Weighted LDA techniques for I-vector based speaker verification

This paper introduces the Weighted Linear Discriminant Analysis (WLDA) technique, based upon the weighted pairwise Fisher criterion, for the purposes of improving i-vector speaker verification in the presence of high intersession variability. By taking advantage of the speaker discriminative information that is available in the distances between pairs of speakers clustered in the development i-vector space, the WLDA technique is shown to provide an improvement in speaker verification performance over traditional Linear Discriminant Analysis (LDA) approaches. A similar approach is also taken to extend the recently developed Source Normalised LDA (SNLDA) into Weighted SNLDA (WSNLDA) which, similarly, shows an improvement in speaker verification performance in both matched and mismatched enrolment/verification conditions. Based upon the results presented within this paper using the NIST 2008 Speaker Recognition Evaluation dataset, we believe that both WLDA and WSNLDA are viable as replacement techniques to improve the performance of LDA and SNLDA-based i-vector speaker verification

Integration of speech biometrics in a phone payment system: text-independent speaker verification

Integration of a speaker recognition system in a payment system by phone.Nowadays, the integration of biometrics in security systems is a prominent research and application field. Also, it is clear that speech is the most common form of communication, which makes a swell candidate. While using speech as a biometric, one could say there are two types of systems that should be analyzed: those systems which do know what the speaker is going to say upon verification and those that do not. This degree thesis offers an overview of both systems, focusing on those that do not know what the speaker is going to say beforehand, also known as textindependent systems. To be able to determine which would be the best approach to integrate speech biometrics into a security system, both types of systems are compared; and two methodologies are also analyzed for the text-independent system. To conclude, one of those methodologies is implemented in a software library which allows the creation a text-independent speaker verification system.En l’actualitat, la integració de biometries en els sistemes de seguretat és una branca d’investigació i aplicacions prominent. A més a més, la veu és un dels mitjans més comuns de comunicació, cosa que fa que sigui una bona candidata per a aquests sistemes. Si prenem la parla com a biometria, es pot dir que hi ha dos tipus de sistemes bastant diferenciats a analitzar: aquells sistemes els quals saben el que dirà la persona que s’intenta verificar i aquells que no saben el que dirà. Aquest treball ofereix una visió àmplia dels dos tipus de sistemes, centrant-se en els sistemes on no es sap el que es dirà, també coneguts com sistemes de text independent. Per decidir quin seria la millor manera d’integrar la parla com a biometria en un sistema de seguretat, es comparen ambdós sistemes i, en el cas del sistema de text independent, es comparen també dues metodologies diferents. Per acabar, s’implementa una d’aquestes metodologies a unes llibreries de software per dur a terme un sistema de verificació de locutor amb text independent.En la actualidad, la integración de biometrías en los sistemas de seguridad es una rama de investigación y de aplicaciones prominente. Además, está claro que la voz es el medio más común de comunicación y es por eso que es una buena candidata. Usando el habla como biometría, se podría decir que hay dos tipos de sistemas diferentes a analizar: aquellos sistemas que saben de antemano aquello que va a decir el locutor que intenta verificarse y aquellos que no lo saben. Este trabajo ofrece una visión amplia de los dos tipos de sistemas, centrándose en los sistemas donde aquello que se va a decir no se sabe, también conocidos como sistemas de texto independiente. Para decir cuál sería la mejor manera de integrar el habla como biometría en un sistema de seguridad se comparan ambos sistemas y, en el caso del sistema de texto independiente, se comparan también dos metodologías diferentes. Para finalizar, se implementa una de estas últimas en unas librerías de software para poder llevar a cabo un sistema de verificación de locutor de texto independiente

Memory and computation effective approaches for i-vector extraction

This paper focuses on the extraction of i-vectors, a compact representation of spoken utterances that is used by most of the state-of-the-art speaker recognition systems. This work was mainly motivated by the need of reducing the memory demand of the huge data structures that are usually precomputed for fast computation of the i-vectors. We propose a set of new approaches allowing accurate i-vector extraction but requiring less memory, showing their relations with the standard computation method introduced for eigenvoices. We analyze the time and memory resources required by these solutions, which are suited to different fields of application, and we show that it is possible to get accurate results with solutions that reduce both computation time and memory demand compared with the standard solutio

Open-set Speaker Identification

This study is motivated by the growing need for effective extraction of intelligence and evidence from audio recordings in the fight against crime, a need made ever more apparent with the recent expansion of criminal and terrorist organisations. The main focus is to enhance open-set speaker identification process within the speaker identification systems, which are affected by noisy audio data obtained under uncontrolled environments such as in the street, in restaurants or other places of businesses. Consequently, two investigations are initially carried out including the effects of environmental noise on the accuracy of open-set speaker recognition, which thoroughly cover relevant conditions in the considered application areas, such as variable training data length, background noise and real world noise, and the effects of short and varied duration reference data in open-set speaker recognition. The investigations led to a novel method termed “vowel boosting” to enhance the reliability in speaker identification when operating with varied duration speech data under uncontrolled conditions. Vowels naturally contain more speaker specific information. Therefore, by emphasising this natural phenomenon in speech data, it enables better identification performance. The traditional state-of-the-art GMM-UBMs and i-vectors are used to evaluate “vowel boosting”. The proposed approach boosts the impact of the vowels on the speaker scores, which improves the recognition accuracy for the specific case of open-set identification with short and varied duration of speech material

