Advanced Digital Signal Processing

Part 5: Digital Filters

Gerhard Schmidt

Christian-Albrechts-Universität zu Kiel
Faculty of Engineering
Institute of Electrical and Information Engineering
Digital Signal Processing and System Theory
Digital Filters

•Contents
❑ Introduction
❑ Digital processing of continuous-time signals
❑ Efficient FIR structures
❑ DFT and FFT
❑ Digital filters
❑ Structures for FIR systems
❑ Structures for IIR systems
❑ Coefficient quantization and round-off effects
❑ Design of FIR filters
❑ Design of IIR filters

❑ Multi-rate digital signal processing

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Short Excurse on Digital Signal Processors

•Fixed-Point DSP Hardware

X data bus (read and write)
Example for a 16-bit fixed- Y data bus (only read)
point DSP architecture: X register Y register
❑ Architecture with 2 busses
Several
Multiplier
❑ Only main components registers
are depicted
Shifting by -1, 0, 1 bit
❑ Architecture varies
among different vendors
Unit for arithmetic and
logic computations

Accumulator 0 Accumulator 1

Limitation

Saturation

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Digital Filters

•General Remarks – Part 1

Digital filters:
❑ Linear-time-invariant (LTI) causal system with a rational transfer function (without loss of generality: numerator degree =
denominator degree = )

with without loss of generality.

: parameters of the LTI system (coefficients of the digital filter)
: filter order
❑ Product notation:

where the are the zeros, and the are the poles of the transfer function (latter are responsible for stability).

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Digital Filters

•General Remarks – Part 2

❑ Difference equation:

with denoting the input signal and the resulting signal after filtering.

Remarks:
❑ Generally the above equation describes a recursive filter with an infinite impulse
response (IIR filter): is calculated from and recursively from

❑ The calculation of requires some memory elements in order to store and

Dynamic system.

❑ If

Filter has no zeros all-pole or autoregressive (AR) filter.

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Digital Filters

•General Remarks – Part 3

The difference equation is purely recursive:

❑ If (causal filter required!):

The difference equation is purely non-recursive:

Non-recursive filter
Transfer function:

❑ Poles but not relevant for stability all-zero filter.

❑ According to the difference equation: is obtained by a weighted average of the last input values
Moving average (MA) filter (as opposite to the AR filter).
❑ From the transfer function it can be seen that the impulse response has finite length.
Finite impulse response (FIR) filter of length and order .

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Digital Filters

•Structures for FIR systems – Part 1

The impulse response is equal to the coefficients :

With the difference equation of the FIR systems and the relation above we get

which is the linear convolution sum (with ). A possible realization is given in the

Direct form structure: Tapped-delay or transversal filter – Part 1

First direct form

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Digital Filters

•Structures for FIR systems – Part 2

Direct form structure: Tapped-delay or transversal filter – Part 2
By transposing the flow graph, that means
❑ reversing the direction of all branches,
❑ exchanging the input and output of the flow graph and
❑ exchanging summation points with branching points and vice versa,
we get the second direct form (below redrawn version):

If the unit impulse is chosen as the input signal, all samples of the impulse response appear
successively at the output of the structure.

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Digital Filters

•Structures for FIR systems – Part 3

Direct form structure: Tapped-delay or transversal filter – Part 3
The number of multiplications can be reduced if the impulse response of the system is symmetric, e.g. if we have:

Then the number of multiplication can be reduced from to in the even case and to in the odd case.
Below the flow graph for the odd case is depicted:

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Digital Filters

•Structures for FIR systems – Part 4

Cascade-form structures
We obtain the cascade realization by factorizing the transfer function into a cascade of shorter length filters:

Remarks:
❑ The reason for the cascade structure is that the shorter-length filters can be implemented with improved
robustness in one of the direct forms than the overall filter.
❑ The are usually second order filters with real coefficients. Therefore the poles and zeros have to be real or appear
in conjugate complex pairs.
❑ For linear phase filters the zeros have to appear in quadruples.

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Digital Filters

•Structures for IIR systems – Part 1

Direct form structures – Part 1
A rational system function can be divided into two parts – an all-zero part and in an all-pole part .

The all-zero filter can be realized with the direct form. By attaching the all-pole system in cascade we obtain the
direct form I realization.

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Digital Filters

•Structures for IIR systems – Part 2

Direct form structures – Part 2

Signal flow graph

of the direct
form I realization:

All-zero All-pole
system system

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Digital Filters

•Structures for IIR systems – Part 3

Direct form structures – Part 3
Another realization can be obtained by exchanging the position of the all-pole and the all-zero filter.

Difference equation for the all-pole part:

where the sequence is an intermediate result and is the input of the all-zero part.

Difference equation of the all-zero part:

The resulting structure is called direct form II realization. Furthermore, it is said to be canonic since it minimizes the
number of memory elements. Only one single delay line is required for storing the delayed versions of the sequence .

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Digital Filters

•Structures for IIR systems – Part 4

Direct form structures – Part 4
Signal flow graph
of the direct form II
realization:

Exchanging the order

of the subfilters

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Digital Filters

•Structures for IIR systems – Part 5

Direct form structures – Part 5
Transposing the direct
form II realization
leads to the following
structure:

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Digital Filters

•Structures for IIR systems – Part 6

Direct form structures – Part 6
Partner work – Please think about the following question and try to find answers
(first group discussions, afterwards broad discussion in the whole group).
❑ What are the advantages and disadvantages of the two structures depicted below?

…………………………………………………………………………………………………………………………………………………………………………………….
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Digital Filters

•Structures for IIR systems – Part 7

Cascade-form structures – Part 1
As for FIR systems we can cascade Subsystems of first or second order to the desired system :

First order subsystem:

Canonical direct form for a first order filter (“bi-linear” filter):

Every first order transfer function can be realized with the above flow graph.
Corresponding transfer function:

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•Structures for IIR systems – Part 8

Cascade-form structures – Part 2
Second order subsystem:
Canonical direct form for a second order filter (“bi-quad” filter):

Corresponding transfer function:

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Digital Filters

•Structures for IIR systems – Part 9

Cascade-form structures – Part 3
Example:
Given is a so-called Chebyshev lowpass filter of 5th order and the cut-off frequency ( is the sampling frequency).
A filter design approach yields the transfer function below. The corresponding filter design algorithms will be discussed later on:

❑ The zeros are all at for .

The poles are
❑ By grouping the poles and we get three subsystems – two second order subsystems and one first order
subsystem with the pole :

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Digital Filters

•Structures for IIR systems – Part 10

Cascade-form structures – Part 4
Example (continued):
❑ For the implementation on a fixed-point DSP it is advantageous to ensure that all stages have similar amplification in
order to avoid numerical problems. Therefore, all subsystems are scaled such that they have approximately the same
amplification for low frequencies:

Remark:
The position of the subsystems in the cascade is in principle arbitrary. However, here the poles of are closest to
the unit circle. Thus, using a fixed-point DSP may lead more likely to numerical overflow compared to and .
Therefore, it is advisable to realize the most sensible filter as the last subsystem.

