Mel-Generalized Cepstral Representation of Speech A Unified Approach to Speech Spectral Estimation. Keiichi Tokuda

Size: px
Start display at page:

Download "Mel-Generalized Cepstral Representation of Speech A Unified Approach to Speech Spectral Estimation. Keiichi Tokuda"

Transcription

1 Mel-Generalized Cepstral Representation of Speech A Unified Approach to Speech Spectral Estimation Keiichi Tokuda Nagoya Institute of Technology Carnegie Mellon University Tamkang University March 13,

2 Conventional Speech Spectral Estimation Linear prediction (LPC) Cepstral analysis Subband filter bank Autoregressive (AR) model Exponential (EX) model Nonparametric Variations Model Pole-zero (ARMA) model Analysis window Adaptive analysis (sample by sample basis) Auditory characteristics Warped LPC, PLP, etc. Auditory frequency scales (mel, Bark) Loudness scales (log, sone) 2

3 Structure of This Talk 1. Conventional cepstral analysis 2. Introduction of generalized logarithmic function Generalized cepstral analysis 3. Introduction of auditory frequency scale Mel-generalized cepstral analysis 4. Applications to speech recognition and coding 3

4 History of Cepstral Analysis B.P. Bogert, M.J.R. Healy, J.W. Tukey (1963) Analysis of seismic signals decomposition into direct wave and echo Cepstrum, Quefrency, Lifter A.M. Noll (1964, 1967) Pitch extraction based on cepstrum A.V. Oppenheim (1966, 1968) Homomorphic deconvolution decomposition into source and vocal tract function Complex cepstrum 4

5 Definition of Cepstrum Fourier transform of signal s(n) Cepstrum S(e jω )=F [ s(n) ] C(m) =F 1 [ log S(e jω ) 2 ] (Bogert et al., Noll) C(m) =F 1 [ log S(e jω ) ] (Oppenheim) 5

6 Complex Cepstrum z-transform of signal s(n) Complex cepstrum S(z) =Z [ s(n) ] c(m) = Z 1 [ log S(z) ] = F 1 [ log S(e jω ) ] = F 1 [ log S(e jω ) + j arg S(e jω ) ] 6

7 Cepstrum and Complex Cepstrum log S(e jω ) = F [ C(m) ] =Re[ F [ c(m) ]] When it is minimum phase (all poles and zeros are located in the unit circle) c(m) = 0, m < 0 C(m), m =0 2C(m), m > 0 7

8 Homomorphic Deconvolution Pulse train White noise e(n) Linear time-invariant system h(n) s(n) =e(n) h(n) Speech s(n) =h(n) e(n) F S(e jω )=H(e jω )E(e jω ) log log S(e jω ) =log H(e jω ) +log E(e jω ) F 1 C(m) =C h (m)+c e (m) 8

9 Amplitude (db) C(m) s(n) Time (100μs) (a) Quefrency (100μs) (b) Frequency (khz) (d) Amplitude (db) Amplitude (db) Frequency (khz) (c) Frequency (khz) (e) 9

10 Spectrum of Periodic Signal e(n) w(n) log E w (e jω ) W(e jω ) 2π/N p N p 0 π Frequency e(n) h(n) s(n) log S w (e jω ) H(e jω ) 0 π Frequency 10

11 Problems of Homomorphic Processing (Cepstral Analysis) Linear smoothing of log spectrum affected by fine structure of FFT spectrum results in a large bias and variance Voiced speech (periodic) Envelope of peaks of spectral fine structure Improved cepstral analysis, PSE: Biased 11

12 Cost Function P (ω): I N (ω): Estimate of Power Spectrum Periodogram E = 1 2π π π { IN (ω) P (ω) log I } N(ω) P (ω) 1 dω min x: Gaussian Process Maximizing p(x c) Unbiased estimation of log spectrum equivalent to one used in LPC Minimization of energy of inverse filter output 12

13 Analysis of Natural Speech Log magnitude (db) Log magnitude (db) Frequency (khz) (a) Unbiased cepstral analysis Frequency (khz) (b) Linear prediction 13

14 Generalized Cepstrum Complex Cepstrum c(m) = Z 1 [ log S(z) ] log S(z) = Z [ c(m) ] Generalized Cepstrum c γ (m) = Z 1 [ s γ (S(z))] s γ (S(z)) = Z [ c γ (m) ] 14

15 Generalized logarithmic function s γ (w) = (w γ 1)/γ, 0 < γ 1 log w, γ =0 s ( x ) γ γ = < γ < 1 γ = 0 (log x ) 1 < γ < 0 γ = 1 x 15

16 Spectral Model Generalized Cepstrum: c γ (m) H(z) = s 1 γ = M m=0 1+γ exp c γ (m) z m M m=0 M m=0 c γ (m) z m 1/γ, 0 < γ 1 c γ (m) z m, γ =0 Inverse function of Generalized logarithm s 1 γ (w) = (1 + γw) 1/γ, 0 < γ 1 exp w, γ =0 16

17 Cost Function E = 1 2π π π { IN (ω) P (ω) log I } N(ω) P (ω) 1 dω min Estimate of Power Spectrum P (ω) = H(e jω ) 2 = σ 2 D(e jω ) 2 Interpretation in time-domain ε = E [ e 2 (n) ] min x(n) 1/D(z) e(n) 17

18 Advantage 1 γ 0: Convex function Global solutioncan easily be obtained The obtained system H(z) is minimum phase, e.g., stable γ = 1 Linear Prediction H(z) = 1 γ =0 Cepstrum H(z) = exp M m=0 M m=0 1 c γ (m)z m c γ (m)z m 18

19 Prediction Gain D(z) is minimum phase Gain of D(z) is one Predictor: Q(z) = k=1 a(k)z k Cost Function: ε = E [ e 2 (n) ] Prediction Gain: G = E [ x 2 (n) ] E [ e 2 (n) ] x(n) x(n) 1/D(z) Q(z) + e(n) e(n) 19

20 Analysis of synthetic signal (Generalized Cepstral Analysis) Log magnitude 10dB True γ =-1-3/4-1/2-1/ Frequency(Hz) Prediction gain(db) 8 7 (a) Example 1 M= (LPC) (UELS) γ 20

21 Analysis of synthetic signal (Generalized Cepstral Analysis) Log magnitude 10dB True γ =-1-3/4-1/2-1/ Frequency(Hz) Prediction gain(db) 5 4 (c) Example 3 M= (LPC) (UELS) γ 21

