Streaming Automatic Speech Recognition with Hybrid Architectures and Deep Neural Network Models

Tesis por compendio[ES] Durante la última década, los medios de comunicación han experimentado una revolución, alejándose de la televisión convencional hacia las plataformas de contenido bajo demanda. Además, esta revolución no ha cambiado solamente la manera en la que nos entretenemos, si no también la manera en la que aprendemos. En este sentido, las plataformas de contenido educativo bajo demanda también han proliferado para proporcionar recursos educativos de diversos tipos. Estas nuevas vías de distribución de contenido han llegado con nuevos requisitos para mejorar la accesibilidad, en particular las relacionadas con las dificultades de audición y las barreras lingüísticas. Aquí radica la oportunidad para el reconocimiento automático del habla (RAH) para cumplir estos requisitos, proporcionando subtitulado automático de alta calidad. Este subtitulado proporciona una base sólida para reducir esta brecha de accesibilidad, especialmente para contenido en directo o streaming. Estos sistemas de streaming deben trabajar bajo estrictas condiciones de tiempo real, proporcionando la subtitulación tan rápido como sea posible, trabajando con un contexto limitado. Sin embargo, esta limitación puede conllevar una degradación de la calidad cuando se compara con los sistemas para contenido en diferido u offline. Esta tesis propone un sistema de RAH en streaming con baja latencia, con una calidad similar a un sistema offline. Concretamente, este trabajo describe el camino seguido desde el sistema offline híbrido inicial hasta el eficiente sistema final de reconocimiento en streaming. El primer paso es la adaptación del sistema para efectuar una sola iteración de reconocimiento haciendo uso de modelos de lenguaje estado del arte basados en redes neuronales. En los sistemas basados en múltiples iteraciones estos modelos son relegados a una segunda (o posterior) iteración por su gran coste computacional. Tras adaptar el modelo de lenguaje, el modelo acústico basado en redes neuronales también tiene que adaptarse para trabajar con un contexto limitado. La integración y la adaptación de estos modelos es ampliamente descrita en esta tesis, evaluando el sistema RAH resultante, completamente adaptado para streaming, en conjuntos de datos académicos extensamente utilizados y desafiantes tareas basadas en contenidos audiovisuales reales. Como resultado, el sistema proporciona bajas tasas de error con un reducido tiempo de respuesta, comparables al sistema offline.[CA] Durant l'última dècada, els mitjans de comunicació han experimentat una revolució, allunyant-se de la televisió convencional cap a les plataformes de contingut sota demanda. A més a més, aquesta revolució no ha canviat només la manera en la que ens entretenim, si no també la manera en la que aprenem. En aquest sentit, les plataformes de contingut educatiu sota demanda també han proliferat pera proporcionar recursos educatius de diversos tipus. Aquestes noves vies de distribució de contingut han arribat amb nous requisits per a millorar l'accessibilitat, en particular les relacionades amb les dificultats d'audició i les barreres lingüístiques. Aquí radica l'oportunitat per al reconeixement automàtic de la parla (RAH) per a complir aquests requisits, proporcionant subtitulat automàtic d'alta qualitat. Aquest subtitulat proporciona una base sòlida per a reduir aquesta bretxa d'accessibilitat, especialment per a contingut en directe o streaming. Aquests sistemes han de treballar sota estrictes condicions de temps real, proporcionant la subtitulació tan ràpid com sigui possible, treballant en un context limitat. Aquesta limitació, però, pot comportar una degradació de la qualitat quan es compara amb els sistemes per a contingut en diferit o offline. Aquesta tesi proposa un sistema de RAH en streaming amb baixa latència, amb una qualitat similar a un sistema offline. Concretament, aquest treball descriu el camí seguit des del sistema offline híbrid inicial fins l'eficient sistema final de reconeixement en streaming. El primer pas és l'adaptació del sistema per a efectuar una sola iteració de reconeixement fent servir els models de llenguatge de l'estat de l'art basat en xarxes neuronals. En els sistemes basats en múltiples iteracions aquests models son relegades a una segona (o posterior) iteració pel seu gran cost computacional. Un cop el model de llenguatge s'ha adaptat, el model acústic basat en xarxes neuronals també s'ha d'adaptar per a treballar amb un context limitat. La integració i l'adaptació d'aquests models és àmpliament descrita en aquesta tesi, avaluant el sistema RAH resultant, completament adaptat per streaming, en conjunts de dades acadèmiques àmpliament utilitzades i desafiants tasques basades en continguts audiovisuals reals. Com a resultat, el sistema proporciona baixes taxes d'error amb un reduït temps de resposta, comparables al sistema offline.[EN] Over the last decade, the media have experienced a revolution, turning away from the conventional TV in favor of on-demand platforms. In addition, this media revolution not only changed the way entertainment is conceived but also how learning is conducted. Indeed, on-demand educational platforms have also proliferated and are now providing educational resources on diverse topics. These new ways to distribute content have come along with requirements to improve accessibility, particularly related to hearing difficulties and language barriers. Here is the opportunity for automatic speech recognition (ASR) to comply with these requirements by providing high-quality automatic captioning. Automatic captioning provides a sound basis for diminishing the accessibility gap, especially for live or streaming content. To this end, streaming ASR must work under strict real-time conditions, providing captions as fast as possible, and working with limited context. However, this limited context usually leads to a quality degradation as compared to the pre-recorded or offline content. This thesis is aimed at developing low-latency streaming ASR with a quality similar to offline ASR. More precisely, it describes the path followed from an initial hybrid offline system to an efficient streaming-adapted system. The first step is to perform a single recognition pass using a state-of-the-art neural network-based language model. In conventional multi-pass systems, this model is often deferred to the second or later pass due to its computational complexity. As with the language model, the neural-based acoustic model is also properly adapted to work with limited context. The adaptation and integration of these models is thoroughly described and assessed using fully-fledged streaming systems on well-known academic and challenging real-world benchmarks. In brief, it is shown that the proposed adaptation of the language and acoustic models allows the streaming-adapted system to reach the accuracy of the initial offline system with low latency.Jorge Cano, J. (2022). Streaming Automatic Speech Recognition with Hybrid Architectures and Deep Neural Network Models [Tesis doctoral]. Universitat Politècnica de València. https://doi.org/10.4995/Thesis/10251/191001Compendi

