# Auto-Tuning Using Fourier Coefficients

Size: px
Start display at page:

Transcription

1 Auto-Tuning Using Fourier Coefficients Math 56 Tom Whalen May 20, 2013 The Fourier transform is an integral part of signal processing of any kind. To be able to analyze an input signal as a superposition of infinitely many signals of different frequency allows for manipulation of the coefficients and, after completing the inverse Fourier transform, interesting effects on the initial signal. Thus, in my project I focused on creating pitch correction effects as seen in software Auto-Tune, and in this paper I will talk about the process of creating a Matlab code that performs this effect. The Fast Fourier Transform algorithm is effective at breaking down mathematical functions into a Fourier series of different frequency complex exponentials and the corresponding coefficients. However, when analyzing audio signals it is impractical and inefficient to take one Fouier Transform of an entire audio clip; for a band-limited signal, most of the 44100t Fourier coefficients will contain no imformation about the original signal and there is no way to tell which frequencies are dominant at different times in the signal. Thus, the way that audio signals are most often processed is using something called the Short-Time Fourier Transform (STFT). The basic idea of the STFT is that one can window the actual signal multiple times and take the FFT of the resulting modified input. The windowing process is done simply by multiplying the signal by another function that is mostly zero and has a maximum of 1. This will kill off the signal at other times that are not of interest and allow for getting the frequencies at a certain point in time via the FFT. There are several types of windowing functions that can be used in the STFT process. The most commonly used set of windows are of the form ( ) 2πn α (1 α) cos, 0 n < N N 1 where α is a constant that determines the shape of the function and N is the length of the windowed signal. The windowing function that I used most in my analysis was the Hanning window, which is the above equation with α =.5. Some other possible widowing functions are the Hamming window (α( =.54) ) and the square function (α = 1). One can even form functions like cos 3 πn N 1 where 0 n < N that are very similar in shape to the Hanning window but have 1

2 πn Figure 1: The Hanning window: 2 (1 cos( N 1 )), 0 n < N a more defined peak near N/2. Windows like the Hanning are much prefered to the square window because the jumps in the square window cause the Fourier coefficients to decay much more gradually than those obtained via one of the smooth windows. However, when using the cosine-based window, it is necessary to have a jump size between windows that enables full reconstruction of the original signal. This reconstruction condition is f j w(j t 0 ) + f j w(j t 1 ) +... = 1 which means that each spectral frame will be weighted evenly. For the square, the obvious choice for the jump size is N, the number of samples in the windowed signal, which will prevent any component of the signal from being processed more than once. For the other windows, the jumpsize is N/4, but this will cause the signal to be interpreted as having twice the actual value, hence a scaling factor of 1/2 to recover the signal. Now this brings us to the actual form of the STFT: ˆf m,t = w(j t)f j ω mj j=0 In practice the result of the running the ST-FFT algorithm is a matrix with each row representing the Fourier coefficients of the signal at some point in time. The final step in the STFT process is choosing a suitable value for N, the number of samples of the signal included in each window. For a typical sampling rate of 44.1 khz, the most common N is 1024, which is large enough to ensure that the Fourier coefficients will decay and there will be no aliasing effects. It is also a small enough that the time resolution of the signal is decent, as there will be roughly 43 spectral frames per second of signal. Here is a generic audio signal and its STFT displayed as a colorscale image with time on the vertical axis and the frequency on the horizontal. To invert the STFT matrix, one must loop through all of the spectral frames and essentially add up all of the resuting signals that are produced by each set 2

