Communications Theory and Engineering
|
|
- Alexia Gregory
- 5 years ago
- Views:
Transcription
1 Communications Theory and Engineering Master's Degree in Electronic Engineering Sapienza University of Rome A.A
2 Speech and telephone speech
3 Based on a voice production model Parametric representation of speech signals Vocal tract Vocal folds Excitation signal Model match Excitation signal FILTER H(f) FILTER H(f) Speech signal Glottal signal VOCAL TRACT
4 Parametric representation The idea: the signal can be considered as the output of a system excited by an appropriate excitation signal QUESTION IS:? How should the excitation signal be defined? How should the system be characterized? THE ANSWER IS IN THE STRUCTURE OF THE SPEECH SIGNAL
5 Parametric representation The speech signal is quasi-stationary: This means that it can be considered as stationary for short-time intervals Typically 10~20 ms 10 ms Time Implying that: the model parameters must be updated every ms A segment of duration ms is referred to as FRAME This analysis is referred to as short-term analysis
6 Parametric representation Source model (excitation signal) Two main categories of sounds were identified: Voiced sounds: Vocal folds start vibrating; the corresponding excitation signal is a pulse train with repetition period T: the pitch interval Air from lungs Voiceless sounds: Vocal folds are open, while the vocal tract closes at a specific point, causing the air coming from lungs to create a turbulence at the constriction. The corresponding excitation signal is noise. Air from lungs Vocal cords tight and vibrating The vocal tract narrows Vocal folds are open
7 Parametric representation Reminder: voiced sounds vs. voiceless sounds The waveform of a voiced sound is almost periodic T T: pitch interval The waveform of a voiceless sound is noise-like
8 Source model Voiced sounds: pulse train generator, with period T PULSE TRAIN GENERATOR Voiceless sounds: noise generator T time NOISE GENERATOR time A voiced/voiceless detector is thus required in order to select which excitation signal shall be used
9 Vocal tract model The filter H(f) must be characterized by a transfer function that mimics the action performed by the vocal tract on the excitation signal A LINEAR PREDICTION filter is adopted The parameters of the filter must be updated every ms But what is linear prediction?
10 Vocal tract model The analysis relies on the idea that a sample predicted by previous signal samples: ( ) s n of the signal can be The prediction of s( n) is a LINEAR combination of previous samples s( n i) i =1,..., p that is: ( ) = α k s( n k)!s n p k=1 Linear Prediction of s(n) p is referred to as PREDICTION ORDER
11 Linear prediction The adopted approach is to determine the coefficients that minimize the difference between sample and the prediction, i.e. minimize the PREDICTION ERROR α k ( ) s! ( n) s n PREDICTION ERROR e( n) = s( n)!s ( n) where BUT ( ) = α k s( n k)!s n p k=1 REMEMBER THAT The analysis is short-term
12 Linear prediction α k In particular the coefficients can be determined by minimizing the SHORT TERM quadratic error N m=1 ( ) E n = e n 2 m for each analysis window n, where N is the number of samples per window Since a window has a typical duration of ms At a 10 khz sampling frequency the corresponding number of samples is
13 Linear prediction N N E n = e 2 ( n m) = ( s ( n m)!s ( n m) ) 2 N p = s ( n m) α k s n m k m=1 m=1 m=1 k=1 ( ) 2 FIND THE MINIMUM We are searching for a set of α k such that: E n α i = 0 for i=1,, p Order of prediction
14 Linear prediction Leading to the the following set of equations: R n YULE-WALKER EQUATIONS R ( n 0) R ( n 1)! R ( n p 1) R ( n 1) R ( n 0)! R ( n p 2) " " # " p 1 ( ) R ( n p 2)! R ( n 0) α 1 α 2 " α p = R ( n 1) R ( n 2) R n " p ( ) where: R n ( i) = s ( n m)s n m+i m ( ) SHORT-TERM AUTOCORRELATION FUNCTION
15 Linear prediction It can be observed that the matrix R n R ( n 0) R ( n 1) R ( n 2)! R ( n p 1) R ( n 1) R ( n 0) R ( n 1)! R ( n p 2) R ( n 2) " R ( n 1) " R ( n 0) is a Toeplitz matrix, and as such: It is symmetric All the elements on the main diagonal have same value "! # ( p 1) R ( n p 2) R ( n p 3)! R ( n 0) " "
