• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.35-38
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• 2004

In this paper, we investigate and analyze the problems encountered in frame-based estimation of TDOA(Time Difference of Arrival) using CPSP function. Spectral leakage occurring in framing of a speech signal by a rectangular window makes estimation of CPSP spectrum inaccurate. Framing with a Hamming window to reduce the spectral leakage effect distorts the signal due to the different weighting at temporally same sample, which make the TDOA estimation using CPSP function inaccurate. In this paper, we solve this problem by reducing the dynamic range of the spectrum of a speech signal with preemphasis. Experimental results confirm that the framing of pre-emphasized microphone output with a rectangular window shows higher success ratio of TDOA estimation than any other framing methods.

• The Journal of Society for e-Business Studies
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• v.23 no.2
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• pp.111-121
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• 2018

Many problems in rotating machinery such as aircraft engines, wind turbines and motors are caused by bearing defects. The abnormalities of the bearing can be detected by analyzing signal data such as vibration or noise, proper pre-processing through a few signal processing techniques is required to analyze their frequencies. In this paper, we introduce the condition monitoring method for diagnosing the failure of the rotating machines by analyzing the vibration signal of the bearing. From the collected signal data, the normal states are trained, and then normal or abnormal state data are classified based on the trained normal state. For preprocessing, a Hamming window is applied to eliminate leakage generated in this process, and the cepstrum analysis is performed to obtain the original signal of the signal data, called the formant. From the vibration data of the IMS bearing dataset, we have extracted 6 statistic indicators using the cepstral coefficients and showed that the application of the Mahalanobis distance classifier can monitor the bearing status and detect the failure in advance.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.7 no.4
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• pp.673-677
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• 2003

In this study, we propose the new filter bank that is adaptive filter bank using neural networks in time domain. Also, we proposed a new filter neuron as neuron with filter window, the structure and algorithm for filter banks. The performance of neural filter banks is shown from two examples. It show characteristics the simple structure and higher speed processing than traditional methods (filter banks in frequency domain, etc.). In many applications, the proposed method will provide the high performance to features detection of signals in time domain.

• The Journal of the Korea Contents Association
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• v.10 no.7
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• pp.31-39
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• 2010

In this study, we prepared the real environmental sound data sets arising from people's movement comprising 9 different environment types. The environmental sounds are pre-processed with pre-emphasis and Hamming window, then go into the classification experiments with the extracted features using MFCC (Mel-Frequency Cepstral Coefficients). The GMM (Gaussian Mixture Model) classifier without post processing tends to yield abruptly changing classification results since it does not consider the results of the neighboring frames. Hence we proposed the post processing methods which suppress abruptly changing classification results by taking the probability or the rank of the neighboring frames into account. According to the experimental results, the method using the probability of neighboring frames improve the recognition performance by more than 10% when compared with the method without post processing.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.41 no.6
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• pp.271-278
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• 2004

In this paper, we propose a content-based audio genre classification algorithm that automatically classifies the query audio into five genres such as Classic, Hiphop, Jazz, Rock, Speech using digital sign processing approach. From the 20 seconds query audio file, the audio signal is segmented into 23ms frame with non-overlapped hamming window and 54 dimensional feature vectors, including Spectral Centroid, Rolloff, Flux, LPC, MFCC, is extracted from each query audio. For the classification algorithm, k-NN, Gaussian, GMM classifier is used. In order to choose optimum features from the 54 dimension feature vectors, SFS(Sequential Forward Selection) method is applied to draw 10 dimension optimum features and these are used for the genre classification algorithm. From the experimental result, we can verify the superior performance of the proposed method that provides near 90% success rate for the genre classification which means 10%∼20% improvements over the previous methods. For the case of actual user system environment, feature vector is extracted from the random interval of the query audio and it shows overall 80% success rate except extreme cases of beginning and ending portion of the query audio file.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.29 no.10
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• pp.783-790
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• 2018

The frequency-hopping technique is one of the spread-spectrum techniques. Frequency hopping is a communication system in which the carrier frequency channel is hopped within the wideband. Therefore, a frequency-hopping system has such advantages as antijamming and low probability of intercept. This system is often used in military communications. Because frequency-hopping signal detection is difficult, it is an important research issue. A novel detection technique is proposed that can improve detection probability. When the received signal is transformed to a frequency domain sample by fast Fourier transform, spectral leakage lowers the detection probability. This problem can be solved by using the Hamming window, and the detection probability can be increased. However, in a frequency-hopping environment, the windowing technique lowers the detection probability. The proposed method solves this weakness. The simulation results show that the proposed detection technique improves the detection probability by as much as 13 %.

