Well, depending on the sound I'd say that 4 to 5 whole notes (with the sharps / flats in between) can sound acceptable to be sourced from one pitched sound. That is, half the way down and half the way up.
It will probably still sound pretty unrealistic, but one has to consider memory usage, so if you're going to have sixteen simultaneous patches (with the percussion channel of course needing to have one sound per key), this seems like a reasonable solution.
Certain sounds (like the power sawtooth in my example) actually use just about one sample per octave I think and still sounds prtty well, but the same naturally doesn't work for "unsynthetical" sounds such as a piano.
Actually most professional samplers have several samples per key (for example a piano instrument I have uses 574 samples with a total filesize of almost 2Gb).
Naturally that leads to the result that you can't have very many patches loaded at once unless you have a great deal of RAM available. It is only recently that a solution has been created (that works properly, that is); DFD (direct from disc).
This is a very complicated process, seeing as it took professional audio software creators years to get it to work satisfactory from what I've heard, in which you load a small buffer of each sound into memory. Then whenever a sound is actually played, it loads the rest of it in in realtime.
That's something I think everybody around here would applaud if you managed to do in DBPro
(No seriously, don't bother, it would need tremendous speed).
(On a side note, this DFD method requires a lot from the processor instead).
If you mean vst's that doesn't use samples stored somewhere in external files, they actually generate their sounds themselves during runtime (mimicing an analogue synthesizer, begining with a simple waveform and then applying filters to this in a effect chain).
About the clicking, I don't think I've ever experienced that myself
As far as I know, I didn't do anything special, at least not with the intention to solve that, in this project.
For altering the frequency, you would want to multiply / divide the previous frequency with 2^ln(1 / 12). It gives the correct frequency difference between each halftone, so yeah, you'll get the sharps and flats as well.
Hm... getting a delay when changing the speed?
May I have a look at that function?
About trying only to change the pitch, but not the speed - this is possible through some complicated digital calculations.
Believe me when I say that the result is worse than the standard frequency change (it basically means that, should you wish to set the pitch to half the initial value, every sample (here I mean the actual byte / word / etc. data of each block within the audio data by "sample") would be written twice, hence really "digitalizing" the sound, if you get what I mean).
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