Robust speaker recognition in presence of non-trivial environmental noise (toward greater biometric security)

The aim of this thesis is to investigate speaker recognition in the presence of environmental noise, and to develop a robust speaker recognition method. Recently, Speaker Recognition has been the object of considerable research due to its wide use in various areas. Despite major developments in this field, there are still many limitations and challenges. Environmental noises and their variations are high up in the list of challenges since it impossible to provide a noise free environment. A novel approach is proposed to address the issue of performance degradation in environmental noise. This approach is based on the estimation of signal-to-noise ratio (SNR) and detection of ambient noise from the recognition signal to re-train the reference model for the claimed speaker and to generate a new adapted noisy model to decrease the noise mismatch with recognition utterances. This approach is termed “Training on the fly” for robustness of speaker recognition under noisy environments. To detect the noise in the recognition signal two different techniques are proposed: the first technique including generating an emulated noise depending on estimated power spectrum of the original noise using 1/3 octave band filter bank and white noise signal. This emulated noise become close enough to original one that includes in the input signal (recognition signal). The second technique deals with extracting the noise from the input signal using one of speech enhancement algorithm with spectral subtraction to find the noise in the signal. Training on the fly approach (using both techniques) has been examined using two feature approaches and two different kinds of artificial clean and noisy speech databases collected in different environments. Furthermore, the speech samples were text independent. The training on the fly approach is a significant improvement in performance when compared with the performance of conventional speaker recognition (based on clean reference models). Moreover, the training on the fly based on noise extraction showed the best results for all types of noisy data

Acoustic model selection for recognition of regional accented speech

Accent is cited as an issue for speech recognition systems. Our experiments showed that the ASR word error rate is up to seven times greater for accented speech compared with standard British English. The main objective of this research is to develop Automatic Speech Recognition (ASR) techniques that are robust to accent variation. We applied different acoustic modelling techniques to compensate for the effects of regional accents on the ASR performance. For conventional GMM-HMM based ASR systems, we showed that using a small amount of data from a test speaker to choose an accent dependent model using an accent identification system, or building a model using the data from N neighbouring speakers in AID space, will result in superior performance compared to that obtained with unsupervised or supervised speaker adaptation. In addition we showed that using a DNN-HMM rather than a GMM-HMM based acoustic model would improve the recognition accuracy considerably. Even if we apply two stages of accent followed by speaker adaptation to the GMM-HMM baseline system, the GMM-HMM based system will not outperform the baseline DNN-HMM based system. For more contemporary DNN-HMM based ASR systems we investigated how adding different types of accented data to the training set can provide better recognition accuracy on accented speech. Finally, we proposed a new approach for visualisation of the AID feature space. This is helpful in analysing the AID recognition accuracies and analysing AID confusion matrices

Advances in Binary and Multiclass Audio Segmentation with Deep Learning Techniques

