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Design of Analog Integrated Circuits

ECE611
Lecture 10

Active Filters
Ayman H. Ismail
ICL
Ain Shams University

Outline

Introduction
Filter Implementations
Filter specifications
Phase delay and group delay
Quality factor
Frequency Response visualized
Filter approximations
Butterworth
Chebychev
Elliptic
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Outline(cont.)
Bessel
All pass
Scaling
Filter realization
Biquads/Cascaded biquads/ Taw Thomson
Multiple loop feedback
LC ladder
SFG
Gyration
State space synthesis
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Introduction
Filters are two port circuits used to process the magnitude
and/or phase of the source signal in a certain predefined way.
Filters find various applications in modern communication
systems and most instrumentation systems.

From 1920 to 1960s filters were made of passive components

such as resistors, capacitors and inductors. In 1950s it was
recognized that size and cost reduction could be achieved by
replacing the large and costly passive inductors with active
circuitry. Since then active filters became used intensively in
many signal processing applications.
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Introduction
In mid 1970 s sub system integration started and fully
integrated active filters have been realized. Since then many
techniques have been used for filter implementation: Analog Filters
Continuous-time analog filters
Discrete-time analog filters
Digital Filters

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Digital Filters
The dynamic range of the filter depends on the number of
bits used to represent the signal.
Cons: If a digital filter is to process an analog signal, an A/D converter and
D/A converter are needed at its input and output respectively.
high power dissipation especially if high dynamic range is required
(lower power for modern technologies).
Not adequate for high frequencies due to the sampling operation
required before the A/D converter.

Pros: In general, the advantages of digital filter are that of digital circuits
namely its low sensitivity to process parameter variation and its noise
and interference immunity.
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Continuous-Time Filters
Two popular methods for implementing continuous time
filters, namely R-C active filters and Gm-C filters.
Cons: filter coefficients in continuous time filters is determined by the
product of dissimilar elements such as capacitors and resistors
(transconductors), therefore coefficients are sensitive to process
variations. The coefficients are of 30% accuracy. For this reasons
continuous time filters have on chip tuning circuit for adjustment of
the filter characteristics.

Pros: Higher operating speed compared to DT analog filters

Gm-C filters have higher operating speed (few 100MHz) than RC active
filters, whose speed is limited by Opamp GBW to few 10MHz.
RC filters can achieve higher dynamic range (90dB) than Gm-C filters
(40-70dB)
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Discrete-Time Filters
Switched capacitor filters can attain relatively high dynamic
range ( 90dB)
Cons: Can not be used for high speed signals due to:

Sampling freq. need to exceed Nyquist

OTA limited GBW
Charge injection and clock feedthrough
Requires anti-aliasing filter

Harder to switch the CMOS switches ON in deep submicron

technologies. This is solved by using switched op amp or enhanced
switch adding extra complexity to switched capacitor circuits.

Pros: coefficients high accuracy, which is determined by capacitor ratio that

has typical matching accuracy value of 0.1%.
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Filter Implementations

B. Murmann, EE315A(VLSI Signal conditioning

circuits) course slides, Stanford University
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Filter Implementations

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Lowpass Filter Magnitude

Response Characteristics

B. Murmann, EE315A(VLSI Signal conditioning

circuits) course slides, Stanford University
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Phase Delay
Assume that a signal composed of two sine waves

The signal is applied to a filter

Then

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Linear Phase Systems

To avoid distortion of the output signal

This is satisfied for linear phase systems

Phase distortion occurs when the phase is nonlinear , i.e. the
derivative of the phase is not constant. In general, the
condition for no phase distortion is
d ( ) ( )

0
d

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Phase Delay and Group Delay

Phase delay
p

( )

Group delay

d ( )
d
If ()=k, k a constant no phase distortion
For a linear phase filter g=p=k
Note that filters with ()=k +c are also called linear phase
filters, but introduce phase distortion. Note that such phase
response does not meet above condition for phase linearity.

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Quality Factor
The term Quality Factor (Q) has different definitions:
Component quality factor (inductor and capacitor)
Pole quality factor
Bandpass filter quality factor
For any component with the transfer function
Quality factor is defined as

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Quality Factor: Component

For Inductor

For a capacitor

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Quality Factor: Pole

B. Murmann, EE315A(VLSI Signal conditioning

circuits) course slides, Stanford University
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Quality Factor: Bandpass Filter

Quality Factor for Bandpass filter

Figure: H.K, EECS 247 (Analog-Digital Interface

Integrated Circuits) course slides, University of
California Berkeley

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Frequency Response Visualized

(CT)

Real poles
Complex poles
Complex zeros

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Magnitude Response

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Phase Response

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Frequency Response Visualized

(DT)

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Filter Approximation
The first step involved in the design of filters is to find a
magnitude characteristic, |H (j)| that satisfies the given
specifications. There are commonly used approximations
namely Butterworth, Chebyshev, Inverse Chebyshev, Elliptic
and Bessel approximations.
Therefore, the problem of finding an adequate |H (j)| that
satisfies the specifications reduces to the selection of one or a
combination of these approximations and determining its
order.

