Search on speech from spoken queries: the Multi-domain International ALBAYZIN 2018 Query-by-Example Spoken Term Detection Evaluation

[Abstract] The huge amount of information stored in audio and video repositories makes search on speech (SoS) a priority area nowadays. Within SoS, Query-by-Example Spoken Term Detection (QbE STD) aims to retrieve data from a speech repository given a spoken query. Research on this area is continuously fostered with the organization of QbE STD evaluations. This paper presents a multi-domain internationally open evaluation for QbE STD in Spanish. The evaluation aims at retrieving the speech files that contain the queries, providing their start and end times, and a score that reflects the confidence given to the detection. Three different Spanish speech databases that encompass different domains have been employed in the evaluation: MAVIR database, which comprises a set of talks from workshops; RTVE database, which includes broadcast television (TV) shows; and COREMAH database, which contains 2-people spontaneous speech conversations about different topics. The evaluation has been designed carefully so that several analyses of the main results can be carried out. We present the evaluation itself, the three databases, the evaluation metrics, the systems submitted to the evaluation, the results, and the detailed post-evaluation analyses based on some query properties (within-vocabulary/out-of-vocabulary queries, single-word/multi-word queries, and native/foreign queries). Fusion results of the primary systems submitted to the evaluation are also presented. Three different teams took part in the evaluation, and ten different systems were submitted. The results suggest that the QbE STD task is still in progress, and the performance of these systems is highly sensitive to changes in the data domain. Nevertheless, QbE STD strategies are able to outperform text-based STD in unseen data domains.Centro singular de investigación de Galicia; ED431G/04Universidad del País Vasco; GIU16/68Ministerio de Economía y Competitividad; TEC2015-68172-C2-1-PMinisterio de Ciencia, Innovación y Competitividad; RTI2018-098091-B-I00Xunta de Galicia; ED431G/0

