Fast and Accurate OOV Decoder on High-Level Features

This work proposes a novel approach to out-of-vocabulary (OOV) keyword search (KWS) task. The proposed approach is based on using high-level features from an automatic speech recognition (ASR) system, so called phoneme posterior based (PPB) features, for decoding. These features are obtained by calculating time-dependent phoneme posterior probabilities from word lattices, followed by their smoothing. For the PPB features we developed a special novel very fast, simple and efficient OOV decoder. Experimental results are presented on the Georgian language from the IARPA Babel Program, which was the test language in the OpenKWS 2016 evaluation campaign. The results show that in terms of maximum term weighted value (MTWV) metric and computational speed, for single ASR systems, the proposed approach significantly outperforms the state-of-the-art approach based on using in-vocabulary proxies for OOV keywords in the indexed database. The comparison of the two OOV KWS approaches on the fusion results of the nine different ASR systems demonstrates that the proposed OOV decoder outperforms the proxy-based approach in terms of MTWV metric given the comparable processing speed. Other important advantages of the OOV decoder include extremely low memory consumption and simplicity of its implementation and parameter optimization.Comment: Interspeech 2017, August 2017, Stockholm, Sweden. 201

Anonymizing Speech: Evaluating and Designing Speaker Anonymization Techniques

The growing use of voice user interfaces has led to a surge in the collection and storage of speech data. While data collection allows for the development of efficient tools powering most speech services, it also poses serious privacy issues for users as centralized storage makes private personal speech data vulnerable to cyber threats. With the increasing use of voice-based digital assistants like Amazon's Alexa, Google's Home, and Apple's Siri, and with the increasing ease with which personal speech data can be collected, the risk of malicious use of voice-cloning and speaker/gender/pathological/etc. recognition has increased. This thesis proposes solutions for anonymizing speech and evaluating the degree of the anonymization. In this work, anonymization refers to making personal speech data unlinkable to an identity while maintaining the usefulness (utility) of the speech signal (e.g., access to linguistic content). We start by identifying several challenges that evaluation protocols need to consider to evaluate the degree of privacy protection properly. We clarify how anonymization systems must be configured for evaluation purposes and highlight that many practical deployment configurations do not permit privacy evaluation. Furthermore, we study and examine the most common voice conversion-based anonymization system and identify its weak points before suggesting new methods to overcome some limitations. We isolate all components of the anonymization system to evaluate the degree of speaker PPI associated with each of them. Then, we propose several transformation methods for each component to reduce as much as possible speaker PPI while maintaining utility. We promote anonymization algorithms based on quantization-based transformation as an alternative to the most-used and well-known noise-based approach. Finally, we endeavor a new attack method to invert anonymization.Comment: PhD Thesis Pierre Champion | Universit\'e de Lorraine - INRIA Nancy | for associated source code, see https://github.com/deep-privacy/SA-toolki

Data-driven approach for synchrotron X-ray Laue microdiffraction scan analysis

We propose a novel data-driven approach for analyzing synchrotron Laue X-ray microdiffraction scans based on machine learning algorithms. The basic architecture and major components of the method are formulated mathematically. We demonstrate it through typical examples including polycrystalline BaTiO$_3$, multiphase transforming alloys and finely twinned martensite. The computational pipeline is implemented for beamline 12.3.2 at the Advanced Light Source, Lawrence Berkeley National Lab. The conventional analytical pathway for X-ray diffraction scans is based on a slow pattern by pattern crystal indexing process. This work provides a new way for analyzing X-ray diffraction 2D patterns, independent of the indexing process, and motivates further studies of X-ray diffraction patterns from the machine learning prospective for the development of suitable feature extraction, clustering and labeling algorithms.Comment: 29 pages, 25 figures under the second round of review by Acta Crystallographica

ニューラルネットワークを用いた非線形主成分分析に関する研究

制度:新 ; 文部省報告番号:甲2050号 ; 学位の種類:博士(工学) ; 授与年月日:2005/3/3 ; 早大学位記番号:新400

Maximal Figure-of-Merit Framework to Detect Multi-label Phonetic Features for Spoken Language Recognition

Bottleneck features (BNFs) generated with a deep neural network (DNN) have proven to boost spoken language recognition accuracy over basic spectral features significantly. However, BNFs are commonly extracted using language-dependent tied-context phone states as learning targets. Moreover, BNFs are less phonetically expressive than the output layer in a DNN, which is usually not used as a speech feature because of its very high dimensionality hindering further post-processing. In this work, we put forth a novel deep learning framework to overcome all of the above issues and evaluate it on the 2017 NIST Language Recognition Evaluation (LRE) challenge. We use manner and place of articulation as speech attributes, which lead to low-dimensional “universal” phonetic features that can be defined across all spoken languages. To model the asynchronous nature of the speech attributes while capturing their intrinsic relationships in a given speech segment, we introduce a new training scheme for deep architectures based on a Maximal Figure of Merit (MFoM) objective. MFoM introduces non-differentiable metrics into the backpropagation-based approach, which is elegantly solved in the proposed framework. The experimental evidence collected on the recent NIST LRE 2017 challenge demonstrates the effectiveness of our solution. In fact, the performance of speech language recognition (SLR) systems based on spectral features is improved for more than 5% absolute Cavg. Finally, the F1 metric can be brought from 77.6% up to 78.1% by combining the conventional baseline phonetic BNFs with the proposed articulatory attribute features