16 Parametric representation Example: [a] VOWEL for several values of prediction order p AMPLITUDE db TIME FREQUENCY FREQUENCY FREQUENCY FREQUENCY FREQUENCY FREQUENCY INPUT SIGNAL SHORT TERM SPECTRUM LPC p=4 LPC p=8 LPC p=12 LPC p=16 LPC p=20
17 Parametric representation Summary VOCAL TRACT PARAMETERS: Coefficientsα k Gain G SOURCE PARAMETERS: PITCH VOICED/VOICELESS decision
18 Parametric representation The complete model is thus as follows: PITCH SOURCE PARAMETER: PITCH + VOICED/VOICELESS decision PULSE TRAIN GENERATOR RANDOM NOISE GENERATOR u n Voiced/voiceless switch x G TIME-VARYING FILTER ( ) ( ) s n α k COEFFICIENTS Transmission rate up to bit/s
19 Parametric representation The predictor order p is typically about 12~14 ENERGY (db) Red: original signal Blue: LPC FREQUENCY
20 Parametric representation VOCODER scheme s( n) LPC analysis filter Coder Channel LPC synthesis filter PITCH detector Decoder ŝ( n) TRANSMITTER CHANNEL RECEIVER
21 Mixed systems Mixed systems are only in part based on speech production models Best example: Multipulse The multipulse method achieves excellent quality far transmission rates around bit/s In this method the vocal tract is represented by a LPC filter, but the source is determined without relying on specific properties of the speech signal ( ) u n LPC filter ( )!s n ( ) u ( n ) ( ) Given a signal!s n, one searches for the system input that makes as close as possible tos n ( ) α k The input u n and the coefficients are then sent to the receiver!s ( n)
22 What is ( ) u n like? Multipulse systems Let us assume a signal window of length 100 samples: it is obvious that, if u( n) had length 100 samples, the synthesis would be perfect The maximum number of available samples depends however on the Depending on the transmission rate, pulses (samples) TRANSMISSION RATE u( n) will thus include the right number of u( n)
23 Multipulse systems Optimal positions and amplitudes of pulses forming the input sequence must then be determined Example: for a bit rate of about 16 kbits/s, one can transmit ~ 30 pulses for a signal frame of 128 samples. u n will contain ~ 30 samples of NON-ZERO amplitude ( ) Finding optimal positions would require to analyze ALL possible positions, with an unacceptable computational cost Sub-optimal solutions are typically adopted
24 Multipulse systems In the search for pulse positions, positions are explored ONE AT THE TIME u( n) In the search for pulse amplitudes, a system of linear equations can be defined
25 Multipulse systems Which information is transmitted? Coefficients Positions Amplitudes Amplitude quantization step NOTE THAT OPPOSITELY to the VOCODER, here there is no information on the structure of the source signal (neither voiced/voiceless decision, nor pitch extraction)
26 Mixed methods GSM system This method was standardized for early digital RADIO-MOBILE voice transmissions STRUCTURE: similar to the one described or the multipulse system, but the search for optimal positions is carried out with a resolution of three samples u( n) POSITION SEARCH Computational cost much lower than in multipulse system 13 kb/s standard
27 MPEG1 Audio Compression Input Critical bands filtering (sub-band filtering) Bit allocation (quantization) Bitstream formatting Output Estimation of masking effects (Psychoacoustic model) MPEG1 audio compression works in the frequency domain It takes advantage of limitations in the human auditory system in order to reduce the bit rate without significant effect on perceived audio quality MPEG1 audio compression evolved in 3 different layers: Layer 1, Layer 2 and Layer 3 (MPEG1 Layer 3, known as MP3)
28 Frequency masking: the Bark scale The perception of a sound at a given frequency reduces the capability of the ear of perceiving other sounds at nearby frequencies The higher the intensity of the sound, the stronger the masking effect The frequency interval affected by a sound is referred to as critical band The width of critical bands grows with frequency
29 Time masking A tone at high intensity affects the capability of human ear to perceive another tone at nearby frequencies even after the perception of the first tone ends.
