made

© All Rights Reserved

0 views

Digital Filters

made

© All Rights Reserved

- Designing a Low- Pass Fir Digital Filter By Using Bartlett Hanning and Blackman Harris Window Technique
- Labview - Digital Filter Design Toolkit Reference Manual
- Attendance System by Biometric Authorization by Speech
- Chapter 2_ FIR filters - Digital Filter Design - mikroElektronika.pdf
- 1391969106
- 8 m.tech Ece Communication 09 11 Verifi
- Origin Tutorials
- Designing of IIR Filter using Radix-4 Multiplier by Precharging Technique
- Altera Implementing FIR Filters and FFTs With 28-Nm Variable-Precision DSP Architecture
- DSP Basics
- IIR Filters
- dsf
- 25.108_Digital Filter Design Using FDATool
- Morales L.G. (Ed.) Adaptive Filtering Applications
- Ch6(2)
- impz function in matlab help
- ECE341 Fall 2016 Syllabus
- fwhomfbrhi27435FWHOMFBRHI
- IJETR032560
- Papers

You are on page 1of 38

Digital Filters

Dr. H. M. Wijekoon

Chief Engineer (Transmission Planning)-R3

644, Sri Jayawardanapura Mawatha

Ethulkotte

1

Filters

Filters may be classified as either digital or analog.

Digital filters are implemented using a digital computer or

special purpose digital hardware.

Analog filters may be classified as either passive or active and

are usually implemented with R, L, and C components and

operational amplifiers.

An active filter is one that, along with R, L, and C components,

also contains an energy source, such as that derived from an

operational amplifier.

A passive filter is one that contains only R, L, and C components.

It is not necessary that all three be present. L is often omitted

(on purpose) from passive filter design because of the size and

cost of inductors – and they also carry along an R that must

be included in the design.

Generally speaking, digital filters have become the focus

of attention in the last 40 years. The interest in digital

filters started with the advent of the digital computer,

especially the affordable PC and special purpose signal

processing boards. People who led the way in the work

(the analysis part) were Kaiser, Gold and Radar.

equation(s) in computer software. There are no R, L, C

components as such. However, digital filters can also be

built directly into special purpose computers in hardware

form. But the execution is still in software.

Signals

Real signals are comprised of a number of frequencies. Some

signals may contain both high frequency and low frequency

components.

Depending on the application, some frequencies may be

undesirable, such as a low frequency AC power supply hum or

interference from some other source.

Filters can be used to remove these undesirable frequency

components.

As an example, the signal shown on the above diagram has five

components, marked in increasing frequency order, f1 to f5.

A filter could be used to remove f1 and f2.

The circuit shown at the top right acts as a filter that will remove

most of the frequencies f1 and f2 so that only the higher

frequencies f3, f4 and f5 remain..

The main filter types are as follows:

Low-pass Filters (LPF) - These filters pass low frequencies and

stop high frequencies.

lowpass

Ideal Practical

stop low frequencies.

highpass highpass

Band pass Filters (BPF) - These filters pass a range of frequencies

and stop frequencies below and above the set range.

bandpass bandpass

Band-Stop Filters (BSF) - These filters pass all frequencies except the

ones within a defined range.

bandstop bandstop

All-Pass Filters (APF) - These filters pass all frequencies, but they

modify the phase of the frequency components.

High Pass

H(w) is a frequency-dependent complex function. This means

that the gain and phase will vary with frequency.

dependent components, it is natural to expect both to be

frequency- dependent.

Cut-Off Frequency

Cut-off frequency is defined as the frequency where the gain of

the filter falls to 1/ 2 = 1/ 1.414 = 0.707 of its value in the pass

band. It is also referred to as the -3dB point since (20log10 (0.707)

= -3).