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Digital Filters

•Structures for IIR systems – Part 11

Cascade-form structures – Part 5
Example (continued):
❑ Frequency responses:

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Digital Filters

•Structures for IIR systems – Part 12

Cascade-form structures – Part 6
Example (continued):
❑ Resulting signal flow graph:

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Digital Filters

•Structures for IIR systems – Part 13

Parallel-form structures – Part 1
An alternative to the factorization of a general transfer function is to use a partial-fraction expansion, which leads to a
parallel-form structure.
❑ We assume distinct poles (which is quite well satisfied in practice). Then, the partial fraction expansion of a
transfer function with numerator degree is given as

where are the coefficients (residues) in the partial fraction expansion and

❑ We further assume that we have only real-valued coefficients, such that we can combine pairs of complex-
conjugate poles to form a second order subsystem:

with

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Digital Filters

•Structures for IIR systems – Part 14

Parallel-form structures – Part 2
❑ Two real-valued poles can also be combined to a second order transfer function:

with

❑ If is odd, there is one real-valued pole left, which leads to one first order partial fraction see example).

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Digital Filters

•Structures for IIR systems – Part 15

Parallel-form structures – Part 3

Signal flow graph of the parallel structure: Signal flow graph of a second order section:

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Digital Filters

•Structures for IIR systems – Part 16

Parallel-form structures – Part 4
Example:
Consider again the 5th order Chebyshev filter with the transfer function

The partial fraction expansion can be given as:

with the poles and residues

The resulting transfer function is:

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Digital Filters

•Structures for IIR systems – Part 17

Parallel-form structures – Part 5
Example (continued): The resulting signal flow graph

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Digital Filters

•Structures for IIR systems – Part 18

Cascaded and Parallel-form structures
Partner work – Please think about the following question and try to find answers
(first group discussions, afterwards broad discussion in the whole group).
❑ What are the differences of the cascaded and parallel form structures?

…………………………………………………………………………………………………………………………………………………………………………………….
…………………………………………………………………………………………………………………………………………………………………………………….

❑ Can you think of applications / hardware architectures where

you would prefer on of the structures?
What do you need to know about the hardware in order to make such a decision?

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Digital Filters

•Coefficient Quantization and Rounding Effects – Part 1

Errors resulting from rounding and truncation – Part 1
In this section we discuss the effects of a fixed-point digital filter implementation on the system performance.

Number representation in fixed-point format:

A real number can be represented as

where is the digit, is the radix (base), the number of integer digits, and the number of fractional digits. Example:

Most important in digital signal processing:

❑ Binary representation with and , most significant bit (MSB) and least significant bit (LSB).
❑ -bit fraction format: binary point between and numbers between and
are possible.

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•Coefficient Quantization and Rounding Effects – Part 2

Errors resulting from rounding and truncation – Part 2
Number representation in fixed-point format (continued):
Positive numbers are represented as

The negative fraction

can be represented with one of the three following formats:

❑ Signs-magnitude format: … with …

❑ One’s-complement format:
Most DSPs use two’s-complement arithmetic
(because of a good “temporary overflow” handling)
❑ Two’s complement format:

where denotes a binary addition.

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•Coefficient Quantization and Rounding Effects – Part 3

Errors resulting from rounding and truncation – Part 3

Number representation in fixed-point format (continued):

Example:
Express the fraction and in sign-magnitude, two’s complement and one’s complement.
❑ can be represented as such that
❑ can be represented
❑ in sign-magnitude format as
❑ in one’s complement format as
❑ in two’s complement format as

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Digital Filters

•Coefficient Quantization and Rounding Effects – Part 4

Errors resulting from rounding and truncation – Part 4
Truncation and rounding:
Problem: Multiplication of two -bit numbers yield a result of length
truncation/rounding necessary
can again be regarded as quantization of the (filter) coefficient
Suppose that we have a fixed-point realization in which a number is quantized from
to bits.

We first discuss the truncation case. Let the truncation error be defined as .
❑ For positive numbers the error is

Truncation leads to a number smaller than the non-quantized number.

❑ For negative numbers and the sign-magnitude representation the error is

Truncation reduces the magnitude of the number.

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Digital Filters

•Coefficient Quantization and Rounding Effects – Part 5

Errors resulting from rounding and truncation – Part 5
Truncation and rounding (continued):
❑ For negative numbers in the two’s complement case the error is

❑ Quantization characteristics for a continuous input signal :

Sign-magnitude Two’s-complement
representation representation

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Digital Filters

•Coefficient Quantization and Rounding Effects – Part 6

Errors resulting from rounding and truncation – Part 6
Truncation and rounding (continued):
Rounding case: The Rounding error is defined as
❑ Rounding affects only the magnitude of the number and is independent from the type of fixed-point realization.
❑ Rounding error is symmetric around zero and falls in the range

❑ Quantization characteristic function:

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Digital Filters

•Coefficient Quantization and Rounding Effects – Part 7

Numerical overflow – Part 1:
If a number is larger/smaller than the maximal/minimal possible number representation,
❑ for sign magnitude and one’s-complement arithmetic,
❑ and , resp., for two’s-complement arithmetic,
we speak of an overflow/underflow condition.

Overflow example in two’s-complement arithmetic (range )

The resulting error can be very large when overflow/underflow occurs.

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Digital Filters

•Coefficient Quantization and Rounding Effects – Part 8

Numerical overflow – Part 2
Two’s-complement quantizer for :

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Digital Filters

•Coefficient Quantization and Rounding Effects – Part 9

Numerical overflow – Part 3

Alternative: saturation or clipping. The error does not increase abruptly in magnitude when overflow/underflow occurs:

Disadvantage: “Summation property” of the two’s-complement representation is violated.

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Digital Filters

•Coefficient Quantization and Rounding Effects – Part 10

Coefficient Quantization and Rounding Effects
Partner work – Please think about the following question and try to find answers
(first group discussions, afterwards broad discussion in the whole group).
❑ What are the most prominent representations in fixed-point arithmetic?
…………………………………………………………………………………………………………………………………………………………………………………….
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❑ How large / small can be the result of an addition / multiplication of two fixed-point numbers
(e.g. each being represented by a 16 bit value)?
…………………………………………………………………………………………………………………………………………………………………………………….
…………………………………………………………………………………………………………………………………………………………………………………….
❑ What do you know about number representations in floating-point arithmetic?
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Digital Filters

•Coefficient Quantization and Rounding Effects – Part 11

Coefficient quantization errors – Part 1
❑ In a DSP/hardware realization of an FIR/IIR filter the accuracy is limited by the word length of the computer
Coefficients obtained from a design algorithm have to be quantized.
❑ Word length reduction of the coefficients leads to different poles and zeros to the desired ones. This may lead to
❑ modified frequency response with decreased selectivity,
❑ stability problems.

Sensitivity to quantization of filter coefficients

Direct form realization, quantized coefficients:

and represent the quantization errors.

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Digital Filters

•Coefficient Quantization and Rounding Effects – Part 12

Effect of quantization of coefficients:

Matlab example for

“robust” filter
design …

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•Coefficient Quantization and Rounding Effects – Part 13

Coefficient quantization errors – Part 2
Sensitivity to quantization of filter coefficients (continued)
As an example, we are interested in the deviation , when the denominator coefficients are quantized
( denotes the resulting pole after quantization). It can be shown that this expression can be expressed as
(Proakis, Manolakis, 1996, pp. 569):

Basic derivation on the

blackboard!

