• Journal of the Korea Society of Computer and Information
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• v.22 no.7
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• pp.83-91
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• 2017

In this paper, we propose a thorough solution for adjusting virtual sound field with different kinds of devices and software in preliminary procedure and late stage of music processing. The basic process of music production includes composing, arranging and recording at pre-production stage as well as sound mixing and mastering at post-production stage. At the initial stage of music creation, it should be checked whether the design of virtual sound field, the choice of the tone and the instrument used in the arrangement match the virtual sound field required for the final work. In later recording, mixing and mastering, elaborate adjustments should be done to the virtual sound field. This study also analyzed how to apply the parameter of the effectors to the design and adjustment of the virtual sound field, making it the source of our creation.

• The Journal of the Acoustical Society of Korea
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• v.18 no.3
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• pp.79-87
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• 1999

In this paper, we reconstruct an algorithm of sound image control for real-time processing and implement a real-time system using digital signal processing board based on TMS320C40. The performance of real-time sound image control system was evaluated by a listening test. The results of the test showed that localized sound image can be perceived by headphone and speaker, and result of test by headphone was better than that of speaker. In the results of test of elevation perception, result of perception between left and right were better than those for differentiating between front and back, and the moving sound image was better than the fixed sound image.

• Journal of Advanced Marine Engineering and Technology
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• v.38 no.9
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• pp.1125-1130
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• 2014

In this paper, we propose a sound reception system against surrounding noise for ships based on digital signal processing technologies. In order to suppress unwanted surrounding noises, a digital band-pass filter is designed, which the pass-band of the filter is between 70Hz to 820Hz. Also, we develope a sound direction indicating algorithm with 4 microphones. After filtering the audio signals from 4 microphones, the developed sound direction indicating algorithm can indicate 8 directions. In addition, we implement prototype board for the sound reception using a digital signal processor chip and audio codecs, and verify the proposed algorithm.

• Proceedings of the Korean Society of Marine Engineers Conference
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• 2006.06a
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• pp.169-170
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• 2006

The main purpose of this study is to develop a program for sound field visualization system which gets noise signals in microphones array for incoming noise signals and it uses to operate noise signals and to store data in multi-channel FFT and is consisted to visualize noise signals with a image which is got by camera in center of array by using beamforming algorithm of the array signal processing.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.28B no.12
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• pp.9-18
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• 1991

In this Paper, a rule-based method for the phenomenon of Korean vocal sound change is proposed. This method could be used to solve a problem between symbolic(Hangul)and phonetic language(Korean) for the study of Korean speech processing. A rule on the phenomenon of vocal sound rearranged for the rule base with a end-consonents on the authority of standard pronunciation rule. The proposed rule base system is simplified by the implementation for the vocal sound change. Also, it is useful to create the data base with phonetic value for the Korean voice processing by syllable unit.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2004.11a
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• pp.441-444
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• 2004

In the sound insulation measurements, the influence of background (extraneous) noise is often serious problem and how to reduce its effect and to improve the signal-to-noise(S/N) ratio is an important theme. As the background noise, such extraneous noises as road traffic noise and machine noise often disturb the measurement. In laboratory measurements on specimens with high sound insulation performances, even the internal noise of the measurement system can become a problem. To improve the signal-to-noise ratio and to improve the measurement accuracy, various kinds of digital signal processing techniques can be applied. In this paper, four kinds of digital signal processing techniques are applied and their effectiveness is examined through field measurements.

• Proceedings of the IEEK Conference
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• 1999.06a
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• pp.1107-1110
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• 1999

In speech signal processing, the accurate decision of the voiced/unvoiced sound is important for robust word recognition and analysis and a high coding efficiency. In this paper, we propose the mehod of the voiced/unvoiced decision using the LSP parameter which represents the spectrum characteristics of the speech signal. The voiced sound has many more LSP parameters in low frequency region. To the contrary, the unvoiced sound has many more LSP parameters in high frequency region. That is, the LSP parameter distribution of the voiced sound is different to that of the unvoiced sound. Also, the voiced sound has the minimun value of sequantial intervals of the LSP parameters in low frequency region. The unvoiced sound has it in high frequency region. we decide the voiced/unvoiced sound by using this charateristics. We used the proposed method to some continuous speech and then achieved good performance.

• Proceedings of the IEEK Conference
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• 2003.07e
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• pp.2244-2247
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• 2003

In this paper, we present a method which can minimize distortion from desired signal in thoracic sound signal processing. We firstly chose the proper wavelet mother function to reduce noise components. Secondly, we chose a clean thoracic sound, then added Gaussian noise and 3 step(10, 15, 20db) uniform noise to it. Finally, the various wavelet functions are applied for noise cancellation. To evaluate the efficiency of this study, we computed SNR and RSE value. Then we found the optimal mother wavelet function for thoracic sound.

• Proceedings of the Korean Information Science Society Conference
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• 2003.10a
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• pp.4-6
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• 2003

A holistic representation, such as sparse ceding or independent component analysis (ICA), was successfully applied to explain early auditory processing and sound classification. In contrast, Part-based representation is an alternative way of understanding object recognition in brain. In this paper. we employ the non-negative matrix factorization (NMF)［1］which learns parts-based representation for sound classification. Feature extraction methods from spectrogram using NMF are explained. Experimental results show that NMF-based features improve the performance of sound classification over ICA-based features.

• The Journal of the Acoustical Society of Korea
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• v.26 no.4
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• pp.144-152
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• 2007

Conventional research works on the classification of the heart sound signal have been done mainly with the artificial neural networks. But the analysis results on the statistical characteristic of the heart sound signal have shown that the HMM is suitable for modeling the heart sound signal. In this paper, we model the various heart sound signals representing different heart diseases with the HMM and find that the classification rate is much affected by the clustering of the heart sound signal. Also, the heart sound signal acquired in real environments is a continuous signal without any specified starting and ending points of time. Hence, for the classification based on the HMM, the continuous cyclic heart sound signal needs to be manually segmented to obtain isolated cycles of the signal. As the manual segmentation will incur the errors in the segmentation and will not be adequate for real time processing, we propose a variant of the ergodic HMM which does not need segmentation procedures. Simulation results show that the proposed method successfully classifies continuous heart sounds with high accuracy.