• Proceedings of the Korea Information Processing Society Conference
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• 2018.10a
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• pp.805-807
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• 2018

본 논문은 인공 대역 확장에서 스펙트럼 포락선 확장 시 발생하는 스펙트럼 왜곡을 줄이는 개선된 조합 매핑(Improved combined mapping) 알고리즘을 제안한다. 벡터양자화를 기반으로 하는 코드북 매핑(Codebook mapping)과 스펙트럼 포락선(Spectrum Envelope)의 선형 의존도를 이용한 선형 매핑(Linear mapping)을 사용하여 각각 확장된 광대역 LSP(Line Spectrum Pair)를 추론하고, 연판정(Soft-decision)을 통해 최적화된 LSP를 추론한다. 제안된 알고리즘으로 합성된 음성신호의 스펙트럼 왜곡(Spectrum Distortion)이 기존 조합매핑으로 얻은 음성 신호의 스펙트럼 왜곡보다 더 적은 왜곡을 갖는 결과를 나타내었다.

• The Journal of the Acoustical Society of Korea
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• v.42 no.6
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• pp.544-551
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• 2023

Speech enhancement used to improve the perceptual quality and intelligibility of noise speech has been studied as a method using a complex-valued spectrum that can improve both magnitude and phase in a method using a magnitude spectrum. In this paper, a study was conducted on how to apply attention mechanism to complex-valued spectrum-based speech enhancement systems to further improve the intelligibility and quality of noise speech. The attention is performed based on additive attention and allows the attention weight to be calculated in consideration of the complex-valued spectrum. In addition, the global average pooling was used to consider the importance of the feature map. Complex-valued spectrum-based speech enhancement was performed based on the Deep Complex U-Net (DCUNET) model, and additive attention was conducted based on the proposed method in the Attention U-Net model. The results of the experiments on noise speech in a living room environment showed that the proposed method is improved performance over the baseline model according to evaluation metrics such as Source to Distortion Ratio (SDR), Perceptual Evaluation of Speech Quality (PESQ), and Short Time Object Intelligence (STOI), and consistently improved performance across various background noise environments and low Signal-to-Noise Ratio (SNR) conditions. Through this, the proposed speech enhancement system demonstrated its effectiveness in improving the intelligibility and quality of noisy speech.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.11 no.3
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• pp.1633-1649
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• 2017

In remote sensing image processing, the traditional fusion algorithm is based on the Intensity-Hue-Saturation (IHS) transformation. This method does not take into account the texture or spectrum information, spatial resolution and statistical information of the photos adequately, which leads to spectrum distortion of the image. Although traditional solutions in such application combine manifold methods, the fusion procedure is rather complicated and not suitable for practical operation. In this paper, an improved IHS transformation fusion algorithm based on the local variance weighting scheme is proposed for remote sensing images. In our proposal, firstly, the local variance of the SPOT (which comes from French "Systeme Probatoire d'Observation dela Tarre" and means "earth observing system") image is calculated by using different sliding windows. The optimal window size is then selected with the images being normalized with the optimal window local variance. Secondly, the power exponent is chosen as the mapping function, and the local variance is used to obtain the weight of the I component and match SPOT images. Then we obtain the I' component with the weight, the I component and the matched SPOT images. Finally, the final fusion image is obtained by the inverse Intensity-Hue-Saturation transformation of the I', H and S components. The proposed algorithm has been tested and compared with some other image fusion methods well known in the literature. Simulation result indicates that the proposed algorithm could obtain a superior fused image based on quantitative fusion evaluation indices.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.20 no.10
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• pp.1867-1872
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• 2016

In the area of digital signal processing, it is necessary to improve the quality of the speech signal after removing the background noise which exists in a various real environments. The important thing to consider when removing the background noise acoustically is that to solve the problem, depending on the information of the human auditory mechanism is mainly the amplitude spectrum of the speech signal. This paper introduces the characteristics of a frequency-weighted spectrum shaping filter for the extraction of the amplitude spectrum of the speech signal with the primary purpose. Therefore, this paper proposes an algorithm using the methods of a Wiener filter and the frequency-weighted spectrum shaping filter according to the acoustic model, after extracted the amplitude spectral information in the noisy speech signal. The spectral distortion (SD) output of the proposed algorithm is experimentally improved more than 5.28 dB compared to a conventional method.

