Automatic Quality Estimation for ASR System Combination

Recognizer Output Voting Error Reduction (ROVER) has been widely used for system combination in automatic speech recognition (ASR). In order to select the most appropriate words to insert at each position in the output transcriptions, some ROVER extensions rely on critical information such as confidence scores and other ASR decoder features. This information, which is not always available, highly depends on the decoding process and sometimes tends to over estimate the real quality of the recognized words. In this paper we propose a novel variant of ROVER that takes advantage of ASR quality estimation (QE) for ranking the transcriptions at "segment level" instead of: i) relying on confidence scores, or ii) feeding ROVER with randomly ordered hypotheses. We first introduce an effective set of features to compensate for the absence of ASR decoder information. Then, we apply QE techniques to perform accurate hypothesis ranking at segment-level before starting the fusion process. The evaluation is carried out on two different tasks, in which we respectively combine hypotheses coming from independent ASR systems and multi-microphone recordings. In both tasks, it is assumed that the ASR decoder information is not available. The proposed approach significantly outperforms standard ROVER and it is competitive with two strong oracles that e xploit prior knowledge about the real quality of the hypotheses to be combined. Compared to standard ROVER, the abs olute WER improvements in the two evaluation scenarios range from 0.5% to 7.3%

Novel Architectures and Optimization Algorithms for Training Neural Networks and Applications

The two main areas of Deep Learning are Unsupervised and Supervised Learning. Unsupervised Learning studies a class of data processing problems in which only descriptions of objects are known, without label information. Generative Adversarial Networks (GANs) have become among the most widely used unsupervised neural net models. GAN combines two neural nets, generative and discriminative, that work simultaneously. We introduce a new family of discriminator loss functions that adopts a weighted sum of real and fake parts, which we call adaptive weighted loss functions. Using the gradient information, we can adaptively choose weights to train a discriminator in the direction that benefits the GAN\u27s stability. Also, we propose several improvements to the GAN training schemes. One is self-correcting optimization for training a GAN discriminator on Speech Enhancement tasks, which helps avoid ``harmful\u27\u27 training directions for parts of the discriminator loss. The other improvement is a consistency loss, which targets the inconsistency in time and time-frequency domains caused by Fourier Transforms. Contrary to Unsupervised Learning, Supervised Learning uses labels for each object, and it is required to find the relationship between objects and labels. Building computing methods to interpret and represent human language automatically is known as Natural Language Processing which includes tasks such as word prediction, machine translation, etc. In this area, we propose a novel Neumann-Cayley Gated Recurrent Unit (NC-GRU) architecture based on a Neumann series-based Scaled Cayley transformation. The NC-GRU uses orthogonal matrices to prevent exploding gradient problems and enhance long-term memory on various prediction tasks. In addition, we propose using our newly introduced NC-GRU unit inside Neural Nets model to create neural molecular fingerprints. Integrating novel NC-GRU fingerprints and Multi-Task Deep Neural Networks schematics help to improve the performance of several molecular-related tasks. We also introduce a new normalization method - Assorted-Time Normalization, that helps to preserve information from multiple consecutive time steps and normalize using them in Recurrent Nets like architectures. Finally, we propose a Symmetry Structured Convolutional Neural Network (SCNN), an architecture with 2D structured symmetric features over spatial dimensions, that generates and preserves the symmetry structure in the network\u27s convolutional layers

Deep Learning for Distant Speech Recognition

Deep learning is an emerging technology that is considered one of the most promising directions for reaching higher levels of artificial intelligence. Among the other achievements, building computers that understand speech represents a crucial leap towards intelligent machines. Despite the great efforts of the past decades, however, a natural and robust human-machine speech interaction still appears to be out of reach, especially when users interact with a distant microphone in noisy and reverberant environments. The latter disturbances severely hamper the intelligibility of a speech signal, making Distant Speech Recognition (DSR) one of the major open challenges in the field. This thesis addresses the latter scenario and proposes some novel techniques, architectures, and algorithms to improve the robustness of distant-talking acoustic models. We first elaborate on methodologies for realistic data contamination, with a particular emphasis on DNN training with simulated data. We then investigate on approaches for better exploiting speech contexts, proposing some original methodologies for both feed-forward and recurrent neural networks. Lastly, inspired by the idea that cooperation across different DNNs could be the key for counteracting the harmful effects of noise and reverberation, we propose a novel deep learning paradigm called network of deep neural networks. The analysis of the original concepts were based on extensive experimental validations conducted on both real and simulated data, considering different corpora, microphone configurations, environments, noisy conditions, and ASR tasks.Comment: PhD Thesis Unitn, 201

