DDS Notes from Chp. 16, ARRL HB 2005

Summary of this will show that the DDS is a polluted method of signal 
generation!

It might be interesting but it surely is not efficient.  
Too many processes and things to over come here!
So argue amongst yourselves. As you read this, count all of the problems and 
technical difficulties this method is plagued with!
Then vote!

Note that there is no such thing in this article as good DDS phase noise 
characteristics!  Very little appears to be good once you begin reading this!


"A DDS system generates digital samples
of a sine wave and converts them to an
analog signal using a DAC. 

In a DDS chip, a phase accumulator is
incremented at each clock time; the phase
information is used to look up a sine-wave
amplitude from a table. This value is passed
to the DAC, which outputs a step-wise sine
wave. *As we saw before, the spectrum of
this sine wave is seasoned with aliases and
contains other minor pollutants. Since the
phase is represented by a binary number
with a fixed number of bits, p, *errors develop
because the number is truncated to
that number of bits. Truncation generates
PM spurs in the DDS output. This occurs
prior to the DAC. 

Further errors are related
to the output resolution of the look-up table.
Table values representing the amplitudes
are truncated to some number of bits, a.
*This mechanism produces AM spurs in the
output.

*Phase noise at the output is that of the
DDS clock source times the ratio of the
output frequency to the clock frequency,
as limited by divider noise. *Spurious levels
also tend to grow as the DDS output
frequency approaches the Nyquist limit.
Strange spurs at the output are usually
related to IMD and harmonics of the desired
signal and their aliases. Remember
that frequencies exceeding half the sampling
frequency “fold back” into the signal
spectrum at a position determined by
their frequency, modulo fs/2. High-order
harmonics are liable to find their way into
one’s band of interest. *Traps at the DAC
output have been known to suppress these
responses. See Project F in the Appendix
for the schematic of a DDS project.
In the analog signal we generate, *the
DAC introduces more AM spurs, harmonics
and IMD because of its inherent nonlinearity,
as discussed above. *Spurs are
also likely at the clock frequency, its harmonics
and sub-harmonics. A higherorder
LPF will take care of these, but we
must see what we can do about the others.
It turns out we may eliminate all the AM
spurs by squaring the DDS output. *We can
do nothing about the remaining PM spurs.
Cranking through Eq 28 will show that
they can be made very low: –113 dBc for
a 20-bit-address sine look-up table and
32-bit phase accumulator. This parameter
is critical in case we want to use the DDS
as the reference to a high-frequency PLL
circuit. *The PLL will multiply the phase
noise and PM spurs by the ratio of the PLL
output frequency to the PLL reference frequency
within the PLL loop BW. Outside
the loop BW, the VCO itself is responsible
for establishing spectral purity. *So while
dividing the DDS to the PLL reference frequency
lowers phase noise and PM spurs,
the PLL multiplies them back upward. A
trade-off exists between spur levels and
reference frequency, hence lock time.
A PLL reference frequency near 100 kHz
has been found to be sufficient for the
desired lock times, with an output-toreference
ratio of 1000. Such a loop should
achieve very fast lock times, as it can be
expected to lock within 500 cycles of the
reference input. The DDS tuning time is at
least three orders of magnitude faster than
this. In the example, the VCO output is
near 100 MHz. DDS energy is injected at
the reference input to the PLL chip, squaring
it and dividing it by 10; the DDS runs
near 1000 kHz. The block diagram of a
PLL using a DDS as its reference is shown
in Fig 16.20. Spurs and phase noise inside
a loop BW of, say, 1 kHz are amplified by
the PLL by the factor:
dB 40
f
f
log 20 N
REF
VCO = ? ?
?
?
? ??
?
= (30)
Of course, we tune the hybrid synthesizer
by programming the DDS; the PLL
programming is fixed. Let’s say we want
1-Hz tuning resolution at the VCO output.
As the DDS frequency is 1/100 of the output,
we must tune the DDS in 10 millihertz
steps! Tuning resolution in a DDS circuit
is determined by the phase accumulator’s
bit resolution, p, and the DDS clock’s frequency,
fclk:
p
clk
DDS
2
f
df = (31)
A clock frequency around 10 MHz and
p = 32 easily satisfy our conditions, producing
a step size of 2.3 millihertz. As
noted above, making the DDS output frequency
a small fraction of the clock frequency
makes it easier to get a clean
output. A range of about half an octave
eases the design of the LPF or BPF used at
the DDS output to limit spurs, aliases and
clock feed-through.
The phase-accumulator/look-up-table
approach is equally useful in generating
numeric BFOs in software. One of the first
things to emerge when considering this
scheme is the potentially large size of the
look-up table. To maintain the full dynamic
range of a DSP system requires
BFO phase and amplitude performance,
as limited by Eqs 28 and 29, at least as
good as the rest of the system. In 16-bit
systems, we are shooting for about 90-
100 dB of dynamic range. A table with 216
= 65,536 entries is not much of a problem
for DDS chip manufacturers to include onboard,
but it may tax available memory
space in embedded systems.
Fortunately, a couple of ways around
the problem have been uncovered. The
first involves the process of interpolation,
very much like the artificial increase of
sampling frequencies we examined above.
In this method, we restrict the number of
table entries to some arbitrary number,
M << 216, while keeping the bit-resolution
of the entries themselves, a, high enough
to satisfy the limits of Eq 29 for the spur
levels we can tolerate. Take the case where
? = 28 = 256 and a = 16. The phase accumulator,
incremented at each sample time
by an amount df that is directly proportional
to the output frequency, forms the
address into the look-up table. Let this
address have bit-resolution p = 16. According
to Eq 28, PM spurs will not exceed
–91 dBc. Since there are only 256
table entries, we may use the most-significant
byte (MSB) of the address to find the
table entries that straddle the correct output
value. We then use the least-significant
byte (LSB) as an unsigned fraction to
find out how far between the two table
entries we must go to reach the correct
output value. If, in order of increasing
address in the table, our two adjacent table
entries are d1 and d2, we may perform a
first-order interpolation between the entries
using: etc..."




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