Modeling analog circuits in DSP?

[abraxalito]

Originally Posted by rsdio

If I stimulate with an impulse, then I should be able to use the response file to create an FIR filter that implements the exact transfer function of the analog circuit in real time. Mathematically, this FIR should be identical to the analog circuit except for any truncation in the response.

I think that's a really nifty technique and I've been playing with it myself in MATLAB. It will work (assuming the filter length is sufficient) for a circuit which is LTI (like your FM transmitter filters, which might be elliptical btw) but you also mentioned transistors and diodes...

[rsdio]

Originally Posted by counter culture

I was referring to Scilab or Octave which are alternatives to Matlab. I think you need the supporting environment to make use of the filters in a simulation without a lot of effort. I believe Octave is closest in terms of interoperability of code, but I haven't had the need to explore either.

Thanks again. I run across Matlab all the time in my research, and I'm mindful that I might need to someday purchase a license. Instead, I tend to write my own software every time I need such calculations, but that can be time-consuming compared to the Matlab environment.

I'll take a look at Octave and Scilab. Good to know about these alternatives, as I'd never heard of them before.

[hochopeper]

I like where this is going and have had similar ideas previously but lacked programming skills/time to implement.

Another open source alternative to matlab is Python with iPython and the SciPy/Numpy/matplotlib tools.

The SciPy library has some filter design tools that seem as though they may be handy, documentation are here: Module SciPy.signal.filter_design

[rsdio]

Originally Posted by hochopeper

I like where this is going and have had similar ideas previously but lacked programming skills/time to implement.

Another open source alternative to matlab is Python with iPython and the SciPy/Numpy/matplotlib tools.

The SciPy library has some filter design tools that seem as though they may be handy, documentation are here: Module SciPy.signal.filter_design

It's entirely possible that Python might host a solution, but I would like to clarify that I am not looking for filter design tools. Filter design tools start with specifications and end up with code or circuits. I can find plenty of tools - including instant web tools online - that will design either an analog circuit or a digital filter that meets a given set of starting specifications.

What I'm looking for are tools to analyze an existing (already designed) circuit for which the original design specifications may not be available, but where the individual analog components are known (either via schematic or reverse-engineering the circuit from the PCB layout and probing around with a meter). That's not something you can do with a standard filter design tool.

[funkathustra]

MATLAB's filter design and analysis toolkit (fdatool) is bulletproof for this stuff. You select the type of filter, the parameters for it, and hit "generate" -- it'll spit out all the coefficients, already broken down into biquads (which is what's usually used for actually implementing filters on DSPs). The process is bulletproof.

You didn't specify the DSP platform you're planning -- is it embedded, or running on a computer? Getting a filter going in software on a computer is pretty trivial; getting it implemented on an embedded DSP processor is a little more convoluted and requires an embedded skillset that most people don't have (do you know what a debugger is and how to use it? How familiar are you with writing straight ANSI C and/or assembly?)

[triode_al]

Originally Posted by rsdio

It's easy to transcribe an analog circuit into SPICE and get a plot of the frequency response, impulse response, distortion or other useful measurements. What I would like to do is model a particular analog circuit in DSP for real time processing of audio.

I today stumbled on the site about TRASH, a utility that allows tweaking and modeling of classic amps, distortions, delays and filters to get a specific flavor of sound.
In fact it feeds into a software DSP from iZotope (running on the PC/Mac).
And the software is expensive (at 200 USD) For that you can now 'spoil' any recording into your beloved sound style, based on e.g. amp modeling (including over-driven tube stages, just missing is 'rolling of tubes'), sweepable filters and delays.

The direction is there.
Now the next step is have a module and feed that into a board from e.g. XMOS or MiniDSP.
MiniDSP has a utility to calculate frequencies, so though a lot of manual, it should be possible to insert many types of circuit.
albert

[s3tup]

Check this out, especially the bilinear transform section:
http://www-sigproc.eng.cam.ac.uk/~op...IR_Filters.pdf