• The Journal of the Acoustical Society of Korea
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• v.23 no.6
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• pp.437-445
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• 2004

In this paper. we describe a novel method for transforming audio contents according to user's preference on sound field. Sound field effect technologies. which transform or simulate acoustic environments as user's preference, are very important for enlarging the reality of acoustic scene. However huge amount of computational power is required to process sound field effect in real time. so it is hard to implement this functionality at the portable audio devices such as MP3 player. In this paper, we propose an efficient method for providing sound field effect to audio contents independent of terminal's computational power through processing this functionality at the server using user's sound field preference, which is transfered from terminal side. To describe sound field preference, user can use perceptual acoustic parameters as well as the URI address of room impulse response signal. In addition, a novel fast convolution method is presented to implement a sound field effect engine as a result of convoluting with a room impulse response signal at the realtime application. and verified to be applicable to real-time applications through experiments. To verify the evidence of benefit of proposed method we performed two subjective listening tests about sound field descrimitive ability and preference on sound field processed sounds. The results showed that the proposed sound field preference can be applicable to the public.

• Journal of the Korean Society of Laryngology, Phoniatrics and Logopedics
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• v.31 no.2
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• pp.71-77
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• 2020

Background and Objectives OperaVOXTM (Oxford Wave Research Ltd.) is a portable voice analysis software package designed for use with iOS devices. As a relatively cheap, portable and easily accessible form of acoustic analysis, OperaVOXTM may be more clinically useful than laboratory-based software in many situations. The aim of this study was to evaluate the agreement between OperaVOXTM and Multi-Dimensional Voice Program (MDVP; Computerized Speech Lab) to assess voice quality before and after laryngeal microsurgery in patient with vocal polyp. Materials and Method Twenty patients who had undergone laryngeal microsurgery for vocal polyp were enrolled in this study. Preoperative and postoperative voices were assessed by acoustic analysis using MDVP and OperaVOXTM. A five-seconds recording of vowel /a/ was used to measure fundamental frequency (F0), jitter, shimmer and noise-to-harmonic ratio (NHR). Results Several acoustic parameters of MDVP and OperaVOXTM related to short-term variability showed significant improvement. While pre-operative value of F0, jitter, shimmer, NHR was 155.75 Hz (male: 125.37 Hz, female: 183.37 Hz), 2.20%, 6.28%, 0.16, post-operative values of these parameter was 164.34 Hz (male: 129.42 Hz, female: 199.26 Hz), 2.15%, 5.18%, 0.14 Hz in MDVP. While pre-operative value of F0, jitter, shimmer, NHR was 168.26 Hz (male: 135.16 Hz, female: 201.37 Hz), 2.27%, 6.95%, 0.26, post-operative values of these parameters was 162.72 Hz (male: 128.267 Hz, female: 197.18 Hz), 1.71%, 5.36%, 0.20 in OperaVOXTM. There was high intersoftware agreement for F0, jitter, shimmer with intraclass correlation coefficient. Conclusion Our results showed that the short-term variability of acoustic parameters in both MDVP and OperaVOXTM were useful for the objective assessment of voice quality in patients who received laryngeal microsurgery. OperaVOXTM is comparable to MDVP and has high intersoftware reliability with MDVP in measuring the F0, jitter, and shimmer

• The Journal of the Acoustical Society of Korea
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• v.41 no.2
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• pp.184-191
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• 2022

Underwater acoustic wave in shallow water is propagated through multipath that has a large delay spread causing Inter-Symbol Interference (ISI) and these characteristics deteriorate the performance in the communication system. In order to analyze the communication performance and investigate the correlation with multipath delay spread in a reverberant environment, an underwater acoustic communication experiment using Binary Phase-Shift Keying (BPSK) signals with symbol rates from 100 sym/s to 8000 sym/s was conducted in a 5 × 5 × 5 m³ water tank. The acoustic channels in a well-controlled tank environment had the characteristics of dense multipath delay spread due to multiple reflections from the interfaces and walls within the tank and showed the maximum excess delay of 40 ms or less, and the Root Mean Squared (RMS) delay spread of 8 ms or less. In this paper, the performances of Bit Error Rate (BER) and output Signal-to-Noise Ratio (SNR) were analyzed using four types of communication demodulation techniques. And the parameter, Symbol interval to Delay spread Ratio in reverberant environment (SDR_rev), which is the ratio of symbol interval to RMS delay spread in the reverberant environment is defined. Finally, the SDR_rev was compared to the BER and the output SNR. The results present the reference symbol rate in which high communication performance can be guaranteed.

