Using stacked transformations for recognizing foreign accented speech

A common problem in speech recognition for foreign accented speech is that there is not enough training data for an accent-specific or a speaker-specific recognizer. Speaker adaptation can be used to improve the accuracy of a speaker independent recognizer, but a lot of adaptation data is needed for speakers with a strong foreign accent. In this paper we propose a rather simple and successful technique of stacked transformations where the baseline models trained for native speakers are first adapted by using accent-specific data and then by another transformation using speaker-specific data. Because the accent-specific data can be collected offline, the first transformation can be more detailed and comprehensive, and the second one less detailed and fast. Experimental results are provided for speaker adaptation in English spoken by Finnish speakers. The evaluation results confirm that the stacked transformations are very helpful for fast speaker adaptation.Peer reviewe

Stacked transformations for foreign accented speech recognition

Nowadays, large vocabulary speech recognizers exist that are performing reasonably well for specific conditions and environments. When the conditions change however, performance degrades quickly. For example, when the person to be recognized has a foreign accent the conditions could mismatch with the model, resulting in high error rates. The problem in recognizing foreign accented speech is the lack of sufficient training data. If enough data would be available of the same accent, from numerous different speakers, a well performing accented speech model could be built. Besides the lack of speech data, there are more problems with training a complete new model. It costs a lot of computational resources and storage space to train a new model. If speakers with different accents must be recognized, these costs explode as every accent needs retraining. A common solution for preventing retraining is to adapt (transform) an existing model, such that it better matches the recognition conditions. In this thesis multiple different adaptation transformations are considered. Speaker Transformations are using speech data from the target speaker, Accent Transformations use speech data from different speakers, who have the same accent as the speech that needs to be recognized. Neighbour Transformations are estimated with speech from different speakers that are automatically determined to be similar to the target speaker. Novelty in this work is the stack wise combination of these adaptations. Instead of using a single transformation, multiple transformations are 'stacked together'. Because all adaptations except the speaker specific adaptation can be precomputed, no extra computational costs at recognition time occur compared to normal speaker adaptation and the adaptations that can be precomputed are much more refined as they can use more and better adaptation data. In addition, they need only a very small amount storage space, compared to a retrained model. The effect of Stacked Transformations is that the models have a better fit for the recognition utterances. When compared to no adaptation, improvements up to 30% in Word Error Rate can be achieved. In adaptation with a small number (5) of sentences, improvements up to 15% are gained

Modeling DNN as human learner

In previous experiments, human listeners demonstrated that they had the ability to adapt to unheard, ambiguous phonemes after some initial, relatively short exposures. At the same time, previous work in the speech community has shown that pre-trained deep neural network-based (DNN) ASR systems, like humans, also have the ability to adapt to unseen, ambiguous phonemes after retuning their parameters on a relatively small set. In the first part of this thesis, the time-course of phoneme category adaptation in a DNN is investigated in more detail. By retuning the DNNs with more and more tokens with ambiguous sounds and comparing classification accuracy of the ambiguous phonemes in a held-out test across the time-course, we found out that DNNs, like human listeners, also demonstrated fast adaptation: the accuracy curves were step-like in almost all cases, showing very little adaptation after seeing only one (out of ten) training bins. However, unlike our experimental setup mentioned above, in a typical lexically guided perceptual learning experiment, listeners are trained with individual words instead of individual phones, and thus to truly model such a scenario, we would require a model that could take the context of a whole utterance into account. Traditional speech recognition systems accomplish this through the use of hidden Markov models (HMM) and WFST decoding. In recent years, bidirectional long short-term memory (Bi-LSTM) trained under connectionist temporal classification (CTC) criterion has also attracted much attention. In the second part of this thesis, previous experiments on ambiguous phoneme recognition were carried out again on a new Bi-LSTM model, and phonetic transcriptions of words ending with ambiguous phonemes were used as training targets, instead of individual sounds that consisted of a single phoneme. We found out that despite the vastly different architecture, the new model showed highly similar behavior in terms of classification rate over the time course of incremental retuning. This indicated that ambiguous phonemes in a continuous context could also be quickly adapted by neural network-based models. In the last part of this thesis, our pre-trained Dutch Bi-LSTM from the previous part was treated as a Dutch second language learner and was asked to transcribe English utterances in a self-adaptation scheme. In other words, we used the Dutch model to generate phonetic transcriptions directly and retune the model on the transcriptions it generated, although ground truth transcriptions were used to choose a subset of all self-labeled transcriptions. Self-adaptation is of interest as a model of human second language learning, but also has great practical engineering value, e.g., it could be used to adapt speech recognition to a lowr-resource language. We investigated two ways to improve the adaptation scheme, with the first being multi-task learning with articulatory feature detection during training the model on Dutch and self-labeled adaptation, and the second being first letting the model adapt to isolated short words before feeding it with longer utterances.Ope