Los avances tecnológicos acaecidos en la última década han cambiado completamente la forma en la que la población interactúa con el contenido multimedia. Esto ha propiciado un aumento significativo tanto en la generación como el consumo de dicho contenido. El análisis y la anotación manual de toda esta información no son factibles dado el gran volumen actual, lo que releva la necesidad de herramientas automáticas que ayuden en la transición hacia flujos de trabajo asistidos o parcialmente automáticos. En los últimos años, la mayoría de estas herramientas están basadas en el uso de redes neuronales y deep learning. En este contexto, el trabajo que se describe en esta tesis se centra en el ámbito de la extracción de información a partir de señales de audio. Particularmente, se estudia la tarea de segmentación de audio, cuyo principal objetivo es obtener una secuencia de etiquetas que aíslen diferentes regiones en una señal de entrada de acuerdo con una serie de características descritas en un conjunto predefinido de clases, como por ejemplo voz, música o ruido.La primera parte de esta memoria esta centrada en la tarea de detección de actividad de voz. Recientemente, diferentes campañas de evaluación internacionales han propuesto esta tarea como uno de sus retos. Entre ellas se encuentra el reto Fearless steps, que trabaja con audios de las grabaciones de las misiones Apollo de la NASA. Para este reto, se propone una solución basada en aprendizaje supervisado usando una red convolucional recurrente como clasificador. La principal contribución es un método que combina información de filtros de 1D y 2D en la etapa convolucional para que sea procesada posteriormente por la etapa recurrente. Motivado por la introducción de los datos del reto Fearless steps, se plantea una evaluación de diferentes técnicas de adaptación de dominio, con el objetivo de comprobar las prestaciones de un sistema entrenado con datos de dominios habituales y evaluado en este nuevo dominio presentado en el reto. Los métodos descritos no requieren de etiquetas en el dominio objetivo, lo que facilita su uso en aplicaciones prácticas. En términos generales, se observa que los métodos que buscan minimizar el cambio en las distribuciones estadísticas entre los dominios fuente y objetivo obtienen los resultados mas prometedores. Los avances recientes en técnicas de representación obtenidas mediante aprendizaje auto-supervisado han demostrado grandes mejoras en prestaciones en varias tareas relacionadas con el procesado de voz. Siguiendo esta línea, se plantea la incorporación de dichas representaciones en la tarea de detección de actividad de voz. Las ediciones más recientes del reto Fearless steps modificaron su propósito, buscando ahora evaluar las capacidades de generalización de los sistemas. El objetivo entonces con las técnicas introducidas es poder beneficiarse de grandes cantidades de datos no etiquetados para mejorar la robustez del sistema. Los resultados experimentales sugieren que el aprendizaje auto-supervisado de representaciones permite obtener sistemas que son mucho menos sensibles al cambio de dominio.En la segunda parte de este documento se analiza una tarea de segmentación de audio más genérica que busca clasificar de manera simultanea una señal de audio como voz, música, ruido o una combinación de estas. En el contexto de los datos propuesto para el reto de segmentación de audio Albayzín 2010, se presenta un enfoque basado en el uso de redes neuronales recurrentes como clasificador principal, y un modelo de postprocesado integrado por modelos ocultos de Markov. Se introduce un nuevo bloque en la arquitectura neuronal con el objetivo de eliminar la información temporal redundante, mejorando las prestaciones y reduciendo el numero de operaciones por segundo al mismo tiempo. Esta propuesta obtuvo mejores prestaciones que soluciones presentadas anteriormenteen la literatura, y que aproximaciones similares basadas en redes neuronales profundas. Mientras que los resultados con aprendizaje auto-supervisado de representaciones eran prometedores en tareas de segmentación binaria, si se aplican en tareas de segmentación multiclase surgen una serie de cuestiones. Las técnicas habituales de aumento de datos que se aplican en el entrenamiento fuerzan al modelo a compensar el ruido de fondo o la música. En estas condiciones las características obtenidas podrían no representar de manera precisa aquellas clases generadas de manera similar a las versiones aumentadas vistas en el entrenamiento. Este hecho limita la mejora global de prestaciones observada al aplicar estas técnicas en tareas como la propuesta en la evaluación Albayzín 2010.La última parte de este trabajo ha investigado la aplicación de nuevas funciones de coste en la tarea de segmentación de audio, con el principal objetivo de mitigar los problemas que se derivan de utilizar un conjunto de datos de entrenamiento limitado. Se ha demostrado que nuevas técnicas de optimización basadas en las métricas AUC y AUC parcial pueden mejorar objetivos de entrenamiento tradicionales como la entropía cruzada en varias tareas de detección. Con esta idea en mente, en esta tesis se introducen dichas técnicas en la tarea de detección de música. Considerando que la cantidad de datos etiquetados para esta tarea es limitada comparado con otras tareas, las funciones de coste basadas en la métrica AUC se aplican con el objetivo de mejorar las prestaciones cuando el conjunto de datos de entrenamiento es relativamente pequeño. La mayoría de los sistemas que utilizan las técnicas de optimización basadas en métricas AUC se limitan a tareas binarias ya que ese el ámbito de aplicación habitual de la métrica AUC. Además, el etiquetado de audios con taxonomías más detalladas en las que hay múltiples opciones posibles es más complejo, por lo que la cantidad de audio etiquetada en algunas tareas de segmentación multiclase es limitada. Como una extensión natural, se propone una generalización de las técnicas de optimización basadas en la métrica AUC binaria, de tal manera que se puedan aplicar con un número arbitrario de clases. Dos funciones de coste distintas se introducen, usando como base para su formulación las variaciones multiclase de la métrica AUC propuestas en la literatura: una basada en un enfoque uno contra uno, y otra basada en un enfoque uno contra el resto.<br /

Towards Automatic Speech-Language Assessment for Aphasia Rehabilitation

Speech-based technology has the potential to reinforce traditional aphasia therapy through the development of automatic speech-language assessment systems. Such systems can provide clinicians with supplementary information to assist with progress monitoring and treatment planning, and can provide support for on-demand auxiliary treatment. However, current technology cannot support this type of application due to the difficulties associated with aphasic speech processing. The focus of this dissertation is on the development of computational methods that can accurately assess aphasic speech across a range of clinically-relevant dimensions. The first part of the dissertation focuses on novel techniques for assessing aphasic speech intelligibility in constrained contexts. The second part investigates acoustic modeling methods that lead to significant improvement in aphasic speech recognition and allow the system to work with unconstrained speech samples. The final part demonstrates the efficacy of speech recognition-based analysis in automatic paraphasia detection, extraction of clinically-motivated quantitative measures, and estimation of aphasia severity. The methods and results presented in this work will enable robust technologies for accurately recognizing and assessing aphasic speech, and will provide insights into the link between computational methods and clinical understanding of aphasia.PHDComputer Science & EngineeringUniversity of Michigan, Horace H. Rackham School of Graduate Studieshttps://deepblue.lib.umich.edu/bitstream/2027.42/140840/1/ducle_1.pd

