Investigation of Frame Alignments for GMM-based Digit-prompted Speaker Verification

Frame alignments can be computed by different methods in GMM-based speaker verification. By incorporating a phonetic Gaussian mixture model (PGMM), we are able to compare the performance using alignments extracted from the deep neural networks (DNN) and the conventional hidden Markov model (HMM) in digit-prompted speaker verification. Based on the different characteristics of these two alignments, we present a novel content verification method to improve the system security without much computational overhead. Our experiments on the RSR2015 Part-3 digit-prompted task show that, the DNN based alignment performs on par with the HMM alignment. The results also demonstrate the effectiveness of the proposed Kullback-Leibler (KL) divergence based scoring to reject speech with incorrect pass-phrases.Comment: accepted by APSIPA ASC 201

Speech Recognition

Chapters in the first part of the book cover all the essential speech processing techniques for building robust, automatic speech recognition systems: the representation for speech signals and the methods for speech-features extraction, acoustic and language modeling, efficient algorithms for searching the hypothesis space, and multimodal approaches to speech recognition. The last part of the book is devoted to other speech processing applications that can use the information from automatic speech recognition for speaker identification and tracking, for prosody modeling in emotion-detection systems and in other speech processing applications that are able to operate in real-world environments, like mobile communication services and smart homes

Deep Neural Network Architectures for Large-scale, Robust and Small-Footprint Speaker and Language Recognition

Tesis doctoral inédita leída en la Universidad Autónoma de Madrid, Escuela Politécnica Superior, Departamento de Tecnología Electrónica y de las Comunicaciones. Fecha de lectura : 27-04-2017Artificial neural networks are powerful learners of the information embedded in speech signals. They can provide compact, multi-level, nonlinear representations of temporal sequences and holistic optimization algorithms capable of surpassing former leading paradigms. Artificial neural networks are, therefore, a promising technology that can be used to enhance our ability to recognize speakers and languages–an ability increasingly in demand in the context of new, voice-enabled interfaces used today by millions of users. The aim of this thesis is to advance the state-of-the-art of language and speaker recognition through the formulation, implementation and empirical analysis of novel approaches for large-scale and portable speech interfaces. Its major contributions are: (1) novel, compact network architectures for language and speaker recognition, including a variety of network topologies based on fully-connected, recurrent, convolutional, and locally connected layers; (2) a bottleneck combination strategy for classical and neural network approaches for long speech sequences; (3) the architectural design of the first, public, multilingual, large vocabulary continuous speech recognition system; and (4) a novel, end-to-end optimization algorithm for text-dependent speaker recognition that is applicable to a range of verification tasks. Experimental results have demonstrated that artificial neural networks can substantially reduce the number of model parameters and surpass the performance of previous approaches to language and speaker recognition, particularly in the cases of long short-term memory recurrent networks (used to model the input speech signal), end-to-end optimization algorithms (used to predict languages or speakers), short testing utterances, and large training data collections.Las redes neuronales artificiales son sistemas de aprendizaje capaces de extraer la información embebida en las señales de voz. Son capaces de modelar de forma eficiente secuencias temporales complejas, con información no lineal y distribuida en distintos niveles semanticos, mediante el uso de algoritmos de optimización integral con la capacidad potencial de mejorar los sistemas aprendizaje automático existentes. Las redes neuronales artificiales son, pues, una tecnología prometedora para mejorar el reconocimiento automático de locutores e idiomas; siendo el reconocimiento de de locutores e idiomas, tareas con cada vez más demanda en los nuevos sistemas de control por voz, que ya utilizan millones de personas. Esta tesis tiene como objetivo la mejora del estado del arte de las tecnologías de reconocimiento de locutor y de idioma mediante la formulación, implementación y análisis empírico de nuevos enfoques basados en redes neuronales, aplicables a dispositivos portátiles y a su uso en gran escala. Las principales contribuciones de esta tesis incluyen la propuesta original de: (1) arquitecturas eficientes que hacen uso de capas neuronales densas, localmente densas, recurrentes y convolucionales; (2) una nueva estrategia de combinación de enfoques clásicos y enfoques basados en el uso de las denominadas redes de cuello de botella; (3) el diseño del primer sistema público de reconocimiento de voz, de vocabulario abierto y continuo, que es además multilingüe; y (4) la propuesta de un nuevo algoritmo de optimización integral para tareas de reconocimiento de locutor, aplicable también a otras tareas de verificación. Los resultados experimentales extraídos de esta tesis han demostrado que las redes neuronales artificiales son capaces de reducir el número de parámetros usados por los algoritmos de reconocimiento tradicionales, así como de mejorar el rendimiento de dichos sistemas de forma substancial. Dicha mejora relativa puede acentuarse a través del modelado de voz mediante redes recurrentes de memoria a largo plazo, el uso de algoritmos de optimización integral, el uso de locuciones de evaluation de corta duración y mediante la optimización del sistema con grandes cantidades de datos de entrenamiento

