• Proceedings of the IEEK Conference
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• 2000.09a
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• pp.827-830
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• 2000

3GPP and ETSI adopted AMR(Adaptive Multi-Rate) as a standard for next generation IMT-2000 service. In this paper, we analyzed algorithm about AMR and optimized ANSI C source on the C complier and assembly language of Texas Instrument . The implemented AMR speech codec requires 28.2MIPS of complexity for encoder and 5.5MIPS for decoder. we performed real-time implementation of AMR speech codec using 82% of TMS320C5402 with 40 MIPS specification. We give proof that the output speech of the implemented speech codec on DSP board is identical with result of C source program simulation. Also the reconstructed speech is verified in the real-time environment consisted of microphone and speaker.

• The Journal of the Acoustical Society of Korea
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• v.14 no.3
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• pp.28-36
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• 1995

In this paper, a double loop control model using leaky delay LMS algorithm are proposed for active noise control. The proposed double loop control model estimates the loudspeaker characteristic and the error path transfer function with on-line using only gain and acoustic time delay to reduce computation burden. The control of error signal through double loop control scheme makes the more robust cntrol system. The input signal of filter to estimate acoustic time delay is used difference between input signal of input microphone and adaptive filter output. And also, in nonstationary environments, the leaky delay LMS algorithm is employed to counteract parameter drift of delay LMS algorithm. For practical noise signal, the proposed double loop control model reduces noise level about 12.9 dB.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.17 no.2
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• pp.280-286
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• 2013

In this paper, a noise reduction system is proposed for the speech interface in intelligent TV applications. To reduce TV speaker sound which are very serious noises degrading recognition performance, a noise reduction algorithm utilizing the direct TV sound as the reference noise input is implemented. In the proposed algorithm, transfer functions are estimated to compensate for the difference between the direct TV sound and that recorded with the microphone installed on the TV frame. Then, the noise power spectrum in the received signal is calculated to perform Wiener filter-based noise cancellation. Additionally, a postprocessing step is applied to reduce remaining noises. Experimental results show that the proposed algorithm shows 88% recognition rate for isolated Korean words at 5 dB input SNR.

• The Transactions of the Korean Institute of Power Electronics
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• v.2 no.2
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• pp.35-40
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• 1997

The hysteresis control method has been frequently used for current control of power conversion equipments or motor drive systems. This method makes the measured signal follow the reference signal by changing the control signal whenever the error signal exceeds the preset band width. In this paper, hysteresis control method with fast response characteristics is applied for active noise control to suppress acoustic noise. Both Pentium processor and sound blaster 16 are used for experimental implementation, which executes A/D, D/A conversion and also is used as operating source of loudspeaker for audible noise cancellation.

• Journal of the Korean Society for Precision Engineering
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• v.26 no.7
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• pp.51-57
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• 2009

In this paper, we studied on an attenuation effect of automobile exhaust noise according to the direction of canceling speaker in duct-acoustic ANC system. Automobile exhaust noise was recorded at 800rpm, 3500rpm and 5000rpm of a diesel engine. Directions of canceling speaker can be set to $30^{\circ}$, $90^{\circ}$ and $150^{\circ}$ against the primary noise flow by acrylic ducts to be made for the experimentation. DSP board used to control the ANC system. The algorithm of this ANC system applied the Filtered-x-LMS algorithm that is modified to compensate for a property of DSP input signal and the secondary-path effect. As an experiment result, the direction of canceling speaker was proved to influence the reduction effect of noise. The $150^{\circ}$ duct in the attenuation effect of noise showed a better result than the $90^{\circ}$ or $30^{\circ}$ duct.

