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Volterra, Lec

,

ons sur les e´quations inte´grales et les e´quat-

ions integro-differentielles, Gauthier-Villars, Paris, 1913.

17. V. Volterra, Lec

,

ons sur la the´orie mathe´matique de la lutte

pour la vie, Gauthier-Villars, Paris, 1931.

18. F. Bloom, Ill-Posed Problems for Integrodifferential Equations

in Mechanics and Electromagnetic Theory, SIAM, Philadel-

phia, 1981.

INTERMEDIATE-FREQUENCY AMPLIFIERS

H. R. WALKER

Pegasus Data Systems, Inc.

Edison, New Jersey

The intermediate-frequency (IF) ampliﬁer is the circuitry

used to process the information bearing signal between

the ﬁrst converter, or mixer, and the decision making cir-

cuit, or detector. It can consist of a very few or a great

many component parts. Generally, it consists of an ampli-

fying stage or device to provide gain, plus a bandpass ﬁl-

ter, or ﬁlters, to limit the frequency band to be passed. The

signal to be processed can be audio, video, digital, or

pulsed, using amplitude modulation, frequency modula-

tion, phase modulation, or combinations thereof. Several

examples are shown in Figs. 13–19.

Bandpass IF ampliﬁers are also used in radio trans-

mitters to limit the occupied bandwidth of the transmitted

signal. Certain modulation methods create a very broad

frequency spectrum, which can interfere with adjacent

channels. Regulatory agencies, such as the FCC (Federal

Communications Commission), require that these out-of-

band signals be reduced below a certain permissible level,

so they must undergo processing through a bandwidth-

limiting ﬁlter and ampliﬁer at the transmitter.

For each application there are certain design restric-

tions or rules that must be followed to achieve optimum

results.

1. GENERAL IF AMPLIFIER FUNCTIONS AND

RESTRICTIONS

The ﬁve basic IFampliﬁer functions and requirements are

as follows:

1. Image Rejection. The mixer stages in a superhet-

erodyne receiver can convert any frequency below or

above the local oscillator frequency to an intermediate

frequency. Only one of these frequencies is desired. The

intermediate frequency must be chosen so that undesir-

able frequencies or images are removed by the RF ampli-

ﬁer ﬁlter (preﬁltering) and are rejected by the mixer. This

may mean that two or three different intermediate fre-

quencies must be used within the same receiver. The in-

termediate frequencies in common use range from 0 Hz to

approximately 2.0GHz.

2. Selectivity. Selectivity is required to reject as much

as possible of any adjacent channel interfering signal.

Generally this means attempting to obtain a bandpass

ﬁlter characteristic as close to that of the ideal ﬁlter as

possible that will pass the necessary Nyquist bandwidth

(the baseband bandwidth from 0Hz to the highest

frequency to be passed) without introducing harmful

amplitude or phase distortion.

3. Gain. Gain is required to amplify a weak signal to a

useful level for the decisionmaking circuit. This gain must

be provided by means of a stable ampliﬁer that introduces

a minimum of noise, so as not to degrade the receiver noise

ﬁgure. All circuit input and output impedances should be

properly matched for optimum power transfer and circuit

stability.

4. Automatic Gain Control. The ampliﬁer gain must

vary automatically with signal strength so that the deci-

sionmaking circuit receives a signal of as nearly constant

level as possible. The stages of the IFampliﬁer must not be

overdriven, or go into limiting, until after the last band-

pass ﬁlter, to prevent ‘‘splattering,’’ or broadening and dis-

tortion of the signal.

5. Linearity. The ampliﬁer should be linear in phase or

amplitude to prevent distortion of the recovered informa-

tion. AM receivers should be linear in amplitude, while

FM or PM receivers should be linear in phase. Some mod-

ulation methods can tolerate more linearity distortion

than others.

2. SELECTING THE INTERMEDIATE FREQUENCY

Image rejection and signal selectivity are the primary rea-

sons for selecting an intermediate frequency. Most cur-

rently manufactured bandpass ﬁlters of the crystal, or

resonator type, have become standardized so that the de-

signer can obtain off-the-shelf components at reasonable

cost for these standard frequencies. The standard AM

broadcast receiver utilizes a 455-MHz IF ﬁlter because

extensive experience has shown that this will reject all but

the strongest images. Assume the desired signal is at

600 kHz. A local oscillator operating at 1055kHz will

have an image frequency at 1510 kHz, which the RF in-

put ﬁlter can easily reject. Similarly, an FM receiver op-

erating at 90.1MHz with an intermediate frequency of

10.7 MHz will have an image at 111.5 MHz, which will be

rejected by the RF ampliﬁer. In both of these cases, a sin-

gle intermediate frequency can be used.

A receiver operating at 450 MHz will require two In-

termediate frequencies obtained by using ﬁrst and second

mixers, as in Fig. 16. The ﬁrst IFampliﬁer may consist of a

relatively broadband ﬁlter operating at 10.7 or 21.4 MHz,

followed by a second converter and IF stage operating at

455 kHz. The ﬁrst IF ﬁlter is narrow enough to reject any

455-kHz images, and the second IF ﬁlter is a narrowband

ﬁlter that passes only the desired signal bandwidth. If

the 455-kHz ﬁlter had been used as the ﬁrst IF ﬁlter, the

450-MHz RF ﬁlter, which is relatively broad, would not

have eliminated the image frequency, which is 455kHz

above or below the local oscillator (LO) frequency.

Television receivers use a video intermediate frequency

of approximately 45 MHz, since this permits a relatively

broad RF ﬁlter to pass the broadband TV signal, while still

INTERMEDIATE-FREQUENCY AMPLIFIERS 2175

Previous Page

rejecting the images. The video signal from the IF ampli-

ﬁer is AM, with an FM sound carrier riding on it. Televi-

sion sound is generally obtained from a beat, or difference

frequency between the video and sound carriers, which is

at 4.5MHz.

Satellite receivers use a broadband ﬁrst intermediate

frequency covering a frequency block from 900 MHz to

2.1 GHz. This is done by means of a low-noise block (LNB)

converter. The second mixer is made tunable so that any

frequency in the block can be converted to the second in-

termediate frequency, which is usually ﬁxed at 70 or

140MHz. The second intermediate frequency, which

drives the detector, has a narrower bandwidth to reduce

noise and reject adjacent channel interference.

Crystal, ceramic resonator, and SAW ﬁlters are massed

produced at relatively low cost for the frequencies men-

tioned above, so that most consumer products employ one

or more of the abovementioned standard frequencies and

standard mass-produced ﬁlters.

3. SELECTIVITY

Carson’s rule, and the Nyquist sampling theorem on

which it is based, state that a certain bandwidth is re-

quired to transmit a signal undistorted. The necessary

bandwidth for an AM signal is given as follows:

BW¼2f

m

ð1Þ

Thus an AM broadcast receiver will require 10 kHz of

bandwidth to pass a 5kHz ¼f

m

audio tone. ( f

m

¼Fre-

quency of modulation.) In data transmission systems, the

frequency f

m

corresponding to the data rate f

b

, is f

m

¼

1

2

f

b

.

The data clock frequency is twice the frequency of the data

in ones and zeros. This means that a baud rate f

b

of 9,600

bits per second (bps) will require a bandwidth of 9.6kHz.

For FM, the necessary bandwidth required for trans-

mission is

BW¼2ðf

m

þDf Þ ð2Þ

A 15-kHz audio tone ( ¼f

m

) and an FM transmitter being

deviated with a modulation index of 5 will require 2 (15 þ

(15 Â5)) ¼180kHz of bandwidth. Df is (5 Â15) and f

m

is

15 kHz. Narrowband FM, or phase modulation (PM) (with

a modulation index of o0.7), is somewhat different in that

the bandwidth actually required is the same as that for

AM. This is due to the fact the higher J

n

Bessel products

are missing [Eq. (1) applies].

These values are for double-sideband transmission.

Single-sideband transmission will require half as much

bandwidth. The required baseband bandwidth is the same

as the value for f

m

. This is also known as the Nyquist

bandwidth, or the minimum bandwidth that can carry the

signal undistorted at baseband.

Ideally, the IF ﬁlter, or the equivalent baseband ﬁlter,

need pass only this bandwidth and no more. This requires

the use of an ‘‘ideal’’ bandpass or lowpass ﬁlter, which does

not exist in practice, but can be approached by various

means. The ﬁlter must be as narrow as conditions permit

to reduce the noise bandwidth and any adjacent channel

interference, since noise power rises linearly with increas-

ing ﬁlter bandwidth [14]:

S

o

N

o

¼b

2

bit rate

filter BW

S

i

N

i

ð3aÞ

S

o

N

o

¼modulationÂ

gain

loss

Âprocessing gain

S

i

N

i

ð3bÞ

These two equations show a generalized relationship be-

tween the signal-to-noise ratio (SNR) at the receiver input

and the SNR at the receiver output. The term b

2

repre-

sents a gain, or loss, in power due to the modulation meth-

od. In FM or PM it is the modulation angle. The term [(bit

rate)/(ﬁlter bandwidth)] is generally known as processing

gain. Narrowing the bandwidth improves the S

o

/N

o

ratio,

but this improvement is not always available, depending

on the modulation method. The Nyquist bandwidth rules

state that it should be (symbol rate)/BW¼1.

Pulse modulation, as in radar (radio detection and

ranging), generally requires a much broader ﬁlter band-

width than the other modulation methods. A condition

called envelope delay or group delay must also be ob-

served. This is discussed later along with the transfer

functions of the ﬁlters. For optimum results, the ﬁlter

bandwidth (Df) must be equal to [1/(pulsewidth)]. If the

ﬁlter bandwidth is too narrow, the amplitude detected is

reduced and the SNR is adversely affected. In this case,

the processing gain is ideally ¼1 [14].

S

o

N

o

¼ðprocessing gainÞ

S

i

N

i

¼

E

b

N

o

ð4Þ

4. GAIN

The IF ampliﬁer must provide sufﬁcient gain to raise a

weak signal at the RF input to the level required, or de-

sired, by the decisionmaking circuit or detector. This re-

ceiver gain can vary from 0 up to 130 dB, most of which is

usually provided by the IFampliﬁer. The RFampliﬁer and

mixer circuits preceding the IF ampliﬁer usually provide

Z20 dB of gain so that the IF ampliﬁer generally contrib-

utes little to the receiver noise ﬁgure. (See NOISE FIGURE

article elsewhere in this encyclopedia.) Ampliﬁers with

very high gain have a tendency to oscillate; hence two

different intermediate frequencies may be used to reduce

the gain on any one frequency, or more of the gain may be

obtained from the RF section.

Gain is provided by an amplifying device, such as a

transistor, or vacuum tube (in older equipment). These

devices have input and output impedances of a complex

nature that must be matched to the ﬁltering circuits for

best power transfer, stability, and lowest noise. Current

practice is often to use a ‘‘gain stage,’’ which consists of

multiple amplifying devices in an integrated circuit

2176 INTERMEDIATE-FREQUENCY AMPLIFIERS

package. These packages often contain the mixer stages

and detectors as well.

5. AUTOMATIC GAIN CONTROL

Receivers must respond to a wide range of input levels

while maintaining a nearly constant level at the detector

or decisionmaking circuit. The user or operator does not

wish to manually adjust the gain to obtain a constant

sound or picture level when changing stations. This func-

tion is performed by detecting the output level of the IF

ampliﬁer and correcting it by means of a feedback circuit

that adjusts the gain to keep the level as constant as pos-

sible. Since this detected level can vary rapidly, it is

passed through a lowpass ﬁlter [usually an RC (resis-

tance Âcapacitance) pair] to integrate or slow down the

changes, then ampliﬁed by a DC (direct-current) ampliﬁer

and applied to an IF ampliﬁer circuit or gain stage that

has variable gain characteristics. Some receivers, such as

those used in an automobile, require relatively rapid act-

ing AGC circuits, while ﬁxed receivers can use a much

slower AGC time constant. Dual-gate ﬁeld-effect transis-

tors use the second gate to control the gain. Bipolar or

single-gate ﬁeld-effect transistors vary the gain by means

of a bias voltage or current applied to the input terminal

along with the signal. Special integrated circuit gain stag-

es for IF ampliﬁcation are available, such as the Motorola

MC1350, which both amplify and provide a variable gain

control function.

6. FILTERS FOR IF AMPLIFIERS

Except for block conversions, which convert wide frequen-

cy bandwidths, such as those used on satellite receivers,

IF ampliﬁers in general use a narrow bandpass, or a low-

pass ﬁlter, to limit the bandwidth to the Nyquist band-

width. Block conversion, on the other hand, can use a

highpass–lowpass ﬁlter pair, where the bandwidth to be

passed lies between the high and low cutoff frequencies.

The traditional bandpass ﬁlter requires one or more

resonant elements. Although the actual resonator may be

a coil and capacitor, ceramic resonator, or SAW ﬁlter, the

principles are basically the same. Digital ﬁlters, which do

not use resonators, have been employed more recently.

These will be discussed later in brief. They are discussed

in more detail elsewhere in this encyclopedia.

The inductance/capacitance resonator was the ﬁrst

used, and is still a comparison standard. Figures 1a and

1b show series resonant circuits, and Fig. 1b shows a par-

allel resonant circuit. These circuits will pass a signal at

the resonant peak and reject a signal off resonance. Re-

sistances R

s

and R

p

are naturally occurring losses that

reduce the circuit efﬁciency. Figure 2 shows the universal

resonance curve, which is applicable to both series and

parallel resonant circuits. It is important to note that the

signal rejection never goes to a zero level in the area of

interest, but reaches an asymptotic value between 0.1 and

0.2 or about À17 dB. If it is necessary to reject a signal on

the shoulders of this curve by 60 dB, then four cascaded

stages of this ﬁlter must be used to obtain the necessary

rejection. Note also that there is a nonlinear phase shift

that reaches a maximum in the area of interest, then

changes to 7701. When stages are cascaded, this phase

shift is multiplied by the number of stages. A nonlinear

phase shift can cause distortion in FM receivers. The

phase shift curve plotted is for a parallel resonant circuit.

The phase reverses for a series circuit. The phase at any

point on the curve is obtained by plotting horizontally

from the vertical amplitude/phase scale: a ¼Q (cycles off

resonance/resonant frequency).

A frequency f

0

at which the response of a parallel res-

onant LC ﬁlter is a maximum, that is, the point at which

the parallel impedance is a maximum, is deﬁned as a pole.

A frequency at which the impedance is a minimum, as in

the series LC circuit, is deﬁned as a zero. Thus the as-

sumed four cascaded stages above would constitute a four-

pole ﬁlter, since it contains four resonant poles. The fre-

quency of resonance is given by Eq. (5); this is the fre-

quency at which [X

c

¼1/ ÀjoC] and [X

L

¼joL] are equal:

f

0

¼

1

2pðLCÞ

1=2

ð5Þ

The bandwidth that an analog LC ﬁlter can pass is altered

by the circuit efﬁciency, or circuit Q, given in Eqs. (6).

C

R

p

R

s

R

p

C

C

L L

L

(a) (b) (c)

Figure 1. Series (a,b) and parallel (c) resonant circuits.

1.0

0.6

–3dB

–6dB

Phase lag

Amplitude

Phase lead

2.5 2.0 1.5 1.0 .5 .5 1.0 1.5 2.0 2.5 α

+75°

+50°

45°

60°

0°

25°

50°

75°

0.4

0.3

0.2

0.1

0.8

0.9

Figure 2. Universal resonance curve (BT¼bandwidthÂbit

period).

INTERMEDIATE-FREQUENCY AMPLIFIERS 2177

Generally the bandwidth is speciﬁed as the bandwidth

between the À3dB points, where the phase shift is 7451.

Q¼

X

c

R

s

for a series circuit ð6aÞ

Q¼

R

p

X

c

for aparallel circuit ð6bÞ

Q¼

f

0

3dBBW

ð6cÞ

For simplicity in analyzing the following circuits, the Q

determining R will be assumed to be a parallel resistance

R

p

across the inductance.

Figure 3 shows a typical IF ampliﬁer stage as used in

earlier transistor radios [1,2]. In this circuit R

p

(the total

shunting resistive load) is actually three resistances in

parallel; one is the equivalent R

p

of the coil itself (repre-

senting the coil losses), another is the input resistance of

the following stage, as reﬂected, and the third is the out-

put resistance of the driving transistor as reﬂected. It

cannot be assumed that the resulting coil Q, and hence the

selectivity of the circuit, is that of the unloaded coil and

capacitor alone. Dual-gate ﬁeld effect transistors have the

highest shunting resistance values, bipolar transistors the

lowest. The gain can be varied by increasing or decreasing

the bias voltage V

b

applied to the input terminal.

Manufacturers of amplifying devices often provide the

impedances, or admittances of their products on their data

sheets. Formerly this was done in the form of h parame-

ters. The more common practice today is to provide the

information in the form of S parameters. These values can

be converted to impedances and admittances, but the

manual process is rather complicated. An easier method

is to use the various software programs (see Available

Software section at end of this article) to make the con-

version. Matrix algebra, h and S parameters are discussed

elsewhere in this encyclopedia and also in the Refs. 3 and

4 in this article. Unfortunately, S parameters for bandpass

ﬁlters are rarely available.

Figure 4a shows the equivalent circuit of the transistor

as the tuned LC ‘‘sees’’ it. The transistor ampliﬁes a cur-

rent, which is passed through a relatively low driving re-

sistance R

s

, to the outside. At the same time, the attached

LC sees an equivalent shunting resistance R

c

and capac-

itance C

c

, which must be added in parallel to R

p

, L, and

C. The input to the following stage, assumed to be an

identical transistor, will have a relatively low shunting

resistance R

i

, and capacitance C

i

, which must be added.

Unless the added capacitances are large compared to the

resonant C, they merely add to it without greatly detuning

the circuit. When tuned, the total C plus L will determine

the frequency and the resulting total R

0

p

will determine

the Q of the LC circuit, and hence the bandwidth. Thus

the complex components can be tuned out and the remain-

ing design problem consists of matching the real or resis-

tive part of the input and output impedances to the best

advantage.

The desired end result is to couple the output of the

driving stage to the input of the following stage with the

least loss by matching the differing impedances. An addi-

tional desired result is to narrow the band of frequencies

passed by means of a ﬁlter. These objectives are accom-

plished by transforming the input and output impedances

to a higher or lower shunting impedance that maintains

the desired bandpass characteristic of the ﬁlter. A low

driving or load impedance can be stepped up to become a

very high impedance, which maintains the circuit Q at the

desired value.

Impedance matching enables the designer to change

the actual impedance to a different apparent value, which

V

cc

V

b

Input

Output

Figure 3. Typical IF ampliﬁer stage.

R

c

Rp

Xc

XC′

i

XC′

c

XL

Rc′ Ri′

R

s

R

i

C

c

C

12

C

i

(a)

(b)

Figure 4. Equivalent circuit of transistor as seen by tuned LC.

2178 INTERMEDIATE-FREQUENCY AMPLIFIERS

is optimum for the circuit. Figure 5 shows how impedanc-

es are matched by transformer action. A transformer with

a 3 : 1 turns ratio is shown as an example. The output im-

pedance relative to the input impedance is given by Eq.

(7), where N

i

and N

o

are the input and output numbers of

turns on the winding.

Z

i

Z

o

¼

ﬃﬃﬃﬃﬃﬃ

N

i

N

o

¸

ð7Þ

Thus 90 O at the input is seen as 10 O at the output with a

3 : 1 stepdown turns ratio. The automatic transformer

(tapped coil in Fig. 5) has the same relationship.

When all the reactances and resistances from the tuned

circuit and the transistor input and output as modiﬁed by

the stepup/stepdown process of the impedance-matching

networks are added, the network in Fig. 4b results. Cal-

culation of the resonant frequency and circuit Q from

these reactances and resistances in parallel is complicated

unless they are converted to admittances. Software is

available at reasonable cost to perform these calculations

(see Available Software section at end of this article).

Stock, or mass-produced IF transformers, which are

used to provide bandpass ﬁltering as well as impedance

matching, seldom have the desired turns ratio to match

the impedances properly. An additional Z-matched circuit

using capacitors enables the available transformers to

match almost any impedance while preserving the circuit

Q. This capacitor divider circuit is often used instead of a

tapped coil or transformer as shown in Fig. 6.

The formulas used to calculate the matching conditions

using capacitors are more complex than those used for

transformer coupling, since there are more variables. In

this circuit R

i

is assumed to be lower than R

p

. Although R

p

is the equivalent parallel resistance of the LC circuit in

Fig. 6, it could also be the reduced resistance or reﬂected

R

p2

at a transformer tap. N in these equations is equal to

the loaded resonator Q, or to a lower arbitrary value if

total shunting R

p

is lowered by transformer action as in

Fig. 6, or if the component ratios become unwieldy [12]:

X

C2

¼

R

i

R

i

ðN

2

þ1Þ

R

p

À1

_ _

1=2

ð8Þ

X

C1

¼

R

p

N

N

2

þ1

1 À

R

i

NX

C2

_ _

ð9Þ

X

C2

%

ﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃ

R

i

R

p

Q

¸

ð10Þ

X

C1

%

R

p

Q

¼X

L

ð11Þ

Equations (8) and (9) calculate the reactances of the two

capacitors. Note that NX

L

is the same as QX

L

. Starting

with a value of N¼Q, ﬁnd X

C1

; then X

C2

:

If N is large in Eq. (8), the equations reduce to the

approximate values in Eqs. (10) and (11). Unless Q is

less than 10, these approximate equations are accurate

enough for general use. As an example, let R

i

¼100 O and

R

p

¼10,000O with Q¼100. Then, using Eq. (11), X

C2

be-

comes 10 O and X

C1

becomes 100O. C

2

is approximately 10

times larger than C

1

. Note the similarity of this ratio to

Eq. (7). If a transformer is involved, N becomes much

smaller and the full formulas (8) and (9) should be used.

Equations (8)–(10) apply for R

i

oR

p

and N4(R

p

/

R

i

À1)

1/2

.

7. DOUBLE-TUNED CIRCUITS

When two identical LC circuits are coupled together as

shown in Fig. 7, a number of responses are possible as

shown in Fig. 8. The amplitude response depends on the

coupling coefﬁcient K. Undercoupling results in a two-pole

ﬁlter with the sharpest selectivity. Critical coupling re-

sults in the narrowest bandwidth with the highest gain.

Transitional coupling is slightly greater than critical cou-

pling and results in a ﬂat-topped response with a wider

90 ohms

10 ohms 10 ohms

Figure 5. Impedance matching by transformer action.

R

p

R

i

C

2

C

1

Figure 6. Lowering of total shunting by transformer action.

C

c

C

c

M

C

1

C

1

C

1

(a) (b) (c)

Figure 7. Coupling of identical LC circuits.

INTERMEDIATE-FREQUENCY AMPLIFIERS 2179

bandwidth. Overcoupling results in a double-humped re-

sponse with sharper skirts and broad bandwidth. The cou-

pling coefﬁcient can be calculated using Eqs. (12).

Equation (12a) applies to mutual inductive coupling and

(12b)–(12d), to capacitive coupling.

K ¼

M

ﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃ

L

1

L

2

p ð12aÞ

K

c

¼

1

ﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃ

Q

1

Q

2

_ ð12bÞ

K ¼

ﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃ

C

c

C

c

þC

1

¸

ð12cÞ

K ¼

ﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃ

C

1

C

c

þC

1

¸

ð12dÞ

Equation (12a) calculates the coupling coefﬁcient for two

identical LC tuned circuits that are coupled together by

leakage inductance (Fig. 7a), often obtained by using

shielded coils with holes in the sides of the shield cans to

allow the magnetic ﬁelds to interact. The size of the hole

determines the value of the mutual inductance M. Since

this is difﬁcult to control, a coupling capacitor is often used

as shown in Figs. 7b and 7c. The critical coupling value is

given by Eq. (12b). The coupling coefﬁcients for Figs. 7b

and 7c are given in Eqs. (12c) and (12d).

The amplitude response curves in Fig. 8 do not yield

any information as to the phase shifts that take place

through the ﬁlter. In AM circuits, phase is generally of

little concern, with most attention paid to the amplitude

ripple and linearity. In FM circuits, nonlinear phase shift

or a related term, differential group delay, becomes more of

a problem and efforts are made to keep the phase shift as

linear as possible. In data transmission circuits using

phase modulation, or amplitude modulation, any nonlin-

earity must be avoided. For these reasons, the coupling

coefﬁcients are carefully adjusted and cascaded IF ampli-

ﬁer stages are used to get the desired transfer function for

the IF ampliﬁer.

8. CASCADING IF AMPLIFIER STAGES AND FILTERS

All ﬁltering actions that take place between the RF input

of the receiver and the decisionmaking circuit are parts of

the IF ampliﬁer bandpass ﬁlter. Since the ﬁnal decision-

making circuit is at baseband, or 0 Hz, all ﬁltering prior to

the decisionmaking circuit is part of the IF bandpass ﬁl-

tering, which should be treated as a whole.

A single LC circuit seldom has the desired bandpass

characteristic for an IF ampliﬁer. Cascading IF ampliﬁer

stages with differing coupling and Q values enables the

designer to obtain the desired transfer response. One com-

bination of LC ﬁlters uses an overcoupled double-tuned

stage followed by a single-tuned stage with a lower Q. The

result is a three-pole ﬁlter with relatively steep skirt

slopes. Cascading these stages results in ﬁlters with re-

sponses resembling Butterworth, Chebyshev, elliptical, or

equal-ripple ﬁlters, which are noted for their rejection of

adjacent channel interference (see Figs. 9 and 10).

When additional ﬁltering is required at baseband,

simple RC ﬁlters, lowpass LC ﬁlters, or digital ﬁnite

impulse response (FIR) ﬁlters are used. These and other

ﬁlters are discussed in greater detail elsewhere in this

encyclopedia.

9. CRYSTAL AND CERAMIC FILTERS

Figure 10a shows the equivalent circuit of a crystal or a

ceramic resonator. These devices have both a pole and a

(a) (b)

Figure 9. Curves resulting from cascad-

ing IF ampliﬁer stages.

A

B

C

D

Figure 8. Results of LC circuit coupling: critical (curve A); tran-

sitional (curve B); overcoupled (curve C); undercoupled (curve D).

2180 INTERMEDIATE-FREQUENCY AMPLIFIERS

zero that are located relatively close to each other in fre-

quency. Quartz crystals have Q values ranging from 2000

to 10,000 depending on the mechanical loading of the

crystal. Ceramic resonators usually have Q values be-

tween 100 and 2000. The higher the Q, the narrower the

ﬁlter bandpass. When two of these devices are connected

as shown in Fig. 10b, the result is a bandpass ﬁlter with

steep skirts as shown in Figure 11. These resonators are

used in pairs to create a two-pole ﬁlter, which can then be

combined in a single container with other pairs, to create a

ﬁlter with as many as eight or more poles. They usually

have excellent adjacent-channel rejection characteristics.

When using these devices, care must be taken to care-

fully match the speciﬁed impedance. Any impedance mis-

match can seriously alter the response curve of the ﬁlter.

The impedance-matching techniques discussed previously

will enable the designer to obtain a very close match,

which will optimize the circuit performance. Typical input

and output impedances range from 50 to 4000O. Crystal

ﬁlter manufacturers often build in transformer or other

tuned matching circuits so that the user does not need to

provide a matching circuit outside the crystal ﬁlter.

SAW (surface acoustic wave) ﬁlters utilize a crystal os-

cillating in a longitudinal mode with many ﬁngers or taps

placed along the surface. They can be made with very

broad bandpass characteristics, which makes them well

suited for TV IF ampliﬁers, spread-spectrum IF ﬁlters,

and other uses requiring a wide RF bandwidth. They have

losses, which are typically about 8–20 dB, so they must

have ampliﬁers with adequate gain ahead of them if the

receiver noise ﬁgure is not to be degraded. They are not

suitable for use in ultranarrowband or low-frequency ap-

plications. The group delay quoted in the speciﬁcations is

usually the differential group delay and not the actual

group delay, which is much higher.

10. BASEBAND IF FILTERING

IF bandpass ﬁlters with specific response characteristics

are sometimes very difﬁcult to obtain, whereas the desired

characteristic is easily and inexpensively obtainable at

baseband. This concept is often applied to transmitters

where a sharp-baseband-cutoff ﬁlter can be obtained using

simple components, such as the switched ﬁlter. An 8-pole

equivalent at baseband becomes a 16-pole ﬁlter at the

modulation intermediate frequency. For example, a sharp-

cutoff ﬁlter for voice with a 4-kHz audio cutoff results in a

bandpass ﬁlter 8kHz wide at RF after modulation, with

the same sharp cutoff. The same cutoff characteristics at

RF would be almost impossible to obtain in a crystal ﬁlter,

which would also be very costly and beyond the manufac-

turing budget for a low-cost transmitter such as a cordless

telephone. By using baseband ﬁltering, a poor-quality RF

ﬁlter that only rejects the opposite image can be used.

Similarly, a wideband, or poor-quality IF ﬁlter, can be used

ahead of a detector, if the undesired signal components

can be ﬁltered off after detection at baseband, by using a

sharp-cutoff ﬁlter.

Switched-capacitor ﬁlters are available as packaged in-

tegrated circuits that can be used at baseband and some

lower intermediate frequencies. They have internal oper-

ational ampliﬁers with a switched feedback capacitor, the

combinations of which determine the ﬁlter characteristics.

Since they are dependent on the speed of the operational

ampliﬁers and the values of the feedback capacitors, they

seldom function much above 100kHz. They can be conﬁg-

ured as Bessel, equal-ripple, and Butterworth ﬁlters. Typ-

ical of this type of ﬁlter are the LTC1060 family

manufactured by Linear Technology Corporation (a) and

the MAX274 from Maxim (b) [see items (a) and (b) in

Available Software list at end of this article].

1

As Bessel

ﬁlters they perform well out to about 0.7 times the cutoff

bandwidth, after which the phase changes rapidly and the

Bessel characteristic is lost.

Digital signal processing (DSP) at baseband is widely

used to reduce the component count and size of the base-

band ﬁlters in very small radio receivers, such as cordless

and cellular telephones. Almost any desired ﬁlter response

can be obtained from DSP and FIR ﬁlters without using

inductors and capacitors, which would require factory

tuning (c,d).

Separate FIR ﬁlters have a ﬂat group delay response

and are the best choice for FM or PM ﬁltering, or ﬁlters

at baseband. Commercially available software design

Figure 11. Steep-skirted bandpass ﬁlter.

(a)

(b)

Series Parallel

Figure 10. Equivalent circuit of a crystal or ceramic reasonator.

1

In the remainder of this article, all lowercase letters in paren-

theses refer to entries in the Available Software list following the

Bibliography. Numbers in brackets refer to Bibliography entries

(references) as usual.

INTERMEDIATE-FREQUENCY AMPLIFIERS 2181

packages permit the design of trial circuits to investigate

phase shift and group delay (e,f).

Unfortunately, digital ﬁltering of any type is frequency-

limited. The ﬁlter must use a sampling rate that is much

higher than the frequency to be passed. To use a digital

ﬁlter, such as a FIR ﬁlter, or DSP as a bandpass ﬁlter at

10.7 MHz, requires an analog-to-digital converter (ADC)

operating at 160 MHz or higher. Filtering at baseband

means the sampling rate can be much lower.

11. AMPLIFYING DEVICES FOR IF AMPLIFIERS

Transistors in one form or another have become the stan-

dard for IF ampliﬁers. The single bipolar or ﬁeld-effect

transistor used as an individual component, was formerly

the preferred device. For very-high-Q circuits, the dual-

gate FET performs best, since it is the most stable and

offers the lowest shunt resistance. Single-gate FET devic-

es often have too much drain to gate capacitance for good

stability. Modern bipolar transistors usually have good

stability, but lower shunt resistances than dual-gate

FETs. Stability is discussed later in this section.

Monolithic ampliﬁers [MMIC (monolithic microwave

integrated circuit) devices] are stable and have good

gain, but the shunt load impedance is too low for most

bandpass ﬁlters other than a crystal or SAW ﬁlter

matched to 50 O.

The most recent practice for IF ampliﬁers is to use in-

tegrated circuit blocks containing more than one transis-

tor in a gain stage. These are then packaged together in an

integrated circuit with other circuit components to form

an almost complete radio. Integrated circuits of this type

are shown below.

12. TYPICAL CONSUMER IF AMPLIFIERS

Consumer radio and TV equipment is mass-produced for

the lowest possible cost consistent with reasonable quality.

Manufacturers of integrated circuits now produce single-

chip IF ampliﬁers that can be combined with mass-pro-

duced stock ﬁlters to produce a uniform product with a

minimum of adjustment and tuning on the assembly line.

In the examples that follow, some circuit components in-

side and outside the IC have been omitted to emphasize

the IF ampliﬁer sections.

Figure 12 shows a single-chip AM receiver that uses

the Philips TDA1072 [7] integrated circuit and ceramic IF

ﬁlters at 455kHz. The input impedance of the ceramic ﬁl-

ter is too low to match the output impedance of the mixer,

so a tuned matching transformer is used to both reduce

the passed bandwidth (preﬁlter) and match the imped-

ances. The input impedance of the IF ampliﬁer was de-

signed to match the average impedance of the ceramic

ﬁlters available. This integrated circuit has a built in au-

tomatic gain control that keeps the received audio output

level relatively constant at 250mV as long as the input

signal level to the chip exceeds 30 mV.

Figure 13 shows a single-chip FM radio based on the

Phillips NE605 integrated circuit (g) that uses ceramic IF

ﬁlters at 10.7 MHz. The input and output impedance of the

IF ampliﬁer sections is approximately 1500 O, to match

the ceramic ﬁlter impedance, so no matching transformer

is required. The audio output is maintained level at

175 mV for all signal levels at the input level from À110

to 0 dBm. An automatic frequency control (AFC) voltage

can be obtained from the quadrature detector output. This

circuit can also be used for narrow-angle phase modula-

tion if a crystal discriminator is used for a phase reference

at the quadrature input.

AGC is available from all FM integrated circuits so that

the gain of the mixer and RF stages can be controlled at a

level that does not allow these stages to be saturated by a

strong incoming signal. Saturation, or nonlinearity before

ﬁltering, results in undesirable signal spreading. The

NE605 has a ‘‘received-signal strength indicator’’ (RSSI)

output that can be ampliﬁed and inverted if necessary to

provide an AGC voltage, or current, for the RF ampliﬁer

and mixer.

Figure 14 shows a TV IF ampliﬁer using the Motorola

MC44301/2 video IF integrated circuit (h) with a SAW

ﬁlter at 45 MHz. The SAW ﬁlter bandpass is made

V

cc

V

cc

z Match

Ceramic filter

RF Input

Oscillator LC Ac output

AF det.

Mixer

AGC det./amp

IF amp

TDA 1072

AGC

Figure 12. Layout of a single-chip AM receiver.

Audio out RSSI/AGC Oscillator

LC

RF

input

IF LIM

Filter Filter

NE605

Mixer

RSSI level det.

Quad

det.

Figure 13. Conﬁguration of a single-chip FM radio.

2182 INTERMEDIATE-FREQUENCY AMPLIFIERS

approximately 6 MHz wide to pass the video and sound.

The circuit has both automatic frequency control (AFC)

and automatic gain control (AGC) features built in. Unlike

the IF ampliﬁers used for AM and FM audio broadcast

applications, the TV IF ampliﬁer includes a phase-locked

loop (PLL) and synchronous detector that locks the fre-

quency of an internal oscillator to the intermediate fre-

quency. This locked, or synchronous, oscillator output is

then mixed with the information-bearing portion of the

signal to create a baseband signal.

The system shown in Fig. 14 is one of a family of 0-Hz

IF ampliﬁers, are becoming more popular in wireless de-

signs, since they permit most or additional signal process-

ing at baseband. In Fig. 14, the video and sound carriers

are both passed by the SAW ﬁlter. They beat together at

4.5 MHz in the detector, providing a second IF stage with

the sound information. This 4.5-MHz IF information is

then ﬁltered by a ceramic bandpass ﬁlter approximately

50 kHz wide to remove any video components, limited and

detected as a standard FM signal to provide the TV sound.

The video portion, consisting of signals from 15 kHz to

approximately 4.25 MHz, is then further processed to

separate the color information at 3.58 MHz from the

black-and-white information. The video output level is de-

tected to provide the AGC voltage.

The phase-locked oscillator, operating at the interme-

diate frequency, can also be used to provide automatic fre-

quency control to the ﬁrst mixer stage local oscillator.

Figure 15 shows a dual-conversion receiver in a single

integrated circuit for communications use, utilizing the

Motorola MC13135 integrated circuit (h). When the

receiver is operated at 450 or 850MHz, as was men-

tioned above, single-conversion IF stages do not offer

the necessary image rejection. This receiver is for narrow-

band FM as opposed to wideband FM for entertain-

ment purposes. The ﬁrst IF ﬁlter is a low-cost ceramic

ﬁlter at 10.7 MHz. The second ﬁlter is a multipole crystal

or ceramic ﬁlter with a bandpass just wide enough to pass

the signal with a small FM deviation ratio. Receivers

of this type can be used with 12.5 and 25 kHz of chan-

nel separation for voice-quality audio. Analog cellular

telephones, aircraft, marine, police, and taxicab radios

are typical examples.

13. DIRECT CONVERSION AND OSCILLATING FILTERS

Direct conversion converts the RF frequency directly to

baseband by using a local oscillator at the RF carrier fre-

quency. The TV IFampliﬁer with the detector circuit given

in Fig. 14 illustrates some of the reasons. Conversion to

baseband can occur at the intermediate frequency or di-

rectly from the RF frequency.

There is a noticeable trend in integrated circuit design

to utilize synchronous detection [5] with the carrier

restored by means of a phase-locked loop, as shown in

Fig. 14, or by means of regenerative IF ampliﬁers [6],

to accomplish several desirable features that cannot

be obtained from the classical circuits with square-law

detectors.

In the case of direct RF-to-baseband conversion, there

is no IF stage in the usual sense, and all ﬁltering occurs at

baseband. For this reason direct-conversion receivers are

referred to as zero-hertz (0-Hz) IF radios. Integrated cir-

cuits for direct RF conversion are available that operate

well above 2.1 GHz at the RF input. The Maxim 2820 (b)

and the AMD1771 (i) are examples. DSP and FIR ﬁlters

are the preferred lowpass ﬁlters at baseband, where they

are referred to as ‘‘windows’’.

It was discovered in the 1940s that the performance of

a TV receiver could be improved by using a reconstructed

synchronous or exalted carrier, as occurs in the TV IF

ampliﬁer depicted in Fig. 14. The carrier is reduced by

vestigial sideband ﬁltering at the transmitter and con-

tains undesirable AM and PM signal components. By

causing an oscillator to be locked to, or to be synchronized

with the carrier, and then to be used by the detector, a

significant improvement in the received signal can be

achieved. Prior to using circuits of this type, the intercar-

rier sound at 4.5MHz in earlier TV sets had a character-

istic 60 Hz buzz due to the AM and PM on the carrier. By

substituting the recovered synchronous carrier instead,

this buzz was removed. Figure 14 illustrates an example.

The earliest direct-conversion receivers using locked

oscillators or synchronous detectors were built in the

Saw FL

Video det.

Video out

Audio out

Phase shifter

Phase det.

Sound IF

4.5 MHz

cer. fil

4.5 MHz det. VCO

AGC

amp

IF

amp

Limiter

IF input

AFC/AFT

out

I

Vcc Vcc

Quadrature L-C

VCO L-C

Q

Figure 14. Layout of a television IF ampliﬁer.

10.7 MHz

cer. fil

455 kHz

cer. fil

IF

amp

Limiter

1st mix

2nd mix

RF input

RSSI

Quad det.

MC13136

Quadrature

L-C

Vcc

1st LO.

L-C

2nd LO.

L-C

RSSI

out

Audio out

Figure 15. Conﬁguration of a dual-conversion receiver in a sin-

gle IC.

INTERMEDIATE-FREQUENCY AMPLIFIERS 2183

1920s, when they were known as synchrodyne or homo-

dyne receivers. The theory is relatively simple. A signal

from the RF ampliﬁer is coupled to an oscillator, causing a

beat or difference frequency. As the frequencies of the two

sources come closer together, the oscillator is pulled to

match the incoming signal and locks to it. The lock range

depends on the strength of the incoming signal. The two

signals are then mixed to provide a signal at baseband,

which can be further ﬁltered by means of a lowpass ﬁlter.

In this way, a relatively broad RF ﬁlter can be used, while

the resulting AM signal bandwidth after detection and

baseband ﬁltering can be very narrow. The Q of the oscil-

lator tank circuit rises dramatically with oscillation so

that Q values of 6000–10,000 are not unusual and selec-

tivity is greatly improved. AGC can be obtained from the

audio signal to maintain an input signal level that is con-

stant to ensure a good lock range. An undesirable charac-

teristic is the whistle or squeal that occurs between

stations. Later receivers used a squelch circuit to make

the signal audible only after locking has occurred. High-

quality receivers for entertainment and communications

use were produced in the 1990s using this principle. They

offer higher sensitivity, better ﬁdelity, and more controlled

response. Integrated circuits for receivers of this type (di-

rect conversion) are now being produced for paging, wi-ﬁ

(wireless ﬁdelity), direct-broadcast TV, and cellular and

cordless telephones. The Maxim 2820 (b) and the AMD

1771 (i) are examples.

Oscillating ﬁlters and phase-locked loops are similar in

principle. An intermediate frequency is applied to a phase/

frequency detector that compares the intermediate fre-

quency with the oscillator frequency. An error voltage is

created that changes the oscillator frequency to match, or

become coherent with, that of the incoming IF carrier fre-

quency. In some cases the phase-locked loop signal is 901

out of phase with the carrier, so a phase shifter is used to

restore the phase and make the signal from the oscillator

coherent in phase with the incoming signal. (See Figs. 14

and 18, where phase-locked loops and phase shifters are

employed.)

Synchronous oscillators and phase-locked loops not

only extend the lower signal-to-noise ratio but also have

a bandwidth ﬁltering effect. The noise bandwidth of the

PLL ﬁlter is the loop bandwidth, while the actual signal

ﬁlter bandwidth is the lock range of the PLL, which is

much greater. Figure 16 shows the amplitude and linear

phase response of a synchronous oscillator. The PLL is not

always the optimum circuit for this use because its fre-

quency/phase-tracking response is that of the loop ﬁlter.

The locked oscillator [6] performs much better than the

PLL since it has a loop bandwidth equal to the lock range

without sacriﬁcing noise bandwidth, although with some

phase distortion. Some authors hold that the synchronous

oscillator and locked oscillator are variations of the PLL in

which the phase detection occurs in the nonlinear region

of the oscillating device and the voltage-controlled oscil-

lator (VCO) frequency change characteristic comes from

the biasing of the oscillator.

Both the PLL and the locked oscillator can introduce

phase distortion in the detected signal if the feedback

loop is nonlinear. A later circuit shown in Fig. 17 has two

feedback loops and is considered to be nearly free of phase

distortion [5]. This circuit has the amplitude/phase res-

ponse given in Fig. 16.

Phase-locked loops have been used for many years for

FM ﬁltering, ampliﬁcation, and detection. They are in

common use with satellite communications links for audio

and video reception. A 74HC4046 phase-locked loop inte-

grated circuit operating at 10.7 MHz (the FM intermediate

frequency) can be used to make an FM receiver for broad-

cast use [7]. The phase-locked loop extends the lower sig-

nal-to-noise limit of the FM receiver by several decibels

while simultaneously limiting bandwidth selectivity to the

lock range of the PLL. The detected audio signal is taken

from the loop ﬁlter.