From this equation we can observe the following:

❑ By using the direct form, each single pole deviation depends on all quantized denominator coefficients .
❑ The error can be minimized by maximizing the distance between the poles and

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Digital Filters

•Coefficient Quantization and Rounding Effects – Part 14

Coefficient quantization errors – Part 3
Sensitivity to quantization of filter coefficients (continued)
Splitting the filter into single or double pole sections (first or second order transfer functions):
❑ Combining the poles and into a second order section leads to a small perturbation error ,
since complex conjugate poles are normally sufficient far apart.
❑ Realization in cascade or parallel form:
The error of a particular pole pair and is independent of its distance from the other poles of the transfer function.

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Digital Filters

•Coefficient Quantization and Rounding Effects – Part 15

Coefficient quantization errors – Part 4
Example: Effects of coefficient quantization
(a) (b)

(c), (d), and (e) are quantized with 16 bits

(c) (d) (e)

Elliptic filter of order (Example taken from [Oppenheim, Schafer 1999])

Unquantized: (a) Magnitude frequency response Quantized: (c) Passband detail for cascade structure
(b) Passband detail (d) Passband detail for parallel structure
(e) Magnitude frequency response for direct structure
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•Coefficient Quantization and Rounding Effects – Part 16

Coefficient quantization errors – Part 5
Pole locations of quantized second order sections
Consider a two-pole filter with the transfer function

Poles: , coefficients: , stability condition:

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•Coefficient Quantization and Rounding Effects – Part 17

Coefficient quantization errors – Part 6
Pole locations of quantized second order sections (continued)
Quantization of and with bits possible pole positions:

Low density for poles (at low frequencies)

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•Coefficient Quantization and Rounding Effects – Part 18

Coefficient quantization errors – Part 7
Pole locations of quantized second order sections (continued)
❑ Non-uniformity of the pole position is due to the fact that is quantized, while the pole locations
are proportional .
❑ Sparse set of possible pole locations around and . Disadvantage for realizing lowpass filters where the
poles are normally clustered near .
Alternative: Coupled-form realization

Analysis on the blackboard

Which corresponds to the following signal flow graph:

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Digital Filters

•Coefficient Quantization and Rounding Effects – Part 19

Coefficient quantization errors – Part 8
Pole locations of quantized second order sections (continued)
By transforming the equations into the z-domain, the transfer function of the filter can be obtained as

❑ We can see from the signal flow graph that the two coefficients and are now linear in , such that a
quantization of these parameters lead to equally spaced pole locations in the z-plane:

Equally distributed density for poles

(now better behavior at low frequencies)

❑ Disadvantage. Increased computational complexity compared to the direct form.

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•Coefficient Quantization and Rounding Effects – Part 20

Coefficient quantization errors – Part 9
Cascade or parallel form
Cascade form:

Parallel form:

❑ Cascade form: Only the numerator coefficients of an individual section determine the perturbation of the
corresponding zero locations direct control over the poles and zeros
❑ Parallel form: A particular zero is affected by quantization errors in the numerator and denominator coefficients of all
individual sections numerator coefficients and do not specify the position of a zero directly, direct control over
the poles only.

Cascaded structures are more robust against coefficient quantization and should be used in most cases.

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•Coefficient Quantization and Rounding Effects – Part 21

Coefficient quantization errors – Part 10
Cascade or parallel form (continued)
Example: Elliptic filter of order , frequency and phase response ([Proakis, Manolakis 96])

Cascade form (3 digits bits) Parallel form ( bits)

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•Coefficient Quantization and Rounding Effects – Part 22

Coefficient quantization errors – Part 11
Coefficient quantization in FIR systems
In FIR systems we only have to deal with the locations of the zeros, since for causal filters all poles are at .

Remarks:
❑ For FIR filters an expression analogous to the deviation and the original and quantized poles can be derived for the zeros.
FIR filters might also be realized in cascade form according to

with second order subsections, in order to limit the effects of coefficient quantization to zeros of the actual subsection only.
❑ However, since the zeros are more or less uniformly spread in the z-plane, in many cases the direct form is also used with
quantized coefficients.
❑ For a linear-phase filter that has a symmetric or asymmetric impulse response, quantization does not affect the phase
characteristics, but only the magnitude.

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Digital Filters

•Coefficient Quantization and Rounding Effects – Part 23

Coefficient quantization errors – Part 12
Partner work – Please think about the following question and try to find answers
(first group discussions, afterwards broad discussion in the whole group).
❑ What are the drawbacks of parallel filter structures? Are there also advantages?

…………………………………………………………………………………………………………………………………………………………………………………….
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❑ Why are FIR filters not as critical in terms of precision compared to IIR filters?

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❑ Why are in today’s processors sometimes the direct structures better than cascaded structures for FIR filters
(answer can not be found in the slides)?

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•Coefficient Quantization and Rounding Effects – Part 24

Zero-input limit cycles – Part 1
❑ Stable IIR filters implemented with infinite-precision arithmetic: If the excitation becomes zero and remains zero for
then the output of the filter will decay asymptotically towards zero.
❑ Same system implemented with fixed-point arithmetic: Output may oscillate indefinitely with a periodic pattern while
the input remains equal to zero: Zero-input limit cycle behavior, due to nonlinear quantizers in the feedback loop or
overflow of additions.
In the following the effects are shown with two examples:

Limit cycles due to round-off truncation

Given: First-order system with the difference equation

Register length for storing and the intermediate results: 4 bits (sign bit plus 3 fractional bits)
product must be rounded or truncated to 4 bits, before adding to .

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•Coefficient Quantization and Rounding Effects – Part 25

Zero-input limit cycles – Part 2
Limit cycles due to round-off truncation (continued)
Signal flow graphs:

Infinite-precision
system:

Nonlinear system due to

quantization:

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•Coefficient Quantization and Rounding Effects – Part 26

Zero-input limit cycles – Part 3
Limit cycles due to round-off truncation (continued)
Nonlinear difference equation ( represents two‘s-complement rounding):

Suppose we have

Then:

Quantization with rounding (+ 0.000100)

A constant steady value is obtained for .

For we have a periodic steady-state oscillation between and .
Such periodic outputs are called limit cycles.

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•Coefficient Quantization and Rounding Effects – Part 27

Zero-input limit cycles – Part 4
Limit cycles due to round-off truncation (continued)

From [Oppenheim, Schafer, 1999]

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•Coefficient Quantization and Rounding Effects – Part 28

Zero-input limit cycles – Part 5
Limit cycles due to overflow
Consider a second-order system realized by the difference equation:

represents two‘s-complement rounding with one sign and 3 fractional digits.

Overflow can occur with the two‘s-complement addition of the products.
Suppose that

Then we have:

continues to oscillate unless an input is applied.

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•Coefficient Quantization and Rounding Effects – Part 29

Zero-input limit cycles – Part 6
Remarks

❑ Some solutions for avoiding limit cycles:

❑ Use of structures which do not support limit-cycle oscillations.
❑ Increasing the word length.
❑ Use of a double-length accumulator and quantization after the accumulation of products.
❑ FIR-filters are limit-cycle free since there is no feedback involved in its signal flow graph.

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•Coefficient Quantization and Rounding Effects – Part 30

Zero-input limit cycles – Part 7

Partner work – Please think about the following question and try to find answers
(first group discussions, afterwards broad discussion in the whole group).
❑ What kind of limit cycles is more critical? Please, give reasons for your answer!

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❑ What can you do to avoid overflow-based limit cycles?

