Monaural speech separation using source-adapted models

Size: px
Start display at page:

Download "Monaural speech separation using source-adapted models"

Transcription

1 Monaural speech separation using source-adapted models Ron Weiss, Dan Ellis LabROSA Department of Electrical Enginering Columbia University 007 IEEE Workshop on Applications of Signal Processing to Audio and Acoustics Ron Weiss, Dan Ellis (Columbia University) Monaural speech separation using source-adapted models WASPAA / 15

2 Monaural speech separation Given single channel recording of multiple talkers Infer the original source signals from mixture Under-determined - more unknowns (sources) than observations Ron Weiss, Dan Ellis (Columbia University) Monaural speech separation using source-adapted models WASPAA 007 / 15

3 Speech separation challenge [Cooke and Lee, 00] Single channel, two-talker mixtures of utterances from 3 speakers Constrained grammar: command() color() preposition() letter(5) digit(10) adverb() Task: determine letter and digit for source that said white -9 to db TMR Ron Weiss, Dan Ellis (Columbia University) Monaural speech separation using source-adapted models WASPAA / 15

4 Model-based separation Frequency (khz) 0 Model means State index Use constraints from prior signal models to guide separation HMM, log spectral features Factorial model inference Explain each frame of mixed signal as combination of model states e.g. Iroquois [Kristjansson et al., 00] Speaker-dependent models Acoustic dynamics and grammar constraints Superhuman performance Ron Weiss, Dan Ellis (Columbia University) Monaural speech separation using source-adapted models WASPAA 007 / 15

5 Model-based separation - Limitations Rely on speaker-dependent models to disambiguate sources What if the task isn t so well defined? No a priori knowledge of speaker identities or grammar Adapt speaker-independent source model [Ozerov et al., 005] Problems 1 Want to adapt to a single utterance, not enough data for MLLR Use PCA to reduce number of adaptation parameters - Eigenvoices Only observation is mixed signal Iterative separation/adaptation algorithm Ron Weiss, Dan Ellis (Columbia University) Monaural speech separation using source-adapted models WASPAA / 15

6 Eigenvoices [Kuhn et al., 000] Train N speaker-dependent models priors on space of speaker variation Pack model parameters (Gaussian means) into speaker supervector Principal component analysis to find orthonormal bases Speaker model is a linear combination of bases: µ = µ + w U + g adapted model mean voice weights eigenvoice bases gain Ron Weiss, Dan Ellis (Columbia University) Monaural speech separation using source-adapted models WASPAA 007 / 15

7 Eigenvoice example Frequency (khz) Frequency (khz) Frequency (khz) Frequency (khz) Mean voice b d g p t k jh ch s z f th v dh m n l r w y iy ih eh eyaeaaawayahaoowuwax Eigenvoice dimension 1 b d g p t k jh ch s z f th v dh m n l r w y iy ih eh eyaeaaawayahaoowuwax Eigenvoice dimension b d g p t k jh ch s z f th v dh m n l r w y iy ih eh eyaeaaawayahaoowuwax Eigenvoice dimension 3 b d g p t k jh ch s z f th v dh m n l r w y iy ih eh eyaeaaawayahaoowuwax Ron Weiss, Dan Ellis (Columbia University) Monaural speech separation using source-adapted models WASPAA / 15

8 Separation algorithm - Signal separation Compose factorial HMM from adapted models Find maximum likelihood path using Viterbi algorithm Reconstruct source signals from Viterbi path model model 1 observations / time Ron Weiss, Dan Ellis (Columbia University) Monaural speech separation using source-adapted models WASPAA 007 / 15

9 Separation algorithm - Model adaptation Find projection of reconstructed source signals onto eigenvoice bases But state sequence is hidden, need EM E-step: HMM forward-backward M-step: for each possible state sequence, project signal frames onto corresponding sequence of states from each eigenvoice basis vector Iterate... Ron Weiss, Dan Ellis (Columbia University) Monaural speech separation using source-adapted models WASPAA / 15

10 Separation example Mixture: t3_swila_m1_sbar9n Adaptation iteration 1 Frequency (khz) Adaptation iteration 3 Adaptation iteration SD model separation Time (sec) Ron Weiss, Dan Ellis (Columbia University) Monaural speech separation using source-adapted models WASPAA / 15

11 Performance 0 Diff Gender Same Gender Same Talker Accuracy Iteration Letter-digit accuracy averaged across all TMRs Adaptation improves separation Same talker case hard - source permutations Ron Weiss, Dan Ellis (Columbia University) Monaural speech separation using source-adapted models WASPAA / 15

12 Performance - Adapted vs. source-dependent models Accuracy Diff Gender db 3dB 0dB 3dB db 9dB Same Gender db 3dB 0dB 3dB db 9dB Same Talker SD SA SI Baseline db 3dB 0dB 3dB db 9dB Ron Weiss, Dan Ellis (Columbia University) Monaural speech separation using source-adapted models WASPAA / 15

13 Performance - Held out speakers Accuracy SA SD Same Gender Diff Gender Num training speakers Num training speakers 3 Trained models on subset of speakers Tested on mixtures from held out speakers Performance suffers for both systems Relative decrease significantly bigger for SD than SA Open question: scale Ron Weiss, Dan Ellis (Columbia University) Monaural speech separation using source-adapted models WASPAA / 15

14 Summary Limitations of model-based source separation Algorithm for model adaptation from mixed signal Significant improvement over speaker-independent models Source-dependent models better on matched training/testing data Adaptation helps generalize better to held out speakers Ron Weiss, Dan Ellis (Columbia University) Monaural speech separation using source-adapted models WASPAA / 15

