Deep Learning for Audio Signal Processing

Given the recent surge in developments of deep learning, this article provides a review of the state-of-the-art deep learning techniques for audio signal processing. Speech, music, and environmental sound processing are considered side-by-side, in order to point out similarities and differences between the domains, highlighting general methods, problems, key references, and potential for cross-fertilization between areas. The dominant feature representations (in particular, log-mel spectra and raw waveform) and deep learning models are reviewed, including convolutional neural networks, variants of the long short-term memory architecture, as well as more audio-specific neural network models. Subsequently, prominent deep learning application areas are covered, i.e. audio recognition (automatic speech recognition, music information retrieval, environmental sound detection, localization and tracking) and synthesis and transformation (source separation, audio enhancement, generative models for speech, sound, and music synthesis). Finally, key issues and future questions regarding deep learning applied to audio signal processing are identified.Comment: 15 pages, 2 pdf figure

WAKE WORD DETECTION AND ITS APPLICATIONS

Always-on spoken language interfaces, e.g. personal digital assistants, rely on a wake word to start processing spoken input. Novel methods are proposed to train a wake word detection system from partially labeled training data, and to use it in on-line applications. In the system, the prerequisite of frame-level alignment is removed, permitting the use of un-transcribed training examples that are annotated only for the presence/absence of the wake word. Also, an FST-based decoder is presented to perform online detection. The suite of methods greatly improve the wake word detection performance across several datasets. A novel neural network for acoustic modeling in wake word detection is also investigated. Specifically, the performance of several variants of chunk-wise streaming Transformers tailored for wake word detection is explored, including looking-ahead to the next chunk, gradient stopping, different positional embedding methods and adding same-layer dependency between chunks. Experiments demonstrate that the proposed Transformer model outperforms the baseline convolutional network significantly with a comparable model size, while still maintaining linear complexity w.r.t. the input length. For the application of the detected wake word in ASR, the problem of improving speech recognition with the help of the detected wake word is investigated. Voice-controlled house-hold devices face the difficulty of performing speech recognition of device-directed speech in the presence of interfering background speech. Two end-to-end models are proposed to tackle this problem with information extracted from the anchored segment. The anchored segment refers to the wake word segment of the audio stream, which contains valuable speaker information that can be used to suppress interfering speech and background noise. A multi-task learning setup is also explored where the ideal mask, obtained from a data synthesis procedure, is used to guide the model training. In addition, a way to synthesize "noisy" speech from "clean" speech is also proposed to mitigate the mismatch between training and test data. The proposed methods show large word error reduction for Amazon Alexa live data with interfering background speech, without sacrificing the performance on clean speech

Automatic Screening of Childhood Speech Sound Disorders and Detection of Associated Pronunciation Errors

Speech disorders in children can affect their fluency and intelligibility. Delay in their diagnosis and treatment increases the risk of social impairment and learning disabilities. With the significant shortage of Speech and Language Pathologists (SLPs), there is an increasing interest in Computer-Aided Speech Therapy tools with automatic detection and diagnosis capability. However, the scarcity and unreliable annotation of disordered child speech corpora along with the high acoustic variations in the child speech data has impeded the development of reliable automatic detection and diagnosis of childhood speech sound disorders. Therefore, this thesis investigates two types of detection systems that can be achieved with minimum dependency on annotated mispronounced speech data. First, a novel approach that adopts paralinguistic features which represent the prosodic, spectral, and voice quality characteristics of the speech was proposed to perform segment- and subject-level classification of Typically Developing (TD) and Speech Sound Disordered (SSD) child speech using a binary Support Vector Machine (SVM) classifier. As paralinguistic features are both language- and content-independent, they can be extracted from an unannotated speech signal. Second, a novel Mispronunciation Detection and Diagnosis (MDD) approach was introduced to detect the pronunciation errors made due to SSDs and provide low-level diagnostic information that can be used in constructing formative feedback and a detailed diagnostic report. Unlike existing MDD methods where detection and diagnosis are performed at the phoneme level, the proposed method achieved MDD at the speech attribute level, namely the manners and places of articulations. The speech attribute features describe the involved articulators and their interactions when making a speech sound allowing a low-level description of the pronunciation error to be provided. Two novel methods to model speech attributes are further proposed in this thesis, a frame-based (phoneme-alignment) method leveraging the Multi-Task Learning (MTL) criterion and training a separate model for each attribute, and an alignment-free jointly-learnt method based on the Connectionist Temporal Classification (CTC) sequence to sequence criterion. The proposed techniques have been evaluated using standard and publicly accessible adult and child speech corpora, while the MDD method has been validated using L2 speech corpora

Transformer Models for Machine Translation and Streaming Automatic Speech Recognition

