I/O switching speed of Xilinx spartan 6 or Altera EP4CE10

Hi,
I am doing a small exercise to learn verilog on FPGAs trying to create a
FSK31 (ham-radio digital mode) from a FPGA using DDS.
The boards I have use either a Spartan6 (XC6SLX9) or an Cyclon IV (EP4CE10).
The problem to create a digital signal for even the lowest ham-bands
(137 Khz) using DDS, the I/O pin of the lowest bit needs to switch at
35.6 Mhz (if using a nice 8bit DDS). For higher bands, the problem is
even worse, or I need to reduce the resolution of the DDS
My question:
I have looked at the documents by altera or xilinx that describe
switching-speed of I/O pins of their chips, but -from what I understand
it all- FPGAs seams to have special I/O driving hardware to drive
very-high speed interfaces and this does make it all a bit "muddy".
Can somebody explain in (relative) simple and "beginners-lingo" how I/O
of an FPGA works really works, what kinds of I/O ports there are, and
how I can know (or change) what is the typical maximum switching-speed
of an I/O port on a spartan6 or a Cyclone-IV.
Cheerio!
Kristoff
Reply to
kristoff
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First of all, do you mean PSK31 or FSK31? I see both are used, but PSK31 is much more common.
Are you looking to generate an audio signal that you can feed into the mic input of a transmitter? Or are you looking to produce the RF signal directly?
I have no idea what you mean by this. A DDS typically creates sine waves although there are times a square wave is appropriate. Or is this an all digital process? Are you talking about the digital output which will drive a DAC?
What part is muddy?
Better would be if you can explain what you intend to do with the output from the FPGA? Is this a digital output or an analog output?
--

Rick C
Reply to
rickman
In general, you would have the I/O pad driven by a FF clocked at some system frequency. So, you could only change to output state at one of those clock edges. At least on Xilinx, it is possible to run an output directly from some signal from the FPGA fabric. This is generally frowned upon as the delay between internal and I/O can be less well defined.
Xilinx does have SERDES components that can run much faster than the rest of the chip, and some of them can get up to the GHz clock range. These can serialize a number of parallel bits running at some lower rate into a serial stream at much higher rates. These FPGAs have a limited number of these SERDES components. You instantiate them using a tool provided by Xilinx.
I only know Xilinx. Each I/O pad has a receiver and a transmitter that can be set up as tri-state. There are also FFs that can be selected into the input and output paths. You can define in the HDL how you want the I/O pad to operate. Also, the Xilinx parts have controllable delays that be used to adjust for timing issues between external parts and the FPGA. The outputs are VERY fast, you can probably run a 300 MHz clock on the output FFs and thus get 150 MHz square waves out.
You also have selectable voltage standards on these I/Os, generally selected in banks and powered by different voltages. So, you can have an FPGA with 3.3 V signals on one side, and 1.8 V signals on the other. This is all set up in the tools, with a constraint file that defines the I/O standard for each pin.
Jon
Reply to
Jon Elson
The maximum switching rate on a pin is very dependent on the IO standard selected. To get the GHz rates you need to use a differential output standard like LVDS. Single-ended like LVCMOS defaults to using a slew-rate limited driver, which will further reduce the effective switching speed. Make sure you set the slew rate to FAST on the high-speed pin if it is running single-ended.
For Spartan-6 you also need to be aware that the device is not homogenous. Typically these parts have "top and bottom" banks of I/O and "left and right" banks of I/O. Some banks have more drive capability on single-ended standards, but don't have the capability to do differential output. Check your board to make sure that the I/O bank type you need is actually brought out to a connector.
Xilinx documentation is somewhat fragmented. You don't get it all in one "datasheet" like you would on simpler devices. They have a data sheet with electrical characteristics, and a lot of "user guides" describing functionality like I/O, configuration, and clocking. As far as I can remember, they generally don't publish maximum switching rate numbers on all I/O types. You'd need to run IBIS simulations to get most of them. LVDS maximum bit rates may be specified in the family overview document, because they will be the highest rates possible using general I/O. If you have a Spartan-6 LXT device (with gigabit transceivers) you also have the option to use those to achieve much higher switching rates, but the interface insn't so simple.