Deep learning for i-vector speaker and language recognition

Over the last few years, i-vectors have been the state-of-the-art technique in speaker and language recognition. Recent advances in Deep Learning (DL) technology have improved the quality of i-vectors but the DL techniques in use are computationally expensive and need speaker or/and phonetic labels for the background data, which are not easily accessible in practice. On the other hand, the lack of speaker-labeled background data makes a big performance gap, in speaker recognition, between two well-known cosine and Probabilistic Linear Discriminant Analysis (PLDA) i-vector scoring techniques. It has recently been a challenge how to fill this gap without speaker labels, which are expensive in practice. Although some unsupervised clustering techniques are proposed to estimate the speaker labels, they cannot accurately estimate the labels. This thesis tries to solve the problems above by using the DL technology in different ways, without any need of speaker or phonetic labels. In order to fill the performance gap between cosine and PLDA scoring given unlabeled background data, we have proposed an impostor selection algorithm and a universal model adaptation process in a hybrid system based on Deep Belief Networks (DBNs) and Deep Neural Networks (DNNs) to discriminatively model each target speaker. In order to have more insight into the behavior of DL techniques in both single and multi-session speaker enrollment tasks, some experiments have been carried out in both scenarios. Experiments on the National Institute of Standard and Technology (NIST) 2014 i-vector challenge show that 46% of this performance gap, in terms of minDCF, is filled by the proposed DL-based system. Furthermore, the score combination of the proposed DL-based system and PLDA with estimated labels covers 79% of this gap. In the second line of the research, we have developed an efficient alternative vector representation of speech by keeping the computational cost as low as possible and avoiding phonetic labels, which are not always accessible. The proposed vectors will be based on both Gaussian Mixture Models (GMMs) and Restricted Boltzmann Machines (RBMs) and will be referred to as GMM-RBM vectors. The role of RBM is to learn the total speaker and session variability among background GMM supervectors. This RBM, which will be referred to as Universal RBM (URBM), will then be used to transform unseen supervectors to the proposed low dimensional vectors. The use of different activation functions for training the URBM and different transformation functions for extracting the proposed vectors are investigated. At the end, a variant of Rectified Linear Unit (ReLU) which is referred to as Variable ReLU (VReLU) is proposed. Experiments on the core test condition 5 of the NIST Speaker Recognition Evaluation (SRE) 2010 show that comparable results with conventional i-vectors are achieved with a clearly lower computational load in the vector extraction process. Finally, for the Language Identification (LID) application, we have proposed a DNN architecture to model effectively the i-vector space of four languages, English, Spanish, German, and Finnish, in the car environment. Both raw i-vectors and session variability compensated i-vectors are evaluated as input vectors to DNN. The performance of the proposed DNN architecture is compared with both conventional GMM-UBM and i-vector/Linear Discriminant Analysis (LDA) systems considering the effect of duration of signals. It is shown that the signals with duration between 2 and 3 sec meet the accuracy and speed requirements of this application, in which the proposed DNN architecture outperforms GMM-UBM and i-vector/LDA systems by 37% and 28%, respectively.En los últimos años, los i-vectores han sido la técnica de referencia en el reconocimiento de hablantes y de idioma. Los últimos avances en la tecnología de Aprendizaje Profundo (Deep Learning. DL) han mejorado la calidad de los i-vectores, pero las técnicas DL en uso son computacionalmente costosas y necesitan datos etiquetados para cada hablante y/o unidad fon ética, los cuales no son fácilmente accesibles en la práctica. La falta de datos etiquetados provoca una gran diferencia de los resultados en el reconocimiento de hablante con i-vectors entre las dos técnicas de evaluación más utilizados: distancia coseno y Análisis Lineal Discriminante Probabilístico (PLDA). Por el momento, sigue siendo un reto cómo reducir esta brecha sin disponer de las etiquetas de los hablantes, que son costosas de obtener. Aunque se han propuesto algunas técnicas de agrupamiento sin supervisión para estimar las etiquetas de los hablantes, no pueden estimar las etiquetas con precisión. Esta tesis trata de resolver los problemas mencionados usando la tecnología DL de diferentes maneras, sin necesidad de etiquetas de hablante o fon éticas. Con el fin de reducir la diferencia de resultados entre distancia coseno y PLDA a partir de datos no etiquetados, hemos propuesto un algoritmo selección de impostores y la adaptación a un modelo universal en un sistema hibrido basado en Deep Belief Networks (DBN) y Deep Neural Networks (DNN) para modelar a cada hablante objetivo de forma discriminativa. Con el fin de tener más información sobre el comportamiento de las técnicas DL en las tareas de identificación de hablante en una única sesión y en varias sesiones, se han llevado a cabo algunos experimentos en ambos escenarios. Los experimentos utilizando los datos del National Institute of Standard and Technology (NIST) 2014 i-vector Challenge muestran que el 46% de esta diferencia de resultados, en términos de minDCF, se reduce con el sistema propuesto basado en DL. Además, la combinación de evaluaciones del sistema propuesto basado en DL y PLDA con etiquetas estimadas reduce el 79% de esta diferencia. En la segunda línea de la investigación, hemos desarrollado una representación vectorial alternativa eficiente de la voz manteniendo el coste computacional lo más bajo posible y evitando las etiquetas fon éticas, Los vectores propuestos se basan tanto en el Modelo de Mezcla de Gaussianas (GMM) y en las Maquinas Boltzmann Restringidas (RBM), a los que se hacer referencia como vectores GMM-RBM. El papel de la RBM es aprender la variabilidad total del hablante y de la sesión entre los supervectores del GMM gen érico. Este RBM, al que se hará referencia como RBM Universal (URBM), se utilizará para transformar supervectores ocultos en los vectores propuestos, de menor dimensión. Además, se estudia el uso de diferentes funciones de activación para el entrenamiento de la URBM y diferentes funciones de transformación para extraer los vectores propuestos. Finalmente, se propone una variante de la Unidad Lineal Rectificada (ReLU) a la que se hace referencia como Variable ReLU (VReLU). Los experimentos sobre los datos de la condición 5 del test de la NIST Speaker Recognition Evaluation (SRE) 2010 muestran que se han conseguidos resultados comparables con los i-vectores convencionales, con una carga computacional claramente inferior en el proceso de extracción de vectores. Por último, para la aplicación de Identificación de Idioma (LID), hemos propuesto una arquitectura DNN para modelar eficazmente en el entorno del coche el espacio i-vector de cuatro idiomas: inglés, español, alemán y finlandés. Tanto los i-vectores originales como los i-vectores propuestos son evaluados como vectores de entrada a DNN. El rendimiento de la arquitectura DNN propuesta se compara con los sistemas convencionales GMM-UBM y i-vector/Análisis Discriminante Lineal (LDA) considerando el efecto de la duración de las señales. Se muestra que en caso de señales con una duración entre 2 y 3 se obtienen resultados satisfactorios en cuanto a precisión y resultados, superando a los sistemas GMM-UBM y i-vector/LDA en un 37% y 28%, respectivament

