ceg@nova.stanford.edu (Chris Gronbeck) (07/21/90)
I'm looking for suggestions as to the best way to design a circuit to drop music one or more octaves. A typlical application being making an electric guitar perform like a bass. Any hints would be useful. Thanks. Christopher Gronbeck ceg@nova.stanford.edu
jthornto@fs1.ee.ubc.ca (THORNTON JOHAN A) (07/21/90)
In article <1990Jul20.223615.4305@portia.Stanford.EDU> ceg@nova.stanford.edu (Chris Gronbeck) writes: > >I'm looking for suggestions as to the best way to design a circuit to >drop music >one or more octaves. A typlical application being making an electric guitar >perform like a bass. Any hints would be useful. Thanks. > The basic octavo circuit is: - send the signal into a clipper __|~~ - feed output into a divide by two flip-flop - wave-shape the output (filter...) ------- _/__/ ----------------------------------------------------- _| ___| E l e c t r i c a l | Johan Thornton, Esq. | | |_/ E n g i n E E r i n g |------------------------- |/| __| U n i v e r s i t y | jthornto@fs1.ee.ubc.ca |-| |/__ o f B r i t i s h |------------------------- | |_____| C o l u m b i a | This space for rent ---- |__|/_| ------------------------------------------------------
crs@lanl.gov (Charlie Sorsby) (08/03/90)
In article <1332@fs1.ee.ubc.ca>, jthornto@fs1.ee.ubc.ca (THORNTON JOHAN A) writes: = In article <1990Jul20.223615.4305@portia.Stanford.EDU> ceg@nova.stanford.edu (Chris Gronbeck) writes: => =>I'm looking for suggestions as to the best way to design a circuit to =>drop music =>one or more octaves. A typlical application being making an electric guitar =>perform like a bass. Any hints would be useful. Thanks. => = = The basic octavo circuit is: = - send the signal into a clipper __|~~ = - feed output into a divide by two flip-flop = - wave-shape the output (filter...) Not quite... Several problems: 1) Even assuming only a single note at a time (i.e. no chords, etc.) this will only give the fundamental pitch of each note--no harmonics--thus the "color" of each note will be lost. 2) If more than one note is played at a time, even more information will be lost. Depending on the clip levels, you will obtain (using a much oversimplified description) only the lowest frequency available at any given moment. 3) It will double the length (duration) of the tune so you get twice as much music for your time :) Nevertheless, subject to its limitations, the method described will lower the notes *it sees* by an octave. Best, Charlie Sorsby "I'm the NRA!" crs@lambda.lanl.gov sorsby@pprg.unm.edu -- Charlie Sorsby crs@lambda.lanl.gov crs@agps.lanl.gov
FC138001@ysub.ysu.edu (Phil Munro) (08/04/90)
I thought this was easy to do with digital techniques. First do an A/D, then a D/A with varying clock frequency. Or something like that.
R_Tim_Coslet@cup.portal.com (08/04/90)
In article <1332@fs1.ee.ubc.ca>, jthornto@fs1.ee.ubc.ca (THORNTON JOHAN A) writ es: = In article <1990Jul20.223615.4305@portia.Stanford.EDU> ceg@nova.stanford.edu (Chris Gronbeck) writes: => =>I'm looking for suggestions as to the best way to design a circuit to =>drop music =>one or more octaves. A typlical application being making an electric guitar =>perform like a bass. Any hints would be useful. Thanks. => = = The basic octavo circuit is: = - send the signal into a clipper __|~~ = - feed output into a divide by two flip-flop = - wave-shape the output (filter...) NO! The clipper by itself will destroy the "signal" information. The only way I know of that will definitely do what you want is a Digital Signal Processor doing a Fast Fourier Transform on the input, divide the "frequencies" obtained by 2, and inverse Fourier Transform to generate the output. There is probably some less expensive "Analog" method (probably using "modulators" of some sort) to rearange the spectrum, but I am not that familiar with this. R. Tim Coslet Usenet: R_Tim_Coslet@cup.portal.com BIX: r.tim_coslet
tuv@pmafire.UUCP (Mark Tovey) (08/06/90)
In article <1990Jul20.223615.4305@portia.Stanford.EDU> ceg@nova.stanford.edu (Chris Gronbeck) writes: > >I'm looking for suggestions as to the best way to design a circuit to >drop music >one or more octaves. A typlical application being making an electric guitar >perform like a bass. Any hints would be useful. Thanks. > > My memory is very dim at this point but I recall reading years ago in something like Popular Electronis or something like that about plans for an ultrasonic sniffer. This device utilized the receiver transducer for the old style television remote control units to pickup ultrasonic sounds. This is where my memory gets real hazy. If I remember correctly, the signal was passed to a circuit that resembled the front end of a radio receiver and reduced the signal down to some intermediate bandwidth. From here it was converted to normal audio and passed into an audio amplifier. I have no idea how well it worked or how to go about building one, but it seems to me that the two problems are similar: shifting a signal down a known amount in the spectrum. Any ideas or thoughts?