If you decide to use the Xilinx board rather than Altera, you can get a lot more insight going on the Xilinx forums, where even the Xilinx employees are available to answer questions.
Good luck on your project, it sounds interesting.
--
Gabor
Reply to
GaborSzakacs
If he's doing what I think he's doing he wants to produce RF directly from a DAC that is connected to the FPGA.
--
Tim Wescott 
Control systems, embedded software and circuit design 
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Reply to
Tim Wescott
How are you arriving at that number? In a world where perfect reconstruction filters exist, you can run the output frequency all the way up to half of the clock frequency. If you want to make the reconstruction filtering much easier you'd limit the output to something like Fs/4 -- and 548kHz is a hell of a lot less than 35.6MHz.
This article should help to clarify your thinking:
formatting link

As for your question about I/O rates: probably up to hundreds of MHz, but it somewhat depends on your abilities. If you don't want to spread your output spectrum from the DDS you may want to make a stable clock source external of the FPGA and use that both to drive the DAC's sample clock and the FPGA. At least the last time I checked (which, granted, was 20 years ago), internal FPGA circuitry was too noisy to use as sampling clocks in communications applications.
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Tim Wescott 
Control systems, embedded software and circuit design 
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Reply to
Tim Wescott
I think the first step here is to select the right DAC, that is able to be be driven from the FPGA and has the necessary properties on the analogue end.
For instance, a randomly selected example might be:
formatting link
which looks to have generous output bandwidth.
That takes 16 bits of LVDS plus an LVDS clock. An FPGA should be able to generate that.
However the next question is: can your selected FPGA manage it, or do you need a more fancy FPGA? That's basically a case of pin-counting and looking at the specs for LVDS drivers. Though it's worth doing a dummy run before you commit (before designing your board, write some code to generate maximum-frequency square wave samples and drive them out the correct pins), because the tool will tell you if you overlooked something or misunderstood.
Or, if you already have an eval board, what DAC do you have?
Theo
Reply to
Theo Markettos
Hi Tim,
I am sorry but I am a bit short on time today. (to much fun things to do, to little time), so I will just reply to your message.
However, I do also like to thank everybody who replied (you, Gabor, Jon, Rickman). I'll try to answers to all the remarks at once.
First, an answer to your other message, and to explain my setup.
Yes, indeed, the goal is to create the RF-signal directly from the FPGA.
As said, this is at this point just an exercise, by trying to make something that is more-or-less "usefull" and -at the same time- a way for me to learn more about FPGAs. At the same time, it's also my first project in verilog.
So, I know, you can do a lot by adding analog circuits after the DAC, but I am at this point trying to learn what exactly a FPGA is capable of doing.
It started after a fellow-ham in our radio-club have a talk on "digital modes" a couple of weeks ago, mentioning PSK31. (yes, "rickman", you are correct, it is PSK31, not FSK. My error).
As the talk also mentioned building your own beacon for CW, QRSS (very slow CW) or WSPR, mainly based on AD9850, I thought "it should be possible to do this for PSK31 too.
The problem is that PSK31 is not just phase-shift keying. To reduce splatter, it reduces the amplitute of the signal to zero just before change is phase-shift is done. (in fact, the signal is multiplied with a cosine signal) (see here:
formatting link
So it was not possible to use a AD9850 for that and -I think- a FPGA would be ideal for this.
My idea was to port the design of (I think) a 2011 "elektor" project for a DDS from using a microcontroller (ATtiny) to using a FPGA. The design is very basic, using a R/2R ladder as DAC.
The DDS is based on a 256-value sinewave table with 8 bits of resolution. This drives 8 output-ports connected to the R/2R ladder.
So, for the best "quality" (stepping 1 value per clock-cycle in a 256 value table) the frequency of the resulting sinewave is fclock/256. Or, going the other way, for a 137 Khz sinewave, you need a 35.072 Mhz clock. (which should of course not be a problem for a FPGA).
But, as said, this does mean that if you have a 8 bit R/2R ladder, you risk have the port of the lowest bit switch at 35.072 Mhz (hum ... come to think of it, probably only half of that).