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Butterworth Approximation
Butterworth approximation was first proposed by Butterworth
in 1930
The normalized magnitude square-characteristics of
Butterworth approximation is

where n is an integer (filter order)

The normalized magnitude square characteristics 3-dB point
(half power point) occurs at = 1.
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Butterworth Approximation(cont.)
The |H(j)| can be expanded to

It can be seen that the first 2n-1 derivatives of |H(j)| are

zero at = 0. This characteristic is always referred to as the
maximally flat magnitude characteristics.
For >>1
Therefore the attenuation
The gain drops by 20n dB/decade in the stop band.
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Butterworth Filter Poles Location

The Butterworth pole are located on a circle with radius = p

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Butterworth Approximation Freq.

Response

Maximally flat response

Moderate phase distortion
Figure: H.K, EECS 247 (Analog-Digital Interface
Integrated Circuits) course slides, University of
California Berkeley
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Chebyshev-I Approximation
The normalized magnitude square-characteristics of
Chebyshev approximation is

Where Cn() is the Chebyshev polynomial (of the first kind) of

the nth order is defined as
Actually the above expression can be
proved to be a polynomial
This function will have values ranging
between 1 and 1/(1+)2 (ripples) in
the range 0 1.
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Chebyshev-I Approximation(cont.)
For >>1

Therefore the attenuation

Chebyshev I filter has ripples in the passband and poorer

group delay, but sharper transition band compared to
Butterworth
As more ripple is allowed in the passband, sharper transition
band and poorer phase response
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Chebyshev-I Approximation Poles

Location
Transfer function has poles, but no zeros
Poles located on an ellipse inside the unit circle
The more allowed ripples (corresponding to higher Q poles) in
the passband, the narrower transition band, the sharper cutoff, and the poorer the phase response

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Chebyshev I Approximation Freq.

Response

Ripple in passband

Figure: Haideh Khorramabadi EECS 247

(Analog-Digital Interface Integrated Circuits)
course slides, University of California Berkeley

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Chebyshev-II Approximation
Contrary to Chebyshev I, Chebyshev II filter approximation has
ripples in the stopband not in passband, sharper transition
band compared to Butterworth, and zeros in stopband.
Passband phase more linear compared to Chebyshev I

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Chebyshev-II Approximation Freq.

Response

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Elliptic Approximation
The elliptic filter approximation function has finite zeros
leading to a very steep transition band.
The elliptic approximation is characterized by equal-ripple
variation in both the pass band and stop band.
The magnitude square-characteristics of elliptic
approximation is
Where

or
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Elliptic Approximation Poles and

Zeros Locations
The elliptic filter approximation has imaginary zeros to sharpen
the transition band

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Elliptic Approximation Freq.

Response

Zeros substantially sharpen transition band at the expense of

reduced stop-band attenuation at high frequency

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Bessel Approximations
Thomson obtained the low pass
functions such that the group
delay is maximally flat at the
origin.
As expected the magnitude
characteristics of Bessel
Thomson filter is far inferior to
the previous three types (poor
stopband attenuation)
Poles of the Bessel approximation
are relatively low Q
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Comparison of Filter
Approximations

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Comparison of Filter
Approximations(cont.)

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All Pass Filters

In some cases the magnitude response is of the prime
concern and the phase response or the delay accompanying is
simply accepted.
If both magnitude and delay function are important, it is
usually preferable to realize the given magnitude function as
possible. If the accompanying delay is not satisfactory an
additional block, known as phase linearizer or delay equalizer
is introduced.
The group delay directly affects the waveform of the signal in
the time domain. For audio applications, the different delays
experienced by signals of different frequencies are not
noticeable to human ears.
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All Pass Filters(cont.)

On the other hand, in the transmission of video signals, if
different frequencies of the signal do not experience the same
delay the picture gets distorted. This means that the phase
response of video filters is of great importance.
Delay equalizers are all-pass network functions that do not
affect the magnitude response of the filter that they equalize
its delay. Usually delay equalizers are of second order sections
having transfer function

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All Pass Filters delay

The value of Q
determines the shape
of the delay function
against frequency.
When Q is below 0.577
approximately, the
delay is monotonically
decreasing.
When Q is high the
delay function will have
a peak. The higher Q
the sharper the peak.