15 References Cooke, M. and Lee, T. W. (00). The speech separation challenge. Kristjansson, T., Hershey, J., Olsen, P., Rennie, S., and Gopinath, R. (00). Super-human multi-talker speech recognition: The IBM 00 speech separation challenge system. In Proceedings of Interspeech. Kuhn, R., Junqua, J., Nguyen, P., and Niedzielski, N. (000). Rapid speaker adaptation in eigenvoice space. IEEE Transations on Speech and Audio Processing, (): Ozerov, A., Philippe, P., Gribonval, R., and Bimbot, F. (005). One microphone singing voice separation using source-adapted models. In Proceedings of IEEE Workshop on Applications of Signal Processing to Audio and Acoustics. Ron Weiss, Dan Ellis (Columbia University) Monaural speech separation using source-adapted models WASPAA / 15

16 Separation algorithm - Initialization Fast convergence needs good initialization 1000 Want to differentiate source models to get 500 best separation 0 Get initial coefficient for each eigenvoice 500 dimension independently Coarsely quantize eigenvoice weights Find most likely combination in mixture w 1 w Eigenvoice weights vs speaker gender Male Female

Eigenvoice Speaker Adaptation via Composite Kernel PCA

Eigenvoice Speaker Adaptation via Composite Kernel PCA Eigenvoice Speaker Adaptation via Composite Kernel PCA James T. Kwok, Brian Mak and Simon Ho Department of Computer Science Hong Kong University of Science and Technology Clear Water Bay, Hong Kong [jamesk,mak,csho]@cs.ust.hk

More information

Independent Component Analysis and Unsupervised Learning. Jen-Tzung Chien

Independent Component Analysis and Unsupervised Learning. Jen-Tzung Chien Independent Component Analysis and Unsupervised Learning Jen-Tzung Chien TABLE OF CONTENTS 1. Independent Component Analysis 2. Case Study I: Speech Recognition Independent voices Nonparametric likelihood

More information

Independent Component Analysis and Unsupervised Learning

Independent Component Analysis and Unsupervised Learning Independent Component Analysis and Unsupervised Learning Jen-Tzung Chien National Cheng Kung University TABLE OF CONTENTS 1. Independent Component Analysis 2. Case Study I: Speech Recognition Independent

More information

Heeyoul (Henry) Choi. Dept. of Computer Science Texas A&M University

Heeyoul (Henry) Choi. Dept. of Computer Science Texas A&M University Heeyoul (Henry) Choi Dept. of Computer Science Texas A&M University hchoi@cs.tamu.edu Introduction Speaker Adaptation Eigenvoice Comparison with others MAP, MLLR, EMAP, RMP, CAT, RSW Experiments Future

More information

The Noisy Channel Model. Statistical NLP Spring Mel Freq. Cepstral Coefficients. Frame Extraction ... Lecture 10: Acoustic Models

The Noisy Channel Model. Statistical NLP Spring Mel Freq. Cepstral Coefficients. Frame Extraction ... Lecture 10: Acoustic Models Statistical NLP Spring 2009 The Noisy Channel Model Lecture 10: Acoustic Models Dan Klein UC Berkeley Search through space of all possible sentences. Pick the one that is most probable given the waveform.

More information

Statistical NLP Spring The Noisy Channel Model

Statistical NLP Spring The Noisy Channel Model Statistical NLP Spring 2009 Lecture 10: Acoustic Models Dan Klein UC Berkeley The Noisy Channel Model Search through space of all possible sentences. Pick the one that is most probable given the waveform.

More information

The Noisy Channel Model. Statistical NLP Spring Mel Freq. Cepstral Coefficients. Frame Extraction ... Lecture 9: Acoustic Models

The Noisy Channel Model. Statistical NLP Spring Mel Freq. Cepstral Coefficients. Frame Extraction ... Lecture 9: Acoustic Models Statistical NLP Spring 2010 The Noisy Channel Model Lecture 9: Acoustic Models Dan Klein UC Berkeley Acoustic model: HMMs over word positions with mixtures of Gaussians as emissions Language model: Distributions

More information

SINGLE CHANNEL SPEECH MUSIC SEPARATION USING NONNEGATIVE MATRIX FACTORIZATION AND SPECTRAL MASKS. Emad M. Grais and Hakan Erdogan

SINGLE CHANNEL SPEECH MUSIC SEPARATION USING NONNEGATIVE MATRIX FACTORIZATION AND SPECTRAL MASKS. Emad M. Grais and Hakan Erdogan SINGLE CHANNEL SPEECH MUSIC SEPARATION USING NONNEGATIVE MATRIX FACTORIZATION AND SPECTRAL MASKS Emad M. Grais and Hakan Erdogan Faculty of Engineering and Natural Sciences, Sabanci University, Orhanli

More information

Statistical NLP Spring Digitizing Speech

Statistical NLP Spring Digitizing Speech Statistical NLP Spring 2008 Lecture 10: Acoustic Models Dan Klein UC Berkeley Digitizing Speech 1 Frame Extraction A frame (25 ms wide) extracted every 10 ms 25 ms 10ms... a 1 a 2 a 3 Figure from Simon

More information

Digitizing Speech. Statistical NLP Spring Frame Extraction. Gaussian Emissions. Vector Quantization. HMMs for Continuous Observations? ...

