• The Journal of the Acoustical Society of Korea
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• v.30 no.2
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• pp.73-79
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• 2011

This paper presents a new impulsive noise detection algorithm in speech signal. The proposed method employs the frequency domain characteristic of the impulsive noise to improve the detection accuracy while avoiding the false-alarm problem by the pitch of the speech signal. Furthermore, we proposed time-frequency domain impulsive noise detector that utilizes both the time and frequency domain parameters which minimizes the false-alarm problem by mutually complementing each other. As the result, the proposed time-frequency domain detector shows the best performance with 99.33 % of detection accuracy and 1.49 % of false-alarm rate.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.18 no.6
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• pp.75-82
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• 2018

In this paper, we consider a precoder design for downlink multiuser multiple-input multiple-output (MU-MIMO) in distributed antenna systems (DAS). In DAS, remote radio heads (RRHs) are placed at geographically different locations within a cell area. Three different precoder design schemes are proposed to maximize the separate or joint signal-to-leakage-plus-noise ratio (SLNR) metrics by considering RRH sum power or per-RRH power constraints. The analytical closed-form form solution for each optimization problem is presented. Through computer simulation, we show that the joint SLNR based precoding schemes have better signal-to-interference-plus-noise ratio (SINR) and bit error rate (BER) performances than the separate SLNR based schemes. Also, it is shown that the precoding scheme with RRH sum power constraint has better performance than the precoding scheme with per-RRH power constraint.

• Proceedings of the IEEK Conference
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• summer
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• pp.199-203
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• 2004

In this paper, we only describe the digital block of two-channel 18-bit analog-to-digital (A/D) converter employing sigma-delta method and xl28 decimation. The device contains two fourth comb filters with 1-bit input from sigma­delta modulator. each followed by a digital half band FIR(Finite Impulse Response) filters. The external analog sigma-delta modulators are sampled at 6.144MHz and the digital words are output at 48kHz. The fourth-order comb filter has designed 3 types of ways for optimal power consumption and signal-to-noise ratio. The following 3 digital filters are designed with 12tap, 22tap and 116tap to meet the specification. These filters eliminate images of the base band audio signal that exist at multiples of the input sample rate. We also designed these filters with 8bit and 16bit filter coefficient to analysis signal-to-noise ratio and hardware complexity. It also included digital output interface block for I2S serial data protocol, test circuit and internal input vector generator. It is fabricated with 0.35um HYNIX standard CMOS cell library with 3.3V supply voltage and the chip size is 2000um by 2000um. The function and the performance have been verified using Verilog XL logic simulator and Matlab tool.

• International Journal of Computer Science & Network Security
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• v.24 no.3
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• pp.182-188
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• 2024

Hardware components are an integral part of Hardware Define Radio (HDR) for seamless operations and optimal performance. On the other hand, Software Define Radio (SDR) is a program that does not rely on any hardware components for its performance. Both of the latter radio programmers utilize modulation functions to make their core components from signal processing viewpoint. The following paper concentrates on SDR based modulation and their performance under different modulations. The bit error rate (BER) of modulations such as PSK, QAM, and PSAM were used as indicators to test channel quality estimation in planar Rayleigh fading. Though it is not commonly used for channel fading, the method of the adder determines the regionally segmented channel fading. Thus, the estimation error of the channel change substantially reduces the performance of the signal, hence, proving to be an effective option. Moreover, this paper also elaborates that BER is calculated as a function of the sample size (signal length) with an average of 20 decibels. Consequently, the size of the results for different modulation schemes has been explored. The analytical results through derivations have been verified through computer simulation. The results focused on parameters of amplitude estimation error for 1dB reduction in the average signal-to-noise ratio, while the combined amplitude deviation estimation error results are obtained for a 3.5 dB reduction

• Journal of Sensor Science and Technology
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• v.9 no.4
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• pp.316-326
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• 2000

A new signal processing algorithm for white light interferometry has been proposed and investigated theoretically. The goal of the algorithm is to determine the absolute optical path length of an interferometer with very high precision (<< one optical wavelength). The algorithm features cross-correlation of interferometer fringe scans and hypothesis testing. The hypothesis test looks for a zero order fringe peak candidate about which the cross-correlation is symmetric minimizing the uncertainty of misidentification. The shot noise limited performance of the proposed signal processing algorithm has been analyzed using computer simulations. Simulation results were extrapolated to predict the misidentification rate at Signal to-Shot noise ratio (SNR) higher than 31 dB. Root-mean-square phase error between the computer-generated zero order fringe peak and the estimated zero order fringe peak has been calculated for the changes of three different parameters (SNR, fringe scan sampling rate, coherence length of light source). Results of computer simulations showed the ability of the proposed signal processing algorithm to identify the zero order fringe peak correctly. The proposed signal processing algorithm uses a software approach, which is potentially inexpensive, simple and fast.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2003.10a
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• pp.606-609
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• 2003

It is known that 1/f noise of MOSFET is generated by superposition of single Random Telelgraph Signal (RTS). In this study, jitter from 1/f noise of MOSFET is analysed with RTS supplied to one of the nodes of the ring oscillator under investigation. Jitter rates are investigated as the number of stage, power supply voltage, and the amplitude of RTS change.

