Discriminative Training for Automatic Speech Recognition

Size: px
Start display at page:

Download "Discriminative Training for Automatic Speech Recognition"

Transcription

1 Discriminative Training for Automatic Speech Recognition 22 nd April 2013 Advanced Signal Processing Seminar

2 Article Heigold, G.; Ney, H.; Schluter, R.; Wiesler, S. Signal Processing Magazine, IEEE, vol.29, no.6, pp.58,69, Nov Covered Topics Statistical Speech Recognition Discriminative Training Criteria Parameter Models Optimisation Implementation Experimental Results Summary and Outlook

3 4 Components of an LVCSR System from [6]

4 Features Usable with discriminative Training Short-term power spectrum Mel frequency cepstral coefficients (MFCC) Perceptual linear prediction (PLP) Further enhancement methods (LDA,...) Other Approaches Feature Extraction through Neural Networks

5 Problem Given Sequence of Feature-Vectors x1 T Find Sequence of Words/Phonemes w1 N Statistical Model [ ] w1 T opt = argmax w N 1 p(w N 1 x T 1 )

6 Problem Statistical Model cont. [ ] w1 T opt = argmax w N 1 = argmax w N 1 = argmax w N 1 = argmax w N 1 p(w N 1 x T 1 ) p(x T 1, w N 1 ) p(x T 1 ) }{{} indep. of w N 1 p(x T 1, w N 1 ) p(x T 1 w N 1 ) p(w N 1 )

7 Problem Statistical Model cont. [ ] w1 T opt = argmax w N 1 p(x1 T w1 N ) }{{} p(w1 N ) }{{} Acoustic Model Language Model

8 Standard Acoustic Model Gaussian Hidden Markov Model p(x T 1 w N 1 ) = = = p(x1 T, s1 T ) S1 T HMM(w 1 N) T p(x t s t ) p(s t s t 1 ) S1 T HMM(w 1 N) t=1 T L s p(s t s t 1 ) c st,ln (x t µ st,l, Σ st,l) S1 T HMM(w 1 N) t=1 l=1 Parameterset for GHMM: p(s1 T 1 st 0 ), c s T 1,l Ls, µ 1 s T 1,l Ls, Σ 1 s T 1,l Ls 1 Λ

9 Maximum Likelihood Standard Approach Find the most probable parameters for the model given the training data set (x T 1, w N 1 ). Training Criterion F (Λ) = log p Λ (x T 1, w N 1 )

10 Maximum Mutual Information (MMI) Motivation Maximizes directly the posterior probability Takes also all competing sentences w N 1 Training Criterion into account F (Λ) = log p Λ (w N 1 x T 1 ) = log p Λ (x T 1, w N 1 ) w N 1 p Λ(x T 1, w N 1 )

11 Maximum Mutual Information (MMI) MMI vs. ML from [1]

12 Minimum Classification Error (MCE) Motivation Assumption that GMMs are not the real distribution Aims to maximize the classification error (WER) Derived by minimizing the expected loss Training Criterion F (Λ) = σ β (log ) p Λ (x1 T, w 1 N ) w 1 N w 1 N p Λ(x1 T, w 1 N)

13 Minimum Phone Error (MPE) Motivation Similar motivation as for MCE Aims to minimize the phone error rate (Levenshtein-Distance) Hypotheses weighted by phone accuracy A( w N 1, w N 1 ) Training Criterion F (Λ) = w N 1 p Λ ( w 1 N x1 T )A( w 1 N, w1 N ) w N 1

14 I-smoothing Overfitting Problem Discriminative training criteria prone to overfitting Critical on less training data Introducing a prior to each Gaussian based on the ML statistics Essentially for MCE/MPE

15 Margin Term Focus on decision boundary Generalization reached through closest training samples to the decision boundary (Margin, SVM) Focus more on these samples by adding a term to the training criterion: exp( A( w N 1, w N 1 )) Applied on MMI: equal to boosted MMI (BMMI)

16 Language Model Scale HMM p(x T 1 w N 1 ) = S T 1 HMM(w N 1 ) p(x T 1, s T 1 ) 1 κ Language Model p(w N 1 ) = p(w N 1 ) 1 κ : language model scale

17 Feature Transform Features containing more information Each training criterion can also be used in the feature space. for instance: y t = x t + M h t

18 Comparison Unified Training Criterion ( w 1 F (Λ) = f N p Λ(x1 T, w 1 N)A( w 1 N, w 1 N ) ) w 1 N p Λ(x1 T, w 1 N)B( w 1 N, w 1 N) MMI setup f (x) = log(x) A( w N 1, w N 1 ) = δ( w N 1, w N 1 ) B( w N 1, w N 1 ) = 1

19 Comparison Binary Classification from [1]

20 Extended Baum-Welch Strong-sense auxillary function G(λ, λ ) G(λ, λ ) F(λ) F(λ ) Principe of Expectation Maximization Algorithm Weak-sense auxillary function λ G(λ, λ ) λ=λ = λ F(λ) λ=λ Principe of Extended Baum-Welch Algorithm

21 Rprop Properties Only sign of the partial derivatives is needed Separate step size for each parameter Simple heuristic Good alternative to EBW Roughly the same number of iterations as EBW for convergence on conservative initial step size