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❑ What can you do to avoid truncation-based limit cycles?

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•Design of FIR Filters – Part 1

General remarks (IIR and FIR filters) – Part 1
❑ Ideal filters are non-causal, and thus physically unrealizable for real-time signal processing applications.
❑ Causality implies that the filter response cannot have an infinitely sharp cut-off from passband to stopband,
and that the stopband amplification can only be zero for a finite number of frequencies .

Magnitude characteristics of physically realizable filter ( ):

: passband ripple,
: stopband ripple,
: passband edge frequency,
: stopband edge frequency

From [Proakis, Manolakis, 1996]

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Digital Filters

•Design of FIR Filters – Part 2

General remarks (IIR and FIR filters) – Part 2
Filter design problem:
❑ Specify and corresponding to the desired application,
❑ Select the coefficients and (free parameters), such that the resulting frequency response best
satisfies the requirements for and .
❑ The degree which approximates the specifications depends on the criterion for selecting the and the
and also on the numerator and denominator degree (the number of coefficients).

How we will continue:

❑ Before we will start of “optimal” design procedures, we will first focus on very simple design schemes.
❑ However, due to their low complexity they are suitable for real-time filter design.
❑ In addition, we will first focus on linear-phase FIR filters.

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 60
Digital Filters

•Design of FIR Filters – Part 3

Linear-phase filters – Part 1
Important class of FIR filters, which we will mainly consider in the following.

Definition:
A filter is said to be a linear-phase filter, if its impulse response satisfies the condition :

With the definition and odd, this leads to a z-transform:

For even we have

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 61
Digital Filters

•Design of FIR Filters – Part 4

Linear-phase filters – Part 2
Result from the last slide for an even filter length and :

When we now substitute with and multiply both sides both sides by we obtain with the definition of a
linear-phase filter:

… multiplication of both sides with …

… simplification and exchanging the order of the addends …

… inserting the result from above …

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 62
Digital Filters

•Design of FIR Filters – Part 5

Linear-phase filters – Part 3

Generalizing the result of the previous slide for all four cases, leads to

which is the z-transform equivalent to the definition of a linear-phase filter.

Consequences:
❑ The roots of the polynomial are identical to the roots of the polynomial :
If is a zero of then is also a zero.
❑ If additionally the impulse response is real-valued, the roots must occur in complex-conjugate pairs:
If is a zero of then is also a zero.

The zeros of a real-valued linear-phase filter occur in quadruples in the z-plane

(exception: zeros on the real axis, zeros on the unit circle).

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 63
Digital Filters

•Design of FIR Filters – Part 6

Linear-phase filters – Part 4
Consequences (continued):
Example: Pole-zero-diagram of a linear-phase filter

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 64
Digital Filters

•Design of FIR Filters – Part 7

Linear-phase filters – Part 5
(a) Type-1 linear-phase system
Definition: Odd length , even symmetry . Frequency response:

… using that …

… abbreviating the term in brackets …

Real term, thus we have a linear phase due to !

As a result we get for the phase of that filter type:

Remember:

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 65
Digital Filters

•Design of FIR Filters – Part 8

Linear phase filters – Part 6
(a) Type-1 linear phase system (continued)
Impulse and (amplitude)
frequency response:

On the following slides

equivalent derivations
for the other cases
(even/odd, type of
symmetry) will be
derived! The next
seven slides are for
reading at home!

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 66
Digital Filters

•Design of FIR Filters – Part 9

Linear-phase filters – Part 7
(b) Type-3 linear-phase system
Odd length , odd symmetry .
Frequency response:

… using that and since …

… abbreviating the term in brackets with and using …

Result:
❑ Linear phase:
❑

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 67
Digital Filters

•Design of FIR Filters – Part 10

Linear-phase filters – Part 8
(b) Type-3 linear-phase system (continued)
Impulse and (amplitude)
frequency response:

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 68
Digital Filters

•Design of FIR Filters – Part 11

Linear-phase filters – Part 9
(c) Type-2 linear-phase system
Even length , even symmetry .
Frequency response:

… using that …

… abbreviating the term in brackets with …

Result:
❑ Linear phase:
❑

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 69
Digital Filters

•Design of FIR Filters – Part 12

Linear-phase filters – Part 10
(c) Type-2 linear-phase system (continued)
Impulse and (amplitude)
frequency response:

Note that is not

periodic with . That’s true
only for ! The phase term
makes again
periodic with !

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 70
Digital Filters

•Design of FIR Filters – Part 13

Linear-phase filters – Part 11
(d) Type-4 linear-phase system
Even length , odd symmetry .
Frequency response:

… using that …

… abbreviating the term in brackets with and using …

Result:
❑ Linear phase:
❑

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 71
Digital Filters

•Design of FIR Filters – Part 14

Linear-phase filters – Part 12
(d) Type-4 linear-phase system (continued)
Impulse and (amplitude)
frequency response:

Note that also is not

periodic with . That’s true
only for ! The phase term
makes again
periodic with !

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 72
Digital Filters

•Design of FIR Filters – Part 15

Linear-phase filters – Part 13
Applications:
❑ Type-1 and type-2 filters are used for “ordinary” filtering, however type-2 filters are not suitable for high-pass filtering.
❑ Type-3 and type-4 filters for example are used for 90 degree phase shifters and so-called Hilbert transformers.

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 73
Digital Filters

•Design of FIR Filters – Part 16

Linear-phase filters – Part 14

Partner work – Please think about the following question and try to find answers
(first group discussions, afterwards broad discussion in the whole group).
❑ What types of linear-phase filters do we have? How do they differ?

…………………………………………………………………………………………………………………………………………………………………………………….
…………………………………………………………………………………………………………………………………………………………………………………….

❑ Why is the term not always periodic with ?

…………………………………………………………………………………………………………………………………………………………………………………….
…………………………………………………………………………………………………………………………………………………………………………………….

❑ Do you know applications where linear-phase filters would be beneficial (compared to other filter types)?

…………………………………………………………………………………………………………………………………………………………………………………….
…………………………………………………………………………………………………………………………………………………………………………………….

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 74
Digital Filters

•FIR Filters
C Code for FIR Filters

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 75
Digital Filters

•Design of FIR Filters – Part 17

Linear-phase filters – Part 15
Design of linear-phase filters using a window function
Given: Desired frequency response

Thus, the impulse response can be obtained using the inverse Fourier-transform:

Examples for “desired” filters:

❑ Ideal lowpass, highpass, or bandpass filters
❑ Delay filters (delaying a signal by a non-integer amount of samples, “fractional delay”)
❑ Hilbert filters (e.g. for frequency shifting)

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 76
Digital Filters

•Design of FIR Filters – Part 18

Linear phase filters – Part 16
Design of linear-phase filters using a window function (continued)
The impulse response has generally infinite length.
Truncation to the length by multiplication with a window function is necessary:

Rectangular window:

Frequency response of the rectangular window (see section about “Frequency analysis of stationary signals”
in the “DFT and FFT” chapter):

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 77
Digital Filters

•Design of FIR Filters – Part 19

Linear phase filters – Part 17
Design of linear-phase filters using a window function (continued)
Suppose, we want to design a linear-phase filter of length with the desired frequency response

where is denoting the cut-off frequency. For the corresponding impulse response we get:

Multiplication with a rectangular window of length leads to

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 78
Digital Filters

•Design of FIR Filters – Part 20

Linear phase filters – Part 18
Design of linear-phase filters using a window function (continued)

Examples for and :

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 79
Digital Filters

•Design of FIR Filters – Part 21

Linear phase filters – Part 19
Design of linear-phase filters using a window function (continued)
Disadvantage of using a rectangular window:
Large sidelobes lead to an undesirable ringing effects (overshoot at the boundary between pass- and stopband)
in the frequency response of the resulting FIR filter.
Gibbs phenomenon:
❑ Result of approximating a discontinuity in the frequency response with a finite number of filter coefficients and a
mean square error criterion
❑ The relation between and can be interpreted as a Fourier series representation with the Fourier
coefficients Gibbs phenomenon results from a Fourier series approximation.
❑ The squared integral error

approaches zero with increasing length of . However, the maximum value of the error
approaches a constant value (independent of the filter length).

