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.
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"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."
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.
eromlignod snipped-for-privacy@aol.com posted to sci.electronics.design:
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.
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 figure.
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.
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.
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. Regards Ken
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.
Ken S. Tucker snipped-for-privacy@vianet.on.ca posted to sci.electronics.design:
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.
Thanks for pointing that out. The noise is mainly from a nearby waterfall, which generates white noise, we should put that on the site. 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! Regards Ken S. Tucker
Ken S. Tucker snipped-for-privacy@vianet.on.ca posted to sci.electronics.design:
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.
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. ;-)
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. Regards Ken
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.
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