22 Experimental Results EBW vs. Rprop from [1]

23 Experimental Results Training Criteria from [1]

24 Experimental Results Margin Term from [1]

25 References [1] Heigold, G.; Ney, H.; Schluter, R.; Wiesler, S., : Modeling, Criteria, Optimization, Implementation, and Performance, Signal Processing Magazine, IEEE, vol.29, no.6, pp.58,69, Nov [2] Povey, D.; Woodland, P.C.; Gales, M. J F, Discriminative map for acoustic model adaptation, Acoustics, Speech, and Signal Processing, Proceedings. (ICASSP 03) IEEE International Conference on, vol.1, no., pp.i-312,i-315 vol.1, 6-10 April 2003 [3] Povey, D.; Woodland, P.C., Minimum Phone Error and I-smoothing for improved discriminative training, Acoustics, Speech, and Signal Processing (ICASSP), 2002 IEEE International Conference on, vol.1, no., pp.i-105,i-108, May 2002 [4] Biing-Hwang Juang; Wu Hou; Chin-Hui Lee, Minimum classification error rate methods for speech recognition, Speech and Audio Processing, IEEE Transactions on, vol.5, no.3, pp.257,265, May 1997 [5] Bahl, L.; Brown, P.; De Souza, P.V.; Mercer, R., Maximum mutual information estimation of hidden Markov model parameters for speech recognition, Acoustics, Speech, and Signal Processing, IEEE International Conference on ICASSP 86., vol.11, no., pp.49,52, Apr 1986 [6] Saon, G.; Jen-Tzung Chien, Large-Vocabulary Continuous Speech Recognition Systems: A Look at Some Recent Advances, Signal Processing Magazine, IEEE, vol.29, no.6, pp.18,33, Nov [7] Hermansky, Hynek. Perceptual linear predictive (PLP) analysis of speech. The Journal of the Acoustical Society of America 87 (1990): [8] Hermansky, H.; Ellis, D.P.W.; Sharma, S., Tandem connectionist feature extraction for conventional HMM systems, Acoustics, Speech, and Signal Processing, ICASSP 00. Proceedings IEEE International Conference on, vol.3, no., pp.1635,1638 vol.3, 2000

Using RASTA in task independent TANDEM feature extraction

Using RASTA in task independent TANDEM feature extraction R E S E A R C H R E P O R T I D I A P Using RASTA in task independent TANDEM feature extraction Guillermo Aradilla a John Dines a Sunil Sivadas a b IDIAP RR 04-22 April 2004 D a l l e M o l l e I n s t

More information

High-speed Noise Cancellation with Microphone Array

High-speed Noise Cancellation with Microphone Array Noise Cancellation a Posteriori Probability, Maximum Criteria Independent Component Analysis High-speed Noise Cancellation with Microphone Array We propose the use of a microphone array based on independent

More information

AN ANALYSIS OF SPEECH RECOGNITION PERFORMANCE BASED UPON NETWORK LAYERS AND TRANSFER FUNCTIONS

AN ANALYSIS OF SPEECH RECOGNITION PERFORMANCE BASED UPON NETWORK LAYERS AND TRANSFER FUNCTIONS AN ANALYSIS OF SPEECH RECOGNITION PERFORMANCE BASED UPON NETWORK LAYERS AND TRANSFER FUNCTIONS Kuldeep Kumar 1, R. K. Aggarwal 1 and Ankita Jain 2 1 Department of Computer Engineering, National Institute

More information

FEATURE COMBINATION AND STACKING OF RECURRENT AND NON-RECURRENT NEURAL NETWORKS FOR LVCSR

FEATURE COMBINATION AND STACKING OF RECURRENT AND NON-RECURRENT NEURAL NETWORKS FOR LVCSR FEATURE COMBINATION AND STACKING OF RECURRENT AND NON-RECURRENT NEURAL NETWORKS FOR LVCSR Christian Plahl 1, Michael Kozielski 1, Ralf Schlüter 1 and Hermann Ney 1,2 1 Human Language Technology and Pattern

More information

Reverse Correlation for analyzing MLP Posterior Features in ASR

Reverse Correlation for analyzing MLP Posterior Features in ASR Reverse Correlation for analyzing MLP Posterior Features in ASR Joel Pinto, G.S.V.S. Sivaram, and Hynek Hermansky IDIAP Research Institute, Martigny École Polytechnique Fédérale de Lausanne (EPFL), Switzerland

More information

Acoustic modelling from the signal domain using CNNs

Acoustic modelling from the signal domain using CNNs Acoustic modelling from the signal domain using CNNs Pegah Ghahremani 1, Vimal Manohar 1, Daniel Povey 1,2, Sanjeev Khudanpur 1,2 1 Center of Language and Speech Processing 2 Human Language Technology

More information

Mikko Myllymäki and Tuomas Virtanen

Mikko Myllymäki and Tuomas Virtanen NON-STATIONARY NOISE MODEL COMPENSATION IN VOICE ACTIVITY DETECTION Mikko Myllymäki and Tuomas Virtanen Department of Signal Processing, Tampere University of Technology Korkeakoulunkatu 1, 3370, Tampere,

More information

Joint recognition and direction-of-arrival estimation of simultaneous meetingroom acoustic events

Joint recognition and direction-of-arrival estimation of simultaneous meetingroom acoustic events INTERSPEECH 2013 Joint recognition and direction-of-arrival estimation of simultaneous meetingroom acoustic events Rupayan Chakraborty and Climent Nadeu TALP Research Centre, Department of Signal Theory

More information

Automatic Morse Code Recognition Under Low SNR

Automatic Morse Code Recognition Under Low SNR 2nd International Conference on Mechanical, Electronic, Control and Automation Engineering (MECAE 2018) Automatic Morse Code Recognition Under Low SNR Xianyu Wanga, Qi Zhaob, Cheng Mac, * and Jianping

More information

Speech Synthesis using Mel-Cepstral Coefficient Feature

Speech 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 information

Applications of Music Processing

Applications of Music Processing Lecture Music Processing Applications of Music Processing Christian Dittmar International Audio Laboratories Erlangen christian.dittmar@audiolabs-erlangen.de Singing Voice Detection Important pre-requisite

More information

I D I A P. Hierarchical and Parallel Processing of Modulation Spectrum for ASR applications Fabio Valente a and Hynek Hermansky a

I D I A P. Hierarchical and Parallel Processing of Modulation Spectrum for ASR applications Fabio Valente a and Hynek Hermansky a R E S E A R C H R E P O R T I D I A P Hierarchical and Parallel Processing of Modulation Spectrum for ASR applications Fabio Valente a and Hynek Hermansky a IDIAP RR 07-45 January 2008 published in ICASSP

More information

IMPROVING WIDEBAND SPEECH RECOGNITION USING MIXED-BANDWIDTH TRAINING DATA IN CD-DNN-HMM

IMPROVING WIDEBAND SPEECH RECOGNITION USING MIXED-BANDWIDTH TRAINING DATA IN CD-DNN-HMM IMPROVING WIDEBAND SPEECH RECOGNITION USING MIXED-BANDWIDTH TRAINING DATA IN CD-DNN-HMM Jinyu Li, Dong Yu, Jui-Ting Huang, and Yifan Gong Microsoft Corporation, One Microsoft Way, Redmond, WA 98052 ABSTRACT

