[time-nuts] DDS - higher frequecies
Suppose I have an A/D running at 1 MHz. The standard simple minded approach is that it will work for any input signal with a bandwidth up to 1/2 MHz. We usually think of that in the baseband, but it also works for, say 1.25 to 1.5 MHz. The input signal gets aliased down into the baseband. (and if you are unlucky, which is easy, some of the aliasing reflects back and overlaps so you can't tell X-y from X+y) There is similar math for D/A, the reverse direction. I think this applies for a DDS making higher frequencies than simple arithmetic would allow it to generate. Does anybody have a good web page for how that works? My simple expectations are that it would have to generate lots of harmonics and then go through a filter to get rid of all the wrong stuff. I'm missing the step where all the harmonics come from. Are they just really tiny and I have to do a lot of good filtering and amplification? Do I need something other than a traditional DDS for this sort of stuff? -- These are my opinions. I hate spam. ___ time-nuts mailing list -- time-nuts@febo.com To unsubscribe, go to https://www.febo.com/cgi-bin/mailman/listinfo/time-nuts and follow the instructions there.
Re: [time-nuts] DDS - higher frequecies
Hi The output spectrum is modified by the usual sin(x)/x based on the actual speed of the DAC. It's like any digital signal, the rise time of the edge and the spectrum are related to each other. Depending on exactly what sort of DAC architecture you have it may work ok, or it may not. If some bits propagate faster than others … not so good. Bob On Nov 25, 2012, at 7:30 PM, Hal Murray wrote: > > Suppose I have an A/D running at 1 MHz. The standard simple minded approach > is that it will work for any input signal with a bandwidth up to 1/2 MHz. We > usually think of that in the baseband, but it also works for, say 1.25 to > 1.5 MHz. The input signal gets aliased down into the baseband. (and if you > are unlucky, which is easy, some of the aliasing reflects back and overlaps > so you can't tell X-y from X+y) > > There is similar math for D/A, the reverse direction. I think this applies > for a DDS making higher frequencies than simple arithmetic would allow it to > generate. > > Does anybody have a good web page for how that works? My simple expectations > are that it would have to generate lots of harmonics and then go through a > filter to get rid of all the wrong stuff. I'm missing the step where all the > harmonics come from. > > Are they just really tiny and I have to do a lot of good filtering and > amplification? > > Do I need something other than a traditional DDS for this sort of stuff? > > > > -- > These are my opinions. I hate spam. > > > > > ___ > time-nuts mailing list -- time-nuts@febo.com > To unsubscribe, go to https://www.febo.com/cgi-bin/mailman/listinfo/time-nuts > and follow the instructions there. ___ time-nuts mailing list -- time-nuts@febo.com To unsubscribe, go to https://www.febo.com/cgi-bin/mailman/listinfo/time-nuts and follow the instructions there.
Re: [time-nuts] DDS - higher frequecies
On 11/25/12 4:30 PM, Hal Murray wrote: Suppose I have an A/D running at 1 MHz. The standard simple minded approach is that it will work for any input signal with a bandwidth up to 1/2 MHz. We usually think of that in the baseband, but it also works for, say 1.25 to 1.5 MHz. The input signal gets aliased down into the baseband. (and if you are unlucky, which is easy, some of the aliasing reflects back and overlaps so you can't tell X-y from X+y) There is similar math for D/A, the reverse direction. I think this applies for a DDS making higher frequencies than simple arithmetic would allow it to generate. Yes.. you generate all the aliases.. Fx, Fs-Fx, Fs+Fx, 2Fs-Fx, 2Fs+Fx, etc. Does anybody have a good web page for how that works? My simple expectations are that it would have to generate lots of harmonics and then go through a filter to get rid of all the wrong stuff. I'm missing the step where all the harmonics come from. Not exactly harmonics, but aliases. What you are doing is convolving the sampling function (a series of ideal impulses, either in frequency or time domain) with the other signal. It's basically the opposite of an undersampling IF ANd, because the typical output has a sample/hold, it's not a series of impulses, but a staircase, so the frequency domain has a sin x/x shape to it. Are they just really tiny and I have to do a lot of good filtering and amplification? Yes.. And all the usual things about timing jitter aggravating the higher order outputs more than the first order, etc. Do I need something other than a traditional DDS for this sort of stuff? ___ time-nuts mailing list -- time-nuts@febo.com To unsubscribe, go to https://www.febo.com/cgi-bin/mailman/listinfo/time-nuts and follow the instructions there.