22 Analysis of synthetic signal (Generalized Cepstral Analysis) Log magnitude 10dB True γ =-1-3/4-1/2-1/ Frequency(Hz) Prediction gain(db) 7 6 (b) Example 2 M= (LPC) (UELS) γ 22

23 Analysis of natural speech (Generalized Cepstral Analysis) /e/ 80 Magnitude(dB) γ = -1 γ = -1/2 γ = -1/3 γ = Frequency(kHz) Frequency(kHz) Frequency(kHz) Frequency(kHz) Prediction gain(db) M= (LPC) (UELS) γ (a) male /e/ 23

24 Analysis of natural speech (Generalized Cepstral Analysis) /N/ 80 Magnitude(dB) γ = -1 γ = -1/2 γ = -1/3 γ = Frequency(kHz) Frequency(kHz) Frequency(kHz) Frequency(kHz) Prediction gain(db) M= (LPC) (UELS) γ (b) male /N/ 24

25 Structure of synthesis filter H(z) (γ = 1/n) 1st 3nd... n-th input 1 1 σ 1 C(z) C(z) C(z) output H(z) =σd(z) =σ { 1 C(z) } n C(z) = 1+γ M c γ (m) z m m=0 25

26 Structure of synthesis filter H(z) (γ =0) LMA filter Input F (z) F (z) F (z) F (z) A L,1 A L,2 A L,3 A L,4 Output + D(z) = exp F (z) R L (F (z)) = L l=1 L l=1 A L,l {F (z)} l A L,l { F (z)} l F (z) = M m=1 c γ (m) z m 26

27 Introduction of auditory frequency scale First-order all-pass function: z 1 α =Ψ(z) = z 1 α 1 αz 1 Phase Characteristics can be used for Frequency Transformation: ω = tan 1 (1 α 2 )sinω (1 + α 2 ) cos ω 2α where Ψ(e jω )=e j ω Warped frequency ω (rad) π π /2 mel scale 10kHz sampling α = π/2 π Frequency ω (rad) 27

28 Mel-Generalized Cepstral Analysis Mel-generalized cepstrum: c α,γ (m) H(z) = s 1 γ = M m=0 1+γ exp c α,γ (m) z m M m=0 M m=0 α c α,γ (m) z m α 1/γ, 0 < γ 1 c α,γ (m) z m α, γ =0 z 1 α = z 1 α 1 αz 1 28

29 (α, γ) =(0, 0) Cepstral model: H(z) = exp (α, γ) =(0, 1) AR model: M m=0 c α,γ (m)z m H(z) = 1 M m=0 1 c α,γ (m)z m (α, γ) =(0.35, 0) Mel-cepstral model: H(z) = exp M m=0 c α,γ (m) z m α (α, γ) =(0.47, 1) Warped AR model: H(z) = 1 M m=0 1 c α,γ (m) z m α 29

30 A Unified Approach to Speech Spectral Estimation α < 1, 1 γ 0 α =0 Generalized cepstral analysis Mel-generalized cepstral analysis γ = 1 Linear prediction Warped Linear Prediction γ =0 Unbiased Cepstral analysis Mel-cepstral analysis 30

31 Mel-generalized analysis of natural speech /N/ M =12 Magnitude(dB) Magnitude(dB) Magnitude(dB) α = 0 α = 0.35 α = Frequency(kHz) Frequency(kHz) Frequency(kHz) γ = 0 γ = 0.5 γ = 1 31

32 Example α =0 γ = 1 γ = 1/3 γ =0 Frequency(kHz) Frequency(kHz) Frequency(kHz) n a N b u d e w a (ms) (a) Waveform 40(dB) 40(dB) (α, γ, M) = (0, -1, 12) LP (α, γ, M) = (0, -1/3, 12) GCEP 40(dB) (α, γ, M) = (0, 0, 12) UELS (b) Spectral estimates ( α = 0) 32

33 Example α =0.35 γ = 1 γ = 1/3 γ =0 Frequency(kHz) Frequency(kHz) Frequency(kHz) n a N b u d e w a (ms) (a) Waveform 40(dB) 40(dB) (α, γ, M) = (0.35, -1, 12) WLP (α, γ, M) = (0.35, -1/3, 12) MGCEP 40(dB) (α, γ, M) = (0.35, 0, 12) MCEP (b) Spectral estimates ( α = 0.35) 33

34 Structure of synthesis filter H(z) (γ = 1/n) 1st 2nd... n-th input 1 1 σ 1 C(z) C(z) C(z) output Structure of H(z) H(z) =σd(z) =σ C(z) = ( 1+γ M m=0 { 1 C(z) } n c α,γ(m) z m α Output ) Input + α z 1 + z 1 + z 1 + z 1 γ(1 α 2 ) + b(1) + α + b(2) Structure of C(z) (M =3) + α b(3) 34

35 Structure of synthesis filter H(z) (γ =0) MLSA filter D(z) = exp F (z) R L (F (z)) F (z) = M m=0 c α,γ(m) z m α sufficient accuracy: maximum spectral error 0.24dB O(8M) multiply-add operations a sample guaranteed stability M multiply-add operations for filter coefficients calculation 35

36 Structure of MLSA filter Input F (z) F (z) F (z) F (z) A L,1 A L,2 A L,3 A L,4 Output + R L (F (z)) exp F (z) =D(z), L =4 Input 1 α 2 α - + α α z 1 + z 1 + z 1 + z b(1) b(2) b(3) + Basic filter F (z), M =3 + Output 36

37 The choice of α, γ for speech analysis/synthesis Analysis/synthesis system with fixed α and γ speech quality change with γ γ 1 Clear γ 0 Smooth When γ = 0, speech quality with (α, M) =(0.35, 15) is almost equivalent to that with (α,m) =(0, 30). When the analysis order is high enough, the difference becomes small. 37

38 Feature of Unified Approach Linear prediction analysis, Cepstral analysis are the special cases. Mathematically well-defined Physical interpretation Minimization of energy of inverse filter output x is Gaussian Minimization of p(x c) (ML estimation) Global solution, stability of the system function Synthesis filter for direct synthesis from the estimated coefficients LMA/MLSA/GMSLA filter Extension to adaptive analysis (sample by sample basis) Parameter transformation for speech recognition 38