More information

Audio Similarity. Mark Zadel MUMT 611 March 8, Audio Similarity p.1/23

Audio Similarity. Mark Zadel MUMT 611 March 8, Audio Similarity p.1/23 Audio Similarity Mark Zadel MUMT 611 March 8, 2004 Audio Similarity p.1/23 Overview MFCCs Foote Content-Based Retrieval of Music and Audio (1997) Logan, Salomon A Music Similarity Function Based On Signal

More information

LEVERAGING JOINTLY SPATIAL, TEMPORAL AND MODULATION ENHANCEMENT IN CREATING NOISE-ROBUST FEATURES FOR SPEECH RECOGNITION

LEVERAGING JOINTLY SPATIAL, TEMPORAL AND MODULATION ENHANCEMENT IN CREATING NOISE-ROBUST FEATURES FOR SPEECH RECOGNITION LEVERAGING JOINTLY SPATIAL, TEMPORAL AND MODULATION ENHANCEMENT IN CREATING NOISE-ROBUST FEATURES FOR SPEECH RECOGNITION 1 HSIN-JU HSIEH, 2 HAO-TENG FAN, 3 JEIH-WEIH HUNG 1,2,3 Dept of Electrical Engineering,

More information

Dimension Reduction of the Modulation Spectrogram for Speaker Verification

Dimension Reduction of the Modulation Spectrogram for Speaker Verification Dimension Reduction of the Modulation Spectrogram for Speaker Verification Tomi Kinnunen Speech and Image Processing Unit Department of Computer Science University of Joensuu, Finland Kong Aik Lee and

More information

ANALYSIS-BY-SYNTHESIS FEATURE ESTIMATION FOR ROBUST AUTOMATIC SPEECH RECOGNITION USING SPECTRAL MASKS. Michael I Mandel and Arun Narayanan

ANALYSIS-BY-SYNTHESIS FEATURE ESTIMATION FOR ROBUST AUTOMATIC SPEECH RECOGNITION USING SPECTRAL MASKS. Michael I Mandel and Arun Narayanan ANALYSIS-BY-SYNTHESIS FEATURE ESTIMATION FOR ROBUST AUTOMATIC SPEECH RECOGNITION USING SPECTRAL MASKS Michael I Mandel and Arun Narayanan The Ohio State University, Computer Science and Engineering {mandelm,narayaar}@cse.osu.edu

More information

SONG RETRIEVAL SYSTEM USING HIDDEN MARKOV MODELS

SONG RETRIEVAL SYSTEM USING HIDDEN MARKOV MODELS SONG RETRIEVAL SYSTEM USING HIDDEN MARKOV MODELS AKSHAY CHANDRASHEKARAN ANOOP RAMAKRISHNA akshayc@cmu.edu anoopr@andrew.cmu.edu ABHISHEK JAIN GE YANG ajain2@andrew.cmu.edu younger@cmu.edu NIDHI KOHLI R

More information

SEMANTIC ANNOTATION AND RETRIEVAL OF MUSIC USING A BAG OF SYSTEMS REPRESENTATION

SEMANTIC ANNOTATION AND RETRIEVAL OF MUSIC USING A BAG OF SYSTEMS REPRESENTATION SEMANTIC ANNOTATION AND RETRIEVAL OF MUSIC USING A BAG OF SYSTEMS REPRESENTATION Katherine Ellis University of California, San Diego kellis@ucsd.edu Emanuele Coviello University of California, San Diego

More information

I D I A P. On Factorizing Spectral Dynamics for Robust Speech Recognition R E S E A R C H R E P O R T. Iain McCowan a Hemant Misra a,b

I D I A P. On Factorizing Spectral Dynamics for Robust Speech Recognition R E S E A R C H R E P O R T. Iain McCowan a Hemant Misra a,b R E S E A R C H R E P O R T I D I A P On Factorizing Spectral Dynamics for Robust Speech Recognition a Vivek Tyagi Hervé Bourlard a,b IDIAP RR 3-33 June 23 Iain McCowan a Hemant Misra a,b to appear in

More information

Performance Analysis of MFCC and LPCC Techniques in Automatic Speech Recognition

Performance Analysis of MFCC and LPCC Techniques in Automatic Speech Recognition www.ijecs.in International Journal Of Engineering And Computer Science ISSN:2319-7242 Volume - 3 Issue - 8 August, 2014 Page No. 7727-7732 Performance Analysis of MFCC and LPCC Techniques in Automatic

More information

Classification of ships using autocorrelation technique for feature extraction of the underwater acoustic noise

Classification of ships using autocorrelation technique for feature extraction of the underwater acoustic noise Classification of ships using autocorrelation technique for feature extraction of the underwater acoustic noise Noha KORANY 1 Alexandria University, Egypt ABSTRACT The paper applies spectral analysis to

More information

IEEE TRANSACTIONS ON AUDIO, SPEECH, AND LANGUAGE PROCESSING, VOL. 19, NO. 8, NOVEMBER

IEEE TRANSACTIONS ON AUDIO, SPEECH, AND LANGUAGE PROCESSING, VOL. 19, NO. 8, NOVEMBER IEEE TRANSACTIONS ON AUDIO, SPEECH, AND LANGUAGE PROCESSING, VOL. 19, NO. 8, NOVEMBER 2011 2439 Transcribing Mandarin Broadcast Speech Using Multi-Layer Perceptron Acoustic Features Fabio Valente, Member,

More information

24 IEEE TRANSACTIONS ON AUDIO, SPEECH, AND LANGUAGE PROCESSING, VOL. 17, NO. 1, JANUARY /$ IEEE

24 IEEE TRANSACTIONS ON AUDIO, SPEECH, AND LANGUAGE PROCESSING, VOL. 17, NO. 1, JANUARY /$ IEEE 24 IEEE TRANSACTIONS ON AUDIO, SPEECH, AND LANGUAGE PROCESSING, VOL. 17, NO. 1, JANUARY 2009 Speech Enhancement, Gain, and Noise Spectrum Adaptation Using Approximate Bayesian Estimation Jiucang Hao, Hagai

More information

IMPROVEMENTS TO THE IBM SPEECH ACTIVITY DETECTION SYSTEM FOR THE DARPA RATS PROGRAM

IMPROVEMENTS TO THE IBM SPEECH ACTIVITY DETECTION SYSTEM FOR THE DARPA RATS PROGRAM IMPROVEMENTS TO THE IBM SPEECH ACTIVITY DETECTION SYSTEM FOR THE DARPA RATS PROGRAM Samuel Thomas 1, George Saon 1, Maarten Van Segbroeck 2 and Shrikanth S. Narayanan 2 1 IBM T.J. Watson Research Center,

More information

DERIVATION OF TRAPS IN AUDITORY DOMAIN

DERIVATION OF TRAPS IN AUDITORY DOMAIN DERIVATION OF TRAPS IN AUDITORY DOMAIN Petr Motlíček, Doctoral Degree Programme (4) Dept. of Computer Graphics and Multimedia, FIT, BUT E-mail: motlicek@fit.vutbr.cz Supervised by: Dr. Jan Černocký, Prof.