• Proceedings of the Korean Institute of Communication Sciences Conference
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• 1984.04a
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• pp.37-41
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• 1984

The error rate performance of amplitude phase keying system has been studied in the environment of cochannel interference and impulsive noise. We have derived the error probability equations of amplitude shift keying signal and phase shift keying signal, and combining the results, we have evaluated the circular APK signal which is the one of the several cases of APK arrays. Using the derived equations, the circular APK system has been evaluated in terms of carrier-to-noise power ratio(CNR), carrier-to-interferer power ratio(CIR), and impulsive index. The graphic results show us the best case and worst case of APK system, and good performance compared to the other systems in cochannel interference and impulsive noise.

• Journal of Convergence for Information Technology
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• v.11 no.6
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• pp.1-6
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• 2021

In the field of speech recognition, as the DNN is applied, the use of speech recognition is increasing, but the amount of calculation for parallel training needs to be larger than that of the conventional GMM, and if the amount of data is small, overfitting occurs. To solve this problem, we propose an efficient method for robust voice feature extraction and voice signal noise removal even when the amount of data is small. Speech feature extraction efficiently extracts speech energy by applying the difference in frame energy for speech and the zero-crossing ratio and level-crossing ratio that are affected by the speech signal. In addition, in order to remove noise, the noise of the speech signal is removed by removing the noise of the speech signal with an average predictive improved LMS filter with little loss of speech information while maintaining the intrinsic characteristics of speech in detection of the speech signal. The improved LMS filter uses a method of processing noise on the input speech signal by adjusting the active parameter threshold for the input signal. As a result of comparing the method proposed in this paper with the conventional frame energy method, it was confirmed that the error rate at the start point of speech is 7% and the error rate at the end point is improved by 11%.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.13 no.2
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• pp.962-977
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• 2019

The human voice is a convenient method of information transfer between different objects such as between men, men and machine, between machines. The development of information and communication technology, the voice has been able to transfer farther than before. The way to communicate, it is to convert the voice to another form, transmit it, and then reconvert it back to sound. In such a communication process, a vocoder is a method of converting and re-converting a voice and sound. The CELP (Code-Excited Linear Prediction) type vocoder, one of the voice codecs, is adapted as a standard codec since it provides high quality sound even though its transmission speed is relatively low. The EVRC (Enhanced Variable Rate CODEC) and QCELP (Qualcomm Code-Excited Linear Prediction), variable bit rate vocoders, are used for mobile phones in 3G environment. For the real-time implementation of a vocoder, the reduction of sound quality is a typical problem. To improve the sound quality, that is important to know the size and shape of noise. In the existing sound quality improvement method, the voice activated is detected or used, or statistical methods are used by the large mount of data. However, there is a disadvantage in that no noise can be detected, when there is a continuous signal or when a change in noise is large.This paper focused on finding a better way to decrease the reduction of sound quality in lower bit transmission environments. Based on simulation results, this study proposed a preprocessor application that estimates the SNR (Signal to Noise Ratio) using the spectral SNR estimation method. The SNR estimation method adopted the IMBE (Improved Multi-Band Excitation) instead of using the SNR, which is a continuous speech signal. Finally, this application improves the quality of the vocoder by enhancing sound quality adaptively.

• The Journal of the Acoustical Society of Korea
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• v.26 no.6
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• pp.301-305
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• 2007

In this paper, we propose a simple algorithm for detecting underwater transient signals on the fact that the frequency range of underwater transient signals is similar to audio frequency. For this, we use a preprocessing module of EVRC speech codec that is the standard speech codec of the mobile communications. If a signal is entered into EVRC noise suppression module, we can get some parameters such as the update flag, the energy of each channel, the noise suppressed signal, the energy of input signal, the energy of background noise, and the energy of enhanced signal. Therefore the energy of the enhanced signal that is normalized with the energy of the background noise is compared with the pre-defined detection threshold, and then we can detect the transient signal. And the detection threshold is updated using the previous value in the noisy period. The experimental result shows that the proposed algorithm has $0{\sim}4% error rate in the AWGN or the colored noise environment.