Multimodal spoofing and adversarial examples countermeasure for speaker verification

Authentication mechanisms have always been prevalent in our society — even as far back as Ancient Mesopotamia in the form of seals. Since the advent of the digital age, the need for a good digital authentication technique has soared stemming from the widespread adoption of online platforms and digitized content. Audio-based authentication like speaker verification has been explored as another mechanism for achieving this goal. Specifically, an audio template belonging to the authorized user is stored with the authentication system. This template is later compared with the current input voice to authenticate the current user. Audio spoofing refers to attacks used to fool the authentication system to gain access to restricted resources. This has been proven to effectively degrade the performance of a variety of audio-authentication methods. In response to this, spoofing countermeasures for the task of anti-spoofing have been developed that can detect and successfully thwart these types of attacks. The advent of deep learning techniques and their usage in real-life applications has led to the research and development of various techniques for purposes ranging from exploiting weaknesses in the deep learning model to stealing confidential information. One of the ways in which the deep learning-based audio authentication model can be evaded is the usage of a set of attacks that are known as adversarial attacks. These adversarial attacks consist of adding a carefully crafted perturbation to the input to elicit a wrong inference from the model. We first explore the performance that multimodality brings to the anti-spoofing task. We aim to augment a unimodal spoofing countermeasure with visual information to identify whether it can improve performance. Since visuals can serve as an additional domain of information, we experiment with whether the existing paradigm of using unimodal spoofing countermeasures for anti-spoofing can benefit from this new information. Our results indicate that augmenting an existing unimodal countermeasure with visual information does not provide any performance benefits. Future work can explore more tightly coupled multimodal models that use objectives like contrastive loss. We then study the vulnerability of deep learning-based multimodal speaker verification to adversarial attacks. In multimodal speaker verification, the vulnerability has not been established and we aim to accomplish this. We find that the multimodal models are heavily reliant on the visual modality and that attacking both modalities lead to a higher attack success rate. Future work can move on to stronger attacks by applying adversarial attacks to bypass the spoofing countermeasure and speaker verification. Finally, we investigate the feasibility of a generic evasion detector that can block both adversarial and spoofing attacks. Since both the spoofing and adversarial attacks target speaker verification models, we aim to add an adversarial attack detection mechanism — feature squeezing — onto the spoofing countermeasure to achieve this. We find that such a detector is feasible but involves a significant reduction in the identification of genuine samples. Future work can explore combining adversarial training as a defense for attacks that target the complete spoofing countermeasure and speaker verification pipeline

Essential Speech and Language Technology for Dutch: Results by the STEVIN-programme

Computational Linguistics; Germanic Languages; Artificial Intelligence (incl. Robotics); Computing Methodologie

