Automatic speech recognition: from study to practice

Today, automatic speech recognition (ASR) is widely used for different purposes such as robotics, multimedia, medical and industrial application. Although many researches have been performed in this field in the past decades, there is still a lot of room to work. In order to start working in this area, complete knowledge of ASR systems as well as their weak points and problems is inevitable. Besides that, practical experience improves the theoretical knowledge understanding in a reliable way. Regarding to these facts, in this master thesis, we have first reviewed the principal structure of the standard HMM-based ASR systems from technical point of view. This includes, feature extraction, acoustic modeling, language modeling and decoding. Then, the most significant challenging points in ASR systems is discussed. These challenging points address different internal components characteristics or external agents which affect the ASR systems performance. Furthermore, we have implemented a Spanish language recognizer using HTK toolkit. Finally, two open research lines according to the studies of different sources in the field of ASR has been suggested for future work

Implementation and Evaluation of Acoustic Distance Measures for Syllables

Munier C. Implementation and Evaluation of Acoustic Distance Measures for Syllables. Bielefeld (Germany): Bielefeld University; 2011.In dieser Arbeit werden verschiedene akustische Ähnlichkeitsmaße für Silben motiviert und anschließend evaluiert. Der Mahalanobisabstand als lokales Abstandsmaß für einen Dynamic-Time-Warping-Ansatz zum Messen von akustischen Abständen hat die Fähigkeit, Silben zu unterscheiden. Als solcher erlaubt er die Klassifizierung von Silben mit einer Genauigkeit, die für die Klassifizierung von kleinen akustischen Einheiten üblich ist (60 Prozent für eine Nächster-Nachbar-Klassifizierung auf einem Satz von zehn Silben für Samples eines einzelnen Sprechers). Dieses Maß kann durch verschiedene Techniken verbessert werden, die jedoch seine Ausführungsgeschwindigkeit verschlechtern (Benutzen von mehr Mischverteilungskomponenten für die Schätzung von Kovarianzen auf einer Gaußschen Mischverteilung, Benutzen von voll besetzten Kovarianzmatrizen anstelle von diagonalen Kovarianzmatrizen). Durch experimentelle Evaluierung wird deutlich, dass ein gut funktionierender Algorithmus zur Silbensegmentierung, welcher eine akkurate Schätzung von Silbengrenzen erlaubt, für die korrekte Berechnung von akustischen Abständen durch die in dieser Arbeit entwickelten Ähnlichkeitsmaße unabdingbar ist. Weitere Ansätze für Ähnlichkeitsmaße, die durch ihre Anwendung in der Timbre-Klassifizierung von Musikstücken motiviert sind, zeigen keine adäquate Fähigkeit zur Silbenunterscheidung.In this work, several acoustic similarity measures for syllables are motivated and successively evaluated. The Mahalanobis distance as local distance measure for a dynamic time warping approach to measure acoustic distances is a measure that is able to discriminate syllables and thus allows for syllable classification with an accuracy that is common to the classification of small acoustic units (60 percent for a nearest neighbor classification of a set of ten syllables using samples of a single speaker). This measure can be improved using several techniques that however impair the execution speed of the distance measure (usage of more mixture density components for the estimation of covariances from a Gaussian mixture model, usage of fully occupied covariance matrices instead of diagonal covariance matrices). Through experimental evaluation it becomes evident that a decently working syllable segmentation algorithm allowing for accurate syllable border estimations is essential to the correct computation of acoustic distances by the similarity measures developed in this work. Further approaches for similarity measures which are motivated by their usage in timbre classification of music pieces do not show adequate syllable discrimination abilities

Particle Filter Design Using Importance Sampling for Acoustic Source Localisation and Tracking in Reverberant Environments

Sequential Monte Carlo methods have been recently proposed to deal with the problem of acoustic source localisation and tracking using an array of microphones. Previous implementations make use of the basic bootstrap particle filter, whereas a more general approach involves the concept of importance sampling. In this paper, we develop a new particle filter for acoustic source localisation using importance sampling, and compare its tracking ability with that of a bootstrap algorithm proposed previously in the literature. Experimental results obtained with simulated reverberant samples and real audio recordings demonstrate that the new algorithm is more suitable for practical applications due to its reinitialisation capabilities, despite showing a slightly lower average tracking accuracy. A real-time implementation of the algorithm also shows that the proposed particle filter can reliably track a person talking in real reverberant rooms.This paper was performed while Eric A. Lehmann was working with National ICT Australia. National ICT Australia is funded by the Australian Government’s Department of Communications, Information Technology, and the Arts, the Australian Research Council, through Backing Australia’s Ability, and the ICT Centre of Excellence programs