Digital Filters

Mapping from s-plane to z-plane (from last lecture)

𝑆𝑖𝑛𝑐𝑒 𝑧 = 𝑒 𝑠𝑇 = 𝑒 (𝜎+𝑗𝜔)𝑇 = 𝑒 𝜎𝑇 𝑒 𝑗𝜔𝑇 𝑤ℎ𝑒𝑟𝑒 𝑇 = 2𝜋/𝜔𝑠

we can map the s-plane to the z-plane as below:

Frequency Response from pole-zero locations

The transfer function H[z] can be expressed in factorized polynomial

o Therefore we can compute the frequency be evaluating H[z] at

z=ejΩ, which is the unity circle.

o Each term (z-zi) can be evaluated as shown:

Therefore, for all the poles and zeros, we can use the graphical

method (similar to the s-plane case):

𝑧 − 𝑧1 𝑧 − 𝑧2 𝑧 − 𝑧3 … … … (𝑧 − 𝑧𝑛 )

𝐻 𝑍 = 𝑏0

𝑧 − 𝛾1 𝑧 − 𝛾2 𝑧 − 𝛾3 … . . (𝑧 − 𝛾𝑛 )

𝑟1 𝑟2 𝑟3 …………………..𝑟𝑛

H[𝑒 𝑗𝜔 ] = 𝑏0

𝑑1 𝑑2 𝑑3 …………….𝑑𝑛

Pole-Zero locations & Filtering

Therefore, for all the poles and zeros, we can use the graphical

method (similar to the s-plane case):

Digital Filters

We will now examine digital filters, starting with a simple example -

a moving average filter (see flow diagram below). Let us first

identify the major components of a digital filter

The input of a digital filter is a series of discrete samples obtained by sampling

the input waveform. The sampling rate must meet the Nyquist criteria that we

covered in our sampling lecture (highest frequency of input signal < = 1/2 x

sampling frequency). The term x(n) means the input at a time (n).

Z -1 Unit Delay

Z-1 represents a time delay that is equal to the sampling period. This is also

called a unit delay. Therefore, each z box delays the samples for one sampling

period. In the diagram, this is shown by the input going into the delay box as

x(n) and coming out as x(n-1). We see this because x(n) means the input at a

time (n), and x(n-1) means the input at time (n-1). What actually happens is

that x(n-1) is the previous input that has been

saved in the memory of the DSP

The output of each delay box is called a tap. Taps are usually fed into scalers

which scale the value of the delayed sample to the required value by multiplying

the input (or delayed input) by a coefficient. In the diagram, these are marked as

b0, b1 and b2. The scaling factor is called the weight. In mathematical terms, the

weight is multiplied by the delayed input, so the output of the first tap is

b0*x(n). The next tap output will be b1*x(n-1), and the output of the last tap is

b2*x(n-2).

Summing Junctions

The output of the weights are fed into summing junctions, which

add the weighted, delayed, forward-fed forward outputs from

taps. So in this example, the output of the first summing junction

is b0*x(n) + b1*x(n-1). At the next summing junction, this is

added to the output of the final tap, giving b0*x(n) + b1*x(n-1) +

b2*x(n-2), which is the output.

The Output y(n)

delayed and weighted samples, and is usually called y(n).

Linear Time-Invariant Digital Filters

Linear time-invariant (LTI) filters are a class of filters whose

output is a linear combination of the input signal samples and

whose coefficients do not vary with time.

The linear property entails that the filter response to a weighted

sum of a number of signals, is the weighted sum of the filter

responses to the individual signals.

This is the principle of superposition. The term time invariant

implies that the filter coefficients and hence its frequency

response is fixed and does not vary with time.

In the time domain the input-output relationship of a discrete-

time linear filter is given by the following linear difference

equation:

Recursive and non-Recursive Filters

There are two categories of digital filter: the recursive filter and

the non-recursive filter.

filters and finite impulse response (FIR) filters, respectively

its outputs as an input.

(commonly referred to as infinite impulse response (IIR).

response. Some implementations of moving average filter are

recursive filters but with a finite impulse response

Recursive and non-Recursive Filters

The transfer function of the filter is the ratio of two

polynomials in the variable z and may be written in a cascade

form as

zero, filter given by

recursive filter given by

Non-Recursive or Finite Impulse Response (FIR) Filters

A non-recursive filter has no feedback and its input-output

relation is given by

of the input signal x(m).

The response of such a filter to an impulse consists of a finite

sequence of M+1 samples, where M is the filter order.

Hence, the filter is known as a Finite-Duration Impulse Response

(FIR) filter.

Other names for a non-recursive filter include all-zero filter,

feed-forward filter or moving average (MA) filter a term usually

used in statistical signal processing literature.