30 Overall masking effect The combination of time masking and frequency masking leads to specific frequency intervals that are not audible for specific time intervals
31 MP3 coding MP3 coding uses information on time and frequency masking effects to achieve efficient bit allocation for quantization Bands affected by masking effects are coded with a low number of bits (higher quantization noise): this module is proprietary PCM bitstream Filter bank (32 sub-bands) Modified Discrete Cosine Transform Non-uniform Quantization FFT (1024 points) Psychoacoustic model (proprietary) Additional control signalling MP3 coded bitstream Bitstream creation Huffman coding
Analysis/synthesis coding
TSBK06 speech coding p.1/32 Analysis/synthesis coding Many speech coders are based on a principle called analysis/synthesis coding. Instead of coding a waveform, as is normally done in general audio coders
More informationEE482: Digital Signal Processing Applications
Professor Brendan Morris, SEB 3216, brendan.morris@unlv.edu EE482: Digital Signal Processing Applications Spring 2014 TTh 14:30-15:45 CBC C222 Lecture 12 Speech Signal Processing 14/03/25 http://www.ee.unlv.edu/~b1morris/ee482/
More informationAPPLICATIONS OF DSP OBJECTIVES
APPLICATIONS OF DSP OBJECTIVES This lecture will discuss the following: Introduce analog and digital waveform coding Introduce Pulse Coded Modulation Consider speech-coding principles Introduce the channel
More informationCellular systems & GSM Wireless Systems, a.a. 2014/2015
Cellular systems & GSM Wireless Systems, a.a. 2014/2015 Un. of Rome La Sapienza Chiara Petrioli Department of Computer Science University of Rome Sapienza Italy 2 Voice Coding 3 Speech signals Voice coding:
More informationSPEECH AND SPECTRAL ANALYSIS
SPEECH AND SPECTRAL ANALYSIS 1 Sound waves: production in general: acoustic interference vibration (carried by some propagation medium) variations in air pressure speech: actions of the articulatory organs
More informationspeech signal S(n). This involves a transformation of S(n) into another signal or a set of signals
16 3. SPEECH ANALYSIS 3.1 INTRODUCTION TO SPEECH ANALYSIS Many speech processing [22] applications exploits speech production and perception to accomplish speech analysis. By speech analysis we extract
More informationDigital Speech Processing and Coding
ENEE408G Spring 2006 Lecture-2 Digital Speech Processing and Coding Spring 06 Instructor: Shihab Shamma Electrical & Computer Engineering University of Maryland, College Park http://www.ece.umd.edu/class/enee408g/
More informationSpeech Compression Using Voice Excited Linear Predictive Coding
Speech Compression Using Voice Excited Linear Predictive Coding Ms.Tosha Sen, Ms.Kruti Jay Pancholi PG Student, Asst. Professor, L J I E T, Ahmedabad Abstract : The aim of the thesis is design good quality
More informationSpeech Synthesis using Mel-Cepstral Coefficient Feature
Speech Synthesis using Mel-Cepstral Coefficient Feature By Lu Wang Senior Thesis in Electrical Engineering University of Illinois at Urbana-Champaign Advisor: Professor Mark Hasegawa-Johnson May 2018 Abstract
More informationLinguistic Phonetics. Spectral Analysis
24.963 Linguistic Phonetics Spectral Analysis 4 4 Frequency (Hz) 1 Reading for next week: Liljencrants & Lindblom 1972. Assignment: Lip-rounding assignment, due 1/15. 2 Spectral analysis techniques There
More informationVoice Excited Lpc for Speech Compression by V/Uv Classification
IOSR Journal of VLSI and Signal Processing (IOSR-JVSP) Volume 6, Issue 3, Ver. II (May. -Jun. 2016), PP 65-69 e-issn: 2319 4200, p-issn No. : 2319 4197 www.iosrjournals.org Voice Excited Lpc for Speech
More informationOverview of Code Excited Linear Predictive Coder
Overview of Code Excited Linear Predictive Coder Minal Mulye 1, Sonal Jagtap 2 1 PG Student, 2 Assistant Professor, Department of E&TC, Smt. Kashibai Navale College of Engg, Pune, India Abstract Advances
More informationEC 6501 DIGITAL COMMUNICATION UNIT - II PART A
EC 6501 DIGITAL COMMUNICATION 1.What is the need of prediction filtering? UNIT - II PART A [N/D-16] Prediction filtering is used mostly in audio signal processing and speech processing for representing
More informationUniversity of Washington Department of Electrical Engineering Computer Speech Processing EE516 Winter 2005
University of Washington Department of Electrical Engineering Computer Speech Processing EE516 Winter 2005 Lecture 5 Slides Jan 26 th, 2005 Outline of Today s Lecture Announcements Filter-bank analysis
More informationEE 225D LECTURE ON MEDIUM AND HIGH RATE CODING. University of California Berkeley
University of California Berkeley College of Engineering Department of Electrical Engineering and Computer Sciences Professors : N.Morgan / B.Gold EE225D Spring,1999 Medium & High Rate Coding Lecture 26
More informationtechniques are means of reducing the bandwidth needed to represent the human voice. In mobile
8 2. LITERATURE SURVEY The available radio spectrum for the wireless radio communication is very limited hence to accommodate maximum number of users the speech is compressed. The speech compression techniques
More informationEE482: Digital Signal Processing Applications
