• Journal of information and communication convergence engineering
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• v.13 no.4
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• pp.235-240
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• 2015

A decimation chain for multi-standard reconfigurable radios is presented for 900-MHz and 1,900-MHz dual-band cellular standards with a data interpolator based on the Lagrange method for adjusting the variable data rate to a fixed data rate appropriate for each standard. The two proposed configurations are analyzed and compared to provide insight into aliasing and the signal bandwidth by means of a newly introduced measure called interpolation error. The average interpolation error is reduced as the ratio of the sampling frequency to the signal BW is increased. The decimation chain and the multi-rate analog-to-digital converter are simulated to compute the interpolation error and the output signal-to-noise ratio. Further, a method to operate the above-mentioned chain under a compressed mode of operation is proposed in order to guarantee continuous packet connectivity for inter-radio-access technologies. The presented decimation chain can be applied to LTE, WCDMA, GSM multi-mode multi-band digital front-end which will ultimately lead to the software-defined radio.

• Journal of the Korean Institute of Telematics and Electronics
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• v.25 no.7
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• pp.826-835
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• 1988

In this paper, we analyze the generation of an image pyramid as a 2-dimensional decimation-interpolation process, and suggest a performance index of FIR filter for decimation and interpolation filter. Until now, most deciamtion and interpolation filters are designed via the approximation of the impulse response of an ideal filter. In this paper, however, we propose a new performance index that minimizes the maximum frequency-weighted mean square error between the desired and the generated interpolated signal, and propose an optimal half-band filter based on the proposed performance index as an example. Some simulation results with real images show that the proposed optimal half-band filter yields a higher PSNR as well as the more preferable image quality, in comparison with other currently used filters with the same computational complexity.

• MALSORI
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• no.45
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• pp.63-77
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• 2003

The purpose of voice color conversion is to change the speaker identity perceived from the speech signal. In this paper, we propose a new voice color conversion algorithm through the formant shifting and the spectrum-tilt modification in the frequency domain. The basic idea of this technique is to convert the positions of source formants into those of target speaker's formants through interpolation and decimation and to modify the spectrum-tilt by utilizing the information of both speakers' spectrum envelops. The LPC spectrum is adopted to evaluate the position of formant and the information of spectrum-tilt. Our algorithm enables us to convert the speaker identity rather successfully while maintaining good speech quality, since it modifies speech waveforms directly in the frequency domain.

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.37 no.5
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• pp.16-22
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• 2000

This paper presents a compression algorithm applicable for transmitting a HPO hologram data. The proposed algorithm exploits a modulating function to compress the bandwidth of the hologram pattern, resulting in decimation due to relaxed Nyquist sampling constraints. At the receiver, the compressed data will be interpolated and compensated via being divided by the modulating function. We also present compression rate and analyze the resolution of a reconstructed image and the periodicity of harmonic interferences. Finally, we shows the validity of the proposed algorithm by simulation where a reconstructed image from undersampled data is compared with a reconstructed image obtained through decimatioin by modulating function, interpolation and compensation.

• The Journal of the Acoustical Society of Korea
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• v.20 no.1
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• pp.13-20
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• 2001

Adaptive digital filters with long impulse response such as acoustic echo canceller and active noise controller suffer from slow convergence and computational burden. Subband techniques and multirate signal processing have been recently developed to improve the problem of computational complexity and slow convergence in conventional adaptive filter. Any FIR transfer function can be realized as a serial connection of interpolators followed by subfilters with a sparse impulse response. In this case, each interpolator which is related to the column vector of Hadamard matrix has band-pass magnitude response characteristics shifted uniformly. Subband technique using Hadamard transform and decimation of subband signal to reduce sampling rate are adapted to system modeling and acoustic noise cancellation In this paper, delayless subband structure with nonuniform bandwidth has been proposed to improve the performance of the convergence speed without aliasing due to decimation, where input signal is split into subband one using tree-structured filter bank, and the subband signal is decimated by a decimator to reduce the sampling rate in each channel, then subfilter with sparse impulse response is transformed to full band adaptive filter coefficient using Hadamard transform. It is shown by computer simulations that the proposed method can be adapted to general adaptive filtering.

