Segmentation, Diarization and Speech Transcription: Surprise Data Unraveled

In this thesis, research on large vocabulary continuous speech recognition for unknown audio conditions is presented. For automatic speech recognition systems based on statistical methods, it is important that the conditions of the audio used for training the statistical models match the conditions of the audio to be processed. Any mismatch will decrease the accuracy of the recognition. If it is unpredictable what kind of data can be expected, or in other words if the conditions of the audio to be processed are unknown, it is impossible to tune the models. If the material consists of `surprise data' the output of the system is likely to be poor. In this thesis methods are presented for which no external training data is required for training models. These novel methods have been implemented in a large vocabulary continuous speech recognition system called SHoUT. This system consists of three subsystems: speech/non-speech classification, speaker diarization and automatic speech recognition. The speech/non-speech classification subsystem separates speech from silence and unknown audible non-speech events. The type of non-speech present in audio recordings can vary from paper shuffling in recordings of meetings to sound effects in television shows. Because it is unknown what type of non-speech needs to be detected, it is not possible to train high quality statistical models for each type of non-speech sound. The speech/non-speech classification subsystem, also called the speech activity detection subsystem, does not attempt to classify all audible non-speech in a single run. Instead, first a bootstrap speech/silence classification is obtained using a standard speech activity component. Next, the models for speech, silence and audible non-speech are trained on the target audio using the bootstrap classification. This approach makes it possible to classify speech and non-speech with high accuracy, without the need to know what kinds of sound are present in the audio recording. Once all non-speech is filtered out of the audio, it is the task of the speaker diarization subsystem to determine how many speakers occur in the recording and exactly when they are speaking. The speaker diarization subsystem applies agglomerative clustering to create clusters of speech fragments for each speaker in the recording. First, statistical speaker models are created on random chunks of the recording and by iteratively realigning the data, retraining the models and merging models that represent the same speaker, accurate speaker models are obtained for speaker clustering. This method does not require any statistical models developed on a training set, which makes the diarization subsystem insensitive for variation in audio conditions. Unfortunately, because the algorithm is of complexity $O(n^3)$, this clustering method is slow for long recordings. Two variations of the subsystem are presented that reduce the needed computational effort, so that the subsystem is applicable for long audio recordings as well. The automatic speech recognition subsystem developed for this research, is based on Viterbi decoding on a fixed pronunciation prefix tree. Using the fixed tree, a flexible modular decoder could be developed, but it was not straightforward to apply full language model look-ahead efficiently. In this thesis a novel method is discussed that makes it possible to apply language model look-ahead effectively on the fixed tree. Also, to obtain higher speech recognition accuracy on audio with unknown acoustical conditions, a selection from the numerous known methods that exist for robust automatic speech recognition is applied and evaluated in this thesis. The three individual subsystems as well as the entire system have been successfully evaluated on three international benchmarks. The diarization subsystem has been evaluated at the NIST RT06s benchmark and the speech activity detection subsystem has been tested at RT07s. The entire system was evaluated at N-Best, the first automatic speech recognition benchmark for Dutch

Towards multi-domain speech understanding with flexible and dynamic vocabulary

Thesis (Ph.D.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 2001.Includes bibliographical references (p. 201-208).In developing telephone-based conversational systems, we foresee future systems capable of supporting multiple domains and flexible vocabulary. Users can pursue several topics of interest within a single telephone call, and the system is able to switch transparently among domains within a single dialog. This system is able to detect the presence of any out-of-vocabulary (OOV) words, and automatically hypothesizes each of their pronunciation, spelling and meaning. These can be confirmed with the user and the new words are subsequently incorporated into the recognizer lexicon for future use. This thesis will describe our work towards realizing such a vision, using a multi-stage architecture. Our work is focused on organizing the application of linguistic constraints in order to accommodate multiple domain topics and dynamic vocabulary at the spoken input. The philosophy is to exclusively apply below word-level linguistic knowledge at the initial stage. Such knowledge is domain-independent and general to all of the English language. Hence, this is broad enough to support any unknown words that may appear at the input, as well as input from several topic domains. At the same time, the initial pass narrows the search space for the next stage, where domain-specific knowledge that resides at the word-level or above is applied. In the second stage, we envision several parallel recognizers, each with higher order language models tailored specifically to its domain. A final decision algorithm selects a final hypothesis from the set of parallel recognizers.(cont.) Part of our contribution is the development of a novel first stage which attempts to maximize linguistic constraints, using only below word-level information. The goals are to prevent sequences of unknown words from being pruned away prematurely while maintaining performance on in-vocabulary items, as well as reducing the search space for later stages. Our solution coordinates the application of various subword level knowledge sources. The recognizer lexicon is implemented with an inventory of linguistically motivated units called morphs, which are syllables augmented with spelling and word position. This first stage is designed to output a phonetic network so that we are not committed to the initial hypotheses. This adds robustness, as later stages can propose words directly from phones. To maximize performance on the first stage, much of our focus has centered on the integration of a set of hierarchical sublexical models into this first pass. To do this, we utilize the ANGIE framework which supports a trainable context-free grammar, and is designed to acquire subword-level and phonological information statistically. Its models can generalize knowledge about word structure, learned from in-vocabulary data, to previously unseen words. We explore methods for collapsing the ANGIE models into a finite-state transducer (FST) representation which enables these complex models to be efficiently integrated into recognition. The ANGIE-FST needs to encapsulate the hierarchical knowledge of ANGIE and replicate ANGIE's ability to support previously unobserved phonetic sequences ...by Grace Chung.Ph.D

