A102 Signals and Systems for Hearing and Speech: Final exam answers

Similar documents
AUDL Final exam page 1/7 Please answer all of the following questions.

Acoustics, signals & systems for audiology. Week 4. Signals through Systems

Signals & Systems for Speech & Hearing. Week 6. Practical spectral analysis. Bandpass filters & filterbanks. Try this out on an old friend

Week 1. Signals & Systems for Speech & Hearing. Sound is a SIGNAL 3. You may find this course demanding! How to get through it:

You know about adding up waves, e.g. from two loudspeakers. AUDL 4007 Auditory Perception. Week 2½. Mathematical prelude: Adding up levels

AUDL 4007 Auditory Perception. Week 1. The cochlea & auditory nerve: Obligatory stages of auditory processing

Week I AUDL Signals & Systems for Speech & Hearing. Sound is a SIGNAL. You may find this course demanding! How to get through it: What is sound?

Hearing and Deafness 2. Ear as a frequency analyzer. Chris Darwin

Digitally controlled Active Noise Reduction with integrated Speech Communication

AUDL GS08/GAV1 Signals, systems, acoustics and the ear. Loudness & Temporal resolution

Imagine the cochlea unrolled

Perception of pitch. Importance of pitch: 2. mother hemp horse. scold. Definitions. Why is pitch important? AUDL4007: 11 Feb A. Faulkner.

Perception of pitch. Definitions. Why is pitch important? BSc Audiology/MSc SHS Psychoacoustics wk 4: 7 Feb A. Faulkner.

Signals, systems, acoustics and the ear. Week 3. Frequency characterisations of systems & signals

Acoustics, signals & systems for audiology. Week 3. Frequency characterisations of systems & signals

MUSC 316 Sound & Digital Audio Basics Worksheet

AUDL GS08/GAV1 Auditory Perception. Envelope and temporal fine structure (TFS)

Sampling and Reconstruction

Lab week 4: Harmonic Synthesis

3.2 Measuring Frequency Response Of Low-Pass Filter :

Perception of pitch. Definitions. Why is pitch important? BSc Audiology/MSc SHS Psychoacoustics wk 5: 12 Feb A. Faulkner.

PA System in a Box. Edwin Africano, Nathan Gutierrez, Tuan Phan

Phase and Feedback in the Nonlinear Brain. Malcolm Slaney (IBM and Stanford) Hiroko Shiraiwa-Terasawa (Stanford) Regaip Sen (Stanford)

PHYS225 Lecture 15. Electronic Circuits

Chapter 3. Meeting 3, Psychoacoustics, Hearing, and Reflections

Chapter 2. Meeting 2, Measures and Visualizations of Sounds and Signals

19 th INTERNATIONAL CONGRESS ON ACOUSTICS MADRID, 2-7 SEPTEMBER 2007

Acoustical Active Noise Control

8A. ANALYSIS OF COMPLEX SOUNDS. Amplitude, loudness, and decibels

Spectrum Analysis: The FFT Display

Definition of Sound. Sound. Vibration. Period - Frequency. Waveform. Parameters. SPA Lundeen

The quality of the transmission signal The characteristics of the transmission medium. Some type of transmission medium is required for transmission:

Data Communication. Chapter 3 Data Transmission

Experiment Five: The Noisy Channel Model

Subtractive Synthesis. Describing a Filter. Filters. CMPT 468: Subtractive Synthesis

Signal Characteristics

Laboratory Assignment 2 Signal Sampling, Manipulation, and Playback

Chapter 12. Preview. Objectives The Production of Sound Waves Frequency of Sound Waves The Doppler Effect. Section 1 Sound Waves

INDIANA UNIVERSITY, DEPT. OF PHYSICS P105, Basic Physics of Sound, Spring 2010

UNIVERSITY OF TORONTO Faculty of Arts and Science MOCK EXAMINATION PHY207H1S. Duration 3 hours NO AIDS ALLOWED

