A Chinese word dividing algorithm based on statistical language models

Japanese large-vocabulary continuous-speech recognition using a newspaper corpus and broadcast news

Code-switching is a commonly occurring phenomenon in many multilingual communities, wherein a speaker switches between languages within a single utterance. Conventional Word Error Rate (WER) is not sufficient for measuring the performance of code-mixed languages due to ambiguities in transcription, misspellings and borrowing of words from two different writing systems. These rendering errors artificially inflate the WER of an Automated Speech Recognition (ASR) system and complicate its evaluation. Furthermore, these errors make it harder to accurately evaluate modeling errors originating from code-switched language and acoustic models. In this work, we propose the use of a new metric, transliteration-optimized Word Error Rate (toWER) that smoothes out many of these irregularities by mapping all text to one writing system and demonstrate a correlation with the amount of code-switching present in a language. We also present a novel approach to acoustic and language modeling for bilingual code-switched Indic languages using the same transliteration approach to normalize the data for three types of language models, namely, a conventional n-gram language model, a maximum entropy based language model and a Long Short Term Memory (LSTM) language model, and a state-of-the-art Connectionist Temporal Classification (CTC) acoustic model. We demonstrate the robustness of the proposed approach on several Indic languages from Google Voice Search traffic with significant gains in ASR performance up to 10% relative over the state-of-the-art baseline.

This paper presents two different approaches utilizing sta-tisticallanguagemodel(SLM)andsupportvectormachines(SVM) for sentence boundary detection of spontaneousJapanese. In the SLM-based approach, linguistic likeli-hoods and occurrence of pause are used to determine sen-tence boundaries. To suppress false alarms, heuristic pat-terns of end-of-sentence expressions are also incorporated.On the other hand, SVM is adopted to realize robust classi-fication against a wide variety of expressions and speechrecognition errors. Detection is performed by an SVM-based text chunker using lexical and pause information asfeatures. We evaluated these approaches on manual and au-tomatic transcription of spontaneous lectures and speeches,and achieved F-measures of 0.85 and 0.78, respectively. Index Terms : sentence boundary detection, spontaneousspeech, statistical language model, support vector ma-chines. 1. Introduction Recent advance of automatic speech recognition (ASR)technology, especially for spontaneous speech, enables var-ious applications such as spoken document archiving andretrieval, speech summarization and speech translation. Toorganizeaspokendocumentinastructuredformandtogiveuseful indices, transcriptions should be segmented into ap-propriate units like sentences. Moreover, these applicationsare usually built by combining an ASR system with naturallanguage processing (NLP) systems such as a parser and amachine translator, which often assume that input text is asentence. However,sentencesinspontaneousspeechareill-formed, and sentence boundaries are indistinct. Output textby ASR systems is just a sequence of words and has no ex-plicitsentenceboundaries, sothefurtherstepofsegmentingthe ASR output is required for these applications.Automatic boundary detection of spoken sentences hasbeen explored mainly on broadcast news (BN)tasks[1,2,3]and conversational telephone speech (CTS) tasks[3, 4, 5] inEnglish. As features for detection, pause, prosodic and lin-guistic information is often used. Most popular approach isa combination of prosodic and linguistic information[2, 3],which realizes high performance on BN and CTS tasks.Prosody-based approaches[1, 5] have also been investi-gated. Meanwhile, linguistic information is not used by it-self, since most of these works were performed on Englishdata, where cue words or expressions of sentence bound-aries are not easily defined.In Japanese, cue expressions are typically observed atthe end of sentences and expected to be useful for bound-ary detection. However, variety of such expressions is solarge in spoken Japanese, that it is hard to collect sufficientamountofdatafortrainingstatisticalmodelssuchasamax-imum entropy (ME) model. Moreover, many of cue expres-sions consist of particles, which are apparently difficult tobedetectedinASR.Thus,robustnessforASRerrorsshouldalso be investigated.In this paper, we address two approaches of sen-tence boundary detection for spontaneous Japanese. Asframeworks of detection, we adopt and compare statisti-cal language model (SLM) and support vector machines(SVM). The proposed approaches are evaluated with reallectures and speeches included in the Corpus of Sponta-neous Japanese (CSJ).