tell@oscar.cs.unc.edu (Stephen Tell) (08/07/90)
In article <1990Aug06.150222.23167@pmafire.UUCP> tuv@pmafire.UUCP (Mark Tovey) writes: > > In article <1990Jul20.223615.4305@portia.Stanford.EDU> ceg@nova.stanford.edu >(Chris Gronbeck) writes: >> >>I'm looking for suggestions as to the best way to design a circuit to >>drop music >>one or more octaves. A typlical application being making an electric guitar >>perform like a bass. Any hints would be useful. Thanks. >> >> > My memory is very dim at this point but I recall reading years ago in >something like Popular Electronis or something like that about plans for >an ultrasonic sniffer. This device utilized the receiver transducer for >the old style television remote control units to pickup ultrasonic sounds. >This is where my memory gets real hazy. If I remember correctly, the signal >was passed to a circuit that resembled the front end of a radio receiver and >reduced the signal down to some intermediate bandwidth. From here it was >converted to normal audio and passed into an audio amplifier. This is perfectly reasonable thing to do; I recall a similar article fairly recently. The device is also useful for listening to bats. You are correct in your description until you mentioned "bandwidth." Substitute "frequency" for a true statement, or read on... > I have no idea how well it worked or how to go about building one, but >it seems to me that the two problems are similar: shifting a signal down >a known amount in the spectrum. Any ideas or thoughts? Since no one else has jumped in, I'll try to remember my communications theory class of a few years ago. The action of the heterodyne converter you described above is qualitatively different from the "reduce by one or more octaves" function. The heterodyne converter "mixes" the input signal with a constant sinewave in a nonlinear element, ideally, a multiplier. The result is a signal at the frequency (or frequencies) which are the _sum_ and _difference_ of the input and the constant sinewave. Note that the only change in "bandwidth" that has occured is a doubling because both the sum and difference are involved. This is essentialy AM modulation. One of the two "sidebands" can be discarded and the result can still be demodulated ("single-sideband"); the bandwidth doubling is not a required. For more info: Any good communications theory textbook for EE's. (warning: calculus required) an OK one: Roden, Martin S. _Analog and Digital Communication Systems_, Prentice Hall 1979, 1985. ISBN 0-13-032822-7 There are likely better ones. Summary: mixing with a local oscilator results addition and subtraction of frequencies. Now, in music (and elsewhere) the boundaries of an "octave" differ by a factor of two; the required operation to reduce a frequency by an octave is a division. If you were to take an octave-wide chunk of the audio spectrum, say from 1000 to 2000Hz, mix it with a 500Hz signal and filter appropriately, an 1000Hz input would be reduced an octave to 500Hz. but, a 2KHz input ends up at 1500Hz, which is not an octave down. One octave has become more than an octave. That doesn't make for music that sounds right. The technique of clip, devide with a flip-flop, and filter will reduce a single signal by an octave. It will also introduce a lot of distortion, which may not be bad if you're playing certain kinds of guitar music :-) To do better, you might take a set of notch filters and device up the spectrum of interest into lots of little pieces. In each one, split the signal two ways. One path gets clipped, and has only frequency information. Rectify and filter the other to get amplitude information. Devide the frequency information by two with a flip-flop, and use a voltage-controlled-amplifier controlled by the amplitude half of the signal to get the amplitude right. Finally, sum up all these little parts of the signal and you've reduced the input by an octave. For better results, split into more frequency bands. I would guess that eight or ten bands per (input) octave might even sound good. For perfect results, split into infinitely many frequency bands. This is essentialy the same as taking the fourier transform of the signal, changing the frequency information, and taking the inverse transform, which someone already suggested. This is done with a digital signal processor. One