David Weenink. First semester 2007

Similar documents
Damped Oscillators (revisited)

Responses of Digital Filters Chapter Intended Learning Outcomes:

Lecture 19 IIR Filters

Recursive, Infinite Impulse Response (IIR) Digital Filters:

Lecture 3 Matlab Simulink Minimum Phase, Maximum Phase and Linear Phase Systems

2. Typical Discrete-Time Systems All-Pass Systems (5.5) 2.2. Minimum-Phase Systems (5.6) 2.3. Generalized Linear-Phase Systems (5.

w n = c k v n k (1.226) w n = c k v n k + d k w n k (1.227) Clearly non-recursive filters are a special case of recursive filters where M=0.

E : Lecture 1 Introduction

Poles and Zeros in z-plane

EE482: Digital Signal Processing Applications

Chapter 3 Data Acquisition and Manipulation

SPEECH ANALYSIS AND SYNTHESIS

Like bilateral Laplace transforms, ROC must be used to determine a unique inverse z-transform.

Chapter 9. Linear Predictive Analysis of Speech Signals 语音信号的线性预测分析

Digital Signal Processing

Let H(z) = P(z)/Q(z) be the system function of a rational form. Let us represent both P(z) and Q(z) as polynomials of z (not z -1 )

Lecture 7 - IIR Filters

Discrete Time Systems

EECE 301 Signals & Systems Prof. Mark Fowler

# FIR. [ ] = b k. # [ ]x[ n " k] [ ] = h k. x[ n] = Ae j" e j# ˆ n Complex exponential input. [ ]Ae j" e j ˆ. ˆ )Ae j# e j ˆ. y n. y n.

Time Series Analysis: 4. Linear filters. P. F. Góra

Your solutions for time-domain waveforms should all be expressed as real-valued functions.

Signals and Systems. Problem Set: The z-transform and DT Fourier Transform

Introduction to DSP Time Domain Representation of Signals and Systems

2.161 Signal Processing: Continuous and Discrete Fall 2008

Lecture 04: Discrete Frequency Domain Analysis (z-transform)

Use: Analysis of systems, simple convolution, shorthand for e jw, stability. Motivation easier to write. Or X(z) = Z {x(n)}

Enhanced Steiglitz-McBride Procedure for. Minimax IIR Digital Filters

INFINITE-IMPULSE RESPONSE DIGITAL FILTERS Classical analog filters and their conversion to digital filters 4. THE BUTTERWORTH ANALOG FILTER

Discrete-time signals and systems

The Impulse Signal. A signal I(k), which is defined as zero everywhere except at a single point, where its value is equal to 1

21.4. Engineering Applications of z-transforms. Introduction. Prerequisites. Learning Outcomes

Transform Analysis of Linear Time-Invariant Systems

CMPT 889: Lecture 5 Filters

Time Series Analysis: 4. Digital Linear Filters. P. F. Góra

Z Transform (Part - II)

Roll No. :... Invigilator s Signature :.. CS/B.Tech (EE-N)/SEM-6/EC-611/ DIGITAL SIGNAL PROCESSING. Time Allotted : 3 Hours Full Marks : 70

Digital Filters. Linearity and Time Invariance. Linear Time-Invariant (LTI) Filters: CMPT 889: Lecture 5 Filters

Analog LTI system Digital LTI system

UNIVERSITY OF OSLO. Please make sure that your copy of the problem set is complete before you attempt to answer anything.

2.161 Signal Processing: Continuous and Discrete Fall 2008

Discrete Time Systems

LECTURE NOTES DIGITAL SIGNAL PROCESSING III B.TECH II SEMESTER (JNTUK R 13)

( ) John A. Quinn Lecture. ESE 531: Digital Signal Processing. Lecture Outline. Frequency Response of LTI System. Example: Zero on Real Axis

Discrete-Time Signals and Systems. The z-transform and Its Application. The Direct z-transform. Region of Convergence. Reference: Sections

Analysis of Finite Wordlength Effects

Lecture 2 OKAN UNIVERSITY FACULTY OF ENGINEERING AND ARCHITECTURE

CHAPTER 2 RANDOM PROCESSES IN DISCRETE TIME

Stability Condition in Terms of the Pole Locations

PS403 - Digital Signal processing

(Refer Slide Time: )

