• The Journal of the Acoustical Society of Korea
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• v.19 no.5
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• pp.65-70
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• 2000

It is blown that the loudness is one of the most important metrics in assessing the sound quality and a calculation method for loudness has been standardized for steady sounds. In this study, a new loudness model is suggested for dealing with the transient sound for a unified analysis of various practical sounds. A signal processing technique is introduced for this purpose, which is required for the band subdivision and the prediction of band-level change of transient sounds. In addition, models for the post-masking and the temporal integration are adopted in the analysis of the loudness of transient sounds. In order to solve the problem of the conventional loudness model in the pure-tone signal processing, a critical band filter is employed in the analysis, which consists of 47 critical filters having a filter spacing of a half of the critical bandwidth. For testing the effectiveness of the present model, the predicted responses are compared with the experimental data and it is observed that they are in good agreements.

• Journal of the Institute of Convergence Signal Processing
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• v.10 no.4
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• pp.214-219
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• 2009

This paper proposes a watermarking algorithm considering the frequency characteristics of Haegeum sounds for copyright protection of digital Haegeum sound contents. The harmonics of Haegeum sounds commonly have large magnitude values in 1500Hz~2000Hz and 2800Hz~3500Hz so that those bands are selected to embed a watermark. The proposed method computes the FFT (fast Fourier transform) of the original sound signal and embeds the watermark bits generated by PN (pseudo noise) sequence into the harmonics in the selected bands. Furthermore, the proposed method is robust to lowpass filter, bandpass filter, cropping, noise addition, MP3 compression attacks and the maximum BER (bit error rate) is 1.41% after lowpass filter attack. To measure the quality of the watermarked sound, subjective listening test, MUSHRA (multiple stimuli with hidden reference and anchor), was conducted. The mean value of MUSHRA listening test is bigger than 98 and 96.67 for every Haegeum sounds and Korean classical music with Haeguem, respectively.

• The Journal of Korea Robotics Society
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• v.3 no.3
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• pp.237-244
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• 2008

Nowadays, research on human-robot interaction has been getting increasing attention. In the research field of human-robot interaction, speech signal processing in particular is the source of much interest. In this paper, we report a speaker localization system with six microphones for a humanoid robot called MAHRU from KIST and propose a time delay of arrival (TDOA)-based feature matrix with its algorithm based on the minimum sum of absolute errors (MSAE) for sound source localization. The TDOA-based feature matrix is defined as a simple database matrix calculated from pairs of microphones installed on a humanoid robot. The proposed method, using the TDOA-based feature matrix and its algorithm based on MSAE, effortlessly localizes a sound source without any requirement for calculating approximate nonlinear equations. To verify the solid performance of our speaker localization system for a humanoid robot, we present various experimental results for the speech sources at all directions within 5 m distance and the height divided into three parts.

• Proceedings of the IEEK Conference
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• 1998.10a
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• pp.1227-1230
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• 1998

Phase control methods are used to expand the sound image in general AV system. However, these methods are effective only to the signal under 1kHz, and the listener must be located in front center of the speaker system. In this paper, we realize the realtime processing system in which phase shifting method is dominant at low frequency and precedence effect is dominant at high frequency. Two sound cards are used to process the audio signal in realtime with 16 bits stereo channel of 44.1 kHz sampling frequency. And the analog circuit is designed to process the phase shifting. In experiments the usefulness of the proposed stereo system is confirmed.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.275-278
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• 2004

In TV sound system, loudspeakers are subject to severe size constraints. The small size of the transducer affects the low frequency signal performance of the system. Bass signal performance contributes significantly to the user perceived sound quality and a good bass signal reproduction is essential. Increasing the sound energy in the bass signal range is an unviable solution since the gain required are exceedingly high and signal distortion occurs because of the speaker overload. Recently methods are being proposed to invoke low frequency illusion using psychoacoustic phenomena of the missing fundamental. This paper proposes a simple and effective signal processing method to create bass signal illusion in TV speakers using the missing fundamental effect, at a complexity of 12 MIPS on Motorola 56371 audio DSP.

