1Introduction Signal Processing Haslerid 19014 (2)

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Signal Processing Design of

Integrated Analog and Digital

Filters

Prof. Paul Hasler

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Types of Integrated Filters

Integrated Filters

Digital Filters

(Binary valued)

Analog Filters

(Continuous or multivalued)

Analog

Continuous-Time

Filters

Analog

Sampled-Data

Filters

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Where to divide Analog and Digital?

A/D

Converter

Real
world
(analog)

DSP

Processor

Computer
(digital)

Real
world
(analog)

DSP

Processor

Computer
(digital)

ASP

IC

A/D

Specialized A/D

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Analog-Digital Comparison

Low SNR: Analog / High SNR: Digital

[Sarpeskar 1997]

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Analog-Digital Comparison

16bit

A/D

FFT

DSP

10 bit A/D

10

10 bit A/D

10

10 bit A/D

10

10 bit A/D

10

filter

filter

filter

filter

Input

Input

DSP

Application

Program

DSP

Application

Program

DSP

SNR < 10bits

• What if exponentially spaced FFT?

Practical Interpretation of Cost

Low SNR: Analog / High SNR: Digital

[Sarpeskar 1997]

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Circuit Issues for Filters

Programmability / Tunability: flexibility and complexity

Available for digital (clocks/ crystals) as well as

some analog (e.g. Floating-Gate) filters

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Circuit Issues for Filters

High Signal-to-Noise Ratio (resolution):

Ratio of the largest signal and the smallest signal

Largest signal: Harmonic Distortion (continuous-time filters), Range limitations

Smallest Signal: Noise

Programmability / Tunability: flexibility and complexity

Available for digital (clocks/ crystals) as well as

some analog (e.g. Floating-Gate) filters

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Circuit Issues for Filters

High Signal-to-Noise Ratio (resolution):

Ratio of the largest signal and the smallest signal

Largest signal: Harmonic Distortion (continuous-time filters), Range limitations

Smallest Signal: Noise

Insensitivity to environmental fluctuations:

Power-supply:

Power Supply Rejection Ratio

(PSRR)

Temperature, etc.

Programmability / Tunability: flexibility and complexity

Available for digital (clocks/ crystals) as well as

some analog (e.g. Floating-Gate) filters

Not a problem for digital filters,

but can be the cause of several
problems to other analog circuits

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Circuit Issues for Filters

High Signal-to-Noise Ratio (resolution):

Ratio of the largest signal and the smallest signal

Largest signal: Harmonic Distortion (continuous-time filters), Range limitations

Smallest Signal: Noise

Insensitivity to environmental fluctuations:

Power-supply:

Power Supply Rejection Ratio

(PSRR)

Temperature, etc.

Programmability / Tunability: flexibility and complexity

Available for digital (clocks/ crystals) as well as

some analog (e.g. Floating-Gate) filters

Not a problem for digital filters,

but can be the cause of several
problems to other analog circuits

Typically, sampling in amplitude / time results in

,

• the more complexity is needed ( S/H blocks, anti-aliasing filters),
• more power / lower-frequency / area

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Design of Analog Filters

• Find the transfer function for a given filter

• “Partition” the transfer function,

or an approximation,
into simple parts that can be implemented.

• Implement the transfer function in

a particular circuit technology

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Design of Analog Filters

• Find the transfer function for a given filter

• “Partition” the transfer function,

or an approximation,
into simple parts that can be implemented.

• Implement the transfer function in

a particular circuit technology

H(s) or H(z)

H

1

(s) H

2

(s) H

3

(s)

GND

V

1

[n]

V

out

[n]

GND

GND

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Design of Analog Filters

H(s) or H(z)

H

1

(s) H

2

(s) H

3

(s)

GND

V

1

[n]

V

out

[n]

GND

GND

As basic building blocks we have
• integrators, delay elements
• first-order (low-pass / bandpass)
• second order functions

(low-pass / bandpass / highpass)

The “circuit” design question is

how to make these functions
what inputs / outputs / internal variables

should be voltages / currents, etc.

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Design of Analog Filters

H(s) or H(z)

H

1

(s) H

2

(s) H

3

(s)

GND

V

1

[n]

V

out

[n]

GND

GND

Design style constrains

partition

Design style

constrains
choice of
transfer function

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Choosing H(s) or H(z) for a filter

Ideal lowpass filter

we can get other filters from lowpass

|H(s)|

frequency

Gain in db = 20 log

10

( amplitude )

= 10 log

10

(signal power)

f

pb

Passband

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H(s) or H(z) for a lowpass filter

Lowpass

log |H(s)|

log(frequency)

f

pb

f

sb

T

pb

T

sb

1

Passband

Transition Region

Stopband

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H(s) or H(z) for a lowpass filter

Lowpass

log |H(s)|

log(frequency)

f

pb

f

sb

T

pb

T

sb

1

Normalized gain

StopBand Frequency

PassBand Frequency

StopBand

Minimum

Gain

PassBand

Minimum

Gain

Passband

Transition Region

Stopband

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H(s) or H(z) for a Highpass filter