Lecture 3 - Design of Digital Filters

Digital Signal Processing

DIGITAL SIGNAL PROCESSING UNIT III INFINITE IMPULSE RESPONSE DIGITAL FILTERS. 3.6 Design of Digital Filter using Digital to Digital

Transform analysis of LTI systems Oppenheim and Schafer, Second edition pp For LTI systems we can write

Problem Set 2: Solution Due on Wed. 25th Sept. Fall 2013

UNIT - III PART A. 2. Mention any two techniques for digitizing the transfer function of an analog filter?

Z-Transform. x (n) Sampler

EE482: Digital Signal Processing Applications

Lecture 18: Stability

Digital Signal Processing:

An Autoregressive Recurrent Mixture Density Network for Parametric Speech Synthesis

Research Article Design of One-Dimensional Linear Phase Digital IIR Filters Using Orthogonal Polynomials

Nonparametric and Parametric Defined This text distinguishes between systems and the sequences (processes) that result when a WN input is applied

Discrete-Time David Johns and Ken Martin University of Toronto

Experiment 13 Poles and zeros in the z plane: IIR systems

Lecture 13: Pole/Zero Diagrams and All Pass Systems

Digital Filters Ying Sun

Review of Discrete-Time System

Today. ESE 531: Digital Signal Processing. IIR Filter Design. Impulse Invariance. Impulse Invariance. Impulse Invariance. ω < π.

Need for transformation?

Grades will be determined by the correctness of your answers (explanations are not required).

Detailed Solutions to Exercises

VALLIAMMAI ENGINEERING COLLEGE. SRM Nagar, Kattankulathur DEPARTMENT OF INFORMATION TECHNOLOGY. Academic Year

Professor Fearing EECS150/Problem Set Solution Fall 2013 Due at 10 am, Thu. Oct. 3 (homework box under stairs)

Chapter 7: Filter Design 7.1 Practical Filter Terminology

Formant Analysis using LPC

Finite Word Length Effects and Quantisation Noise. Professors A G Constantinides & L R Arnaut

Analysis Of Ill-Conditioning Of Multi-Channel Deconvolution Problems

ELEG 305: Digital Signal Processing


Lesson 1. Optimal signalbehandling LTH. September Statistical Digital Signal Processing and Modeling, Hayes, M:

EE 521: Instrumentation and Measurements

Massachusetts Institute of Technology Department of Electrical Engineering and Computer Science. Fall Solutions for Problem Set 2

UNIT 4: DIGITAL SYSTEM MODELS

DIGITAL SIGNAL PROCESSING. Chapter 6 IIR Filter Design

ECE503: Digital Signal Processing Lecture 5

Linear Prediction 1 / 41

/ (2π) X(e jω ) dω. 4. An 8 point sequence is given by x(n) = {2,2,2,2,1,1,1,1}. Compute 8 point DFT of x(n) by

z-transforms Definition of the z-transform Chapter

Introduction to Digital Signal Processing

EEL3135: Homework #4

LINEAR-PHASE FIR FILTERS DESIGN

Question Bank. UNIT 1 Part-A

How to manipulate Frequencies in Discrete-time Domain? Two Main Approaches

VISA: Versatile Impulse Structure Approximation for Time-Domain Linear Macromodeling

2.161 Signal Processing: Continuous and Discrete Fall 2008

An Fir-Filter Example: Hanning Filter

Solutions: Homework Set # 5

DSP. Chapter-3 : Filter Design. Marc Moonen. Dept. E.E./ESAT-STADIUS, KU Leuven

Transcription:

Institute of Phonetic Sciences University of Amsterdam First semester 2007

Digital s What is a digital filter? An algorithm that calculates with sample values Formant /machine H 1 (z) that: Given input value x n Calculates output value y n (no delay) One value per sampling period T (the clock)

Digital s What is a digital filter? An algorithm that calculates with sample values Formant /machine H 1 (z) that: Given input value x n Calculates output value y n (no delay) One value per sampling period T (the clock)

Digital s What is a digital filter? An algorithm that calculates with sample values Formant /machine H 1 (z) that: Given input value x n Calculates output value y n (no delay) One value per sampling period T (the clock)

Digital s What is a digital filter? An algorithm that calculates with sample values Formant /machine H 1 (z) that: Given input value x n Calculates output value y n (no delay) One value per sampling period T (the clock)

Usage? Formant Pre-emphasis od speech signals Formant synthesis ing in the time domain...

