In this work, we present an adaptive non-dataaided compensator for the in-phase/quadrature-phase (I/Q)
mismatch in low-intermediate frequency (heterodyne) receivers.
In particular, the adaptive I/Q mismatch algorithm is derived by exploiting the uncorrelatedness between the desired and the image signals, and is used to compensate for gain imbalance and phase-offset error. Furthermore, the frequencyoffset error is separately compensated by using an adaptive minimization of a fourth-order statistics based cost function.

© All Rights Reserved

22 tayangan

In this work, we present an adaptive non-dataaided compensator for the in-phase/quadrature-phase (I/Q)
mismatch in low-intermediate frequency (heterodyne) receivers.
In particular, the adaptive I/Q mismatch algorithm is derived by exploiting the uncorrelatedness between the desired and the image signals, and is used to compensate for gain imbalance and phase-offset error. Furthermore, the frequencyoffset error is separately compensated by using an adaptive minimization of a fourth-order statistics based cost function.

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Shafayat Abrar

Department of Electrical Engineering

COMSATS Institute of Information Technology

Islamabad 44000, Pakistan. sabrar@comsats.edu.pk

Azzedine Zerguine

Department of Electrical Engineering

King Fahd University of Petroleum and Minerals

Dhahran 31261, Saudi Arabia. azzedine@kfupm.edu.sa

Asoke K. Nandi

Department of Electronic and Computer Engineering

Brunel University, Uxbridge UB8 3PH, UK

asoke.nandi@brunel.ac.uk

AbstractIn this work, we present an adaptive non-dataaided compensator for the in-phase/quadrature-phase (I/Q)

mismatch in low-intermediate frequency (heterodyne) receivers. In particular, the adaptive I/Q mismatch algorithm is

derived by exploiting the uncorrelatedness between the desired

and the image signals, and is used to compensate for gain

imbalance and phase-offset error. Furthermore, the frequencyoffset error is separately compensated by using an adaptive

minimization of a fourth-order statistics based cost function.

signal model for an imbalanced analog front-end is presented. In Section III, a cost function sensitive to imbalance

is proposed and optimized to yield into an adaptive algorithm

for imbalance compensation. In Section IV, an iterative

method is derived for the estimation of frequency-offset.

Simulation results are presented in Section V and conclusion

are drawn in Section VI.

decorrelation, blind calibration, frequency-offset recovery,

fourth-order statistics

The system model is shown in Fig. 1. In the absence of

additive noise, the received RF signal is given by

I. I NTRODUCTION

A low-IF (heterodyne) receiver architecture is very attractive because of its inherent immunity to DC offsets

and 1/f noise. Traditionally, IF less than or equal to half

of the channel band-width is employed. Larger values of

IF impose higher requirement on the image rejection ratio

(IRR) specification of the receiver. A GSM receiver, for

example, requires IRR of larger than 32 dB with IF of 100

kHz and 50 dB with IF of 200 kHz. It is highly desirable to

increase IF beyond 100 kHz, because it then allows simple

high-pass filtering to remove DC [1].

Due to components mismatches in analogue electronics

and resulting in-phase/quadrature-phase (I/Q) imbalances,

the IRR performance of a heterodyne receiver may degrade

significantly [2]. The dynamic nature of the mismatches

entails dynamic or on-line compensation. Without any compensation, IRR of as low as 25 dB is possible in existing

designs. To compensate these imbalances and remove image signal from the desired band, statistical independence

based adaptive methods were introduced (refer to [3] and

references therein). Here, in this work, we discuss a simple

algorithm for non-data-aided I/Q compensation which is

derived by exploiting the correlation properties of desired

and image signals.

2fLO fC is the central frequency of the image band, and

s(t) and q(t) are the baseband desired and image signals,

respectively. The quadrature mixture is assumed to suffer

Analogue

Processing

cos(2 fLO t)

Digital

Processing

(1)

xn

LPF

LPF

A/D

rn

rRF(t)

LPF

LPF

A/D

zn

i

exp(+i2 fIF nT)

xn

zn

Adaptive

gain/phase

mismatch

removal

Figure 1.