In large vocabulary continuous speech recognition (LVCSR) for agglutinative and inflectional languages, we encounter problems due to theoretically infinite full-word lexicon size. Sub-word lexicon units may be utilized to dramatically reduce the out-of-vocabulary rate in test data. One can develop language models based on sub-word units to perform LVCSR. However, it has not always been beneficial to use sub-word lexicon units, since shorter units have higher acoustic confusability among them and language model history is effectively shorter as compared to the history in full-word language models. To reduce the aforementioned problems, we propose using the longest possible sub-word units in our lexicon, namely half-words and full-words only. We also incorporate linguistic rules of half-word combination into our statistical language model. The language constraints are represented with a rule-based WFSM which can be combined with an N-gram language model to yield a better and smaller language model. We study the performance of the proposed system for Turkish LVCSR, when the language constraint takes the form of enforcing vowel harmony between stems and endings. We also introduce novel error-rate metrics that are more appropriate than word-error-rate for agglutinative languages. Using half-words with a bi-gram model yields a significant reduction in word-error-rate as compared to a bi-gram full-word model. In addition, combining a tri-gram half-word language model with the vowel-harmony WFSM improves the accuracy further when rescoring the bi-gram lattices.

Nowadays automatic speech recognition (ASR) is an important task for machines. Several applications such as speech translation, virtual assistants and voice bot systems use ASR to understand human speech. Most of the research and available models are for widely used languages, such as English, German, French, Chinese and Spanish. This paper presents the Armenian speech recognition system. As a result of this research developed acoustic and language models for the Armenian language (modern ASR systems combine acoustic and language models to achieve higher accuracy). RNN-based Baidu’s Deep Speech deep neural network was used to train the acoustic model, and the KenLM toolkit was used to train the probabilistic language model. The acoustic model was trained and validated on ArmSpeech Armenian native speech corpus using transfer-learning and data augmentation techniques and tested on the Common Voice Armenian database. The language model was built based on the texts scraped from Armenian news websites. Final models are small in size and can be run and do real-time speech-to-text tasks on IoT devices. Testing on the Common Voice Armenian database the model gave 0.902565 WER and 0.305321 CER without the language model, and 0.552975 WER and 0.285904 CER with the language model. The paper aims to describe environment setup, data collection, acoustic and language models training processes, as well as final results and benchmarks.

In this paper, we describe the work developed in the creation of syllable-based language model for continuous speech recognition system for Myanmar language. Speech recognition systems contain language model as the part of prediction word order sequence. In English and other languages, speech recognition system can use word-based language model. Since Myanmar is monosyllabic and syllable-timed language, the syllable-based language model is more suitable in speech recognition system. At the first step, this paper explains the structure of traditional (Phrase based) language model and describes how to normalized Myanmar sentences and building of normalized Myanmar language model. In the second step, the evaluation of the two language models is expressed. Finally, the comparison of these two language models with the perplexity values is described. The syllable-based Myanmar language model has lower perplexity value than the traditional language model. This language model can support the creation of word-based Myanmar language model.

We present speech experiments that were carried out to evaluate a topically focusing language model in large vocabulary speech recognition. An ordered topical clustering is first computed as a self-organized mapping of a large document collection. Language models are then trained for each text cluster or for several neighboring clusters. The obtained organized collection of language models is efficiently utilized in continuous speech recognition to concentrate on the model that corresponds closest to the current topic of discussion. The speech recognition experiments are carried out on a novel Finnish speech database. A property of Finnish that is particularly challenging for speech recognition is the extremely fast vocabulary growth that makes many of the standard word-based language modeling methods impractical for large vocabulary tasks.

Statistical language models (LM) play a key role in Automatic Speech Recognition (ASR) systems used by conversational agents. These ASR systems should provide a high accuracy under a variety of speaking styles, domains, vocabulary and argots. In this paper, we present a DNN-based method to adapt the LM to each user-agent interaction based on generalized contextual information, by predicting an optimal, context-dependent set of LM interpolation weights. We show that this framework for contextual adaptation provides accuracy improvements under different possible mixture LM partitions that are relevant for both (1) Goal-oriented conversational agents where it's natural to partition the data by the requested application and for (2) Non-goal oriented conversational agents where the data can be partitioned using topic labels that come from predictions of a topic classifier. We obtain a relative WER improvement of 3% with a 1-pass decoding strategy and 6% in a 2-pass decoding framework, over an unadapted model. We also show up to a 15% relative improvement in recognizing named entities which is of significant value for conversational ASR systems.

An efficient, scalable speech recognition architecture is proposed for multidomain dialog systems by combining topic detection and topic-dependent language modeling. The inferred domain is automatically detected from the user's utterance, and speech recognition is then performed with an appropriate domain-dependent language model. The architecture improves accuracy and efficiency over current approaches and is scaleable to a large number of domains. In this paper, a novel framework using a multilayer hierarchy of language models is introduced in order to improve robustness against topic detection errors. The proposed system provides a relative reduction in WER of 10.5% over a single language model system. Furthermore it achieves an accuracy that is comparable to using multiple language models in parallel while using only a fraction of the computational cost.