And this got me thinking about how fast you can actually switch a I/O pin of a FPGA. I know that FPGAs are used for "very high speed" devices, .. so I guess that the pins are able to switch very fast, ... but how fast. And, ... how do you actually interface it?
I noticed that the pinplanner in the Altera software allows you to select a number of different modes for I/O pins: different voltages, HSTL class I and II, LVCMOS, LVTTL, LVDS/PPDS/RSDS (in different variants), SSTL.
So I started reading documentation about all these different modes, what speed they offer, how to interface them, etc. but the information was -at least to me- quite daunting.
(hence my question).
BTW. My apologies to everybody for not making my original question more clear. But then, the result was that I got some information that was not planned, but not less interesting never-the-less :-)
Thanks for the link. (actually quite an interesting read) :-)
That is interesting information. Thanks. :-)
Again thanks to you and to all who replied.
As said, currently this is just an exercise for myself, but the goal does is to actually build this thing: get the analog backend of the this done correctly and use it -as a portable FSK31-enabled beacon-, perhaps for SOTA-operations.
Cheerio! Kr. Bonne.
Reply to
kristoff
The best quality by what measure? You're really not going to improve things much by clocking that fast. And what if you get your hands on a 16-bit DAC? Will you be asking how to do this at several GHz?
Rolling your own DAC from resistors and the output pins of a ginormous digital device is not a recipe for high quality. Unless you really, really know what you're doing here you're giving up a lot more than you're going to get by over-clocking your DDS so severely.
My suggestion: If you stick to LVCMOS, then you won't be able to run super-blazing-fast, but you'll be plenty fast to clock a DAC at 50 or 100MHz. That should be plenty, and will, in fact, probably be challenging your PCB layout abilities (it gets ever-less trivial as the frequencies go up, although keeping the parts close together should help).
You will be far happier if you just buy a DAC. Get something that's good for a sampling rate of around 50MHz, but that has an internal latch. I suspect that there will be affordable 12-bit ones, even.
And then, since you need to put a good bandlimit filter in there _anyway_ to conform to good RF practice -- call it a "reconstruction filter" and be happy.
--
Tim Wescott 
Control systems, embedded software and circuit design 
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Reply to
Tim Wescott
Ok, so some things have been explained more clearly, other things are more muddy.
All the stuff about the PSK31 signal and other background is fine, but not terribly important to what you need from us.
You are asking how fast an FPGA I/O can be switched which I think is also not terribly relevant to your real problem, but I believe someone answered that... "it depends".
An I/O in an FPGA typically is programmable both in drive strength and in slew rate. This is to allow support of multiple I/O standards while also allowing a minimum of RF interference being created by the digital outputs. Very fast edges create high frequency harmonics. Slower edges reduce these harmonics. So this setting ultimately sets the limit to how fast you can switch an I/O pin. However, FPGAs are intended to be synchronously clocked devices, so usually what is more important is the internal maximum clock rate which will set the maximum rate for changing the value driven to the I/O pin. This is specified in most FPGA data sheets, but does not reflect a terribly useful parameter when the FPGA is doing much logic work. This logic slows the maximum clock rate and so the I/O toggle rate in any useful design.
Now, to the DDS. A DDS is typically designed with two sections, a phase accumulator and a sine generator. The phase accumulator does not always count by one. It can be programmable with a variable step size to generate a variable frequency sine wave from a fixed frequency digital clock. The formula is Fout = Fclk * N / M where Fclk is the digital clock rate, N is the step size added to the phase accumulator on each clock and M is the modulus of the phase accumulator which does not have to be a binary number. It is only important that the counter count from 0 to M-1 and then wrap around to 0 again.
What is not clear to me is if the PSK31 signal is a PSK31 modulated audio tone that is then modulated on a carrier, or if the carrier is directly modulated with PSK31. The articles I've seen talk about using a PC sound card output to generate the audio signal but they don't say how this is modulated on the carrier... perhaps I'm showing my ignorance of ham radio. lol
The design of a directly modulated PSK31 signal at RF means you will generate a carrier, but it needs to be modulated both in phase and amplitude. I don't know how they accomplished that in the article you read, but in the FPGA the phase modulation is just an increment that is added to the output of the phase accumulator in the DDS, equal to half the modulus M of the accumulator (180 degrees). The amplitude would be adjusted by a multiplier after the sine wave generator.