~

sn

~

qn

Iterative

frequency

mismatch

removal

sn

fIF is desired IF freq.

T is symbol period.

qn

fLO fIF and phase mismatch . The impairments and

signal rn , sampled at the rate of 1/T , is expressed as:

rn = (sn + qn ) e+i2n(fIF T +)

+ (qn + sn ) ei2n(fIF T +) ,

wz,n = wz,n1 + qen sen

(2)

where = 0.5 1 (1 + )e , = 0.5 1 + (1 + )ei ,

= T f , sn = s(t)|t=nT and qn = q(t)|t=nT . After

down-conversion and low-pass filtering, we obtain the baseband signals {xn } and {zn }:

i

xn = (sn + qn ) e+i2n ,

(3a)

sn ) ei2n .

(3b)

zn = (qn +

as follows:

{xn } and {zn } to estimate the impairments, , and .

Assuming that impairments are perfectly known, then the

desired and image signals are expressed as

1+

(xn zn ) ei2n , (4a)

sbn = sen ei2n = 1||

2

1+

(zn xn ) e+i2n , (4b)

qbn = qen e+i2n = 1||

2

we can show:

1

1

1.

= angle

, and =

(5)

1+

1+

III. E XISTING S OLUTIONS FOR M ISMATCH

C OMPENSATION

phase imbalances and the bias errors of the in-phase and

quadrature channels of a coherent signal processor by means

of coefficients which are derived from measurements of a

test signal. The residual errors after correction depend upon

the signal-to-noise ratio (S/N) of the test signal and the degree of filtering used in deriving the correction coefficients.

According to [6] this correction can be applied only one

frequency at a time. When the I and Q channels cover a

wide bandwidth, the imbalance is a function of frequency.

This method might be impractical to apply because it needs

an off-line test input and can not track time variation.

A solution mentioned in [7] is to move the generation of

I and Q signals to the digital domain by using the Hilbert

transform. In this approach, the data for one channel (say the

I channel) are obtained from a single-path A/D converter,

while the data of the Q channel are generated by processing

the channel data through a Hilbert filter. According to [6],

however, the operating speed is limited by the complexity

of Hilbert filters. In order to get balanced I and Q signals, a

high-order filter is needed to perform the Hilbert transform.

The resulting solution has high power consumption, low

processing speed, and high hardware complexity.

In [6], authors proposed a two-tap (wx,n and wz,n )

butterfly-structure adaptive filter which worked on the principle of noise cancellation (de-correlation) and it was realized

(6a)

(6b)

and qen = zn xn wx,n1 . The above algorithm was simple

but heuristic in nature. Though it was explained to be a

de-correlation process but it was not explained what cost

function was exactly being minimized or maximized. It also

which could have been

utilized to implement a simpler update.

In the year 2007, authors in [8] proposed a gradient-based

adaptive algorithm to compensate for I/Q imbalance in lowIF receivers. Here the compensation algorithm developments

build on the so-called circular nature of complex-valued

communications waveforms which is known to hold only

under perfect I/Q balance. A well-behaving non-circularity

measure is first formed which is then minimized iteratively

using gradient-descent type approach. The derived compensator is computationally simple and operates blindly on the

received signal, meaning that no known training or pilot

data is needed. In particular, they suggested to minimize the

following non-circularity measure:

2

= arg min Eu2n , where un := xn xn .

(7)

mentioned to be biased. Ignoring biased terms, however, an

approximate unbiased algorithm was obtained as follows:

(8)

n

b

bn1 + rn cn 2rn2 b

n1

bn1

c2n

(1 r ) |xn |2 . Detailed discussion on (7) appeared in [9].

Note that a simplified form of the update (8) appeared earlier

in [1] (and later in [15]) but without the notion of circularity;

they proposed two real-valued updates: (below and

denote the real and imaginary parts, respectively)

[b

n ] = [b

n1 ] + [un ]2 [un ]2 ,

(9)

[b

n ] = [b

n1 ] + 2[un ][un ],

where un is as specified in (7). Note that (9) is equivalent

to the following single-step complex-valued update:

bn = b

n1 + u2n .

