Wavelet-based techniques for speech recognition

In this thesis, new wavelet-based techniques have been developed for the extraction of features from speech signals for the purpose of automatic speech recognition (ASR). One of the advantages of the wavelet transform over the short time Fourier transform (STFT) is its capability to process non-stationary signals. Since speech signals are not strictly stationary the wavelet transform is a better choice for time-frequency transformation of these signals. In addition it has compactly supported basis functions, thereby reducing the amount of computation as opposed to STFT where an overlapping window is needed. [Continues.

Acoustic Features for Environmental Sound Analysis

International audienceMost of the time it is nearly impossible to differentiate between particular type of sound events from a waveform only. Therefore, frequency domain and time-frequency domain representations have been used for years providing representations of the sound signals that are more inline with the human perception. However, these representations are usually too generic and often fail to describe specific content that is present in a sound recording. A lot of work have been devoted to design features that could allow extracting such specific information leading to a wide variety of hand-crafted features. During the past years, owing to the increasing availability of medium scale and large scale sound datasets, an alternative approach to feature extraction has become popular, the so-called feature learning. Finally, processing the amount of data that is at hand nowadays can quickly become overwhelming. It is therefore of paramount importance to be able to reduce the size of the dataset in the feature space. The general processing chain to convert an sound signal to a feature vector that can be efficiently exploited by a classifier and the relation to features used for speech and music processing are described is this chapter

A survey on artificial intelligence-based acoustic source identification

The concept of Acoustic Source Identification (ASI), which refers to the process of identifying noise sources has attracted increasing attention in recent years. The ASI technology can be used for surveillance, monitoring, and maintenance applications in a wide range of sectors, such as defence, manufacturing, healthcare, and agriculture. Acoustic signature analysis and pattern recognition remain the core technologies for noise source identification. Manual identification of acoustic signatures, however, has become increasingly challenging as dataset sizes grow. As a result, the use of Artificial Intelligence (AI) techniques for identifying noise sources has become increasingly relevant and useful. In this paper, we provide a comprehensive review of AI-based acoustic source identification techniques. We analyze the strengths and weaknesses of AI-based ASI processes and associated methods proposed by researchers in the literature. Additionally, we did a detailed survey of ASI applications in machinery, underwater applications, environment/event source recognition, healthcare, and other fields. We also highlight relevant research directions

Recent Advances in Signal Processing

The signal processing task is a very critical issue in the majority of new technological inventions and challenges in a variety of applications in both science and engineering fields. Classical signal processing techniques have largely worked with mathematical models that are linear, local, stationary, and Gaussian. They have always favored closed-form tractability over real-world accuracy. These constraints were imposed by the lack of powerful computing tools. During the last few decades, signal processing theories, developments, and applications have matured rapidly and now include tools from many areas of mathematics, computer science, physics, and engineering. This book is targeted primarily toward both students and researchers who want to be exposed to a wide variety of signal processing techniques and algorithms. It includes 27 chapters that can be categorized into five different areas depending on the application at hand. These five categories are ordered to address image processing, speech processing, communication systems, time-series analysis, and educational packages respectively. The book has the advantage of providing a collection of applications that are completely independent and self-contained; thus, the interested reader can choose any chapter and skip to another without losing continuity

Models and analysis of vocal emissions for biomedical applications

This book of Proceedings collects the papers presented at the 3rd International Workshop on Models and Analysis of Vocal Emissions for Biomedical Applications, MAVEBA 2003, held 10-12 December 2003, Firenze, Italy. The workshop is organised every two years, and aims to stimulate contacts between specialists active in research and industrial developments, in the area of voice analysis for biomedical applications. The scope of the Workshop includes all aspects of voice modelling and analysis, ranging from fundamental research to all kinds of biomedical applications and related established and advanced technologies

Increasing Accuracy Performance through Optimal Feature Extraction Algorithms

This research developed models and techniques to improve the three key modules of popular recognition systems: preprocessing, feature extraction, and classification. Improvements were made in four key areas: processing speed, algorithm complexity, storage space, and accuracy. The focus was on the application areas of the face, traffic sign, and speaker recognition. In the preprocessing module of facial and traffic sign recognition, improvements were made through the utilization of grayscaling and anisotropic diffusion. In the feature extraction module, improvements were made in two different ways; first, through the use of mixed transforms and second through a convolutional neural network (CNN) that best fits specific datasets. The mixed transform system consists of various combinations of the Discrete Wavelet Transform (DWT) and Discrete Cosine Transform (DCT), which have a reliable track record for image feature extraction. In terms of the proposed CNN, a neuroevolution system was used to determine the characteristics and layout of a CNN to best extract image features for particular datasets. In the speaker recognition system, the improvement to the feature extraction module comprised of a quantized spectral covariance matrix and a two-dimensional Principal Component Analysis (2DPCA) function. In the classification module, enhancements were made in visual recognition through the use of two neural networks: the multilayer sigmoid and convolutional neural network. Results show that the proposed improvements in the three modules led to an increase in accuracy as well as reduced algorithmic complexity, with corresponding reductions in storage space and processing time

