• Korean Architects
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• no.3 s.144
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• pp.56-60
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• 1981

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2008.05a
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• pp.763-767
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• 2008

In a speech coding system using excitation source of voiced and unvoiced, it would be involved a distortion of speech qualify in case coexist with a voiced and an unvoiced consonants in a frame. So, We proposed TS(Transition Segment) including unvoiced consonant searching and extraction method in order to uncoexistent with a voiced and unvoiced consonants in a frame. This research present a new method of TS approximate-synthesis by using Least Mean Square and frequency band division. As a result, this method obtain a high quality approximation-synthesis waveforms within TS by using frequency information of 0.547kHz below and 2.813kHz above. The important thing is that the maximum error signal can be made with low distortion approximation-synthesis waveform within TS. This method has the capability of being applied to a new speech coding of Voiced/Silence/TS, speech analysis and speech synthesis.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.38 no.5
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• pp.576-582
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• 2001

The Multi-Band Excitation(MBE) speech coder uses a different approach for the representation of the excitation signal. It replaces the frame-based single voiced/unvoiced classification of a classical speech coder with a set of such decision over harmonic intervals in the frequency domain. This enables each speech segment to be a mixture of voiced and unvoiced, and improves the synthetic speech quality by reducing decision errors that might occur on the frame-based single voiced and unvoiced decision process when input speech is degraded with noise. The IMBE-LP, improved version of MBE with linear prediction, represents the spectral information of MBE model with linear prediction coefficients to obtain low bit rate of 2.4 kbps. In this Paper, we proposed a variable rate IMBE-LP vocoder that has lower bit rate than IMBE-LP without degrading the synthetic speech quality. To determine the LP order, it uses the spectral band information of the MBE model that has something to do with he input speech's characteristics. Experimental results are riven with our findings and discussions.

• Journal of the Korean Institute of Illuminating and Electrical Installation Engineers
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• v.20 no.7
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• pp.58-64
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• 2006

Recently deep interests have been paid on the effective generation of ozone, which has been widely used for water treatment, deodorization, color removal, and chemical processing of exhausted smoke. The silent discharge reaction has been proposed as the most effective one in the many ozone generation methods, because the silent discharge can be generated under the conditions of lower applied voltage and power consumption, compared other ones. In this paper, in order to improve the ozone generation and ozone generation efficiency, the conventional silent discharge chamber with $Al_2O_3$ dielectric layer and tubular ferroelectric bed discharge reactor packed with $Al_2O_3$ pellets were made, and the silent discharge of the reactors were studied experimentally. The ozone generation characteristics are also discussed based on the discharge characteristics, especially on the wall charge accumulation properties and power consumption. The results show that the electric charges of discharge tube with bead are around 2.5 times as large as those without bead. In the discharge chamber packed with dielectric beads, the ozone concentration and the energy yield characteristics were also improved, compared with those in the conventional silent discharge reactor.

• The Transactions of the Korea Information Processing Society
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• v.6 no.1
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• pp.247-252
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• 1999

In order to realize a speech coding at low bit rates, a pitch information is useful parameter. In case of extracting an average pitch information form continuous speech, the several pitch errors appear in a frame which consonant and vowel are coexistent; in the boundary between adjoining frames and beginning or ending of a sentence. In this paper, I propose an Individual Pitch (IP) extraction method using residual signals of the FIR-STREAK digital filter in order to restrict the pitch extraction errors. This method is based on not averaging pitch intervals in order to accomodate the changes in each pitch interval. As a result, in case of Ip extraction method suing FIR-STREAK digital filter, I can't find the pitch errors in a frame which consonant and vowel are consistent; in the boundary between adjoining frames and beginning or ending of a sentence. This method has the capability of being applied to many fields, such as speech coding, speech analysis, speech synthesis and speech recognition.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.13 no.2
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• pp.413-418
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• 2009

In a speech coding system using excitation source of voiced and unvoiced, it would be involved a distortion of speech qualify in case coexist with a voiced and an unvoiced consonants in a frame. So, We proposed TS(Transition Segment) including unvoiced consonant searching and extraction method in order to uncoexistent with a voiced and unvoiced consonants in a frame. This research present a new method of TS approximate-synthesis by using Least Mean Square and frequency band division. As a result, this method obtain a high qualify approximation-synthesis waveforms within TS by using frequency information of 0.547kHz below and 2.813kHz above. The important thing is that the maximum error signal can be made with low distortion approximation-synthesis waveform within TS. This method has the capability of being applied to a new speech coding of Voiced/Silence/TS, speech analysis and speech synthesis.

