• The Journal of the Acoustical Society of Korea
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• v.10 no.5
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• pp.11-19
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• 1991

음성 신호의 합성기법들 중에서 파형코딩법은 음질이 우수하기 때문에 분석에 의한 합성법으로 많이 사용하고 있다. 그렇지만 음원과 성도의특성을 분리하지 않고 파형의 잉여분만을 제거한 후에 파 형자체를 저장하기 때문에 규칙에 의한 합성기법으로 사용하기에는 어려움이 많다. 본 논문은 파형코딩 법 중 선형 PCM 코딩법으로 저장된 음성파형에 대해 피치를 양분할 수 있는 주기반분법을 제안하여 파형자체의 음원을 분리하지 않고 피치 주기를 변경시킬 수 있는 새로운 피치 변경법을 제안하였다. 따 라서 음질이 우수한 파형코딩 합성법으로 규칙에 의한 합성을 수행할 수 있다.

• The Journal of the Acoustical Society of Korea
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• v.17 no.8
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• pp.82-87
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• 1998

음성합성은 합성방식에 따라 파형부호화법, 신호원부호화법, 혼성부호화법으로 분류 할 수 있다. 특히 고음질 합성을 위해서는 파형부호화를 이용한 합성방식이 적합하다. 그렇 지만, 파형부호화를 이용한 합성법은 여기 성분과 여파기 성분을 분리하지 않고 처리하기 때문에 음절단위나 음소단위의 합성기법으로는 바람직하지 못하다. 따라서 파형부호화법을 규칙에 의한 합성에 적용되도록 음원피치를 변경시키기 위한 피치 변경법이 필요하게 된다. 본 논문에서는 스펙트럼 왜곡을 최소화하기 위해 켑스트럼의 성질을 이용하여 피치를 변경 하는 방법에 대하여 제안하였다. 이 방법은 주파수영역상에서 여기 스펙트럼과 여파기 스펙 트럼을 분리하여 여기 스펙트럼을 여기 켑스트럼으로 변환한 후 영값 삽입이나 삭제에 의해 피치를 변경하고 스펙트럼영역에서 피치 변경된 스펙트럼을 재구성하는 기법을 적용하였다. 제안한 방법의 성능을 평가하기 위해 스펙트럼 왜곡율을 측정하여 본 결과 평균 스펙트럼 왜곡율은 평균 2.29%이하로 유지되었으며 주관적인 음질도 평균 3.74로 우수하였다.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.11 no.6
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• pp.200-207
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• 2001

In this paper, a new method for Partial field decomposition is developed that is based on the beamforming algorithm for the application of acoustical holography to a composite sound field generated by multiple incoherent sound sources. In the proposed method, source Positions are first predicted by MUSIC(multiple signal classification) algorithm. The composite sound fields can then be decomposed into each partial field by the beamforming. Results of both numerical simulations and experiments show that the method can find each partial field very accurately and effectively, and that it also has Potential to be used for application to distributed sources.

• The Journal of the Acoustical Society of Korea
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• v.18 no.7
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• pp.80-84
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• 1999

In this paper, the split model analysis algorithm, which can generate the wideband speech signal from the spectral information of narrowband signal, is developed. The split model analysis algorithm deals with the separation of the 10/sup th/ order LPC model into five cascade-connected 2/sup nd/ order model. The use of the less complex 2/sup nd/ order models allows for the exclusion of the complicated nonlinear relationships between model parameters and all the poles of the LPC model. The relationships between the model parameters and its corresponding analog poles is proved and applied to each 2/sup nd/ order model. The wideband speech signal is obtained by changing only the sampling rate.

