LOL. Going rate? For the MN3007 I suppose it's a seller's market. Some companies (here in the US) have been known to sell them for $1.95 which by itself ain't bad-- but then they have a $250 minimum order that *is* bad. Some companies in the UK sell them for $9.95 each (British equivalent); and on eBay they can be anything from $0.99 cents each up--- not including shipping charges. But since this IC has a bad reputation for noise, there are newer and better ICs that do the same thing, and it's sure not a hot item any more I don't suppose anyone would pay too much for them.
** Hate to be the one that breaks the news - but that device is NOT digital.
The technique used is called " delta modulation " and is strictly an analogue method - high audio quality is barely even possible and needs a clock frequency in the MHz range.
There is NO conversion of a changing quantity to a binary number stream as is the case with all modern digital audio systems.
Compared to the BBD chip the OP is after - that is no practical alternative whatsoever.
Although I realize the above comments are all in fun, please note that the OP wants to build a chorus unit. The "conventional" way to do that is with a *variable* delay line, where the delay is modulated (at several Hz). That would be tough to do with the above methods, which are
*fixed* delay lines (though they make dandy reverb units).
The phaser/flanger approach I mentioned in my prior post skips the actual delay line and just uses the fact that when you combine multiple versions of an original, each version having a different (and variable) delay (ie, voices or instruments trying to track each other but not quite making it), what you hear sounds a lot like the original with a bunch of roving peaks and dips. So it just creates the peaks and dips, sans delay.
Though I suppose you could use the same technique to create a real variable delay, by cascading a bunch of Bessel LP stages instead of all-pass stages, I've never heard of anyone doing that. I suspect you get a lot less "effect" bang for the buck.
Best regards,
Bob Masta DAQARTA v6.02 Data AcQuisition And Real-Time Analysis
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Scope, Spectrum, Spectrogram, Sound Level Meter Frequency Counter, FREE Signal Generator Pitch Track, Pitch-to-MIDI Science with your sound card!
Well, it's certainly "digital" (data stream is all 1s and
0s), just not the conventional Pulse Code Modulation (PCM) that produces a binary number stream.
The given circuit is a crude version of the Sigma-Delta technique (formerly known as Delta-Sigma!) which is used in all modern digital audio systems. The main difference for audio quality has to do with the sophistication of the feedback from the encoder output back to the input. Modern converters do indeed operate in the MHz range, then down-convert to "normal" audio sample rates when they convert to PCM.
Agreed. In the "olden days" I built a simple converter like this (but a tad more sophisticated) for the purpose of making a reverb. But without a fancy feedback scheme (or a ridiculous clock rate) you get all sorts of whistles and "birdies". As I recall, these sounded like undersampling aliases (which I guess they really were) of a PCM audio without the anti-alias filters ... *far* more annoying than the BBD noise floor.
Best regards,
Bob Masta DAQARTA v6.02 Data AcQuisition And Real-Time Analysis
formatting link
Scope, Spectrum, Spectrogram, Sound Level Meter Frequency Counter, FREE Signal Generator Pitch Track, Pitch-to-MIDI Science with your sound card!
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