English Lexical Stress Recognition Using Recurrent Neural Networks

Lexical stress is an integral part of English pronunciation. The command of lexical stress has an effect on the perceived fluency of the speaker. Moreover, it serves as a cue to recognize words. Methods that can automatically recognize lexical stress in spoken audio can be used to help English learners improve their pronunciation. This thesis evaluated lexical stress recognition methods based on recurrent neural networks. The purpose was to compare two sets of features: a set of prosodic features making use of existing speech recognition technologies, and simple spectral features. Using the latter feature set would allow for an end-to-end model, significantly simplifying the overall process. The problem was formulated as one of locating the primary stress, the most prominently stressed syllable in the word, in an isolated word. Datasets of both native and non-native speech were used in the experiments. The results show that models using the prosodic features outperform models using the spectral features. The difference between the two was particularly stark on the non-native dataset. It is possible that the datasets were too small to enable training end-to-end models. There was a considerable variation in performance among different words. It was also observed that the presence of a secondary stress made it more difficult to detect the primary stress.Sanapaino on olennainen osa englannin kielen ääntämistä. Sen osaaminen vaikuttaa puhujan havaittuun sujuvuuteen, ja se toimii vihjeenä sanojen tunnistamiselle. Menetelmiä, joilla sanapaino voidaan automaattisesti tunnistaa puheesta, voidaan käyttää apuna englannin oppijoiden ääntämisen parantamisessa. Tämä diplomityö arvioi takaisinkytkeytyviin neuroverkkoihin perustuvia menetelmiä sanapainon tunnistukseen. Tarkoitus oli vertailla kahdenlaisia piirteitä: joukkoa prosodisia piirteitä, jotka hyödyntävät olemassa olevia puheentunnistusteknologioita, ja yksinkertaisia äänen spektriin perustuvia piirteitä. Jälkimmäisten piirteiden käyttö mahdollistaisi päästä-päähän -mallien käyttämisen, mikä yksinkertaistaisi kokonaisprosessia merkittävästi. Ongelma esitettiin muodossa, jossa tarkoitus oli löytää pääpainon sijainti, eli sanan voimakkaiten erottuva tavu, yksittäisestä sanasta. Tutkimuksessa käytettiin dataa sekä englantia äidinkielenään että ei-äidinkielenään puhuvilta. Tulosten mukaan prosodisia piirteitä käyttävät mallit suoriutuvat tehtävästä paremmin kuin äänen spektriin perustuvia piirteitä käyttävät mallit. Erot olivat erityisen suuria datajoukossa, joka koostui englantia ei-äidinkielenään puhuvien puheesta. On mahdollista, että käytetyt datajoukot olivat liian pieniä päästä-päähän -mallien opettamista varten. Mallien suorituskyvyssä oli huomattavaa vaihtelua eri sanojen välillä. Tutkimuksessa havaittiin myös, että sivupainon läsnäolo vaikeutti pääpainon tunnistamista

A Sound Approach to Language Matters: In Honor of Ocke-Schwen Bohn

The contributions in this Festschrift were written by Ocke’s current and former PhD-students, colleagues and research collaborators. The Festschrift is divided into six sections, moving from the smallest building blocks of language, through gradually expanding objects of linguistic inquiry to the highest levels of description - all of which have formed a part of Ocke’s career, in connection with his teaching and/or his academic productions: “Segments”, “Perception of Accent”, “Between Sounds and Graphemes”, “Prosody”, “Morphology and Syntax” and “Second Language Acquisition”. Each one of these illustrates a sound approach to language matters

Acoustic model selection for recognition of regional accented speech

Accent is cited as an issue for speech recognition systems. Our experiments showed that the ASR word error rate is up to seven times greater for accented speech compared with standard British English. The main objective of this research is to develop Automatic Speech Recognition (ASR) techniques that are robust to accent variation. We applied different acoustic modelling techniques to compensate for the effects of regional accents on the ASR performance. For conventional GMM-HMM based ASR systems, we showed that using a small amount of data from a test speaker to choose an accent dependent model using an accent identification system, or building a model using the data from N neighbouring speakers in AID space, will result in superior performance compared to that obtained with unsupervised or supervised speaker adaptation. In addition we showed that using a DNN-HMM rather than a GMM-HMM based acoustic model would improve the recognition accuracy considerably. Even if we apply two stages of accent followed by speaker adaptation to the GMM-HMM baseline system, the GMM-HMM based system will not outperform the baseline DNN-HMM based system. For more contemporary DNN-HMM based ASR systems we investigated how adding different types of accented data to the training set can provide better recognition accuracy on accented speech. Finally, we proposed a new approach for visualisation of the AID feature space. This is helpful in analysing the AID recognition accuracies and analysing AID confusion matrices