Modelo acústico de língua inglesa falada por portugueses

Trabalho de projecto de mestrado em Engenharia Informática, apresentado à Universidade de Lisboa, através da Faculdade de Ciências, 2007No contexto do reconhecimento robusto de fala baseado em modelos de Markov não observáveis (do inglês Hidden Markov Models - HMMs) este trabalho descreve algumas metodologias e experiências tendo em vista o reconhecimento de oradores estrangeiros. Quando falamos em Reconhecimento de Fala falamos obrigatoriamente em Modelos Acústicos também. Os modelos acústicos reflectem a maneira como pronunciamos/articulamos uma língua, modelando a sequência de sons emitidos aquando da fala. Essa modelação assenta em segmentos de fala mínimos, os fones, para os quais existe um conjunto de símbolos/alfabetos que representam a sua pronunciação. É no campo da fonética articulatória e acústica que se estuda a representação desses símbolos, sua articulação e pronunciação. Conseguimos descrever palavras analisando as unidades que as constituem, os fones. Um reconhecedor de fala interpreta o sinal de entrada, a fala, como uma sequência de símbolos codificados. Para isso, o sinal é fragmentado em observações de sensivelmente 10 milissegundos cada, reduzindo assim o factor de análise ao intervalo de tempo onde as características de um segmento de som não variam. Os modelos acústicos dão-nos uma noção sobre a probabilidade de uma determinada observação corresponder a uma determinada entidade. É, portanto, através de modelos sobre as entidades do vocabulário a reconhecer que é possível voltar a juntar esses fragmentos de som. Os modelos desenvolvidos neste trabalho são baseados em HMMs. Chamam-se assim por se fundamentarem nas cadeias de Markov (1856 - 1922): sequências de estados onde cada estado é condicionado pelo seu anterior. Localizando esta abordagem no nosso domínio, há que construir um conjunto de modelos - um para cada classe de sons a reconhecer - que serão treinados por dados de treino. Os dados são ficheiros áudio e respectivas transcrições (ao nível da palavra) de modo a que seja possível decompor essa transcrição em fones e alinhá-la a cada som do ficheiro áudio correspondente. Usando um modelo de estados, onde cada estado representa uma observação ou segmento de fala descrita, os dados vão-se reagrupando de maneira a criar modelos estatísticos, cada vez mais fidedignos, que consistam em representações das entidades da fala de uma determinada língua. O reconhecimento por parte de oradores estrangeiros com pronuncias diferentes da língua para qual o reconhecedor foi concebido, pode ser um grande problema para precisão de um reconhecedor. Esta variação pode ser ainda mais problemática que a variação dialectal de uma determinada língua, isto porque depende do conhecimento que cada orador têm relativamente à língua estrangeira. Usando para uma pequena quantidade áudio de oradores estrangeiros para o treino de novos modelos acústicos, foram efectuadas diversas experiências usando corpora de Portugueses a falar Inglês, de Português Europeu e de Inglês. Inicialmente foi explorado o comportamento, separadamente, dos modelos de Ingleses nativos e Portugueses nativos, quando testados com os corpora de teste (teste com nativos e teste com não nativos). De seguida foi treinado um outro modelo usando em simultâneo como corpus de treino, o áudio de Portugueses a falar Inglês e o de Ingleses nativos. Uma outra experiência levada a cabo teve em conta o uso de técnicas de adaptação, tal como a técnica MLLR, do inglês Maximum Likelihood Linear Regression. Esta última permite a adaptação de uma determinada característica do orador, neste caso o sotaque estrangeiro, a um determinado modelo inicial. Com uma pequena quantidade de dados representando a característica que se quer modelar, esta técnica calcula um conjunto de transformações que serão aplicadas ao modelo que se quer adaptar. Foi também explorado o campo da modelação fonética onde estudou-se como é que o orador estrangeiro pronuncia a língua estrangeira, neste caso um Português a falar Inglês. Este estudo foi feito com a ajuda de um linguista, o qual definiu um conjunto de fones, resultado do mapeamento do inventário de fones do Inglês para o Português, que representam o Inglês falado por Portugueses de um determinado grupo de prestígio. Dada a grande variabilidade de pronúncias teve de se definir este grupo tendo em conta o nível de literacia dos oradores. Este estudo foi posteriormente usado na criação de um novo modelo treinado com os corpora de Portugueses a falar Inglês e de Portugueses nativos. Desta forma representamos um reconhecedor de Português nativo onde o reconhecimento de termos ingleses é possível. Tendo em conta a temática do reconhecimento de fala este projecto focou também a recolha de corpora para português europeu e a compilação de um léxico de Português europeu. Na área de aquisição de corpora o autor esteve envolvido na extracção e preparação dos dados de fala telefónica, para posterior treino de novos modelos acústicos de português europeu. Para compilação do léxico de português europeu usou-se um método incremental semi-automático. Este método consistiu em gerar automaticamente a pronunciação de grupos de 10 mil palavras, sendo cada grupo revisto e corrigido por um linguista. Cada grupo de palavras revistas era posteriormente usado para melhorar as regras de geração automática de pronunciações.The tremendous growth of technology has increased the need of integration of spoken language technologies into our daily applications, providing an easy and natural access to information. These applications are of different nature with different user’s interfaces. Besides voice enabled Internet portals or tourist information systems, automatic speech recognition systems can be used in home user’s experiences where TV and other appliances could be voice controlled, discarding keyboards or mouse interfaces, or in mobile phones and palm-sized computers for a hands-free and eyes-free manipulation. The development of these systems causes several known difficulties. One of them concerns the recognizer accuracy on dealing with non-native speakers with different phonetic pronunciations of a given language. The non-native accent can be more problematic than a dialect variation on the language. This mismatch depends on the individual speaking proficiency and speaker’s mother tongue. Consequently, when the speaker’s native language is not the same as the one that was used to train the recognizer, there is a considerable loss in recognition performance. In this thesis, we examine the problem of non-native speech in a speaker-independent and large-vocabulary recognizer in which a small amount of non-native data was used for training. Several experiments were performed using Hidden Markov models, trained with speech corpora containing European Portuguese native speakers, English native speakers and English spoken by European Portuguese native speakers. Initially it was explored the behaviour of an English native model and non-native English speakers’ model. Then using different corpus weights for the English native speakers and English spoken by Portuguese speakers it was trained a model as a pool of accents. Through adaptation techniques it was used the Maximum Likelihood Linear Regression method. It was also explored how European Portuguese speakers pronounce English language studying the correspondences between the phone sets of the foreign and target languages. The result was a new phone set, consequence of the mapping between the English and the Portuguese phone sets. Then a new model was trained with English Spoken by Portuguese speakers’ data and Portuguese native data. Concerning the speech recognition subject this work has other two purposes: collecting Portuguese corpora and supporting the compilation of a Portuguese lexicon, adopting some methods and algorithms to generate automatic phonetic pronunciations. The collected corpora was processed in order to train acoustic models to be used in the Exchange 2007 domain, namely in Outlook Voice Access