Professor Brendan Morris, SEB 3216, brendan.morris@unlv.edu EE482: Digital Signal Processing Applications Spring 2014 TTh 14:30-15:45 CBC C222 Lecture 14 Quiz 04 Review 14/04/07 http://www.ee.unlv.edu/~b1morris/ee482/
More informationSpeech Synthesis; Pitch Detection and Vocoders
Speech Synthesis; Pitch Detection and Vocoders Tai-Shih Chi ( 冀泰石 ) Department of Communication Engineering National Chiao Tung University May. 29, 2008 Speech Synthesis Basic components of the text-to-speech
More informationDigital Signal Representation of Speech Signal
Digital Signal Representation of Speech Signal Mrs. Smita Chopde 1, Mrs. Pushpa U S 2 1,2. EXTC Department, Mumbai University Abstract Delta modulation is a waveform coding techniques which the data rate
More informationComparison of CELP speech coder with a wavelet method
University of Kentucky UKnowledge University of Kentucky Master's Theses Graduate School 2006 Comparison of CELP speech coder with a wavelet method Sriram Nagaswamy University of Kentucky, sriramn@gmail.com
More informationEnhanced Waveform Interpolative Coding at 4 kbps
Enhanced Waveform Interpolative Coding at 4 kbps Oded Gottesman, and Allen Gersho Signal Compression Lab. University of California, Santa Barbara E-mail: [oded, gersho]@scl.ece.ucsb.edu Signal Compression
More informationBasic Characteristics of Speech Signal Analysis
www.ijird.com March, 2016 Vol 5 Issue 4 ISSN 2278 0211 (Online) Basic Characteristics of Speech Signal Analysis S. Poornima Assistant Professor, VlbJanakiammal College of Arts and Science, Coimbatore,
More informationA Physiologically Produced Impulsive UWB signal: Speech
A Physiologically Produced Impulsive UWB signal: Speech Maria-Gabriella Di Benedetto University of Rome La Sapienza Faculty of Engineering Rome, Italy gaby@acts.ing.uniroma1.it http://acts.ing.uniroma1.it
More informationThe Channel Vocoder (analyzer):
Vocoders 1 The Channel Vocoder (analyzer): The channel vocoder employs a bank of bandpass filters, Each having a bandwidth between 100 Hz and 300 Hz. Typically, 16-20 linear phase FIR filter are used.
More informationRobust Linear Prediction Analysis for Low Bit-Rate Speech Coding
Robust Linear Prediction Analysis for Low Bit-Rate Speech Coding Nanda Prasetiyo Koestoer B. Eng (Hon) (1998) School of Microelectronic Engineering Faculty of Engineering and Information Technology Griffith
More informationTelecommunication Electronics
Politecnico di Torino ICT School Telecommunication Electronics C5 - Special A/D converters» Logarithmic conversion» Approximation, A and µ laws» Differential converters» Oversampling, noise shaping Logarithmic
More informationDEPARTMENT OF INFORMATION TECHNOLOGY QUESTION BANK. Subject Name: Information Coding Techniques UNIT I INFORMATION ENTROPY FUNDAMENTALS
DEPARTMENT OF INFORMATION TECHNOLOGY QUESTION BANK Subject Name: Year /Sem: II / IV UNIT I INFORMATION ENTROPY FUNDAMENTALS PART A (2 MARKS) 1. What is uncertainty? 2. What is prefix coding? 3. State the
More informationAudio Signal Compression using DCT and LPC Techniques
Audio Signal Compression using DCT and LPC Techniques P. Sandhya Rani#1, D.Nanaji#2, V.Ramesh#3,K.V.S. Kiran#4 #Student, Department of ECE, Lendi Institute Of Engineering And Technology, Vizianagaram,
More informationInternational Journal of Modern Trends in Engineering and Research e-issn No.: , Date: 2-4 July, 2015
International Journal of Modern Trends in Engineering and Research www.ijmter.com e-issn No.:2349-9745, Date: 2-4 July, 2015 Analysis of Speech Signal Using Graphic User Interface Solly Joy 1, Savitha
More informationAnalog and Telecommunication Electronics
Politecnico di Torino - ICT School Analog and Telecommunication Electronics D5 - Special A/D converters» Differential converters» Oversampling, noise shaping» Logarithmic conversion» Approximation, A and
More informationGeneral outline of HF digital radiotelephone systems
Rec. ITU-R F.111-1 1 RECOMMENDATION ITU-R F.111-1* DIGITIZED SPEECH TRANSMISSIONS FOR SYSTEMS OPERATING BELOW ABOUT 30 MHz (Question ITU-R 164/9) Rec. ITU-R F.111-1 (1994-1995) The ITU Radiocommunication
More informationVoice Transmission --Basic Concepts--
Voice Transmission --Basic Concepts-- Voice---is analog in character and moves in the form of waves. 3-important wave-characteristics: Amplitude Frequency Phase Telephone Handset (has 2-parts) 2 1. Transmitter
More informationCOMPRESSIVE SAMPLING OF SPEECH SIGNALS. Mona Hussein Ramadan. BS, Sebha University, Submitted to the Graduate Faculty of
COMPRESSIVE SAMPLING OF SPEECH SIGNALS by Mona Hussein Ramadan BS, Sebha University, 25 Submitted to the Graduate Faculty of Swanson School of Engineering in partial fulfillment of the requirements for
More informationEC 2301 Digital communication Question bank
EC 2301 Digital communication Question bank UNIT I Digital communication system 2 marks 1.Draw block diagram of digital communication system. Information source and input transducer formatter Source encoder
More informationSpeech Coding using Linear Prediction
Speech Coding using Linear Prediction Jesper Kjær Nielsen Aalborg University and Bang & Olufsen jkn@es.aau.dk September 10, 2015 1 Background Speech is generated when air is pushed from the lungs through
More informationReading: Johnson Ch , Ch.5.5 (today); Liljencrants & Lindblom; Stevens (Tues) reminder: no class on Thursday.
