• The Journal of the Acoustical Society of Korea
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• v.29 no.1
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• pp.69-75
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• 2010

In this paper, the pitch is detected after the elimination of formant ingredients by flattening the spectrum in frequency domain. In order to remove impact of formant and transition frequency in the signal spectrum, formant envelop is made by linear interpolation with any points each sub-band and the spectrum of speech signal is compensated by the reverse of the envelop interpolated linearly after we divide frequency band into several segment based on LSP and detect the points. The experimental result showed the proposed method appeared an outstanding performance in compared with LPC, Cepstrum, Lifter methods. The method reduced the gross error rate 1.30% than the LPC method which appeared a good performance except the proposed method. Also, the proposed method showed low error rate in noise environment.

• The Journal of the Acoustical Society of Korea
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• v.29 no.4
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• pp.243-250
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• 2010

This paper proposes a spectral modeling of Korean traditional instrument, Haegeum, using cepstral analysis to naturally describe Haegeum sounds varying with time. To get a precise result of cepstral analysis, we set the frame size to 3 periods of input signal and more cepstral coefficients are used to extract formants. The performance is enhanced by flexibly controlling the cutoff frequency of bandpass filter depending on the resonances in the synthesis process of sinusoidal components and the deleting peaks remained in the residual signal. To detect the change of pitch, we divide the input frames into silence, attack, and sustain region and determine which region the current frame is involved in. Then, the proposed method readjusts the frame size according to the fundamental frequency in the case of the current frame is in attack region and corrects the extraction errors of the fundamental frequency for the frames in sustain region. With these processes, the synthesized sounds are much more similar to the originals. The evaluation result through the listening test by a Haegeum player says that the synthesized sounds are almost similar to originals (96~100 % similar to the original sounds).

• The Journal of the Acoustical Society of Korea
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• v.8 no.5
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• pp.78-85
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• 1989

This paper describes a statistical methods and perception test for extracting the parameters to be used for the synthesis-by-rule of Korean plosives. Formant synthesizer is chosen for the synthesis of the phonemes. Speech materials for the analysis consists of 72 CV monosyllables from the single male speaker. The analysis is done mainly focused on the variation of parameters in time and frequency domain, then perception tests are executed to estimate the effects of variations of the formant transitions.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.06e
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• pp.411-414
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• 1998

음성의 음질 향상(Speech Enhancement)을 위한 여러 가지 방법 중에서 주파수 차감법(Spectral Subtraction)은 계산량이 적기 때문에 현재 실시간으로 Speech Enhancement를 할 수 있는 가장 적절한 방법이다. 그러나, 이 방법은 원래의 입력음성에 없던 새로운 잡음을 만들어내는 큰 단점이 있는데, 이를 제거하기 위해 많은 연구가 되어오고 있다. 이러한 연구의 방향은 대부분 주변프레임 또는 주변의 주파수 성분과의 평균을 통해 피크값을 무디게 해 줌으로써 새로 생긴 튀는 잡음을 감소시키는 것이다. 이런 방법은 음성자체의 정보 또한 평균이 되어버리게 하는 새로운 단점을 낳는데, 이런 현상은 무성음구간에서 특히 심각해진다. 본 논문에서는 입력음성의 LPC 분석으로 백색필터(Whitening Filter)를 구성하여 이를 통과시킨 잔류신호(Residual)를 주파수 차감하여 얻은 새로운 잔류신호를 역 필터링하여(Synthesis Filter) 개선된 음성을 얻는 방법을 제안하였다. 제안된 알고리듬은, 주파수 차감시 포만트(Formant)의 정보가 더 유지 될 수 있기 때문에 잔류잡음을 줄일 수 있다. 청취 테스트 결과 제안한 방법이 기존의 방법보다 잔류잡음을 더 줄이는 사실을 확인할 수 있었다.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.49-52
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• 2001

