Blind Adaptive Compensation of I/Q Mismatch

and Frequency Offset in Low-IF Receivers

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.

The paper is organized as follows. In Section II, a general

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.

Keywords-I/Q imbalance, adaptive filter, low-IF, adaptive

decorrelation, blind calibration, frequency-offset recovery,
fourth-order statistics

II. S YSTEM M ODEL

The system model is shown in Fig. 1. In the absence of
additive noise, the received RF signal is given by

I. I NTRODUCTION

rRF (t) = 2{s(t)ei2fC t } + 2{q(t)ei2fI t },

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.

where fC is the carrier frequency of desired band, fI =

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)

exp( i2 fIF nT)

xn
LPF

LPF

A/D

rn

rRF(t)
LPF

LPF

A/D

zn

i
exp(+i2 fIF nT)

(1+ ) sin(2 fLO t+ )

xn
zn

Adaptive
gain/phase
mismatch
removal

Figure 1.

~
sn
~
qn

Iterative
frequency
mismatch
removal

sn

fLO is local oscillator freq.

fIF is desired IF freq.
T is symbol period.

qn

A heterodyne receiver and mismatch compensators.

with amplitude mismatch , frequency mismatch f = fC

fLO fIF and phase mismatch . The impairments and

 are assumed to be frequency independent. The digital IF

signal rn , sampled at the rate of 1/T , is expressed as:
rn = (sn + qn ) e+i2n(fIF T +)
+ (qn + sn ) ei2n(fIF T +) ,

wx,n = wx,n1 + sen qen

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:

The digital processor at receiver uses the baseband signals

{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

where = / . The impairments and are related to ,

we can show:




1 
1

 1.
= angle
, and = 
(5)
1+
1+
III. E XISTING S OLUTIONS FOR M ISMATCH
C OMPENSATION

A method is presented in [5] for correcting the gain and

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)

where sen and qen were obtained as sen = xn zn wz,n1

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

overlooked the fact that wx,n = wz,n

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)

The true gradient of this self-cancellation cost (7) was

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

where cn := c cn1 + (1 c ) x2n and rn := r rn1 +

(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)

The rational behind (9) is to design an update which ensures

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)

Note that this cost is insensitive to frequency offset error,

, 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)

where g is a positive step-size and 0 < g < 1 is

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)

where = 2|A|2 and 0.5(n1 + n1 ) is the estimate

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

A. For PSK Signals:

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

sbn = sen ei2nn1

1+
bn1
b
(xn b
n1 zn ) ei2nn1 ,
=
1 |b
n1 |2

(18)

If sbn is an m-PSK, then the maximum likelihood approach

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 )) ,

Further assuming n n1 , we obtain

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

are positive forgetting factors.

B. For QAM Signals:

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)

b we can show that the fourth-order

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)

These statistics may be computed iteratively and lead to the

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 is known, a fine estimate of

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)

where t is a positive step-size and t is a positive

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
,

The optimization of (28) may be realized as separate minb and ;

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

Taking the z-transform of (32), we get

1
b
b
(z)
= (z)z
+ 2(z)z 1 + t (z),
(z) = (z)z 1 + s (z),

(32)

(33)

Combining the two expressions in (33), we obtain

2 s (z)z 1
1
b
b
(z)
= (z)z
+ t (z) +
1 z 1

(34)

Denoting (z) := (z)z 1 /(1z 1), we obtain an alternate

form of (33) as follows:
o

n = n1 + n1 ,
o

bn =
b n1 + t n + 2 s n ,

(35)

Experimentally, we have found that a suitable value of

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

proposed adaptive scheme can successfully compensate for

frequency-independent imbalances.
o and + indicate saddle point and minima, resp.

VI. S IMULATION R ESULTS AND C ONCLUSIONS

(2)

(1)

opt

opt

[]

+
o
+

(1)

(2)

0.5(opt + opt )
2

[]

= 5x10

and Iterations = 30000

[]

0
2
4

3
4

= 1x10

[]

and Iterations = 8000

2
0

[]

We carry out simulations to evaluate the performance of

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,

bn is found to be always converging to the root with

(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

[]

Figure 2. (a) Contours of cost for = 0.8 and = 10 , and (b)-(c)

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

2
2

(d) sbn

20

One-Step
Solution
t=
210 4
.
.
1-step

Adaptive Frequency Offset Recovery

16QAM : = 0.8, = 10 ,
= 1 104 , SNR = 30 dB,
t = 0.98, g = 0.998, g =1104

30

0
40

1
1

1
1

(e) |b
n |
0.3

0
4

x 10
1

50
0

bn
(f )

2000

4000

6000

8000

10000

Iterations

Figure 4. Frequency-offset recovery: normalized MSE traces for 16QAM.

0.2
0.5
0.1
0
0

Simulated
True value
0
2000 4000 6000 0

Simulated
True value
2000 4000 6000

(g) MSE and SC traces

10

|Eb
sn qbn |2
E|b
sn sn |2

[dB]

0
10
20
30
0

1000

2000

3000

4000

5000

6000

Iterations

Figure 3.

Scatter plots and convergence traces for QPSK.

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