Scaling and Bias Codes for Modeling Speaker-Adaptive DNN-based Speech Synthesis Systems

Most neural-network based speaker-adaptive acoustic models for speech synthesis can be categorized into either layer-based or input-code approaches. Although both approaches have their own pros and cons, most existing works on speaker adaptation focus on improving one or the other. In this paper, after we first systematically overview the common principles of neural-network based speaker-adaptive models, we show that these approaches can be represented in a unified framework and can be generalized further. More specifically, we introduce the use of scaling and bias codes as generalized means for speaker-adaptive transformation. By utilizing these codes, we can create a more efficient factorized speaker-adaptive model and capture advantages of both approaches while reducing their disadvantages. The experiments show that the proposed method can improve the performance of speaker adaptation compared with speaker adaptation based on the conventional input code.Comment: Accepted for 2018 IEEE Workshop on Spoken Language Technology (SLT), Athens, Greec

Losses Can Be Blessings: Routing Self-Supervised Speech Representations Towards Efficient Multilingual and Multitask Speech Processing

Self-supervised learning (SSL) for rich speech representations has achieved empirical success in low-resource Automatic Speech Recognition (ASR) and other speech processing tasks, which can mitigate the necessity of a large amount of transcribed speech and thus has driven a growing demand for on-device ASR and other speech processing. However, advanced speech SSL models have become increasingly large, which contradicts the limited on-device resources. This gap could be more severe in multilingual/multitask scenarios requiring simultaneously recognizing multiple languages or executing multiple speech processing tasks. Additionally, strongly overparameterized speech SSL models tend to suffer from overfitting when being finetuned on low-resource speech corpus. This work aims to enhance the practical usage of speech SSL models towards a win-win in both enhanced efficiency and alleviated overfitting via our proposed S$^3$-Router framework, which for the first time discovers that simply discarding no more than 10\% of model weights via only finetuning model connections of speech SSL models can achieve better accuracy over standard weight finetuning on downstream speech processing tasks. More importantly, S$^3$-Router can serve as an all-in-one technique to enable (1) a new finetuning scheme, (2) an efficient multilingual/multitask solution, (3) a state-of-the-art ASR pruning technique, and (4) a new tool to quantitatively analyze the learned speech representation. We believe S$^3$-Router has provided a new perspective for practical deployment of speech SSL models. Our codes are available at: https://github.com/GATECH-EIC/S3-Router.Comment: Accepted at NeurIPS 202

Conditioning Text-to-Speech synthesis on dialect accent: a case study

Modern text-to-speech systems are modular in many different ways. In recent years, end-users gained the ability to control speech attributes such as degree of emotion, rhythm and timbre, along with other suprasegmental features. More ambitious objectives are related to modelling a combination of speakers and languages, e.g. to enable cross-speaker language transfer. Though, no prior work has been done on the more fine-grained analysis of regional accents. To fill this gap, in this thesis we present practical end-to-end solutions to synthesise speech while controlling within-country variations of the same language, and we do so for 6 different dialects of the British Isles. In particular, we first conduct an extensive study of the speaker verification field and tweak state-of-the-art embedding models to work with dialect accents. Then, we adapt standard acoustic models and voice conversion systems by conditioning them on dialect accent representations and finally compare our custom pipelines with a cutting-edge end-to-end architecture from the multi-lingual world. Results show that the adopted models are suitable and have enough capacity to accomplish the task of regional accent conversion. Indeed, we are able to produce speech closely resembling the selected speaker and dialect accent, where the most accurate synthesis is obtained via careful fine-tuning of the multi-lingual model to the multi-dialect case. Finally, we delineate limitations of our multi-stage approach and propose practical mitigations, to be explored in future work

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

Deep Transfer Learning for Automatic Speech Recognition: Towards Better Generalization

Automatic speech recognition (ASR) has recently become an important challenge when using deep learning (DL). It requires large-scale training datasets and high computational and storage resources. Moreover, DL techniques and machine learning (ML) approaches in general, hypothesize that training and testing data come from the same domain, with the same input feature space and data distribution characteristics. This assumption, however, is not applicable in some real-world artificial intelligence (AI) applications. Moreover, there are situations where gathering real data is challenging, expensive, or rarely occurring, which can not meet the data requirements of DL models. deep transfer learning (DTL) has been introduced to overcome these issues, which helps develop high-performing models using real datasets that are small or slightly different but related to the training data. This paper presents a comprehensive survey of DTL-based ASR frameworks to shed light on the latest developments and helps academics and professionals understand current challenges. Specifically, after presenting the DTL background, a well-designed taxonomy is adopted to inform the state-of-the-art. A critical analysis is then conducted to identify the limitations and advantages of each framework. Moving on, a comparative study is introduced to highlight the current challenges before deriving opportunities for future research