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 80
Digital Filters

•Design of FIR Filters – Part 22

Linear phase filters – Part 20
Design of linear-phase filters using a window function (continued)
Use of other appropriate window functions with lower sidelobes in their frequency responses.

From [Proakis, Manolakis, 1996]

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 81
Digital Filters

•Design of FIR Filters – Part 23

Linear phase filters – Part 21
Design of linear-phase filters using a window function (continued)
Frequency-domain characteristics of some window functions [Proakis, Manolakis, 1996]:

Type of window Approximate transition Peak sidelobe

width of main lobe in dB
Rectangular -13
Bartlett -27
Hann -32
Hamming -43
Blackman -58

The parameter in the Kaiser window allows to adjust the width of the main lobe, and thus also to adjust the compromise
between overshoot reduction and increased transition bandwidth in the resulting FIR filter. denotes the Bessel function
of the first kind of order zero.

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 82
Digital Filters

•Design of FIR Filters – Part 24

Linear phase filters – Part 22
Design of linear-phase filters using a window function (continued)
Magnitude frequency response of the resulting linear-phase FIR filter, when different window functions
are used to truncate the infinite-length impulse response with the desired frequency response :

Achieved with a
Hamming window

Achieved with a
rectangular window

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 83
Digital Filters

•Design of FIR Filters – Part 25

Linear phase filters – Part 23
Design of linear-phase filters using a window function (continued)
Magnitude frequency response of the resulting linear-phase FIR filter, when different window functions
are used to truncate the infinite-length impulse response with the desired frequency response :

Achieved with a
Kaiser window
Achieved with a
Blackman window

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 84
Digital Filters

•Design of FIR Filters – Part 26

Linear phase filters – Part 24

Matlab
example

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 85
Digital Filters

•Design of FIR Filters – Part 27

Linear-phase filters – Part 24

Partner work – Please think about the following questions and try to find answers
(first group discussions, afterwards broad discussion in the whole group).
❑ What are the basic steps to get a stable, causal, finite, and linear-phase filter from a “desired” filter?
……………………………………………………………………………………………………………………………………………………………………………………..
……………………………………………………………………………………………………………………………………………………………………………………..

❑ What does the multiplication with a window function corresponds to in the frequency domain?
How should the spectrum of an “optimal” window function look like?
……………………………………………………………………………………………………………………………………………………………………………………..
……………………………………………………………………………………………………………………………………………………………………………………..

❑ What are the basic parameters that describe window functions in the frequency domain?

……………………………………………………………………………………………………………………………………………………………………………………..
……………………………………………………………………………………………………………………………………………………………………………………..

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 86
Digital Filters

•Design of FIR Filters – Part 28

Linear phase filters – Part 25
Frequency sampling design
The desired frequency response is specified at a set of equally spaced frequencies:

We could now design an FIR filter with a frequency response equal to the desired one at the above mentioned
frequency supporting points:

By combining both equations we obtain for :

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 87
Digital Filters

•Design of FIR Filters – Part 29

Linear phase filters – Part 26
Frequency sampling design (continued)
Multiplication with and summation over yields to

… multiplication with the exponential term mentioned above and summation …

… exchanging the summation order and rearranging the exponential …

… exploiting the properties of sums of exponentials …

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 88
Digital Filters

•Design of FIR Filters – Part 30

Linear phase filters – Part 27
Frequency sampling design (continued)
Resolving the result from the last slide to leads to

… dividing by and multiplication with …

Some remarks:
❑ The result can be computed efficiently using an IFFT!
❑ Note that only frequency supporting point are specified, the filter characteristic in between these supporting
points might be “not as expected”.
❑ This type of design is sometimes used in real-time applications (due to its low complexity)!

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 89
Digital Filters

•Design of FIR Filters – Part 31

Linear phase filters – Part 28

Optimum equiripple design (Chebyshev approximation)

❑ Window design techniques try to reduce the difference between the desired and the actual frequency response
(error function) by choosing suitable windows.
❑ How far can the maximum error be reduced?
The theory of Chebyshev approximation answers this question and provides us with algorithms to find the
coefficients of linear-phase FIR filters, where the maximum of the frequency response error is minimized.
❑ Chebyshev approximation:
Approximation that minimizes the maximum errors over a set of frequencies.
❑ The resulting filters exhibit an equiripple behavior in their frequency responses
equiripple filters.

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 90
Digital Filters

•Design of FIR Filters – Part 32

Linear phase filters – Part 29
Optimum equiripple design (Chebyshev approximation) (continued)

As we have shown before, every linear-phase filter has a frequency response of the form

where is a real-valued positive or negative function (amplitude frequency response).

It can be shown that for all types of linear-phase symmetry can always be written as a weighted sum of cosines.
For example, for type 1 linear-phase filters we have

with

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 91
Digital Filters

•Design of FIR Filters – Part 33

Linear phase filters – Part 30
Optimum equiripple design (Chebyshev approximation) (continued)
Problem definition:
Acceptable frequency response for the FIR filter:
❑ Linear phase,
❑ transition bandwidth between pass- and stopband,
❑ passband deviation from unity,
❑ stopband deviation from zero.
(Multiple bands are possible as well.)

In the following we will restrict ourselves to lowpass type 1 linear-phase filters.

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 92
Digital Filters

•Design of FIR Filters – Part 34

Linear phase filters – Part 31
Optimum equiripple design (Chebyshev approximation) (continued)
Approximation Problem: Given
❑ a compact subset of in the frequency domain (consisting of pass- and stop-band in the lowpass filter case),
❑ a desired real-valued frequency response , defined on ,
❑ a positive weight function , defined on , and
❑ the form of , here (type-1 linear phase)

This is a so-called “minimax” criterion.

Goal: Minimization of the error

over by the choice of .

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 93
Digital Filters

•Design of FIR Filters – Part 35

Linear phase filters – Part 32
Optimum equiripple design (Chebyshev approximation) (continued)
Alternation theorem (without proof):
If is a linear combination of cosine functions,

then a necessary and sufficient condition is that is the unique and best weighted Chebyshev approximation to a
given continuous function on is:
The weighted error function exhibits at least extremal frequencies in .
These frequencies are supporting points for which hold:

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 94
Digital Filters

•Design of FIR Filters – Part 36

Linear phase filters – Part 33
Optimum equiripple design (Chebyshev approximation) (continued)
❑ Consequences from the alternation theorem:
Best Chebyshev approximation must have an equiripple error function and is unique.
❑ Example: Amplitude frequency response of an optimum type 1 linear-phase filter with

[Parks, Burrus: Digital Filter Design, 1987]

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 95
Digital Filters

•Design of FIR Filters – Part 37

Linear phase filters – Part 34
Optimum equiripple design (Chebyshev approximation) (continued)
❑ If the extremal frequencies were known, we could use the frequency-sampling design from above to specify
the desired values at the extremal frequencies in the passband, and in the stopband, respectively.
How to find the set of extremal frequencies?