More information

Acoustic Modeling from Frequency-Domain Representations of Speech

Acoustic Modeling from Frequency-Domain Representations of Speech Acoustic Modeling from Frequency-Domain Representations of Speech Pegah Ghahremani 1, Hossein Hadian 1,3, Hang Lv 1,4, Daniel Povey 1,2, Sanjeev Khudanpur 1,2 1 Center of Language and Speech Processing

More information

Calibration of Microphone Arrays for Improved Speech Recognition

Calibration of Microphone Arrays for Improved Speech Recognition MITSUBISHI ELECTRIC RESEARCH LABORATORIES http://www.merl.com Calibration of Microphone Arrays for Improved Speech Recognition Michael L. Seltzer, Bhiksha Raj TR-2001-43 December 2001 Abstract We present

More information

The Automatic Classification Problem. Perceptrons, SVMs, and Friends: Some Discriminative Models for Classification

The Automatic Classification Problem. Perceptrons, SVMs, and Friends: Some Discriminative Models for Classification Perceptrons, SVMs, and Friends: Some Discriminative Models for Classification Parallel to AIMA 8., 8., 8.6.3, 8.9 The Automatic Classification Problem Assign object/event or sequence of objects/events

More information

Robust Speaker Identification for Meetings: UPC CLEAR 07 Meeting Room Evaluation System

Robust Speaker Identification for Meetings: UPC CLEAR 07 Meeting Room Evaluation System Robust Speaker Identification for Meetings: UPC CLEAR 07 Meeting Room Evaluation System Jordi Luque and Javier Hernando Technical University of Catalonia (UPC) Jordi Girona, 1-3 D5, 08034 Barcelona, Spain

More information

Modulation Spectrum Power-law Expansion for Robust Speech Recognition

Modulation Spectrum Power-law Expansion for Robust Speech Recognition Modulation Spectrum Power-law Expansion for Robust Speech Recognition Hao-Teng Fan, Zi-Hao Ye and Jeih-weih Hung Department of Electrical Engineering, National Chi Nan University, Nantou, Taiwan E-mail:

More information

Enhanced voice recognition to reduce fraudulence in ATM machine

Enhanced voice recognition to reduce fraudulence in ATM machine Enhanced voice recognition to reduce fraudulence in ATM machine 1 Hridya Venugopal, Hema.U, Kalaiselvi.S, Mahalakshmi.M Department of Information Technology Alpha college of Engineering Email:hridya.nbr@gmail.com,hemau5490@gmail.com,kalaika3@gmail.com,

More information

Neural Network Acoustic Models for the DARPA RATS Program

Neural Network Acoustic Models for the DARPA RATS Program INTERSPEECH 2013 Neural Network Acoustic Models for the DARPA RATS Program Hagen Soltau, Hong-Kwang Kuo, Lidia Mangu, George Saon, Tomas Beran IBM T. J. Watson Research Center, Yorktown Heights, NY 10598,

More information

An Improved Voice Activity Detection Based on Deep Belief Networks

An Improved Voice Activity Detection Based on Deep Belief Networks e-issn 2455 1392 Volume 2 Issue 4, April 2016 pp. 676-683 Scientific Journal Impact Factor : 3.468 http://www.ijcter.com An Improved Voice Activity Detection Based on Deep Belief Networks Shabeeba T. K.

More information

Statistical Modeling of Speaker s Voice with Temporal Co-Location for Active Voice Authentication

Statistical Modeling of Speaker s Voice with Temporal Co-Location for Active Voice Authentication INTERSPEECH 2016 September 8 12, 2016, San Francisco, USA Statistical Modeling of Speaker s Voice with Temporal Co-Location for Active Voice Authentication Zhong Meng, Biing-Hwang (Fred) Juang School of

More information

The 2010 CMU GALE Speech-to-Text System

The 2010 CMU GALE Speech-to-Text System Research Showcase @ CMU Language Technologies Institute School of Computer Science 9-2 The 2 CMU GALE Speech-to-Text System Florian Metze, fmetze@andrew.cmu.edu Roger Hsiao Qin Jin Udhyakumar Nallasamy

More information

A Correlation-Maximization Denoising Filter Used as An Enhancement Frontend for Noise Robust Bird Call Classification

A Correlation-Maximization Denoising Filter Used as An Enhancement Frontend for Noise Robust Bird Call Classification A Correlation-Maximization Denoising Filter Used as An Enhancement Frontend for Noise Robust Bird Call Classification Wei Chu and Abeer Alwan Speech Processing and Auditory Perception Laboratory Department

More information

Separating Voiced Segments from Music File using MFCC, ZCR and GMM

Separating Voiced Segments from Music File using MFCC, ZCR and GMM Separating Voiced Segments from Music File using MFCC, ZCR and GMM Mr. Prashant P. Zirmite 1, Mr. Mahesh K. Patil 2, Mr. Santosh P. Salgar 3,Mr. Veeresh M. Metigoudar 4 1,2,3,4Assistant Professor, Dept.

