331 research outputs found

    Contributions and applications around low resource deep learning modeling

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    El aprendizaje profundo representa la vanguardia del aprendizaje automático en multitud de aplicaciones. Muchas de estas tareas requieren una gran cantidad de recursos computacionales, lo que limita su adopción en dispositivos integrados. El objetivo principal de esta tesis es estudiar métodos y algoritmos que permiten abordar problemas utilizando aprendizaje profundo con bajos recursos computacionales. Este trabajo también tiene como objetivo presentar aplicaciones de aprendizaje profundo en la industria. La primera contribución es una nueva función de activación para redes de aprendizaje profundo: la función de módulo. Los experimentos muestran que la función de activación propuesta logra resultados superiores en tareas de visión artificial cuando se compara con las alternativas encontradas en la literatura. La segunda contribución es una nueva estrategia para combinar modelos preentrenados usando destilación de conocimiento. Los resultados de este capítulo muestran que es posible aumentar significativamente la precisión de los modelos preentrenados más pequeños, lo que permite un alto rendimiento a un menor costo computacional. La siguiente contribución de esta tesis aborda el problema de la previsión de ventas en el campo de la logística. Se proponen dos sistemas de extremo a extremo con dos técnicas diferentes de aprendizaje profundo (modelos de secuencia a secuencia y transformadores). Los resultados de este capítulo concluyen que es posible construir sistemas integrales para predecir las ventas de múltiples productos individuales, en múltiples puntos de venta y en diferentes momentos con un único modelo de aprendizaje automático. El modelo propuesto supera las alternativas encontradas en la literatura. Finalmente, las dos últimas contribuciones pertenecen al campo de la tecnología del habla. El primero estudia cómo construir un sistema de reconocimiento de voz Keyword Spotting utilizando una versión eficiente de una red neuronal convolucional. En este estudio, el sistema propuesto es capaz de superar el rendimiento de todos los puntos de referencia encontrados en la literatura cuando se prueba contra las subtareas más complejas. El último estudio propone un modelo independiente de texto a voz de última generación capaz de sintetizar voz inteligible en miles de perfiles de voz, mientras genera un discurso con variaciones de prosodia significativas y expresivas. El enfoque propuesto elimina la dependencia de los modelos anteriores de un sistema de voz adicional, lo que hace que el sistema propuesto sea más eficiente en el tiempo de entrenamiento e inferencia, y permite operaciones fuera de línea y en el dispositivo.Deep learning is the state of the art for several machine learning tasks. Many of these tasks require large amount of computational resources, which limits their adoption in embedded devices. The main goal of this dissertation is to study methods and algorithms that allow to approach problems using deep learning with restricted computational resources. This work also aims at presenting applications of deep learning in industry. The first contribution is a new activation function for deep learning networks: the modulus function. The experiments show that the proposed activation function achieves superior results in computer vision tasks when compared with the alternatives found in the literature. The second contribution is a new strategy to combine pre-trained models using knowledge distillation. The results of this chapter show that it is possible to significantly increase the accuracy of the smallest pre-trained models, allowing high performance at a lower computational cost. The following contribution in this thesis tackles the problem of sales fore- casting in the field of logistics. Two end-to-end systems with two different deep learning techniques (sequence-to-sequence models and transformers) are pro- posed. The results of this chapter conclude that it is possible to build end-to-end systems to predict the sales of multiple individual products, at multiple points of sale and different times with a single machine learning model. The proposed model outperforms the alternatives found in the literature. Finally, the last two contributions belong to the speech technology field. The former, studies how to build a Keyword Spotting speech recognition system using an efficient version of a convolutional neural network. In this study, the proposed system is able to beat the performance of all the benchmarks found in the literature when tested against the most complex subtasks. The latter study proposes a standalone state-of-the-art text-to-speech model capable of synthesizing intelligible voice in thousands of voice profiles, while generating speech with meaningful and expressive prosody variations. The proposed approach removes the dependency of previous models on an additional voice system, which makes the proposed system more efficient at training and inference time, and enables offline and on-device operations

    Leveraging audio-visual speech effectively via deep learning

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    The rising popularity of neural networks, combined with the recent proliferation of online audio-visual media, has led to a revolution in the way machines encode, recognize, and generate acoustic and visual speech. Despite the ubiquity of naturally paired audio-visual data, only a limited number of works have applied recent advances in deep learning to leverage the duality between audio and video within this domain. This thesis considers the use of neural networks to learn from large unlabelled datasets of audio-visual speech to enable new practical applications. We begin by training a visual speech encoder that predicts latent features extracted from the corresponding audio on a large unlabelled audio-visual corpus. We apply the trained visual encoder to improve performance on lip reading in real-world scenarios. Following this, we extend the idea of video learning from audio by training a model to synthesize raw speech directly from raw video, without the need for text transcriptions. Remarkably, we find that this framework is capable of reconstructing intelligible audio from videos of new, previously unseen speakers. We also experiment with a separate speech reconstruction framework, which leverages recent advances in sequence modeling and spectrogram inversion to improve the realism of the generated speech. We then apply our research in video-to-speech synthesis to advance the state-of-the-art in audio-visual speech enhancement, by proposing a new vocoder-based model that performs particularly well under extremely noisy scenarios. Lastly, we aim to fully realize the potential of paired audio-visual data by proposing two novel frameworks that leverage acoustic and visual speech to train two encoders that learn from each other simultaneously. We leverage these pre-trained encoders for deepfake detection, speech recognition, and lip reading, and find that they consistently yield improvements over training from scratch.Open Acces

