• Journal of Communications and Networks
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• v.1 no.2
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• pp.86-88
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• 1999

The Discrete Fourier Transform (DFT) and the Inverse Discrete Fourier Transform (IDFT) are commonly used in signal processing applications, in particular in digital communication sys-tems using the multi-carrier modulation principle. In such systems an IDFT is computed at the transmitter end, and a DFT at the re-ceiver end. This paper examines a technique of computations, for which only negligible differences appear between the DFT and the IDFT calculations while the number of arithmetic operations re-quired is substantially reduced. This offers significant advantages for the design of an IDFT/DFT processor for Discrete Multitone(DMT) systems.

• The Journal of the Acoustical Society of Korea
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• v.20 no.4E
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• pp.39-44
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• 2001

We investigate the numerical instability of linear prediction for discrete harmonic spectra of audio signals. It is identified that the eigenvalue spread is very large when discrete harmonic spectra are confined only in a lower part of the entire signal bandwidth. A simple method that redefines the sampling frequency and associate harmonic frequencies is proposed to improve the numerical stability. Simulation results using real audio signals indicate its superior stabilizing ability and improved accuracy in the discrete spectral estimation for both LP and DAP.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.45 no.3
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• pp.80-92
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• 2008

Many predominant video coding tools in terms of coding efficiency were adopted in the latest video coding standard, H.264/AVC. Regardless of development of these predominant video coding tools such as the variable block-size motion estimation/compensation, intra prediction based on various directions, and so on, the discrete cosine transform has been continuously used starting from the early video coding standards. Generally, the correlation coefficient of the residual signal is usually less than 0.5 when this residual signal is actually encoded. In this interval of correlation coefficient, the discrete cosine transform does not show the optimal coding gain, and the discrete sine transform which is a sub-optimal transform when the correlation coefficient is in the interval from -0.5 to 0.5 can be used in conjunction with the discrete cosine transform in the video coding. In this paper, an alternative transform that alternatively uses the discrete sine transform and integer cosine transform in H.264/AVC by using rate-distortion optimization is proposed. The proposed method achieves a BD-PSNR gain of up to 0.71 dB compared to H.264/AVC JM 10.2 at relatively high bitrates.

• Proceedings of the IEEK Conference
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• 2000.09a
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• pp.959-962
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• 2000

A heart sound algorithm, which separates the heart sound signal into four parts; the first heart sound, the systolic period, the second heart sound, and the diastolic period has been developed. The algorithm uses discrete intensity envelopes of approximations of the wavelet transform analysis method to the phonocard-iogram(PCG)signal. Heart sound a highly nonstation-ary signal, so in the analysis of heart sound, it is important to study the frequency and time information. Further more, Wavelet Transform provides more features and characteristics of the PCG signal that will help physician to obtain qualitative and quantitative measurements of the heart sound.

• 전기의세계
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• v.19 no.5
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• pp.1-7
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• 1970

Starting from the review of signal flow graphs and flow graphs, this paper gives an example of sampled-data systems for Sedlar & Bekey's formulation. In this purpose it discussed the difference between Mason's signal-flow graphs and Coates flow graphs for drawing th flow graph of a linear system, and then a new flow-graph symbol introduced in order to distinguish between continuous and discrete systems. Thus, the paper is analysed and compared with a sampled-data systems between conventional methods and new method of signal flow graphs.

• Journal of Korea Society of Digital Industry and Information Management
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• v.9 no.1
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• pp.165-176
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• 2013

This paper introduces the high density discrete wavelet transform using quincunx sampling, which is a discrete wavelet transformation that combines the high density discrete transformation and non-separable processing method, each of which has its own characteristics and advantages. The high density discrete wavelet transformation is one that expands an N point signal to M transform coefficients with M > N. The high density discrete wavelet transformation is a new set of dyadic wavelet transformation with two generators. The construction provides a higher sampling in both time and frequency. This new transform is approximately shift-invariant and has intermediate scales. In two dimensions, this transform outperforms the standard discrete wavelet transformation in terms of shift-invariant. Although the transformation utilizes more wavelets, sampling rates are high costs and some lack a dominant spatial orientation, which prevents them from being able to isolate those directions. A solution to this problem is a non separable method. The quincunx lattice is a non-separable sampling method in image processing. It treats the different directions more homogeneously than the separable two dimensional schemes. Proposed wavelet transformation can generate sub-images of multiple degrees rotated versions. Therefore, This method services good performance in image processing fields.

• 제어로봇시스템학회:학술대회논문집
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• 2002.10a
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• pp.97.1-97
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• 2002

$\textbullet$ Abstract $\textbullet$ Introduction $\textbullet$ Wire Rope Detecting System $\textbullet$ Signal Processing $\textbullet$ Experiment $\textbullet$ Conclusion $\textbullet$ Reference

• The Journal of the Acoustical Society of Korea
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• v.27 no.7
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• pp.372-378
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• 2008

Voice activity detectors (VAD) are important in wireless communication and speech signal processing, In the conventional VAD methods, an expression for the likelihood ratio test (LRT) based on statistical models is derived in discrete Fourier transform (DFT) domain, Then, speech or noise is decided by comparing the value of the expression with a threshold, This paper presents a new statistical VAD method based on a signal subspace approach, The probabilistic principal component analysis (PPCA) is employed to obtain a signal subspace model that incorporates probabilistic model of noisy signal to the signal subspace method, The proposed approach provides a novel decision rule based on LRT in the signal subspace domain, Experimental results show that the proposed signal subspace model based VAD method outperforms those based on the widely used Gaussian distribution in DFT domain.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.12 no.1
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• pp.1-6
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• 2012

In this paper, the performance comparison of the TR-UWB System using Wavelet Pulse, DPSS pulse, and second Gaussian pulse in multi-user environment is analyzed. The data signal and the reference signal is sent with some time intervals in TR-UWB system. At this time the two signals are through the same channel. In the receiver the reference signal is used as a template. This advantage results in demodulation without channel estimation and low complexity of it. However the conventional TR-UWB system based on a Gaussian signal in a multi-user environment has the disadvantage of poor performance due to the interferences between the users. To overcome this disadvantage, DPSS (Discrete Prolate Shperoidal Sequence) is used to reduce the interferences between the users. We propose the system with multiresolution function of orthogonal wavelet reducing the interferences, which has a better performance in multi-user environment when they are sent in transmission side.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.2 no.2
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• pp.37-43
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• 2009

In this study, multi-channel digital hearing aid algorithm for low power system is proposed. First, MDCT(Modified Discrete Cosine Transform) method converts time domain of input speech signal into frequency domain of it. Output signal from MDCT makes a group about each channel, and then each channel signal adjusts a gain using LCF(Loudness Compensation Function) table depending on hearing loss of an auditory person. Finally, compensation signal is composed by TDAC and IMDCT. Its all of process make progress 16-bit fixed-point operation. We use fast-MDCT instead of MDCT for reducing system complexity and previously computed tables instead of log computation for estimating a gain. This algorithm evaluate through computer simulation.