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This special issue of Analog Dialogue is offered as a bonus to our faithful readers and an encouragement to new readers. We have
reprinted here the popular series, entitled “Ask The Applications Engineer”, from its inception in 1988 through Number 23* in
1996.
We are celebrating Analog Dialogue’s 30th sequential year in print, just concluded. During those years, Analog Dialogue has informed,
enlightened, and occasionally educated more than a generation of engineers, scientists, and technicians about Analog Devices’s
products, practice, and ideas for high-performance signal processing by analog, digital, and mixed means.
This 30th anniversary Selection, Ask The Applications Engineer, joins the collection of articles in our 25th anniversary bonus
edition, The Best of Analog Dialogue—1967 to 1991. Readers of that 224-page collection (still available upon request) will recall its
many articles relating to practice—as well as articles about landmark products. Our intent in originating the “Ask The Applications
Engineer” series has been to supplement that mine of information with a source of what has come to be called FAQ—frequently
asked questions (and answers). They are chronicled by a few of our Applications Engineers, but are based on the accumulated
wisdom and experience of our worldwide Applications Engineering staff, interacting with tens of thousands of customers worldwide.
Throughout the pages of this volume you will find references to Analog Devices products and publications, just as they appeared
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The articles in this publication appear in chronological order, starting with “Ask The Applications Engineer—1”. But in order for
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questions to be answered in the future.
*With one exception: We have not included “Ask The Applications Engineer—4”, which was devoted to a description of a 5.25" Component Selection
Guide diskette that was novel and useful in its time; however, it has long been obsolete—in this day of the Web site, the CD-ROM, and the selection tree
in print. Just about everything else has turned out to include information of timeless value.
Q. What about circuits requiring both “input” and “output” offset very large value of summing resistance, compared to the
trim? signal-input resistances, in order to (i) avoid loading the
A. Circuits such as instrumentation and isolation amplifiers often summing point excessively, (ii) scale the correction voltage
have two stages of dc gain, and the gain of the input stages can properly and produce enough attenuation to minimize the
be variable. Thus a two stage amplifier, with an input offset, effects of differential supply-voltage drifts. It is often helpful
IOS, an output offset, OOS, a first stage gain of K, and a to use resistances between the supplies and the potentiometer
unity-gain output stage, has (for zero input) an output, OP, to increase trim resolution and reduce dissipation.
of: Where gain trim is provided for in a circuit, it will generally
OP = OOS + K × IOS (6) consist of a variable resistor. Details of its value and connection
will appear on the data sheet of the device. Where gain trim is
From (6) it is evident that if the gain is constant we need only not required, this resistor may be replaced by a fixed resistor
adjust either IOS or OOS to null the total offset (although if having half the resistance of the maximum value of the
the input uses a long-tailed pair of bipolar transistors we will recommended trim potentiometer.
get a better offset temperature coefficient if we trim both—for
FET long-tailed pairs this is not necessarily the case). If the Where gain trim is not provided it is not always achievable
first stage gain is to be varied, both offsets must be trimmed. externally without an additional variable-gain stage. For
example, consider a DAC using a ladder network. If the ladder
This is done by an iterative procedure. With zero input, and network is used in the current mode (Figure 2a), the input
gain set to maximum, the input offset is adjusted until the impedance at the reference terminal does not vary with digital
output is also zero. The gain is then reduced to its minimum code, and the gain of the DAC may be trimmed with a small
value and the output offset adjusted until the output is zero variable resistor in series with either the reference input or the
again. The two steps are repeated until no further adjustment feedback resistor. However, if the DAC is used in the voltage
is necessary. Gain trimming should not be done until both mode (Fig 2b), then the reference input impedance is code
IOS and OOS are pulled; the actual values of the high and low dependent, and gain may only be trimmed by varying the
gains used in offset trim are unimportant. reference voltage—which is not always possible—or the gain
Q. What circuitry should I use for gain and offset trims? of the buffer amplifier.
A. Many amplifiers (and a few converters) have terminals for The possibility of trimming gain in circuits not furnished with
trimming gain and offset. Many more do not. gain-trim circuitry, therefore, will depend on individual cases;
Offset trim is normally performed with a potentiometer each must be assessed on its own merits. b
connected between two assigned terminals, and its wiper is
connected (sometimes via a resistor) to one of the supplies.
The correct connections and component values will be given
on the device data sheet. One of the commonest differences
between op-amps is the value of offset correction potentiometer
and which supply it should be connected to.
Where separate terminals are not provided for offset trim, an
offset-adjusting constant can usually be added to the input
signal. Two basic possibilities are shown in Figures 1a and 1b.
Where the correction is being made to a system where a
differential input op amp is used as an inverter (the commonest
case) the method of la is best to correct for device offsets—but
not system offsets. In the single-ended connection, method 1b
will work for system offsets but should be avoided where a. CMOS DAC connected for current steering. Input imped-
possible for small device offsets, because it often requires a ance is constant.
b. Current summed at inver ting input. b. The same DAC connected for voltage output.
Figure 1. Two connections for offset adjustment. Figure 2. Comparing basic DAC circuits.
*Data sheets are available for any of the Analog Devices products mentioned
here. An Application Note: “Operation and Applications of the AD654 V-to-F
Converter,” is also available without charge.
Q. What circuit precautions are necessary when using a VFC? The phase-locked-loop FVC illustrated differs from any other
A. Apart from the usual precautions necessary with any precision PLL in only one respect: the voltage-controlled oscillator of
analog circuitry (grounding, decoupling, current routing, the normal PLL, which must be monotonic but not necessarily
isolation of noise, etc., a subject for a book, not a paragraph) linear, has been replaced by a VFC with a linear control law.
the main precautions necessary when using a VFC are the In the servo system, negative feedback keeps the VFC’s output
choice of capacitor and separation of the input and output. frequency equal to the input frequency. The output voltage,
The critical capacitors in a precision VFC (the multivibrator’s the VFC’s input, is accurately proportional to the input
timing capacitor, and the monostable timing capacitor in a frequency.
charge-balance type) must be stable with temperature variation. Designing PLL systems is beyond the scope of this discussion,1
Furthermore, if they suffer from dielectric absorption, the VFC but if a 4000-series CMOS PLL, the 4046, is used just as a
will be nonlinear and may have poor settling time. phase detector (its VCO’s transfer characteristic is not sufficiently
If a capacitor is charged, discharged and then open-circuited linear), we can build the FVC shown here, with an AD654
it may recover some charge. This effect, known as dielectric VFC.
absorption (DA), can reduce the precision of VFCs or sample-
hold amplifiers using such capacitors. VFCs and SHAs should
therefore use Teflon or polypropylene, or zero- temperature-
coefficient (NPO, COG) ceramic capacitors with low DA.
Coupling between output and input of a VFC can also affect
its linearity. To prevent problems, decoupling practices and
the usual layout precautions should be observed. This is
critically important with opto couplers, which require high Q. What is a synchronous VFC?
current drive (10-30 mA).
A. A charge-balance VFC with improved linearity and stability,
Q. How do you make a frequency-to-voltage converter? where the monostable is replaced by a bistable, driven by an
A. There are two popular methods: the input frequency triggers external clock. The fixed time during which the precision
the monostable of a charge-balance VFC that has a resistor in current discharges the integrator is one clock period of the
parallel with its integration capacitor; or the input frequency external clock.
can be applied to the phase/frequency comparator of a phase- A further advantage of the SVFC is that the discharge does
locked loop (PLL), which uses a VFC (of either type) as its not start when the integrator passes the comparator threshold
oscillator. The basic principle of the first type is illustrated (at a non-critical rate), but on the next clock cycle. The SVFC
below. output is synchronous with a clock, so it is easier to interface
with counters, µPs, etc.; it is especially useful in multichannel
systems: it eliminates problems of interference from multiple
asynchronous frequency sources.
There are two disadvantages. Since the output pulses are
synchronized to a clock they are not equally spaced but have
substantial jitter. This need not affect the user of a SVFC for
For each cycle of the input frequency, a charge, ∆Q, is delivered a/d conversion, but it does prevent its use as a precision
to the leaky integrator formed by R and C. At equilibrium, an oscillator. Also, capacitive coupling of the clock into the
equal charge must leak away during each period, T (= 1/f), of the comparator causes injection-lock effects when the SVFC is at
input, at an average rate, I = V/R. Thus, V = ∆Q?f?R. 2/3 or 1/2 FS, causing a small (4-6 bit at 18-bit resolution at
1-MHz clock) dead zone in its response. Poor layout or device
Though the mean voltage is independent of C, the output ripple design can worsen this effect.
is inversely dependent on C. The peak-to-peak ripple voltage,
∆V, is given by the equation, ∆V= ∆Q/C. This indicates that Despite these difficulties the improvement in performance
ripple is independent of frequency (assuming that the charge, produced by the abolition of the timing monostable makes the
Q, is delivered in a short time relative to the period of the SVFC ideal for the majority of high-resolution VFC
input). The settling time of this type of FVC is determined by applications.
the exponential time constant, RC, from which the time to Q. Can you have a synchronized FVC?
settle within a particular error band may be calculated. A. Yes, and with very good performance; it is best done with an
From these equations, we see that the characteristics of this FVC- connected SVFC and a clock that is common to both
type of FVC are interdependent, and it is not possible to ends of the transmission path. If the input signal to a
optimize ripple and settling time separately. To do this we synchronized FVC is not phase related to the clock, severe
must use a PLL. timing problems can arise, which can only be solved by the
use of additional logic (two D flip-flops) to establish the correct
phase relationship. b
1
See Gardner, F. M, Phase-lock Techniques, 2nd ed., New York: Wiley, 1979, for
more detail; also Analog Devices’ Analog-Digital Conversion Handbook.
1 5 mV external hysteresis
A useful discussion of comparator foibles can be found in Troubleshooting
Techniques Quash Spurious Oscillations, by Bob Pease, EDN, September 14, Figure 2. Hysteresis helps clean up comparator response.
1989, pp. 152-6
A problem encountered with external hysteresis is that output Can you suggest a circuit that performs autozeroing when the comparator
voltage depends on supply voltage and loading. This means the is off-line to minimize drift?
hysteresis voltage can vary from application to application; though Try the circuit shown in Figures 3 and 4. In the Calibrate mode,
this affects resolution, it need not be a serious problem, since the the input is disconnected and the positive input of the comparator
hysteresis is usually a very small fraction of the range and can is switched to ground. The comparator is connected in a loop with
tolerate a safety margin of two or three (or more) times what one a pair of low-voltage sources of opposing polarity charging a
might calculate. Swapping in a few comparators can help buffered capacitor in response to the comparator’s output state.
confidence in the safety margin. Don’t use wirewound resistors
for feedback; their inductance can make matters worse. If the comparator’s minus input terminal is above ground, then
the comparator output will be low, the 1-µF capacitor will be
What’s the difference between propagation delay and prop-delay connected to the negative voltage (–365 mV) and the voltage from
dispersion? Which of the two specifications is of most concern? the buffer amplifier will ramp down until it is below the plus input
Propagation delay is the time from when the input signal crosses (ground)—plus hysteresis and any offsets—at which point the
the transition point to when the output of the comparator actually comparator switches. If it is below ground, the comparator’s output
switches. Propagation-delay dispersion is the variation in prop delay will be high, the capacitor will be connected to the positive voltage
as a function of overdrive level. If you’re using a comparator in (+365 mV), the output from the buffer amplifier ramps up. In the
pin-drive electronics in an automatic test system, then prop-delay final state, each time the comparator switches (when the ramped
dispersion will determine the maximum edge resolution. In change exceeds the hysteresis voltage), the polarity of the current
contrast, propagation delay can be considered as a fixed time offset is reversed; thus the capacitor voltage averages out the offsets of
and therefore compensated for by other techniques. the buffer and comparator.
I have a +5-volt system and don’t want to add an additional supply At the end of the Calibrate cycle, the JFET switch is opened, with
voltage; can I use my comparator with a single supply? the capacitor charged to a voltage equal to the offsets of the
Yes, but to establish the threshold use an adequately bypassed comparator and buffer ± the hysteresis voltage. At the same time,
stable reference source well within the common-mode range of the Calibrate signal goes low, disabling the feedback to the polarity
the device. The signal level is also to be referenced to this source. switch and connecting the input signal to the comparator. b
specified common-mode range (even though within the specified –5V ph-ph
signal range), the comparator may respond erroneously. For proper TRIWAVE (5kHz)
(AD790 +INPUT)
operation, ensure that both input signals do not exceed the
common-mode range of the specific comparator. For example,
the AD790 has a +VS differential input range, but its common- Figure 4. Comparator output, buffer output, and comparator
mode range is from – VS to 2 volts below +VS. input.
input stages, although modern processing techniques are inverting and non-inverting input terminals reverse functions.
overcoming this drawback. Since MOSFETS also tend to have Negative feedback becomes positive feedback and the circuit
relatively high offset voltages, junction FETs (JFETs) are used may latch up. This latchup is unlikely to be destructive, but
for high-performance low-bias-current op amps. A typical JFET power may have to be switched off to correct it. This figure
op-amp input stage is shown in this figure. shows the effect of such phase inversion in a circuit where
latch-up does not occur. The problem may be avoided by using
bipolar amplifiers, or by restricting the common-mode range
of the signal in some way.
their inverting and non-inverting inputs. Current-feedback, or on the horizontal scale. Let’s read the chart for the ADOP27:
transimpedance, op-amps, which have different input structures The horizontal line indicates the ADOP27’s voltage noise level
at these two inputs, do not. Their data sheets must be consulted of about 3 nV/√Hz is equivalent to a source resistance of less
for details of the noise on the two inputs. than about 500 Ω. Noise will not be reduced by (say) a 100-Ω
The noise of op-amps is Gaussian with constant spectral source impedance, but it will be increased by a 2-kΩ source
density, or “white”, over a wide range of frequencies, but as impedance. The vertical line for the ADOP27 indicates that,
frequency decreases the spectral density starts to rise at about for source resistances above about 100 kΩ, the noise voltage
3 dB/octave. This low-frequency noise characteristic is known produced by amplifier’s current noise will exceed that contributed
as “1/f noise” since the noise power spectral density goes by the source resistance; it has become the dominant source.
inversely with frequency (actually 1/f g). It has a –1 slope on a
log plot (the noise voltage (or current) 1/√f spectral density
slopes at –1/2). The frequency at which an extrapolated –3 dB/
octave spectral density line intersects the midfrequency
constant spectral density value is known as the “l/f corner
frequency” and is a figure of merit for the amplifier. Early
monolithic IC op-amps had 1/f corners at over 500 Hz, but
today values of 20–50 Hz are usual, and the best amplifiers
(such as the AD-OP27 and the AD-OP37) have corner
frequencies as low as 2.7 Hz. 1/f noise has equal increments
for frequency intervals having equal ratios, i.e., per octave or
per decade.