Digitizing Speech. Statistical NLP Spring Frame Extraction. Gaussian Emissions. Vector Quantization. HMMs for Continuous Observations? ... Statistical NLP Spring 2008 Digitizing Speech Lecture 10: Acoustic Models Dan Klein UC Berkeley Frame Extraction A frame (25 ms wide extracted every 10 ms 25 ms 10ms... a 1 a 2 a 3 Figure from Simon Arnfield

More information

Experiments with a Gaussian Merging-Splitting Algorithm for HMM Training for Speech Recognition

Experiments with a Gaussian Merging-Splitting Algorithm for HMM Training for Speech Recognition Experiments with a Gaussian Merging-Splitting Algorithm for HMM Training for Speech Recognition ABSTRACT It is well known that the expectation-maximization (EM) algorithm, commonly used to estimate hidden

More information

Speech Separation Using Gain-Adapted Factorial Hidden Markov Models

Speech Separation Using Gain-Adapted Factorial Hidden Markov Models 1 Speech Separation Using Gain-Adapted Factorial Hidden Markov Models Martin H. Radfar 1,, Richard. M. Dansereau, and Willy Wong 3 1 Department of Computer Science, Stony Brook University, NY, USA Department

More information

Embedded Kernel Eigenvoice Speaker Adaptation and its Implication to Reference Speaker Weighting

Embedded Kernel Eigenvoice Speaker Adaptation and its Implication to Reference Speaker Weighting IEEE TRANSACTIONS ON SPEECH AND AUDIO PROCESSING, December 21, 2005 1 Embedded Kernel Eigenvoice Speaker Adaptation and its Implication to Reference Speaker Weighting Brian Mak, Roger Hsiao, Simon Ho,

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

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

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

Automatic Speech Recognition (CS753)

Automatic Speech Recognition (CS753) Automatic Speech Recognition (CS753) Lecture 21: Speaker Adaptation Instructor: Preethi Jyothi Oct 23, 2017 Speaker variations Major cause of variability in speech is the differences between speakers Speaking

More information

PHONEME CLASSIFICATION OVER THE RECONSTRUCTED PHASE SPACE USING PRINCIPAL COMPONENT ANALYSIS

PHONEME CLASSIFICATION OVER THE RECONSTRUCTED PHASE SPACE USING PRINCIPAL COMPONENT ANALYSIS PHONEME CLASSIFICATION OVER THE RECONSTRUCTED PHASE SPACE USING PRINCIPAL COMPONENT ANALYSIS Jinjin Ye jinjin.ye@mu.edu Michael T. Johnson mike.johnson@mu.edu Richard J. Povinelli richard.povinelli@mu.edu

More information

The Noisy Channel Model. CS 294-5: Statistical Natural Language Processing. Speech Recognition Architecture. Digitizing Speech

The Noisy Channel Model. CS 294-5: Statistical Natural Language Processing. Speech Recognition Architecture. Digitizing Speech CS 294-5: Statistical Natural Language Processing The Noisy Channel Model Speech Recognition II Lecture 21: 11/29/05 Search through space of all possible sentences. Pick the one that is most probable given

More information

Single Channel Signal Separation Using MAP-based Subspace Decomposition

Single Channel Signal Separation Using MAP-based Subspace Decomposition Single Channel Signal Separation Using MAP-based Subspace Decomposition Gil-Jin Jang, Te-Won Lee, and Yung-Hwan Oh 1 Spoken Language Laboratory, Department of Computer Science, KAIST 373-1 Gusong-dong,

More information

Temporal Modeling and Basic Speech Recognition

Temporal Modeling and Basic Speech Recognition UNIVERSITY ILLINOIS @ URBANA-CHAMPAIGN OF CS 498PS Audio Computing Lab Temporal Modeling and Basic Speech Recognition Paris Smaragdis paris@illinois.edu paris.cs.illinois.edu Today s lecture Recognizing

More information

Dept. Electronics and Electrical Engineering, Keio University, Japan. NTT Communication Science Laboratories, NTT Corporation, Japan.

Dept. Electronics and Electrical Engineering, Keio University, Japan. NTT Communication Science Laboratories, NTT Corporation, Japan. JOINT SEPARATION AND DEREVERBERATION OF REVERBERANT MIXTURES WITH DETERMINED MULTICHANNEL NON-NEGATIVE MATRIX FACTORIZATION Hideaki Kagami, Hirokazu Kameoka, Masahiro Yukawa Dept. Electronics and Electrical

More information

Joint Factor Analysis for Speaker Verification

Joint Factor Analysis for Speaker Verification Joint Factor Analysis for Speaker Verification Mengke HU ASPITRG Group, ECE Department Drexel University mengke.hu@gmail.com October 12, 2012 1/37 Outline 1 Speaker Verification Baseline System Session

More information

Speaker recognition by means of Deep Belief Networks

Speaker recognition by means of Deep Belief Networks Speaker recognition by means of Deep Belief Networks Vasileios Vasilakakis, Sandro Cumani, Pietro Laface, Politecnico di Torino, Italy {first.lastname}@polito.it 1. Abstract Most state of the art speaker

More information

On the Influence of the Delta Coefficients in a HMM-based Speech Recognition System

On the Influence of the Delta Coefficients in a HMM-based Speech Recognition System On the Influence of the Delta Coefficients in a HMM-based Speech Recognition System Fabrice Lefèvre, Claude Montacié and Marie-José Caraty Laboratoire d'informatique de Paris VI 4, place Jussieu 755 PARIS

