Hi Philipp, > But it sounds like an disordered signal and I don't > understand how to unfluence the volume or the of frequency this signal. > I actually have an algorythm for sine signals but it's too complex > for undestanding.
It seems like you're missing the basic understanding how digitial audio works at all. For example take your CD player at home. The music has to be recorded digitally in a studio in the first step. The analog signal of the music is quantized in a certain resolution. That can be an arbitrary number of bits, usually 8, 16, 24 bits. For a CD that's 16bits. Thus the former analog signal is now represented by 2^16 different voltages or given as a signed integer you have a possible range between -32768 and 32767 for your audio signal or in other words, that are the maximum values for the amplitude of your sound signal. Ok, now I could go on about the sampling frequency and other stuff. But I'm too lazy :*) > Signal description system which is also used in Windows Soundcard > interface libraries ( or not? ). Perhaps someone knows good (E)books > or tutorials about that. Have a look into the source of small audio applications. And for basic information search the internet. A good resource for dsp/audio programming is http://www.harmony-central.com Cheers, Alex