• Journal of Biomedical Engineering Research
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• v.17 no.2
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• pp.209-214
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• 1996

The display of biological signals in raster scan monitors often involves a multirate sampling operation which consists of decimation .and interpolation. All electroencephalouaphic (EEG) samples of 10 to 30 seconds (2, 500 to 7, 500 samples at 250[Hz] sampling frequency) must be displayed in the computer screen to keep the aspect ratio of the paper polygraph output. Since the current afrorclable display technology Plots at most 2, 000 Pixels Per row, sDme signal samples need to be discarde4 This Paper studies methods to perform this operation characterizing them from the signal processing viewpoint and compares the display quality among several decimation techniques. Experimental results show that a nonlinear operation such as the peak detection method could be preferable to the canonical linear filtering to reduce aliasing.

• Journal of the Korea Institute of Information Security & Cryptology
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• v.18 no.5
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• pp.93-101
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• 2008

Side channel attacks are well known as one of the most powerful physical attacks against low-power cryptographic devices and do not take into account of the target's theoretical security. As an important succeeding factor in side channel attacks (specifically in DPAs), exact time-axis alignment methods are used to overcome misalignments caused by trigger jittering, noise and even some countermeasures intentionally applied to defend against side channel attacks such as random clock generation. However, the currently existing alignment methods consider only on the position of signals on time-axis, which is ineffective for certain countermeasures based on time-axis misalignments. This paper proposes a new signal alignment method based on interpolation and decimation techniques. Our proposal can align the size as well as the signals' position on time-axis. The validity of our proposed method is then evaluated experimentally with a smart card chip, and the results demonstrated that the proposed method is more efficient than the existing alignment methods.

• Journal of Korea Multimedia Society
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• v.9 no.4
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• pp.443-450
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• 2006

In digital image communication systems, YUV 4:2:0 or YUV 4:2:2 images filtered and subsampled versions of YUV 4:4:4 images are utilized and these images recover their size by an interpolation filter. Low pass filters and interpolation filters in the image communication systems are generally utilized. Thus, to improve the image quality, efficient low pass filters and interpolation filters are still required. In this paper, we propose new and efficient low pass filters and interpolation filters and their design method. The low pass filters and interpolation filters used in the MPEG-2 system were developed independently. We utilize wavelet transforms to jointly design low pass filters and interpolation filters. Simulation results show that the proposed filters are superior to the filters used in MPEG-2 in terms of PSNR. In addition, the length of the proposed interpolation filters is shorter than that of the filters used in the MPEG 2 system.

• Journal of Broadcast Engineering
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• v.9 no.1
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• pp.54-63
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• 2004

A new motion compensated frame Interpolation (MCI) algorithm by block based motion estimation (BME) is proposed. The block for the BME is composed of a large overlapped block for practical object motion estimation (ME) and a small block (which has a coinciding center with the ME-block) for the more precise motion compensated image description. Pixels in the block for the ME are sub-sampled to reduce computational complexity. The proposed method is executed with the various ME-blocks which have different size and sub-sampling ratio, and compared to the conventional method.

• Proceedings of the Acoustical Society of Korea Conference
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• 1993.06a
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• pp.109-112
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• 1993

In this paper, we propose the modified VQ, applied correlation between codewords in order to reduce the error rate due to personal and speakers' temporal variation. Such a modified VQ is used in the stage of preprocessing of HMM and the temporal variation is absorbed by nonlinear Decimation and Interpolation of vowel part that we obtain higher recognition rate than not so case. The objects of experiment are Korea 142 DDD regional names and we show that the proposed method increase the recognition rate.