BBD Questions

Discussion in 'General Electronics Chat' started by tesseract, Aug 14, 2009.

  1. tesseract

    Thread Starter New Member

    Aug 14, 2009
    Hey guys. This is my first post, but I'm sure there will be many more.

    I'm a musician, and over the past couple of years I've been experimenting with building my own effects (stompboxes and the like). I don't have any formal education in this area, but so far, I've managed fairly well on my own. That said, I'm starting on a new effect (still in the pre-planning stages), and I have a few questions.

    The effect is an analog audio delay. (I'd prefer to implement the effect as a digital delay, but I think I would need to use a full-blown DSP for that, and that's just beyond my means at this point.) I've done some reading, and I've learned a bit about how I should go about it, but I'm a little fuzzy on BBDs. I can't really find much helpful info on the internet, so I'm hoping that some of you have had some experience with BBDs and will be able to shed some light on the subject for me.

    So here are my questions:

    1. What would be a good BBD to use? None of the major distributors (Mouser, DigiKey, etc.) seem to have ANY in stock, so I'm not sure what kind of BBDs are out there or how they compare to each other. The Panasonic MN3205 looks good, but I haven't been able to find a pinout or much info aside from an old basic datasheet. (Keep in mind that this is for an audio delay effect, so a high maximum delay time [up to 1 or 2 seconds] is preferred.)
    2. How difficult is it to use a BBD for this purpose? Can I just connect an analog signal source and a clock to the chip, then run the output through a low-pass filter, or is it more involved than that? (I understand the concept of feedback, but I'm just concerned with delaying the signal for now.)
    3. Can I use any clock source, or do I have to use a special BBD driver?
    4. Do most recently produced BBDs have their own LP filter built in, or will I need to make my own? Also, it seems to me that the cutoff frequency for the filter should be the same as (or slightly lower than) the frequency of the clock. Is this so?
    5. I've noticed that some analog delays use a compander as well. Is this a necessity, or does it just make the signal smoother/cleaner somehow?
    I think that's all for now. I'd greatly appreciate any help or advice, even if it's not directly related to my questions. Thanks in advance.
    Last edited: Aug 14, 2009
  2. KL7AJ

    Senior Member

    Nov 4, 2008
    :( Alas the SDA512 which was a readily availlable BBD (you could even get it from Radio Shack) has been discontinued. I've seen a couple for sale at obscenely inflated "vintage" prices, but nothing remotely reasonable.

    I think your best bet is sythesizing the action with a FIR filter in software. Not a trivial task, but it's been done.

  3. tesseract

    Thread Starter New Member

    Aug 14, 2009
    So you're saying a delay effect could be achieved with just a FIR filter?

    I'm certainly open to going the digital route (I've even done some PoC stuff with a PIC MCU), but I just don't have the time or money to invest in a DSP-based solution. If I could do the same thing with some digital ICs, though, that would be perfect.
  4. beenthere

    Retired Moderator

    Apr 20, 2004
    To implement a variable delay, you should be able to digitize the audio and store it into RAM with a microcontroller. The delay gets implemented by letting the digitized audio sit for a period before reading it out and applying it to a D to A converter. Not quite a full-blown DSP situation. Should be simple to set an input thumbwheel for the delay.
  5. tesseract

    Thread Starter New Member

    Aug 14, 2009
    I might end up using an MCU anyway, but I'd need to find a good ADC first. I'll do some searching, but would you happen to know of a good parallel ADC for audio (around 40khz, 16 bits or so)?
  6. rjenkins

    AAC Fanatic!

    Nov 6, 2005
    Have a look at chips like the Microchip DSPIC30F4014

    This has built-in 12 Bit (at up to 100KHz sample rate) ADCs and a high speed 16 bit PWM which you can use with simple filtering as a D to A.

    It also has an interface for a PC standard AC97 audio device which I believe would give you full 16 bit stereo 44/48 KHz facilities.

    (I'd try the internal 12 bit first, for effects it should be pretty good).

    That DSPIC device only has 2KB RAM internally so you'd only be able to do 'short' effects, but using some spare I/O pins with a couple of latch chips and an external static RAM (eg. 64 or 128K) to expand the buffer memory, you could do a decent echo and many other effects.

    The advantage of the PIC series is that a lot of development software is free (Microchip MPLab), you can get cheap programmers / in circuit debuggers (PicKit 2) and quite a few devices are available in 'conventional' (ie. not surface mount) packages which means you can prototype easily and cheaply.

    There are many other options, but that would be my first choice to try.
  7. t06afre

    AAC Fanatic!

    May 11, 2009
    You will find a lot of good ADC/DAC units for sound. The sound card industry use them a lot. 24 bit should not be a problem.