BTW, in an FPGA there is no reason to limit yourself to a 256 entry look up table (LUT) for the sine wave generator. There are also shortcuts you can use to cut the size of this table by 4. So using a 2048 entry table you can use 8192 points per cycle of the sine wave. These take advantage of the redundancy of the values in a sine wave cycle, ramping up vs ramping down and positive values vs. negative values. You also don't even need to use a LUT. There are approximations using multiplies that can get you 18 bits of resolution on the input to the sine generator. This reduces what is called phase truncation which creates close in spurs to the carrier which are hard to filter out. More phase resolution reduces these spurs and gives you a cleaner signal.
I spent some time looking at DDS designs in FPGAs and found that most designs stop well short of squeezing the best performance available.
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Rick C
Reply to
rickman
If it's coming out of a sound card then the assumption is that you're plugging it into a single-sideband transceiver.
Which should be easy-peasy, even for a beginner.
I'm not sure that he needs the best performance available, but given that he's a beginner I think he needs something simple.
Later on he can get more complicated.
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Tim Wescott 
Wescott Design Services 
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Reply to
Tim Wescott
Apples and oranges. The sound card method is what many do because they don't need to build anything. I can't tell if the design being transcribed from an MCU to an FPGA is intended to work that way or rather it would seem RF will be generated directly.
It has been a while since I looked much at single side band. Is that just AM with filtering applied or is some other method used to generate the RF signal?
I was studying to get a ham license a half year ago. I should remember this.
*If* this is the method intended.
Maybe, maybe not. Amateur radio has specs on unintended emissions. I don't know if an 8 bit input/360 degree sine table would provide low enough noise. That's only 6 bits of magnitude resolution. This degree of truncation in the phase word will generate lots of close in spurs which can't be easily filtered.
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Rick C
Reply to
rickman
I did a little reading an found that SSB is a bit more complex to generate than the PSK31 signal. So directly generating a PSK31 modulated SSB signal will be a bit harder to do than just using a DDS circuit. But then nothing says the PSK31 signal *has* to be SSB modulated.
Still not sure what the OP intends here. I hope he is coming up to speed on how a DDS works.
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Rick C
Reply to
rickman
If all you want to do is generate PSK31 and drive an antenna with it, you do exactly the same thing at RF that you would do at audio to drive it out a sound card, only with a higher carrier.
If you treat a SSB transmitter as a black box, it's just shifting up the frequency of the stuff coming in by many MHz, and pumping the result out the antenna. How that's _done_ is complicated, but it's conceptually simple.
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Tim Wescott 
Wescott Design Services 
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Reply to
Tim Wescott
It is my impression simply mixing the baseband signal up to RF is *not* what SSB is.
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Rick C
Reply to
rickman
I suspect all the quadrature stuff is what Tim's putting in that black box. I could be wrong.
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Les Cargill
Reply to
Les Cargill
Yup. Or the modulate to IF and filter, if you want to be mid-fashioned about it (_old_ fashioned, i.e. early 1950's is "quadrature stuff" all done in analog. Filtering came later).
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Tim Wescott 
Wescott Design Services 
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Reply to
Tim Wescott
Not _simply_ mixing, no -- just doing that gets you a double sideband signal, with few advantages over AM. You need to remove one sideband in the mixing process. But that's all inside the black box.
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Tim Wescott 
Wescott Design Services 
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Reply to
Tim Wescott
Yes, I know. I was taking exception with your statement, "it's /just/ shifting up the frequency". Also, I'm pretty sure to get SSB you *don't* do exactly the same thing at RF that you would do at audio to generate a PSK31 signal. SSB seems to be the convention for transmitting PSK31 since this is mostly used for QRP and SSB will get a better range.
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Rick C
Reply to
rickman
Right, you mix the audio with a carrier, then use a sharp filter to cut off the carrier and everything on one side of the carrier. So, only the frequencies above (for upper SSB) or below (for lower SSB) are sent through. This is usually done at some fixed frequency, and then hetrodyned up to whatever frequency you actually want to transmit at.
Jon
Reply to
Jon Elson

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