Jim, Maybe shape factor is the wrong term to use. All I know is that I took a relative low phase noise source, a HP 8640B, and tuned the SDR so that the signal was in the bandpass with the filter set to 1000 Hz. I then retuned the SDR so that the signal was 6db down. I then retuned the SDR again so that the signal was 60 db down. If I divided the 6 db bandpass by the 60 db bandpass, the number was about 1.1. If I followed the same proceedure with the filter set to 100 Hz , the number was 2.7. What causes the difference?
Tom W0IVJ Jim Lux wrote: > At 08:18 AM 2/18/2007, Tom Thompson wrote: > >> Mark and Bill, >> >> I made some measurements and got similar results as Mark. The one thing >> that confused me was the difference in shape factor between the narrow >> filters and the wide filters, but I think you just cleared that up for >> me, Bill. It has to be a function of the bin resolution and the bin >> bleed. Thanks, Mark for bringing this up, and thanks Bill for clearing >> my confusion...very interesting. > > > > This is somewhat confusing because you are using a conceptual model > (shape factor) that is really derived from analog filter design in a > domain (digital filters with a lot of samples) that it isn't as well > suited to. > > In analog filters, we talk about how many sections or poles it might > have, and knowing that number tells you what the ultimate rolloff is > going to be (12 dB/octave per section, eh?). The close in rolloff in a > high q filter (say a crystal lattice) is still determined by combining > a relatively small number of tuned circuits (albeit high q ones).. > Essentially, you stack up a bunch of stagger tuned sections so that > you get a "bart's head" type frequency domain response. You have to > worry about interacctions between the tuned circuits (some deliberate, > as in a double tuned IF stage, some not), drifting in component > parameters, and non-ideal components, so Q isn't infinite. > > > But in the digital domain, you can (easily) build a filter that is the > equivalent of 4000 ideal lossless LC tuned networks with infinite Q. > Yowza!.. Sure, there are tradeoffs, and there are some peculiarities > (roundoff, truncation, etc.) but it's easy to build filters that have > "desirable" properties but which don't fit the usual analog filter > metrics and design tradeoffs. For instance, it's pretty easy to build > a "linear phase" filter in the digital world (one that has the same > time delay for all frequencies in the passband, which has minimal > pulse shape distortion).. something that is quite challenging with > analog filters (as anyone who has agonized over group delay properties > has dealt with). > > In the digital world, one could build a dynamically adjusting CW > keying envelope that is precisely limited in it's bandwidth to the > current keying rate, without "ringing". Heck, in the digital world, > one can have non-physically realizable filters (i.e. that have an > output before the input is applied, in some senses) > > > So the challenge we all face when working with digital filters is that > a lot of the traditional measurements and tradeoffs change. > Sometimes, a measurement (e.g. swept response) gives results that, if > an analog filter were being measured, would mean that the measurement > system is broken. Other times, we make measurements that "mean > something" in terms of an analog design (3rd order intercept is a good > example) that doesn't necessarily have the same interpretation in the > digital world (or more correctly in the hybrid digital analog world). > For instance, Spurious Free Dynamic Range is a very different thing > when applied to A/Ds than when applied to a LNA and mixer. > > Shape Factor for filters is another such metric.. It's a shorthand way > of describing a certain kind of filter (bandpass with symmetric > skirts). A shape factor of 6 is a lot different from 2, but the > difference between 1.1 and 1.05 is less so, in terms of practical > significance. if you really want to specify adjacent channel > rejection, then that's the spec you should be working with (i.e. 3dB > bandwidth of X kHz, 60 dB down at X+Y kHz) > > Also, watch out for stopband bounce.. I work with a variety of analog > filters that have fairly steep rolloffs, a deep null at about 2.5-3x > cutoff frequency, but that only have 30 dB of rejection far out. > Why? Because other stages provide the far away attenuation, but I'm > concerned about suppressing the spur at the clock rate from the glitch > energy in the dac. The filter might have a fair amount of phase > ripple in the passband, but I can compensate that in the equalization > in the digital data stream going to the DAC. But, if I were to look > at just the digital filter characteristics, it would look terrible. > It's the overall system performance that you're concerned about. > > A similar strategy is used in consumer audio DACs. They take the > digital stream at 44.1 kS/s, interpolate it it up to 192k, then run it > to the DAC. The analog filter can then use relatively few sections > with low Q, because the 192 is almost 10 times the filter cutoff of > 20-25 kHz, so you don't need an extreme shape factor to get good > performance. > > Jim, W6RMK > > > > > > _______________________________________________ FlexRadio mailing list FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archive Link: http://www.mail-archive.com/flexradio%40flex-radio.biz/ FlexRadio Homepage: http://www.flex-radio.com/ FlexRadio Knowledge Base: http://kb.flex-radio.com/
