• Earthquakes and Structures
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• v.3 no.3_4
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• pp.341-363
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• 2012

The spectral representation method is a quick and versatile tool for the generation of spatially variable, response-spectrum-compatible simulations to be used in the nonlinear seismic response evaluation of extended structures, such as bridges. However, just as recorded data, these simulated accelerations require processing, but, unlike recorded data, the reasons for their processing are purely numerical. Hence, the criteria for the processing of acceleration simulations need to be tied to the effect of processing on the structural response. This paper presents a framework for processing acceleration simulations that is based on seismological approaches for processing recorded data, but establishes the corner frequency of the high-pass filter by minimizing the effect of processing on the response of the structural system, for the response evaluation of which the ground motions were generated. The proposed two-step criterion selects the filter corner frequency by considering both the dynamic and the pseudo-static response of the systems. First, it ensures that the linear/nonlinear dynamic structural response induced by the processed simulations captures the characteristics of the system's dynamic response caused by the unprocessed simulations, the frequency content of which is fully compatible with the target response spectrum. Second, it examines the adequacy of the selected estimate for the filter corner frequency by evaluating the pseudo-static response of the system subjected to spatially variable excitations. It is noted that the first step of this two-fold criterion suffices for the establishment of the corner frequency for the processing of acceleration time series generated at a single ground-surface location to be used in the seismic response evaluation of, e.g. a building structure. Furthermore, the concept also applies for the processing of acceleration time series generated by means of any approach that does not provide physical considerations for the selection of the corner frequency of the high-pass filter.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.10A
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• pp.930-937
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• 2005

In this paper, a processing time and traffic capacity analysis algorithm for RFID system using LBT-serial searching scheme is proposed. Service time, carrier sensing time, additional delay time required for contiguous frequency channel occupancy, and additional delay time required for the contiguous using the same frequency channel are considered and the processing delay and frequency channel capacity are analyzed for the steady state operation of the system. The simulation results showing maximum capacity of the system and explaining the accuracy of the algorithm are provided.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.1A
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• pp.71-79
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• 2010

The algorithm of multiple channel signal processing requires the flexibility of variable frequency band, efficient allocation of transmission power, and flexible frequency band reallocation to satisfy various service types which requires different transmission rates and frequency band. This paper proposes an improved multiple channel signal processing for converting the frequency band of multiple carrier signals efficiently using a window function and DFT in the time domain. In contrast to the previous algorithm of multiple-channel signal processing performing band-pass signal processing in the frequency domain, the proposed algorithm is a method of block signal processing using a window function in the time domain. In addition, the complexity of proposed algorithm of the window function is lower than that of the previous algorithm performing signal processing in the frequency domain, and it performs the frequency band transform efficiently. The computer simulation result shows that the perfect reconstruction of output signal and the flexible frequency band reallocation is performed efficiently by the proposed algorithm.

• Proceedings of the IEEK Conference
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• 2000.07b
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• pp.835-838
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• 2000

Popular methods for extracting a text region in video images are in general based on analysis of a whole image such as merge and split method, and comparison of two frames. Thus, they take long computing time due to the use of a whole image. Therefore, this paper suggests the faster method of extracting a text region without processing a whole image. The proposed method uses line sampling methods, FFT and neural networks in order to extract texts in real time. In general, text areas are found in the higher frequency domain, thus, can be characterized using FFT The candidate text areas can be thus found by applying the higher frequency characteristics to neural network. Therefore, the final text area is extracted by verifying the candidate areas. Experimental results show a perfect candidate extraction rate and about 92% text extraction rate. The strength of the proposed algorithm is its simplicity, real-time processing by not processing the entire image, and fast skipping of the images that do not contain a text.

• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2005.11a
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• pp.302-305
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• 2005

Recently, many methods to analyze signal have been proposed and representative methods are the Fourier transform and wavelet transform. In these methods, the Fourier transform represents signal with combination cosine and sine at all locations in the frequency domain. However, it doesn't provide time information that particular frequency occurs in signal and denpends on only the global feature of the signal. So, to improve these points the wavelet transform which is capable of multiresolution analysis has been applied to many fields such as speech processing, image processing and computer vision. And the wavelet transform, which uses changing window according to scale parameter, presents time-frequency localization. In this paper, we proposed a new approach using a wavelet of cosine and sine type and analyzed features of signal in a limited point of frequency-time plane.

• The Journal of the Acoustical Society of Korea
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• v.29 no.1
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• pp.41-47
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• 2010

The pitch alteration technique for voice conversion is classified in time domain, frequency domain and hybrid domain. The Hybrid domain method has a merit of clearness and natural-ness of pitch altered speech but has the major drawback of long processing time. In this paper, we proposed a new method that can reduce the processing time of pitch alteration in time-frequency hybrid domain. We omitted the bit-reversing process of FFT and IFFT in changing the processing domain. Therefore we can reduce the processing time by 86.26% to the conventional method with same quality.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.9A
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• pp.822-829
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• 2005

In this paper, a processing time and trafnc capacity analysis algorithm for RFID system using LBT-Random Searching scheme is proposed. Service time, carrier sensing time, additional delay time required for contiguous frequency channel occupancy, and additional delay time required for the contiguous using the same frequency channel are considered and the processing delay and frequency channel capacity are analyzed for the steady state operation of the system. The simulation results showing maximum capacity of the system and explaining the accuracy of the algorithm are provided.

• Korean Journal of Fisheries and Aquatic Sciences
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• v.51 no.2
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• pp.178-186
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• 2018

A chirp-echo data acquisition and processing system was developed for use as a simplified, PC-based chirp echo-sounder with some data processing software modules. The design of the software and hardware system was implemented via a field-programmable gate array (FPGA). Digital signal processing algorithms for driving a single-channel chirp transmitter and dual-channel receivers with independent TVG (time varied gain) amplifier modules were incorporated into the FPGA for better real-time performance. The chirp-echo data acquisition and processing system consisted of a notebook PC, an FPGA board, and chirp sonar transmitter and receiver modules, which were constructed using three chirp transducers operating over a frequency range of 35-210 kHz. The functionality of this PC-based chirp echo-sounder was tested in various field experiments. The results of these experiments showed that the developed PC-based chirp echo-sounder could be used in the acquisition, processing and analysis of broadband acoustic echoes related to fish species identification.

• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2000.08a
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• pp.45-48
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• 2000

In this paper algorithms of frequency hopping sequences generation such as Lempel-Greenberger, optimum Lempel-Greenberger and Kumar sequences for spread spectrum communications are described. Using the scalogram based on wavelet transform, time-frequency characteristics of frequency hopped signals corresponding to the considered hopping sequences are analyzed.

• ETRI Journal
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• v.44 no.5
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• pp.849-858
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• 2022

Dynamic voltage frequency scaling (DVFS) has been widely adopted for runtime power management of various processing units. In the case of neural processing units (NPUs), power management of neural network applications is required to adjust the frequency and voltage every layer to consider the power behavior and performance of each layer. Unfortunately, DVFS is inappropriate for layer-wise run-time power management of NPUs due to the long latency of voltage scaling compared with each layer execution time. Because the frequency scaling is fast enough to keep up with each layer, we propose a layerwise dynamic frequency scaling (DFS) technique for an NPU. Our proposed DFS exploits the highest frequency under the power limit of an NPU for each layer. To determine the highest allowable frequency, we build a power model to predict the power consumption of an NPU based on a real measurement on the fabricated NPU. Our evaluation results show that our proposed DFS improves frame per second (FPS) by 33% and saves energy by 14% on average, compared with DVFS.