39 Word Recognition based on HMM Spectral Analysis: 1. (α 1,γ 1,M 1 )=(0, 1, 12) Linear Prediction 2. (α 1,γ 1,M 1 )=(0.35, 1/3, 12) Mel-generalized cepstral analysis 3. (α 1,γ 1,M 1 )=(0.35, 0, 12) Mel-cepstral analysis Output vector of HMM: (α 2,γ 2,M 2 )=(0.35, 0, 12) Mel-cepstral coefficients and Δ (dynamic coefficients) H(z) =s 1 γ 1 M 1 m=0 c α1,γ 1 (m) z m α 1 = s 1 γ 2 m=0 c α2,γ 2 (m) z m α 2 39

40 Recognition Accuracy (Continuous HMM, 33 phoneme models, 2618 words) Linear Prediction (α 1, γ 1, M 1)= (0, -1, 12) Mel-Generalized Cepstral Analysis (α 1, γ 1, M 1)= (0.35, -1/3, 12) Mel-Cepstral Analysis (α 1, γ 1, M 1)= (0.35, 0, 12) 77.5% 80.7% 79.6% Recognition accuracy (%) 40

41 Application to 16kb/s wideband CELP coder Input Quantization of MGC Coef. MGC Analysis encoder CELP Excitation Generator Minimize MSE Synthesis Filter MGC coef. MGC coef. Perceptual Weighting Filter MGC coef. MGC coef. decoder CELP Excitation Generator Synthesis Filter Postfilter Output 41

42 Speech quality as a function of α (γ = 1/2) D M O S 3.5 average female male α 42

43 Subjective Evaluation MNRU(dB) G kb/s G kb/s G kb/s Conv.CELP 16kb/s MGC-CELP 16kb/s D M O S average female male 43

44 Summary A unified approach to speech spectral estimation A unified approach to Linear predicton and Cepstral analysis Introduction of auditory frequency scale Efficint representation of speech spectrum with an appropriate choice of α and γ Application to speech anaylysis/synthesis, speech coding, speech recognition Future work: Optimal α and γ (Phoneme/Speaker dependent?) Speech Signal Processing Toolkit: 44

Feature extraction 2

Feature extraction 2 Centre for Vision Speech & Signal Processing University of Surrey, Guildford GU2 7XH. Feature extraction 2 Dr Philip Jackson Linear prediction Perceptual linear prediction Comparison of feature methods

More information

Signal representations: Cepstrum

Signal representations: Cepstrum Signal representations: Cepstrum Source-filter separation for sound production For speech, source corresponds to excitation by a pulse train for voiced phonemes and to turbulence (noise) for unvoiced phonemes,

More information

CEPSTRAL ANALYSIS SYNTHESIS ON THE MEL FREQUENCY SCALE, AND AN ADAPTATIVE ALGORITHM FOR IT.

CEPSTRAL ANALYSIS SYNTHESIS ON THE MEL FREQUENCY SCALE, AND AN ADAPTATIVE ALGORITHM FOR IT. CEPSTRAL ANALYSIS SYNTHESIS ON THE EL FREQUENCY SCALE, AND AN ADAPTATIVE ALGORITH FOR IT. Summarized overview of the IEEE-publicated papers Cepstral analysis synthesis on the mel frequency scale by Satochi

More information

Linear Prediction 1 / 41

Linear Prediction 1 / 41 Linear Prediction 1 / 41 A map of speech signal processing Natural signals Models Artificial signals Inference Speech synthesis Hidden Markov Inference Homomorphic processing Dereverberation, Deconvolution

More information

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

Chapter 9. Linear Predictive Analysis of Speech Signals 语音信号的线性预测分析 Chapter 9 Linear Predictive Analysis of Speech Signals 语音信号的线性预测分析 1 LPC Methods LPC methods are the most widely used in speech coding, speech synthesis, speech recognition, speaker recognition and verification

More information

Feature extraction 1

Feature extraction 1 Centre for Vision Speech & Signal Processing University of Surrey, Guildford GU2 7XH. Feature extraction 1 Dr Philip Jackson Cepstral analysis - Real & complex cepstra - Homomorphic decomposition Filter

More information

Voiced Speech. Unvoiced Speech

Voiced Speech. Unvoiced Speech Digital Speech Processing Lecture 2 Homomorphic Speech Processing General Discrete-Time Model of Speech Production p [ n] = p[ n] h [ n] Voiced Speech L h [ n] = A g[ n] v[ n] r[ n] V V V p [ n ] = u [

More information

Speech Signal Representations

Speech Signal Representations Speech Signal Representations Berlin Chen 2003 References: 1. X. Huang et. al., Spoken Language Processing, Chapters 5, 6 2. J. R. Deller et. al., Discrete-Time Processing of Speech Signals, Chapters 4-6

More information

Signal Modeling Techniques in Speech Recognition. Hassan A. Kingravi

Signal Modeling Techniques in Speech Recognition. Hassan A. Kingravi Signal Modeling Techniques in Speech Recognition Hassan A. Kingravi Outline Introduction Spectral Shaping Spectral Analysis Parameter Transforms Statistical Modeling Discussion Conclusions 1: Introduction

More information

L7: Linear prediction of speech

L7: Linear prediction of speech L7: Linear prediction of speech Introduction Linear prediction Finding the linear prediction coefficients Alternative representations This lecture is based on [Dutoit and Marques, 2009, ch1; Taylor, 2009,

More information

Automatic Speech Recognition (CS753)

Automatic Speech Recognition (CS753) Automatic Speech Recognition (CS753) Lecture 12: Acoustic Feature Extraction for ASR Instructor: Preethi Jyothi Feb 13, 2017 Speech Signal Analysis Generate discrete samples A frame Need to focus on short

More information

Frequency Domain Speech Analysis

Frequency Domain Speech Analysis Frequency Domain Speech Analysis Short Time Fourier Analysis Cepstral Analysis Windowed (short time) Fourier Transform Spectrogram of speech signals Filter bank implementation* (Real) cepstrum and complex

More information

SPEECH ANALYSIS AND SYNTHESIS

SPEECH ANALYSIS AND SYNTHESIS 16 Chapter 2 SPEECH ANALYSIS AND SYNTHESIS 2.1 INTRODUCTION: Speech signal analysis is used to characterize the spectral information of an input speech signal. Speech signal analysis [52-53] techniques

More information

Topic 6. Timbre Representations

Topic 6. Timbre Representations Topic 6 Timbre Representations We often say that singer s voice is magnetic the violin sounds bright this French horn sounds solid that drum sounds dull What aspect(s) of sound are these words describing?

