To implement square waves, I'm using a wavetable that literally jumps between -1.0 and 1.0. I know this isn't how squares work in the physical world, and it certainly doesn't sound very good. I'm wondering how un-ideal the typical square wave is in an analog synth. I know you can approach a square b...

I've seen them going for 400-500 USD in Japan.
Having yet another analog poly always seemed redundant to me, but if I knew the markup was that insane in other countries, I'd get a little hustle going.

I saw an interview with Steve Albini and this precise concern seemed to be the root of his analog fundamentalism, as opposed to concerns over audio itself.

Sorry for bumping my own thread, but I thought it might be worthwhile to throw in additional observations since people sometimes find this board through search engines. I've done more experimenting, and the algorithm I've described indeed modulates pitch periodically. There needs to be an additional...

Hello comrades, I'm back with another general synthesis question. I'm not asking to be spoonfed, but any knowledge/suggested reading material that will nudge me in the right direction is greatly appreciated. At the moment I'm using wavetables with linear interpolation to represent oscillators, and e...

Thanks for the answers. I have a much clearer idea of where to go in my programming now. I already have a variable that increments for every sample-writing cycle and "timestamps" a given component with the last cycle it was active, so it should be easy to calculate the phase of waves without actuall...

Sorry, I meant "what frequency is a VCO oscillating at before any key is pressed and a sample and hold circuit locks it into the last played pitch's frequency?" I'm guessing something like this varies between hardware, but is it generally higher/lower than the keys' pitches? Also, when simulating a ...

That's pretty cool. What's the oscillator doing before any keys are pressed though?

So I know lots of LFOs are free running, where they are always cycling through their waveform regardless of whether or not a note is being played. This behavior is easy to observe since the F of the O is so L, and the frequency is usually independent of any pitch that is played. I think I've read be...

Well I'm a remarkably stupid person sometimes. It seems that somewhere in the header file, the buffer that writes each byte to the soundcard was inadvertently changed from type uint8_t to uint32_t. Can't believe that slipped past me. For whatever reason, storing a near-silent 1 byte amplitude value ...

Thanks for the reply. It's good to know that my stopgap measure of returning a straight 0 is ok. But you're right; I shouldn't be able to hear the dither noise anyway. I put some printfs in the function and it looks like for inputs of 0.0, the final dithered int sometimes wraps around to the neighbo...

So I'm on/off writing an audio program in C, just for learning's sake. All internal sound values are 32 bit floats, but they are converted to 16 bit integers before being written to the sound card. I've been told that I need to dither so that I may avoid distortion caused by rounding errors. I've us...

I kinda regret getting an AX-80 over a 60. At the time they were both floating around 400 USD and figured the one with more oscillators would automatically be better, not realizing the 60 is a whole lot more interesting.

Now I see them listed for considerably more. I blame OPN.

Thanks for the info. I imagine I'll get into filters soon enough as I'm planning on doing lots of audio programming over the break, but I've found out why my pitch-shifting was so noisy: a sign error! The PCM should have been read, manipulated, and written as signed rather than unsigned. So I'm afra...

I wasn't aware of that. I'm kind of stumbling through all this blindly. Maybe that's the next direction I should head in, thanks.