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Hi. I've working with al TMS320C613 DSK building some guitar effects.
I was suscesfully whit all of these but I had no idea about build a
valve guitar preamp on DSP. In other forums, people said I have to
decimate, interpolate and check out curves in the valve 12AX7's
datasheet and process the signal with that. The thing is .... I'm
still have not much idea about it. 

Is someone could give me some tips about it?....Thanx a lot

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On Tuesday 26 September 2006 07:09, williamluisterry wrote:
> Hi. I've working with al TMS320C613 DSK building some guitar effects.
> I was suscesfully whit all of these but I had no idea about build a
> valve guitar preamp on DSP.

This is far from a solution, but might give you an idea:
Valve amplifiers differ from others mainly in that they can handle a wider 
frequency range. 
While a common transistor or IC amp 's frequency range may operate up to 
15..20 kHz, a good valve amp might exceed 100kHz.

For a good approximation, your whole system must be able to work up to high 
frequencies. An example:
You use an analogue input (with amplifier stage, impedance matcher, 
anti-aliasing filter, etc.). Since this is usually transistor based, the 
frequency content of your signal is already rather limited, when it enters 
the ADC.
Now, your A/D-converter samples the signal (maybe with 48kS/s), which applies 
an additional hard frequency limit, and which requires rigid filtering in the 
stage before or inside the ADC to avoid aliasing.
Then you output your signal which means: DAC, output amp, transmission lines.
Yes, even the transmission lines are of interest, since the high-voltage valve 
signals are usually coupled with a transformer, which is most often not 
present in transistor output stages.

So, what you have to do is this:
1) Check the whole transmission system around the valve stage (you cannot 
consider it as an isolated component) - amplitude and phase .
2) Check the system around your DSP.
3) Now compare both, compare their signal response.
If you find systematic differences, you might add a stage in your DSP code 
which removes this difference - and you're done.

I guess that phase response will make the difference.
And this is the most difficult thing - because you can handle phase only in 
quantities of your sample rate - a valve amplifier does this continuously.
To get so close to the continuous case that you cannot hear the difference 
would certainly require a system with high sample rate (192kS/s?), very high 
precision/resolution math (64bit processing width ?) and a fairly powerful 
DSP.

You can easily demonstrate this, since you have a DSP system already:
1) reduce the sampling rate to 8kS/s ( or even 4kS/s)
  ( this might be done artificially by taking only every 8th sample or so)
   now you can hear some of the artifacts
2) reduce the calculation precision 
  (by cutting some LSBs after every calculation stage)
  you'll find that some filters might fail, especially IIR filters at the low 
frequencies.

Bernhard
I

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