Speech Recognition

Chapters in the first part of the book cover all the essential speech processing techniques for building robust, automatic speech recognition systems: the representation for speech signals and the methods for speech-features extraction, acoustic and language modeling, efficient algorithms for searching the hypothesis space, and multimodal approaches to speech recognition. The last part of the book is devoted to other speech processing applications that can use the information from automatic speech recognition for speaker identification and tracking, for prosody modeling in emotion-detection systems and in other speech processing applications that are able to operate in real-world environments, like mobile communication services and smart homes

The Design and Application of an Acoustic Front-End for Use in Speech Interfaces

This thesis describes the design, implementation, and application of an acoustic front-end. Such front-ends constitute the core of automatic speech recognition systems. The front-end whose development is reported here has been designed for speaker-independent large vocabulary recognition. The emphasis of this thesis is more one of design than of application. This work exploits the current state-of-the-art in speech recognition research, for example, the use of Hidden Markov Models. It describes the steps taken to build a speaker-independent large vocabulary system from signal processing, through pattern matching, to language modelling. An acoustic front-end can be considered as a multi-stage process, each of which requires the specification of many parameters. Some parameters have fundamental consequences for the ultimate application of the front-end. Therefore, a major part of this thesis is concerned with their analysis and specification. Experiments were carried out to determine the characteristics of individual parameters, the results of which were then used to motivate particular parameter settings. The thesis concludes with some applications that point out, not only the power of the resulting acoustic front-end, but also its limitations

Reinforcement Learning

Brains rule the world, and brain-like computation is increasingly used in computers and electronic devices. Brain-like computation is about processing and interpreting data or directly putting forward and performing actions. Learning is a very important aspect. This book is on reinforcement learning which involves performing actions to achieve a goal. The first 11 chapters of this book describe and extend the scope of reinforcement learning. The remaining 11 chapters show that there is already wide usage in numerous fields. Reinforcement learning can tackle control tasks that are too complex for traditional, hand-designed, non-learning controllers. As learning computers can deal with technical complexities, the tasks of human operators remain to specify goals on increasingly higher levels. This book shows that reinforcement learning is a very dynamic area in terms of theory and applications and it shall stimulate and encourage new research in this field

