'm serious about this
    subject, I/we designed this unit,

    and I respect the problem of acoustic transducers.
    At that site are recorded wave forms of Loons,
    (let me know if you have any problems getting
    their call, my current system hasn't got audio).

    I'd get the Loons to yell by recording them and
    then replaying their call over the lake. They'd
    show at my dock yelling back. So I relied on
    my "tin ear" (and others) to inform me of distortion.
    I wasn't crazy about the science of the test but
    what choice did I have?
    Ken S. Tucker
  6. Andrew

    Andrew Guest

    Adobe Audition,(Previously known as Cool Edit Pro) I believe has this
    sort of thing built in and IIRC you can write your own plugins.,239029154,240001669,00.htm

    "Beat detection, tempo and pitch shifting, and vocal/instrumental
    channel extraction (for a cappella and karaoke) are just a sampling of
    the powerful audio-manipulation tools you'll find within Audition. You
    can also restore and sweeten individual tracks with filters and effects
    such as high-quality click/pop eliminators, noise and hiss reduction,
    time stretching, sample rate conversion, and even pitch correction (for
    fixing off-key notes). We're particularly impressed with the app's
    Spectral view, which isolates individual instruments and transients and
    permits full editing in the frequency space. All audio is processed
    internally in high-fidelity 32-bit and sample rates up to 10MHz."

  7. Tom Bruhns

    Tom Bruhns Guest

    No problem getting the loon sounds to play, though there's a
    tremendous amount of echo in them, it seems like!

    I hadn't directly addressed your original posting where you asked how
    good modern ADCs are, and about analyzing the digitized sounds. The
    best off-the-shelf audio converters I know about are 24 bit, can
    digitize with output rate of 96k and 192k samples/second in addition
    to the old 48k, and have distortion products typically at the part per
    million level. The noise is pretty darned good too. I suspect it's
    unlikely that you'll find a transducer that linear, at least not with
    loud sounds, and it's not trivial by any means to make a preamp with
    such low distortion (though some of the modern op amps have helped a
    lot with that).

    One way to view an FFT is that it compares the input signal with sines
    and cosines at the frequencies corresponding to the FFT points (also
    commonly called "bins"). An advantage of the DFT is that you can do
    that comparison for any spot frequency, and you're not limited to the
    linear frequency spacing of the FFT; but of course, it's much slower
    if you want to do a LOT of points. On the other hand, there's a DFT
    algorithm that lets you calculate as the data comes in, and as soon as
    you've finished collecting the data, just a very few operations gives
    you the final answer from the DFT -- you can run several of those in
    parallel if you want.

    And of course, you can design a filter or detector that is "optimal"
    in some sense, using things you know about the waveform you're
    analyzing. An FFT is a good general-purpose spectral analysis tool,
    but it likely won't be the _best_ tool for some specific application.


  8. JosephKK

    JosephKK Guest

    eromlignod posted to

    In that case you could get a studio grade microphone and 24-bit
    digitizer and sample for about 5 minutes for the lowest note. And
    feed that to an FFT program. The file should be over 300 MiB.
    For higher notes use correspondingly less time and get smaller files
    as a bonus.
  9. Pieter

    Pieter Guest

    If you only do this on one string at one frequency, I would use
    filters. I have an old but nice B&K 2110 filter (Audio Frequency
    Spectrometer) that can be used for such things. 1/3 octave filtering
    should do, removes enough and keeps the signal intact enough. You can
    compare the filtered output to a reference, precise signal on a scope
    and see the difference (one sine will "walk", or make a lissajous

    Or read several seconds of data into a computer and do FFT etc.

    You will need a reference somewhere.

    You can also do it the other way around: with a coil driven by a
    precise sine source, you can bring the string into resonance and
    measure the peaks. Notice that you must place the coil where you
    expect the peak: at exactly the middle of the string, you can generate
    the base tone, NOT the double frequency - for that the coil must be
    app. at 1/4th of the string length.

  10. Phil Allison

    Phil Allison Guest


    ** Like what ???????????????

    Crypic advice is just as worthless as the purest of bullshit.

    Be prepared to justify the use of magnetic force drive.

    ....... Phil
  11. Glen Walpert

    Glen Walpert Guest

    I suspect that a good part of the echo and the general poor quality of
    the recording (compared to other loon recordings or the real thing)
    are in part due to putting the microphone in the birdhouse-like box.
    Realizing the full capabilities of a microphone requires that the
    sound being measured directly hit the microphone without passing
    through any apertures or reflecting from nearby surfaces. For
    precision lab measurements the protective screen over the microphone
    diaphragm is usually removed. (Measurement microphones come with
    callibration curves with and without the screen, and they are
    significantly different, without always being much better.)