14. AM STEREO (C-QUAM)

AM stereo radio is another application of the phase-locked

oscillator at the intermediate frequency. AM stereo radio

is dependent on two programs being transmitted at the

same time at the same frequency. They arrive at the re-

ceiver detector circuitry through a common IF ampliﬁer

V

cc

L-C tank

Input

Level adjust

Output

Figure 17. Circuit with same amplitude and phase response as

in Fig. 16 but with two feedback loops and markedly decreased

phase distortion.

Amplitude

Noise BW

Filter signal

bandwidth

Phase

Figure 16. Amplitude and linear phase response of a synchro-

nous oscillator.

2184 INTERMEDIATE-FREQUENCY AMPLIFIERS

operating at 455kHz. The normal program heard by all

listeners is the LþR program. The stereo information

(LÀR) is transmitted at the same frequency, but in quad-

rature phase to the LþR program. Quadrature, or or-

thogonal transmission, is used because the orthogonal

channels do not interfere with one another.

Each program section requires a reference carrier,

that is coherent with its own sideband data. The LþR

program, which has a carrier, may use an ordinary

square-law detector or a synchronous detector. This is

the program heard over monaural radios. To obtain the

LÀR program that is transmitted without a carrier, a

phase-locked loop is used at the intermediate frequency

to lock a voltage controlled oscillator to the carrier of the

LþR program. This carrier is then shifted 901 in phase

and becomes the reference carrier for the LþR segment.

The output of the PLL has the proper phase for the LÀR

detector, so no phase shifting is necessary. The LÀR

detector is a coherent or synchronous detector that

ignores the orthogonal LþR information. By adding,

then inverting and adding, the left and right channels

are separated. Figure 18 shows a simpliﬁed block diagram

of the C-QUAM receiver.

The Motorola MC1032X (h) series of integrated circuits

are designed for AM stereo use. The MC10322 and

MC10325 have most of the components required, includ-

ing the IF ampliﬁers, for a complete AM stereo receiver in

two integrated circuit packages.

15. SUBCARRIERS

Subcarriers are used to carry two or more signals on the

same carrier. They differ from the orthogonal signals used

with C-QUAM in that they are carried as separate signals

superimposed over the main carrier information, as in the

video sound carrier shown in Fig. 14. In Fig. 14, a fre-

quency-modulated subcarrier at 4.5MHz is carried on top

of the main video signal information, which extends from

0 to 4.25 MHz. This is an example of an AM/FM subcar-

rier. Nondigital satellites utilize a frequency-modulated

video carrier with as many as 12 subcarriers at frequen-

cies ranging from 4.5 to 8.0 MHz. Normal FM stereo

broadcasting utilizes a FM/AM subcarrier at 38 kHz to

carry the LÀR portion of the stereo program. FM stations

frequently carry additional subcarriers at 67 and 92 kHz.

These FM/FM subcarriers are used to carry background

music, ethnic audio programs, and digital data.

To detect a subcarrier, the signal is ﬁrst reduced to

baseband, then a bandpass ﬁlter is used that separates

only the subcarrier frequencies. The subcarrier frequen-

cies are then passed to a second detector, which must be of

the type appropriate for the subcarrier modulation. This

can be seen in Fig. 14, where a 4.5-MHz ﬁlter is used. This

is followed by a limiter and quadrature detector, which is

appropriate for the FM signal. In the case of a 67-kHz FM/

FM subcarrier, the ﬁlter is 15 kHz wide at 67 kHz. Detec-

tion can be accomplished by a discriminator, quadrature

detector, or PLL.

16. CELLULAR AND CORDLESS TELEPHONES

Analog cellular telephones employ the circuits shown

in Figs. 13 and 15. Digital telephones utilizing Gaussian

minimum shift keying (GMSK) also use these circuits.

Digital telephones using quadrature amplitude modula-

tion (QAM) or phase shift keying (PSK) employ circuits

similar to that used for C-QUAM with digital ﬁltering

and signal processing instead of audio ﬁltering at base-

band. The PLL used for digital receivers is a more

complex circuit known as the Costas loop, which is neces-

sary to restore a coherent carrier for digital data recovery.

Some cellular phones are dual-mode; that is, they can

transmit and receive analog voice or digital GMSK

modulation using circuits similar to those shown in

Figs. 13 and 15.

17. NEUTRALIZATION, FEEDBACK, AND AMPLIFIER

STABILITY

Earlier transistors and triode vacuum tubes had consid-

erable capacitance between the output element (collector

or plate) and the input side of the device (see Fig. 4). Feed-

back due to this capacitance is multiplied by the gain of

the stage so that enough signal from the output was often

coupled back to the input to cause the stage to oscillate

unintentionally, as opposed to the planned oscillation of

the locked oscillator, synchronous oscillator, or PLL. To

prevent this, feedback of an opposite phase was deliber-

ately introduced to cancel the undesired feed back. A neu-

tralized IF ampliﬁer is shown in Fig. 19. Transistors and

integrated circuits made since 1985 are rarely unstable

and generally do not require neutralization unless seri-

ously mismatched. A better solution than neutralization is

usually to improve the matching of the components and

the circuit layout.

By carefully controlling the feedback, a regenerative IF

ampliﬁer can be constructed that operates on the verge

of oscillation. This greatly increases the Q of the tuned

circuit, thus narrowing the IF bandwidth. Circuits of

this type were once used in communication receivers for

L + R out

L – R out

Phase shifter

Phase det.

455 kHz

cer. fil

I det.

Q det. VCO

AGC

amp

IF

amp

Limiter

IF input

AFC

out

I

Vcc VCO L-C

Q

AGC

Carrier

MC10322

Figure 18. Simpliﬁed block diagram of the C-QUAM receiver.

INTERMEDIATE-FREQUENCY AMPLIFIERS 2185

commercial and amateur use, where they were referred to

as ‘‘Q multipliers.’’

The maximum stable gain (MSG) that can be achieved

from a potentially unstable ampliﬁer stage without neu-

tralization is obtainable from the S parameters and can be

calculated from Eq. (13). This equation assumes that the

input and output impedances are matched and there is

little or no scattering reﬂection at either the input or out-

put. The stability factor K, usually given with the S pa-

rameters, must be 41. A failure to match the impedances

can result in an unstable ampliﬁer, but does not necessar-

ily do so. A higher gain can be obtained, but at the risk of

instability.

K ¼MSG¼

S

21

S

12

ð13Þ

In addition to impedance mismatch, the most frequent

cause of ampliﬁer instability, or oscillation, is poor circuit-

board layout or inadequate grounding and shielding, not

the device parameters. The wiring, whether printed or

handwired, forms inductive or capacitive coupling loops

between the input and output terminals of the amplifying

device. This is particularly noticeable when high-gain ICs

such as the NE605 are used. These integrated circuits

have IF gains of 4100dB and require very careful board

layouts for best results. Undesirable feedback can greatly

decrease the usable gain of the circuit.

18. SOFTWARE RADIO

Digital radios, or radios based on digital signal processing

(DSP), offer some technical advantages over their analog

predecessors. Digital radios can be used not only for dig-

ital modulation but also for AM and FM. One receiver can

simultaneously detect both digital and analog modulation;

thus they can be used for cellular telephones in environ-

ments where multiple modulation standards are used. As

a class, they belong to the 0-Hz intermediate-frequency

group.

The typical receiver consists of a conventional RF front

end and a mixer stage that converts the signal to a lower

frequency, as in the dual conversion radios discussed

above (Fig. 16). The signal at this stage is broadband in

nature, but not broadband enough to include the image

frequencies. The signal is then fed to an analog-to-digital

converter (ADC), which is sampled at several times f

m

.

This converts the portion of interest of the signal to base-

band (or 0 Hz) instead of a higher intermediate frequency.

The actual ﬁltering to remove unwanted interfering sig-

nals then takes place at baseband, using digital ﬁltering.

Digital signal processing and decimation are covered else-

where in this work. The ADC (c) performs the same func-

tions as do the oscillating detectors shown above.

Noise ﬁgure, ampliﬁcation, and AGC considerations of

the ﬁrst IF ampliﬁer are the same as those for a conven-

tional receiver. The ADC and the DSP ﬁlters function best

with a constant signal input level.

The term ‘‘software radio’’ has been adopted because

the tuning function is done in software by changing the

sampling frequency at the ADC. The sampling frequency

is obtained from a digitally controlled frequency synthe-

sizer instead of tuned LC circuits.

19. SPREAD-SPECTRUM RADIOS

The spread-spectrum receiver also uses a conventional

front end with a wideband ﬁrst IF stage. The same con-

ditions apply as to software radios and dual-conversion

receivers. The ﬁrst IF stage must have the necessary

bandwidth to accommodate the spread bandwidth, ampli-

fy it with minimum added noise, and match the output to

the despreading circuitry. Spread-spectrum technology is

covered elsewhere in this encyclopedia. While usually as-

sociated with digital reception, spread-spectrum technol-

ogy can also be used for analog audio.

20. ORTHOGONAL FREQUENCY-DIVISION

MULTIPLEXING (OFDM) AND CODED OFDM (COFDM)

These modulation methods could be considered similar to

spread-spectrum techniques, or to methods requiring dual

conversion, in that they use a very broad spectrum as a

ﬁrst level, followed by a narrowband ﬁlter to extract an

individual channel. SAW ﬁlters are generally used at RF,

while second-stage processing can use digital ﬁltering, as

in the software radio, or be done at baseband.

21. TRANSFER FUNCTIONS

The amplitude response, plotted relative to frequency of a

ﬁlter, is usually given in terms of the transfer function

H( f ). Some typical transfer functions are as follows. For

the LC ﬁlter of Fig. 2

H

LC

ðf Þ ¼ expÀ

Qot

2

ð14Þ

The LaPlace transform equivalent is

HðsÞ ¼

K

s

2

þBs þo

2

0

ð15Þ

V

cc

Figure 19. Conﬁguration of a neutralized IF ampliﬁer.

2186 INTERMEDIATE-FREQUENCY AMPLIFIERS

A similar curve obtainable with digital ﬁlters is the Gauss-

ian ﬁlter:

H

Gauss

ðf Þ ¼ expÀ1:38

1

ðBTÞ

2

_ _

ð16Þ

A generalized Nyquist IF ﬁlter bandpass spectrum is seen

in Fig. 20.

In Fig. 20 the centerline represents either the carrier

frequency, or 0 Hz. The portion of the spectrum to the right

of the centerline is the baseband response, while both ex-

tremes represent the RF double-sideband response with

the carrier at the center. The B region is the baseband re-

sponse of an ‘‘ideal’’ ﬁlter, which does not exist in practice.

Practical ﬁlters have a rolloff, or excess bandwidth, shown

in a. Outside the desired bandpass, there is a ‘‘comeback’’

in region C. The ‘‘ideal’’ ﬁlter has no rolloff and no come-

back. The region B is the required Nyquist bandwidth.

Multilevel digital modulation methods such as quad-

rature amplitude modulation (QAM) and multiple phase

shift keying (MPSK) require ﬁlters that are free of ampli-

tude and phase distortion within the Nyquist bandwidth,

then having a rolloff a as abrupt as reasonably possible

outside that distortion-free bandwidth. The optimum ﬁlter

for this is considered to be the raised-cosine ﬁlter, so called

because the region after the uniform response is half-cycle

of a cosine wave squared (cosine raised to second power).

The transfer function for the raised-cosine ﬁlter is as fol-

lows. In the central bandpass region B, we obtain

Hð f Þ ¼1 for jf j > ¼ Àf

m

ð1 ÀaÞ;

or o¼f

m

ð1þaÞ ðor Æf

m

Þ

ð17Þ

When a ¼0, the ﬁlter is said to be the ‘‘ideal’’ ﬁlter. In the

transition region a; since cos 2A¼cos

2

AÀ1 or 1 þcos 2A

¼cos

2

A¼0 elsewhere, we obtain the following forms of

Eq. (17):

1. H( f) ¼cos

2

[(p|f|T)/2a) Àp(1 Àa)/4a], for Àf

m

(1Àa)

o¼|f|o¼f

m

(1 þa)

2. H( f) ¼

1

2

{1þcos [(p|f|T)/a) Àp(1Àa)/2a]}

In practice, there is always some comeback as seen in re-

gion C.

Figure 20 shows the double-sided RF bandwidth when

the center reference is the carrier. The right-hand side is

the baseband bandwidth with the reference at 0 Hz. When

used as a lowpass ﬁlter at baseband, the ﬁlter is referred

to as a ‘‘window.’’ There are many rolloff curves associated

with windows, which are realizable with DSPs or ﬁeld-

programmable gated arrays (FPGAs) used as FIR ﬁlters.

Designing an RF bandpass ﬁlter with these rolloff curves

is very difﬁcult; therefore, the preferred practice is to do

the ﬁltering at baseband where numerous windowing

curves are available.

Some popular rolloff curves for FIR ﬁlters used as ‘‘win-

dows’’ are the Bartlett, Blackman, Hamming, Hanning,

Elanix, Truncated Sinx/x, and Kaiser. These are usually

realized by changing the multipliers in the 2 of Eq. (17)

[above; after text following Eq. (17)] form of the raised-

cosine equation. For example, using this form of Eq. (17),

the Hamming window has the equation H(f) ¼{0.54 þ

0.46cos[(p|f|T)/a) Àp(1Àa)/2a]}. The ‘‘ideal’’ ﬁlter shape

(a ¼0) at baseband is called a ‘‘rectangular’’ window.

22. GROUP DELAY, ENVELOPE DELAY, AND RISE TIME

The group delay for conventional ﬁlters is traditionally

calculated to be [11]:

T

g

¼

DF

2pDf

ð18Þ

For LC or Gaussian ﬁlters (Fig. 2), this is

T

g

¼

1

ð4Df Þ

and T

g

¼

QDF

o

ð19Þ

Obviously, a very narrow bandwidth ﬁlter [Df] has a very

large group delay, which will adversely affect pulse mod-

ulation.

There is an associated equation for the risetime of the

conventional ﬁlter: T

r

¼0.7/B, where B is the 3-dB band-

width [Df] of the ﬁlter. This is the time interval from 10%

to 90% on the RC curve. Bandwidth, risetime, and sam-

pling rate are mathematically linked.

A radar system with a narrow pulse must have a RC

risetime that allows the pulse to pass. This necessarily

means a very broad ﬁlter bandwidth and an accompany-

ing high noise level.

Two-level modulation methods, such as BPSK, QPSK,

GMSK, NBFM, and NBPM (binary, quadrature, Gaussian

minimum shift keying and narrowband frequency and

phase modulation), can use a narrower-than-usual

bandpass ﬁlter. The bandpass can be as low as 0.2[Df] in

Eqs. (18) and (19).

Refer to Fig. 2 and Eq. (4), which demonstrate a reduc-

tion of the output level of the high-frequency portion of the

signal (sidebands) that pass through the ﬁlter and a si-

multaneously reduction of the noise power bandwidth.

The result is an overall S

o

/N

o

improvement. The notation

BT is used for this concept B¼bandwidth and T¼bit pe-

riod ¼1/f

b

. The value of T is ﬁxed, but B can be altered.

The effect is to raise the processing gain in Eq. (4) by 1/BT.

Certain newer modulation concepts (ultranarrowband)

require a ﬁlter that does not conform to the group delay

B

+f

m

(1+)

+f

m

-f

m

(1−)

−f

m

C C

Reference

Figure 20. A simpliﬁed Nyquist ﬁlter bandpass spectrum (from

Sklar [13] and Feher [14]).

INTERMEDIATE-FREQUENCY AMPLIFIERS 2187

equation [Eq. (18)]. These so called zero-group-delay ﬁlters

have a very narrow bandwidth with almost instantaneous

pulse response at a single frequency. Figure 21 shows the

circuit of a half-lattice, or bridge-type, ﬁlter, that has near-

zero group delay to a single-pulsed frequency. At the par-

allel resonant frequency of the crystal, the crystal appears

to have a very high resistance and the signal passes via

the phasing capacitor in the opposite bridge arm. This

circuit has a frequency response similar to that of the uni-

versal curve (Fig. 2) with shoulders that extend from 0 Hz

to inﬁnity. Therefore, it must be used together with pre-

ﬁlters to narrow the total noise bandwidth. A small ca-

pacitor or inductor can be used at z to extend the tuning

range of the crystal [11].

23. COMPUTER-AIDED DESIGN AND ENGINEERING

Digital ﬁlters are easily designed using commercially

available software packages and information provided by

the IC manufacturers (d–g, l).

For IF ﬁlter design using discrete components, the ad-

mittances rather than the impedances are easiest to use,

since most components are in parallel as shown in the

equivalent circuit of Fig. 4b. Unfortunately, most available

data are in the form of S parameters, which are very dif-

ﬁcult to convert manually to impedances or admittances.

Parameters for the ﬁlters are rarely available, so calcu-

lated values based on assumed input and output imped-

ances must be used unless test equipment capable of

measuring return losses or standing waves is available,

in which case the S parameters can be measured or cal-

culated.

Smith and Linville charts have been used by some au-

thors to design IF ampliﬁers, but these methods are not

totally satisfactory for IF ampliﬁer design, since a high-Q

circuit has its plot near the outer edge of the circle and

changes are difﬁcult to observe. The network admittance

values shown in Fig. 4 would be used.

Computer-aided programs for linear or analog designs,

such as the various ‘‘SPICE’’ programs are readily avail-

able (j). Other programs which concentrate specifically on

ﬁlter design (f, k–m) can simplify the ﬁlter design. They

have outputs that then interface with the SPICE pro-

grams if desired. Most semiconductor manufacturers pro-

vide scattering parameters (S parameters) or SPICE input

data on disk for use with these programs. Some design

software sources are listed below (after the Bibliography).

Some of the IF ampliﬁer integrated circuit manufacturers

also provide software specific to their products.

BIBLIOGRAPHY

(References 8, 9, 10, and 13 contains applicable

software).

1. J. M. Petitt and M. M. McWhorter, Electronic Ampliﬁer Cir-

cuits, McGraw-Hill, New York, 1961.

2. W. Th. Hetterscheid, Transistor Bandpass Ampliﬁers, Philips

Technical Library, N.V. Philips, Netherlands/Philips Semicon-

ductors, 1964.

3. Roy Hejhall, RF Small Signal Design Using Two-Port Param-

eters, Motorola Applications Note AN 215A.

4. F. Davis, Matching Network Designs with Computer Solu-

tions, Motorola Applications Note AN 267.

5. V. Uzunoglu and M. White, Synchronous oscillators and co-

herent phase locked oscillators, IEEE Trans. Circuits Syst.

36(7) (1989).

6. H. R. Walker, Regenerative IFampliﬁers improve noise band-

width, Microwaves RF Mag. (Dec. 1995, Jan. 1996).

7. R. E. Best, Phase Locked Loops, McGraw-Hill, New York,

1984.

8. R. W. Goody, P-Spice for Windows, Prentice-Hall, Englewood

Cliffs, NJ, 2001.

9. M. E. Herniter, MicroSim P-Spice, Prentice-Hall, Englewood

Cliffs, NJ, 2000.

10. J. Keown, Orcad PSpice and Circuit Analysis, Prentice-Hall,

Englewood Cliffs, NJ, 2001.

11. W.-K. Chen, The Circuits and Filters Handbook, IEEE Press,

New York, 1995.

12. ARRL Handbook, Amateur Radio Relay League, Newington,

CT, 2000.

13. B. Sklar, Digital Communications, Prentice-Hall, Englewood

Cliffs, NJ, 2001. (Contains the Elanix SysView design soft-

ware on CD.)

14. K. Feher, Wireless Digital Communications, Prentice-Hall,

Englewood Cliffs, NJ, 1995.

AVAILABLE SOFTWARE

The following companies are representative of those pro-

viding packaged IFampliﬁers as integrated circuits #, and

those offering development software packages *.

(a) #*Linear Technology Corporation, 720 Sycamore Drive, Mil-

pitas, CA 95035 (www.linear-tech.com).

(b) #Maxim Integrated Products, 120 San Gabriel Drive, Sunny-

vale, CA 94086 (www.maxim-ic.com).

(c) #*Analog Devices, One Technology Way, P.O. Box 9106, Nor-

wood, MA 02062 (www.analog.com).

(d) *#Texas Instruments, P.O. Box 954, Santa Clarita CA 91380

(www.ti.com/sc or www.ti.com/sc/expressdsp).

(e) #*Altera Corp., 101 Innovation Drive, San Jose, CA 95134

(www.altera.com).

Z

Figure 21. Circuit of a half-lattice (bridge-type) ﬁlter with near-

zero group delay to a single-pulsed frequency (from Chen [12]).

2188 INTERMEDIATE-FREQUENCY AMPLIFIERS

(f) *#Xilinx, Inc., 2100 Logic Drive, San Jose, CA 95124 (www.

xilinx.com).

(g) *# Philips Semiconductors, 811 E. Arques Avenue, P.O. Box

3409; Sunnyvale, CA 94088 (www.semiconductors.philips.

com).

(h) *#Motorola Literature Distribution Center, P.O. Box 5405;

Denver, CO 80217 (www.motorola.com/semiconductors/ or

www.Design-net.com).

(i) *#AMD, One AMD Place, P.O. Box 3453; Sunnyvale, CA 94088

(www.amd.com).

(j) *MicroSim Corp., 20 Fairbanks, Irvine, CA 92618 (www.

orcad.com).

(k) *Eagleware Corp., 1750 Mountain Glen, Stone Mountain, GA

30087 (www.eagleware.com).

(l) *Elanix Inc., 5655 Lindero Canyon Road, Suite 721, Westlake

Village, CA 91362 (www.elanix.com).

(m) *The Math Works (MatLab), 3 Apple Hill Drive, Natick, MA

01760-2098 (www.mathworks.com).

(n) *Intusoft, P.O. Box 710, San Pedro, CA 90733 (www.intusoft.

com).

(o) *#Hewlett-Packard Company, P.O. Box 58199, Santa Clara, CA

95052 (www.hp.com).

(p) *Z Domain Technologies, 555 Sun Valley Drive, Roswell, GA

30076 (www.zdt.com/Bdsp).

(q) #Rockwell Semiconductor Systems, 4311 Jamboree Road,

Newport Beach, CA 92660 (www.rockwell.com).

(r) #RF Micro Devices, 7628 Thorndike Road, Greensboro, NC

27409-9421 (www.rfmd.com).

INTERMODULATION

JOSE

´

CARLOS PEDRO

University of Aveiro

Portugal

1. INTRODUCTION

1.1. What Is Intermodulation Distortion?

Although the term intermodulation is used by some

authors to describe a speciﬁc manifestation of nonlinear

distortion, in this text we will adopt the wide-sense mean-

ing of intermodulation as any form of nonlinear distortion,

unless otherwise explicitly stated. So, it is convenient to

start an introduction to intermodulation by saying a few

words about distortion.

In the ﬁeld of telecommunication systems, distortion is

understood as any form of signal impairment. In this way,

distortion takes the broad sense of all differences between

the received and the transmitted information signals,

speciﬁcally, those added or signal-dependent perturba-

tions.

In the ﬁrst set of added, or signal-independent, pertur-

bations, we should include random noise and determinis-

tic interferences. Typical examples of the former are the

always present thermal noise or shot noise of electronic

circuits. The second could be illustrated by some man-

made (synthetic) repetitive impulsive noise or simply

another telecommunications channel that shares the

same transmission medium but that does not carry any

useful information.

The set of signal-dependent perturbations can also be

divided into two major parts—linear distortion and non-

linear distortion—according to whether what distin-

guishes the received signal from its transmitted version

is due to a linear or a nonlinear process. The reason for this

organization stands in the easiness with which we correct

linear distortion and the difﬁculty we have in dealing with

nonlinearity. In fact, since linear distortion describes all

differences in time-domain waveform, or frequency-domain

spectrum, as the ones caused by any usual ﬁlter or dis-

persive transmission medium, it can be corrected by an-

other inverse ﬁlter, with a methodology usually known as

pre- or postequalization. On the other hand, nonlinear

distortion cannot be corrected this way, remaining nowa-

days as a very tough engineering problem.

So, from a purely theoretical point of view, what

distinguishes linear distortion from nonlinear distortion

is simply the essence of the mapping corresponding to the

telecommunication system, from the signal source to the

detected signal. If that mapping responds to scaled ver-

sions of two different signals with two scaled versions of

the responses to these two signals, when they are pro-

cessed individually, we say that our transmission system

obeys superposition, and is thus linear [1]. In any other

case, we say that the system is a source of nonlinear

distortion. Nonlinear distortion can, therefore, manifest

itself in many different forms that range from the obvious

signal clipping of saturated ampliﬁers, to the almost

unnoticeable total harmonic distortion present in our

high-ﬁdelity audio ampliﬁers.

Because nonlinear distortion is a property not shared

by our more familiar linear systems, we could think of it as

something visible only in some special-purpose systems or

poorly designed circuits. Unfortunately, that is not the

case. To a greater or lesser extent, nonlinear distortion is

present in the vast majority of electronic systems. Because

nonlinear distortion is associated with PN and PIN diodes

or varactors [2], it is found in many control devices such as

solid-state switches, controlled attenuators, phase shif-

ters, or tunable ﬁlters—and, because of the recognized

nonlinearity of magnetic-core inductors, it can also arise

from other passive ﬁlters and diplexers. However, prob-

ably more surprising, is the fact that it can even arise from

devices usually assumed as linear.

One example is the nonlinear distortion produced by

some RF MEM (radiofrequency micromachined electro-

mechanical) switches [3]. Another is passive intermodula-

tion (PIM), which is frequently observed when loose

connections, or junctions made of different metals or of

similar but oxidized metals are subject to high power

levels [4]. So, PIM is generated in many RF connectors,

antennas, antenna pylons, wire fences, and other compo-

nents. Finally, intermodulation can even arise from our

supposedly linear electronic circuits as it is inherent to the

operation of all electronic active transducers. To under-

stand this, let us take the example of the general ampliﬁer

described in Fig. 1.

Because our ampliﬁer is a physical system, it must obey

energy conservation, which implies that the sum of all

INTERMODULATION 2189

forms of input power—either signal power P

in

or DC

supply power P

DC

—must equal the sum of all forms of

output power, whether it is signal power delivered to the

load P

out

or dissipated power P

diss

such as heat or harmonic

distortion:

P

in

þP

DC

¼P

out

þP

diss

ð1Þ

On the other hand, we know that output signal power

should be a scaled replica of the input signal power,

deﬁning, in this way, a certain ampliﬁer power gain:

G

P

P

out

P

in

ð2Þ

However, (1) also implies that

G

P

¼1þ

P

DC

ÀP

diss

P

in

ð3Þ

which shows that no ampliﬁer that relies on a real supply

of ﬁnite power can keep its gain constant for any increasing

input signal level. Sooner or later, it will have to show gain

compression, presenting, therefore, nonlinearity.

1.2. Characterizing Intermodulation Distortion

After this brief introduction to the concept of intermodula-

tion distortion, let us now see in more detail which forms

of distortion it describes. For that, we will assume a simple

system represented by the following cubic model:

yðtÞ ¼a

1

xðt Àt

1

Þ þa

2

xðt Àt

2

Þ

2

þa

3

xðt Àt

3

Þ

3

þ Á Á Á ð4Þ

in which the input x(t) is nonlinearly transformed into an

output y(t). Note that this system not only shows non-

linearity as it also has memory, since it does not respond

instantaneously to the input, but to certain past versions

of it. This dynamic behavior is due to the presence of the

delays t

1

, t

2

, and t

3

.

1.2.1. Single-Tone Distortion Characterization. Suppos-

ing the input is initially composed of one amplitude A(t)

and phase y(t) modulated RF carrier of frequency o

c

xðtÞ ¼AðtÞ cos½o

c

t þyðtÞ ð5Þ

the output will be composed of three sets of terms, say,

y

1

(t), y

2

(t), and y

3

(t), each one corresponding to a certain

polynomial degree. Illustrations of the time-domain wave-

forms and frequency-domain spectra of the input x(t) and

output y(t) are depicted in Figs. 2a and 2b and in Fig. 3a

and 3b, respectively.

The ﬁrst term of the output is given by

y

1

ðtÞ ¼a

1

Aðt Àt

1

Þ cos½o

c

t þyðt Àt

1

Þ Àf

1

ð6Þ

(where f

1

¼o

c

t

1

) and corresponds to the expected linear

response. Note that it includes exactly the same frequency

components already present at the input.

The second term

y

2

ðtÞ ¼

1

2

a

2

Aðt Àt

2

Þ

2

þ

1

2

a

2

Aðt Àt

2

Þ

2

cos½2o

c

t þ2yðt Àt

2

Þ À2f

2

ð7Þ

(where f

2

¼o

c

t

2

) involves baseband products whose fre-

quency falls near DC and some other products whose

frequencies are located around the second harmonic,

2o

c

. The ﬁrst ones consist of second-order intermodulation

products of the form o

x

¼o

1

Ào

2

(in which o

x

is the

resulting frequency, while o

1

and o

2

are any two distinct

frequencies already present at the input), and describe the

demodulation generally provided by even-order nonlinea-

rities. When o

1

¼o

2

, then o

x

¼0, and the terms fall

exactly at DC. So, they also describe the circuit’s DC

bias shift. Because they are what is sought in AC to DC

converters, in ampliﬁers they model the variation of the

y(t) mean value from the quiescent point, to the mean

value shown in presence of a signiﬁcant RF excitation—

the large-signal bias point. The second type of even-order

products is again second-order intermodulation distortion

whose frequency now falls at o

x

¼o

1

þo

2

. For o

1

¼o

2

,

Signal

source

Signal

load

Power supply

P

diss

P

diss

P

diss

P

diss

P

diss

P

diss

P

diss

P

diss

P

diss P

diss

P

diss

P

diss

P

diss

P

diss

P

diss

P

diss

P

diss P

diss

P

dc

P

in

P

out

V

DD

V

GG

Figure 1. Conceptual ampliﬁer showing input/output

power relations.

2190 INTERMODULATION

0 1000 3000 6000 8000

−300

−200

−100

0

Frequency (MHz)

|S

xx

(f )| (dB)

2000 4000 7000 5000

(b)

1.0 1.1 1.2

1.3 1.4 1.5

−1

−0.5

0

0.5

1

x(t ) ( )

1.05 1.15 1.25 1.35 1.45

(a)

Time ( s) µ

Figure 2. (a) Time-domain waveform of the in-

put signal x(t); (b) corresponding frequency-

domain spectrum.

−10

−5

0

5

10

y(t ) ( )

1.0 1.1 1.2

1.3 1.4 1.5 1.05 1.15 1.25 1.35 1.45

Time ( s)

(a)

0 1000 2000 3000 4000 5000 6000 7000 8000

−300

−200

−100

0

Frequency (MHz)

|S

yy

(f )| (dB)

(b)

µ

Figure 3. (a) Time-domain waveform of the

output signal y(t); (b) corresponding frequency-

domain spectrum.

INTERMODULATION 2191

then o

x

¼2o

1

, and the products are known as second-

order harmonic distortion.

Finally, the third term is

y

3

ðtÞ ¼

3

4

a

3

Aðt Àt

3

Þ

3

cos½o

c

t þyðt Àt

3

Þ Àf

3

þ

1

4

a

3

Aðt Àt

3

Þ

3

cos½3o

c

t þ3yðt Àt

3

Þ À3f

3

ð8Þ

(where f

3

¼o

c

t

3

) and also involves two different sets of

products located near the input frequency band (or funda-

mental band) o

c

and the third harmonic 3o

c

.

As are the even-order products, the third-order pro-

ducts falling near the third harmonic 3o

c

are classiﬁed as

out-of-band products. Appearing at o

x

¼o

1

þo

2

þo

3

, that

is, out of the fundamental signal band in RF systems of

narrow bandpass characteristics, these products seldom

constitute a major source of nonlinear signal impairment

as they can be easily ﬁltered out. Note, however, that they

may also constitute in-band products in ultra-wide-band

systems such as cable television (CATV).

Third-order products falling exactly over, or in the

vicinity of o

c

, in which the resulting frequencies can be

either o

x

¼o

1

þo

2

Ào

3

, o

x

¼2o

1

Ào

2

, or even o

x

¼o

1

(whether they arise from the combination of three distinct,

two equal and one different, or three equal input frequen-

cies, respectively), are obviously called in-band products.

Contrary to the products treated above, they cannot be

eliminated by linear ﬁltering, constituting the principal

object of intermodulation distortion studies in microwave

and wireless systems. In fact, some authors even reserve

the term intermodulation distortion for this particular

form of nonlinear signal perturbation.

To analyze these in-band distortion products in more

detail, we will now consider two different situations of

system memory. In the ﬁrst case, it is assumed that the

time delays of (4) are due only to the active device’s

reactive components or to the input and output matching

networks. In this way, they may be comparable to the RF

carrier period, but negligible when compared to the much

slower modulation timescale. Therefore, the in-band pro-

ducts can be approximated by

3

4

a

3

AðtÞ

3

cos½o

c

t þyðtÞ Àf

3

ð9Þ

which shows that, although the system kept its dynamic

behavior to the RF carrier, it became memoryless (i.e.,

responds instantaneously) to the modulation envelope.

Since general amplitude and phase modulations have

frequency components that start at DC, we have already

seen that these products include spectral lines falling

exactly over the ones already present at the input, and

some other new components named as spectral regrowth.

The third-order signal components that are coincident

with the input are given by o

x

¼o

1

þo

1

Ào

1

¼o

1

þ

(o

1

Ào

1

) ¼o

1

and can be understood as being generated

by mixing second-order products at DC with ﬁrst-order (or

linear) ones. Except for their associated gain, which is no

longer a

1

, but

3

4

a

3

multiplied by the input amplitude-

averaged power A

2

, these products are indistinguishable

from the linear components of (6). They carry the same

information content, and are, therefore, termed signal-

correlated products. Although, in a strict sense, they

should be considered as nonlinear distortion products (as

their signal power rises at a slope of 3 dB/dB against the

1 dB/dB that characterizes truly linear components), from

an information content viewpoint, they may also be con-

sidered as linear products. In fact, since, for a constant-

input-averaged power, they cannot be distinguished from

the ﬁrst-order components, it all happens as if the ampli-

ﬁer had remained linear but with a gain that changed

from its small-signal value of G¼a

1

exp( Àjf

1

) to an

amplitude-dependent large-signal gain of G(A) ¼

a

1

exp( Àjf

1

) þ(3/4)A

2

a

3

exp( Àjf

3

). So, input amplitude

signal variations [or amplitude modulation (AM)] produce

different output amplitude variations, according to the so-

called ampliﬁer AM–AM conversion. But, since the gain is

also characterized by a certain phase, it is obvious that

input amplitude signal variations will also generate out-

put phase variations. In conclusion, and as illustrated in

Figs. 4a and 4b, the ampliﬁer will show not only AM–AM

but also AM–PM conversion.

Figure 5 depicts a possible block diagram of a labora-

tory setup intended to measure these static AM–AM and

AM–PM characteristics [6]. As shown, it relies on a usual

microwave vector network analyzer whose signal source is

swept in power.

As a curious aside from this analysis, we should point

out that, although our nonlinearity manifests a signal

amplitude-dependent gain, it is completely insensitive to

P

in

(dBm)

P

out

(dBm)

−15 −10 −5 0 5 10 15 20 25

−5

0

5

10

15

20

25

AM-AM

(a)

−1°

0

1°

2°

3°

4°

5°

( °) φ

P

in

(dBm)

AM-PM

−15 −10 −5 0 5 10 15 20 25

(b)

Figure 4. (a) Ampliﬁer’s AM–AM conversion;

(b) AM–PM conversion.

2192 INTERMODULATION

the input signal phase. In fact, as can be concluded from

(9), the bandpass characteristics of our ampliﬁer would be

completely transparent to a phase-modulated signal of

constant amplitude, in the sense that the phase informa-

tion present at the output would be exactly equal to the

phase information present at its input.

In the second case, it is supposed that, beyond the usual

time constants of the order of the RF carrier period, our

system may even present time delays, t

1

0

, t

2

0

, and t

3

0

,

comparable to the modulation period (e.g., determined by

the bias circuitry, active-device charge carrier traps, self-

heating). Such time constants are no longer irrelevant

for the envelope evolution with time, and the system is

said to present long-term or envelope memory effects. The

in-band output distortion becomes

3

4

a

3

Aðt Àt

3

0

Þ

3

cos½o

c

t þyðt Àt

3

0

Þ Àf

3

0

ð10Þ

and the output envelope will show a phase shift that is

dynamically dependent on the rate of amplitude varia-

tions. In this case, the output AM–AM or AM–PM is no

longer static, and dynamic (or hysteretic) AM–AM and

AM–PM conversions are observed, as shown in Figs. 6a

and 6b.

This shows that, if our nonlinear system only presents

short-term memory effects, and thus is memoryless for the

envelope, it may be characterized by a set of gain and

AM-AM

P

in

(dBm)

P

out

(dBm)

(a)

−15 −10 −5 0 5 10 15 20 25

−5

0

5

10

15

20

25

AM-PM

(°) φ

P

in

(dBm)

(b)

−1°

0

1°

2°

3°

4°

5°

−10 −5 0 5 10 15 20 25 −15

Figure 6. Typical hysteretic AM–AM (a)

and AM–PM (b) characteristics shown by

nonlinear dynamic ampliﬁers suffering

from both short-term and long-term

memory effects.

1.800

GHz dBm

DC

AMP

Power and Frequency

control

Figure 5. AM–AM and AM–PM characteriza-

tion setup based on a microwave vector network

analyzer.

INTERMODULATION 2193

phase shift tests made with a sinusoidal, or CW (contin-

uous-wave), excitation with swept amplitude and, even-

tually, with varying frequency. However, if the system is

also dynamic to the envelope, then the observed AM–AM/

AM–PM varies with the speed of the input amplitude

sweep, and such a test becomes questionable. Since each

of the tested CW signals can be seen as a carrier modu-

lated by a constant (DC) envelope, it becomes obvious that

we cannot fully characterize a dynamic system using only

these simple DC excitations.

Moreover, it is clear that testing in-band intermodula-

tion products with a CW signal will never be an easy task,

as the output will only have signal-correlated components

where o

x

¼o

c

, which all overlap onto the usually much

higher linear output. Obviously, in-band intermodulation

characterization requires more complex stimuli.

1.2.2. Two-Tone Distortion Characterization. One way

to increase the complexity of our test signal is to use a

two-tone excitation:

xðtÞ ¼A

1

cosðo

1

tÞ þA

2

cosðo

2

tÞ ð11Þ

The in-band output components of (4) when subject to this

new stimulus will be

a

1

A

1

cos½o

1

t Àf

110

þa

1

A

2

cosðo

2

t Àf

101

Þ

þ

3

4

a

3

A

2

1

A

2

cos½ð2o

1

Ào

2

Þt Àf

32À1

þ

3

4

a

3

A

3

1

þ

6

4

a

3

A

1

A

2

2

_ _

cos½o

1

t Àf

310

þ

6

4

a

3

A

2

1

A

2

þ

3

4

a

3

A

3

2

_ _

cosðo

2

t Àf

301

Þ

þ

3

4

a

3

A

1

A

2

2

cos½ð2o

2

Ào

1

Þt Àf

3À12

ð12Þ

Beyond the expected linear components arising at o

1

and

o

2

, (12) is also composed of other third-order products at

o

1

, o

2

, 2o

1

Ào

2

, and 2o

2

Ào

1

. They constitute again the

signal-correlated (o

1

and o

2

) and signal-uncorrelated

(2o

1

Ào

2

and 2o

2

Ào

1

) components. The terms at o

1

(o

2

) that are dependent only on A

1

(A

2

) constitute the

AM–AM/AM–PM conversion discussed above. But now

there are some new terms at o

1

(o

2

) whose amplitude is

also controlled by A

2

(A

1

). They model two different, but

obviously related, nonlinear effects. One is cross-modula-

tion, a nonlinear effect in which amplitude modulation of

one RF carrier is converted into amplitude modulation of

the other; the other is known as desensitization, the loss of

receiver sensitivity to one signal when in presence of an

incoming strong perturbation (e.g., a jammer).

The terms at 2o

1

Ào

2

and 2o

2

Ào

1

are spectral re-

growth components that appear as sidebands located side

by side to the fundamentals at a distance equal to their

frequency separation o

2

Ào

1

. These in-band intermodula-

tion distortion (IMD) sidebands rise at a constant slope of

3 dB per dB of input level rise, until higher-order compo-

nents (in the case of our polynomial model, output con-

tributions due to higher-degree terms) show up. Since

ﬁrst-order components rise at a slope of only 1 dB per dB,

we could conceive of an extrapolated (never reached in

practice) output power where the output IMD and funda-

mentals would take the same value. As illustrated in

Fig. 7, this is the so-called third-order intercept point IP

3

.

Although meaningful only for small-signal regimes,

where the fundamental and IMD components follow their

idealized straight-line characteristics, IP

3

is still the most

widely used (some times erroneously) intermodulation

distortion ﬁgure of merit.

Figure 8 shows a block diagram of the most popular

laboratory setup used for two-tone intermodulation tests.

It relies on a two-tone generator of high-spectral purity,

and a high-dynamic range microwave spectrum analyzer.

Although, for many years, two-tone intermodulation

characterization has been restricted to these amplitude

measurements, more recently we have seen an increasing

interest to also identify the IMD components’ phase. The

reason for this can be traced to the efforts devoted to

extract behavioral models capable of representing the

device’s IMD characteristics and to the design of ampliﬁer

linearizers that must be effective even when the main

nonlinear device presents long-term memory effects. In

fact, since most of the linearizers can be understood as

auxiliary circuits capable of generating IMD components

that will cancel the ones arising from the main ampliﬁer,

it is obvious that those linearizing circuits must be

designed to meet both IMD amplitude and phase require-

ments.

Unfortunately, the ﬁrst problem that arises when try-

ing to measure the IMD components’ phase is that, despite

phase is a relative entity, we have no phase reference for

IMD. Contrary to what happens to the output fundamen-

tals in which we can refer their phases to the phases at the

input (usually arbitrarily assumed zero), the problem is

that now there are no input components at the IMD

frequencies. So, we ﬁrst need to create a reference signal

at that IMD frequency. That is usually done with a

−10 20

−100

−80

−60

−40

−20

0

20

40

P

Fund

(

2

)

1dB/dB

3dB/dB

P

out

(dBm)

P

in

(dBm)

0

P

IMD

(2

2

–

1

)

IP

3i

10 30

IP

3

ω

ω ω

Figure 7. Typical fundamental and third-order intermodulation

power versus input power plots. Note the deﬁnition of the extra-

polated third-order intercept point IP

3

.

2194 INTERMODULATION

reference nonlinearity; thus IMD phase measurement

results become relative to the reference nonlinearity

used in the setup. For example, in the setup depicted in

Fig. 9, the reference nonlinearity is based in the nonlinear

characteristic of broadband Schottky diodes, and the

phase value is acquired from the variable phase shift

necessary to balance the device under test (DUT) and

reference arms.

1.2.3. Multitone Distortion Characterization. For com-

pleteness, let us now brieﬂy introduce intermodulation

characterization under multitone excitations. A detailed

analysis of this important and up-to-date subject can be

found in various references [e.g. 5,6].