L105/205 Phonetics Scarborough Handout 7 10/18/05 Reading: Johnson Ch.2.3.3-2.3.6, Ch.5.5 (today); Liljencrants & Lindblom; Stevens (Tues) reminder: no class on Thursday Spectral Analysis 1. There are
More informationAdaptive Filters Application of Linear Prediction
Adaptive Filters Application of Linear Prediction Gerhard Schmidt Christian-Albrechts-Universität zu Kiel Faculty of Engineering Electrical Engineering and Information Technology Digital Signal Processing
More informationSignal Processing for Speech Applications - Part 2-1. Signal Processing For Speech Applications - Part 2
Signal Processing for Speech Applications - Part 2-1 Signal Processing For Speech Applications - Part 2 May 14, 2013 Signal Processing for Speech Applications - Part 2-2 References Huang et al., Chapter
More informationMel Spectrum Analysis of Speech Recognition using Single Microphone
International Journal of Engineering Research in Electronics and Communication Mel Spectrum Analysis of Speech Recognition using Single Microphone [1] Lakshmi S.A, [2] Cholavendan M [1] PG Scholar, Sree
More informationSpeech Signal Analysis
Speech Signal Analysis Hiroshi Shimodaira and Steve Renals Automatic Speech Recognition ASR Lectures 2&3 14,18 January 216 ASR Lectures 2&3 Speech Signal Analysis 1 Overview Speech Signal Analysis for
More informationPulse Code Modulation
Pulse Code Modulation EE 44 Spring Semester Lecture 9 Analog signal Pulse Amplitude Modulation Pulse Width Modulation Pulse Position Modulation Pulse Code Modulation (3-bit coding) 1 Advantages of Digital
More informationThe source-filter model of speech production"
24.915/24.963! Linguistic Phonetics! The source-filter model of speech production" Glottal airflow Output from lips 400 200 0.1 0.2 0.3 Time (in secs) 30 20 10 0 0 1000 2000 3000 Frequency (Hz) Source
More informationAuditory modelling for speech processing in the perceptual domain
ANZIAM J. 45 (E) ppc964 C980, 2004 C964 Auditory modelling for speech processing in the perceptual domain L. Lin E. Ambikairajah W. H. Holmes (Received 8 August 2003; revised 28 January 2004) Abstract
More informationPitch Period of Speech Signals Preface, Determination and Transformation
Pitch Period of Speech Signals Preface, Determination and Transformation Mohammad Hossein Saeidinezhad 1, Bahareh Karamsichani 2, Ehsan Movahedi 3 1 Islamic Azad university, Najafabad Branch, Saidinezhad@yahoo.com
More informationWideband Speech Coding & Its Application
Wideband Speech Coding & Its Application Apeksha B. landge. M.E. [student] Aditya Engineering College Beed Prof. Amir Lodhi. Guide & HOD, Aditya Engineering College Beed ABSTRACT: Increasing the bandwidth
More informationSpeech Coding in the Frequency Domain
Speech Coding in the Frequency Domain Speech Processing Advanced Topics Tom Bäckström Aalto University October 215 Introduction The speech production model can be used to efficiently encode speech signals.
