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you're right, though rewriting this for double shouldn't be that complicated

okay, so i guess this is more complicated than expected...

i think i will redo from start with some tutorials like this one here: http://www.dsptutor.freeuk.com/ and see if this will help me to understand the basics. it's very hard to take the first steps in signal processing. but i'm quite happy what i already can produce with my currently limited set of functions.

this leads me to another question: is it common when designing a 4k soft synth to have all of this stuff including filters and a complete sequencer, patterns and such? i could imagine that some people rather build up a small set of oscillators modifiers and filters and simply hardcode their sound? i believe its painful to implement a complete toolset (editors are needed for the synth modules, pattern and sequencer, etc)

thanks

hi!

thanks a lot for your pointers... i have read them all already, and i'm still stuck somewhere. so maybe i need to precise my questions (sorry for the late weak post yesterday)

i found some example stuff and i'm sure that i understand what they are doing, but i'm not sure how to 'use' them correctly.

like:
short filter_fir(short s, float a0, float a1, a2)
{
     return signal(s) * a0 + signal(s - 1) * a1 + signal(s - 2) * a2;
}

okay, nice. That's some kind of filter. Good, but how to set the coefficients on a (unknown) stream that was generated in advance by lets say a mix(sinus(440), square(200)) or whatever signal (anything is possible). now i need to do / am requested to do:  implement a high pass and low pass filter. so i need so set those coefficients, right?

thanks for your help!

cheers

 

hi folks!

i'm currently working on my first software synthesizer... i run into more and more problems right now, said, tackling them for days...

what i have: i coded a modular synthesizer that can make some oscillators (sine, square, noise), i have ADSR envelopes, mixers, etc. i can do filters like echo and stuff. pretty much fun

now for what i don't understand: how to design frequency based filters like high and low pass? from theory, it looks like doing a FFT, masking out the filtered frequencies and finally doing some kind of inverse FFT. am i right?

but somehow i have troubles with doing this (beside understanding how FFT works). some more questions:

how does a standard 4k high/low pass filter work? how is it implemented?

are there any good resources for getting the 100% basics for coding this effective? I had no success in finding anything good over the net (even musicdsp has their forum down, so no chance)?

i found some stuff like FIR filtering, but even this didn't give me any clue... there's one nice tutorial at in4k.untergrund.net but i don't get it...

so, you see, i am in trouble, or maybe even stuck somewhere i hope you guys can point me into the right direction.

okay, so for the last question: what i understood is that in a sequencer, the musician is playing notes. those notes are basically my samples, so i need to modify my oscillators in order to get the correct sound, right? pitching may destroy the whole sound...

or am i overengineering my whole stuff?

best wishes