3 Signal Time x 10 4 Figure 2: The input signal Figure 3: Its STFT on a color-scale map 3

4 of coefficients, which is commonly referred to as the overlap-add method of reconstruction. Shifting by the jumpsize each iteration, multiplying the ISTFT output by the window, and then adding it to the total output signal will perfectly reconstruct a signal from its set of STFT coefficients. The problem with the STFT is that, for any given spectral frame, it doesn t tell us the instantaneous frequencies due to the fact that there isn t much energy resolution. The solution to this problem relies in the fact that a complex coefficient a + ib can be written in the polar form βe φ where β = a 2 + b 2 and φ = tan 1 b a. The φ term acts as a phase of the signal, due to the shift of a particular frequency in the windowing process. Looking at the term in the Fourier series for which the coefficient is a + ib, we can now write this as βe iφ e im = βe i(φ+m). The new interpretation of this is that the new amplitude of the wave is the magnitude of the original coefficient and the new frequency is actually the old frequency shifted by the phase φ. Thus, by converting to polar coordinates we can keep track of actual values of the frequencies in an audio signal rather than those determined by the length of the DFT matrix. The implementation of the phase vocoder is illustrated by finding the change in phase for a specified energy bin between successive spectral frames, as shown in the following equations: ω(k, t) = φ(k, t) φ(k, t) t ω(k, t) ω wrapped (k, t) = [( ω(k, t) + π)%2π] π ω new (k, t) = ω wrapped (k, t) ω(k, t) Because the phases are wrapped from -π to π, we can t simply calculate the new frequency by finding the phase difference and adding it to the frequency denoted by k. We have to instead find the phase difference, subtract the expected value, make it positive and mod it with 2π and finally add it with the frequency value of the bin to get the true frequency of the signal. Now that we know exactly which frequencies we are dealing with, it is possible to directly manipulate the coefficients. Pitch correction is carried out by first finding out the frequency values for the musical notes and making them target bins for the signal coefficients. Once, this has been done the next step is to find the dominant overtone of the input signal. Since pitch correction is employed mainly for human singing, there will normally only be one tone that dominates the spectrum of Fourier coefficients. It is also necessary to convert the frequencies, which have now been processed by the phase vocoder, into Hertz in order to correctly coorelate the target bins, which is given by the equation: ω Hz (k) = k(sampling rate) where k is the bin number and ranges from 1 to the length of the window length of window Once this has been done, the next step is to loop through all of the spectral frames, identifying the dominant tone in each one, and pushing each into the target bin of the nearest musical note. However, because Matlab s implementation 4

5 of the FFT doesn t allow for changing the actual frequencies that correspond to the coefficients, I had to effectively change the frequencies by altering the values of the amplitudes. Since we want to rebin the dominant tone of the signal, we want to go from frequency m to frequency m + φ, where φ is some phase factor that moves this frequency to the desired one. Now writing out the series term as: fˆ m e i(m+φ) = f ˆ m e iφ e im This tells us that we should multiply the Fourier coefficients by a factor e iφ that depends on the distance from the nearest desired frequency. This will shift all the frequencies to output the pitch corrected signal that should, in theory, produce a much more euphonic sound than the original clip. There are a few restrictions, however. For a real-valued input like an audio clip, we have the relation that f m = fm, which means that the coefficient with the maximum amplitude will appear twice. Thus, we need only analyze the coefficients f 0 to f N/2, which will give us the correct difference between the actual frequency corresponding to the coefficient and the goal musical note. Furthermore, there isn t enough energy resolution to separate the lower frequencies present in the signal. With a window size of 1024 and a sampling rate of approximately 43 Hz, the information of the entire lowest musical octave, with values below 43 Hz, is all contained in the first Fourier coefficient, so it is impossible to resolve out which notes are dominant in the signal. Thus, it is best just to avoid this difficulty and begin analyzing the dominant tones starting with the second coefficient. This all sounds easy enough in theory, but the actual implementation was a bit more difficult. After a long amount of time put into making the code work, I wasn t able to get a functional auto-tune script up and running. Though the computational aspect of this project was not completely fulfilled, the theory behind pitch correction programs is very interesting creates a basis for other neat effects to implemented on audio signals. References [1] Boulanger, Richard; Lazzarini, Victor. The Audio Programming Book. Massachusetts Institute of Technology,

### FOURIER TRANSFORM BASED SIMPLE CHORD ANALYSIS. UIUC Physics 193 POM

FOURIER TRANSFORM BASED SIMPLE CHORD ANALYSIS Fanbo Xiang UIUC Physics 193 POM Professor Steven M. Errede Fall 2014 1 Introduction Chords, an essential part of music, have long been analyzed. Different

### B3. Short Time Fourier Transform (STFT)

B3. Short Time Fourier Transform (STFT) Objectives: Understand the concept of a time varying frequency spectrum and the spectrogram Understand the effect of different windows on the spectrogram; Understand

### Analysis/resynthesis with the short time Fourier transform

Analysis/resynthesis with the short time Fourier transform summer 2006 lecture on analysis, modeling and transformation of audio signals Axel Röbel Institute of communication science TU-Berlin IRCAM Analysis/Synthesis

### Analog and Digital Signals, Time and Frequency Representation of Signals

1 Analog and Digital Signals, Time and Frequency Representation of Signals Required reading: Garcia 3.1, 3.2 CSE 3213, Fall 2010 Instructor: N. Vlajic 2 Data vs. Signal Analog vs. Digital Analog Signals