Remez exchange algorithm (Parks, McLellan, 1972)

❑ It can be shown that the error function

can be forced to take on some values for any given set of frequency points

Simplification
Restriction to and

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 96
Digital Filters

•Design of FIR Filters – Part 38

Linear phase filters – Part 35
Optimum equiripple design (Chebyshev approximation) (continued)
Remez exchange algorithm (continued)

This can be written as a set of linear equations according to R+1 equations!

We obtain a unique solution for the coefficients , and the error magnitude .
1 unknown!
R unknowns!
Finding the new set of extremal frequencies can be obtained using an FFT with zero padding:

❑ The frequency point are usually chosen in an equally spaced grid.

The number of the frequency points is approximately .
❑ The algorithm is initialized with a trial set of arbitrarily chosen frequencies

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 97
Digital Filters

•Design of FIR Filters – Part 39

Linear phase filters – Part 36
Optimum equiripple design (Chebyshev approximation) (continued)
Remez exchange algorithm (continued)

The steps of the Remez algorithm:

1. Solve the linear equation for the desired frequency response , yielding an error magnitude in the -th iteration.
2. Interpolate to find the frequency response on the entire grid of frequencies.
3. Search over the entire grid of frequencies for a larger magnitude error than obtained in step 1.
4. Stop, if no larger magnitude error can be found.
Otherwise, take the frequencies, where the error attains its maximum magnitude as a new trial set of extremal
frequencies and go to step 1.

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 98
Digital Filters

•Design of FIR Filters – Part 40

Linear phase filters – Part 37
Optimum equiripple design (Chebyshev approximation) (continued)
Remez exchange algorithm (continued)

[From: Parks, Burrus:

Digital Filter Design, 1987]

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 99
Digital Filters

•Design of FIR Filters – Part 41

Linear phase filters – Part 38
Remez exchange algorithm (continued)
Example:
Desired:

Problem: Choose the two coefficients and such that they minimize the Chebyshev error

(approximation of a parabola by a straight line).

Approach/ solution:
three extremal points
the resulting equations to be solved:

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 100
Digital Filters

•Design of FIR Filters – Part 42

Linear phase filters – Part 39
Remez exchange algorithm (continued)
Example:
1. Arbitrarily chosen trial set:
Matrix version of the linear equations:

2. Next trial set chosen as those three points, where the error

achieves its maximum magnitude

Linear equations to solve:

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 101
Digital Filters

•Design of FIR Filters – Part 43

Linear phase filters – Part 40
Remez exchange algorithm (continued)
Example:
3. Next trial set:
Linear equations to solve:

is the extremal point set.

After the third step the parameter does not change any more. Now the coefficients and are used for the final solution.

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 102
Digital Filters

•Design of FIR Filters – Part 44

Linear phase filters – Part 41
Remez exchange algorithm (continued)
Example:

[From: Parks, Burrus: Digital Filter Design, 1987]

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Digital Filters

•Design of FIR Filters – Part 45

Linear phase filters – Part 42
Remez exchange algorithm (continued)
Estimation of the filter length:
Given the stop-/ passband ripple and the transition bandwidth the necessary filter order can be
estimated as (Kaiser, 1974)

Design example:
Design a linear-phase lowpass filter with the specifications

weighting in the stopband.

The filter order estimate gives . Rounding up yields a filter length of

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 104
Digital Filters

•Design of FIR Filters – Part 46

Linear phase filters – Part 43
Remez exchange algorithm (continued)
Design example (continued):

In the passband the specifications are not satisfied.

Increasing the filter length by one,

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 105
Digital Filters

•Design of FIR Filters – Part 47

Linear phase filters – Part 44
Remez exchange algorithm (continued)
Design example (continued):

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 106
Digital Filters

•Design of FIR Filters – Part 48

Linear-phase filters – Part 45

Partner work – Please think about the following questions and try to find answers
(first group discussions, afterwards broad discussion in the whole group).
❑ What are the problems when designing an FIR filter using only frequency supporting points?
……………………………………………………………………………………………………………………………………………………………………………………..
……………………………………………………………………………………………………………………………………………………………………………………..

❑ What is optimized with a “minimax” criterion? What other criteria do you know?
……………………………………………………………………………………………………………………………………………………………………………………..
……………………………………………………………………………………………………………………………………………………………………………………..

❑ What are the basic steps of the Remez exchange algorithm?

……………………………………………………………………………………………………………………………………………………………………………………..
……………………………………………………………………………………………………………………………………………………………………………………..

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Digital Filters

•Predictor-based Filter Design

Magnitude frequency response with and without equalization
Equalization with FIR and IIR filters

❑ Predictor-based filter design

❑ Linear-phase extension

Without eq.
With eq.

Magnitude frequency response of the equalization filter

Frequency in Hz
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Digital Filters

•Predictor-based Filter Design

Effect of a Prediction-Error Filter in the Frequency Domain
Estimated power spectral densities
Input signal (speech)
Decorrelated signal (filter order = 16)

Frequency in Hz

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 109
Digital Filters

•Predictor-based Filter Design

Effect of a Prediction-Error Filter in the Frequency Domain

Prediction filter

Prediction-error Power adjustment

filter

Prediction filter

Inverse prediction-error filter Inverse power adjustment

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 110
Digital Filters

•Predictor-based Filter Design

Minimization without side conditions (they are included in the filter structure)
❑ Cost function:

The resulting prediction-error filter is minimum phase:

❑ An FIR-filter is computed, whose zeros are all inside the unit circle.
❑ Signals can pass the filter “maximally fast“.
❑ The inverse prediction filter (an IIR filter) is therefore automatically stable because all zeros turn into poles which are now
inside the unit circle as well.

The resulting filters are normalized:

❑ Frequency response of the prediction-error filter:

❑ Frequency response of the inverse filter:

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 111
Digital Filters

•Predictor-based Filter Design

Cost function:

❑ Minimization of the average error power

Lösung:
❑ Yule-Walker equation system

Robust and computationally efficient solution:

❑ Levinson-Durbin recursion

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 112
Digital Filters

•Predictor-based Filter Design

Interpretation of the impulse response as a signal:
With this point of view, the autocorrelation function can be estimated directly out of the impulse response

Magnitude frequency response can be interpreted as a power density spectrum:

With this point of view, the autocorrelation function can be estimated directly out of the magnitude frequency response by

This gives the option to modify the frequency response prior to computing the IDFT.