First, we will assume that our stimulus can be de-

scribed as a sum of Q sinusoids of different frequencies:

xðtÞ ¼

Q

q¼1

A

q

cosðo

q

tÞ ¼

1

2

Q

q¼ÀQ

A

q

e

jo

q

t

ð13Þ

The output of a general power series such as (4) to the

excitation of (13) will be

yðtÞ ¼

N

n¼1

y

n

ðtÞ ð14aÞ

where each of the orders can be expressed as

y

n

ðtÞ ¼

1

2

n

a

n

Q

q ¼ÀQ

A

q

e

jo

q

t

_ _

n

¼

1

2

n

a

n

Q

q1 ¼ÀQ

Á Á Á

Q

qn ¼ÀQ

A

q

1

Á Á Á A

q

n

e

jðo

q

1

þÁÁÁ þo

qn

Þt

ð14bÞ

which contains various frequencies at o

x

¼o

q1

þ?þo

qn

,

originating from many different mixing products.

Since there is, in general, more than one mixing

product—that is, more than one combination of input

frequencies—falling at the same frequency, the calcula-

tion of their output amplitude requires that ﬁrst we are

able to determine the number of those different combina-

tions. One systematic way to do this is to recognize that

their frequencies must obey [7]

o

n;m

¼o

q

1

þ Á Á Á þo

qn

¼m

ÀQ

o

ÀQ

þ Á Á Á

þm

À1

o

À1

þm

1

o

1

þ Á Á Á þm

Q

o

Q

ð15Þ

2

1

1

2

1

2

AMP

DC − 00.53 + 15.00

1.709

1.801

GHz

GHz

dBm

dBm

Isolators

Low-Pass

Filters

Figure 8. The most popular laboratory setup used for two-tone intermodulation tests.

DC

AMP

1.709

GHz dBm

Reference

Nonlinearity

Phase Shifter

Attenuator

DC

Isolator

Low-Pass

Filters

Isolator

1.801

GHz dBm

**Figure 9. Possible IMD phase measurement setup based on a reference nonlinearity, a spectrum
**

analyzer, and an IMD cancellation loop.

INTERMODULATION 2195

where

Q

q ¼ÀQ

m

q

¼m

ÀQ

þ Á Á Á þm

À1

þm

1

þ Á Á Á þm

Q

¼n ð16Þ

deﬁning the following mixing vector:

v¼½m

ÀQ

Á Á Á m

À1

m

1

Á Á Á m

Q

ð17Þ

Then, the number of different ways of generating the same

mixing vector is given by the multinomial coefﬁcient [7]:

t

n;n

¼

n!

m

ÀQ

! . . . m

À1

!m

1

! . . . m

Q

!

ð18Þ

These Q-tone distortion components allow a generaliza-

tion of the two-tone signal-to-intermodulation distortion

ratio [IMR; sometimes also known as carrier-to-IMD ratio

(C/I)] to various multitone distortion ﬁgures of merit. One

of these is deﬁned as the ratio between the constant-

amplitude output fundamental signals and the highest

sideband IMD component (M-IMR).

Another measure is the ratio between integrated fun-

damental output power and integrated upper or lower

sideband distortion. As this sideband spectral regrowth

falls exactly over the location of a potentially present

adjacent channel, it is called the adjacent-channel power

ratio (ACPR).

Finally, a measure of the ratio of the fundamentals to

the signal-uncorrelated distortion components that fall

exactly among the fundamental components is given by

the so-called noise-power-ratio (NPR). The reason for this

denomination comes from the fact that, although that

ﬁgure of merit is being introduced in this text for a

multitone excitation, it was traditionally measured with

a bandlimited white-noise stimulus—a generalized multi-

tone excitation with an inﬁnite number of tones.

Besides all these ﬁgures are measures of nonlinear

effects that share a common physical origin, and it has

not been easy to relate them, except for very particular

situations. First, we [5] presented relations between

various multitone distortion ﬁgures and IMR, obtained

for a third-degree polynomial memoryless model. Then,

Boulejfen et al. [8] extended those results for a ﬁfth-degree

polynomial. As a summary of these results, Fig. 10 pre-

sents the ratio of IMR to the above-deﬁned multi-tone

distortion ﬁgures versus the number of tones Q for a

memoryless cubic polynomial.

A laboratory setup for multitone distortion tests is

similar to the one already shown for two-tone tests, except,

obviously, with respect to the signal generator [6]. How-

ever, since a NPR test focuses on the distortion that falls

exactly over the output fundamentals, something must be

done to separate the desired distortion components from

the much higher fundamental signals. The usual way to

solve that problem consists in creating a very narrow

measurement window within the input signal bandwidth.

This is accomplished by either shutting down a few input

tones—when a multitone signal generator is used—or

introducing a notch ﬁlter between the bandlimited

white-noise generator and the nonlinear device under

test [6].

2. CAD TOOLS FOR INTERMODULATION

DISTORTION PREDICTION

Because intermodulation distortion is a nonlinear effect,

any attempt to predict its behavior by hand for even the

most simple practical circuits or devices becomes extre-

mely difﬁcult, if not impossible. So, intermodulation dis-

tortion prediction relies heavily on good device models and

appropriate computer simulation algorithms. Unfortu-

nately, these subjects are so vast that we have to restrict

this text to a ﬁrst guiding overview. So, we will concen-

trate our discussion on a set of criteria for model quality

(for this speciﬁc purpose) and give some hints concerning

usual simulation tools.

2.1. Nonlinear Device Modeling for Distortion Analysis:

General Considerations

Starting with nonlinear device models, we can divide them

into four general groups: (1) physical and empirical mod-

els and (2) global and local models.

Physical models are mathematical descriptions of the

internal device operation that are drawn from the know-

ledge of the device’s geometric and physical structure, and

ACPR

M-IMR

NPR

0 10 20 30 40 50 60 70 80 90 100

−4

−2

0

2

4

6

8

10

Number of Tones, Q

IMR/Q-Tone Distortion (dB)

110 120 130

IMR (6.0 dB)

1

4

4

3

IMR (1.3 dB)

IMR (7.8 dB)

6

1

140 150 160

Figure 10. Ratio of two-tone IMR to NPR, M-

IMR and ACPR versus the number of tones Q

for a memoryless cubic polynomial.

2196 INTERMODULATION

from the application of a certain set of basic physics laws.

Although relying on extremely complex formulations that

require an enormous number of parameters and are

computationally expensive to evaluate, they can provide

much better accuracy than the empirical models as they

necessarily mimic the basic device operation. On the other

hand, empirical models do not require any information

about the internal structure of the device, relying com-

pletely on input–output behavioral observations. Hence,

they are also known as blackbox models or behavioral

models.

Typical examples of physical models are the Schottky

diode equation and the device models described by a set of

coupled partial-differential equations of electric potential,

charge, and charge carrier mobility. Examples of purely

behavioral models are the linear scattering matrix, table-

based device models, or even the abovementioned AM–

AM/AM–PM models.

Local models can be distinguished from global models

for their approximation range. Because they are beha-

vioral in nature, they constitute two different compro-

mises between the domain of ﬁtting and the level of

accuracy. While local models are very good in representing

mild nonlinear behavior in the vicinity of some quiescent

point, global models are conceived as valid for any possible

operation regime, but at the expense of an increased error.

It is therefore natural that they are also known as small-

signal or large-signal models, respectively. The Gummel–

Poon model of BJTs, the quadratic model of FETs, or an

AM–AM/AM–PM representation are examples of global

models, while the poor extrapolation capability usually

associated with polynomial approximators tends to grant

them a distinct local behavior. For example, the cubic

polynomial that could be extracted from the third-order

intercept point is necessarily a local model valid only for

small-signal excitation levels.

The polynomial example given above is not accidental

as it plays a fundamental role in all nonlinear distortion

analysis. In fact, for some unique reason we began this

article using exactly the same polynomial of expression

(4). As we then concluded, if the model is a polynomial, we

have a direct and easy way to calculate the various

intermodulation products to any signal that can be de-

scribed as a sum of sinusoids. Furthermore, by simply

selecting its coefﬁcients, we can tailor the polynomial for

very different approximation goals. To understand that,

let us use an illustration example. Figures 11 and 12

depict the approximation of one typical transfer function

characteristic by two different polynomials: a Taylor series

and a Chebyshev polynomial series, both of 10th degree.

As seen from Figs. 11a and 11b, the coefﬁcients of the

Chebyshev polynomial were selected so that the polyno-

mial could produce an optimum approximation to the

response of our original nonlinearity to a sinusoid of

1.5 V peak amplitude, centered at a quiescent point of

0 V (something close to what is found in typical class AB

power ampliﬁcation regimes).

On the other hand, the coefﬁcients of the Taylor series

were taken as the appropriately scaled derivatives of the

nonlinearity at the same quiescent voltage of 0V. It

constitutes, therefore, the optimum polynomial approxi-

mation to the nonlinear response to any signal of inﬁni-

tesimal input amplitude (see Fig. 11a).

When excited by sinusoids of variable amplitude, the

output DC component (Fig. 12a), the fundamental compo-

nent (Fig 12b), and the second- and third-harmonic dis-

tortion components (Figs. 12c and 12d, respectively)

reveal that these two polynomial approximators present,

indeed, very distinct properties.

The Taylor series is clearly a local approximator that

produces optimum results in the vicinity of the quiescent

point, but then suffers from a catastrophic degradation

when the excitation exceeds B0.3V of amplitude. On the

−2 −1.5 −1 −0.5 0 0.5 1 1.5 2

−2

−1

0

1

2

3

4

5

6

(a)

(b)

f (x)( )

x (V)

0 20 40 60 80 100 120 140 160 180 200

1

2

3

4

5

Time (ns)

f (x(t ))( )

0

Figure 11. (a) Nonlinear memoryless transfer function, f(x) (- -)

and its 10th-order Taylor series approximation around x ¼0 V

(. . .) and Chebyshev polynomial optimized for a sinusoidal input

amplitude of A¼1.5V (—); (b) time-domain waveform of the

output of the transfer function f[x(t)] (- -) and of its 10th-order

Taylor series approximation (. . .) and Chebyshev polynomial (—),

when excited by a CW input of amplitude A¼1.5 V.

INTERMODULATION 2197

contrary, the Chebyshev series is worse at those small-

signal levels, but performs much better up to excitation

amplitudes of 1.5 V. It behaves, therefore, as a global

approximator. (In fact, the Chebyshev series is still a local

approximator whose domain is no longer deﬁned around

the ﬁxed quiescent point of 0 V, but around a new general-

ized dynamic quiescent point imposed by an input sinu-

soid of 1.5 V amplitude.) As any other mean-square error

approximator, the Chebyshev polynomial wanders around

the original nonlinearity (see Figs. 11a and 11b), obviously

failing the higher-order derivatives of the function. This is

why, contrary to the Taylor series, which, by construction,

osculates these derivatives, the Chebyshev series does not

show good small-signal distortion behavior.

What we have just seen in this example is common to

almost all empirical models and results in an important

message as far as intermodulation distortion calculations

are concerned. Simultaneously reproducing the device’s

mild nonlinear details (local characteristics) and the gen-

eral trends (global properties) is so difﬁcult that we should

–35 –30 –25 –20 –15 –10 –5 0 5

4

4.5

5

5.5

6

6.5

7

7.5

8

A(dB

v

)

–35 –30 –25 –20 –15 –10 –5 0 5

A(dB

v

)

F

0

(A) (dB)

(a) (b)

–30

–25

–20

–15

–10

–5

0

5

F

1

(A) (dB)

–35 –30 –25 –20 –15 –10 –5 0 5

–80

–70

–60

–50

–40

–30

–20

–10

–80

–70

–60

–50

–40

–30

–20

–10

0

A (dB

v

)

A (dB

v

)

F

2

( A) (dB)

(c) (d)

3

F (A) (dB)

–35 –30 –25 –20 –15 –10 –5 0 5

Figure 12. (a) DC component of the output of the transfer function f[x(t)] (- -) of its 10th-order

Taylor series approximation (Á Á Á) and Chebyshev polynomial (—), when excited by a CW input of

amplitude 0.015VoAo1.95V; (b) fundamental component of output of transfer function f[x(t)] (- -)

of its 10th-order Taylor series approximation (Á Á Á) and Chebyshev polynomial (—), when excited by

a CW input of amplitude 0.015VoAo1.95V; (c) second-harmonic component of output of transfer

function f[x(t)] (- -) of its 10th-order Taylor series approximation (Á Á Á) and Chebyshev polynomial (—

), when excited by a CW input of amplitude 0.015VoAo1.95V; (d) third-harmonic component of

output of unit transfer function f[x(t)] (- -) and its 10th-order Taylor series approximation (y) and

Chebyshev polynomial (—), when excited by a CW input of amplitude 0.015VoAo1.95V.

2198 INTERMODULATION

never trust an empirical model unless we have guarantees

that it was speciﬁcally tested for nonlinear distortion.

2.2. Nonlinear Models for Distortion Analysis at the

Circuit Level

To perform intermodulation analysis at the circuit level,

that is, to compute the distortion arising from a certain

electronic circuit subject to a speciﬁc bandpass RF input

signal stimulus, the device must be represented by some

equivalent-circuit model [6]. This is the normal modeling

requirement for using either time-marching algorithms,

like the ones used by SPICE, or frequency-domain simu-

lators, such as the harmonic-balance solvers. Such equiva-

lent circuits have topologies and parameter sets usually

supported from both physical and empirical data.

Linear, or bias-independent, elements are usually ex-

tracted from a broadband small-signal AC characteriza-

tion. Nonlinear elements can be either voltage-controlled

current sources (nonlinear device currents) i(v), voltage-

controlled electric charge sources (nonlinear capacitances)

q(v), or current-controlled magnetic ﬂux sources (non-

linear inductances) f(i). Each of these is assumed to be

described by a static, or memoryless, function of its

controlling variable(s), which can, again, be supported

by both physical device knowledge or by empirical obser-

vations. Mostly in this latter case, it is the selection of

these functions that determines the quality of the model

for nonlinear distortion predictions. A small mean-square

error between measured and modeled data in the whole

range of device operation guarantees good global proper-

ties, but says nothing about local properties. To be able to

also provide good predictability under small-signal re-

gimes, the model must osculate at least the ﬁrst three

derivatives of the actual device function, which requires

special model extraction procedures.

Although those derivatives can be obtained from suc-

cessive differentiation of measured i(v), q(v), or f(i) data,

this is not recommended for at least two important

reasons: (1) since most of the microwave transistors

show low-frequency dispersion effects, differentiating DC

data may not lead to the real AC behavior; and (2) the

aggravation of measurement noise produced by numerical

differentiation. If we rely on averages (data integration) to

reduce random measurement errors, it is natural to expect

an aggravation of those errors if we go backward, that is,

numerically differentiating measurement data. So, the

best way to obtain these device derivatives is to measure

entities that directly depend on them; and one good

example of those entities is exactly the harmonic or

intermodulation distortion produced by the device under

a CW or a two-tone excitation.

As an example, the laboratory setup depicted in Fig. 13

uses exactly this principle to acquire the nine coefﬁcients

of the Taylor series expansion of the drain–source current

of a FET:

i

ds

ðv

ds

; v

ds

Þ ¼G

m

v

gs

þG

ds

v

ds

þG

m2

v

2

gs

þG

md

v

gs

v

ds

þG

d2

v

2

ds

þG

m3

v

3

gs

þG

m2d

v

2

gs

v

ds

þG

md2

v

gs

v

2

ds

þG

d3

v

3

ds

ð19Þ

Exciting the FET at the gate side with a sinusoid of

frequency o

1

and at the drain side with a sinusoid of

frequency o

2

allows the extraction of G

m

from the output

current component at o

1

, G

ds

from the component at o

2

,

G

m2

from the component at 2o

1

, G

md

from the component

at o

1

þo

2

, G

d2

from the component at 2o

2

, and so on.

Unfortunately, the actual procedure is not that simple.

Although the unilateral properties presented by micro-

wave FETs at low frequencies guarantee that v

gs

will have

only the o

1

component, the requirement that the device is

terminated at the drain side by a nonnull impedance

determines that v

ds

will have components at o

1

, at o

2

,

and at all their mixing products. This impedes the ortho-

gonal (or one-to-one) extraction just explained, demanding

the solution of a 2 Â2 linear system for G

m

and G

ds

; a 3 Â3

linear system for G

m2

, G

md

, and G

d2

; and a 4 Â4 linear

system for extracting G

m3

, G

m2d

, G

md2

, and G

d3

[9]. Since

the concept supporting this setup is general, it can be

extended to other nonlinear current sources present in

any nonlinear device equivalent-circuit model, or even to

charge sources [10].

As an illustrative example, Fig. 14 shows all nine

coefﬁcients of (19) extracted with the setup of Fig. 13,

FET

LPF

LPF

Diplexer

V

DS

V

L

(2)

V

GS

ATTN ATTN

ATN

Spectrum

analyzer

Double power

supply

Vs(1)

Figure 13. Laboratory setup used to extract

the Taylor series coefﬁcients of a bidimensional

nonlinearity such as the i

DS

(v

GS

,v

DS

) of a FET.

INTERMODULATION 2199

from a medium-power microwave GaAs MESFET biased

in the saturation region.

2.3. Nonlinear Models for Distortion Analysis at the

System Level

Although system simulation for the modulated bandpass

RF signals has already taken the ﬁrst steps, system

simulation at the complex envelope level is, by far, the

most usual way to assess distortion performance of entire

communication systems. It assumes that the amplitude/

phase-modulated RF signal of (5) can be given by

xðtÞ ¼AðtÞ cos½o

c

t þyðtÞ

¼Re½AðtÞe

jyðtÞ

e

jo

c

t

¼Re½ ~ xxðtÞe

jo

c

t

ð20Þ

in which ~ xxðtÞ is the complex envelope—the lowpass equiva-

lent signal of x(t) [11]—and that we are interested only in

the system’s in-band characteristics. Thus, the object of

the analysis ceases to be the real bandpass RF-modulated

signal to become only the complex lowpass envelope. In

this way, a signiﬁcant improvement in simulation efﬁ-

ciency is achieved because time-domain simulations no

longer need to be carried on with sampling rates imposed

by the RF carrier and its harmonics, but only by the much

slower envelope. So, the models required for these envel-

ope-level system simulators are lowpass complex equiva-

lent behavioral models of the original bandpass RF

components [11]. They are, therefore, single-input/single-

output maps, which may be either linear on nonlinear.

Linear maps are easily implemented as gain factors in

the memoryless case, or as ﬁnite or inﬁnite impulse

responses, FIR or IIR, digital ﬁlters [11,12], when in

presence of dynamic elements.

A linear dynamic complex envelope ﬁlter whose fre-

quency response function is

~

HHðj ~ ooÞ can be directly derived

from the corresponding circuit level ﬁlter HðjoÞ by simply

going through the following bandpass–lowpass transfor-

mation [11]

~

HHðj ~ ooÞ ¼H½jð ~ ooþo

c

Þuð ~ ooþo

c

Þ ð21Þ

where u(o) is the unity step function.

G

m

(mS) G

m2

(mS/V), G

m3

(mS/V

2

)

200

150

100

50

0

–50

200

150

100

50

0

–50

V

GS

(V)

4.0 –3.5 –3.0 –2.5 –2.0 –1.5 –1.0 –0.5

G

ds

(mS) G

md

(mS/V), G

m2d

(mS/V

2

)

G

d3

(mS/V

2

) G

d2

(mS/V), G

md2

(mS/V

2

)

50

40

30

20

0

–20

15

10

5

–5

10

–10

0

–0.4

1

0

–3

–0.3

2

–2

–1

3

–0.2

–0.1

0.1

0.2

0.3

0.4

0

V

GS

(V)

4.0 –3.5 –3.0 –2.5 –2.0 –1.5 –1.0 –0.5

V

GS

(V)

4.0 –3.5 –3.0 –2.5 –2.0 –1.5 –1.0 –0.5

(a) (b)

(c)

Figure 14. Taylor series coefﬁcients of the bidimensional voltage-controlled i

DS

(v

GS

,v

DS

) current

source of a GaAs MESFET for a constant V

DS

in the saturation zone: (a) G

m

(—), G

m2

(–K–), and

G

m3

(– þ–); (b) G

ds

(—), G

md

(–K–) and G

m2d

(– þ–); (c) G

d2

(—), G

md2

(–K–), and G

d3

(– þ–).

2200 INTERMODULATION

2.3.1. Memoryless AM–AM/AM–PM Models. In their

most basic form, nonlinear complex envelope models

simply try to describe the amplitude-dependent memory-

less nonlinear effects observed for the amplitude and

phase modulation content. They are the AM–AM/AM–

PM models discussed above, and of which the quadra-

ture Saleh model of Fig. 15 is one of the most widely

known [13].

When modeled as a polynomial nonlinearity, this AM–

AM/AM–PM model can include only odd-degree (2nþ1)

terms involving n negative carrier frequencies plus (nþ1)

positive ones [11]

~ yyðtÞ ¼

ðNÀ1Þ=2

n¼0

p

2nþ1

2

2n

2nþ1

nþ1

_ _

~ xxðtÞj ~ xxðtÞj

2n

ð22Þ

where

m

r

_ _

stands for the number of different combinations

of r elements taken from a population of size m and p

2nþ1

are the polynomial coefﬁcients, now having real and

imaginary parts.

2.3.2. Dynamic AM–AM/AM–PM Models. As already

seen in the introduction, when the system presents mem-

ory not only to the RF signal (as indicated by the AM–PM

effect) but also to the slowly varying lowpass envelope, this

AM–AM/AM–PM model becomes unsatisfactory and a

true dynamic model is required. For example, one possi-

bility for such an extension could be to make the in-phase

A

I

(.) and quadrature A

Q

(.) static nonlinear functions

dependent not on the amplitude envelope but on some

dynamic version of it. In this way, the AM–AM and

AM–PM conversions would no longer be instantaneous

functions of A(t), but, as shown in Fig. 16, become instan-

taneous functions of an auxiliary dynamic variable ~ zzðtÞ,

and thus dynamically varying with A(t).

2.3.3. Memoryless Nonlinearity: Linear Filter Cascade

Models. Beyond the methods described above, several

other approximated topologies have been tried for build-

ing nonlinear dynamic models [14]. Some of those, like the

two- or three-box models shown in Fig. 17, deserve men-

tion because of their practical relevance. In fact, they

somehow mimic the internal structure of typical RF

devices (as microwave power ampliﬁers), which are

usually constituted by a broadband (memoryless) non-

linear active device sandwiched between two linear dy-

namic input and output matching networks.

As shown in Fig. 17, these two- or three-box nonlinear

dynamic models can be cascades of a linear ﬁlter followed

by the measured memoryless AM–AM/AM–PM nonlinear

model (known as the Wiener model), be cascades of this

AM–AM/AM–PM memoryless nonlinearity followed by a

linear ﬁlter (the Hammerstein model), or even be consti-

tuted by a combination of both (the Wiener–Hammerstein

model). Other parallel combinations of memoryless non-

linearities and linear ﬁlters also became popular when an

optimal extraction procedure was shown to be practically

possible [15].

2.3.4. General Nonlinear Dynamic Models. Unfortu-

nately, these three-box models become hopelessly inaccu-

rate when the dynamic effects presented to the envelope

are not due to the bandwidth limitations of the linear

matching networks, but are intrinsically mixed with the

nonlinearity [14]. That is the case, for example, with

wireless power ampliﬁers whose nonlinear dynamic ef-

fects cannot obviously arise from bandwidth limitations—

the RF signal can have bandwidths as narrow as 1% or

0.01%, but from the active device self-heating or from

reactive (to the envelope) bias paths. In such cases, more

general nonlinear dynamic models, as the ones brieﬂy

explained in the following paragraphs, must be attempted.

When the lowpass equivalent system is stable, contin-

uous, and of fading memory (i.e., its response cannot keep

memory from an inﬁnitely remote past), mathematics

operator theory has shown that its response ~ yyðtÞ to any

input ~ xxðtÞ can be approximated, within any desired error

margin, by

~ yyðsÞ ¼f

NL

½ ~ xxðsÞ; ~ xxðs À1Þ; . . . ; ~ xxðs ÀQÞ ð23Þ

where f

NL

(.) is a (Qþ1)-to-one static nonlinear function, s

is the time instant in which the output is being calculated;

~ xxðs À1Þ; . . . ; ~ xxðs ÀQÞ are delayed, or past versions of the

x(t ) y(t )

Nonlinear / Memoryless

Nonlinear / Memoryless

jA

Q

[A(t)]sin

y

[A(t)]

|x(t)|

|x(t )|

x(t)

~ ~

~

~

~

e

j (t)

A

y

[A(t )]e

j [A(t )]

A(t)

A

I

[A(t)]cos

y

[A(t)]

**Figure 15. AM–AM and AM–PM memoryless lowpass equiva-
**

lent behavioral model known as the Saleh quadrature model.

Nonlinear/Memoryless

e

j (t )

A(t )

Linear Filter

y(t )

~

x(t )

~

z(t )

~

|x(t )|

x(t)

~

~

|x(t )|

~

A

y

[A(t ),z(t)]e

j

y

[A(t ),z(t )] ~

~

H

z

()

Figure 16. An AM–AM/AM–PM model in which the ampliﬁer is

modeled as a dynamic gain function of the envelope amplitude.

x(t ) y(t )

AM-AM/AM-PM O() H()

Figure 17. A three-box, or Wiener–Hammerstein, lowpass

equivalent model.

INTERMODULATION 2201

input; ~ xxðsÞ; and Q is the system’s ﬁnite memory span.

Indeed, expression (23) simply states that the system

output at a certain instant can be calculated as the non-

linear combination of the input at that instant and all its

past versions within the memory span. There are basically

two ways of implementing this nonlinear and dynamic

input–output mapping, depending on whether f

NL

(.) is

approximated by a (Qþ1)-to-one polynomial or by a

neural network: polynomial ﬁlters [12] and artiﬁcial

neural networks (ANNs) [16].

In the ﬁrst case, (23) becomes

~ yyðsÞ ¼

N

n¼1

~ yy

n

ðsÞ ð24aÞ

where

~ yy

n

ðsÞ ¼

Q

q

1

¼0

Á Á Á

Q

q

2nþ1

¼0

~

hh

2nþ1

ðq

1

; . . . ; q

2nþ1

Þ ~ xxðs Àq

1

Þ

Á Á Á ~ xxðs Àq

nþ1

Þ ~ xxðs Àq

nþ2

Þ

Ã

Á Á Á ~ xxðs Àq

2nþ1

Þ

Ã

ð24bÞ

Such a dynamic polynomial formulation (also known as a

Volterra ﬁlter [12]) presents two important advantages:

1. Its various output components can be traced to a

particular coefﬁcient or term. Therefore, it leads

to useful concepts as nonlinear order and gives

insights into parameter extraction. In fact, this

immediately allows model implementations such as

the ones depicted in Figs. 18a and 18b for the ﬁrst-

and third-order outputs, ~ yy

1

ðsÞ and ~ yy

3

ðsÞ, respectively.

2. The second advantage, shared with all polynomial

approximators, is that the formulation is linear in

the parameters (although obviously nonlinear in the

inputs). Thus, it allows a direct model parameter

extraction based on the solution of a system of

simultaneous linear equations.

Unfortunately, it also presents an important disadvan-

tage. Like any other polynomial approximator, it is a local

model.

It is mostly this drawback that justiﬁes the alternative

ANN formulation. A single hidden-layer ANN can be

expressed as [16]

~ uu

k

ðsÞ ¼

Q

q ¼0

½w

k

ðqÞ ~ xxðs ÀqÞ þb

k

ð25aÞ

~ yyðsÞ ¼b

o

þ

K

k ¼1

w

o

ðkÞf

s

½ ~ uu

k

ðsÞ ð25bÞ

in which the w

k

(q) and w

o

(k) are weighting factors and b

o

and b

k

are bias values, constituting the model parameter

set. f

s

(.) are static single-input/single-output nonlinear

functions (the so-called activating functions) of sigmoid

shape. Because a sigmoid is an output-bounded function,

an ANN is well behaved for all inputs.

...

...

...

y

3

(s)

...

...

...

...

...

...

... ...

...

x(s)

(b)

x

3

a

3,000

a

3,001

a

3,00Q

a

3,011

a

3,01Q

a

3,0QQ

a

3,112

a

3,1QQ

a

3,QQQ

a

3,111

x

3

z

−1

z

−1

z

−1 ..

..

x

3

~

~

y

1

(s)

x(s)

(a)

z

−1

z

−1

z

−1

a

1,0

a

1,1

a

1,2

a

1,Q

...

...

~

~

Figure 18. Implementation examples of ﬁrst- (a) and third- (b) order kernels of a general

polynomial ﬁlter.

2202 INTERMODULATION

A direct implementation of a dynamic ANN is shown in

Fig. 19. However, recognizing that (25a) constitutes a

biased linear FIR ﬁlter, whose bias is b

k

and impulse

response is w

k

(.), this dynamic ANN can also be imple-

mented as a set of parallel branches of the Wiener type, as

depicted in Fig. 20 [14].

Unfortunately, since all terms of the ANN are similar,

there is no way to identify relations between the system’s

output properties and any particular ANN terms.

Furthermore, as the model is now also nonlinear for the

w

k

(q) and b

k

parameters, the parameter extraction process

must rely on some form of optimization. This optimization

process, called ‘‘ANN training,’’ is known to give results

that are highly dependent on the input–output training

data. Moreover, there is no guarantee that the parameter

set found is unique or even optimum, which can constitute

a severe limitation to the model’s predictability.

2.4. A Glimpse of Nonlinear Simulation Algorithms for

Distortion Prediction

In circuit-level simulators [17], the mathematical repre-

sentation of the circuit is built by substituting each

electronic element with its constitutive relation [e.g., a

linear resistor can be represented as Ohm’s law, i ¼v/R; a

nonlinear resistor would be given by a voltage-controlled

current source, i(v); while a capacitor would be given by a

linear or nonlinear charge, q(v)] and then applying

Kirchhoff ’s current and voltage laws to the complete

circuit. This leads to a system of ordinary nonlinear

differential equations (ODEs) in time such as

i½yðtÞ þ

dq½yðtÞ

dt

¼xðtÞ ð26Þ

where x(t) and y(t) stand for the time-domain waveform of

the excitation and the state-variable vectors, respectively;

i[y(t)] represents memoryless linear or nonlinear ele-

ments, while q[y(t)] models memoryless linear or non-

linear charges (capacitors) or ﬂuxes (inductors). The

objective of the simulation is to ﬁnd the y(t) circuit

solution vector given a known x(t) input excitation.

On the other hand, system-level simulators are usually

implemented as either event-driven or envelope-driven

machines. In both cases the simulator treats the system in

the time domain, computing a set of time samples of the

information signal.

Event-driven machines operate at a very high logic

level, in which the information is simply a set of successive

logic states. They are, therefore, state ﬂow simulators,

without enough subsystem description detail to allow

distortion calculations.

Envelope-driven simulators operate with the analogue

complex envelope. Hence, they do not handle the true

bandpass RF blocks but simply their complex lowpass

equivalents. Nevertheless, since these blocks are still

nonlinear dynamic blocks, the lowpass equivalent system

mathematical representation will again be an ordinary

differential equation similar to (26) with the only differ-

ence that now both the excitation vector x(t) and the state

variable vector y(t) are, in general, complex entities.

So, except for the type of signals handled, an ODEsuch as

(26) can be used to represent bandpass RF circuits, bandpass

RF systems, or even complex lowpass equivalent systems.

2.4.1. Time-Domain Techniques. The most intuitive

way to solve (26) is to covert it into a difference equation

i½yðsÞ þ

q½yðsÞ Àq½yðs À1Þ

T

s

¼xðsÞ ð27aÞ

or

i½yðsÞT

s

þq½yðsÞ ¼xðsÞT

s

þq½yðs À1Þ ð27bÞ

in which T

s

is the sampling period, and then determine all

time samples of y(t), y(s), starting from a known initial

state y(0). Because we are integrating the nonlinear ODE

in a set of discretized timesteps, this is known as timestep

integration, and constitutes the basic approach adopted

in all time-domain circuit simulators (time-marching

machines) such as SPICE, or system simulators like

Simulink

1

.

z

−1

z

−1

z

−1

u

k

b

k

w

o

(k)

w

k

(q)

f (u

k

)

u

k

b

o

y

s

~

x

s

~

Figure 19. Implementation of a nonlinear dynamic artiﬁcial

neural network.

+

+

+

+

+

+

+ +

+

f [u

1

(s)] W

1

( )

u

1

(s)

[u

k

(s)]

u

k

(s)

f [u

K

(s)]

u

K

(s)

b

1

b

o

b

k

b

k

W

k

( )

W

k

( )

w

o

(1)

w

o

(k)

w

o

(k)

y(s)

~

~

~

~

~

~

~

f

x(s)

~

Figure 20. Alternative implementation of the model of Fig. 19, in

which the ANN is rebuilt as a parallel combination of several

biased linear ﬁlter/memoryless nonlinearity branches.

1

Simulink is a general-purpose system simulation package that is

supported by the Matlab scientiﬁc computation software plat-

form.

INTERMODULATION 2203

Although timestep integration is still the nonlinear

analysis method of wider acceptance, it suffers from

several disadvantages in the RF distortion circuit simula-

tion ﬁeld. First, since it was conceived to compute the

circuit’s transient response, while our interest normally

resides in the steady state, it becomes quite inefﬁcient as it

has to wait until all transients have vanished. Also, by

operating in the time domain, it cannot handle linear

elements having a frequency-domain description, such as

dispersive distributed transmission media. Finally, even if

that drawback is circumvented (e.g., by approximating

these elements by lumped networks of reduced order), the

necessity of operating in the time domain, while the input

and resulting signals are usually handled in frequency

domain, would end up in all difﬁculties associated with the

discrete Fourier transform (DFT), namely, spectral leak-

age when transforming quasiperiodic multitone signals.

Fortunately, some time-domain alternatives to the initial

timestep integration method, like the ‘‘shooting Newton’’

[17], can bypass the transient response, therefore obviat-

ing the waste of time needed to let it vanish.

Furthermore, time-domain methods beneﬁt from two

important advantages: (1) since they rely on the SPICE

simulator engine, they are well known and available in

many electronic design automation tools; and (2) as they

use time as a natural continuation parameter [17], they

are especially suitable for supporting strong nonlinear

regimes. Envelope-driven system-level simulators must

handle the information envelopes, which are aperiodic

by nature. So, timestep integration does not suffer from

the inefﬁciency attributed to the calculation of the periodic

steady-state response, becoming the obvious choice in

solving (26).

2.4.2. Frequency-Domain Techniques. Frequency-do-

main techniques no longer seek a set of time samples of

the circuit output or the state variables’ waveforms but a

spectral representation of them. In their most simple

form, they assume that both the steady state of the

excitation and the ODE solution are periodic in time, so

that they can be expanded in a truncated DFT of (2Kþ1)

frequency points. For example, the state variables vector

would be represented by

yðtÞ ¼

K

k¼ÀK

Yðko

0

Þe

jko

0

t

ð28Þ

Since, in the frequency domain, time-domain derivatives

are transformed into products by jo, substituting (28) into

(26) leads to

I½YðoÞ þjXQ½YðoÞ ¼XðoÞ ð29Þ

which is a nonlinear algebraic function in the DFT coefﬁ-

cients Y(ko

0

). The orthogonality between different fre-

quency components provided by the DFT determines

that, despite its appearance, this is not a single equation

but can be expanded in a set of (2Kþ1) equations, each of

these must be fulﬁlled for its harmonic component; in

other words, the LHS and RHS (left- and right-hand side)

components must be in equilibrium, which is why (29) is

known as the ‘‘harmonic-balance equation.’’

Since this harmonic-balance (HB) technique computes

the periodic steady state directly, it circumvents most of

the disadvantages attributed to time-marching techni-

ques. Its only drawbacks are that, depending on the

DFT, it can handle only moderate nonlinear regimes,

where the y(t) can be described by a relatively small

number of harmonics, and that it requires both the ex-

citation and the vector of state variables to be periodic. As

we have already seen in Section 1, the excitations used for

intermodulation distortion analysis are often of the two-

tone or multitone type. In general, the frequencies of these

tones do not constitute any harmonic set (they cannot be

made harmonics of a common fundamental), and the

corresponding waveform is aperiodic. (Such multitone

signals are actually said to be quasiperiodic waveforms.)

One way to circumvent this problem consists in imagining

that a multitone time-domain waveform is evolving, not in

the natural time t, but in a number of artiﬁcial timescales

equal to the number of nonharmonically related tones,

t

1

,y,t

Q

. For example, for a two-tone regime, the ODE in

time becomes a multirate partial-differential equation

(MPDE) in t

1

and t

2

:

i½yðt

1

; t

2

Þ þ

@q½yðt

1

; t

2

Þ

@t

1

þ

@q½yðt

1

; t

2

Þ

@t

2

¼xðt

1

; t

2

Þ ð30Þ

Since y(t

1

, t

2

) is now double-periodic in t

1

and t

2

, it admits

a bidimensional Fourier expansion

yðt

1

; t

2

Þ ¼

K

k

1

¼ÀK

K

k

2

¼ÀK

Yðk

1

o

1

; k

2

o

2

Þe

jðk

1

o

1

t

1

þk

2

o

2

t

2

Þ

ð31Þ

which, substituted in (30), results in a new bidimensional

HB equation. This is the technique known as the multi-

dimensional discrete Fourier transform harmonic-balance

(MDFT HB).

2.4.3. Time-Domain/Frequency-Domain Hybrid Techni-

ques. When the excitation is a RF carrier of frequency

o

c

, modulated by some independent baseband modulation

signal, like the one expressed in (5), it can be again

conceived as varying according to two independent time-

scales: one, t

1

, with fast evolution, for the carrier; and

another, t

2

, slower, for the modulation. So, the circuit can

again be described by a bidimensional MPDE such as (30).

If we now recognize that this regime is periodic for the

carrier but aperiodic for the modulation, we immediately

conclude that simulation efﬁciency would be maximized if

we treated the carrier evolution in t

1

in the frequency

domain, but kept the baseband evolution t

2

in time. This

supposes a solution in which the vector of state variables

is decomposed in a t

2

time-varying Fourier series

yðt

1

; t

2

Þ ¼

K

k ¼ÀK

Yðko

c

; t

2

Þe

jðkoct

1

Þ

ð32Þ

2204 INTERMODULATION

which, substituted in (30), leads to

I½Yðko

c

; t

2

Þ þjX

c

Q½Yðko

c

; t

2

Þ

þ

@Q½Yðko

c

; t

2

Þ

@t

2

¼Xðko

c

; t

2

Þ

ð33Þ

Solving (33) for the envelope, with a timestep integration

scheme, and for the carrier, with harmonic balance, leads

to the following recursive HB equation:

I½Yðko

c

; sÞT

s

þjX

c

Q½Yðko

c

; sÞT

s

þQ½Yðko

c

; sÞ

¼Xðko

c

; sÞT

s

þQ½Yðko

c

; s À1Þ

ð34Þ

By handling the RF signal components in the frequency

domain and the envelope in the time domain, (34) is

particularly appropriate to bridge the gap between circuit

and envelope-driven system simulation. In fact, we can

conceive of a simulator in which all except a few circuits of

a communication system are treated as system-level com-

plex equivalent lowpass behavioral input–output blocks—

for maximized computational efﬁciency—while the re-

maining circuits are treated at the RF bandpass circuit

level—for maximum accuracy.

2.4.4. Volterra Series. Although the Volterra series

method is not very widely used outside the intermodula-

tion prediction ﬁeld, it plays a determinant role for the

analysis and design of very-low-distortion circuits.

In comparison with the previously mentioned methods,

Volterra series no longer tries to ﬁnd a solution in an

iterative and numerical way, but seeks for an analytic

solution of a polynomial approximation of the original

circuit or system. In fact, it assumes that if the nonlinea-

rities of the original circuit or system can be decomposed

in a Taylor series around a certain ﬁxed quiescent point

iðyÞ ¼g

1

yþg

2

y

2

þg

3

y

3

ð35Þ

qðyÞ ¼c

1

yþc

2

y

2

þc

3

y

3

ð36Þ

then, the solution can be approximated by the following

functional series in the time domain:

yðtÞ ¼y

1

ðtÞ þy

2

ðtÞ þy

3

ðtÞ

¼

_

1

À1

h

1

ðtÞxðt ÀtÞdt

þ

_

1

À1

_

1

À1

h

2

ðt

1

; t

2

Þxðt Àt

1

Þxðt Àt

2

Þdt

1

dt

2

þ

_

1

À1

_

1

À1

_

1

À1

h

3

ðt

1

; t

2

; t

3

Þxðt Àt

1

Þ

Âxðt Àt

2

Þxðt Àt

3

Þ dt

1

dt

2

dt

3

ð37Þ

If the excitation can be expressed as a frequency-domain

sum of complex exponentials (possibly, but not necessarily,

harmonically related sinusoids)

xðtÞ ¼

Q

q ¼ÀQ

Xðo

q

Þe

joqt

ð38Þ

then we obtain a frequency-domain version of (37)

yðtÞ ¼

Q

q¼ÀQ

H

1

ðoÞXðo

q

Þe

joqt

þ

Q

q

1

¼ÀQ

Q

q

2

¼ÀQ

H

2

ðo

q

1

; o

q

2

ÞXðo

q

1

ÞXðo

q

2

Þe

jðoq

1

þoq

2

Þt

þ

Q

q

1

¼ÀQ

Q

q

2

¼ÀQ

Q

q

3

¼ÀQ

H

3

ðo

q

1

; o

q

2

; o

q

3

Þ

ÂXðo

q

1

ÞXðo

q

2

ÞXðo

q

3

Þe

jðo

q

1

þo

q

2

þo

q

3

Þt

ð39Þ

in which the h

n

(t

1

,y,t

n

) of (37) and the H

n

(o

1

,y,o

n

) of

(39) are the nth-order impulse responses and the nth-

order nonlinear transfer functions, respectively. Each of

these sets can be obtained from the other by the direct

application of a n-dimensional Fourier transform pair.

The Volterra series method consists in determining the

set of h

n

(.) or of H

n

(.) (as occurs with conventional linear

systems, the frequency-domain version is usually pre-

ferred), which then becomes a true nonlinear dynamic

model of the system. In fact, note that if one knows all the

H

n

(o

1

,y,o

n

) of a circuit or system, up to a certain order,

one immediately knows its response up to that order [from

(39)] to any multitone input represented by (38).

To show how these nonlinear transfer functions can be

determined, let us consider again the general circuit or

system described by the ODE of (26). Substituting (35),

(36), and (39) into (26), and assuming that the input is now

a ﬁrst-order elementary excitation of

xðtÞ ¼e

jot

ð40Þ

the orthogonality of the complex exponentials leads us to

the conclusion that H

1

(o) must be given by

H

1

ðoÞ ¼

1

g

1

þjoc

1

ð41Þ

In fact, this H

1

(o) is merely the usual transfer function of

the linear circuit or system obtained from a linearization

around the quiescent point.

