• Proceedings of the Acoustical Society of Korea Conference
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• 1992.06a
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• pp.54-57
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• 1992

음성합성에 있어서 음의 자연성을 합성시키는 문제는 크게 두가지로 나누어진다. 첫째는 합성음을 원음에 가깝게 구현하려는 합성방법 자체의 문제로, 언어 합성이 가지고 있는 일반적인 문제이다. 또 다른 문제는 운율에 관한 것으로 낱말 또는 문장 내에서의 운율에 따라 합성음의 자연성이 좌우된다. 이러한 운율에 따라 합성음의 자연성이 좌우된다. 이러한 운율의 조절에는 지속시간, 피치, 그리고 음의 세기 등이 이용된다. 켑스트럼을 이용하여 분석합성을 하는 경우, pole-zero 모델로 스펙트럼 포락을 근사하므로 원음에 충실하고, 필터계수와 구동정보를 분리하여 분석, 합성하므로 인위적인 운율의 조절이 용이하여 음성합성이 가지는 위의 두가지 문제를 해결하는데 적합하다고 판단된다. 본 연구에서는 켑스트럼을 이용하여 분석합성 시스템을 구성하였다. 음성 합성 과정에서, 운율 조절 파라미터중의 하나인 피치 주기의 변경에 따라 스펙트럼 포락의 왜곡에 대해 살펴보고, 왜곡을 최소화하는 방안을 제안한다.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.06c
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• pp.147-150
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• 1998

In the case of speech synthesis, the waveform coding method with high quality is mainly used to the synthesis by analysis. Because the parameters of this coding method are not classified as both excitation and vocal tract parameters, it is difficult to apply the waveform coding method to the synthesis by rule. Thus, in order to apply the waveform coding method to the synthesis by rule, a pitch alteration is required for the prosody control. In the speech synthesis method by the conventional PSOLA technique, applying symmetric window function to asymmetric speech waveform, it occurs the unbalance phenomenon of energy according to the overlapped degree of pitch interval adjustment. In this paper to overcome the unbalance phenomenon of energy, we proposed a new method that can convert asymmetric waveform to symmetric one by time-frequency conversion. As a result, we can obtain an average spectrum distortion ratio with 6.38% according to the pitch alteration ratio.

• Journal of the Korean Magnetic Resonance Society
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• v.17 no.1
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• pp.19-23
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• 2013

The temperature dependences of the NMR spectrum and the spin-lattice relaxation times in $KH_2PO_4$ were investigated via single-crystal NMR and MAS NMR. The stretched-exponential relaxation that occurred because of the distribution of correlation times was indicative of the degree of the distribution of the double-well potential on the hydrogen bond. The behaviors responsible for the strong temperature dependences of the $^1H$ and $^{31}P$ spin-lattice relaxation times in the rotating frame $T_{1{\rho}}$ in $KH_2PO_4$ are likely related to the reorientational motion of the hydrogen-bond geometry and the $PO_4$ tetrahedral distortion.

• Journal of the Korean Institute of Telematics and Electronics
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• v.25 no.4
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• pp.430-436
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• 1988

LSF(Line Spectral Frequency) derived from LPC has known as a very useful transmission parameter of speech signals, for it has a good linear interpolation characteristics and a low spectrum distortion at low bit rates coding. This paper presents that it is possible to extract directly the formant frequencies of speech signals from LSF parameter without application of FFT algorithm by comparing the distribution of LSF parameter with the frequency distribution of analysis filter. This paper suggests the advanced algorithm that results in improving the speed of convergence at analytic solution method. Also, for the flexibility of parameters, the process that transforms from LSF to LPC is presented.

• Journal of Electrical Engineering and information Science
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• v.3 no.1
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• pp.126-130
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• 1998

In this paper, a new interference rejection filter using the data-recycling technique is presented. The reference signal of the interference rejection filter is formed by using chip decisions, which is correlated with the narrowband interference components of the received signal. The decision feedback techniques reduce the distortion of the desired signal, which is introduced by the interference rejection filter through he use of feedback chip decisions. In order to update the filter coefficients a simple and efficient data-recycling technique that has improved performance over the conventional LMS algorithm is used. The performance of the proposed interference rejection filter is compared to filter with conventional LMS linear filter. The results show that the convergence speed of the proposed interference rejection filter using decision feedback and data-recycling technique is significantly faster than that of the conventional filters. Also, BER performance of the interference rejection filter demonstrates the superiority of the proposed filter algorithm.