Recursive or Infinite Impulse Response (IIR) Filters

its output is a function of the previous output samples and the

present and past input samples as described by the following

equation

output persists forever.

Impulse Response (IIR) filter.

Figure shows a particular case of an IIR filter when the output is a

function of N previous output samples and the present input

sample given by

Filtering, Convolution

it follows that the operation of filtering of a signal x(m) can be

mathematically expressed as the convolution of the input

signal and the impulse response of the filter h(m) as

following four sub-operations:

(1) Fold the signal x(k) to yield x(-k), this is done because the samples with

the earliest-time index (i.e. most distance past) go into filter first.

(2) Shift the folded input signal x(-k) to obtain x(m-k).

(3) Multiply x(m-k) by the impulse response of the filter h(k).

(4) Sum the results of the vector product h(k)x(m-k) to obtain the filter

output y(m).

𝑧

Ex: Using only the fact that 𝛾𝑘𝑢 𝑘 ↔ and properties of the

𝑧−𝛾

z−transform, find the z−transform of:

transform of the signal x[n] as shown in Fig. Q2.

Hint: x[n] = n{u[n] – u[n-6]}.

Ex03: Show a canonical realization of the following transfer functions:

whose transfer function is given by:

shown in Fig.Q5(a) and (b).

Ex06: a) Realize a digital filter whose transfer function is give by

has a maximum value of Ω = 0, and it decreases monotonically with

frequency until Ω = π. The 3-dB bandwidth is the frequency where

the amplitude response drops to 0.707. Determine the 3-dB

bandwidth of this filter when a = 0.2.

Ex07:Pole-zero configurations of two filters are shown in Fig.

Q7(a) and (b). Sketch roughly the amplitude and the phase

responses of these filters

completely, and to have a sharp recovery on either sides of 5000 Hz

to a gain of unity. Assume that the sampling frequency is 40 kHz.

Ex09: a) Show that the amplitude response of a system with a pole

at z = r and a zero at z = 1/r (r is less than or equal to 1) is constant

with frequency (this is called an “allpass” filter).

1 ±𝑗𝜃

with two poles at 𝑧 = 𝑟𝑒 ±𝑗𝜃 and two zeros at 𝑧 = 𝑒 (r 1) is

𝑟

also an allpass filter.

Example

Derive the amplitude response of a discrete-time system having a

sampling frequency of 1000 Hz, and with zeros at (z=1) and (z=-1),

and poles at

- Designing a Low- Pass Fir Digital Filter By Using Bartlett Hanning and Blackman Harris Window TechniqueUploaded byijsret
- Labview - Digital Filter Design Toolkit Reference ManualUploaded byJhon Jairo Anaya
- Attendance System by Biometric Authorization by SpeechUploaded byshu3h0jee7
- Chapter 2_ FIR filters - Digital Filter Design - mikroElektronika.pdfUploaded byAnimesh Nagrare
- 1391969106Uploaded bySriram Kumar
- 8 m.tech Ece Communication 09 11 VerifiUploaded byAnil Marturi
- Origin TutorialsUploaded byEric Doctore Krage
- Designing of IIR Filter using Radix-4 Multiplier by Precharging TechniqueUploaded byEditor IJTSRD
- Altera Implementing FIR Filters and FFTs With 28-Nm Variable-Precision DSP ArchitectureUploaded bykn65238859
- DSP BasicsUploaded byaditi_nangia_1
- IIR FiltersUploaded byHari Unnikrishnan
- dsfUploaded byMuhammad Bilal Junaid
- 25.108_Digital Filter Design Using FDAToolUploaded byJibran Siddiqui
- Morales L.G. (Ed.) Adaptive Filtering ApplicationsUploaded byArash Torkaman
- Ch6(2)Uploaded byanon_682302459
- impz function in matlab helpUploaded byhasan1100
- ECE341 Fall 2016 SyllabusUploaded byMuhammad Abdul Jabbar
- fwhomfbrhi27435FWHOMFBRHIUploaded byAnu Anusha
- IJETR032560Uploaded byerpublication
- PapersUploaded byanandh30
- be309f04r4Uploaded by29377
- DSP U1 LNUploaded byKarthick Sekar
- Lab 2 Cancel NoiseUploaded bymalikireddy36
- fwldsp.pdfUploaded byAlex
- 34741Uploaded byDhanarasiSyam
- Qp Cs2403Uploaded byMaalavica Ramakrishnan
- 4414ijbb01Uploaded byijbbjournal
- 5F56D51Bd01Uploaded byguyoaser
- Lec-2-Systems v4.0Uploaded byNikesh Bajaj
- Filter TypesUploaded byBency Abraham