More information

Machine Recognition of Sounds in Mixtures

Machine Recognition of Sounds in Mixtures Machine Recognition of Sounds in Mixtures Outline 1 2 3 4 Computational Auditory Scene Analysis Speech Recognition as Source Formation Sound Fragment Decoding Results & Conclusions Dan Ellis

More information

Non-Negative Matrix Factorization And Its Application to Audio. Tuomas Virtanen Tampere University of Technology

Non-Negative Matrix Factorization And Its Application to Audio. Tuomas Virtanen Tampere University of Technology Non-Negative Matrix Factorization And Its Application to Audio Tuomas Virtanen Tampere University of Technology tuomas.virtanen@tut.fi 2 Contents Introduction to audio signals Spectrogram representation

More information

A Small Footprint i-vector Extractor

A Small Footprint i-vector Extractor A Small Footprint i-vector Extractor Patrick Kenny Odyssey Speaker and Language Recognition Workshop June 25, 2012 1 / 25 Patrick Kenny A Small Footprint i-vector Extractor Outline Introduction Review

More information

ENHANCEMENTS OF MAXIMUM LIKELIHOOD EIGEN-DECOMPOSITION USING FUZZY LOGIC CONTROL FOR EIGENVOICE-BASED SPEAKER ADAPTATION.

ENHANCEMENTS OF MAXIMUM LIKELIHOOD EIGEN-DECOMPOSITION USING FUZZY LOGIC CONTROL FOR EIGENVOICE-BASED SPEAKER ADAPTATION. International Journal of Innovative Computing, Information and Control ICIC International c 2011 ISSN 1349-4198 Volume 7, Number 7(B), July 2011 pp. 4207 4222 ENHANCEMENTS OF MAXIMUM LIKELIHOOD EIGEN-DECOMPOSITION

More information

Uncertainty Modeling without Subspace Methods for Text-Dependent Speaker Recognition

Uncertainty Modeling without Subspace Methods for Text-Dependent Speaker Recognition Uncertainty Modeling without Subspace Methods for Text-Dependent Speaker Recognition Patrick Kenny, Themos Stafylakis, Md. Jahangir Alam and Marcel Kockmann Odyssey Speaker and Language Recognition Workshop

More information

STA 414/2104: Machine Learning

STA 414/2104: Machine Learning STA 414/2104: Machine Learning Russ Salakhutdinov Department of Computer Science! Department of Statistics! rsalakhu@cs.toronto.edu! http://www.cs.toronto.edu/~rsalakhu/ Lecture 9 Sequential Data So far

More information

Eigenvoice Modeling With Sparse Training Data

Eigenvoice Modeling With Sparse Training Data IEEE TRANSACTIONS ON SPEECH AND AUDIO PROCESSING, VOL. 13, NO. 3, MAY 2005 345 Eigenvoice Modeling With Sparse Training Data Patrick Kenny, Member, IEEE, Gilles Boulianne, Member, IEEE, and Pierre Dumouchel,

More information

Augmented Statistical Models for Speech Recognition

Augmented Statistical Models for Speech Recognition Augmented Statistical Models for Speech Recognition Mark Gales & Martin Layton 31 August 2005 Trajectory Models For Speech Processing Workshop Overview Dependency Modelling in Speech Recognition: latent

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

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

Speaker Verification Using Accumulative Vectors with Support Vector Machines

Speaker Verification Using Accumulative Vectors with Support Vector Machines Speaker Verification Using Accumulative Vectors with Support Vector Machines Manuel Aguado Martínez, Gabriel Hernández-Sierra, and José Ramón Calvo de Lara Advanced Technologies Application Center, Havana,

More information

Hidden Markov Model and Speech Recognition

Hidden Markov Model and Speech Recognition 1 Dec,2006 Outline Introduction 1 Introduction 2 3 4 5 Introduction What is Speech Recognition? Understanding what is being said Mapping speech data to textual information Speech Recognition is indeed

More information

A TWO-LAYER NON-NEGATIVE MATRIX FACTORIZATION MODEL FOR VOCABULARY DISCOVERY. MengSun,HugoVanhamme

A TWO-LAYER NON-NEGATIVE MATRIX FACTORIZATION MODEL FOR VOCABULARY DISCOVERY. MengSun,HugoVanhamme A TWO-LAYER NON-NEGATIVE MATRIX FACTORIZATION MODEL FOR VOCABULARY DISCOVERY MengSun,HugoVanhamme Department of Electrical Engineering-ESAT, Katholieke Universiteit Leuven, Kasteelpark Arenberg 10, Bus

More information

STA 4273H: Statistical Machine Learning

STA 4273H: Statistical Machine Learning STA 4273H: Statistical Machine Learning Russ Salakhutdinov Department of Statistics! rsalakhu@utstat.toronto.edu! http://www.utstat.utoronto.ca/~rsalakhu/ Sidney Smith Hall, Room 6002 Lecture 11 Project

More information

Support Vector Machines using GMM Supervectors for Speaker Verification

Support Vector Machines using GMM Supervectors for Speaker Verification 1 Support Vector Machines using GMM Supervectors for Speaker Verification W. M. Campbell, D. E. Sturim, D. A. Reynolds MIT Lincoln Laboratory 244 Wood Street Lexington, MA 02420 Corresponding author e-mail:

More information

A Variance Modeling Framework Based on Variational Autoencoders for Speech Enhancement

A Variance Modeling Framework Based on Variational Autoencoders for Speech Enhancement A Variance Modeling Framework Based on Variational Autoencoders for Speech Enhancement Simon Leglaive 1 Laurent Girin 1,2 Radu Horaud 1 1: Inria Grenoble Rhône-Alpes 2: Univ. Grenoble Alpes, Grenoble INP,