More information

Department of Electrical and Computer Engineering Digital Speech Processing Homework No. 6 Solutions

Department of Electrical and Computer Engineering Digital Speech Processing Homework No. 6 Solutions Problem 1 Department of Electrical and Computer Engineering Digital Speech Processing Homework No. 6 Solutions The complex cepstrum, ˆx[n], of a sequence x[n] is the inverse Fourier transform of the complex

More information

CS578- Speech Signal Processing

CS578- Speech Signal Processing CS578- Speech Signal Processing Lecture 7: Speech Coding Yannis Stylianou University of Crete, Computer Science Dept., Multimedia Informatics Lab yannis@csd.uoc.gr Univ. of Crete Outline 1 Introduction

More information

Chapter 10 Applications in Communications

Chapter 10 Applications in Communications Chapter 10 Applications in Communications School of Information Science and Engineering, SDU. 1/ 47 Introduction Some methods for digitizing analog waveforms: Pulse-code modulation (PCM) Differential PCM

More information

Vocoding approaches for statistical parametric speech synthesis

Vocoding approaches for statistical parametric speech synthesis Vocoding approaches for statistical parametric speech synthesis Ranniery Maia Toshiba Research Europe Limited Cambridge Research Laboratory Speech Synthesis Seminar Series CUED, University of Cambridge,

More information

Linear Prediction Coding. Nimrod Peleg Update: Aug. 2007

Linear Prediction Coding. Nimrod Peleg Update: Aug. 2007 Linear Prediction Coding Nimrod Peleg Update: Aug. 2007 1 Linear Prediction and Speech Coding The earliest papers on applying LPC to speech: Atal 1968, 1970, 1971 Markel 1971, 1972 Makhoul 1975 This is

More information

Sinusoidal Modeling. Yannis Stylianou SPCC University of Crete, Computer Science Dept., Greece,

Sinusoidal Modeling. Yannis Stylianou SPCC University of Crete, Computer Science Dept., Greece, Sinusoidal Modeling Yannis Stylianou University of Crete, Computer Science Dept., Greece, yannis@csd.uoc.gr SPCC 2016 1 Speech Production 2 Modulators 3 Sinusoidal Modeling Sinusoidal Models Voiced Speech

More information

LECTURE NOTES IN AUDIO ANALYSIS: PITCH ESTIMATION FOR DUMMIES

LECTURE NOTES IN AUDIO ANALYSIS: PITCH ESTIMATION FOR DUMMIES LECTURE NOTES IN AUDIO ANALYSIS: PITCH ESTIMATION FOR DUMMIES Abstract March, 3 Mads Græsbøll Christensen Audio Analysis Lab, AD:MT Aalborg University This document contains a brief introduction to pitch

More information

Lab 9a. Linear Predictive Coding for Speech Processing

Lab 9a. Linear Predictive Coding for Speech Processing EE275Lab October 27, 2007 Lab 9a. Linear Predictive Coding for Speech Processing Pitch Period Impulse Train Generator Voiced/Unvoiced Speech Switch Vocal Tract Parameters Time-Varying Digital Filter H(z)

More information

Class of waveform coders can be represented in this manner

Class of waveform coders can be represented in this manner Digital Speech Processing Lecture 15 Speech Coding Methods Based on Speech Waveform Representations ti and Speech Models Uniform and Non- Uniform Coding Methods 1 Analog-to-Digital Conversion (Sampling

More information

Speaker Identification Based On Discriminative Vector Quantization And Data Fusion

Speaker Identification Based On Discriminative Vector Quantization And Data Fusion University of Central Florida Electronic Theses and Dissertations Doctoral Dissertation (Open Access) Speaker Identification Based On Discriminative Vector Quantization And Data Fusion 2005 Guangyu Zhou

More information

Zeros of z-transform(zzt) representation and chirp group delay processing for analysis of source and filter characteristics of speech signals

Zeros of z-transform(zzt) representation and chirp group delay processing for analysis of source and filter characteristics of speech signals Zeros of z-transformzzt representation and chirp group delay processing for analysis of source and filter characteristics of speech signals Baris Bozkurt 1 Collaboration with LIMSI-CNRS, France 07/03/2017

More information

SPEECH RECOGNITION USING TIME DOMAIN FEATURES FROM PHASE SPACE RECONSTRUCTIONS

SPEECH RECOGNITION USING TIME DOMAIN FEATURES FROM PHASE SPACE RECONSTRUCTIONS SPEECH RECOGNITION USING TIME DOMAIN FEATURES FROM PHASE SPACE RECONSTRUCTIONS by Jinjin Ye, B.S. A THESIS SUBMITTED TO THE FACULTY OF THE GRADUATE SCHOOL IN PARTIAL FULFILLMENT OF THE REQUIREMENTS for

More information

Source modeling (block processing)

Source modeling (block processing) Digital Speech Processing Lecture 17 Speech Coding Methods Based on Speech Models 1 Waveform Coding versus Block Waveform coding Processing sample-by-sample matching of waveforms coding gquality measured

More information

L8: Source estimation

L8: Source estimation L8: Source estimation Glottal and lip radiation models Closed-phase residual analysis Voicing/unvoicing detection Pitch detection Epoch detection This lecture is based on [Taylor, 2009, ch. 11-12] Introduction

More information

Lecture 9: Speech Recognition. Recognizing Speech

Lecture 9: Speech Recognition. Recognizing Speech EE E68: Speech & Audio Processing & Recognition Lecture 9: Speech Recognition 3 4 Recognizing Speech Feature Calculation Sequence Recognition Hidden Markov Models Dan Ellis http://www.ee.columbia.edu/~dpwe/e68/

More information

The Equivalence of ADPCM and CELP Coding

The Equivalence of ADPCM and CELP Coding The Equivalence of ADPCM and CELP Coding Peter Kabal Department of Electrical & Computer Engineering McGill University Montreal, Canada Version.2 March 20 c 20 Peter Kabal 20/03/ You are free: to Share

More information

Lecture 9: Speech Recognition

Lecture 9: Speech Recognition EE E682: Speech & Audio Processing & Recognition Lecture 9: Speech Recognition 1 2 3 4 Recognizing Speech Feature Calculation Sequence Recognition Hidden Markov Models Dan Ellis

More information

Modeling Prosody for Speaker Recognition: Why Estimating Pitch May Be a Red Herring

Modeling Prosody for Speaker Recognition: Why Estimating Pitch May Be a Red Herring Modeling Prosody for Speaker Recognition: Why Estimating Pitch May Be a Red Herring Kornel Laskowski & Qin Jin Carnegie Mellon University Pittsburgh PA, USA 28 June, 2010 Laskowski & Jin ODYSSEY 2010,

More information

Text-to-speech synthesizer based on combination of composite wavelet and hidden Markov models