The performance of the statistical context-sensitive spelling error correction depends on the quality and quantity of the data for statistical language model. In general, the size and quality of data in a statistical language model are proportional. However, as the amount of data increases, the processing speed becomes slower and storage space also takes up a lot. We suggest the improved statistical language model to solve this problem. And we propose an effective spelling error candidate generation method based on a new statistical language model. The proposed statistical model and the correction method based on it improve the performance of the spelling error correction and processing speed.

Statistical language modeling (LM) that purports to quantify the acceptability of a given piece of text has long been an interesting yet challenging research area. In particular, language modeling for information retrieval (IR) has enjoyed remarkable empirical success; one emerging stream of the LM approach for IR is to employ the pseudo-relevance feedback process to enhance the representation of an input query so as to improve retrieval effectiveness. This paper presents a continuation of such a general line of research and the main contribution is threefold. First, we propose a principled framework which can unify the relationships among several widely-used query modeling formulations. Second, on top of the successfully developed framework, we propose an extended query modeling formulation by incorporating critical query-specific information cues to guide the model estimation. Third, we further adopt and formalize such a framework to the speech recognition and summarization tasks. A series of empirical experiments reveal the feasibility of such an LM framework and the performance merits of the deduced models on these two tasks.

In statistical language models, how to integrate diverse linguistic knowledge in a general framework for long-distance dependencies is a challenging issue. In this paper, an improved language model incorporating linguistic structure into maximum entropy framework is presented. The proposed model combines trigram with the structure knowledge of base phrase in which trigram is used to capture the local relation between words, while the structure knowledge of base phrase is considered to represent the long-distance relations between syntactical structures. The knowledge of syntax, semantics and vocabulary is integrated into the maximum entropy framework. Experimental results show that the proposed model improves by 24% for language model perplexity and increases about 3% for sign language recognition rate compared with the trigram model.

In this paper, a generic Symbol Input Correction Method for Human-Machine Interfaces, especially useful for embedded devices where the input subsystem is often size-constrained, combining a Language Model, the Input Hypothesis information and an Error Model is proposed. The approach can be seen as a flexible and efficient way to perform Stochastic Error-Correcting Language Modeling. We use Weighted Finite-State Transducers (WFSTs) to represent the Language Model, the complete set of symbol-input Hypotheses interpreted as a sequence of vectors of symbol probabilities, and an Error Model. This approach is different to other methods since it does not involve an explicit parsing process and also because it combines the practical advantages of a de-coupled (input-system + post-processor) model with the error-recovery power of a integrated approach where the capture and the interpretation are performed in the same element (e.g. in a HMM). The symbol-input subsystem can be a physical, "soft" (touch screen-based) or reduced (as in a mobile phone) keyboard, a speech or gesture-based recognizer, an Off-line or On-line OCR system or any other Human-Machine Interface consisting of a sequence of symbols conveying information belonging to a language where the segmentation of the input is known (although some segmentation errors can be recovered by the Error Model).

End-to-end speech recognition is a promising technology for enabling compact automatic speech recognition (ASR) systems since it can unify the acoustic and language model into a single neural network. However, as a drawback, training of end-to-end speech recognizers always requires transcribed utterances. Since end-to-end models are also known to be severely data hungry, this constraint is crucial especially because obtaining transcribed utterances is costly and can possibly be impractical or impossible. This paper proposes a method for alleviating this issue by transferring knowledge from a language model neural network that can be pretrained with text-only data. Specifically, this paper attempts to transfer semantic knowledge acquired in embedding vectors of large-scale language models. Since embedding vectors can be assumed as implicit representations of linguistic information such as part-of-speech, intent, and so on, those are also expected to be useful modeling cues for ASR decoders. This paper extends two types of ASR decoders, attention-based decoders and neural transducers, by modifying training loss functions to include embedding prediction terms. The proposed systems were shown to be effective for error rate reduction without incurring extra computational costs in the decoding phase.

An obvious problem with automatic speech recognition (ASR) for Taigi is that the amount of training data is far from enough to build a practical ASR system. Collecting speech data with reliable transcripts for training the acoustic model (AM) is feasible but expensive. Moreover, text data used for language model (LM) training is extremely scarce and difficult to collect because Taigi is a spoken language, not a commonly used written language. Interestingly, the subtitles of Taigi drama in Taiwan have long been in Chinese characters for Mandarin. Since a large amount of Taigi drama episodes with Mandarin Chinese subtitles are available on YouTube, we propose a method to augment the training data for AM and LM of Taigi ASR. The idea is to use an initial Taigi ASR system to convert a Mandarin Chinese subtitle into the most likely Taigi word sequence by referring to the speech. Experimental results show that our ASR system can be remarkably improved by such training data augmentation.