More information

HIDDEN MARKOV MODELS IN SPEECH RECOGNITION

HIDDEN MARKOV MODELS IN SPEECH RECOGNITION HIDDEN MARKOV MODELS IN SPEECH RECOGNITION Wayne Ward Carnegie Mellon University Pittsburgh, PA 1 Acknowledgements Much of this talk is derived from the paper "An Introduction to Hidden Markov Models",

More information

Segmental Recurrent Neural Networks for End-to-end Speech Recognition

Segmental Recurrent Neural Networks for End-to-end Speech Recognition Segmental Recurrent Neural Networks for End-to-end Speech Recognition Liang Lu, Lingpeng Kong, Chris Dyer, Noah Smith and Steve Renals TTI-Chicago, UoE, CMU and UW 9 September 2016 Background A new wave

More information

Nonnegative Matrix Factorization with Markov-Chained Bases for Modeling Time-Varying Patterns in Music Spectrograms

Nonnegative Matrix Factorization with Markov-Chained Bases for Modeling Time-Varying Patterns in Music Spectrograms Nonnegative Matrix Factorization with Markov-Chained Bases for Modeling Time-Varying Patterns in Music Spectrograms Masahiro Nakano 1, Jonathan Le Roux 2, Hirokazu Kameoka 2,YuKitano 1, Nobutaka Ono 1,

More information

How to Deal with Multiple-Targets in Speaker Identification Systems?

How to Deal with Multiple-Targets in Speaker Identification Systems? How to Deal with Multiple-Targets in Speaker Identification Systems? Yaniv Zigel and Moshe Wasserblat ICE Systems Ltd., Audio Analysis Group, P.O.B. 690 Ra anana 4307, Israel yanivz@nice.com Abstract In

More information

Session Variability Compensation in Automatic Speaker Recognition

Session Variability Compensation in Automatic Speaker Recognition Session Variability Compensation in Automatic Speaker Recognition Javier González Domínguez VII Jornadas MAVIR Universidad Autónoma de Madrid November 2012 Outline 1. The Inter-session Variability Problem

More information

ALTERNATIVE OBJECTIVE FUNCTIONS FOR DEEP CLUSTERING

ALTERNATIVE OBJECTIVE FUNCTIONS FOR DEEP CLUSTERING ALTERNATIVE OBJECTIVE FUNCTIONS FOR DEEP CLUSTERING Zhong-Qiu Wang,2, Jonathan Le Roux, John R. Hershey Mitsubishi Electric Research Laboratories (MERL), USA 2 Department of Computer Science and Engineering,

More information

PCA & ICA. CE-717: Machine Learning Sharif University of Technology Spring Soleymani

PCA & ICA. CE-717: Machine Learning Sharif University of Technology Spring Soleymani PCA & ICA CE-717: Machine Learning Sharif University of Technology Spring 2015 Soleymani Dimensionality Reduction: Feature Selection vs. Feature Extraction Feature selection Select a subset of a given

More information

CS 5522: Artificial Intelligence II

CS 5522: Artificial Intelligence II CS 5522: Artificial Intelligence II Particle Filters and Applications of HMMs Instructor: Wei Xu Ohio State University [These slides were adapted from CS188 Intro to AI at UC Berkeley.] Recap: Reasoning

More information

Hidden Markov Models. Dr. Naomi Harte

Hidden Markov Models. Dr. Naomi Harte Hidden Markov Models Dr. Naomi Harte The Talk Hidden Markov Models What are they? Why are they useful? The maths part Probability calculations Training optimising parameters Viterbi unseen sequences Real

More information

CS 5522: Artificial Intelligence II

CS 5522: Artificial Intelligence II CS 5522: Artificial Intelligence II Particle Filters and Applications of HMMs Instructor: Alan Ritter Ohio State University [These slides were adapted from CS188 Intro to AI at UC Berkeley. All materials

More information

An Asynchronous Hidden Markov Model for Audio-Visual Speech Recognition

An Asynchronous Hidden Markov Model for Audio-Visual Speech Recognition An Asynchronous Hidden Markov Model for Audio-Visual Speech Recognition Samy Bengio Dalle Molle Institute for Perceptual Artificial Intelligence (IDIAP) CP 592, rue du Simplon 4, 1920 Martigny, Switzerland

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

An EM Algorithm for Localizing Multiple Sound Sources in Reverberant Environments

An EM Algorithm for Localizing Multiple Sound Sources in Reverberant Environments An EM Algorithm for Localizing Multiple Sound Sources in Reverberant Environments Michael I. Mandel, Daniel P. W. Ellis LabROSA, Dept. of Electrical Engineering Columbia University New York, NY {mim,dpwe}@ee.columbia.edu

More information

Hidden Markov Models in Language Processing

Hidden Markov Models in Language Processing Hidden Markov Models in Language Processing Dustin Hillard Lecture notes courtesy of Prof. Mari Ostendorf Outline Review of Markov models What is an HMM? Examples General idea of hidden variables: implications

More information

MULTISENSORY SPEECH ENHANCEMENT IN NOISY ENVIRONMENTS USING BONE-CONDUCTED AND AIR-CONDUCTED MICROPHONES. Mingzi Li,Israel Cohen and Saman Mousazadeh

MULTISENSORY SPEECH ENHANCEMENT IN NOISY ENVIRONMENTS USING BONE-CONDUCTED AND AIR-CONDUCTED MICROPHONES. Mingzi Li,Israel Cohen and Saman Mousazadeh MULTISENSORY SPEECH ENHANCEMENT IN NOISY ENVIRONMENTS USING BONE-CONDUCTED AND AIR-CONDUCTED MICROPHONES Mingzi Li,Israel Cohen and Saman Mousazadeh Department of Electrical Engineering, Technion - Israel

More information

Hidden Markov Models, I. Examples. Steven R. Dunbar. Toy Models. Standard Mathematical Models. Realistic Hidden Markov Models.