Live Streaming Speech Recognition Using Deep Bidirectional LSTM Acoustic Models and Interpolated Language Models

[EN] Although Long-Short Term Memory (LSTM) networks and deep Transformers are now extensively used in offline ASR, it is unclear how best offline systems can be adapted to work with them under the streaming setup. After gaining considerable experience on this regard in recent years, in this paper we show how an optimized, low-latency streaming decoder can be built in which bidirectional LSTM acoustic models, together with general interpolated language models, can be nicely integrated with minimal performance degradation. In brief, our streaming decoder consists of a one-pass, real-time search engine relying on a limited-duration window sliding over time and a number of ad hoc acoustic and language model pruning techniques. Extensive empirical assessment is provided on truly streaming tasks derived from the well-known LibriSpeech and TED talks datasets, as well as from TV shows on a main Spanish broadcasting station.This work was supported in part by European Union's Horizon 2020 Research and Innovation Programme under Grant 761758 (X5gon), and 952215 (TAILOR) and Erasmus+ Education Program under Grant Agreement 20-226-093604-SCH, in part by MCIN/AEI/10.13039/501100011033 ERDF A way of making Europe under Grant RTI2018-094879-B-I00, and in part by Generalitat Valenciana's Research Project Classroom Activity Recognition under Grant PROMETEO/2019/111. Funding for open access charge: CRUE-Universitat Politecnica de Valencia. The associate editor coordinating the review of this manuscript and approving it for publication was Prof. Lei Xie.Jorge-Cano, J.; Giménez Pastor, A.; Silvestre Cerdà, JA.; Civera Saiz, J.; Sanchis Navarro, JA.; Juan, A. (2022). Live Streaming Speech Recognition Using Deep Bidirectional LSTM Acoustic Models and Interpolated Language Models. IEEE/ACM Transactions on Audio Speech and Language Processing. 30:148-161. https://doi.org/10.1109/TASLP.2021.3133216S1481613