To obtain the second-order nonlinear transfer function,

we would now assume that the system is excited by a

second-order elementary excitation of

xðtÞ ¼e

jo

1

t

þe

jo

2

t

ð42Þ

Substituting (35), (36), (39), and (42) into (26), and collect-

ing components in the second-order mixing product of

INTERMODULATION 2205

o

1

þo

2

would lead to

H

2

ðo

1

; o

2

Þ ¼ À

g

2

þjðo

1

þo

2

Þc

2

g

1

þjðo

1

þo

2

Þc

1

H

1

ðo

1

ÞH

1

ðo

2

Þ ð43Þ

Similarly, the calculation of the third-order nonlinear

transfer function assumes an input of

xðtÞ ¼e

jo

1

t

þe

jo

2

t

þe

jo

3

t

ð44Þ

and leads to

H

3

ðo

1

; o

2

; o

3

Þ ¼ À

2

3

g

2

þjðo

1

þo

2

þo

3

Þc

2

g

1

þjðo

1

þo

2

þo

3

Þc

1

Â½H

1

ðo

1

ÞH

2

ðo

2

; o

3

Þ þH

1

ðo

2

Þ

ÂH

2

ðo

1

; o

3

Þ þH

1

ðo

3

ÞH

2

ðo

1

; o

2

Þ

À

g

3

þjðo

1

þo

2

þo

3

Þc

3

g

1

þjðo

1

þo

2

þo

3

Þc

1

ÂH

1

ðo

1

ÞH

1

ðo

2

ÞH

1

ðo

3

Þ

ð45Þ

The terms À[g

2

þj(o

1

þo

2

)c

2

]H

1

(o

1

)H

1

(o

2

) in (43) and

the terms

À

2

3

½g

2

þjðo

1

þo

2

þo

3

Þc

2

Â½H

1

ðo

1

ÞH

2

ðo

2

; o

3

Þ þH

1

ðo

2

ÞH

2

ðo

1

; o

3

Þ

þH

1

ðo

3

ÞH

2

ðo

1

; o

2

Þ

and À½g

3

þjðo

1

þo

2

þo

3

Þc

3

H

1

ðo

1

ÞH

1

ðo

2

ÞH

1

ðo

3

Þ

in (45) are known as the elementary second-order and

third-order nonlinear sources, respectively. In fact, com-

paring (43) and (45) with (41), we immediately conclude

that for the calculation of ﬁrst-, second-, and third-order

solutions, what we have been doing was to always analyze

the same linearized version of the original ODE with the

appropriate elementary nonlinear sources at o for ﬁrst

order, at o

1

þo

2

for second order, and at o

1

þo

2

þo

3

for

third order. That is why the method of Volterra series

analysis is known to solve a forced nth-order nonlinear

dynamic problem, solving n times the same linear pro-

blem, with the appropriate (1st, 2nd,y,nth)-order forcing

functions, in a recursive way. As it is based on a poly-

nomial approximation of the nonlinearities, the Volterra

series is a local model restricted to small-signal levels, or

mildly nonlinear regimes. In practice, it can be used only

for calculating the distortion in nonsaturated mixers,

small-signal ampliﬁers, or nonsaturated power ampliﬁers,

that is, well below the 1-dB compression point. However,

because it is a closed-form nonlinear model, it provides

qualitative information on the nonlinear circuit or sys-

tem’s operation, giving, for instance, insight into the

physical origins of nonlinear distortion, and can be di-

rectly used for circuit and system’s design.

3. INTERMODULATION DISTORTION IN SMALL-SIGNAL

AMPLIFIERS

First, let us clarify the meaning of ‘‘small-signal ampli-

ﬁers,’’ as most of us would expect no appreciable nonlinear

distortion from these circuits. This term is used to distin-

guish low-noise or high-gain ampliﬁers from the essen-

tially different power ampliﬁers, treated in the next

section. While the small-signal ampliﬁers referred to

here always supposedly operate in highly backed-off class

A regimes, power ampliﬁers are operated close to satura-

tion, usually even in strongly nonlinear modes as class

AB, B, or C.

So, now, one question comes to our minds: ‘‘What are

the mechanisms capable of causing signiﬁcant nonlinear

distortion in small-signal ampliﬁers?’’ To advance with an

answer, let us consider the case of the low-noise ampliﬁer

of a wireless communication receiver front end. This is a

circuit that faces, beyond the very weak desired channel,

many other incoming channels present in the same com-

munication system’s band. For example, a low-noise front

end of a handset can be simultaneously excited by the

desired channel coming from a remote base station, and by

another channel coming from a nearby transmitter hand-

set. Since the ratio of distances between our receiver and

the base station, and our receiver and the perturbing

transmitter, can easily be on the order of several kilo-

meters to one meter, the ratio of incoming powers can be

higher than 10

8

to 1. Therefore, the signal-to-perturbation

ratio can be as poor as À70 or À80 dB, and, if it is true

that a desired signal of, say, À70 dBm cannot generate

any signiﬁcant nonlinear distortion, that is no longer the

case for the þ10 dBm perturbation.

Indeed, as seen in Section 1, this high-level perturba-

tion can produce nonlinear distortion sidebands falling

over the desired channel, cross-modulation, and desensi-

tization. These effects are illustrated in Fig. 21 and can

constitute a severe limitation to the performance of RF

front ends. In fact, they allow the deﬁnition of a very

important signal ﬁdelity ﬁgure of merit, the dynamic

range, which is the ratio between the amplitudes of the

highest and lowest incoming detectable signals that still

guarantee the speciﬁed signal-to-noise ratio SNR. The

lowest-amplitude detectable signal—the receiver’s sensi-

tivity S

i

—is the one that stands the desired SNR over the

noise ﬂoor. The highest-amplitude detectable signal, P

max

,

is deﬁned as the one that generates a nonlinear distortion

perturbation whose power equals the noise ﬂoor. So

DR

P

max

S

i

or DR

dB

P

max

dBm

ÀS

i

dBm

ð46Þ

where DR is the dynamic range.

Since signal excursions appearing at the nonlinear

active device are always kept much smaller than the

applied DC voltages and currents, the ampliﬁer can be

approximately described by a local model. For example,

considering the ideal (low-frequency) situation in which

the only nonlinear effects can be attributed to the

i

ds

(v

gs

,v

ds

) current of a FET, the ampliﬁer would be

described by the equivalent circuit depicted in Fig. 22,

2206 INTERMODULATION

while the nonlinearity would be represented by the Taylor

series of (19).

Although the model shown in Fig. 22 is very simpliﬁed,

it will already give us some insight onto the mechanisms

controlling IMD generation in these small-signal ampli-

ﬁers. For that, we will ﬁrst redraw this circuit as the one of

Fig. 23 over which a Volterra series analysis will then be

performed.

Note that ports 1 and 2 in Fig. 23 handle the ampliﬁer’s

input and output, respectively, but were deﬁned after the

terminal admittances Y

S

and Y

L

were incorporated into

the main circuit. port 3 serves to deﬁne v

gs

, one of the

control voltages of the i

ds

nonlinearity, and port 4 serves to

deﬁne v

ds

, the other control voltage. Furthermore, since

Fig. 23 is the linearized equivalent-circuit version of the

original circuit of Fig. 22, its only i

ds

(v

gs

,v

ds

) components

are the ﬁrst-order ones: G

m

v

gs

and G

ds

v

ds

. All the other

nonlinear terms of (19) will behave as nonlinear sources

that will be incorporated as independent current sources

applied to port 4 [6,7].

Assuming that the equivalent Norton current excita-

tion corresponds to a narrowband two-tone stimulus, this

circuit can be represented by the following [Z] matrix and

input and output boundary conditions:

V

1

ðoÞ

V

2

ðoÞ

V

3

ðoÞ

V

4

ðoÞ

_

¸

¸

¸

¸

¸

_

_

¸

¸

¸

¸

¸

_

¼

Z

11

Z

12

Z

13

Z

14

Z

21

Z

22

Z

23

Z

24

Z

31

Z

32

Z

33

Z

34

Z

41

Z

42

Z

43

Z

44

_

¸

¸

¸

¸

¸

_

_

¸

¸

¸

¸

¸

_

.

I

1

ðoÞ

I

2

ðoÞ

I

3

ðoÞ

I

4

ðoÞ

_

¸

¸

¸

¸

¸

_

_

¸

¸

¸

¸

¸

_

ð47aÞ

I

1

ðoÞ ¼I

s

ðoÞ : i

s

ðtÞ ¼I

s

1

cosðo

1

tÞ þI

s

2

cosðo

2

tÞ ð47bÞ

I

2

ðoÞ ¼0 ð47cÞ

I

3

ðoÞ ¼0 ð47dÞ

If the two-tones are closely separated in frequency, the

circuit reactances are similar for all in-band products. So,

using o

0

as the center frequency [o

0

¼(o

1

þo

2

)/2], Z

i-

ij

(o

1

)EZ

ij

(o

2

)EZ

ij

(2o

1

Ào

2

)EZ

ij

(2o

2

Ào

1

)EZ

ij

(o

0

) for

any i, j ¼1,2,3,4.

After determining the linear equivalent-circuit [Z]

matrix, the nonlinear currents’ method of Volterra series

analysis [6,7] proceeds by determining ﬁrst-order control

voltages V

3,1

(o

0

) and V

4,1

(o

0

) (q¼1,2) and ﬁrst-order

output voltage V

2,1

(o

0

), from the excitation I

s

(o

0

):

V

gs;1

ðo

1

Þ; V

gs;1

ðo

2

Þ : V

3;1

ðo

0

Þ ¼Z

31

ðo

0

Þ

I

S

ðo

0

Þ

2

ð48aÞ

V

ds;1

ðo

1

Þ; V

ds;1

ðo

2

Þ : V

4;1

ðo

0

Þ ¼Z

41

ðo

0

Þ

I

S

ðo

0

Þ

2

ð48bÞ

V

L;1

ðo

1

Þ; V

L;1

ðo

2

Þ : V

2;1

ðo

0

Þ ¼Z

21

ðo

0

Þ

I

S

ðo

0

Þ

2

ð49Þ

From these ﬁrst-order control variables, the second-order

nonlinear current of i

ds

at o

1

Ào

2

Do and o

1

þo

2

So,

SNR

DR

P

Max

S

i

N

i

S

xx

()

**Figure 21. Nonlinear distortion impairments in a
**

mildly nonlinear receiver system: illustration of the

concepts of receiver’s desensitization and dynamic range.

+

−

+

−

+

−

+

−

i

DS

(v

GS

,v

DS

)

v

S

(t )

v

DS

(t )

R

0

v

o

(t)

R

0

Z

L

( )

v

GS

(t )

V

DS

V

GS

Input

matching

network

Output

matching

network

Figure 22. Model of a mildly nonlinear ampliﬁer for

small-signal distortion studies.

INTERMODULATION 2207

I

4,2

(o) should now be determined:

I

4;2

ðDoÞ ¼ À½2G

m2

jZ

31

ðo

0

Þj

2

þG

md

Z

31

ðo

0

ÞZ

41

ðo

0

Þ

Ã

þG

md

Z

31

ðo

0

Þ

Ã

Z

41

ðo

0

Þ þ2Gd

2

jZ

41

ðo

0

Þj

2

jI

S

j

2

4

ð50Þ

I

4;2

ðSoÞ ¼ À2½G

m2

Z

31

ðo

0

Þ

2

þG

md

Z

31

ðo

0

ÞZ

41

ðo

0

Þ

þG

d2

Z

41

ðo

0

Þ

2

I

2

S

4

ð51Þ

Then, the linear circuit should be analyzed again for this

new second-order current source, determining the second-

order control voltages at the difference o

1

Ào

2

Do and

sum o

1

þo

2

So frequencies, V

3,2

(o) and V

4,2

(o):

V

3;2

ðDoÞ ¼Z

34

ðDoÞI

4;2

ðDoÞ ð52Þ

V

3;2

ðSoÞ ¼Z

33

ðSoÞI

3;2

ðSoÞ þZ

34

ðSoÞI

4;2

ðSoÞ ð53Þ

V

4;2

ðDoÞ ¼Z

44

ðDoÞI

4;2

ðDoÞ ð54Þ

V

4;2

ðSoÞ ¼Z

43

ðSoÞI

3;2

ðSoÞ þZ

44

ðSoÞI

4;2

ðSoÞ ð55Þ

The last step consists in calculating the third-order non-

linear current of i

ds

at 2o

1

Ào

2

, I

4,3

(2o

1

Ào

2

) from ﬁrst-

and second-order control voltages V

3,1

(o), V

3,2

(o), V

4,1

(o),

and V

4,2

(o):

I

4;3

ð2o

1

Ào

2

Þ ¼ À½2G

m2

Z

31

ðo

0

Þ

Ã

Z

34

ð2o

0

ÞI

4;2

ð2o

0

Þ

þG

md

Z

31

ðo

0

Þ

Ã

Z

44

ð2o

0

ÞI

4;2

ð2o

0

Þ

þG

md

Z

41

ðo

0

Þ

Ã

Z

34

ð2o

0

ÞI

4;2

ð2o

0

Þ

þ2G

d2

Z

41

ðo

0

Þ

Ã

Z

44

ð2o

0

ÞI

4;2

ð2o

0

Þ

I

Ã

S

2

À½2G

m2

Z

31

ðo

0

ÞZ

34

ðDoÞI

4;2

ðDoÞ

þG

md

Z

31

ðo

0

ÞZ

44

ðDoÞI

4;2

ðDoÞ

þG

md

Z

41

ðo

0

ÞZ

34

ðDoÞI

4;2

ðDoÞ

þ2G

d2

Z

41

ðo

0

ÞZ

44

ðDoÞI

4;2

ðDoÞ

I

S

2

À 3G

m3

½ Z

31

ðo

0

ÞjZ

31

ðo

0

Þj

2

þG

m2d

Z

31

ðo

0

Þ

2

Z

41

ðo

0

Þ

Ã

þ2G

m2d

jZ

31

ðo

0

Þj

2

Z

41

ðo

0

Þ

þG

md2

Z

31

ðo

0

Þ

Ã

Z

41

ðo

0

Þ

2

þ2G

md2

Z

31

ðo

0

ÞjZ

41

ðo

0

Þj

2

þ3G

d3

Z

41

ðo

0

ÞjZ

41

ðo

0

Þj

2

I

S

jI

S

j

2

8

ð56Þ

and then, ﬁnally, determine third-order output voltage at

the IMD frequency 2o

1

Ào

2

:

V

2;3

ð2o

1

Ào

2

Þ ¼Z

24

ðo

0

ÞI

4;3

ð2o

1

Ào

2

Þ ð57Þ

Now we are in a position to calculate the ampliﬁer’s

signal-to-IMD ratio (IMR) by ﬁrst determining output

power at the fundamental

P

L

¼

1

2

G

L

ðo

0

ÞjV

L

ðo

0

Þj

2

¼

1

2

G

L

ðo

0

ÞjZ

21

ðo

0

Þj

2

jI

S

j

2

ð58Þ

and IMD components

P

L

3

¼2G

L

ðo

0

ÞjV

2;3

ð2o

1

Ào

2

Þj

2

ð59Þ

where G

L

(o) is the real part of load admittance Y

L

(o). A

full expression for this IMR would be very complex. But,

under the assumptions that internal feedback is negligible

[Z

34

(o)E0] and that both second-harmonic and o

1

Ào

2

distortion will be very small (usually veriﬁed in small-

signal ampliﬁers), it can be approximated by

IMR %16

A

v

ðo

0

Þ

Z

D

ðo

0

Þ

¸

¸

¸

¸

¸

¸

¸

¸

2

.

j3G

m3

þG

m2d

A

v

ðo

0

Þ

Ã

þ2G

m2d

A

v

ðo

0

Þ þG

md2

A

v

ðo

0

Þ

2

þ2G

md2

jA

v

ðo

0

Þj

2

þ3G

d3

A

v

ðo

0

ÞjA

v

ðo

0

Þj

2

j

À2

jV

S

j

À4

ð60Þ

where Z

D

(o)1/[G

ds

þY

L

(o)], A

v

(o) is the intrinsic voltage

gain deﬁned by A

v

(o)V

ds

(o)/V

gs

(o), and V

S

is the voltage

amplitude of the signal source, V

S

(o) ¼Z

S

(o)I

S

(o).

If we now study the variation of various third-order

current components with V

GS

bias, as shown in Fig. 24 for

a typical general-purpose small-signal MESFET, we con-

clude that there are two very good points of IMD perfor-

mance, the so-called small-signal IMD ‘‘sweet spots.’’ The

ﬁrst one is located at the FET’s threshold voltage [6] and

thus has a very small associated power gain. The other is

located in high-V

GS

regions, and although it may not

correspond to very good noise ﬁgures, it is deﬁnitely useful

for designing high-gain, highly linear small-signal ampli-

ﬁers. Unfortunately, this latter small-signal IMD ‘‘sweet

spot’’ is a peculiarity of only some GaAs MESFETs, and

was never observed on HEMTs, MOSFETs or BJTs.

C

gs

C

gd

R

i

R

s

L

s

I

ds,1

V

gs

,

( ) V

ds

I

4

V

3

=V

gs

I

3

V

2

I

2

I

S

Y

S

I

1

V

1

V

4

=V

ds

Y

L

+

−

+

−

+

−

+

−

Figure 23. Linearized equivalent-circuit model of a FET-based

mildly nonlinear ampliﬁer used in Volterra series calculations.

2208 INTERMODULATION

Turning now our attention to the IMR variation with

source impedance, we can conclude that since P

L

can also

be given by P

L

¼

1

2

G

L

ðo

0

ÞjA

v

ðo

0

Þj

2

jV

S

j

2

, Eq. (60) conﬁrms

the empirical observation that, for constant output power,

third-order distortion in FET-based small-signal ampli-

ﬁers is almost independent of input termination Z

S

(o).

As far as the IMR variation with load impedance is

concerned, Fig. 24 and (60) indicate that, for typical V

GS

bias, the nonlinear current contributions of G

m3

and G

m2d

have effects on IMR that are important but, fortunately,

opposite in sign. Since NIG

m2d

is proportional to voltage

gain, and thus to Z

L

(o), this implies that a maximization

of voltage gain can also be beneﬁcial to IMR. This hypoth-

esis was indeed fully conﬁrmed by the measured and

simulated IMR

3

load-pull data [9], showing that and

optimum Z

L

(o) really exists and it tends to coincide with

the one that maximizes small-signal voltage and power

gains in MESFET-based small-signal ampliﬁers [6,9].

Since BJTs and HBTs have mildly nonlinear character-

istics that are completely different from those of FETs,

these results cannot be directly extrapolated to bipolar-

based small-signal ampliﬁers. For example, while the

most important nonlinearity source, i

DS

(v

GS

,v

DS

), is lo-

cated at the FET’s output, in bipolars it is manifested in

both the input, i

B

(v

BE

,v

CE

), and the output, i

C

(v

BE

,v

CE

) [6].

4. INTERMODULATION DISTORTION IN HIGH-POWER

AMPLIFIERS

Let us now turn our attention to power ampliﬁers (PAs).

Contrary to small-signal ampliﬁers where noise ﬁgure and

gain are of primary concern, power ampliﬁers are de-

signed for high output power P

out

and power-added efﬁ-

ciency (PAE).

In a well-designed PA, maximum output power is

determined by the loadline (load impedance termination)

and available output signal excursion. Power-added efﬁ-

ciency is dependent mostly on the PA operation class

(quiescent point) and on a convenient output voltage and

current waveform shaping, speciﬁcally, selection of har-

monic terminations. Therefore, it seems that little is left

for optimizing intermodulation distortion. Fortunately, as

we will see, that is not necessarily the case.

Since real devices do not present abrupt turnon points,

it is difﬁcult to precisely deﬁne the PA operation class. So,

to prevent any ambiguity in the following discussion, we

will ﬁrst deﬁne classes A, AB, B, and C. Taking into

account the discussion in Ref. 6, we will adopt the follow-

ing deﬁnitions: (1) the turnon bias V

T

is deﬁned as the

input quiescent point to which the turnon small-signal

IMD ‘‘sweet spot’’ corresponds (see Fig. 24); (2) biasing the

device below V

T

corresponds to class C (G

m3

40); (3)

biasing it exactly at V

T

corresponds to class B (G

m3

¼0);

and (4) biasing it above V

T

will determine the usual class

AB or class A (G

m3

o0).

The ﬁrst design step to be taken when designing a PA

is to decide whether precedence should be given to PAE

or to IMD specs, as they generally lead to opposite design

solutions. The traditional PA design rules state that a

PA optimized for IMD requires unsaturated class A opera-

tion; that is, the device should be biased and always

kept comfortably inside the linear ampliﬁcation zone

(saturation region of FETs and the active region of BJTs

or HBTs).

On the other hand, a PA optimized for PAE is usually

biased near class B or slightly into class C—that is, with a

quiescent point where output voltage is halfway between

knee voltage and breakdown, and output current is close

to turnon—and then is allowed to be driven into satura-

tion. This leads to saturated classes such as classes E and

F [18]. Unfortunately, as such operation classes achieve

their high efﬁciencies by operating the active device in an

almost switching mode, their associated nonlinear distor-

tion is also huge. In fact, recognizing that a switching

power ampliﬁer turns any waveform into a constant-

amplitude square wave, it is easy to conclude that those

class E or F PAs cannot be used when the amplitude of the

RF-modulated signal also carries information content

(modulation formats of non-constant-amplitude envelope).

The basic goal when designing linear PAs is to get class

B PAE with class A IMD—and, although this is seldom

possible, there are some particular PA features that

provide a means to escape from this apparent deadend.

One that has been receiving a great deal of attention is the

so-called large-signal IMD sweet spots [19]. Contrary to

their small-signal counterparts studied above, which were

associated to a particular quiescent point and found

effective only at very-small-signal levels, these are pecu-

liar points of the IMD–input power characteristic (see

Fig. 25) where only a few decibels of output-power backoff

(and thus a few percent of efﬁciency degradation) can lead

to astonishingly high levels of IMD reduction.

To understand how this curious effect takes place, we

need to abandon our small-signal local model, since, for the

signal levels where these large-signal IMD sweet spots are

observed, the Taylor expansion of (19) presents an unac-

ceptable error or simply may not converge. Instead, we are

–40

–20

0

20

40

60

80

NIG

m3

, NIG

m2d

, NIG

md2

, NIG

d3

, NI

d3

(mS/V

2

)

–2 –1.8 –1.6 –1.4 –1.2 –1 –0.8 –0.6 –0.4 –0.2 0

V

GS

(V)

–40

–20

0

20

40

60

80

Figure 24. Magnitude of total i

ds

(v

gs

,v

ds

) third-order current

components NI

d3

(—) and its various components due to G

m3

NIG

m3

(–K–), G

m2d

NIG

m2d

(– þ–), G

md2

NIG

md2

(Á Á Á), and G

d3

NIG

d3

(- - -) as a function of V

GS

bias.

INTERMODULATION 2209

forced to rely on qualitative solutions of approximated

global models. For that, we will transform the bidimen-

sional dependence of (19) on the input and output control

voltages, v

GS

(t) and v

DS

(t), into a one-dimensional model,

generating in this way an equivalent single-input/single-

output transfer function (TF), i

DS

[v

GS

(t)]. This assumes an

output boundary condition imposed by load impedance,

Z

L

(o), V

ds

(o) ¼V

DC

ÀZ

L

(o)I

ds

(o), where V

DC

is the applied

output quiescent voltage, beyond knowledge of the active

device nonlinear model i

DS

[v

GS

(t), v

DS

(t)].

In order to describe, with enough generality, the global

nonlinearities of the device, we will also consider that

turnon can be represented by an exponential of input

voltage. This method is commonly adopted to represent

subthreshold conduction in FETs, and is even more faith-

ful for describing i

C

(v

BE

) in common-base or common-

emitter bipolar devices. Then, for increasing v

GS

voltages,

it is assumed that the FET passes through a nearly

quadratic zone, which, because of short-channel effects,

tends to become linear for even higher v

GS

. This i

DS

(v

GS

)

behavior was shown to be well reproduced by the following

empirical expression [20]

i

DS

ðv

GS

Þ ¼b

½smtðv

GS

Þ

2

1 þy smtðv

GS

Þ

ð61aÞ

where smt(v

GS

) is a smooth turnon function of v

GS

given by

smtðv

GS

Þ ¼K

V

ln 1þe

v

GS

K

V

_ _

ð61bÞ

and b, y and K

V

are empirical scaling parameters.

If we now take the output boundary into account,

i

DS

(v

DS

) will be almost unchanged unless v

GS

is so high

that R

L

Á i

DS

becomes close to V

DC

. There, v

DS

is so small

that the FET enters the triode region (the saturation

region for a bipolar based PA). v

GS

rapidly looses control

over i

DS

, and the TF saturates (the PA enters into strong

compression). So, the global transfer characteristic

i

DS

(v

GS

) presents a distinct sigmoid shape.

Assuming again a two-tone stimulus, several qualita-

tive conclusions may be drawn for large-signal operation.

One of the most important is that when the ampliﬁer is

driven into harder and harder nonlinear regimes, its gain

goes into compression, which means that the phase of the

in-band nonlinear distortion components must oppose

those of the fundamentals. So, PA energy balance con-

siderations derived in Section 1 show that the large-signal

asymptotic phase value of the IMD sidebands, at 2o

1

Ào

2

and 2o

2

Ào

1

, must tend to a constant value of 1801 [19].

On the other hand, we also know that small-signal IMD

phase is determined by the sign of the TF local derivatives,

determined by the active device’s quiescent point. As seen

above, G

m3

is positive below V

T

(class C operation), is null

exactly at V

T

(class B) and is negative above V

T

(classes

AB and A). So, since small-signal IMD sign can be made

positive, and tends to negative values in the large-signal

asymptotic regime, the Bolzano–Weierstrass theorem

guarantees the existence of at least one IMD null some-

where in between. This will be observed as a more or less

pronounced notch in an IMD–P

in

plot, constituting a large-

signal IMD sweet spot.

From these general conclusions it is clear that the

existence of a large-signal IMD sweet spot depends on

the small-signal IMD phase and on the physical effects

that determine large-signal gain compression. So, each

operation class will have its own particular IMD behavior.

Under class C, V

GS

oV

T

, G

m3

40, the PA presents gain

expansion and a high IMD level with 01 phase. When the

signal excursion reaches a strong nonlinearity as the

gate–channel junction conduction, gate–channel break-

down, or, more likely, the saturation–triode region transi-

tion, the PA enters into compression and an IMD notch is

observed. So, a large-signal IMD sweet spot should be

expected for class C ampliﬁers when the signal excursion

is at the onset of saturation, not far from the PA’s 1-dB

compression point. Although the PAE is not yet at its

maximum, it may present an interesting value.

In class A, V

GS

4V

T

, G

m3

o0, the PA starts at small

signal with almost unnoticeable gain compression and a

very small level of IMD with 1801 phase. As this phase is

maintained when the device enters strong compression,

no IMD sweet spot will be generated. Thus, and unless the

PA is biased above the small-signal IMD sweet spot found

for high V

GS

bias in certain MESFETs [9], no large-signal

IMD sweet spot will be observed. On the contrary, a

sudden increase of IMD power at the on-set of saturation

is the usual outcome of class A PAs.

In class AB, where V

GS

is only slightly higher than V

T

and G

m3

o0, the PA again shows a very shallow gain

compression and a low-level IMD of 1801 phase. Hence,

similar to what was concluded for class A operation, no

IMD sweet spot should be expected. Nevertheless, depend-

ing on the abruptness of turnon and succeeding lineariza-

tion of the TF characteristic, it can be shown that a

transition from 1801 to 01 IMD phase can occur at lower

values of output power [20,21] generating an IMD sweet

spot. At this stage, the circuit begins to behave as a class

C PA, with 01 IMD phase and gain expansion. Conse-

quently, a new IMD sweet spot will have to occur at large

signal when gain compression will ﬁnally take place. In

–80

V

GS

= –0.35V

V

GS

= –0.41V

V

GS

= –0.53V

–90

–70

–60

–50

–40

–30

–20

–10

–15 –10 –5 0 5

P

in

(dBm)

P

nout

(dBm)

Figure 25. Different IMD versus P

in

patterns showing large-

signal IMD ‘‘sweet spots’’ for a HEMT device.

2210 INTERMODULATION

summary, depending on the actual device’s transfer char-

acteristic and on the adopted quiescent point, class AB may

be signiﬁcantly different from class A in that it may even

present two IMD sweet spots, one for small to moderate

levels of P

in

and another for the onset of saturation.

When the device is biased for class B (i.e., V

GS

¼V

T

and

G

m3

¼0), there is no small-signal IMD to be compensated

by the large-signal distortion behavior. The PA presents

very low levels of small-signal distortion (remember that it

was biased at a small-signal IMD sweet spot) and then

presents a sudden rise of distortion power at the onset

of saturation. To illustrate the results of this analysis,

Fig. 26 shows three IMR–P

in

patterns typically observed

in MOSFET-based PAs biased for classes A, AB, and C,

respectively.

To close this qualitative analysis, let us draw some

conclusions about the dependence of large-signal IMD

sweet spots on impedance terminations. Starting with

source impedance, it is intuitive to realize that, because

the i

DS

(v

GS

,v

DS

) nonlinearity is located at the output, the

large-signal IMD behavior will be mostly invariant with

Z

S

(o), as was the case studied earlier for small-signal

ampliﬁers. Note that this conclusion may not be extra-

polated to bipolar based ampliﬁers, in which there is an

input nonlinearity due to the base–emitter junction, and

an output nonlinearity due to the active-to-saturation

region transition [6,22].

As seen from the small-signal Volterra series analysis

above, the dependence of i

DS

on v

DS

should also produce its

own impact on the large-signal IMD sweet spot, via Z

L

(o).

In fact, since these sweet spots were related to the output

signal excursion that crosses the saturation-to-triode re-

gion transition, and as the loadline slope determines that

signal level (see Fig. 27), it should be expected that the P

in

for which the IMD sweet spot is observed will be strongly

dependent on load termination. This is illustrated in Figs.

27a and 27b, where a shift of the simulated IMD sweet-

spot position is evident when loadline slope 1/R

L

is varied.

Furthermore, if the PA output is not perfectly matched,

the intrinsic load impedance actually presented to the

nonlinear current source may have a certain nonnull

phase. The output-induced large-signal distortion compo-

nents will no longer be in exact opposite phase with the

small-signal ones, and the previously observed large-

signal IMD sweet spots cease to be sharp dips of IMD

power to become more or less smooth valleys.

Further conclusions can also be drawn about the im-

pact of out-of-band terminations on the large-signal IMD

sweet spots [6,22]. As was seen above from the small-

signal analysis, the presence of even-order mixing pro-

ducts—which, as we have already seen, can be remixed

with the fundamentals—will generate new odd-order pro-

ducts. But, contrary to the small-signal case in which it

was assumed that the quasilinear operation of the ampli-

ﬁer would determine a minor effect to these indirect odd-

order products, that is no longer valid for a PA, and its

analysis becomes again much more complex:

1. Efﬁciency considerations may have previously

dictated a certain second-harmonic termination.

Further, if in most usual situations we seek a

squared output voltage waveform, that is, without

even-order harmonics, there are situations (e.g., the

I

Max

V

K

V

DC

V

BR

= V

Max

v

DS

i

DS

(v

GS

,v

DS

) P

IMD

(dBm)

Q

P

in

(dBm)

−180

−40 −30 −20 −10 0 10 20

−160

−140

−120

−100

−80

−60

−40

−20

0

20

(a) (b)

R

L

R

L

Q

R

L

Figure 27. Impact of PA loadline slope 1/R

L

on large-signal IMD ‘‘sweet spots’’.

IMR (dBc)

70

60

50

40

30

20

10

–10 –5 0 5 10 15 20

P

out

(dBm)

Class C

Class AB

Class A

Figure 26. IMR versus P

out

power plots of typical MOSFET-

based PAs at the three operation classes studied: C ( Á Á Á Á ), AB

(- - -) and A (—).

INTERMODULATION 2211

so-called inverse class F [18,23]) in which those even

harmonics are indeed maximized.

2. If, in small-signal ampliﬁers, there would be no

difﬁculty in designing bias networks presenting a

very low impedance to the modulation baseband

(o

2

Ào

1

), in PAs that is again incomparably more

difﬁcult. Indeed, as output currents may be on the

order of several amperes, any parasitic resistance or

inductance may immediately develop a nonnegligi-

ble output voltage.

3. There will be even additional, contributing base-

band reactances in PAs from more or less unex-

pected physical origins. That is the case of trap-

induced low-frequency dispersion presented by some

microwave active devices [24], and dynamic self-

heating, common to almost all PAs [25].

Depending on the phase of the out-of-band terminations,

these new indirect odd-order products may have a phase

that either reinforces or reduces the directly generated

IMD. As far as the even-harmonics-induced products are

concerned, since the modulation bandwidth (or the two-

tone separation Do) is usually much smaller than the PA

bandwidth, it may be assumed that Z

L

(2o

1

)EZ

L

(2o

2

)E

Z

L

(2o

0

). So, no important IMD behavior variation within

the bandwidth should be expected; that is, the indirect

odd-order distortion products may reduce, reinforce, or be

in quadrature with the direct ones, but their impact will

be the same along the whole modulation bandwidth.

The situation regarding the baseband-induced pro-

ducts is completely different. Now, Z

L

(Do) may vary

signiﬁcantly within the modulation bandwidth, especially

if the bias networks present resonances. Therefore, it is

likely that IMD power will vary within that bandwidth,

and the ampliﬁer will show (undesirable) long-term mem-

ory effects. Moreover, the complex conjugate symmetry of

load impedance requires that the imaginary part of

Z

L

(o

2

Ào

1

) have a sign opposite that of Z

L

(o

1

Ào

2

). So,

if some other odd-order products (e.g., the ones due to the

presence of second harmonics) also have signiﬁcant ima-

ginary parts, their addition will even produce asymmetric

IMD sidebands [22].

These strange IMD effects have received a lot of atten-

tion more recently as their induced long-term memory

immensely complicates the design of PA linearizers. For-

tunately, since direct static IMD usually dominates this

indirect dynamic distortion, those long-term memory ef-

fects are seldom noticed. They would be evident only if the

direct static odd-order products were reduced. Unfortu-

nately, IMD sweet spots are, by nature, exactly one of

these situations, and so the selection of these out-of-band

impedances should not be overlooked during the PA de-

sign and implementation phases.

5. INTERMODULATION DISTORTION IN MICROWAVE

AND RF MIXERS

A mixer can be viewed as a special kind of ampliﬁer in

which the bias supply no longer provides a constant

voltage or current, but one that varies in time—the local

oscillator. In the same way, an ampliﬁer is a device where

the constant quiescent point is perturbed by a certain

dynamic signal; a mixer is a similar device where the

local-oscillator (LO) time-varying ‘‘quiescent point’’ is

perturbed by a dynamic radiofrequency (RF) excitation.

Assuming that the mixer is operated in an unsaturated

mode, as is the case of most practical interest, the RF

signal level is much smaller than the LO level, and the

mixer can be analyzed, for the RF signal, as a mild

nonlinearity. Thus, it admits a low-degree polynomial

expansion in the vicinity of the time-varying LO quiescent

point. That constitutes the standard large-signal/small-

signal analysis of mixers [7,26]. Mixer distortion analysis

can thus follow exactly the one already carried out for

small-signal ampliﬁers, with the exception that now we

must start by determining the strong nonlinear regime

imposed by the LO and, eventually, some DC bias. The

voltage and current waveforms calculated in this way

constitute the time-varying quiescent point. Despite the

sinusoidal form of the LO excitation, the device’s strong

nonlinearities will determine a periodic regime composed

by the LO frequency o

LO

and its harmonics. So, referring

to the illustrative case of the active FET mixer depicted in

Fig. 28, the time-varying quiescent voltages that control

the FET’s i

DS

(v

GS

,v

DS

) nonlinearity will be given by

v

GS

ðtÞ ¼

K

k ¼ÀK

V

gs

ðko

LO

Þe

jko

LO

t

ð62aÞ

v

DS

ðtÞ ¼

K

k¼ÀK

V

ds

ðko

LO

Þe

jko

LO

t

ð62bÞ

Then, the nonlinearity must be approximated by a local

polynomial model. For instance, a Taylor series such as

v

RF

(t )

v

IF

(t )

v

LO

(t )

V

GS

RF/LO

Diplexer

+ −

V

DS

+ −

+

−

+

−

Figure 28. Simpliﬁed schematic of the active

FET mixer used in the mixer distortion analysis.

2212 INTERMODULATION

(19), in the vicinity of this time-varying LO quiescent

point, [v

GS

(t), v

DS

(t)], where the small-signal component,

i

ds

(v

gs

,v

ds

), is determined by the small-signal RF excita-

tion. Since the coefﬁcients of such Taylor series depend on

the control voltages v

GS

(t) and v

DS

(t), they will also be

time-variant:

i

ds

ðv

gs

; v

ds

Þ ¼G

m

ðtÞv

gs

ðtÞ þG

ds

ðtÞv

ds

ðtÞ

þG

m2

ðtÞv

gs

ðtÞ

2

þG

md

ðtÞv

gs

ðtÞv

ds

ðtÞ

þG

d2

ðtÞv

ds

ðtÞ

2

þG

m3

ðtÞv

gs

ðtÞ

3

þG

m2d

ðtÞv

gs

ðtÞ

2

v

ds

ðtÞ þG

md2

ðtÞv

gs

ðtÞv

ds

ðtÞ

2

þG

d3

ðtÞv

ds

ðtÞ

3

ð63Þ

As v

GS

(t) and v

DS

(t) are periodic, the coefﬁcients of (63) are

again periodic obeying a Fourier expansion of the form

gðtÞ ¼

K

k ¼ÀK

Gðko

LO

Þe

jko

LO

t

ð64Þ

Assuming that the RF signal is a two-tone signal

v

RF

ðtÞ ¼V

RF

cosðo

1

tÞ þV

RF

cosðo

2

tÞ ð65Þ

(in which we consider, without any lack of generality, that

o

1

oo

2

oo

LO

), the small-signal components of v

GS

(t),

v

DS

(t) Àv

gs

(t), v

ds

(t) À are again two-tone signals. Substi-

tuting (64) and (65) in (63) determines a small-signal

current i

ds

(t), whose components obey ko

LO

Æm

1

o

RF

1

Æm

2

o

RF

2

(k is any integer number and m

1

,m

2

A{ À3, À2,

À1,0,1,2,3}, |m

1

|þ|m

2

|r3) and thus include the input

tone frequencies at o

RF

1;2

(k¼0), the intermediate fre-

quencies IF at o

IF

1;2

¼o

LO

Ào

RF

1;2

(k ¼1), its second and

third harmonics at 2o

IF

1;2

¼2o

LO

À2o

RF

1;2

(k ¼2), and

3o

IF

1;2

¼3o

LO

À3o

RF

1;2

(k ¼3), respectively, and second-

and third-order intermodulation products at o

IF

2D

¼o

IF

2

À

o

IF

1

¼o

RF

2

Ào

RF

1

(k ¼0) and o

IF

3

¼2o

IF

2

Ào

IF

1

¼o

LO

À

ð2o

RF

2

Ào

RF

1

Þ (k ¼1), respectively.

One surprising conclusion that may be drawn from this

analysis is that, contrary to an ampliﬁer in which a single

Taylor coefﬁcient determines both nth-order harmonics

and intermodulation products, in a mixer, for example, the

baseband second-order products are determined by the

DC component of the Fourier expansion of a coefﬁcient

while the second harmonic is determined by the compo-

nent at 2o

LO

. Similarly, it is the o

LO

Fourier component

that determines the in-band third-order products, while

the third harmonic is controlled by the Fourier component

at 3o

LO

. Therefore, contrary to what happens in a mem-

oryless ampliﬁer, the behavior of the harmonics of a

memoryless mixer may say nothing about the behavior

of the corresponding in-band distortion products. A de-

tailed analysis of the distortion arising in a mixer is quite

laborious and requires a full small-signal/large-signal

analysis using the conversion matrix formalism [6,7,26].

However, some qualitative insight can already be obtained

if we consider the ideal situation of a unilateral gate mixer

(total absence of feedback) where the input is tuned for

o

RF

and o

LO

and the output is tuned for o

IF

. v

gs

(t) will

have only o

RF

components, while v

ds

(t) will have only the

resulting o

IF

components and its in-band distortion pro-

ducts o

IF

3

.

In such an ideal case the FET’s i

ds

(t) current component

at the IF fundamental frequencies will be given by

I

ds

ðo

IF

Þ %G

m1

V

gs

ðÀo

IM

Þ

þG

mÀ1

V

gs

ðo

RF

Þ þG

ds

0

V

ds

ðo

IF

Þ

ð66Þ

where G

m

k

and G

ds

k

stand for the kth-order harmonic of

the Fourier expansion of G

m

(t) and G

ds

(t), as expressed by

(64), and V

gs

(o) and V

ds

(o) represent the v

gs

(t) and v

ds

(t)

components at o. o

IM

is the so-called image frequency.

Because it is symmetrically located near the RF compo-

nents, taking o

LO

as the symmetry axis (since, in the

present case, o

RF

¼o

LO

Ào

IF

, then o

IM

¼o

LO

þo

IF

), it

will be also converted to the IF output, thus constituting

additive interference.

If now the third-order intermodulation product compo-

nents of i

ds

(t), I

ds

ðo

IF

3

Þ, were calculated, we would have

I

ds3

ðo

IF

3

Þ % G

m3À1

V

ð3Þ

gs

ð2o

RF

1

Ào

RF

2

Þ ð67Þ

in which V

ð3Þ

gs

ð2o

RF

1

Ào

RF

2

Þ stands for the terms at o

IF

3

that result from the frequency-domain convolutions of

V

gs

(o)

*

V

gs

(o)

*

V

gs

(o) or the time-domain products of

v

gs

(t)

3

. Expressions (66) and (67) show that a mixer

designed for high linearity, namely, one in which conver-

sion gain is maximized and IMD is minimized, requires a

(V

GS

,V

DS

) bias point and a LO drive level that maximize

ﬁrst-order Fourier component of the time-varying trans-

conductance G

m

(t) and minimize ﬁrst-order Fourier com-

ponent of G

m3

(t). Unfortunately, these are conﬂicting

requirements since maximizing G

m1

or G

mÀ1

means

searching for a G

m

(t) waveform of highest amplitude and

odd symmetry, while reducing G

m3 À1

implies reducing

G

m3

(t) swing and a G

m3

(t) waveform of even symmetry,

which, as we will see next, cannot be accomplished

simultaneously. So, a compromise should be sought in

terms of conversion gain and linearity optimization.

To illustrate this simpliﬁed analysis, Fig. 29a shows

three G

m

(t) waveforms for three distinct V

GS

bias points,

Fig. 29b shows the corresponding G

m3

(t) waveforms, and,

ﬁnally, Fig. 29c shows the resulting conversion gain

and IMD ratio ½I

ds

ðo

IF

Þ=I

ds

ðo

IF

3

Þ for the whole range of

V

GS

bias.

As stated above, there is indeed a compromise between

linearity and conversion gain. Although very high IMR

values can be obtained for particular bias points, none of

them coincides with the zone of highest conversion gain.

In a typical sigmoidal G

m

(v

GS

) (such that depicted in Fig.

14a), conversion efﬁciency is optimized when the FET is

biased for maximum G

m

(v

GS

) variation, that is, for the

G

m2

(v

GS

) peak. Unfortunately, that maximized variation

of G

m

(v

GS

) is accompanied by an also nearly ideal odd

symmetry of G

m3

(v

GS

), which is responsible for the

INTERMODULATION 2213

observed IMD impairment. Furthermore, this simpliﬁed

analysis also shows that, as was previously studied for

ampliﬁers, IMD behavior of mixers strongly depends on

the actual shape of the device’s nonlinearity. (For example,

the very sharp peaks of IMR shown in Fig. 29c are due to

the ideal symmetric sigmoidal model used for the simu-

lated FET’s transconductance.) So, different devices will

show quite distinct IMR patterns, impeding a straightfor-

ward extrapolation of these active FET mixer results to

diode mixers [6,27] or even resistive FET mixers [28,29].