More information

Research Article Implementation of a Tour Guide Robot System Using RFID Technology and Viterbi Algorithm-Based HMM for Speech Recognition

Research Article Implementation of a Tour Guide Robot System Using RFID Technology and Viterbi Algorithm-Based HMM for Speech Recognition Mathematical Problems in Engineering, Article ID 262791, 7 pages http://dx.doi.org/10.1155/2014/262791 Research Article Implementation of a Tour Guide Robot System Using RFID Technology and Viterbi Algorithm-Based

More information

SOUND SOURCE RECOGNITION AND MODELING

SOUND SOURCE RECOGNITION AND MODELING SOUND SOURCE RECOGNITION AND MODELING CASA seminar, summer 2000 Antti Eronen antti.eronen@tut.fi Contents: Basics of human sound source recognition Timbre Voice recognition Recognition of environmental

More information

Simultaneous Recognition of Speech Commands by a Robot using a Small Microphone Array

Simultaneous Recognition of Speech Commands by a Robot using a Small Microphone Array 2012 2nd International Conference on Computer Design and Engineering (ICCDE 2012) IPCSIT vol. 49 (2012) (2012) IACSIT Press, Singapore DOI: 10.7763/IPCSIT.2012.V49.14 Simultaneous Recognition of Speech

More information

VQ Source Models: Perceptual & Phase Issues

VQ Source Models: Perceptual & Phase Issues VQ Source Models: Perceptual & Phase Issues Dan Ellis & Ron Weiss Laboratory for Recognition and Organization of Speech and Audio Dept. Electrical Eng., Columbia Univ., NY USA {dpwe,ronw}@ee.columbia.edu

More information

Hierarchical and parallel processing of auditory and modulation frequencies for automatic speech recognition

Hierarchical and parallel processing of auditory and modulation frequencies for automatic speech recognition Available online at www.sciencedirect.com Speech Communication 52 (2010) 790 800 www.elsevier.com/locate/specom Hierarchical and parallel processing of auditory and modulation frequencies for automatic

More information

Robustness (cont.); End-to-end systems

Robustness (cont.); End-to-end systems Robustness (cont.); End-to-end systems Steve Renals Automatic Speech Recognition ASR Lecture 18 27 March 2017 ASR Lecture 18 Robustness (cont.); End-to-end systems 1 Robust Speech Recognition ASR Lecture

More information

An Investigation on the Use of i-vectors for Robust ASR

An Investigation on the Use of i-vectors for Robust ASR An Investigation on the Use of i-vectors for Robust ASR Dimitrios Dimitriadis, Samuel Thomas IBM T.J. Watson Research Center Yorktown Heights, NY 1598 [dbdimitr, sthomas]@us.ibm.com Sriram Ganapathy Department

More information

Relative phase information for detecting human speech and spoofed speech

Relative phase information for detecting human speech and spoofed speech Relative phase information for detecting human speech and spoofed speech Longbiao Wang 1, Yohei Yoshida 1, Yuta Kawakami 1 and Seiichi Nakagawa 2 1 Nagaoka University of Technology, Japan 2 Toyohashi University

More information

SIMULATION VOICE RECOGNITION SYSTEM FOR CONTROLING ROBOTIC APPLICATIONS

SIMULATION VOICE RECOGNITION SYSTEM FOR CONTROLING ROBOTIC APPLICATIONS SIMULATION VOICE RECOGNITION SYSTEM FOR CONTROLING ROBOTIC APPLICATIONS 1 WAHYU KUSUMA R., 2 PRINCE BRAVE GUHYAPATI V 1 Computer Laboratory Staff., Department of Information Systems, Gunadarma University,

More information

Combined Features and Kernel Design for Noise Robust Phoneme Classification Using Support Vector Machines

Combined Features and Kernel Design for Noise Robust Phoneme Classification Using Support Vector Machines 1 Combined Features and Kernel Design for Noise Robust Phoneme Classification Using Support Vector Machines Jibran Yousafzai, Student Member, IEEE Peter Sollich Zoran Cvetković, Senior Member, IEEE Bin

More information

Speech Signal Analysis

Speech 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 information

Audio Augmentation for Speech Recognition

Audio Augmentation for Speech Recognition Audio Augmentation for Speech Recognition Tom Ko 1, Vijayaditya Peddinti 2, Daniel Povey 2,3, Sanjeev Khudanpur 2,3 1 Huawei Noah s Ark Research Lab, Hong Kong, China 2 Center for Language and Speech Processing

More information

Artificial Bandwidth Extension Using Deep Neural Networks for Spectral Envelope Estimation

Artificial Bandwidth Extension Using Deep Neural Networks for Spectral Envelope Estimation Platzhalter für Bild, Bild auf Titelfolie hinter das Logo einsetzen Artificial Bandwidth Extension Using Deep Neural Networks for Spectral Envelope Estimation Johannes Abel and Tim Fingscheidt Institute

More information

UNSUPERVISED SPEAKER CHANGE DETECTION FOR BROADCAST NEWS SEGMENTATION

UNSUPERVISED SPEAKER CHANGE DETECTION FOR BROADCAST NEWS SEGMENTATION 4th European Signal Processing Conference (EUSIPCO 26), Florence, Italy, September 4-8, 26, copyright by EURASIP UNSUPERVISED SPEAKER CHANGE DETECTION FOR BROADCAST NEWS SEGMENTATION Kasper Jørgensen,

More information

Performance analysis of voice activity detection algorithm for robust speech recognition system under different noisy environment

Performance analysis of voice activity detection algorithm for robust speech recognition system under different noisy environment BABU et al: VOICE ACTIVITY DETECTION ALGORITHM FOR ROBUST SPEECH RECOGNITION SYSTEM Journal of Scientific & Industrial Research Vol. 69, July 2010, pp. 515-522 515 Performance analysis of voice activity

More information

Progress in the BBN Keyword Search System for the DARPA RATS Program

Progress in the BBN Keyword Search System for the DARPA RATS Program INTERSPEECH 2014 Progress in the BBN Keyword Search System for the DARPA RATS Program Tim Ng 1, Roger Hsiao 1, Le Zhang 1, Damianos Karakos 1, Sri Harish Mallidi 2, Martin Karafiát 3,KarelVeselý 3, Igor

More information

Single Channel Speaker Segregation using Sinusoidal Residual Modeling

Single Channel Speaker Segregation using Sinusoidal Residual Modeling NCC 2009, January 16-18, IIT Guwahati 294 Single Channel Speaker Segregation using Sinusoidal Residual Modeling Rajesh M Hegde and A. Srinivas Dept. of Electrical Engineering Indian Institute of Technology

