• MALSORI
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• no.52
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• pp.111-120
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• 2004

This paper focuses on the DSP implementation of an HMM-based speech recognizer that can handle several hundred words of vocabulary size as well as speaker independency. First, we develop an HMM-based speech recognition system on the PC that operates on the frame basis with parallel processing of feature extraction and Viterbi decoding to make the processing delay as small as possible. Many techniques such as linear discriminant analysis, state-based Gaussian selection, and phonetic tied mixture model are employed for reduction of computational burden and memory size. The system is then properly optimized and compiled on the TMS320C6711 DSP for real-time operation. The implemented system uses 486kbytes of memory for data and acoustic models, and 24.5 kbytes for program code. Maximum required time of 29.2 ms for processing a frame of 32 ms of speech validates real-time operation of the implemented system.

• The Journal of the Acoustical Society of Korea
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• v.16 no.1E
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• pp.57-63
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• 1997

In this paper, we investigated performance of a vocabulary-independent speech recognizer with speaker adaptation. The vocabulary-independent speech recognizer does not require task-oriented speech databases to estimate HMM parameters, but adapts the parameters recursively by using input speech and recognition results. The recognizer has the advantage that it relieves efforts to record the speech databases and can be easily adapted to a new task and a new speaker with different recognition vocabulary without losing recognition accuracies. Experimental results showed that the vocabulary-independent speech recognizer with supervised offline speaker adaptation reduced 40% of recognition errors when 80 words from the same vocabulary as test data were used as adaptation data. The recognizer with unsupervised online speaker adaptation reduced abut 43% of recognition errors. This performance is comparable to that of a speaker-independent speech recognizer trained by a task-oriented speech database.

• Proceedings of the KOSOMBE Conference
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• v.1992 no.05
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• pp.218-218
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• 1992

• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.6
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• pp.1443-1453
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• 1998

This paper presents the speech synthesis engine that converts the character strings kept in a computer memory into the synthesized speech sounds with enhancing the intelligibility and the naturalness by adapting the waveform processing method. The speech engine using demisyllable speech segments receives command streams for pitch modification, duration and energy control. The command based engine isolates the high level processing of text normalization, letter-to-sound and the lexical analysis and the low level processing of signal filtering and pitch processing. The TTS(Text-to-Speech) system implemented by using the speech synthesis engine has three independent object modules of the Text-Normalizer, the Commander and the said Speech Synthesis Engine those of which are easily replaced by other compatible modules. The architecture separating the high level and the low level processing has the advantage of the expandibility and the portability because of the mix-and-match nature.

• MALSORI
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• no.57
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• pp.99-112
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• 2006

In this paper, we introduce two retrieval methods for photos with speech documents. We compare the pattern of speech query with those of speech documents recorded in digital cameras, and measure the similarities, and retrieve photos corresponding to the speech documents which have high similarity scores. As the first approach, a phoneme recognition scheme is used as the pre-processor for the pattern matching, and in the second one, the vector quantization (VQ) and the dynamic time warping (DTW) are applied to match the speech query with the documents in signal domain itself. Experimental results show that the performance of the first approach is highly dependent on that of phoneme recognition while the processing time is short. The second method provides a great improvement of performance. While the processing time is longer than that of the first method due to DTW, but we can reduce it by taking approximated methods.

• Phonetics and Speech Sciences
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• v.2 no.1
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• pp.77-85
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• 2010

This paper proposes an efficient feature vector processing technique to guard the Speech Emotion Recognition (SER) system against a variety of noises. In the proposed approach, emotional feature vectors are extracted from speech processed by comb filtering. Then, these extracts are used in a robust model construction based on feature vector classification. We modify conventional comb filtering by using speech presence probability to minimize drawbacks due to incorrect pitch estimation under background noise conditions. The modified comb filtering can correctly enhance the harmonics, which is an important factor used in SER. Feature vector classification technique categorizes feature vectors into either discriminative vectors or non-discriminative vectors based on a log-likelihood criterion. This method can successfully select the discriminative vectors while preserving correct emotional characteristics. Thus, robust emotion models can be constructed by only using such discriminative vectors. On SER experiment using an emotional speech corpus contaminated by various noises, our approach exhibited superior performance to the baseline system.

• The Transactions of the Korea Information Processing Society
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• v.4 no.6
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• pp.1531-1540
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• 1997

In this paper, we presents automatic Korean continuous speech recognizer which is improved by the speech rate estimation and the compensation methods. Automatic continuous speech recognition is significantly more difficult than isolated word recognition because of coarticulatory effects and variations in speech rate. In order to recognize continuous speech, modeling methods of coarticulatory effects and variations in speech rate are needed. In this paper, the speech rate is measured by change of format, and the compensation is peformed by extracting relatively many feature vectors in fast speech. Coarticulatory effects are modeled by defining 514 Korean diphone set, and ETRI's 445 word DB is used for training speech material. With combining above methods, we implement automatic Korean continuous speech recognizer, which shows improved recognition rate, based on DHMM(Discrete Hidden Markov Model).

• Journal of Korea Society of Industrial Information Systems
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• v.7 no.3
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• pp.74-80
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• 2002

There are many researches on post-processing methods for the Korean continuous speech recognition enhancement using natural language processing techniques. It is very difficult to use a formal morphological analyzer for improving the speech recognition because the analysis technique of natural language processing is mainly for formal written languages. In this paper, we propose a speech recognition enhancement model using syntactic unit of morphemes. This approach uses the functional word level longest match which dose not consider spacing words. We describe the post-processing mechanism for the improving speech recognition by using proposed model which uses the relationship of phonological structure information between predicates md auxiliary predicates or bound nouns that are frequently occurred in Korean sentences.

• Phonetics and Speech Sciences
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• v.9 no.3
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• pp.57-65
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• 2017

The current study explored how Korean-speaking children processed the multiple acoustic cues (VOT and f0) for the stop laryngeal contrast (/t'/, /t/, and /$t^h$/) and examined whether individual perceptual strategies could be related to a general cognitive ability performing executive functions (EF). 15 children (aged from 7 to 8) participated in the speech perception task identifying the three Korean laryngeal stops (3AFC) on listening to the auditory stimuli of C-/a/ with synthetically varying VOT and f0. They completed a series of EF tasks to measure working memory, inhibition, and cognitive shifting ability. The findings showed that children used the two cues in a highly correlated manner. While children utilized VOT consistently for the three laryngeal categories, their use of f0 was either reduced or enhanced depending on the phonetic categories. Importantly, the children's processing strategies of a f0 suppression for a tense-aspirated contrast were meaningfully associated with children's better cognitive abilities such as working memory, inhibition, and attentional shifting. As a preliminary experimental investigation, the current research demonstrated that listeners with inefficient processing strategies were poor at the EF skills, suggesting that cognitive skills might be responsible for developmental variations of processing sub-phonemic information for the linguistic contrast.

• MALSORI
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• no.50
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• pp.41-50
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• 2004

Recently, there has been considerable progress in the natural language processing and digital signal processing components and this progress has led to the improved synthetic speech qualify of many commercial TTS systems. But there still remain many obstacles to overcome for the practical application of TTS. To resolve the problems, the cooperative research among the related areas is highly required and a common Korean TTS platform is essential to promote these activities. This platform offers a general framework for building Korean speech synthesis systems and a full C/C++ source for modules supports to implement and test his own algorithm. In this paper we described the aspect of a Korean TTS platform to be developed and a developing plan.