부산지역 청소년 음성의 지속시간, 피치주파수 포만트 주파수 특성을 연령별, 성별로 분석하였다. 실제 발음 환경에서의 음성 패턴은 발성화자 개인 및 화자별로 다양하게 변화한다. 이를 모델 화하기 위해서는 다량의 음성 데이터로부터 통계적 방법에 의한 변화 요인별 파라미터 분석이 선행되어야 할 것이다. 실험에 사용된 데이터는 부산지역에 거주하는 청소년(초등학생, 중학생, 고등학생)들이 연령별로 3회 발성한 우화의 일부와 단모음(/아/,/이/,/우/,/에/,/오/)이다 실험 결과로부터 얻어진 지속시간, 주파수 특성 변화 패턴을 연령별, 성별로 구분하여 통계적으로 분석한 뒤 이를 정량화 하였다. 실험 결과로부터 부산 지역 청소년 음성의 지속시간, 주파수 특성은 예측된 바와 같이 기 연구된 성인 음성과 많은 차이를 보였으며 이는 부산 지역 방언의 DB 구축 시 설계자가 고려해야 할 기초자료로 활용 될 수 있을 것이다.

• Proceedings of the IEEK Conference
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• 1998.06a
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• pp.611-614
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• 1998

윈도우즈 95와 같은 멀티미디어 환경 하에서 개인 신분 확인을 위한 방법은 비밀번호를 키보드로 입력받는 것이었으나, 본 논문에서는 음성을 이용하는 방법으로 기존의 방법이 기준패턴의 시간에 따라 변하는 특성을 보상하지 못한다는 단점을 보완하는 방법이다. 즉, 이를 위해 음성신호의 특징인 기본주파수와 제1포만트의 비율을 이용하여 기준패턴을 형성화하는 방법에 관한 것이다. 제안한 방법으로 실험한 결과, 98%의 전체 인식율을 얻게 되었고, 윈도우즈 환경에서 비밀번호 사용 대신 음성 사용에 대한 가능성을 보여 주었다.

• The Journal of the Acoustical Society of Korea
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• v.21 no.7
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• pp.665-670
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• 2002

The most common form of pre-emphasis is y(n)=s(n)-As(n-1), where A typically lies between 0.9 and 1.0 in voiced signal. Also, this value reflects the degree of pre-emphasis and equals R(1)/R(0) in conventional method. This paper proposes a new flattening method to compensate the weaked high frequency components that occur by vocal cord characteristic. We used interval information of LSP to estimate formant frequency, After obtaining the value of slope and inverse slope using linear interpolation among formant frequency, flattening process is followed. Experimental results show that the proposed method flattened the weaked high frequency components effectively. That is, we could improve the flattening characteristics by using interval information of LSP as flattening factor at the process that compensates weaked high frequency components.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.11 no.9
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• pp.3558-3562
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• 2010

The exact fundamental frequency (pitch) extraction is important in speech signal processing. However the exact pitch extraction from speech signal is very difficult due to the effect of formant and transitional amplitude. So in this paper, the pitch is detected after flattening the spectrum in frequency region by proposed algorithm for minimized harmonics variance value. Experimental result showed the proposed method appeared an outstanding performance in compared with LPC, Cepstrum. Also, the results show the proposed method is better than conventional method.

• Journal of the Institute of Convergence Signal Processing
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• v.5 no.1
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• pp.1-5
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• 2004

A new method for the estimation of pitch-frequency contour of voiced speech is presented. The method is based on the double application of Kaiser's energy operator［1］, which has the capabilities of extracting amplitude and frequency of a sinusoidal waveform. According to the modulation model, a vowel can be represented by a combination of damped sinusoids representing formants, modulated by pitch pulses. Therefore, the amplitude envelope of each of the components will give a pitch-like waveform and the pitch can be obtained by averaging the frequencies of this waveform. The first part is the same as Gopalan's approach［9］, but by substituting the LPC based spectral analysis with the second application of energy operator, the algorithm becomes very simple and can be processed on-line. Although the estimation is rather coarse, the suggested algorithm can be useful for getting a general sketch of pitch contour on-line.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.8 no.5
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• pp.1122-1127
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• 2007

We propose the complexity reduction algorithm of real root method that is mainly used in the Vocoder. The real root method is that if polynomial equations have the real roots, we are able to find those and transform them into LSP(Line Spectrum Pairs). However, this method takes much time to compute, because the root searching is processed sequentially in frequency region. The important characteristic of LSP is that most of coefficients are occurred in specific frequency region. So, the searching frequency region is ordered and adjusted by each coefficient's distribution in this paper. Transformation time can be reduced by proposed algorithm than the sequential searching method in frequency region. When we compare this proposed method with the conventional real root method, the experimental result is that the searching time was reduced about 48% in average.