Group delay for second order delay equalizers

with o = 1 and different values for Q
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All Pass Filter Used to Equalize

delay of A Butterworth Filter
It is required to equalize the delay of an 3rd order low pass
Butterworth filter.
A second order delay equalizer with Q = 0.5 and o = 1 is used.

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All Pass Filters Freq. Response

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Resultant Delay
Using the 3rd order Butterworth filter cascaded with the
second order delay equalizer leads to a nearly constant delay
throughout pass band.

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Active Filter Realization

Cascaded connection
of biquads and first
order secions

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Cascaded Connection of Biquads

and first order sections
In this realization method, second order (and first order if
filter order is odd) sections are cascaded to implement the
high order transfer functions. The general transfer function of
a second order section is

The required transfer function is decomposed to second order

functions then each of them is implemented.

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Cascaded Connection of Biquads

and first order sections
Biquads cascading is easy to design and tune, but the
resulting filter suffers from high sensitivity to component
variations. Therefore this technique not suitable for
implementation of high Q or high order filters. Typically,
integrated continuous time filters use biquads to realize filters
only up to ~5th order

B. Murmann, EE315A(VLSI Signal conditioning

circuits) course slides, Stanford University
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Cascaded Connection of Biquads

and First Order Sections(cont.)
There are three main degrees of freedom for the design of
cascaded biquads: Pole-zero pairing, i.e., which poles with which zeros of H(s)
will be paired to form each biquad
The position of each biquad in the cascade (biquad
ordering)
The distribution of the overall gain among the various
biquads
The above degrees of freedom greatly affect the dynamic
range of the overall filter, so they must be considered
seriously in the design phase.
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Pole-zero Pairing
Recall: complex pole near the j axis creates an elevation in
the magnitude response at frequencies around the imaginary
part of the pole. A zero at a similar location creates a deep
notch in the magnitude response at frequencies around the
imaginary part of the zero.
If such a pole and zero are very much apart in the s-plane, and
they are paired together to form a biquadratic function, the
minimum value in the magnitude response inside the pass
band will be much lower than the maximum value

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Pole-zero Pairing
Therefore, the dynamic range of the input signal is squeezed,
because the input signal can not be very large in order to
avoid the nonlinear operation, and it can not be very small at
the same time to avoid burying the output signal in noise at
frequencies near zero.
Therefore, to decompose a high order filter function it is
recommended to pair each pole with nearest zero starting
with the pole with highest Q.

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Biquad Ordering
The determination of the best sequence may be a very difficult
task especially in the case of having a larger number of
sections. For N sections there exist N! different combinations
for the biquad sections.
For correct arrangement of biquads, the frequency of
maximum magnitude of each biquad should be determined.
Then the biquads should be arranged such that neighboring
biquads have their maxima frequency as far apart as possible.
Low pass biquad sections and band pass sections should be
placed in front, while high pass biquad sections should be
placed last. This is carried to eliminate the low frequency noise
as possible.
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Biquad Ordering(cont.)
Arranging the biquads from low-Q to high-Q provides smooth
transfer functions from the input to the intermediate nodes,
and hence, helps to minimize harmonic distortion, but the
output will suffer from significant noise peaking near the
corner frequency due to the last stage with high-Q

Reversing the ordering will allow the later stages to filter out
the noise peaking near corner frequency, and may also filter
out harmonics (but not intermodulation).

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Gain Distribution
The overall gain of the filter should be distributed such that all
stages utilize the maximum available swing (or less than max.
available swing) as the input tone is swept across all
frequencies

This reduces non-linearity

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Passive LC Ladder
The passband response of ladder filters is much less sensitive
to component variations when compared to a biquad cascade
Poles tend to move together

The gain in the passband is 0.5 (-6dB)

The normalized values of Ls and Cs corresponding to a filter
approximation is obtained from filter tables of CAD tools
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Gyration
Gyrators are 2 port networks first introduced in 1948 by
Tellegan. The matrix describing the gyrator 2 port network is

The gyrator is represented symbolically as follows

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Gyration
Gyrators are used in active filter implementation due to their
impedance inversion property. This property can be used to
realize inductor-less filters starting from LC ladder.