Simple Integrator : y n = 0.5x n + 0.5x n 1 (n = 1, 2,... ) x 1 =1.00 Formant y 1 =0.50 x 1 y 1 Calculations y 1 = 0.5x 1 + 0.5x 0 = 0.5 1 + 0.5 0 = 0.5 y 2 = 0.5x 2 + 0.5x 1 = 0.5 0 + 0.5 1 = 0.5 y 3 = 0.5x 3 + 0.5x 2 = 0.5 0 + 0.5 0 = 0 y 4 = 0.5x 4 + 0.5x 3 = 0.5 0 + 0.5 0 = 0

Simple Integrator : y n = 0.5x n + 0.5x n 1 (n = 1, 2,... ) x 2 =0 Formant y 2 =0.50 x 2 y 2 Calculations y 1 = 0.5x 1 + 0.5x 0 = 0.5 1 + 0.5 0 = 0.5 y 2 = 0.5x 2 + 0.5x 1 = 0.5 0 + 0.5 1 = 0.5 y 3 = 0.5x 3 + 0.5x 2 = 0.5 0 + 0.5 0 = 0 y 4 = 0.5x 4 + 0.5x 3 = 0.5 0 + 0.5 0 = 0

Simple Integrator : y n = 0.5x n + 0.5x n 1 (n = 1, 2,... ) x 3 =0 Formant y 3 =0 x 3 y 3 Calculations y 1 = 0.5x 1 + 0.5x 0 = 0.5 1 + 0.5 0 = 0.5 y 2 = 0.5x 2 + 0.5x 1 = 0.5 0 + 0.5 1 = 0.5 y 3 = 0.5x 3 + 0.5x 2 = 0.5 0 + 0.5 0 = 0 y 4 = 0.5x 4 + 0.5x 3 = 0.5 0 + 0.5 0 = 0

Simple Integrator : y n = 0.5x n + 0.5x n 1 (n = 1, 2,... ) x 4 =0 Formant y 4 =0 x 4 y 4 Calculations y 1 = 0.5x 1 + 0.5x 0 = 0.5 1 + 0.5 0 = 0.5 y 2 = 0.5x 2 + 0.5x 1 = 0.5 0 + 0.5 1 = 0.5 y 3 = 0.5x 3 + 0.5x 2 = 0.5 0 + 0.5 0 = 0 y 4 = 0.5x 4 + 0.5x 3 = 0.5 0 + 0.5 0 = 0

Simple Integrator : y n = 0.5x n + 0.5x n 1 (n = 1, 2,... ) x 4 =0 y 4 =0 Formant x 4 y 4 Calculations y 1 = 0.5x 1 + 0.5x 0 = 0.5 1 + 0.5 0 = 0.5 y 2 = 0.5x 2 + 0.5x 1 = 0.5 0 + 0.5 1 = 0.5 y 3 = 0.5x 3 + 0.5x 2 = 0.5 0 + 0.5 0 = 0 y 4 = 0.5x 4 + 0.5x 3 = 0.5 0 + 0.5 0 = 0 If x n = (1, 0, 0, 0,... ) then y n = (0.5, 0.5, 0, 0,... ) (Impulse response)

Simple Differentiator : y n = 0.5x n 0.5x n 1 (n = 1, 2,... ) x 1 =1.00 Formant y 1 =0.50 x 1 y 1 Calculations y 1 = 0.5x 1 0.5x 0 = 0.5 1 0.5 0 = 0.5 y 2 = 0.5x 2 0.5x 1 = 0.5 0 0.5 1 = 0.5 y 3 = 0.5x 3 0.5x 2 = 0.5 0 0.5 0 = 0 y 4 = 0.5x 4 0.5x 3 = 0.5 0 0.5 0 = 0

