• The Journal of the Acoustical Society of Korea
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• v.6 no.3
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• pp.52-59
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• 1987

In this paper, we present a method of realizing an infinite impulse response (IIR) digital filter (DF)using linear delta modulation (LDM) as a simple analog/digital (A/D) converter. This method makes the realization of IIR digital filters much simpler than that of conventional ones because it does not require hardware multipliers and a pulse code modulation (PCM) A/D converter. Compared to the finite impulse respponse (FIR) LDMDF of Lee and Un [1] , this IIR LDMDF requires significantly less computation time.

• Journal of Electrical Engineering and Technology
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• v.3 no.3
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• pp.430-435
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• 2008

In the paper, an algorithm called a Constraint algorithm is proposed to solve the fan-in problem occurred in ADC encoding circuits. The Flash ADC architecture uses a double-base number system (DBNS). The DBNS has known to represent the multi-dimensional logarithmic number system (MDLNS) used for implementing the multiplier accumulator architecture of FIR filter in digital signal processing (DSP) applications. The authors use the DBNS with the base 2 and 3 to represent binary output of ADC. A symmetric map is analyzed first, and then asymmetric map is followed to provide addition read DBNS to DSP circuitry. The simulation results are shown for the Double-Base Integer Encoder (DBIE) of the 6-bit ADC to demonstrate an effectiveness of the Constraint algorithm, using $0.18{\mu}\;m$ CMOS technology. The DBIE’s processing speed of the ADC is fast compared to the FAT tree encoder circuit by 0.95 GHz.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.22 no.2
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• pp.306-317
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• 1997

This ppaer presents a bit-rate conversion system for efficient communications between 3 channel CVSD systems with different bit-rates. The proposed conversion system is implemented in the digital domain and specially, the conversion problem between 32 Kbps and 16 Kbps CVSD systems is studied. The conventional conversion system implemented in the analog domain allows signals to be easily degraded by external noises. To overcome this problem, a digital CVSD bit-rate conversion system robust to external noises is developed. the new systemdecodes CVSD bit sequences and converts sampling rates of decoded signals, then encodes signals at target bit-rates. Since linear phase property does not matter in this application, instead of FIR filters a IIR filter is employed to reduce the system complexity. Therefore, a 3 channel digital CVSD bit-rate conversion system was successfully real-time implemented using a general purpose DSP. In addition, conversion problems with unkown time constants were experimented and good experimental results were obtained.

• Journal of Institute of Control, Robotics and Systems
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• v.16 no.12
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• pp.1143-1149
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• 2010

Because the noise is the main cause for decreasing the performance at speech recognition, the place or environment is very important in speech recognition. To improve the speech recognition performance in the real situations where various extraneous noises are abundant, a novel combination of FIR and Wiener filters is proposed and experimented. The combination resulted in improved accuracy and reduced processing time, enabling fast analysis and response in emergency situations. Usually, there are many dangerous situations in our city life, so for the smarter city it is necessary to detect many types of sound in various environment. Therefore this paper is about how to detect many types of sound in real city, especially on CCTV. This paper is for implementing the smarter city by detecting many types of sounds and filtering one of the emergency sound in this sound stream. And then it can be possible to handle with the emergency or dangerous situation.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.33B no.2
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• pp.95-102
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• 1996

This paper describes the realization of a rela-time adaptive acoustic echo canceller, which adopts a microprogramming method, for removing acoustical echoes in speakerphone systems using th eADSP-2101 microprocessor with a pipeline and modified harvard architecture. We apply the LMS (least mean square) algorithm to estimate the coefficients of a transversal FIR filter. For the acustic adaptive echo canceller, we propose a parallel operation programming to imrove algorithm execution speed and apply a nonlinear quantization to reduce the quantization error caused by large dynamic range of voice signal.

• The Journal of the Acoustical Society of Korea
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• v.11 no.1E
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• pp.56-63
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• 1992

The poblem of active oise control has been analysed using a adaptive signal processing technique. In this methods, the adaptive signal processor or model predicts the primary sound wave travelling along the acoustic plant and generates the secondary source 180° out of phase which attempts to attempts to attenuate the undesired noise by destructive interference. In the solutions presented here, acoustic propagation delay is considered as a part of the model which used the FIR filter. The effects of error path and auxiliary path transfer functioin are anayzed and a new on=-line technique for error path modeling, adaptive delayed inverse modeling is presented. In this study, using these new concepts, our system can more reduce the noise level in duct to 5dB-15dB than only using LMS algorithm system.

• Proceedings of the Earthquake Engineering Society of Korea Conference
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• 2001.04a
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• pp.58-65
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• 2001

It is essential to know exactly what the response of the seismograph is inclusive of characteristic of the seismic sensors before using it for detailed seismic study. This is because the recorded earthquake data can be more or less affected by the overall system and need to be corrected properly to the analysis`s best to obtain the right results. In this respect, two basic earthquake data processing techniques are introduced and applied, for validation purpose, to real data from KEPRI seismic monitoring system which were established for determining the site-specific characteristics of the earthquakes around the Nuclear Power Plants. One is conventional instrumental correction technique for velocity data and the other is for removing acausal ringing originate from using linear phase FIR filter. These techniques are all implemented in the time domain using digital filtering process and shows the desired results when applied to real earthquake data.

• Proceedings of the KIEE Conference
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• 2001.05a
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• pp.38-41
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• 2001

This paper presents a digital current differential protection algorithm for a transformer in power system. This algorithm uses an FIR filter to improve the performance of the relay. This paper presents a practical method setting the operating slope of the relay and reduce ct mismatch. A series of EMTP simulation results have shown effectiveness of the algorithm under various type of transformers and conditions.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 1994.04a
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• pp.16-21
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• 1994

For the systems such as ANC(Active Noise Control) systems having auxiliary path after FIR type adaptive filter, Filtered-X LMS algorithm is effective. However behaviors of this algorithm has not been fully understood. The convergence property of this algorithm depends on not only cross correlation matrix between the filtered signals through model and real auxiliary path state solution of weight vector in Filtered-X LMS algorithm is investigated for under-determined case, over-determined case, and nonsingular case. Also, the convergence speed in case of two tone noise is investigated based on the eigenvalue spread of cross correlation matrix.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2008.04a
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• pp.714-720
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• 2008

This research presents a laboratory study about an subjective response of impulsive sound caused by leisure shooting. The sources are sampled from outdoor noise and their levels range from 40 to 75 dB at the interval of 5dB. The noise unit is based on A-weighted sound exposure level (ASEL; $L_{AE}$). To make equal ASEL of outdoor noise, finite impulse response (FIR) filter is applied to the originally sampled source to include the effect of distance attenuation. The evaluation method of the jury test adopted a Semantic Difference(SD) Method. In the result of the jury test for impulsive noise, the mean response rating expressed a linear relation and %HA(percent highly annoyed) displayed a exponential growth relation.