Butterworth Active Bandpass Filter using Sallen-Key Topology

Introduction to Equalization

ALTERNATING CURRENT (AC)

Analysis on Acoustic Attenuation by Periodic Array Structure EH KWEE DOE 1, WIN PA PA MYO 2

Structure of Speech. Physical acoustics Time-domain representation Frequency domain representation Sound shaping

Brief Introduction to Signals & Systems. Phani Chavali

6.551j/HST.714j Acoustics of Speech and Hearing: Exam 2

Computational Perception. Sound localization 2

Introduction to cochlear implants Philipos C. Loizou Figure Captions

Appendix B. Design Implementation Description For The Digital Frequency Demodulator

LAB #7: Digital Signal Processing

University Tunku Abdul Rahman LABORATORY REPORT 1

Fundamentals of Digital Audio *

Standard Octaves and Sound Pressure. The superposition of several independent sound sources produces multifrequency noise: i=1

Outline. Communications Engineering 1

Sampling and Reconstruction of Analog Signals

CHAPTER 14. Introduction to Frequency Selective Circuits

HCS 7367 Speech Perception

SHOCK AND VIBRATION RESPONSE SPECTRA COURSE Unit 17. Aliasing. Again, engineers collect accelerometer data in a variety of settings.

Temporal resolution AUDL Domain of temporal resolution. Fine structure and envelope. Modulating a sinusoid. Fine structure and envelope

3D Distortion Measurement (DIS)

E40M Sound and Music. M. Horowitz, J. Plummer, R. Howe 1

Filter Banks I. Prof. Dr. Gerald Schuller. Fraunhofer IDMT & Ilmenau University of Technology Ilmenau, Germany. Fraunhofer IDMT

When you have completed this exercise, you will be able to determine the frequency response of a

Physics 101. Lecture 21 Doppler Effect Loudness Human Hearing Interference of Sound Waves Reflection & Refraction of Sound

Measuring procedures for the environmental parameters: Acoustic comfort

Terminology (1) Chapter 3. Terminology (3) Terminology (2) Transmitter Receiver Medium. Data Transmission. Direct link. Point-to-point.

Module 3 : Sampling and Reconstruction Problem Set 3

EE 422G - Signals and Systems Laboratory

Case study for voice amplification in a highly absorptive conference room using negative absorption tuning by the YAMAHA Active Field Control system

Modulation. Digital Data Transmission. COMP476 Networked Computer Systems. Analog and Digital Signals. Analog and Digital Examples.

END-OF-YEAR EXAMINATIONS ELEC321 Communication Systems (D2) Tuesday, 22 November 2005, 9:20 a.m. Three hours plus 10 minutes reading time.

Chapter 15: Active Filters

Signal Processing. Introduction

Data Communications & Computer Networks

Transfer Function (TRF)

MUS 302 ENGINEERING SECTION

Application Note AN-13 Copyright October, 2002

INTRODUCTION TO COMPUTER MUSIC SAMPLING SYNTHESIS AND FILTERS. Professor of Computer Science, Art, and Music

Acoustics, signals & systems for audiology. Week 9. Basic Psychoacoustic Phenomena: Temporal resolution

Sound/Audio. Slides courtesy of Tay Vaughan Making Multimedia Work

UNIT TEST I Digital Communication

Application Note 7. Digital Audio FIR Crossover. Highlights Importing Transducer Response Data FIR Window Functions FIR Approximation Methods

STATION NUMBER: LAB SECTION: Filters. LAB 6: Filters ELECTRICAL ENGINEERING 43/100 INTRODUCTION TO MICROELECTRONIC CIRCUITS

6.101 Project Proposal April 9, 2014 Kayla Esquivel and Jason Yang. General Outline

Chapter 2: Digitization of Sound

Sound waves. septembre 2014 Audio signals and systems 1

describe sound as the transmission of energy via longitudinal pressure waves;

Advanced Audiovisual Processing Expected Background

4. Digital Measurement of Electrical Quantities

Distortion products and the perceived pitch of harmonic complex tones

Lab 8. ANALYSIS OF COMPLEX SOUNDS AND SPEECH ANALYSIS Amplitude, loudness, and decibels

PROBLEM SET 6. Note: This version is preliminary in that it does not yet have instructions for uploading the MATLAB problems.