### Lab 1. The Fourier Transform

Lab 1. The Fourier Transform Introduction In the Communication Labs you will be given the opportunity to apply the theory learned in Communication Systems. Since this is your first time to work in the

### The Calculation of G rms

The Calculation of G rms QualMark Corp. Neill Doertenbach The metric of G rms is typically used to specify and compare the energy in repetitive shock vibration systems. However, the method of arriving

### Trigonometric functions and sound

Trigonometric functions and sound The sounds we hear are caused by vibrations that send pressure waves through the air. Our ears respond to these pressure waves and signal the brain about their amplitude

### Lecture - 4 Diode Rectifier Circuits

Basic Electronics (Module 1 Semiconductor Diodes) Dr. Chitralekha Mahanta Department of Electronics and Communication Engineering Indian Institute of Technology, Guwahati Lecture - 4 Diode Rectifier Circuits

### The Fourier Analysis Tool in Microsoft Excel

The Fourier Analysis Tool in Microsoft Excel Douglas A. Kerr Issue March 4, 2009 ABSTRACT AD ITRODUCTIO The spreadsheet application Microsoft Excel includes a tool that will calculate the discrete Fourier

### Due Wednesday, December 2. You can submit your answers to the analytical questions via either

ELEN 4810 Homework 6 Due Wednesday, December 2. You can submit your answers to the analytical questions via either - Hardcopy submission at the beginning of class on Wednesday, December 2, or - Electronic

### SGN-1158 Introduction to Signal Processing Test. Solutions

SGN-1158 Introduction to Signal Processing Test. Solutions 1. Convolve the function ( ) with itself and show that the Fourier transform of the result is the square of the Fourier transform of ( ). (Hints:

### S. Boyd EE102. Lecture 1 Signals. notation and meaning. common signals. size of a signal. qualitative properties of signals.

S. Boyd EE102 Lecture 1 Signals notation and meaning common signals size of a signal qualitative properties of signals impulsive signals 1 1 Signals a signal is a function of time, e.g., f is the force

### THE SPECTRAL MODELING TOOLBOX: A SOUND ANALYSIS/SYNTHESIS SYSTEM. A Thesis. Submitted to the Faculty

THE SPECTRAL MODELING TOOLBOX: A SOUND ANALYSIS/SYNTHESIS SYSTEM A Thesis Submitted to the Faculty in partial fulfillment of the requirements for the degree of Master of Arts in ELECTRO-ACOUSTIC MUSIC

### SR2000 FREQUENCY MONITOR

SR2000 FREQUENCY MONITOR THE FFT SEARCH FUNCTION IN DETAILS FFT Search is a signal search using FFT (Fast Fourier Transform) technology. The FFT search function first appeared with the SR2000 Frequency

### 7. Beats. sin( + λ) + sin( λ) = 2 cos(λ) sin( )

34 7. Beats 7.1. What beats are. Musicians tune their instruments using beats. Beats occur when two very nearby pitches are sounded simultaneously. We ll make a mathematical study of this effect, using

### Electronic Communications Committee (ECC) within the European Conference of Postal and Telecommunications Administrations (CEPT)

Page 1 Electronic Communications Committee (ECC) within the European Conference of Postal and Telecommunications Administrations (CEPT) ECC RECOMMENDATION (06)01 Bandwidth measurements using FFT techniques

### L9: Cepstral analysis

L9: Cepstral analysis The cepstrum Homomorphic filtering The cepstrum and voicing/pitch detection Linear prediction cepstral coefficients Mel frequency cepstral coefficients This lecture is based on [Taylor,

### Doppler. Doppler. Doppler shift. Doppler Frequency. Doppler shift. Doppler shift. Chapter 19

Doppler Doppler Chapter 19 A moving train with a trumpet player holding the same tone for a very long time travels from your left to your right. The tone changes relative the motion of you (receiver) and

### CHAPTER 6 Frequency Response, Bode Plots, and Resonance

ELECTRICAL CHAPTER 6 Frequency Response, Bode Plots, and Resonance 1. State the fundamental concepts of Fourier analysis. 2. Determine the output of a filter for a given input consisting of sinusoidal

### Electrical Resonance

Electrical Resonance (R-L-C series circuit) APPARATUS 1. R-L-C Circuit board 2. Signal generator 3. Oscilloscope Tektronix TDS1002 with two sets of leads (see Introduction to the Oscilloscope ) INTRODUCTION