Remember that any resistance in the non-inverting input will
Q. Why don’t you publish a noise figure? have Johnson noise and will also convert current noise to a
A. The noise figure (NF) of an amplifier (expressed in dB) is a noise voltage; and Johnson noise in feedback resistors can be
measure of the ratio of the amplifier noise to the thermal noise significant in high-resistance circuits. All potential noise sources
of the source resistance. must be considered when evaluating op amp performance.
Vn = 20 log { [Vn(amp)+Vn(source)]/Vn(source)} Q. You were going to tell me about Johnson noise.
It is a useful concept for r-f amplifiers, which are almost always A. At temperatures above absolute zero, all resistances have noise
used with the same source resistance driving them (usually due to thermal movement of charge carriers. This is called
50 Ω or 75 Ω), but it would be misleading when applied to op Johnson noise. The phenomenon is sometimes used to measure
amps, since they are used in many different applications with cryogenic temperatures. The voltage and current noise in a
widely varying source impedances (which may or may not be resistance of R ohms, for a bandwidth of B Hz, at a temperature
resistive). of T kelvins, are given by:
Q. What difference does the source impedance make?
V n = 4kTRB and I n = 4kTB / R
A. At temperatures above absolute zero all resistances are noise
sources; their noise increases with resistance, temperature, and Where k is Boltzmann’s Constant (1.38 × 10–23 J/K). A handy
bandwidth (we’ll discuss basic resistance noise, or Johnson noise, rule of thumb is that a 1-kΩ resistor has noise of 4 nV/√Hz at
in a moment). Reactances don’t generate noise, but noise room temperature.
currents through them will develop noise voltages. All resistors in a circuit generate noise, and its effect must always
If we drive an op-amp from a source resistance, the equivalent be considered. In practice, only resistors in the input(s) and,
noise input will be the RSS sum of the amplifier’s noise voltage, perhaps, feedback, of high-gain, front-end circuitry are likely
the voltage generated by the source resistance, and the voltage to have an appreciable effect on total circuit noise.
caused by the amplifier’s I n flowing through the source Noise can be reduced by reducing resistance or bandwidth,
impedance. For very low source resistance, the noise generated but temperature reduction is generally not very helpful unless
by the source resistance and amplifier current noise would a resistor can be made very cold—since noise power is
contribute insignificantly to the total. In this case, the noise at proportional to the absolute temperature, T = °C + 273°. b
the input will effectively be just the voltage noise of the op-amp. (to be continued)
If the source resistance is higher, the Johnson noise of the source REFERENCES
resistance may dominate both the op-amp voltage noise and Barrow, J., and A. Paul Brokaw, “Grounding for Low- and High-Frequency
the voltage due to the current noise; but it’s worth noting that, Circuits,” Analog Dialogue 23-3, 1989.
since the Johnson noise only increases with the square root of Bennett, W. R., Electrical Noise. New York: McGraw-Hill, 1960.
Freeman, J. J., Principles of Noise. New York: John Wiley & Sons, Inc., 1958.
the resistance, while the noise voltage due to the current noise Gupta, Madhu S., ed., Electrical Noise: Fundamentals & Sources. New York: IEEE
is directly proportional to the input impedance, the amplifier’s Press, 1977. Collection of classical reprints.
current noise will always dominate for a high enough value of Motchenbacher, C. D., and F. C. Fitchen, Low-Noise Electronic Design. New
input impedance.When an amplifier’s voltage and current noise York: John Wiley & Sons, Inc., 1973.
are high enough, there may be no value of input resistance for Rice, S.O., “Math Analysis for Random Noise” Bell System Technical Journal 23
July, 1944 (pp 282–332).
which Johnson noise dominates. Rich, Alan, “Understanding Interference-Type Noise,” Analog Dialogue 16-3,
This is demonstrated by the figure nearby, which compares 1982.
- - - , “Shielding and Guarding,” Analog Dialogue 17-1, 1983.
voltage and current noise noise for several Analog Devices op Ryan, Al, and Tim Scranton, “DC Amplifier Noise Revisited,” Analog Dialogue
amp types, for a range of source-resistance values. The diagonal 18-1, 1984.
line plots vertically the Johnson noise associated with resistances van der Ziel, A. Noise. Englewood Cliffs, NJ: Prentice-Hall, Inc., 1954. b
It may be cured by using an op amp with sufficient bandwidth that any current, I, is a flow of electrons or holes, which carry
to have a low output impedance at the ADC’s clock frequency, discrete charges, and the noise given in the formula is just an
or by choosing an ADC containing an input buffer or one whose expression of the graininess of the flow.
input impedance is not modulated by its internal clock (many He concludes that the omission of this noise component in
sampling ADCs are free of this problem). In cases where the any circuit carrying current, including purely resistive circuits,
op amp can drive a capacitive load without instability, and the can lead to serious design problems. And he illustrates its
reduction of system bandwidth is unimportant, a shunt significance by pointing out that this noise current, calculated
capacitor decoupling the ADC input may be sufficient to effect from the flow of dc through any ideal resistor, becomes equal
a cure. to the thermal Johnson noise current at room temperature when
Q. Are there any other interesting noise phenomena in high-precision only 52 mV is applied to the resistor—and it would become
analog circuits? the dominant current noise source for applied voltages higher
A. The tendency of high-precision circuitry to drift with time is a than about 200 mV.
noise-like phenomenon (in fact, it might be argued that, at a A. Since designers of low-noise op amps have blithely ignored
minimum, it is identical to the lower end of 1/f noise). When this putative phenomenon, what’s wrong? The assumption that
we specify long-term stability, we normally do so in terms of the above shot noise equation is valid for conductors.
µV/1,000 hr or ppm/1,000 hr. Many users assume that, since Actually, the shot noise equation is developed under the
there are, on the average, 8,766 hours in a year, an instability assumption that the carriers are independent of one another.
of x/1,000 hr is equal to 8.8 x/yr. While this is indeed the case for currents made up of discrete
This is not the case. Long-term instability (assuming no long- charges crossing a barrier, as in a junction diode (or a vacuum
term steady deterioration of some damaged component within tube), it is not true for metallic conductors. Currents in
the device), is a “drunkard’s walk” function; what a device did conductors are made up of very much larger numbers of
during its last 1,000 hours is no guide to its behavior during carriers (individually flowing much more slowly), and the noise
the next thousand. The long-term error mounts as the square- associated with the flow of current is accordingly very much
root of the elapsed time, which implies that, for a figure of smaller—and generally lost in the circuit’s Johnson noise.
x/l,000 hr, the drift will actually be multiplied by √8.766, or Here’s what Horowitz and Hill2 have to say on the subject:
about 3× per year, or 9× per 10 years. Perhaps the spec should
be in µV/1,000 √hr. “An electric current is the flow of discrete electric charges,
not a smooth fluidlike flow. The finiteness of the charge
In fact, for many devices, things are a bit better even than this. quantum results in statistical fluctuations of the current. If the
The “drunkard’s walk” model, as noted above, assumes that charges act independently of each other,* the fluctuating current
the properties of the device don’t change. In fact, as the device is . . .
gets older, the stresses of manufacture tend to diminish and
the device becomes more stable (except for incipient failure I noise (rms) = InR = (2 qIdc B)1/2
sources). While this is hard to quantify, it is safe to say that— where q is the electron charge (1.60 × 10–19 C) and B is the
provided that a device is operated in a low-stress environment— measurement bandwidth. For example, a “steady” current of
its rate of long-term drift will tend to reduce during its lifetime. 1 A actually has an rms fluctuation of 57 nA, measured in a
The limiting value is probably the 1/f noise, which builds up as 10-kHz bandwidth; i.e., it fluctuates by about 0.000006%.
the square-root of the natural logarithm of the ratio, i.e., √1n 8.8 The relative fluctuations are larger for smaller currents: A
for time ratios of 8.8, or 1.47 × for 1 year, 2.94 × for 8.8 years, “steady” current of 1 µA actually has an rms current-noise
4.4 × for 77 years, etc. fluctuation, over 10 kHz, of 0.006%, i.e., –85 dB. At 1 pA dc,
the rms current fluctuation (same bandwidth) is 56 fA, i.e., a
A READER’S CHALLENGE: 5.6% variation! Shot noise is ‘rain on a tin roof.’ This noise,
Q. A reader sent us a letter that is just a wee bit too long to quote like resistor Johnson noise, is Gaussian and white.
directly, so we’ll summarize it here. He was responding to the
mention in these columns (Analog Dialogue 24-2, pp. 20-21) “The shot noise formula given earlier assumes that the charge
of the shot effect, or Schottky noise (Schottky was the first to carriers making up the current act independently. That is
note and correctly interpret shot effect—originally in vacuum indeed the case for charges crossing a barrier, as for example
tubes1). Our reader particularly objected to the designation of the current in a junction diode, where the charges move by
shot noise as solely a junction phenomenon, and commented diffusion; but it is not true for the important case of metallic
that we have joined the rest of the semiconductor and op-amp conductors, where there are long-range correlations between charge
engineering fraternity in disseminating misinformation. carriers. Thus the current in a simple resistive circuit has far less
noise than is predicted by the shot noise formula.* Another
In particular, he pointed out that the shot noise formula— important exception to the shot-noise formula is provided by
In = √2q IB amperes, our standard transistor current-source circuit, in which negative
where In is the rms shot-noise current, I is the current flowing feedback acts to quiet the shot noise.”
through a region, q is the charge of an electron, and B is the *Italics ours
1
Goldman, Stanford, Frequency Analysis, Modulation, and Noise. New York:
bandwidth—does not seem to contain any terms that depend McGraw-Hill Book Company, 1948, p. 352.
on the physical properties of the region. Hence (he goes on) 2
Horowitz, Paul and Winfield Hill, The Art of Electronics, 2nd edition. Cam-
shot noise is a universal phenomenon associated with the fact bridge (UK): Cambridge University Press, 1989, pp. 431-2. b
C IDEAL HF DECOUPLING HAS These reactances must be considered carefully when designing
1. LOW INDUCTANCE CAPACITOR (MONOLITHIC CERAMIC)
GROUND 2. MOUNTED VERY CLOSE TO THE IC
3. WITH SHORT LEADS
high frequency circuits which contain resistors.
IC PLANE
4. AND SHORT, WIDE PC TRACKS
Q. But many of the circuits you describe are for making precision
IT MAY BE SHUNTED WITH A TANTALUM BEAD ELECTROLYTIC
TO PROVIDE GOOD LF DECOUPLING AS WELL. measurements at DC or very low frequencies. Stray inductance and
capacitance don’t matter in such applications, do they?
C
A. They actually do. Since transistors (either discrete or within
ICs) have very wide bandwidths, if such circuits are terminated
with reactive loads, they may sometimes oscillate at frequencies
of hundreds or thousands of MHz; bias shifts and rectification
THIS SORT OF THING IS USELESS! associated with the oscillations can have devastating effects on
low-frequency precision and stability.
Supply decoupling does more than prevent instability. An op-
amp is a four-terminal device (at least), since there must be a Even worse, this oscillation may not appear on an oscilloscope,
return path for both input signals and the output circuit. It is either because the oscilloscope bandwidth is too low for such
customary to consider the common terminal of both op-amp a high frequency to be displayed, or because the scope probe’s
supplies (for op-amps using + supplies) as the output signal capacitance is sufficient to stop the oscillation. It is always wise
return path, but in fact, one of the supplies will be the de facto to use a wideband (LF to 1.5 GHz or more) spectrum analyzer
return path at higher frequencies, and the decoupling of the to verify the absence of parasitic oscillations in a system. Such
amplifier’s supply terminal for this supply must take into checks should be made while the input is varied throughout
consideration both the necessity of normal high-frequency its whole dynamic range, since parasitic oscillations may
decoupling and the routing of the output ground.* sometimes occur over a narrow range of inputs.
Q. In “Ask the Application Engineer,” you’re always describing Q. Are there any problems with the resistance of resistors?
non-ideal behavior of integrated circuits. It must be a relief to use a A. The resistance of a resistor is not fixed but varies with
simple component like a resistor and know that you have a near-ideal temperature. The temperature coefficient (TC) varies from a
component. few parts per million per degree Celsius (ppm/°C) to thousands
A. I only wish that a resistor was an ideal component, and that of ppm/°C. The resistors with the best stability are wirewound
that little cylinder with wire ends behaved just like a pure or metal film types, and the worst are carbon composition.
resistance. Real resistors also contain imaginary resistance Large temperature coefficients are sometimes useful (an
components—in other words they’re reactive. Most resistors earlier “Ask the Applications Engineer”† mentioned how a
have a small capacitance, typically 1-3 pF, in parallel with their +3,500-ppm/°C resistor can be used to compensate for the
resistance, although some types of film resistors, which have a kT/q term in the equation for the behavior of a junction diode).
spiral groove cut in their resistive film, may be inductive, with But in general, the variation of resistance with temperature is
inductances of a few tens or hundreds of nH. likely to be a source of error in precision circuits.
