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On 6/2/12 4:40 AM, Benjamin S. wrote:
> I'm familiar with stationary (FIR/IIR) and adaptive filtering. My matlab
> skills are OK but I've never done any practical DSP in matlab. There is a
> book called "Digital Signal Processing using Matlab" by Ingle and Proakis.
> It starts with very simple things which I find extremely boring so I
> wonder whether it's the best way to learn how to process some signals.
>
> Any useful tutorials or advice would be welcome.
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On 6/2/12 1:18 PM, Fred Marshall wrote:
> I find the signal processing tools in Matlab to be hard to understand
> and, thus, use. Part of that's because I would so rarely use them and so
> I rarely use them. Positive feedback! ... or is that being negative??
>
> So, I just write my own functions. It's not that hard and then you know
> what you've got.
>
> For filter design, I use other tools and transfer the coefficients into
> Matlab.
i found that for FIR filters, both firpm() (used to be called remez())
and firls() work pretty good for design. and also the Kaiser window
(together with fft()) works pretty good for the "windowing method" of
FIR design.
they should have a berchin() function along with prony() to design IIRs
in the signal processing toolbox.
i wonder what Ingle and Proakis would say about the STUPID FUCKING
requirement that all indices are positive (not ever zero, not ever
negative). how do they integrate that inflexible indexing convention
into the accepted and concise convention we use in DSP that has negative
and zero indices, both for samples and for FFT bins? and about the
latter, *what* do they tell the reader about the DC bin in the FFT? is
it X(1)? totally stupid, and you would think these authors would start
rattling Cleve's telephone and tell him how that unfixable indexing
convention makes it harder to integrate MATLAB with their DSP book.
--
r b-j rbj@audioimagination.com
"Imagination is more important than knowledge."
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