• MALSORI
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• no.53
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• pp.157-169
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• 2005

This paper proposes an efficiently representing method of spatial cues on multichannel spatial audio coding. The Binaural Cue Coding (BCC) method introduced recently represents multichannel audio signals by means of Inter Channel Level Difference (ICLD) or Source Index (SI). We tried to express more efficiently ICLD and SI information based on Inter Channel Correlation in this paper. We adopt different spatial cues according to ICC and propose a compensation method of empty spectrums created by using SI. We performed a MOS test and measuring spectral distortion. The results show that the proposed method can reduce the bitrate of side information without large degradation of the audio quality.

• The Journal of the Acoustical Society of Korea
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• v.22 no.4
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• pp.269-275
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• 2003

In this paper, we proposed the robust watermark diminishing distortion of the original data inserting the watermark in specific coefficients of the frequency domain. In case the alpha is more than 0.5. we found that proposed watermark is detected by experiment of MP3, FFT, Cropping and Echo attack. Our proposed method improved the Cox's method in the SNR aspect.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.34S no.4
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• pp.50-56
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• 1997

In this paper, efficient quantization methods of line spectrum pairs (LSP) which has good performances and low complexity and memory are proosed for vocoder of mobile communication system. The adaptive quantization method utilizing the ordering property of LSP parameters is used in a scalar quantizer and a vector-scalar hybrid quantizer. The proposed scalar quantization algorithm needs 31 bits/frame to maintain the transparent quality of speech. The improved vector-scalar quantizer achieves an average spectral distortion of 1dB using 26 bits/frame. The proposed methods are evaluated in the channel errors and changed the predictor structure to maintain the robustness to channel errors.

• Proceedings of the Acoustical Society of Korea Conference
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• 1995.06a
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• pp.159-163
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• 1995

음성 코딩에 있어서 스펙트럼 정보의 효율적인 방법으로써 다중 형태 프레임 분할 방법을 제안하였다. 일반적인 보코더의 경우 엔코더에서 전송된 스펙트럼 정보를 인접 프레임 계수간의 선형보간에 의해 하부프레임 단위의 스펙트럼 계수간의 선형보간에 의해 하부프레임 단위의 스펙트럼 정보를 재생해 낸다. 이 때 비선형적 스펙트럼 변화가 존재하는 경우에는 스펙트럼 왜곡이 발생하며 본 논문에서는 이를 intra frame spectral distortion 이라 정의한다. 제안된 다중 형태 프레임 분할 방법은 한 프레임내에서 스펙트럼 계수의 변화를 잘 나타낼 수 있도록 차수마다 다른 분할 형태를 적용하여 IFSD을 최소화시키는 방법이다. 스펙트럼 정보로써 line spectrum pair 계수를 이용하였으며, 분할 개수 및 분할 형태의 개수에 따른 IFSD을 분석하여 이를 기존의 방법과 비교하였다. 그 결과 기존의 방법보다 적은 정보량으로 왜곡이 적은 스펙트럼을 재생할 수 있었다.

• Journal of the Korean Magnetic Resonance Society
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• v.5 no.2
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• pp.66-72
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• 2001

The noise suppression of time domain NMR data by discrete wavelet transform with high order Daubechies wavelet coefficients exhibits severe peak distortion and incomplete noise suppression near real signal. However, the fact that even a shift averaged Harr wavelet transform with a set of Daubechies wavelet coefficients (1/2, -l/2) can be used as a new and excellent tool to distinguish real peaks from the noise contaminated NMR signal is introduced. New algorithms of shift averaged Harr wavelet were developed and quantitatively evaluated in terms of threshold and signal to noise ratio (SNR).