V-Cloak: Intelligibility-, Naturalness- & Timbre-Preserving Real-Time Voice Anonymization

Voice data generated on instant messaging or social media applications contains unique user voiceprints that may be abused by malicious adversaries for identity inference or identity theft. Existing voice anonymization techniques, e.g., signal processing and voice conversion/synthesis, suffer from degradation of perceptual quality. In this paper, we develop a voice anonymization system, named V-Cloak, which attains real-time voice anonymization while preserving the intelligibility, naturalness and timbre of the audio. Our designed anonymizer features a one-shot generative model that modulates the features of the original audio at different frequency levels. We train the anonymizer with a carefully-designed loss function. Apart from the anonymity loss, we further incorporate the intelligibility loss and the psychoacoustics-based naturalness loss. The anonymizer can realize untargeted and targeted anonymization to achieve the anonymity goals of unidentifiability and unlinkability. We have conducted extensive experiments on four datasets, i.e., LibriSpeech (English), AISHELL (Chinese), CommonVoice (French) and CommonVoice (Italian), five Automatic Speaker Verification (ASV) systems (including two DNN-based, two statistical and one commercial ASV), and eleven Automatic Speech Recognition (ASR) systems (for different languages). Experiment results confirm that V-Cloak outperforms five baselines in terms of anonymity performance. We also demonstrate that V-Cloak trained only on the VoxCeleb1 dataset against ECAPA-TDNN ASV and DeepSpeech2 ASR has transferable anonymity against other ASVs and cross-language intelligibility for other ASRs. Furthermore, we verify the robustness of V-Cloak against various de-noising techniques and adaptive attacks. Hopefully, V-Cloak may provide a cloak for us in a prism world.Comment: Accepted by USENIX Security Symposium 202

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

Edge-centric multimodal authentication system using encrypted biometric templates

Data security, complete system control, and missed storage and computing opportunities in personal portable devices are some of the major limitations of the centralized cloud environment. Among these limitations, security is a prime concern due to potential unauthorized access to private data. Biometrics, in particular, is considered sensitive data, and its usage is subject to the privacy protection law. To address this issue, a multimodal authentication system using encrypted biometrics for the edge-centric cloud environment is proposed in this study. Personal portable devices are utilized for encrypting biometrics in the proposed system, which optimizes the use of resources and tackles another limitation of the cloud environment. Biometrics is encrypted using a new method. In the proposed system, the edges transmit the encrypted speech and face for processing in the cloud. The cloud then decrypts the biometrics and performs authentication to confirm the identity of an individual. The model for speech authentication is based on two types of features, namely, Mel-frequency cepstral coefficients and perceptual linear prediction coefficients. The model for face authentication is implemented by determining the eigenfaces. The final decision about the identity of a user is based on majority voting. Experimental results show that the new encryption method can reliably hide the identity of an individual and accurately decrypt the biometrics, which is vital for errorless authentication

The definition of the relevant population and the collection of data for likelihood ratio-based forensic voice comparison

Within the field of forensic speech science there is increasing acceptance of the likelihood ratio (LR) as the logically and legally correct framework for evaluating forensic voice comparison (FVC) evidence. However, only a small proportion of experts cur- rently use the numerical LR in casework. This is due primarily to the difficulties involved in accounting for the inherent, and arguably unique, complexity of speech in a fully data-driven, numerical LR analysis. This thesis addresses two such issues: the definition of the relevant population and the amount of data required for system testing. Firstly, experiments are presented which explore the extent to which LRs are affected by different definitions of the relevant population with regard to sources of systematic sociolinguistic between-speaker variation (regional background, socio-economic class and age) using both linguistic-phonetic and ASR variables. Results show that different definitions of the relevant population can have a substantial effect on the magnitude of LRs, depending on the input variable. However, system validity results suggest that narrow controls over sociolinguistic sources of variation should be preferred to general controls. Secondly, experiments are presented which evaluate the effects of development, test and reference sample size on LRs. Consistent with general principles in statistics, more precise results are found using more data across all experiments. There is also considerable evidence of a relationship between sample size sensitivity and the dimensionality and speaker discriminatory power of the input variable. Further, there are potential trade-offs in the size of each set depending on which element of LR output the analyst is interested in. The results in this thesis will contribute towards im- proving the extent to which LR methods account for the linguistic-phonetic complexity of speech evidence. In accounting for this complexity, this work will also increase the practical viability of applying the numerical LR to FVC casework