Highpass

log |H(s)|

log(frequency)

f

pb

f

sb

T

pb

T

sb

1

Normalized gain

StopBand Frequency

PassBand Frequency

StopBand

Minimum

Gain

PassBand

Minimum

Gain

Stopband

Transition Region

Passband

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H(s) or H(z) for a Highpass filter

Bandpass

log |H(s)|

log(frequency)

f

pb1

f

sb1

T

pb

T

sb

1

Normalized gain

Low StopBand

Frequency

Low PassBand

Frequency

StopBand

Minimum

Gain

PassBand

Minimum

Gain

f

sb1

High StopBand

Frequency

f

pb2

High PassBand

Frequency

Passband

Transition Region

Transition Region

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Four Canonical Cont-Time

Filters

Butterworth: Maximally flat in passband…moderate rolloff

Chebyshev : Faster rolloff by allowing ripples in passband or stopband

Elliptic: Fastest rollff by appowing ripples in both passband and stopband

Bessel: Near linear phase, slow rolloff

Classic Analog Filters (IIR digital filters):

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Four Canonical Cont-Time

Filters

Butterworth: Maximally flat in passband…moderate rolloff

Chebyshev : Faster rolloff by allowing ripples in passband or stopband

Elliptic: Fastest rollff by appowing ripples in both passband and stopband

Bessel: Near linear phase, slow rolloff

Other FIR (digital) filters….
Other filter design (H(s) or H(z)) techniques: Optimization approaches

Should we choose H(s) or H(z) for our representation?

Partially due to particular circuit tradeoffs

(tunability? tools? continuous tunability? Accuracy? power consumption?)

Classic Analog Filters (IIR digital filters):

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What happens if we just cascade first

order stages?

T(s) =

1

(1 + j s

τ

)

N

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What happens if we just cascade first

order stages?

T(s) =

1

(1 + j s

τ

)

N

10

2

10

3

10

4

10

-3

10

-2

10

-1

10

0

Frequency (Hz)

Filter gain

N=1

N=9

N=33

17

25

Corner shifts with N

Rolloff is not initially

sharp…..

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Butterworth Filter Design

T

pb

=

1

1 +

ε

2

Definition of

ε

:

Transfer Function: (low-pass)

T(s) =

1

1 + (-1)

N+1

ε

s

N

τ

N

T

pb

= 1/

2 for

ε

= 1

(

f

sb

τ

= 2

π

)

1

1 +

ε

2

ω

2N

τ

2N

|T(j

ω

)| =

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Butterworth Filter Design

T

pb

=

1

1 +

ε

2

Definition of

ε

:

Transfer Function: (low-pass)

T(s) =

1

1 + j(-1)

N+1

ε

2

s

N

τ

N

T

pb

= 1/

2 for

ε

= 1

Need to solve to meet the specification of T

sb

at f

sb

: Filter Order (N)

(

f

pb

τ

= 2

π

)

T

sb

2

( 1 +

ε

2

(

f

sb

/

f

pb

)

2N

) = 1

1

1 +

ε

2

ω

2N

τ

2N

|T(j

ω

)| =

To meet specifications, one chooses

the next largest integer

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Butterworth Filter Design

T

pb

=

1

1 +

ε

2

Definition of

ε

:

Transfer Function: (low-pass)

T(s) =

1

1 + j(-1)

N+1

ε

2

s

N

τ

N

T

pb

= 1/

2 for

ε

= 1

Need to solve to meet the specification of T

sb

at f

sb

: Filter Order (N)

(

f

sb

τ

= 2

π

)

T

sb

2

( 1 +

ε

2

(

f

sb

/

f

pb

)

2N

) = 1

1

1 +

ε

2

ω

2N

τ

2N

|T(j

ω

)| =

To meet specifications, one chooses

the next largest integer

Pole locations: (

ε

= 1)

1/τ

k

= (

1/τ

)

(

sin(

(2k-1)

π

/(2N)

) + j cos(

(2k-1)

π

/(2N)

)

),

k=1…N

T(j

ω

) =

1

(1 + s

τ

1

) (1 + s

τ

2

) … (1 + s

τ

N

)

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Butterworth Filter Design

10

2

10

3

10

4

10

-1

10

0

10

1

Frequency (Hz)

Sub

-Filter gain

τ

= 1.5915e-005

Q = 3.83, 1.31, 0.821,

0.630, 0.541, 0.504

Specs:

ε

= 1, f

pb

=1e4,

f

sb

=1.5e4, T

sb

= 1e-2

10

2

10

3

10

4

10

-6

10

-5

10

-4

10

-3

10

-2

10

-1

10

0

Frequency (Hz)

Filter gain

N = 12

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Chebyshev Filter Design

T

pb

=

1

1 +

ε

2

Definition of

ε

:

T

pb

= 1/

2 for

ε

= 1

(

f

pb

τ

= 2

π

)

Transfer Function(low-pass)