(10)

E[un ]2 = E[un ]2 and E[un ][un ] = 0 in steady-state

(which is nothing but circularity) and thus successfully

removing the gain/phase related imbalances. The update

(9) was proposed on heuristic ground and algorithmic development arising from some cost was not discussed. An

improved version of (10) is obtained in [10] by exploiting

the normalized least mean square filtering, as given by

bn =

bn1 +

u2n

.

|un |2

(11)

where the normalization factor serves to provide a dependent factor. There are number of other adaptive compensators which are suitable for higher baud-rate block data

transfer; interested readers may refer to [11]-[22].

IV. E STIMATION OF G AIN I MBALANCE

Exploiting the fact that the desired and image signals {sn }

and {qn } are mutually uncorrelated, optimum closed-form

estimates were obtained in [4] as follows:

p

B B 2 4|A|2

(1)

,

(12a)

opt =

p2A

B + B 2 4|A|2

(2)

,

(12b)

opt =

2A

where A := Exn zn , and B := E |xn |2 + |zn |2 . For

(1)

vanishing imbalance, i.e., A 0, we have opt 0 and

(2)

opt . Note that authors in [4] preferred to use the root

(1)

with smaller magnitude, i.e., opt .

In this work, we propose to obtain the value of

adaptively by minimizing a cost which is measure of the

correlation between the estimated signals, sbn and qbn , mathematically it is expressed as

2

= arg min E (xn zn ) (zn xn ) ,

(13)

, which facilitates separate estimation of . To obtain a

gradient-based adaptive algorithm for , we use

(14)

bn = b

n1 |C|2 ,

n1 xn zn ).

for = n1 and C := E(xn

bn1 zn )(b

Note that the auxiliary variable C can be expressed as C =

2

ABb

n1 + A

bn1

, where A and B are as specified in

(12); next, we find

|C|2

|C|2 C

=

= C (2A b

n1 B) , (15)

b

n1

C b

n1

Replacing the statistics A, B and C with their respective

estimates, we get the following gradient-based algorithm:

An = g An1 + (1 g ) xn zn ,

Bn = g Bn1 + (1 g ) |xn |2 + |zn |2 ,

2

Cn = An Bn

bn1 + An

bn1

,

bn =

bn1 + g Cn Bn

2An

bn1

(16)

a forgetting-factor. Substituting An and Bn in (12), we

(1)

(2)

(1)

(2)

obtain n1 and n1 as the estimates of opt and opt ,

(1)

(2)

respectively. Using n1 and n1 , we can express (16) as

follows:

(2)

(1)

n1 n1

bn =

bn1 b

n1 n1 b

(1)

(2)

(17)

n1 + n1

,

bn1

2

(1)

(2)

of saddle point (see Fig. 2(a)). This implies that, depending

(1)

on initialization, the update may converge either to opt and

(2)

opt . Under no imbalance condition, however, as one of the

roots is required to be zero, the update has a natural tendency

to converge to the root with smaller magnitude provided that

the step-size is large enough to help escape the other root

(see Fig. 2(b)-(c)). Also note that, unlike [8], the proposed

algorithm (16) is unbiased.

V. E STIMATION

OF

F REQUENCY O FFSET

The presence of frequency-offset error contaminates the

b n1 is

estimated signal sbn by the factor e+i2n . Suppose

the available estimate of , then sbn is expressed as

b

1+

bn1

b

(xn b

n1 zn ) ei2nn1 ,

=

1 |b

n1 |2

(18)

estimates , as given by,

m

PN

bnk sbn1k

angle

k=0 s

b=

,

(19)

2m

where N denotes number of symbols. Note that this estimator assumes that the signal has constant modulus; in

the presence of gain imbalance, however, we would need

gain normalization to ensure this property. Denoting n :=

b n1 , and assuming no additive noise, note that

m

sbn1

sbn

|b

sn1 | |b

sn |

!m

b

b

sen1

ei2(n1)n2 sen ei2nn1

=

(20)

|b

sn1 | |b

sn |

m

sn1 ei2(n1)n1 sn ei2nn

= ei2m(n1 +n(n n1 )) ,

m

sbn1

sbn

1

,

angle

n

2m

|b

sn1 | |b

sn |

(21)

b n is obtained

With the aid of (21), an iterative estimate of

as

sbn1 sbn m

b

b

,

n = d n1 + (1 d )

|b

sn1 sbn |

n o

bn

angle

b n = o

b n1 + (1 o )