Text-to-speech synthesizer based on combination of composite wavelet and hidden Markov models 8th ISCA Speech Synthesis Workshop August 31 September 2, 2013 Barcelona, Spain Text-to-speech synthesizer based on combination of composite wavelet and hidden Markov models Nobukatsu Hojo 1, Kota Yoshizato

More information

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

Nonparametric and Parametric Defined This text distinguishes between systems and the sequences (processes) that result when a WN input is applied Linear Signal Models Overview Introduction Linear nonparametric vs. parametric models Equivalent representations Spectral flatness measure PZ vs. ARMA models Wold decomposition Introduction Many researchers

More information

Evaluation of the modified group delay feature for isolated word recognition

Evaluation of the modified group delay feature for isolated word recognition Evaluation of the modified group delay feature for isolated word recognition Author Alsteris, Leigh, Paliwal, Kuldip Published 25 Conference Title The 8th International Symposium on Signal Processing and

More information

Lecture 7: Feature Extraction

Lecture 7: Feature Extraction Lecture 7: Feature Extraction Kai Yu SpeechLab Department of Computer Science & Engineering Shanghai Jiao Tong University Autumn 2014 Kai Yu Lecture 7: Feature Extraction SJTU Speech Lab 1 / 28 Table of

More information

An Autoregressive Recurrent Mixture Density Network for Parametric Speech Synthesis

An Autoregressive Recurrent Mixture Density Network for Parametric Speech Synthesis ICASSP 07 New Orleans, USA An Autoregressive Recurrent Mixture Density Network for Parametric Speech Synthesis Xin WANG, Shinji TAKAKI, Junichi YAMAGISHI National Institute of Informatics, Japan 07-03-07

More information

Design of a CELP coder and analysis of various quantization techniques

Design of a CELP coder and analysis of various quantization techniques EECS 65 Project Report Design of a CELP coder and analysis of various quantization techniques Prof. David L. Neuhoff By: Awais M. Kamboh Krispian C. Lawrence Aditya M. Thomas Philip I. Tsai Winter 005

More information

Quadrature-Mirror Filter Bank

Quadrature-Mirror Filter Bank Quadrature-Mirror Filter Bank In many applications, a discrete-time signal x[n] is split into a number of subband signals { v k [ n]} by means of an analysis filter bank The subband signals are then processed

More information

Allpass Modeling of LP Residual for Speaker Recognition

Allpass Modeling of LP Residual for Speaker Recognition Allpass Modeling of LP Residual for Speaker Recognition K. Sri Rama Murty, Vivek Boominathan and Karthika Vijayan Department of Electrical Engineering, Indian Institute of Technology Hyderabad, India email:

More information

Design Criteria for the Quadratically Interpolated FFT Method (I): Bias due to Interpolation

Design Criteria for the Quadratically Interpolated FFT Method (I): Bias due to Interpolation CENTER FOR COMPUTER RESEARCH IN MUSIC AND ACOUSTICS DEPARTMENT OF MUSIC, STANFORD UNIVERSITY REPORT NO. STAN-M-4 Design Criteria for the Quadratically Interpolated FFT Method (I): Bias due to Interpolation

More information

Time-domain representations

Time-domain representations Time-domain representations Speech Processing Tom Bäckström Aalto University Fall 2016 Basics of Signal Processing in the Time-domain Time-domain signals Before we can describe speech signals or modelling

More information

CEPSTRAL analysis has been widely used in signal processing

CEPSTRAL analysis has been widely used in signal processing 162 IEEE TRANSACTIONS ON SPEECH AND AUDIO PROCESSING, VOL. 7, NO. 2, MARCH 1999 On Second-Order Statistics and Linear Estimation of Cepstral Coefficients Yariv Ephraim, Fellow, IEEE, and Mazin Rahim, Senior

More information

Reformulating the HMM as a trajectory model by imposing explicit relationship between static and dynamic features

Reformulating the HMM as a trajectory model by imposing explicit relationship between static and dynamic features Reformulating the HMM as a trajectory model by imposing explicit relationship between static and dynamic features Heiga ZEN (Byung Ha CHUN) Nagoya Inst. of Tech., Japan Overview. Research backgrounds 2.

More information

Pulse-Code Modulation (PCM) :

Pulse-Code Modulation (PCM) : PCM & DPCM & DM 1 Pulse-Code Modulation (PCM) : In PCM each sample of the signal is quantized to one of the amplitude levels, where B is the number of bits used to represent each sample. The rate from

More information

Optimal Design of Real and Complex Minimum Phase Digital FIR Filters

Optimal Design of Real and Complex Minimum Phase Digital FIR Filters Optimal Design of Real and Complex Minimum Phase Digital FIR Filters Niranjan Damera-Venkata and Brian L. Evans Embedded Signal Processing Laboratory Dept. of Electrical and Computer Engineering The University

More information

Audio Coding. Fundamentals Quantization Waveform Coding Subband Coding P NCTU/CSIE DSPLAB C.M..LIU

Audio Coding. Fundamentals Quantization Waveform Coding Subband Coding P NCTU/CSIE DSPLAB C.M..LIU Audio Coding P.1 Fundamentals Quantization Waveform Coding Subband Coding 1. Fundamentals P.2 Introduction Data Redundancy Coding Redundancy Spatial/Temporal Redundancy Perceptual Redundancy Compression

More information

ECE 8440 Unit Applica,ons (of Homomorphic Deconvolu,on) to Speech Processing

ECE 8440 Unit Applica,ons (of Homomorphic Deconvolu,on) to Speech Processing ECE 8440 Unit 24 1 13.2 Applica,ons (of Homomorphic Deconvolu,on) to Speech Processing Speech produc,on can be modeled as the convolu,on of an excita,on signal with the unit sample response of a linear

More information

6.003 (Fall 2011) Quiz #3 November 16, 2011

6.003 (Fall 2011) Quiz #3 November 16, 2011 6.003 (Fall 2011) Quiz #3 November 16, 2011 Name: Kerberos Username: Please circle your section number: Section Time 2 11 am 3 1 pm 4 2 pm Grades will be determined by the correctness of your answers (explanations

More information

Practical Spectral Estimation

Practical Spectral Estimation Digital Signal Processing/F.G. Meyer Lecture 4 Copyright 2015 François G. Meyer. All Rights Reserved. Practical Spectral Estimation 1 Introduction The goal of spectral estimation is to estimate how the