• Journal of the Korean Institute of Illuminating and Electrical Installation Engineers
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• v.20 no.5
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• pp.99-106
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• 2006

In this paper, implementation of active mufflers using multiple channel adaptive IIR filter is presented. Usually, recursive LMS(RLMS) algorithms for adaptive IIR filters are highly efficient than filtered-X LMS(FXLMS) algorithms, when the order of both algorithms are the same. However, RLMS algorithms usually diverge before the algorithms arenot yet converged. So, the prefilters are presented to improve the stability by pulling the poles of feedback control transfer function in the beginning of active noise control and returning the original poles after the filters converge. The engine noises of diesel engine automobiles and gasoline engine automobiles are analyzed and the mathematical model of an active muffler is derived. Computer simulations and experiments are performed to show the effectiveness of the proposed systems.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.13 no.11
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• pp.2311-2316
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• 2009

Lately, in the automotive market appeared keywords such as convenience, safety in presentation and increase importance of pan of vehicle. Accordingly, the use of many electronic devices was required essentially and communication between electronic devices is being highlighted. Various devices such as controllers, sensors and multimedia device(audio, speakers, video, navigation) in-vehicle connected car network such as CAN, MOST. Modem in-vehicle network managed and operated as purpose of each other. In this Paper, intelligent car navigation considering convenience and safety implement on MOST Network and present system to control CAN Network in vehicle.

• Korean Journal of Fisheries and Aquatic Sciences
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• v.44 no.5
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• pp.543-549
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• 2011

An underwater speaker was developed for use as an acoustic deterrent device that transmits acoustic energy through the water omnidirectionally over a broadband frequency range to eliminate marine mammal attacks and to prevent physical damage to the inshore and coastal fishing grounds of Korea. The underwater speaker was constructed of two vibration caps machined from 6061-T6 aluminum alloy and a stack of PZ 26 piezoelectric ceramic rings (Ferroperm Piezoceramics A/S) connected mechanically in series and electrically in parallel. The performance characteristics of the underwater speaker were measured and analyzed in an experimental water tank of $5\;m{\times}5\;m{\times}6\;m$. The peak transmitting voltage response (TVR) was measured at 11.16 kHz with 163.45 dB re $1\;{\mu}Pa$/V at 1m. The underwater speaker showed a near omnidirectional beam pattern at the peak TVR resonance frequency. The usable frequency range was 4-25 kHz with a lower TVR limit of approximately 140 dB. We conclude that this underwater speaker could be satisfactorily used as an acoustic deterrent device against marine mammals, particularly the bottlenose dolphin, to protect catches and fishing grounds as well as the mammals themselves, for example, by keeping them away from fishing gear and/or vessels.

• The Journal of the Korea institute of electronic communication sciences
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• v.4 no.1
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• pp.66-69
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• 2009

An adaptive echo canceller is necessary for an application such as a speakerphone, 3G image telephony and VoIP service system. These echo cancellers need to have many taps for filtering echo signals. Many taps cause computation data to increase and convergence speed to be low. To overcome these problems, An adaptive echo canceller with the advanced convergence speed is proposed in this paper. To improve the speed, we divide an echo band into subbands and place a subband filter to be adaptive for each subband. Each subband filter recognizes the echo signal as subband echo signals. So, dynamic range of subband is small, the convergence speed is fast. Moreover, as the number of Tap and weight update are estimated in each subband, the implementation complex of a adaptive filter is low.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.23 no.11
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• pp.951-956
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• 2013

This paper describes techniques used to analyze the sweet spot of sound field reproduced by ear-level linear arrays of loudspeakers by geometrical approach method. Previous researches have introduced various sweet spot definitions in their own way. In general, sweet spot is defined as an area whose stereophonic sound effect is valid. Its size is affected by the geometrical arrangement of the system. In this paper, a case when plane waves are generated by linear arrays of loudspeakers in the horizontal plane is considered. So the sweet spot is defined as an area in which the listener can perceive the desired azimuth angle. Because there are many loudspeakers, impulse responses at listener's ears are in the form of pulse-train and the time-duration of the pulse-train affects the localization performance of the listener. So we calculated the maximum time duration of pulse-train by geometrical approach method and identified with the results of impulse response simulation. This paper also includes parameter analysis with respect to aperture size, so it suggests a tool for sound engineers to expect the sweet spot size and listener's sound perception.