An investigation of supervector regression for forensic voice comparison on small data

International audienceThe present paper deals with an observer design for a nonlinear lateral vehicle model. The nonlinear model is represented by an exact Takagi-Sugeno (TS) model via the sector nonlinearity transformation. A proportional multiple integral observer (PMIO) based on the TS model is designed to estimate simultaneously the state vector and the unknown input (road curvature). The convergence conditions of the estimation error are expressed under LMI formulation using the Lyapunov theory which guaranties bounded error. Simulations are carried out and experimental results are provided to illustrate the proposed observer

Speaker recognition by means of Deep Belief Networks

Most state-of-the-art speaker recognition systems are based on Gaussian Mixture Models (GMMs), where a speech segment is represented by a compact representation, referred to as "identity vector" (ivector for short), extracted by means of Factor Analysis. The main advantage of this representation is that the problem of intersession variability is deferred to a second stage, dealing with low-dimensional vectors rather than with the high-dimensional space of the GMM means. In this paper, we propose to use as a pseudo-ivector extractor a Deep Belief Network (DBN) architecture, trained with the utterances of several hundred speakers. In this approach, the DBN performs a non-linear transformation of the input features, which produces the probability that an output unit is on, given the input features. We model the distribution of the output units, given an utterance, by a reduced set of parameters that embed the speaker characteristics. Tested on the dataset exploited for training the systems that have been used for the NIST 2012 Speaker Recognition Evaluation, this approach shows promising result

Analysis of Large-Scale SVM Training Algorithms for Language and Speaker Recognition

This paper compares a set of large scale support vector machine (SVM) training algorithms for language and speaker recognition tasks.We analyze five approaches for training phonetic and acoustic SVM models for language recognition. We compare the performance of these approaches as a function of the training time required by each of them to reach convergence, and we discuss their scalability towards large corpora. Two of these algorithms can be used in speaker recognition to train a SVM that classifies pairs of utterances as either belonging to the same speaker or to two different speakers. Our results show that the accuracy of these algorithms is asymptotically equivalent, but they have different behavior with respect to the time required to converge. Some of these algorithms not only scale linearly with the training set size, but are also able to give their best results after just a few iterations. State-of-the-art performance has been obtained in the female subset of the NIST 2010 Speaker Recognition Evaluation extended core test using a single SVM syste