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 113
Digital Filters

•Predictor-based Filter Design

Impulse response
measurement
Autocorrelation
computation
Levinson-Durbin
recursion
Prediction-error filter
(FIR filter)

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 114
Digital Filters

•Predictor-based Filter Design

Design of the magnitude response, DFT, and
(magnitude) squaring

Smoothing

Modification

IDFT

Levinson-Durbin
recursion
Prediction-error filter
(FIR filter)

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 115
Digital Filters

•Predictor-based Filter Design

Design of the magnitude response, DFT, and
(magnitude) squaring

Smoothing

Robust inversion
Smoothing
(avoid division by zero)

Modification Modification

IDFT IDFT

Levinson-Durbin Levinson-Durbin
recursion recursion
Prediction-error filter Prediction-error filter
(FIR filter) (IIR filter)
Comparison

Filter selection
(FIR or IIR)

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 116
Digital Filters

•Predictor-based Filter Design

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 117
Digital Filters

•Predictor-based Filter Design

Linear-Phase Filter Structures – Part 1
The system properties of FIR filters can be used to design a linear-phase equalization filter. If an FIR filter with frequency response

is mirrored with respect to time, i.e.,

the frequency response of the mirrored filter is

Both filters connected in series result therefore in a linear-phase system. The attenuation (or gain) properties of the entire filter is
doubled compared to a single filter.

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Digital Filters

•Predictor-based Filter Design

Linear-Phase Filter Structures – Part 2

The property described on the previous slide can be exploited. In order not to do a “double” equalization, the filter is designed only
with the magnitude spectrum (instead of the power density spectrum of the filter). This is a simple way to achieve halving (in the
logarithmic domain) or taking the square root (in the linear domain) of the desired frequency response

The block diagram of the filter structure looks like this:

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Digital Filters

•Predictor-based Filter Design

Magnitude frequency responses with and without equalization
Example for FIR-based equalization filters
❑ 32 coefficients
Without eq.
With minimum-phase eq. ❑ No additional modifications in the frequency domain
With linear-phase eq.
❑ Minimum-phase and linear-phase approach
Magnitude frequency responses of the equalization filters
Minimum-phase eq.
Linear-phase eq.

Group delays of the equalization filters

Minimum-phase eq.
Linear-phase eq.

Frequency in Hz

Digital Signal Processing and System Theory | Advanced Digital Signal Processing| Digital Filters Slide 120
Digital Filters

•Design of IIR Filters – Part 1

Basics of IIR-Filter Design – Part 1

❑ In the following only design algorithms are discussed which convert an analog into a digital filter. However, there are also
numerous algorithms for directly designing an IIR filter in the z-domain (frequency sampling method, least-squares design).
❑ Why starting with an analog filter?
Analog filter design is a well developed field (lots of existing design catalogs).
❑ The problem can be defined in the z-domain, transformed into the s-domain and solved there, and finally transformed back
into the z-domain.
❑ Analog filter: Transfer function

with the filter coefficients and the filter order .

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Digital Filters

•Design of IIR Filters – Part 2

Basics of IIR-Filter Design – Part 2
Furthermore: Definition of the Laplace Transform

❑ Note that linear-phase designs are not possible for causal and stable IIR Filters, since the condition

has to be satisfied.
Mirror-image pole outside the unit-circle for every pole inside the unit circle.
Unstable filter.

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•Design of IIR Filters – Part 3

Filter design by impulse invariance – Part 1
Goal:
Design an IIR filter with an impulse response being the sampled version of the impulse response of a given analog filter

with being the sampling interval. For the frequency response (ideal sampling assumed) we obtain:

Remarks:
❑ should be selected sufficiently small to avoid aliasing.
❑ The method is not suitable to design highpass filters due to the large amount of possible aliasing.

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•Design of IIR Filters – Part 4

Filter design by impulse invariance – Part 2
Suppose that the poles of the analog filter are distinct. In that case we can transform the transfer function into a
partial-fraction expansion of :

with : coefficients of the partial-fraction expansion,

: poles of the analog filter.
The inverse Laplace transform yields

Sampling of yields

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•Design of IIR Filters – Part 5

Filter design by impulse invariance – Part 3

We obtain for the transfer function of :

… inserting the computation of (see last slide) …

… changing the summation order …

… using the summation for …

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•Design of IIR Filters – Part 6

Filter design by impulse invariance – Part 4

Thus, given an analog filter with poles the transfer function of the corresponding digital filter using the impulse
invariant transform is:

with poles at .

Note: This result holds only for distinct poles. The generalization to multiple-order poles is possible.

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•Design of IIR Filters – Part 7

Filter design by impulse invariance – Part 5
Example:
Problem:
Convert the analog filter with the transfer function

into a digital filter using the impulse invariant method.

Solution:
The poles of Partial fraction expansion yields:

We finally have:

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•Design of IIR Filters – Part 8

Filter design by impulse invariance – Part 6
Example (continued):
Magnitude frequency responses:

Digital filter: Analog filter:

[Proakis, Manolakis, 1996]

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Digital Filters

•Design of IIR Filters – Part 9

Bilinear transform – Part 1
Algebraic transform between the variables and .
Mapping of the entire -axis of the s-plane to one revolution of the unit circle in the z-plane.

Definition:

denoting the sampling interval.

The transfer function of the corresponding digital filter can be obtained from the transfer function of the analog filter
according to

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Digital Filters

•Design of IIR Filters – Part 10

Bilinear transform – Part 2
Properties:
❑ Rearranging the definition for yields

By substituting we obtain

If and if
causal, stable continuous-time filters map into causal stable discrete-time filters.

❑ By inserting into the above expression, it can be seen that for all values of on the -axis.
The -axis maps onto the unit circle (meaning that the bilinear transform is unique).

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•Design of IIR Filters – Part 11

Bilinear transform – Part 3
Properties (continued):
❑ Relationship between and :
Inserting and into the definition

Nonlinear mapping between and (warping of the frequency axis) according to

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•Design of IIR Filters – Part 12

Bilinear transform – Part 4
Properties (continued):
As a result we obtain for the nonlinear mapping between and :

This could be interpreted

as a warping of the
frequency axis.

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•Design of IIR Filters – Part 13

Bilinear transform – Part 5

s plane z plane

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•Design of IIR Filters – Part 14

Bilinear transform – Part 6
Remarks:
❑ The design of a digital filter often begins with
frequency specifications in the digital domain, which are converted to the
analog domain. The analog filter is then designed considering these specifications (i.e. using the classical approaches
from the following section) and converted back into the digital domain using the bilinear transform.
❑ When using this procedure, the parameter cancels out and thus can be set to an arbitrary value ( ).

Example:
Problem:
Design a digital single-pole lowpass filter with –3 dB frequency (cutoff frequency) of , using the bilinear
transform applied to the analog filter with the transfer function

with denoting the analog cut-off frequency.

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•Design of IIR Filters – Part 15

Bilinear transform – Part 7
Example (continued):
Solution:
is obtained from with

The analog filter has now the transfer function

which is transformed back into the digital domain by using the bilinear transform

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•Design of IIR Filters – Part 16

Bilinear transform – Part 8
Example (continued):
Solution:
The transfer function of the digital filter is

Note that the parameter has been divided out.

The frequency response is

Especially we have , and , which is the desired response.

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•Design of IIR Filters – Part 17

Questions:

Partner work – Please think about the following question and try to find answers
(first group discussions, afterwards broad discussion in the whole group).
❑ What can you set/adjust when using the impulse invariance method?

……………………………………………………………………………………………………………………………………………………………………………………..
……………………………………………………………………………………………………………………………………………………………………………………..

❑ If you would have to specify properties of a method that maps the Laplace domain into the z-domain,
what would you mention?
……………………………………………………………………………………………………………………………………………………………………………………..
……………………………………………………………………………………………………………………………………………………………………………………..