*This issue is developed in detail in the free application note, “An IC amplifier
user’s guide to decoupling, grounding, and making things go right for a change,”
by Paul Brokaw. [AN-202]
ground plane material (standard 1 oz copper has resistance of For example, 1 cm of 0.25-mm track has an inductance of
0.45 mΩ/square) and the length through which currents flow, 10 nH.
but the calculation can be complicated. At DC and low
frequencies (dc-50 kHz), voltage drops can be measured 2R L, R in mm
with an instrumentation amplifier such as the AMP-02 or
L
the AD620.
2L
WIRE INDUCTANCE = 0.0002L In – 0.75 µH
R
powered from the same supply as the circuit being tested—or EXAMPLE: 1cm of 0.25mm PC track has an inductance of 9.59 nH
from its own supply—but the grounds of the amplifier, its (H = 0.038mm, W = 0.25mm, L = 1cm)
supply if separate, and the oscilloscope must be connected to But inductive reactance is generally much less of a problem
the power ground of the circuit under test at the power supply. than stray flux cutting inductive loops and inducing voltages;
The voltage between any two points on the ground plane may loop area must be minimized, since voltage is proportional to
then be measured by applying the probes to those points. The it. In wired circuits this is easily done using twisted pairs.
combination of the amplifier gain and oscilloscope sensitivity REDUCING LOOP INDUCTANCE
give a measurement sensitivity of 5 µV/div. Amplifier noise
will swell the oscilloscope trace to a band about 3 µV wide but
LOAD
it is still possible to make measurements with about 1-µV SOURCE LOAD
resolution—sufficient to identify most low-frequency ground SOURCE
with two coils of 6-10 turns each. One coil is connected to the
input of a spectrum analyzer, the other to the probes, to make
a high-frequency isolating transformer. Reducing area and increasing separation will minimize the
effect.
The test is similar to the LF one but the spectrum analyzer
displays noise as an amplitude-frequency plot.While this differs Usually, all that is necessary is to minimize loop area and
from time-domain information, sources of noise may be easier maximize the distance between potentially interfering loops.
to identify by their frequency signatures; in addition, the use Occasionally magnetic shielding is required, but it is expensive
of a spectrum analyzer provides at least 60 dB more sensitivity and liable to mechanical damage; avoid it whenever possible.
than is possible with a broadband oscilloscope. REFERENCES
Q. What about the inductance of wires? The Best of Analog Dialogue 1967-1991. Norwood MA: Analog
A. The inductance of wire- and PC-track leads should not Devices (1991), pp. 120-129, 193-195. Contains many additional
be overlooked at higher frequencies. Here are some references.
approximations for calculating the inductance of straight wires Mixed-Signal Design Seminar Notes. Norwood MA: Analog Devices
and runs. (1991). Contains additional References. b
SIMPLE ZENER DIODE BURIED (OR SUBSURFACE) *From Analog Dialogue 9-1 (1975) also The Best of Analog Dialogue, 1967 to
ZENER DIODE 1991, p. 72.
loop around the sources of error, these connections avoid the RS (or a current source) must be chosen so that for all expected
effects of voltage drops; they also correct gain and offset errors values of negative supply and reference load current the ground-
when current-buffer amplifiers are used to drive substantial and output-terminal currents are within ratings.
loads or sink currents flowing in the wrong direction. The sense Q. What about capacitive loads?
terminal should be connected to the output side of the buffer
amplifier, preferably at the load. A. Many references have output amplifiers that become unstable
and may oscillate when operated with large capacitive loads;
so it is inadvisable to connect high capacitance (several µF or
more) to the output of a reference to reduce noise, but 1-10 nF
capacitors are often recommended—and some references (e.g.,
AD588) have noise-reduction terminals to which capacitance
can be safely connected. If force-sense terminals are available,
it may be possible to tailor loop dynamics under capacitive
load. Consult data sheets and manufacturers’ Application
Engineers to be sure. Even if the circuit is stable, it may not be
advisable to use large capacitive loads since they increase the
turn-on time of the reference.
Q. Don’t references turn on as soon as power is applied?
A. By no means. In many references the current that drives the
reference element (Zener or bandgap) is derived from the
Q. What do you mean by flow in the wrong direction”? stabilized output. This positive feedback increases dc stability
A. Consider a +5-V reference operated from a +10-V supply. If but leads to a stable “off ” state that resists startup. On-chip
its 5-volt output terminal is loaded by a resistor to ground, circuitry to deal with this and facilitate startup is normally
current will flow out of the terminal. If the resistor is instead designed to draw minimal current, so many references come
connected to the +10-V supply, current will flow into the up somewhat slowly (1-10 ms is typical). Some devices are
terminal. Most references will allow net current flow in either indeed specified for faster turn-on; but some are even slower.
direction; but some will source current but not sink it—or will If the designer needs reference voltage very quickly after power
sink much less than can be sourced. Such devices, identifiable is applied to the circuit, the reference chosen must have a
by the way their output current is specified on the data sheet, sufficiently fast turn-on specification; and noise reduction
may not be used in applications where substantial net current capacitance should be minimized. Reference turn-on delay may
must flow into the reference terminal. A common example is limit the opportunities for strobing the supplies of data
the use of a positive reference as a negative reference. conversion systems in order to conserve system power. The
+6V → +30V
problem must still be considered even if the reference is built
into the converter chip; it is also important in systems of this
2 type to consider the power-up characteristics of the converter
VIN as well [discussed in “Ask The Applications Engineer—1,”
VOUT 6 Analog Dialogue 22-2 (1988), p. 29].
AD586
High-precision references may require an additional period of
GND thermal stabilization after turn-on before the chip reaches
4 thermal stability and thermally induced offsets arrive at their
–5V final values. Such effects will be mentioned on the data sheet
RS
4kΩ
and are unlikely to exceed a few seconds.
–15V
Q. Does using these high precision references instead of its internal
Q. Why not just buy a negative reference? reference make a converter more accurate?
A. Because most single voltage-output references are positive A. Not necessarily. For example, the AD674B, a high-speed
references. Two-pin active references, of course, can be used descendant of the classical AD574, has a factory-trimmed
for either polarity; they are used in the same way as Zener calibration error of 0.25% (± l0 LSB) max, with an internal
diodes (and they are usually bandgap devices). reference guaranteed accurate to within +100 mV (1%). Since
0.25% of 10 V = 25 mV, full scale is 10.000 V + 25 mV. Suppose
For a three-terminal positive reference to be used as a negative that an AD674B with a 1% high internal reference (10.1 V)
reference, it must be able to sink current. Its output terminal had been factory-trimmed for 10.000 V full scale, by a 1%
is connected to ground and its ground terminal (which becomes gain increase. If an accurate 10.00-V AD588 system reference
the negative-reference terminal) is connected to the negative were to be connected to the device’s reference input, full scale
supply via a resistor (or a constant current source). The positive would become 10.100 V, at 4 times the specified max error.
supply pin must generally be connected to a positive supply at
least a few volts above ground. But some devices can provide Q. Please discuss the role of the clock as a system reference.
negative reference in the two-terminal mode: the positive and A. Oops, we’re out of space! This question introduces a topic that
output terminals are connected together to ground. merits thoughtful discussion. We’ll do it in a future issue. b
IA
CSTRAY
ID
You will incur some reduction in digital noise margin, but it is
A usually acceptable with TTL or CMOS logic if it’s less than a
few hundred millivolts or so. If your ADC has single-ended
AGND DGND
GND ECL outputs, you may want to put a push-pull gate on each
digital output—i.e., one with both true and complementary
A A ∆V D
outputs. Tie the grounds of this gate package to the analog
ground plane and connect the logic signals differentially across
= ANALOG = DIGITAL
A
GROUND PLANE
D
GROUND PLANE the interface. Use a differential line receiver at the other end
Q. O.K., you’ve told me to join the AGND and DGND pins of the IC which is grounded to the digital ground plane. The noise
together to the same ground plane—but I am maintaining separate between the analog and digital ground planes is now common-
analog and digital ground planes in my system. I want them tied mode—most of it will be rejected at the output of the differential
together only at one point: the common point where the power supply line receiver. You could use the same technique with TTL or
returns are all joined together and connected to chassis ground. Now CMOS, but there is usually enough noise margin not to require
what do I do? differential transmission techniques.
However, one thing you said troubles me greatly. In general, it analog ground plane to prevent digital noise from coupling
is unwise to connect the ADC outputs directly to a noisy data into the analog output.
bus. The bus noise may couple back into the ADC analog input Q. What about mixed-signal chips which contain ADCs, DACs, and
through the stray internal capacitance—which may range from DSPs such as your ADSP-21msp5O voiceband processor?
0.1 to 0.5 pF. It is much better to connect the ADC outputs
directly to an intermediate buffer latch located close to the A. The same philosophy applies. You should never think of a
ADC. The buffer latch is grounded to your digital ground plane, complex mixed-signal chip, such as the ADSP-21msp50, as
so its output logic levels are now compatible with those of the being only a digital chip! The same guidelines we’ve just been
rest of your system. discussing should be applied. Even though the effective
sampling rate of the 16-bit sigma-delta ADC and DAC is only
VA VD1 V D2
0.1µF 0.1µF 0.1µF 8 ksps, the converters operate at an oversampling frequency of
1 MHz. The device requires an external 13-MHz clock, and
A A D
an internal 52- MHz processor clock is generated from it with
a phase-locked loop. So you see, successful application of this
device requires an understanding of design techniques for both
BUFFER
ADC
LATCH precision- and high-speed circuits.
TO NOISY
DATA BUS
Q. What about the analog and digital power-supply requirements of
A these devices? Should I buy separate analog and digital power supplies
A D or use the same supply?
Q. I think I understand now, but why on earth didn’t you just call all A. This really depends on how much noise is on your digital
the ground pins of your ADC AGND in the first place; then none of supply. The ADSP-21msp50, for example, has separate pins
this would have come up in the first place? for the +5- V analog supply and the +5-V digital supply. If you
have a relatively quiet digital supply, you can probably get away
A. Perhaps. But what if the incoming-inspection person connects
with using it for the analog supply too. Be sure to properly
an ohmmeter between these pins and finds out that they are
decouple each supply pin at the device with a 0.1-µF ceramic
not actually connected together inside the package? The whole
capacitor. Remember to decouple to the analog ground plane,
lot will probably be rejected—and the IC may be blown!
not the digital ground plane! You may also want to use ferrite
Furthermore, there is a tradition associated with ADC data
beads for further isolation. The diagram below shows the proper
sheets which says we must label the pins to indicate their true
arrangement. A much safer solution is to use a separate +5-V
function, not what we would like them to be.
analog supply. You can generate the +5 V from a quiet +15-V
Q. O. K. Now, here comes a question I’ve been saving as your ultimate or +12-V supply using a three-terminal regulator, if you can
test! I have a colleague who designed a system with separate analog tolerate the extra power dissipation. b
and digital ground systems. My colleague says that, with the ADC’s
AGND pin connected to the analog ground plane and the DGND REFERENCES [not available from Analog Devices unless noted]
pin connected to the digital ground plane, the system is working 1. Ralph Morrison, Grounding and Shielding Techniques in Instrumentation, Third
Edition. New York: Wiley-lnterscience, 1986.
fine! How do you explain this? 2. Henry W. Ott, Noise Reduction Techniques in Electronic Systems, Second
A. First of all, just because a practice is not recommended doesn’t
necessarily mean you can’t get away with it some of the time 0.1µF 0.1µF
ODDS ‘N’ ENDS (Continued from earlier issues) A. The phase noise of the sampling clock is often ignored, because
by James Bryant the limiting factor on system performance used to be the
aperture jitter of the of the sample-hold—but if we consider
TIME REFERENCES (continued from 26-1—AA-11) the system as a whole, aperture jitter is just one component of
Q. Why do you say that the clock of a system is a reference? the total phase noise in the sampling clock chain.With modern
A. This comment does not necessarily apply to the conversion sampling ADCs the aperture jitter may be less important than
clock of an ADC; it applies principally to the sampling clock other components of phase noise.
of a sampled-data system. In these systems, the signal is
required to be sampled repeatedly at predictable (usually equal) 1
SNR = 20 log 10
intervals for storage, communication, computational analysis, 90
2πfta
tph = 2ps
or other types of processing. The quality of the sampling clock 14
80
is a system-performance-limiting factor. tph = 10ps
70 12
Q. But crystal oscillators are very stable, aren’t they? 60 10
tph = 50ps
ENOB
SNR – dB
A. They have good long-term stability, but they are often used in 50 8
ways which introduce short-term phase noise. Phase noise is 40
tph = 250ps
6
also introduced by designers who, instead of using crystal 30 tph = 1250ps 4
oscillators, use R-C relaxation oscillators (such as the 555 or 20
the 4046)—which have a great deal of phase noise. 10
Q. How can I ensure that my sampling clock has low phase noise? 0
1 2 3 5 7 10 20 30 50 70 100
A. Don’t use the crystal oscillator circuitry in your microprocessor FREQUENCY OF FULLSCALE SINEWAVE INPUT – MHz
or DSP processor as the source of your sampling clock. If at all
The diagram shows the effect of the total phase jitter of the
possible, do not use a logic gate in a crystal oscillator. Crystal
sampling clock on signal-to-noise ratio (SNR) or effective
oscillators made with logic gates generally overdrive the crystal;
number of bits (ENOB). This jitter has the rms value of tph,
this is bad for its long term stability, and usually introduces
which is made up of the root-sum-of-squares of the phase jitter
worse phase noise than would a simple transistor oscillator. In
on the sampling clock oscillator, the phase jitter introduced by
addition, digital noise from the processor—or from other gates
pickup during transmission of the sampling clock through the
in the package if a logic gate is used as an oscillator—will appear
system, and the aperture jitter of the SHA in the sampling
as phase noise on the oscillator output.