### Mathematics. (www.tiwariacademy.com : Focus on free Education) (Chapter 5) (Complex Numbers and Quadratic Equations) (Class XI)

( : Focus on free Education) Miscellaneous Exercise on chapter 5 Question 1: Evaluate: Answer 1: 1 ( : Focus on free Education) Question 2: For any two complex numbers z1 and z2, prove that Re (z1z2) =

### Introduction to Complex Numbers in Physics/Engineering

Introduction to Complex Numbers in Physics/Engineering ference: Mary L. Boas, Mathematical Methods in the Physical Sciences Chapter 2 & 14 George Arfken, Mathematical Methods for Physicists Chapter 6 The

### Sound absorption and acoustic surface impedance

Sound absorption and acoustic surface impedance CHRISTER HEED SD2165 Stockholm October 2008 Marcus Wallenberg Laboratoriet för Ljud- och Vibrationsforskning Sound absorption and acoustic surface impedance

### Design of FIR Filters

Design of FIR Filters Elena Punskaya www-sigproc.eng.cam.ac.uk/~op205 Some material adapted from courses by Prof. Simon Godsill, Dr. Arnaud Doucet, Dr. Malcolm Macleod and Prof. Peter Rayner 68 FIR as

### The continuous and discrete Fourier transforms

FYSA21 Mathematical Tools in Science The continuous and discrete Fourier transforms Lennart Lindegren Lund Observatory (Department of Astronomy, Lund University) 1 The continuous Fourier transform 1.1

### Introduction to Digital Audio

Introduction to Digital Audio Before the development of high-speed, low-cost digital computers and analog-to-digital conversion circuits, all recording and manipulation of sound was done using analog techniques.

### ε: Voltage output of Signal Generator (also called the Source voltage or Applied

Experiment #10: LR & RC Circuits Frequency Response EQUIPMENT NEEDED Science Workshop Interface Power Amplifier (2) Voltage Sensor graph paper (optional) (3) Patch Cords Decade resistor, capacitor, and

### CIRCUITS LABORATORY EXPERIMENT 3. AC Circuit Analysis

CIRCUITS LABORATORY EXPERIMENT 3 AC Circuit Analysis 3.1 Introduction The steady-state behavior of circuits energized by sinusoidal sources is an important area of study for several reasons. First, the

### Speech Signal Processing: An Overview

Speech Signal Processing: An Overview S. R. M. Prasanna Department of Electronics and Electrical Engineering Indian Institute of Technology Guwahati December, 2012 Prasanna (EMST Lab, EEE, IITG) Speech

### Little LFO. Little LFO. User Manual. by Little IO Co.

1 Little LFO User Manual Little LFO by Little IO Co. 2 Contents Overview Oscillator Status Switch Status Light Oscillator Label Volume and Envelope Volume Envelope Attack (ATT) Decay (DEC) Sustain (SUS)

### SIGNAL PROCESSING & SIMULATION NEWSLETTER

1 of 10 1/25/2008 3:38 AM SIGNAL PROCESSING & SIMULATION NEWSLETTER Note: This is not a particularly interesting topic for anyone other than those who ar e involved in simulation. So if you have difficulty

### ANALYZER BASICS WHAT IS AN FFT SPECTRUM ANALYZER? 2-1

WHAT IS AN FFT SPECTRUM ANALYZER? ANALYZER BASICS The SR760 FFT Spectrum Analyzer takes a time varying input signal, like you would see on an oscilloscope trace, and computes its frequency spectrum. Fourier's

### Matlab GUI for WFB spectral analysis

Matlab GUI for WFB spectral analysis Jan Nováček Department of Radio Engineering K13137, CTU FEE Prague Abstract In the case of the sound signals analysis we usually use logarithmic scale on the frequency

1 25. AM radio receiver The chapter describes the programming of a microcontroller to demodulate a signal from a local radio station. To keep the circuit simple the signal from the local amplitude modulated

### AN-007 APPLICATION NOTE MEASURING MAXIMUM SUBWOOFER OUTPUT ACCORDING ANSI/CEA-2010 STANDARD INTRODUCTION CEA-2010 (ANSI) TEST PROCEDURE

AUDIOMATICA AN-007 APPLICATION NOTE MEASURING MAXIMUM SUBWOOFER OUTPUT ACCORDING ANSI/CEA-2010 STANDARD by Daniele Ponteggia - dp@audiomatica.com INTRODUCTION The Consumer Electronics Association (CEA),