Detecting early signs of dementia in conversation

Dementia can affect a person's speech, language and conversational interaction capabilities. The early diagnosis of dementia is of great clinical importance. Recent studies using the qualitative methodology of Conversation Analysis (CA) demonstrated that communication problems may be picked up during conversations between patients and neurologists and that this can be used to differentiate between patients with Neuro-degenerative Disorders (ND) and those with non-progressive Functional Memory Disorder (FMD). However, conducting manual CA is expensive and difficult to scale up for routine clinical use.\ud This study introduces an automatic approach for processing such conversations which can help in identifying the early signs of dementia and distinguishing them from the other clinical categories (FMD, Mild Cognitive Impairment (MCI), and Healthy Control (HC)). The dementia detection system starts with a speaker diarisation module to segment an input audio file (determining who talks when). Then the segmented files are passed to an automatic speech recogniser (ASR) to transcribe the utterances of each speaker. Next, the feature extraction unit extracts a number of features (CA-inspired, acoustic, lexical and word vector) from the transcripts and audio files. Finally, a classifier is trained by the features to determine the clinical category of the input conversation. Moreover, we investigate replacing the role of a neurologist in the conversation with an Intelligent Virtual Agent (IVA) (asking similar questions). We show that despite differences between the IVA-led and the neurologist-led conversations, the results achieved by the IVA are as good as those gained by the neurologists. Furthermore, the IVA can be used for administering more standard cognitive tests, like the verbal fluency tests and produce automatic scores, which then can boost the performance of the classifier. The final blind evaluation of the system shows that the classifier can identify early signs of dementia with an acceptable level of accuracy and robustness (considering both sensitivity and specificity)

Affective Speech Recognition

Speech, as a medium of interaction, carries two different streams of information. Whereas one stream carries explicit messages, the other one contains implicit information about speakers themselves. Affective speech recognition is a set of theories and tools that intend to automate unfolding the part of the implicit stream that has to do with humans emotion. Application of affective speech recognition is to human computer interaction; a machine that is able to recognize humans emotion could engage the user in a more effective interaction. This thesis proposes a set of analyses and methodologies that advance automatic recognition of affect from speech. The proposed solution spans two dimensions of the problem: speech signal processing, and statistical learning. At the speech signal processing dimension, extraction of speech low-level descriptors is dis- cussed, and a set of descriptors that exploit the spectrum of the signal are proposed, which have shown to be particularly practical for capturing affective qualities of speech. Moreover, consider- ing the non-stationary property of the speech signal, further proposed is a measure of dynamicity that captures that property of speech by quantifying changes of the signal over time. Furthermore, based on the proposed set of low-level descriptors, it is shown that individual human beings are different in conveying emotions, and that parts of the spectrum that hold the affective information are different from one person to another. Therefore, the concept of emotion profile is proposed that formalizes those differences by taking into account different factors such as cultural and gender-specific differences, as well as those distinctions that have to do with individual human beings. At the statistical learning dimension, variable selection is performed to identify speech features that are most imperative to extracting affective information. In doing so, low-level descriptors are distinguished from statistical functionals, therefore, effectiveness of each of the two are studied dependently and independently. The major importance of variable selection as a standalone component of a solution is to real-time application of affective speech recognition. Although thousands of speech features are commonly used to tackle this problem in theory, extracting that many features in a real-time manner is unrealistic, especially for mobile applications. Results of the conducted investigations show that the required number of speech features is far less than the number that is commonly used in the literature of the problem. At the core of an affective speech recognition solution is a statistical model that uses speech features to recognize emotions. Such a model comes with a set of parameters that are estimated through a learning process. Proposed in this thesis is a learning algorithm, developed based on the notion of Hilbert-Schmidt independence criterion and named max-dependence regression, that maximizes the dependence between predicted and actual values of affective qualities. Pearson’s correlation coefficient is commonly used as the measure of goodness of a fit in the literature of affective computing, therefore max-dependence regression is proposed to make the learning and hypothesis testing criteria consistent with one another. Results of this research show that doing so results in higher prediction accuracy. Lastly, sparse representation for affective speech datasets is considered in this thesis. For this purpose, the application of a dictionary learning algorithm based on Hilbert-Schmidt independence criterion is proposed. Dictionary learning is used to identify the most important bases of the data in order to improve the generalization capability of the proposed solution to affective speech recognition. Based on the dictionary learning approach of choice, fusion of feature vectors is proposed. It is shown that sparse representation leads to higher generalization capability for affective speech recognition