last method, for guitars anyway: Sell the guitar, and get a MIDI guitar controller, which you play like a guitar and it produces MIDI signals corresponding to the notes you hit. Run the MIDI data through a computer to mess with the note data and drop things down an octave, and then to a synthesizer set up to sound like a guitar. Some synths can probably do the translation without an external computer. (1/2 :-) Sorry this got so long, but I hate to see incomplete information go uncompleted. Someone may get really confused. Someday it may be me who needs the info. -------------------------------------------------------------------- Steve Tell e-mail: tell@wsmail.cs.unc.edu usmail: #5L Estes Park apts CS Grad Student, UNC Chapel Hill. Carrboro NC 27510
vekurpan@tekred.CNA.TEK.COM (Vincent E Kurpan) (08/07/90)
In a previous article an author suggests using a mixer sceme to convert frequency. this can not be used for anything like this because it will only subtract frequency and not divide. If you did this the spacing between the notes would be all wrong and would sound very strange... unless you had a different channel for each note in which case you might as well build the whole source. Unfortunately, the simplest scheme may be to digitize one cycle of the waveform and then play back with a clock that is synthesized for each note. This is not totally trivial and works only one note at a time. Theres no free lunch.
weimin@zeus.ece.jhu.edu (weimin liu) (08/08/90)
In article <32424@cup.portal.com> R_Tim_Coslet@cup.portal.com writes: >The only way I know of that will definitely do what you want is a >Digital Signal Processor doing a Fast Fourier Transform on the input, >divide the "frequencies" obtained by 2, and inverse Fourier Transform >to generate the output. This will change the "color" of the notes, i.e., the resonant properties of a particular instrument. What you really want is to change the fundamental frequency (f0) but not the resonant frequencies. I don't know much about music but for speech there are several methods, such as linear predictive coding (LPC) and cepstrum (not spectrum) analysis, to separate f0 from the higher resonant stuff. They did something like this once on Late Night with David Letterman in which Paul Shaefer's voice pitch was significantly lowered through the microphone. weimin@zeus.ece.jhu.edu
paul@hpldola.HP.COM (Paul Bame) (08/09/90)
I think Rat Shack still sells their tape recorder which allows you to up to double the playback speed of a tape and have the pitch corrected. I believe they use the technique described years ago in one of the popular electronics magazines using two "bucket-brigade" audio fifos and some fancy clocking which amounts to some type of re-sampling. I'm surprised some DSP type hasn't chimed in to describe this in detail. -Paul "Spice is the Variety of Life" paul@hpldola.hp.com N0KCL
bobt@pogo.WV.TEK.COM (Bob Tidrick) (08/10/90)
In article <6101@tekred.CNA.TEK.COM> vekurpan@tekred.CNA.TEK.COM (Vincent E Kurpan) writes: >Unfortunately, the simplest scheme may be to digitize one cycle of the >waveform and then play back with a clock that is synthesized for each >note. This is not totally trivial and works only one note at a time. >Theres no free lunch. This is not exactly true. When one waveform is digitized it can be played back at half speed. If you are sampling at say 50 Khz send it out again at 25 Khz and you will have the wave form at half speed. A reasonably fast micro processor should be able to keep up with the playing. All you will lose is half of the waveforms. but you will need to lose these anyway. What needs to be done is sense the highest peak of the wave form and look for it again. This should give you one period. send this to a circuit that will clock it out again at half of the sample rate. Another waveform can be caught while this one is going out. Higher notes will take less samples and lower ones more. You would want to be sure to allow enough memory in your playback circuit for the lowest notes. Something similar to this has been done to compress speech. The wave forms are put out on a tape running at half speed. When it is played back the speech is at the proper frequency but twice as fast. Just my $.04 worth (inflation you know) -- Bob Tidrick GPID Engineering Tektronix Inc. Wilsonville OR.