More informationAudio Compression using the MLT and SPIHT
Audio Compression using the MLT and SPIHT Mohammed Raad, Alfred Mertins and Ian Burnett School of Electrical, Computer and Telecommunications Engineering University Of Wollongong Northfields Ave Wollongong
More informationSpeech Coding Technique And Analysis Of Speech Codec Using CS-ACELP
Speech Coding Technique And Analysis Of Speech Codec Using CS-ACELP Monika S.Yadav Vidarbha Institute of Technology Rashtrasant Tukdoji Maharaj Nagpur University, Nagpur, India monika.yadav@rediffmail.com
More informationVocoder (LPC) Analysis by Variation of Input Parameters and Signals
ISCA Journal of Engineering Sciences ISCA J. Engineering Sci. Vocoder (LPC) Analysis by Variation of Input Parameters and Signals Abstract Gupta Rajani, Mehta Alok K. and Tiwari Vebhav Truba College of
More informationMultimedia Signal Processing: Theory and Applications in Speech, Music and Communications
Brochure More information from http://www.researchandmarkets.com/reports/569388/ Multimedia Signal Processing: Theory and Applications in Speech, Music and Communications Description: Multimedia Signal
More informationImproving Sound Quality by Bandwidth Extension
International Journal of Scientific & Engineering Research, Volume 3, Issue 9, September-212 Improving Sound Quality by Bandwidth Extension M. Pradeepa, M.Tech, Assistant Professor Abstract - In recent
More informationDigital Audio. Lecture-6
Digital Audio Lecture-6 Topics today Digitization of sound PCM Lossless predictive coding 2 Sound Sound is a pressure wave, taking continuous values Increase / decrease in pressure can be measured in amplitude,
More informationSpeech/Music Change Point Detection using Sonogram and AANN
International Journal of Information & Computation Technology. ISSN 0974-2239 Volume 6, Number 1 (2016), pp. 45-49 International Research Publications House http://www. irphouse.com Speech/Music Change
More informationSGN Audio and Speech Processing
Introduction 1 Course goals Introduction 2 SGN 14006 Audio and Speech Processing Lectures, Fall 2014 Anssi Klapuri Tampere University of Technology! Learn basics of audio signal processing Basic operations
More informationTechniques for low-rate scalable compression of speech signals
University of Wollongong Research Online University of Wollongong Thesis Collection University of Wollongong Thesis Collections 2002 Techniques for low-rate scalable compression of speech signals Jason
More informationChapter IV THEORY OF CELP CODING
Chapter IV THEORY OF CELP CODING CHAPTER IV THEORY OF CELP CODING 4.1 Introduction Wavefonn coders fail to produce high quality speech at bit rate lower than 16 kbps. Source coders, such as LPC vocoders,
More informationDepartment of Electronics and Communication Engineering 1
UNIT I SAMPLING AND QUANTIZATION Pulse Modulation 1. Explain in detail the generation of PWM and PPM signals (16) (M/J 2011) 2. Explain in detail the concept of PWM and PAM (16) (N/D 2012) 3. What is the
More informationOverview of Signal Processing
Overview of Signal Processing Chapter Intended Learning Outcomes: (i) Understand basic terminology in signal processing (ii) Differentiate digital signal processing and analog signal processing (iii) Describe
More informationDifferent Approaches of Spectral Subtraction Method for Speech Enhancement
ISSN 2249 5460 Available online at www.internationalejournals.com International ejournals International Journal of Mathematical Sciences, Technology and Humanities 95 (2013 1056 1062 Different Approaches
More informationAcoustic Phonetics. Chapter 8
Acoustic Phonetics Chapter 8 1 1. Sound waves Vocal folds/cords: Frequency: 300 Hz 0 0 0.01 0.02 0.03 2 1.1 Sound waves: The parts of waves We will be considering the parts of a wave with the wave represented
More informationOn a Classification of Voiced/Unvoiced by using SNR for Speech Recognition
International Conference on Advanced Computer Science and Electronics Information (ICACSEI 03) On a Classification of Voiced/Unvoiced by using SNR for Speech Recognition Jongkuk Kim, Hernsoo Hahn Department
More informationECE 556 BASICS OF DIGITAL SPEECH PROCESSING. Assıst.Prof.Dr. Selma ÖZAYDIN Spring Term-2017 Lecture 2
ECE 556 BASICS OF DIGITAL SPEECH PROCESSING Assıst.Prof.Dr. Selma ÖZAYDIN Spring Term-2017 Lecture 2 Analog Sound to Digital Sound Characteristics of Sound Amplitude Wavelength (w) Frequency ( ) Timbre
More informationINTRODUCTION TO ACOUSTIC PHONETICS 2 Hilary Term, week 6 22 February 2006
1. Resonators and Filters INTRODUCTION TO ACOUSTIC PHONETICS 2 Hilary Term, week 6 22 February 2006 Different vibrating objects are tuned to specific frequencies; these frequencies at which a particular
More informationNOVEL PITCH DETECTION ALGORITHM WITH APPLICATION TO SPEECH CODING
NOVEL PITCH DETECTION ALGORITHM WITH APPLICATION TO SPEECH CODING A Thesis Submitted to the Graduate Faculty of the University of New Orleans in partial fulfillment of the requirements for the degree of
More informationENEE408G Multimedia Signal Processing