• International Journal of Control, Automation, and Systems
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• v.3 no.2
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• pp.217-224
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• 2005

Generally, the underwater flight vehicle (UFV) depth control system operates with the following problems: it is a multi-input multi-output (MIMO) system because the UFV contains both pitch and depth angle variables as well as multiple control planes, it requires robustness because of the possibility that it may encounter uncertainties such as parameter variations and disturbances, it requires a continuous control input because the system that has reduced power consumption and acoustic noise is more practical, and further, it has the speed dependency of controller parameters because the control forces of control planes depend on the operating speed. To solve these problems, an adaptive fuzzy sliding mode controller (AFSMC), which is based on the decomposition method using expert knowledge in the UFV depth control and utilizes a fuzzy basis function expansion (FBFE) and a proportional integral augmented sliding signal, is proposed. To verify the performance of the AFSMC, UFV depth control is performed. Simulation results show that the AFSMC solves all problems experienced in the UFV depth control system online.

• The Journal of the Acoustical Society of Korea
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• v.17 no.2E
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• pp.10-15
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• 1998

Application of the Filtered-X LMS adaptive filter to active noise cancellation requires to estimate the transfer characteristics between the output and the error signal of the adaptive canceler. In this paper, we derive an adaptive cancellation algorithm and analyze is convergence behavior when the acoustic noise is assumed to consist of multiple sinusoids. The results of the convergence analysis of the Filtered-X LMS algorithm indicate that the effects of parameter estimation inaccuracy on the convergence behavior of the algorithm are characterize by two distinct components : Phase estimation error and estimated magnitude. In particular, the convergence of the Filtered-X LMS algorithm is show to be strongly affected by the accuracy of the phase response estimate. Simulation results of the algorithm are presented which support the theoretical convergence analysis.

• Proceedings of the Korean Institute of Electrical and Electronic Material Engineers Conference
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• 2003.05d
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• pp.29-32
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• 2003

Recently, the development of diagnosis technique with high confidence is important on power equipment, for this reason is to use for measurement and analysis of PD with prior appearance of insulation breakdown In this paper, we presents the analysis of trouble signal to use both method of general analysis of $\Phi$-Q-N in PD and statistical parameter by this interpretation Equipment of simulation has made independently DS for 170kV GIS of one phase with same on field. The detected signal through the sensor of Induction and Acoustic Emission is classified which used to characteristic neural network algorithm and then it is analysis.

• Proceedings of the KSPS conference
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• 1996.02a
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• pp.23-40
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• 1996

The communicative disorders in cleft palate patients have relationship with the acoustic and He physiological phenomena. Particularily hypernasality is a parameter of cleft palate speech that has been studied by many clinicians and speech pathologists. The degree of hypernasality has been assessed by the listener,s judgement, but perceptual assessements have poor scientific reliability, so objective instruments have been needed to test hypernasality with diagnostics accuracy. This study was analyzed the nasalance score using a Nasometer for cleft palate patients. The simple vowels /a/, /i/, /e/ and the approximants /j/, /w/ were tested for the degree of hypernasality after operation. The phrases containing long and short duration times were used in this study to asses hypeernasality. Fiberopic views shows the open velopharyngeal port that resulted in hypernasality of cleft palate patients. The authors assert the important of the management of cleft palate patients.

• Journal of the Korean Society of Laryngology, Phoniatrics and Logopedics
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• v.12 no.2
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• pp.115-120
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• 2001

In this paper, we synthesized vowel /ae/ with voice synthesis program of Dr. Speech Science, and we also synthesized pathologic vowel /ae/ by some parameters such as high frequency gain (HFG), low frequency gain(LFG), pitch flutter(PF) which represents jitter value and flutter of amplitude(FA) which represents shimmer value, and grade ranked as mild, moderate and severe respectively. And then we analysed all pathologic voice by analysis program of Dr. Speech Science. We expect that this synthesized pathologic voices are useful for understanding the parameter such as noise, jitter and shimmer and feedback effect to patient with voice disorder.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.13 no.4
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• pp.239-246
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• 2003

Application of the filtered-x Least Mean Square(LMS) adaptive filter to active noise control requires to estimate the transfer characteristics between the output and the error signal of the adaptive controller. In this paper, we derive the filtered-x adaptive noise control algorithm and analyze its convergence behavior when the acoustic noise consists of multiple sinusoids. The results of the convergence analysis of the filtered-x LMS algorithm indicate that the effects of the parameter estimation inaccuracy on the convergence behavior of the algorithm are characterized by two distinct components Phase estimation error and estimated gain. In particular, the convergence is shown to be strongly affected by the accuracy of the phase response estimate. Simulation results are presented to support the theoretical convergence analysis.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2007.11a
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• pp.23-27
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• 2007

Many trials have been used to model room impulse responses, all attempting to provide efficient representations of room acoustics. The traditional model designs for room impulse response seem to fail in accuracy, controllability, or computational efficiency. In time domain, a room impulse response is generally considered as the combination of three parts having different acoustic characteristics, initial time delay, early reflection, and late reverberation. This paper introduces new learning hybrid model for the room impulse response. In this proposed model, those three parts are modeled using different models with learning algorithms that determine the length or boundary of each model in the hybrid model. By the simulation with measured room impulse responses, it was examined that the performance of proposed model shows the best efficiency in views of both the parameter numbers and modeling error.