ADC. This diagram may be somewhat unsettling, as it shows
Q. But crystal oscillators are very stable, aren’t they? just how little phase noise is required to corrupt a
A. Ideally, use a single transistor or FET as your crystal oscillator high-resolution sampled-data system.
and buffer it with a logic gate. This logic gate, and the oscillator
itself, should have a well-decoupled supply; the other gates in MORE ON TRIMMING
the package should not be used because logic noise from them Q. I don’t have enough range to adjust the offset of my circuit—and it
will phase-modulate the signal. (They may be used for dc seems to have rather more drift than I’d expected.
applications but not for fast-switching operations.) A. I’ll bet the amplifier is a bipolar type and you are using its
If there is a divider between the crystal oscillator and the offset-trim terminals to trim other circuit voltages.
sampling clock input of the various ADCs, the divider power Q. How did you guess?
supply should be decoupled separately from the system logic
A. The range of offset adjustment of an op amp is normally 2 to
to keep power supply noise from phase-modulating the clock.
5 times the maximum expected offset of the lowest grade of
The sampling clock line should be kept away from all logic the device (in some early op amps, it was much larger, but
signals to prevent pickup from introducing phase noise. Equally, such a wide range is not ideal). If the lowest grade has a VOS
it should be kept away from low-level analog signals lest it (max) of ± 1 mV, then the likely adjustment range with the
corrupt them. recommended circuit is ± 2 to ± 5 mV.
Q. You have told me not to use the clock oscillator of my processor as If the external voltage you are attempting to compensate for is
the sampling clock source.Why not? Isn’t it sensible to use the same larger than this (referred to the op amp’s input), you will not
oscillator for both, since there will then be a constant phase be able to do so with the amplifier’s offset-trim terminals.
relationship between the signals?
Furthermore, if you are using a bipolar-input op amp, it is
A. True. But in such cases, it is often better to use a single discrete
inadvisable to use these terminals for external offset correction
low-noise oscillator to drive the processor clock input and the
because drift will be increased. Here’s why: the input stage
sampling clock divider through separate buffers (though they may
thermal drift is proportional to the internal offset; if this
share a package) than to use the oscillator in the processor. In
has been trimmed to a minimum, the drift will also be a
medium-accuracy systems with low sampling rates it may be
minimum. If you then trim the amplifier to compensate for an
possible to use the processor’s internal oscillator—but check
external offset, drift will no longer be minimized. However,
with the diagram below).
FET-input op amps have separately trimmed offset and drift,
Q. Just how serious is this problem of noise on a sampling clock? I their offset adjustment terminals may thus be used for small
hardly ever see it mentioned in articles on sampled data systems. system adjustments. b
DISTORTION – dBc
–60
rms. For a 3-V rms signal level, the corresponding signal-to- RL = 50 or 100Ω
THD – %
the THD figure. The legend on the graph indicates that the f1 f2 2 = SECOND-ORDER IMD PRODUCTS
measurement-equipment “floor” is about –120 dB; hence at 3 = THIRD-ORDER IMD PRODUCTS
AMPLITUDE
frequencies below 10 kHz, the THD may be even less. NOTE: f 1 = 5MHz, f 2 = 6MHz
A. Because of the increasing need for wide dynamic range at high Two-tone intermodulation-distortion specifications are of
frequencies, most wideband amplifiers now have distortion especial interest in r-f applications and are a major concern in
specifications. The data sheet may give individual values for the design of communications receivers. IMD products can
the second and third harmonic components, or it may give mask out small signals in the presence of larger ones. Although
THD. If THD is specified, only the first few harmonics IMD has been rarely specified in op amps operating at
contribute significantly to the result. At high frequencies, it is frequencies less than 1 MHz, many of today’s dc op amps are
often useful to show the individual distortion components wideband types that can operate usefully at radio frequencies.
separately rather than specifying THD. The AD9620 is a For this reason, it is becoming common to see IMD
600-MHz (typical –3-dB bandwidth) low distortion unity-gain specifications on fast op amps.
buffer. Here are graphs of the AD9620’s second and third Q. What are the second- and third-order intercept points, and what is
harmonic distortion as a function of frequency for various their significance?
loading conditions.
A. Usually associated with r-f applications, these specs provide the IMD products with respect to the output power actually
figures of merit to characterize the IMD performance of the delivered into the 50-Ω load rather than the actual op-amp output
amplifier. The higher the intercept power, the higher the input power.
level at which IMD becomes significant—and the lower the Another parameter that may be of interest is the 1-dB
IMD at a given signal level. compression point, shown in the figure. This is the point at
Here’s how it is derived: Two spectrally pure tones are applied which the output signal has started to limit and is attenuated
to the amplifier. The output signal power in a single tone (in by l dB from the ideal input/output transfer function.
dBm) and the relative amplitudes of the second-order and third- The figure below is a plot of the third-order intercept power
order products (referenced to a single tone) are plotted (and values for the AD9620 buffer amplifier as a function of input
extrapolated) here as a function of input signal power. frequency. Its data can be used to approximate the actual value
INTERCEPT POINTS, GAIN COMPRESSION, IMD of the third order intermodulation products at various
SECOND-ORDER frequencies and signal levels.
INT 2 INTERCEPT
OUTPUT POWER (PER TONE) – dBm
THIRD-ORDER
INT 3 50
INTERCEPT
1dB 50Ω
1dB COMPRESSION
POINT
50Ω
INTERCEPT – +dBm
FUNDAMENTAL 40
(SLOPE = 1)
TEST CIRCUIT
SECOND-ORDER IMD
(SLOPE = 2)
THIRD-ORDER IMD 30
(SLOPE = 3)
THIRD-ORDER
POWER – dBm
For a higher intercept, the line moves to the right (same slope), IMD N = N TH -ORDER IMD PRODUCT (dBm)
INT N = N TH -ORDER INTERCEPT POINT (dBm)
–40
showing lower 3rd-order products for a given input level. 3dB/dB P OUT = OUTPUT POWER INTO THE LOAD (dBm)
SLOPE 2ND ORDER IMD: –IMD 2 = INT 2 – 2•P OUT
Many r-f mixers and “gain blocks” have 50-Ω input and output –60 3RD ORDER IMD: –IMD 3 = 2•INT 3 – 3•P OUT
–68dBm IN THIS EXAMPLE:
impedances. The output power is simply the power that the –IMD 3 = 2 × 40dBm – 3 × 4dBm = 68dBm
by squaring the rms output voltage (Vo) and dividing by the load
resistance, RL. The power is converted into dBm as follows: This analysis assumes that the op-amp distortion can be
2 modeled with a simple power series expansion as described in
V0
Reference 1. Unfortunately, op amps don’t always follow simple
RL
Output power = 10 log 10 d Bm models (especially at high frequencies), so the third-order
1mW
intercept specification should primarily be used as a figure of
Since an op amp, on the other hand, is a low-output-impedance merit, rather than a substitute for measurements. b
device, for most r-f applications, the output of the op amp
must be source- and load-terminated. This means that the REFERENCES
1. Robert A. Witte, “Distortion Measurements Using a Spectrum Analyzer,”
actual op amp output power has to be 3 dB higher than the RF Design, September 1992, pp. 75-84. (not available from ADI)
power delivered to the load, as calculated from the above 2. High Speed Design Seminar, 1996. Norwood, MA: Analog Devices, Inc.
formula. In this type of application, it is customary to define 3. 1992 Amplifier Applications Guide. Norwood, MA: Analog Devices, Inc.
FERRITE
TOROID
SIGNAL
TO MEASUREMENT TO OSCILLOSCOPE
POINT OR SPECTRUM ANALYSER
GUARD
PIN
GUARD
Common-mode signals can be observed quite easily by
disabling any sensor excitation and connecting the oscilloscope
ground to the ground at the board input and joining all the The guard line is connected to ground at the source end and
sensor leads together and to the oscilloscope input. All too at the other end to the amplifier’s guard pin (or a comparable
often this signal will have an amplitude of several hundred derived voltage), which represents what the amplifier “thinks”
millivolts and contain components from low frequencies to tens is common mode, via a capacitor. The high-frequency common-
or hundreds of MHz. mode signal will appear (by definition) across the bottom
The world is full of HF noise sources: ham radio operators, winding, and will induce an equal common-mode voltage in
police, people with portable phones, garage door openers, the the other two, subtracting the common-mode voltage in series
sun, supernovas, switching power-supply and logic signals with each line and effectively cancelling the HF common-mode
(e.g., PCs). Since we cannot eliminate HF noise in the signal at the amplifier inputs.
environment, we must filter it out of low-frequency signals There are, of course, potential problems. A capacitor in series
before they arrive at precision amplifiers. with the transformer is almost essential in the guard circuit to
The simplest type of protection can be used when the signal block DC and LF and prevent transformer core saturation by
bandwidth is only a few Hz. A simple RC low-pass filter low-frequency currents in the guard circuit. The impedance
inserted ahead of an amplifier will afford both normal-mode looking into the amplifier guard terminal must be much lower
and common-mode HF protection. A suitable circuit is shown than the impedance of the transformer windings; and at very
in the figure. There are two important issues to be considered high frequencies the capacitances of the transformer will allow
in the choice of components: the resistances R and R9 (shown signal leakage or may cause phase shift. These issues set
as 1 kΩ in the diagram, a value suitable for amplifier bias incompatible constraints on the design of the transformer, if it
currents of a few nA or less) must be chosen so that they do must deal with a very wide range of common-mode frequencies.
not increase the offset appreciably as the amplifier bias current In such a case double cancellation using two separate
flows in them. The normal-mode time constant, (R + R9)C2, transformers as shown might be considered—the one nearer
must be much larger than the common-mode time constants, the amplifier having high inductance (and correspondingly high
RC1 and R9C19, otherwise the common-mode time constants capacitance) and the other having good VHF efficiency.
would have to be very carefully matched to avoid an imbalance HF TRANSFORMER
that would convert the common-mode to a signal between the (LOW C) MF TRANSFORMER
GUARD
R 1kΩ PIN
C1
GUARD
0.01µF C2
C1´ 0.1µF
0.01µF
R´ 1kΩ Other approaches are also possible: the amplifier can be sited
closer to the sensor and the long leads be replaced by leads (or
If the signal bandwidth is wider, such simple filters will not be optical fibre) carrying digital data, which is less vulnerable;
suitable because they remove the desired HF normal-mode more shielding is often (but not always) helpful; and sometimes
signals as well as the unwanted HF common-mode signals. (but rarely) it is possible to reduce the possibility of unexpected
Large HF common-mode signals are very likely to suffer HF signals (even if you keep away the hams and police, there
common-mode→normal mode conversion (as well as minor is always the possibility of the unexpected pizza delivery truck
rectification, producing low-frequency errors) if they get to radioing to its base . . .)
the amplifier, so it is necessary to use a filter which will reject The most important consideration, though, is awareness of
HF common-mode signals but will pass DC and HF the possibility of HF interference and readiness to tackle it. If
normal-mode signals. designs are always made in the expectation of unwanted HF,
Such a filter is shown below. It was devised many years ago by chances are excellent that precautions will be adequate—it’s
Bill Gunning of Astrodata and is related to the “phantom when you don’t expect it that the trouble starts.
circuit” used in long-distance telephone circuits. It uses a tightly Q. How did it work out with the French customer?
coupled “trifilar” transformer having three windings in an
A. His problem was cured with two resistors, three capacitors and
accurate 1:1:1 ratio. An AC voltage across any winding will
a piece of grounded copper foil. We went off to “La Cognette”
also be present on the others.
to celebrate. b
Now that the attenuation is in hand, we can consider the noise MODULATOR
magnitude itself: Let’s be conservative (by about 50%) and OUTPUT
SPECTRUM
MAGNITUDE – dB
take the effective filter attenuation to be 20 dB (i.e., 0.1 V/V).
To be able to calculate the maximum allowed noise spectral INPUT SHAPED QUANTIZATION
density when using a single pole filter, an estimate should be SIGNAL NOISE
SPECTRUM
made of the maximum performance degradation that aliased
noise can contribute. From the dynamic specs of the AD1877
we find that the total noise power internal to the converter is
90 dB below (32 ppm of) full-scale input. If the whole system 0 fi fb Fms Fms – fb Fms – fi Fms f
is to be within, say, 0.5 dB of this spec, the total aliased noise 2
power can’t exceed the rss difference between –90 dB and
–89.5 dB or –99.1 dB (11.1 × 10–6). Using this information, As shown, the spectral “sticks” (single frequencies) at fi and
and the fact that the input scale of the AD1877 is 3 V p-p, we Fms – fi correspond to an input signal, while the shaded area
find that aliased noise must not exceed 3/(2√2) V × 11.1 × 10–6 shows how the quantization noise has been pushed (shaped)
= 11.8 µV rms. If all this noise were assumed lumped in a single beyond the bandwidth of interest, fb.
aliased band, and noting that rms noise = noise spectral density The digital filter, which is often an n-tap FIR filter, takes the
(N.S.D.) × √BW, high-speed low-resolution (1-bit) modulator output and
11.8 µ V performs a weighted average of n modulator outputs in a
N.S.D. < = 59 nV / Hz manner dictated by the desired filter characteristics. The output
3.092 MHz– 3.052 MHz
of the filter is a high-resolution word, which becomes the A/D
This is the maximum post-filter spectral density allowed. To output. The digital filter is designed to filter out “everything”
find the maximum prefilter spectral density (MPSD), with the between fb and Fms–fb, where Fms is the sampling rate of the
effective filter attenuation of 20 dB (i.e., × 0.1) established modulator. Cleaning out all the noise in between fb and Fms–fb
previously, M.P.S.D. = 10 × 59 nV/√Hz = 0.59 µV/√Hz. makes it possible to reduce the sampling rate to values between
Clearly your system has to be pretty noisy in the 3-6-9-12-MHz Fms and 2fb without causing any spectra to overlap (i.e., aliasing).
regions in order for a simple RC filter not to suffice; however, Conceptually, reducing the sample rate, i.e., decimation, can
as always, one must be careful of ambient rf pickup. be thought of as only sending every dth digital filter output to
Q: As I understand it, the noise floor of sigma-delta converters may the A/D output, where d is the decimation factor. This will
exhibit some irregularities. Any thoughts on that? bring the spectral images close together, as shown in the figure,
which makes the output look like an output from a non
A: Most sigma-delta converters exhibit some spikes in the noise
oversampled converter. The upper figure shows the output of
floor, called idle tones. In general, these spikes have low energy,
the modulator after digital filtering but prior to decimation.
not enough to substantially affect the S/N of the converter.