Re: [time-nuts] DDS - higher frequecies
On 11/26/2012 01:30 AM, Hal Murray wrote: Suppose I have an A/D running at 1 MHz. The standard simple minded approach is that it will work for any input signal with a bandwidth up to 1/2 MHz. We usually think of that in the baseband, but it also works for, say 1.25 to 1.5 MHz. The input signal gets aliased down into the baseband. (and if you are unlucky, which is easy, some of the aliasing reflects back and overlaps so you can't tell X-y from X+y) There is similar math for D/A, the reverse direction. I think this applies for a DDS making higher frequencies than simple arithmetic would allow it to generate. Does anybody have a good web page for how that works? My simple expectations are that it would have to generate lots of harmonics and then go through a filter to get rid of all the wrong stuff. I'm missing the step where all the harmonics come from. Are they just really tiny and I have to do a lot of good filtering and amplification? Do I need something other than a traditional DDS for this sort of stuff? I think you would enjoy digging up the Analog Devices DDS material, which goes into this and many other things. http://www.analog.com/static/imported-files/tutorials/450968421DDS_Tutorial_rev12-2-99.pdf See section 10. Cheers, Magnus ___ time-nuts mailing list -- time-nuts@febo.com To unsubscribe, go to https://www.febo.com/cgi-bin/mailman/listinfo/time-nuts and follow the instructions there.
Re: [time-nuts] DDS - higher frequecies
Hal, Check out the Analog Devices website. Good info on DDS Dacs there. You want to stay a bit away from the 1/2fs Nyquist limit in your DA. The reason is the image coming down from your 1MHz clock. If you output say 0.45MHz, you have an image at 0.55 MHz already (1MHz - 0.45MHz) so your filter has to be extremely steep to make that work and remove the spur at 0.55 MHz.. Check out the Mini Circuits LFCN low pass filters, they work at higher frequencies, and are very steep.. Your filter quality is going to determine how close you can get to Nyquist. Bye Said Sent From iPhone On Nov 25, 2012, at 16:30, Hal Murray wrote: > > Suppose I have an A/D running at 1 MHz. The standard simple minded approach > is that it will work for any input signal with a bandwidth up to 1/2 MHz. We > usually think of that in the baseband, but it also works for, say 1.25 to > 1.5 MHz. The input signal gets aliased down into the baseband. (and if you > are unlucky, which is easy, some of the aliasing reflects back and overlaps > so you can't tell X-y from X+y) > > There is similar math for D/A, the reverse direction. I think this applies > for a DDS making higher frequencies than simple arithmetic would allow it to > generate. > > Does anybody have a good web page for how that works? My simple expectations > are that it would have to generate lots of harmonics and then go through a > filter to get rid of all the wrong stuff. I'm missing the step where all the > harmonics come from. > > Are they just really tiny and I have to do a lot of good filtering and > amplification? > > Do I need something other than a traditional DDS for this sort of stuff? > > > > -- > These are my opinions. I hate spam. > > > > > ___ > time-nuts mailing list -- time-nuts@febo.com > To unsubscribe, go to https://www.febo.com/cgi-bin/mailman/listinfo/time-nuts > and follow the instructions there. ___ time-nuts mailing list -- time-nuts@febo.com To unsubscribe, go to https://www.febo.com/cgi-bin/mailman/listinfo/time-nuts and follow the instructions there.
Re: [time-nuts] DDS - higher frequecies
On 11/25/12 5:19 PM, Said Jackson wrote: Hal, Check out the Analog Devices website. Good info on DDS Dacs there. You want to stay a bit away from the 1/2fs Nyquist limit in your DA. The reason is the image coming down from your 1MHz clock. If you output say 0.45MHz, you have an image at 0.55 MHz already (1MHz - 0.45MHz) so your filter has to be extremely steep to make that work and remove the spur at 0.55 MHz.. Check out the Mini Circuits LFCN low pass filters, they work at higher frequencies, and are very steep.. Your filter quality is going to determine how close you can get to Nyquist. There's also a variety of intermod type products that show up, particularly when you talk about spurs from phase truncation and the like. So you get not only the phase truncation spurs, but also all the aliases of those spurs. There's a fair amount of literature around about this, especially from about 10-20 years ago, when 1GHz ADCs and the logic to drive them weren't easy to come by. People wanted to generate signals in the hundreds of MHz range, but with logic and DACs that were slower. There's a reason that people do dithering in these sorts of applications: it degrades the peak performance, but at least it keeps you from having a big spur in the wrong place. There's a nice PhD dissertation out there (which name escapes me right now) that has a whole matlab code to simulate/analyze it. ___ time-nuts mailing list -- time-nuts@febo.com To unsubscribe, go to https://www.febo.com/cgi-bin/mailman/listinfo/time-nuts and follow the instructions there.