Hidden Markov Models, I. Examples. Steven R. Dunbar. Toy Models. Standard Mathematical Models. Realistic Hidden Markov Models. , I. Toy Markov, I. February 17, 2017 1 / 39 Outline, I. Toy Markov 1 Toy 2 3 Markov 2 / 39 , I. Toy Markov A good stack of examples, as large as possible, is indispensable for a thorough understanding

More information

CS 136a Lecture 7 Speech Recognition Architecture: Training models with the Forward backward algorithm

CS 136a Lecture 7 Speech Recognition Architecture: Training models with the Forward backward algorithm + September13, 2016 Professor Meteer CS 136a Lecture 7 Speech Recognition Architecture: Training models with the Forward backward algorithm Thanks to Dan Jurafsky for these slides + ASR components n Feature

More information

Efficient Target Activity Detection Based on Recurrent Neural Networks

Efficient Target Activity Detection Based on Recurrent Neural Networks Efficient Target Activity Detection Based on Recurrent Neural Networks D. Gerber, S. Meier, and W. Kellermann Friedrich-Alexander-Universität Erlangen-Nürnberg (FAU) Motivation Target φ tar oise 1 / 15

More information

Deep NMF for Speech Separation

Deep NMF for Speech Separation MITSUBISHI ELECTRIC RESEARCH LABORATORIES http://www.merl.com Deep NMF for Speech Separation Le Roux, J.; Hershey, J.R.; Weninger, F.J. TR2015-029 April 2015 Abstract Non-negative matrix factorization

More information

Maximum A Posteriori Estimation for Multivariate Gaussian Mixture Observations of Markov Chains

Maximum A Posteriori Estimation for Multivariate Gaussian Mixture Observations of Markov Chains Maximum A Posteriori Estimation for Multivariate Gaussian Mixture Observations of Markov Chains Jean-Luc Gauvain 1 and Chin-Hui Lee Speech Research Department AT&T Bell Laboratories Murray Hill, NJ 07974

More information

Sound Recognition in Mixtures

Sound Recognition in Mixtures Sound Recognition in Mixtures Juhan Nam, Gautham J. Mysore 2, and Paris Smaragdis 2,3 Center for Computer Research in Music and Acoustics, Stanford University, 2 Advanced Technology Labs, Adobe Systems

More information

Brief Introduction of Machine Learning Techniques for Content Analysis

Brief Introduction of Machine Learning Techniques for Content Analysis 1 Brief Introduction of Machine Learning Techniques for Content Analysis Wei-Ta Chu 2008/11/20 Outline 2 Overview Gaussian Mixture Model (GMM) Hidden Markov Model (HMM) Support Vector Machine (SVM) Overview

More information

CURRENT state-of-the-art automatic speech recognition

CURRENT state-of-the-art automatic speech recognition 1850 IEEE TRANSACTIONS ON AUDIO, SPEECH, AND LANGUAGE PROCESSING, VOL. 15, NO. 6, AUGUST 2007 Switching Linear Dynamical Systems for Noise Robust Speech Recognition Bertrand Mesot and David Barber Abstract

More information

MAP adaptation with SphinxTrain

MAP adaptation with SphinxTrain MAP adaptation with SphinxTrain David Huggins-Daines dhuggins@cs.cmu.edu Language Technologies Institute Carnegie Mellon University MAP adaptation with SphinxTrain p.1/12 Theory of MAP adaptation Standard

More information

(3) where the mixing vector is the Fourier transform of are the STFT coefficients of the sources I. INTRODUCTION

(3) where the mixing vector is the Fourier transform of are the STFT coefficients of the sources I. INTRODUCTION 1830 IEEE TRANSACTIONS ON AUDIO, SPEECH, AND LANGUAGE PROCESSING, VOL. 18, NO. 7, SEPTEMBER 2010 Under-Determined Reverberant Audio Source Separation Using a Full-Rank Spatial Covariance Model Ngoc Q.

More information

Engineering Part IIB: Module 4F11 Speech and Language Processing Lectures 4/5 : Speech Recognition Basics

Engineering Part IIB: Module 4F11 Speech and Language Processing Lectures 4/5 : Speech Recognition Basics Engineering Part IIB: Module 4F11 Speech and Language Processing Lectures 4/5 : Speech Recognition Basics Phil Woodland: pcw@eng.cam.ac.uk Lent 2013 Engineering Part IIB: Module 4F11 What is Speech Recognition?