• Journal of KSNVE
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• v.8 no.1
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• pp.180-186
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• 1998

Vibration test fixture is used in random vibration control testing. The specified reference spectrum should be transmitted equally to the specimen attachment points on the fixture. In most practical cases, however, spectrum at each of specimen attachment points may be quite different from the specified reference spectrum because of the dynamic characteristics of vibration test fixture. This paper proposes the method of experimental dynamic modification of fixture system for vibration test so that the reference spectrum can be transmitted to the specimen attachment points without distortion. The stiffness of mounts of specimen and the thickness of fixture are considered as design variables. The frequency response functions of specimen used for input data are obtained from vibration testing, and the frequency response functions of fixture are obtained from finite element modeling. The sensitivities of frequency response functions at specimen attachment points to the mount stiffness are derived from synthesis method of transfer function. And the sensitivities to the thickness of fixture are also derived from finite element modeling. The presented method is verified by computer simulation and vibration testing.

• Journal of the Korean Geotechnical Society
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• v.24 no.4
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• pp.115-123
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• 2008

The dispersive phase velocity of a wave propagating through multilayered systems such as a soil site is an important parameter and carries valuable information in non-destructive site characterization tests. The dispersive phase velocity of a wave can be determined using the phase spectrum, which is easily evaluated through the cross power spectrum. However, the phase spectrum determined using the cross power spectrum is easily distorted by background noise which always exists in the field. This causes distortion of measured signal and difficulties in the determination of the dispersive phase velocities. In this paper, a new method to evaluate the phase spectrum using the harmonic wavelet transform is proposed and the phase spectrum by the proposed method is applied to the determination of dispersion curve. The proposed method can successfully remove background noise effects. To evaluate the validity of the proposed method, numerical simulations of multi-layered systems were performed. Phase spectrums and dispersion curves determined by the proposed method were found to be in good agreement with the actual phase spectrums and dispersion curves biased by heavy background noise. The comparison manifests the proposed method to be a very useful tool to overcome noise effects.

• The Journal of the Acoustical Society of Korea
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• v.24 no.7
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• pp.416-422
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• 2005

Speech recognition in digital communication systems has very low performance due to the spectral distortion caused by speech codecs. In this paper, the spectral distortion by speech codecs is analyzed and a pre-processing method which compensates for the spectral distortion is proposed for performance enhancement of speech recognition. Three standard speech codecs. IS-127 EVRC. ITU G.729 CS-ACELP and IS-96 QCELP. are considered for algorithm development and evaluation, and a single method which can be applied commonly to all codecs is developed. The performance of the proposed method is evaluated for three codecs, and by using the speech features extracted from the compensated spectrum. the recognition rate is improved by the maximum of $15.6\%$ compared with that using the degraded speech features.

• Journal of Advanced Navigation Technology
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• v.13 no.6
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• pp.846-853
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• 2009

This paper presents the design of a high efficiency frequency multiplier with load-pull simulation and analyses the nonlinear distortion of the frequency multiplier. The frequency multiplier shows serious distortion of multiplying signal bandwidth because of nonlinearity when modulated signal is applied, so a digital predistortion with look up table (LUT) is applied to compensate for the distortion of the frequency multiplier. The frequency multiplier is designed to produce 5.8GHz output by doubling the input frequency to be operating at IEEE 802.11a standard wireless LAN. The output spectrum shows 12dB ACPR improvement both at +11MHz, +20MHz offset from center frequency after linearization.

• Journal of the Korean Institute of Electrical and Electronic Material Engineers
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• v.17 no.10
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• pp.1132-1141
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• 2004

The waveforms of the leakage currents for the surface discharge according to the degree of salt contamination and the variation of environmental condition on the EPDM polymer insulators, were shown in this paper. The variation phenomena of fundamental wave, 3rd and 5th harmonic waveforms were also shown from the beginning of the applied voltage and to the flashover voltage. To develop the technique of percentage contamination degree of EPDM polymer insulator according to the variation of environmental condition, the distortion degree of 3rd and 5th harmonic wave to the fundamental wave was utilized through the spectrum analysis for the waveforms of leakage current in the simulation of salt and fog test. The fact that distortion degree % of 3rd and 5th harmonics for fundamental wave is a necessary indicator for the assessment of contamination degree for the polymer insulators, was known.

• The Journal of the Acoustical Society of Korea
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• v.17 no.4E
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• pp.35-40
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• 1998

In this paper, an efficient quantization method of line spectrum pairs(LSP) with cascaded structure of vector quantizer and scalar quantizer is proposed. First, input LSP parameters is vector-quantized using a codebook a with a moderate number of entries. In the second stage of quantization, the components of residual vector are individually quantized by the scalar quantizer. The utilization of ordering property of LSP parameters and the inclusion of interframe prediction improve the quantizer performance and remove the stability check routine after quantization procedure. The new vector-scalar hybrid quantizer using 26 bits/frame shows a transparent quality of speech that an average spectral distortion is 1 dB and the frame proportion with above 2 dB spectral distortion is less than 2%. The performances of proposed quantization method is evaluated in the transmission errors.