• Speech Sciences
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• v.10 no.4
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• pp.125-136
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• 2003

Speech coding techniques have been studied to reduce the complexity and bit rate but also to improve the sound quality. CELP type vocoder, has used as a one of standard, supports the great sound quality even low bit rate. In this paper, the preprocessing of input speech to reduce the bit rate is the different with the conventional vocoder. The different kinds of parameter are used for the preprocessing so this paper is compared with theses parameters for finding the more appropriate parameter for the vocoder. The parameters are used to synthesize the speech not to encode or decode for coding technique so we proposed the simple algorithm not to have the influence on the processing time or the computation time. The parameters in used the preprocessing step are speaking rate, duration and PSOLA technique.

• Journal of Biomedical Engineering Research
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• v.24 no.5
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• pp.465-471
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• 2003

The purpose of this research was to evaluate chronic stress using physiological parameters. Wistar rats were exposed to the sound stress for 14 days. Biosignals were acquired hourly. To develop a fuzzy inference system which can integrate physiological parameters. the parameters of the system were adjusted by the adaptive neuro-fuzzy inference system. Of the training dataset, input dataset was the physiological parameters from the biosignals and output dataset was the target values from the cortisol production. Physiological parameters were integrated using the fuzzy inference system. then 24-hour results were analyzed by the Cosinor method. Chronic stress was evaluated from the degree of circadian rhythm disturbance. Suppose that the degree of stress for initial rest period is 1. Then. the degree of stress after 14-day sound stress increased to 1.37, and increased to 1.47 after the 7-day recovery period. That is, the rat was exposed to 37%-increased amount of stress by the 14-day sound and did not recover after the 7-day recovery period.

• Korean Journal of Air-Conditioning and Refrigeration Engineering
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• v.17 no.2
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• pp.95-100
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• 2005

Numerical simulations and experiments were conducted to analyze the design parameters for a slim room air-conditioner. These design parameters included a fan shape, a front panel, a scroll shape, a bell mouth, a distance between a fan and a heat exchanger, etc. Each design parameter was analyzed numerically and/or experimentally in terms of the flow rate and the sound pressure level, which should be the most influential factors for developing the slim room air-conditioner. The fan with a uniform height showed a better performance than that with a linearly varying height. It is recommended to use a front grill rather than a front panel according to sound pressure levels since the front panel itself is a huge resistance to the inlet flow. A redesigned scroll shape by changing the rotational direction of a fan also contributed a lot to lowering the sound pressure level. There existed a distance between a fan and a heat exchanger, where flow rates increased effectively.

• International Journal of Highway Engineering
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• v.20 no.3
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• pp.59-64
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• 2018

PURPOSES : We propose a parameter-setting-free harmony-search (PSF-HS) algorithm to determine the individual sound power levels of noise sources in the cases of industrial or road noise environment. METHODS :In terms of using methods, we use PSF-HS algorithm because the optimization parameters cannot be fixed through finding the global minimum. RESULTS:We found that the main advantage of the PSF-HS heuristic algorithm is its ability to find the best global solution of individual sound power levels through a nonlinear complex function, even though the parameters of the original harmony-search (HS) algorithm are not fixed. In an industrial and road environment, high noise exposure is harmful, and can cause nonauditory effects that endanger worker and passenger safety. This study proposes the PSF-HS algorithm for determining the PWL of an individual machine (or vehicle), which is a useful technique for industrial (or road) engineers to identify the dominant noise source in the workplace (or road field testing case). CONCLUSIONS : This study focuses on providing an efficient method to determine sound power levels (PWLs) and the dominant noise source while multiple machines (or vehicles) are operating, for comparison with the results of previous research. This paper can extend the state-of-the-art in a heuristic search algorithm to determine the individual PWLs of machines as well as loud machines (or vehicles), based on the parameter-setting-free harmony-search (PSF-HS) algorithm. This algorithm can be applied into determining the dominant noise sources of several vehicles in the cases of road cross sections and congested housing complex.