ZL=1/SC then

This means that the circuit formed by loading the gyrator with
a capacitor is an inductor with inductance
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Gyrator Realization Using

Transconductors
It is straight forward to show that two cross transcoductors
implements the gyrator matrix
A grounded gyrator implementation using transconductors ,
where the gyrator is loaded with grounded impedance ZL

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Gyrator Realization Using

Transconductors(cont.)
A floating gyrator can be implemented with 2 grounded
gyrators

Also
Va Gm1 Vb Gm3

Therefore

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Active Filter Synthesis Using

Gyrators
Active circuits
are used to
simulate the
impedance of
coils and
resistors
(gyration).
Hence,
transforming
the LC ladder to
an active filter.
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Signal Flow Graph Synthesis

In this filter realization technique, it is the operation or
function, i.e., the equations that describe the topology of the
LC ladder, that is simulated. That is to say that instead of using
the active circuits to simulate the impedance of coils and
resistors (gyration), they are used to simulate the voltages and
currents in the LC ladder
Both voltages and currents in the LC ladder are treated as
voltage signals and they are considered state variables.
The number of independent state variables is equal to the
order of the filter.
The signal flow graph (SFG) realization technique is best
understood using an example
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Signal Flow Graph Synthesis

Example (Gm-C Filter)
It is required to implement an active filter (G m-C filter)
corresponding to the 5th order passive LC ladder shown
using SFG. The equations describing the operation of the
LC ladder are

Typically Rs=RL,
and hence DC
gain =-6dB

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Signal Flow Graph Synthesis

Example (Gm-C Filter)

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Signal Flow Graph Synthesis

Example (Gm-C Filter)
Note that the number of state variables is five [V1, V2, V3, I1,
I2] and equal to the order of the filter)
To simplify the design the transconductors used to implement
the active filter are chosen equal to 1/Rs = 1/RL = Gm.
The currents in all the equations above are divided by G m to
transform all state variables to voltages. This yields the
following set of equations

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Signal Flow Graph Synthesis

Example (Gm-C Filter)

where

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Signal Flow Graph Synthesis

Example (Gm-C Filter)
Previous equations can be represented by the following block
diagram. The active filter can be implemented by replacing
each block with an integrator (5 integrators).
The SFG simulation is called also leapfrog simulation due to
the similarity between the block diagram and the so-named
childrens game.

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Signal Flow Graph Synthesis

Example (Gm-C Filter)
The main problem with signal flow graph simulation is the
difficulty of obtaining simple equations that can be
implemented with first order integrators in some cases,
especially in case of signal flow graphs representing bandpass
filters.
Typically Gm is selected =1/Rs=1/RL

Gm=1/Rs=1/RL

CL1= L1 Gm2

CL2= L2 Gm2

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Signal Flow Graph Synthesis

Example (Op-Amp RC Filter)
Similarly an active Opamp-RC filter corresponding to the 5th
order passive LC ladder can be implemented using SFG
V V
1

1
R*
V1 s 1 I1
V1 (Vs V1 ) I1R*
*
Rs
Rs
Sc1

Sc1 R

V V
V1 V2
*
I1 R 1 2
I1
S L1
SL1

R*

( I R* I 2 R* )

V2 1
Sc2 R*

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Signal Flow Graph Synthesis

Example (Op-Amp RC Filter)

V V
V2 Vo
*
2
o

R I 2
I 2
S L2
SL2

R*

1 1
R* 1

Vo I 2 Vo
Vo I 2 R* Vo
RL Sc3
RL Sc3 R*

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Signal Flow Graph Synthesis

Example (Op-Amp RC Filter)

R*
1
*
V1 (Vs V1 ) V1
*
Rs

Sc1 R

V V
*
1
2

V1
S L1 R *

R *2

(V V2
V2 1
*
Sc2 R
*

V V
2
o

V2
S L2 R*

R *2

*
R* 1

Vo V2 Vo
RL Sc3 R*

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Signal Flow Graph Synthesis

Example (Op-Amp RC Filter)

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Signal Flow Graph Synthesis

Example (Op-Amp RC Filter)

In this slide R* is
mentioned as
simply R.

RL=Rs=R

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Scaling
Previous expressions had the frequency normalized. (cutoff
frequency of the filter is assumed 1 rad/sec)
This is done to standardize the basic theory and design of the
filter, and then the filter is adapted to different applications in
a process called scaling.
Frequency scaling refers to dividing each inductance and
capacitance in the obtained passive network with a scaling
factor Kf, such that what occurs at in the original network
occurs at Kf.
A similar process, is impedance scaling, carried to set the
values of the resistors, capacitors and inductors in the
obtained passive network to realizable values.
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Scaling
Impedance scaling is carried out by multiplying all resistors
and inductors with a scaling factor Kz and dividing all
capacitors with the same factor. This does not affect the
transfer function of the passive filter.
Node scaling (voltage scaling) is used to optimize dynamic
range. Scaling filter nodes means scaling the voltage levels at
the filter nodes such that the maximum value of voltage levels
at all nodes is the same. Usually this requires using unequal
transcoductors, which can not be easily matched. Therefore,
in practice, only filter nodes, which contribute to nonlinearity, are scaled. (more about node scaling later)
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LC ladder Denormalization

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