### This document is downloaded from DR-NTU, Nanyang Technological University Library, Singapore.

This document is downloaded from DR-NTU, Nanyang Technological University Library, Singapore. Title Transcription of polyphonic signals using fast filter bank( Accepted version ) Author(s) Foo, Say Wei;

### Scientific Programming

1 The wave equation Scientific Programming Wave Equation The wave equation describes how waves propagate: light waves, sound waves, oscillating strings, wave in a pond,... Suppose that the function h(x,t)

### Convolution, Correlation, & Fourier Transforms. James R. Graham 10/25/2005

Convolution, Correlation, & Fourier Transforms James R. Graham 10/25/2005 Introduction A large class of signal processing techniques fall under the category of Fourier transform methods These methods fall

### Wavelet analysis. Wavelet requirements. Example signals. Stationary signal 2 Hz + 10 Hz + 20Hz. Zero mean, oscillatory (wave) Fast decay (let)

Wavelet analysis In the case of Fourier series, the orthonormal basis is generated by integral dilation of a single function e jx Every 2π-periodic square-integrable function is generated by a superposition

### L and C connected together. To be able: To analyse some basic circuits.

circuits: Sinusoidal Voltages and urrents Aims: To appreciate: Similarities between oscillation in circuit and mechanical pendulum. Role of energy loss mechanisms in damping. Why we study sinusoidal signals

### Frequency Response of FIR Filters

Frequency Response of FIR Filters Chapter 6 This chapter continues the study of FIR filters from Chapter 5, but the emphasis is frequency response, which relates to how the filter responds to an input

### High Quality Integrated Data Reconstruction for Medical Applications

High Quality Integrated Data Reconstruction for Medical Applications A.K.M Fazlul Haque Md. Hanif Ali M Adnan Kiber Department of Computer Science Department of Computer Science Department of Applied Physics,

### Correlation and Convolution Class Notes for CMSC 426, Fall 2005 David Jacobs

Correlation and Convolution Class otes for CMSC 46, Fall 5 David Jacobs Introduction Correlation and Convolution are basic operations that we will perform to extract information from images. They are in

### SWISS ARMY KNIFE INDICATOR John F. Ehlers

SWISS ARMY KNIFE INDICATOR John F. Ehlers The indicator I describe in this article does all the common functions of the usual indicators, such as smoothing and momentum generation. It also does some unusual

### MATH 4330/5330, Fourier Analysis Section 11, The Discrete Fourier Transform

MATH 433/533, Fourier Analysis Section 11, The Discrete Fourier Transform Now, instead of considering functions defined on a continuous domain, like the interval [, 1) or the whole real line R, we wish

### Adding Sinusoids of the Same Frequency. Additive Synthesis. Spectrum. Music 270a: Modulation

Adding Sinusoids of the Same Frequency Music 7a: Modulation Tamara Smyth, trsmyth@ucsd.edu Department of Music, University of California, San Diego (UCSD) February 9, 5 Recall, that adding sinusoids of

### SIGNAL PROCESSING FOR EFFECTIVE VIBRATION ANALYSIS

SIGNAL PROCESSING FOR EFFECTIVE VIBRATION ANALYSIS Dennis H. Shreve IRD Mechanalysis, Inc Columbus, Ohio November 1995 ABSTRACT Effective vibration analysis first begins with acquiring an accurate time-varying

### Sampling Theorem Notes. Recall: That a time sampled signal is like taking a snap shot or picture of signal periodically.

Sampling Theorem We will show that a band limited signal can be reconstructed exactly from its discrete time samples. Recall: That a time sampled signal is like taking a snap shot or picture of signal

### Unified Lecture # 4 Vectors

Fall 2005 Unified Lecture # 4 Vectors These notes were written by J. Peraire as a review of vectors for Dynamics 16.07. They have been adapted for Unified Engineering by R. Radovitzky. References [1] Feynmann,

### Time Series Analysis: Introduction to Signal Processing Concepts. Liam Kilmartin Discipline of Electrical & Electronic Engineering, NUI, Galway

Time Series Analysis: Introduction to Signal Processing Concepts Liam Kilmartin Discipline of Electrical & Electronic Engineering, NUI, Galway Aims of Course To introduce some of the basic concepts of

### FFT Algorithms. Chapter 6. Contents 6.1

Chapter 6 FFT Algorithms Contents Efficient computation of the DFT............................................ 6.2 Applications of FFT................................................... 6.6 Computing DFT

