Fluency-related Temporal Features and Syllable Prominence as Prosodic Proficiency Predictors for Learners of English with Different Language Backgrounds

Prosodic features are important in achieving intelligibility, comprehensibility, and fluency in a second or foreign language (L2). However, research on the assessment of prosody as part of oral proficiency remains scarce. Moreover, the acoustic analysis of L2 prosody has often focused on fluency-related temporal measures, neglecting language-dependent stress features that can be quantified in terms of syllable prominence. Introducing the evaluation of prominence-related measures can be of use in developing both teaching and assessment of L2 speaking skills. In this study we compare temporal measures and syllable prominence estimates as predictors of prosodic proficiency in non-native speakers of English with respect to the speaker's native language (L1). The predictive power of temporal and prominence measures was evaluated for utterance-sized samples produced by language learners from four different L1 backgrounds: Czech, Slovak, Polish, and Hungarian. Firstly, the speech samples were assessed using the revised Common European Framework of Reference scale for prosodic features. The assessed speech samples were then analyzed to derive articulation rate and three fluency measures. Syllable-level prominence was estimated by a continuous wavelet transform analysis using combinations of F0, energy, and syllable duration. The results show that the temporal measures serve as reliable predictors of prosodic proficiency in the L2, with prominence measures providing a small but significant improvement to prosodic proficiency predictions. The predictive power of the individual measures varies both quantitatively and qualitatively depending on the L1 of the speaker. We conclude that the possible effects of the speaker's L1 on the production of L2 prosody in terms of temporal features as well as syllable prominence deserve more attention in applied research and developing teaching and assessment methods for spoken L2.Peer reviewe

Maximal Figure-of-Merit Framework to Detect Multi-label Phonetic Features for Spoken Language Recognition

Bottleneck features (BNFs) generated with a deep neural network (DNN) have proven to boost spoken language recognition accuracy over basic spectral features significantly. However, BNFs are commonly extracted using language-dependent tied-context phone states as learning targets. Moreover, BNFs are less phonetically expressive than the output layer in a DNN, which is usually not used as a speech feature because of its very high dimensionality hindering further post-processing. In this work, we put forth a novel deep learning framework to overcome all of the above issues and evaluate it on the 2017 NIST Language Recognition Evaluation (LRE) challenge. We use manner and place of articulation as speech attributes, which lead to low-dimensional “universal” phonetic features that can be defined across all spoken languages. To model the asynchronous nature of the speech attributes while capturing their intrinsic relationships in a given speech segment, we introduce a new training scheme for deep architectures based on a Maximal Figure of Merit (MFoM) objective. MFoM introduces non-differentiable metrics into the backpropagation-based approach, which is elegantly solved in the proposed framework. The experimental evidence collected on the recent NIST LRE 2017 challenge demonstrates the effectiveness of our solution. In fact, the performance of speech language recognition (SLR) systems based on spectral features is improved for more than 5% absolute Cavg. Finally, the F1 metric can be brought from 77.6% up to 78.1% by combining the conventional baseline phonetic BNFs with the proposed articulatory attribute features

Äärelliset tilamallit lukupuheen tunnistamisessa ja tarkastamisessa

An automatic speech recognition system has to combine acoustic and linguistic information. Therefore the search space spans multiple layers. Finite state models and weighted finite state transducers in particular can efficiently represent this search space by modeling each layer as a transducer and combining them using generic weighted finite state transducer algorithms. When recognising a text prompt being read aloud, the prompt gives a good estimate of what is going to be said. However human reading naturally produces some deviations from the text, called miscues. The purpose of this thesis is to create a system which accurately recognises recordings of reading. A miscue tolerant finite state language model is implemented and compared against two traditional approaches, an N-gram model and forced alignment. The recognition result will ultimately be used to validate the recording as fit for further automatic processing in a spoken foreign language exam, which Project DigiTala is designing for the Finnish matriculation examination. The computerization of the matriculation examination in Finland makes the use of such automatic tools possible. This thesis first introduces the context for the task of recognising and validating reading. Then it explores three methodologies needed to solve the task: automatic speech recognition, finite state models, and the modeling of reading. Next it recounts the implementation of the miscue tolerant finite state language models and the two baseline methods. After that it describes experiments which show that the miscue tolerant finite state language models solve the task of this thesis significantly better than the baseline methods. Finally the thesis concludes with a discussion of the results and future work.Automaattinen puheentunnistusjärjestelmä yhdistää akustista ja kielellistä tietoa, joten sen hakuavaruus on monitasoinen. Tämän hakuavaruuden voi esittää tehokkaasti äärellisillä tilamalleilla. Erityisesti painotetut äärelliset tilamuuttajat voivat esittää jokaista hakuavaruuden tasoa ja nämä muuttajat voidaan yhdistää yleisillä muuttaja-algoritmeilla. Kun tunnistetaan ääneen lukemista syötteestä, syöte rajaa hakuavaruutta hyvin. Ihmiset kuitenkin poikkeavat tekstistä hieman. Kutsun näitä lukupoikkeamiksi, koska ne ovat luonnollinen osa taitavaakin lukemista, eivätkä siis suoranaisesti lukuvirheitä. Tämän diplomityön tavoite on luoda järjestelmä, joka tunnistaa lukupuheäänitteitä tarkasti. Tätä varten toteutetaan lukupoikkeamia sietävä äärellisen tilan kielimalli, jota verrataan kahteen perinteiseen menetelmään, N-gram malleihin ja pakotettuun kohdistukseen. Lukupuheen tunnistustulosta käytetään, kun tarkastetaan, sopiiko äänite seuraaviin automaattisiin käsittelyvaiheisiin puhutussa vieraan kielen kokeessa. DigiTalaprojekti muotoilee puhuttua osiota vieraan kielen ylioppilaskokeisiin. Ylioppilaskokeiden sähköistäminen mahdollistaa tällaisten automaattisten menetelmien käytön. Kokeet sekä englanninkielisellä simuloidulla aineistolla että ruotsinkielisellä tosimaailman aineistolla osoittavat, että lukupoikkeamia sietävä äärellisen tilan kielimalli ratkaisee diplomityön ongelmanasettelun. Vaikealla tosimaailman aineistolla saadaan 3.77 ± 0.47 prosentuaalinen sanavirhemäärä