More information

Voice Activity Detection

Voice Activity Detection Voice Activity Detection Speech Processing Tom Bäckström Aalto University October 2015 Introduction Voice activity detection (VAD) (or speech activity detection, or speech detection) refers to a class

More information

SPEECH ENHANCEMENT BASED ON A LOG-SPECTRAL AMPLITUDE ESTIMATOR AND A POSTFILTER DERIVED FROM CLEAN SPEECH CODEBOOK

SPEECH ENHANCEMENT BASED ON A LOG-SPECTRAL AMPLITUDE ESTIMATOR AND A POSTFILTER DERIVED FROM CLEAN SPEECH CODEBOOK 18th European Signal Processing Conference (EUSIPCO-2010) Aalborg, Denmar, August 23-27, 2010 SPEECH ENHANCEMENT BASED ON A LOG-SPECTRAL AMPLITUDE ESTIMATOR AND A POSTFILTER DERIVED FROM CLEAN SPEECH CODEBOOK

More information

Campus Location Recognition using Audio Signals

Campus Location Recognition using Audio Signals 1 Campus Location Recognition using Audio Signals James Sun,Reid Westwood SUNetID:jsun2015,rwestwoo Email: jsun2015@stanford.edu, rwestwoo@stanford.edu I. INTRODUCTION People use sound both consciously

More information

Implementation of Text to Speech Conversion

Implementation of Text to Speech Conversion Implementation of Text to Speech Conversion Chaw Su Thu Thu 1, Theingi Zin 2 1 Department of Electronic Engineering, Mandalay Technological University, Mandalay 2 Department of Electronic Engineering,

More information

Auditory Based Feature Vectors for Speech Recognition Systems

Auditory Based Feature Vectors for Speech Recognition Systems Auditory Based Feature Vectors for Speech Recognition Systems Dr. Waleed H. Abdulla Electrical & Computer Engineering Department The University of Auckland, New Zealand [w.abdulla@auckland.ac.nz] 1 Outlines

More information

Binaural Speaker Recognition for Humanoid Robots

Binaural Speaker Recognition for Humanoid Robots Binaural Speaker Recognition for Humanoid Robots Karim Youssef, Sylvain Argentieri and Jean-Luc Zarader Université Pierre et Marie Curie Institut des Systèmes Intelligents et de Robotique, CNRS UMR 7222

More information

Effective post-processing for single-channel frequency-domain speech enhancement Weifeng Li a

Effective post-processing for single-channel frequency-domain speech enhancement Weifeng Li a R E S E A R C H R E P O R T I D I A P Effective post-processing for single-channel frequency-domain speech enhancement Weifeng Li a IDIAP RR 7-7 January 8 submitted for publication a IDIAP Research Institute,

More information

The fundamentals of detection theory

The fundamentals of detection theory Advanced Signal Processing: The fundamentals of detection theory Side 1 of 18 Index of contents: Advanced Signal Processing: The fundamentals of detection theory... 3 1 Problem Statements... 3 2 Detection

More information

Estimating Single-Channel Source Separation Masks: Relevance Vector Machine Classifiers vs. Pitch-Based Masking

Estimating Single-Channel Source Separation Masks: Relevance Vector Machine Classifiers vs. Pitch-Based Masking Estimating Single-Channel Source Separation Masks: Relevance Vector Machine Classifiers vs. Pitch-Based Masking Ron J. Weiss and Daniel P. W. Ellis LabROSA, Dept. of Elec. Eng. Columbia University New

More information

An Efficient Extraction of Vocal Portion from Music Accompaniment Using Trend Estimation

An Efficient Extraction of Vocal Portion from Music Accompaniment Using Trend Estimation An Efficient Extraction of Vocal Portion from Music Accompaniment Using Trend Estimation Aisvarya V 1, Suganthy M 2 PG Student [Comm. Systems], Dept. of ECE, Sree Sastha Institute of Engg. & Tech., Chennai,

More information

MFCC AND GMM BASED TAMIL LANGUAGE SPEAKER IDENTIFICATION SYSTEM

MFCC AND GMM BASED TAMIL LANGUAGE SPEAKER IDENTIFICATION SYSTEM www.advancejournals.org Open Access Scientific Publisher MFCC AND GMM BASED TAMIL LANGUAGE SPEAKER IDENTIFICATION SYSTEM ABSTRACT- P. Santhiya 1, T. Jayasankar 1 1 AUT (BIT campus), Tiruchirappalli, India

More information

IMPULSIVE NOISE MITIGATION IN OFDM SYSTEMS USING SPARSE BAYESIAN LEARNING

IMPULSIVE NOISE MITIGATION IN OFDM SYSTEMS USING SPARSE BAYESIAN LEARNING IMPULSIVE NOISE MITIGATION IN OFDM SYSTEMS USING SPARSE BAYESIAN LEARNING Jing Lin, Marcel Nassar and Brian L. Evans Department of Electrical and Computer Engineering The University of Texas at Austin

More information

Automatic Text-Independent. Speaker. Recognition Approaches Using Binaural Inputs

Automatic Text-Independent. Speaker. Recognition Approaches Using Binaural Inputs Automatic Text-Independent Speaker Recognition Approaches Using Binaural Inputs Karim Youssef, Sylvain Argentieri and Jean-Luc Zarader 1 Outline Automatic speaker recognition: introduction Designed systems

More information

An Optimization of Audio Classification and Segmentation using GASOM Algorithm

An Optimization of Audio Classification and Segmentation using GASOM Algorithm An Optimization of Audio Classification and Segmentation using GASOM Algorithm Dabbabi Karim, Cherif Adnen Research Unity of Processing and Analysis of Electrical and Energetic Systems Faculty of Sciences

More information

Perceptually Motivated Linear Prediction Cepstral Features for Network Speech Recognition

Perceptually Motivated Linear Prediction Cepstral Features for Network Speech Recognition Perceptually Motivated Linear Prediction Cepstral Features for Network Speech Recognition Aadel Alatwi, Stephen So, Kuldip K. Paliwal Signal Processing Laboratory Griffith University, Brisbane, QLD, 4111,