    Speech Enhancement and Dereverberation with Diffusion-based Generative Models

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    In this work, we build upon our previous publication and use diffusion-based generative models for speech enhancement. We present a detailed overview of the diffusion process that is based on a stochastic differential equation and delve into an extensive theoretical examination of its implications. Opposed to usual conditional generation tasks, we do not start the reverse process from pure Gaussian noise but from a mixture of noisy speech and Gaussian noise. This matches our forward process which moves from clean speech to noisy speech by including a drift term. We show that this procedure enables using only 30 diffusion steps to generate high-quality clean speech estimates. By adapting the network architecture, we are able to significantly improve the speech enhancement performance, indicating that the network, rather than the formalism, was the main limitation of our original approach. In an extensive cross-dataset evaluation, we show that the improved method can compete with recent discriminative models and achieves better generalization when evaluating on a different corpus than used for training. We complement the results with an instrumental evaluation using real-world noisy recordings and a listening experiment, in which our proposed method is rated best. Examining different sampler configurations for solving the reverse process allows us to balance the performance and computational speed of the proposed method. Moreover, we show that the proposed method is also suitable for dereverberation and thus not limited to additive background noise removal. Code and audio examples are available online, see https://github.com/sp-uhh/sgmseComment: Accepted versio

    Audiovisual speech perception in cochlear implant patients

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    Hearing with a cochlear implant (CI) is very different compared to a normal-hearing (NH) experience, as the CI can only provide limited auditory input. Nevertheless, the central auditory system is capable of learning how to interpret such limited auditory input such that it can extract meaningful information within a few months after implant switch-on. The capacity of the auditory cortex to adapt to new auditory stimuli is an example of intra-modal plasticity — changes within a sensory cortical region as a result of altered statistics of the respective sensory input. However, hearing deprivation before implantation and restoration of hearing capacities after implantation can also induce cross-modal plasticity — changes within a sensory cortical region as a result of altered statistics of a different sensory input. Thereby, a preserved cortical region can, for example, support a deprived cortical region, as in the case of CI users which have been shown to exhibit cross-modal visual-cortex activation for purely auditory stimuli. Before implantation, during the period of hearing deprivation, CI users typically rely on additional visual cues like lip-movements for understanding speech. Therefore, it has been suggested that CI users show a pronounced binding of the auditory and visual systems, which may allow them to integrate auditory and visual speech information more efficiently. The projects included in this thesis investigate auditory, and particularly audiovisual speech processing in CI users. Four event-related potential (ERP) studies approach the matter from different perspectives, each with a distinct focus. The first project investigates how audiovisually presented syllables are processed by CI users with bilateral hearing loss compared to NH controls. Previous ERP studies employing non-linguistic stimuli and studies using different neuroimaging techniques found distinct audiovisual interactions in CI users. However, the precise timecourse of cross-modal visual-cortex recruitment and enhanced audiovisual interaction for speech related stimuli is unknown. With our ERP study we fill this gap, and we present differences in the timecourse of audiovisual interactions as well as in cortical source configurations between CI users and NH controls. The second study focuses on auditory processing in single-sided deaf (SSD) CI users. SSD CI patients experience a maximally asymmetric hearing condition, as they have a CI on one ear and a contralateral NH ear. Despite the intact ear, several behavioural studies have demonstrated a variety of beneficial effects of restoring binaural hearing, but there are only few ERP studies which investigate auditory processing in SSD CI users. Our study investigates whether the side of implantation affects auditory processing and whether auditory processing via the NH ear of SSD CI users works similarly as in NH controls. Given the distinct hearing conditions of SSD CI users, the question arises whether there are any quantifiable differences between CI user with unilateral hearing loss and bilateral hearing loss. In general, ERP studies on SSD CI users are rather scarce, and there is no study on audiovisual processing in particular. Furthermore, there are no reports on lip-reading abilities of SSD CI users. To this end, in the third project we extend the first study by including SSD CI users as a third experimental group. The study discusses both differences and similarities between CI users with bilateral hearing loss and CI users with unilateral hearing loss as well as NH controls and provides — for the first time — insights into audiovisual interactions in SSD CI users. The fourth project investigates the influence of background noise on audiovisual interactions in CI users and whether a noise-reduction algorithm can modulate these interactions. It is known that in environments with competing background noise listeners generally rely more strongly on visual cues for understanding speech and that such situations are particularly difficult for CI users. As shown in previous auditory behavioural studies, the recently introduced noise-reduction algorithm "ForwardFocus" can be a useful aid in such cases. However, the questions whether employing the algorithm is beneficial in audiovisual conditions as well and whether using the algorithm has a measurable effect on cortical processing have not been investigated yet. In this ERP study, we address these questions with an auditory and audiovisual syllable discrimination task. Taken together, the projects included in this thesis contribute to a better understanding of auditory and especially audiovisual speech processing in CI users, revealing distinct processing strategies employed to overcome the limited input provided by a CI. The results have clinical implications, as they suggest that clinical hearing assessments, which are currently purely auditory, should be extended to audiovisual assessments. Furthermore, they imply that rehabilitation including audiovisual training methods may be beneficial for all CI user groups for quickly achieving the most effective CI implantation outcome