USING DEEP LEARNING-BASED FRAMEWORK FOR CHILD SPEECH EMOTION RECOGNITION

Biological languages of the body through which human emotion can be detected abound including heart rate, facial expressions, movement of the eyelids and dilation of the eyes, body postures, skin conductance, and even the speech we make. Speech emotion recognition research started some three decades ago, and the popular Interspeech Emotion Challenge has helped to propagate this research area. However, most speech recognition research is focused on adults and there is very little research on child speech. This dissertation is a description of the development and evaluation of a child speech emotion recognition framework. The higher-level components of the framework are designed to sort and separate speech based on the speaker’s age, ensuring that focus is only on speeches made by children. The framework uses Baddeley’s Theory of Working Memory to model a Working Memory Recurrent Network that can process and recognize emotions from speech. Baddeley’s Theory of Working Memory offers one of the best explanations on how the human brain holds and manipulates temporary information which is very crucial in the development of neural networks that learns effectively. Experiments were designed and performed to provide answers to the research questions, evaluate the proposed framework, and benchmark the performance of the framework with other methods. Satisfactory results were obtained from the experiments and in many cases, our framework was able to outperform other popular approaches. This study has implications for various applications of child speech emotion recognition such as child abuse detection and child learning robots

The effect of sampling variability on overall performance and individual speakers’ behaviour in likelihood ratio-based forensic voice comparison

In the past years, there is increasing awareness and acceptance among forensic speech scientists of using Bayesian reasoning and likelihood ratio (LR) framework for forensic voice comparison (FVC) and expressing expert conclusions. Numerous studies have explored overall performance using numerical LRs. Given that the data used for validation is a sample coming from an unknown distribution, little attention has been paid to the effect of sampling variability or individuals’ behaviour. This thesis investigates these issues using linguistic-phonetic variables. First, it investigates how different configurations of training, test and reference speakers affect overall performance. The results show that variability in overall performance is mostly caused by varying the test speakers, while less variability is caused by sampling variability in the reference and training speakers. Second, this thesis explores the effect of sampling variability on overall performance and individuals’ behaviour in relation to the use of linguistic-phonetic features. Results show that sampling variability affects overall performance to different extents using different features, while combining more features does not always improve overall performance. Sampling variability has limited effects on individuals in same-speaker comparisons, and most speakers are less affected by sampling variability in different-speaker comparisons when four or more features are used. Third, this thesis explores the effect of sampling variability on overall performance in relation to score distributions. Results reveal that system validity and reliability are more affected by different- speaker score skewness, and less affected by same-speaker score skewness. Using different calibration methods reduces the effect of sampling variability to different extents. The results in this thesis have implications for both FVC using numerical LRs and FVC in general, as experts need to make pragmatic decisions whether numerical LR is used or not, and every decision made has implication to final evaluation results. Further, the results on score skewness and different calibration methods have potential contribution for improving FVC performance using automatic systems

Biometrics

Biometrics uses methods for unique recognition of humans based upon one or more intrinsic physical or behavioral traits. In computer science, particularly, biometrics is used as a form of identity access management and access control. It is also used to identify individuals in groups that are under surveillance. The book consists of 13 chapters, each focusing on a certain aspect of the problem. The book chapters are divided into three sections: physical biometrics, behavioral biometrics and medical biometrics. The key objective of the book is to provide comprehensive reference and text on human authentication and people identity verification from both physiological, behavioural and other points of view. It aims to publish new insights into current innovations in computer systems and technology for biometrics development and its applications. The book was reviewed by the editor Dr. Jucheng Yang, and many of the guest editors, such as Dr. Girija Chetty, Dr. Norman Poh, Dr. Loris Nanni, Dr. Jianjiang Feng, Dr. Dongsun Park, Dr. Sook Yoon and so on, who also made a significant contribution to the book

Speech and natural language processing for the assessment of customer satisfaction and neuro-degenerative diseases

ABSTRACT: Nowadays, the interest in the automatic analysis of speech and text in different scenarios have been increasing. Currently, acoustic analysis is frequently used to extract non-verbal information related to para-linguistic aspects such as articulation and prosody. The linguistic analysis focuses on capturing verbal information from written sources, which can be suitable to evaluate customer satisfaction, or in health-care applications to assess the state of patients under depression or other cognitive states. In the case of call-centers many of the speech recordings collected are related to the opinion of the customers in different industry sectors. Only a small proportion of these calls are evaluated, whereby these processes can be automated using acoustic and linguistic analysis. In the assessment of neuro-degenerative diseases such as Alzheimer's Disease (AD) and Parkinson's Disease (PD), the symptoms are progressive, directly linked to dementia, cognitive decline, and motor impairments. This implies a continuous evaluation of the neurological state since the patients become dependent and need intensive care, showing a decrease of the ability from individual activities of daily life. This thesis proposes methodologies for acoustic and linguistic analyses in different scenarios related to customer satisfaction, cognitive disorders in AD, and depression in PD. The experiments include the evaluation of customer satisfaction, the assessment of genetic AD, linguistic analysis to discriminate PD, depression assessment in PD, and user state modeling based on the arousal-plane for the evaluation of customer satisfaction, AD, and depression in PD. The acoustic features are mainly focused on articulation and prosody analyses, while linguistic features are based on natural language processing techniques. Deep learning approaches based on convolutional and recurrent neural networks are also considered in this thesis

Exploring variabilities through factor analysis in automatic acoustic language recognition