More information

Review of Fundamentals of Digital Signal Processing

Review of Fundamentals of Digital Signal Processing Solution Manual for Theory and Applications of Digital Speech Processing by Lawrence Rabiner and Ronald Schafer Click here to Purchase full Solution Manual at http://solutionmanuals.info Link download

More information

Singer Identification using MFCC and LPC and its comparison for ANN and Naïve Bayes Classifiers

Singer Identification using MFCC and LPC and its comparison for ANN and Naïve Bayes Classifiers Singer Identification using MFCC and LPC and its comparison for ANN and Naïve Bayes Classifiers Kumari Rambha Ranjan, Kartik Mahto, Dipti Kumari,S.S.Solanki Dept. of Electronics and Communication Birla

More information

Parametric Signal Modeling and Linear Prediction Theory 1. Discrete-time Stochastic Processes (cont d)

Parametric Signal Modeling and Linear Prediction Theory 1. Discrete-time Stochastic Processes (cont d) Parametric Signal Modeling and Linear Prediction Theory 1. Discrete-time Stochastic Processes (cont d) Electrical & Computer Engineering North Carolina State University Acknowledgment: ECE792-41 slides

More information

Chirp Decomposition of Speech Signals for Glottal Source Estimation

Chirp Decomposition of Speech Signals for Glottal Source Estimation Chirp Decomposition of Speech Signals for Glottal Source Estimation Thomas Drugman 1, Baris Bozkurt 2, Thierry Dutoit 1 1 TCTS Lab, Faculté Polytechnique de Mons, Belgium 2 Department of Electrical & Electronics

More information

Discrete-Time Signals and Systems. Frequency Domain Analysis of LTI Systems. The Frequency Response Function. The Frequency Response Function

Discrete-Time Signals and Systems. Frequency Domain Analysis of LTI Systems. The Frequency Response Function. The Frequency Response Function Discrete-Time Signals and s Frequency Domain Analysis of LTI s Dr. Deepa Kundur University of Toronto Reference: Sections 5., 5.2-5.5 of John G. Proakis and Dimitris G. Manolakis, Digital Signal Processing:

More information

DFT & Fast Fourier Transform PART-A. 7. Calculate the number of multiplications needed in the calculation of DFT and FFT with 64 point sequence.

DFT & Fast Fourier Transform PART-A. 7. Calculate the number of multiplications needed in the calculation of DFT and FFT with 64 point sequence. SHRI ANGALAMMAN COLLEGE OF ENGINEERING & TECHNOLOGY (An ISO 9001:2008 Certified Institution) SIRUGANOOR,TRICHY-621105. DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING UNIT I DFT & Fast Fourier

More information

Random signals II. ÚPGM FIT VUT Brno,

Random signals II. ÚPGM FIT VUT Brno, Random signals II. Jan Černocký ÚPGM FIT VUT Brno, cernocky@fit.vutbr.cz 1 Temporal estimate of autocorrelation coefficients for ergodic discrete-time random process. ˆR[k] = 1 N N 1 n=0 x[n]x[n + k],

More information

Glottal Source Estimation using an Automatic Chirp Decomposition

Glottal Source Estimation using an Automatic Chirp Decomposition Glottal Source Estimation using an Automatic Chirp Decomposition Thomas Drugman 1, Baris Bozkurt 2, Thierry Dutoit 1 1 TCTS Lab, Faculté Polytechnique de Mons, Belgium 2 Department of Electrical & Electronics

More information

1. Probability density function for speech samples. Gamma. Laplacian. 2. Coding paradigms. =(2X max /2 B ) for a B-bit quantizer Δ Δ Δ Δ Δ

1. Probability density function for speech samples. Gamma. Laplacian. 2. Coding paradigms. =(2X max /2 B ) for a B-bit quantizer Δ Δ Δ Δ Δ Digital Speech Processing Lecture 16 Speech Coding Methods Based on Speech Waveform Representations and Speech Models Adaptive and Differential Coding 1 Speech Waveform Coding-Summary of Part 1 1. Probability

More information

Noise Robust Isolated Words Recognition Problem Solving Based on Simultaneous Perturbation Stochastic Approximation Algorithm

Noise Robust Isolated Words Recognition Problem Solving Based on Simultaneous Perturbation Stochastic Approximation Algorithm EngOpt 2008 - International Conference on Engineering Optimization Rio de Janeiro, Brazil, 0-05 June 2008. Noise Robust Isolated Words Recognition Problem Solving Based on Simultaneous Perturbation Stochastic

More information

Proc. of NCC 2010, Chennai, India

Proc. of NCC 2010, Chennai, India Proc. of NCC 2010, Chennai, India Trajectory and surface modeling of LSF for low rate speech coding M. Deepak and Preeti Rao Department of Electrical Engineering Indian Institute of Technology, Bombay

More information

An Excitation Model for HMM-Based Speech Synthesis Based on Residual Modeling

An Excitation Model for HMM-Based Speech Synthesis Based on Residual Modeling An Model for HMM-Based Speech Synthesis Based on Residual Modeling Ranniery Maia,, Tomoki Toda,, Heiga Zen, Yoshihiko Nankaku, Keiichi Tokuda, National Inst. of Inform. and Comm. Technology (NiCT), Japan

More information

Efficient Block Quantisation for Image and Speech Coding

Efficient Block Quantisation for Image and Speech Coding Efficient Block Quantisation for Image and Speech Coding Stephen So, BEng (Hons) School of Microelectronic Engineering Faculty of Engineering and Information Technology Griffith University, Brisbane, Australia

More information

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

Finite Word Length Effects and Quantisation Noise. Professors A G Constantinides & L R Arnaut Finite Word Length Effects and Quantisation Noise 1 Finite Word Length Effects Finite register lengths and A/D converters cause errors at different levels: (i) input: Input quantisation (ii) system: Coefficient

More information

Gaussian Processes for Audio Feature Extraction

Gaussian Processes for Audio Feature Extraction Gaussian Processes for Audio Feature Extraction Dr. Richard E. Turner (ret26@cam.ac.uk) Computational and Biological Learning Lab Department of Engineering University of Cambridge Machine hearing pipeline

More information

CHAPTER 2 RANDOM PROCESSES IN DISCRETE TIME

CHAPTER 2 RANDOM PROCESSES IN DISCRETE TIME CHAPTER 2 RANDOM PROCESSES IN DISCRETE TIME Shri Mata Vaishno Devi University, (SMVDU), 2013 Page 13 CHAPTER 2 RANDOM PROCESSES IN DISCRETE TIME When characterizing or modeling a random variable, estimates