    Imagining & Sensing: Understanding and Extending the Vocalist-Voice Relationship Through Biosignal Feedback

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    The voice is body and instrument. Third-person interpretation of the voice by listeners, vocal teachers, and digital agents is centred largely around audio feedback. For a vocalist, physical feedback from within the body provides an additional interaction. The vocalist’s understanding of their multi-sensory experiences is through tacit knowledge of the body. This knowledge is difficult to articulate, yet awareness and control of the body are innate. In the ever-increasing emergence of technology which quantifies or interprets physiological processes, we must remain conscious also of embodiment and human perception of these processes. Focusing on the vocalist-voice relationship, this thesis expands knowledge of human interaction and how technology influences our perception of our bodies. To unite these different perspectives in the vocal context, I draw on mixed methods from cog- nitive science, psychology, music information retrieval, and interactive system design. Objective methods such as vocal audio analysis provide a third-person observation. Subjective practices such as micro-phenomenology capture the experiential, first-person perspectives of the vocalists them- selves. Quantitative-qualitative blend provides details not only on novel interaction, but also an understanding of how technology influences existing understanding of the body. I worked with vocalists to understand how they use their voice through abstract representations, use mental imagery to adapt to altered auditory feedback, and teach fundamental practice to others. Vocalists use multi-modal imagery, for instance understanding physical sensations through auditory sensations. The understanding of the voice exists in a pre-linguistic representation which draws on embodied knowledge and lived experience from outside contexts. I developed a novel vocal interaction method which uses measurement of laryngeal muscular activations through surface electromyography. Biofeedback was presented to vocalists through soni- fication. Acting as an indicator of vocal activity for both conscious and unconscious gestures, this feedback allowed vocalists to explore their movement through sound. This formed new perceptions but also questioned existing understanding of the body. The thesis also uncovers ways in which vocalists are in control and controlled by, work with and against their bodies, and feel as a single entity at times and totally separate entities at others. I conclude this thesis by demonstrating a nuanced account of human interaction and perception of the body through vocal practice, as an example of how technological intervention enables exploration and influence over embodied understanding. This further highlights the need for understanding of the human experience in embodied interaction, rather than solely on digital interpretation, when introducing technology into these relationships

    Proceedings of the 8th Workshop on Detection and Classification of Acoustic Scenes and Events (DCASE 2023)

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    This volume gathers the papers presented at the Detection and Classification of Acoustic Scenes and Events 2023 Workshop (DCASE2023), Tampere, Finland, during 21–22 September 2023

    Speech wave-form driven motion synthesis for embodied agents

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    The main objective of this thesis is to synthesise motion from speech, especially in conversation. Based on previous research into different acoustic features or the combination of them were investigated, no one has investigated in estimating head motion from waveform directly, which is the stem of the speech. Thus, we study the direct use of speech waveform to generate head motion. We claim that creating a task-specific feature from waveform to generate head motion leads to better performance than using standard acoustic features to generate head motion overall. At the same time, we completely abandon the handcrafted feature extraction process, leading to more effectiveness. However, there are a few problems if we would like to apply speech waveform, 1) high dimensional, where the dimension of the waveform data is much higher than those common acoustic features and thus making the training of the model more difficult, and 2) irrelevant information, which refers to the full information in the original waveform implicating potential cumbrance for neural network training. To resolve these problems, we applied a deep canonical correlated constrainted auto-encoder (DCCCAE) to compress the waveform into low dimensional and highly correlated embedded features with head motion. The estimated head motion was evaluated both objectively and subjectively. In objective evaluation, the result confirmed that DCCCAE enables the creation of a more correlated feature with the head motion than standard AE and other popular spectral features such as MFCC and FBank, and is capable of being used in achieving state-of-the-art results for predicting natural head motion with the advantage of the DCCCAE. Besides investigating the representation learning of the feature, we also explored the LSTM-based regression model for the proposed feature. The LSTM-based models were able to boost the overall performance in the objective evaluation and adapt better to the proposed feature than MFCC. MUSHRA-liked subjective evaluation results suggest that the animations generated by models with the proposed feature were chosen to be better than the other models by the participants of MUSHRA-liked test. A/B test further that the LSTM-based regression model adapts better to the proposed feature. Furthermore, we extended the architecture to estimate the upper body motion as well. We submitted our result to GENEA2020 and our model achieved a higher score than BA in both aspects (human-likeness and appropriateness) according to the participant’s preference, suggesting that the highly correlated feature pair and the sequential estimation helped in improving the model generalisation

    Anonymizing Speech: Evaluating and Designing Speaker Anonymization Techniques

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    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
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