Hi all.
What I need is at least a 10th order 20kHz zero phase-shift LPF. The reason I need zero phase shift is that I want to put it in the forward path of an amplifier to roll off the response well before the inherent phase shift at high frequencies of the other stages of the amplifier start to kick in. This way I can (maybe) apply a big heap of negative feedback around the whole amp without such a great risk of HF instability because the amplifier gain has dropped to unity or less at HF by our little filter.
Now I reveal how lazy I am ;-) Is there some software that will enable me to specify the filter parameters I want e.g corner frequency, slope etc just by pushing a few buttons, and will generate some suitable ready-to-run code that can then be burnt? Is there a chip that has both 16 bit A/D & D/A for this application?
I looked at the TI website but was just overwhelmed by the volume of stuff :-( Everything I have looked at so far assumes you are already a rocket scientist, but in this case I don't yet at least want to learn all the in's and out's of DSP, just make a filter so I can get on with my project.
TIA G.P.
What I need is at least a 10th order 20kHz zero phase-shift LPF. The reason I need zero phase shift is that I want to put it in the forward path of an amplifier to roll off the response well before the inherent phase shift at high frequencies of the other stages of the amplifier start to kick in. This way I can (maybe) apply a big heap of negative feedback around the whole amp without such a great risk of HF instability because the amplifier gain has dropped to unity or less at HF by our little filter.
Now I reveal how lazy I am ;-) Is there some software that will enable me to specify the filter parameters I want e.g corner frequency, slope etc just by pushing a few buttons, and will generate some suitable ready-to-run code that can then be burnt? Is there a chip that has both 16 bit A/D & D/A for this application?
I looked at the TI website but was just overwhelmed by the volume of stuff :-( Everything I have looked at so far assumes you are already a rocket scientist, but in this case I don't yet at least want to learn all the in's and out's of DSP, just make a filter so I can get on with my project.
TIA G.P.
Check out Alesis semiconductor's AL3101. It is far from a turnkey solution, but it is easily interfaced and addressed. There is even a cheap ($200?) demo board to play with.
The main limitation is 48kHz sample rate.
The main limitation is 48kHz sample rate.
DSP etc
Hi Circlotron,
You may want to take a look at the 2-channel DSP filter which was described recently in EW. It is 2-channel, consisting of a pair of ADC and DAC'S with standard CD quality, interfacing to a DSP. Nice thing is that there is Windows software to spec the filter characteristics and download them to the card. This may not be the solution to put in your amps, but it is very versatile (IRR/FIR with up to 1024 taps!) that you could use it for testing. Not really cheap, I think some US $ 600.
Unfortunately, I cannot give you the ref right now, will have to wait until tomorrow afternoon. Sorry. But maybe you already know about it?
Cheers, Jan Didden
Hi Circlotron,
You may want to take a look at the 2-channel DSP filter which was described recently in EW. It is 2-channel, consisting of a pair of ADC and DAC'S with standard CD quality, interfacing to a DSP. Nice thing is that there is Windows software to spec the filter characteristics and download them to the card. This may not be the solution to put in your amps, but it is very versatile (IRR/FIR with up to 1024 taps!) that you could use it for testing. Not really cheap, I think some US $ 600.
Unfortunately, I cannot give you the ref right now, will have to wait until tomorrow afternoon. Sorry. But maybe you already know about it?
Cheers, Jan Didden
Digital filters imply delay. Delay is basically the same as phase shift. I would be weary of this approach.
BTW DAC chips often have digital filters built in. If you want a separate unit, consider looking at the TI PCM1704 datasheet where an external digital filter is referenced. Other manufacturers make such filters as well.
Petter
BTW DAC chips often have digital filters built in. If you want a separate unit, consider looking at the TI PCM1704 datasheet where an external digital filter is referenced. Other manufacturers make such filters as well.
Petter
Rocket science
I too would be wary of this approach. A high order LPF has a lot of delay, even if it is "linear phase" (which means it has only delay as its phase response), and you will have "fun" making something stable.
But perhaps there is another similar approach that would give the results you want. If you let the DSP have two inputs, one is the input signal and one is the feedback signal from the amplifier output, you might get somewhere.
However, I think that to do this well, you will need to learn some of that rocket science. Control theory really IS rocket science.
I'm not sure of a good cookbook type reference, because what you want to do is digital control, not just digital signal processing, so you would need to use aspects of both control theory and DSP to get it right, and you would presumably have to model the response of the amplifier to design the compensator for it.
Good books on Rocket Science are unfortunately rather expensive. Oppenheim and Schafer have a good one called Discrete Time Signal Processing. There is a book called something like Digital Control of Dynamic Systems by Franklin, Powell, and Workman.
Maybe you can find some more cookbook type references by browsing Amazon starting with those books, and looking at the lists and "people who bought this also bought ..." links.
Also look at this one: http://www.dspguru.com/info/faqs/firfaq.htm
Have fun,
-- mirlo
I too would be wary of this approach. A high order LPF has a lot of delay, even if it is "linear phase" (which means it has only delay as its phase response), and you will have "fun" making something stable.
But perhaps there is another similar approach that would give the results you want. If you let the DSP have two inputs, one is the input signal and one is the feedback signal from the amplifier output, you might get somewhere.
However, I think that to do this well, you will need to learn some of that rocket science. Control theory really IS rocket science.
I'm not sure of a good cookbook type reference, because what you want to do is digital control, not just digital signal processing, so you would need to use aspects of both control theory and DSP to get it right, and you would presumably have to model the response of the amplifier to design the compensator for it.
Good books on Rocket Science are unfortunately rather expensive. Oppenheim and Schafer have a good one called Discrete Time Signal Processing. There is a book called something like Digital Control of Dynamic Systems by Franklin, Powell, and Workman.
Maybe you can find some more cookbook type references by browsing Amazon starting with those books, and looking at the lists and "people who bought this also bought ..." links.
Also look at this one: http://www.dspguru.com/info/faqs/firfaq.htm
Have fun,
-- mirlo
DSP LPF etc
Circlotron,
The reference I promised is a real-time digital filter on www.umist.ac.uk/dias/pag/signalwizard.htm.
In view of the above you may already have changed your mind. BUT, the signalwizard is dual channel so it might still work with parallel paths as described above. Proce is steep though at UKP 499.
Jan Didden
Circlotron,
The reference I promised is a real-time digital filter on www.umist.ac.uk/dias/pag/signalwizard.htm.
In view of the above you may already have changed your mind. BUT, the signalwizard is dual channel so it might still work with parallel paths as described above. Proce is steep though at UKP 499.
Jan Didden
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