• The Journal of the Acoustical Society of Korea
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• v.18 no.8
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• pp.48-53
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• 1999

This paper presents an analytic derivation of the erroneous effect when the sliding-DFT is implemented in a recursive way with the finite-bit approximation of the twiddle factors. The analysis result is obtained in a closed form equation of the noise-to-signal power ratio(NSR) employing the zero-mean white Gaussian signal as the target input of the DFT. The parameters of the wordlength used in representing the twiddle factors and the blocklength of the DFT appear in the NSR explicitly as its function variables. The derivation is based on the error dynamic equation which is derived from the recursive SDFT, and on the analytic exploration of the statistical characteristics of the approximation coefficients treating them as random variables of having spatial distributions. The analytically derived results are verified through the comparison with the data actually measured from the computer simulation experiment.

• Proceedings of the Korea Contents Association Conference
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• 2005.11a
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• pp.580-584
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• 2005

One of the problems of panoramic image stitching methods is that its computational cost is so high that the image processing required usually cannot be done in real-time. Real-time performance is important in applications such as video surveillance becausewe must see current scenes. But it takes more than several seconds to calculate transform coefficients between images. Panoramic VR technologies such as Apple QuickTime VR have problem that distorts images of top and bottom. This paper presents a fast stitching method and a methpd reducing distortions of top and bottom in cubic panoramic image.

• Journal of Electrical Engineering and Technology
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• v.6 no.3
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• pp.319-327
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• 2011

An iterative technique based on orthogonal filters and frequency tracking is proposed to estimate harmonic components in power systems. The technique uses frequency interpolation to estimate fundamental frequency and harmonics when the nominal frequency of the signal is a non-integer value. Due to the number of computations involved during the generation of filter coefficients, an offline computation is suggested. Beneficial features of the proposed technique include fixed sampling rate and fixed data window size. The performance of the proposed technique is examined by simulating different power system operating conditions and evaluating the data from these simulations. A technique based on Fast Fourier Transform is also used to estimate the harmonic components for all the simulated signals. These estimates are compared with those obtained from the proposed technique. Results show that the proposed technique can converge to the accurate fundamental frequency and therefore, provide accurate harmonic components even when the fundamental frequency is not equal to the nominal frequency.

• Journal of the Institute of Convergence Signal Processing
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• v.6 no.3
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• pp.107-111
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• 2005

With the increasing use of multimedia technologies, image compression requires higher performance as well as new functionality in the informationized society. Specially, in the specific area of still image encoding in embedded system, a new standard, JPEG2000 that improve various problem of JPEG was developed. This paper proposed a method that reduce quantity of data delivered in EBCOT(Embedded Block Coding with Optimized Truncation) process using SPIHT(Set Partitioning in Hierarchical Trees) Algorithm to optimize selection of threshold from feature of wavelet transform coefficients and to remove sign bit in LL area for the increment of compression efficiency on JPEG2000. The experimental results showed the proposed algorithm achieves more improved bit rate in embedded system.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.9C
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• pp.1298-1305
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• 2004

Most watermarking techniques embed watermarks in the middle frequency range for robustness and invisibility. In our proposed watermarking algorithms embed the gaussian sequence watermark into low frequency area of the wavelet transform domain because the histogram of low sub-band area is composed by similar coefficients. Also, our proposed scheme doesn't need the original image in extraction procedure The experimental results show good robustness against the Check Mark benchmarking tools.

• The Transactions of the Korean Institute of Electrical Engineers D
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• v.54 no.2
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• pp.124-130
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• 2005

In this paper, low-power design and implementation techniques for DWT(Discrete Wavelet Transform) of the JPEG2000 compression are proposed. In DWT block of the JPEG2000, linear phase 9 tap and 7 tap filters are used. For low-power implementation of those filters, processor technique for DA(Distributed Arithmetic) filter and minimization technique for number of addition in CSD(Canonic Signed Digit) filter are utilized. Proposed filter structure consists of 3 blocks. In the first CSD coefficient block, every possible 4 bit CSD coefficients are calculated and stored. In second processor block, multiplication is done by MUX and addition processor in terms of the binary values of filter coefficient. Finally, in third block, multiplied values are output and stored in flip-flop train. For comparison of the implementation area and power dissipation, proposed and conventional structures are implemented by using Verilog-HDL coding. In simulation, it is shown that 53.1% of the implementation area can be reduced comparison with those of the conventional structure.

• The Journal of the Acoustical Society of Korea
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• v.24 no.8
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• pp.492-497
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• 2005

In this paper, we suggest that the new volume of robust audio watermarking method protects form the MP3 compression attack, mostly used in audio field. Watermarks were inserted at the coefficients repeatedly from low frequencies to high frequencies after DCT transform in commonly used Cox's spread spectrum method. However, our method could be investigated losing frequency when MP3 attacks and make order by small amount of losing frequency and then insert watermarks rightly through statistical analysis. In the result about various sound clips, the suggested method have better performances than Cox's method by preserving watermarks and reducing distortions of the original sounds.

• Proceedings of the IEEK Conference
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• summer
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• pp.237-243
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• 2004

A multichannel smart sound sensor capable to detect and identify sound events in noisy conditions is presented in this paper. Sound information extraction is a complex task and the main difficulty consists is the extraction of high­level information from an one-dimensional signal. The input of smart sound sensor is composed of data collected by 5 microphones and its output data is sent through a network. For a real time working purpose, the sound analysis is divided in three steps: sound event detection for each sound channel, fusion between simultaneously events and sound identification. The event detection module find impulsive signals in the noise and extracts them from the signal flow. Our smart sensor must be capable to identify impulsive signals but also speech presence too, in a noisy environment. The classification module is launched in a parallel task on the channel chosen by data fusion process. It looks to identify the event sound between seven predefined sound classes and uses a Gaussian Mixture Model (GMM) method. Mel Frequency Cepstral Coefficients are used in combination with new ones like zero crossing rate, centroid and roll-off point. This smart sound sensor is a part of a medical telemonitoring project with the aim of detecting serious accidents.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.44 no.5
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• pp.94-102
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• 2007

In de-noising, it is wellknown that wavelet-thresholding algorithm shows near-optimal performances in the minimax sense. However, the wavelet-thresholding algorithm is difficult in realization it on hardware, such as FPGA, because of wavelet function complexity. In this paper, we present a new do-noising technique with the binary tree structured filter bank, which is based on the signal power ratio of each subbands to the total signal power. And we realize it on FPGA. For simple realization, the filter banks are designed by Hadamard transform coefficients. The simulation and hardware experimental results show that the performance of the proposed method is similar with that of soft thresholding de-noising algorithm based on wavelets, nevertheless it is simple.

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.44 no.9
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• pp.48-53
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• 2007

Some digital signal processing applications, such as FFT, request multiplications with a group(or, groups) of a few predetermined coefficients. In this paper, based on the modified CSD algorithm, an efficient multiplier design method for predetermined coefficient groups is proposed. In the multiplier design for sine-cosine generator used in direct digital frequency synthesizer(DDFS), and in the multiplier design used in 128 point $radix-2^4$ FFT, it is shown that the area, power and delay time can be reduced up to 34%.