A Soft Computing Based Approach for Multi-Accent Classification in IVR Systems

A speaker's accent is the most important factor affecting the performance of Natural Language Call Routing (NLCR) systems because accents vary widely, even within the same country or community. This variation also occurs when non-native speakers start to learn a second language, the substitution of native language phonology being a common process. Such substitution leads to fuzziness between the phoneme boundaries and phoneme classes, which reduces out-of-class variations, and increases the similarities between the different sets of phonemes. Thus, this fuzziness is the main cause of reduced NLCR system performance. The main requirement for commercial enterprises using an NLCR system is to have a robust NLCR system that provides call understanding and routing to appropriate destinations. The chief motivation for this present work is to develop an NLCR system that eliminates multilayered menus and employs a sophisticated speaker accent-based automated voice response system around the clock. Currently, NLCRs are not fully equipped with accent classification capability. Our main objective is to develop both speaker-independent and speaker-dependent accent classification systems that understand a caller's query, classify the caller's accent, and route the call to the acoustic model that has been thoroughly trained on a database of speech utterances recorded by such speakers. In the field of accent classification, the dominant approaches are the Gaussian Mixture Model (GMM) and Hidden Markov Model (HMM). Of the two, GMM is the most widely implemented for accent classification. However, GMM performance depends on the initial partitions and number of Gaussian mixtures, both of which can reduce performance if poorly chosen. To overcome these shortcomings, we propose a speaker-independent accent classification system based on a distance metric learning approach and evolution strategy. This approach depends on side information from dissimilar pairs of accent groups to transfer data points to a new feature space where the Euclidean distances between similar and dissimilar points are at their minimum and maximum, respectively. Finally, a Non-dominated Sorting Evolution Strategy (NSES)-based k-means clustering algorithm is employed on the training data set processed by the distance metric learning approach. The main objectives of the NSES-based k-means approach are to find the cluster centroids as well as the optimal number of clusters for a GMM classifier. In the case of a speaker-dependent application, a new method is proposed based on the fuzzy canonical correlation analysis to find appropriate Gaussian mixtures for a GMM-based accent classification system. In our proposed method, we implement a fuzzy clustering approach to minimize the within-group sum-of-square-error and canonical correlation analysis to maximize the correlation between the speech feature vectors and cluster centroids. We conducted a number of experiments using the TIMIT database, the speech accent archive, and the foreign accent English databases for evaluating the performance of speaker-independent and speaker-dependent applications. Assessment of the applications and analysis shows that our proposed methodologies outperform the HMM, GMM, vector quantization GMM, and radial basis neural networks

SYNTHESIZING DYSARTHRIC SPEECH USING MULTI-SPEAKER TTS FOR DSYARTHRIC SPEECH RECOGNITION

Dysarthria is a motor speech disorder often characterized by reduced speech intelligibility through slow, uncoordinated control of speech production muscles. Automatic Speech recognition (ASR) systems may help dysarthric talkers communicate more effectively. However, robust dysarthria-specific ASR requires a significant amount of training speech is required, which is not readily available for dysarthric talkers. In this dissertation, we investigate dysarthric speech augmentation and synthesis methods. To better understand differences in prosodic and acoustic characteristics of dysarthric spontaneous speech at varying severity levels, a comparative study between typical and dysarthric speech was conducted. These characteristics are important components for dysarthric speech modeling, synthesis, and augmentation. For augmentation, prosodic transformation and time-feature masking have been proposed. For dysarthric speech synthesis, this dissertation has introduced a modified neural multi-talker TTS by adding a dysarthria severity level coefficient and a pause insertion model to synthesize dysarthric speech for varying severity levels. In addition, we have extended this work by using a label propagation technique to create more meaningful control variables such as a continuous Respiration, Laryngeal and Tongue (RLT) parameter, even for datasets that only provide discrete dysarthria severity level information. This approach increases the controllability of the system, so we are able to generate more dysarthric speech with a broader range. To evaluate their effectiveness for synthesis of training data, dysarthria-specific speech recognition was used. Results show that a DNN-HMM model trained on additional synthetic dysarthric speech achieves WER improvement of 12.2% compared to the baseline, and that the addition of the severity level and pause insertion controls decrease WER by 6.5%, showing the effectiveness of adding these parameters. Overall results on the TORGO database demonstrate that using dysarthric synthetic speech to increase the amount of dysarthric-patterned speech for training has a significant impact on the dysarthric ASR systems