More information

speaker recognition using gmm-ubm semester project presentation

speaker recognition using gmm-ubm semester project presentation speaker recognition using gmm-ubm semester project presentation OBJECTIVES OF THE PROJECT study the GMM-UBM speaker recognition system implement this system with matlab document the code and how it interfaces

More information

Estimating Correlation Coefficient Between Two Complex Signals Without Phase Observation

Estimating Correlation Coefficient Between Two Complex Signals Without Phase Observation Estimating Correlation Coefficient Between Two Complex Signals Without Phase Observation Shigeki Miyabe 1B, Notubaka Ono 2, and Shoji Makino 1 1 University of Tsukuba, 1-1-1 Tennodai, Tsukuba, Ibaraki

More information

Deep Learning for Speech Recognition. Hung-yi Lee

Deep Learning for Speech Recognition. Hung-yi Lee Deep Learning for Speech Recognition Hung-yi Lee Outline Conventional Speech Recognition How to use Deep Learning in acoustic modeling? Why Deep Learning? Speaker Adaptation Multi-task Deep Learning New

More information

p(d θ ) l(θ ) 1.2 x x x

p(d θ ) l(θ ) 1.2 x x x p(d θ ).2 x 0-7 0.8 x 0-7 0.4 x 0-7 l(θ ) -20-40 -60-80 -00 2 3 4 5 6 7 θ ˆ 2 3 4 5 6 7 θ ˆ 2 3 4 5 6 7 θ θ x FIGURE 3.. The top graph shows several training points in one dimension, known or assumed to

More information

Human Mobility Pattern Prediction Algorithm using Mobile Device Location and Time Data

Human Mobility Pattern Prediction Algorithm using Mobile Device Location and Time Data Human Mobility Pattern Prediction Algorithm using Mobile Device Location and Time Data 0. Notations Myungjun Choi, Yonghyun Ro, Han Lee N = number of states in the model T = length of observation sequence

More information

Hidden Markov Models. Aarti Singh Slides courtesy: Eric Xing. Machine Learning / Nov 8, 2010

Hidden Markov Models. Aarti Singh Slides courtesy: Eric Xing. Machine Learning / Nov 8, 2010 Hidden Markov Models Aarti Singh Slides courtesy: Eric Xing Machine Learning 10-701/15-781 Nov 8, 2010 i.i.d to sequential data So far we assumed independent, identically distributed data Sequential data

More information

Improved Speech Presence Probabilities Using HMM-Based Inference, with Applications to Speech Enhancement and ASR

Improved Speech Presence Probabilities Using HMM-Based Inference, with Applications to Speech Enhancement and ASR Improved Speech Presence Probabilities Using HMM-Based Inference, with Applications to Speech Enhancement and ASR Bengt J. Borgström, Student Member, IEEE, and Abeer Alwan, IEEE Fellow Abstract This paper

More information

ISOLATED WORD RECOGNITION FOR ENGLISH LANGUAGE USING LPC,VQ AND HMM

ISOLATED WORD RECOGNITION FOR ENGLISH LANGUAGE USING LPC,VQ AND HMM ISOLATED WORD RECOGNITION FOR ENGLISH LANGUAGE USING LPC,VQ AND HMM Mayukh Bhaowal and Kunal Chawla (Students)Indian Institute of Information Technology, Allahabad, India Abstract: Key words: Speech recognition

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

Hidden Markov Models

Hidden Markov Models Hidden Markov Models Dr Philip Jackson Centre for Vision, Speech & Signal Processing University of Surrey, UK 1 3 2 http://www.ee.surrey.ac.uk/personal/p.jackson/isspr/ Outline 1. Recognizing patterns

More information

Low development cost, high quality speech recognition for new languages and domains. Cheap ASR

Low development cost, high quality speech recognition for new languages and domains. Cheap ASR Low development cost, high quality speech recognition for new languages and domains Cheap ASR Participants: Senior members : Lukas Burget, Nagendra Kumar Goel, Daniel Povey, Richard Rose Graduate students:

More information

Estimation of Relative Operating Characteristics of Text Independent Speaker Verification

Estimation of Relative Operating Characteristics of Text Independent Speaker Verification International Journal of Engineering Science Invention Volume 1 Issue 1 December. 2012 PP.18-23 Estimation of Relative Operating Characteristics of Text Independent Speaker Verification Palivela Hema 1,

More information

"Robust Automatic Speech Recognition through on-line Semi Blind Source Extraction"

Robust Automatic Speech Recognition through on-line Semi Blind Source Extraction "Robust Automatic Speech Recognition through on-line Semi Blind Source Extraction" Francesco Nesta, Marco Matassoni {nesta, matassoni}@fbk.eu Fondazione Bruno Kessler-Irst, Trento (ITALY) For contacts:

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

Noise Compensation for Subspace Gaussian Mixture Models

Noise Compensation for Subspace Gaussian Mixture Models Noise ompensation for ubspace Gaussian Mixture Models Liang Lu University of Edinburgh Joint work with KK hin, A. Ghoshal and. enals Liang Lu, Interspeech, eptember, 2012 Outline Motivation ubspace GMM

More information

Hidden Markov Modelling

Hidden Markov Modelling Hidden Markov Modelling Introduction Problem formulation Forward-Backward algorithm Viterbi search Baum-Welch parameter estimation Other considerations Multiple observation sequences Phone-based models

More information

EUSIPCO

EUSIPCO EUSIPCO 213 1569744273 GAMMA HIDDEN MARKOV MODEL AS A PROBABILISTIC NONNEGATIVE MATRIX FACTORIZATION Nasser Mohammadiha, W. Bastiaan Kleijn, Arne Leijon KTH Royal Institute of Technology, Department of

More information

CSC321 Lecture 20: Autoencoders

CSC321 Lecture 20: Autoencoders CSC321 Lecture 20: Autoencoders Roger Grosse Roger Grosse CSC321 Lecture 20: Autoencoders 1 / 16 Overview Latent variable models so far: mixture models Boltzmann machines Both of these involve discrete