6. CONCLUSIONS

This article showed that the study of nonlinear distortion

mechanisms is a subject of fundamental interest that

spreads through almost all microwave and RF signal

processing circuits and systems. Involving various scien-

tiﬁc disciplines that range from the physical level of the

active-device modeling, to the circuit and system’s level of

communication links, it requires a broad range of micro-

wave knowledge. Hence, and despite the now more than

40 years of continued progress, intermodulation distortion

is still an exciting and challenging ﬁeld of strong active

research both in industry and academia.

Acknowledgements

The author would like to express his gratitude to several

of his colleagues and graduate students who contributed

with some of the knowledge presented in this article. Of

Time (ps) Time (ps)

IMR (dB)

V

GS

(V)

Conversion gain (dB)

G

m

(t ) (mS)

(a) (b)

(c)

G

m3

(t) (mS/ V

2

)

10

0 100 200 300 400 500 600 700 800 900 1000

0

20

30

40

50

60

0 100 200 300 400 500 600 700 800 900 1000

25

20

−25

−20

−15

−10

−10

−20

−30

−40

−2.5 −2.0 −1.5 −1.0 −0.5

−5

15

10

5

0

10

20 100

90

80

70

60

50

40

30

0

Figure 29. (a) Time-domain waveforms of G

m

(t) for three different V

GS

bias points: V

GS

¼ À1.5 V

(- - -), V

GS

¼ À1.0 V (—), and V

GS

¼ À0.6V (Á Á Á); (b) corresponding time-domain waveforms of

G

m3

(t) for the same three V

GS

bias points; (c) conversion gain (—) and IMD ratio (– þ–) for the

whole range of the FET’s V

GS

bias.

2214 INTERMODULATION

these, Nuno B. Carvalho, Jose A. Garcia, and Christian

Fager deserve a special mention.

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INTERMODULATION MEASUREMENT

MUHAMMAD TAHER

ABUELMA’ATTI

King Fahd University of

Petroleum and Minerals

Dhahran, Saudi Arabia

1. INTRODUCTION

Virtually all electronic circuits and systems exhibit non-

linear input–output transfer characteristic. Mixers, fre-

quency multipliers, modulators, and square-law detectors

represent examples of intentional class members, while

linear power ampliﬁers, active ﬁlters, and microwave

transmitters, in which nonlinearity represents an unde-

sirable deviation of the system from ideal, linear opera-

tion, are examples of unintentional members.

Whenever a number of signals of differing frequencies

pass through a nonlinear device, energy is transferred to

frequencies that are sums and differences of the original

frequencies. These are the intermodulation products

(IMPs). In such cases, the instantaneous level of one sig-

nal may effectively modulate the level of another signal;

INTERMODULATION MEASUREMENT 2215

hence the term intermodulation. In a transmitting

system, the results of excessive intermodulation are

unwanted signals that may cause interference. In a

receiver, internally generated intermodulation can hinder

reception of the desired signals. It is interesting to note

that the ear’s cochlea has a similar nonlinear response

and produces sums and differences of the input frequen-

cies in the same way, particularly with loud sounds [1].

It has also been found that passive components, nor-

mally considered to be linear, can also generate IMPs. A

variety of situations can arise in which nonlinear resis-

tance junctions can be formed at metallic mating surfaces.

Such junctions may result from salt or chemical deposi-

tions or from corrosion. The result is sometimes known as

the ‘‘rusty bolt effect’’ because rusted bolts in structures

have been known to exhibit such nonlinearities. This phe-

nomenon is referred to as passive intermodulation (PIM).

Sources of PIM include waveguides, directional couplers,

duplexers, and antennas [2–6].

Intermodulation may also occur at the ampliﬁer–loud-

speaker interface [7], or in general as a result of the non-

linear interaction between the input signal of a two-port

and a signal injected to the output port and propagating

into the input via a feedback network [8]. Externally

induced transmitter intermodulation, also known as

reverse intermodulation, backward intermodulation, and

antenna-induced intermodulation, is the mixing of a

carrier frequency with one or more interfering signals in

a transmitter’s ﬁnal stage [9]. Moreover, lack of screening

of open-wire transmission lines can result in significant

coupling to adjacent lines frequently giving rise to inter-

modulation products [10]. Furthermore, intermodulation

may arise when an array of receiving antennas is illumi-

nated with a transient impulsive electromagnetic plane

wave [11].

In discussing the sources of IMPs it is convenient to

divide nonlinear mechanisms yielding IMPs into two prin-

cipal forms. The ﬁrst is due to a nonlinear amplitude in-

put/output characteristic (AM/AM), which causes

amplitude compression with increasing input amplitude.

The second mechanism occurs because of the variation of

phase shift through the device, or the system, as the input

amplitude is changed (AM/PM).

Depending on the signal characteristics, sources of

IMPs can be divided into two categories: (1) static nonlin-

earity, depending solely on the amplitude of the signal,

and (2) dynamic nonlinearity, depending not only on the

amplitude but also on the time properties or frequency

composition of the signal.

Static nonlinearities usually encountered in electronic

circuits and systems can be classiﬁed into clipping, cross-

over, and soft nonlinearities [12] as shown in Fig. 1.

Among the hard nonlinearities of clipping (which is sig-

nificant near maximum input amplitudes) and crossover

(significant mostly at small input amplitudes), the soft

nonlinearity is usually the most important in the transfer

characteristic of an electronic circuit. If the frequency con-

tent or the time properties of the input signal affect the

transfer characteristic of the circuit or the system, the re-

sulting nonlinearities may be called dynamic. Intermodu-

lation products resulting from dynamic nonlinearities are

referred to as transient intermodulation (TIM), slew-in-

duced distortion (SID), or dynamic intermodulation dis-

tortion (DIM) [13–16].

2. SIMPLE INTERMODULATION THEORY

IMPs occur when two or more signals exist simultaneously

in a nonlinear environment. In general, if N signals with

frequencies f

1

to f

N

are combined in a static nonlinearity,

the output will contain spectral components at frequencies

given by

N

n¼1

k

n

f

n

where k

n

is a positive integer, a negative integer, or zero,

and

N

n¼1

jk

n

j is the order of the IMP. Even with a small

number of input signals N, a very large number of IMPs

are generated. Fortunately, not all products are equally

troublesome. Depending on the system involved, some of

these IMPs can be neglected since they will be ﬁltered out

at some point. For example, most of the communication

systems operate over a limited frequency band. Thus,

IMPs falling out of the band will be attenuated. Moreover,

amplitudes of the IMPs generally decrease with the order

of the products, and high-order products can often be ne-

glected. Low-order intermodulation components such as

the second-order component f

m

Àf

n

and f

m

þf

n

and the

third-order components occurring at frequencies 2f

m

Àf

n

and f

m

þf

n

Àf

q

are usually the most troublesome, having

the largest magnitudes and/or lying close to the originat-

ing frequencies, making their removal by ﬁltering practi-

cally difﬁcult. However, a salient characteristic of PIM, as

distinguished from the conventional IM counterpart, dis-

cussed above, is that the PIMs causing trouble are of a

high order, say, 11th–21st.

Analysis of nonlinear systems differs from that of linear

systems in several respects: (1) there is no single analyt-

ical approach that is generally applicable (such as Fourier

a

a

b

b

c

c

Input

Output

Figure 1. Different types of static nonlinearities: (a) clipping;

(b) soft; (c) crossover.

2216 INTERMODULATION MEASUREMENT

or Laplace transforms in linear systems); (2) closed-form

analytical solutions of nonlinear equations are not ordi-

narily possible; and (3) there is rarely sufﬁcient informa-

tion available to enable a set of equations that accurately

model the system to be derived. These factors preclude the

exact analytical determination of nonlinear effects, such

as IMPs, in the general case. In order to get anything done

at all, it is usually necessary to make various simplifying

assumptions and then use an approximate model that will

provide results of acceptable accuracy for the problem in

hand.

A simple approach, therefore, is to use frequency-do-

main techniques that provide a separate solution for each

frequency present in the output. In general, such methods

are (1) centered around a description of the nonlinear

mechanism by a continuous function type of characteris-

tic, for example, a polynomial or a Fourier series repre-

sentation of the output in terms of the input; and (2) based

on the simplifying assumption that this characteristic

does not vary with frequency, in other words, that it is a

memoryless characteristic.

Memoryless nonlinear circuits are oftenly modeled

with a power series of the form

V

out

¼

N

n¼0

k

n

V

n

i

ð1Þ

The ﬁrst coefﬁcient, k

0

, represents the DC offset in the

circuit. The second coefﬁcient, k

1

, is the gain of the circuit

associated with linear circuit theory. The remaining coef-

ﬁcients, k

2

and above, represent the nonlinear behavior of

the circuit. If the circuit were completely linear, all the

coefﬁcients except k

1

would be zero.

The model can be simpliﬁed by ignoring the terms that

come after the k

3

term. For soft nonlinearities, the size of

k

n

decreases rapidly as n gets larger. For many applica-

tions the reduced model of Eq. (2) is sufﬁcient, since the

second-order and third-order effects dominate. However,

many devices, circuits, and systems present difﬁculties for

the polynomial approximation:

V

out

¼k

0

þk

1

V

i

þk

2

V

2

i

þk

3

V

3

i

ð2Þ

Assuming that the input signal is a two-tone of the form

V

i

¼V

1

cos o

1

t þV

2

cos o

2

t ð3Þ

then combining Eqs. (2) and (3), yields

V

out

¼a

0

þb

1

cos o

1

t þc

1

cos o

2

t þb

2

cos 2o

1

t

þc

2

cos 2o

2

t þb

3

cosðo

1

þo

2

Þt þc

3

cosðo

1

Ào

2

Þt

þb

4

cos 3o

1

t þc

4

cos 3o

2

t þb

5

ðcosð2o

1

þo

2

Þt

þ cosð2o

1

Ào

2

ÞtÞ þc

5

ðcosð2o

2

þo

1

Þt

þ cosð2o

2

Ào

1

ÞtÞ ð4Þ

where

a

0

¼k

0

þ

k

2

2

ðV

2

1

þV

2

2

Þ

b

1

¼k

1

V

1

þ

3

4

k

3

V

3

1

þ

3

2

k

3

V

1

V

2

2

c

1

¼k

1

V

2

þ

3

4

k

3

V

3

2

þ

3

2

k

3

V

2

1

V

2

b

2

¼

1

2

k

2

V

2

1

c

2

¼

1

2

k

2

V

2

2

b

3

¼c

3

¼k

2

V

1

V

2

b

4

¼

1

4

k

3

V

3

1

c

4

¼

1

4

k

3

V

3

2

b

5

¼

3

4

k

3

V

2

1

V

2

c

5

¼

3

4

k

3

V

1

V

2

2

For equal-amplitude input tones, Eq. (4) shows that the

second-order terms, of amplitudes b

2

, c

2

, b

3

, c

3

will be in-

creased 2 dB in amplitude when input tones are increased

by 1dB. The third-order terms, of amplitudes b

4

, c

4

, b

5

, c

5

,

are increased by 3 dB in amplitude when the input tones

are increased by 1 dB.

While Eq. (1) is adequate, and widely used, to predict

the intermodulation performance of a wide range of de-

vices, circuits, and systems, it seldom can be used. Exam-

ples include, but are not restricted to, prediction of

spectral regrowth in digital communication systems, tran-

sient intermodulation and frequency-dependent nonlin-

earities, and passive intermodulation.

3. SPECTRAL REGROWTH

When a modulated signal passes through a nonlinear de-

vice, its bandwidth is broadened by odd-order nonlinear-

ities. This phenomenon, called spectral regrowth or

spectral regeneration, is a result of mixing products (in-

termodulation) between the individual frequency compo-

nents of the spectrum [17]. The spectral regrowth can be

classiﬁed in the two following categories: (1) in-band in-

termodulations and (2) out-of-band intermodulations. The

ﬁrst cannot be eliminated by linear ﬁltering and are re-

sponsible for the signal-to-noise ratio degradation and,

consequently, for the bit error rate (BER) degradation in

digital communication systems. The second generates the

interference between adjacent channels and can be ﬁl-

tered out at the nonlinear device output with certain out-

put power penalty that is caused by the ﬁlter insertion

losses. This spectral regrowth causes adjacent-channel

INTERMODULATION MEASUREMENT 2217

interference (ACI), which is measured by the adjacent-

channel power ratio (ACPR).

The ACPR is the power in the main channel divided by

the power in the lower plus upper adjacent channels. Con-

sidering just the lower channel yields ACPR

lower

and the

upper channel alone yields ACPR

upper

. Analog cellular ra-

dio uses frequency or phase modulation, and the ACPR is

adequately characterized by intermodulation distortion of

discrete tones. Typically, third-order intermodulation

product (IMP3) generation, in a two-tone test, is adequate

to describe spectral regrowth. Thus, distortion in analog

radio is accurately modeled using discrete-tone steady-

state simulation. Digital radio, however, uses complex

modulation, and adjacent-channel distortion has little re-

lationship to intermodulation in a two-tone test [18,19]. A

modulated input signal applied to radiofrequency (RF)

electronics in digital radio is a sophisticated waveform re-

sulting from coding, ﬁltering, and quadrature generation.

Neither can it be represented by a small number of dis-

crete tones (or frequencies), nor can the waveform be rep-

resented in a simple analytic form. Thus, in digital radio,

ACPR is more difﬁcult to predict than one- or two-tone

responses since it depends not only on the intrinsic non-

linear behavior of the device (e.g. ampliﬁer) but also on the

encoding method (i.e., the statistics of the input stream)

and the modulation format being used. The only way the

input stream can conveniently and accurately be repre-

sented is by its statistics, and transforming these using an

appropriate behavioral model provides accurate and efﬁ-

cient modeling of ACPR [20]. While in Ref. 20 the input

signal is assumed Gaussian, digital communication sig-

nals are often far from being Gaussian. In Ref. 21 the in-

put is assumed stationary but not necessarily Gaussian.

ACPR is, therefore, deﬁned differently in the various

wireless standards. The main difference is the way in

which adjacent-channel power affects the performance of

another wireless receiver for which the offending signal is

cochannel interference [20]. In general the ACPR can be

deﬁned as [20]

ACPR¼

_

f

4

f

3

Sðf Þdf

_

f

2

f

1

Sðf Þdf

ð5Þ

where S(f) is the power spectral density (PSD) of a signal

whose channel allocation is between frequencies f

1

and f

2

,

and its adjacent channel occupies frequencies between f

3

and f

4

. Regulatory authorities impose strict constraints on

ACPR and accurate methods of its determination are of par-

ticular interest to those involved in wireless system design.

4. SIMPLE TRANSIENT INTERMODULATION THEORY

To illustrate how TIM distortion arises, consider a differ-

ential ampliﬁer with negative feedback applied between

the output and the inverting input and a voltage step ap-

plied to the noninverting input. If the open-loop gain of the

ampliﬁer were ﬂat and the time delay through it were zero,

the voltage step would instantaneously propagate undis-

torted through the ampliﬁer, back through the feedback

loop, and into the inverting input, where it would be sub-

tracted from the input signal, and the difference signal,

which is a voltage step occurring at the same time that the

input voltage does, would be ampliﬁed by the ampliﬁer.

However, this is not the case when the open-loop gain of the

ampliﬁer is not ﬂat and the time delay through it is not

zero. When the voltage step occurs, the limited high-fre-

quency response of the ampliﬁer prevents the appearance

of a signal at the ampliﬁer output terminal until the inter-

nal capacitors of the ampliﬁer can charge or discharge. This

causes the momentary absence of a feedback signal at the

inverting input to the ampliﬁer, possibly causing the am-

pliﬁer to severely overload until the feedback signal arrives.

If the input signal to the differential ampliﬁer is formed

of a sine wave superimposed on a square wave, the am-

pliﬁer will exhibit the same response to the abrupt level

changes in the square wave as it did to the voltage step

discussed above. During the momentary absence of the

feedback when the square wave changes level, the ampli-

ﬁer can either saturate or cut off. If this occurs, the sine

wave momentarily disappears from the signal at the out-

put terminal of the ampliﬁer, or it momentarily decreases

in amplitude. This happens because the saturated or cut-

off ampliﬁer appears as a short circuit or open circuit, re-

spectively, to the sine wave, and this component of the

input signal is interrupted from the output signal, thus

resulting in TIM [16].

A point to be noted is that if the term were understood

literally, this would imply transients of both high and low

frequencies and/or high or low operating levels, in other

words, all transients. In actual practice, however, TIM oc-

curs only for signals with simultaneous high level and

high frequencies—not lower levels or lower frequencies.

The key parameter of such signals is that they are char-

acterized by high signal slopes, not just high frequencies

or high levels. Neither high frequencies nor high levels in

themselves necessarily result in distortion, unless their

combination is such that a high effective signal slope is

produced. TIM is actually generated when the signal slope

approaches or exceeds the ampliﬁer slew rate. This can

happen for either transient or steady-state signals. Thus,

a more easily understood term to what actually happens

would be one that relates both slew rate and signal slope.

A more descriptive term to describe the mechanism would,

therefore, be the slew-induced distortion (SID); other de-

scriptive variations of this term are ‘‘slew rate distortion’’

or ‘‘slewing distortion’’ [22].

Because of the complexity of the mechanism resulting

in TIM, especially handling the frequency dependence of

the ampliﬁer nonlinearity and incorporation of the feed-

back, Eq. (1) cannot be used to predict the TIM perfor-

mance of nonlinear devices, and recourse to other

analytical techniques, for example, Volterra series or har-

monic balance analysis, would be inevitable.

5. VOLTERRA SERIES ANDHARMONIC BALANCE ANALYSIS

Volterra series describes a system with frequency-depen-

dent nonlinearity in a way that is equivalent to the

2218 INTERMODULATION MEASUREMENT

manner in which Taylor series approximates an analytic

function. Depending on the amplitude of the exciting sig-

nal, a nonlinear system can be described by a truncated

Volterra series. Similar to the Taylor series representa-

tion, for very high amplitudes the Volterra series diverges.

Volterra series describe the output of a nonlinear system

as the sum of the response of a ﬁrst-order operator, a sec-

ond-order one, a third-order one, and so on [23]. Every

operator is described in either the time domain or the fre-

quency domain with a kind of transfer function called a

Volterra kernel.

In Volterra series analysis the nonlinear circuit is

treated purely as an AC problem. Assuming that none of

the input signals are harmonically related, an iterative

solution can be applied for circuits not operated under

distortion saturation conditions. First the circuit is solved

for the input signals. These results are then used to cal-

culate the second-order distortion products, and these are

treated as generators at a different frequency to the input

signals and the network is again solved. This is then re-

peated for higher-order distortion products. This leads to

extremely fast calculation of distortion behavior. Simula-

tion at higher power levels can be achieved by feeding

back contributions from higher-order distortion products

[24]. The use of Volterra series to characterize the output

as a function of the input [25,26] can, therefore, provide

closed-form expressions for all the distortion products of a

frequency-dependent nonlinearity excited by a multisinu-

soidal signal.

However, techniques using Volterra series suffer from

the disadvantage that a complex mathematical procedure

is required to obtain a closed-form expression for the out-

put amplitude associated with a single component of the

output spectrum. Moreover, the problem of obtaining out-

put products of orders higher than the third becomes pro-

hibitively difﬁcult unless it may be assumed that higher-

order contributions vanish rapidly [27]. The Volterra se-

ries approach is, therefore, most applicable to mild non-

linearities where low-order Volterra kernels can

adequately model the circuit behavior. With appropriate

assumptions and simpliﬁcations, many useful features of

the Volterra series technique can be used to ﬁnd approx-

imate expressions for TIM (SID). These are quite accurate

for relatively small distortion conditions [28,29].

Alternatively, most RF and microwave circuit analysis

are based on the harmonic balance analysis [30]. The har-

monic balance technique works by processing the linear

part of the circuit in the frequency domain and the non-

linear part in the time domain. Computation in the fre-

quency domain is very fast and efﬁcient, especially for

frequency-selective components such as transmission

lines and resonant circuits. Computations in the time do-

main are followed by Fourier transform. Harmonic bal-

ance analysis can, therefore, handle intermodulation

distortion provided there are not too many excitation

tones. In the harmonic balance technique an initial esti-

mate is required for the ﬁnal waveshape, and this is re-

ﬁned interactively during analysis. The harmonic balance

method computes the response of a nonlinear circuit by

iteration, and the ﬁnal result is a list of numbers that do

not indicate which nonlinearities in the circuit are mainly

responsible for the observed nonlinear behavior. Hence

such a method is suitable for veriﬁcation of circuits that

have already been designed. This method does not present

information from which designers can derive which circuit

parameters or circuit elements they have to modify in or-

der to obtain the required speciﬁcations [31]. While Vol-

terra series analysis can provide such information, it is

applicable only to weak nonlinearities.

While viewed as a universal solution, and has been

widely used, the harmonic balance analysis may be un-

necessarily slow, cumbersome, and prone to subtle errors

[32], especially for weak nonlinearities or when a nonlin-

ear device is excited by very small signals. Volterra series

analysis is generally more accurate than harmonic bal-

ance for these types of problems, and it is several orders of

magnitude faster than a harmonic balance analysis [32].

Moreover, Volterra series analysis integrates well with

linear analysis tools, supporting simultaneous optimiza-

tion of several parameters of the nonlinear system. There-

fore, Volterra theory appears to be an ideal tool for circuits

and systems that are not strongly nonlinear but have as-

pects of linear and nonlinear circuits [32]. However, Vol-

terra series analysis becomes very cumbersome above

third-order products, and for products above ﬁfth order,

it loses most of its advantages over the harmonic balance

analysis. The major disadvantage of Volterra series is the

occasional difﬁculty in deciding whether the limitations to

weakly nonlinear operation have been exceeded.

In fact, Volterra-series analysis and the harmonic bal-

ance technique complement each other [32]. Thus, while

the Volterra series analysis works well in those cases

where harmonic balance works poorly, the harmonic bal-

ance works well where the Volterra series works poorly.

Volterra series analysis is, therefore, not appropriate for

mixers, frequency multipliers, saturated power ampliﬁers,

and similar strongly driven and/or hard nonlinearities.

Volterra series analysis is suitable for small-signal ampli-

ﬁers, phase shifters, attenuators, and similar small-signal

and/or soft nonlinearities.

Another technique for analyzing nonlinear systems is

the describing function. This approach can yield closed-

form expressions for a feedback system that contains an

isolated static nonlinearity in the feedback loop [33]. Since

it is not possible to map all nonlinear circuits and systems

to such a feedback system, the describing function method

has restricted applications.

6. PASSIVE INTERMODULATION (PIM)

While the concept of intermodulation in active devices

such as ampliﬁers, ﬁlters, and mixers is familiar and well

documented, the effects of intermodulation in passive com-

ponents such as directional couplers, cables, coaxial con-

nectors, power splitters, antennas, and electromechanical

and solid-state programmable attenuators are less famil-

iar and less documented. More recently, evidence has

emerged that PIM has an impact in other system equip-

ment, such as ampliﬁers and extenders, ﬁber nodes, and

interface units [34]. Poor mechanical contact, dissimilar

metals in direct contact, ferrous content in the conductors,

INTERMODULATION MEASUREMENT 2219

debris within the connector, poor surface ﬁnish, corrosion,

vibration, and temperature variations are among the

many possible causes of PIM. The sources of PIM have

been studied extensively; see Refs. 35–43 and the refer-

ences cited therein. Similar to the intermodulation prod-

ucts in active devices, PIM is generated when two or more

RF signals pass through RF passive devices having non-

linear characteristics [41,42]. Generally the nonlinearities

of RF passive devices consist of contact nonlinearity and

material nonlinearity [43]. Contact nonlinearity refers to

all metal contact nonlinearities causing nonlinear cur-

rent–voltage behavior, such as the tunneling effect, micro-

discharge, and contact resistance. Material nonlinearity

refers to the bulk material itself. Magnetoresistivity of the

transmission line, thermal resistivity, and nonlinear

hystresis of ferromagnetic material are good examples

[43]. PIM generation in RF passive devices is caused

by the simultaneous appearance of one or more of these

PIM sources, and the overall performance is often domi-

nated by one principal PIM source [43]. In the case of

antennas, PIM is generated not only by the same PIM

sources as in general RF passive components but also by

the external working environment, such as conducting

metal materials.

Over the years Eq. (1) was used to describe the nonlin-

ear current/voltage conduction characteristics of passive

components, (see, e.g., Refs. 37–39 and the references cited

therein). While this approach results in simple expres-

sions for the magnitudes of the harmonics and intermod-

ulations products resulting from multisinusoidal

excitations, it suffers from the following shortcomings. In

order to predict high-order harmonic or intermodulation

product magnitudes, it is necessary to determine coefﬁ-

cients of terms of similar order in the polynomial. A pre-

requisite to obtaining coefﬁcients of high-order polynomial

terms is measurement of output products of the same or-

der. For example, to obtain the coefﬁcients of a ﬁfth-order

polynomial, it is necessary to measure the output ﬁfth-or-

der components. With increasing use of narrowband com-

ponents in multicouplers used in base stations of mobile

radio systems, it becomes difﬁcult to determine high-order

coefﬁcients in the nonlinear characteristic because the

measured high-order product amplitudes from which

they are computed are inﬂuenced to an unknown extent

by the system selectivity [44]. To overcome these prob-

lems, an exponential method has been used to predict the

intermodulation arising from corrosion [45].

7. INTERMODULATION CHARACTERIZATION

Although it is important to understand the origin of in-

termodulation and the engineering techniques for avoid-

ing it, it is equally important to be able to characterize it

objectively, preferably in a way that correlates well with

the subjective perception of the intermodulation. The abil-

ity to characterize an imperfection in this way is an

important step toward eliminating it as a system perfor-

mance degradation.

Several techniques for characterizing intermodula-

tion distortion have been proposed. While some of these

techniques measure the total intermodulation distortion,

others distinguish between the various intermodulation

products. The latter are preferred, for subjective percep-

tion of intermodulation shows that equal amounts of total

intermodulation disortion differ widely in their effect ac-

cording to how the total is made up.

Depending on the signal characteristics, techniques for

characterization of intermodulation distortion can be clas-

siﬁed into two categories: (1) steady-state techniques,

where characterization is performed on the assumption

that the input to the system under consideration is a mul-

tisinusoidal signal, and (2) dynamic techniques, where

characterization is performed on the assumption that

the input to the system under consideration is formed of

a sinusoidal signal superimposed on another signal char-

acterized by rapid changes of state, for example, a square

wave or a sawtooth wave. While steady-state techniques

can be used to characterize both RF and audio systems,

dynamic techniques are generally used for characterizing

only audio systems.

7.1. Steady-State Techniques

7.1.1. The Intercept Point. Increasing the signal level at

the input to a weakly nonlinear device will cause the IMPs

to increase at the output. In fact, the increase in IMP am-

plitudes is faster than the increase in the output version of

the input signal. For increasing fundamental input power,

the fundamental output power increases in a linear man-

ner, according to the gain or loss of the device. At some

point, gain compression occurs and the fundamental out-

put power no longer increases with input power. The out-

put power of the second-order intermodulation products

also increases with fundamental input power, but at a

faster rate. Recall that, according to the simple intermod-

ulation theory, the second-order intermodulation changes

by 2dB per 1dB of change in the fundamental. Similarly,

the third-order intermodulation changes by 3 dB per 1 dB

of change in the fundamental. Thus, on a logarithmic

scale, as shown in Fig. 2, the lines representing the

second- and third-order intermodulation products have

twice and three times, respectively, the slope of the

fundamental line.

If there were no gain compression, the fundamental

input power could be increased until the second-order in-

termodulation eventually caught up with it, and the two

output power levels would be equal. This point is referred

to as the second-order intercept point (IP2). The third-or-

der intermodulation product also increases faster than the

fundamental, and those two lines will intersect at the

third-order intercept point (IP3). Rarely can either of

these two points be measured directly, due to the gain

compression of the fundamental. Instead, the intercept

points are extrapolated from measurements of the funda-

mental and intermodulation products at power levels

below the point where gain compression occurs. The

intercept points are usually speciﬁed in dBm and may

refer to either the output or the input; the two points

will differ by the gain of the system under consideration.

The second-order and third-order intercept points are ﬁg-

ures of merit that are independent of the signal level.

2220 INTERMODULATION MEASUREMENT

Therefore, the intermodulation performance of two differ-

ent systems can be compared quite easily if their intercept

points are known [46].

Using the intercept point it is easy to calculate the rel-

ative intermodulation level corresponding to a given input

signal level. In fact, the difference between the level of the

second-order intermodulation and the fundamental signal

level is the same as the difference between the fundamen-

tal signal level and the intercept point. Thus, if the second-

order intercept point is þ15 dBm and the fundamental

signal level is À10 dBm (both referred to the output of the

device), the difference between these two values is 25 dB.

Therefore, the second-order intermodulation products will

be 25 dB below the fundamental, or À35 dBm. So the in-

tercept point allows easy conversion between fundamental

signal level and the intermodulation level.

The difference between the level of the third-order in-

termodulation products and the fundamental signal level

is twice the difference between the fundamental signal

level and the third-order intercept point. (Note that the

second-order intercept point is not the same as the third-

order intercept point.) Suppose that the third-order inter-

cept point is þ5 dBm and the fundamental signal is

À25 dBm, both referred to the output of the device. The

difference between the intercept point and the fundamen-

tal is 30 dB, so the third-order intermodulation products

will be 2 times 30 dB down from the fundamental. The

relative distortion level is À60 dB, and the absolute power

of the intermodulation products is À85 dBm.

It is important, however, to note that the preceding

analyses assume that the second-order and third-order

intermodulation curves have slopes of 2 and 3dB/dB, re-

spectively. Thus, theoretically, the intercept points are not

functions of the input power level. If a power sweep

is performed, it is expected that the intercept points will

remain constant. The intercept points can, therefore, be

calculated from measurements at only one power level.

However, if the input signal exceeds a certain limit, the

amplitudes of the output fundamentals and the resulting

intermodulation products will start to saturate, and the

intercept points will usually drop off, indicating an invalid

measurement. It is essential to know this limit. It is par-

ticularly useful for high-dynamic-range circuits and sys-

tems with relatively low output powers where the

intermodulation is low, but only for signals that are low

enough. Expanding the model of Eq. (2) to include fourth-

and ﬁfth-order terms [47] can achieve this.

Moreover, at low power levels, the intercept points will

start to change as the noise ﬂoor of the measuring instru-

ment, usually a spectrum analyzer, is approached, thus

indicating an invalid measurement. It is important, there-

fore, to look at the variation of the intercept points as

functions of power as this provides a good way of checking

the valid measurement range.

7.1.2. Two-Tone Test. The two-tone test is extensively

used in characterizing a wide range of devices. Magnetic

tapes [48]; microwave and millimeter-wave diode detec-

tors [49]; analog-to-digital converters [50,51]; gamma cor-

rectors [52]; and electrical components such as resistors,

capacitors, inductors, as well as contacts of switches, con-

nectors, and relays [53] are a few examples. The two-tone

test is also used to characterize the performance of the

basilar membrane of the cochlea [54].

The two-tone test can also be used to determine the

transfer characteristic of a nonlinear device modeled by

the polynomial approximation of Eq. (2). With the input

formed of two properly selected frequencies o

1

and o

2

, and

if the second-order and third-order intermodulation prod-

ucts are measured separately, it is possible to ﬁnd, from

the measured data, the coefﬁcients of the quadratic and

cubic terms k

2

and k

3

, respectively, in the polynomial ap-

proximation of Eq. (2). If in addition, the IMPs are mea-

sured at two sets of values of o

1

and o

2

, it is possible to

identify the dominant physical nonlinear process from the

variation of IMPs with test frequencies [13].

The two-tone test can also be used to determine the

complex transfer characteristic of a nonlinear device

exhibiting AM/AM nonlinearity only with ﬁxed phase

shift between the output and the input. In this case a

complete set of measurement for all the two-tone inter-

modulation products produced by the nonlinearity at two

different power levels is necessary [55]. If the device under

consideration exhibits both AM/AM and AM/PM non-

linearities, then determination of a unique set of polyno-

mial coefﬁcients requires a complete set of intermodula-

tion measurements at three different power levels [55].

The set obtained at the highest power level will decide

the amplitude range within which the characterization

will be valid.

•

•

IP2

IP3

Input power (dBm/tone)

O

u

t

p

u

t

p

o

w

e

r

(

d

B

m

)

−100 −60 −20

−80

0

−40

0

40

80

a

b

c

Figure 2. Third-order and second-order intercept points are de-

termined by extending the fundamental, the second- and the

third-order intermodulation transfer function lines: (a) Funda-

mental transfer function, slope ¼1; (b) second-order intermodu-

lation, slope ¼2; (c) third-order intermodulation, slope ¼3. IP3—

third-order intercept point; IP2—second-order intercept point.

INTERMODULATION MEASUREMENT 2221

According to the basic assumption that the nonlinear-

ities are represented by polynomials, high-accuracy rep-

resentation of the device characteristics will require

difﬁcult accurate measurements of higher-order intermod-

ulation products, in addition to increased complications

and considerable efforts involved in the analysis [55].

Another difﬁculty from which this method suffers arises

from the necessity of measuring complete sets of two-tone

intermodulation products spread over a relatively wide

frequency range, which consequently may impose strin-

gent speciﬁcations on the measuring instruments and

techniques if accurate measurements are to be achieved.

In the two-tone test the inband IMPs are used to de-

scribe a device, a circuit or a system nonlinearity. Mea-

surements are made in or near the frequency range of

interest. In this test, the input signal consists of two fre-

quencies, o

1

and o

2

of equal amplitude and a ﬁxed amount

of frequency spacing. At the output of the circuit or the

system under test the amplitudes of the third-order inter-

modulation products 2o

1

Ào

2

and 2o

2

Ào

1

are measured.

The intermodulation distortion is deﬁned as the ratio be-

tween the root sum square of the intermodulation prod-

ucts and the root sum square of the twin-tone amplitudes.

Unless a wave analyzer or a spectrum analyzer is avail-

able, the implementation of the two-tone test invariably

require ampliﬁcation of the whole output spectrum to ob-

tain components o

1

and o

2

on a normalized value (100%).

Then, o

1

and o

2

are suppressed, and the remaining com-

ponents 2o

1

Ào

2

and 2o

2

Ào

1

are measured with an AC

voltmeter or oscilloscope. Especially at audiofrequencies,

this approach requires steep ﬁlters, one set of ﬁlters for

each set of o

1

and o

2

. For the same reason o

2

Ào

1

cannot

be too low, so it will never be a really narrowband system.

This narrowband aspect is particularly important for

higher frequencies, where equalizers, in the reproduction

audio channel, may give unequal ampliﬁcation of the com-

ponents in the spectrum [56]. In the audiofrequency range

several versions of the two-tone test are available [56–59].

7.1.3. Three-Tone Test. In this test, again, specific in-

band IMPs are selected to characterize the overall system

nonlinearities [60]. The more even spectral distribution

and ﬂexibility, while still allowing discrete frequency eval-

uation, make this an attractive test for multifrequency

systems such as communication and cable television sys-

tems. In this test three equal-amplitude tones are applied

to the input of the nonlinear system under consideration.

Thus

V

i

¼Vðcos o

1

t þ cos o

2

t þ cos o

3

tÞ ð6Þ

Combining Eqs. (2) and (6), and using simple trigonomet-

ric identities, it is easy to show that the third-order term,

k

3

V

i

3

will contribute, to the output spectrum, the following:

1. Three components at frequencies o

1

, o

2

and o

3

each

with amplitude given by

A

1

¼

15

4

k

3

V

3

ð7Þ

2. Three components at frequencies 3o

1

, 3o

2

, 3o

3

each

with amplitude given by

A

3

¼

1

4

k

3

V

3

ð8Þ

3. Twelve components at frequencies 2o

m

Æo

n

; m; n¼

1–3, each with amplitude given by

A

21

¼

3

4

k

3

V

3

ð9Þ

4. Four components at frequencies o

m

Æo

n

Æo

p

; m; n;

p¼1–3, each with amplitude given by

A

111

¼

3

2

k

3

V

3

ð10Þ

Equations (9) and (10) show that an intermodulation prod-

uct of frequency o

m

Æo

n

Æo

p

is 6 dB higher in level than

an intermodulation product of frequency 2o

m

Æo

n

: Inter-

modulation distortion is deﬁned as the ratio between the

amplitude of one of the intermodulation products of fre-

quency o

m

Æo

n

Æo

p

and the amplitude of one of the three

output tones. In this test the choice of frequencies o

1

,o

2

,o

3

used to perform the measurement is important. This is

because a system’s intermodulation performance may not

be constant over its operating frequency range.

The three-tone test is widely used to characterize the

performance of RF ampliﬁers used in television broadcast

transposers, where the vision carrier, color subcarrier, and

sound carrier frequency components interact in the pres-

ence of ampliﬁer nonlinearities. If the three frequency

components are represented as single frequencies (o the

vision carrier, o

sc

the color subcarrier, and o

s

the sound

carrier with amplitudes V

v

, V

sc

, and V

s

, respectively), then

the input signal can be expressed as

V

i

¼V

v

cos o

v

t þV

sc

cos o

sc

t þV

s

cos o

s

t ð11Þ

Combining Eqs. (2) and (11), and using simple trigono-

metric identities, it is easy to show that the third-order

term of Eq. (2) produces, among others, two in-band in-

termodulation components given by

V

ip

¼

3

2

k

3

V

v

V

sc

V

s

cosðo

v

þo

s

Ào

sc

Þt

þ

3

4

k

3

V

s

V

2

sc

cosð2o

sc

Ào

s

Þt

ð12Þ

Intermodulation performance of the transposer is mea-

sured by taking the transposer out of service and using the

three-tone simulation of a composite video and sound sig-

nal, given by Eq. (11), as its input. The three levels and

frequencies vary from system to system. Typical levels,

below the peak synchronous pulse level, are V

v

¼ À6 dB,

V

sc

¼17 dB, and V

s

¼ À10 dB. Under these conditions, the

ﬁrst term of Eq. (12) is the most visible, and the second

term will be much lower in amplitude, typically 17 dB less.

Using a spectrum analyzer, the relative amplitude of

the major in-band intermodulation is measured and

2222 INTERMODULATION MEASUREMENT

referenced to the level of peak synchronous pulse. Usually,

the permissible level of the major in-band intermodulation

component is À53 dB below the reference level. This

three-tone test method is slow and requires spectrum an-

alyzers with relatively wide dynamic ranges. Moreover, it

measures the system performance at one luminance level

and one chrominance level. Thus, it does not test the sys-

tem over its full operating range [61].

The inadequacy of the internationally accepted three-

tone test method can be overcome by using a modiﬁed col-

orbar test signal [61]. The colorbars are applied to the

transposer via a test transmitter. The colorbars and sound

carrier therefore apply the three tones to the transposer,

changing levels in rapid succession. With suitable pro-

cessing, based on sampling the demodulated colorbar sig-

nal for short intervals corresponding to a selected color,

intermodulation levels can be measured simultaneously at

seven different luminance levels and can be shown in his-

togram form [61].

7.1.4. Noise Power Ratio (NPR) Test. In the NPR test,

the input to the device under test is obtained from a white-

noise source that is bandlimited to the instantaneous fre-

quency range of interest. This emulates a situation with

many simultaneous input signals. Provided that none of

the signals dominate, according to the central-limit theo-

rem, the resulting voltage obtained when many uncorre-

lated signals are added will approach a Gaussian

distribution. True white noise covers a frequency range

of interest continuously, unlike discrete signals.

The NPR test measures the amount of intermodulation

products power between two frequency ranges of white

Gaussian noise. A white-noise generator is used with its

output frequency range limited by a bandpass ﬁlter ac-

cording to the bandwidth of the device under test. A quiet

channel is formed by a switchable band-reject ﬁlter, as

shown in Fig. 3. Then, the resulting white-noise signal is

applied to the input of the device under test. At the output

of the device under test is a receiver which is switch-tuned

to the frequency of the band-reject ﬁlter used to produce

the quiet channel. The NPR test is widely used for eval-

uating the intermodulation performance of systems whose

input signal spectrum distribution can be approximated

by that of white noise. However, the NPR may be degraded

by the noise ﬂoor of the system under test, especially

under very low loading conditions. It may also be degraded

by the distortion products produced under high loading

conditions [62].

7.1.5. Cross-Modulation. Cross-modulation occurs when

modulation from a single unwanted modulated signal

transfers itself across and modulates the wanted signal.

Cross-modulation is troublesome primarily if the desired

signal is weak and is adjacent to a strong unwanted signal.

Even when the carrier of the strong unwanted signal is not

passed through the system, the modulation on the unde-

sired carrier will be transferred to the desired carrier.

Cross-modulation is, therefore, a special case of intermod-

ulation. Recall that when the input to a non-

linear system is formed of a two-tone signal of the form of

Eq. (3), then the amplitudes of the output components at

frequencies o

1

and o

2

will be given by

b

1

¼k

1

V

1

þ

3

4

k

3

V

3

1

þ

3

2

k

3

V

1

V

2

2

ð13Þ

and

c

1

¼k

1

V

2

þ

3

4

k

3

V

3

2

þ

3

2

k

3

V

2

1

V

2

ð14Þ

respectively. Thus, the output obtained at each frequency

o

1

and o

2

, is dependent on the amplitude of the signal

component of the other frequency. If the amplitude of

the wanted unmodulated carrier is V

1

and the instanta-

neous amplitude of the unwanted amplitude-modulated

carrier is

V

2

ðtÞ ¼V

2

ð1 þmcos o

m

tÞ ð15Þ

then, using Eq. (13), the amplitude of the wanted carrier

will be

b

1

¼k

1

V

1

þ

3

4

k

3

V

3

1

þ

3

2

k

3

V

1

V

2

2

ð1 þmcos o

m

tÞ

2

ð16Þ

For small values of m and with k

3

5k

1

, Eq. (16) can be ap-

proximated by

b

1

ﬃk

1

V

1

þ3k

3

V

1

V

2

2

mcos o

m

t ð17Þ

Thus the wanted carrier will be modulated by a modulation

index

p¼3

k

3

k

1

V

2

2

m ð18Þ

The cross-modulation factor is then deﬁned as

K ¼

p

m

ð19Þ

Thus, one frequency will be modulated by the modulation of

the other frequency. Similar results can be obtained if the

unwanted carrier is FM-modulated.

Frequency

P

o

w

e

r

(

d

B

)

∆

δω

o

ω

b

a a

A

B

Figure 3. The output spectrum of a noise–power ratio measure-

ment. (a) injected noise; (b) noise and intermodulation generated

in the measurement bandwidth do by the DUT. NPR¼AÀB.

INTERMODULATION MEASUREMENT 2223

Cross-modulation can be measured as the change in the

amplitude of the wanted unmodulated carrier as a func-

tion of the amplitude of the unwanted unmodulated car-

rier. This is the procedure recommended by the NCTA

(National Cable Television Association) standard cross-

modulation measurement [63]. Alternatively, cross-modu-

lation can be measured using the definition of Eq. (19):

measuring percentage modulation that appears on an un-

modulated desired carrier due to the presence of an un-

desired modulated carrier, divided by the percentage

modulation on the undesired carrier [64].