❑ When using the bilinear transform, where is the imaginary axis mapped in the z-domain?
……………………………………………………………………………………………………………………………………………………………………………………..
……………………………………………………………………………………………………………………………………………………………………………………..

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Digital Filters

•Design of IIR Filters – Part 18

Characteristics of commonly used analog filters – Part 1
❑ The design of a digital filter can be reduced to the design of an appropriate analog filter and then performing the
conversion from to .
❑ In the following we briefly discuss the characteristics of commonly used analog (lowpass) filters.
We will focus here on four different types of IIR filters:

❑ Butterworth filters
❑ Type 1 Chebyshev filters
❑ Type 2 Chebyshev filters
❑ Cauer filters

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Digital Filters

•Design of IIR Filters – Part 19

Characteristics of commonly used analog filters – Part 2

Examples for the

different filter types:

All filter are of order 4.

The bilinear transform

has been used to
create discrete filters.

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•Design of IIR Filters – Part 20

Characteristics of commonly used analog filters – Part 3
Butterworth filters
Lowpass Butterworth filters are
allpole-filters characterized by
the squared magnitude
frequency response

is the order of the filter,

is the -3 dB frequency
(cut-off frequency).

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•Design of IIR Filters – Part 21

Characteristics of commonly used analog filters – Part 4
Butterworth filters (continued)

Since , we get by analytic continuation into the whole s-plane

Poles of :

❑ The poles of occur on a circle of radius at equally spaced points in the s-plane.
❑ poles are located in the left half of the s-plane and belong to
❑ The remaining poles lie in the right half of the s-plane and belong to (stability!).

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•Design of IIR Filters – Part 22

Characteristics of commonly used analog filters – Part 5
Butterworth filters (continued)
Pole locations in the s-plane for :
Poles that belong to

Poles that belong to

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•Design of IIR Filters – Part 23

Characteristics of commonly used analog filters – Part 6
Butterworth filters (continued)
Frequency responses ( ):

[Proakis, Manolakis, 1996]

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Digital Filters

•Design of IIR Filters – Part 24

Characteristics of commonly used analog filters – Part 7
Butterworth filters (continued)
Estimation of the required filter order:
At the stopband edge frequency the squared magnitude frequency response can be written as

which leads to

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Digital Filters

•Design of IIR Filters – Part 25

Characteristics of commonly used analog filters – Part 8
Butterworth filters (continued)
Example:
Problem:
Determine the order and the poles of a lowpass Butterworth filter that has a -3 dB bandwidth of 500 Hz and an
attenuation of 40 dB at 1000 Hz,

❑ -3 dB frequency
❑ stopband frequency
❑ attenuation of 40 dB

Solution:
For the order we obtain

In order to be “on the safe side” we choose

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Digital Filters

•Design of IIR Filters – Part 26

Characteristics of commonly used analog filters – Part 9
Butterworth filters (continued)
Example (continued):
Properties of the resulting digital filter (transformation by the bilinear transform, )

Magnitude frequency response Transition band

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Digital Filters

•Design of IIR Filters – Part 27

Characteristics of commonly used analog filters – Part 10
Butterworth filters (continued)
Example (continued):
Properties of the resulting digital filter (transformation by the bilinear transform, )

Phase response Pole/zero locations

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Digital Filters

•Design of IIR Filters – Part 28

Characteristics of commonly used analog filters – Part 11
Chebyshev filters
Two types of Chebyshev filters:
❑ Type 1 filters are all-pole filters with equiripple behavior in the passband and monotonic characteristic
(similar to a Butterworth filter) in the stopband.
❑ Type 2 filters have poles and zeros (for finite s), and equiripple behavior in the stopband, but a monotonic
characteristic in the passband.

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Digital Filters

•Design of IIR Filters – Part 29

Characteristics of commonly used analog filters – Part 12
Type 1 Chebyshev filters
Squared magnitude
frequency response:

where is a parameter
related to the passband
ripple, and is the
-th order Chebyshev
polynomial (see next slide).

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•Design of IIR Filters – Part 30

Characteristics of commonly used analog filters – Part 13
Type 1 Chebyshev filter (continued)
The Chebyshev polynomial is defined as

and can be obtained by the recursive equation

Examples:
❑
❑
❑

represents a polynom of degree in .

Chebyshev behavior (minimizing the maximal error) in the passband (or in the stopband for 2 type filters)

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•Design of IIR Filters – Part 31

Characteristics of commonly used analog filters – Part 14
Type 1 Chebyshev filter (continued)
The filter parameter is related to the passband ripple:
For odd:

For even:

At the passband edge frequency we have such that

which establishes a relation between the passband ripple and the parameter .

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Digital Filters

•Design of IIR Filters – Part 32

Characteristics of commonly used analog filters – Part 15
Type 1 Chebyshev filter (continued)
Typical squared magnitude frequency responses for a Chebyshev type 1 filter ( ):

[Proakis, Manolakis, 1996]

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•Design of IIR Filters – Part 33

Characteristics of commonly used analog filters – Part 16
Type 2 Chebyshev filter
Squared magnitude
frequency response:

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Digital Filters

•Design of IIR Filters – Part 34

Characteristics of commonly used analog filters – Part 17
Type 2 Chebyshev filter
Estimation of the filter order:
Chebyshev filters only depend on the parameters and the ratio .
Using these values, it can be shown that the required order can be estimated as

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Digital Filters

•Design of IIR Filters – Part 35

Characteristics of commonly used analog filters – Part 18
Type 2 Chebyshev filter
Typical squared magnitude frequency responses for a Chebyshev type 2 filter ( ):

[Proakis, Manolakis, 1996]

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Digital Filters

•Design of IIR Filters – Part 36

Characteristics of commonly used analog filters – Part 19
Elliptic (Cauer) filters
❑ Elliptic filters have equiripple (Chebyshev) behavior in both pass- and stopband.
❑ The transfer function contains both poles and zeros, where the zeros are located on the -axis.
❑ The squared magnitude frequency response

where denotes the Jacobian elliptic function of order , and the parameter controls the passband ripple.
❑ The filter design is optimal in pass- and stopband in the equiripple sense:
However, other types of filters may be preferred due to their better phase response characteristics
(i.e. approximately linear-phase), for example the Butterworth filter.

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Digital Filters

•Design of IIR Filters – Part 37

Characteristics of commonly used analog filters – Part 20
Elliptic (Cauer) filters

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Digital Filters

•Design of IIR Filters – Part 38

Characteristics of commonly used analog filters – Part 21
Elliptic (Cauer) filters (continued)
Characteristic squared magnitude frequency responses for a elliptic filter ( ):

[Proakis, Manolakis, 1996]

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•Design of IIR Filters – Part 39

Characteristics of commonly used analog filters – Part 22
Elliptic (Cauer) filters (continued)
Estimation of the filter order:
Required order to achieve the specifications with the parameters and
( ):

where denotes the complete elliptic integral of the first kind (tabulated)

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•Summary

❑ Introduction
❑ Digital processing of continuous-time signals
❑ DFT and FFT
❑ Digital filters
❑ Structures for FIR systems
❑ Structures for IIR systems
❑ Coefficient quantization and round-off effects
❑ Design of FIR filters
❑ Design of IIR filters

❑ Multi-rate digital signal processing

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