More information

Computational Genomics and Molecular Biology, Fall

Computational Genomics and Molecular Biology, Fall Computational Genomics and Molecular Biology, Fall 2011 1 HMM Lecture Notes Dannie Durand and Rose Hoberman October 11th 1 Hidden Markov Models In the last few lectures, we have focussed on three problems

More information

Gaussian Mixture Model Uncertainty Learning (GMMUL) Version 1.0 User Guide

Gaussian Mixture Model Uncertainty Learning (GMMUL) Version 1.0 User Guide Gaussian Mixture Model Uncertainty Learning (GMMUL) Version 1. User Guide Alexey Ozerov 1, Mathieu Lagrange and Emmanuel Vincent 1 1 INRIA, Centre de Rennes - Bretagne Atlantique Campus de Beaulieu, 3

More information

1. Markov models. 1.1 Markov-chain

1. Markov models. 1.1 Markov-chain 1. Markov models 1.1 Markov-chain Let X be a random variable X = (X 1,..., X t ) taking values in some set S = {s 1,..., s N }. The sequence is Markov chain if it has the following properties: 1. Limited

More information

10. Hidden Markov Models (HMM) for Speech Processing. (some slides taken from Glass and Zue course)

10. Hidden Markov Models (HMM) for Speech Processing. (some slides taken from Glass and Zue course) 10. Hidden Markov Models (HMM) for Speech Processing (some slides taken from Glass and Zue course) Definition of an HMM The HMM are powerful statistical methods to characterize the observed samples of

More information

Acoustic Vector Sensor based Speech Source Separation with Mixed Gaussian-Laplacian Distributions

Acoustic Vector Sensor based Speech Source Separation with Mixed Gaussian-Laplacian Distributions Acoustic Vector Sensor based Speech Source Separation with Mixed Gaussian-Laplacian Distributions Xiaoyi Chen, Atiyeh Alinaghi, Xionghu Zhong and Wenwu Wang Department of Acoustic Engineering, School of

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

Robust Speaker Identification

Robust Speaker Identification Robust Speaker Identification by Smarajit Bose Interdisciplinary Statistical Research Unit Indian Statistical Institute, Kolkata Joint work with Amita Pal and Ayanendranath Basu Overview } } } } } } }

More information

This is the author s version of a work that was submitted/accepted for publication in the following source:

This is the author s version of a work that was submitted/accepted for publication in the following source: This is the author s version of a work that was submitted/accepted for publication in the following source: Wallace, R., Baker, B., Vogt, R., & Sridharan, S. (2010) Discriminative optimisation of the figure

More information

IDIAP. Martigny - Valais - Suisse ADJUSTMENT FOR THE COMPENSATION OF MODEL MISMATCH IN SPEAKER VERIFICATION. Frederic BIMBOT + Dominique GENOUD *

IDIAP. Martigny - Valais - Suisse ADJUSTMENT FOR THE COMPENSATION OF MODEL MISMATCH IN SPEAKER VERIFICATION. Frederic BIMBOT + Dominique GENOUD * R E S E A R C H R E P O R T IDIAP IDIAP Martigny - Valais - Suisse LIKELIHOOD RATIO ADJUSTMENT FOR THE COMPENSATION OF MODEL MISMATCH IN SPEAKER VERIFICATION Frederic BIMBOT + Dominique GENOUD * IDIAP{RR

More information

Diagonal Priors for Full Covariance Speech Recognition

Diagonal Priors for Full Covariance Speech Recognition Diagonal Priors for Full Covariance Speech Recognition Peter Bell 1, Simon King 2 Centre for Speech Technology Research, University of Edinburgh Informatics Forum, 10 Crichton St, Edinburgh, EH8 9AB, UK

More information

Design and Implementation of Speech Recognition Systems

Design and Implementation of Speech Recognition Systems Design and Implementation of Speech Recognition Systems Spring 2013 Class 7: Templates to HMMs 13 Feb 2013 1 Recap Thus far, we have looked at dynamic programming for string matching, And derived DTW from

More information

Boundary Contraction Training for Acoustic Models based on Discrete Deep Neural Networks

Boundary Contraction Training for Acoustic Models based on Discrete Deep Neural Networks INTERSPEECH 2014 Boundary Contraction Training for Acoustic Models based on Discrete Deep Neural Networks Ryu Takeda, Naoyuki Kanda, and Nobuo Nukaga Central Research Laboratory, Hitachi Ltd., 1-280, Kokubunji-shi,

More information

Hidden Markov Models and Gaussian Mixture Models

Hidden Markov Models and Gaussian Mixture Models Hidden Markov Models and Gaussian Mixture Models Hiroshi Shimodaira and Steve Renals Automatic Speech Recognition ASR Lectures 4&5 23&27 January 2014 ASR Lectures 4&5 Hidden Markov Models and Gaussian

More information

Front-End Factor Analysis For Speaker Verification

Front-End Factor Analysis For Speaker Verification IEEE TRANSACTIONS ON AUDIO, SPEECH AND LANGUAGE PROCESSING Front-End Factor Analysis For Speaker Verification Najim Dehak, Patrick Kenny, Réda Dehak, Pierre Dumouchel, and Pierre Ouellet, Abstract This

More information

An Evolutionary Programming Based Algorithm for HMM training

An Evolutionary Programming Based Algorithm for HMM training An Evolutionary Programming Based Algorithm for HMM training Ewa Figielska,Wlodzimierz Kasprzak Institute of Control and Computation Engineering, Warsaw University of Technology ul. Nowowiejska 15/19,

More information