• The Journal of the Acoustical Society of Korea
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• v.28 no.8
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• pp.761-767
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• 2009

The detection of target echoes in a sonar image is usually difficult since reverberation is originated by the returns reflected around the boundary and volumes. Under the scenario of the target presence around the reverberation, the detection performance of existing algorithms is degraded. Since they have a similar statistical features. But proposed detector gives improvement existing algorithms Under this scenario. In this paper, 4 segmentation contrast box algorithm using maximum contrast value is proposed based on statistical segmentation, which gives better detection performance in the sense of reducing false alarms. The simulations validate the effectiveness of the proposed algorithm.

• The Journal of the Acoustical Society of Korea
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• v.32 no.1
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• pp.79-85
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• 2013

Feature extraction of CASA system uses time continuity and channel similarity and makes correlogram of auditory elements for the use. In case of using feature extraction with cross correlation coefficient for channel similarity, it has much computational complexity in order to display correlation quantitatively. Therefore, this paper suggests feature extraction method using non-parametric correlation coefficient in order to reduce computational complexity when extracting the feature and tests to segregate target speech by CASA system. As a result of measuring SNR (Signal to Noise Ratio) for the performance evaluation of target speech segregation, the proposed method shows a slight improvement of 0.14 dB on average over the conventional method.

• The Journal of the Acoustical Society of Korea
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• v.31 no.5
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• pp.332-339
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• 2012

Beamspace transform algorithm transforms spatial-domain data - such as x, y, z dimension - into incidence-angle-domain data, which is called beamspace-domain data. The beamspace transform method is generally used in source localization and tracking, and adaptive beamforming problem. When the beamspace transform method is used in multichannel audio source separation, the inverse beamspace transform is also important because the source image have to be reconstructed. This paper studies the beamspace transform and inverse transform algorithms for multichannel audio source separation system, especially for the beamspace-domain multichannel NMF algorithm.

• The Journal of the Acoustical Society of Korea
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• v.19 no.3
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• pp.26-34
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• 2000

In this paper, we propose a 4kbps speech coder that combines the harmonic vector excitation coding with time-separated transition coding. The harmonic vector excitation coding uses the harmonic excitation coding in the voiced frame and uses the vector excitation coding with the structure of analysis-by-synthesis in the unvoiced frame, respectively. But two mode coding method is not effective for transition frame mixed in voiced and unvoiced signal and a new method beyond using unvoiced/voiced mode coding is needed. Thus, we designed a time-separated transition coding method for transition frame in which a voiced/unvoiced decision algorithm separates unvoiced and voiced duration in a frame, and harmonic-harmonic excitation coding and vector-harmonic excitation coding method is selectively used depending on the previous frame U/V decision. In the decoder, the voiced excitation signals are generated efficiently through the inverse FFT of harmonic magnitudes and the unvoiced excitation signals are made by the inverse vector quantization. The reconstructed speech signal are synthesized by the Overlap/Add method.

• The Journal of the Acoustical Society of Korea
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• v.27 no.1
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• pp.40-46
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• 2008

Weak target detection and localization in the presence of loud surface ship noise is a critical problem for matched field processing (MFP) in shallow water. For stationary sources, each signal component of received signal can be separated and interference can be suppressed using eigen space analysis schemes. However, source motion, in realistic cases, causes spreading of signal energies in their subspace. In this case, eigenvalues of target and interfere signal components are mixed and hard to be separated with usual phone space eigenvector decomposition (EVD) approaches. Our technique is based on mode space and utilizes the difference in their physical characteristics of surface and submerged sources. Performing EVD for modal cross spectral density matrix, interference components in the mode amplitude subspace can be classified and eliminated. This technique is demonstrated with synthetic data, and results are discussed.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.14 no.10
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• pp.2207-2214
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• 2010

Underwater acoustic(UWA) communication has multipath error because of reflection by sea-level and sea-bottom. The multipath of UWA channel causes signal distortion and error floor. In this paper, we proposed split input bits of channel decoder using method of maximum value, average value, LLR value for optimal estimation. Channel coding method is LDPC(N size=16000) standard in DVB-S2. As shown in simulation results, the performance of LLR value method is better than other methods.