Digital Processing of Continuous-Time Signals

ECE 2111 Signals and Systems Spring 2009, UMD Experiment 3: The Spectrum Analyzer

Chapter 3. Data Transmission

Threshold Noise-Cancelling Headphones

Psycho-acoustics (Sound characteristics, Masking, and Loudness)

Active Filter Design Techniques

Digital Processing of

Transcription:

A12 Signals and Systems for Hearing and Speech: Final exam answers 1) Take two sinusoids of 4 khz, both with a phase of. One has a peak level of.8 Pa while the other has a peak level of. Pa. Draw the spectrum of the larger sinusoid (on linear scales) and 3 cycles of its waveform. What are the peak levels of the two sinusoids in db SPL? What would be their peak level (in db SPL) if added? What if one had a phase of 18 while the other remained at? What level would the sum of these two sinusoids be, in Pa and db SPL? (1 points) sound pressure (Pa).9.8.7.6..4.3.2.1 1 2 3 4 frequency (Hz) 1 amplitude (Pa)..1.2.3.4..6.7 -. -1 time (s) Pa db SPL larger.8 92. smaller. 88. sum (in phase) 1.3 96.3 sum (out of phase).3 83. 1

2) A complex periodic waveform is made up from the following components. For each combination, calculate the fundamental frequency and fundamental period. (1 points) answer a) frequencies 2 4 6 2 Hz ms b) periods 2. 1 66.67 Hz 1 ms c) frequencies 2 3 6 Hz 2 ms d) periods 2. 2 1 1 Hz 1 ms e) frequencies 1 14 2 4 Hz 2 ms 3) Consider a wave which consists of the first 16 harmonics of a sawtooth wave whose fundamental period is 4 ms, and whose fundamental component has a level of 3 Pa. (2 points) a) Draw its spectrum (on db SPL and logarithmic frequency scales over the frequency range 12 Hz to 2 khz). b) This wave is then put through a System X which results in the output wave having a spectrum in which all components are of equal amplitude at 2 Pa. Draw this spectrum on db SPL and logarithmic frequency scales over the frequency range 12 Hz to 2 khz. c) Over the same frequency range, and again using db and logarithmic frequency scales, draw the amplitude response of System X. db SPL gain (db) 12 1 8 6 4 2 1 1 1 2 1 1 1 1 1 - -1 Note: x-axis to go from 12 Hz, 2 Hz, Hz, 1 khz, 2kHz. Section b) essentially the same graph as Section a) but all points at the same level of 1 db SPL. 2

4) Suppose that a pair of ear plugs reduced the amplitude of sound outside the ear by a factor of four (i.e. so that the sound pressure in the ear was one-quarter of what it was outside the ear), what would be the db SPL level inside the ear given an external noise level of 16 db SPL? How many Pa does this correspond to? (1 Points) original one quarter result (db SPL) result (Pa) 16-12.412 93.96.997631 ) Sketch, on db and logarithmic frequency scales (over a frequency range of Hz 12.8 khz), a bandpass filter which has a gain of 2 db in the passband which ranges from 8 Hz to 3.2 khz, and which rolls off at 1 db/octave on the low frequency side, and 6 db/octave on the high frequency side. (1 Points) frequency gain (db) -2 1-1 2 4 1 8 2 16 2 32 2 64 14 128 8 gain (db) 2 2 1 1-1 1 1 1 1-1 -1 frequency (Hz) 6) Give three examples of the way in which the notion of a bandpass filter can be useful in characterizing the functioning of the auditory periphery. Be sure to specify the input and 3