1

1 +

ε

2

cos

2

( N cos

-1

(

ωτ

) )

|T(j

ω

)| =

ωτ

>1

1 +

ε

2

cosh

2

( N cosh

-1

(

ωτ

) )

=

ωτ

<1

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Chebyshev Filter Design

T

pb

=

1

1 +

ε

2

Definition of

ε

:

T

pb

= 1/

2 for

ε

= 1

(

f

sb

τ

= 2

π

)

Transfer Function(low-pass)

Need to solve to meet the specification of T

sb

at f

sb

: Filter Order (N)

1

1 +

ε

2

cos

2

( N cos

-1

(

ωτ

) )

|T(j

ω

)| =

ωτ

>1

1 +

ε

2

cosh

2

( N cosh

-1

(

ωτ

) )

=

ωτ

<1

( 1 +

ε

2

cosh

2

( N cosh

-1

(f

sb

/ f

pb

) ) )

T

sb

= 1

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Chebyshev Filter Design

T

pb

=

1

1 +

ε

2

Definition of

ε

:

T

pb

= 1/

2 for

ε

= 1

(

f

sb

τ

= 2

π

)

Transfer Function(low-pass)

Need to solve to meet the specification of T

sb

at f

sb

: Filter Order (N)

Pole locations:

Where k goes from 1, 2, ….N

1

1 +

ε

2

cos

2

( N cos

-1

(

ωτ

) )

|T(j

ω

)| =

ωτ

>1

1 +

ε

2

cosh

2

( N cosh

-1

(

ωτ

) )

=

ωτ

<1

( 1 +

ε

2

cosh

2

( N cosh

-1

(f

sb

/ f

pb

) ) )

T

sb

2

= 1

1/τ

k

= (

1/τ

)

(

sin(

(2k-1)

π

/(2N)

) sinh(

(1/N) sinh

-1

(1/

ε

)

)

+ j cos(

(2k-1)

π

/(2N)

) cosh(

(1/N) sinh

-1

(1/

ε

)

)

)

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Chebyshev Filter Design

10

2

10

3

10

4

10

-3

10

-2

10

-1

10

0

Frequency (Hz)

Filter gain

10

2

10

3

10

4

10

-2

10

-1

10

0

10

1

Frequency (Hz)

Sub

-Filter gain

Specs:

ε

= 1, f

pb

=1e4,

f

sb

=1.5e4, T

sb

= 1e-2

N = 6

τ

= 22.034

µ

s, Q = 3.4645

τ

= 16.288

µ

s, Q = 12.804

τ

= 53.433

µ

s, Q = 1.0459

Fewer stages, higher Qs….

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Transformations between s and z

z

-1

=

e

-sT

Simple Transformation

s ~

z

-1

~ 1 - sT

(1 - z

-1

)

T

1

T = sampling period

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Transformations between s and z

Bilinear transform

s ~ ~

z

-1

~

1 - z

-1

1 + z

-1

1 – s T

1 + s T

T

1

z

-1

=

e

-sT

~ s

1 - e

-sT

1 + e

-sT

T

1

1 - ( 1 – sT +

0.5*

(sT)

2

-…)

1 + 1 – sT +

0.5*

(sT)

2

-…

T

1

Simple Transformation

s ~

z

-1

~ 1 - sT

(1 - z

-1

)

T

1

T = sampling period

~

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General Filter Topology

1

st

Order

Σ

1

st

Order

Σ

1

st

Order

Σ

1

st

Order

Σ

V

out

V

in

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General Filter Topology

N*N parameters

N poles, N zeros for N-th order filter

Problem is underspecified…therefore can optimize

for SNR, complexity, etc…

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Partitioning H(s) for Circuit

Implementation

“Partition” the transfer function into simple parts

Factorization into first order and second order terms

Nearest neighbor feedback (~LC ladder filter network)

Addition of factors

(maybe out of an approximation of an FIR filter)

Additional feedback / feedforward terms

Similar for either H(s) or H(z)

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Typical Filter Topologies

2

nd

Order

2

nd

Order

2

nd

Order

2

nd

Order

1

st

Order

V

in

V

out

Cascade of First and Second-Order Sections

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Typical Filter Topologies

2

nd

Order

2

nd

Order

2

nd

Order

2

nd

Order

1

st

Order

V

out

V

in

1

st

Order

1

st

Order

Σ

1

st

Order

Σ

1

st

Order

Σ

1

st

Order

Σ

Σ

Σ

1

st

Order

1

st

Order

Σ

Σ

1

st

Order

1

st

Order

Σ

Σ

1

st

Order

V

out

V

in

Nearest Neighbor Feedback (

Inspired by LC ladder network filters

)

V

in

V

out

Cascade of First and Second-Order Sections

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Conclusions

Basic directions for integrated circuit filters

• Continuous or Discrete: Time and/or Amplitude

• High level specifications of filters

• Obtaining a filter function (H(s) or H(z))

• Implementing the filter function into basic blocks

(first and second-order filter sections, integrators, delays, etc.)


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