(22)

,

2m

where

and

The estimator (22) is not useful for frequency-offset estimation in quadrature amplitude modulation due to its multimodulus constellation. Assuming that the gain imbalance has

been compensated and denoting = 2n, we have

sbn = sen ei = sn ei()

b

(23)

Denoting e := ,

statistics of sbn contains the information of unknown e [23]:

E sb4n,I + sb4n,Q = constant+

(24)

1

E s4n,I + s4n,Q 6s2n,I s2n,Q cos(4e )

4

Note that cos(4e ) is maximum (that is equal to +1) when

e = 0 and it is minimum (that is equal to 1) when

e = /4. So the unknown phase is compensated if

it is between /4 and +/4. For phase ambiguity due

to the multiples of 90 degree may be compensated using

differential encoding. Further

note that, for QAM signals,

E s4n,I + s4n,Q 6s2n,I s2n,Q is a negative quantity which

helps us formulate minimization of the following cost for

the recovery of unknown phase:

4

4

min E sbn,I

+ sbn,Q

(25)

b

b + sen,Q sin

b and

Notice that sbn,I = [b

sn ] = sen,I cos

b

b

sbn,Q = [b

sn ] = e

sn,I sin + sen,Q cos , these relations

help us obtain the following:

4

3

Eb

sn,I

= +4Eb

sn,I

sbn,Q ,

b

3

Eb

s 4 = 4Eb

sn,Q

sbn,I ,

b n,Q

(26a)

(26b)

following gradient-based algorithm:

3

sbn,I ,

Gn = t Gn1 + (1 t ) sbn,Q

3

Hn = t Hn1 + (1 t ) sbn,I

sbn,Q ,

b

b

n = n1 + t n , (n := Gn Hn ),

b

obtain a coarse (but gradient-based adaptive) estimate of .

Since the relation = 2n can equivalently be expressed

as n = n1 + 2, where n is the true value of at

time n. With these considerations, we suggest to solve

bn =

b n1 + 2n1 t J ,

bn

4

(29)

s J

,

n = n1

bn

4

b n is

bn =

b n /(2n);

Once

b

however, in practice, the n is not explicitly required to be

b n is equivalently sufficient

computed as the knowledge of

b n = n , where the

for the purpose. Note that J /

statistical error quantity n is as specified in (32). The

derivative of cost J w.r.t. n requires attention; note that

b J

J

=

,

(30)

b

bn

b n1 +

The constraint in (28) allows us to express

2n1 , which gives

b n1

b0

bn

+ 2

+ 2n 2n, (31)

n1

n1

n1

b

Note that the gradient /

is growing linearly in time

which is analytically correct but its use in the update expression may cause divergence. One possible way to handle this

situation is to use a diminishing step-size to overcome the

b

linear growth of /.

However, a diminishing step size

usually leads to slow convergence and requires exhaustive

experimentation to determine how rapidly the step-size must

decrease in order to prevent scenarios in which it (the stepsize) becomes too small when the iterates are far from the

required estimate. The other solution is to simply drop this

gradient factor as it is always positive and has no role in

determining the direction of the update. We prefer to adopt

the latter proposal while using a fixed but very small stepsize s for n to ensure the stability and low jitter.

bn =

b n1 + 2n1 + t n ,

n = n1 + s n ,

(27)

forgetting-factor less that one. Note that the algorithm (27)

does not (explicity) exploit the fact that = 2n . Exploiting this information, we modify the problem (25) as

follows:

4

4

b = 2n

b

, s.t.

J := min E sbn,I

+ sbn,Q

(28)

b

b

,

b however, the resulting

imizations with respect to

two updates must also satisfy the constraint in (28). To

realize such an optimization, we introduce an auxiliary (or

intermediate) variable n , minimize the cost w.r.t. it, and

1

b

b

(z)

= (z)z

+ 2(z)z 1 + t (z),

(z) = (z)z 1 + s (z),

(32)

(33)

2 s (z)z 1

1

b

b

(z)

= (z)z

+ t (z) +

1 z 1

(34)

form of (33) as follows:

o

n = n1 + n1 ,

o

bn =

b n1 + t n + 2 s n ,

(35)

is close to the square of t , i.e., s ( t )2 .

frequency-independent imbalances.

o and + indicate saddle point and minima, resp.