output signals, and the units in which they would be measured, for each system you name (1 points) Ear canal: input = sound at ear canal entrance; output = sound at tympanic membrane; both measure in pascals Middle ear: input = movement of tympanic membrane (meters) or sound pressure at TM (Pa); output = movement of stapes (m) or pressure in cochlear fluids (Pa) Basilar membrane: input = movement of stapes (m) or pressure in cochlear fluids (Pa); output = movement of BM (m) 7) Suppose you were told that someone had a threshold that was 7 db better than the average normal-hearing listener at 12 Hz, and that the normal threshold of hearing at this frequency was 632. µpa. What sound pressure (in Pa or µpa) would be the least intense that person could hear at 12 Hz? What is the normal average threshold in db SPL at 12 Hz? ( points) 632. µpa = 3 db SPL 7 db better means a threshold of 23 db SPL = 282. µpa or.282 Pa 8) Consider a cascade of three systems with the following amplitude responses. Draw the amplitude response of the complete cascade. What would be the amplitude response of the cascade if the position of Systems A and B were reversed? (1 points) It doesn t matter if the order of the systems is reversed. total 2 gain or response (db) 1-1 -2-3 -4 1 1 2 frequency (Hz) 4

9) Draw the input and output spectra of white noise passed through the first system in the cascade of question 8. (1 points) Input spectrum must be a continuous flat line; output spectrum identical to frequency response of 1 st system. Must be labeled db or some appropriate linear measure by frequency. 1) Imagine you had a microphone that could be considered a perfect LTI system. Draw its input/output function (on linear scales) for a 1 khz sinusoid, assuming that a sound pressure level of 1 db SPL leads to an output level of 2 mv. (1 points) 1 db SPL = 2 Pa. Graph must be a straight line with the given point. output level (mv) 4 4 3 3 2 2 1 1 1 2 3 4 input level (Pa) 11) After digitising a signal, you find that its spectrum of the digital signal only contains frequencies up to Hz. What is the most likely reason for that? (1 points) The sound card is sampling at 1 khz, so the anti-aliasing low-pass filter is set to khz.

12) Suppose you had a system that half-wave rectified the input, which is the same as setting all negative values of the wave to zero, but leaving the positive values untouched. Here are sample input and output waveforms for a sinusoid of peak amplitude 1 V and frequency of 2 Hz: What s the simplest way of knowing that this is not an LTI system? Is this system homogeneous? Additive? Time-invariant? Give reasons for your answers. (2 points) An input sinusoid resulted in a non-sinusoidal output. Homogeneous: yes Additive: no Time-invariant: yes 13) Draw frequency domain diagrams to show how a narrow band-pass filter can be used to extract a single harmonic from a complex periodic input signal of fundamental period 1 ms (for example, a sawtooth). How would you decide what bandwidth to use for the band-pass filter? (2 Points) Need: Sensible graph of input spectrum (many harmonics at multiples of 1 Hz) Sensible graph of a bandpass filter whose bandwidth is less than 1 Hz Sensible graph of output spectrum (a single harmonics at some multiple of 1 Hz) 14) A sound card is a device within, or inserted into a computer, that changes analogue signals into digital ones (and back). One crucial function of a sound card is to filter a signal at its input. What is this filtering process called? What type of filter is used? Sketch a possible frequency response for this filter. Which frequencies are filtered and why? Make sure you explain the Nyquist frequency in this respect. (1 Points) Anti-aliasing filtering. A low pass filter. Need sensible graph of a lowpass filter. Only frequencies less than half the sampling frequency can be represented accurately. Higher frequencies than this are aliased into lower frequencies. The Nyquist rate is twice the highest frequency in the signal, and is the minimum rate needed for accurate representation. 1) A compressor-limiter is a tool used widely in speech and music amplification systems to normalise the amplitudes, i.e. attenuate high amplitudes and boost low amplitudes. Without a compressor-limiter we would always need a volume control at hand when watching TV to increase volume when people speak with a low-voice and decrease it in a loud passage. Explain whether a compressor-limiter is a linear time-invariant system. (1 Points) 6

It is not LTI because it is not homogeneous. The gain of the system changes with the amplitude of the input. 7