Cross-modulation can also be measured using two

equal-amplitude carriers. The wanted carrier, o

2

, is un-

modulated while the unwanted carrier, o

1

, is FM-modu-

lated. The output spectrum clearly shows the frequency

deviation of the wanted carrier. Moreover, it can be shown

that the frequency deviation of the intermodulation com-

ponents, of the output spectrum, is larger than that of the

original FM-modulated unwanted carrier. For the inter-

modulation product of frequency ao

1

7bo

2

, the deviation

will be multiplied by a. Thus, it may be easier to measure

the cross-modulation by measuring the deviation of an in-

termodulation product rather than the deviation of the

wanted unmodulated carrier [65].

7.1.6. Differential Gain. Differential gain (DG), a pa-

rameter of special interest in color-TVengineering, is con-

ventionally deﬁned as the difference in gain encountered

by a low-level high-frequency sinusoid at two stated in-

stantaneous amplitudes of a superimposed slowly varying

sweep signal. In video signal transmission, the high-fre-

quency sinusoid represents the chromatic signal and the

low-frequency sinusoid represents the luminance signal.

Corresponding to the theoretical conditions of the differ-

ential measurement, DG measurement is performed by a

signal of the form of Eq. (3) with o

2

bo

1

and V

2

!0:0 at

V

1

¼0.0 and X [66]. Therefore, recalling that when the in-

put to a nonlinear system is formed of a two-tone signal of

the form of Eq. (3), the amplitude of the output component

at frequency o

2

will be given by

c

1

¼k

1

V

2

þ

3

4

k

3

V

3

2

þ

3

2

k

3

V

2

1

V

2

ð20Þ

Thus, DG can be expressed as

DG¼1 À

k

1

þ

3

4

k

3

V

2

2

k

1

þ

3

4

k

3

V

2

2

þ

3

2

k

3

X

2

ð21Þ

DG can, therefore, be considered to some extent as a mea-

sure of the intermodulation performance of a system

under test.

7.1.7. Dynamic Range. Dynamic range can be deﬁned

as the amplitude range over which a circuit or a system

can operate without performance degradation. The mini-

mum amplitude is dictated by the input thermal noise and

the noise contributed by the system. The maximum am-

plitude is dictated by the distortion mechanisms of the

system under consideration. In general, the amount of

tolerable distortion will depend on the type of signal and

the system under test. However, for the purpose of an ob-

jective definition the maximum amplitude will be consid-

ered the input signal level at which the intermodulation

distortion is equal to the minimum amplitude [67]. The

dynamic range can, therefore, be considered to some ex-

tent as a measure of the intermodulation performance of a

system under test.

A useful working definition of the dynamic range is

that it is (1) two-thirds of the difference in level between

the noise ﬂoor and the intercept point in a 3 kHz band-

width [68] or (2) the difference between the fundamental

response input level and the third-order response input as

measured along the noise ﬂoor (sometimes deﬁned as 3 dB

bandwidth above the noise ﬂoor) in a 3 kHz bandwidth, as

shown in Fig. 4. Reducing the bandwidth improves dy-

namic range because of the effect on noise.

Because the power level at which distortion becomes

intolerable varies with signal type and application, a ge-

neric definition has evolved. The upper limit of a net-

work’s power span is the level at which the power of one

IM product of a speciﬁed order is equal to the network’s

noise ﬂoor. The ratio of the noise ﬂoor power to the upper-

limit signal power is referred to as the network’s dynamic

range (DR). Thus the DR can be determined from [69]

DR

n

¼

n À1

n

½IP

n;in

ÀMDS ð22Þ

where DR

n

is the dynamic range in decibels, n is the order,

IP

in

is the input intercept power in dBm, and MDS is the

minimum detectable signal power in dBm.

Alternatively, in receiver circuits the spurious-free

dynamic range (SFDR) and the intermodulation-free

• IP3

(a) (b)

(c)

Dynamic

range

−40 0.0 40 80 120 160

−40

0.0

40

80

120

160

O

u

t

p

u

t

l

e

v

e

l

(

d

B

a

b

o

v

e

i

n

p

u

t

)

Input level (dBuV)

Figure 4. The dynamic range is the difference between the fun-

damental response input level and the third-order response input

as measured along the noise ﬂoor: (a) fundamental response;

(b) third-order intermodulation response; (c) noise ﬂoor.

2224 INTERMODULATION MEASUREMENT

dynamic range (IFDR) are widely used to quantify the

capability of the receiver to listen to a weak station,

without disturbance from an intermodulation product

generated by strong stations on other frequencies. The

SFDR and the IFDR are in fact measures of how strong

two signals can be before the level of their intermodulation

products can reach the noise ﬂoor of the receiver. The

SFDR, or the IFDR, is deﬁned as the difference in decibels

between the power levels of the third-order intermodula-

tion IM3 (assuming that there is only a third-order non-

linearity) and the carrier when the IM3 power level equals

the noise ﬂoor at a given noise bandwidth. It can be

expressed as [70]

SFDR¼

2

3

½IIP3 ÀEINÀ10 log

10

ðNBWÞ ð23Þ

where IIP3 is the third-order input intercept point, EIN in

(dB/Hz) is the equivalent input noise, and NBW (in Hz) is

the noise bandwidth.

7.1.8. Adjacent- and Cochannel Power Ratio Tests. In

modern telecommunication circuits, signals constituting

one or more modulated carriers are handled. Character-

ization of the intermodulation performance of such cir-

cuits cannot, therefore, be performed using two-tone and

three-tone input signals; a combination of equally spaced

tones—in practice, more than B10 sinusoids [71], with

constant power and correlated or uncorrelated phases—

would be more appropriate [72].

Because of the nonlinearity of the device under test,

intermodulation products will be generated. These inter-

modulation products can be classiﬁed as adjacent-channel

distortion when their frequencies are located to the right

or to the left of the original spectrum, or cochannel dis-

tortion when their frequencies are located exactly over the

original spectrum. The adjacent-channel power ratio

(ACPR) is deﬁned as the ratio between the total linear

output power and the total output power collected in the

upper and lower adjacent channels [73]. The cochannel

power ratio (CCPR) is deﬁned as the ratio between total

linear output power and total distortion power collected

in the input bandwidth [73]. The intermodulation distor-

tion ratio (IMR) is the ratio between the linear output

power per tone and the output power of adjacent-channel

tones [73].

In fact, the ACPR, CCPR, and IMR distortion measure-

ments are simple extensions of the two-tone intermodula-

tion measurement [74]. However, it is important to ﬁrst

generate a very clean multitone signal. This can be easily

achieved using the technique described in Ref. 75.

8. INTERMODULATION MEASUREMENT

8.1. Measurement Equipment

8.1.1. Multitone Tests. A block diagram of the system

used for multitone intermodulation measurement is

shown in Fig. 5. The multiple-frequency source can be

implemented from two or three synthesized sine/square/

triangular-wave generators. Ampliﬁer/attenuator pairs

can be added at the output of each generator. Bandpass

ﬁlters can also be added to suppress the harmonic con-

tents at the output of each generator. For RF measure-

ments, harmonic suppression and isolation between

different generators is achieved by using ampliﬁer/circu-

lator combinations and cavity resonators [76]. The syn-

thesized sources are combined using hybrids or combiners

of adequate isolation. Spectral purity at this point is cru-

cial to the accuracy of the measurement. The multitone

output is fed to the device under test (DUT). The output of

the DUT is fed to the spectrum analyzer. For RF measure-

ments, the output of the DUT can be fed to directional

couplers. The outputs of the directional couplers are fed to

a television oscilloscope and/or a spectrum analyzer.

Alternatively, for audiofrequency measurements, the

intermodulation components of interest can be ﬁltered

out, using bandpass ﬁlters, and fed to AC voltmeters.

For audiofrequency measurements, resistive combiners

are widely used for combining the outputs of two or

more signal generators.

8.1.2. Measurement Using a Microcomputer. Intermod-

ulation can also be measured using a microcomputer [77].

A block diagram of this technique is shown in Fig. 6. This

technique is based on measuring the single-tone input–

output characteristic of the DUT using a vector voltmeter.

The output of the vector voltmeter is fed to a microcom-

puter that converts it into three digital data lines repre-

senting the input amplitude, the output amplitude, and

the phase lag between the input and output signals. After

storing the data, the microcomputer increments the am-

plitude of the input signal. After storing all the necessary

data, the microcomputer, using a stochastic method, cal-

culates the amplitudes of the intermodulation components

of the DUT. Although the procedure reported in Ref. 77

uses a stochastic method for calculating the amplitudes of

the intermodulation components resulting from a two-

tone input signal, the same procedure can be applied to

any number of input tones using different analytic tech-

niques for modeling the nonlinear characteristics of the

DUT.

Alternatively, microcomputers can be added to the

measurement setup of Fig. 5 to

1. Control the frequencies of the signal sources, espe-

cially in the millimeter-wave range, where the

A

A

BPF

BPF

BPF C DUT

SG1

SG2

SA C

Figure 5. Block diagram of the two-tone test setup; multitone

tests require additional signal generators, combiners, ampliﬁers,

and bandpass ﬁlters (SG—signal generator; A—ampliﬁer; BPF—

bandpass ﬁlter; C—combiner; DUT—device under test; SA—spec-

trum analyzer).

INTERMODULATION MEASUREMENT 2225

difference in frequencies between the signal sources

may be less than 0.001 of the base signal frequency

[78].

2. Scan the base signal frequency over the measure-

ment range of interest in predeﬁned steps [79].

3. Correct the power from each source so that power

delivery to the DUT will be the same across the

whole frequency range scanned.

4. Read and calculate the parameters of interest dur-

ing the measurements [80,81].

8.1.3. Noise Power Ratio Test. Figure 7 shows a block

diagram of a noise power ratio test setup [62]. The setup

consists of a white-noise generator that applies an accu-

rate level of white Gaussian noise power with known

bandwidth (equaling Do and centered around o

0

) to the

DUT. The output of the DUT is measured with the band-

reject ﬁlter out. When the band-reject ﬁlter, with band-

width¼do and centered around o

0

, is switched in, a nar-

row band of frequencies is attenuated by about 70 dB, and

a quiet channel, of width do and centered around o

0

, is

formed as shown in Fig. 3. At the output of the DUT, the

noise power is measured in the quiet channel, using a

bandpass ﬁlter with bandwidth do and centered around

o

0

. This noise power is due to the thermal noise and the

intermodulation introduced by the DUT. The NPR is the

ratio between the noise power measured without the

band-reject ﬁlter inserted before the DUT to that mea-

sured with the band-reject ﬁlter inserted. The white-noise

generator corrects the loading power level for the inser-

tion loss of the band-reject ﬁlter.

8.1.4. Noise Floor and SFDR Test. Figure 8 shows a test

setup for measurement of noise ﬂoor and the SFDR of a

communication link [70]. To measure the noise ﬂoor of the

communication link, the transmitter is switched off. Then

the noises of the low-noise ampliﬁer and the spectrum an-

alyzer are measured. Switching the transmitter on in-

creases the noise ﬂoor by the transmitter noise and

therefore the difference between the two noise measure-

ments is the noise generated by the transmitter.

To measure the SFDR, the input power is decreased

until the IM3 level equals the noise ﬂoor. Recall that de-

creasing the input power by 1dB decreases the IM3 level

by 3 dB. However, this is true only if the third-order non-

linearity is dominant. Higher-order nonlinearities will

contribute to the third-order intermodulation (IM3), and

DC DC DUT

VV

MC

SG

Figure 6. Block diagram of a microcomputer-based intermodu-

lation measurement setup (SG—signal generator; DC—direction-

al coupler; DUT—device under test; VV—vector voltmeter; MC—

microcomputer).

•

•

•

• •

• •

•

•

•

•

• WNG BPF1

BRF

DUT

BPF2

PM

Figure 7. Block diagram of the noise power

ratio test setup (WNG—white-noise genera-

tor; BPF1—bandpass ﬁlter with bandwidth

do centered around o

0

; BRF—band-reject ﬁl-

ter with bandwidth do centered around o

0

;

DUT—device under test; BPF2—bandpass ﬁl-

ter with bandwidth do centered around o

0

;

PM—power meter).

Communication

Link

R

LNA

SA

SG1 SG2 CIR2 CIR1 C

T

Figure 8. Setup for noise ﬂoor and SFDR measurement (SG—

signal generator; CIR—circulator; C—combiner; T—transmitter;

R—receiver; LNA—low-noise ampliﬁer; SA—spectrum analyzer).

2226 INTERMODULATION MEASUREMENT

in such cases the measured SFDR will be different from

calculations obtained using Eq. (23).

8.1.5. Externally Induced Intermodulation Test. This is a

two-tone test with one signal applied to the input and the

other signal applied to the output [9]. A test setup is

shown in Fig. 9. Two directional couplers are used to

gauge both the forward-carrier power and the intermod-

ulation product levels. Two more directional couplers are

added to inject the interfering signal and to measure the

actual injected value using the spectrum analyzer.

8.2. Measurement Accuracy

8.2.1. Multitone Tests. For accurate measurements of

the intermodulation products using multitone tests, it is

essential to reduce, or remove, the nonlinear distortion

originating in the signal sources and/or the measurement

equipment. Measurement accuracy may, therefore, be

affected by the purity of the signal sources, the linearity

of the combiners, and the performance of the spectrum

analyzer.

8.2.2. Signal Sources. Measurement of the amplitudes

of the intermodulation components requires the use of two

or more signals. The frequencies of these signals must be

noncommensurate. Otherwise, harmonics in one source

might interfere with the fundamental(s) of other signal(s)

and thus interfere with the desired intermodulation com-

ponents.

Ideally the signal generators would produce perfect si-

nusoids, but in reality all signals have imperfections. Of

particular interest here is the spectral purity, which is a

measure of the inherent frequency stability of the signal.

Perhaps the most common method used to quantify the

spectral purity of a signal generator is its phase noise [82].

In the time domain, the phase noise manifests itself as a

jitter in the zero crossings of a sine wave. In the frequency

domain, the phase noise appears as sidebands surround-

ing the original frequency. Thus, mixing with other fre-

quencies, due to the nonlinearities of the DUT, would

result in additional intermodulation products. It is, there-

fore, important to consider the intermodulation due to

phase noise when calculating the intermodulation perfor-

mance of the DUT [83].

Signal generators with automatic level control (ALC)

may produce signals with unwanted modulation. The ALC

is implemented by rectifying the output signal of the gen-

erator and feeding back the resulting DC voltage to drive

an amplitude modulator. If a second signal is applied to

the output of the signal generator, the detector will pro-

duce a signal at the point of difference between the two

frequencies. This signal will modulate the generator’s out-

put. The frequency of the modulation sidebands will share

the same spectral lines as the intermodulation products of

interest. Isolating the signal generators and the combin-

ers can minimize such an effect. This can be achieved by

ensuring that there is as much attenuation as possible

between them.

8.2.3. Combiners. Measurement of intermodulation

products is performed by applying to the input of the cir-

cuit, or the system, under test a signal consisting of two or

more different frequencies obtained from different signal

generators. The outputs of the signal generators are,

therefore, combined by a combiner. The combiner must

provide sufﬁcient isolation between the signal sources to

reduce the possibility of producing intermodulation prod-

ucts before the combined input signal is applied to the

circuit or the system under test. While resistive combiners

are adequate for input signal levels up to a few millivolts,

for larger voltage levels the use of power combiners may

be inevitable [84]. Insertion of an attenuator in each arm

of the combiner helps minimize the distortion components

resulting from the interaction between the two signal

sources. Such components, if generated, should be at least

80 dB below the fundamental components.

A simple test to determine whether adequate isolation

has been achieved can be effected by introducing a vari-

able attenuator between the signal source combiner and

the DUT in Fig. 6. This is set to a low value during mea-

surements, but at setup, when IMPs have been located on

the spectrum analyzer, increasing the attenuation by 3 dB

will result in a reduction in the observed IMP level. If this

reduction is only 3 dB, it has to be assumed that the IMP

observed has originated in the signal sources, not in the

DUT. If, however, the reduction is 6 dB for a second-order

IMP or 9dB for a third-order IMP [see Eq. (4)], then it is

safe to assume that the IMP has originated in the DUT or

the spectrum analyzer.

Alternatively, a technique that attenuates the parasitic

intermodulation products that result from the interaction

between the generators of the fundamental components,

before the input of the spectrum analyzer, was described

in Ref. 85. A block diagram of the technique is shown in

Fig. 10. The input to the system under test is formed by

SG2 A BPF

DC

SA

DUT

SG1

DC •

PM

Figure 9. Measurement of externally induced intermodulation

can be performed by using two tones: one injected at the input and

one injected at the output of the DUT (SG—signal generator;

DC—directional coupler; PM—power meter; SA—spectrum

analyzer; BPF—bandpass ﬁlter; A—ampliﬁer).

INTERMODULATION MEASUREMENT 2227

combining the outputs of two signal generators at fre-

quencies o

1

and o

2

in the combiner. The ﬁrst hybrid com-

biner/splitter (HCS1) splits the combined signal into two

branches with voltage transfer ratio a¼a and b¼

ﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃ

1 Àa

2

p

at the ﬁrst and second outputs. Using Eq. (1), and assum-

ing that the system under test and the compensator have

identical nonlinear characteristics, the inputs of the sec-

ond hybrid combiner/splitter (HCS2) can be expressed as

V

a

¼

N

n¼0

k

n

ðaV

i

Þ

n

ð24Þ

and

V

b

¼

N

n¼0

k

n

ﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃ

1 Àa

2

p

V

n

i

_ _

ð25Þ

Using Eqs. (24) and (25), the output of the second hybrid

combiner/splitter (HCS2), with voltage transfer ratio op-

posite in sign and equal to the reciprocal of that of HCS1,

can be expressed as

V

out

¼

N

n¼0

Àk

n

ﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃ

1 Àa

2

p

ðaV

i

Þ

n

Àa

ﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃ

1 Àa

2

p

V

n

i

_ _ _ _

ð26Þ

According to Eq. (26), broadband compensation occurs for

the linear components of the combined signal, with n¼1.

Thus, all the linearly transformed spectral components

are eliminated. This is also true for the intermodulation

components that may result from the nonlinear interac-

tion between the two signal generators. The output of

HCS2 can, therefore, be applied directly to the spectrum

analyzer.

This technique does not require complicated high-order

selective ﬁlters and can attenuate the parasitic intermod-

ulation components and the fundamental frequency com-

ponents by about 50 dB over a wide range of frequencies

differing by 7–10 octaves. However, it requires a compen-

sator with a nonlinear characteristic similar to that of the

system under test.

8.2.4. Spectrum Analyzers. Spectrum analyzers are

widely used in measuring the intermodulation perfor-

mance of electronic circuits and systems. Internal circuits

of the spectrum analyzers are, themselves, imperfect and

will also produce distortion products [46]. The distortion

performance of the analyzers is usually speciﬁed by the

manufacturers, either directly or lumped into a dynamic

range speciﬁcation. The performance of the analyzer can

be stretched, however, if the nature of these distortion

products is understood.

Amplitudes of the distortion products, resulting from

the internal circuits of the analyzer, can be reduced by

reducing the signal levels at the analyzer’s input. Thus,

using internal and/or external attenuators can reduce the

input signal levels to the analyzer and hence reduce its

distortion products and improve the intermodulation mea-

surement range of the spectrum analyzer. However, re-

duced input levels to the analyzer mean reduced signal-to-

noise ratio, and the distortion component to be measured

may be buried in the noise. While reducing the resolution

bandwidth of the analyzer can reduce noise, this may lead

to slower sweep rate. Thus, achieving an optimum dynam-

ic range involves tradeoffs between input signal levels and

analyzer distortion. Usually, datasheets of analyzers will

contain information about noise level in each resolution

bandwidth and distortion products generated by the ana-

lyzer for each input level. Using this information, one can

determine the dynamic range of the analyzer for various

input levels [86].

Whenever good selectivity, as well as sensitivity and

dynamic range, are of prime importance, test receivers

may be used in preference to spectrum analyzers [6]. Al-

ternatively, if the frequencies of the intermodulation com-

ponents of interest are sufﬁciently lower (or higher) than

the fundamental frequencies, then lowpass (or highpass)

ﬁlters can be used to remove the fundamental components

that would give rise to other nonlinear distortion compo-

nents in the spectrum analyzer. Attenuation factors of

80 dB or more, at frequencies outside the band of interest,

are recommended. The insertion loss of the lowpass (or the

highpass) ﬁlter should be as small as possible; 0.4dB or

less is recommended.

If the frequency of the intermodulation component of

interest is not sufﬁciently higher (or lower) than the fun-

damental frequencies, then it would be necessary to have

complicated multiple-section high-order ﬁlters with am-

plitude–frequency characteristics that are nearly rectan-

gular. Such ﬁlters will change, to some extent, the

amplitude of the intermodulation components, and this

will complicate calculation of the intermodulation perfor-

mance of the system under test. A method for compensat-

ing for a large fundamental component, thus allowing the

measurement of small intermodulation components in its

presence, was described in Ref. 87.

A block diagram of the compensation method is shown

in Fig. 11. The input to the system under test is formed of

one large amplitude signal at frequency o

1

and one small

amplitude signal at frequency o

2

with o

1

5o

2

. The output

of the system under test contains fundamental compo-

nents at frequencies o

1

and o

2

, and intermodulation com-

ponents at frequencies o

2

7no

1

, n¼1, 2, y, N. In order to

measure the small amplitude intermodulation compo-

nents, it is necessary to avoid applying to the analyzer

the fundamental component at frequency o

2

. This can be

achieved as follows. The output of the system under test is

SG1

SG2

C HSC1

DUT

CO

HSC2 SA

a

b

V

i

•

Figure 10. A technique for attenuating the intermodulation

products resulting from interaction between the signal genera-

tors of the fundamental components (SG—signal generator; C—

combiner; DUT—device under test; BRF—band-reject ﬁlter; PS—

phase shifter; DA—differential ampliﬁer).

2228 INTERMODULATION MEASUREMENT

fed to the second band-reject ﬁlter BRF2 to suppress the

fundamental component at o

1

. The output of the signal

generator of frequency o

2

is fed to the ﬁrst band-reject ﬁl-

ter BRF1 to suppress any component at frequency o

1

be-

fore reaching the phase shifter through the combiner. The

phase shifter compensates, at the frequency o

2

, the phase

shift through the system under test.

Ideally, the voltages of frequency o

2

at the inputs of the

differential ampliﬁer are equal. Thus, the output of the

differential ampliﬁer at frequency o

2

is ideally zero. In

practice, the output voltage at o

2

will be attenuated by

50–60 dB [6]. The output of the differential ampliﬁer,

with suppressed fundamental component at frequency

o

2

, can be applied to the spectrum analyzer. This compen-

sation technique, which entails additional ﬁlters and

matching units, can be used only for broadband measure-

ments with o

1

5o

2

.

Although spectrum analyzers using digital IF sections

may not suffer from the internally generated distortion,

discussed above, they may suffer from the relatively

low-level distortion products resulting from the analog-

to-digital conversion. The amplitudes of these products

is usually less sensitive to the amplitude of the signal

components.

8.2.5. Noise Power Ratio Test. The accuracy of the noise

power ratio (NPR) test is affected mainly by two factors:

(1) the noise ﬂoor of the ampliﬁer that will dominate under

very low loading conditions and (2) the distortion products

produced under very high loading conditions. It is, there-

fore, recommended to sweep the loading between two pre-

speciﬁed start and stop levels. The NPR is measured at

different levels, and the largest measured value of NPR is

considered as the worst case.

8.2.6. Microcomputer-Based Tests. Quantization errors

associated with the analog-to-digital conversion of the

data in microcomputer-based intermodulation test must

be taken into account. Measurement errors due to quan-

tization are affected by the length of the binary digits and

determine the dynamic range of operation [77].

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ITERATIVE METHODS

ROBERT J. BURKHOLDER

JIN-FA LEE

The Ohio State University

Columbus, Ohio

1. INTRODUCTION

Iterative methods are used in RF and microwave engi-

neering to solve complex systems by repeatedly reﬁning

an approximate solution until a desired level of accuracy

or performance is achieved. In many such problems, an

exact solution does not exist and a direct numerical solu-

tion may not be feasible because of the very large number

of degrees of freedom. Typical applications include the so-

lution of large systems of differential and integral equa-

tions that may involve thousands or millions of unknown

variables. For these problems it may not be possible to

generate and store a full system matrix, and then solve it

directly (e.g., by inversion, factorization, or Gauss elimi-

nation). Iterative methods only need to apply an operator

(or system matrix) to the solution at each iteration. They

are particularly well suited for the solution of sparse ma-

trix systems because a large percentage of the operations

involved are negligible.

Mathematically, an iterative algorithm starts with an

initial approximate solution, and repeatedly applies an

operator to the solution to improve its accuracy at each

iteration. Eventually the solution should converge to a

given level of accuracy. Convergence is the primary issue

associated with any iterative method. The solution may

converge very slowly if the iterative operator is not well

conditioned, or it may even diverge. Figure 1 illustrates

the basic iterative loop.

Initial

approximate

solution

End

Yes

No

Is solution

sufficiently

accurate?

Apply iterative

operator to obtain

improved solution

from previous

solution

Figure 1. Schematic diagram of an iterative solution.

ITERATIVE METHODS 2231

There are two broad categories of iterative methods:

stationary and nonstationary. Stationary methods are

characterized by an operator that does not change with

each iteration. Classical iterative methods are included in

this category, such as Jacobi and Gauss–Seidel. Conjugate-

gradient methods are included in the class of nonstation-

ary methods, wherein some parameters in the operator

change with each iteration [1].

In general, conjugate-gradient iterative methods have

better convergence properties than do classical iterative

methods when compared over a wide range of problems.

The convergence of classical methods tends to be very

problem-dependent. In fact, classical methods are often

based on the underlying physics of a particular scenario.

For example, a classical iterative algorithm may be de-

signed to model the multiple electromagnetic (EM) wave

scattering between two or more objects. Such an algorithm

could be very rapidly convergent for that problem, but

slowly convergent or even divergent for a different prob-

lem. On the other hand, conjugate-gradient methods have

theoretically guaranteed convergence if the system matrix

is nonsingular, although in practice the limited numerical

precision of a computer may cause the algorithm to stall.

The convergence of any iterative method may be im-

proved by altering the formulation so that it is better con-

ditioned. This is referred to as preconditioning the

operator or system of equations. The accuracy of the solu-

tion at each iteration may be gauged in terms of the re-

sidual error, which is a measure of how well the solution

satisﬁes the original system of equations.

2. HISTORICAL REVIEW OF ITERATIVE METHODS IN

ELECTROMAGNETICS

Iterative methods in EM did not become popular until ad-

vances in computer technology made it possible to solve

large systems of equations. Classical iterative methods

were developed to model physical EM interactions be-

tween different parts of a geometry. Thiele et al. ﬁrst de-

veloped a hybrid technique to combine physical optics and

the method of moments in 1982 [2–4]. The solution iter-

ates between the optically lit region and the shadow re-

gion of an arbitrary scattering geometry. This method was

extended further and made more general by Hodges and

Rahmat-Samii [5], including the interactions between an-

tennas and their supporting platform. Domain decompo-

sition was used by Sullivan and Carin to break up a

method-of-moments (MoM) problem into multiple, sim-

pler, solution regions [6]. Iterative method of moments and

iterative physical optics have been used to solve multi-

bounce problems such as the EM scattering from large

open-ended cavities [7,8]. Classical iterative methods have

also been applied extensively to compute the scattering

from rough surfaces. The forward-backward method de-

veloped by Holliday et al. [9], and the method of ordered

multiple interactions of Kapp and Brown [10], take ad-

vantage of the dominant forward and backward propaga-

tion of EM waves over a rough surface. The generalized

forward–backward method extended this work to include

an obstacle on the rough surface by modifying the matrix

splitting used in the forward–backward method [11]. Com-

parisons of stationary with nonstationary iterative meth-

ods are presented in Refs. 12–14.

The conjugate gradient (CG) method was developed in

1952 by Hestenes and Stiefel [15]. However, like the clas-

sical iterative methods, it was not used in the area of elec-

tromagnetics until advances in computers made it possible

to solve large linear systems. Sarkar and Rao used the CG

method to solve method of moments problems in 1984 [16],

and Sarkar and Arvas presented a more general CG de-

velopment for eletromagnetics problems in 1985 [17]. The

CG–fast Fourier transform method (CG-FFT) became pop-

ular for solving quasiplanar geometries in the late 1980s

[18,19]. The development of fast integral equation meth-

ods, such as the CG-FFT, the fast multipole method [20],

the adaptive integral method [21], and the precorrected

FFT method [22] gave CG methods a boost. These methods

greatly reduce the computational cost of applying the in-

tegral equation operator, thereby allowing very large sys-

tems of equations to be solved.

3. MATRIX NOTATION FOR ITERATIVE METHODS

The solution of a system of equations with N degrees of

freedom, or unknown variables, may be expressed in ma-

trix format as

Ax ¼b ð1Þ

where A is an NÂN system matrix, x is a column vector

containing the N unknown coefﬁcients, and b is a known

excitation-dependent column vector. The individual ele-

ments of this equation may be expressed as

b

m

¼

N

n¼1

A

mn

x

n

ð2Þ

This matrix equation is obtained by discretizing the EM

operator governing the problem of interest, whether it is

from a differential equation or integral equation formula-

tion. The unknown quantity, such as the EM ﬁelds or

equivalent currents, are expanded into a set of N

known basis functions with unknown coefﬁcients compris-

ing the column vector x. The N basis functions are tested

(or sampled) with N test functions to yield a system of N

equations.

The preceding equation for the unknown coefﬁcients

may be solved using direct matrix inversion or factoriza-

tion. However, the operational complexity for the direct

approach is O(N

3

), that is, of order N-cubed. This means

that the number of computations necessary to solve the

system is proportional to N

3

, which may be too costly when

there are thousands or millions of unknowns. Iterative

methods have an operational complexity of no more than

O(N

2

) per iteration, which is the cost of doing one matrix–

vector multiplication. So as long as the solution converges

quickly, the iterative method is much more efﬁcient.

Iterative methods seek to solve Eq. (1) by succes-

sively improving an initial solution to a desired degree of

2232 ITERATIVE METHODS

accuracy. The residual error vector is a measure of the

accuracy of the solution after k iterations and is deﬁned by

r

ðkÞ

¼b ÀAx

ðkÞ

ð3Þ

The residual error norm, or simply the residual error, is

the length of this vector normalized to the length of the

excitation vector, r

ðkÞ

_

_

_

_

= b k k, where jjr

ðkÞ

jj ¼

ﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃ

r

ðkÞ

r

ðkÞ

p

. Here,

the inner product (or vector product) of two column vectors

is Hermitian, deﬁned by

ab ¼

N

n¼1

a

Ã

n

b

n

ð4Þ

where the * superscript denotes the complex conjugate.

(Note: The CG algorithm described later does not use the

complex conjugate in the inner product, as will be made

apparent.) The residual error tells us how well the solu-

tion satisﬁes the system of equations, and is most often

used as the criterion for halting the iterations. The abso-

lute error vector is deﬁned by

e

ðkÞ

¼x

ðkÞ

Àx ð5Þ

where x is the exact solution to (1). The spectral radius of a

matrix is deﬁned as the magnitude of its largest eigenval-

ue. This quantity is important for determining conver-

gence of classical iterative methods, whereas the

eigenvalue spectrum of a matrix determines the conver-

gence of CG methods.

4. CLASSICAL ITERATIVE METHODS

As is apparent from the historical review presented ear-

lier, classical iterative methods are often used to solve

problems via a physical decomposition of the geometry,

sometimes even using a different solution technique for

each region. All of these methods can be cast in the form of

matrix splittings, where the original system matrix is de-

composed in some manner that makes the problem easier

to solve. Figure 2 shows some common matrix splittings.

We will focus on the lower–upper (LU) triangular split-

ting. The block-diagonal and banded matrix splittings are

extensions of the LU splitting, where the diagonal D is

replaced by the block-diagonal or banded portion of the

matrices. Likewise, the hybrid decomposition is a special

case of the block-diagonal splitting with only two blocks on

the diagonal.

All the matrix splittings of Fig. 2 have the general form

A¼MÀN. We may then write an iterative equation from

Eq. (1) as

Mx

ðkÞ

¼Nx

ðkÀ1Þ

þb ð6Þ

starting with some initial solution candidate x

(0)

and solv-

ing repeatedly. It is easy to show that if x

(k)

¼x

(k À1)

, then

Eq. (1) is satisﬁed and x

(k)

¼x. To solve (6) for x

(k)

, we need

Mto be easily invertible or factorizable. Diagonal matrices

are trivial to invert, and block-diagonal matrices are

easily inverted by inverting each block independently of

the other blocks. Lower triangular and upper triangular

matrices are also easy to invert via backward and forward

substitutions, respectively [1]. All of these types of inver-

sions are computed much more efﬁciently than inverting

the entire system matrix A.

The absolute error vector at the kth iteration may be

shown to be

e

ðkÞ

¼ðM

À1

NÞ

k

ðx

ð0Þ

ÀxÞ ð7Þ

Therefore, the spectral radius of the matrix ðM

À1

NÞ must

be less than unity to guarantee convergence [1]. This en-

sures that the absolute error approaches zero as k goes to

inﬁnity. The residual error vector for the kth iteration may

be shown to be

r

ðkÞ

¼Nðx

ðkÞ

Àx

ðkÀ1Þ

Þ

which is easily computed by saving the matrix–vector

product Nx

ðkÀ1Þ

from the previous iteration. Some com-

mon iterative algorithms based on the matrix splittings of

Fig. 2 are discussed next.

4.1. Jacobi Iteration

This is the simplest classical iteration algorithm. We

choose M¼D and N¼ À(LþU), so the iterative equation

becomes

Dx

ðkÞ

¼b ÀðLþUÞx

ðkÀ1Þ

ð8Þ

The magnetic ﬁeld integral equation (MFIE) has this

form, which is also used in the iterative physical optics

L

U

D

A

11

A

21

A

22

A

12

(a) (b)

(c) (d)

Figure 2. Some common matrix splittings. (a) hybrid decomposi-

tion; (b) lower–upper triangular; (c) block-diagonal; (d) banded.

ITERATIVE METHODS 2233

technique [8]. The operational cost is O(N

2

), which is the

cost of computing the matrix–vector product ðLþUÞx

ðkÀ1Þ

on the right-hand side (RHS) of (8).

4.2. Gauss–Seidel Method

This is an improvement over simple Jacobi iteration [1].

Here we choose M¼DþL and N¼ ÀU, resulting in

ðDþLÞx

ðkÞ

¼b ÀUx

ðkÀ1Þ

ð9Þ

This equation is solved using forward substitution. This is

easy to see by writing the expression for the individual

elements as

x

ðkÞ

m

¼

b

m

À

mÀ1

n¼1

x

ðkÞ

n

À

N

n¼mþ1

x

ðkÀ1Þ

n

D

m

ð10Þ

The elements x

ðkÞ

m

are updated sequentially for m¼

1,2,y,N, so the updated values can be used on the RHS

of (10). The convergence of Gauss–Seidel is expected to be

somewhat better than Jacobi, and with the same opera-

tional cost.

4.3. Symmetric Gauss–Seidel Method

The Gauss–Seidel method can be formulated using for-

ward or backward substitution. A symmetric form of

Gauss–Seidel iteration is obtained using both forward

and backward substitution in the following two-step algo-

rithm:

ðDþLÞx

ðkÀ1=2Þ

¼b ÀUx

ðkÀ1Þ

ðDþUÞx

ðkÞ

¼b ÀLx

ðkÀ1=2Þ

ð11Þ

This is the form of the forward–backward method [9],

or the method of multiple ordered interactions [10]. This

two-step algorithm has the same operational cost of a one-

step algorithm because the half-matrix–vector product

Lx

ðkÀ1=2Þ

is reused in step 2 of each iteration, and Ux

ðkÀ1Þ

may be saved from the previous iteration and reused.

4.4. Relaxation

Unless the problem geometry is very well ordered, or the

system matrix is strongly diagonally dominant, the clas-

sical iterative algorithms above will probably have poor

convergence properties. To improve convergence, a relax-

ation parameter [1] (or damping coefﬁcient), o may be in-

troduced. This is a constant usually in the range 0ooo2

such that the relaxed iterative equations reduce to the

basic equations above for o¼1. The relaxed Jacobi itera-

tive equation is given by

Dx

ðkÞ

¼ob þ ð1 ÀoÞDÀoðLþUÞ ½ x

ðkÀ1Þ

ð12Þ

It is easy to show that (12) reduces to (8) for o¼1, and

if x

ðkÞ

¼x

ðkÀ1Þ

then (1) is satisﬁed and x

(k)

¼x for any non-

zero o.

Likewise, the relaxed Gauss–Seidel method, also

known as successive overrelaxation (SOR) [1], is given by

ðDþoLÞx

ðkÞ

¼ob þ ð1 ÀoÞDÀoU ½ x

ðkÀ1Þ

ð13Þ

The relaxed form of symmetric Gauss–Seidel is known as

symmetric successive overrelaxation (SSOR) [1], and is

given by

ðDþoLÞx

ðkÀ1=2Þ

¼ob þ ð1 ÀoÞDÀoU ½ x

ðkÀ1Þ

ðDþoUÞx

ðkÞ

¼ob þ ð1 ÀoÞDÀoL ½ x

ðkÀ1=2Þ

ð14Þ

Figure 3 shows a plot of the convergence of relaxed Jacobi,

SOR, SSOR, and the biconjugate gradient stabilized

(BCGS) algorithms for the problem of radar scattering

from a perfect electrically conducting cylinder computed

by the method of moments [14]. For this relatively simple

problem, the SSOR method has the best convergence and

the BCGS, the worst. However, for more arbitrary geom-

etries the classical iterations may fail to converge, and

may eventually diverge.

5. CONJUGATE-GRADIENT (CG) METHODS

CG methods are superior to classical iterative methods

in the sense that they are theoretically guaranteed to

converge in no more than N iterations if the matrix is

0 10 20 30 40 50 60 70 80 90 100

0.001

0.002

0.003

0.005

0.01

0.02

0.03

0.04

0.05

0.1

0.2

0.3

0.4

0.5

1

R

e

s

i

d

u

a

l

e

r

r

o

r

n

o

r

m

Iteration units

SSOR

SOR

FB

Jacobi

Bi-CGStab

2m

2m

x

z

y

Frequency 300 MHz

Vertical polarization

Elevation angle 30 deg

E

−−i

Figure 3. Convergence of classical iterative methods compared

with BCGS.

2234 ITERATIVE METHODS

nonsingular. (In practice, the numerical precision of the

computer may limit this theoretical convergence.) They

are not limited to specific types of problems or physics, and

can be used as general matrix solvers. This is achieved by

generating a sequence of search vectors that will eventu-

ally span the entire N-space. The only difference between

the various CG versions is how these search vectors are

generated. In general, the search vectors are chosen such

that each new vector is linearly independent of all previ-

ous vectors, and the residual error is minimized. The basic

CG method is applicable only to symmetric systems and is

presented here ﬁrst, followed by two popular methods for

nonsymmetric systems, the modiﬁed biconjugate gradient

(BCG) and the generalized minimum residual (GMRES)

methods. An excellent resource for these and other CG

algorithms is the Templates book [23], for which associat-

ed computer subroutines are readily available.

5.1. CG Algorithm for Complex Symmetric Matrices

The basic CG algorithm for solving the complex symmetric

matrix equation A~ xx ¼

~

bb is listed below. A complete deriva-

tion is included later in this article. In the following algo-

rithm the vector products are not Hermitian, that is, there

is no complex conjugation as in (4).

Conjugate Gradient Algorithm 1

Initialization:

~ vv ¼

~

bb

ﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃ

b

~

A

~

bb

_ ; ~ xx ¼0; ~ rr ¼

~

bb

Iteration:

1. ~ xx ¼ ~ xx þa~ vv; a ¼ v

~

~

bb

2. ~ uu¼A~ vv; ~ rr ¼ ~ rr Àa ~ uu

3. Check jj ~ rrjj=jj

~

bbjj e; if yes, then stop, and ~ xx is a good

approximation; if no, then continue.

4. ~ pp¼ ~ rr Àb~ vv; b¼ u

~

~ rr

5. ~ vv ¼ ~ pp=

ﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃ

p

~

A~ pp

_

6. Go to step 1.

Each iteration of the CG algorithm involves one matrix–

vector multiplication, which is at most an O(N

2

) operation.

The residual error of the solution is checked in step 3. If it

is less than some threshold error e, the iterations are halt-

ed. This threshold level determines the accuracy of the

solution. For most engineering applications a threshold

level in the range 0.0001–0.01 yields sufﬁcient accuracy.

Of course, greater accuracy requires more iterations, and

it is possible for the algorithm to stall before reaching a

given threshold. The convergence properties of the CG

method are discussed later in this article.

5.2. Modiﬁed Biconjugate-Gradient Method for

Nonsymmetric Matrices

In solving the magnetic ﬁeld integral equation or combined

ﬁeld integral equations or many other RF engineering

applications (such as hybrid ﬁnite-element/integral equa-

tion formulation), we quite often end up with a nonsym-

metric matrix equation A~ xx ¼

~

bb, where A is an NÂN

complex nonsymmetric matrix. There are many variants

of Krylov-based methods for solving this equation; here we

shall list the modiﬁed BCG method, followed by the

GMRES method. In the following algorithm the vector

products are not Hermitian; that is, there is no complex

conjugation as in (4).

Modiﬁed BCG Algorithm for Solving Complex Nonsymmetric

Matrix Equations

Initialization:

~ xx ¼ ~ xx

T

¼0 ~ rr ¼

~

bb ~ rr

T

¼

~

bb

~ pp¼ ~ rr ~ pp

T

¼ ~ rr

T

~ vv ¼

~ pp

ﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃ

p

~

T

A~ pp

_ ~ vv

T

¼

~ pp

T

ﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃ

p

~

A

T

~ pp

T

_

Iteration:

1.

~ xx ¼ ~ xx þa~ vv ¼ ~ xx þ v

~

T

~

bb

_ _

~ vv

~ xx

T

¼ ~ xx

T

þa

T

~ vv

T

¼ ~ xx

T

þ v

~

~

bb

_ _

~ vv

T

2. Compute

~ uu¼A~ vv

~ uu

T

¼A

T

~ vv

T

3.

~ rr ¼ ~ rr Àa ~ uu

~ rr

T

¼ ~ rr

T

Àa

T

~ uu

T

4. Check convergence. If jj ~ rrjj=jj

~

bbjj e, then stop and ~ xx

is a good approximation; if not, continue.

5.

~ pp¼ ~ rr Àb ~ pp b¼

u

~

T

~ rr

u

~

T

~ pp

~ pp

T

¼ ~ rr

T

Àb

T

~ pp

T

b

T

¼

u

~

~ rr

T

u

~

~ pp

T

6.

~ vv ¼

~ pp

ﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃ

p

~

T

A~ pp

_

~ vv

T

¼

~ pp

T

ﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃ

p

~

A

T

~ pp

T

_

7. Go to step 1.

In this BCG algorithm it is noted that two matrix–vector

products need to be computed for each iteration, one with

the original matrix A and the other with its transpose A

T

.

This makes the BCG methods roughly twice as computa-

tionally expensive as the CG method for symmetric sys-

tems. The difference between the basic BCG and the

modiﬁed BCG is that the former uses the Hermitian ma-

trix A

H

(i.e., complex conjugate–transpose) instead of the

transpose matrix A

T

.