The lower figure shows the spectral output after decimation—
Despite that, however, many applications cannot tolerate spikes
the final A/D output.
in the frequency spectrum that extend much beyond the white
noise floor. In audio applications, the human ear, for example, In real converters, digital filtering and decimation are intimately
does an excellent job of detecting tones in the absence of large combined for economy in design and manufacture. Thus, the
input signals even though the tones are well below the terms “digital filter” and “decimator” are used interchangeably
integrated (0-20-kHz) noise of the system. to describe the digital circuitry processing the modulator output
to produce the output of the converter.
There are two sources of idle tones. Their most common cause
is voltage-reference modulation.To understand this mechanism
a basic understanding of sigma-delta converters is needed. Here DIGITAL FILTER OUTPUT
MAGNITUDE
two states (high and low, or 0 and 1, or +1 and –1), and the
MAGNITUDE
+
INTEGRATION COMPARATOR O.K., now back to “idle tones”. Let’s start by looking at the
ANALOG ∑
INPUT
–
1/s DIGITAL
FILTERING
output of the modulator when a dc signal is applied to the input.
1-BIT AND n-BIT DIGITAL
AT F ms DECIMATION OUTPUT For an exact mid-scale dc input level, the output of the mod-
1-BIT AT F s << F ms
± V REF DAC ulator is equally likely to be high (1) or low (0), in other words,
the pulse density is 0.5, very likely to result in bitstream patterns Q: So what can I do to minimize the chances of idle tones interfering
like 010101. These regular patterns mean that the output spec- with my A/D conversion?
trum will have a spike at Fms/2 (upper figure). If the dc input A: Follow the layout recommendations and bypassing schemes
now moves somewhat off midscale, the modulator output bit recommended by the manufacturer of the converter. This
pattern will change accordingly. The spectrum of the modulator applies not only to the voltage reference, but to power supplies
output will now show spikes at Fms/2– ∂F and Fms/2 + ∂F, with and grounding as well. It is the manufacturer’s responsibility
∂F proportional to the dc change from midscale (lower figure). to minimize the voltage-reference corruption that takes place
inside the converter, but it is up to the system designer to
minimize the external coupling. By following those guidelines,
CONCENTRATION OF the user should be able to reduce the coupling to a negligible
MAGNITUDE
ENERGY AT Fms /2
level. If, despite the proper design precautions, idle tones are
still an issue, there is yet another option that can be pursued.
As I explained previously, frequency of the idle tones is a
function of the dc input. This opens up the possibility of
introducing enough dc offset on the A/D input to move the
Fms Fms f idle tones out of the bandwidth of interest to where they will
2 be filtered out by the decimation filter. If the user does not
want the dc offset to propagate through the system it can be
Fms
– ∂f
subtracted out by the processor that handles the data from the
2 A/D.
MAGNITUDE
Fms
+ ∂f
2
Q: What kind of a load does the input of sigma-delta converters present
to my signal conditioning circuitry?
A: It depends on the converter. Some sigma-delta converters have
a buffer at the input, in which case the input impedance is very
Fms Fms f high and loading is negligible. But in many cases the input is
2 connected directly to the modulator of the converter. A
With effective digital filtering, how can such tones possibly switched-capacitor sigma-delta modulator will have a simplified
find their way down to baseband? The answer is via the voltage equivalent circuit like that shown in the figure.
reference. The digital output is a measure of the ratio of the
S1 S2
analog input to the voltage reference. An x% change in the
magnitude of the voltage reference will result in a –x% change
C
in the magnitude of the digital output word. Voltage-reference
change will, in effect, amplitude modulate the A/D output. Now,
we have clocks internal to the converter, and possibly also Switches S1 and S2 are controlled by the two phases of a clock
externally, running at Fms/2. If small amounts of these clock to produce alternating closures. While S1 is closed, the input
pulses get coupled onto the voltage reference line, they will capacitor samples the input voltage. When S1 is opened, S2 is
change it slightly and, in effect, modulate the tones at Fms/2 – ∂F closed and the charge on C is dumped into the integrator,
and Fms/2 + ∂F. One of the difference frequencies created by this thus discharging the capacitor. The input impedance can be
modulation is at ∂F, and it is clearly in the bandwidth of interest. computed by calculating the average charge that gets drawn
Nonlinearities may also create tones at multiples of ∂F. by C from the external circuitry. It can be shown that if C is
Q: From your explanation it seems that if I apply an ac signal to the allowed to fully charge up to the input voltage before S1 is
converter I do not have to worry about idle tones? opened then the average current into the input is the same as
if there were a resistor of 1/(FswC) ohms connected between
A: Well, any ac signal generally has a dc component associated the input and ground, where Fsw is the rate at which the input
with it, which will have to be represented by the modulator capacitor is sampling the input voltage. F sw is directly
output, so the explanation above still applies. But if the total proportional to the frequency of the clock applied to the
dc input offset (i.e., internal converter offset plus external converter. This means that the input impedance is inversely
offsets) in your system is exactly 0, the tones will be at dc (0 Hz). proportional to the converter output sample rate.
There is another source of idle tones in lower-order (<3rd- Sometimes other factors, such as gain, can influence the input
order) modulators. The order of the modulator (number of impedance. This is the case for the 16/24-bit AD771x family
integrations) is a measure of how much quantization-noise of signal conditioning A/Ds. The inputs of these converters
shaping takes place. Second-order modulators can actually can be programmed for gains of 1 to 128 V/V. The gain is
exhibit bit patterns that show up directly in the baseband, even adjusted using a patented technique that effectively increases
if voltage-reference modulation is not occurring. This is one of Fsw (but keeps the converter output sample rate constant) and
the reasons why sigma-delta converters from Analog Devices combines the charges from multiple samples. The input
that are designed for ac applications use higher-order (≥3) impedance of these converters is, for example, 2.3 MΩ when
sigma-delta modulators. the device’s external clock is 10 MHz and the input gain is 1.
With input gain of 8, the input impedance is reduced to 288 kΩ.
These impedances, as noted earlier, represent the average For converters that have a differential input, a differential
current flow into or out of the converters. However, they are version of this circuit may be used, as shown in the figure below.
not the impedances to consider when determining the Since one input is positive with respect to ground while the
maximum allowable output impedance of the A/D driver other is negative, one input (the negative one) needs to be
circuitry. Instead, one needs to consider the charging time of supplied negative charge while the other needs to get rid of
the capacitor, C, when S1 is closed. For dc applications the negative charge when the input capacitors are switched on line.
driver circuit impedance has only to be low enough so that the Connecting a capacitor between the two inputs enables most
capacitor, C, will be charged to a value within the required of the charge that is needed by one input to be effectively
accuracy before S1 is opened.The impedance will be a function supplied by the other input. This minimizes undesirable charge
of how long S1 is closed (proportional to the sampling rate), transfers to and from the analog ground.
the capacitance, C and CEXT in parallel with the input (unless
CEXT >> C).The table shows allowable values of external series
resistance with fCLKIN = 10 MHz which will avoid gain error of
1 LSB of 20 bits—for various values of gain and external
capacitance on the AD7710. VIN +
∆∑ A/D CONVERTER
DIFFERENTIAL WITH A DIFFERENTIAL
Typical external series resistance which will not INPUT SWITCHED CAPACITOR
INPUT
introduce 20-bit gain error VIN –
Another issue has to do with interference from clock signals If Nrms is the rms value of the converter noise and VLSB is the
affecting the A/D conversion. As I noted earlier, the digital size of the LSB in volts (=␣ Vspan/2b, where b is the number of
decimation filter can’t provide any filtering of signals whose bits in the output word) the peak to peak noise in terms of
frequencies are close to multiples of the modulator sampling LSBs, NB, is
rate. To be precise, the passbands are [kFms ± fb]s where k is an
6 × Nrms 6 × 2b × Nrms
integer, Fms is the modulator sampling rate, and fb is the NB = =
VLSB Vspan
decimator cutoff frequency.
Besides the consequences for anti-aliasing discussed earlier, If the signal-to-noise ratio of a converter expresses noise power
the decimator cutoff frequencies have a bearing on the selection relative to full scale, rms signal V span / (2 2) , we have
of clock frequencies for devices that operate in the same system
as the converter. These frequency bands (i.e., the passbands) NB = 3 × 2b × 10–SNR/20
embody the converter’s greatest vulnerability to interference 2
(inductive or capacitive coupling, power supply noise, etc.), How many codes show up at the output depends where the
because any signals in these frequency bands that manage to mean of the input, i.e., the dc input value, is with respect to
get into the modulator will not be subjected to attenuation in code transitions. If the mean is close to the boundary between
the filter. Therefore one is wise to avoid using clock frequencies two output codes, more codes are likely to appear than if the
that fall in these bands to minimize the possibility of interfering mean is half way between two output codes. It can easily be
with the conversion—unless they are synchronous with the shown that NC, the number of codes appearing for a particular
converter clock. value of NB, is either INT(NB)+1 or INT(NB)+2, depending
on the dc input value [INT(NB) is the integer portion of NB].
QUESTIONS ON NOISE IN CONVERTERS And don’t be surprised to see even more codes from the less-
Q:␣ I recently evaluated a dual-supply A/D converter; one of the tests I probable noise amplitudes >± 3␣ standard deviations.
did was to ground the input and look at the output codes on a LED How many bits will NC cause to toggle on the output? The
register. To my big surprise I got a range of output codes instead of a number of bits needed to represent NC codes is
single code output as I expected?
log NC
A:␣ The cause is circuit noise. When the dc input is at the transition INT + 0.5
log 2
between two output codes, just a little circuit noise in even the
finest dc converters will ensure that two codes will appear at We can, however, see many more bits toggle, since the number
the output. This is a fact of life in the converter world. In many of bits toggling is a function of the actual value of the converter’s
instances, as in your case, the internal noise may be large dc input. Consider, for example, that a one-code transition
enough to cause several output codes to appear. Consider, for from an output word of –1 to 0 on a 2s-complement-coded
example, a converter with peak-to-peak noise of just over converter involves inverting all the output bits.
2␣ LSB. When the input of this converter is grounded, or a Lets look at an example using the AD1879, an 18-bit sigma-
clean dc source is connected to the input, we will always see delta converter with dynamic range of 103␣ dB. From the
three—and sometimes even four—codes appear at the output. definition of dynamic range we have
The circuit noise prevents the voltage being sampled from being
confined to a voltage bin that corresponds to one digital code. 103 = 20 log S
Nrms
Any external noise on the A/D input (including a noisy signal),
From the AD1879 data sheet, we find that the rms value of a
on the power supplies, or on the control lines will add to the
full-scale input signal, S, is 6/√2␣ V rms. This allows us to solve
internal circuit noise—and possibly result in more bits toggling.
for Nrms which turns out to be 30␣ µV. We next find the LSB
Q: Is there a way to determine how many codes I can expect to appear size by dividing the full input range by the number of possible
when I apply a dc signal to a converter? output codes:
A: It would not be hard in the ideal case where you knew the VLSB = 12 = 45.8 µV
noise distribution, the exact size of the codes where the dc 218
input is at and where within a code quantum the input lies (in Thus NB is 3.9. We can therefore expect either 4 or 5 different
the center, on the edge of two codes, etc.). But in reality you codes to appear at the AD1879 output when the input is
don’t have this information. However, knowing some of the ac grounded (ground corresponds to a midscale input for the
specifications (S/N, dynamic range, etc.) of the converter, you AD1879).
can make an estimate. From these specs you can find the
magnitude of the rms converter noise relative to full scale. The
noise will in all likelihood have a Gaussian amplitude
PROBABILITY
One can take this estimation one step further: If the standard actual applied dc input is slightly above the border between
deviation (the rms value) of a Gaussian distribution and the the two codes, whereas the calculations assume it is exactly on
mean (the mean of the noise is 0 in this case) are known, one the border.
can use standard tables for the Gaussian distribution to The biggest weakness of this estimating technique is the fact
calculate what percent of the time the noise will fall into a that in conventional converters the code width (the amount
voltage interval corresponding to a specific output code. A the dc input has to be increased to increase the digital output
histogram can be estimated, showing the distribution of codes by one bit) varies from code to code. This means that if the dc
at the output. Also the process can be reversed: a histogram input is in an area where codes are narrow, we can expect more
showing the distribution of noise codes at a given value of dc bits to be toggling than in an area where the codes are wide.
output permits one to estimate the S/N ratio for a converter. This method also assumes that the circuit noise within the
To make all this real, let’s continue our example involving the converter stays constant, whether the applied signal is ac or
AD1879. Consider two cases, one where the input lies midway dc. This is not exactly true in many cases.
between two output codes and one when the input is on the The estimate will probably be more accurate when used with
transition between two codes. From the calculations above, sigma-delta converters (except for “dead bands”), because
we found that the standard deviation (sd) of the noise (the neither of the two factors mentioned above is an issue in such
rms value) was 30␣ µV. The size of one LSB in terms of sd is converters.