• The Journal of the Acoustical Society of Korea
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• v.42 no.5
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• pp.395-401
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• 2023

Sound Event Detection (SED) aims to identify not only sound category but also time interval for target sounds in an audio waveform. It is a critical technique in field of acoustic surveillance system and monitoring system. Recently, various models have introduced through Detection and Classification of Acoustic Scenes and Events (DCASE) Task 4. This paper explored how to design optimal parameters of DenseNet based model, which has led to outstanding performance in other recognition system. In experiment, DenseRNN as an SED model consists of DensNet-BC and bi-directional Gated Recurrent Units (GRU). This model is trained with Mean teacher model. With an event-based f-score, evaluation is performed depending on parameters, related to model architecture as well as model training, under the assessment protocol of DCASE task4. Experimental result shows that the performance goes up and has been saturated to near the best. Also, DenseRNN would be trained more effectively without dropout technique.

• The Journal of the Acoustical Society of Korea
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• v.28 no.3
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• pp.260-267
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• 2009

It is very difficult to form a desired complex sound field at a designated region precisely as an application of acoustic arrays, which is one of important objects of array systems. To solve the problem, a filter design method was suggested, which employed the concept of an inverse method using the acoustical holography based on the boundary element method. In the acoustical holography used for the source identification, the measured field data are employed to reconstruct the vibro-acoustic parameters on the source surface. In the analogous problem of source array design, the desired field data at some specific points in the sound field was set as constraints and the volume velocity at the surface points of the source plane became the source signal to satisfy the desired sound field. In the filter design, the constraints for the desired sound field are set, first. The array source and given space are modelled by the boundary elements. Then, the desired source parameters are inversely calculated in a way similar to the holographic source identification method. As a test example, a target field comprised of a quiet region and a plane wave propagation region was simultaneously realized by using the array with 16 loudspeakers.

• 제어로봇시스템학회:학술대회논문집
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• 2003.10a
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• pp.1398-1403
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• 2003

In this paper, a simple method to design an MIMO sound field control system was proposed. The control system was designed to achieve 1) noise attenuation and 2) sound equalization by utilizing feedback and feedforward controllers. The method was based on partial model matching on frequency domain which only required measured frequency response data, or impulse response data in order to tune parameters of the controller. The proposed method was applied to a normal office room and results of experiment showed effectiveness of the proposed method.

• Proceedings of the IEEK Conference
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• 2002.07a
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• pp.117-120
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• 2002

A single DSP implementation of 3D sound movement is described. With the use of a realtime 3D acoustic image localization algorithm, an efficient approach is devised for synthesizing the 3D sound movement by interpolating only two parameters of "delay" and "gain". Based on this algorithm, the realtime 3D sound synthesis is performed by a commercially available 16-bit fixed-point DSP with computational labor of 65 MIPS and memory space of 9.6k words, which demonstrates that the algorithm call be used even for the mobile applications.

• Transactions of the Korean Society of Mechanical Engineers A
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• v.22 no.4
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• pp.879-886
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• 1998

It is very important to understand the vibration and noise characteristics of a structure to developed quiet machines and lessen their noise. In this paper, the vibration and sound radiation characteristics of a simple and a bar-stiffened plate have been investigated using numerical and experimental techniques. In numerical process, FEM analysis has been performed for the vibration level ; the time-space squared and averaged velocity and BEM analysis for sound radiation parameters ; sound power and radiation efficiency. In experimental process, FFT signal processing method has been used. While a power from an exiciter is applied to the structure by using a point contact, sound intensity and vibration level has been measured. Based on these two data, the radiation efficiency has been calculated. Results show that the radiation efficiency for the stiffened structure increases compared to the simple plate, due to the extra edges provided by the stiffener.

• Phonetics and Speech Sciences
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• v.1 no.4
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• pp.105-109
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• 2009

This thesis describes the method for well voice announcement using the TTS(Text-To-Speech) technology in the shop music broadcasting service. Offering a high quality TTS sound service for each shop requires a great expense. According to a report on the architectural acoustics the room acoustic indexes such as reverberation time and early decay time are closely connected with a subjective awareness about acoustics. By using the result the customers will be able to recognize better the voice announcement by applying sound effect to speech files made by TTS. The result of an aural comprehension examination has shown better about almost all of the parameters by applying reverb effect to TTS sound.