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

Streaming cascade-based speech translation leveraged by a direct segmentation model

[EN] The cascade approach to Speech Translation (ST) is based on a pipeline that concatenates an Automatic Speech Recognition (ASR) system followed by a Machine Translation (MT) system. Nowadays, state-of-the-art ST systems are populated with deep neural networks that are conceived to work in an offline setup in which the audio input to be translated is fully available in advance. However, a streaming setup defines a completely different picture, in which an unbounded audio input gradually becomes available and at the same time the translation needs to be generated under real-time constraints. In this work, we present a state-of-the-art streaming ST system in which neural-based models integrated in the ASR and MT components are carefully adapted in terms of their training and decoding procedures in order to run under a streaming setup. In addition, a direct segmentation model that adapts the continuous ASR output to the capacity of simultaneous MT systems trained at the sentence level is introduced to guarantee low latency while preserving the translation quality of the complete ST system. The resulting ST system is thoroughly evaluated on the real-life streaming Europarl-ST benchmark to gauge the trade-off between quality and latency for each component individually as well as for the complete ST system.The research leading to these results has received funding from the European Union's Horizon 2020 research and innovation program under grant agreement no. 761758 (X5Gon) and 952215 (TAILOR); the Government of Spain's research project Multisub, ref. RTI2018-094879-B-I00 (MCIU/AEI/FEDER,EU) and FPU scholarships FPU14/03981 and FPU18/04135; and the Generalitat Valenciana's research project Classroom Activity Recognition, ref. PROMETEO/2019/111 and predoctoral research scholarship ACIF/2017/055.Iranzo-Sánchez, J.; Jorge-Cano, J.; Baquero-Arnal, P.; Silvestre Cerdà, JA.; Giménez Pastor, A.; Civera Saiz, J.; Sanchis Navarro, JA.... (2021). Streaming cascade-based speech translation leveraged by a direct segmentation model. Neural Networks. 142:303-315. https://doi.org/10.1016/j.neunet.2021.05.013S30331514

Adapting Language Models When Training on Privacy-Transformed Data

International audienceIn recent years, voice-controlled personal assistants have revolutionized the interaction with smart devices and mobileapplications. The collected data are then used by system providers to train language models (LMs). Each spoken message reveals personal information, hence removing private information from the input sentences is necessary. Our data sanitization process relies on recognizing and replacing named entities by other words from the same class. However, this may harm LM training because privacy-transformed data is unlikely to match the test distribution. This paper aims to fill the gap by focusing on the adaptation of LMs initially trained on privacy-transformed sentences using a small amount of original untransformed data. To do so, we combine class-based LMs, which provide an effective approach to overcome data sparsity in the context of n-gram LMs, and neural LMs, which handle longer contexts and can yield better predictions. Our experiments show that training an LM on privacy-transformed data result in a relative 11% word error rate (WER) increase compared to training on the original untransformed data, and adapting that model on a limited amount of original untransformed data leads to a relative 8% WER improvement over the model trained solely on privacy-transformed data

Tackling Sequence to Sequence Mapping Problems with Neural Networks

In Natural Language Processing (NLP), it is important to detect the relationship between two sequences or to generate a sequence of tokens given another observed sequence. We call the type of problems on modelling sequence pairs as sequence to sequence (seq2seq) mapping problems. A lot of research has been devoted to finding ways of tackling these problems, with traditional approaches relying on a combination of hand-crafted features, alignment models, segmentation heuristics, and external linguistic resources. Although great progress has been made, these traditional approaches suffer from various drawbacks, such as complicated pipeline, laborious feature engineering, and the difficulty for domain adaptation. Recently, neural networks emerged as a promising solution to many problems in NLP, speech recognition, and computer vision. Neural models are powerful because they can be trained end to end, generalise well to unseen examples, and the same framework can be easily adapted to a new domain. The aim of this thesis is to advance the state-of-the-art in seq2seq mapping problems with neural networks. We explore solutions from three major aspects: investigating neural models for representing sequences, modelling interactions between sequences, and using unpaired data to boost the performance of neural models. For each aspect, we propose novel models and evaluate their efficacy on various tasks of seq2seq mapping.Comment: PhD thesi