Deep learning-based automatic analysis of social interactions from wearable data for healthcare applications

PhD ThesisSocial interactions of people with Late Life Depression (LLD) could be an objective measure of social functioning due to the association between LLD and poor social functioning. The utilisation of wearable computing technologies is a relatively new approach within healthcare and well-being application sectors. Recently, the design and development of wearable technologies and systems for health and well-being monitoring have attracted attention both of the clinical and scientific communities. Mainly because the current clinical practice of – typically rather sporadic – clinical behaviour assessments are often administered in artificial settings. As a result, it does not provide a realistic impression of a patient’s condition and thus does not lead to sufficient diagnosis and care. However, wearable behaviour monitors have the potential for continuous, objective assessment of behaviour and wider social interactions and thereby allowing for capturing naturalistic data without any constraints on the place of recording or any typical limitations of the lab-setting research. Such data from naturalistic ambient environments would facilitate automated transmission and analysis by having no constraints on the recordings, allowing for a more timely and accurate assessment of depressive symptoms. In response to this artificial setting issue, this thesis focuses on the analysis and assessment of the different aspects of social interactions in naturalistic environments using deep learning algorithms. That could lead to improvements in both diagnosis and treatment. The advantages of using deep learning are that there is no need for hand-crafted features engineering and this leads to using the raw data with minimal pre-processing compared to classical machine learning approaches and also its scalability and ability to generalise. The main dataset used in this thesis is recorded by a wrist worn device designed at Newcastle University. This device has multiple sensors including microphone, tri-axial accelerometer, light sensor and proximity sensor. In this thesis, only microphone and tri-axial accelerometer are used for the social interaction analysis. The other sensors are not used since they need more calibration from the user which in this will be the elderly people with depression. Hence, it was not feasible in this scenario. Novel deep learning models are proposed to automatically analyse two aspects of social interactions (the verbal interactions/acoustic communications and physical activities/movement patterns). Verbal Interactions include the total quantity of speech, who is talking to whom and when and how much engagement the wearer contributed in the conversations. The physical activity analysis includes activity recognition and the quantity of each activity and sleep patterns. This thesis is composed of three main stages, two of them discuss the acoustic analysis and the third stage describes the movement pattern analysis. The acoustic analysis starts with speech detection in which each segment of the recording is categorised as speech or non-speech. This segment classification is achieved by a novel deep learning model that leverages bi-directional Long Short-Term Memory with gated activation units combined with Maxout Networks as well as a combination of two optimisers. After detecting speech segments from audio data, the next stage is detecting how much engagement the wearer has in any conversation throughout these speech events based on detecting the wearer of the device using a variant model of the previous one that combines the convolutional autoencoder with bi-directional Long Short-Term Memory. Following this, the system then detects the spoken parts of the main speaker/wearer and therefore detects the conversational turn-taking but only includes the turn taking between the wearer and other speakers and not every speaker in the conversation. This stage did not take into account the semantics of the speakers due to the ethical constraints of the main dataset (Depression dataset) and therefore it was not possible to listen to the data by any means or even have any information about the contents. So, it is a good idea to be considered for future work. Stage 3 involves the physical activity analysis that is inferring the elementary physical activities and movement patterns. These elementary patterns include sedentary actions, walking, mixed activities, cycling, using vehicles as well as the sleep patterns. The predictive model used is based on Random Forests and Hidden Markov Models. In all stages the methods presented in this thesis have been compared to the state-of-the-art in processing audio, accelerometer data, respectively, to thoroughly assess their contribution. Following these stages is a thorough analysis of the interplay between acoustic interaction and physical movement patterns and the depression key clinical variables resulting to the outcomes of the previous stages. The main reason for not using deep learning in this stage unlike the previous stages is that the main dataset (Depression dataset) did not have any annotations for the speech or even the activity due to the ethical constraints as mentioned. Furthermore, the training dataset (Discussion dataset) did not have any annotations for the accelerometer data where the data is recorded freely and there is no camera attached to device to make it possible to be annotated afterwards.Newton-Mosharafa Fund and the mission sector and cultural affairs, ministry of Higher Education in Egypt

Principled methods for mixtures processing

This document is my thesis for getting the habilitation à diriger des recherches, which is the french diploma that is required to fully supervise Ph.D. students. It summarizes the research I did in the last 15 years and also provides the short­term research directions and applications I want to investigate. Regarding my past research, I first describe the work I did on probabilistic audio modeling, including the separation of Gaussian and α­stable stochastic processes. Then, I mention my work on deep learning applied to audio, which rapidly turned into a large effort for community service. Finally, I present my contributions in machine learning, with some works on hardware compressed sensing and probabilistic generative models.My research programme involves a theoretical part that revolves around probabilistic machine learning, and an applied part that concerns the processing of time series arising in both audio and life sciences