More information

Estimation of Cepstral Coefficients for Robust Speech Recognition

Estimation of Cepstral Coefficients for Robust Speech Recognition Estimation of Cepstral Coefficients for Robust Speech Recognition by Kevin M. Indrebo, B.S., M.S. A Dissertation submitted to the Faculty of the Graduate School, Marquette University, in Partial Fulfillment

More information

Exemplar-based voice conversion using non-negative spectrogram deconvolution

Exemplar-based voice conversion using non-negative spectrogram deconvolution Exemplar-based voice conversion using non-negative spectrogram deconvolution Zhizheng Wu 1, Tuomas Virtanen 2, Tomi Kinnunen 3, Eng Siong Chng 1, Haizhou Li 1,4 1 Nanyang Technological University, Singapore

More information

AUTOREGRESSIVE (AR) modeling identifies and exploits

AUTOREGRESSIVE (AR) modeling identifies and exploits IEEE TRANSACTIONS ON SIGNAL PROCESSING, VOL. 55, NO. 11, NOVEMBER 2007 5237 Autoregressive Modeling of Temporal Envelopes Marios Athineos, Student Member, IEEE, and Daniel P. W. Ellis, Senior Member, IEEE

More information

Multidimensional digital signal processing

Multidimensional digital signal processing PSfrag replacements Two-dimensional discrete signals N 1 A 2-D discrete signal (also N called a sequence or array) is a function 2 defined over thex(n set 1 of, n 2 ordered ) pairs of integers: y(nx 1,

More information

Signal Modeling Techniques In Speech Recognition

Signal Modeling Techniques In Speech Recognition Picone: Signal Modeling... 1 Signal Modeling Techniques In Speech Recognition by, Joseph Picone Texas Instruments Systems and Information Sciences Laboratory Tsukuba Research and Development Center Tsukuba,

More information

Nonparametric Deconvolution of Seismic Depth Phases

Nonparametric Deconvolution of Seismic Depth Phases IMA Workshop Time Series Analysis and Applications to Geophysical Systems November 1-16, 1 Minneapolis, Minnesota Nonparametric Deconvolution of Seismic Depth Phases Robert H. Shumway 1 Jessie L. Bonner

More information

[Omer* et al., 5(7): July, 2016] ISSN: IC Value: 3.00 Impact Factor: 4.116

[Omer* et al., 5(7): July, 2016] ISSN: IC Value: 3.00 Impact Factor: 4.116 IJESRT INTERNATIONAL JOURNAL OF ENGINEERING SCIENCES & RESEARCH TECHNOLOGY TAJWEED UTOMATION SYSTEM USING HIDDEN MARKOUV MODEL AND NURAL NETWORK Safaa Omer Mohammed Nssr*, Hoida Ali Abdelgader SUDAN UNIVERSITY

More information

Discrete-Time David Johns and Ken Martin University of Toronto

Discrete-Time David Johns and Ken Martin University of Toronto Discrete-Time David Johns and Ken Martin University of Toronto (johns@eecg.toronto.edu) (martin@eecg.toronto.edu) University of Toronto 1 of 40 Overview of Some Signal Spectra x c () t st () x s () t xn

More information

Blind Deconvolution of Ultrasonic Signals Using High-Order Spectral Analysis and Wavelets

Blind Deconvolution of Ultrasonic Signals Using High-Order Spectral Analysis and Wavelets Blind Deconvolution of Ultrasonic Signals Using High-Order Spectral Analysis and Wavelets Roberto H. Herrera, Eduardo Moreno, Héctor Calas, and Rubén Orozco 3 University of Cienfuegos, Cuatro Caminos,

More information

Lecture 5: GMM Acoustic Modeling and Feature Extraction

Lecture 5: GMM Acoustic Modeling and Feature Extraction CS 224S / LINGUIST 285 Spoken Language Processing Andrew Maas Stanford University Spring 2017 Lecture 5: GMM Acoustic Modeling and Feature Extraction Original slides by Dan Jurafsky Outline for Today Acoustic

More information

Bich Ngoc Do. Neural Networks for Automatic Speaker, Language and Sex Identification

Bich Ngoc Do. Neural Networks for Automatic Speaker, Language and Sex Identification Charles University in Prague Faculty of Mathematics and Physics University of Groningen Faculty of Arts MASTER THESIS Bich Ngoc Do Neural Networks for Automatic Speaker, Language and Sex Identification

More information

/ (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

/ (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 Code No: RR320402 Set No. 1 III B.Tech II Semester Regular Examinations, Apr/May 2006 DIGITAL SIGNAL PROCESSING ( Common to Electronics & Communication Engineering, Electronics & Instrumentation Engineering,

More information

Speech Coding. Speech Processing. Tom Bäckström. October Aalto University

Speech Coding. Speech Processing. Tom Bäckström. October Aalto University Speech Coding Speech Processing Tom Bäckström Aalto University October 2015 Introduction Speech coding refers to the digital compression of speech signals for telecommunication (and storage) applications.

More information

CS 188: Artificial Intelligence Fall 2011

CS 188: Artificial Intelligence Fall 2011 CS 188: Artificial Intelligence Fall 2011 Lecture 20: HMMs / Speech / ML 11/8/2011 Dan Klein UC Berkeley Today HMMs Demo bonanza! Most likely explanation queries Speech recognition A massive HMM! Details

More information

EE538 Final Exam Fall :20 pm -5:20 pm PHYS 223 Dec. 17, Cover Sheet

EE538 Final Exam Fall :20 pm -5:20 pm PHYS 223 Dec. 17, Cover Sheet EE538 Final Exam Fall 005 3:0 pm -5:0 pm PHYS 3 Dec. 17, 005 Cover Sheet Test Duration: 10 minutes. Open Book but Closed Notes. Calculators ARE allowed!! This test contains five problems. Each of the five

More information

INTRODUCTION Noise is present in many situations of daily life for ex: Microphones will record noise and speech. Goal: Reconstruct original signal Wie

INTRODUCTION Noise is present in many situations of daily life for ex: Microphones will record noise and speech. Goal: Reconstruct original signal Wie WIENER FILTERING Presented by N.Srikanth(Y8104060), M.Manikanta PhaniKumar(Y8104031). INDIAN INSTITUTE OF TECHNOLOGY KANPUR Electrical Engineering dept. INTRODUCTION Noise is present in many situations

More information

Lecture 7 Discrete Systems

Lecture 7 Discrete Systems Lecture 7 Discrete Systems EE 52: Instrumentation and Measurements Lecture Notes Update on November, 29 Aly El-Osery, Electrical Engineering Dept., New Mexico Tech 7. Contents The z-transform 2 Linear

