home *** CD-ROM | disk | FTP | other *** search
- Newsgroups: rec.audio
- Path: sparky!uunet!think.com!ames!data.nas.nasa.gov!win31.nas.nasa.gov!fineberg
- From: fineberg@win31.nas.nasa.gov (Samuel A. Fineberg)
- Subject: Re: Why filter D/A output?
- References: <1992Nov8.222911.27702@doug.cae.wisc.edu> <1992Nov20.000415.1811@lugb.latrobe.edu.au>
- Sender: news@nas.nasa.gov (News Administrator)
- Organization: CSC, NASA Ames Research Center, NAS Division
- Date: Fri, 20 Nov 92 21:19:18 GMT
- Message-ID: <1992Nov20.211918.15482@nas.nasa.gov>
- Reply-To: fineberg@nas.nasa.gov
- Lines: 43
-
- In article <1992Nov20.000415.1811@lugb.latrobe.edu.au>, MATGBB@LURE.LATROBE.EDU.AU (BYRNES,Graham) writes:
- |> In <1992Nov12.201144.28397@en.ecn.purdue.edu> syd@en.ecn.purdue.edu writes:
- |>
- |> > philg@martigny.ai.mit.edu writes:
- |> >
- |> > >Driving a nonlinear, i.e. distorting, system, i.e. amplifier/speaker,
- |> > >with a single frequency produces an output made up only of that
- |> > >frequency and its harmonics. This can be shown easily by looking at a
- |> > >Taylor series expansion of a nonlinear function and seeing what
- |> > >happens to cosine squared and cosine cubed terms (use trig
- |> > >identities).
- |> >
- |> > This is [mathematically] not true in general.
- |> > Example: f(x) = arccos (x). The output of such a system
- |> > when x = cos(t) is simply t, which is not periodic, and
- |> > hence does not have a Fourier series expansion.
- |> >
- |> > One could be more obnoxious by letting f(x) = cos (1.5 arccos(x)).
- |> >
- |> > The Taylor series must converge at every point in order for this
- |> > argument to work. (Note that arccos (x) does not meet this requirement.)
- |> > A sufficient condition for this is that the nonlinearity be continuous.
- |> > (In the real world, this is probably a valid assumption.)
- |> >
- |> > --
- |> > Dennis Hilgenberg
- |> > syd@ecn.purdue.edu
- |> > LaRouche in '96
- |> I guess we can at least assume that the transfer function is bounded
- |> over any compact interval ? :-) One might even hope that it would be
- |> continuous....so we can use Taylor's theorem to get an arbitrarily
- |> accurate approximation over the domain (ie possible input values).
- |> Cheers, GB
- |> (Yeah, I know, what a wank)
- Actually you're both wrong. The reason why we need filtering is because
- real functions are sampled over finite intervals, i.e., we can't really
- produce a delta function. This causes distortion, however, it is all outside
- of the frequency range of the original function (remember we filtered the
- original function to prevent aliasing). Try sampling and recovering a
- signal, then display it on a scope or spectrum analyzer, it is pretty obvious.
- (Dennis, you should have done this or will do this in EE440 lab).
-
- Sam
-