(2)

(1)

opt

opt

[]

+

o

+

(1)

(2)

0.5(opt + opt )

2

[]

= 5x10

[]

0

2

4

3

4

= 1x10

[]

2

0

[]

the proposed estimators. The baseband signals in the desired

and image bands are expressed as sn = an + wn and qn =

bn + vn , respectively, where {an } and {bn } are transmitted

quadrature phase-shift keying (QPSK) symbols, and {wn }

and {vn } denote additive white Gaussian noise. The signalto-noise ratios (SNRs) of the received signals {sn } and {qn }

are taken as 30 dB. The forgetting factors were selected as

g = d = o = 0.998 and the step-size = 6 104 .

At time zero, adaptive/iterative parameters were initialized

b 0 = 1, b

b 0 = 0. The

as A0 = 1, B0 = 2,

0 = 0, and

4

frequency offset = 1 10 , the amplitude mismatch

= 0.8, and the phase mismatch = 10 (this gives =

(1)

0.3863i0.1563, = 1.3863i0.1563 resulting in opt =

(2)

0.2877 i0.0803 and opt = 3.2245 i0.8999).

In this experiment, we study convergence behaviour of

update (16) for small and relatively large step-sizes (for

QPSK signal). Refer to Fig. 2(a) for the contour plot of the

(1)

(2)

cost where the global minima, opt and opt , and the saddle

(1)

(2)

point 0.5(opt + opt ) are labeled. Next in Fig. 2(a) and (b),

we provide traces of convergence for small and relatively

large step sizes, respectively. It can be noticed that for small

bn is initialized near

step-size (i.e., g = 5 105 ), when

(2)

(2)

opt , it converged to opt ; however, for relatively large stepsize (i.e., g = 1 104 ), regardless of the initialization,

(1)

smaller magnitude, i.e., opt . Further, with g = 1 104,

refer to Fig. 3(a)-(d) and Fig. 3(e)-(f) for scatter plots and

convergence traces, respectively; both estimators can be

noticed to be converging steadily to true values. Refer to

Fig. 3(g) for the traces of empirically obtained mean square

error E|b

sn sn |2 and squared absolute correlation |Eb

sn qbn |2 .

Both indices are decreasing along iteration and attaining a

stable floor in steady-state; this means that, as a result of

successful convergence, estimated signal sbn is getting close

to desired signal sn and image qbn is rejected from sbn . Note

that 1000 symbol points are used in each scatter plot (for

a single realization) and traces (in Fig. 3) were averaged

over 500 independent realizations. In the second experiment,

we compare the performance of adaptive frequency-offset

recovery algorithms (27) and (35) by simulating the traces of

mean square errors in estimation error. We observe clearly in

Fig. 4 that the two-step algorithm (35) is far more superior

to one-step algorithm (27). All simulation parameters are

clearly depicted in Fig. 4.

In this work, an adaptive non-data aided in-phase /

quadrature-phase imbalance compensator for heterodyne receiver was developed. Simulation results showed that the

2

4

[]

convergence trajectories of b

n for small and relatively large step-sizes.

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(a) sn

(b) xn

10

Two-Step

Solution

t = 1.510 4

s = 2t

2-step

2

b n/ 1

NMSE: E

10

1

1

(c) sen

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(d) sbn

20

One-Step

Solution

t=

210 4

.

.

1-step

16QAM : = 0.8, = 10 ,

= 1 104 , SNR = 30 dB,

t = 0.98, g = 0.998, g =1104

30

0

40

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1

1

1

(e) |b

n |

0.3

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x 10

1

50

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bn

(f )

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Iterations

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0.1

0

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Simulated

True value

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Simulated

True value

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|Eb

sn qbn |2

E|b

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[dB]

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Iterations

Figure 3.

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