A Text Rewriting Decoder with Application to Machine Translation

Ph.DDOCTOR OF PHILOSOPH

Forgetting Exceptions is Harmful in Language Learning

We show that in language learning, contrary to received wisdom, keeping exceptional training instances in memory can be beneficial for generalization accuracy. We investigate this phenomenon empirically on a selection of benchmark natural language processing tasks: grapheme-to-phoneme conversion, part-of-speech tagging, prepositional-phrase attachment, and base noun phrase chunking. In a first series of experiments we combine memory-based learning with training set editing techniques, in which instances are edited based on their typicality and class prediction strength. Results show that editing exceptional instances (with low typicality or low class prediction strength) tends to harm generalization accuracy. In a second series of experiments we compare memory-based learning and decision-tree learning methods on the same selection of tasks, and find that decision-tree learning often performs worse than memory-based learning. Moreover, the decrease in performance can be linked to the degree of abstraction from exceptions (i.e., pruning or eagerness). We provide explanations for both results in terms of the properties of the natural language processing tasks and the learning algorithms.Comment: 31 pages, 7 figures, 10 tables. uses 11pt, fullname, a4wide tex styles. Pre-print version of article to appear in Machine Learning 11:1-3, Special Issue on Natural Language Learning. Figures on page 22 slightly compressed to avoid page overloa

From seen to unseen: Designing keyboard-less interfaces for text entry on the constrained screen real estate of Augmented Reality headsets

Text input is a very challenging task in the constrained screen real-estate of Augmented Reality headsets. Typical keyboards spread over multiple lines and occupy a significant portion of the screen. In this article, we explore the feasibility of single-line text entry systems for smartglasses. We first design FITE, a dynamic keyboard where the characters are positioned depending on their probability within the current input. However, the dynamic layout leads to mediocre text input and low accuracy. We then introduce HIBEY, a fixed 1-line solution that further decreases the screen real-estate usage by hiding the layout. Despite its hidden layout, HIBEY surprisingly performs much better than FITE, and achieves a mean text entry rate of 9.95 words per minute (WPM) with 96.06% accuracy, which is comparable to other state-of-the-art approaches. After 8 days, participants achieve an average of 13.19 WPM. In addition, HIBEY only occupies 13.14% of the screen real estate at the edge region, which is 62.80% smaller than the default keyboard layout on Microsoft Hololens.Peer reviewe

Development of a stemmer for the isiXhosa language

IsiXhosa language is one of the eleven official languages and the second most widely spoken language in South Africa. However, in terms of computational linguistics, the language did not get attention and natural language related work is almost non-existent. Document retrieval using unstructured queries requires some kind of language processing, and an efficient retrieval of documents can be achieved if we use a technique called stemming. The area that involves document storage and retrieval is called Information Retrieval (IR). Basically, IR systems make use of a Stemmer to index document representations and also terms in users’ queries to retrieve matching documents. In this dissertation, we present the developed Stemmer that can be used in both conditions. The Stemmer is used in IR systems, like Google to retrieve documents written in isiXhosa. In the Eastern Cape Province of South Africa many public schools take isiXhosa as a subject and also a number of Universities in South Africa teach isiXhosa. Therefore, for a language important such as this, it is important to make valuable information that is available online accessible to users through the use of IR systems. In our efforts to develop a Stemmer for the isiXhosa language, an investigation on how others have developed Stemmers for other languages was carried out. From the investigation we came to realize that the Porter stemming algorithm in particular was the main algorithm that many of other Stemmers make use of as a reference. We found that Porter’s algorithm could not be used in its totality in the development of the isiXhosa Stemmer because of the morphological complexity of the language. We developed an affix removal that is embedded with rules that determine which order should be followed in stripping the affixes. The rule is that, the word under consideration is checked against the exceptions, if it’s not in the exceptions list then the stripping continue in the following order; Prefix removal, Suffix removal and finally save the result as stem. The Stemmer was successfully developed and was tested and evaluated in a sample data that was randomly collected from the isiXhosa text books and isiXhosa dictionary. From the results obtained we concluded that the Stemmer can be used in IR systems as it showed 91 percent accuracy. The errors were 9 percent and therefore these results are within the accepted range and therefore the Stemmer can be used to help in retrieval of isiXhosa documents. This is only a noun Stemmer and in the future it can be extended to also stem verbs as well. The Stemmer can also be used in the development of spell-checkers of isiXhosa

Acoustic Modelling for Under-Resourced Languages

Automatic speech recognition systems have so far been developed only for very few languages out of the 4,000-7,000 existing ones. In this thesis we examine methods to rapidly create acoustic models in new, possibly under-resourced languages, in a time and cost effective manner. For this we examine the use of multilingual models, the application of articulatory features across languages, and the automatic discovery of word-like units in unwritten languages

Perceptual learning of context-sensitive phonetic detail

[Abstract abbreviated due to inability of DSpace@Cambridge to display phonetic symbols. Please see the full abstract in the attached pdf file.] Although familiarity with a talker or accent is known to facilitate perception, it is not clear what underlies this phenomenon. Previous research has focused primarily on whether listeners can learn to associate novel phonetic characteristics with low-level units such as features or phonemes. However, this neglects the potential role of phonetic information at many other levels of representation. To address this shortcoming, this thesis investigated perceptual learning of systematic phonetic detail relating to higher levels of linguistic structure, including prosodic, grammatical and morphological contexts. Furthermore, in contrast to many previous studies, this research used relatively natural stimuli and tasks, thus maximising its relevance to perceptual learning in ordinary listening situations. This research shows that listeners can update their phonetic representations in response to incoming information and its relation to linguistic-structural context. In addition, certain patterns of systematic phonetic detail were more learnable than others. These findings are used to inform an account of how new information is integrated with prior experience in speech processing, within a framework that emphasises the importance of phonetic detail at multiple levels of representation.This work was funded by an AHRC grant