### Measuring Line Edge Roughness: Fluctuations in Uncertainty

Tutor6.doc: Version 5/6/08 T h e L i t h o g r a p h y E x p e r t (August 008) Measuring Line Edge Roughness: Fluctuations in Uncertainty Line edge roughness () is the deviation of a feature edge (as

### Sampling and Interpolation. Yao Wang Polytechnic University, Brooklyn, NY11201

Sampling and Interpolation Yao Wang Polytechnic University, Brooklyn, NY1121 http://eeweb.poly.edu/~yao Outline Basics of sampling and quantization A/D and D/A converters Sampling Nyquist sampling theorem

### Short-time FFT, Multi-taper analysis & Filtering in SPM12

Short-time FFT, Multi-taper analysis & Filtering in SPM12 Computational Psychiatry Seminar, FS 2015 Daniel Renz, Translational Neuromodeling Unit, ETHZ & UZH 20.03.2015 Overview Refresher Short-time Fourier

### Understanding Poles and Zeros

MASSACHUSETTS INSTITUTE OF TECHNOLOGY DEPARTMENT OF MECHANICAL ENGINEERING 2.14 Analysis and Design of Feedback Control Systems Understanding Poles and Zeros 1 System Poles and Zeros The transfer function

### Chapter 4 Online Appendix: The Mathematics of Utility Functions

Chapter 4 Online Appendix: The Mathematics of Utility Functions We saw in the text that utility functions and indifference curves are different ways to represent a consumer s preferences. Calculus can

### Experimental Modal Analysis

Experimental Analysis A Simple Non-Mathematical Presentation Peter Avitabile, University of Massachusetts Lowell, Lowell, Massachusetts Often times, people ask some simple questions regarding modal analysis

### EDEXCEL NATIONAL CERTIFICATE/DIPLOMA UNIT 5 - ELECTRICAL AND ELECTRONIC PRINCIPLES NQF LEVEL 3 OUTCOME 4 - ALTERNATING CURRENT

EDEXCEL NATIONAL CERTIFICATE/DIPLOMA UNIT 5 - ELECTRICAL AND ELECTRONIC PRINCIPLES NQF LEVEL 3 OUTCOME 4 - ALTERNATING CURRENT 4 Understand single-phase alternating current (ac) theory Single phase AC

### Introduction to Matrices for Engineers

Introduction to Matrices for Engineers C.T.J. Dodson, School of Mathematics, Manchester Universit 1 What is a Matrix? A matrix is a rectangular arra of elements, usuall numbers, e.g. 1 0-8 4 0-1 1 0 11

### MUSICAL INSTRUMENT FAMILY CLASSIFICATION

MUSICAL INSTRUMENT FAMILY CLASSIFICATION Ricardo A. Garcia Media Lab, Massachusetts Institute of Technology 0 Ames Street Room E5-40, Cambridge, MA 039 USA PH: 67-53-0 FAX: 67-58-664 e-mail: rago @ media.

### Lecture L3 - Vectors, Matrices and Coordinate Transformations

S. Widnall 16.07 Dynamics Fall 2009 Lecture notes based on J. Peraire Version 2.0 Lecture L3 - Vectors, Matrices and Coordinate Transformations By using vectors and defining appropriate operations between

### Shaft. Application of full spectrum to rotating machinery diagnostics. Centerlines. Paul Goldman, Ph.D. and Agnes Muszynska, Ph.D.

Shaft Centerlines Application of full spectrum to rotating machinery diagnostics Benefits of full spectrum plots Before we answer these questions, we d like to start with the following observation: The

### Lesson 3 DIRECT AND ALTERNATING CURRENTS. Task. The skills and knowledge taught in this lesson are common to all missile repairer tasks.

Lesson 3 DIRECT AND ALTERNATING CURRENTS Task. The skills and knowledge taught in this lesson are common to all missile repairer tasks. Objectives. When you have completed this lesson, you should be able

### Analog Representations of Sound

Analog Representations of Sound Magnified phonograph grooves, viewed from above: The shape of the grooves encodes the continuously varying audio signal. Analog to Digital Recording Chain ADC Microphone

### Measuring Impedance and Frequency Response of Guitar Pickups

Measuring Impedance and Frequency Response of Guitar Pickups Peter D. Hiscocks Syscomp Electronic Design Limited phiscock@ee.ryerson.ca www.syscompdesign.com April 30, 2011 Introduction The CircuitGear

### 10.3. The Exponential Form of a Complex Number. Introduction. Prerequisites. Learning Outcomes

The Exponential Form of a Complex Number 10.3 Introduction In this Section we introduce a third way of expressing a complex number: the exponential form. We shall discover, through the use of the complex

### The Time Constant of an RC Circuit

The Time Constant of an RC Circuit 1 Objectives 1. To determine the time constant of an RC Circuit, and 2. To determine the capacitance of an unknown capacitor. 2 Introduction What the heck is a capacitor?