IberSPEECH 2020: XI Jornadas en Tecnología del Habla and VII Iberian SLTech

IberSPEECH2020 is a two-day event, bringing together the best researchers and practitioners in speech and language technologies in Iberian languages to promote interaction and discussion. The organizing committee has planned a wide variety of scientific and social activities, including technical paper presentations, keynote lectures, presentation of projects, laboratories activities, recent PhD thesis, discussion panels, a round table, and awards to the best thesis and papers. The program of IberSPEECH2020 includes a total of 32 contributions that will be presented distributed among 5 oral sessions, a PhD session, and a projects session. To ensure the quality of all the contributions, each submitted paper was reviewed by three members of the scientific review committee. All the papers in the conference will be accessible through the International Speech Communication Association (ISCA) Online Archive. Paper selection was based on the scores and comments provided by the scientific review committee, which includes 73 researchers from different institutions (mainly from Spain and Portugal, but also from France, Germany, Brazil, Iran, Greece, Hungary, Czech Republic, Ucrania, Slovenia). Furthermore, it is confirmed to publish an extension of selected papers as a special issue of the Journal of Applied Sciences, “IberSPEECH 2020: Speech and Language Technologies for Iberian Languages”, published by MDPI with fully open access. In addition to regular paper sessions, the IberSPEECH2020 scientific program features the following activities: the ALBAYZIN evaluation challenge session.Red Española de Tecnologías del Habla. Universidad de Valladoli

Speech technologies for the audiovisual and multimedia interaction environments

361 p

EVALITA Evaluation of NLP and Speech Tools for Italian Proceedings of the Final Workshop

Editor of the proceedings of EVALITA 2016

Learning Attention Mechanisms and Context: An Investigation into Vision and Emotion

Attention mechanisms for context modelling are becoming ubiquitous in neural architectures in machine learning. The attention mechanism is a technique that filters out information that is irrelevant to a given task and focuses on learning task-dependent fixation points or regions. Furthermore, attention mechanisms suggest a question about a given task, i.e. `what' to learn and `where/how' to learn for task-specific context modelling. The context is the conditional variables instrumental in deciding the categorical distribution for the given data. Also, why is learning task-specific context necessary? In order to answer these questions, context modelling with attention in the vision and emotion domains is explored in this thesis using attention mechanisms with different hierarchical structures. The three main goals of this thesis are building superior classifiers using attention-based deep neural networks~(DNNs), investigating the role of context modelling in the given tasks, and developing a framework for interpreting hierarchies and attention in deep attention networks. In the vision domain, gesture and posture recognition tasks in diverse environments, are chosen. In emotion, visual and speech emotion recognition tasks are chosen. These tasks are selected for their sequential properties for modelling a spatiotemporal context. One of the key challenges from a machine learning standpoint is to extract patterns which bear maximum correlation with the information encoded in its signal while being as insensitive as possible to other types of information carried by the signal. A possible way to overcome this problem is to learn task-dependent representations. In order to achieve that, novel spatiotemporal context modelling networks and the mixture of multi-view attention~(MOMA) networks are proposed using bidirectional long-short-term memory network (BLSTM), convolutional neural network~(CNN), Capsule and attention networks. A framework has been proposed to interpret the internal attention states with respect to the given task. The results of the classifiers in the assigned tasks are compared with the \textit{state-of-the-art} DNNs, and the proposed classifiers achieve superior results. The context in speech emotion recognition is explored deeply with the attention interpretation framework, and it shows that the proposed model can assign word importance based on acoustic context. Furthermore, it has been observed that the internal states of the attention bear correlation with human perception of acoustic cues for speech emotion recognition. Overall, the results demonstrate superior classifiers and context learning models with interpretable frameworks. The findings are very important for speech emotion recognition systems. In this thesis, not only better models are produced, but also the interpretability of those models are explored, and their internal states are analysed. The phones and words are aligned with the attention vectors, and it is seen that the vowel sounds are more important for defining emotion acoustic cues than the consonants, and the model can assign word importance based on acoustic context. Also, how these approaches for emotion recognition using word importance for predicting emotions are demonstrated by the attention weight visualisation over the words. In a broader perspective, the findings from the thesis about gesture, posture and emotion recognition may be helpful in tasks like human-robot interaction~(HRI) and conversational artificial agents (such as Siri, Alexa). The communication is grounded with the symbolic and sub-symbolic cues of intent either from visual, audio or haptics. The understanding of intent is much dependent on the reasoning about the situational context. Emotion, i.e.\ speech and visual emotion, provides context to a situation, and it is a deciding factor in the response generation. Emotional intelligence and information from vision, audio and other modalities are essential for making human-human and human-robot communication more natural and feedback-driven