PCN: Point Completion Network

Shape completion, the problem of estimating the complete geometry of objects from partial observations, lies at the core of many vision and robotics applications. In this work, we propose Point Completion Network (PCN), a novel learning-based approach for shape completion. Unlike existing shape completion methods, PCN directly operates on raw point clouds without any structural assumption (e.g. symmetry) or annotation (e.g. semantic class) about the underlying shape. It features a decoder design that enables the generation of fine-grained completions while maintaining a small number of parameters. Our experiments show that PCN produces dense, complete point clouds with realistic structures in the missing regions on inputs with various levels of incompleteness and noise, including cars from LiDAR scans in the KITTI dataset.Comment: 3DV 2018 oral. Honorable mention for Best Paper awar

End-to-End Open Vocabulary Keyword Search With Multilingual Neural Representations

Conventional keyword search systems operate on automatic speech recognition (ASR) outputs, which causes them to have a complex indexing and search pipeline. This has led to interest in ASR-free approaches to simplify the search procedure. We recently proposed a neural ASR-free keyword search model which achieves competitive performance while maintaining an efficient and simplified pipeline, where queries and documents are encoded with a pair of recurrent neural network encoders and the encodings are combined with a dot-product. In this article, we extend this work with multilingual pretraining and detailed analysis of the model. Our experiments show that the proposed multilingual training significantly improves the model performance and that despite not matching a strong ASR-based conventional keyword search system for short queries and queries comprising in-vocabulary words, the proposed model outperforms the ASR-based system for long queries and queries that do not appear in the training data.Comment: Accepted by IEEE/ACM Transactions on Audio, Speech and Language Processing (TASLP), 202

Advances in Subspace-based Solutions for Diarization in the Broadcast Domain

La motivación de esta tesis es la necesidad de soluciones robustas al problema de diarización. Estas técnicas de diarización deben proporcionar valor añadido a la creciente cantidad disponible de datos multimedia mediante la precisa discriminación de los locutores presentes en la señal de audio. Desafortunadamente, hasta tiempos recientes este tipo de tecnologías solamente era viable en condiciones restringidas, quedando por tanto lejos de una solución general. Las razones detrás de las limitadas prestaciones de los sistemas de diarización son múltiples. La primera causa a tener en cuenta es la alta complejidad de la producción de la voz humana, en particular acerca de los procesos fisiológicos necesarios para incluir las características discriminativas de locutor en la señal de voz. Esta complejidad hace del proceso inverso, la estimación de dichas características a partir del audio, una tarea ineficiente por medio de las técnicas actuales del estado del arte. Consecuentemente, en su lugar deberán tenerse en cuenta aproximaciones. Los esfuerzos en la tarea de modelado han proporcionado modelos cada vez más elaborados, aunque no buscando la explicación última de naturaleza fisiológica de la señal de voz. En su lugar estos modelos aprenden relaciones entre la señales acústicas a partir de un gran conjunto de datos de entrenamiento. El desarrollo de modelos aproximados genera a su vez una segunda razón, la variabilidad de dominio. Debido al uso de relaciones aprendidas a partir de un conjunto de entrenamiento concreto, cualquier cambio de dominio que modifique las condiciones acústicas con respecto a los datos de entrenamiento condiciona las relaciones asumidas, pudiendo causar fallos consistentes en los sistemas.Nuestra contribución a las tecnologías de diarización se ha centrado en el entorno de radiodifusión. Este dominio es actualmente un entorno todavía complejo para los sistemas de diarización donde ninguna simplificación de la tarea puede ser tenida en cuenta. Por tanto, se deberá desarrollar un modelado eficiente del audio para extraer la información de locutor y como inferir el etiquetado correspondiente. Además, la presencia de múltiples condiciones acústicas debido a la existencia de diferentes programas y/o géneros en el domino requiere el desarrollo de técnicas capaces de adaptar el conocimiento adquirido en un determinado escenario donde la información está disponible a aquellos entornos donde dicha información es limitada o sencillamente no disponible.Para este propósito el trabajo desarrollado a lo largo de la tesis se ha centrado en tres subtareas: caracterización de locutor, agrupamiento y adaptación de modelos. La primera subtarea busca el modelado de un fragmento de audio para obtener representaciones precisas de los locutores involucrados, poniendo de manifiesto sus propiedades discriminativas. En este área se ha llevado a cabo un estudio acerca de las actuales estrategias de modelado, especialmente atendiendo a las limitaciones de las representaciones extraídas y poniendo de manifiesto el tipo de errores que pueden generar. Además, se han propuesto alternativas basadas en redes neuronales haciendo uso del conocimiento adquirido. La segunda tarea es el agrupamiento, encargado de desarrollar estrategias que busquen el etiquetado óptimo de los locutores. La investigación desarrollada durante esta tesis ha propuesto nuevas estrategias para estimar el mejor reparto de locutores basadas en técnicas de subespacios, especialmente PLDA. Finalmente, la tarea de adaptación de modelos busca transferir el conocimiento obtenido de un conjunto de entrenamiento a dominios alternativos donde no hay datos para extraerlo. Para este propósito los esfuerzos se han centrado en la extracción no supervisada de información de locutor del propio audio a diarizar, sinedo posteriormente usada en la adaptación de los modelos involucrados.<br /