On the Utility of Representation Learning Algorithms for Myoelectric Interfacing

Electrical activity produced by muscles during voluntary movement is a reflection of the firing patterns of relevant motor neurons and, by extension, the latent motor intent driving the movement. Once transduced via electromyography (EMG) and converted into digital form, this activity can be processed to provide an estimate of the original motor intent and is as such a feasible basis for non-invasive efferent neural interfacing. EMG-based motor intent decoding has so far received the most attention in the field of upper-limb prosthetics, where alternative means of interfacing are scarce and the utility of better control apparent. Whereas myoelectric prostheses have been available since the 1960s, available EMG control interfaces still lag behind the mechanical capabilities of the artificial limbs they are intended to steer—a gap at least partially due to limitations in current methods for translating EMG into appropriate motion commands. As the relationship between EMG signals and concurrent effector kinematics is highly non-linear and apparently stochastic, finding ways to accurately extract and combine relevant information from across electrode sites is still an active area of inquiry.This dissertation comprises an introduction and eight papers that explore issues afflicting the status quo of myoelectric decoding and possible solutions, all related through their use of learning algorithms and deep Artificial Neural Network (ANN) models. Paper I presents a Convolutional Neural Network (CNN) for multi-label movement decoding of high-density surface EMG (HD-sEMG) signals. Inspired by the successful use of CNNs in Paper I and the work of others, Paper II presents a method for automatic design of CNN architectures for use in myocontrol. Paper III introduces an ANN architecture with an appertaining training framework from which simultaneous and proportional control emerges. Paper Iv introduce a dataset of HD-sEMG signals for use with learning algorithms. Paper v applies a Recurrent Neural Network (RNN) model to decode finger forces from intramuscular EMG. Paper vI introduces a Transformer model for myoelectric interfacing that do not need additional training data to function with previously unseen users. Paper vII compares the performance of a Long Short-Term Memory (LSTM) network to that of classical pattern recognition algorithms. Lastly, paper vIII describes a framework for synthesizing EMG from multi-articulate gestures intended to reduce training burden

Contributions to speech processing and ambient sound analysis

We are constantly surrounded by sounds that we continuously exploit to adapt our actions to situations we are facing. Some of the sounds like speech can have a particular structure from which we can infer some information, explicit or not. This is one reason why speech is possibly that is the most intuitive way to communicate between humans. Within the last decade, there has been significant progress in the domain of speech andaudio processing and in particular in the domain of machine learning applied to speech and audio processing. Thanks to these progresses, speech has become a central element in many human to human distant communication tools as well as in human to machine communication systems. These solutions work pretty well on clean speech or under controlled condition. However, in scenarios that involve the presence of acoustic perturbation such as noise or reverberation systems performance tends to degrade severely. In this thesis we focus on processing speech and its environments from an audio perspective. The algorithms proposed here are relying on a variety of solutions from signal processing based approaches to data-driven solutions based on supervised matrix factorization or deep neural networks. We propose solutions to problems ranging from speech recognition, to speech enhancement or ambient sound analysis. The target is to offer a panorama of the different aspects that could improve a speech processing algorithm working in a real environments. We start by describing automatic speech recognition as a potential end application and progressively unravel the limitations and the proposed solutions ending-up to the more general ambient sound analysis.Nous sommes constamment entourés de sons que nous exploitons pour adapter nos actions aux situations auxquelles nous sommes confrontés. Certains sons comme la parole peuvent avoir une structure particulière à partir de laquelle nous pouvons déduire des informations, explicites ou non. C’est l’une des raisons pour lesquelles la parole est peut-être le moyen le plus intuitif de communiquer entre humains. Au cours de la décennie écoulée, des progrès significatifs ont été réalisés dans le domaine du traitement de la parole et du son et en particulier dans le domaine de l’apprentissage automatique appliqué au traitement de la parole et du son. Grâce à ces progrès, la parole est devenue un élément central de nombreux outils de communication à distance d’humain à humain ainsi que dans les systèmes de communication humain-machine. Ces solutions fonctionnent bien sur un signal de parole propre ou dans des conditions contrôlées. Cependant, dans les scénarios qui impliquent la présence de perturbations acoustiques telles que du bruit ou de la réverbération les performances peuvent avoir tendance à se dégrader gravement. Dans cette HDR, nous nous concentrons sur le traitement de la parole et de son environnement d’un point de vue audio. Les algorithmes proposés ici reposent sur une variété de solutions allant des approches basées sur le traitement du signal aux solutions orientées données à base de factorisation matricielle supervisée ou de réseaux de neurones profonds. Nous proposons des solutions à des problèmes allant de la reconnaissance vocale au rehaussement de la parole ou à l’analyse des sons ambiants. L’objectif est d’offrir un panorama des différents aspects qui pourraient être améliorer un algorithme de traitement de la parole fonctionnant dans un environnement réel. Nous commençons par décrire la reconnaissance automatique de la parole comme une application finale potentielle et analysons progressivement les limites et les solutions proposées aboutissant à l’analyse plus générale des sons ambiants

