Building a synthesizer with no prior experience

Question

I've got an extensive music background and have been infatuated with synthesizers since I was young. I had dreamed of going into an EE program to make my dream a reality, but I couldn't afford school (long story).

Basically, I'd like to build a 3 oscillator analog synthesizer, but I have zero experience building boards. I know basic signal flow and process and have a strong design sense, but I really lack the knowledge otherwise to get started.

I think starting simple with a 1 oscillator and building upwards would be an awesome learning experience, and I would love to get to the level that some of you are at and actually make a fully usable synthesizer.

My question is, for audio-oriented electronics, where can I get started for this full analog synthesizer? Am I reaching too far? What will I need to make this happen?

Answer 1

There is plenty of interesting info at the Music From Outer Space website. Including circuit schematics accompanied by detailed explanations of the designs. As well as a range of advice and links relevant to building synthesizers.

You'd be better off first gaining experience with some simpler audio electronics projects, and building up the skills and equipment that a major project like this would require. Then you'd be in a better position to evaluate how/whether to proceed further.

Answer 2

Another place to start is the TB303 schematics! the machine that started it all :) http://machines.hyperreal.org/manufacturers/Roland/TB-303/schematics/

Start off by understanding a portion of the circuit and go ahead and design something based on it on small piece of breadboard, and build up from there.

Once you have a few of the elements working, try designing your own pcb with ground planes (this will improve the noise levels) or check out some kits as Rob has already linked to.

Answer 3

I'd also recommend they synth DIY pages at http://www.yusynth.net. That's a great resource I've used for my synth projects.

EDIT: Sound Synthesis: Analog and Digital Techniques by Terence Thomas. Awesome book. First chapter starts with how to build a variable output power supply, then moves into a "test box" which is basically a speaker, a couple pots and a breadboard so you can test before soldering.

Copious schematics and foil diagrams throughout, and an excellent explanation of why certain components are placed where they are to achieve particular results. HIGHLY recommended!

Answer 4

I would suggest that even if you want to use analogue filtering stages (they can give sound a warmth that can be hard to achieve via other means) it may be a good idea to generate the starting waveforms digitally. Many Williams' Electronics arcade machines in the 1980's generates sound using a board that contained a 6800 microprocessor, a small amount of RAM and ROM, and a little bit of I/O including a DAC. All of the sound effects were generated using tight program loops which generated samples and fed them to the DAC. Since the processor was used for nothing but sound generation, loop execution speed could be used for timing.

In practice, even the simplest microcontrollers have some sort of timer resource, which could be helpful if you want to be able to change the audio parameters while playing sounds. Using something like 6805 code, one would start by writing a poll routine for each voice; for speed these routines would live in RAM--something like:

poll1:
        brclr TMR_CONTROL,TMR_READY,poll ; Wait for start of next 'tick'
        bclr TMR_CONTROL,TMR_READY
FRQ1L:  lda #PATCH
PH1L:   add #PATCH
        sta PH1L+1 ; Patch value for LSB of phase
FRQ1M:  lda #PATCH
PH1M:   add #PATCH
        sta PH1M+1 ; Patch code
FRQ1H:  lda #PATCH
PH1H:   add #PATCH
        sta PH1H+1 ; Patch code
        sta FETCH+2 ; Patch LSB of target
FETCH:  lda TABLE_BASE ; 16-bit address
        clr DAC_ENABLES
        sta DAC_OUTPUT
        lda #ENABLE_1
        sta DAC_ENABLES
        rts

Next, one would have a main loop which would repeatedly call the poll routine for each voice in sequence and, between calls, perform whatever other logic needed to be done (e.g. seeing if any voice parameters needed to be updated). Using this approach, it's possible to update a fair number of voices with a high sample rate.

While it's possible to do the initial wave generation entirely using analogue circuitry, it's difficult to have multiple independent analog generators whose frequency characteristics are absolutely identical within a fraction of a percent. The human ear is very sensitive to variations in pitch--far more so than to variations in amplitude--so whatever is used for signal generation must be very consistent. Using a simple microcontroller as a starting point is a good way to get such consistency, even if one then feeds the generated signal through analog shaping circuitry.