More information

IMPACT OF DEEP MLP ARCHITECTURE ON DIFFERENT ACOUSTIC MODELING TECHNIQUES FOR UNDER-RESOURCED SPEECH RECOGNITION

IMPACT OF DEEP MLP ARCHITECTURE ON DIFFERENT ACOUSTIC MODELING TECHNIQUES FOR UNDER-RESOURCED SPEECH RECOGNITION IMPACT OF DEEP MLP ARCHITECTURE ON DIFFERENT ACOUSTIC MODELING TECHNIQUES FOR UNDER-RESOURCED SPEECH RECOGNITION David Imseng 1, Petr Motlicek 1, Philip N. Garner 1, Hervé Bourlard 1,2 1 Idiap Research

More information

IMPROVING MICROPHONE ARRAY SPEECH RECOGNITION WITH COCHLEAR IMPLANT-LIKE SPECTRALLY REDUCED SPEECH

IMPROVING MICROPHONE ARRAY SPEECH RECOGNITION WITH COCHLEAR IMPLANT-LIKE SPECTRALLY REDUCED SPEECH RESEARCH REPORT IDIAP IMPROVING MICROPHONE ARRAY SPEECH RECOGNITION WITH COCHLEAR IMPLANT-LIKE SPECTRALLY REDUCED SPEECH Cong-Thanh Do Mohammad J. Taghizadeh Philip N. Garner Idiap-RR-40-2011 DECEMBER

More information

Signal Analysis Using Autoregressive Models of Amplitude Modulation. Sriram Ganapathy

Signal Analysis Using Autoregressive Models of Amplitude Modulation. Sriram Ganapathy Signal Analysis Using Autoregressive Models of Amplitude Modulation Sriram Ganapathy Advisor - Hynek Hermansky Johns Hopkins University 11-18-2011 Overview Introduction AR Model of Hilbert Envelopes FDLP

More information

Electronic disguised voice identification based on Mel- Frequency Cepstral Coefficient analysis

Electronic disguised voice identification based on Mel- Frequency Cepstral Coefficient analysis International Journal of Scientific and Research Publications, Volume 5, Issue 11, November 2015 412 Electronic disguised voice identification based on Mel- Frequency Cepstral Coefficient analysis Shalate

More information

Adaptive noise level estimation

Adaptive noise level estimation Adaptive noise level estimation Chunghsin Yeh, Axel Roebel To cite this version: Chunghsin Yeh, Axel Roebel. Adaptive noise level estimation. Workshop on Computer Music and Audio Technology (WOCMAT 6),

More information

A STUDY ON CEPSTRAL SUB-BAND NORMALIZATION FOR ROBUST ASR

A STUDY ON CEPSTRAL SUB-BAND NORMALIZATION FOR ROBUST ASR A STUDY ON CEPSTRAL SUB-BAND NORMALIZATION FOR ROBUST ASR Syu-Siang Wang 1, Jeih-weih Hung, Yu Tsao 1 1 Research Center for Information Technology Innovation, Academia Sinica, Taipei, Taiwan Dept. of Electrical

More information

SpeakerID - Voice Activity Detection

SpeakerID - Voice Activity Detection SpeakerID - Voice Activity Detection Victor Lenoir Technical Report n o 1112, June 2011 revision 2288 Voice Activity Detection has many applications. It s for example a mandatory front-end process in speech

More information

Isolated Word Recognition Based on Combination of Multiple Noise-Robust Techniques

Isolated Word Recognition Based on Combination of Multiple Noise-Robust Techniques Isolated Word Recognition Based on Combination of Multiple Noise-Robust Techniques 81 Isolated Word Recognition Based on Combination of Multiple Noise-Robust Techniques Noboru Hayasaka 1, Non-member ABSTRACT

More information

Announcements. Today. Speech and Language. State Path Trellis. HMMs: MLE Queries. Introduction to Artificial Intelligence. V22.

Announcements. Today. Speech and Language. State Path Trellis. HMMs: MLE Queries. Introduction to Artificial Intelligence. V22. Introduction to Artificial Intelligence Announcements V22.0472-001 Fall 2009 Lecture 19: Speech Recognition & Viterbi Decoding Rob Fergus Dept of Computer Science, Courant Institute, NYU Slides from John

More information

SIGNAL PROCESSING FOR ROBUST SPEECH RECOGNITION MOTIVATED BY AUDITORY PROCESSING CHANWOO KIM

SIGNAL PROCESSING FOR ROBUST SPEECH RECOGNITION MOTIVATED BY AUDITORY PROCESSING CHANWOO KIM SIGNAL PROCESSING FOR ROBUST SPEECH RECOGNITION MOTIVATED BY AUDITORY PROCESSING CHANWOO KIM MAY 21 ABSTRACT Although automatic speech recognition systems have dramatically improved in recent decades,

More information

Collaborative Classification of Multiple Ground Vehicles in Wireless Sensor Networks Based on Acoustic Signals

Collaborative Classification of Multiple Ground Vehicles in Wireless Sensor Networks Based on Acoustic Signals Western Michigan University ScholarWorks at WMU Dissertations Graduate College 1-1-2011 Collaborative Classification of Multiple Ground Vehicles in Wireless Sensor Networks Based on Acoustic Signals Ahmad

More information

A CONSTRUCTION OF COMPACT MFCC-TYPE FEATURES USING SHORT-TIME STATISTICS FOR APPLICATIONS IN AUDIO SEGMENTATION

A CONSTRUCTION OF COMPACT MFCC-TYPE FEATURES USING SHORT-TIME STATISTICS FOR APPLICATIONS IN AUDIO SEGMENTATION 17th European Signal Processing Conference (EUSIPCO 2009) Glasgow, Scotland, August 24-28, 2009 A CONSTRUCTION OF COMPACT MFCC-TYPE FEATURES USING SHORT-TIME STATISTICS FOR APPLICATIONS IN AUDIO SEGMENTATION

More information

Mobile Wireless Channel Dispersion State Model

Mobile Wireless Channel Dispersion State Model Mobile Wireless Channel Dispersion State Model Enabling Cognitive Processing Situational Awareness Kenneth D. Brown Ph.D. Candidate EECS University of Kansas kenneth.brown@jhuapl.edu Dr. Glenn Prescott

More information

Introduction to HTK Toolkit

Introduction to HTK Toolkit Introduction to HTK Toolkit Berlin Chen 2004 Reference: - Steve Young et al. The HTK Book. Version 3.2, 2002. Outline An Overview of HTK HTK Processing Stages Data Preparation Tools Training Tools Testing

