Hi,

I live in both worlds (more or less :) ), and the tools seems somewhat similar (I suspect that when one of them includes a feature, the other tries to catch the rythm and viceversa). I've had a look to the Xilinx DDS compiler, and it is somewhat different. You can download the free version of the Xilinx ISE Design Suite and play a bit around.

For the project where I will implement the DDCON I selected the Altera mainly due to similarities with other project that also uses Altera, availability of development boards around, and also because I'm more familiar with Linux on the Nios-II processor than on the Microblaze (and that for this project I prefer to be in the low end - Cyclone or Spartan - rather than in the higher end - Stratix or Virtex).

Regards,

Javier

El 21/06/2011 12:34, Luis Cupido escribió:
Hi,

I'm an Altera user and would say the DDS core generator
is really very good, I would expect it to be not too different from
Xilinx these days. (does anyone that lives on both worlds know better ?)

lc
ct1dmk.


On 6/21/2011 10:03 AM, Ulrich Bangert wrote:
John,

as usual I second your opinion and I did have already on my mind to suggest
XILINX's DDS compiler to the group too.

However your statement

to provide SFDR up to 150 dB (and I'd notice it if I were
getting much less than that in practice.)

has pushed me up! When I tell the compiler to generate me a 150 dB SFDR DDS then it produces an block with 28 (!) bits output witdh for the DAC. So, I am asking myself what wonder-chips you may be using as DAC for your DDS that
features a dynamic range high enough to really measure a 150 dB SFDR?

Best regards
Ulrich Bangert, DF6JB

-----Ursprüngliche Nachricht-----
Von: time-nuts-boun...@febo.com
[mailto:time-nuts-boun...@febo.com] Im Auftrag von John Miles
Gesendet: Dienstag, 21. Juni 2011 00:52
An: 'Discussion of precise time and frequency measurement'
Betreff: Re: [time-nuts] DDS'ery


I'm not familiar with Altera's DDS options, but I will say
that Xilinx's DDS compiler is superb.  It can be configured
to provide SFDR up to 150 dB (and I'd notice it if I were
getting much less than that in practice.)

As Javier hinted, the reason you can't use the MSB directly
is that its transition point is not necessarily stationary
between cycles of the frequency you're trying to synthesize.
It will flop around all over the place.  You need at least a
few more bits in most applications -- remember that in an
n-bit word, the magnitude represented by the n-1 LSBs is
almost as much as the bit-n MSB.

When DDS technology was first becoming popular in the 1980s,
Qualcomm was one of the main vendors, and they required
external DACs.  High-speed DACs were pricy and used a lot of
power, so I imagine that a great many people tried feeding
the MSB directly to the filter, as I did.  It could be
feasible at some selected frequencies or at very high
clock/output ratios, but in the general case the output
signal is just comically awful.

You would need a truly massive filter to provide the needed
flywheel effect to make up for those missing bits.  And it
would need to be a BPF, not just an LPF, because not all of
the artifacts associated with output quantization are above
the desired carrier frequency.  Sometimes the MSB's toggle
period is going to be shorter than it should be, and
sometimes it's going to be longer.

-- john, KE5FX


-----Original Message-----
From: time-nuts-boun...@febo.com [mailto:time-nuts-
boun...@febo.com]
On Behalf Of Luis Cupido
Sent: Monday, June 20, 2011 9:46 AM
To: Discussion of precise time and frequency measurement
Subject: Re: [time-nuts] DDS'ery

Gracias, Javier.

As you read in my previous email I'm basically
worried about close-in spurs (those that
will pass through the PLL loop filter).

will digest that 4th section... tks.

....

Since I'm inside an FPGA... I'm eager to get
spurs down without leaving the digital world...
Anyone knows any literature covering that ?

Thanks.

Luis cupido.
ct1dmk.





On 6/20/2011 4:52 PM, Javier Herrero wrote:
To reduce the spurii due to quantization distortion. Here is an
explanation, in Section 4

http://www.analog.com/static/imported-
files/tutorials/450968421DDS_Tutorial_rev12-2-99.pdf


Regards,

Javier

El 20/06/2011 17:39, Luis Cupido escribió:
Well, if we really need to filter it out
we better filter the MSB and square it
again...

Why having a DAC for ???

Right ?

Luis Cupido.
ct1dmk.

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Chief Technology Officer
HV Sistemas S.L.                          PHONE:         +34 949 336 806
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