Subject: Re: [linux-audio-dev] Re: Exact desicption of PCM system for ALSA
From: Alexander Ehlert (alexander.ehlert_AT_uni-tuebingen.de)
Date: Fri Jul 12 2002 - 15:56:01 EEST
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
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