45.78 µV Q: Ah, now I understand why there are multiple codes at the output.
= 1.524
30.0 µV But why not discard the bits that toggle and only bring out the bits
that stay steady, since the others are really indeterminate? Isn’t that
In the case where the dc input is midway between code
the real resolution of the converter?
transitions, as shown below, it is clear that any noise that falls
within –0.5␣ LSBs to +0.5␣ LSBs from the input will result in A: Many converters are designed for ac or dynamic applications
the correct code at the A/D output. This corresponds to the where THD (total harmonic distortion) and THD+N (total
noise being confined to a range of (–0.5␣ ×␣ 1.524)␣ sd to harmonic distortion+noise) are the most important specs. The
(+0.5␣ ×␣ 1.524)␣ sd from its mean (0). From standard tables one design therefore focuses on minimizing harmonic distortion
can find that the noise will fall in this range 55.4% of the time. for high- and low-level input signals, while keeping the noise
If the noise falls within 0.5␣ LSBs to 1.5␣ LSBs, the output will to acceptable levels. As it turns out, these requirements
be one code too high. Again from standard tables one can find somewhat contradict the requirements for a good dc converter,
that this will occur 21.2% of the time. Continuing in this which is optimized for precision conversion of slow moving
manner one can calculate the whole histogram showing the signals where harmonic distortion is not an issue. It is actually
distribution of output codes. desirable to have some noise (called dither) superimposed on
57.4% the input signal to minimize distortion at very low input signal
(55.4%)
600 levels; dither can also be used to improve dc accuracy where
NUMBER OF INSTANTS
agree well with the calculated values. The lower figure shows –0.1
–0.2
a case where the dc input is close to the boundary between
–0.3
two codes. By following a similar procedure, one can calculate
–0.4
how the histogram should look. Again the experimental and –0.5
calculated values are in excellent agreement. Note that the
When the A/D input is very low in amplitude, so that the signal is often about 1/3␣ LSB rms (2␣ LSBs peak-to-peak if the
amplitude does not change more than a fraction of a LSB noise is Gaussian). Clearly, this will result in a converter that
between samples, the samples stay in the same quantum, and will have more than two codes at the output when the input is
are therefore constant for a few sample periods. This is depicted grounded. We saw an example earlier involving the AD1879
in the figure below, which shows a sinusoidal input signal that which had either four or five codes appear on the output
has an amplitude of only 1.5␣ LSBs, the A/D output and the depending on the dc input level.
quantization noise. Note that the quantization error follows The figure below shows the simulated output of an A/D
the input waveform exactly while the samples are staying converter with an undithered low level input signal. The
constant. The longer the samples stay constant, the more the quantization noise is a function of the input signal magnitude
quantization noise looks like the input waveform, i.e., the at the sample instant.This correlation between the quantization
correlation between the input signal and the quantization noise noise and the input signal shows up as a cluster of harmonically
increases. While the rms of the quantization error may not have related sticks in the A/D output spectrum. Note that the
changed, the quantization error will take on a non-uniform magnitude scale in the figure is referenced to the input signal
spectral shape. In fact, the correlated quantization noise shows (not full scale input).
up as harmonics in the A/D spectrum. 0
1.5 –5
–10
INPUT VOLTAGE IN LSBs
1.0
–15
0.5 –20
–25
0
–30
–0.5 –35
–40
–1.0
–45
–1.5 –50
1.5 –55
–60
1.0 0.0 0.2 0.4 0.6 0.8 1.0 1.2 1.4 1.6 1.8 2.0 0.0 0.2 0.4 0.6 0.8 1.0 1.2 1.4 1.6 1.8 2.0
0.5 The right-hand figure shows the A/D output after a dither signal
OUTPUT CODE
0
that is 4␣ dB above the quantization noise floor is added to the
input. In this case the quantization noise depends on the
–0.5
magnitude of the dither signal at the instant when a sample is
–1.0 taken. Since the value of the dither doesn’t depend on the input
–1.5
signal, the quantization noise becomes uncorrelated to the input
1.5 and the harmonics in the A/D spectrum are eliminated, but at
INPUT SIGNAL
the cost of an overall increase in the noise floor.
1.0
QUANTIZATION Instead of actually adding noise to the A/D input, dithering
0.5 ERROR
can be accomplished by using the thermal noise of the converter
LSBs
close to the upper transition, etc.). That may be sufficiently the actual imperfections suffices, but these can only suggest ac
accurate for your application, but if you add noise to the input performance. For example, integral nonlinearity is a major
of the converter—so that several codes can appear at the factor in determining large-signal distortion (along with glitch
output—you will find that the code distribution contains energy for D/As) while differential nonlinearity governs small-
information to place the dc value of the input more exactly. signal distortion. To accurately determine the ac performance,
In the earlier examples involving the AD1879, we saw how the at least two types of tests are performed in the case of A/Ds.
code distribution looks when the input is in the vicinity of a The tests are as follows:
code transition; the two most-frequent output codes are the a) Full-scale sine
ones on either side of the transition. Their average is therefore A sinusoidal signal approaching full-scale is applied to the
a good estimate of where the input lies. In fact, taking the converter. The signal is large enough so that converter’s
average of a lot of conversions, while the input stays put, is an imperfections cause significant harmonic components to occur
excellent way of enhancing the resolution of the converter. One at multiples of the input signal frequency. The harmonics will
has to be careful, when processing the converter output, to show up in the output spectrum, along with noise. A common
allow the output word length to grow without introducing performance measure is the relative magnitude of the harmonic
roundoff errors. Otherwise one actually injects unwanted components, usually expressed in dB. Relative to what? Two
noise—called requantization noise—into the final output. Note possibilities are the applied input signal and the full scale of
that filtering out the noise is only just that; it will have no effect the converter (which in most cases is different from the applied
on other error sources of the converter, such as integral and input signal). Referring the harmonics to full scale will clearly
differential nonlinearity. yield a lower (more attractive) number than referring them to
This concept of resolution enhancement is an interesting one the rms value of the actual input signal. This reference issue
and is not restricted to the dc domain. One can actually trade causes a lot of confusion when dynamic specifications are
resolution for bandwidth in the ac domain and combine the evaluated, because there is no universally accepted standard
outputs of several converters or to construct a more-accurate for what each performance measure should be referred to. The
output. The basic principle is that signal repetitions (which best advice I can give you is: never assume anything; read
are self-correlated) add linearly, while repetitions of random manufacturers’ data sheets very carefully.
noise produce root-square increases. Thus, a fourfold increase Sometimes the magnitudes of the individual harmonics are
in number of samples increases S/N by 6␣ dB. specified, but most often only the total harmonic distortion
Q:␣ You mentioned a couple of converter ac specifications above. I am (THD) is specified. The THD measures the total power of the
somewhat confused about how S/N, THD+N,THD, S/THD, harmonics and is found by adding the individual harmonics in
S/THD+N, and dynamic range are measured on A/D and D/A rss fashion. The formula then for THD when referred to the
converters and how they relate to each other. Can you shed any light input signal is
on this?
Σ H (i) Σ H (i)
m m
2 2
A:␣ Your confusion is quite understandable. There is unfortunately rms rms
i=2 i=2
no industry standard on exactly how these quantities are 20 log10 or 10 log10
S S2
measured and therefore, what exactly they mean. Sometimes
manufacturers are guilty of choosing the definition that portrays where H(i)rms refers to the rms value of ith harmonic component
their part favorably. and S to the rms value of the input signal. Usually, harmonics
Most often data sheets include a note on the testing conditions 2 through 5 are sufficient. Note that the input-frequency, or
and how the different specs were calculated. The best advice I fundamental, component is the first harmonic. To refer any
can give is to read these very carefully. By simple calculations harmonic to full scale, add x␣ dB to the formula above, where x
you can often convert a specification for one part to a number is the magnitude of the input signal relative to full scale. This
that allows a fair comparison to a specification for another part. simple conversion formula can be applied to other
specifications, but take care to observe proper polarity of the
Most specifications are not expressed in absolute units, but as
log quantities.
relative measurements or ratios. Noise, for example, is not
specified in rms volts, but as SNR, or the ratio between signal Nowadays, clear distinction is usually made between total
power and noise power under particular test conditions. These harmonic distortion plus noise (THD+N) and THD. This has
ratios are usually expressed in decibels, dB, and occasionally not always been the case. THD+N includes not only the
as percentages (%). A power ratio, x, expressed in bels, is harmonics that are generated in the conversion, but also the
defined as log10x; multiply by 10 if expressed in decibels (one noise. The formula for THD+N when referred to the input
tenth of a bel): 10␣ log 10 x. SNR is therefore equal to signal is:
10␣ log10␣ (signal power/noise power)␣ dB. Evaluated in terms of
Σ H (i)
m
rms voltage quantities, SNR␣ =␣ 20␣ log10(Vsignal/Vnoise). N 2rms + 2
rms
i=2
Armed with this knowledge, let’s see whether we can make 20 log10
S
sense out of the multiple specifications you mentioned above
or
(many of which are redundant). Those specifications seek to
Σ H (i)
describe how the imperfections of the converter affect the m
where Nrms is the rms value of the integrated noise in the It’s important to note that the performance measures above
bandwidth specified for the measurement. are affected by: bandwidth of the measurement, the sampling
Another commonly used specification is signal to noise-plus- frequency, and the input signal frequency. For a fair comparison
distortion (S/[N+D], or S/[THD+N]), also called sinad. of two converters, one has to make sure that these test
This is essentially the inverse of THD+N, when referred to the conditions are similar for both.
signal; its dB number is the same, but with opposite polarity. Image Filtering Question
Another performance measure describing the test results is Q: I intend to use Analog’s AD1800 family of audio D/A converters
the signal to noise ratio, S/N or SNR, which is a measure of for a digital audio playback application. I understand that using an
the relative noise power, most useful for estimating response interpolator ahead of the D/A will make it easier to filter the
to small signals in the absence of harmonics. If S/N is not D/A output, assuming I want to get rid of all the images at the D/A
specified, but THD and THD+N are provided, relative to the output. But is it really necessary to filter the output, since all the
input signal, THD can be rss-subtracted from THD+N to obtain images will be above the audible range as long as sampling is at
the noise to signal ratio [=␣ 1/(S/N]. If the numbers are given >40␣ kHz?
in dB, the rss subtraction formula for logarithmic quantities in A: Good question. The audio equipment (audio amplifiers,
the Appendix can be used as follows equalizers, power amplifiers, etc.) that may eventually receive
the output of your D/As are typically built to handle 20-Hz to
20-kHz signals. Since they are not intended to respond at
SNR = –10 log10 10 THD+N /10 – 10THD/10 frequencies much beyond 20␣ kHz—and in effect themselves
function as filters—they may not have the necessary slew rate
to yield the input signal power relative to noise power expressed
and gain to handle incoming signals from an unfiltered D/A
in dB.
output having significant energy well above 20␣ kHz. With their
b) Low-level sine slew-rate and gain limitations, the amplifiers are driven into
The second test usually performed is to apply a sinusoidal signal nonlinear regions, generating distortion. These distortion
well below full scale to the converter (usually –60␣ dB). At this products are not limited to high frequencies but can affect the
input level, sigma-delta converters usually exhibit negligible 20-Hz to 20-kHz range as well. Attenuating the high frequency
nonlinearities, so only noise (no harmonic components) signals at the DAC will therefore reduce the possibility of
appears in the spectrum. At this level, S/N␣ =␣ S/N+D distortion. CD players often include filters steep enough to
= –THD+N␣ =␣ –THD, when all are referred to the same level. reduce the total out-of-band energy to >80␣ dB below full scale.
As a result, one specification indicating the noise level suffices APPENDIX
to describe the result of this test. This specification called RSS addition of logarithmic quantities: The root-square sum
dynamic range (inversely, dynamic-range distortion), specifies
the magnitude of the integrated noise (and harmonics if they of two rms signals, S1 and S2, has an rms value of S12+ S22 .
exist) over a specific bandwidth relative to full scale, when a One often needs to calculate the rss sum of two numbers that are
–60-dB input signal is applied to the converter. expressed in dB relative a given reference. To do this one has to
Conventional (i.e. not sigma-delta) converters can exhibit take the antilogs, perform the rss addition, then convert the result
harmonics in their output spectrum even with low-level input back to dB. These three operations can be combined into one
signals because all the codes may not have equal width convenient formula: If D1 and D2 are ratios expressed in dB, their
(differential nonlinearity). In some such instances, the S/N, sum, expressed in dB, is
which ignores harmonics, measured with a –60-dB input signal, 10 log10 10D1/10 + 10D2/10
is different from dynamic range.
Frequently one sees THD+N at –60-dB and dynamic range Similarly, to find the difference between two rms quantities,
specified for the same converter. These really are, as explained x = S22 – S12
above, redundant since they only differ in the reference used.
the result, x, expressed in dB, is
The only twist on dynamic range is that sometimes, when audio
converters are specified, a filter that mimics the frequency 10 log10 10D2/10 – 10D1/10 b
response of the human ear is applied to the converter output.
References:
This processing of the converter output is called A-weighting
[1] Oversampling Delta-Sigma Data Converters—Theory, Design, and
(because an A-weighting filter is used); it will effectively
Simulation, edited by J.C. Candy and G.C. Temes, IEEE Press,
decrease the noise floor, and therefore increase the signal-to-
Piscataway, NJ, 1991.
noise ratio, if the noise is white.