Soft margin estimation for automatic speech recognition

In this study, a new discriminative learning framework, called soft margin estimation (SME), is proposed for estimating the parameters of continuous density hidden Markov models (HMMs). The proposed method makes direct use of the successful ideas of margin in support vector machines to improve generalization capability and decision feedback learning in discriminative training to enhance model separation in classifier design. SME directly maximizes the separation of competing models to enhance the testing samples to approach a correct decision if the deviation from training samples is within a safe margin. Frame and utterance selections are integrated into a unified framework to select the training utterances and frames critical for discriminating competing models. SME offers a flexible and rigorous framework to facilitate the incorporation of new margin-based optimization criteria into HMMs training. The choice of various loss functions is illustrated and different kinds of separation measures are defined under a unified SME framework. SME is also shown to be able to jointly optimize feature extraction and HMMs. Both the generalized probabilistic descent algorithm and the Extended Baum-Welch algorithm are applied to solve SME. SME has demonstrated its great advantage over other discriminative training methods in several speech recognition tasks. Tested on the TIDIGITS digit recognition task, the proposed SME approach achieves a string accuracy of 99.61%, the best result ever reported in literature. On the 5k-word Wall Street Journal task, SME reduced the word error rate (WER) from 5.06% of MLE models to 3.81%, with relative 25% WER reduction. This is the first attempt to show the effectiveness of margin-based acoustic modeling for large vocabulary continuous speech recognition in a HMMs framework. The generalization of SME was also well demonstrated on the Aurora 2 robust speech recognition task, with around 30% relative WER reduction from the clean-trained baseline.Ph.D.Committee Chair: Dr. Chin-Hui Lee; Committee Member: Dr. Anthony Joseph Yezzi; Committee Member: Dr. Biing-Hwang (Fred) Juang; Committee Member: Dr. Mark Clements; Committee Member: Dr. Ming Yua

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

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

Dealing with linguistic mismatches for automatic speech recognition

Recent breakthroughs in automatic speech recognition (ASR) have resulted in a word error rate (WER) on par with human transcribers on the English Switchboard benchmark. However, dealing with linguistic mismatches between the training and testing data is still a significant challenge that remains unsolved. Under the monolingual environment, it is well-known that the performance of ASR systems degrades significantly when presented with the speech from speakers with different accents, dialects, and speaking styles than those encountered during system training. Under the multi-lingual environment, ASR systems trained on a source language achieve even worse performance when tested on another target language because of mismatches in terms of the number of phonemes, lexical ambiguity, and power of phonotactic constraints provided by phone-level n-grams. In order to address the issues of linguistic mismatches for current ASR systems, my dissertation investigates both knowledge-gnostic and knowledge-agnostic solutions. In the first part, classic theories relevant to acoustics and articulatory phonetics that present capability of being transferred across a dialect continuum from local dialects to another standardized language are re-visited. Experiments demonstrate the potentials that acoustic correlates in the vicinity of landmarks could help to build a bridge for dealing with mismatches across difference local or global varieties in a dialect continuum. In the second part, we design an end-to-end acoustic modeling approach based on connectionist temporal classification loss and propose to link the training of acoustics and accent altogether in a manner similar to the learning process in human speech perception. This joint model not only performed well on ASR with multiple accents but also boosted accuracies of accent identification task in comparison to separately-trained models

The direction of technical change in AI and the trajectory effects of government funding

Government funding of innovation can have a significant impact not only on the rate of technical change, but also on its direction. In this paper, we examine the role that government grants and government departments played in the development of artificial intelligence (AI), an emergent general purpose technology with the potential to revolutionize many aspects of the economy and society. We analyze all AI patents filed at the US Patent and Trademark Office and develop network measures that capture each patent’s influence on all possible sequences of follow-on innovation. By identifying the effect of patents on technological trajectories, we are able to account for the long-term cumulative impact of new knowledge that is not captured by standard patent citation measures. We show that patents funded by government grants, but above all patents filed by federal agencies and state departments, profoundly influenced the development of AI. These long-term effects were especially significant in early phases, and weakened over time as private incentives took over. These results are robust to alternative specifications and controlling for endogeneity

Measuring context dependency in birdsong using artificial neural networks

Context dependency is a key feature in sequential structures of human language, which requires reference between words far apart in the produced sequence. Assessing how long the past context has an effect on the current status provides crucial information to understand the mechanism for complex sequential behaviors. Birdsongs serve as a representative model for studying the context dependency in sequential signals produced by non-human animals, while previous reports were upper-bounded by methodological limitations. Here, we newly estimated the context dependency in birdsongs in a more scalable way using a modern neural-network-based language model whose accessible context length is sufficiently long. The detected context dependency was beyond the order of traditional Markovian models of birdsong, but was consistent with previous experimental investigations. We also studied the relation between the assumed/auto-detected vocabulary size of birdsong (i.e., fine- vs. coarse-grained syllable classifications) and the context dependency. It turned out that the larger vocabulary (or the more fine-grained classification) is assumed, the shorter context dependency is detected