Compensation of Nuisance Factors for Speaker and Language Recognition

The variability of the channel and environment is one of the most important factors affecting the performance of text-independent speaker verification systems. The best techniques for channel compensation are model based. Most of them have been proposed for Gaussian mixture models, while in the feature domain blind channel compensation is usually performed. The aim of this work is to explore techniques that allow more accurate intersession compensation in the feature domain. Compensating the features rather than the models has the advantage that the transformed parameters can be used with models of a different nature and complexity and for different tasks. In this paper, we evaluate the effects of the compensation of the intersession variability obtained by means of the channel factors approach. In particular, we compare channel variability modeling in the usual Gaussian mixture model domain, and our proposed feature domain compensation technique. We show that the two approaches lead to similar results on the NIST 2005 Speaker Recognition Evaluation data with a reduced computation cost. We also report the results of a system, based on the intersession compensation technique in the feature space that was among the best participants in the NIST 2006 Speaker Recognition Evaluation. Moreover, we show how we obtained significant performance improvement in language recognition by estimating and compensating, in the feature domain, the distortions due to interspeaker variability within the same language. Index Terms—Factor anal

A Comparative Re-Assessment of Feature Extractors for Deep Speaker Embeddings

Modern automatic speaker verification relies largely on deep neural networks (DNNs) trained on mel-frequency cepstral coefficient (MFCC) features. While there are alternative feature extraction methods based on phase, prosody and long-term temporal operations, they have not been extensively studied with DNN-based methods. We aim to fill this gap by providing extensive re-assessment of 14 feature extractors on VoxCeleb and SITW datasets. Our findings reveal that features equipped with techniques such as spectral centroids, group delay function, and integrated noise suppression provide promising alternatives to MFCCs for deep speaker embeddings extraction. Experimental results demonstrate up to 16.3\% (VoxCeleb) and 25.1\% (SITW) relative decrease in equal error rate (EER) to the baseline.Comment: Accepted to Interspeech 202

Intersession Variability Compensation in Language and Speaker Identification

Variabilita kanálu a hovoru je velmi důležitým problémem v úloze rozpoznávání mluvčího. V současné době je ve velkém množství vědeckých článků uvedeno několik technik pro kompenzaci vlivu kanálu. Kompenzace vlivu kanálu může být implementována jak v doméně modelu, tak i v doménách příznaků i skóre. Relativně nová výkoná technika je takzvaná eigenchannel adaptace pro GMM (Gaussian Mixture Models). Mevýhodou této metody je nemožnost její aplikace na jiné klasifikátory, jako napřílad takzvané SVM (Support Vector Machines), GMM s různým počtem Gausových komponent nebo v rozpoznávání řeči s použitím skrytých markovových modelů (HMM). Řešením může být aproximace této metody, eigenchannel adaptace v doméně příznaků. Obě tyto techniky, eigenchannel adaptace v doméně modelu a doméně příznaků v systémech rozpoznávání mluvčího, jsou uvedeny v této práci. Po dosažení dobrých výsledků v rozpoznávání mluvčího, byl přínos těchto technik zkoumán pro akustický systém rozpoznávání jazyka zahrnující 14 jazyků. V této úloze má nežádoucí vliv nejen variabilita kanálu, ale i variabilita mluvčího. Výsledky jsou prezentovány na datech definovaných pro evaluaci rozpoznávání mluvčího z roku 2006 a evaluaci rozpoznávání jazyka v roce 2007, obě organizované Amerických Národním Institutem pro Standard a Technologie (NIST)Varibiality in the channel and session is an important issue in the text-independent speaker recognition task. To date, several techniques providing channel and session variability compensation were introduced in a number of scientic papers. Such implementation can be done in feature, model and score domain. Relatively new and powerful approach to remove channel distortion is so-called eigenchannel adaptation for Gaussian Mixture Models (GMM). The drawback of the technique is that it is not applicable in its original implementation to different types of classifiers, eg. Support Vector Machines (SVM), GMM with different number of Gaussians or in speech recognition task using Hidden Markov Models (HMM). The solution can be the approximation of the technique, eigenchannel adaptation in feature domain. Both, the original eigenchannel adaptation and eigenchannel adaptation on features in task of speaker recognition are presented. After achieving good results in speaker recognition, contribution of the same techniques was examined in acoustic language identification system with $14$ languages. In this task undesired factors are channel and speaker variability. Presented results are presented on the NIST Speaker Recognition Evaluation 2006 data and NIST Language Recognition Evaluation 2007 data.

Speech Recognition in noisy environment using Deep Learning Neural Network

Recent researches in the field of automatic speaker recognition have shown that methods based on deep learning neural networks provide better performance than other statistical classifiers. On the other hand, these methods usually require adjustment of a significant number of parameters. The goal of this thesis is to show that selecting appropriate value of parameters can significantly improve speaker recognition performance of methods based on deep learning neural networks. The reported study introduces an approach to automatic speaker recognition based on deep neural networks and the stochastic gradient descent algorithm. It particularly focuses on three parameters of the stochastic gradient descent algorithm: the learning rate, and the hidden and input layer dropout rates. Additional attention was devoted to the research question of speaker recognition under noisy conditions. Thus, two experiments were conducted in the scope of this thesis. The first experiment was intended to demonstrate that the optimization of the observed parameters of the stochastic gradient descent algorithm can improve speaker recognition performance under no presence of noise. This experiment was conducted in two phases. In the first phase, the recognition rate is observed when the hidden layer dropout rate and the learning rate are varied, while the input layer dropout rate was constant. In the second phase of this experiment, the recognition rate is observed when the input layers dropout rate and learning rate are varied, while the hidden layer dropout rate was constant. The second experiment was intended to show that the optimization of the observed parameters of the stochastic gradient descent algorithm can improve speaker recognition performance even under noisy conditions. Thus, different noise levels were artificially applied on the original speech signal

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