• Journal of Internet Computing and Services
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• v.2 no.1
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• pp.31-38
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• 2001

Speech database built of phonemes is significant in the studies of speech recognition, speech synthesis and analysis, Phoneme, consist of voiced sounds and unvoiced ones, Though there are many feature differences in voiced and unvoiced sounds, the traditional algorithms for detecting the boundary between phonemes do not reflect on them and determine the boundary between phonemes by comparing parameters of current frame with those of previous frame in time domain, In this paper, we propose the assort algorithm, which is based on a block and reflecting upon the feature differences between voiced and unvoiced sounds for phoneme segmentation, The assort algorithm uses the distance measure based upon MFCC(Mel-Frequency Cepstrum Coefficient) as a comparing spectrum measure, and uses the energy, zero crossing rate, spectral energy ratio, the formant frequency to separate voiced sounds from unvoiced sounds, N, the result of out experiment, the proposed system showed about 79 percents precision subject to the 3 or 4 syllables isolated words, and improved about 8 percents in the precision over the existing phonemes segmentation system.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.18 no.6
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• pp.1294-1301
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• 2014

In this paper, separate probabilistic distribution models for voiced and unvoiced speech are estimated and utilized to improve speaker recognition performance. Also, in addition to the conventional mel-frequency cepstral coefficient, skewness, kurtosis, and harmonic-to-noise ratio are extracted and used for voiced speech intervals. Two kinds of scores for voiced and unvoiced speech are linearly fused with the optimal weight found by exhaustive search. The performance of the proposed speaker recognizer is compared with that of the conventional recognizer which uses mel-frequency cepstral coefficient and a unified probabilistic distribution function based on the Gassian mixture model. Experimental results show that the lower the number of Gaussian mixture, the greater the performance improvement by the proposed algorithm.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.3C
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• pp.239-247
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• 2003

This paper presents a dual-rate (2.4/4.0 kbps) Improved Harmonic-CELP(IHC) speech coder based on the EHC(Efficient Harmonic-CELP) which was presented by the authors. The proposed IHC employs the harmonic coding for voiced and the CELP for unvoiced segments. In the IHC, an initial voiced/unvoiced estimate is obtained by the pitch gain and energy. Then, the final V/UV mode is decided by using the frame energy contour. A new harmonic estimation combining peak picking and delta adjustment provides a more reliable harmonic estimation than that in the EHC. In addition, a noise mixing scheme in conjunction with an improved band voicing measurement provides the naturalness of the synthesized speech. To demonstrate the performance of the proposed IHC coder, the coder has been implemented and compared with the 2.0/4.0 kbps HVXC(Harmonic excitation Vector Coding) standardized by MPEG-4. Results of subjective evaluation showed that the proposed IHC coder and produce better speech quality than the HVXC, with only 40% complexity of the HVXC.

• Korean Journal of Microbiology
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• v.45 no.4
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• pp.312-317
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• 2009

Genome analysis of a model filamentous fungus, Aspergillus nidulans, revealed that there are six putative forkhead genes. Among them, fkhF (AN8949.2) showed A. nidulans-specific. fkhF gene is located in chromosome VII and composed of 2,337 bp coding region for 778 amino acid. Since little is known about the involvement of the forkhead proteins in the developmental process of the filamentous fungi, including A. nidulans, we generated a deletion mutant of fkhF gene and analyzed. Deletion of fkhF resulted in less-dense conidiophore formation in a solid culture. However, the sexual developmental process or cleistothecia formation was normal. Furthermore, fkhF deletion mutant produced conidiophores and conidia under the submerged culture, suggesting that the fkhF gene is involved in repression of inappropriated induction and maturation of asexual developmental process but not in sexual development.