ITERATIVE METHODS 2235

5.3. GMRES Method for Nonsymmetric Matrix Equations

Another useful Krylov space iterative method for nonsym-

metric systems is the generalized minimum residual

(GMRES) method [24]. Like the CG method, a sequence

of linearly independent search vectors is generated. How-

ever, unlike the CG method, the entire set of search vec-

tors is saved in memory. Coefﬁcients are found that give

the minimum residual error over the complete set of

search vectors. In essence, it is a ‘‘brute force’’ CG meth-

od. The advantages are that only one matrix–vector prod-

uct is computed per iteration and the transpose of the

matrix is not needed. Furthermore, the GMRES method

truly minimizes the residual at each iteration, so its con-

vergence is monotonic. The disadvantage is that all the

previous search vectors must be stored in memory. There-

fore, the memory requirement grows with the number of

iterations. This may not be a problem for dense system

matrices for which the matrix storage is generally much

larger than the storage of a set of search vectors (depend-

ing, of course, on how many search vectors are stored).

To alleviate the memory requirement, the GMRES al-

gorithm may be restarted after a certain number of iter-

ations. The solution vector after one set of iterations is

used as the initial solution for the next set of iterations.

However, the restarted version of the GMRES algorithm

is not guaranteed to converge because the reduced set

of expansion vectors may not span the entire solution

space. The GMRES algorithm is listed in the third-edition

book by Golub and Van Loan [1] and in Templates [23].

A simpliﬁed algorithm that is conceptually equivalent

to GMRES, the generalized conjugate residual (GCR)

method [25], is listed below. In the following algorithm

the vector products are Hermitian, using complex conju-

gation as in (4).

Generalized Conjugate Residual Algorithm

Initialization: x ¼0, r ¼b, p

1

¼b, u

1

¼Ap

1

Iteration: k¼1; 2; . . .:

1. x ¼x þap

k

, a ¼u

k

r= u

k

k k

2

.

2. r ¼r Àau

k

.

3. Check jjrjj=jjbjj e. If yes, then stop, and x is a good

approximation; if no, then continue.

4. b

i

¼

u

i

Ar

u

i

k k

2

, for i ¼1; 2; . . . ; k.

5. p

k þ1

¼r À

k

i ¼1

b

i

p

i

.

6. u

k þ1

¼Ar À

k

i ¼1

b

i

u

i

.

7. Go to step 1.

This algorithm is very similar to the basic conjugate-

gradient method. Note that only one matrix–vector prod-

uct is used per iteration (in step 4) if we store all the

vectors p

i

and u

i

for i ¼1; 2; . . . ; k: It is also helpful to store

u

i

k k

2

to avoid repeated computation in step 4. If storage

becomes excessive, the algorithm may be restarted after

the mth iteration starting with p

1

¼p

m

and u

1

¼u

m

.

6. PRECONDITIONERS FOR ITERATIVE METHODS

The convergence rate of iterative methods, both classical

and conjugate-gradient, can be very slow if the system

matrix is not well-conditioned. As mentioned in the sec-

tion on classical iterative methods, the convergence of

these methods depends on the spectral radius of the iter-

ation matrix ðM

À1

NÞ. Similarly, the convergence rate of

CG methods depends on the spectral properties of the sys-

tem matrix (see Section 7 for a discussion). Certain for-

mulations in electromagnetics give rise to poorly

conditioned systems, such as the electric ﬁeld integral

equation (EFIE). Sometimes the formulation may be

altered to give a better conditioned system, such as by

converting the EFIE to the combined ﬁeld integral equa-

tion. The choice of basis functions may also affect the con-

ditioning. Alternatively, one may apply a preconditioner

matrix M to the original system as

M

À1

Ax ¼M

À1

b

Clearly, if the inverse of M approximates the inverse of A,

then the solution of this system should be easier, or, math-

ematically speaking, the matrix M

À1

A should have better

spectral properties than the original matrix. The precon-

ditioner may be implemented in any iterative algorithm

by replacing matrix–vector products of the form Ap with

M

À1

Ap, and the excitation vector b with M

À1

b. There are

cleverer ways to do this as described in Section 7.

The preconditioner should improve convergence, while

its inverse M

À1

(or factorization) must be computed efﬁ-

ciently. It is not a coincidence that the preconditioner ma-

trix M uses the same symbol as the classical iterative

matrix splitting M. In fact, the M matrix of all of the clas-

sical iteration matrix splittings discussed here and shown

in Fig. 2 may be used as a preconditioner, namely, diago-

nal, block-diagonal, lower or upper triangular, and banded.

Classical splittings often mimic wave interactions, which

make them useful as preconditioners. From the matrix-

splitting point of view, we want the matrix Mto contain the

‘‘dominant’’ portion of the system matrix A. Then the in-

verse of M will approximate the inverse of A, and the it-

erative algorithm should therefore converge rapidly.

A very effective preconditioner for the EFIE with sub-

sectional basis functions is described in Ref. 26. The pre-

conditioner M is a sparse version of A, which contains the

matrix entries corresponding to basis interactions within

a speciﬁed distance. Incomplete factorization is used to

compute a sparse factorization of M. In fact, there is a

large class of preconditioners that use incomplete factor-

ization. Some common preconditioning approaches for

iterative algorithms are discussed in Refs. 1 and 23.

7. THEORY OF THE BASIC CG METHOD

The basic CG method is applicable only to symmetric sys-

tems. Consider the following complex symmetric matrix

equation

Ax ¼b ð15Þ

2236 ITERATIVE METHODS

where A is an NÂN complex symmetric matrix, x is the

solution column vector, and b is the RHS excitation col-

umn vector. Before we derive the CG method, let us try to

answer a few related questions ﬁrst.

A-Conjugate Condition. Given a set of basis column vectors

v

0

; v

1

; . . . ; v

nÀ1

f g with noN, how to determine the best ap-

proximate solution

~ xx

app

¼

nÀ1

i ¼0

c

i

~ vv

i

ð16Þ

that solves (15). The answer is the Galerkin method, or

weighted residual. For each column vector, we shall form a

residual column vector

~

RR¼

~

bb ÀA~ xx

app

ð17Þ

Note also that equation (16) can be written in matrix

form as

~ xx

app

¼V~ cc

V¼ ~ vv

0

~ vv

1

Á Á Á ~ vv

nÀ1

_ ¸

; ~ cc ¼

c

0

c

1

.

.

.

c

nÀ1

_

¸

¸

¸

¸

¸

¸

¸

¸

¸

¸

_

_

¸

¸

¸

¸

¸

¸

¸

¸

¸

¸

_

ð18Þ

Subsequently, by requiring that the residual vector

~

RR be

orthogonal to all the basis vectors is equivalent to solving

the coefﬁcient vector ~ cc through the following reduced ma-

trix equation

V

t

AV

_ _

~ cc ¼V

t

~

bb ð19Þ

Let us take a closer look at Eq. (19). If the reduced matrix

V

t

AV turns out to be an identity matrix, then the coefﬁ-

cients can be simply computed by

c

i

¼v

~

i

~

bb; v

~

¼ð ~ vvÞ

t

ð20Þ

What is more is that, as will be seen later in this section,

there is no need to store all these basis vectors in order to

ﬁnd the approximate matrix solution ~ xx

app

. Requiring

V

t

AV I implies that the basis vectors need to satisfy

v

~

i

A~ vv

j

¼d

ij

ð21Þ

which is called the A-conjugate condition.

Before moving on to derive the CG methods, lets take a

few moments to restate what we have discussed in a more

fundamental way. You see, as in many applications, to

solve equations, whether inﬁnite dimensional problems

(integral equation formulations), or ﬁnite-dimensional

problems (like matrix equations), the Galerkin method is

a very good method of choice. Once again, in applying the

Galerkin method, we shall need to establish what the trial

and test function spaces are. When the operators are sym-

metric, some would argue that they need to be positive

definite as well, we can simply have both the trial and test

functions be the same. The next logical question will be

how to generate these basis vectors that span the trial and

test function spaces. As basic linear algebra taught us,

these basis vectors at least need to be linearly indepen-

dent, preferably orthonormal. This is where the A-conju-

gate condition comes in. When the operator is symmetric,

we can, with some violation when the operator is not pos-

itive definite, deﬁne the vector inner product as

~ vv

i

; ~ vv

j

¸ _

¼v

~

i

A~ vv

j

ð22Þ

As you shall see, different definitions of the ‘‘inner prod-

uct’’ lead to different variants of CG methods.

For the matrix equation A~ xx ¼

~

bb with a nonzero initial

solution ~ xx

i

, it is always possible to solve for the correction

equation A~ xx

0

¼

~

bb

0

¼

~

bb ÀA~ xx

i

_ _

, and with ~ xx ¼ ~ xx

0

þ ~ xx

i

. There-

fore, without loss of generality, we shall assume that we

will solve A~ xx ¼

~

bb with initial guess zero. We shall derive

the CG method by induction.

k ¼0: With the initial solution ~ xx

ð0Þ

¼0, the residual vector

is simply ~ rr

ð0Þ

¼

~

bb. The trial space for solving the matrix

equation can now be established as

V

0

¼ ~ vv

0

½ ¼MGS

A

~ rr

0

f g ð23Þ

The notation MGS

A

~ aa

0

~ aa

1

Á Á Á ~ aa

nÀ1

_ _

means making

a orthonormal basis from the n column vectors,

~ aa

0

~ aa

1

Á Á Á ~ aa

nÀ1

, through the modiﬁed Gram–Schmidt

(MGS) process and the inner product is deﬁned by the

A-conjugate condition:

~ vv

0

¼

~ rr

0

ﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃ

r

~

0

A~ rr

0

_ ð24Þ

Notice that, in equation (24), we have an expression r

~

0

A~ rr

0

.

If the matrix A is positive definite, the expression r

~

0

A~ rr

0

will always be a positive nonzero number, thus Eq. (24)

will always be valid. Since in our case A is a complex sym-

metric matrix, it is possible that r

~

0

A~ rr

0

0 even though

~ rr

0

O0. This is referred to as ‘‘breakdown’’ in the CG meth-

od. Although, in practical computation, it rarely occurs,

but when the matrix A is poorly conditioned, it is possible

that r

~

0

A~ rr

0

% 0 and thus causes slow and even failure to

converge in the CG process. It should be emphasized here

that many researchers object to the use of CG method to

non-positive-definite matrix equations; in reality, with

good preconditioners (a topic which is of paramount im-

portance) the CG method may be used to solve complex

symmetric matrix equations.

k ¼1: The best solution in the trial space V

0

¼span ~ vv

0

f g,

from the Galerkin method, for the matrix equation

ITERATIVE METHODS 2237

A~ xx ¼

~

bb is

~ xx

ð1Þ

¼c

0

~ vv

0

; c

0

¼v

~

0

~

bb ð25Þ

Subsequently, the residual vector ~ rr

ð1Þ

can be obtained as

~ rr

ð1Þ

¼

~

bb ÀA~ xx

ð1Þ

¼ ~ rr

ð0Þ

Àc

0

A~ vv

0

ð26Þ

Since the solution is solved through Galerkin method, and

v

~

0

~ rr

ð1Þ

¼0, it is certain that ~ rr

ð1Þ

is linearly independent with

vectors from V

0

¼span ~ vv

0

f g; therefore it would be a good

idea to have

V

1

¼V

0

[ span ~ rr

ð1Þ

_ _

¼MGS

A

~ vv

0

~ rr

ð1Þ

_ _

ð27Þ

Consequently, our new basis vector is determined through

the modiﬁed Gram–Schmidt (MGS) process:

~ ww¼ ~ rr

ð1Þ

Àb~ vv

0

; b¼v

~

0

A~ rr

ð1Þ

¼r

~

ð1Þ

A~ vv

0

~ vv

1

¼

~ ww

ﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃ

w

~

A ~ ww

_

ð28Þ

It is easy to verify that v

~

0

A~ vv

1

¼v

~

1

A~ vv

0

¼0 and v

~

0

A~ vv

0

¼

v

~

1

A~ vv

1

¼1. Moreover, we see from equation (26), A~ vv

0

2 V

1

.

To summarize, at k ¼1, we have the following condi-

tions:

1. V

1

¼span ~ vv

0

~ vv

1

_ _

¼MGS

A

~ vv

0

~ rr

ð1Þ

_ _

2. V

t

1

AV

1

¼I

3. A~ vv

0

2 V

1

kth iteration: At this moment, we have the trial space

V

kÀ1

¼span ~ vv

0

~ vv

1

Á Á Á ~ vv

kÀ1

_ _

and it satisﬁes

1. v

~

i

A~ vv

j

¼d

ij

; i; j ¼0; 1; . . . ðk À1Þ

2. A~ vv

i

2 V

kÀ1

; i ¼0; 1; . . . k À2 ð Þ

The best matrix solution in the trial space V

kÀ1

¼

span ~ vv

0

~ vv

1

Á Á Á ~ vv

kÀ1

_ _

is then

~ xx

ðkÞ

¼

kÀ1

i ¼0

c

i

~ vv

i

¼

kÀ1

i ¼0

v

~

i

~

bb

_ _

~ vv

i

¼ ~ xx

ðkÀ1Þ

þ v

~

kÀ1

~

bb

_ _

~ vv

kÀ1

¼ ~ xx

ðkÀ1Þ

þa~ vv

kÀ1

ð29Þ

and of course, the residual vector is computed through

~ rr

ðkÞ

¼

~

bb ÀA~ xx

ðkÞ

¼ ~ rr

ðkÀ1Þ

ÀaA~ vv

kÀ1

ð30Þ

From the Galerkin method it follows that

v

~

i

~ rr

ðkÞ

¼0; i ¼0; 1; . . . k À1 ð Þ ð31Þ

Since A~ vv

i

2 V

kÀ1

; i ¼0; 1; . . . k À2 ð Þ, we also have

v

~

i

A~ rr

ðkÞ

¼0; i ¼0; 1; . . . k À2 ð Þ ð32Þ

Subsequently, the next basis vector will be computed by

~ pp¼ ~ rr

ðkÞ

À

kÀ1

i ¼0

b

i

~ vv

i

¼ ~ rr

ðkÞ

À

kÀ1

i ¼0

v

~

i

A~ rr

ðkÞ

_ _

~ vv

i

¼ ~ rr

ðkÞ

À v

~

kÀ1

A~ rr

ðkÞ

_ _

~ vv

kÀ1

¼ ~ rr

ðkÞ

Àb~ vv

kÀ1

ð33Þ

and

~ vv

k

¼

~ pp

ﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃ

p

~

A~ pp

_ ð34Þ

Consequently, ~ vv

0

~ vv

1

Á Á Á ~ vv

k

_ _

is an A-conjugate basis

for the trial space V

k

. This process continues until ~ rr

ðkÞ

_

_

_

_

is

very small at a certain iteration k; it then implies, for all

practical purposes, that ~ xx

ðkÞ

is the solution to the matrix

equation A~ xx ¼

~

bb. Note that the process is extremely simple

and the recursive nature of the process makes it possible

not to store all the basis vectors.

The detailed induction argument above leads directly

to the basic CG algorithm 1 listed earlier in this article.

7.1. Convergence Rate of Conjugate-Gradient Methods

There are two features that can make CG converge fast:

(1) eigenvalue clusters and (2), a good condition number of

the matrix. To see why eigenvalue clusters are good for CG

method, let’s look at the following theorem.

Theorem 1. Assume that matrix A, which is a diagonal-

izable NÂN symmetric matrix, has only k distinctive

eigenvalues, namely

lðAÞ ¼ l

0

l

0

. . . l

0

¸ﬄﬄﬄﬄﬄﬄﬄﬄﬄﬄﬄﬄﬄﬄ..ﬄﬄﬄﬄﬄﬄﬄﬄﬄﬄﬄﬄﬄﬄ¸

n

0

l

1

l

1

. . . l

1

¸ﬄﬄﬄﬄﬄﬄﬄﬄﬄﬄﬄﬄﬄﬄ..ﬄﬄﬄﬄﬄﬄﬄﬄﬄﬄﬄﬄﬄﬄ¸

n

1

. . . l

kÀ1

l

kÀ1

. . . l

kÀ1

¸ﬄﬄﬄﬄﬄﬄﬄﬄﬄﬄﬄﬄﬄﬄﬄﬄﬄﬄﬄﬄ..ﬄﬄﬄﬄﬄﬄﬄﬄﬄﬄﬄﬄﬄﬄﬄﬄﬄﬄﬄﬄ¸

n

kÀ1

_

_

_

_

_

_

n

0

þn

1

þ . . . þn

kÀ1

¼N

ð35Þ

then the dimension of the Krylov space

K

m

~ vv

0

; A ð Þ ¼ ~ vv

0

A~ vv

0

Á Á Á A

mÀ1

~ vv

0

_ _

ð36Þ

will always be bounded by k regardless of the initial vector

~ vv

0

and m:

dim K

m

~ vv

0

; A ð Þ ½ k ð37Þ

Proof: Let ~ ee

p

i

be the ith eigenvector corresponding to

eigenvalue l

p

of the matrix A:

A~ ee

p

i

¼l

p

~ ee

p

i

; i ¼0; 1; . . . n

p

À1

_ _

ð38Þ

2238 ITERATIVE METHODS

Since these eigenvectors form a complete set of basis vec-

tors, any column vector ~ vv

0

can be written as a linear com-

bination of these eigenvectors:

~ vv

0

¼

kÀ1

p¼0

n

p

À1

i ¼0

c

p

i

~ ee

p

i

¼

kÀ1

p¼0

~ vv

ðpÞ

0

~ vv

ðpÞ

0

¼

npÀ1

i ¼0

c

p

i

~ ee

p

i

ð39Þ

It then follows that

A~ vv

0

¼

kÀ1

p¼0

n

p

À1

i ¼0

l

p

c

p

i

~ ee

p

i

¼

kÀ1

p¼0

l

p

n

p

À1

i ¼0

c

p

i

~ ee

p

i

¼

kÀ1

p¼0

l

p

~ vv

ðpÞ

0

ð40Þ

Moreover, we have

A

n

~ vv

0

¼

kÀ1

p¼0

l

n

p

~ vv

ðpÞ

0

ð41Þ

This means that for any Krylov vector A

n

~ vv

0

, it can always

be written as a linear combination of k independent vec-

tors ~ vv

ð0Þ

0

; ~ vv

ð1Þ

0

Á Á Á ~ vv

ðkÀ1Þ

0

. Thus, we conclude that

dim½K

m

ð ~ vv

0

; AÞ ¼ dimf ~ vv

0

A~ vv

0

. . . A

mÀ1

~ vv

0

g k ð42Þ

regardless of the initial vector and the iteration number

m.

Consequently, in applying the CG method, or any

Krylov-based methods, to solve a matrix equation with

k distinctive eigenvalues, CG converges in at most k

iterations.

Next, let’s examine the effect of condition number on

the convergence rate of the CG methods. To gain more in-

sight, let us assume further that matrix A is an NÂN

symmetric positive definite (SPD) matrix. With this as-

sumption, we can state a fact that at the mth iteration, the

CG method produces the same solution as the following

minimization problem.

Minimization: Seek ~ xx

ðmÞ

2 K

m

~

bb; A

_ _

¼

~

bb A

~

bb Á Á Á A

mÀ1

~

bb

_ _

such that the quadratic form

x

~

Àx

~

ðmÞ

_ _

A ~ xx À ~ xx

ðmÞ

_ _

ð43Þ

is minimized.

Since A is SPD, and its eigenvectors form a complete

set of basis vectors, we can express the RHS vector

~

bb as

follows:

~

bb¼b

0

~ ee

0

þb

1

~ ee

1

þ Á Á Á þb

NÀ1

~ ee

NÀ1

¼

NÀ1

i ¼0

b

i

~ ee

i

ð44Þ

It is easy to show then the exact solution ~ xx is

~ xx ¼

b

0

l

0

~ ee

0

þ

b

1

l

1

~ ee

1

þ Á Á Á þ

b

NÀ1

l

NÀ1

~ ee

NÀ1

¼

NÀ1

i ¼0

b

i

l

i

~ ee

i

ð45Þ

Furthermore, a general trial vector in the Krylov space in

the mth iteration is of the form

~ vv ¼

mÀ1

i ¼0

c

i

A

i

~

bb

_ _

¼

NÀ1

i ¼0

c

0

þc

1

l

i

þ Á Á Á þc

mÀ1

l

mÀ1

i

_ _

b

i

~ ee

i

ð46Þ

Subsequently

~ xx À ~ vv ¼

NÀ1

i ¼0

1

l

i

1 Àc

0

l

i

Àc

1

l

2

i

ÀÁ Á Á Àc

mÀ1

l

m

i

_ _

b

i

~ ee

i

ð47Þ

and a quadratic functional F ~ vv ð Þ can be deﬁned as

F ~ vv ð Þ ¼ðx

~

Àv

~

ÞA ~ xx À ~ vv ð Þ ð48Þ

Substituting (46) and (47) into Eq. (48), we have

Fð ~ vvÞ ¼

NÀ1

i ¼0

½1 Àc

0

l

i

Àc

1

l

2

i

ÀÁ Á Á Àc

mÀ1

l

m

i

2

1

l

i

b

2

i

max

0iNÀ1

l

i

½1 Àc

0

l

i

Àc

1

l

2

i

ÀÁ Á Á Àc

mÀ1

l

m

i

2

NÀ1

i ¼0

1

l

i

b

2

i

¼ max

0iNÀ1

l

i

½1 Àc

0

l

i

Àc

1

l

2

i

ÀÁ Á Á Àc

mÀ1

l

m

i

2

x

~

A~ xx

¸

¸

¸

¸

¸

¸

¸

¸

ð49Þ

Since the CG solution is the same as the one that mini-

mizes the quadratic functional, we have

F ~ xx

ðmÞ

_ _

¼ min

~ vv2K

m

ð

~

bb;AÞ

Fð ~ vvÞ x

$

A~ xx

¸

¸

¸

¸

¸

¸ min

f c

0

c

1

Á Á Á c

mÀ1

g

max

0iNÀ1

l

i

½1 Àc

0

l

i

Àc

1

l

2

i

ÀÁ Á Á Àc

mÀ1

l

m

i

2

¼ x

$

A~ xx

¸

¸

¸

¸

¸

¸ min

P

m

ð0Þ ¼1

max

0iNÀ1

P

m

ðl

i

Þ

¸

¸

¸

¸

2

ð50Þ

where P

m

l ð Þ is the mth polynomial in l. If we arrange

the eigenvalues of A in ascending manner, namely,

l

0

l

1

Á Á Á l

NÀ1

, then we can replace the best appro-

ximation problem on the discrete set with the best

ITERATIVE METHODS 2239

approximation problem on the interval l

0

l

NÀ1

_ ¸

. Note

that we have

min

Pmð0Þ ¼1

max

0iNÀ1

l

i

P

m

ðl

i

Þ

¸

¸

¸

¸

min

Pmð0Þ ¼1

max

l

0

ll

NÀ1

P

m

ðlÞ

¸

¸

¸

¸

ð51Þ

The solution to the minmax problem on an interval is

known; namely

min

P

m

ð0Þ ¼1

max

l

0

ll

NÀ1

P

m

ðlÞ

¸

¸

¸

¸

¼

1

T

m

l

NÀ1

þl

0

l

NÀ1

Àl

0

_ _ max

l

0

ll

NÀ1

T

m

l

NÀ1

þl

0

À2l

l

NÀ1

Àl

0

_ _ ¸

¸

¸

¸

¸

¸

¸

¸

ð52Þ

where T

m

x ð Þ ¼

1

2

x þ

ﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃ

x

2

À1

p

_ _

m

þ x À

ﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃ

x

2

À1

p

_ _

m

_ _

is the

Chebyshev polynomial. Also, since max

À1x1

T

m

x ð Þ

¸

¸

¸

¸

¼1 and

À1 ½ðl

NÀ1

þl

0

À2lÞ=ðl

NÀ1

Àl

0

Þ 1, we then ﬁnd

min

P

m

ð0Þ ¼1

max

l

0

ll

NÀ1

P

m

ðlÞ

¸

¸

¸

¸

¼

1

T

m

l

NÀ1

þl

0

l

NÀ1

Àl

0

_ _

¼2

s

m

1þs

2m

;

s ¼

1 À

ﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃ

l

0

l

NÀ1

_

1 þ

ﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃ

l

0

l

NÀ1

_

ð53Þ

In conclusion, the convergence rate of CG method, mea-

sured in A norm is

ﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃ

x

~

Àx

~

ðmÞ

_ _

A ~ xx À ~ xx

ðmÞ

ð Þ

¸

2

s

m

1þs

2m

ﬃﬃﬃﬃﬃﬃﬃﬃﬃ

x

~

A~ xx

_

ð54Þ

7.2. Preconditioned CG method

We conclude the previous section by observing that CG

method works well on matrices that are either well-con-

ditioned or have just a few distinct eigenvalues. For many

RF engineering applications (such as the electric ﬁeld in-

tegral equation), the system matrix equations are usually

not suitable directly for CG method. However, if a proper

preconditioning matrix, M¼C

t

C, can be found, then the

system matrix can be transformed into

A~ xx ¼

~

bb )A

0

~ zz ¼

~

bb

0

A

0

¼ C

t

_ _

À1

AC

À1

_ _

; ~ xx ¼C~ zz;

~

bb

0

¼ C

t

_ _

À1

~

bb

ð55Þ

Applying the CG algorithm 1 to the transformed matrix

equation results in the following algorithm.

CG Algorithm 2

Initialization: ~ vv

0

¼

~

bb

0

ﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃ

b

0

~

A

0 ~

bb

0

_ ; ~ zz ¼0; ~ rr

0

¼

~

bb

0

Iteration:

1. ~ zz ¼ ~ zz þa~ vv

0

; a ¼v

~

0 ~

bb

0

.

2. ~ uu

0

¼A

0

~ vv

0

; ~ rr

0

¼ ~ rr

0

Àa ~ uu

0

.

3. Check jj ~ rr

0

jj=jj

~

bb

0

jj e. If yes, then stop, and ~ zz is a good

approximation; if no, then continue.

4. ~ pp

0

¼ ~ rr

0

Àb~ vv

0

; b¼u

~

0

~ rr

0

.

5. ~ vv

0

¼ ~ pp

0

=

ﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃ

p

~

0

A

0

~ pp

0

_

.

6. Go to step 1.

Of course, once we have ~ zz, then we can obtain ~ xx via

~ xx ¼C

À1

~ zz. However, it is possible to avoid explicit refer-

ence to the matrix C

À1

by deﬁning ~ pp

0

¼C~ pp; ~ zz ¼C~ xx and

~ rr

0

¼ðC

t

Þ

À1

~ rr in every CG iteration. Indeed, if we substi-

tute these definitions into CG algorithm 2 and recall

~

bb

0

¼ðC

t

Þ

À1

~

bb, then we obtain

CG Algorithm 3

Initialization:

C~ vv ¼

C

t

_ _

À1

~

bb

ﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃ

b

~

C

À1

C

t

_ _

À1

_ _

A C

À1

C

t

_ _

À1

_ _

~

bb

¸ ; ~ zz ¼0; ~ rr ¼

~

bb

Iteration:

1. C~ xx ¼C~ xx þaC~ vv; a ¼v

~

~

bb.

2. C

t

_ _

À1

~ uu¼ C

t

_ _

À1

A~ vv; C

t

_ _

À1

~ rr ¼

C

t

_ _

À1

~ rr Àa C

t

_ _

À1

~ uu.

3. Check jj ~ rr

0

jj=jj

~

bb

0

jj e, If yes, then stop, and ~ xx is a good

approximation; if no, then continue.

4. C~ pp¼ C

t

_ _

À1

~ rr ÀbC~ vv; b ¼ u

~

C

À1

C

t

_ _

À1

~ rr.

5. C~ vv ¼C~ pp=

ﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃ

~ ppA~ pp

_

.

6. Go to step 1.

Finally, the entire algorithm can be simpliﬁed by using

the preconditioner M¼C

t

C directly instead of referring to

C or C

t

. This is then the preconditioned CG algorithm.

Preconditioned CG Algorithm

Initialization: ~ zz ¼0; ~ rr ¼

~

bb; ~ pp¼M

À1

~ rr; ~ vv ¼ ~ pp=

ﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃ

p

~

A~ p p

_

Iteration:

1. ~ xx ¼ ~ xx þa~ vv; a ¼v

~

~

bb.

2. ~ uu¼A~ vv; ~ rr ¼ ~ rr Àa ~ uu.

3. Check jj ~ rrjj=jj

~

bbjj e. If yes, then stop, and ~ xx is a good

approximation; if no, then continue.

4. ~ pp¼M

À1

~ rr Àb~ vv; b ¼u

~

M

À1

~ rr.

5. ~ vv ¼ ~ pp=

ﬃﬃﬃﬃﬃﬃﬃﬃﬃﬃ

p

~

A~ pp

_

.

6. Go to step 1.

2240 ITERATIVE METHODS

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ed., Johns Hopkins Univ. Press, Baltimore, 1996.

2. T. J. Kim and G. A. Thiele, A Hybrid diffraction technique—

general theory and applications, IEEE Trans. Anten. Propag.

30:888–897 (1982).

3. M. Kaye, P. K. Murthy, and G. A. Thiele, An iterative method

for solving scattering problems, IEEE Trans. Anten. Propag.

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(1997).

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Dongarra, V. Eijkhout, R. Pozo, C. Romine, and H. van der

Vorst, Templates for the Solution of Linear Systems: Building

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FURTHER READING

A. F. Peterson, S. L. Ray, and R. Mittra, Computational Methods

for Electromagnetics, Wiley, New York, 1997 (a good general

resource for computational electromagnetics, including the ﬁ-

nite-element and ﬁnite-difference methods, the method of mo-

ments, basis expansions, and solution methods).

O. Axelsson, Iterative Solution Methods, Cambridge Univ. Press,

1996 (a good source for iterative methods, in general). Matrix

Computations by Golub and Van Loan [1] is an excellent ref-

erence on matrix theory, solution of matrix systems, and iter-

ative algorithms.

The Templates book [23] presents many iterative algorithms and

their underlying theories, along with a discussion of precondi-

tioners and parallelization; it is available online at http://

www.netlib.org/linalg/html_templates/Templates.html, and

the associated software may be downloaded from http://

www.netlib.org/templates/.

ITS RADIO SERVICE STANDARDS AND

WIRELESS ACCESS IN VEHICULAR

ENVIRONMENTS (ITS-WAVE) AT 5.9GHz

RAMEZ L. GERGES

IEEE-ITSC Standards

Committee

Goleta, California

1. INTRODUCTION

This article describes ongoing activities to create a new

family of standards that supports the emerging Intelligent

ITS RADIO SERVICE STANDARDS AND WIRELESS ACCESS IN VEHICULAR ENVIRONMENTS (ITS-WAVE) AT 5.9 GHz 2241

Transportation Systems (ITS) and telematics wireless

markets. ITS-WAVE is a radiocommunication system in-

tended to provide seamless, interoperable services to sur-

face transportation systems. After an initial overview of

the ITS-WAVE family of standards, more emphasis will be

given to the radio (lower layers) part of the system, and

the use of orthogonal frequency-division multiplexing

(OFDM) for the physical layer [1].

1.1. ITS, Telematics, and Wireless Interoperability

The Intelligent Transportation Systems (ITS) initiative

was created by Congress in the Intermodal Surface Trans-

portation Efﬁciency Act of 1991 (ISTEA) to improve the

mobility and safety of the surface transportation system.

ITS is deﬁned as those systems utilizing synergistic tech-

nologies and systems engineering concepts to develop and

improve transportation systems of all kinds. Communica-

tion and information technologies are at the core of road-

side infrastructure and in-vehicle systems. These

technologies promise to enhance mobility by improving

the way we monitor and manage trafﬁc ﬂow, clear inci-

dents, reduce congestion, and provide alternate routes to

travelers. The telematics industry is focused on driver

comfort and safety, and while ‘‘telematics’’ in general

has meant the blending of computers and telecommuni-

cations, it is used within the ITS community with the

connotation of ‘‘automotive telematics’’ or the in-vehicle

subsystem of ITS.

In 1999, the Federal Communications Commission

(FCC) allocated the 5.850–5.925-GHz band for use by the

ITS radio service for both public safety and commercial

ITS applications. Many standards development organiza-

tions (e.g., IEEE, IETF, ISO) are engaged in the process of

achieving an end-to-end ITS wireless interoperability.

This article addresses Wireless Access in Vehicular Envi-

ronments (WAVE), which is currently being developed un-

der the IEEE WG 802.11, WAVE Study Group.

1.2. ITS Radio Services

The proposed ITS-WAVE standard addresses broadband

wireless communications that operate in a long range

(r1000m) and at a high data rate [27Mbps (megabits

per second)] for all ITS applications. The proposed lower-

layer standard currently addresses communications

between roadside units and mostly high-speed, but occa-

sionally stopped and slow-moving, vehicles or between

high-speed vehicles. The ITS new spectrum will be used to

support multiple applications to enhance public safety and

transportation mobility and can be categorized as follows:

1. Public Safety: The primary use of this band is to offer

services such as emergency vehicle signal preemp-

tion and announcements for work zones. While the

FCC has allocated the 4.9GHz for communications

between ﬁrst responders, the 5.9-GHz band is ex-

pected to allow ﬁrst responders to communicate with

the general driving public on roads and freeways.

2. Mobility: Services such as electronic toll, vehicle

probes, traveler information, and public transporta-

tion integration are expected to enhance the trans-

portation system performance.

3. Driver Safety: New features such as support of col-

lision avoidance and warnings for excessive speed

and railroad crossings are expected to improve sys-

tem performance. More recently, vehicle manufac-

turers and telematics providers have shown interest

in the ITS-WAVE standards. There is no other ra-

diocommunication technology that can support the

real-time requirements for vehicle-to-vehicle com-

munications.

1.3. ITS-WAVE Development History

Attempts to develop standards for the wireless ITS envi-

ronment date back to the early 1990s, when California

adopted the Title 21 regulation to achieve a common stan-

dard for ‘‘toll collections.’’ The dedicated short-range com-

munications (DSRC) standard at 900 MHz, and Title 21

[2], predated the ITS initiative, and addressed only the

electronic toll collection; it was not intended to support a

national interoperable wireless ITS standard.

The Intermodal Surface Transportation Efﬁciency Act

of 1991 (ISTEA) funded many research ITS programs. In

the mid-1990s, the author (then with the New Technology

program at Caltrans) initiated some of the ﬁrst technical

studies to develop an integrated wireless communications

system for all ITS applications [3]. In 1996, the U.S. Na-

tional System Architecture identiﬁed wireless communi-

cations as one of the critical enabling technologies needed

to support many of the ITS services. Later, the USDoT

funded more studies, and the California Department of

Transportation (Caltrans) established the Testbed Center

for Interoperability (TCFI) to study and test end-to-end

wireless interoperability. In May 1997, the Intelligent

Transportation Society of America (ITSA) ﬁled a Petition

for Rulemaking, requesting that the FCC allocate 75 MHz

of spectrum in the 5.850–5.925-GHz band on a coprimary

basis for DSRC-based ITS services. In 1998 at the IEEE–

Vehicular Technology Conference, the author suggested to

leverage the economical feasibility of the IEEE 802.11 to

achieve wireless ITS interoperability [4]. In 1999, the FCC

amended Parts 2 and 90 of the Commission’s Rules to al-

locate the 5.850–5.925-GHz band to the Mobile Service for

Dedicated Short Range Communications of Intelligent

Transportation Services.

1

The USDoT funded the Ameri-

can Society for Testing and Materials (ASTM) to initiate

the standard writing group for the DSRC at 5.9 GHz. In

2000, TCFI tested the ﬁrst video relay to a moving vehicle

at highway speed using OFDM technology.

2

The success-

ful test paved the way to use broadband technologies for

wireless ITS. Later, the ASTM selected the OFDM Forum

proposal to use the IEEE 802.11a [5,6] as the basis for the

new standard.

3

The new DSRC standard is now being

1

ET Docket 98-95, 14 FCC Record 18221.

2

Wireless LAN provided the OFDM equipment at 2.4 GHz; Fur-

ther infromation is available at http://www.wi-lan.com.

3

The OFDM-Forum proposal (802.11 RA) suggested changing the

physical layer of the IEEE 802.11 to match the requirements of

the ‘‘road access’’ environment. (http://www.ofdm-forum.com).

2242 ITS RADIO SERVICE STANDARDS AND WIRELESS ACCESS IN VEHICULAR ENVIRONMENTS (ITS-WAVE) AT 5.9 GHz

completed within the IEEE WG 802.11. The Study Group

(SG) decided to use the ASTM standard [7] as the basis for

the ITS-WAVE proposal.

4

As the proposed standard is not

limited to short-range applications, the SG has named it

the Wireless Access in Vehicular Environments (WAVE)

instead of DSRC. This will also avoid any confusion with

the single carrier technology in use in the United States

(900-MHz band) or in Japan and Europe (at 5.8 GHz but

different standards).

On December 17, 2003 the FCC adopted the rules

for the ITS band. It is expected that the new standard

(possibly 802.11p) will be completed by the end of 2005.

2. ITS RADIO SERVICES SYSTEM-LEVEL DESCRIPTION

2.1. Spectrum Allocation for ITS, Telematics, and Public

Safety

The Broadband ITS Radio Service (ITS-RS) establishes a

common framework for providing wireless services in the

5.850–5.925-GHz band. This band is allocated for ITS-RS

applications by the FCC.

5

Figure 1 shows the spectrum

allocation in the 4.9–5.9-GHz band. The differences be-

tween the ITS-WAVE and the IEEE 802.11 WLAN sys-

tems stem from the fact that the ITS-WAVE operates in a

licensed band, and it establishes reliable communications

between units operating at full vehicle mobility, a different

environment than the indoor WLAN.

These communications may occur with other units that

are (1) ﬁxed along the roadside or above the roadway, (2)

mounted in other high-speed moving vehicles, (3) mounted

in stationary vehicles, (4) mounted on mobile platforms

(e.g., watercraft, buoy, or a robotic platform), or (5) porta-

ble or handheld. In-vehicle communications units are

called onboard units (OBUs). Communication units ﬁxed

along the freeways, over the road on gantries or poles, or

off the road in private or public areas, are called roadside

units (RSUs). The WAVE RSUs may function as stations

or as access points (APs) and the WAVE OBUs only have

functions consistent with those of stations (STAs). The

common function between all RSUs is that these station-

ary units control access to the radiofrequency medium for

OBUs in their communication zone or relinquish control to

broadcast data only.

The vehicular mobility environment requires that we

design a system that can survive both the time-dispersive

(frequency-selective) multipath fading and the frequency-

dispersive (time-selective) fading environment. Tests con-

ducted at the Testbed Center For Interoperability (TCFI)

at UCSB show that we may encounter up to 400ns of

delay spread and up to 2200 Hz of Doppler spread as

explained later. Single-carrier transmission, with a time-

domain equalizer, has an inherent limitation due to con-

vergence and tracking problems which arises as the

number of taps increase. A coded OFDM (COFDM)

approach similar to the IEEE 802.11a/g standard offered

a more robust, as well as economically feasible solution

ITS-RS

China

Japan

USA

Europe

4.900 5.000

5.150 5.250

5.030 5.091

5.150

5.150

5.350

5.350 5.470

Indoor 200 mW / Outdoor 1 W EIRP

DFS & TPC DFS & TPC

4.900 5.000 5.100 5.200 5.300 5.400 5.500 5.600 5.700 5.800 5.900 6.000

DFS: Dynamic Frequency Selection

TPC: Transmit Power Control

Freq./GHz

Homeland Security

Indoor 200 mW EIRP Outdoor 1W EIRP

ITS-RS

Max mean

Tx power

Max peak

Tx power

5.725

5.725

5.85

5.825

5.850 5.725

5.925

Outdoor 4W EIRP

Frequency allocations 4.9-5.9 GHz:

Figure 1. Spectrum allocation at the 5-GHz Band. (This ﬁgure is available in full color at http://

www.mrw.interscience.wiley.com/erfme.)

4

The ASTM E2213-02 was approved but not published because of

copyright issues with the IEEE.

5

Title 47, Code of Federal Regulations (CFR), Part 90, Subpart M.

ITS RADIO SERVICE STANDARDS AND WIRELESS ACCESS IN VEHICULAR ENVIRONMENTS (ITS-WAVE) AT 5.9 GHz 2243

based on the success of the WLAN industry. This economic

feasibility also made the COFDM approach a better can-

didate than the single-carrier transmission with a fre-

quency-domain equalizer. Although, the latter has the

same complexity and may have some good features [e.g.,

avoids the PAPR (peak-to-average power ratio) issues].

The 802.11a scheme would not be able to tolerate the

delay spread expected in the WAVE environment. Figure 2

shows the impact of delay spread on a 16-QAM signal

constellation for a 64-subcarrier OFDM system [1]. The

channel has a two-ray multipath; the second ray is 6dB

lower than the ﬁrst one: (1) delay spread less than guard

time (Fig. 2a), (2) delay spread greater than guard time

by 3% of the FFT interval (Fig. 2b), and (3) delay

spread greater than guard time by 9% of the FFT inter-

val (Fig. 2c).

We proposed to double the guard interval (GI) to be

more multipath-tolerant; in principle, using half the mas-

ter clock should double the GI and scale down the channel

bandwidth to 10 MHz, a desired outcome to increase the

number of channels within the allocated spectrum. Of

course the maximum data rate will be reduced to 27 Mbps,

which is still adequate for demanding ITS applications

(e.g., video relay). WLAN chips manufacturers (e.g., In-

tersil and Atheros) conﬁrmed the feasibility of the ap-

proach using their current 802.11a implementations.

6

It is

expected that products with the correct front end operat-

ing at 5.9GHz (10 MHz bandwidth) will be available as the

market develops.

In order to accommodate the more dynamic vehicle en-

vironment with essentially the same radio technology, and

provide priority to public safety communications, the ITS

community is proposing a complementing set of standards

under the IEEE SCC32. These standards address the

upper layers including a different channel structure and

access mechanism than that of the IEEE 802.11 as ex-

plained later.

2.2. System Architecture and SDO Coordination

The International Standards Organization (ISO) and

the IEEE are coordinating their standards development

efforts to achieve ITS wireless interoperability. To this end

a common CALM/WAVE architecture has been developed

as shown in Figs. 3 and 4.

The current scope of the IEEE-WAVE proposed project

is to create an amendment of IEEE 802.11 to support com-

munication between vehicles and the roadside and be-

tween vehicles while operating at speeds up to a minimum

of 200km/h for communication ranges up to 1000 m. The

amendment will support communications in the 5-GHz

bands; specifically, the 5.850–5.925-GHz band within

North America with the aim to enhance the mobility and

safety of all forms of surface transportation, including rail

and maritime transportation. Amendments to the PHY

and MAC will be limited to those required to support com-

munications under these operating environments within

the 5-GHz bands.

The IEEE SCC32 sponsors the IEEE P1556, DSRC

Security and Privacy and the ITS-WAVE (upper layers).

The WG P1556 is proposing a dual-certificate system for

public safety and vehicle safety to balance security and

anonymity requirements. The IEEE WG P1609 architec-

ture adopted IPV6 as the method of handling upper-layer

applications. It consists of a series of four standards:

1. P1609.4 deﬁnes the channelization approach and

considers integration issue with the IEEE 802.11e

and IEEE 802.11 h.

2. P1609.3 is based on the IPv6 speciﬁcation and may

include a broad range of supporting standards de-

ﬁned by the Internet Engineering Task Force

(IETF). It deﬁnes IPv6 addressing and conﬁgura-

tion issues, network services (e.g., WAVE router ad-

vertisement), and all the WAVE management

entities needed for registration and ‘‘service table’’

exchanges.