[ES] El procesamiento del lenguaje natural (NLP) es un conjunto de problemas computacionales con aplicaciones de máxima relevancia, que junto con otras tecnologías informáticas se ha beneficiado de la revolución que ha significado el aprendizaje profundo. Esta tesis se centra en dos problemas fundamentales para el NLP: la traducción automática (MT) y el reconocimiento automático del habla o transcripción automática (ASR); así como en una arquitectura neuronal profunda, el Transformer, que pondremos en práctica para mejorar las soluciones de MT y ASR en algunas de sus aplicaciones. El ASR y MT pueden servir para obtener textos multilingües de alta calidad a un coste razonable para una diversidad de contenidos audiovisuales. Concre- tamente, esta tesis aborda problemas como el de traducción de noticias o el de subtitulación automática de televisión. El ASR y MT también se pueden com- binar entre sí, generando automáticamente subtítulos traducidos, o con otras soluciones de NLP: resumen de textos para producir resúmenes de discursos, o síntesis del habla para crear doblajes automáticos. Estas aplicaciones quedan fuera del alcance de esta tesis pero pueden aprovechar las contribuciones que contiene, en la meduda que ayudan a mejorar el rendimiento de los sistemas automáticos de los que dependen. Esta tesis contiene una aplicación de la arquitectura Transformer al MT tal y como fue concebida, mediante la que obtenemos resultados de primer nivel en traducción de lenguas semejantes. En capítulos subsecuentes, esta tesis aborda la adaptación del Transformer como modelo de lenguaje para sistemas híbri- dos de ASR en vivo. Posteriormente, describe la aplicación de este tipus de sistemas al caso de uso de subtitulación de televisión, participando en una com- petición pública de RTVE donde obtenemos la primera posición con un marge importante. También demostramos que la mejora se debe principalmenta a la tecnología desarrollada y no tanto a la parte de los datos.[CA] El processament del llenguage natural (NLP) és un conjunt de problemes com- putacionals amb aplicacions de màxima rellevància, que juntament amb al- tres tecnologies informàtiques s'ha beneficiat de la revolució que ha significat l'impacte de l'aprenentatge profund. Aquesta tesi se centra en dos problemes fonamentals per al NLP: la traducció automàtica (MT) i el reconeixement automàtic de la parla o transcripció automàtica (ASR); així com en una ar- quitectura neuronal profunda, el Transformer, que posarem en pràctica per a millorar les solucions de MT i ASR en algunes de les seues aplicacions. l'ASR i MT poden servir per obtindre textos multilingües d'alta qualitat a un cost raonable per a un gran ventall de continguts audiovisuals. Concretament, aquesta tesi aborda problemes com el de traducció de notícies o el de subtitu- lació automàtica de televisió. l'ASR i MT també es poden combinar entre ells, generant automàticament subtítols traduïts, o amb altres solucions de NLP: amb resum de textos per produir resums de discursos, o amb síntesi de la parla per crear doblatges automàtics. Aquestes altres aplicacions es troben fora de l'abast d'aquesta tesi però poden aprofitar les contribucions que conté, en la mesura que ajuden a millorar els resultats dels sistemes automàtics dels quals depenen. Aquesta tesi conté una aplicació de l'arquitectura Transformer al MT tal com va ser concebuda, mitjançant la qual obtenim resultats de primer nivell en traducció de llengües semblants. En capítols subseqüents, aquesta tesi aborda l'adaptació del Transformer com a model de llenguatge per a sistemes híbrids d'ASR en viu. Posteriorment, descriu l'aplicació d'aquest tipus de sistemes al cas d'ús de subtitulació de continguts televisius, participant en una competició pública de RTVE on obtenim la primera posició amb un marge significant. També demostrem que la millora es deu principalment a la tecnologia desen- volupada i no tant a la part de les dades[EN] Natural language processing (NLP) is a set of fundamental computing prob- lems with immense applicability, as language is the natural communication vehicle for people. NLP, along with many other computer technologies, has been revolutionized in recent years by the impact of deep learning. This thesis is centered around two keystone problems for NLP: machine translation (MT) and automatic speech recognition (ASR); and a common deep neural architec- ture, the Transformer, that is leveraged to improve the technical solutions for some MT and ASR applications. ASR and MT can be utilized to produce cost-effective, high-quality multilin- gual texts for a wide array of media. Particular applications pursued in this thesis are that of news translation or that of automatic live captioning of tele- vision broadcasts. ASR and MT can also be combined with each other, for instance generating automatic translated subtitles from audio, or augmented with other NLP solutions: text summarization to produce a summary of a speech, or speech synthesis to create an automatic translated dubbing, for in- stance. These other applications fall out of the scope of this thesis, but can profit from the contributions that it contains, as they help to improve the performance of the automatic systems on which they depend. This thesis contains an application of the Transformer architecture to MT as it was originally conceived, achieving state-of-the-art results in similar language translation. In successive chapters, this thesis covers the adaptation of the Transformer as a language model for streaming hybrid ASR systems. After- wards, it describes how we applied the developed technology for a specific use case in television captioning by participating in a competitive challenge and achieving the first position by a large margin. We also show that the gains came mostly from the improvement in technology capabilities over two years including that of the Transformer language model adapted for streaming, and the data component was minor.Baquero Arnal, P. (2023). Transformer Models for Machine Translation and Streaming Automatic Speech Recognition [Tesis doctoral]. Universitat Politècnica de València. https://doi.org/10.4995/Thesis/10251/19368

Attention-Inspired Artificial Neural Networks for Speech Processing: A Systematic Review

Artificial Neural Networks (ANNs) were created inspired by the neural networks in the human brain and have been widely applied in speech processing. The application areas of ANN include: Speech recognition, speech emotion recognition, language identification, speech enhancement, and speech separation, amongst others. Likewise, given that speech processing performed by humans involves complex cognitive processes known as auditory attention, there has been a growing amount of papers proposing ANNs supported by deep learning algorithms in conjunction with some mechanism to achieve symmetry with the human attention process. However, while these ANN approaches include attention, there is no categorization of attention integrated into the deep learning algorithms and their relation with human auditory attention. Therefore, we consider it necessary to have a review of the different ANN approaches inspired in attention to show both academic and industry experts the available models for a wide variety of applications. Based on the PRISMA methodology, we present a systematic review of the literature published since 2000, in which deep learning algorithms are applied to diverse problems related to speech processing. In this paper 133 research works are selected and the following aspects are described: (i) Most relevant features, (ii) ways in which attention has been implemented, (iii) their hypothetical relationship with human attention, and (iv) the evaluation metrics used. Additionally, the four publications most related with human attention were analyzed and their strengths and weaknesses were determined