Interaction intermodale dans les réseaux neuronaux profonds pour la classification et la localisation d'évènements audiovisuels

La compréhension automatique du monde environnant a de nombreuses applications telles que la surveillance et sécurité, l'interaction Homme-Machine, la robotique, les soins de santé, etc. Plus précisément, la compréhension peut s'exprimer par le biais de différentes taches telles que la classification et localisation dans l'espace d'évènements. Les êtres vivants exploitent un maximum de l'information disponible pour comprendre ce qui les entoure. En s'inspirant du comportement des êtres vivants, les réseaux de neurones artificiels devraient également utiliser conjointement plusieurs modalités, par exemple, la vision et l'audition. Premièrement, les modèles de classification et localisation, basés sur l'information audio-visuelle, doivent être évalués de façon objective. Nous avons donc enregistré une nouvelle base de données pour compléter les bases actuellement disponibles. Comme aucun modèle audio-visuel de classification et localisation n'existe, seule la partie sonore de la base est évaluée avec un modèle de la littérature. Deuxièmement, nous nous concentrons sur le cœur de la thèse: comment utiliser conjointement de l'information visuelle et sonore pour résoudre une tâche spécifique, la reconnaissance d'évènements. Le cerveau n'est pas constitué d'une "simple" fusion mais comprend de multiples interactions entre les deux modalités. Il y a un couplage important entre le traitement de l'information visuelle et sonore. Les réseaux de neurones offrent la possibilité de créer des interactions entre les modalités en plus de la fusion. Dans cette thèse, nous explorons plusieurs stratégies pour fusionner les modalités visuelles et sonores et pour créer des interactions entre les modalités. Ces techniques ont les meilleures performances en comparaison aux architectures de l'état de l'art au moment de la publication. Ces techniques montrent l'utilité de la fusion audio-visuelle mais surtout l'importance des interactions entre les modalités. Pour conclure la thèse, nous proposons un réseau de référence pour la classification et localisation d'évènements audio-visuels. Ce réseau a été testé avec la nouvelle base de données. Les modèles précédents de classification sont modifiés pour prendre en compte la localisation dans l'espace en plus de la classification.Abstract: The automatic understanding of the surrounding world has a wide range of applications, including surveillance, human-computer interaction, robotics, health care, etc. The understanding can be expressed in several ways such as event classification and its localization in space. Living beings exploit a maximum of the available information to understand the surrounding world. Artificial neural networks should build on this behavior and jointly use several modalities such as vision and hearing. First, audio-visual networks for classification and localization must be evaluated objectively. We recorded a new audio-visual dataset to fill a gap in the current available datasets. We were not able to find audio-visual models for classification and localization. Only the dataset audio part is evaluated with a state-of-the-art model. Secondly, we focus on the main challenge of the thesis: How to jointly use visual and audio information to solve a specific task, event recognition. The brain does not comprise a simple fusion but has multiple interactions between the two modalities to create a strong coupling between them. The neural networks offer the possibility to create interactions between the two modalities in addition to the fusion. We explore several strategies to fuse the audio and visual modalities and to create interactions between modalities. These techniques have the best performance compared to the state-of-the-art architectures at the time of publishing. They show the usefulness of audio-visual fusion but above all the contribution of the interaction between modalities. To conclude, we propose a benchmark for audio-visual classification and localization on the new dataset. Previous models for the audio-visual classification are modified to address the localization in addition to the classification

EMG-to-Speech: Direct Generation of Speech from Facial Electromyographic Signals

The general objective of this work is the design, implementation, improvement and evaluation of a system that uses surface electromyographic (EMG) signals and directly synthesizes an audible speech output: EMG-to-speech