Current trends in multilingual speech processing

In this paper, we describe recent work at Idiap Research Institute in the domain of multilingual speech processing and provide some insights into emerging challenges for the research community. Multilingual speech processing has been a topic of ongoing interest to the research community for many years and the field is now receiving renewed interest owing to two strong driving forces. Firstly, technical advances in speech recognition and synthesis are posing new challenges and opportunities to researchers. For example, discriminative features are seeing wide application by the speech recognition community, but additional issues arise when using such features in a multilingual setting. Another example is the apparent convergence of speech recognition and speech synthesis technologies in the form of statistical parametric methodologies. This convergence enables the investigation of new approaches to unified modelling for automatic speech recognition and text-to-speech synthesis (TTS) as well as cross-lingual speaker adaptation for TTS. The second driving force is the impetus being provided by both government and industry for technologies to help break down domestic and international language barriers, these also being barriers to the expansion of policy and commerce. Speech-to-speech and speech-to-text translation are thus emerging as key technologies at the heart of which lies multilingual speech processin

The effects of adverse conditions on speech recognition by non-native listeners: Electrophysiological and behavioural evidence

This thesis investigated speech recognition by native (L1) and non-native (L2) listeners (i.e., native English and Korean speakers) in diverse adverse conditions using electroencephalography (EEG) and behavioural measures. Study 1 investigated speech recognition in noise for read and casually produced, spontaneous speech using behavioural measures. The results showed that the detrimental effect of casual speech was greater for L2 than L1 listeners, demonstrating real-life L2 speech recognition problems caused by casual speech. Intelligibility was also shown to decrease when the accents of the talker and listener did not match when listening to casual speech as well as read speech. Study 2 set out to develop EEG methods to measure L2 speech processing difficulties for natural, continuous speech. This study thus examined neural entrainment to the amplitude envelope of speech (i.e., slow amplitude fluctuations in speech) while subjects listened to their L1, L2 and a language that they did not understand. The results demonstrate that neural entrainment to the speech envelope is not modulated by whether or not listeners understand the language, opposite to previously reported positive relationships between speech entrainment and intelligibility. Study 3 investigated speech processing in a two-talker situation using measures of neural entrainment and N400, combined with a behavioural speech recognition task. L2 listeners had greater entrainment for target talkers than did L1 listeners, likely because their difficulty with L2 speech comprehension caused them to focus greater attention on the speech signal. L2 listeners also had a greater degree of lexical processing (i.e., larger N400) for highly predictable words than did native listeners, while native listeners had greater lexical processing when listening to foreign-accented speech. The results suggest that the increased listening effort experienced by L2 listeners during speech recognition modulates their auditory and lexical processing

A computational model for studying L1’s effect on L2 speech learning

abstract: Much evidence has shown that first language (L1) plays an important role in the formation of L2 phonological system during second language (L2) learning process. This combines with the fact that different L1s have distinct phonological patterns to indicate the diverse L2 speech learning outcomes for speakers from different L1 backgrounds. This dissertation hypothesizes that phonological distances between accented speech and speakers' L1 speech are also correlated with perceived accentedness, and the correlations are negative for some phonological properties. Moreover, contrastive phonological distinctions between L1s and L2 will manifest themselves in the accented speech produced by speaker from these L1s. To test the hypotheses, this study comes up with a computational model to analyze the accented speech properties in both segmental (short-term speech measurements on short-segment or phoneme level) and suprasegmental (long-term speech measurements on word, long-segment, or sentence level) feature space. The benefit of using a computational model is that it enables quantitative analysis of L1's effect on accent in terms of different phonological properties. The core parts of this computational model are feature extraction schemes to extract pronunciation and prosody representation of accented speech based on existing techniques in speech processing field. Correlation analysis on both segmental and suprasegmental feature space is conducted to look into the relationship between acoustic measurements related to L1s and perceived accentedness across several L1s. Multiple regression analysis is employed to investigate how the L1's effect impacts the perception of foreign accent, and how accented speech produced by speakers from different L1s behaves distinctly on segmental and suprasegmental feature spaces. Results unveil the potential application of the methodology in this study to provide quantitative analysis of accented speech, and extend current studies in L2 speech learning theory to large scale. Practically, this study further shows that the computational model proposed in this study can benefit automatic accentedness evaluation system by adding features related to speakers' L1s.Dissertation/ThesisDoctoral Dissertation Speech and Hearing Science 201

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

Automatic Speech Recognition for Low-resource Languages and Accents Using Multilingual and Crosslingual Information

This thesis explores methods to rapidly bootstrap automatic speech recognition systems for languages, which lack resources for speech and language processing. We focus on finding approaches which allow using data from multiple languages to improve the performance for those languages on different levels, such as feature extraction, acoustic modeling and language modeling. Under application aspects, this thesis also includes research work on non-native and Code-Switching speech