La problématique traitée par la Reconnaissance de la Langue (LR) porte sur la définition découverte de la langue contenue dans un segment de parole. Cette thèse se base sur des paramètres acoustiques de courte durée, utilisés dans une approche d adaptation de mélanges de Gaussiennes (GMM-UBM). Le problème majeur de nombreuses applications du vaste domaine de la re- problème connaissance de formes consiste en la variabilité des données observées. Dans le contexte de la Reconnaissance de la Langue (LR), cette variabilité nuisible est due à des causes diverses, notamment les caractéristiques du locuteur, l évolution de la parole et de la voix, ainsi que les canaux d acquisition et de transmission. Dans le contexte de la reconnaissance du locuteur, l impact de la variabilité solution peut sensiblement être réduit par la technique d Analyse Factorielle (Joint Factor Analysis, JFA). Dans ce travail, nous introduisons ce paradigme à la Reconnaissance de la Langue. Le succès de la JFA repose sur plusieurs hypothèses. La première est que l information observée est décomposable en une partie universelle, une partie dépendante de la langue et une partie de variabilité, qui elle est indépendante de la langue. La deuxième hypothèse, plus technique, est que la variabilité nuisible se situe dans un sous-espace de faible dimension, qui est défini de manière globale.Dans ce travail, nous analysons le comportement de la JFA dans le contexte d un dispositif de LR du type GMM-UBM. Nous introduisons et analysons également sa combinaison avec des Machines à Vecteurs Support (SVM). Les premières publications sur la JFA regroupaient toute information qui est amélioration nuisible à la tâche (donc ladite variabilité) dans un seul composant. Celui-ci est supposé suivre une distribution Gaussienne. Cette approche permet de traiter les différentes sortes de variabilités d une manière unique. En pratique, nous observons que cette hypothèse n est pas toujours vérifiée. Nous avons, par exemple, le cas où les données peuvent être groupées de manière logique en deux sous-parties clairement distinctes, notamment en données de sources téléphoniques et d émissions radio. Dans ce cas-ci, nos recherches détaillées montrent un certain avantage à traiter les deux types de données par deux systèmes spécifiques et d élire comme score de sortie celui du système qui correspond à la catégorie source du segment testé. Afin de sélectionner le score de l un des systèmes, nous avons besoin d un analyses détecteur de canal source. Nous proposons ici différents nouveaux designs pour engendrées de tels détecteurs automatiques. Dans ce cadre, nous montrons que les facteurs de variabilité (du sous-espace) de la JFA peuvent être utilisés avec succès pour la détection de la source. Ceci ouvre la perspective intéressante de subdiviser les5données en catégories de canal source qui sont établies de manière automatique. En plus de pouvoir s adapter à des nouvelles conditions de source, cette propriété permettrait de pouvoir travailler avec des données d entraînement qui ne sont pas accompagnées d étiquettes sur le canal de source. L approche JFA permet une réduction de la mesure de coûts allant jusqu à généraux 72% relatives, comparé au système GMM-UBM de base. En utilisant des systèmes spécifiques à la source, suivis d un sélecteur de scores, nous obtenons une amélioration relative de 81%.Language Recognition is the problem of discovering the language of a spoken definitionutterance. This thesis achieves this goal by using short term acoustic information within a GMM-UBM approach.The main problem of many pattern recognition applications is the variability of problemthe observed data. In the context of Language Recognition (LR), this troublesomevariability is due to the speaker characteristics, speech evolution, acquisition and transmission channels.In the context of Speaker Recognition, the variability problem is solved by solutionthe Joint Factor Analysis (JFA) technique. Here, we introduce this paradigm toLanguage Recognition. The success of JFA relies on several assumptions: The globalJFA assumption is that the observed information can be decomposed into a universalglobal part, a language-dependent part and the language-independent variabilitypart. The second, more technical assumption consists in the unwanted variability part to be thought to live in a low-dimensional, globally defined subspace. In this work, we analyze how JFA behaves in the context of a GMM-UBM LR framework. We also introduce and analyze its combination with Support Vector Machines(SVMs).The first JFA publications put all unwanted information (hence the variability) improvemen tinto one and the same component, which is thought to follow a Gaussian distribution.This handles diverse kinds of variability in a unique manner. But in practice,we observe that this hypothesis is not always verified. We have for example thecase, where the data can be divided into two clearly separate subsets, namely datafrom telephony and from broadcast sources. In this case, our detailed investigations show that there is some benefit of handling the two kinds of data with two separatesystems and then to elect the output score of the system, which corresponds to the source of the testing utterance.For selecting the score of one or the other system, we need a channel source related analyses detector. We propose here different novel designs for such automatic detectors.In this framework, we show that JFA s variability factors (of the subspace) can beused with success for detecting the source. This opens the interesting perspectiveof partitioning the data into automatically determined channel source categories,avoiding the need of source-labeled training data, which is not always available.The JFA approach results in up to 72% relative cost reduction, compared to the overall resultsGMM-UBM baseline system. Using source specific systems followed by a scoreselector, we achieve 81% relative improvement.AVIGNON-Bib. numérique (840079901) / SudocSudocFranceF