• Journal of the Korean Institute of Intelligent Systems
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• v.14 no.2
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• pp.201-206
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• 2004

This study attempted to develop a speech recognition module applied to a wheelchair for the physically handicapped. In the proposed speech recognition module, TMS320C32 was used as a main processor and Mel-Cepstrum 12 Order was applied to the pro-processor step to increase the recognition rate in a noisy environment. DTW (Dynamic Time Warping) was used and proven to be excellent output for the speaker-dependent recognition part. In order to utilize this algorithm more effectively, the reference data was compressed to 1/12 using vector quantization so as to decrease memory. In this paper, the necessary diverse technology (End-point detection, DMA processing, etc.) was managed so as to utilize the speech recognition system in real time

• Proceedings of the Acoustical Society of Korea Conference
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• 1991.06a
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• pp.7-10
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• 1991

In a text-to-speech system, sound units (phonemes, words, or phrases, etc.) can be concatenated together to produce required utterance. The quality of the resulting speech is dependent on factors including the phonological/prosodic contour, the quality of basic concatenation units, and how well the units join together. Thus although the quality of each basic sound unit is high, if occur the discontinuity in the concatenation part then the quality of synthesis speech is decrease. To solve this problem, a smoothing operation should be carried out in concatenation parts. But a major problem is that, as yet, no method of parameter smoothing is available for joining the segment together. Thus in this paper, we proposed a new aigorithm that smoothing the unnatural discountinuous parts which can be occured in speech waveform editing. This algorithm used the exponential smoothing method.

• Speech Sciences
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• v.10 no.4
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• pp.137-147
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• 2003

Speech synthesis can classify by synthesis way, that is waveform coding, source coding and mixture coding. Specially, waveform coding is suitable for high quality synthesis. However, it is not desirable by synthesis techniques of syllable or phoneme unit because it do not separate and handles excitation and formant part. Therefore, there is a need for pitch alteration method applied in synthesis by the rule in waveform coding. This study propose about pitch alteration method that use spectrum scaling after do to flatten spectra by subband linear approximation to minimize spectrum distortion. This paper show evaluation whether show excellency of some measure compared with LPC, Cepstrum, lifter function and method that propose. estimation method seeks distribution of each flattened signal and measured degree of flattened spectra Signal flattened is normalized, So that highest point amounts to zero, and distribution of signal ,whose average is zero, is calculated. this show result that measure the spectrum distortion rate to estimate performance of method that propose. The average spectrum distortion rate was kept below the average 2.12%, so the method that propose is superiors than existent method.

• The Journal of the Acoustical Society of Korea
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• v.15 no.4
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• pp.98-104
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• 1996

This paper propose the alteraton rule to compensate a formant trasition of several connected vowels for improving an unnatural synthesized continuous speech which is concatenated by each demisyllable without coarticulated formant transition for use in dmisyllable based synthesis by rule. To fullfill each formant transition part, the database of 42 stationary vowels which are segmented from the stable part of each vowels is appended to the one of Korean demisyllables, and the resonance circuit used in formant synthesis is employed to change the formant frequency of speech signals. To evaluate the synthesied speech by this rule, we carried out the alteration rule for connected vowels of the synthesized speech based on demisyllable, and compare spectrogram and MOS tested scores with the original and the demisyllable based synthesized speech without this rule. The result shows that this proposed rule can synthesize the more natural speech.

• The Journal of the Korea Contents Association
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• v.9 no.1
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• pp.139-146
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• 2009

This paper addresses a modified spectral subtraction schemes which is suitable to speech recognition under low signal-to-noise ratio (SNR) noisy environment such as the automatic speech recognition (ASR) system in car. The conventional spectral subtraction schemes rely on the SNR such that attenuation is imposed on that part of the spectrum that appears to have low SNR, and accentuation is made on that part of high SNR. However, such postulation is adequate for high SNR environment, it is grossly inadequate for low SNR scenarios such as that of car environment. Proposed methods focused specifically to low SNR noisy environment by using weighting function for enhancing speech dominant region in speech spectrum. Experimental results by using voice commands for car show the superior performance of the proposed method over conventional methods.

• Proceedings of the IEEK Conference
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• 1998.10a
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• pp.527-530
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• 1998

In this paper, a transformation-based English part of speech tagging system is designed and implemented. The tagging system tags raw corpus at first and the transformation rule correct the errors. Apart from traditional rule based tagging system, this system makes rules automatically. Using 60,000 words of corpus as a training corpus, the transformation rules are generated automatically by iterative training. The idea how to calculate positive effect of transformation and select transformation rules is proposed to generate more effective and correct transformations. In this paper, part of the Brown corpus and English text is used for experimental data. And the performance of transformation based tagging system is demonstrated by the calculation of accuracy.

• The KIPS Transactions:PartB
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• v.10B no.6
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• pp.665-672
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• 2003

This paper describes a speech synthesis technique by concatenating unit phoneme. At that time, a major problem is that discontinuity is happened from connection part between unit phonemes, especially from connection part between unit phonemes recorded by different persons. To solve the problem, this paper uses clustered diphone, and proposes a spectral smoothing technique, not only using formant trajectory and distribution characteristic of spectrum but also reflecting human's acoustic characteristic. That is, the proposed technique performs unit phoneme clustering using distribution characteristic of spectrum at connection part between unit phonemes and decides a quantity and a scope for the smoothing by considering human's acoustic characteristic at the connection part of unit phonemes, and then performs the spectral smoothing using weights calculated along a time axes at the border of two diphones. The proposed technique removes the discontinuity and minimizes the distortion which can be occurred by spectrum smoothing. For the purpose of the performance evaluation, we test on five hundred diphones which are extracted from twenty sentences recorded by five persons, and show the experimental results.

• Cross-Cultural Studies
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• v.42
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• pp.371-392
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• 2016

This study analyzed the aspect of the decision of the Korean part of speech and the properties of the grammatical items based on "韓語通" which was published in 1909. "韓語通" is a Korean grammar book written by 前間恭作 who also published "校訂交隣須知" in 1904. "韓語通" is known for influencing of 'Otsuki grmmar(大槻 文法),' dividing Korean part of speech into eleven. Based on 'mood' and 'voice' we can assume that "韓語通" adopted Otsuki's grammar. '存在詞' is another clue that "韓語通" adopted Yamada's grammar. However, 前間恭作 persisted that Korean language is different from Japanese language. This view is different from 寶迫繁勝, 高橋亨, 藥師寺知? etc. This study tried to investigate the interchange of the two languages in historical study of Korean and Japanese linguistics during modern and contemporary period. For this purpose, we searched the aspect of the part of speech and analyzed the grammar items. In conclusion, we was able to light on how Japanese scholars approached to Korean grammar system in late 19th and early 20th centuries.

• Korean Journal of Cognitive Science
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• v.7 no.2
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• pp.57-73
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• 1996

• Proceedings of the Korean Society for Cognitive Science Conference
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• 2000.01a
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• pp.376-380
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• 2000