🚀 go-pugleaf

I've been reading about some of the low-level mechanisms used in the
synthesis of speech.  As so often happens when I stray away from my areas
of expertise, the jargon is a little out of my domain and references to
simpler topics have been difficult to track down.

Answers, or pointers to answers to the following would be appreciated:

a) What is a "second order" filter?

   From contextual information, am I correct in assuming that it is a
   bandpass filter that is defined in terms of base frequency, amplitude,
   and bandwidth?

b) For the purposes of insight, I would like to model its behavior with
   software.  Can I model the desired behavior as a series of iterative,
   incremental calculations instead of (apparently) using a reverse
   Fourier Transform?

   example:

	Something like what I want:
		sample[t0] = filter(sample[t0], freq, amp, bandw)
		sample[t1] = filter(sample[t1], freq, amp, bandw)
		...
		sample[tN] = filter(sample[tN], freq, amp, bandw)

	What I can do now, but not what I think I want:
		generate(input, N, freq, amp, bandw);
		fft(input, N, output)


c) What do I read to learn more?  There must be a definitive reference work
   on these sorts of topics...

email responses preferred, I'll share info with those that ask.

Thanks.

	-- Andy Pfiffer
	   "...I'm just not that bright." -- Homer Simpson