- Heat Load Calculation (Cold Storage Plant)Uploaded byKenny Graiven
- sdg7Uploaded bysiamae
- assin 2Uploaded bysiamae
- assin 2.docxUploaded bysiamae
- humies 1Uploaded bysiamae
- Sample 2Uploaded bythareenda
- cvcreativeUploaded byIvona Băbărelu
- Assignment No 2Uploaded bysiamae
- The transportation is movement of objects.docxUploaded bysiamae
- run_control.txtUploaded bysiamae
- Us 4616179Uploaded bysiamae
- Pfister2008a ICEM Torque Measurement Methods for Very High Speed Synchronous Motors DownloadéUploaded bysiamae
- Emtdc Manual(1)Uploaded byLe Quang Vinh
- Draft Sorp (Ceb) - 27 May 2016Uploaded bysiamae
- Pscad SimulationUploaded bysiamae
- OBSERVATIO1Uploaded bysiamae
- 2Uploaded bysiamae
- 1 Phase Inverters and RectifiersUploaded bysiamae
- Pre-feasibility Study for Construction of Mini Hydro Power PlantUploaded byMiguel Prieto
- EnergyUploaded bysiamae
- Torque Measurement (Autosaved)Uploaded bysiamae
- Hybrid 1Uploaded bysiamae
- New Document(26) 25-OcUploaded bysiamae
- Drawings Ccps KelanitisseUploaded bysiamae
- OperationalUploaded bysiamae
- InsulationUploaded bysiamae
- AssignmUploaded bysiamae
- 77_43525_EA121_2013_3__2_Uploaded bysiamae
- 02 Boq Ground Floor Super StructureUploaded bykavish malaka
- Motor Sizing CalculationUploaded bymithun46

- Final Zachman and TOGAF Enterprise FrameworkUploaded byMati Chala
- AER124_QPUploaded by3raja
- Racor_DieselFuelFilterUploaded byigize2
- osha hospital standards aUploaded byapi-389205029
- Filter Design 2Uploaded bywww.bhawesh.com.np
- eureka math grade 1 module 1 tips for parentsUploaded byapi-270945269
- 3M Electronic Grade CoatingUploaded byganasaii3882
- Chap6_Looping in MATLABUploaded byhazellllll
- CPIIndico100RadInstallationServiceManualUploaded byEragon_88
- edu555 cni week 7 iUploaded byapi-304225088
- Recovering the Rife MicroscopeUploaded byjkorolas372
- Hydraulic (HEC RAS)Uploaded byEldren Jamee
- TutorialSet3_Sol.pdfUploaded bywill
- NetApp OnCommand 5.0 Sizing GuideUploaded byjensterd2040
- ENG5 Budget of Work 1st.docxUploaded byMyprimadona Ganotisi
- Petzetakis - PVC ManualUploaded byJohn Gordon-Smith
- Rothoblaas.bsi.Technical Data Sheets.enUploaded byAndrei Gheorghica
- Lab 3 EnerconUploaded byALlan ABiang
- 2016-06-13 Rheinmetall at Eurosatory MBT Advanced Technology Demonstrator EnUploaded bypzkw2000
- Michael Buble-The Way You Look TonightUploaded byRichelle Sales
- hms-sci-1204-089Uploaded byshijub_001
- Probook 650 G1.pdfUploaded byGabriel-Ştefan Necoară
- FlangesUploaded byGledis Kallogjeri
- fluorogold.pdfUploaded bynileshmadankar
- unit lesson plan-groupingUploaded byapi-252532158
- ivulitUploaded byscr789
- PrefaceUploaded byaegosmith
- Bihar Development Report 2010Uploaded byIndicus Analytics
- Personal and Professional DevelopmentUploaded bysunnybakliwal
- SEM-9521E Vibrating ScreenUploaded byRexx Mexx