• 한국태양에너지학회:학술대회논문집
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• 2011.04a
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• pp.269-274
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• 2011

A solar cell is primary parts to produce electrical energy from the Sun. And, we can utilize those solar cells as a power generation system in home, factory, and so on. In order to make proper power, the solar cells are configured in series and parallel lay down. In condition of uniform illumination, the solar array will produce an enough power by photovoltaic effects from the solar cells. In case of non-uniform illumination on the solar cells, the power will be dramatically decreased compared to design. Fortunately, there were so many research outputs regarding the illumination effects on solar array. In this work, we tried to find out the non-uniform effects on unit CPV solar cell, because there were no research outputs for unit CPV solar cell considering illumination. The CPV solar cell was used in CPV system to make a power by the Sun. We chosen the triple junction solar cell of GaAsInP2Ge for simulation, which has a 30 % of conversion efficiency. By simulation, we obtained the output performance of CPV solar cells in condition of various illumination by using Hamming Window function. Its performance was degraded by 10 % to 50 % depending illumination conditions.

• The Journal of the Convergence on Culture Technology
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• v.9 no.4
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• pp.563-568
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• 2023

ECG(electrocardiogram) is used to measure the rate and regularity of heartbeats, as well as the size and position of the chambers, the presence of any damage to the heart, and the cause of all heart diseases can be found. Because the ECG signal obtained using the ECG-KIT includes noise in the ECG signal, noise must be removed from the ECG signal to apply to the deep learning. In this paper, Noise included in the ECG signal was removed by using a lowpass filter of the Digital FIR Hamming window function. When the performance evaluation of the three activation functions, sigmoid(), ReLU(), and tanh() functions, which was confirmed that the activation function with the smallest error was the tanh() function, the elapsed time was longer when the batch size was small than large. Also, it was confirmed that result of the performance evaluation for the GRU model was superior to that of the LSTM model.

• The Journal of the Acoustical Society of Korea
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• v.21 no.8
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• pp.744-756
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• 2002

The transient acoustic fields formed by a 3.5 ㎒ curved linear array transducer which is commonly used in ultrasonic medical imaging system for diagnosis of abdomen are systematically analyzed to obtain new design parameters for the better acoustic image. In the analysis with an assumption of radiating waveform, element size, radius of curvature, amplitude apodization are considered as parameters giving constitutive relations with the fields. As simulation results, appropriate new parameters with the reduced curvature and elevation aperture and the apodization of Hamming window, which make an improved acoustic beam with lower side lobe levels than a conventional typical transducer, are obtained.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.12 no.7
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• pp.3455-3474
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• 2018

As a unique form of signed-digit representation, non-adjacent form (NAF) minimizes Hamming weight by removing a stream of non-zero bits from the binary representation of positive integer. Thanks to this strong point, NAF has been used in various applications such as cryptography, packet filtering and so on. In this paper, to improve the NAF conversion speed of the $NAF_w$ algorithm, we propose a new NAF conversion algorithm, called w-bit Shifting Non-Adjacent Form($SNAF_w$), where w is width of scanning window. By skipping some unnecessary bit comparisons, the proposed algorithm improves the NAF conversion speed of the $NAF_w$ algorithm. To verify the excellence of the $SNAF_w$ algorithm, the $NAF_w$ algorithm and the $SNAF_w$ algorithm are implemented in the 8-bit microprocessor ATmega128. By measuring CPU cycle counter for the NAF conversion under various input patterns, we show that the $SNAF_2$ algorithm not only increases the NAF conversion speed by 24% on average but also reduces deviation in the NAF conversion time for each input pattern by 36%, compared to the $NAF_2$ algorithm. In addition, we show that $SNAF_w$ algorithm is always faster than $NAF_w$ algorithm, regardless of the size of w.