• Journal of Biosystems Engineering
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• v.24 no.3
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• pp.253-258
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• 1999

This study was conducted to develop an automatic egg inspection system for detecting creaked eggs based on acoustic impulse response. This system includes a sound generator, a sound sensor with signal conditioner, and a computer. The sound generator that hit the sharp of the dull edges of an egg was constructed with a ceramic ball pendulum attached to a rotary type solenoid. The signal conditioner included a pre-amplifier and a digital signal processing (DSP) board. The parameters for distinguishing cracked and normal eggs were the area, the geometric centroid and the resonance frequency of power spectrum of the acoustic signal generated. An algorithm for on-line detection of the continuous transferring eggs was developed. The performance tests resulted with 91% success rate to separate cracked and normal eggs at the rate of 1 second per an egg.

• Journal of Korea Multimedia Society
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• v.20 no.2
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• pp.108-114
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• 2017

Lip reading is a field of image processing to assist the process of sound recognition. In some environment, the capture of sound signal usually has significant noise and therefore, the recognition rate of sound signal decreases. Lip reading can be a good feature for the increase of recognition rates. Conventional lip extraction methods have been proposed widely. Maia et. al. proposed a method by the sum of Cr and Cb. However, there are two problems as follows: the point with maximum saturation is not always regarded as lips region and the inner part of lips such as oral cavity and teeth can be classified as lips. To solve these problems, this paper proposes a method which adopts the histogram-based classifier for the extraction of lips region. The proposed method consists of two stages, learning and test. The amount of computation is minimized because this method has no color conversion. The performance of proposed method gives 66.8% of detection rate compared to 28% of conventional ones.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2006.05a
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• pp.1306-1311
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• 2006

The root cause of abnormal sound from a heat exchanger of condensate water system in a nuclear power plant is investigated by using the impact signal-processing methodology based on the Hertz theory. The predicted results for the location of impact force and the loose part size meet good agreement with the identified materials during the overhaul period in the plant. Nuclear power plants have experienced several loose parts and the frequency of the loose part will be increased along the aging of the plants. So, this analysis methodology for the impact signal will be widely utilized for the primary and secondary side of the nuclear power plant.

• Proceedings of the Computational Structural Engineering Institute Conference
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• 2007.04a
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• pp.309-314
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• 2007

Optimal layer sequencing of a multi-layered acoustical foam is solved to maximize its sound transmission loss. A foam consisting of air and poroelastic layers can be optimized when a limited amount of a poroelastic material is allowed. By formulating the sound transmission loss maximization problem as a one dimensional topology optimization problem, optimal layer sequencing and thickness were systematically found for several frequencies. For optimization, the transmission losses of air and poroelastic layers were calculated by the transfer matrix derived from Biot's theory. By interpolating five intrinsic parameters among several poroelastic material parameters, dear air-poroelastic layer distributions were obtained; no filtering or post-processing was necessary. The optimized foam layouts by the proposed method were shown to differ depending on the frequency of interest.

• Proceedings of the Korean Society of Computer Information Conference
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• 2014.01a
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• pp.1-4
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• 2014

Although physical modeling synthesis is becoming more and more efficient in rich and natural high-quality sound synthesis, its high computational complexity limits its use in portable devices. This constraint motivated research of single-instruction multiple-data many-core architectures that support the tremendous amount of computations by exploiting massive parallelism inherent in physical modeling synthesis. Since no general consensus has been reached which grain sizes of many-core processors and memories provide the most efficient operation for sound synthesis, design space exploration is conducted for seven processing element (PE) configurations. To find an optimal PE configuration, each PE configuration is evaluated in terms of execution time, area and energy efficiencies. Experimental results show that all PE configurations are satisfied with the system requirements to be implemented in portable devices.