[2] J. Vanderkooy and S.P. Lipshitz, “Resolution Below the Least
Everything discussed above applies to both A/D and D/A Significant Bit in Digital Systems with Dither,” J.Audio Eng.
converters, with the possible exception of signal to noise ratio. Soc., vol. 32, pp. 106-113 (1984 Mar.); correction ibid., p.889
Sometimes (particularly for audio D/A converters) S/N is a (1984 Nov.).
measure of how “quiet” the D/A output is when zero (midscale) [3] A.H. Bowker and G.J. Lieberman, Engineering Statistics,
code is sent to the converter. Under these conditions, the S/N Prentice␣ Hall, Englewood Cliffs, NJ, 1972.
expresses the analog noise power at the D/A output relative to
full scale output.
designers, and its users. A major dilemma, often causing dis- 50Ω
1µF 0.1µF
A. A key requirement over the years has been the need to drive the
1µF 0.1µF
input of the op amp with a fast, precise signal source, often +VS
referred to as a flat-top generator.␣ As the name implies, such a –VS
generator would have a sharp transition between two levels of
known amplitude at time, t0, should have minimal overshoot A1, the DUT in this circuit, is configured for a gain of –1. The
(or undershoot) and then remain flat for the remainder of the voltage divider from input to output forms a second “false”
measurement time.␣ In this case “flat” means significantly flatter summing node that will replicate the signal at the amplifier’s
than the error to be measured in the amplifier. summing node. The 100-Ω potentiometer is used to null the dc
voltage.The wiper of the potentiometer is clamped by the diodes
The great accuracy is required to be certain that any output
at the input of A2 to limit saturation effects in this amplifier.
signal from the op amp is entirely due to its settling response
The output is also similarly clamped.
and not its response to a signal that is present at the input after
the step transition.␣ Any active device in the path of this signal Since the pre- and post-transition voltages at the output of A2
would require better settling characteristics than the DUT. will be the same (i.e., the difference will be zero), the settling
characteristics of this amplifier due to a step change are not strategies to produce thermal symmetry, but this is easier for low-
important for measuring A1. Thus, the output of A2 can be level high-precision devices than those designed for high-speed,
measured to find the settling time of A1. because of the large, rapid swings of power that occur.
This technique requires that the DUT be configured as an In particular, the new dielectrically isolated processes (like
inverting amplifier. The circuit can be made to work at other XFCB) that have worked wonders for improving the raw speed
gains, but the resistor values and setting of the dc balance of the op amps can have some difficulty in minimizing the
potentiometer will have more influence on the measurement. presence of thermal tails.␣ This is because the process provides
Q. Any other techniques? each transistor a separate dielectric “tub.”␣ While this dielectric
isolation reduces the parasitic capacitance and greatly speeds
A. Another technique for measuring settling time uses the up electrical performance, it also provides thermal insulation
computing power of a digital oscilloscope. It calculates a that slows the dissipation of heat to the substrate.
waveform that represents the settling error as the instantaneous
difference between the acquired input and output signals of the The seriousness of long tails depends on the application. For
DUT and compares them with the values for an ideally settling example, some systems sample at rates compatible with the initial
device. The resulting waveform is the error of the DUT. short-term settling time and are not seriously affected by longer
term drifts. Communication systems and others, where the
If there is a gain error in this system, it will show up as a dc frequency domain properties of the converted signal are most
offset in the error waveform. The calculation can be adapted for important, are examples of such systems. Although long-term
a DUT with any gain, either inverting or non inverting. It also settling errors can produce variations in gain and offset, the
can compensate for a signal generator that itself has a low long-term thermal tails will have minimal contribution to the
frequency settling tail. The DUT response to a low frequency distortion products of the digitized signal. For these systems,
input will not be influenced by that settling time. frequency domain measurements—such as distortion
Because such oscilloscopes are designed primarily for speed, in products—are more important than time domain measurements,
order to determine errors at higher resolutions, averaging must such as settling time.
be used. For example, if the A/D used in the oscilloscope has On the other hand, systems such as video and scanners might
only 8␣ bit resolution, but accuracy better than 8␣ bits, a number produce a step input, followed by a long-duration plateau of
of cycles can be averaged to increase the effective resolution of constant value. During this time, repetitive A/D conversions of
the measurement. the op amp output signal will track the long-term settling
Q. Any more? characteristic. For these systems it is important to understand
A. Yet a third way to measure settling time is to look at the output the long term settling characteristics of the op amp.
directly.␣ A Data Precision Data 6000 can directly digitize signals The figures below illustrate the long- and short-term settling
of up to 5␣ V with 16-bit accuracy and 10-ps resolution. The patterns for the AD8036, a unity-gain-stable high-speed clamp
only fly in the ointment is that the instrument relies on repetitive amp that is a good candidate for an A/D driver in high speed
sampling with a comparator probe. The waveform is built up systems. The figure at left shows that after the initial large
one bit at a time for each of the sample points. As a result, transition, the output is still about 0.09% from its long-term
obtaining a settling characteristic can be very time consuming. final value. However, the right-hand figure shows, on a 300×
This is especially so when using a relay-type flat top generator faster scale, that after about 16␣ ns the output has entered a local
with a 1-kHz upper frequency. 0.01% short-term settling region which can be usefully sampled
Q. Why do data sheets sometimes define short term and long term settling by some systems. The distortion of the AD8036 is extremely
characteristics? low (2nd and 3rd harmonics down by more than 65␣ dB with
500-Ω load) so it would be a good candidate in systems where
A. The traditional definition of settling time is the time from the this kind of performance is critical. b
input transition to the time when the amplifier output enters
0.4
the specified error zone and does not leave again. This concept 0.05
0.3 0.04
is relatively uncomplicated and straightforward. However, there 0.2 0.03
are some cases where the initial settling is fast, followed by an 0.1 0.02
0
ERROR – %
0.01
–0.1 0
amplifiers may exhibit this characteristic in the vicinity of the –0.2 –0.01
lower rail. Of greater prevalence for large transients, a “thermal –0.3 –0.02
tail” is a slow drift that continues for a relatively long time after –0.4 –0.03
–0.5
rapid settling to apparently excellent initial accuracy. –0.6
–0.04
–0.05
0 2 4 6 8 10 12 14 16 18 0 5 10 15 20 25 30 35 40 45
Thermal tails are produced when voltage level changes within SETTLING TIME - s SETTLING TIME – ns
DECODE LOGIC
The figure shows an AD7890 8-Channel multiplexed 12-bit
CS 0 CS 1 CS “N”
serial A/D converter (ADC) connected to the serial port of an
DAC 0 DAC 1 ... DAC “N”
ADSP-2105 digital signal processor (DSP). Also shown is the
timing sequence that the DSP uses to communicate with the DATA BUS
ADC. The 12 bits that constitute the conversion result are
transmitted as a serial data stream over a single line. The data Serial converters cannot in general be mapped into a processor’s
stream also includes three additional bits that identify the input memory. But a number of serial DACs could be connected to
channel that the AD7890’s multiplexer is currently selecting. the serial I/O port of the processor. Then, other ports on the
To distinguish the bits of the serial data stream from one processor could be used to generate Chip Select signals to
another, a clock signal (SCLK) must be provided, usually by enable the DACs individually. The Chip Select signals will
the DSP; However, sometimes the ADC supplies this clock as require a line from each device to the interface. But there may
an output. The DSP usually (but not always) supplies an be a limit to the number of lines on the processor that can be
additional framing pulse that is active either for one cycle at configured to transmit Chip Select signals.
the beginning of the communication or, as shown (TFS/RFS), One way of getting around this problem is to use serial DACs
for the duration of the transmission. that can be daisy-chained together. The figure shows how to
In this example, the DSP’s serial port is used to program an connect multiple DACs to a single I/O port. Each DAC has a
internal 5-bit register in the ADC. The register’s bits control Serial Data Out (SDO) pin that connects to the Serial Data In
such functions as selecting the channel to be converted, putting (SDI) pin of the next DAC in the chain. LDAC and SCLK are
the device in power-down mode, and starting a conversion. It fed in parallel to all the DACs in the chain. Because the data
should be evident that the serial interface, in this case, must clocked into SDI eventually appears at SDO (N clock cycles
be bi-directional. later), a single I/O port can address multiple DACs. However,
the port must output a long data stream (N bits per DAC times
A parallel ADC, on the other hand, connects directly (or
the number of devices in the chain). The great advantage of
possibly through buffers) to the data bus of the processor it is
this configuration is that device decoding is not needed. All
interfaced with. The figure shows the AD7892 interfaced to
devices are effectively at the same I/O location. The main
an ADSP-2101. When a conversion is complete, the AD7892
drawback of daisy chaining is accessibility (or latency). To
interrupts the DSP, which responds by doing a single read of
change the state of even a single DAC, the processor must still
the ADC’s decoded memory address.
TIMER
output a complete data stream from the I/O port.
DMA13–DMA0
Q. If serial data converters save so much space and wire, why aren’t (less than 10 effective bits of resolution).There is also an additional
they used in every space-sensitive application? danger that overshoot and noise on the sampling signal will further
A. A major disadvantage of serial interfacing is the tradeoff of degrade the integrity of the analog to digital conversion.
speed for space. For example, to program a parallel DAC, just Q. When should I choose a converter with an asynchronous serial
place the data on the data bus and clock it into the DAC with interface?
a single pulse. However, when writing to a serial DAC, the bits A. An asynchronous link allows devices to exchange unclocked
must be clocked in sequentially (N␣ ␣ clock pulses for an N-bit data with each other. The devices must initially be programmed
converter) and followed by a Load pulse. The processor’s I/O to use the identical data formats. This involves setting a
port spends a relatively large amount of time communicating particular data rate (usually expressed in baud, or bits per
with a serial converter. Consequently, serial converters with second). A convention, that defines how to initiate and end
throughput rates above 500␣ ␣ ksps are uncommon. transmissions, is also necessary. We do this using identifiable
Q. My 8-bit processor doesn’t have a serial port. Is there a way to interface data sequences called start and stop bits. The transmission may
a serial 12-bit ADC like the AD7893 to the processor’s parallel bus? also include parity bits that facilitate error detection.
A. It can of course be done using an external shift register, which RXD RXD
is loaded serially (and asynchronously), then clocked into the TXD TXD
processor’s parallel port. However, if the sense of the question COM PORT
is “without external logic”, the serial ADC can be interfaced AD1B60 ADM232 PC
of the digital input is insufficient. Here again, experimentation The figure also shows how to deal with the increasingly
is necessary—but a good starting point would be about 10␣ pF. common challenge of powering a mixed-signal system with a
single power supply. As in the grounding case, we run separate
ADC/DAC power lines (preferably power planes) to the analog and digital
TO INTERNAL portions of the circuit. We treat the digital power pin of the
DIGITAL 50Ω CIRCUITRY
INPUT converter as analog. But some isolation from the analog power
SIGNAL
CEXT CPAR pin, in the form of an inductor, is appropriate. Remember that
both power pins of the converter should have separate
decoupling capacitors. The data sheet will recommend
appropriate capacitors, but a good rule of thumb is 0.1␣ µF. If
space permits, a single 10-µF capacitor per device should also
Q. You mentioned that clock overshoot can degrade the noise be included.
performance of a converter. Is there anything else I can do from an Q. I want to design an isolated serial interface between an ADC and a
interfacing point of view to get a good signal to noise ratio? microcontroller using opto-isolators.What should I be aware of when
A.␣ Because your system is operating in a mixed-signal environment using these devices?
(i.e., analog and digital), the grounding scheme is critical. You A. Opto-isolators (also known as opto-couplers) can be used to
probably know that—because digital circuitry is noisy—analog create a simple and inexpensive high-voltage isolation barrier.
and digital grounds should be kept separate, joined at only The presence of a galvanic isolation barrier between converter
one point. This connection is usually made at the power supply. and micro also means that analog and digital system grounds
In fact, if the analog and digital devices are powered from a no longer need to be connected. As shown in the figure, an
common supply, as might be the case in a +5␣ V or +3.3␣ V isolated serial interface between the AD7714 precision ADC
single-supply system, there is no choice but to connect the and the popular 68HC11 microcontroller can be implemented
grounds back at the supply. But the data sheet for the converter with as few as three optoisolators.
probably has an instruction to connect the pins AGND and +5V +5V
DGND at the device! So how can one avoid creating a ground
loop that can result if the grounds are connected in two places? 68HC11 10kΩ
AD7714
425Ω
MISO 4N25
The figure below shows how to resolve this apparent dilemma.
The key is that the AGND and DGND labels on the converter’s DATA OUT
pins refer to the parts of the converter to which those pins are 10kΩ
425Ω
connected. The device as a whole should be treated as analog. 4N25 DATA IN
So after the AGND and DGND pins have been connected
together, there should be a single connection to the system’s MOSI
analog ground. True, this will cause the converter’s digital 10kΩ
currents to flow in the analog ground plane, but this is generally 425Ω
4N25 SCLK
a lesser evil than exposing the converter’s DGND pin to a noisy
digital ground plane. This example also shows a digital buffer, SCLK AGND DGND
referred to digital ground, to isolate the converter’s serial data
pins from a noisy serial bus. If the converter is making a point-
to-point connection to a micro, this buffer may be unnecessary.
The designer should be aware, though, that the use of
optoisolators having relatively slow rise and fall times with
TO OTHER
DIGITAL CIRCUITS CMOS converters can cause problems, even when the serial
communication is running at a slow speed.