3. P1609.2 deﬁnes applications services.

4. P1609.1 deﬁnes a resource manager for onboard

units (OBUs).

The ISO-Transport Information & Control (TC204)

Working Group 16 is developing standards for wide-area

wireless communications for transport information and

control. ISO-TC204-WG16 is developing the communica-

tion air interface for long and medium range (or short

media) (CALM) architecture. The CALM scope includes

0

0 2 4 6 −2 −4 −6 0 2 4 6 −2 −4 −6 0 2 4 6 −2 −4 −6

−2

−4

−6

2

4

6

(a)

0

−2

−4

−6

2

4

6

0

−2

−4

−6

2

4

6

(b) (c)

Figure 2. Impact of delay spread. (This ﬁgure is available in full color at http://www.mrw.

interscience.wiley.com/erfme.)

6

WLAN products from Intersil are now part of Conexant (http://

www.conexant.com), Atheros ( http://www.atheros.com).

2244 ITS RADIO SERVICE STANDARDS AND WIRELESS ACCESS IN VEHICULAR ENVIRONMENTS (ITS-WAVE) AT 5.9 GHz

communications between ﬁxed sites and switching be-

tween communication media (e.g., 3G cellular and

WAVE), as well as issues such as handover and mobility

management. CALM mandates end-to-end system inter-

operability at all levels. CALM-M5 is adopting the IEEE-

WAVE proposal for the lower layers at 5 GHz.

2.3. Basic Concept of Operation

The ITS-WAVE typically consists of two types of radio

devices. The ﬁrst type is always used while stationary,

usually permanently mounted along the roadside, and

is referred to as the roadside unit (RSU). The second is

SNMP agent

(RFC 1157)

App 1

App Data

Sockets

App 2

App Data

Sockets

OBU IVN

App 3

App Data

Sockets

UDP

Networking Services

IVN

L2/L1

UDP (RFC 768)

Networking Services (IPv6 – RFC 2460)

SNMP

MIB

SME

(1609.3)

Logical Link Control (802.2)

Channelization (1609.4)

MAC (802.11p)

PHY (802.11p)

WME

(1609.3)

MLME

(802.11p)

PLME

(802.11p)

IVN

L2/L1

IVN: In- Vehicle Network

Figure 3. Wave architecture.

CME

(Commun

ication

Managem

ent

Entity)

ISO

LME

(Link

Managem

ent Entity)

Common

Station,

PHY,

MAC, LLC

Managers

Service Access Point – Management Service Access Point – Data Transfer

CALM M5

CALM M5

3G

cellular

std

CALM 3G

NETWORK INTERFACE

Routing and Media Switching based on IPv6

ISO 21210-2

Directory

Services

Convergence

Layer

Convergence

Layer

Layer 5-7

INTERNET

Non-CALM-

aware

Point-to-point

Non-CALM-

aware

IP (Internet)

CALM-Aware

APPLICATIONS

Figure 4. CALM architecture. (This ﬁgure is available in full color at http://www.mrw.

interscience.wiley.com/erfme.)

ITS RADIO SERVICE STANDARDS AND WIRELESS ACCESS IN VEHICULAR ENVIRONMENTS (ITS-WAVE) AT 5.9 GHz 2245

mobile, mounted on board vehicles, and is referred to as

the onboard unit (OBU). Three types of communication

are supported: a command/response type between a ser-

vice provider and a service user, a broadcast to listener,

and a peer-to-peer type that does not identify either device

as controlling the actions of the other. OBUs and RSUs

can initiate both types of communication. The command/

response type includes various forms of transactions be-

tween a service provider and a user of that service. To en-

sure scalable interoperability between ITS-WAVE units,

the proposed standards deﬁne two levels of implementa-

tions. A minimal implementation only supports the lower

layers, those below the network layer, and will be referred

to as a ‘‘WAVE radio.’’ An implementation that has the full

ITS-WAVE protocol stack is referred to as a ‘‘WAVE de-

vice.’’ Multiple devices interact with each other through

the Networking Services (IEEE P1609).

Figure 5 represents the current ITS-WAVE band plan.

The ITS-WAVE uses a ‘‘control channel’’ and any of six ‘‘ser-

vice channels’’. Licensing of both roadside RSUs and OBUs

are necessary to prevent unauthorized use of the control

channel. OBUs should be licensed by rule, since these

devices are mobile and can operate nationwide, communi-

cating with any other ITS-WAVE devices within range.

The onboard units (OBUs) are required to listen on the

control channel every few hundred milliseconds, in order

to check for public safety messages. The messages on the

control channel are of variable length, but are generally

kept short, to permit maximum access to the channel.

Control channel access will be performed via a standard

IEEE 802.11a, Multiple Access with Collision Avoidance

(CSMA/CA). By default, all devices when turned on are

tuned to the control channel. If an ITS-WAVE device

desires to transmit, but detects another message being

broadcast on the control channel, it must wait before at-

tempting to transmit. A request to send (RTS) is initiated,

and time is granted ﬁrst to high priority (public safety)

broadcasts, then to lower-priority transmissions. The

same control channel is used for roadside-to-vehicle, ve-

hicle-to-roadside, and vehicle-to-vehicle communications.

‘‘Control channel interval’’ and ‘‘service channel interval’’

are controlled by RSU beacon frames. Since the control

channel will be ﬁxed throughout the nation, all ITS-WAVE

devices will be able to access those services in an interop-

erable matter.

A registration process must occur before a WAVE device

can be considered ‘‘ready for operation’’; the RSU broad-

casts beacon frames that include the ‘‘provider service ta-

ble’’ (PST) and the ‘‘WAVE router advertisement’’ (WRA)

on the control channel. Application initialization proce-

dures are based on SNMP, and the designated service

channel, priority, and power level are indicated in the

PST. At the end of the application initialization state,

the RSU commands the OBU to switch to the designated

service channel. The RSU, now on the service channel,

receives UDP datagrams sent by the OBU. The RSU

routes datagrams to and from the applications indicated

by the global IPv6.

The description above is included to give an idea about

the basic concept of operations, with the understanding

that the proposed standards are now under development.

The P1609.3, 1609.4, and the P1556 are currently the

most critical part of the WAVE family of standards as they

require integration and coordination with many other

standards such as IEEE 802.11e/h/i and many of the

IETF recommendations.

Shared public safety/private

Control Med Rng Service Short Rng Service

Dedicated public safety

High avail Intersections

40 dBm

Not currently implemented

Power limit

Power limit

Power limit

Not currently implemented

44.8 dBm

33 dBm

23 dBm

Uplink

Downlink

Public

safety

Veh-Veh

Ch 172

Public

safety/

private

Ch 174

Public

safety/

private

Ch 176

Control

channel

Ch 178

Public

safety/

private

Ch 180

Public

safety/

private

Ch 182 Ch 184

Public safety

intersections

Canadian special license zones*

Frequency (GHz)

5

.

8

2

5

5

.

8

3

0

5

.

8

3

5

5

.

8

4

0

5

.

8

4

5

5

.

8

5

0

5

.

8

6

0

5

.

8

6

5

5

.

8

7

0

5

.

8

7

5

5

.

8

8

0

5

.

8

8

5

5

.

8

9

0

5

.

8

9

5

5

.

9

0

0

5

.

9

0

5

5

.

9

1

0

5

.

9

1

5

5

.

9

2

0

5

.

9

2

5

5

.

8

5

5

Figure 5. ITS-wave band plan. (This ﬁgure is available in full color at http://www.mrw.

interscience.wiley.com/erfme.)

2246 ITS RADIO SERVICE STANDARDS AND WIRELESS ACCESS IN VEHICULAR ENVIRONMENTS (ITS-WAVE) AT 5.9 GHz

It is expected that WAVE radios that implement only

the lower layers will develop ﬁrst, as they leverage the

existing IEEE 802.11a standard and chip technology.

3. DESIGNING FOR WIRELESS VEHICULAR

ENVIRONMENTS

3.1. Channel Impairments

There is extensive literature on different statistical mod-

els of the communication channels at different frequency

bands [8]. However, limited data of actual ﬁeld measure-

ments for vehicle–vehicle and vehicle–roadside communi-

cation are available at the ITS-WAVE frequency band.

Statistical models include large-scale path loss models and

small-scale fading models:

1. Large-scale propagation models characterize the

mean received power over large transmitter–receiv-

er separation distances. It is used to estimate radio

coverage area of a transmitter.

2. Small-scale (fading) models characterize the rapid

ﬂuctuations of the received signal strength and

phase over a very short distance. Multipath struc-

ture (power delay proﬁle) is used to measure and

describe the fading effects.

Both large- and small-scale fading models are needed

for packet error rate characterization.

3.1.1. Time-Dispersive (Frequency-Selective) Multipath

Fading Channel. A time-dispersive channel is deﬁned as a

channel for which the delay spread is much wider than the

signal duration. The classiﬁcation of a channel as time

dispersive is therefore dependent on the data rate of the

system. For a single carrier, high-data-rate systems, time-

dispersive channels are commonly encountered. This type

of fading is often referred to as frequency-selective because

the signal may be simultaneously faded at one frequency

and not at another. OFDMis robust against delay spread by

design because of the longer symbol time and the fact that

each subcarrier experiences a ﬂat-fading channel. Similar

to the IEEE 802.11a, the insertion of guard interval, and

the use of forward error correction (FEC) are essential de-

sign elements to the coded OFDM scheme employed in the

WAVE physical layer. This multipath rejection capability

was one of the main reasons for selecting COFDM instead

of a single-carrier system, especially for ITS applications

that operate at longer ranges and at high data rates.

Short-range systems typically experience significantly

smaller delay spreads than does a longer-range system.

Previous studies show that 90% RMS (root-mean-square)

delay spread is less than 100ns for typical short-range

applications (e.g., toll collection) in urban environments.

7

RMS delay spread could be up to 300 ns [9,10] in a non-

line-of-sight (NLoS) heavy-multipath environment, as

may be expected in a freeway urban environment.

In order for subcarriers to perceive a ﬂat-fading chan-

nel, the bandwidth (subcarrier spacing) must be less than

the ‘‘coherence bandwidth’’ (B

c

) of the channel.

8

B

c

is the

bandwidth of the channel variation in frequency and is

deﬁned [8] as

B

c

¼1=5s

where s is the RMS delay spread of the channel. The ITS-

WAVE has a subcarrier spacing (bandwidth) of 156kHz,

and each subcarrier will encounter ﬂat fading as long as

so100 ns (for rangeo300m). For long ranges (large delay

spread), the pilot channels are available to estimate the

channel in the frequency domain if they are well struc-

tured. In order to use the pilots for channel estimation, the

pilot spacing in frequency has to be less than B

c

(B2 MHz

for s ¼500ns). This may not be the case using the current

pilot structure of the IEEE 802.11a (pilot spacing ¼14 Â

156 kHz ¼2.18 MHz4B

c

). Interpolation of pilot subcar-

rier in the current structure may not be sufﬁcient to track

the frequency selective fades. It is expected that the ﬁrst

generation WAVE radios, those using modiﬁed 802.11a

chips, will be limited in range and may not be suitable for

long-range public safety applications.

3.1.2. Frequency-Dispersive (Time-Selective) Fading

Channel. Frequency-dispersive channels are classiﬁed as

channels that have a Doppler spread larger than the

channel bandwidth. Doppler spread is a direct result of

multiple Doppler shifts which are caused by motion of the

transmit and/or receive antenna. Doppler shifts can also

result from reﬂections off of moving objects.

9

Distortion of

the power spectrum of the received signal results from

Doppler spread, which can be approximated by the Dopp-

ler spread B

d

B

d¼

f

m

.

cos a

where f

m¼

v

.

f

c

/c, where v is the vehicle speed in m/s, f

c

is

the carrier frequency in Hz, c is the speed of light in m/s,

and a is the angle between the direction of vehicle travel

and the ray of the communication path. In the case of the

ITS-WAVE where vehicle speeds of r120 mph (193 km/h)

must be supported (public safety), the maximum Doppler

shift for a vehicle traveling directly toward the roadside

antenna would be about 1100Hz at 5.9 GHz, and much

less for vehicle–vehicle communication (two vehicles head-

ing in the same directions).

Time-selective fading caused by Doppler spread is

described by the coherence time (T

c

) of the channel. T

c

represents the duration over which the channel character-

istics do not change significantly, and is deﬁned [8] as

T

c

¼0:423=f

m

7

For transmit–receive (Tx/Rx) separation of 30–300m, both LoS

and NLoS (Xiongwen, etc.; IEEE JSAC 2002).

8

B

c

is deﬁned as the bandwidth over which the frequency corre-

lation function is above 0.5.

9

If a sinusoidal signal is transmitted over a fading channel (com-

monly referred to as a constant wave), the Doppler spread B

d

is

deﬁned as the range of frequencies over which the received Dopp-

ler spectrum is essentially nonzero.

ITS RADIO SERVICE STANDARDS AND WIRELESS ACCESS IN VEHICULAR ENVIRONMENTS (ITS-WAVE) AT 5.9 GHz 2247

At a vehicle speed of 120 mph and a frequency of 5.9GHz,

T

c

¼400ms. When using pilot symbols at the start of a

packet, the assumption is that channel variations during

the rest of the packet are negligible. This limits the packet

duration to less than T

c

, and places an upper limit on the

packet size. At a data rate of 3Mbps,

1

2

-code-rate BPSK-

modulated signal, the maximum packet size

10

is 135 bytes.

Although this suggests that higher-order modulation

would give better performance, as their transmission

time is shorter, these modulation schemes degrade more

in the presence of channel impairment.

3.2. The ITS-WAVE Physical Layer

The ITS-WAVE physical layer is based on using the

robustness of the coded orthogonal frequency-division

multiplexing (OFDM) signal to achieve the required per-

formance in the wireless vehicular environments. OFDM

is a special case of multicarrier modulation (MCM), which

is the principle of transmitting data by dividing the data-

stream into several parallel bitstreams and modulating

each bitstream onto individual subcarriers. Each subcar-

rier is a narrowband signal, resulting in long bit intervals.

High data rates are achieved by using multiple orthogonal

subcarriers for a single data transmission. The OFDM

system differs from traditional MCM in that the spectra of

the subcarriers were allowed to overlap under the restric-

tion that they were all mutually orthogonal. An orthogo-

nal relationship between subcarriers is achieved if there

are integer numbers of subcarrier frequency cycles over

the symbol interval. This orthogonality guarantees that

each subcarrier has a null at the center frequency of all

other subcarriers as shown in Fig. 6.

Orthogonality is achieved with precision by modulating

the subcarriers with a discrete Fourier transform (DFT),

which is implemented in hardware with the fast Fourier

transform (FFT). By transmitting several symbols in par-

allel, the symbol duration is increased proportionately,

which reduces the effects of intersymbol interference (ISI)

caused by the dispersive fading environment. Additional

multipath rejection and resistance to intercarrier inter-

ference (ICI) is realized by cyclically extending each sym-

bol on each subcarrier. Rather than using an empty guard

space, a cyclic extension of the OFDM symbol is used to

ensure that delayed replicas of the OFDM symbol will

always have an integer number of cycles within the FFT

interval. This effectively converts the linear convolution of

the channel to a circular one, as long as the cyclic preﬁx

(CP) is longer than the impulse response of the channel.

The penalty of using a CP is loss of signal energy propor-

tional to the length of the CP. In order to avoid excessive

bit errors on individual subcarriers that are in a deep fade,

forward error control (FEC) is typically applied.

The ITS-WAVE physical layer organizes the spectrum

into operating channels. Each 10-MHz channel is com-

posed of 52 subcarriers. Four of the subcarriers are used as

pilot carriers for monitoring path shifts and ICI, while the

other 48 subcarriers are used to transmit data symbols.

Subcarriers are spaced 156.25 kHz apart, giving a total

bandwidth of 8.8MHz. The composite waveform, consist-

ing of all 52 subcarriers, is upconverted to one of the seven

channels between 5.850 and 5.925 GHz. As shown

in Fig. 7, channels are numbered from À26 to 26. Sub-

carrier 0 is not used for signal processing reasons, and

pilot subcarriers are assigned to subcarriers À21, À7, 7,

and 21. To avoid strong spectral lines in the Fourier trans-

form, the pilot subcarriers transmit a ﬁxed bit sequence as

speciﬁed in the IEEE 802.11a using a conservative mod-

ulation technique. Table 1 compares the ITS-WAVE and

the IEEE 802.11a parameters. Table 2 lists ITS-WAVE

baseband modulation values.

3.2.1. Structure of the WAVE Physical Layer. The phy-

sical layer is structured as two sublayers: the physical-

layer convergence procedure (PLCP) sublayer and the

physical-medium-dependent (PMD) sublayer. The PLCP

communicates to MAC via primitives through the physi-

cal-layer service access point (SAP); it prepares the PLCP

protocol data unit (PPDU) shown in Fig. 8. The PPDU

provides for asynchronous transfer of the MAC protocol

data unit (MPDU) between stations. The PMD provides

Table 1. Comparison of 802.11a and ITS-WAVE Parameters

Parameter 802.11a ITS-WAVE

Channel bandwidth (MHz) 20 10

Subcarrier spacing (kHz) 312.5 156.25

T

FFT

(ms) 3.2 6.4

T

GI

(ns) 800 1600

T

SYM

(ms) 4 8

Channel symbol rate (Msps) 12 6

Minimum data rate (BPSK) (Mbps) 6 3.0

Maximum data rate (64-QAM) (Mbps) 54 27

A B C D E Tone

Figure 6. Subcarrier Orthogonality in OFDM systems. (This

ﬁgure is available in full color at http://www.mrw.interscience.

wiley.com/erfme.)

Center

frequency

Carrier

number

−25 −20 −15 −10 −5 5 15 20 25 10

0

Figure 7. Structure of an Operating Channel.

10

Packet duration¼[10(16þ2)(80þ1)(80þ135)(8)(2/48)(80)] Â

100ns ¼400ms.

2248 ITS RADIO SERVICE STANDARDS AND WIRELESS ACCESS IN VEHICULAR ENVIRONMENTS (ITS-WAVE) AT 5.9 GHz

actual transmission and reception of the physical layer

entities via the wireless medium, interfaces directly to the

medium, and provides modulation and demodulation of

the transmission frame.

3.2.2. Roles of Preamble, Training Sequences, and Pi-

lots. The ITS-WAVE speciﬁes a preamble at the start of

every packet as shown in Fig. 9.

The PLCP preamble consists of 10 short training sym-

bols, each of which is 1.6 ms, followed by two long training

symbols, each of which is 6.4 ms including a 3.2-ms preﬁx

that precedes the long training symbol. The long training

sequence contains a guard interval, T

GI2

, and two long

training symbols, each 6.4ms in duration. The short sym-

bols are used by the receiver for synchronization [signal

detection, AGC (automatic gain control), diversity selec-

tion, frequency offset estimation, and timing synchroniza-

tion]. The long symbols are used to ﬁne-tune the frequency

offset and channel estimates. This training sequence is

transmitted over all 52 subcarriers and is QPSK-modu-

lated. In terms of algorithmic complexity, carrier frequen-

cy offset and timing recovery are by far the most difﬁcult to

determine. The phase-locked-loop (PLL) on the radio sub-

system is responsible only for maintaining the 5-ppm volt-

age-controlled oscillator (VCO) requirement. Digital

signal processing is used, independent of the VCO, to

remove the carrier frequency offset. It is important

to note that once the carrier frequency offset is deter-

mined by the digital baseband hardware, there is no time

to provide a feedback signal to the WAVE radio’s VCO

since a PLL network will take too long to eliminate the

offset. The training sequences are followed by the SIGNAL

symbol, which is a single BPSK-modulated OFDM data

symbol containing information about the packet such as

data rate.

After preamble transmission, any common frequency

offset is tracked via the four pilot subcarriers as shown in

Fig. 10. It is not necessary to use pilots to estimate the

channel as long as the channel remains fairly stationary

over the duration of a single packet. The four pilot signals

facilitate coherent detection throughout the duration of

the packet. The remaining subcarriers carry the data

body of the packet. The pilot spacing is selected to be

less than the coherent bandwidth of the channel, as

explained later.

3.2.3. ITS-WAVE Performance Issues. The performance

of an OFDM receiver is affected by several factors, most

of which fall into the categories of hardware limita-

tions and channel impairments. Hardware limitations,

Table 2. ITS-WAVE Baseband Modulation

Data Rate (Mbps) Code Rate Modulation N_CBPS N_DBPS

3

1

2

BPSK 48 24

4.5

3

4

BPSK 48 36

6

1

2

QPSK 96 48

9

3

4

QPSK 96 72

24

1

2

16-QAM 192 96

18

3

4

16-QAM 192 144

24

2

3

64-QAM 288 192

27

3

4

64-QAM 288 216

PLCP - Header

Rate

4-bits

Reserved

1-bit

Length

12-bits

Parity

1-bit

Tall

6-bits

Service

16-bits

PSDU

Tall

6-bits

Pad

bits

Coded - OFDM

BPSK rate = 1/2

Coded - OFDM

Rate Indicated by signal symbol

PPDU

Data

Variable number of OFDM symbols

Signal

(1) OFDM symbol

PLCP preamble

12 - symbols

Figure 8. PPDU frame.

Signal detect,

AGC, diversity

selection

Coarse freq.

offset

estimation

timing

synchronize

Channel and

fine frequency

offset estimation

RATE

LENGTH

SERVICE + DATA DATA

16 + 16 = 32 s

10 x 1.6 = 16 s 2 x 1.6 + 2 x 6.4 = 16.0 s

1.6 + 6.4 = 8.0 s 1.6 + 6.4 = 8.0 s 1.6 + 6.4 = 8.0 s

t

1

t

2

t

3

t

4

t

5

t

6

t

7

t

8

t

9

t

10

G12 T

1

T

2 GI GI GI SIGNAL Data 1 Data 2

Figure 9. ITS-WAVE PLCP structure.

ITS RADIO SERVICE STANDARDS AND WIRELESS ACCESS IN VEHICULAR ENVIRONMENTS (ITS-WAVE) AT 5.9 GHz 2249

particularly clock accuracy and oscillator stability, affect

the synchronization accuracy of the receiver. The channel

impairments discussed in Section 3.1 include ‘‘delay

spread’’ and ‘‘Doppler spread,’’ which result in frequency-

selective fading and time-selective fading, respectively.

OFDM is extremely sensitive to receiver synchroniza-

tion imperfections, which can cause degradation of system

throughput and performance. The overlap between sub-

carriers leads to a system that is extremely sensitive to

imperfections in carrier frequency synchronization. Also,

multiplexing symbols onto multiple subcarriers results in

a system that is extremely sensitive to imperfections in

timing synchronization. This requires that the receiver

architecture be structured to correct for frequency, timing,

and sampling. Figure 11 is a simpliﬁed block diagram [1]

depicting the major processing modules associated with

the ITS-WAVE physical layer.

3.2.3.1. Synchronization. Synchronization is a big hur-

dle in OFDM systems. The ITS-WAVE physical layer uses

the same synchronization scheme as in the IEEE 802.11a;

it usually consists of three processes:

1. Frame detection

2. Carrier frequency offset estimation and correction

3. Sampling error correction

Frame detection is used to determine the symbol

boundary so that correct samples of the symbol frame

can be taken. The ﬁrst 10 short symbols are identical and

are used for frame detection. The received signal is corre-

lated with the known short-symbol waveform that pro-

duces correlation peaks. The received signal is also

correlated with itself with a delay of one short symbol,

which creates a plateau for the length of 10 short symbols.

If the correlation peaks are within the plateau, the last

peak is used as the position from where the start of the

next symbol is determined.

Frequency offset estimation uses the long training

symbols, which are two FFT symbols back-to-back. The

Frame detection:

10 short symbol

Frequency offset

estimation:

Two FFT symbol

back-to-back

Data

Pilot

52 sub-carriers

Frequency

S

y

m

b

o

l

Figure 10. Pilot structure. (This ﬁgure is

available in full color at http://www.mrw.

interscience.wiley.com/erfme.)

Coding Interleaving

Binary input data

Binary output data

QAM

mapping

Pilot

insertion

Serial to

parallel

RF TX DAC

Parallel

to serial

Add cyclic

extension and

windowing

IFFT (TX)

FFT (RX)

Deinterleaving Decoding

QAM

demapping

Channel

correction

Parallel

to serial

RF RX ADC

Remove cyclic

extension

Timing and

frequency

synchronization

Serial to

parallel

Symbol timing

Figure 11. Basic OFDM block diagram.

2250 ITS RADIO SERVICE STANDARDS AND WIRELESS ACCESS IN VEHICULAR ENVIRONMENTS (ITS-WAVE) AT 5.9 GHz

corresponding chips of the two FFT symbols are then

correlated to estimate the frequency offset. Channel esti-

mation uses the same two OFDM symbols as the frequen-

cy offset estimation. Once the frame start is detected,

frequency offset is estimated and signal samples are com-

pensated, the two long symbols are transformed into fre-

quency domain by FFT. After performing FFT on the

preambles, the frequency-domain values are compared

with the known preamble values to determine the chan-

nel response.

3.2.3.2. Carrier Frequency Offset. The ITS-WAVE (like

the 802.11a) speciﬁes that the carrier frequency and sym-

bol clock be derived off the same oscillator. This allows the

receiver to compute symbol clock drift directly from the

carrier frequency offset (e.g., ppm error). Frequency syn-

chronization must be applied before the FFT. Without a

carrier frequency offset, the peak of any subcarrier corre-

sponds to the zero crossings of all other subcarriers. When

there is a random frequency offset, there is no longer an

integer number of cycles over T

FFT

, resulting in ICI. The

degradation in SNR that occurs due to random frequency

offset is approximated by D [1] in decibels

D %

10

3 ln 10

ðpDFT

FFT

Þ

2

E

s

N

0

DF is the frequency offset and W ( ¼1/T

FFT

) is the band-

width of the composite OFDM waveform (subcarrier spac-

ing). In essence, any carrier frequency offset results in a

shift of the received signal in the frequency domain. This

frequency error results in energy spillover between sub-

carriers, resulting in loss of their mutual orthogonality.

The approximation states that the degradation increases

with the square of normalized frequency offset. The major

tradeoffs encountered when selecting an appropriate car-

rier frequency offset correction algorithm include speed,

accuracy, and performance under noisy conditions.

Short training symbols can recognize offsets as high as

312.5 kHz [

1

2

Â(1/1.6 ms)]. However, their short duration

results in reduced accuracy since they produce only 16-

point FFT samples per symbol. Although there are 10

short training symbols, 5 or 6 are consumed during RSSI,

AGC, and timing recovery. Long training symbols provide

a much more accurate estimate of the frequency offset

since they produce 4 times as many FFT points compared

to the short training symbol. However, their long extent

limits the discernable frequency offset to 78 kHz [

1

2

Â

(1/6.4 ms)] as shown in Fig. 12. Noise imparts variance on

the ﬁnal offset estimate, thereby mitigating its accuracy.

3.2.3.3. Symbol Timing. Errors in symbol timing syn-

chronization manifest as ISI and nonuniform phase shift

to the constellation points. Both of these effects naturally

lead to degradation of bit error rate (BER). The fast fourier

transform (FFT) demodulation process accumulates over

exactly one 6.4-ms OFDM interval. If the start of the sym-

bol time is not accurately established, the FFT demodu-

lation process will operate on two adjacent symbols

leading to ISI as shown in Fig. 13. Coarse synchroniza-

tion can resolve to within half the sampling period and

remove ISI. However, the residual sampling time offsets

must be identiﬁed, or a nonuniform phase shift will be

imparted to the constellation points.

The 6.4-ms FFT window is divided up into 64 time in-

stants separated by 100ns. Each point of the FFT is com-

puted at a rate of 10 Msps (megasamples, i.e., 10,000

samples, per second), which corresponds to 64 discrete

frequency-domain samples of the composite 6.4-ms symbol.

Since these 64 samples are 100ns apart [T

s

¼100ns], the

Long sync and

Data symbol spectrum:

52 sub-carriers

0 freq

Coarse frequency estimate must place 52 sub-carriers

To within ½ frequency-bin of their true location: +/− 78.125 KHz

0.15625 MHz

freq

0

Short sync spectrum:

12 sub-carriers

+1 +1 +1 −1 −1 −1 −1 −1 +1 +1 +1 +1

0.625 MHz

Figure 12. Carrier frequency offset.

T

Multi-path components

T

g

Sampling start

max

T

x

T

Figure 13. ISI and sampling-time offset. (This ﬁgure is available

in full color at http://www.mrw.interscience.wiley.com/erfme.)

ITS RADIO SERVICE STANDARDS AND WIRELESS ACCESS IN VEHICULAR ENVIRONMENTS (ITS-WAVE) AT 5.9 GHz 2251

range of the maximum detectable sampling offset ranges

from À50 to þ50 ns. This sampling time offset manifests

itself in the frequency domain as a phase shift, which is

proportional to the subcarrier frequency. Subcarriers at

the high end of the frequency range are affected dispro-

portionately relative to those subcarriers at the low end.

The effect of this phase shift on BER can be devastating, as

symbols that map to subcarriers at the edges will experi-

ence a phase shift that rotates the constellation point out

of its reliable detection region.

Sampling frequency offset does not negatively impact

performance on a ‘‘symbol per symbol’’ basis. However, it

can have harmful effects over large numbers of symbols.

The ITS-WAVE proposal calls out a 5ppm static center

frequency offset from the VCO for analog-to-digital/digi-

tal-to-analog clocks and carrier VCOs. At 10 MHz, a 5ppm

ﬁgure corresponds to a 50 Hz offset, which means that one

of the clocks is toggling 50 Hz faster than the other. In the

period of one 10-MHz clock (100 ns), one clock will advance

past the other by 0.5 psc. If we take into account the num-

ber of samples per symbol and the number of symbols in a

large packet, we ﬁnd that over a time span of 50 symbols

the sampling instants for symbols will have shifted by

2 ns.

11

This timeshift will manifest itself in the frequency

domain as a phase shift proportional to the subcarrier fre-

quency. This is clearly a receiver steady-state issue, and

can’t be detected during training. During receiver track-

ing, this offset is taken care of by processing the pilots and

feeding back corrections to an interpolator.

3.2.4. ITS-WAVE Adjacent-channel and Cochannel Inter-

ference. Effects of adjacent-channel and cochannel inter-

ference has been studied using simulation [11], a Simulink

model developed to evaluate these types of interference as

shown in Fig. 14.

In the model shown in Fig. 14 we consider the type of

the device and apply the corresponding spectrum mask as

Bernoulli random

binary generator

Bernoulli

binary

Bernoulli random

binary generator1

Bernoulli

binary

Transmitter

In1

In1 Out1

Out1

Interferer

Int_gain

Tx_gain

K−

[80x1]

[96×1]

[96×1]

[80x1] [80x1]

K− K−

[80x1]

K−

[80x1] [80x1] [80x1]

Tx_path_loss

[80x1]

Receiver sensitivity

[80x1]

In 1 Out 1

[96x1]

[96x1]

Tx

Rx

Error rate

calculation

In1 Out1

Receiver

[96x1]

In_path_loss

[80x1]

AWGN

AWGN

Channel

OFDM demo:

Initial settings

Figure 14. Enhanced simulation model. (This ﬁgure is available in full color at http://

www.mrw.interscience.wiley.com/erfme.)

Table 3. ITS-WAVE Device Class Spectral Mask

Device

Class

7 4.5 MHz

Offset

75MHz

Offset

75.5MHz

Offset

710MHz

Offset

715MHz

Offset

A 0 À10 À20 À28 À40

B 0 À16 À20 À28 À40

C 0 À26 À32 À40 À50

D 0 À35 À45 À55 À65

Table 4. ITS-WAVE Classes and Transmit Power Levels

Device Class Maximum Device Output Power (dBm)

A 0

B 10

C 20

D 28.8

Table 5. ITS-WAVE Receiver Sensitivity

Data Rate (Mbps) Minimum Sensitivity (dBm)

3 À85

4.5 À84

6 À82

9 À80

12 À77

18 À70

24 À69

27 À67

11

0.5psc/sample Â80 samples/symbol Â50 symbols ¼2 ns.

2252 ITS RADIO SERVICE STANDARDS AND WIRELESS ACCESS IN VEHICULAR ENVIRONMENTS (ITS-WAVE) AT 5.9 GHz

given in Table 3. The model also considers the fact that the

devices operate at the maximum power output according

to Table 4, which reﬂects the increase of the out-of-band

attenuation for higher power devices. The model takes

into account the minimum receiver sensitivity as per

Table 5. The channel path loss is modeled according

to the two-segment model with a breakpoint of 164m as

given by

LðdÞ

dB

¼20 log ðdÞ þ43:05 ðdo164mÞ

LðdÞ

dB

¼40 log ðdÞ À1:263 ðd

3

64 mÞ

This is typical for models that use ray tracing [12], where

the path loss is generally proportional to 1/d

2

before the

breakpoint and 1/d

4

after the breakpoint. The breakpoint

represents the point at which the ﬁrst Fresnel zone touch-

es the ground, wherein the reﬂected ray off the surface of

the ground cancels some of the power of the direct ray. The

breakpoint is approximated by d

bp

D(4h

t

h

r

)/l, where h

t

is

the transmit antennal height and h

r

is the receive anten-

na height.

3.3. The ITS-WAVE MAC Layer

Generally, for reliable system operation, the MAC must be

properly designed to match the physical layer so that its

impairments do not cause undue degradation at higher

layers. The IEEE 802.11 MAC is a very complex protocol;

it took over 10 years of development with the support of

dozens of corporations developing products for the WLAN

market. The ITS-WAVE Study Group intends to use the

IEEE 802.11a MAC without modiﬁcation, except for

changes to the management information base (MIB).

The management information specific to each layer is rep-

resented as a MIB for that layer. The generic model of

MIB-related management primitives exchanged across

the management SAPs is to allow the SAP user entity

to either GET the value of a MIB attribute, or to SET the

value of a MIB attribute. The invocation of a SET:request

primitive may require that the layer entity perform

certain deﬁned actions. Figure 15 depicts these generic

primitives.

4. CHALLENGES AND FUTURE DEVELOPMENTS

4.1. Validation, Veriﬁcation, and Testing

Developing the ITS-WAVE family of standards is a com-

plex task, and the fact that these standards will be sup-

porting safety applications makes validation, veriﬁcation,

testing, and system integration critical steps for develop-

ing this market. The USDoT has begun this process

through funding the Vehicle Safety Communications Con-

sortium (VSCC) and other industry participants. At the

Caltrans Testbed Center For Interoperability (TCFI), we

have developed lab and ﬁeld infrastructure [11] in order

to support these activities as the standards mature.

Figure 16 shows data collected over the air using Agilent’s

equipment (VSG, VXI, and PSA) at TCFI. In addition, we

have demonstrated passing data between the test equip-

ment and the simulation tool (Simulink). Validation of the

MAC layer will be a special challenge; currently we are

experimenting with Telelogic’s TAU G2, which supports

both the Speciﬁcations and Description Language (SDL)

and the Uniﬁed Modeling Language (UML).

12

SDL is an

ITU formal language that was used to describe the orig-

inal IEEE 802.11 MAC speciﬁcations.

4.2. System Integration: The Santa Barbara Radio

Access Network

Beyond the development of the ITS-WAVE family of stan-

dards and the availability of telematics products, the de-

velopment of the ITS wireless market will require reliable

roadside infrastructure [13]. This infrastructure requires

feeder and backhaul networks that may use both ﬁxed

wireless and landline networks. Figure 17 shows one such

infrastructure that has passed the planning stage: Santa

Barbara’s Radio Access Network (RAN) [14]. The RF plan-

ning of 28 sites has been completed, and some sites are

Data link

Physical layer

MAC

MAC SAP

MAC

Management

Entity

PHY

Management

Entity

PLCP

PMD SAP

MIB

MIB

PLME GET/SET

PLME GET/SET

Station

Management

Entity

MLME GET/SET

PHY SAP

PMD

Figure 15. GET and SET operations.

12

Further information is available at http://www.telelogic.com.

ITS RADIO SERVICE STANDARDS AND WIRELESS ACCESS IN VEHICULAR ENVIRONMENTS (ITS-WAVE) AT 5.9 GHz 2253

being installed through the collaboration of Caltrans,

local governments, and the university (UCSB). SB-RAN

is currently part of a new proposal to develop a public

safety testbed, which addresses wireless infrastructure

interoperability (WII) issues for both the ‘‘ﬁrst responder’’

(4.9 GHz) and the ITS (5.9-GHz) bands.

4.3. Observations and Future Developments

While WAVE standards efforts are progressing within the

IEEE, there are remaining questions yet to be answered

regarding many issues such as the following:

*

Security architecture (P1556 and 802.11i)

*

Multiple-channel devices and the current ‘‘concept of

operations’’

*

Interference mitigation in a real environment

*

MAC extension and its relation to IEEE 802.11e/h

*

‘‘Pilot structure’’ and its impact on dedicated public

safety channels

*

Fast handover

*

IP-based internetworking

*

Wireless infrastructure interoperability

As we move from the descriptive phase of the standard

development to the performance and testing phases, the

need for better protocol development tools will continue to

be a challenge.

Roadside system integration issues are rarely ad-

dressed in the wireless ITS community. Issues such as

cost-effective feeder and backhaul networks to support the

wireless infrastructure are considered implementation

issues and are outside the scope of the national efforts.

4

2

*

6

9

7

8

10

11

13

12

14

15

16

18

17

22

20

19

23 25

27

21 26

24

28

1

3

5

*

TV Hill

*

Monopole

Figure 17. Santa Barbara’s RAN site locations.

(This ﬁgure is available in full color at http://

www.mrw.interscience.wiley.com/erfme.)

Figure 16. Over-the-air ITS-WAVE signal—18 Mbps. (This ﬁgure is available in full color at

http://www.mrw.interscience.wiley.com/erfme.)

2254 ITS RADIO SERVICE STANDARDS AND WIRELESS ACCESS IN VEHICULAR ENVIRONMENTS (ITS-WAVE) AT 5.9 GHz

Presently, the ITS industry is addressing market-enabling

applications such as vehicle safety and toll 0applications.

The more general ITS ‘‘public safety’’ applications such as

work-zone safety (WZS), public protection disaster relief

(PPDR), and homeland security, are assumed to be the

role of public agencies.

The potential gains from considering advanced tech-

nologies such as software-deﬁned radios (SDR), and mul-

tiple-input/multiple-output (MIMO) systems have not yet

been investigated for wireless ITS applications.

5. SUMMARY AND CONCLUSION

This article examined the emerging ITS-WAVE family of

standards, with emphasis on the mobile vehicle environ-

ment and the lower-layer standard. Our ﬁndings con-

ﬁrmed the validity of adopting the IEEE 802.11a as the

basis for the broadband wireless ITS standard. The OFDM

Forum proposal, originally submitted to the ASTM for

wireless road access, is now well understood and it has

been accepted by the ITS industry.

The physical-layer, proposal is ready for standardiza-

tion with the exception of the new pilot structure issue.

Long-range dedicated public safety cannot be realized

without resolving this issue. The data-link layer proves

to be more challenging, as we integrate other IEEE stan-

dards (e.g., 802.11e/h/i).

As new devices and systems are introduced, there will

be a need for demonstration projects, testing standards,

compliance certiﬁcation, and performance benchmarks.

Acronyms

AGC

ASTM

BER

BPSK

CALM

Caltrans

CFR

CP

CSMA/

CA

DFT

DoT

DSRC

FCC

FFT

ICI

IEEE

IETF

ISI

ISO

ISTEA

ITS

ITSA

ITS-RS

Automatic gain control

American Society for Testing and Materials

Bit error rate

Binary phase shift keying

Communications air interface for long and

medium range

California Department of Transportation

Code of Federal Regulations

Cyclic preﬁx

Multiple access with collision avoidance

Discrete Fourier transform

Department of Transportation

Dedicated short-range communications

Federal Communications Commission

Fast Fourier transform

Intercarrier interference

Institute of Electrical and Electronics

Engineers

Internet Engineering Task Force

Intersymbol interference

International Standards Organization

Intermodel Surface Transportation Efﬁciency

Act of 1991

Intelligent Transportation Systems

Intelligent Transportation Society of America

ITS Radio Services

ITU

MAC

MCM

MIB

MME

MPDU

Msps

OBUs

OFDM

OMG

PAPR

PER

PHY

PLCP

PMD

PPDR

PPDU

PSA

QAM

QPSK

RAN

RSSI

RSUs

SAP

SDL

SDO

SNMP

SNR

TCP/IP

UCSB

UML

UNII

VSA

VSCC

WAVE

WG

802.11

WII

WLAN

WZS

International Telecommunication Union

Medium access control

Multicarrier modulation

Management information base

MAC management entity

MAC protocol data unit

Megasymbol per second

Onboard units

Orthogonal frequency-division multiplexing

Object Management Group

Peak-to-average power ratio

Packet error rate

Physical layer (OSI)

Physical-layer convergence procedure

Physical-medium-dependent

Public protection disaster relief

PLCP protocol data unit

Power spectral analyzer

Quadrature amplitude modulation

Quadrature phase shift keying

Radio access network

Received signal strength indicator

Roadside units

Service access point

Speciﬁcation(s) and Description Language

Standards development organization

Simple Network Management Protocol

Signal-to-noise ratio

Transmissions Control Protocol/Internet

Protocol

University of California, Santa Barbara

Uniﬁed Modeling Language

Unlicensed national information infrastruc-

ture

Vector spectrum analyzer

Vehicle Safety Communications Consortium

Wireless Access in Vehicular Environments

Work Group 802.11—WLAN standards

Wireless infrastructure interoperability

Wireless local-area network

Work-zone safety

BIBLIOGRAPHY

1. R. Van Nee and R. Prasad, OFDM for Wireless Multimedia

Communications, Artech House, Boston, 2000.

2. R. Gerges, Communications Technologies for IVHS, UCLA,

1994.

3. A. Polydoros, R. Gerges et al, Integrated layer packet

radio study for AHS, Proc. 3rd IEEEE Mediterranean Symp.

New Directions in Control and Automation, Cypress, July

1995.

4. R. Gerges, Wireless communications and spectrum require-

ments for ITS, Paper presented at IEEE Vehicular Technology

Conf. Ottawa, May 1998.

5. IEEE 802.11 [full title: Information Technology—Telecommu-

nications and Information Exchange between Systems—Local

and Metropolitan Area Networks—Specific Requirements—

ITS RADIO SERVICE STANDARDS AND WIRELESS ACCESS IN VEHICULAR ENVIRONMENTS (ITS-WAVE) AT 5.9 GHz 2255

Part 11: Wireless LAN Medium Access Control (MAC) and

Physical Layer (PHY) Speciﬁcations], ANSI/IEEE Std 802.11,

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Information Technology—Telecommunications and informa-

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Medium Access Control (MAC) and Physical Layer (PHY)

Speciﬁcations, High-Speed Physical Layer in the 5 GHz

Band), IEEE Std 802.11a-1999.

7. ASTM E2213-02, Standard Speciﬁcation for Telecommunica-

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munications (DSRC) Medium Access Control (MAC) and

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8. T. S. Rappaport, Wireless Communications Principles and

Practice, Prentice-Hall, Englewood Cliffs, NJ, 2002.

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10. H. Steendam, and M. Moeneclaey, Analysis and optimization

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2256 ITS RADIO SERVICE STANDARDS AND WIRELESS ACCESS IN VEHICULAR ENVIRONMENTS (ITS-WAVE) AT 5.9 GHz

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