ENEE408G Multimedia Signal Processing Design Project on Digital Speech Processing Goals: 1. Learn how to use the linear predictive model for speech analysis and synthesis. 2. Implement a linear predictive
More informationAssistant Lecturer Sama S. Samaan
MP3 Not only does MPEG define how video is compressed, but it also defines a standard for compressing audio. This standard can be used to compress the audio portion of a movie (in which case the MPEG standard
More informationLow Bit Rate Speech Coding
Low Bit Rate Speech Coding Jaspreet Singh 1, Mayank Kumar 2 1 Asst. Prof.ECE, RIMT Bareilly, 2 Asst. Prof.ECE, RIMT Bareilly ABSTRACT Despite enormous advances in digital communication, the voice is still
More informationMASTER'S THESIS. Speech Compression and Tone Detection in a Real-Time System. Kristina Berglund. MSc Programmes in Engineering
2004:003 CIV MASTER'S THESIS Speech Compression and Tone Detection in a Real-Time System Kristina Berglund MSc Programmes in Engineering Department of Computer Science and Electrical Engineering Division
More informationProblem Sheet 1 Probability, random processes, and noise
Problem Sheet 1 Probability, random processes, and noise 1. If F X (x) is the distribution function of a random variable X and x 1 x 2, show that F X (x 1 ) F X (x 2 ). 2. Use the definition of the cumulative
More informationPattern Recognition. Part 6: Bandwidth Extension. Gerhard Schmidt
Pattern Recognition Part 6: Gerhard Schmidt Christian-Albrechts-Universität zu Kiel Faculty of Engineering Institute of Electrical and Information Engineering Digital Signal Processing and System Theory
More informationDigital Communication (650533) CH 3 Pulse Modulation
Philadelphia University/Faculty of Engineering Communication and Electronics Engineering Digital Communication (650533) CH 3 Pulse Modulation Instructor: Eng. Nada Khatib Website: http://www.philadelphia.edu.jo/academics/nkhatib/
More informationPage 0 of 23. MELP Vocoder
Page 0 of 23 MELP Vocoder Outline Introduction MELP Vocoder Features Algorithm Description Parameters & Comparison Page 1 of 23 Introduction Traditional pitched-excited LPC vocoders use either a periodic
More informationProject 0: Part 2 A second hands-on lab on Speech Processing Frequency-domain processing
Project : Part 2 A second hands-on lab on Speech Processing Frequency-domain processing February 24, 217 During this lab, you will have a first contact on frequency domain analysis of speech signals. You
More informationOverview of Digital Signal Processing
Overview of Digital Signal Processing Chapter Intended Learning Outcomes: (i) Understand basic terminology in digital signal processing (ii) Differentiate digital signal processing and analog signal processing
More informationComplex Sounds. Reading: Yost Ch. 4
Complex Sounds Reading: Yost Ch. 4 Natural Sounds Most sounds in our everyday lives are not simple sinusoidal sounds, but are complex sounds, consisting of a sum of many sinusoids. The amplitude and frequency
More informationPerception of pitch. Definitions. Why is pitch important? BSc Audiology/MSc SHS Psychoacoustics wk 5: 12 Feb A. Faulkner.
Perception of pitch BSc Audiology/MSc SHS Psychoacoustics wk 5: 12 Feb 2009. A. Faulkner. See Moore, BCJ Introduction to the Psychology of Hearing, Chapter 5. Or Plack CJ The Sense of Hearing Lawrence
More informationMicrocomputer Systems 1. Introduction to DSP S
Microcomputer Systems 1 Introduction to DSP S Introduction to DSP s Definition: DSP Digital Signal Processing/Processor It refers to: Theoretical signal processing by digital means (subject of ECE3222,
More informationAp A ril F RRL RRL P ro r gra r m By Dick AH6EZ/W9
April 2013 FRRL Program By Dick AH6EZ/W9 Why Digital Voice? Data speed or RF bandwidth reduction Transmission by shared digital media such as T1s Security and encryption PCM or ADPCM first US Patent in
More informationQUESTION BANK EC 1351 DIGITAL COMMUNICATION YEAR / SEM : III / VI UNIT I- PULSE MODULATION PART-A (2 Marks) 1. What is the purpose of sample and hold
QUESTION BANK EC 1351 DIGITAL COMMUNICATION YEAR / SEM : III / VI UNIT I- PULSE MODULATION PART-A (2 Marks) 1. What is the purpose of sample and hold circuit 2. What is the difference between natural sampling
More informationChapter 4. Digital Audio Representation CS 3570
Chapter 4. Digital Audio Representation CS 3570 1 Objectives Be able to apply the Nyquist theorem to understand digital audio aliasing. Understand how dithering and noise shaping are done. Understand the
More information1) The modulation technique used for mobile communication systems during world war II was a. Amplitude modulation b. Frequency modulation
1) The modulation technique used for mobile communication systems during world war II was a. Amplitude modulation b. Frequency modulation c. ASK d. FSK ANSWER: Frequency modulation 2) introduced Frequency
More informationITM 1010 Computer and Communication Technologies
ITM 1010 Computer and Communication Technologies Lecture #20 Review: Communication Technologies 2003 香港中文大學, 電子工程學系 (Prof. H.K.Tsang) ITM 1010 計算機與通訊技術 1 Review of Communication Technologies! Information