Speaker recognition: current state and experiment

[ANGLÈS] In this thesis the operation of the speaker recognition systems is described and the state of the art of the main working blocks is studied. All the research papers looked through can be found in the References. As voice is unique to the individual, it has emerged as a viable authentication method. There are several problems that should be considered as the presence of noise in the environment and changes in the voice of the speakers due to sickness for example. These systems combine knowledge from signal processing for the feature extraction part and signal modeling for the classification and decision part. There are several techniques for the feature extraction and the pattern matching blocks, so it is quite tricky to establish a unique and optimum solution. MFCC and DTW are the most common techniques for each block, respectively. They are discussed in this document, with a special emphasis on their drawbacks, that motivate new techniques which are also presented here. A search through the Internet is done in order to find commercial working implementations, which are quite rare, then a basic introduction to Praat is presented. Finally, some intra-speaker and inter-speaker tests are done using this software.[CASTELLÀ] En esta tesis, el funcionamento de los sistemas de reconocimiento del hablante es descrito y el estado del arte de los principales bloques de funcionamento es estudiado. Todos los documentos de investigación consultados se encuentran en las referencias. Dado que la voz es única al individuo, se ha vuelto un método viable de identificación. Hay varios problemas que han de ser considerados, como la presencia de ruido en el ambiente y los cambios en la voz de los hablantes, por ejemplo debido a enfermedades. Estos sistemas combinan conocimiento de procesado de señal en la parte de extracción de características de la voz y modelaje de señal en la parte de clasificación y decisión. Hay diferentes técnicas para la extracción de las características, y para el tratamiento de la similitud entre patrones, de tal manera que es complicado establecer una única y óptima solución. MFCC y DTW son las técnicas más comunes para cada bloque, respectivamente. Son tratadas en este documento, haciendo énfasis en sus problemas, que motivan nuevas técnicas, que también son presentadas aquí. Se realiza una búsqueda por Internet, para encontrar productos comerciales implementados, que son pocos, posteriormente se hace una introducción al software Praat. Finalmente, se realizan algunos intra-speaker i inter-speaker tests usando este programa.[CATALÀ] En aquesta tesi, el funcionament dels sistemes de reconeixement del parlant és descrit i l'estat de l'art dels principals blocs de funcionament és estudiat. Tots els documents de recerca consultats es troben a les referències. Donat que la veu és única a l'individu, ha esdevingut un mètode viable d'identificació. Hi ha diversos problemes que han de ser considerats, com ara la presència de soroll en l'ambient i els canvis en la veu dels parlants, per exemple deguts a malalties. Aquests sistemes combinen coneixement de processament de senyal en la part d'extracció de característiques de la veu i modelatge de senyal en la part de classificació i decisió. Hi ha diferents tècniques per a l'extracció de les característiques, i per al tractament de la similitud entre patrons, de tal manera que és complicat establir una única i òptima solució. MFCC i DTW són les tècniques més comunes per a cada bloc, respectivament. Són tractades en aquest document, fent èmfasi en els seus problemes, que motiven noves tècniques, que també són presentades aquí. Es realitza una cerca per Internet, per tal de trobar productes comercials implementats, que són pocs, posteriorment es fa una introducció al software Praat. Finalment, es realitzen alguns intra-speaker i inter-speaker tests fent servir aquest programa

Automatic speech recognition: from study to practice

Today, automatic speech recognition (ASR) is widely used for different purposes such as robotics, multimedia, medical and industrial application. Although many researches have been performed in this field in the past decades, there is still a lot of room to work. In order to start working in this area, complete knowledge of ASR systems as well as their weak points and problems is inevitable. Besides that, practical experience improves the theoretical knowledge understanding in a reliable way. Regarding to these facts, in this master thesis, we have first reviewed the principal structure of the standard HMM-based ASR systems from technical point of view. This includes, feature extraction, acoustic modeling, language modeling and decoding. Then, the most significant challenging points in ASR systems is discussed. These challenging points address different internal components characteristics or external agents which affect the ASR systems performance. Furthermore, we have implemented a Spanish language recognizer using HTK toolkit. Finally, two open research lines according to the studies of different sources in the field of ASR has been suggested for future work

Analysis of constant-Q filterbank based representations for speech emotion recognition

International audienceThis work analyzes the constant-Q filterbank-based time-frequency representations for speech emotion recognition (SER). Constant-Q filterbank provides non-linear spectrotemporal representation with higher frequency resolution at low frequencies. Our investigation reveals how the increased low-frequency resolution benefits SER. The time-domain comparative analysis between short-term mel-frequency spectral coefficients (MFSCs) and constant-Q filterbank-based features, namely constant-Q transform (CQT) and continuous wavelet transform (CWT), reveals that constant-Q representations provide higher time-invariance at low-frequencies. This provides increased robustness against emotion irrelevant temporal variations in pitch, especially for low-arousal emotions. The corresponding frequency-domain analysis over different emotion classes shows better resolution of pitch harmonics in constant-Q-based time-frequency representations than MFSC. These advantages of constant-Q representations are further consolidated by SER performance in the extensive evaluation of features over four publicly available databases with six advanced deep neural network architectures as the back-end classifiers. Our inferences in this study hint toward the suitability and potentiality of constant-Q features for SER