FPGA Central

hello,

i'm trying to program a wavetable-based synth in c++. I have som
waveforms and different envelopes for them and want to switch betwee
settings while playing. I use a 4096 samples buffer. My question is, ho
do i now implement anti-aliasing? I thought of multiplying envelope
waveform, and then lowpass and the resample with linear interpolation at
lower sample rate. Is it right, that this has to be done, everytime
change a note or change the setting for waveform /envelope or pulse width
can anybody recommend a (phase linear) antialiasing filter? i would think
-24dB filter is not enough.

thanks for any suggestions

tom

i did some research and it seems i need some kind of Crochière/Simth
Gosset interpolator. Is there a source for the coeffcients for the filte
table?

tom

On Feb 12, 7:32 pm, "mudskipper" <tomacm...@freenet.de> wrote:
> i did some research and it seems i need some kind of Crochière/Simth &
> Gosset interpolator. Is there a source for the coeffcients for the filter
> table?

dunno what kind of tools you have. (MATLAB, maybe?) probably for your
purposes, a windowed sinc() function would be okay. a Kaiser window
would be best but a Hamming window, which is easier to compute, might
be good enough. for a primer on the theory, maybe take a look at:

http://groups.google.com/group/comp....ef1e2580?fwc=1

r b-j

mudskipper wrote:
> i did some research and it seems i need some kind of Crochière/Simth &
> Gosset interpolator. Is there a source for the coeffcients for the filter
> table?
>
> tom
>
Julius O. Smith's site has a fairly detailed description of how to
compute the filter coefficients.

http://ccrma-www.stanford.edu/~jos/resample/

If that isn't enough info, then a good place to start is looking at an
implementation of the algorithm. libsamplerate implements this
algorithm, and will show fairly well how to implement the algorithm in
the way you need it.

Regards

Marc