More information

Part III Spectrum Estimation

Part III Spectrum Estimation ECE79-4 Part III Part III Spectrum Estimation 3. Parametric Methods for Spectral Estimation Electrical & Computer Engineering North Carolina State University Acnowledgment: ECE79-4 slides were adapted

More information

representation of speech

representation of speech Digital Speech Processing Lectures 7-8 Time Domain Methods in Speech Processing 1 General Synthesis Model voiced sound amplitude Log Areas, Reflection Coefficients, Formants, Vocal Tract Polynomial, l

More information

On Variable-Scale Piecewise Stationary Spectral Analysis of Speech Signals for ASR

On Variable-Scale Piecewise Stationary Spectral Analysis of Speech Signals for ASR On Variable-Scale Piecewise Stationary Spectral Analysis of Speech Signals for ASR Vivek Tyagi a,c, Hervé Bourlard b,c Christian Wellekens a,c a Institute Eurecom, P.O Box: 193, Sophia-Antipolis, France.

More information

Digital Signal Processing. Midterm 1 Solution

Digital Signal Processing. Midterm 1 Solution EE 123 University of California, Berkeley Anant Sahai February 15, 27 Digital Signal Processing Instructions Midterm 1 Solution Total time allowed for the exam is 8 minutes Some useful formulas: Discrete

More information

On Moving Average Parameter Estimation

On Moving Average Parameter Estimation On Moving Average Parameter Estimation Niclas Sandgren and Petre Stoica Contact information: niclas.sandgren@it.uu.se, tel: +46 8 473392 Abstract Estimation of the autoregressive moving average (ARMA)

More information

ADSP ADSP ADSP ADSP. Advanced Digital Signal Processing (18-792) Spring Fall Semester, Department of Electrical and Computer Engineering

ADSP ADSP ADSP ADSP. Advanced Digital Signal Processing (18-792) Spring Fall Semester, Department of Electrical and Computer Engineering Advanced Digital Signal rocessing (18-792) Spring Fall Semester, 201 2012 Department of Electrical and Computer Engineering ROBLEM SET 8 Issued: 10/26/18 Due: 11/2/18 Note: This problem set is due Friday,

More information

Voice Activity Detection Using Pitch Feature

Voice Activity Detection Using Pitch Feature Voice Activity Detection Using Pitch Feature Presented by: Shay Perera 1 CONTENTS Introduction Related work Proposed Improvement References Questions 2 PROBLEM speech Non speech Speech Region Non Speech

More information

Correspondence. Pulse Doppler Radar Target Recognition using a Two-Stage SVM Procedure

Correspondence. Pulse Doppler Radar Target Recognition using a Two-Stage SVM Procedure Correspondence Pulse Doppler Radar Target Recognition using a Two-Stage SVM Procedure It is possible to detect and classify moving and stationary targets using ground surveillance pulse-doppler radars

More information

ETSI EN V7.1.1 ( )

ETSI EN V7.1.1 ( ) European Standard (Telecommunications series) Digital cellular telecommunications system (Phase +); Adaptive Multi-Rate (AMR) speech transcoding GLOBAL SYSTEM FOR MOBILE COMMUNICATIONS R Reference DEN/SMG-110690Q7

More information

Statistical and Adaptive Signal Processing

Statistical and Adaptive Signal Processing r Statistical and Adaptive Signal Processing Spectral Estimation, Signal Modeling, Adaptive Filtering and Array Processing Dimitris G. Manolakis Massachusetts Institute of Technology Lincoln Laboratory

More information

Massachusetts Institute of Technology Department of Electrical Engineering and Computer Science : Discrete-Time Signal Processing

Massachusetts Institute of Technology Department of Electrical Engineering and Computer Science : Discrete-Time Signal Processing Massachusetts Institute of Technology Department of Electrical Engineering and Computer Science 6.34: Discrete-Time Signal Processing OpenCourseWare 006 ecture 8 Periodogram Reading: Sections 0.6 and 0.7

More information

EE 602 TERM PAPER PRESENTATION Richa Tripathi Mounika Boppudi FOURIER SERIES BASED MODEL FOR STATISTICAL SIGNAL PROCESSING - CHONG YUNG CHI

EE 602 TERM PAPER PRESENTATION Richa Tripathi Mounika Boppudi FOURIER SERIES BASED MODEL FOR STATISTICAL SIGNAL PROCESSING - CHONG YUNG CHI EE 602 TERM PAPER PRESENTATION Richa Tripathi Mounika Boppudi FOURIER SERIES BASED MODEL FOR STATISTICAL SIGNAL PROCESSING - CHONG YUNG CHI ABSTRACT The goal of the paper is to present a parametric Fourier

More information

CS 343: Artificial Intelligence

CS 343: Artificial Intelligence CS 343: Artificial Intelligence Particle Filters and Applications of HMMs Prof. Scott Niekum The University of Texas at Austin [These slides based on those of Dan Klein and Pieter Abbeel for CS188 Intro

More information

-Digital Signal Processing- FIR Filter Design. Lecture May-16

-Digital Signal Processing- FIR Filter Design. Lecture May-16 -Digital Signal Processing- FIR Filter Design Lecture-17 24-May-16 FIR Filter Design! FIR filters can also be designed from a frequency response specification.! The equivalent sampled impulse response

More information

EE 5345 Biomedical Instrumentation Lecture 12: slides

EE 5345 Biomedical Instrumentation Lecture 12: slides EE 5345 Biomedical Instrumentation Lecture 1: slides 4-6 Carlos E. Davila, Electrical Engineering Dept. Southern Methodist University slides can be viewed at: http:// www.seas.smu.edu/~cd/ee5345.html EE

More information

Stress detection through emotional speech analysis

Stress detection through emotional speech analysis Stress detection through emotional speech analysis INMA MOHINO inmaculada.mohino@uah.edu.es ROBERTO GIL-PITA roberto.gil@uah.es LORENA ÁLVAREZ PÉREZ loreduna88@hotmail Abstract: Stress is a reaction or

More information

Linear models. Chapter Overview. Linear process: A process {X n } is a linear process if it has the representation.

Linear models. Chapter Overview. Linear process: A process {X n } is a linear process if it has the representation. Chapter 2 Linear models 2.1 Overview Linear process: A process {X n } is a linear process if it has the representation X n = b j ɛ n j j=0 for all n, where ɛ n N(0, σ 2 ) (Gaussian distributed with zero

More information