Data-Driven Enhancement of State Mapping-Based Cross-Lingual Speaker Adaptation

The thesis work was motivated by the goal of developing personalized speech-to-speech translation and focused on one of its key component techniques – cross-lingual speaker adaptation for text-to-speech synthesis. A personalized speech-to-speech translator enables a person’s spoken input to be translated into spoken output in another language while maintaining his/her voice identity. Before addressing any technical issues, work in this thesis set out to understand human perception of speaker identity. Listening tests were conducted in order to determine whether people could differentiate between speakers when they spoke different languages. The results demonstrated that differentiating between speakers across languages was an achievable task. However, it was difficult for listeners to differentiate between speakers across both languages and speech types (original recordings versus synthesized samples). The underlying challenge in cross-lingual speaker adaptation is how to apply speaker adaptation techniques when the language of adaptation data is different from that of synthesis models. The main body of the thesis work was devoted to the analysis and improvement of HMM state mapping-based cross-lingual speaker adaptation. Firstly, the effect of unsupervised cross-lingual adaptation was investigated, as it relates to the application scenario of personalized speech-to-speech translation. The comparison of paired supervised and unsupervised systems shows that the performance of unsupervised cross-lingual speaker adaptation is comparable to that of the supervised fashion, even if the average phoneme error rate of the unsupervised systems is around 75%. Then the effect of the language mismatch between synthesis models and adaptation data was investigated. The mismatch is found to transfer undesirable language information from adaptation data to synthesis models, thereby limiting the effectiveness of generating multiple regression class-specific transforms, using larger quantities of adaptation data and estimating adaptation transforms iteratively. Thirdly, in order to tackle the problems caused by the language mismatch, a data-driven adaptation framework using phonological knowledge is proposed. Its basic idea is to group HMM states according to phonological knowledge in a data-driven manner and then to map each state to a phonologically consistent counterpart in a different language. This framework is also applied to regression class tree construction for transform estimation. It is found that the proposed framework alleviates the negative effect of the language mismatch and gives consistent improvement compared to previous state-of-the-art approaches. Finally, a two-layer hierarchical transformation framework is developed, where one layer captures speaker characteristics and the other compensates for the language mismatch. The most appropriate means to construct the hierarchical arrangement of transforms was investigated in an initial study. While early results show some promise, further in-depth investigation is needed to confirm the validity of this hierarchy

Modeling of Polish Intonation for Statistical-Parametric Speech Synthesis

Wydział NeofilologiiBieżąca praca prezentuje próbę budowy neurobiologicznie umotywowanego modelu mapowań pomiędzy wysokopoziomowymi dyskretnymi kategoriami lingwistycznymi a ciągłym sygnałem częstotliwości podstawowej w polskiej neutralnej mowie czytanej, w oparciu o konwolucyjne sieci neuronowe. Po krótkim wprowadzeniu w problem badawczy w kontekście intonacji, syntezy mowy oraz luki pomiędzy fonetyką a fonologią, praca przedstawia opis uczenia modelu na podstawie specjalnego korpusu mowy oraz ewaluację naturalności konturu F0 generowanego przez wyuczony model za pomocą eksperymentów percepcyjnych typu ABX oraz MOS przy użyciu specjalnie w tym celu zbudowanego resyntezatora Neural Source Filter. Następnie, prezentowane są wyniki eksploracji fonologiczno-fonetycznych mapowań wyuczonych przez model. W tym celu wykorzystana została jedna z tzw. metod wyjaśniających dla sztucznej inteligencji – Layer-wise Relevance Propagation. W pracy przedstawione zostały wyniki powstałej na tej podstawie obszernej analizy ilościowej istotności dla konturu częstotliwości podstawowej każdej z 1297 specjalnie wygenerowanych lingwistycznych kategorii wejściowych modelu oraz ich wielorakich grupowań na różnorodnych poziomach abstrakcji. Pracę kończy dogłębna analiza oraz interpretacja uzyskanych wyników oraz rozważania na temat mocnych oraz słabych stron zastosowanych metod, a także lista proponowanych usprawnień.This work presents an attempt to build a neurobiologically inspired Convolutional Neural Network-based model of the mappings between discrete high-level linguistic categories into a continuous signal of fundamental frequency in Polish neutral read speech. After a brief introduction of the current research problem in the context of intonation, speech synthesis and the phonetic-phonology gap, the work goes on to describe the training of the model on a special speech corpus, and an evaluation of the naturalness of the F0 contour produced by the trained model through ABX and MOS perception experiments conducted with help of a specially built Neural Source Filter resynthesizer. Finally, an in-depth exploration of the phonology-to-phonetics mappings learned by the model is presented; the Layer-wise Relevance Propagation explainability method was used to perform an extensive quantitative analysis of the relevance of 1297 specially engineered linguistic input features and their groupings at various levels of abstraction for the specific contours of the fundamental frequency. The work ends with an in-depth interpretation of these results and a discussion of the advantages and disadvantages of the current method, and lists a number of potential future improvements.Badania przedstawione w pracy zostały cz˛e´sciowo zrealizowane w ramach grantu badawczego Harmonia nr UMO-2014/14/M/HS2/00631 przyznanego przez Narodowe Centrum Nauki