More informationOpen Access Improved Frame Error Concealment Algorithm Based on Transform- Domain Mobile Audio Codec
Send Orders for Reprints to reprints@benthamscience.ae The Open Electrical & Electronic Engineering Journal, 2014, 8, 527-535 527 Open Access Improved Frame Error Concealment Algorithm Based on Transform-
More informationSystems for Audio and Video Broadcasting (part 2 of 2)
Systems for Audio and Video Broadcasting (part 2 of 2) Ing. Karel Ulovec, Ph.D. CTU in Prague, Faculty of Electrical Engineering xulovec@fel.cvut.cz Only for study purposes for students of the! 1/30 Systems
More informationSpeech Processing. Undergraduate course code: LASC10061 Postgraduate course code: LASC11065
Speech Processing Undergraduate course code: LASC10061 Postgraduate course code: LASC11065 All course materials and handouts are the same for both versions. Differences: credits (20 for UG, 10 for PG);
More informationNonuniform multi level crossing for signal reconstruction
6 Nonuniform multi level crossing for signal reconstruction 6.1 Introduction In recent years, there has been considerable interest in level crossing algorithms for sampling continuous time signals. Driven
More informationPulse Code Modulation
Pulse Code Modulation Modulation is the process of varying one or more parameters of a carrier signal in accordance with the instantaneous values of the message signal. The message signal is the signal
More informationDilpreet Singh 1, Parminder Singh 2 1 M.Tech. Student, 2 Associate Professor
A Novel Approach for Waveform Compression Dilpreet Singh 1, Parminder Singh 2 1 M.Tech. Student, 2 Associate Professor CSE Department, Guru Nanak Dev Engineering College, Ludhiana Abstract Waveform Compression
More informationPerception of pitch. Definitions. Why is pitch important? BSc Audiology/MSc SHS Psychoacoustics wk 4: 7 Feb A. Faulkner.
Perception of pitch BSc Audiology/MSc SHS Psychoacoustics wk 4: 7 Feb 2008. A. Faulkner. See Moore, BCJ Introduction to the Psychology of Hearing, Chapter 5. Or Plack CJ The Sense of Hearing Lawrence Erlbaum,
More informationE : Lecture 8 Source-Filter Processing. E : Lecture 8 Source-Filter Processing / 21
E85.267: Lecture 8 Source-Filter Processing E85.267: Lecture 8 Source-Filter Processing 21-4-1 1 / 21 Source-filter analysis/synthesis n f Spectral envelope Spectral envelope Analysis Source signal n 1
More informationAdvanced audio analysis. Martin Gasser
Advanced audio analysis Martin Gasser Motivation Which methods are common in MIR research? How can we parameterize audio signals? Interesting dimensions of audio: Spectral/ time/melody structure, high
More informationVoice mail and office automation
Voice mail and office automation by DOUGLAS L. HOGAN SPARTA, Incorporated McLean, Virginia ABSTRACT Contrary to expectations of a few years ago, voice mail or voice messaging technology has rapidly outpaced
More informationClass 4 ((Communication and Computer Networks))
Class 4 ((Communication and Computer Networks)) Lesson 5... SIGNAL ENCODING TECHNIQUES Abstract Both analog and digital information can be encoded as either analog or digital signals. The particular encoding
More informationSILK Speech Codec. TDP 10/11 Xavier Anguera I Ciro Gracia
SILK Speech Codec TDP 10/11 Xavier Anguera I Ciro Gracia SILK Codec Audio codec desenvolupat per Skype (Febrer 2009) Previament usaven el codec SVOPC (Sinusoidal Voice Over Packet Coder): LPC analysis.
More informationCHAPTER 4. PULSE MODULATION Part 2
CHAPTER 4 PULSE MODULATION Part 2 Pulse Modulation Analog pulse modulation: Sampling, i.e., information is transmitted only at discrete time instants. e.g. PAM, PPM and PDM Digital pulse modulation: Sampling
More informationRealization and Performance Evaluation of New Hybrid Speech Compression Technique
Realization and Performance Evaluation of New Hybrid Speech Compression Technique Javaid A. Sheikh Post Graduate Department of Electronics & IT University of Kashmir Srinagar, India E-mail: sjavaid_29ku@yahoo.co.in
More informationSignal Characteristics
Data Transmission The successful transmission of data depends upon two factors:» The quality of the transmission signal» The characteristics of the transmission medium Some type of transmission medium
More informationL19: Prosodic modification of speech
L19: Prosodic modification of speech Time-domain pitch synchronous overlap add (TD-PSOLA) Linear-prediction PSOLA Frequency-domain PSOLA Sinusoidal models Harmonic + noise models STRAIGHT This lecture
More informationTime division multiplexing The block diagram for TDM is illustrated as shown in the figure
CHAPTER 2 Syllabus: 1) Pulse amplitude modulation 2) TDM 3) Wave form coding techniques 4) PCM 5) Quantization noise and SNR 6) Robust quantization Pulse amplitude modulation In pulse amplitude modulation,
More informationEvaluation of Audio Compression Artifacts M. Herrera Martinez
Evaluation of Audio Compression Artifacts M. Herrera Martinez This paper deals with subjective evaluation of audio-coding systems. From this evaluation, it is found that, depending on the type of signal
More information