More information

Detection of Compound Structures in Very High Spatial Resolution Images

Detection of Compound Structures in Very High Spatial Resolution Images Detection of Compound Structures in Very High Spatial Resolution Images Selim Aksoy Department of Computer Engineering Bilkent University Bilkent, 06800, Ankara, Turkey saksoy@cs.bilkent.edu.tr Joint work

More information

The Munich 2011 CHiME Challenge Contribution: BLSTM-NMF Speech Enhancement and Recognition for Reverberated Multisource Environments

The Munich 2011 CHiME Challenge Contribution: BLSTM-NMF Speech Enhancement and Recognition for Reverberated Multisource Environments The Munich 2011 CHiME Challenge Contribution: BLSTM-NMF Speech Enhancement and Recognition for Reverberated Multisource Environments Felix Weninger, Jürgen Geiger, Martin Wöllmer, Björn Schuller, Gerhard

More information

Dimension Reduction of the Modulation Spectrogram for Speaker Verification

Dimension Reduction of the Modulation Spectrogram for Speaker Verification Dimension Reduction of the Modulation Spectrogram for Speaker Verification Tomi Kinnunen Speech and Image Processing Unit Department of Computer Science University of Joensuu, Finland tkinnu@cs.joensuu.fi

More information

Advanced audio analysis. Martin Gasser

Advanced 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 information

DISTANT speech recognition (DSR) [1] is a challenging

DISTANT speech recognition (DSR) [1] is a challenging 1 Convolutional Neural Networks for Distant Speech Recognition Pawel Swietojanski, Student Member, IEEE, Arnab Ghoshal, Member, IEEE, and Steve Renals, Fellow, IEEE Abstract We investigate convolutional

More information

Change Point Determination in Audio Data Using Auditory Features

Change Point Determination in Audio Data Using Auditory Features INTL JOURNAL OF ELECTRONICS AND TELECOMMUNICATIONS, 0, VOL., NO., PP. 8 90 Manuscript received April, 0; revised June, 0. DOI: /eletel-0-00 Change Point Determination in Audio Data Using Auditory Features

More information

Implementing Speaker Recognition

Implementing Speaker Recognition Implementing Speaker Recognition Chase Zhou Physics 406-11 May 2015 Introduction Machinery has come to replace much of human labor. They are faster, stronger, and more consistent than any human. They ve

More information

Comparison of Spectral Analysis Methods for Automatic Speech Recognition

Comparison of Spectral Analysis Methods for Automatic Speech Recognition INTERSPEECH 2013 Comparison of Spectral Analysis Methods for Automatic Speech Recognition Venkata Neelima Parinam, Chandra Vootkuri, Stephen A. Zahorian Department of Electrical and Computer Engineering

More information

Correspondence. Voiced-Unvoiced-Silence Classifications of Speech Using Hybrid Features and a Network Classifier I. INTRODUCTION

Correspondence. Voiced-Unvoiced-Silence Classifications of Speech Using Hybrid Features and a Network Classifier I. INTRODUCTION 250 IEEE TRANSACTIONS ON SPEECH AND AUDIO PROCESSING, VOL. 1, NO. 2, APRIL 1993 Correspondence Voiced-Unvoiced-Silence Classifications of Speech Using Hybrid Features and a Network Classifier Yingyong

More information

Machine Learning. Classification, Discriminative learning. Marc Toussaint University of Stuttgart Summer 2014

Machine Learning. Classification, Discriminative learning. Marc Toussaint University of Stuttgart Summer 2014 Machine Learning Classification, Discriminative learning Structured output, structured input, discriminative function, joint input-output features, Likelihood Maximization, Logistic regression, binary

More information

A multi-class method for detecting audio events in news broadcasts

A multi-class method for detecting audio events in news broadcasts A multi-class method for detecting audio events in news broadcasts Sergios Petridis, Theodoros Giannakopoulos, and Stavros Perantonis Computational Intelligence Laboratory, Institute of Informatics and

More information

CHORD RECOGNITION USING INSTRUMENT VOICING CONSTRAINTS

CHORD RECOGNITION USING INSTRUMENT VOICING CONSTRAINTS CHORD RECOGNITION USING INSTRUMENT VOICING CONSTRAINTS Xinglin Zhang Dept. of Computer Science University of Regina Regina, SK CANADA S4S 0A2 zhang46x@cs.uregina.ca David Gerhard Dept. of Computer Science,

More information

Tools for Advanced Sound & Vibration Analysis

Tools for Advanced Sound & Vibration Analysis Tools for Advanced Sound & Vibration Ravichandran Raghavan Technical Marketing Engineer Agenda NI Sound and Vibration Measurement Suite Advanced Signal Processing Algorithms Time- Quefrency and Cepstrum

More information

CS 188: Artificial Intelligence Spring Speech in an Hour

CS 188: Artificial Intelligence Spring Speech in an Hour CS 188: Artificial Intelligence Spring 2006 Lecture 19: Speech Recognition 3/23/2006 Dan Klein UC Berkeley Many slides from Dan Jurafsky Speech in an Hour Speech input is an acoustic wave form s p ee ch

More information

I D I A P. Mel-Cepstrum Modulation Spectrum (MCMS) Features for Robust ASR R E S E A R C H R E P O R T. Iain McCowan a Hemant Misra a,b

I D I A P. Mel-Cepstrum Modulation Spectrum (MCMS) Features for Robust ASR R E S E A R C H R E P O R T. Iain McCowan a Hemant Misra a,b R E S E A R C H R E P O R T I D I A P Mel-Cepstrum Modulation Spectrum (MCMS) Features for Robust ASR a Vivek Tyagi Hervé Bourlard a,b IDIAP RR 3-47 September 23 Iain McCowan a Hemant Misra a,b to appear

More information

PLP 2 Autoregressive modeling of auditory-like 2-D spectro-temporal patterns

PLP 2 Autoregressive modeling of auditory-like 2-D spectro-temporal patterns PLP 2 Autoregressive modeling of auditory-like 2-D spectro-temporal patterns Marios Athineos a, Hynek Hermansky b and Daniel P.W. Ellis a a LabROSA, Dept. of Electrical Engineering, Columbia University,

More information