“QUIET” NOISY SERIAL
VD VA VD
SYSTEM DIGITAL DATA BUS CMOS logic inputs are designed to be driven by a definite
POWER
ADC/DAC BUFFER logic zero or logic one. In these states, they source and sink
LATCH
SYSTEM
GROUND
a minimal amount of current. However, when the input
voltage is in transition between logic zero and logic one (0.8␣ V
AGND DGND
to 2.0␣ V), the gate will consume an increased amount of
current. If the opto-isolators used have relatively slow rise
A D
and fall times, the excessive amount of time spent in the
TO OTHER
DIGITAL GROUND/POWER PLANE
DIGITAL CIRCUITS dead-band will cause self-heating in the gate. This self-heating
ANALOG GROUND/POWER PLANE tends to shift the threshold voltage of the logic gate upwards,
which can lead to a single clock edge being interpreted by
the converter as multiple clock pulses. To prevent this
threshold jitter, the lines coming from the optoisolators
should be buffered using Schmitt trigger circuits, to deliver
fast, sharp edges to the converter. b
even be self-resonant, with comparatively high Q, because of Another thing to remember about high frequency decoupling
the low series resistance accompanying their low inductance. is the actual physical placement of the capacitor. Even short
Disc ceramic capacitors, on the other hand, are sometime quite lengths of wire have considerable inductance, so mount the
inductive, although less expensive. HF decoupling capacitors as close as possible to the IC, and
Q. I’ve seen the term “dissipation factor” used in capacitor selection ensure that leads consist of short, wide PC tracks.
charts.What is it? Ideally, HF decoupling capacitors should be surface-mount
A. Good question. Since leakage, ESR, and ESL are almost always parts to eliminate lead inductance, but wire-ended capacitors
difficult to spec separately, many manufacturers will lump are ok, providing the device leads are no longer than 1.5␣ mm.
leakage, ESR and ESL into a single specification known as G
CAP
G
CAP
dissipation factor, or DF, which basically describes the R
O
R
O
U U
inefficiency of the capacitor. DF is defined as the ratio of energy IC N IC N
D D
dissipated per cycle to energy stored per cycle. In practice, this P P
L L
is equal to the power factor for the dielectric, or the cosine of A A
N N
the phase angle. If the dissipation at high frequencies is E E
principally modeled as series resistance, at a critical frequency RIGHT WAY WRONG WAY
Dissipation factor also turns out to be the equivalent to the II. Stray Capacitance:
reciprocal of the capacitor’s figure of merit, or Q, which is also
sometimes included on the manufacturer’s data sheet. A. Now that we’ve talked about the parasitic effects of capacitors
as components, let’s talk about another form of parasitic known
Dielectric Absorption, RDA, CDA: Monolithic ceramic as “stray” capacitance.
capacitors are excellent for HF decoupling, but they have
considerable dielectric absorption, which makes them unsuitable Q. What’s that?
for use as the hold capacitor of a sample-hold amplifier (SHA). A. Well, just like a parallel-plate capacitor, stray capacitors are
Dielectric absorption is a hysteresis-like internal charge formed whenever two conductors are in close proximity to each
distribution that causes a capacitor which is quickly discharged other (especially if they’re running in parallel), and are not
and then open-circuited to appear to recover some of its charge. shorted together or screened by a conductor serving as a
Since the amount of charge recovered is a function of its Faraday shield.
previous charge, this is, in effect, a charge memory and will
cause errors in any SHA where such a capacitor is used as the A
C = 0.0085 ER
d d
hold capacitor.
C = CAPACITANCE IN pF
AREA “A”
ER = DIELECTRIC CONSTANT RELATIVE TO AIR
A = AREA OF PARALLEL CONDUCTORS IN mm2
A
d = DISTANCE BETWEEN CONDUCTORS IN mm
B
C (NC)
Capacitor Model
A B C
TIME Stray or “parasitic” capacitance commonly occurs between
parallel traces on a PC board or between traces/planes on
Capacitors that are recommended for this type of application
opposite sides of a PC board. The occurrence and effects of
include the “poly” type capacitors we spoke about earlier, i.e.,
stray capacitance—especially at very high frequencies—are
polystyrene, polypropylene, or Teflon. These capacitor types
unfortunately often overlooked during circuit modelling and
have very low dielectric absorption (typically <0.01%).
can lead to serious performance problems when the system
The characteristics of capacitors in general are summarized in circuit board is constructed and assembled; examples include
the capacitor comparison chart (page 21). greater noise, reduced frequency response, even instability.
A note about high-frequency decoupling in general: The
PARASITIC CAPACITANCE
best way to insure that an analog circuit is adequately decoupled PC TRACES PC TRACES
at both high and low frequencies is to use an electrolytic-type
capacitor, such as a tantalum bead, in parallel with a monolithic
1.5 mm
ceramic one. The combination will have high capacitance at TYPICAL
low frequency, and will remain capacitive up to quite high GROUND PLANE
frequencies. It’s generally not necessary to have a tantalum
ADJACENT BETWEEN TRACES AND/OR
capacitor on each individual IC, except in critical cases; if there BETWEEN TRACES PLANES ON OPPOSITE SIDES
is less than 10␣ cm of reasonably wide PC track between each (a) (b)
PC BOARD PC BOARD
IC and the tantalum capacitor, it’s possible to share one TOP VIEW CROSS SECTIONAL VIEW
tantalum capacitor among several ICs.
For instance, if the capacitance formula is applied to the case KOVAR® LID
of traces on opposite sides of a board, then for general purpose
CERAMIC
PCB material (ER␣ =␣ 4.7, d␣ =␣ 1.5␣ mm), the capacitance between
conductors on opposite sides of the board is just under
3␣ pF/cm2. At a frequency of 250␣ MHz, 3␣ pF corresponds to a
reactance of 212.2␣ ohms! Whatever the environmental noise level, it is good practice for
Q. So how can I eliminate stray capacitance? the user to ground the lid of any side brazed ceramic IC where
the lid is not grounded by the manufacturer. This can be done
A. You can never actually “eliminate” stray capacitance; the best
with a wire soldered to the lid (this will not damage the device,
you can do is take steps to minimize its effects in the circuit.
as the chip is thermally and electrically isolated from the lid).
Q. How do I do that? If soldering to the lid is unacceptable, a grounded phosphor-
A. Well, one way to minimize the effects of stray coupling is to bronze clip may be used to make the ground connection, or
use a Faraday shield, which is simply a grounded conductor conductive paint can be used to connect the lid to the ground
between the coupling source and the affected circuit. pin. Never attempt to ground such a lid without verifying that it is,
in fact, unconnected; there do exist device types with the lid
Q. How does it work? connected to a supply rail rather than to ground!
A. Look at the Figure; it is an equivalent circuit showing how a One case where a Faraday shield is impracticable is between
high-frequency noise source, VN, is coupled into a system the bond wires of an integrated circuit chip. This has important
impedance, Z, through a stray capacitance, C. If we have little consequences. The stray capacitance between two chip bond
or no control over Vn or the location of Z1, the next best solution wires and their associated leadframes is of the order of 0.2␣ pF;
is to interpose a Faraday shield: observed values generally lie between 0.05 and 0.6␣ pF.
CAPACITANCE, C
Z2 = 1/j C ≈ 0.2pF
VN Z1 VCOUPLED
CIRCUIT IMPEDANCE
Z1 VOLTAGE NOISE
VCOUPLED = VN ( –––––– ) COUPLED THROUGH
Z1 + Z2 STRAY CAPACITANCE
Consider a high-resolution converter (ADC or DAC), which
is connected to a high-speed data bus. Each line of the data
As shown, below, the Faraday shield interrupts the coupling bus, (which will be switching at around 2 to 5␣ V/ns),will be
electric field. Notice how the shield causes the noise and able to influence the converter’s analog port via this stray
coupling currents to return to their source without flowing capacitance; the consequent coupling of digital edges will
through Z1. degrade the performance of the converter.
FARADAY
SHIELD
Z1 VCOUPLED
VN
CIRCUIT HIGH SPEED ANALOG
IMPEDANCE DATA BUS SECTION
CAPACITIVE NOISE
AND FARADAY SHIELDS
PARASITIC (STRAY)
CAPACITANCE
Z1 VCOUPLED
VN
CIRCUIT
IMPEDANCE
This problem may be avoided by isolating the data bus,
interposing a latched buffer as an interface. Although this
solution involves an additional component that occupies board
Another example of capacitive coupling is in side-brazed area, consumes power, and adds cost, it can significantly
ceramic IC packages. These DIP packages have a small, square, improve the converter’s signal-to-noise. b
conducting Kovar lid soldered onto a metallized rim on the
ceramic package top. Package manufacturers offer only two
options: the metallized rim may be connected to one of the BUFFER/
DATA LATCH
D-A
CONVERTER
corner pins of the package, or it may be left unconnected. Most HIGH SPEED ANALOG
logic circuits have a ground pin at one of the package corners, DATA BUS SECTION
and therefore the lid is grounded. But many analog circuits do BUFFER/
DATA LATCH
A-D
CONVERTER
not have a ground pin at a package corner, and the lid is left
floating. Such circuits turn out to be far more vulnerable to
electric field noise than the same chip in a plastic DIP package,
where the chip is unshielded.
LOG – Ω
+1 VO
impedance input, resulting in zero input voltage, and uses Z(s)
LG
RO
BODE PLOT
current feedback to maintain zero input current. VIN–
I ERR R F+R O NG
The transfer function of a transimpedance amplifier is expressed
RF
RG LOG f
as a voltage output with respect to a current input. As the CURRENT FEEDBACK RF O fCL
AMPLIFIER, NONINVERTING
function implies, the open-loop “gain”, vO/iIN, is expressed in GAIN CONNECTION
ohms. Hence a current-feedback op amp can be referred to as
a transimpedance amplifier. It’s interesting to note that the Consider now a simplified model for a current-feedback
closed-loop relationship of a voltage-feedback op amp circuit amplifier. The noninverting input is the high-impedance input
can also be configured as a transimpedance, by driving its of a unity gain buffer, and the inverting input is its low-
dynamically low-impedance summing node with current (e.g., impedance output terminal. The buffer allows an error current
from a photodiode), and thus generating a voltage output equal to flow in or out of the inverting input, and the unity gain
to that input current multiplied by the feedback resistance. forces the inverting input to track the noninverting input. The
Even more interesting, since ideally any op amp application error current is mirrored to a high impedance node, where it is
can be implemented with either voltage or current feedback, converted to a voltage and buffered at the output. The high-
this same I-V converter can be implemented with a current impedance node is a frequency-dependent impedance, Z(s),
feedback op amp. When using the term transimpedance amplifier, analogous to the open-loop gain of a voltage feedback amplifier;
understand the difference between the specific current- it has a high dc value and rolls off at 20␣ dB/decade.
feedback op amp architecture, and any closed-loop I-V
converter circuit that acts like transimpedance. The closed-loop transfer function is found by summing the
currents at the V IN– node, while the buffer maintains
Let’s take a look at the simplified model of a voltage feedback VIN+␣ =␣ VIN–. If we assume, for the moment, that the buffer has
amplifier. The noninverting gain configuration amplifies the zero output resistance, then Ro␣ =␣ 0Ω
difference voltage, (VIN+␣ –␣ VIN–), by the open loop gain A(s)
Vo − V IN − −V IN −
+ Ierr = 0 and Ierr =V 0 / Z (s )
and feeds a portion of the output back to the inverting input
+
through the voltage divider consisting of RF and RG. To derive RF RG
the closed-loop transfer function of this circuit, Vo/VIN+, assume Substituting, and solving for Vo/VIN+
Z (s)
A(s)
Vo R 1
= 1+ F , where LG =
V IN +
VIN+ VO
A(s)
RG 1 RF
GAIN – dB
VIN– LG
BODE PLOT 1+
LG
RF
RG
VOLTAGE FEEDBACK
NG The closed-loop transfer function for the current feedback
LOG f
AMPLIFIER, NONINVERTING fCL amplifier is the same as for the voltage feedback amplifier, but
GAIN CONNECTION
the loop gain (1/LG) expression now depends only on RF, the
LOG Ω
VIN+
+ With the resistor outside the feedback loop, but in series with
ZOL(S)
+1 VO the load capacitance, the amplifier doesn’t directly drive a
RO purely capacitive load. A CF op amp also gives the option of
–
increasing R F to reduce the loop gain. Regardless of the
RF RF
ZF(S) RF + RO 1 +
RG approach taken, there will always be a penalty in bandwidth,
RG slew rate, and settling time. It’s best to experimentally optimize
CF LOG f
fP fZ a particular amplifier circuit, depending on the desired
characteristics, e.g., fastest rise time, fastest settling to a
Another issue to consider is the effect of shunt capacitance at specified accuracy, minimum overshoot, or passband flatness.
the inverting input. Recall that with a voltage feedback VIN+ RS
increasing the rate of closure between the noise gain and open RG
RF
RO
ranges. This calls for level shifting or ac coupling and biasing to
–
the proper range, but this is already a requirement in most
RF
RF ZF(S) RF + RO 1 +
RG
single-supply systems. If the system must operate to one or
CIN RG
both rails, or if the maximum amount of headroom is demanded
CF LOG f
fZ1 fP fZ2 in ac-coupled applications, a current feedback op amp may
simply not be the best choice. Another factor is the rail-to-rail
Load capacitance presents the same problem with a current output swing specifications when driving heavy loads. Many
feedback amplifier as it does with a voltage feedback amplifier: so-called rail-to-rail parts don’t even come close to the rails
increased phase shift of the error signal, resulting in degradation when driving back- terminated 50- or 75-Ω cables, because of
of phase margin and possible instability. There are several well- the increase in VCESAT as output current increases. If you really
documented circuit techniques for dealing with capacitive need true rail-to-rail performance, you don’t want or need a
loads, but the most popular for high speed amplifiers is a resistor current feedback op amp; if you need highest speed and output
in series with the output of the amplifier (as shown below). current, this is where CF op amps excel. b