    All of the really good loon or other bird recordings I have heard were
    made with a microphone at the focus of a parabolic reflector aimed
    directly at the sound source. The pre-microphone signal gain with no
    gain on indirect noise sources cannot be matched in any other way.
    Without a good sound signal to the microphone, the microphone,
    amplifier and A/D performance are almost irrelevant; even perfect
    components would not give you a good recording. Put the microphone in
    a box, and you might as well use the cheapest components you can find,
    it won't make much difference in the recorded sound quality.
  12. Thanks for your "feeback" (pun intended) Tom.

    Thanks for listening.
    The Loons were a few hundred yards from shore
    and recorded at night, so the shoreline trees
    would bounce the sound quite a bit, those guys
    are loud! That haunting echo might be part of the
    uniqueness of the recording.
    My audio frequency ADC experience was with
    audio scramblers for security, like spy stuff.
    The audio was digitized via an ADC-ROM and
    anti-ROM'd at the recieving end.
    I worked with servicing and developing Medical
    Ultrasound equipment, where the audio frequency
    runs to 10Mhz, that's fun stuff. Stuff is ingenious,
    all dedicated to get a good image.
  13. Pieter

    Pieter Guest

    Scanning through frequencies, one can find points where the string
    gets into resonance. It will differ from a piano, as a piano strikes a
    string, more like a pulse and will give a combination of base tone and
    harmonics. With this coil, one can find resonace peaks without the
    need for that base tone. Notice that if the frequency depends a (tiny
    little) bit on the base tone, the harmonics get modulated a little by
    that base tone, what will not happen with the coil measurement.

  14. JosephKK

    JosephKK Guest

    Ken S. Tucker posted to
    I personally found the recordings unacceptable. The noise was far
    larger than the loon calls were. Make a point of checking an audio
    file before placing it on your web site please.
  15. Thanks Joseph & Mr. Walpert as well.

    Thanks for pointing that out.
    The noise is mainly from a nearby waterfall, which
    generates white noise, we should put that on the
    I've tried to eliminate "waterfall" noise using a
    graphics equalizer but with mixed results.
    I designed a filter in the EAR that softly favors
    about 4kHz, it's a trade-off.

    We recorded the sounds on to cassette, then
    fed them into the wavefile back in the 90's.

    Incidentally, we've only had one field failure.
    A Woodpecker started pecking the hole and
    that's were the microphone is, behind a light
    screen, and it damaged the mike.
    The customer (a wealthy bird watcher) makes
    an appointment and comes to the office.
    I swapped in a new PCB at cost ($50), and
    put heavy screening over the hole.
    He bought another one because he wanted
    stereo, putting one at each end of his river
    front property.
    I expected lightning striking the units but that
    hasn't been reported.

    We did echo testing of the mounting and
    minimized that effect with foam.
    All customers are satisfied, but we're working
    on improvements!
    Ken S. Tucker
  16. JosephKK

    JosephKK Guest

    Ken S. Tucker posted to
    What i heard was more like excess gain going into oscillation.
    Squeals and hums and such. It was kind of weird, when the loon calls
    broke through they were clear.
  17. Tom Bruhns

    Tom Bruhns Guest

    Well, the file is 8 bit uncompressed at 11025 samples/second. Don't
    expect a whole lot out of that, of course. I'm looking, right now, at
    a section from near the start of the recording, about 8 seconds from
    the start. I can see some energy at 60Hz and its harmonics, up to
    about 360Hz, but by far the most energy is in the 0.8kHz to 1.8kHz
    range. A typical segment is a warble between 1.0 and 1.2kHz, about 10
    cycles of warble per second, with a very slightly falling average
    frequency. That lasts about half a second, and is followed a very
    short gap later by either about the same thing, or something similar
    but at generally not quite as distinct from about 1.2 to 1.4kHz, same
    sort of warble.

    Perhaps the most remarkable thing to me is that the recording goes on
    for several seconds at a time with only two or three dB variation in
    amplitude. Seems like that would get monotonous very quickly. ;-)

  18. Our estimate is the EAR unit is 20 db better than
    the human ear, so little things like crickets, frogs,
    etc will be heard in the absence of loud sounds,
    but the gain is dynamic, reducing as the output
    goes to high, to prevent clipping. That's a handy
    thing to provide wider dynamic range, via sound
    "compression". I'm guessing that's the effects
    you heard.
  19. Yeah! We worked hard to eliminate 60Hz...etc,
    fancy filtering, shielded cable is standard too,
    it's another trade-off, costs go up to take the 60hz
    down a bit more for increasingly expensive filters.
    Mainly it's a subjective call, boosting the amp
    volume to where 60hz is audiable would also make
    a cricket sound like a "rock concert".
    Hmmm, might make a good cell phone "ringtone".
    Yup, after awhile you just tune them out,
    them Loons will spend all night humping,
    sort of a lullaby.
    Thanks Tom.
  20. JosephKK

    JosephKK Guest

    Ken S. Tucker posted to
    You may never know until you use a couple of different machines to
    play it back with.
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