### Final Year Project Progress Report. Frequency-Domain Adaptive Filtering. Myles Friel. Supervisor: Dr.Edward Jones

Final Year Project Progress Report Frequency-Domain Adaptive Filtering Myles Friel 01510401 Supervisor: Dr.Edward Jones Abstract The Final Year Project is an important part of the final year of the Electronic

### Applications of the DFT

CHAPTER 9 Applications of the DFT The Discrete Fourier Transform (DFT) is one of the most important tools in Digital Signal Processing. This chapter discusses three common ways it is used. First, the DFT

### The Sonometer The Resonant String and Timbre Change after plucking

The Sonometer The Resonant String and Timbre Change after plucking EQUIPMENT Pasco sonometers (pick up 5 from teaching lab) and 5 kits to go with them BK Precision function generators and Tenma oscilloscopes

### Computer Networks and Internets, 5e Chapter 6 Information Sources and Signals. Introduction

Computer Networks and Internets, 5e Chapter 6 Information Sources and Signals Modified from the lecture slides of Lami Kaya (LKaya@ieee.org) for use CECS 474, Fall 2008. 2009 Pearson Education Inc., Upper

### Practical Design of Filter Banks for Automatic Music Transcription

Practical Design of Filter Banks for Automatic Music Transcription Filipe C. da C. B. Diniz, Luiz W. P. Biscainho, and Sergio L. Netto Federal University of Rio de Janeiro PEE-COPPE & DEL-Poli, POBox 6854,

### Introduction to Digital Filters

CHAPTER 14 Introduction to Digital Filters Digital filters are used for two general purposes: (1) separation of signals that have been combined, and (2) restoration of signals that have been distorted

### Introduction to IQ-demodulation of RF-data

Introduction to IQ-demodulation of RF-data by Johan Kirkhorn, IFBT, NTNU September 15, 1999 Table of Contents 1 INTRODUCTION...3 1.1 Abstract...3 1.2 Definitions/Abbreviations/Nomenclature...3 1.3 Referenced

### Signal Processing First Lab 01: Introduction to MATLAB. 3. Learn a little about advanced programming techniques for MATLAB, i.e., vectorization.

Signal Processing First Lab 01: Introduction to MATLAB Pre-Lab and Warm-Up: You should read at least the Pre-Lab and Warm-up sections of this lab assignment and go over all exercises in the Pre-Lab section

### A Sound Analysis and Synthesis System for Generating an Instrumental Piri Song

, pp.347-354 http://dx.doi.org/10.14257/ijmue.2014.9.8.32 A Sound Analysis and Synthesis System for Generating an Instrumental Piri Song Myeongsu Kang and Jong-Myon Kim School of Electrical Engineering,

### AC CIRCUITS - CAPACITORS AND INDUCTORS

EXPRIMENT#8 AC CIRCUITS - CAPACITORS AND INDUCTORS NOTE: Two weeks are allocated for this experiment. Before performing this experiment, review the Proper Oscilloscope Use section of Experiment #7. Objective

### DIODE CIRCUITS LABORATORY. Fig. 8.1a Fig 8.1b

DIODE CIRCUITS LABORATORY A solid state diode consists of a junction of either dissimilar semiconductors (pn junction diode) or a metal and a semiconductor (Schottky barrier diode). Regardless of the type,

### Musical Analysis and Synthesis in Matlab

3. James Stewart, Calculus (5th ed.), Brooks/Cole, 2003. 4. TI-83 Graphing Calculator Guidebook, Texas Instruments,1995. Musical Analysis and Synthesis in Matlab Mark R. Petersen (mark.petersen@colorado.edu),

### Basics of Digital Recording

Basics of Digital Recording CONVERTING SOUND INTO NUMBERS In a digital recording system, sound is stored and manipulated as a stream of discrete numbers, each number representing the air pressure at a

### Unit 1 Number Sense. In this unit, students will study repeating decimals, percents, fractions, decimals, and proportions.