Simple Differentiator : y n = 0.5x n 0.5x n 1 (n = 1, 2,... ) x 2 =0 Formant y 2 =-0.50 x 2 y 2 Calculations y 1 = 0.5x 1 0.5x 0 = 0.5 1 0.5 0 = 0.5 y 2 = 0.5x 2 0.5x 1 = 0.5 0 0.5 1 = 0.5 y 3 = 0.5x 3 0.5x 2 = 0.5 0 0.5 0 = 0 y 4 = 0.5x 4 0.5x 3 = 0.5 0 0.5 0 = 0

Simple Differentiator : y n = 0.5x n 0.5x n 1 (n = 1, 2,... ) x 3 =0 Formant y 3 =0 x 3 y 3 Calculations y 1 = 0.5x 1 0.5x 0 = 0.5 1 0.5 0 = 0.5 y 2 = 0.5x 2 0.5x 1 = 0.5 0 0.5 1 = 0.5 y 3 = 0.5x 3 0.5x 2 = 0.5 0 0.5 0 = 0 y 4 = 0.5x 4 0.5x 3 = 0.5 0 0.5 0 = 0

Simple Differentiator : y n = 0.5x n 0.5x n 1 (n = 1, 2,... ) x 4 =0 Formant y 4 =0 x 4 y 4 Calculations y 1 = 0.5x 1 0.5x 0 = 0.5 1 0.5 0 = 0.5 y 2 = 0.5x 2 0.5x 1 = 0.5 0 0.5 1 = 0.5 y 3 = 0.5x 3 0.5x 2 = 0.5 0 0.5 0 = 0 y 4 = 0.5x 4 0.5x 3 = 0.5 0 0.5 0 = 0

Simple Differentiator : y n = 0.5x n 0.5x n 1 (n = 1, 2,... ) x 4 =0 y 4 =0 Formant x 4 y 4 Calculations y 1 = 0.5x 1 0.5x 0 = 0.5 1 0.5 0 = 0.5 y 2 = 0.5x 2 0.5x 1 = 0.5 0 0.5 1 = 0.5 y 3 = 0.5x 3 0.5x 2 = 0.5 0 0.5 0 = 0 y 4 = 0.5x 4 0.5x 3 = 0.5 0 0.5 0 = 0 If x n = (1, 0, 0, 0,... ) then y n = (0.5, 0.5, 0, 0,... ) (Impulse response)

Pre-Emphasis 1 : y n = x n ax n 1, where a [0, 1] What is the spectrum? take Z-transform on the left and the right sample sequences: k y kz k = k {x k ax k 1 }z k Formant Y (z) = {x k z k a k x k 1z k = {x k z k a k x k 1z k+1 1 = {x k z k az 1 k x k 1z (k 1) = X (z) az 1 X (z) The simplification: Y (z) = (1 az 1 )X (z) We define H(z) = Y (z)/x (z) as the filter s spectrum

Pre-Emphasis 2 : y n = x n ax n 1, where a [0, 1] Spectrum: H(z) = 1 az 1, this equals the sum of the spectrum of δ(n), H(z) = 1, and the spectrum of aδ(n 1), H(z) = az 1! 1em] H(z) shows the filter s amplitude response. We now write z = e +2πifT (f values are equidistant ) H(f ) = H(f )H (f ) = (1 ae 2πifT )(1 ae +2πifT ) = 1 + a 2 a(e +2πifT + e 2πifT ) = 1 + a 2 2a cos 2πfT Formant

Pre-Emphasis Response 20 log H(f ) = 20 log 1 + a 2 2a cos 2πfT 6 = 10 log(1 + a 2 2a cos 2πfT ) Formant Amplitude (db) 0 25 0 0.5 F s Frequency (Hz) a = 0.50 (dotted) a = 0.80 (dashed) a = 0.95 (plain) Extremes: 20 log H(F s /2)/H(0) = 20 log 1+a 1 a

Digital ing in the Time-Domain Most general linear digital filter formula: y n = M k=0 a kx n k + N j=1 b jy n j Output is linear combinantion of M + 1 inputs x k and N outputs y j. Formant Special Cases Finite Impulse response (FIR) filter, all b j = 0. y n = M k=0 a kx n k Non-recursive, output stops exactly M samples after the last input. Also called: Moving Average (MA) filter. Infinite Impulse Response (IIR) filter, some b j 0. Impulse response may be infinite. Auto Regressive (AR) filter if M = 0 ARMA if N 0 and M 0

Digital ing in the Time-Domain Most general linear digital filter formula: y n = M k=0 a kx n k + N j=1 b jy n j Output is linear combinantion of M + 1 inputs x k and N outputs y j. Formant Special Cases Finite Impulse response (FIR) filter, all b j = 0. y n = M k=0 a kx n k Non-recursive, output stops exactly M samples after the last input. Also called: Moving Average (MA) filter. Infinite Impulse Response (IIR) filter, some b j 0. Impulse response may be infinite. Auto Regressive (AR) filter if M = 0 ARMA if N 0 and M 0

Digital ing in the Time-Domain Most general linear digital filter formula: y n = M k=0 a kx n k + N j=1 b jy n j Output is linear combinantion of M + 1 inputs x k and N outputs y j. Formant Special Cases Finite Impulse response (FIR) filter, all b j = 0. y n = M k=0 a kx n k Non-recursive, output stops exactly M samples after the last input. Also called: Moving Average (MA) filter. Infinite Impulse Response (IIR) filter, some b j 0. Impulse response may be infinite. Auto Regressive (AR) filter if M = 0 ARMA if N 0 and M 0

Digital Layout Example Layout of the filter: y n = P 3 k=0 a kx n k + P 3 j=1 b jy n j = a 0x n + a 1x n 1 + a 2x n 2 + a 3x n 3 + b 1y n 1 + b 2y n 2 + b 3y n 3 Formant

Digital Frequency Response y n = M k=0 a kx n k + N j=1 b jy n j has : Formant Y (z) = P M k=0 a kz k 1 P N j=1 b j z j Numerator and denumerator are polynomials in z. A polynomial may become zero for certain combination of its coefficients. Numerator: no problem, i.e. H(z 0 ) = 0 Denumerator: unstable filter.

IIR Example: Formant Formant (1) y n q p = x n py n 1 qy n 2 = e 2πBT = 2 q cos 2πFT

The Formant y n = x n py n 1 qy n 2 A second order recursive filter with response: H(f ) = 1 1 + pz 1 + qz 2 = z 2 z 2 + pz + q Denominator is second degree polynomial in z. Zeros: z 1,2 = p 2 ± p 2 4 q H(f ) = 1 (z z 1 )(z z 2 ) Interesting (resonance) when zeros lie within the unit circle. Then p2 4 q < 0 and z 1,2 = p 2 ± i q p2 4 We see that z 1 = z2 Formant

The Formant Frequency Response F s /2 z 1 z 2 Formant 0 0.5 F s Given p and q, z 1 and z 2 are fixed. 1 To get the frequency response H(f ) = z z 1 (z z 2 we let z trace the upper part of the unit circle. When we start at z = 0 and end in z = 1 the response follows the curve on the right. When z is close to z 1, z z 1 is very small and 1 the response z z 1 very large. We have resonance. z = 0 means frequency f = 0. z = 1 means f = F s /2

Relations Between p, q and F, B p and q and F Solve re iφ = z 1 = p 2 + i p 2 4 q cos φ = p 2 q Because φ = 2πFT we get F = 1 p 2πT arccos 2 q Formant F, B and p, q F & B (p,q): F = 1 2πT arccos p 2 q B = 1 πt ln q p & q (F, B): q = e 2πBT p = 2 q cos 2πFT

More Formants: Serial Formant For the total filter response: H(z) = Y (z) X (z) = H 2(z)Y 1 (z) X (z) = H 2 (z)h 1 (z) If H 1 (z) and H 2 (z) only have poles than H(z) has only poles.

More Formants: Parallel Formant For the total filter response: H(z) = Y (z) X (z) = Y 1(z)+Y 2 (z) X (z = H 1 (z) + H 2 (z) If H 1 (z) and H 2 (z) only have poles than H(z) has poles and also may have zeros. These zeros depend on H 1 (z) and H 2 (z) and are fixed. This is not desireable.