Unit 1 Number Sense In this unit, students will study repeating decimals, percents, fractions, decimals, and proportions. BLM Three Types of Percent Problems (p L-34) is a summary BLM for the material

### Chapter 15, example problems:

Chapter, example problems: (.0) Ultrasound imaging. (Frequenc > 0,000 Hz) v = 00 m/s. λ 00 m/s /.0 mm =.0 0 6 Hz. (Smaller wave length implies larger frequenc, since their product,

### No Solution Equations Let s look at the following equation: 2 +3=2 +7

5.4 Solving Equations with Infinite or No Solutions So far we have looked at equations where there is exactly one solution. It is possible to have more than solution in other types of equations that are

### Lecture 3: Signaling and Clock Recovery. CSE 123: Computer Networks Stefan Savage

Lecture 3: Signaling and Clock Recovery CSE 123: Computer Networks Stefan Savage Last time Protocols and layering Application Presentation Session Transport Network Datalink Physical Application Transport

### Fermi National Accelerator Laboratory. The Measurements and Analysis of Electromagnetic Interference Arising from the Booster GMPS

Fermi National Accelerator Laboratory Beams-Doc-2584 Version 1.0 The Measurements and Analysis of Electromagnetic Interference Arising from the Booster GMPS Phil S. Yoon 1,2, Weiren Chou 1, Howard Pfeffer

### 8.2. Solution by Inverse Matrix Method. Introduction. Prerequisites. Learning Outcomes

Solution by Inverse Matrix Method 8.2 Introduction The power of matrix algebra is seen in the representation of a system of simultaneous linear equations as a matrix equation. Matrix algebra allows us

### PeakVue Analysis for Antifriction Bearing Fault Detection

August 2011 PeakVue Analysis for Antifriction Bearing Fault Detection Peak values (PeakVue) are observed over sequential discrete time intervals, captured, and analyzed. The analyses are the (a) peak values

### A comparison of radio direction-finding technologies. Paul Denisowski, Applications Engineer Rohde & Schwarz

A comparison of radio direction-finding technologies Paul Denisowski, Applications Engineer Rohde & Schwarz Topics General introduction to radiolocation Manual DF techniques Doppler DF Time difference

### Supporting Information

S1 Supporting Information GFT NMR, a New Approach to Rapidly Obtain Precise High Dimensional NMR Spectral Information Seho Kim and Thomas Szyperski * Department of Chemistry, University at Buffalo, The

### Direct and Reflected: Understanding the Truth with Y-S 3

Direct and Reflected: Understanding the Truth with Y-S 3 -Speaker System Design Guide- December 2008 2008 Yamaha Corporation 1 Introduction Y-S 3 is a speaker system design software application. It is

### Transmission Lines. Smith Chart

Smith Chart The Smith chart is one of the most useful graphical tools for high frequency circuit applications. The chart provides a clever way to visualize complex functions and it continues to endure

### RightMark Audio Analyzer 6.0. User s Guide

RightMark Audio Analyzer 6.0 User s Guide About RMAA RightMark Audio Analyzer is intended for testing the quality of analog and digital sound sections of any audio equipment, be it a sound card, portable

### Introduction to FM-Stereo-RDS Modulation

Introduction to FM-Stereo-RDS Modulation Ge, Liang Tan, EK Kelly, Joe Verigy, China Verigy, Singapore Verigy US 1. Introduction Frequency modulation (FM) has a long history of its application and is widely

### Department of Electrical and Computer Engineering Ben-Gurion University of the Negev. LAB 1 - Introduction to USRP

Department of Electrical and Computer Engineering Ben-Gurion University of the Negev LAB 1 - Introduction to USRP - 1-1 Introduction In this lab you will use software reconfigurable RF hardware from National

### The Fundamentals of FFT-Based Audio Measurements in SmaartLive

The Fundamentals of FFT-Based Audio Measurements in SmaartLive Paul D. Henderson This article serves as summary of the Fast-Fourier Transform (FFT) analysis techniques implemented in t h e SIA -SmaartLive

### Review of Fundamental Mathematics

Review of Fundamental Mathematics As explained in the Preface and in Chapter 1 of your textbook, managerial economics applies microeconomic theory to business decision making. The decision-making tools

### CHAPTER 5 Round-off errors

CHAPTER 5 Round-off errors In the two previous chapters we have seen how numbers can be represented in the binary numeral system and how this is the basis for representing numbers in computers. Since any